Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap
That's one of the uses of the SIP_CODEC dialplan variable. Just set it in
the context or the sip.conf or users.conf. In your particular case, just
set up a specific context for the IAX calls
[iax-in]
Exten => _X.,1,Set(SIP_CODEC=G722)
Exten => _X.,n,answer()
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jaap Winius
Sent: Tuesday, November 15, 2011 8:47 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Forcing a CODEC
Hi folks,
How can I take advantage of a high-bandwidth CODEC, like G.722, for internal
communications at my site, but use G.711 (alaw/ulaw) for all other outgoing
calls? I need G.711 to support Inband DTMF signaling.
As my site has multiple locations that are tied together with IAX trunks, I
was hoping that it would be possible to specify alaw and ulaw as the first
two CODEC choices for the SIP phones, as well as in their sip.conf
configurations, but that I could use the IAX trunks (with bandwidth=high) to
force the phones to use their third CODEC choice, g722, because that would
be the only CODEC specified for the IAX trunks (following disallow=all).
Unfortunately, that doesn't work. Although the Asterisk console reports that
g722 is being used, when I listen to the connection it's obvious that a
G.711 CODEC is being used. Curiously, the reverse does
work: if g722 is specified as the first CODEC of choice for the phones, it
is possible to use the IAX trunks to force them to use alaw/ulaw instead.
Is a solution to this problem?
I'm using Debian squeeze with Asterisk 1.6.2.9.
Cheers,
Jaap
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Remove all other codec On Nov 15, 2011 8:17 PM, "Jaap Winius" <jwinius at umrk.nl> wrote:> Hi folks, > > How can I take advantage of a high-bandwidth CODEC, like G.722, for > internal communications at my site, but use G.711 (alaw/ulaw) for all other > outgoing calls? I need G.711 to support Inband DTMF signaling. > > As my site has multiple locations that are tied together with IAX trunks, > I was hoping that it would be possible to specify alaw and ulaw as the > first two CODEC choices for the SIP phones, as well as in their sip.conf > configurations, but that I could use the IAX trunks (with bandwidth=high) > to force the phones to use their third CODEC choice, g722, because that > would be the only CODEC specified for the IAX trunks (following > disallow=all). > > Unfortunately, that doesn't work. Although the Asterisk console reports > that g722 is being used, when I listen to the connection it's obvious that > a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is > specified as the first CODEC of choice for the phones, it is possible to > use the IAX trunks to force them to use alaw/ulaw instead. > > Is a solution to this problem? > > I'm using Debian squeeze with Asterisk 1.6.2.9. > > Cheers, > > Jaap > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111115/cc015148/attachment.htm>