How to force codec transcoding?

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Hi, we are working on a new codec for asterisk. It's early stages and
the main goal is just to be able to "hear it" using normal client end
points. So what I want is to be able to take a couple of normal G711
extensions, but have asterisk internally force a transcode to my new
codec. In this way we can easily run up a bunch of listening tests with
a wide variety of users very easily Ok, so how do I force this transcode in the middle, ideally without
using two asterisk…

Asterisk Users 3.3 years ago 0 Answer

Asterisk 10.3.0 Now Available

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The Asterisk Development Team has announced the release of Asterisk 10.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix potential buffer overrun and memory leak when executing "sip show peers" (Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey) * --- Fix ACK routing for non-2xx responses. (Closes issue ASTERISK-19389.) * --- Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (Closes issue…

VoIP News 3.3 years ago 0 Answer

Asterisk 1.8.11.0 Now Available

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The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you! The following are the issues resolved in this release: * --- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey) * --- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.) * --- Remove possible segfaults…

Asterisk Users 3.3 years ago 0 Answer

Mute DTMF

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I have been breaking my head on this, can't find a solution. Anyone know a way to mute DTMF on SIP? I have already tried changing the
dtmfmode option and messing with different codec/dtmfmode settings but so
far, not having any luck. Not even sure changing the codec is an option but pulling at straws at the
moment. Thanks in advance for any assistance.

Asterisk Users 3.3 years ago 1 Answer

DAHDI works, but returns CHANUNAVAIL ??

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DAHDI 2.6.0, dahdi show status
Description Alarms IRQ bpviol CRC
Fra Codi Options LBO
Wildcard TDM400P REV I Board 5 OK 0 0 0
CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Dahdi 1 is an internal extension, dahdi 4 is pstn. This call completes. But DAHDI comes back with CHANUNAVAIL. This a
problem since we then test for CHANUNAVAIL to use an alternative provider.

Asterisk Users 3.3 years ago 1 Answer