Displaying 20 results from an estimated 600 matches similar to: "Know who's logged in"
2006 Apr 10
1
SIP channel unavailable/busy/really not there
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's in my config or not (if that makes sense, basic automap
of dial-in lines to sip phones, but if they've
2010 Jul 23
2
application call to Gosub affects flow of control, and needs to be re-written using AEL
Hi,
For some reason (outbound call tracking) I've got a few different
outbound call process (using a macro for queuemetrics logging, or direct
call)
i wanted to factorise the routing process so i came up with something
like the following. All in one it's working like expected, however
every "ael reload" command trigger a lot of warning like that
"application call
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
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
2011 Mar 15
4
[1.4] Asterisk doesn't hang up?
Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
========== extensions.conf
;Play MoH for a few seconds, hang up, and
;check ChanIsAvail() able to detect when line idle again
exten => 8888,1,Answer()
exten =>
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
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
2015 Sep 14
2
AgentLogin() on the multiple servers?
Hello,
Let say all the SIP devices will be registered on the proxy like kamailio.
Agent is a member of Support and Billings Queues on the asterisk servers.
Support queue on "Server A" and Billings Queue on "Server B" for example.
This will be done via RealTime Queue.
I want Agent to dial 1234 on a sip device and it will prompt to enter a pin
number to Login via
2004 Aug 12
1
AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060>
--
2010 Aug 09
1
op_div: non-numeric argument
Ladies, Gentlemen
We are experiencing an unusual problem in our asterisk 1.4.34.. We are
attempting to determine if channels are in use before paging to them.
This works correctly, as in it pages the phone.. however, we see the error
message below on the console... after googling, we discovered limited
information regarding the issue...
-- Executing [NPANXX7298 at from-pstn:1]
2003 Jun 04
2
AgentLogin
Hi,
We recently installed asterisk too replace our office PABX, however we
are finding it hard to get documentation on the way Agents login.
In the agents.conf we have setup a user of agent => 1003,4444,Test. In
the extentions.conf file we have added in exten => 9,1,AgentLogin.
We can dial 9 and key in the uid and password which logs us in. The
asterisk CLI confirms it is login, but when
2003 Aug 28
12
Asterisk stops responding
Anyone have any thoughts on why versions of asterisk I try (4 so far)
after CVS-07/18/03 always end up locking up on me... which means no sip
clients can register/re-register and if I type "reload" or "stop now" at
the cli it just returns and does nothing.
I have experienced this same issue on three separate boxes. Two running
RedHat 9 and one running Redhat 8.
I don't
2008 Aug 01
3
Asterisk Queues problem
Hi,
I have Asterisk 1.4.18 and I have been running call center queues on it.
Today it suddenly stopped adding inbound calls to queues. I am facing
with following error: app_queue.c:3939 queue_exec:
unable to join queue "myqueue"
In extension file:
Queue(myqueue|t|||120)
And my agents are joining in following
2007 Jun 25
1
Problems with ChanIsAvail always return status 0
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
[macro-callonlyiffree]
exten => s,1,ChanIsAvail(${ARG1}|s)
exten => s,n,NoOp(${AVAILCHAN})
exten
2004 Jul 16
1
Patch to test: Never say goodbye to an agent :-)
http://bugs.digium.com/bug_view_page.php?bug_id=0001693
This patch adds a lot of options for AgentLogin/AgentCallbackLogin
Please test and respond in the bug tracker!
/O
-------------------------------------------------------------------------------------
"This patch adds quite a few new features into __login_exec () of channels/chan_agent.c for AgentLogin() and AgentCallbackLogin(). Only
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
2009 Mar 11
2
how to configure for incoming message-summary SUBSCRIBE
Hi!
AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE -
but how should I handle the SUBSCRIBE in the context?
thanks
klaus
SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:u+431234567 at 11.111.11.11:39982>
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