[asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 & Mitel SIP phones

Sat Jul 19 14:15:46 CDT 2008

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 <mwiater at cablespeed.com>:

> 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:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>>
> Contact: <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> Contact:"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User" <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>>
> 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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965>
>   set_destination: Parsing
> <"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK100c41c2;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Contact: <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>   set_destination: Parsing
> <"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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
> '2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20' in 32000 ms (Method:
> INVITE)
>   set_destination: Parsing
> <"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Contact: <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:103 BYE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:104 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> Content-Length:0
>
>
> <------------->
>   --- (8 headers 0 lines) ---
>   set_destination: Parsing
> <"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Contact: <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> Contact:"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User" <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>>
> 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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:103 BYE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> Contact:"p:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:102 INVITE
> User" <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>>
> 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:517 at 172.16.1.174 <p%3A517 at 172.16.1.174>>;tag=4881ea36-2ca-6747d965
> SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport
> From: "512" <sip:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To: <sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> Call-ID: 2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20
> 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:512 at 172.16.1.20 <sip%3A512 at 172.16.1.20>>;tag=as7ec9e8af
> To:<sip:517 at 172.16.1.174 <sip%3A517 at 172.16.1.174>
> >;tag=4881ea36-2ca-6747d965
> CSeq:103 BYE
> User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1
> Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20<Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa at 172.16.1.20>
> Content-Length:0
>
>
> <------------->
>   --- (8 headers 0 lines) ---
> greybeamPBX*CLI> sip set debug off
>
>
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