Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm
Sure. Contact me off list... W _____ From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Tuesday, September 28, 2004 2:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/8dc685cc/attachment.htm
Harry, This is just out of consideration for the hundreds of people who are not interested in the same. Since I will be sending attachments, it is just a kindness to spare them the additional download time and keep the list weight light. Again, anyone interested in the files can contact me off list and I will send them a copy of mine, al beit cleansed of the usual security information like passwords and such. Cheers, Wiley -----Original Message----- From: harry gaillac [mailto:gaillacharry@yahoo.fr] Sent: Tuesday, September 28, 2004 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 off list ?? some people could be interested in configuring soundpoint ip. harry --- "Wiley E. Siler" <wsiler@e2020inc.com> a ?crit :> Sure. Contact me off list... > > W > > _____ > > From: Tim Jackson [mailto:tim@angelinacounty.net] > Sent: Tuesday, September 28, 2004 2:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Polycom IP500 > > > Got my first round of IP500s in today. Anybody have any example > sip.cfg files they'd like to share? > > Tim Jackson > Network Engineer > Angelina County, Texas > (936)639-4827 office > (936)414-6723 mobile > > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > information by persons or entities other than the intended recipient > is prohibited. If you received this in error, please contact the > sender and delete the material from any computer > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Cr?ez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Anybody have any example configs for the phones themselves? The sip.conf isn't hard, I just need some help in setting up the phones. -Tim -----Original Message----- From: Wiley E. Siler [mailto:wsiler@e2020inc.com] Sent: Tuesday, September 28, 2004 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Harry, This is just out of consideration for the hundreds of people who are not interested in the same. Since I will be sending attachments, it is just a kindness to spare them the additional download time and keep the list weight light. Again, anyone interested in the files can contact me off list and I will send them a copy of mine, al beit cleansed of the usual security information like passwords and such. Cheers, Wiley -----Original Message----- From: harry gaillac [mailto:gaillacharry@yahoo.fr] Sent: Tuesday, September 28, 2004 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 off list ?? some people could be interested in configuring soundpoint ip. harry --- "Wiley E. Siler" <wsiler@e2020inc.com> a ?crit :> Sure. Contact me off list... > > W > > _____ > > From: Tim Jackson [mailto:tim@angelinacounty.net] > Sent: Tuesday, September 28, 2004 2:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Polycom IP500 > > > Got my first round of IP500s in today. Anybody have any example > sip.cfg files they'd like to share? > > Tim Jackson > Network Engineer > Angelina County, Texas > (936)639-4827 office > (936)414-6723 mobile > > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > information by persons or entities other than the intended recipient > is prohibited. If you received this in error, please contact the > sender and delete the material from any computer > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Cr?ez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
The config files for the phone are in the zip I sent you. Did you not receive the zip? W -----Original Message----- From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Tuesday, September 28, 2004 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Anybody have any example configs for the phones themselves? The sip.conf isn't hard, I just need some help in setting up the phones. -Tim -----Original Message----- From: Wiley E. Siler [mailto:wsiler@e2020inc.com] Sent: Tuesday, September 28, 2004 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Harry, This is just out of consideration for the hundreds of people who are not interested in the same. Since I will be sending attachments, it is just a kindness to spare them the additional download time and keep the list weight light. Again, anyone interested in the files can contact me off list and I will send them a copy of mine, al beit cleansed of the usual security information like passwords and such. Cheers, Wiley -----Original Message----- From: harry gaillac [mailto:gaillacharry@yahoo.fr] Sent: Tuesday, September 28, 2004 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 off list ?? some people could be interested in configuring soundpoint ip. harry --- "Wiley E. Siler" <wsiler@e2020inc.com> a ?crit :> Sure. Contact me off list... > > W > > _____ > > From: Tim Jackson [mailto:tim@angelinacounty.net] > Sent: Tuesday, September 28, 2004 2:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Polycom IP500 > > > Got my first round of IP500s in today. Anybody have any example > sip.cfg files they'd like to share? > > Tim Jackson > Network Engineer > Angelina County, Texas > (936)639-4827 office > (936)414-6723 mobile > > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > information by persons or entities other than the intended recipient > is prohibited. If you received this in error, please contact the > sender and delete the material from any computer > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Cr?ez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Nope, didn't receive a zip file. -Tim -----Original Message----- From: Wiley E. Siler [mailto:wsiler@e2020inc.com] Sent: Wednesday, September 29, 2004 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 The config files for the phone are in the zip I sent you. Did you not receive the zip? W -----Original