IAX to SIP calling problem

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IAX to SIP calling problem

Postby arunvoip » Fri Jun 30, 2006 3:55 am

Hi,

My asterisk,vicidial and my audio codes are in same subnet. I have configured my Audio codes MP124 FXS for my SIP agents. My sip agents are behind NAT/Firewall. I have one IAX2 agent in the same subnet of my sip. When I call from my IAX2 agent to SIP agent it says this message:
***************************************************
Executing Dial("IAX2/2001-1", "SIP/2003|30|to") in new stack
-- Called 2003
-- SIP/2003-a06d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing VoiceMail("IAX2/2001-1", "u2003") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/3' (language 'en')
== Spawn extension (default, 2003, 2) exited non-zero on 'IAX2/2001-1'
-- Executing DeadAGI("IAX2/2001-1", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("IAX2/2001-1", "VD_hangup.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/2001-1'
******************************************
and gets hangup. But when I call from my SIP agent 2003 to my IAX2 agent 2001 its fine. When I call from SIP/2003 to SIP/2004 I get same message and hangup. When I saw my sip show peers it shows :
******************
2004/2004 *.*.*.* D N 5060 OK
2003/2003 *.*.*.* D N 5060 OK

I think its causing some problem b'coz both agents are using the same port. Is there any why to configure FXS to tell for each agent get different port if, its causing a problem.

thanks

./Arun
arunvoip
 
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Postby mflorell » Fri Jun 30, 2006 5:05 am

I am not very familiar with te AudioCodes FXS gateways, but I think that and/or your firewall may be the problem. This is not really a VICIDIAL issue so you may want to post on the asterisk-users or the Digium forums about this.
mflorell
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