[asterisk-users] ooh323 does not work fine, what about h323 channel

[Faisal Hanif]

Hi,

As per my experience YATE is the best option for H323<=>SIP Proxy.

Regards,

Faisal

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, July 07, 2011 2:48 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] ooh323 does not work fine, what about h323 channel

Hi All;

The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by
selecting the add-on). But really does not work in good performance, for
example: if a call came from gnugk to asterisk and the ooh323 handled it,
the performance is bad .. some calls are drop and if it is ringing, then it
rings for small duration and then stop ringing ....

In other words, if the call went from gnugk to the provider directly (all
the path h323), it is better than coming for Asterisk via the ooh323 channel
and then to be translated for SIP to be sent for provider.

I would like to try the h323 channel (and not the ooh323), but I do not know
what I have to do to compile? Any advise?

Did anyone tried yate to do the translation from h323 to sip? How it is?

Regards
Bilal

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