Displaying 20 results from an estimated 3000 matches similar to: "Asterisk and Linphone"
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
2012 Nov 21
1
core show translation - difference in Asterisk Versions
Hello All,
I was wondering if somebody could elaborate the change in
translation of codecs specifically the amount of time increased in Asterisk
11. For example
*Asterisk 11*
* **alaw **speex *
*gsm **15000 **15000 *
*ulaw 9150 15000*
* *
*Asterisk 1.6.x*
* **alaw **speex *
*gsm **2 12002 *
*ulaw 1 12002*
I did recalculate the
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
Hi,
I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.
[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
2016 Dec 10
6
failing to start asterisk on centos7
ive installed asterisk but below is what am getting proces gets
killed.please help
[root at localhost sounds]# asterisk -vvvvc
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit :
> Hi,
If you're not using RTP encryption did you uncheck the option in your
RTP TAB from identity ?
>
> This is the log. ex dialling 0 from snom phone
>
>
> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
> <http://123.231.72.210:33878> --->
> INVITE sip:0 at 54.206.59.252
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
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
>
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also
2010 Nov 11
2
Asterisk Playback sound dropping on linphone
Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn't matter whether I choose
alaw, ulaw, gsm as codec (besides changing cpu load of course).
How can I debug it? I'm using A* 1.6.2 and both linphone
2003 Jul 01
2
Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's "frustrated programmer" award goes to Linphone, which has the
following debug output:
> (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use linphone to connect to asterisk and listen to an
announcement. I suspect that
2005 Jan 12
1
linphone -> NAT -> * -> NAT -> firefly woes.
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:
iax.conf
[100]
type=friend
username=Foo
context=default
auth=md5,plaintext,rsa
secret=secret
host=dynamic
callerid="Foo" <100>
qualify=no
sip.conf
[10]
type=friend
username=Bar
context=default
callerid=Bar <10>
2020 May 26
3
Attempting to get BLF working with linphone
Hi John,
1. Could you get any further, in your quest for working BLF with linphone ?
2. Have you tried with a different Linphone version (4.12 is pending on
Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ?
Best regards
Le mer. 25 mars 2020 à 15:06, John Hughes <john at calva.com> a écrit :
>
> On 23/03/2020 18:51, Joshua C. Colp wrote:
>
> On Mon, Mar
2007 Feb 15
1
error during make while installing Linphone-1.5.1
Hi All,
I am getting this error during make.
please help me./
speexec.c: In function `speex_ec_process':
speexec.c:112: syntax error before "noise"
cc1: warnings being treated as errors
speexec.c:133: warning: implicit declaration of function
`speex_echo_state_reset'
speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes
pointer from integer without a cast
2020 Jun 13
3
Voice "broken" during calls
Am 13.06.2020 09:30, schrieb Luca Bertoncello:
Hi again (again)
I noticed right now another strange detail...
I made a call using my mobile phone (connected to the Asterisk). The
quality was top...
Maybe is the problem in a codec used from our phones at homes?
Could someone suggest me how to check the codec used by my mobile phone
and the codec used by the phones at home?
Thanks
Luca
2010 Nov 03
1
Asterisk linphone call dropping by itself
hi all, please help... I am calling in the simplest way among two
linphone clients connected to one asterisk server... the call ends on
one side without any sign of problem, while on the other side it stays
connected.
I checked the SIP dialogue and at some point the server sends a BYE
message to one party
I have no timeout set, though the duration of a call is always around 20s.
the two
2014 Feb 11
0
g726 transcoding
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
I am running certified-asterisk-11.2-cert2
Thanks
Gareth
> core show translation paths alaw
--- Translation paths SRC Codec "alaw"
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