Message----- From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Tuesday, September 28, 2004 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Anybody have any example configs for the phones themselves? The sip.conf isn't hard, I just need some help in setting up the phones. -Tim -----Original Message----- From: Wiley E. Siler [mailto:wsiler@e2020inc.com] Sent: Tuesday, September 28, 2004 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Harry, This is just out of consideration for the hundreds of people who are not interested in the same. Since I will be sending attachments, it is just a kindness to spare them the additional download time and keep the list weight light. Again, anyone interested in the files can contact me off list and I will send them a copy of mine, al beit cleansed of the usual security information like passwords and such. Cheers, Wiley -----Original Message----- From: harry gaillac [mailto:gaillacharry@yahoo.fr] Sent: Tuesday, September 28, 2004 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 off list ?? some people could be interested in configuring soundpoint ip. harry --- "Wiley E. Siler" <wsiler@e2020inc.com> a ?crit :> Sure. Contact me off list... > > W > > _____ > > From: Tim Jackson [mailto:tim@angelinacounty.net] > Sent: Tuesday, September 28, 2004 2:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Polycom IP500 > > > Got my first round of IP500s in today. Anybody have any example > sip.cfg files they'd like to share? > > Tim Jackson > Network Engineer > Angelina County, Texas > (936)639-4827 office > (936)414-6723 mobile > > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > information by persons or entities other than the intended recipient > is prohibited. If you received this in error, please contact the > sender and delete the material from any computer > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Cr?ez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I've read somewhere that in the new SIP application they added the XML browser to the IP500, is this true? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/6ca65cc1/attachment.htm
Nope. You need to have the IP600 for the mini-browser function. _____ From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Thursday, September 30, 2004 2:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 I've read somewhere that in the new SIP application they added the XML browser to the IP500, is this true? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/153f8489/attachment.htm
Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -----Original Message----- From: Andrei (MPI) [mailto:asterisk@markovprocesses.com] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote:>Hey All, > >First Time Writing. > >I'm trying to set up my IP500 phones to register SIP with *. I input >all the (I assume) correct data in to the fields on the Web Interface. >And I get no notification that the phone is even attempting to >register, no failed messages etc. I have read that the Web interface is>crap and the XML config files is the way to go. Does anyone have a >basic config file that doesn't change any defaults? I couldn't seem tofind one.> >Extra Info: >Server is 192.168.0.3 >Phone name/ext I want to be 301 > >Thanks, >Chris Cherry > > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like "not hearing" one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote:>Any idea if 1.34 fixes the problems with the phones being up for long >periods of time and weird call problems (I cannot hear remote caller, >but they can hear me) ? > >-Tim > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of PaulHales>Sent: Wednesday, December 01, 2004 6:00 PM >To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >Subject: RE: [Asterisk-Users] Polycom IP500 > > >Any idea if 1.34 makes Daylight Savings work for us people inAustralia?> >PaulH > >-----Original Message----- >From: Andrei (MPI) [mailto:asterisk@markovprocesses.com] >Sent: Thursday, 2 December 2004 9:30 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Polycom IP500 > >Hi Chris, > >First of all, you need to configure ftp or tftp and watch syslog >closely - >what the phone is looking for at boot time. You would need to putconfig>files into (t)ftp directory, named according to MAC address of you >phone. >XML and Web is really weird - they do not even share same config data. >For >example, I had to change address of SNTP server (clock) and it stillwas>showing as 'clock' on Web admin page for the phone. I will email youthe>config files I got from a good fellow from this list not so long ago... >and >those config files do really help! > >Also, I suggest that you upgrade your SIP firmware if you have not done >it >yet. I got the Polycom 500 with firmware which was very old and >incapable to >work with asterisk. Mine is 1.3.1 now >(http://www.freedomphones.net/polycom/files/). > >If anyone has SIP firmware 1.3.4 - please make it downloadablesomewhere>on >the internet? > >Thank you, >Andrei > >Chris Cherry wrote: > > > >>Hey All, >> >>First Time Writing. >> >>I'm trying to set up my IP500 phones to register SIP with *. I input >>all the (I assume) correct data in to the fields on the Web Interface.