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 (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] <[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] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965> > set_destination: Parsing > <"p:[EMAIL PROTECTED] <[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] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > Contact: <sip:[EMAIL PROTECTED] <[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] <[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] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[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] <[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] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > Contact: <sip:[EMAIL PROTECTED] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To:<sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > CSeq:103 BYE > 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/2.0 416 Unsupported URI Scheme > Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport > From:"512" <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To:<sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > CSeq:104 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) --- > set_destination: Parsing > <"p:[EMAIL PROTECTED] <[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] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > Contact: <sip:[EMAIL PROTECTED] <[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] <[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) --- > Retransmitting #1 (no NAT) to 172.16.1.174:5060: > BYE "p:[EMAIL PROTECTED] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To:<sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > CSeq:103 BYE > 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 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] <[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) --- > Retransmitting #2 (no NAT) to 172.16.1.174:5060: > BYE "p:[EMAIL PROTECTED] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To: <sip:[EMAIL PROTECTED] <[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] <[EMAIL PROTECTED]>>;tag=as7ec9e8af > To:<sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > >;tag=4881ea36-2ca-6747d965 > CSeq:103 BYE > User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 > Call-ID:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > Content-Length:0 > > > <-------------> > --- (8 headers 0 lines) --- > greybeamPBX*CLI> sip set debug off > > > 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_______________________________________________ -- 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