H323 trunk to SIP problem (asterisk)

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H323 trunk to SIP problem (asterisk)

Postby syedbilalmasaud » Sun Dec 13, 2009 10:07 am

Dear All,

I am trying to configure H323 Trunk on freePBX , first of all freepbx is not supported with h323 properly means it will not listen on 1720(H323 port) unless I will install OOh323 and recompile asterisk , but when I will recompile , systems starts working on h323 but once installation is done , I have following problems

1:- SIP users stop registering to asterisk , (I can see packets are arriving to system but asterisk says that id and pass are incorrect , which is 100% correct)
2:- When I will receive call from H323 trunk and transfer it to SIP , no voice connectivity
3:- Call is coming from Cisco Access Server AS53xx , which is only supported for H323 , In this stage I cannot change or update IOS because I dont want to change IOS just for small change and I am pretty sure that H323 and SIP should work ,I am doing some thing SILLY

any Ideas , I am sure that someone must have tried this kind of stuff on asterisk

Call Flow

Cisco AS5350 (123456) toll free number ==> H323 ==> Asterisk ==> 123456,s,1,Dial(SIP/100,25,tT)

Am I doing some thing wrong ?

Thanks


Cheers:)
Bilal
Cheers :)

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syedbilalmasaud
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Re: H323 trunk to SIP problem (asterisk)

Postby kbukhari » Wed Dec 16, 2009 8:22 pm

syedbilalmasaud wrote:Dear All,

I am trying to configure H323 Trunk on freePBX , first of all freepbx is not supported with h323 properly means it will not listen on 1720(H323 port) unless I will install OOh323 and recompile asterisk , but when I will recompile , systems starts working on h323 but once installation is done , I have following problems

i will recomend you not to use freepbx use asterisk on command line
1:- SIP users stop registering to asterisk , (I can see packets are arriving to system but asterisk says that id and pass are incorrect , which is 100% correct)

check output of " asterisk -rx 'sip show peers'
2:- When I will receive call from H323 trunk and transfer it to SIP , no voice connectivity

need more discriptions
3:- Call is coming from Cisco Access Server AS53xx , which is only supported for H323 , In this stage I cannot change or update IOS because I dont want to change IOS just for small change and I am pretty sure that H323 and SIP should work ,I am doing some thing SILLY

why not?
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Postby syedbilalmasaud » Thu Dec 17, 2009 10:16 am

ok let me do the following

I will install asterisk on command line maually with h323 support and see how it works

for SIP registration asterisk just says bad auth
which is correct and i am 100% sure
Cheers :)



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Re:

Postby LinuxFreaK » Tue Jan 26, 2010 11:41 am

Dear syedbilalmasaud,
Salam,

syedbilalmasaud wrote:for SIP registration asterisk just says bad auth
which is correct and i am 100% sure


Show your /etc/asterisk/sip.conf

Best Regards.
Farrukh Ahmed
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