Dean @ INKnBITs wrote:
I have the same problem here, why does asterisk not use ulaw with Sip1
-> Sip3 ? As it has allow=g729 and allow=ulaw in Sip1, should it not
fallback onto ulaw when the g729 fails?
Thanks,
Dean.
-----Original Message-----
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of *Rosli
Sukri
*Sent:* 08 August 2006 13:38
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Problems with Codecs in Asterisk
either
1)pay digium for g.729 license or
2)allow g.729 for sip3
- sip 1 -> sip2 work cause it will pass thru,
- sip 2 -> sip3 fails because since asterisk wants to do transcoding
to 729<->711 and no license
if bandwidth is a concern just use GSM (if available as a codec on
the phone)
On 8/8/06, *Chan Kwang Mien* < [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Hi,
My test-setup is as follows :
sip1 <--> Asterisk <--> sip2
^
|-------> sip3
In sip.conf,
[sip1]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729
allow=ulaw
[sip2]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729
[sip3]
type=friend
host=dynamic
secret=pass
disallow=all
allow=ulaw
sip1 supports g.729 and g.711u only
sip2 supports g.729 only
sip3 supports g.711u only
sip1 is able to establish a call to sip2.
However, I have problem establishing a call from sip1 to sip3. sip3
rings but when I answered it, it hanged up.
The Logs are :
-- Executing Dial("SIP/2006-389a", "SIP/2003") in new stack
-- Called 2003
Aug 8 09:55:15 WARNING[6937]: channel.c:2725
ast_channel_make_compatible: No path to translate from
SIP/2003-b5f8(4)
to SIP/2006-389a(256)
-- SIP/2003-b5f8 is ringing
-- SIP/2003-b5f8 answered SIP/2006-389a
Aug 8 09:55:16 WARNING[6937]: channel.c:2725
ast_channel_make_compatible: No path to translate from
SIP/2006-389a(256) to SIP/2003-b5f8(4)
Aug 8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full:
Had to
drop call because I couldn't make SIP/2006-389a compatible with
SIP/2003-b5f8
== Spawn extension (phones, 2003, 1) exited non-zero on
'SIP/2006-389a'
I think the codecs used by sip3 and sip1 are incompatible. Does
anyone
know how I could make them compatible ?
I believe the issue is this...
When sip1 initiates a call, a codec is selected based on the sip phone
preference and asterisk codec "ordering". That selection has nothing to
do with "where" the call is going to be directed (eg, sip2 or sip3).
That negotiation happens early, otherwise you would not be able to hear
busy & congested tones, audio messages, etc.
"After" that negotiation happens, then asterisk begins processing the
call by doing the same thing with the destination sip phone. In other
words, asterisk negotiates an appropriate codec with sip2 (or sip3) that
is based on that phone's codec preference and what asterisk's codec
ordering for that sip phone definition.
"After" both of the above steps are completed, asterisk then tries to
bridge the two calls, and if you don't have the g729 codec installed, it
can't bridge ulaw to g729. There is no more codec negotiation going on
after step 1 and 2 above.
The above can easily be verified by simply doing a "sip debug" and
placing a call.
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