[Asterisk-Users] Codec problem from SIP to H323

[Alejandro Mejía Evertsz]

I tried by just upgrading to Ast1.2.4 but same problem.
Then I tried to install OH323 but I have this error when compiling :S

chan_oh323.c: In function `reload_config':
chan_oh323.c:4677: warning: implicit declaration of function `sscanf'
chan_oh323.c: At top level:
chan_oh323.c:3244: warning: 'update_call_ids' defined but not used
gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o \
-L../wrapper -loh323wrap_s \
-L/usr/src/openh323_v1_17_1/lib -lh323_FreeBSD_x86_r_s \
-L/usr/src/pwlib_v1_9_0/lib -lpt_FreeBSD_x86_r_s \
-lstdc++ -lldap -lldap_r -llber -lpthread -lssl -lcrypto -lexpat
/usr/bin/ld: cannot find -lldap
gmake[1]: *** [chan_oh323.so] Error 1
gmake[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.3/asterisk-driver'
gmake: *** [subdirs_build] Error 1

Does anyone know how to get rid of that?

I'm running:
OS: FreeBSD 6.0
Asterisk: 1.2.4
OH323: 0.7.3

Thanks in advance.

Alejandro.

-----Mensaje original-----
De: Oliver Vermeulen [mailto:oliver at wvg-tele.com] 
Enviado el: Wednesday, April 19, 2006 4:09 PM
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
CC: amejia at carmeltelecom.com
Asunto: RE: [Asterisk-Users] Codec problem from SIP to H323

Try to upgrade asterisk to version 1.2.4 

Are you using OH323 or H323 ?

I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323
and everything worked fine.

Cheers,
Oliver



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alejandro
Mejía Evertsz
Sent: Thursday, April 20, 2006 12:44 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Codec problem from SIP to H323

Hello.

I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:

- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf has "disallow=all & allow=g729"
- h323.conf has "disallow=all & allow=g729"

The problem:

[SIPphone]		[sip.conf]		[h323.conf]
[H323gw]
g729		--->	allow=g729	--->	allow=g729	--->	g729

When I dial to the gateway from the SIPphone using g729 as my sip phone's
default codec I get:

    -- Executing Dial("SIP/amejia-8be1", "H323/######@H323gw") in new stack
Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator
path exists for channel type H323 (native 4) to 256
Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to
create channel of type 'H323' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)


I don't get it why is it trying to "translate" anything. There's nothing to
translate, cause I'm using g729 in both ends.
Well, to make it more interesting, I tried this way:

[SIPphone]		[sip.conf]		[h323.conf]
[H323gw]
g711		--->	allow=all	--->	allow=all	--->	g729

This way, it passes the call to the gateway just giving a waring that it
can't find a codec to translate. But at least it passes the call.
It rings on the other side, and of course as I don't have any g729 licenses
installed it drops the call when answered.

    -- Executing Dial("SIP/amejia-1fc8", "H323/######@H323gw") in new stack
    -- Called #######@H323gw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
    -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8
    -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8
    -- H323/H323gw-2 is ringing
    -- H323/H323gw-2 answered SIP/amejia-1fc8
Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible:
No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256)
Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop
call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2
  == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8'


Does anybody know how can I get rid of the problem I get on the first
scenario?
Why does it try to use codec 4 (g711u) if both ends are configured with
g729?

Please give me some light. I don't know what else to try.

Thank you all.

Alejandro Mejia

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