[Asterisk-Users] problems with mediatrix 1204 FXO

Wed Sep 3 09:57:55 MST 2003

> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Martin Pycko
> Sent: Tuesday, September 02, 2003 2:31 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] problems with mediatrix 1204 FXO
> 
> 
> Try canreinvite=no
> 
> Martin

That worked like a champ.  Now I'm facing another issue:

When a sip phone terminates the call, asterisk sends the BYE to the 
wrong SIP URI...  Instead of sending it to pstnline2 at 172.20.16.7
it sends it to 611 at 172.20.16.7

Unless I'm misuderstanding the rfc asterisk is not properly 
handling the CONTACT header.

Sip debug follows:

[root at carbon root]# asterisk -r
Asterisk CVS-09/01/03-11:21:09, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <markster at linux-support.net>
=========================================================================
Connected to Asterisk CVS-09 currently running on carbon (pid = 21270)
    -- Remote UNIX connection
carbon*CLI> sip debug
SIP Debugging Enabled
Sip read: >
INVITE sip:611 at sprackett.com SIP/2.0
Via:SIP/2.0/UDP 66.46.196.156:5060
From:"Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To:<sip:611 at sprackett.com>
Contact:"Zac at Work" <sip:mitel at 66.46.196.156>
Call-ID:e1cc0000-d4c439 at 66.46.196.156
Subject:sip phone call
CSeq:2 INVITE
User-Agent:Mitel-5055-SIP-Phone TST16_12 08000F014455
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY
Max-Forwards:70
Content-Type:application/sdp
Content-Length:263

v=0
o=mitel 1062593444 1062593443 IN IP4 66.46.196.156
s=SIP Call
p=+613-592-2122
c=IN IP4 66.46.196.156
t=0 0
m=audio 20050 RTP/AVP 18 0 8 96
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=sendrecv

13 headers, 12 lines
Using latest request as basis request
Sending to 66.46.196.156 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 66.46.196.156:5060
From: "Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To: <sip:611 at sprackett.com>;tag=as38e73f19
Call-ID: e1cc0000-d4c439 at 66.46.196.156
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="3b29343b"
Content-Length: 0

 to 66.46.196.156:5060
Sip read: >
ACK sip:611 at sprackett.com SIP/2.0
Via:SIP/2.0/UDP 66.46.196.156:5060
From:"Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To:<sip:611 at sprackett.com>;tag=as38e73f19
Contact:"Zac at Work" <sip:mitel at 66.46.196.156>
Call-ID:e1cc0000-d4c439 at 66.46.196.156
CSeq:2 ACK
User-Agent:Mitel-5055-SIP-Phone TST16_12 08000F014455
Max-Forwards:70
Content-Length:0

10 headers, 0 lines
Sip read: >
INVITE sip:611 at sprackett.com SIP/2.0
Via:SIP/2.0/UDP 66.46.196.156:5060
From:"Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To:<sip:611 at sprackett.com>
Contact:"Zac at Work" <sip:mitel at 66.46.196.156>
Call-ID:e1cc0000-d4c439 at 66.46.196.156
Subject:sip phone call
CSeq:3 INVITE
User-Agent:Mitel-5055-SIP-Phone TST16_12 08000F014455
Proxy-Authorization: Digest username="mitel", realm="asterisk", nonce="3b29343b", uri="sip:611 at sprackett.com", response="88938fc85f922b75263f65dfd9286c13", opaque="", algorithm=MD5
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY
Max-Forwards:70
Content-Type:application/sdp
Content-Length:263

v=0
o=mitel 1062593444 1062593443 IN IP4 66.46.196.156
s=SIP Call
p=+613-592-2122
c=IN IP4 66.46.196.156
t=0 0
m=audio 20050 RTP/AVP 18 0 8 96
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=sendrecv

14 headers, 12 lines
Using latest request as basis request
Sending to 66.46.196.156 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 611 in internal
list_route: hop: <sip:mitel at 66.46.196.156>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.46.196.156:5060
From: "Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To: <sip:611 at sprackett.com>;tag=as5e890982
Call-ID: e1cc0000-d4c439 at 66.46.196.156
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:611 at 66.11.172.3>
Content-Length: 0

 to 66.46.196.156:5060
    -- Executing Dial("SIP/mitel-d160", "SIP/611 at mediatrix-1204|20|Tr") in new stack
We're at 172.20.16.1 port 20112
Answering with capability 2
Answering with capability 4
Answering with capability 8
11 headers, 9 lines
Reliably Transmitting:
INVITE sip:611 at 172.20.16.7 SIP/2.0
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>
Contact: <sip:04 at 172.20.16.1>
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 22727 22727 IN IP4 172.20.16.1
s=session
c=IN IP4 172.20.16.1
t=0 0
m=audio 20112 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
 (no NAT) to 172.20.16.7:5060
    -- Called 611 at mediatrix-1204
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.46.196.156:5060
From: "Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To: <sip:611 at sprackett.com>;tag=as5e890982
Call-ID: e1cc0000-d4c439 at 66.46.196.156
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:611 at 66.11.172.3>
Content-Length: 0

 to 66.46.196.156:5060
Sip read: >
SIP/2.0 100 Trying
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 102 INVITE
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>;tag=3be40ad0-c9c9ff1e
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
Content-Length: 0

