Displaying 20 results from an estimated 500 matches similar to: "how to know what codec is being used"
2017 Nov 22
3
Chan Local, Originate and slin
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin
2017 Nov 22
2
Chan Local, Originate and slin
Again - when Originate is run from dialplan, i get:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin at 8000)->(slin at 192000)
ReadTranscode: No
When it's made with a call file (no matter how a call file is created), I
see
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Please
2016 Jun 30
3
how to join 2 channels using AGI/AMI
sorry for top-posting, the two topics started with 2 different reason
subject, but then we finished on the same problem.
btw,the 2 show channel are reported above:
the channel with DTMF working
kcenter*CLI> core show channel SIP/pbx2-000004b9
-- General --
Name: SIP/pbx2-000004b9
Type: SIP
UniqueID: 1467323106.1275
Caller ID: xxxx
Caller ID Name: xxxx
2014 Apr 29
1
"CBAnn" channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter
and then exit a conference room, I see:
-- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c>
-- Channel CBAnn/207-0000067f;2 left
2009 Mar 26
3
Know who's logged in
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the agent
logs in, a channel keeps running all the time that the agent is logged
in to receive the incoming calls. How do I know which agent logged in
(code)? Right now, if I query the login channel, there is no information
about which agent is logged on:
#
2015 Sep 30
3
Change Asterisk MulticastRTP codec
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data
2017 Nov 22
2
Chan Local, Originate and slin
On Wed, 22 Nov 2017, Dmitriy Serov wrote:
> ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
> /var/spool/asterisk/outgoing/${number}-${confnum})
I get:
Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/...
Unknown keyword 'ActionID' at line 2 of
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
2014 May 07
1
asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more.
I tried every combination. silent on both sides.
I compiled pjsip with no resample in pjsip.
./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr
is there a way to force asterisk back to do the codec translation?
Attachment:
sip show channel of the
2016 Jun 30
2
how to join 2 channels using AGI/AMI
this is the point, and the strange thing:
DTMF is set to rfc2833, but is working both on incoming and outgoing calls,
it is not working only on calls generated with the Originate AMI command,
or with the queue member that point to Local dialplan, as you suggested
2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>:
> Looking at your logs it looks like you may need to
2012 Feb 17
0
Calling SIP extension through Local/XXXX negotiates slin codec instead of gsm
I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga
on Fedora 16 x86_64 for my tests.
[root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf
2010 Jul 31
0
MeetMe transcode / format problem
Hi Group,
actual i have a transcode problem and i have no idea to solve this. All my wav files are alaw encoded and i allow only alaw codec.
But sometimes the WriteFormat is slin and if i recall the same number the WriteFormat is alaw for the Channel.
Why the channel has sometimes slin and sometimes alaw?
NativeFormats: 0x8 (alaw)
WriteFormat: 0x40 (slin)
ReadFormat: 0x8 (alaw)
WriteTranscode:
2011 Apr 14
0
Followme() and variables
We have a variable set for each user/peer/whatnot that signals what the
outbound caller-id should be sent as with our carrier.
When someone dials a followme extension, this does not appear to be carried
over for when the calls reach an outside caller, and we see the outbound
caller-id being set as 'asterisk' vs the number desired.
Has anyone else seen this, or found a way to
2009 Feb 25
1
Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and
2023 May 05
0
Calls running forever / CDRs inaccurate
Hi list!
Running Asterisk 20.0.0 on CentOS 7, logging CDRs using
cdr_adaptive_odbc to mariadb-server-5.5.68 (via
mariadb-connector-odbc-3.1.7-ga-rhel7)
Using chan_sip.
I'm facing the problem when there is a sudden spike of calls, some of
the calls that are being made during those spikes hang forever
basically. This looks like this:
[root at voip]# asterisk -rx 'core show channels
2015 Jul 06
0
Unisteam not showing callerid
hi list
can U help me
caller id in USTM if now working
-- Starting switch on '4211 at 4211-1' to 4203
-- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0",
"") in new stack
Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0:
================================================================================
Info:
Name=
2014 Mar 11
1
PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Hello,
I have installed the latest version 12 that has been released (12.1.0.rc3).
I have setup default dtmf mode (rfc47..) but when I am calling to a
endpoint that doesn't support it (no telephony event in the rtpmap) the
asterisk responds OK in the signalling but DTMF is not working.
Is it a known issue?
Below you can see the output of the asterisk monitor.
<--- Received SIP request
2007 Sep 20
0
Video doesn't work for outgoing call?
I've tried to put a call file to /var/spool/asterisk/outgoing/
to make an outgoing video call, but not succeeded.
I could hear the audio, but no video.
The asterisk version is 1.4.10, with videosupport=yes
The client is eyebeam 1.5.7, with h263 support.
Here are some debug messages.
It shows the client and asterisk negotiated the video capabilities
without problem. However, the 'show
2011 Jun 20
2
different format in asterisk
Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables
1. chan->readformat
2. chan->writeformat
3. chan ->rawreadformat
4. chan ->rawwriteformat
5. chan->nativeformats
Thanks
Nikhil
2015 Apr 08
1
Help debugging a possible SIP channel leak in 11.17.0, possible race condition
Have you tried Asterisk 13? The bridging mechanism has been completely
rewritten on Asterisk 12, so there's no longer channel masquerading and
zombie channels. Might be worth a try.
2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>:
> El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?:
>
> I am trying to collect enough information about an