Displaying 20 results from an estimated 3000 matches similar to: "ices low volume"
2006 Apr 13
1
placing call with agi
I'm trying to set up a system so that I can record a conversation over
SIP. Monitor and the like don't work so well for me, because I need to
pipe the conversation to other programs in realtime, rather than record
to a file, so I've been trying to use EAGI instead. (if anyone has any
other suggestions about this, it would be greatly appreciated!)
At this point, I'm a little
2010 Feb 24
2
audio glitches in conference
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few seconds, on average. They also seem to delay
some buffer somewhere, so that if I start recording
2010 Feb 10
1
problems with 1.6
In an attempt to fix problems with EAGI delays in 1.4 (see my other
message for more on that), I've tried upgrading to 1.6, in case it's a
bug that's fixed in the newer version.
Unfortunately, I'm having all kinds of trouble with this new install. My
system relies on conferences, and whenever I add any channel to it
(adding a SIP connection, playing an audio file, activating
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:
2006 Apr 21
2
extension match sip address
Is there a way to have an extension match on a sip address? I've tried
the obvious - _.@. but it seems to behave just like _. which is no
good.
Is there a better way?
--
Jon-o Addleman - http://redowl.dyndns.org
2008 Feb 02
3
IE, flash and icecast
I'm having trouble getting an IE client to hear mp3 streams through a
flash player. It appears to be the same problem as described at
http://icecast.imux.net/viewtopic.php?t=2039 - the flash player connects
to the icecast server and begins downloading the audio data, but never
actually plays it.
I've tried all the suggestions from that thread - I patched icecast so
that the Content-length
2010 Mar 09
1
confbridge manager/cli
I've just started switching my project to use confbridge instead of
meetme and app_conference (because of audio glitches that kept appearing
in those applications).
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's 'meetme list'
command? Anything else I can use through the cli or manager API? I
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
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
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
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
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
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>
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 Jun 30
0
stuck channel
I'm getting this intermittent problem, sometimes a zap channel gets
stuck after a call. Below is a snapshot of the channel. Any ideas what
can be happening?
Name: Zap/1-1
Type: Zap
UniqueID: 1056988772.10
Caller ID: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormat: 68
WriteFormat: 4
ReadFormat: 4
1st File Descriptor:
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=