Re: [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
I'm using P0S3-8-12-00 and things are working great with piaf and asterisk 1.4. Drop me a direct line to email, and I can send you my configs and such if that would help diag things for you. On Feb 3, 2010 3:02 PM, i...@comtek.co.uk i...@comtek.co.uk wrote: David Gibbons wrote: snip I have upgraded the phones to the most recent firmware (POS3-08-11-0... Thats straight out of that section. Its the most recent SIP firmware I could find. Application Load ID: 'POS3-08-11-00'. Boot Load ID: PC03A300 DSP Load ID 4.0(5.0)[A0] It seems to mean 8.11. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/firmware/sip/8_11/english/release/notes/796040sip_811.html#wp1099767 Thanks, Ian -- === Ian Crowther ... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list T... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
Hi, Our Cisco 7940 phones on a single network sometimes seem to drop calls as soon as they are picked up. After a second INVITE the phone sends '500 Internal Error'. One phone thinks its still in a call (though there is no audio) while the other phone is not in a call. The drop happens immediately after connecting and I think it is due to a second INVITE. Turning reinvite off seems like it fixes the problem but I haven't tested enough to be sure yet. The bug only seems to happen on the first time a call is made after a period of time (and even then only occasionally). Successive calls between the same phones are always fine. That makes debugging it a bit of a pain. I have managed to capture a packet trace between * (10.200.4.100) and the 7940 (10.200.4.66) showing the error. 36.437814 10.200.4.100 - 10.200.4.66 SIP/SDP Request: INVITE sip:3...@10.200.4.66:5060;transport=udp, with session description 36.584594 10.200.4.66 - 10.200.4.100 SIP Status: 100 Trying 36.720893 10.200.4.66 - 10.200.4.100 SIP Status: 180 Ringing 43.211744 10.200.4.66 - 10.200.4.100 SIP/SDP Status: 200 OK, with session description 43.212001 10.200.4.100 - 10.200.4.66 SIP Request: ACK sip:3...@10.200.4.66:5060;transport=udp 43.212536 10.200.4.100 - 10.200.4.66 SIP/SDP Request: INVITE sip:3...@10.200.4.66:5060;transport=udp, with session description 43.304295 10.200.4.66 - 10.200.4.100 SIP Status: 500 Internal Server Error 43.304489 10.200.4.100 - 10.200.4.66 SIP Request: ACK sip:3...@10.200.4.66:5060;transport=udp 43.402253 Cisco_76:a5:0f - BroadcastARP Who has 10.200.4.100? Tell 10.200.4.66 43.470520 10.200.4.66 - 10.200.4.100 RTP PT=ITU-T G.711 PCMA, SSRC=0xFA576B7, Seq=2616, Time=414864, Mark There is a description of an error http://lists.iptel.org/pipermail/serdev/2005-November/006344.html which seems similar but since its in 2005 I assume the problem would be fixed by now. If that is the problem, can anybody suggest a workaround? I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). Can anybody offer any advice or suggestions on fixing or debugging this further? If anybody else has encountered the problem and knows of a fix that would be great. Thanks, Ian Crowther -- === Ian Crowther Tel: +44 845 4501626 Unit 108, 10th Avenue, IT Dept, ComtekFax: +44 845 4501627 Zone 3, Deeside Industrial Network Systems UK Ltd Park, CH5 2UA, Flintshire === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
snip I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). snip That doesn't look like cisco firmware to me... Unless I'm mistaken. What version are the phones on? (Settings = Status = Firmware Versions) -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
David Gibbons wrote: snip I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). snip That doesn't look like cisco firmware to me... Unless I'm mistaken. What version are the phones on? (Settings = Status = Firmware Versions) -Dave Thats straight out of that section. Its the most recent SIP firmware I could find. Application Load ID: 'POS3-08-11-00'. Boot Load ID: PC03A300 DSP Load ID 4.0(5.0)[A0] It seems to mean 8.11. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/firmware/sip/8_11/english/release/notes/796040sip_811.html#wp1099767 Thanks, Ian -- === Ian Crowther Tel: +44 845 4501626 Unit 108, 10th Avenue, IT Dept, ComtekFax: +44 845 4501627 Zone 3, Deeside Industrial Network Systems UK Ltd Park, CH5 2UA, Flintshire === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes - and it seems to prevent presence hints from working until the phone is rebooted.. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 26, 2006, at 9:28 AM, Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 ___ --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
RE: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
We're not seeing that behaviour... -Original Message- From: Anthony Rodgers [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 28, 2006 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Yes - and it seems to prevent presence hints from working until the phone is rebooted.. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 26, 2006, at 9:28 AM, Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 ___ --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 ___ --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
[Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while. Happens for us, when you do a transfer via the Polycom's transfer button. Doesn't seem to cause any problems though. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes. It does not seem to cause any problems. Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address DEFANGED_uri=sip:[EMAIL PROTECTED];user=ip DEFANGED_priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
I have been seeing the same errors here with Polycom 501 and 601 phones. Asterisk version is 1.2.9.1 and Polycom SIP version 1.6.3On 6/26/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them?I called Polycom tech support, who where utterly useless.Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6).SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032Reliably Transmitting (no NAT) to xxx.187.128.95:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: sip:[EMAIL PROTECTED];tag=as6fd80d1bTo: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xml Subscription-State: activeContent-Length: 371?xml version=1.0?!DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtdpresence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE /atom id=2944026address uri= sip:[EMAIL PROTECTED];user=ip priority=0.80status status=open /msnsubstatus substatus=online //address/atom /presence-- SIP read from xxx.187.128.95:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: sip:[EMAIL PROTECTED];tag=as6fd80d1bTo: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036Content-Length: 0Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___ --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
RE: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while. Happens for us, when you do a transfer via the Polycom's transfer button. Doesn't seem to cause any problems though. It's bloody annoying though, especially for those type-A's that don't like to see the console cluttered up with junk. :) Doug ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes we have been getting this error message '500 Internal Server' errors back from their Polycom IP-601 (normally IP address). Do not know why. Was able to regenerate the same issue some times, but not all the time. It is not consistent. Symptom: If you have several phones online (10 extns)if for some reason all the phones start to sent the message because several people in the office are transferring and answering new calls and existing calls in a certain manor, after a while the Asterisk reboots, and if at that instance, if you have any lines on park or on hold, all those lines gets dropped, andthenlight gets stuck on the Polycom IP601 phone. The only way you could get rid of this light on the Polycom phone is by rebooting the phones where the lights are stuck (Almost all phones). Symptom regeneration: It happens when a person is talking, then multiple calls come in and then the person tries to transfer the call to some one. If only one or two error message is coming from the IP601 it will not cause any problem. Solution: We do not have any solutions for it yet. Hope that Asterisk or Polycom will come up with a solution/ Patch/ Firmware upgrade soon. If you do find a solution please let us know. Thanks, Ben K. Chennat On 6/26/06, Doug Lytle [EMAIL PROTECTED] wrote: Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while.Happens for us, when you do a transfer via thePolycom's transfer button.Doesn't seem to cause any problems though.Doug--Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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
[Asterisk-Users] 500 Internal Server Error
I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 Internal Server Error back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) The strange thing is that one of the access servers is working fine with the exact same configs in every way. I have moved both routers to the same IOS version which is: IOS (tm) 5400 Software (C5400-JS-M), Version 12.2(2)XB15 I have included a copy of the dial-peers that are specified on the non-functional access server, and I have double checked the configs against the circuitassignments and they are correct. ! dial-peer voice 63201 pots destination-pattern 632 no digit-strip port 1/0:0 ! dial-peer voice 63202 pots destination-pattern 632 no digit-strip port 1/2:0 ! dial-peer voice 63203 pots destination-pattern 632 no digit-strip port 1/3:0 ! dial-peer voice 63204 pots destination-pattern 632 no digit-strip port 1/4:0 ! dial-peer voice 63401 pots destination-pattern 634 no digit-strip port 1/5:0 ! dial-peer voice 63402 pots destination-pattern 634 no digit-strip port 1/6:0 ! dial-peer voice 99701 pots destination-pattern 997 no digit-strip port 1/0:0 ! dial-peer voice 99702 pots destination-pattern 997 no digit-strip port 1/2:0 ! dial-peer voice 99703 pots destination-pattern 997 no digit-strip port 1/3:0 ! dial-peer voice 99704 pots destination-pattern 997 no digit-strip port 1/4:0 ! dial-peer voice 43001 pots destination-pattern 430 no digit-strip port 1/0:0 ! dial-peer voice 43002 pots destination-pattern 430 no digit-strip port 1/2:0 ! dial-peer voice 43003 pots destination-pattern 430 no digit-strip port 1/3:0 ! dial-peer voice 43004 pots destination-pattern 430 no digit-strip port 1/4:0 ! dial-peer voice 67001 pots destination-pattern 670 no digit-strip port 1/0:0 ! dial-peer voice 67002 pots destination-pattern 670 no digit-strip port 1/2:0 ! dial-peer voice 67003 pots destination-pattern 670 no digit-strip port 1/3:0 ! dial-peer voice 67004 pots destination-pattern 670 no digit-strip port 1/4:0 ! sip-ua max-forwards 15 retry invite 10 timers trying 1000 timers expires 30 sip-server ipv4:XXX.XXX.XXX.XXX:5060 no transport tcp ! The following is the debugs I collected from the access server with the problem: 6936: 006932: Dec 29 01:48:05.075: Received: 6937: INVITE sip:[EMAIL PROTECTED] SIP/2.0 6938: Via: SIP/2.0/UDP 65.67.76.41:5060;branch=z9hG4bK72629db3 6939: From: 5462000 sip:[EMAIL PROTECTED];tag=as3e9b26ba 6940: To: sip:[EMAIL PROTECTED] 6941: Contact: sip:[EMAIL PROTECTED] 6942: Call-ID: [EMAIL PROTECTED] 6943: CSeq: 102 INVITE 6944: User-Agent: Asterisk PBX 6945: Date: Wed, 29 Dec 2004 01:47:54 GMT 6946: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 6947: Content-Type: application/sdp 6948: Content-Length: 179 6949: 6950: v=0 6951: o=root 9671 9671 IN IP4 65.67.76.41 6952: s=session 6953: c=IN IP4 65.67.76.41 6954: t=0 0 6955: m=audio 11980 RTP/AVP 0 3 6956: a=rtpmap:0 PCMU/8000 6957: a=rtpmap:3 GSM/8000 6958: a=silenceSupp:off - - - - 6959: 6960: 006933: Dec 29 01:48:05.075: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 65.67.76.41:5060 6961: 006934: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sipSPISipIncomingCall 6962: 006935: Dec 29 01:48:05.075: 0x63CEA7B8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) 6963: 006936: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: act_idle_new_message 6964: 006937: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: Converting TimeZone CST to SIP default timezone = GMT 6965: 006938: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sact_idle_new_message_invite 6966: 006939: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sip_stats_method 6967: 006940: Dec 29 01:48:05.075: sipSPIGetSdpBody : Parse incoming session description 6968: 006941: Dec 29 01:48:05.079: Info: Media ip address/domain name in c line: 65.67.76.41 6969: 6970: 006942: Dec 29 01:48:05.079: sact_idle_new_message_invite: non dial peer leg - using RTP Supported Codecs 6971: 6972: 006943: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 18 6973: 6974: 006944: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 0 6975: 6976: 006945: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 8 6977: 6978: 006946: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 4 6979: 6980: 006947: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 2 6981: 6982: 006948: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 15 6983: 6984: 006949: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP
