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tucker@vplan.co.uk

2004-Jul-01 12:34 UTC

[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@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@192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
To: <sip:2200@192.168.1.103>
Contact: <sip:asterisk@192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@iax-ogateway]/2' failed
localhost*CLI>
                                                                        
       
Sip read:
SIP/2.0 100 Trying
To: <sip:2200@192.168.1.103>
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
To: <sip:2200@192.168.1.103>
Contact: <sip:asterisk@192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@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@iax-ogateway]/2'
                                                                        
       
                                                                        
       
Sip read:
SIP/2.0 487 Request Terminated
To: <sip:2200@192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
To: <sip:2200@192.168.1.103>;tag=830cbfdb36128143
Contact: <sip:asterisk@192.168.1.100:0>
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103>;tag=830cbfdb36128143
From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84
Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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>

Deon Rodden

2004-Jul-01 13:07 UTC

head link

[Asterisk-Users] IAX2 to IAX2 connection problems

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@vplan.co.uk>
To: <asterisk-users@lists.digium.com>
Cc: <tucker@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@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@192.168.1.103 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> To: <sip:2200@192.168.1.103>
> Contact: <sip:asterisk@192.168.1.100:0>
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@iax-ogateway]/2'
failed
> localhost*CLI>
>
>
> Sip read:
> SIP/2.0 100 Trying
> To: <sip:2200@192.168.1.103>
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103>;tag=830cbfdb36128143
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> To: <sip:2200@192.168.1.103>
> Contact: <sip:asterisk@192.168.1.100:0>
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@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@iax-ogateway]/2'
>
>
>
>
> Sip read:
> SIP/2.0 487 Request Terminated
> To: <sip:2200@192.168.1.103>;tag=830cbfdb36128143
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> To: <sip:2200@192.168.1.103>;tag=830cbfdb36128143
> Contact: <sip:asterisk@192.168.1.100:0>
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@192.168.1.103>;tag=830cbfdb36128143
> From: "asterisk"
<sip:asterisk@192.168.1.100:0>;tag=as17b86f84
> Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
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>

tucker

2004-Jul-01 15:07 UTC

head link

[Asterisk-Users] IAX2 to IAX2 connection problems

Only trying to get one way, office to local working at moment

Local calls work ok, can dial out and receive calls from POTS
Console to SIP works fine
Routing will work locally, answer, to sip, to mailbox if unavilable etc

Remote end
extensions.conf
exten => _43,1,Dial(IAX2/iax-gateway/2200,10)

iax.conf
[iax-gateway]
type=friend
allow=localpcip
context=incoming
host=localpcip

sip.conf
no sip installed

Local end
extensions.conf
PHONES1=SIP/2200				
PHONES1VM=2200
[incoming]
exten => 2200,1,Dial(${PHONES1},20,Ttm)
exten => 2200,2,Macro(vmessage,${PHONES1VM})
exten => 2200,2,Wait,5
exten => 2200,3,Hangup

iax.conf
[iax-ogateway]
type=friend
allow=remotepcip
context=incoming
host=remotepcip

sip.conf
[2200]
type=friend
host=dynamic
context=home
callerid="SPA1" <2200>
dtmfmode=rfc2833
nat=0
;

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