Displaying 20 results from an estimated 2000 matches similar to: "[1.4 + FreeBSD 6.2] Playing WAV PCM file?"
2014 Jan 23
1
mixmonitor extension
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
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Marek Cervenka
=======================================
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
2008 Jan 15
3
Meetme recording
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
********************************************************************
This email and any attachments
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
2005 Mar 20
1
I cannot use G711 (ulaw|alaw)
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
I do not understand why because my Asterisk box load these codecs properly!
Does somebody
2010 Dec 06
1
Asterisk 1.6.2.10 video call
Hello list,
I'm trying to set up a video call from my Ekiga client to a Grandstream
GXV3140 IP-phone. The call succeeds but there is no video.
I have in sip.conf :
videosupport=yes
disallow=all
allow=alaw
allow=g726
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
The Grandstream peer has codecs (sip.conf) :
gsm;alaw;g729;h261;h263;h263p;h264
The Ekiga peer has codecs
2011 Apr 11
1
Asterisk codec negotiation and canreinvite=no
Hi all,
I realise that asterisk's codec negotiation has been discussed in
the past multiple times. What I haven't been able to understand is
how asterisk decides which video codecs to advertise to the other
end when canreinvite=no in sip.conf and the initial caller
doesn't support video.
My tests are quite simple, I use an asterisk with 4 peers all on the
same LAN. My sip.conf
2005 Jul 02
1
Sipura SPA2000 behind NAT
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
___________ HOME _______________ ____OFFICE ____
SPA2000 <---> Linux Box <--> Asterisk Box
192.168.0.253 192.168.0.1 eth1 200.93.xxx.a
200.93.xxx.b eth0
My problem is when I try to call to any trunk or extention
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
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
2011 Nov 21
1
video calls not working
Hi list,*
*I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.*
Extensions.conf*
exten => 111,1,Answer()
same => n,Dial(SIP/2206,60,r)
same => n,Hangup()
*SIP.conf*
[2218]
type=friend
secret=*******
callerid="Virendra" <9172341457>
host=dynamic ;
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
> I suppose that you enable the video support on sip.conf, right?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>
2010 Aug 28
4
Asterisk does not translate from wav to alaw
Hello list,
I have a file to be played in wav-format.
I thought Asterisk would automatically take the wav-file and translate
it to the codec used, but I see this :
[Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File
/var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any
format
[Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to
open
2005 Sep 18
2
Asterisk Won't Process Call
We have a basic application that runs a SIP channel to pick up a call
and process it. We are using Broadvoice and it's been working great.
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it. Asterisk seems to send the
call back to Broadvoice. Nothing at all has been changed in the
configuration to warrant this. Below is the
2010 Jul 05
1
Problems with ulaw/g729 translation
Dear Folks,
I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is
simple: connection to the PSTN directly via SIP, using g729 codec, and
connection to the softphones (X-lite 3.0 build 56125) trought local network,
using ulaw codec.
Sometimes, I got messages like:
[Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported
SDP media type in offer: image
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
Hi,
We can't read the messages in our mailbox always getting
-- <SIP/tootaiAUDIO-00000001> Playing
'/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr')
[Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message:
Playback of message
/var/spool/asterisk/voicemail/default/100/Old/msg0002 failed
As you see Asterisk try to read
2004 Mar 30
1
G726 not working ?
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced".
When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
[format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
==