>>And I get no notification that the phone is even attempting to >>register, no failed messages etc. I have read that the Web interfaceis>> >> > > > >>crap and the XML config files is the way to go. Does anyone have a >>basic config file that doesn't change any defaults? I couldn't seem to >> >> >find >one. > > >>Extra Info: >>Server is 192.168.0.3 >>Phone name/ext I want to be 301 >> >>Thanks, >>Chris Cherry >> >> >> >> >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >CAUTION: This email message and accompanying data may contain >information >that is confidential. If you are not the intended recipient, you are >notified that any use, dissemination, distribution or copying of this >message or data is prohibited. If you have received this email message >in >error, please notify us immediately and erase all copies of thismessage>and >attachments. Thank you. > >CAUTION: This email message and accompanying data may contain >information that is confidential. If you are not the intendedrecipient,>you are notified that any use, dissemination, distribution or copyingof>this message or data is prohibited. If you have received this email >message in error, please notify us immediately and erase all copies of >this message and attachments. Thank you. >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone "learns" the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway. Ty -----Original Message----- From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like "not hearing" one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote:>Any idea if 1.34 fixes the problems with the phones being up for long >periods of time and weird call problems (I cannot hear remote caller, >but they can hear me) ? > >-Tim > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of PaulHales>Sent: Wednesday, December 01, 2004 6:00 PM >To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >Subject: RE: [Asterisk-Users] Polycom IP500 > > >Any idea if 1.34 makes Daylight Savings work for us people inAustralia?> >PaulH > >-----Original Message----- >From: Andrei (MPI) [mailto:asterisk@markovprocesses.com] >Sent: Thursday, 2 December 2004 9:30 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Polycom IP500 > >Hi Chris, > >First of all, you need to configure ftp or tftp and watch syslog >closely - >what the phone is looking for at boot time. You would need to putconfig>files into (t)ftp directory, named according to MAC address of you >phone. >XML and Web is really weird - they do not even share same config data. >For >example, I had to change address of SNTP server (clock) and it stillwas>showing as 'clock' on Web admin page for the phone. I will email youthe>config files I got from a good fellow from this list not so long ago... >and >those config files do really help! > >Also, I suggest that you upgrade your SIP firmware if you have not done >it >yet. I got the Polycom 500 with firmware which was very old and >incapable to >work with asterisk. Mine is 1.3.1 now >(http://www.freedomphones.net/polycom/files/). > >If anyone has SIP firmware 1.3.4 - please make it downloadablesomewhere>on >the internet? > >Thank you, >Andrei > >Chris Cherry wrote: > > > >>Hey All, >> >>First Time Writing. >> >>I'm trying to set up my IP500 phones to register SIP with *. I input >>all the (I assume) correct data in to the fields on the Web Interface.>>And I get no notification that the phone is even attempting to >>register, no failed messages etc. I have read that the Web interfaceis>> >> > > > >>crap and the XML config files is the way to go. Does anyone have a >>basic config file that doesn't change any defaults? I couldn't seem to >> >> >find >one. > > >>Extra Info: >>Server is 192.168.0.3 >>Phone name/ext I want to be 301 >> >>Thanks, >>Chris Cherry >> >> >> >> >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >CAUTION: This email message and accompanying data may contain >information >that is confidential. If you are not the intended recipient, you are >notified that any use, dissemination, distribution or copying of this >message or data is prohibited. If you have received this email message >in >error, please notify us immediately and erase all copies of thismessage>and >attachments. Thank you. > >CAUTION: This email message and accompanying data may contain >information that is confidential. If you are not the intendedrecipient,>you are notified that any use, dissemination, distribution or copyingof>this message or data is prohibited. If you have received this email >message in error, please notify us immediately and erase all copies of >this message and attachments. Thank you. >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Routing here isn't an issue. The router that is their gateway has all internal routes learned via OSPF. Connectivity to the * box is fine (all 100mbit, the interface the phones are on is a dot1q sub-interface though). I'm 100% confident that it's not an routing/nat problem (no NAT taking place). But I've given up. No real answers on the polycom issue, I've just taken the phones and the * box down. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ty Purcell Sent: Thursday, December 02, 2004 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone "learns" the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway. Ty -----Original Message----- From: Tim Jackson [mailto:tim@angelinacounty.net] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like "not hearing" one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote:>Any idea if 1.34 fixes the problems with the phones being up for long >periods of time and weird call problems (I cannot hear remote caller, >but they can hear me) ? > >-Tim > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of PaulHales>Sent: Wednesday, December 01, 2004 6:00 PM >To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >Subject: RE: [Asterisk-Users] Polycom IP500 > > >Any idea if 1.34 makes Daylight Savings work for us people inAustralia?> >PaulH > >-----Original Message----- >From: Andrei (MPI) [mailto:asterisk@markovprocesses.com] >Sent: Thursday, 2 December 2004 9:30 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Polycom IP500 > >Hi Chris, > >First of all, you need to configure ftp or tftp and watch syslog >closely - >what the phone is looking for at boot time. You would need to putconfig>files into (t)ftp directory, named according to MAC address of you >phone. >XML and Web is really weird - they do not even share same config data. >For >example, I had to change address of SNTP server (clock) and it stillwas>showing as 'clock' on Web admin page for the phone. I will email youthe>config files I got from a good fellow from this list not so long ago... >and >those config files do really help! > >Also, I suggest that you upgrade your SIP firmware if you have not done >it >yet. I got the Polycom 500 with firmware which was very old and >incapable to >work with asterisk. Mine is 1.3.1 now >(http://www.freedomphones.net/polycom/files/). > >If anyone has SIP firmware 1.3.4 - please make it downloadablesomewhere>on >the internet? > >Thank you, >Andrei > >Chris Cherry wrote: > > > >>Hey All, >> >>First Time Writing. >> >>I'm trying to set up my IP500 phones to register SIP with *. I input >>all the (I assume) correct data in to the fields on the Web Interface.