Displaying 20 results from an estimated 700 matches similar to: "stuck channel"
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
2006 Jan 07
1
choppy music on hold - only on PRI PSTN
Hello to all.
I do not know what is causing choppy music on hold when call comes in
through E1 card (PRI).. but this channel info is somehow strange.. We use
Alaw over PRI (and I think it's format number 8),
But why is WriteFormat at 2 ?????
Thanks!
show channel Zap/1-1
-- General --
Name: Zap/1-1
Type: Zap
UniqueID: 1136667936.0
Caller
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
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>
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:
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:
#
2004 Apr 13
1
DNID Digits - Australia
Hi,
Yet another question, now that I have callerid working correctly, I'm
trying to work out how to utilise the different numbers I have. I have a
100 number range allocated to my E1/PRI/OnRamp service.
My incoming calls are handled like this:
Advertised/published number is an analogue line terminating on a X101P.
If the analog line is busy, it has a call diversion to the PRI on a
TE405P
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
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
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
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
2007 May 10
1
ices low volume
(this was also posted to the asterisk forum, but received no replies...
Maybe someone here can help)
I'm using the ices command to stream a conference to an icecast server.
This is working nicely, for the most part, but the volume is very low.
The streamed ogg vorbis audio is much quieter than what I hear in a SIP
client, for example (on the same machine with the same audio hardware,
of
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=
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
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
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
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.
2009 Jul 20
0
No subject
-- SIP/ vaso -e26c answered Zap/14-1
-- Executing DumpChan("SIP/ vaso -e26c", "") in new stack
-- Executing DumpChan("SIP/vaso-e26c", "") in new stack
Dumping Info For Channel: SIP/vaso-e26c:
============================================================================
====
Info:
Name= SIP/vaso-e26c
Type=