[asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Oh... this does not happen all of the time, maybe 50%. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' or [Jul 19 10:45:03] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '172.16.1.20;tag=as4a1b11c8' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:45:03] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: can't resolve in DNS) : '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' or [Jul 19 10:52:18] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'nt-Length:0' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- SIP/517-09215fb0 answered SIP/512-09258c78 -- Native bridging SIP/512-09258c78 and SIP/517-09215fb0 [Jul 19 10:52:18] WARNING[22054]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- Got SIP response 416 Unsupported URI Scheme back from 172.16.1.157 [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' So it seems that the Mitel phone is sending a bad contact field in SIP. I've confirmed via tcpdump that this is what's in the SIP packet on the wire. I wanted to try a different version of SIP on the Mitel but that doesn't seem to be an option, it's not available for download and the local Mitel vendor can't seem to get his hands on anything newer than 6.0.0.something, though there is supposedly 7.1.x available. These phones are running 06.00.00.19. The Asterisk server has a pretty standard sip.conf, bindaddr=0.0.0.0 pedantic=no; bindport=5060 srvlookup=no tos_video=af41 notifyringing=yes notifyhold=yes allowsubscribe=yes limitonpeer=yes localnet=172.16.1.0/255.255.255.0 Polycom phones on this same asterisk server do not display this behavior. I'm wondering if there is a workaround for this apparent Mitel issue in Asterisk's configuration. Anyone using this combination with success? Thanks in advance for any thoughts Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Matt Watson wrote: On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. Thanks Matt I was also able to test this with Mitel's firmware version 7.0.0.8 with the same results. Mitel phone still acts like it's on a call, Asterisk does not nor does the originating phone. PBX*CLI sip set debug peer 517 SIP Debugging Enabled for IP: 172.16.1.174:5060 Audio is at 172.16.1.20 port 15594 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.1.174:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport From: 512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 19 Jul 2008 17:20:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 2247 2247 IN IP4 172.16.1.20 s=session c=IN IP4 172.16.1.20 t=0 0 m=audio 15594 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Content-Length:0 - --- (8 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 180 Ringing Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Content-Length:0 - --- (9 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 200 OK Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Contact:p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User sip:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE Supported:timer,100rel,replaces Content-Type:application/sdp Content-Length:182 v=0 o=517 1216473942 1216473941 IN IP4 172.16.1.174 s=SIP Call c=IN IP4 172.16.1.174 t=0 0 m=audio 20012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.16.1.174:20012 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.1.174:20012 [Jul 19 13:20:56] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 13:20:56] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '172.16.1.174' list_route: hop: p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 set_destination: Parsing p:[EMAIL
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Maybe stupid solution but, when Mitel phone i called, why dont you pickup put the person on hold, call Mitel phone, and connect them, what i want to say, add some delay. 2008/7/19 Mark Wiater [EMAIL PROTECTED]: Matt Watson wrote: On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. Thanks Matt I was also able to test this with Mitel's firmware version 7.0.0.8 with the same results. Mitel phone still acts like it's on a call, Asterisk does not nor does the originating phone. PBX*CLI sip set debug peer 517 SIP Debugging Enabled for IP: 172.16.1.174:5060 Audio is at 172.16.1.20 port 15594 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.1.174:5060: INVITE sip:[EMAIL PROTECTED] [EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport From: 512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 19 Jul 2008 17:20:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 2247 2247 IN IP4 172.16.1.20 s=session c=IN IP4 172.16.1.20 t=0 0 m=audio 15594 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Content-Length:0 - --- (8 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 180 Ringing Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Allow-Events:talk,hold,conference Content-Length:0 - --- (9 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 200 OK Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Contact:p:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Allow-Events:talk,hold,conference Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE Supported:timer,100rel,replaces Content-Type:application/sdp Content-Length:182 v=0 o=517 1216473942 1216473941 IN IP4 172.16.1.174 s=SIP Call c=IN IP4 172.16.1.174 t=0 0 m=audio 20012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.16.1.174:20012 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
2008/7/19 Mark Wiater [EMAIL PROTECTED]: Your problem is the Contact header coming back in the Mitel's Ok response. Contact:p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 Asterisk is not going to be able to parse that! It should be: Contact: sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 I'd get your Mitel vendor to log a fault about that. It's a big problem as the Contact field is critical for further in-dialogue SIP requests such as transfer related requests and BYEs. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
