Displaying 20 results from an estimated 100 matches similar to: ""CBAnn" channel not going away in Asterisk 12"
2014 Oct 21
1
[asterisk-user] Confbridge Kick Action
Hi All,
I am working on Asterisk 12.6.0 with ConfBridge module, when there are
multiple user like admin and normal participant running with conference.
When I try to kicked 2 user (Normal User), it play file "conf-kicked" and
again join conference
My scenario in confbridge like.
1] Admin User (e.g. SIP/8484-00000000)
2] Normal User (e.g. SIP/8484-00000001)
3] Admin User (e.g.
2018 Feb 21
2
Asterisk crash on core show channel
Hello Asterisk list,
I am facing some Asterisk crashes which are consistently pointing to the
same backtrace, which is the following (using DONT_OPTIMIZE,
BETTER_BACKTRACES and MALLOC_DEBUG):
Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)):
#0 0x00007f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6
#1 0x00000000004a91ca in cdr_object_get_by_name_cb ()
#2 0x0000000000463c60 in
2017 Nov 14
2
Confbridge SFU for Asterisk 15
On 11/14/17 4:27 PM, Joshua Colp wrote:
> On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote:
>> On 11/14/17 3:55 PM, Joshua Colp wrote:
>>
>>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:
>>>> I followed the blog post and I can get video from the conference if
>>>> I configure the bridge as follow_talker so I know everything
2018 Feb 21
2
Asterisk crash on core show channel
Thanks for you answer Marcus,
So maybe this means some bug was fixed? Anyone aware of something related?
>From the release notes, I couldn't find any direct change that could fix
this....
Thanks,
Kind regards,
Patrick Wakano
On 21 February 2018 at 20:29, Marcus Kvarsell <Marcus.Kvarsell at fogwise.se>
wrote:
> Hello, i found upgrading to asterisk 15 helped.
>
>
>
>
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
2020 Aug 18
2
Channels freeze on Confbridge
I am having a strange problem. We have an Asterisk 16.12.0 server
(we have upgraded at least two versions since we found the problem)
where users complain that confbridge calls end after about 30 minutes or
so. The problem is that according to Asterisk the calls are still
active. All users are cut off at the same time but a "core show
channels verbose" still shows channels as
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 May 07
0
Video with asterisk12 and pjsip
Hi,
I tried to turn on Video and get the following cli-WARNING output
-- Executing [8600 at outgoing-kamailio:1] Answer("PJSIP/7000-00000000",
"") in new stack
> 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to
192.168.8.203:17200
-- Executing [8600 at outgoing-kamailio:2]
ConfBridge("PJSIP/7000-00000000", "8600") in new stack
--
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
2008 Aug 09
1
how to know what codec is being used
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does
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:
#
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
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
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
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
2017 Nov 14
2
Confbridge SFU for Asterisk 15
On 11/14/17 3:55 PM, Joshua Colp wrote:
> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:
>> I followed the blog post and I can get video from the conference if
>> I configure the bridge as follow_talker so I know everything is working
>> on the pjsip side. The only problem is that video_mode = sfu is
>> apparently not valid in either confbridge.conf or
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