Asterisk & 3 flavors of h323 - - audio and / or dtmf pro

dt; 12/19/2005
sbj; Asterisk & 3 flavors of h323 - - audio and / or dtmf problems

I need to use an h323 Voip connection from Asterisk s/w pbx to Cisco router / voice gateway, with a low bandwidth codec such as g729.  Thus far I’ve tried out 3 h323 code flavors but all had audio or dtmf difficulties.  These were chan-h323, oh332., and ooh323.  The latter is the most recent, with Asterisk / Asterisk Addons 1.2.1, Fedore core 4 Linux.  (Suse8 Linux was used with the prior 2.  Pwlib, Openh323 versions selections for those 2 followed the documents specs.)  
- - octimotor - -

I have successfully integrated ooh323/Asterisk with an Avaya Prologix and have managed to get DTMF and signalling working with no problem. I am using G711, as it appears the G729 on the Avaya is a bit funky.

sbj; Asterisk & 3 flavors of h323 - - audio and / or dtmf prolem

In Asterisk, /etc/asterisk/ooh323.conf, which of the 4 options for dtmfmode= <value> did your ip phone accept?  With ooh323 connecting Asterisk to a Cisco gateway, employing g711/ulaw for tests, NONE of the 4 value options allowed dtmf digits to be recognized by the router script, although I could hear the ivr audio continuing to ask for the first input.  	

I did not find a counterpart value available in ooh323.conf, as exists for the comparable setting within oh323.conf, userInputMode=TONE - - usage of h245 tone.  My issue with oh323 then was that the caller could hear audio from the ivr only when ulaw codec, not the Digium g729 codec, was employed. (Oh323 and g729 did still pass dtmf, despite audio silence, the debugs proved.)  

	- - octimotor - - 11:06 hr, gmt - 5; 12/20/2005

I too am having issues getting ooh323 & DTMF working with an Avaya PABX.

I’ve read elsewhere to use the Woomera H323 channel driver instead as it allows DTMF inline.

Would appreciate any assistance with this.

With v1.2+ of Asterisk I was able to acheive DTMF with the Avaya and ooh323 (no longer having to use Woomera). I will have a nother look at my config on both sides and post it here.

That would be great MuppetMaster.

I can see the tones being sent by monitoring the Avaya log, but nothing registers when I monitor the ooh323 debug output…

According to the latest version of ooh323 it fixes the following:

[quote]Release 0.8.1

  1. Fixed bug causing dialed digits alias being set as a char string.
  2. Added close for cmdListener
  3. Fixed called party number set.
  4. Other minor FreeBSD support updates.[/quote]

I’m yet to see if this will potentially solve my problem, has anyone used the latest ooh323 & asterisk 1.2.1 with any Avaya systems?

Thanks :laughing: