rendered paste bodyBelow is a 'sip debug' dump from my Asterisk 1.2 (behind NAT) to the TalkShoe service.
The talkshoe peer:
[talkshoe]
type = peer
host = 66.212.134.192
fromuser = 123
disallow = all
allow=ulaw
qualify=yes
dtmfmode=auto
insecure = very
The dialplan line:
exten => 733,n,Dial(SIP/talkshoe,60,)
-- Hungup 'Zap/3-1'
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp("Zap/3-1", "") in new stack
-- Executing Dial("Zap/3-1", "SIP/talkshoe|60|") in new stack
We're at 192.168.67.18 port 14828
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 66.212.134.192:5060:
INVITE sip:66.212.134.192 SIP/2.0
Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK77b0646e;rport
From: "Channel 3" <sip:123@192.168.67.18>;tag=as030a5bf1
To: <sip:66.212.134.192>
Contact: <sip:123@192.168.67.18>
Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Jul 2007 11:29:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 29760 29760 IN IP4 192.168.67.18
s=session
c=IN IP4 192.168.67.18
t=0 0
m=audio 14828 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called talkshoe
pungenday*CLI>
<-- SIP read from 66.212.134.192:5060:
SIP/2.0 180 Ringing
From: "Channel 3"<sip:123@192.168.67.18>;tag=as030a5bf1
To: <sip:66.212.134.192>;tag=a000217-13c4-468df02d-378793b7-532ab8
Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK77b0646e
Contact: <sip:66.212.134.192>
Content-Length: 0
--- (8 headers 0 lines) ---
-- SIP/talkshoe-08149a88 is ringing
pungenday*CLI>
<-- SIP read from 66.212.134.192:5060:
SIP/2.0 403 Forbidden
From: "Channel 3"<sip:123@192.168.67.18>;tag=as030a5bf1
To: <sip:66.212.134.192>;tag=a000217-13c4-468df02d-378793b7-532ab8
Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK77b0646e
Content-Length: 0
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 66.212.134.192:5060:
ACK sip:66.212.134.192 SIP/2.0
Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK77b0646e;rport
From: "Channel 3" <sip:123@192.168.67.18>;tag=as030a5bf1
To: <sip:66.212.134.192>;tag=a000217-13c4-468df02d-378793b7-532ab8
Contact: <sip:123@192.168.67.18>
Call-ID: 2705765123bad3291632549955ad21f3@192.168.67.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Jul 6 14:29:59 WARNING[29774]: chan_sip.c:9847 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Channel 3" <sip:123@192.168.67.18>;tag=as030a5bf1'
-- SIP/talkshoe-08149a88 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/3-1' status is 'CONGESTION'
Destroying call '2705765123bad3291632549955ad21f3@192.168.67.18'
-- Hungup 'Zap/3-1'
pungenday*CLI> sip no debug
SIP Debugging Disabled