7 headers, 0 lines
Sip read: > sip
SIP/2.0 200 OK
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 102 INVITE
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>;tag=3be40ad0-c9c9ff1e
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
Content-Length: 148
Content-Type: application/sdp
Contact: "PSTN LINE 2" <sip:pstnline2 at 172.20.16.7:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Supported: replaces

v=0
o=MxSIP 0 0 IN IP4 172.20.16.7
s=SIP Call
c=IN IP4 172.20.16.7
t=0 0
m=audio 5006 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

11 headers, 8 lines
Found audio format UNKN
Found audio format ALAW
Found description format PCMU
Found description format PCMA
Capabilities: us - 524302, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:pstnline2 at 172.20.16.7:5060>
set_destination: Parsing <sip:pstnline2 at 172.20.16.7:5060> for address/port to send to
set_destination: set destination to 172.20.16.7, port 5060
Transmitting:
ACK sip:611 at 172.20.16.7 SIP/2.0
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>;tag=3be40ad0-c9c9ff1e
Contact: <sip:04 at 172.20.16.1>
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 172.20.16.7:5060
    -- SIP/mediatrix-1204-1ce5 answered SIP/mitel-d160
We're at 66.11.172.3 port 20824
Answering with capability 4
Answering with capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.46.196.156:5060
From: "Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To: <sip:611 at sprackett.com>;tag=as5e890982
Call-ID: e1cc0000-d4c439 at 66.46.196.156
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:611 at 66.11.172.3>
Content-Type: application/sdp
Content-Length: 155

v=0
o=root 22727 22727 IN IP4 66.11.172.3
s=session
c=IN IP4 66.11.172.3
t=0 0
m=audio 20824 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 66.46.196.156:5060
    -- Attempting native bridge of SIP/mitel-d160 and SIP/mediatrix-1204-1ce5
Sip read:
ACK sip:611 at 66.11.172.3 SIP/2.0
Via:SIP/2.0/UDP 66.46.196.156:5060
From:"Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To:<sip:611 at sprackett.com>;tag=as5e890982
Contact:"Zac at Work" <sip:mitel at 66.46.196.156>
Call-ID:e1cc0000-d4c439 at 66.46.196.156
CSeq:3 ACK
User-Agent:Mitel-5055-SIP-Phone TST16_12 08000F014455
Max-Forwards:70
Content-Length:0

10 headers, 0 lines
Sip read:
BYE sip:611 at 66.11.172.3 SIP/2.0
Via:SIP/2.0/UDP 66.46.196.156:5060
From:"Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To:<sip:611 at sprackett.com>;tag=as5e890982
Contact:"Zac at Work" <sip:mitel at 66.46.196.156>
Call-ID:e1cc0000-d4c439 at 66.46.196.156
CSeq:4 BYE
User-Agent:Mitel-5055-SIP-Phone TST16_12 08000F014455
Max-Forwards:70
Content-Length:0

10 headers, 0 lines
Sending to 66.46.196.156 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.46.196.156:5060
From: "Zac at Work" <sip:mitel at sprackett.com>;tag=3f55e1cc-1d7-10d8aff2
To: <sip:611 at sprackett.com>;tag=as5e890982
Call-ID: e1cc0000-d4c439 at 66.46.196.156
CSeq: 4 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:611 at 66.11.172.3>
Content-Length: 0

 to 66.46.196.156:5060
set_destination: Parsing <sip:pstnline2 at 172.20.16.7:5060> for address/port to send to
set_destination: set destination to 172.20.16.7, port 5060
Reliably Transmitting:
BYE sip:611 at 172.20.16.7 SIP/2.0
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>;tag=3be40ad0-c9c9ff1e
Contact: <sip:04 at 172.20.16.1>
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 172.20.16.7:5060
  == Spawn extension (internal, 611, 1) exited non-zero on 'SIP/mitel-d160'
Sip read:
SIP/2.0 404 Not Found
Call-ID: 736d2dcb3433da573130197a6f9624c4 at 172.20.16.1
CSeq: 103 BYE
From: "Zac at Work" <sip:04 at 172.20.16.1>;tag=as592d4b87
To: <sip:611 at 172.20.16.7>;tag=3be40ad0-c9c9ff1e
Via: SIP/2.0/UDP 172.20.16.1:5060;branch=z9hG4bK49103a45
Content-Length: 0

7 headers, 0 lines
    -- Got SIP response 404 "Not Found" back from 172.20.16.7

No such command 'sip' (type 'help' for help)
carbon*CLI> sip no debug
SIP Debugging Disabled