>>And I get no notification that the phone is even attempting to >>register, no failed messages etc. I have read that the Web interfaceis>> >> > > > >>crap and the XML config files is the way to go. Does anyone have a >>basic config file that doesn't change any defaults? I couldn't seem to >> >> >find >one. > > >>Extra Info: >>Server is 192.168.0.3 >>Phone name/ext I want to be 301 >> >>Thanks, >>Chris Cherry >> >> >> >> >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >CAUTION: This email message and accompanying data may contain >information >that is confidential. If you are not the intended recipient, you are >notified that any use, dissemination, distribution or copying of this >message or data is prohibited. If you have received this email message >in >error, please notify us immediately and erase all copies of thismessage>and >attachments. Thank you. > >CAUTION: This email message and accompanying data may contain >information that is confidential. If you are not the intendedrecipient,>you are notified that any use, dissemination, distribution or copyingof>this message or data is prohibited. If you have received this email >message in error, please notify us immediately and erase all copies of >this message and attachments. Thank you. >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on "10.24.102.0/24" and Asterisk resides on "10.24.100.0/24". Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote:>Theres no NAT going on here. Just 1 router in between, phones reside on >10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT >problem. Any other ideas? > >-Tim > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei >(MPI) >Sent: Wednesday, December 01, 2004 11:15 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Polycom IP500 > >Tim, > >You may see description of new 1.3.4 firmware at polycom.com (check - >http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd>f >) released in October. > >Though, it was proven over time that troubles with a SIP phone like"not> >hearing" one side or the other is NAT related problems. You may want to>investigate firewall setup. I am not saying it is not phone related,but> >the phone would be the last one to blame. > >Also, may I express my feelfings about Cisco and Polycom - not allowing>direct firmware download for their phones - that sucks big time. I will>get the firmware this way or the other. They just force me to waste my >time again and again contacting their dealers and searching the >internet. That should just enrage customers, in my opinion. Are they so>big, they do not even care? > >Sincerely, >Andrei > >Tim Jackson wrote: > > > >>Any idea if 1.34 fixes the problems with the phones being up for long >>periods of time and weird call problems (I cannot hear remote caller, >>but they can hear me) ? >> >>-Tim >> >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul >> >> >Hales > > >>Sent: Wednesday, December 01, 2004 6:00 PM >>To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >>Subject: RE: [Asterisk-Users] Polycom IP500 >> >> >>Any idea if 1.34 makes Daylight Savings work for us people in >> >> >Australia? > > >>PaulH >> >>-----Original Message----- >>From: Andrei (MPI) [mailto:asterisk@markovprocesses.com] >>Sent: Thursday, 2 December 2004 9:30 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [Asterisk-Users] Polycom IP500 >> >>Hi Chris, >> >>First of all, you need to configure ftp or tftp and watch syslog >>closely - >>what the phone is looking for at boot time. You would need to put >> >> >config > > >>files into (t)ftp directory, named according to MAC address of you >>phone. >>XML and Web is really weird - they do not even share same config data. >>For >>example, I had to change address of SNTP server (clock) and it still >> >> >was > > >>showing as 'clock' on Web admin page for the phone. I will email you >> >> >the > > >>config files I got from a good fellow from this list not so longago...>>and >>those config files do really help! >> >>Also, I suggest that you upgrade your SIP firmware if you have notdone>>it >>yet. I got the Polycom 500 with firmware which was very old and >>incapable to >>work with asterisk. Mine is 1.3.1 now >>(http://www.freedomphones.net/polycom/files/). >> >>If anyone has SIP firmware 1.3.4 - please make it downloadable >> >> >somewhere > > >>on >>the internet? >> >>Thank you, >>Andrei >> >>Chris Cherry wrote: >> >> >> >> >> >>>Hey All, >>> >>>First Time Writing. >>> >>>I'm trying to set up my IP500 phones to register SIP with *. I input >>>all the (I assume) correct data in to the fields on the WebInterface.>>> >>> > > > >>>And I get no notification that the phone is even attempting to >>>register, no failed messages etc. I have read that the Web interface >>> >>> >is > > >>> >>> >>> >>> >> >> >> >> >>>crap and the XML config files is the way to go. Does anyone have a >>>basic config file that doesn't change any defaults? I couldn't seemto>>> >>> >>> >>> >>find >>one. >> >> >> >> >>>Extra Info: >>>Server is 192.168.0.3 >>>Phone name/ext I want to be 301 >>> >>>Thanks, >>>Chris Cherry >>> >>> >>> >>> >>> >>> >>> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >>CAUTION: This email message and accompanying data may contain >>information >>that is confidential. If you are not the intended recipient, you are >>notified that any use, dissemination, distribution or copying of this >>message or data is prohibited. If you have received this email message >>in >>error, please notify us immediately and erase all copies of this >> >> >message > > >>and >>attachments. Thank you. >> >>CAUTION: This email message and accompanying data may contain >>information that is confidential. If you are not the intended >> >> >recipient, > > >>you are notified that any use, dissemination, distribution or copying >> >> >of > > >>this message or data is prohibited. If you have received this email >>message in error, please notify us immediately and erase all copies of >>this message and attachments. Thank you. >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
