mattf
2004-Feb-04 04:59 UTC
[Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500
What firmware and sip versions are you using? I have several Polycom phones on my system right now and I've never had any registration problems with them. Instead of leaving the host as dynamic try declaring an IP address(that's the only difference I see between your sip.conf and mine). If you are still having problems I've like to see your polycom .cfg files for one of these phones, you might be missing a setting in one of them. MATT--- -----Original Message----- From: David Liu [mailto:dtliu@scu.edu] Sent: Wednesday, February 04, 2004 1:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500 We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user "DavidLiu" <sip:DavidLiu@192.168.0.254>;tag=9F67E426-59D92ED7 Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user "DavidLiu" <sip:DavidLiu@192.168.0.254>;tag=BFDEF35B-1CBC4F2C in sip.conf: canreinvite=yes host=dynamic canreinvite=yes dtmfmode=rfc2833 context=sip port=5060 Usually say after the phone failed to register with Asterisk, I can attempt to place a call. It will fail of course. But then I can try calling again and usually the call will go through and it will successfully re-register itself without needing a restart. What can this be? Surely Polycom is re-registering every 3600 before Asterisk times it out. But Asterisk is just refusing it. By the way, anyone know whether Asterisk is geared towards RFC3261 or RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, will it work better or the same with Asterisk? David
David Liu
2004-Feb-04 18:56 UTC
[Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500
Hi Matt, I did try setting my sip.conf to have host=ip.address. such as the following: [DavidLiu] type=friend username=DavidLiu secret=mypassword host=192.168.3.16 canreinvite=yes dtmfmode=rfc2833 context=sip callerid="David Liu" <1000> mailbox=1000 port=5060 Then asterisk will complain with the following error: Feb 5 09:52:01 NOTICE[278546]: Registration from '"DavidLiu" <sip:DavidLiu@192.168.0.254>' failed for '192.168.3.16' Feb 5 09:52:32 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic Feb 5 09:52:32 NOTICE[278546]: Registration from '"DavidLiu" <sip:DavidLiu@192.168.0.254>' failed for '192.168.3.16' Feb 5 09:52:33 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic etc......(repeat until phone stops registering) David> ----- Original Message -----From: "mattf" <mattf@vicimarketing.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, February 04, 2004 3:59 AM Subject: RE: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500> What firmware and sip versions are you using? I have several Polycomphones> on my system right now and I've never had any registration problems with > them. > > Instead of leaving the host as dynamic try declaring an IP address(that's > the only difference I see between your sip.conf and mine). > > If you are still having problems I've like to see your polycom .cfg files > for one of these phones, you might be missing a setting in one of them. > > MATT--- > > > -----Original Message----- > From: David Liu [mailto:dtliu@scu.edu] > Sent: Wednesday, February 04, 2004 1:06 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Minor Registration Problem With PolycomSoundpoint> IP 500 > > > We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk > environment. So far it has been good. Call Hold, Transfer, DMTF etc. > > However, I do notice every now and then the Polycom fails to register with > Asterisk. Asterisk console outputs the following: > > Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response:Unable> to determine sequence number from '' > Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to > authenticate user "DavidLiu" > <sip:DavidLiu@192.168.0.254>;tag=9F67E426-59D92ED7 > Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to > authenticate user "DavidLiu" > <sip:DavidLiu@192.168.0.254>;tag=BFDEF35B-1CBC4F2C > > in sip.conf: > canreinvite=yes > host=dynamic > canreinvite=yes > dtmfmode=rfc2833 > context=sip > port=5060 > > Usually say after the phone failed to register with Asterisk, I canattempt> to place a call. It will fail of course. But then I can try callingagain> and usually the call will go through and it will successfully re-register > itself without needing a restart. > > What can this be? Surely Polycom is re-registering every 3600 before > Asterisk times it out. But Asterisk is just refusing it. > > By the way, anyone know whether Asterisk is geared towards RFC3261 or > RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP > PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, > will it work better or the same with Asterisk? > > David > > _______________________________________________ > 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
David Liu
2004-Feb-04 18:58 UTC
[Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500
By the way I am using BootRom 2.4.1 and 1.1.0 SIP Would you want me to send all my CFG files to you? David
mattf
2004-Feb-04 20:19 UTC
[Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500
try this first in your sip.conf entry for your Polycom phone: host=dynamic defaultip=10.10.10.10 (put the phone IP address there) I have all of my Polycom's set to friend so I know that's not your problem. if that is still generating bad registration messages, then send me your Polycom .cfg files MATT--- -----Original Message----- From: David Liu [mailto:dtliu@scu.edu] Sent: Wednesday, February 04, 2004 8:57 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500 Hi Matt, I did try setting my sip.conf to have host=ip.address. such as the following: [DavidLiu] type=friend username=DavidLiu secret=mypassword host=192.168.3.16 canreinvite=yes dtmfmode=rfc2833 context=sip callerid="David Liu" <1000> mailbox=1000 port=5060 Then asterisk will complain with the following error: Feb 5 09:52:01 NOTICE[278546]: Registration from '"DavidLiu" <sip:DavidLiu@192.168.0.254>' failed for '192.168.3.16' Feb 5 09:52:32 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic Feb 5 09:52:32 NOTICE[278546]: Registration from '"DavidLiu" <sip:DavidLiu@192.168.0.254>' failed for '192.168.3.16' Feb 5 09:52:33 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic etc......(repeat until phone stops registering) David> ----- Original Message -----From: "mattf" <mattf@vicimarketing.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, February 04, 2004 3:59 AM Subject: RE: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500> What firmware and sip versions are you using? I have several Polycomphones> on my system right now and I've never had any registration problems with > them. > > Instead of leaving the host as dynamic try declaring an IP address(that's > the only difference I see between your sip.conf and mine). > > If you are still having problems I've like to see your polycom .cfg files > for one of these phones, you might be missing a setting in one of them. > > MATT--- > > > -----Original Message----- > From: David Liu [mailto:dtliu@scu.edu] > Sent: Wednesday, February 04, 2004 1:06 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Minor Registration Problem With PolycomSoundpoint> IP 500 > > > We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk > environment. So far it has been good. Call Hold, Transfer, DMTF etc. > > However, I do notice every now and then the Polycom fails to register with > Asterisk. Asterisk console outputs the following: > > Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response:Unable> to determine sequence number from '' > Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to > authenticate user "DavidLiu" > <sip:DavidLiu@192.168.0.254>;tag=9F67E426-59D92ED7 > Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to > authenticate user "DavidLiu" > <sip:DavidLiu@192.168.0.254>;tag=BFDEF35B-1CBC4F2C > > in sip.conf: > canreinvite=yes > host=dynamic > canreinvite=yes > dtmfmode=rfc2833 > context=sip > port=5060 > > Usually say after the phone failed to register with Asterisk, I canattempt> to place a call. It will fail of course. But then I can try callingagain> and usually the call will go through and it will successfully re-register > itself without needing a restart. > > What can this be? Surely Polycom is re-registering every 3600 before > Asterisk times it out. But Asterisk is just refusing it. > > By the way, anyone know whether Asterisk is geared towards RFC3261 or > RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP > PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, > will it work better or the same with Asterisk? > > David > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users