Alejandro Mejía Evertsz
2006-Apr-19 14:44 UTC
[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
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alejandro
Mej?a Evertsz
Sent: Thursday, April 20, 2006 12:44 AM
To: asterisk-users@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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