Summary:ASTERISK-08804: Error at call from h323 to SIP
Reporter:dmoreno (dmoreno)Labels:
Date Opened:2007-02-14 06:25:03.000-0600Date Closed:2011-06-07 14:00:51
Versions:Frequency of
Environment:Attachments:( 0) messages.tar.bz2
( 1) messages-150207.tar.bz2
( 2) messages-190207.tar.bz2
Ubuntu Dapper (Linux 2.6.15-27-686 #1 SMP PREEMPT i686 GNU/Linux)
Asterisk 1.4.0 release

Asterisk configuration as sip gateway with gnugk as h323 gatekeeper (all in the same host -> There is no nat in this scenario.
Some config files:

* h323.conf (complete file):
port = 1730
bindaddr =
gatekeeper =
AllowGKRouted = yes
AcceptAnonymous = yes

* sip.conf (only h323 references):
exten => _.,1,NoOp(Incoming Call from h323 protocol ${CALLERID} for ${EXTEN})
exten => 11,1,NoOp(Call to lolo)
exten => 11,n,Dial(SIP/lolo)
exten => 22,1,NoOp(Call to pera)
exten => 22,n,Dial(SIP/pera)
exten => 33,1,NoOp(Call to diego)
exten => 33,n,Dial(SIP/diego)

* gnugk.ini (default config file with only one addon):

Now, I call from a h323 sotfphone (Polycom PVX 8.0.0;, dialing 33 extension, to a SIP softphone (Ekiga-2.0.1 registered as sip user diego;). Then, the SIP phone rings and when I pick the phone the call is finished with this error:

[Feb 12 17:34:16] ERROR[6311]: chan_h323.c:1848 external_rtp_create: Unable to find call ip$

ERROR: on_external_rtp_create failure


I attach a log file with this options:
core set verbose 3
core set debug 2
h323 set debug
h323 set trace 5
Comments:By: Olle Johansson (oej) 2007-02-14 08:49:25.000-0600

the sip.conf above looks more like extensions.conf

By: dmoreno (dmoreno) 2007-02-14 09:46:44.000-0600

Exactly, but I do not find the way to edit the report...

By: Paul Cadach (pcadach) 2007-02-14 12:41:52.000-0600

As pointed in channels/h323/README, you shoud:
a) point exact bindaddr in /etc/asterisk/h323.conf; or
b) set hostname to point to your networked IP address.

In your trace Asterisk uses address for media streams because there is no bindaddr in /etc/asterisk/h323.conf (or it points to, so address taken from hostname map.

Also, your system don't have G.729 codec required by H.323 side:
[Feb 14 12:21:37] WARNING[29419] channel.c: Unable to find a codec translation path from g729 to ulaw
[Feb 14 12:21:37] WARNING[29419] channel.c: No path to translate from H323/ip$ to SIP/diego-0823c088(4)
[Feb 14 12:21:37] WARNING[29419] app_dial.c: Had to drop call because I couldn't make H323/ip$ compatible with SIP/diego-0823c088

To resolve this, update your /etc/asterisk/h323.conf with list of supported codecs, or order G.729 codec...

By: Paul Cadach (pcadach) 2007-02-14 12:43:46.000-0600

Please, drop a short note if suggestions I made was helpful and this bug can be closed. Otherwise, please provide additional information (trace) after solving existing problems in your installation.

By: dmoreno (dmoreno) 2007-02-15 06:31:16.000-0600

I changed the h323.conf with:
bindaddr =

And the error continues. I attached the log file like messages-150207.tar.bz2

By: Paul Cadach (pcadach) 2007-02-17 02:18:50.000-0600

In second case you don't have G.729 codec installed:
[Feb 15 12:12:38] WARNING[28795] channel.c: No path to translate from H323/ip$10
.10.10.6:3230/5103(256) to SIP/diego-0823c128(4)


By: dmoreno (dmoreno) 2007-02-19 06:48:01.000-0600

I installed the g729 codec and I changed the h323.conf with:
bindaddr =

The call is established and the audio is OK but the video is not transmited. I attached the log file.

Thanks for the attention, Paul.

By: Paul Cadach (pcadach) 2007-02-19 12:09:16.000-0600

chan_h323 isn't support video, but this is in TODO list...