Von L.
2006-Jun-28 09:04 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-30000 UDP. Once they plug the phone it (power and ethernet) I see on the CLI console of the asterisk server that the phones register: Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on bell (pid = 3652) nell*CLI> Verbosity is at least 10 -- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600 Here is the top part of my sip.conf ;_____________________________________________________________ ;sip.conf ;_____________________________________________________________ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes I know it has something to do with the NAT because if I plug my Polycom directly into my cable modem, thus making it sit on the Internet and have a real IP, everything works just fine. I am curious what I am missing. Thanks. Von L.
Tom Vile
2006-Jun-28 09:17 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
You have to lower the registration interval in the phones to under a minute otherwise the NAT hole closes and no calls come in. Polycom has said that they are going to be putting in a keep alive in the firmware at some point. On 6/28/06, Von L. <methodvon@gmail.com> wrote:> Hello, > > Here is a breakdown of the issue I am experiencing. I have three remote > employees, in various states, who have Polycom 501 phones. They are > unable to receive incoming calls after a few minutes of the phones being > plugged in. They work immediately after being plugged in, but they lose > the ability shortly thereafter. They can always make outbound calls, but > only to real phone numbers, not extensions. > > They each have NAT routers, and I have triple checked that they have > opened/forwarded the correct ports, basically 5060-30000 UDP. Once they > plug the phone it (power and ethernet) I see on the CLI console of the > asterisk server that the phones register: > > Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. > Written by Mark Spencer <markster@digium.com> > ========================================================================> Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on > bell (pid = 3652) > nell*CLI> > Verbosity is at least 10 > -- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600 > > Here is the top part of my sip.conf > > ;_____________________________________________________________ > ;sip.conf > ;_____________________________________________________________ > > [general] > port=5060 > bindaddr=0.0.0.0 > externip=XXX.XXX.XXX.XXX > localnet=XXX.XXX.XXX.XXX/255.255.255.248 > canreinvite=no > tos=reliability > srvlookup=yes > disallow=all > allow=ulaw > dtmfmode=rfc2833 > nat=yes > ignoreregexpire=yes > > I know it has something to do with the NAT because if I plug my Polycom > directly into my cable modem, thus making it sit on the Internet and > have a real IP, everything works just fine. > > I am curious what I am missing. > > Thanks. > > Von L. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856
Michiel van Baak
2006-Jun-28 09:19 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
On 12:04, Wed 28 Jun 06, Von L. wrote:> Hello, > ;_____________________________________________________________ > ;sip.conf > ;_____________________________________________________________ > > [general] > port=5060 > bindaddr=0.0.0.0 > externip=XXX.XXX.XXX.XXX > localnet=XXX.XXX.XXX.XXX/255.255.255.248 > canreinvite=no > tos=reliability > srvlookup=yes > disallow=all > allow=ulaw > dtmfmode=rfc2833 > nat=yes > ignoreregexpire=yesShow us one of the phone entries. Basically check if the following is set there: nat=yes qualify=yes The qualify=yes will send packets so the nat states stay open. -- Michiel van Baak michiel@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?"
