Displaying 20 results from an estimated 4000 matches similar to: "How to pick a codec on the fly"
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
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
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
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000
2007 Jun 05
1
g729
I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies
Jun 5
2010 Oct 11
1
Unable to find a codec translation path from ulaw|h261 to slin
I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've
encountered a problem playing back a .wav file to an Ekiga client:
My dialplan looks like:
exten => 730,1,answer
exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
exten => 730,n,hangup
Sovereign.wav is a .wav file that plays nicely on my 1.4 server.
Here is what the console displays:
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2012 Aug 01
2
Problem with callfile and CDR
Good afternoon list.
I am experiencing a problem with the CDR and callfiles. What is happening
is this: When generating a call with a callfile, everything works
perfectly, but the CDR is recorded in the table when they answer the call
destination. The field disposition is being recorded correctly, but the
duration field is marked with the ring time and billsec is marked with 0.
This just happens
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 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from Digium.
This is the error message from asterisk -vvg:
[app_playback.so] => (Sound File
2010 Sep 27
8
Problems compiling Asterisk on Debian
Hello,
I'm trying to compile DAHDI on DEBIAN but i have the following error:
root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make
echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed."
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
exit 1
make: *** [modules] Error 1
What should i do?
Thanks!
-------------- next
2015 Feb 17
4
Callfile problem - Unable to find codec translation path from (nothing)
Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using:
[outbound-swift]
exten => _[a-zA-Z].,1,Answer
exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure)
;exten => _[a-zA-Z].,1,Swift("${EXTEN}")
exten => _[a-zA-Z].,n,Goto(1)
[mis-phone]
exten =>
2007 May 04
2
Asterisk Codec Translation Table
Hello list,
I have always though codec translation table is dircetly connected to system speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
2010 Jun 17
3
Music on Hold problema
Hi people, I have a problem with Music On Hold, it is stopped just after starting...
This is the log:
[Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:1] NoOp("SIP/7PBX-08229d18", "Start") in new stack
[Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold'
[Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:2]
2012 Feb 22
2
codec mismatch on channel
Hi
I am keep getting this warning message when doing attendant transfer:
WARNING[6027] channel.c: Codec mismatch on channel
Local/XX at Inside-1f32;1 setting write format to slin from alaw native
formats (alaw)
What can I do to lose it.
I am using asterisk 10.1.2
Best regards
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
2011 Mar 09
3
Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)
Hello!
Client is using ulaw, however server sometimes fills the log with following:
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP -> SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give