[Asterisk-Users] IAX2 to IAX2 connection problems
Deon Rodden
drodden at webunited.net
Thu Jul 1 13:07:02 MST 2004
What's your iax.conf config files look like on both end? And your dial
statements in the extensions.conf file? Also, what version of Asterisk are
you running locally, remotely?
----- Original Message -----
From: <tucker at vplan.co.uk>
To: <asterisk-users at lists.digium.com>
Cc: <tucker at vplan.co.uk>
Sent: Thursday, July 01, 2004 3:34 PM
Subject: [Asterisk-Users] IAX2 to IAX2 connection problems
>
> 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>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list