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>
