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Tim Jackson

2004-Sep-28 14:31 UTC

[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
 
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Wiley E. Siler

2004-Sep-28 14:34 UTC

head link

[Asterisk-Users] Polycom IP500

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

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Wiley E. Siler

2004-Sep-28 15:21 UTC

head link

[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 


	

	
		
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Tim Jackson

2004-Sep-28 17:18 UTC

head link

[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

_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Wiley E. Siler

2004-Sep-29 07:18 UTC

head link

[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

_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
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Tim Jackson

2004-Sep-29 07:21 UTC

head link

[Asterisk-Users] Polycom IP500

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

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Tim Jackson

2004-Sep-30 14:38 UTC

head link

[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
 
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Wiley E. Siler

2004-Sep-30 14:45 UTC

head link

[Asterisk-Users] Polycom IP500

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

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Tim Jackson

2004-Dec-01 17:38 UTC

head link

[Asterisk-Users] Polycom IP500

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
>
>  
>
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CAUTION: This email message and accompanying data may contain
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error, please notify us immediately and erase all copies of this message
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CAUTION: This email message and accompanying data may contain
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you are notified that any use, dissemination, distribution or copying of
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Tim Jackson

2004-Dec-02 08:06 UTC

head link

[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 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 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 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
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Ty Purcell

2004-Dec-02 08:33 UTC

head link

[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 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 to
>>    
>>
>find
>one.
>  
>
>>Extra Info:
>>Server is 192.168.0.3
>>Phone name/ext I want to be 301
>>
>>Thanks,
>>Chris Cherry
>>
>> 
>>
>>    
>>
>
>_______________________________________________
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>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
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Tim Jackson

2004-Dec-02 08:57 UTC

head link

[Asterisk-Users] Polycom IP500

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 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 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 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.
>_______________________________________________
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>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
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Tim Jackson

2004-Dec-02 10:18 UTC

head link

[Asterisk-Users] Polycom IP500

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.p
d>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 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
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 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
>>
>>_______________________________________________
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>>To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
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>>
>> 
>>
>>    
>>
>
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Adam Robins

2004-Dec-07 06:49 UTC

head link

[Asterisk-Users] Polycom IP500

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

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Tim Jackson

2005-Jan-06 00:08 UTC

head link

[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
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

Wiley Siler

2005-Jan-06 01:36 UTC

head link

[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

_______________________________________________
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Tim Jackson

2005-Jan-06 07:54 UTC

head link

[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

_______________________________________________
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Wiley Siler

2005-Jan-06 08:12 UTC

head link

[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

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Tim Jackson

2005-Jan-06 08:19 UTC

head link

[Asterisk-Users] Polycom IP500

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

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Tim Jackson

2005-Jan-06 14:17 UTC

head link

[Asterisk-Users] Polycom IP500

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 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
>  
>
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Tim Jackson

2005-Jan-07 08:41 UTC

head link

[Asterisk-Users] Polycom IP500

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

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asterisk users - Sep 2004 - Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

[Asterisk-Users] Polycom IP500

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