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 PROTECTED]>;tag=4881ea36-2ca-6747d965> for address/port to send to set_destination: set destination to 172.16.1.174, port 5060 Transmitting (no NAT) to 172.16.1.174:5060: ACK "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK100c41c2;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing <"p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965> for address/port to send to set_destination: set destination to 172.16.1.174, port 5060 Reliably Transmitting (no NAT) to 172.16.1.174:5060: BYE "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) set_destination: Parsing <"p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965> for address/port to send to set_destination: set destination to 172.16.1.174, port 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 "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 2247 2248 IN IP4 172.16.1.156 s=session c=IN IP4 172.16.1.156 t=0 0 m=audio 2224 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/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From:"512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To:<sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 CSeq:103 BYE 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/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport From:"512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To:<sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 CSeq:104 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Content-Length:0 <-------------> --- (8 headers 0 lines) --- set_destination: Parsing <"p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965> for address/port to send to set_destination: set destination to 172.16.1.174, port 5060 Transmitting (no NAT) to 172.16.1.174:5060: ACK "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- 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) --- Retransmitting #1 (no NAT) to 172.16.1.174:5060: BYE "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- PBX*CLI> <--- SIP read from 172.16.1.174:5060 ---> SIP/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From:"512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To:<sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 CSeq:103 BYE 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 set debug peer 517 <--- 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) --- Retransmitting #2 (no NAT) to 172.16.1.174:5060: BYE "p:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From: "512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To: <sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- PBX*CLI> sip set debug off <--- SIP read from 172.16.1.174:5060 ---> SIP/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport From:"512" <sip:[EMAIL PROTECTED]>;tag=as7ec9e8af To:<sip:[EMAIL PROTECTED]>;tag=4881ea36-2ca-6747d965 CSeq:103 BYE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Content-Length:0 <-------------> --- (8 headers 0 lines) --- greybeamPBX*CLI> sip set debug off _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - 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