http://www.freedomphones.net/polycom/files/ -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Sent: Sunday, December 05, 2004 4:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom IP500 Does anyone have a location to download the latest Polycom firmware etc? Other than the extranet site, because I am not a reseller, there fore I have no login. [minirant] And shouldn't end users be granted access to this kind of thing anyway? Geeze [/minirant] Thanks, Chris Cherry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.4.5 - Release Date: 12/3/2004 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays "Line used remotely" and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid="Tim Jackson" <101> secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 0106005724|key |*|00|Initial log entry. Current logging level 4 0106005724|ssps |*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 0106005724|sip |*|00|Initial log entry. Current logging level 4 0106005724|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 0106005724|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang="") 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 1 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 2 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username="101", realm="angelinacounty.net", nonce="243b35d1", uri="sip:192.9.200.9:5060", response="11f3478d812d35993018150f29fb5e81", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.202.2:5060>;expires=3600 Date: Thu, 06 Jan 2005 06:46:36 GMT Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 13 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as77cf03d0 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:100@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@192.9.202.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060> Contact: <sip:asterisk@192.9.200.9> Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 192.9.202.2:5060 Scheduling destruction of call '23c9fa48037fec98416d74650481661e@192.9.200.9' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7d0b7e8a", uri="sip:100@192.9.200.9:5060", response="38d9b121d4ee361e584727823f195810", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 100 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as5a181df1 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:100@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B Contact: <sip:100@192.9.200.9> Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 339 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:101@192.9.200.9;method=SUBSCRIBE" /> <atom id="100"> <address uri="sip:100@192.9.200.9;user=ip" priority="0,800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060>;tag=81B3E4D0-500C78DF CSeq: 102 NOTIFY Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 Contact: <sip:101@192.9.202.2:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Destroying call '23c9fa48037fec98416d74650481661e@192.9.200.9' asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B CSeq: 102 NOTIFY Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Message is NOTIFY asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as1d96eff1 Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7735e16f", uri="sip:101@192.9.200.9:5060", response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 101 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE Contact: <sip:101@192.9.200.9> Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:101@192.9.200.9"> <dialog id="101"> <state>confirmed</state> </dialog> </dialog-info> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE CSeq: 102 NOTIFY Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Destroying call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile
How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays "Line used remotely" and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid="Tim Jackson" <101> secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang="") 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 1 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 2 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username="101", realm="angelinacounty.net", nonce="243b35d1", uri="sip:192.9.200.9:5060", response="11f3478d812d35993018150f29fb5e81", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.202.2:5060>;expires=3600 Date: Thu, 06 Jan 2005 06:46:36 GMT Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 13 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as77cf03d0 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:100@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@192.9.202.