[Asterisk-Users] IAX2 to IAX2 connection problems

Thu Jul 1 12:34:01 MST 2004

Hi

My head hurts... Can anyone help out here, my remote IAX can see my
local IAX and visa versa, conversation starts, I can dial my remote
(POTS) landline number, remote end answers, trys to route to local
iax2, I see it start the conversation here, the extension (SIP) rings
once and then it dies...

Both ends are defined with accept IPADDRESS to keep it in the family and
simple..

Debug info below for those who are good on this stuff..

Any info would be good, working config sections even better.

Thanks in advance

localhost*CLI> sip debug
SIP Debugging Enabled
localhost*CLI> iax2 debug
IAX2 Debugging Enabled
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass:
NEW
   Timestamp: 00018ms  SCall: 00002  DCall: 00000 [80.168.166.208:4569]
   VERSION         : 2
   CALLED NUMBER   : 2200
   LANGUAGE        : en
   FORMAT          : 2
   CAPABILITY      : 65283
   ADSICPE         : 2
   DATE TIME       : 149004861

    -- Accepting unauthenticated call from 80.168.166.208, requested
format = 2, actual format = 2
    -- Executing Dial("IAX2[iax-ogateway at iax-ogateway]/2",
"SIP/2200|20|Ttm") in new stack
We're at 192.168.1.100 port 14058
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:2200 at 192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
To: <sip:2200 at 192.168.1.103>
Contact: <sip:asterisk at 192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 01 Jul 2004 19:26:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3420 3420 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 14058 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 192.168.1.103:5060
    -- Called 2200
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
ACCEPT
   Timestamp: 00008ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
   FORMAT          : 2

Jul  1 20:26:26 WARNING[-309027920]: chan_iax2.c:2504 iax2_send:
timestamp is 0?Jul  1 20:26:26 WARNING[-309027920]: channel.c:1343
ast_prod: Prodding channel 'IAX2[iax-ogateway at iax-ogateway]/2' failed
localhost*CLI>

Sip read:
SIP/2.0 100 Trying
To: <sip:2200 at 192.168.1.103>
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
Server: Sipura/SPA2000-1.0.33
Content-Length: 0

8 headers, 0 lines

Sip read:
SIP/2.0 180 Ringing
To: <sip:2200 at 192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
Server: Sipura/SPA2000-1.0.33
Content-Length: 0

8 headers, 0 lines
    -- SIP/2200-e9d1 is ringing
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass:
ACK
   Timestamp: 00008ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: VOICE   Subclass: 2
   Timestamp: 00020ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass:
ACK
   Timestamp: 00020ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
Reliably Transmitting:
CANCEL sip:2200 at 192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
To: <sip:2200 at 192.168.1.103>
Contact: <sip:asterisk at 192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.103:5060
  == Spawn extension (incoming, 2200, 1) exited non-zero on
'IAX2[iax-ogateway at iax-ogateway]/2'
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass:
HANGUP
   Timestamp: 00068ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
    -- Hungup 'IAX2[iax-ogateway at iax-ogateway]/2'

Sip read:
SIP/2.0 487 Request Terminated
To: <sip:2200 at 192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
Server: Sipura/SPA2000-1.0.33
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:2200 at 192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
To: <sip:2200 at 192.168.1.103>;tag=830cbfdb36128143
Contact: <sip:asterisk at 192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.103:5060

Sip read:
SIP/2.0 200 OK
To: <sip:2200 at 192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk at 192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac at 192.168.1.100
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
Server: Sipura/SPA2000-1.0.33
Content-Length: 0

8 headers, 0 lines
Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass:
ACK
   Timestamp: 00068ms  SCall: 00002  DCall: 00002 [80.168.166.208:4569]
localhost*CLI>