Michael Graves
2006-Jun-28 09:32 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Sounds like the registration interval in the phones is less than the required registration interval of the server. I had this occur when using a SIP phone with an ITSP. Michael On Wed, 28 Jun 2006 12:04:40 -0400, Von L. wrote:>Hello,>Here is a breakdown of the issue I am experiencing. I have three remote >employees, in various states, who have Polycom 501 phones. They are >unable to receive incoming calls after a few minutes of the phones being >plugged in. They work immediately after being plugged in, but they lose >the ability shortly thereafter. They can always make outbound calls, but >only to real phone numbers, not extensions.>They each have NAT routers, and I have triple checked that they have >opened/forwarded the correct ports, basically 5060-30000 UDP. Once they >plug the phone it (power and ethernet) I see on the CLI console of the >asterisk server that the phones register:>Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. >Written by Mark Spencer <markster@digium.com> >========================================================================>Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on >bell (pid = 3652) >nell*CLI> >Verbosity is at least 10 >-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600>Here is the top part of my sip.conf>;_____________________________________________________________ >;sip.conf >;_____________________________________________________________>[general] >port=5060 >bindaddr=0.0.0.0 >externip=XXX.XXX.XXX.XXX >localnet=XXX.XXX.XXX.XXX/255.255.255.248 >canreinvite=no >tos=reliability >srvlookup=yes >disallow=all >allow=ulaw >dtmfmode=rfc2833 >nat=yes >ignoreregexpire=yes>I know it has something to do with the NAT because if I plug my Polycom >directly into my cable modem, thus making it sit on the Internet and >have a real IP, everything works just fine.>I am curious what I am missing.>Thanks.>Von L. >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -->Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/5045b975/attachment.htm
Doug Lytle
2006-Jun-28 09:39 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to
Von L. wrote:> Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. > Written by Mark Spencer <markster@digium.com> > ========================================================================> Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on >I would suggest you upgrade your Asterisk. This is VERY outdated and CVS to boot! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Tom Vile
2006-Jun-28 09:54 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
FYI, when we had NAT routers at both locations setting qualify=yes did not work. On 6/28/06, Michiel van Baak <michiel@vanbaak.info> wrote:> On 12:04, Wed 28 Jun 06, Von L. wrote: > > Hello, > > ;_____________________________________________________________ > > ;sip.conf > > ;_____________________________________________________________ > > > > [general] > > port=5060 > > bindaddr=0.0.0.0 > > externip=XXX.XXX.XXX.XXX > > localnet=XXX.XXX.XXX.XXX/255.255.255.248 > > canreinvite=no > > tos=reliability > > srvlookup=yes > > disallow=all > > allow=ulaw > > dtmfmode=rfc2833 > > nat=yes > > ignoreregexpire=yes > > Show us one of the phone entries. > Basically check if the following is set there: > nat=yes > qualify=yes > > The qualify=yes will send packets so the nat states stay open. > -- > Michiel van Baak > michiel@vanbaak.eu > http://michiel.vanbaak.eu > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD > > "Why is it drug addicts and computer afficionados are both called users?" > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856
Dr. Michael J. Chudobiak
2006-Jun-28 10:29 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote:> plugged in. They work immediately after being plugged in, but they lose > the ability shortly thereafter. They can always make outbound calls, but > only to real phone numbers, not extensions. > > They each have NAT routers, and I have triple checked that they have > opened/forwarded the correct ports, basically 5060-30000 UDP. Once theySee the "NAT Issues" section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP ISP ports too). Basically, some NAT routers "forget" UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike
Dr. Michael J. Chudobiak
2006-Jun-28 10:30 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote:> plugged in. They work immediately after being plugged in, but they lose > the ability shortly thereafter. They can always make outbound calls, but > only to real phone numbers, not extensions. > > They each have NAT routers, and I have triple checked that they have > opened/forwarded the correct ports, basically 5060-30000 UDP. Once theySee the "NAT Issues" section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP SIP ports too). Basically, some NAT routers "forget" UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike
Cullin J. Wible
2006-Jun-28 12:30 UTC
[Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes.
Polycom phones support STUN - that should solve the issue too. Cullin -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dr. Michael J. Chudobiak Sent: Wednesday, June 28, 2006 1:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes. Von L. wrote:> plugged in. They work immediately after being plugged in, but they > lose the ability shortly thereafter. They can always make outbound > calls, but only to real phone numbers, not extensions. > > They each have NAT routers, and I have triple checked that they have > opened/forwarded the correct ports, basically 5060-30000 UDP. Once > theySee the "NAT Issues" section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP SIP ports too). Basically, some NAT routers "forget" UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Greg Kennedy
2006-Jun-28 22:28 UTC
[Asterisk-Users] RE: Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
It is totally nat, first try to port map the ports through your firewall, on the network page set the rtp and sip ports plus the nat ip to use. I had the exact same problem and this was the only solution. Or add the following to your config for the phone: nat.mediaPortStart="5004" nat.signalPort="5060" nat.ip="your.public.ip.address" just change the media and signalport for each phone, and set it to what you port map in your firewall. I tried to do the qualify deal and it only works for a minute or two as well, so just use the port mapping. If only Polycom would pull their collective heads from their rears and give us stun we wouldn't need this bs!You wrote:>Sounds like the registration interval in the phones is less than the required registration interval of the server. I had this occur when using a SIP phone with an ITSP. >Michael >On Wed, 28 Jun 2006 12:04:40 -0400, Von L. wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/72953627/attachment.htm