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060> Contact: <sip:asterisk@192.9.200.9> Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 192.9.202.2:5060 Scheduling destruction of call '23c9fa48037fec98416d74650481661e@192.9.200.9' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7d0b7e8a", uri="sip:100@192.9.200.9:5060", response="38d9b121d4ee361e584727823f195810", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 100 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as5a181df1 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:100@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B Contact: <sip:100@192.9.200.9> Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 339 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:101@192.9.200.9;method=SUBSCRIBE" /> <atom id="100"> <address uri="sip:100@192.9.200.9;user=ip" priority="0,800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060>;tag=81B3E4D0-500C78DF CSeq: 102 NOTIFY Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 Contact: <sip:101@192.9.202.2:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Destroying call '23c9fa48037fec98416d74650481661e@192.9.200.9' asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B CSeq: 102 NOTIFY Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Message is NOTIFY asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as1d96eff1 Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7735e16f", uri="sip:101@192.9.200.9:5060", response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 101 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE Contact: <sip:101@192.9.200.9> Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:101@192.9.200.9"> <dialog id="101"> <state>confirmed</state> </dialog> </dialog-info> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE CSeq: 102 NOTIFY Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Destroying call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
They were updated, to reflect the new card. And I can call in perfectly. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays "Line used remotely" and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid="Tim Jackson" <101> secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang="") 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 1 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 2 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username="101", realm="angelinacounty.net", nonce="243b35d1", uri="sip:192.9.200.9:5060", response="11f3478d812d35993018150f29fb5e81", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.202.2:5060>;expires=3600 Date: Thu, 06 Jan 2005 06:46:36 GMT Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 13 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as77cf03d0 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:100@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@192.9.202.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060> Contact: <sip:asterisk@192.9.200.9> Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 192.9.202.2:5060 Scheduling destruction of call '23c9fa48037fec98416d74650481661e@192.9.200.9' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7d0b7e8a", uri="sip:100@192.9.200.9:5060", response="38d9b121d4ee361e584727823f195810", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 100 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as5a181df1 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:100@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B Contact: <sip:100@192.9.200.9> Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 339 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:101@192.9.200.9;method=SUBSCRIBE" /> <atom id="100"> <address uri="sip:100@192.9.200.9;user=ip" priority="0,800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060>;tag=81B3E4D0-500C78DF CSeq: 102 NOTIFY Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 Contact: <sip:101@192.9.202.2:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Destroying call '23c9fa48037fec98416d74650481661e@192.9.200.9' asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B CSeq: 102 NOTIFY Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Message is NOTIFY asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as1d96eff1 Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7735e16f", uri="sip:101@192.9.200.9:5060", response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 101 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE Contact: <sip:101@192.9.200.9> Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:101@192.9.200.9"> <dialog id="101"> <state>confirmed</state> </dialog> </dialog-info> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE CSeq: 102 NOTIFY Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Destroying call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FROM MY SIP.CONF [1000] type=friend host=dynamic context=local allow=ulaw secret=YESITIS callerid="Front Desk" <1000> mailbox=1000@sip dtmfmode=rfc2833 nat=0 FROM MY EXTENSION.CONF [local] include => mainmenu include => parkedcalls include => trunklocal include => trunktollfree include => trunkld include => trunkint include => sip YOURS sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all May as well just set allow=ulaw unless you are eally using something else. Does your extensions.conf have a context default which is set up with something like... [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9XXXXXXX,2,Congestion exten => _9480NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9480NXXXXXX,2,Congestion exten => _9602NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9602NXXXXXX,2,Congestion exten => _9623NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9623NXXXXXX,2,Congestion Where TRUNK is passed in from a global? MINE GLOBALS ;Trunk Info TRUNK=ZAP/g1 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) On a guess, it seems like your context for incoming could be correct and your context for out may be wrong. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 7:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 They were updated, to reflect the new card. And I can call in perfectly. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays "Line used remotely" and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid="Tim Jackson" <101> secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang="") 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 1 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 2 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username="101", realm="angelinacounty.net", nonce="243b35d1", uri="sip:192.9.200.9:5060", response="11f3478d812d35993018150f29fb5e81", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.202.2:5060>;expires=3600 Date: Thu, 06 Jan 2005 06:46:36 GMT Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 13 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as77cf03d0 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:100@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@192.9.202.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060> Contact: <sip:asterisk@192.9.200.9> Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 192.9.202.2:5060 Scheduling destruction of call '23c9fa48037fec98416d74650481661e@192.9.200.9' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7d0b7e8a", uri="sip:100@192.9.200.9:5060", response="38d9b121d4ee361e584727823f195810", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 100 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as5a181df1 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:100@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B Contact: <sip:100@192.9.200.9> Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 339 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:101@192.9.200.9;method=SUBSCRIBE" /> <atom id="100"> <address uri="sip:100@192.9.200.9;user=ip" priority="0,800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060>;tag=81B3E4D0-500C78DF CSeq: 102 NOTIFY Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 Contact: <sip:101@192.9.202.2:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Destroying call '23c9fa48037fec98416d74650481661e@192.9.200.9' asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B CSeq: 102 NOTIFY Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Message is NOTIFY asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as1d96eff1 Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7735e16f", uri="sip:101@192.9.200.9:5060", response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 101 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE Contact: <sip:101@192.9.200.9> Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:101@192.9.200.9"> <dialog id="101"> <state>confirmed</state> </dialog> </dialog-info> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE CSeq: 102 NOTIFY Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Destroying call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
This isn't a dialplan issue, it's a SIP issue. The same dialplan and sip.conf are working perfectly with the other server. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 FROM MY SIP.CONF [1000] type=friend host=dynamic context=local allow=ulaw secret=YESITIS callerid="Front Desk" <1000> mailbox=1000@sip dtmfmode=rfc2833 nat=0 FROM MY EXTENSION.CONF [local] include => mainmenu include => parkedcalls include => trunklocal include => trunktollfree include => trunkld include => trunkint include => sip YOURS sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all May as well just set allow=ulaw unless you are eally using something else. Does your extensions.conf have a context default which is set up with something like... [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9XXXXXXX,2,Congestion exten => _9480NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9480NXXXXXX,2,Congestion exten => _9602NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9602NXXXXXX,2,Congestion exten => _9623NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9623NXXXXXX,2,Congestion Where TRUNK is passed in from a global? MINE GLOBALS ;Trunk Info TRUNK=ZAP/g1 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) On a guess, it seems like your context for incoming could be correct and your context for out may be wrong. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 7:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 They were updated, to reflect the new card. And I can call in perfectly. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays "Line used remotely" and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid="Tim Jackson - Home" <101> secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid="Tim Jackson" <101> secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang="") 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 1 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9> CSeq: 2 REGISTER Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 Contact: <sip:101@192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username="101", realm="angelinacounty.net", nonce="243b35d1", uri="sip:192.9.200.9:5060", response="11f3478d812d35993018150f29fb5e81", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=36767043-B9FDB2DA To: <sip:101@192.9.200.9>;tag=as024fe72d Call-ID: d8038d0f-22c84c59-3f42a480@192.9.202.2 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.202.2:5060>;expires=3600 Date: Thu, 06 Jan 2005 06:46:36 GMT Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 13 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as77cf03d0 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:100@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@192.9.202.