asterisk users - Jun 2005 - codec preference

Need some help understanding codec preferences:

I have 2 asterisk servers.
Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and 
allow=ulaw in iax.conf
Server 2 receives calls and routes them to server 1. It has the same 
allow lines.
We receive calls from a phone co and route them via server 2 to server 
1. The calls originate in g729 and everything works fine.

Now I want to take calls from FWD, which delivers calls in ulaw format, 
and have them routed by server 2 to server 1, which will play back a 
.gsm file. This doesn't work, and I get these diags on server 1:

       > requested format = ulaw,
       > requested prefs = (),
       > actual format = g729,
       > host prefs = (g729|gsm|ulaw),
       > priority = mine
    -- Executing Playback("IAX2/server1@xxx.xxx.xxx.xxx:4569-12", 
"demo-insruct") in new stack
Jun  8 09:51:55 WARNING[46848]: file.c:489 ast_openstream_full: File 
demo-insruct does not exist in any format
Jun  8 09:51:55 WARNING[46848]: file.c:793 ast_streamfile: Unable to 
open demo-insruct (format g729): No such file or directory
Jun  8 09:51:55 WARNING[46848]: app_playback.c:90 playback_exec: 
ast_streamfile failed on IAX2/server1@xxx.xxx.xxx.xxx:4569-12 for 
demo-insruct

demo-instruct.gsm does exist in the sounds dir.

So why doesn't server1 just accept the call in ulaw and play the 
message? Why does it force the call to g729?

Mark

Someone stop me I'm replying to posts again...

Anyway, your preferences are setup to prefer g729 over ulaw, and the other
end offered g729... so it was used first. Thus, change your order in
iax.conf so ulaw is first and it will magically start working magically!

- Joshua Colp.
(file in #asterisk on Freenode)

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark Willis
Sent: Tuesday, June 07, 2005 10:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] codec preference

Need some help understanding codec preferences:

I have 2 asterisk servers.
Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and 
allow=ulaw in iax.conf
Server 2 receives calls and routes them to server 1. It has the same 
allow lines.
We receive calls from a phone co and route them via server 2 to server 
1. The calls originate in g729 and everything works fine.

Now I want to take calls from FWD, which delivers calls in ulaw format, 
and have them routed by server 2 to server 1, which will play back a 
.gsm file. This doesn't work, and I get these diags on server 1:

       > requested format = ulaw,
       > requested prefs = (),
       > actual format = g729,
       > host prefs = (g729|gsm|ulaw),
       > priority = mine
    -- Executing Playback("IAX2/server1@xxx.xxx.xxx.xxx:4569-12", 
"demo-insruct") in new stack
Jun  8 09:51:55 WARNING[46848]: file.c:489 ast_openstream_full: File 
demo-insruct does not exist in any format
Jun  8 09:51:55 WARNING[46848]: file.c:793 ast_streamfile: Unable to 
open demo-insruct (format g729): No such file or directory
Jun  8 09:51:55 WARNING[46848]: app_playback.c:90 playback_exec: 
ast_streamfile failed on IAX2/server1@xxx.xxx.xxx.xxx:4569-12 for 
demo-insruct

demo-instruct.gsm does exist in the sounds dir.

So why doesn't server1 just accept the call in ulaw and play the 
message? Why does it force the call to g729?

Mark


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

what's the use of enabling multiple codecs for a peer?
can asterisk avoid trancoding when both phones are capable of a common codec
enabled for them?
e.g. first priority codec = g729, 2nd= ulaw,,, if phone 1 calls another ip
phone capable of g729 to use it and when calling through Zap, can asterisk
let the phone use ulaw?
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060428/5b9d5f09/attachment.htm