Displaying 5 results from an estimated 5 matches for "preferred_codec_on".
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preferred_codec_1_
2015 Feb 27
0
Reply to INVITE with 1 codec
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to "yes" the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile.
But in version 13.1 (I think version 11.2 also) is not working like that , it keeps sending all the codecs and sometimes both parties pick a different one causing one way...
2011 Aug 02
1
Codec negotiation issue (no audio format found to offer)
Running build 1.8.5.0 (compiled from source) I seem to be having an issue
with codec negotiation. I have a Grandstream HT503 FXO port connected to a
pstn line, a Polycom SP501, and a SIP trunk with callwithus.
What I'm essentially looking to accomplish is for ulaw or g729 (preferably
ulaw) to be used to the Grandstream FXO or any other internal endpoint, and
for g729 only to be used outbound
2014 Mar 24
1
Problem with TLS/SRTP with Asterisk 11.8.1
...general]
context=guest
allowguest=no
allowoverlap=no
allowtransfer=no
bindaddr=0.0.0.0:5060
udpbindaddr=0.0.0.0:5060
tcpenable=no
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
transport=udp
preferred_codec_only=no
disallow=all
allow=ulaw
language=en
trustrpid=no
dtmfmode=rfc2833
videosupport=no
alwaysauthreject=yes
directmedia=no
jbenable = yes
jbforce = no
[encrypted]
type=friend
secret=1234
context=internal
callerid="Encrypted" <1002>
host=dynamic
qualify=yes
canreinvite=no
dtmfmode=r...
2019 Jul 11
0
AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
...through where a remote
endpoint has requested T.38.
For versions of Asterisk 13 before 13.21.0 and Asterisk 15
before 15.4.0 the “preferred_codec_only” option must also
be set to “yes”. If set to “no” the crash will not occur.
Resolution If T.38 faxing is not required this functionality can be
disabled by ensuring the “t38pt_udptl” is set to “no” so a
T.38 reinvite is not...
2017 May 13
2
pjsip: asterisk can't decide which codec to use
On 05/12/2017 at 08:49 PM, Joshua Colp wrote:
> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
>
> <snip>
>
>>
>> If I'm doing exactly the same call originated with another extension,
>> there can't be seen these frequent changes. But the strange thing is,
>> that in both cases the part between extension and asterisk doesn't show