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060> Contact: <sip:asterisk@192.9.200.9> Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 192.9.202.2:5060 Scheduling destruction of call '23c9fa48037fec98416d74650481661e@192.9.200.9' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:100@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7d0b7e8a", uri="sip:100@192.9.200.9:5060", response="38d9b121d4ee361e584727823f195810", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 100 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B To: <sip:100@192.9.200.9>;tag=as5a181df1 Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:100@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B Contact: <sip:100@192.9.200.9> Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 339 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:101@192.9.200.9;method=SUBSCRIBE" /> <atom id="100"> <address uri="sip:100@192.9.200.9;user=ip" priority="0,800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport From: "asterisk" <sip:asterisk@192.9.200.9>;tag=as00270f99 To: <sip:101@192.9.202.2:5060>;tag=81B3E4D0-500C78DF CSeq: 102 NOTIFY Call-ID: 23c9fa48037fec98416d74650481661e@192.9.200.9 Contact: <sip:101@192.9.202.2:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Destroying call '23c9fa48037fec98416d74650481661e@192.9.200.9' asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport From: <sip:100@192.9.200.9>;tag=as5a181df1 To: "Tim Jackson" <sip:101@192.9.200.9>;tag=A5DE6FC-D938162B CSeq: 102 NOTIFY Call-ID: 70bce7a8-79a1e882-74df3bc1@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Message is NOTIFY asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 1 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Found peer '101' Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as1d96eff1 Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f" Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 15000 ms asterisk*CLI> Sip read: SUBSCRIBE sip:101@192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9> CSeq: 2 SUBSCRIBE Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Accept: application/simple-message-summary Proxy-Authorization: Digest username="101", realm="angelinacounty.net", nonce="7735e16f", uri="sip:101@192.9.200.9:5060", response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 : 5060 (NAT) Found peer '101' Looking for 101 in default Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.9.200.9> Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo rt=5060 From: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE To: <sip:101@192.9.200.9>;tag=as2d05161a Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:101@192.9.200.9>;expires=3600 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '3c57a613-cc4f559d-1ed53124@192.9.202.2' in 3610000 ms Reliably Transmitting: NOTIFY sip:101@192.9.200.9 SIP/2.0 Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE Contact: <sip:101@192.9.200.9> Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:101@192.9.200.9"> <dialog id="101"> <state>confirmed</state> </dialog> </dialog-info> (NAT) to 192.9.202.2:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport From: <sip:101@192.9.200.9>;tag=as2d05161a To: "Tim Jackson" <sip:101@192.9.200.9>;tag=B0756DC7-622969BE CSeq: 102 NOTIFY Call-ID: 3c57a613-cc4f559d-1ed53124@192.9.202.2 Contact: <sip:101@192.9.202.2:5060> Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Destroying call 'd8038d0f-22c84c59-3f42a480@192.9.202.2' Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Asterisk CVS-v1-0-01/06/05-00:11:36 built by root@asterisk on a i686 running Linux [channels] echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=inbound-pots signalling=fxs_ks callerid="Unknown Caller" <> group = 1 channel => 1-2 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=noawnser signalling=fxs_ks callerid="Unknown caller" <> group = 1 channel => 3-4 -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, For what it's worth, from my working sip.conf for Polycoms: [2010] type=friend username=usr2010 callerid="MyName" <2010> secret=nobodyknowswhatitis host=dynamic dtmfmode=inband context=admin defaultip=192.168.1.10 progressinband=no Notes: dtmfmode=inband and progressinband=no - that seems to be recommended from * sample sip.conf file for Polycoms. defaultip= setting helped with network issues, not only with Polycoms, with Cisco 7940 as well. Also in main sip.conf: [general] ... disallow=all ; Allow all codecs allow=ulaw,alaw maxexpirey=7200 defaultexpirey=3600 canreinvite=no Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what is your network infrastructure? Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from October 2004). And of course: what is Asterisk and zaptel version? What is your zapata.conf (just curious)? Andrei Tim Jackson wrote:>Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone >cards to a 1U IBM server with a TDM04B card. I finally got the card >working in the server, but I'm having issues with these Polycom IP500s >now. Using the exact same config from the old server I'm getting weird >errors. Dial a number on the phone and it gives you dialtone but nouser>interaction (if that makes sense) then after about 35-40 seconds it >displays "Line used remotely" and hangs up. Inbound calls ring, but you >can't answer them, registration seems to be ok, but I'm at a loss. > >sip.conf: >[101] >type=friend >callerid="Tim Jackson - Home" <101> >secret=itsasekret >username=101 >host=dynamic >dtmfmode=rfc2833 >nat=yes >canreinvite=no >context=default >allow=all > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
That's what I'm about to try, I keep getting pulled off of this project to go do other things. Thanks for the input. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim Jackson wrote:>Copied your sip.conf and changed the settings and I'm getting the exact >same error. I'm also running 1.3.4 of the SIP app for the IP500. > >Someone has already pointed out that you might have ran into a network problem. What's the network setup between phone and the server?>Asterisk CVS-v1-0-01/06/05-00:11:36 built by root@asterisk on a i686 > > >I was unable to use Asterisk from latest CVS, I am using version from 12/02 CVS. I was getting authorization failed in CLI, and phone could not make calls with CVS-latest Asterisk. Might be something similar in your setup? Just copy /usr/src/asterisk from old server and try make install.. Please, someone, comment on latest changes in CVS for SIP configurations? Might enforced md5 passwords etc? Or anything like that?>context=noawnser > >A typo, right? Andrei _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users