I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I get a complaint from the server:
-- Call accepted by 66.225.202.72 (format g729)
-- Format for call is g729
Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find
a path from g729 to slin
. . . .
I get ringback from Nufone, but as soon as the call answers I get an error:
Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find
a path from g729 to slin
. . .
What am I doing wrong to cause it to want to transcode? I assume that's
where the complaint is coming from. I thought Asterisk could pass
through without transcoding as long as the endpoints are all g729.
Thanks.
B.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brian
Capouch
Sent: Sunday, April 24, 2005 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] g729 passthrough?
I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I get a complaint from the server:
-- Call accepted by 66.225.202.72 (format g729)
-- Format for call is g729
Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find
a path from g729 to slin
. . . .
I get ringback from Nufone, but as soon as the call answers I get an error:
Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find
a path from g729 to slin
. . .
What am I doing wrong to cause it to want to transcode? I assume that's
where the complaint is coming from. I thought Asterisk could pass
through without transcoding as long as the endpoints are all g729.
Thanks.
B.
;;;;;;;;;;;;;;;;;;;;;;;
Brian,
Add to the [general] section in sip.conf the following:
disallow=all
allow=g729
allow=ulaw
allow=alaw
For some reason Asterisk will not pass audio through itself without trying
to transcode unless you have this in your config.
Don't ask me why it will not work with allow=g729 under the individual peer.
This has to go in the [general] section.
James Taylor
MetroTel
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Great info!!!. Thank you all guys. Regards, Sebastian ---------- Forwarded message ---------- From: Sergio Serrano <sergio.serrano@avanzada7.com> Date: Mon, 19 Sep 2005 18:38:55 +0200 Subject: RE: [Asterisk-Users] Asterisk in Spanish To: smilioto@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Try in www.asterisk-es.org -----Mensaje original----- De: Sebastian Milioto [mailto:smilioto@gmail.com] Enviado el: lunes, 19 de septiembre de 2005 15:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Asterisk in Spanish Hi all, I've been installing Asterisk@Home and (of course) all the "answering machine" (I don't sure that's the right word in english, "preatendedora" in spanish) speech is in enlgish languaje. Is there anyway to download all those .gsm files speaked in spanish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: sebamilioto@hotmail.com e-mail: smilioto@GMAIL.com Mobile: 549 3571 543658 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.1/104 - Release Date: 16/09/2005