Asterisk avaya integration - h323 trunk - no audio on avaya


I am trying out with asterisk to avaya communication.

On asterisk side i have installed the following things:

On the avaya side i am using S8300

I have done with installations, setting up the H323 trunk.

I am [color=red]able to make intra asterisk calls[/color], means my one SIP extension
can make a call to other SIP extension on the asterisk. intra asterisk
communication is working fine.
I am able to make a call from avaya to asterisk and asterisk to avaya.

[color=red]Audio is working perfectly from avaya to asterisk.
The problem i am facing is “no audio” from asterisk to avaya.[/color]
Any helping pointer will be highly appreciated.
Is there any problem with the versions that i have installed?

following is the trace from my h323.conf file:

port = 1720
bindaddr =



on both sides i am using the same codecs, like on the asterisk side i am using alaw and asterisk side i am using g.711A that is alaw.

i have tried with ulaw too, tried out with dtmf too like setting to rfc2833,
info too but still the problem persists.

Thanks in advance,
Vishal Bhardwaj.

I am assuming you are calling an IP phone on the Avaya side… On the second page of the Avaya phone form look and see if IP - IP audio connections is set to yes. If so, change this to No. This will force the call to use a Media Resource. I have seen this fix the problem.

If that doesn’t work, try doing a list trace station on the Avaya phone and see what it shows when you make the call with no audio. It might give some clues.
Good Luck!

Hi pspenn,

Thanks a ton, for all your help, it solved my problem. Now:

  1. I am able to make calls from Avaya to asterisk and audio on both directions is working fine.

  2. I am able to make calls from Asterisk to Avaya, but the audio is one sided from Avaya to asterisk only, no audio on Avaya side (please help me further)…:frowning:

  3. How can i send DTMF tones from Avaya (using S8300) to Asterisk and how to detect those on Asterisk side?

thanks and regards,
Vishal Bhardwaj.

Hello Vishal,
For DTMF you will need to change the Signaling Group in Avaya and look on page 1 for something called DTMF over IP. Change this to “Out-of-Band”. This is based on your setting in Asterisk where “dtmfmode” is set to rfc2833.

As for the audio problem, you should do a “list trace station xxxx” The x is for the extension number on Avaya. If you are using ASA then you will need to go in to terminal emulator to do this. Once list trace is running you can then place your test call and all information will start showing about that call. You will probably see a codec mismatch somewhere. That should help point you in the right direction. There is also a Wiki that I found where it walks through setting up h.323 with Asterisk and Avaya. You might want to google that and make sure your settings match with the Wiki. I used the Wiki and my connections are perfect. I am still thinking on what the problem could be and will post a reply if I can think of anything else to try.



Hi Perry,

Thanks a lot again.

  1. I have done with audio from Asterisk to Avaya and (as you said) it was merely the codec mismatch.

  2. I am not able to get how to send DTMF from Avaya and [color=red]how to capture it on asterisk side[/color]. In the Signaling Group - the DTMF over IP is already set to “out-of-band” and on asterisk side we are using DTMF as rfc2833. But how to capture the DTMF on the asterisk side i dont know.
    I am following the following link for my “Extension.conf” file, but this is not working for me.

Need more help on this front

Thanks & regards,
Vishal Bhardwaj.

Hi Perry,

One more querry.

What we are trying is part of a bigger task we want to accomplish: Try to setup a asterisk voice mail with our enterprise Avaya PBX(8300 Definity ACM 3.1). So on the asterisk side, we will need both the caller as well as the called party. We’ve read elsewhere about folks who have tried this using mode codes that some Avaya systems send. But any method of knowing the caller and the called party is fine with us.


  1. 3301 calls 3302 (both on Avaya)
  2. 3302 doesn’t pick the call.
  3. Route the call to an Asterisk (voicemail) number 1111 using a coverage path.

Problem is:

  1. how to send 3301, 3302 (source and destination on Avaya) to asterisk?
  2. how to read 3301 (as source) and 3302 (as destination) from Asterisk?

The h.323 trunk between Avaya and Asterisk is working just fine.

Thanks in advance,
Vishal Bhardwaj.

vishal_with_me, did you get * setup as a Audix replacement for your Avaya system? I am trying to do the same thing.

It sounds as if your remaining issue is more how to send CLI from the Avaya to the Asterisk system rather than specifically to do with DTMF tones.

It seems that you are trying to use the Partner setup for a S8300, which I doubt that the codes are the same for…You will have to do a lot of testing to get it to replace audix, but the asterisk console will help big time, as you can see the call come in to the channel and you can use NoOp to print whats coming in to see what you need to be watching for in your extensions.conf file to match them. The wiki works great for Partner systems, I have used it twice as a vm for a Partner. You will be able to see what he used to print to the console and implement the same thing, in order to get the dtmf coming from S8300