Displaying 20 results from an estimated 2000 matches similar to: "how do I know what codec is being used"
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:
#
2003 Nov 06
3
which channel format number is right?
Hi all,
if i enter a "show codecs" at cli * response with:
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 << 4) MPEG-2 layer 3
32 (1 << 5) ADPCM
64 (1 << 6) 16 bit Signed Linear PCM
128 (1 << 7) LPC10
2004 Dec 21
6
Caller ID - TE405P - Telstra Onramp 10 - Australia
I am having problems getting incoming caller id to work on a Telstra
Onramp 10.
I have changed "/DEFAULT_CIDRINGS 2"/
Is there something i'm missing ?
My Cisco 7960 just shows "asterisk"
Thanks,
Nathan
[zapata.conf]
context=incoming
usecallingpres=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
busydetect=no
pridialplan=local
usecallerid=yes
callerid=asreceived
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
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
2004 Oct 07
1
spandsp RxFAX problems.
Hello,
Anyone else experiencing problems with the latest spandsp (pre3)
and last libtiff beta? I'm getting 8 bytes long file, with the
TIFF header only during such connection:
-- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1
-- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack
-- Executing
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
2005 Oct 05
4
dropped calls when g729 is used on sip leg
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a pretty rigorous torture over the last 4 months, and
they've performed famously. No dropped calls ever.
We invested in some g729 licenses. changed my ipmid.cfg so that g729 is
priority 1 and ulaw is priority 2. I added allow=g729 to my extension's
sip.conf entry, where existed before disallow=all
2003 Aug 26
1
H.323 channel problems
I have hit a problem where chan_h323 sometimes doesn't hang up properly
and stays stuck in the "Up" state, with asterisk consuming 100% of CPU:
*CLI> show channels
Channel (Context Extension Pri ) State Appl. Data
H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None)
1 active channel(s)
*CLI>
2011 Jun 20
2
different format in asterisk
Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables
1. chan->readformat
2. chan->writeformat
3. chan ->rawreadformat
4. chan ->rawwriteformat
5. chan->nativeformats
Thanks
Nikhil
2004 Sep 25
1
How can I dial one unbusy channel of 4 available?
Hi.
I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any
of the 4 PSTN lines connected to the asterisk box from and dial your
number, and asterisk will dial out through one of the 4 sip accounts I
have on a SIP -> PSTN provider. I think of something like this in the
extensions.conf
[incoming]
exten => s,1,Wait,1 ; Wait a second, just for
2014 Apr 29
1
"CBAnn" channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter
and then exit a conference room, I see:
-- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c>
-- Channel CBAnn/207-0000067f;2 left
2017 Nov 22
2
Chan Local, Originate and slin
On Wed, 22 Nov 2017, Dmitriy Serov wrote:
> ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${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
2004 Nov 03
3
zt hook failed: Device or resource busy
Hello,
I ordered the Devel lite kit, and installed it.
I am just trying to get the FXO port to work, and am having trouble.
To load the card I do the following.
modprobe wcfxs
modprobe wcfxo
ztcfg -vv
asterisk -vc
My /var/log/asterisk/messages show
Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy
Here is my /etc/zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
2009 Jan 28
2
How to retrieve a phone number from call forwarding?
Hi,
I'm very new to Asterisk and I have the following scenario.
1. Let's say I have a number of 1-222-222-2222 from my SIP service provider
(VoicePulse).
2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail
to the number provided by SIP service provider (1-222-222-2222).
3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a
voicemail message.
2004 Jun 29
4
Getting Asterisk to automatically dialout
Hi,
I'm trying to get asterisk to auto-dail out. I created a *.call file
with the the top of it being "Channel: Zap/1/2609944", which should have
connected to Zap channel 1 and dial out to 2609944, but It did not do
so, asterisk would say a call was completed to Zap/1/2609944 but I never
heard that phone ring. So I tried just putting "Channel: Zap/1" at the
top of
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
2005 Apr 05
1
problem with remote forward and SSH 2.4.0 server
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I'm having a problem with OpenSSH_4.0p1 when trying to do remote port
forwards to a server running SSH Secure Shell 2.4.0. The problem appears
to be that 2.4.0 chokes on bind addresses that aren't numeric addresses,
such as "localhost" and "".
The following commands are failing for me from the 4.0p1 client to the
2.4.0