[asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 & Mitel SIP phones
Mark Wiater
mwiater at cablespeed.com
Sat Jul 19 12:24:02 CDT 2008
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/2.0
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport
From: "512" <sip:512 at 172.16.1.20>;tag=as7ec9e8af
To: <sip:517 at 172.16.1.174>
Contact: <sip:512 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>;tag=as7ec9e8af
To:<sip:517 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
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>;tag=as7ec9e8af
To:<sip:517 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
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>;tag=as7ec9e8af
To:<sip:517 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
Contact:"p:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
CSeq:102 INVITE
User" <sip:517 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>;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>;tag=4881ea36-2ca-6747d965>
set_destination: Parsing
<"p:517 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>;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>;tag=as7ec9e8af
To: <sip:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
Contact: <sip:512 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>;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>;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>;tag=as7ec9e8af
To: <sip:517 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>;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>;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>;tag=as7ec9e8af
To: <sip:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
Contact: <sip:512 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>;tag=as7ec9e8af
To:<sip:517 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
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>;tag=as7ec9e8af
To:<sip:517 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
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing
<"p:517 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>;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>;tag=as7ec9e8af
To: <sip:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
Contact: <sip:512 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>;tag=as7ec9e8af
To:<sip:517 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
Contact:"p:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
CSeq:102 INVITE
User" <sip:517 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>;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>;tag=as7ec9e8af
To: <sip:517 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>;tag=as7ec9e8af
To:<sip:517 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
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>;tag=as7ec9e8af
To:<sip:517 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
Contact:"p:517 at 172.16.1.174>;tag=4881ea36-2ca-6747d965
CSeq:102 INVITE
User" <sip:517 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>;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>;tag=as7ec9e8af
To: <sip:517 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>;tag=as7ec9e8af
To:<sip:517 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
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
greybeamPBX*CLI> sip set debug off
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