Hi P,
Al the settings in the SIP.conf were copied to me by the provider (they’re not entirely ‘up’ with asterisk, that was just ‘what worked’ for them with other clients…)
Ok, here’s the full SIP debug output:
<-- SIP read from 202.61.13.40:5060:
INVITE sip:s@202.128.109.6 SIP/2.0
To: <sip:0390130863@voice.mibroadband.com.au>
From: <sip:0417055052@202.61.13.52>;tag=ce0e5e7d
Via: SIP/2.0/UDP 202.61.13.40:5060;branch=z9hG4bK-d87543-c63d142ce87de2796bf8-1-cHBhZWE4NWIxNTc2NGU1MjUwM2MzZg..-d87543-
Call-ID: 6e2ffa14150222c50126debcc87120f4
CSeq: 240278331 INVITE
Contact: <sip:2nCvrO0hlpBnLAEazI4Rc5somneKml6QQ..@202.61.13.40:5060>
Expires: 180
Max-Forwards: 68
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Content-Disposition: session;handling=required
Content-Type: application/sdp
Date: Tue, 18 Apr 2006 14:54:55 GMT
Supported: timer, resource-priority, replaces
Timestamp: 1145372095
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow-Events: telephone-event
Content-Length: 426
Remote-Party-ID: <sip:0417055052@202.61.13.52>;party=calling;screen=yes;privacy=off
Min-SE: 1800
Cisco-Guid: 1993281523-3484684762-2284322819-3128418427
v=0
o=CiscoSystemsSIP-GW-UserAgent 9710 1096284119 IN IP4 202.61.13.40
s=SIP Call
c=IN IP4 202.61.13.40
t=0 0
m=audio 16684 RTP/AVP 18 8 0 98 99 102 101 19
c=IN IP4 202.61.13.40
a=fmtp:18 annexb=yes
a=fmtp:101 0-16
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:19 CN/8000
--- (21 headers 17 lines)---
Using INVITE request as basis request - 6e2ffa14150222c50126debcc87120f4
Sending to 202.61.13.40 : 5060 (non-NAT)
Found no matching peer or user for '202.61.13.40:5060'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 102
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port 202.61.13.40:16684
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format telephone-event
Found description format CN
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x11c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for s in default (domain 202.128.109.6)
list_route: hop: <sip:2nCvrO0hlpBnLAEazI4Rc5somneKml6QQ..@202.61.13.40:5060>
Transmitting (no NAT) to 202.61.13.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.61.13.40:5060;branch=z9hG4bK-d87543-c63d142ce87de2796bf8-1-cHBhZWE4NWIxNTc2NGU1MjUwM2MzZg..-d87543-;received=202.61.13.40
From: <sip:0417055052@202.61.13.52>;tag=ce0e5e7d
To: <sip:0390130863@voice.mibroadband.com.au>
Call-ID: 6e2ffa14150222c50126debcc87120f4
CSeq: 240278331 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@202.128.109.6>
Content-Length: 0
---
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "0417055052") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "0417055052") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "0") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "SIP/202.61.13.52-081bb460") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "default") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "19042006-00:54:55") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "1145372095") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "s") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "0") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "en") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "21") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "202.128.109.6") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "6e2ffa14150222c50126debcc87120f4") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "Cisco-SIPGateway/IOS-12.x") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "20060419-005455") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "1145372095.70") in new stack
-- Executing NoOp("SIP/202.61.13.52-081bb460", "") in new stack
-- Executing Dial("SIP/202.61.13.52-081bb460", "SIP/damien") in new stack
-- Called damien
Using INVITE request as basis request - 6e2ffa14150222c50126debcc87120f4
-- SIP/damien-9043 is ringing
Looking at that, I’m guessing it’s something to do with the bit that says "Found no matching peer or user for ‘202.61.13.40:5060’ " (even though that’s one of the two peers that it maps against the engin registration)?, and then goes on to look for s in default context?
Thanks,
Damien