rendered paste bodyAsterisk isn't setting the destination correctly when exchange 2007's call is proxied thru openSER. Is this an openSER issue or should asterisk be handing the call differently?
The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between.
significant differences
========================
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888
-- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8
== Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888'
Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI>
VERSES
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:goodproxy:5060>
set_destination: Parsing <sip:goodproxy:5060> for address/port to send to
set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060:
asterisk debug of openser call
===========
--- (9 headers 0 lines) ---
pbx*CLI>
<-- SIP read from openser:5060:
SIP/2.0 180 Ringing
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078
TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f
CSEQ: 102 INVITE
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk
MAX-FORWARDS: 70
VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060
CONTENT-LENGTH: 0
SERVER: RTCC/2.0.6017.0
--- (9 headers 0 lines) ---
-- Called openser/8886000
-- SIP/openser-082966e8 is ringing
pbx*CLI>
<-- SIP read from openser:5060:
SIP/2.0 200 OK
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078
TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f
CSEQ: 102 INVITE
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk
MAX-FORWARDS: 70
VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060
CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>
CONTENT-LENGTH: 197
CONTENT-TYPE: application/sdp
SERVER: RTCC/2.0.6017.0
v=0
o=- 0 0 IN IP4 exchangeUM
s=Microsoft Exchange Speech Engine
c=IN IP4 exchangeUM
t=0 0
m=audio 30660 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888
-- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8
== Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888'
Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI>
<-- SIP read from openser:5060:
BYE sip:3149@asterisk SIP/2.0
Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>
Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>
FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f
TO: <sip:3149@asterisk>;tag=as4aa53078
CSEQ: 1 BYE
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk
MAX-FORWARDS: 69
Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825
VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65
CONTENT-LENGTH: 0
USER-AGENT: RTCC/2.0.6017.0
P-hint: outbound
--- (13 headers 0 lines) ---
Sending to openser : 5060 (non-NAT)
Transmitting (no NAT) to openser:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser
Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65
Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>
Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>
From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f
To: <sip:3149@asterisk>;tag=as4aa53078
Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3149@asterisk>
Content-Length: 0
ngrep of above call
======================
#
U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing.
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078.
TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f.
CSEQ: 102 INVITE.
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk.
MAX-FORWARDS: 70.
VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060.
CONTENT-LENGTH: 0.
SERVER: RTCC/2.0.6017.0.
.
#
U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK.
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078.
TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f.
CSEQ: 102 INVITE.
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk.
MAX-FORWARDS: 70.
VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060.
CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>.
CONTENT-LENGTH: 197.
CONTENT-TYPE: application/sdp.
SERVER: RTCC/2.0.6017.0.
.
v=0.
o=- 0 0 IN IP4 exchangeUM.
s=Microsoft Exchange Speech Engine.
c=IN IP4 exchangeUM.
t=0 0.
m=audio 30660 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0.
Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>.
Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>.
FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f.
TO: <sip:3149@asterisk>;tag=as4aa53078.
CSEQ: 1 BYE.
CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk.
MAX-FORWARDS: 69.
Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825.
VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65.
CONTENT-LENGTH: 0.
USER-AGENT: RTCC/2.0.6017.0.
P-hint: outbound.
.
#
U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK.
Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser.
Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65.
Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>.
Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>.
From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f.
To: <sip:3149@asterisk>;tag=as4aa53078.
Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk.
CSeq: 1 BYE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:3149@asterisk>.
Content-Length: 0.
.
good call from working proxy
========
We're at asterisk port 17508
Video is at asterisk port 11130
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to goodproxy:5060:
INVITE sip:6000@goodproxy SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport
From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a
To: <sip:6000@goodproxy>
Contact: <sip:3149@asterisk>
Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 05 Feb 2007 14:55:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 381 381 IN IP4 asterisk
s=session
c=IN IP4 asterisk
t=0 0
m=audio 17508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called exchange12/6000
pbx*CLI>
<-- SIP read from goodproxy:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy
From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a
To: <sip:6000@goodproxy>
Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
CSeq: 102 INVITE
User-Agent: M-Networks USR/1.0
Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE
Content-Length: 0
--- (9 headers 0 lines) ---
pbx*CLI>
<-- SIP read from goodproxy:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a
TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98
CSEQ: 102 INVITE
CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
MAX-FORWARDS: 70
CONTENT-LENGTH: 0
SERVER: RTCC/2.0.6017.0
--- (9 headers 0 lines) ---
-- SIP/exchange12-082b8e18 is ringing pbx*CLI>
<-- SIP read from goodproxy:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a
TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98
CSEQ: 102 INVITE
CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
MAX-FORWARDS: 70
CONTACT: <sip:goodproxy:5060>
CONTENT-LENGTH: 197
CONTENT-TYPE: application/sdp
SERVER: RTCC/2.0.6017.0
v=0
o=- 0 0 IN IP4 exchangeUM
s=Microsoft Exchange Speech Engine
c=IN IP4 exchangeUM
t=0 0
m=audio 47688 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:goodproxy:5060>
set_destination: Parsing <sip:goodproxy:5060> for address/port to send to
set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060:
ACK sip:goodproxy:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport
From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a
To: <sip:6000@goodproxy>;tag=f51839ab98
Contact: <sip:3149@asterisk>
Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8
-- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms
set_destination: Parsing <sip:goodproxy:5060> for address/port to send to
set_destination: set de5060 SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport
From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a
To: <sip:6000@goodproxy>;tag=f51839ab98
Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8'
pbx*CLI>
<-- SIP read from goodproxy:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a
TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9
CSEQ: 103 BYE
CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk
MAX-FORWARDS: 70
CONTENT-LENGTH: 0
SERVER: RTCC/2.0.6017.0
--- (9 headers 0 lines) ---
Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk'
pbx*CLI>