[Asterisk-Users] IAX-IAX Trunking not works

Gary Lawrence VOIP at ITcom.Net
Tue May 24 22:00:16 MST 2005


While you have active calls, type at the cli prompt "iax2 trunk debug".

If trunking is working you should get a reply like:

IAX2 Trunk Debug Requested
Beginning trunk processing
Ending trunk processing with 1 peers and 3 calls processed

If you want to free up more bandwidth add "echocancel=no" to your iax.conf

Gary Lawrence

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Clark
Sent: Monday, May 23, 2005 10:01 AM
To: Adnan Ahmed; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX-IAX Trunking not works

Adnan Ahmed wrote:
> Hello ,
> I want some tips guidance i am sure this topic discuss alot in list,i
> try my best to solve it by myself try googling looking wiki everywhere
> but no luck question is iax-iax trunking not working setting,trying
> each n every option
> 
> server2 iax.conf:
> [general]
> bindport=4569
> bandwidth=low
> disallow=all
> allow=gsm
> jitterbuffer=no
> tos=lowdelay
> trunk=yes
> notransfer=yes
> 
> [saim]
> username=saim
> secret=saim
> 
> type=friend
> host=dynamic
> context=from-sip
> 
> disallow=all
> allow=gsm
> 
> [noman]
> username=saim
> secret=noman
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [asteriskser1]
> type=friend
> ;auth=md5
> ;secret=qwerty
> context=local
> ;host=dynamic
> defaultip=192.168.0.51
> notransfer=yes
> qualify=no
> trunk=yes
> canreinvite=no
> 
> server1 iax.conf:
> [general]
> bindport=4569
> bandwidth=low
> disallow=all
> allow=gsm
> jitterbuffer=no
> tos=lowdelay
> trunk=yes
> notransfer=yes
> 
> [user1]
> username=user1
> secret=user1
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [user2]
> username=user2
> secret=user2
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [test2]
> type=friend
> context=local
> defaultip=192.168.0.51
> notransfer=yes
> qualify=no
> trunk=yes
> canreinvite=no
> 
> 
> I am using Kiax soft phone  on both servers using codec GSM asterisk
> latest stable version OS SLES9 ,any help is highly appreciated i had
> look almost every place in wiki regarding iax trunking but all in
> vein.
> Thanks In Advance.
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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> 
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