Displaying 20 results from an estimated 400 matches similar to: "Asterisk proxying a REFER"
2015 Apr 28
0
hi list need your help
facing problem with originating webrtc calls
1-when iam doing call from webrtc iget ice working
<--- SIP read from WS:91.196.158.205:1466 --->
INVITE sip:0669197533 at 77.91.132.9 SIP/2.0
Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
Max-Forwards: 69
To: <sip:0669197533 at 77.91.132.9>
From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43
Call-ID:
2015 Apr 27
1
Asterisk proxying a REFER
Hello,
we are using Asterisk with Adhearsion as our application server, with
another Asterisk box acting as the office PBX, where all office phones are
registered.
A REFER to transfer calls within the office results in the Adhearsion
application call being dropped, because the leg between the PBX and the app
server is terminated by the PBX following the REFER.
Is there a way to configure
2023 Jun 28
1
SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
Hello list
when trying to set up webRTC communications with sipjs client package
(tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file
the following :
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
c=IN IP4 99.88.77.66... OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.
2015 May 15
1
Re-INVITE and bridge breakage
Hello,
as a variation of our issues with Adhearsion calls dropping when an INVITE
comes in for a bridged call, I now have a new issue to contend with.
Our call is in an AsyncAGI application, and has been bridged to another
channel.
The provider that supplies the DID sends a polling reINVITE every 15
minutes (it's a documented Metaswitch behavior amongst others).
The reINVITE is seen as a new
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
Hello,
I was testing with sdp and something came up worth asking:
While calling from a webrtc client to another (chrome, sip.js) Asterisk
receives the following sdp and rejects it with 488 Not Acceptable. Why does
this happen, what's wrong with the sdp? The second sdp body below is
accepted instead. Both have rtp profile RTP/SAVPF, difference is that the
second one was produced by rtpengine,
2016 Aug 09
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
I'm trying for several days now to get ICE support for my Asterisk 11.23
on CentOS 6.
My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230
--> softphone Zoiper
(problem : no audio)
Reverse does not work either.
(problem : failed get local SDP)
I followed this guide :
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
2016 Aug 11
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
Using Asterisk 12.8.2.
I now have the "via ICE" messages in the RTP debug (see below).
If you look in the SIP debug (see below), you also now see the
"ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the
webRTC client.
But still no audio ! None at all ! In both directions.
You can see in the SIP debug that the IP-address in de
2016 Oct 17
2
Streaming for ASR
Matt Riddell wrote:
>
>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com
>> <mailto:luca.pradovera at gmail.com>> wrote:
>>
>> I have been working on designs for two different projects, where both
>> of them would need to use the IBM Watson streaming ASR service.
>>
>> Would it be possible to write out the audio frames
2010 Oct 28
0
Adhearsion 1.0 - Now Showing
Thanks to the hard work of many people in the Adhearsion community, I am pleased to be able to announce the immediate availability of Adhearsion version 1.0. Since Jay Phillips first began work on the project in 2006 Adhearsion has changed the way developers think about telephony applications. Now with several years of operating experience and multitudes of applications deployed to production,
2011 Feb 23
0
Adhearsion 1.0.1 Released
The Adhearsion team announces the release of Adhearsion version 1.0.1. Adhearsion is an open source Ruby-language framework for creating telephony applications. This update primarily addresses compatibility with newer versions of other software but also adds native support for Bundler to newly created Adhearsion applications.
Here are some highlights from the changelog:
Handling of new Asterisk
2007 Dec 03
0
Adhearsion Install Fails.
Not strictly an Asterisk question.
I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems, and I get this...
[root at localhost rubygems-0.9.5]# gem install adhearsion
Bulk updating Gem source index for: http://gems.rubyforge.org
ERROR: While executing gem ... (Errno::ENOENT)
No such file or directory - /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7/bin/ahn
The
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video
2011 Nov 25
1
Install Adhearsion on Debian
Hi,
I'm giving Adhearsion a try on a Debian Squeeze.
I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started)
that the command "sudo gem install adhearsion" should "automatically add
the ahn command to your system".
On mine I can't run ahn without specifying full path
(/var/lib/gems/1.8/bin/ahn).
Did I miss something ?
Regards
-------------- next
2011 Nov 25
0
Install Adhearsion on Debian [SOLVED]
2011/11/25 Olivier <oza_4h07 at yahoo.fr>
>
>
> 2011/11/25 John Knight <john at classiccitytelco.com>
>
>> Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps?
>>
>
> No I didn't.
> I would have thought that rubygems installation should car of this (adding
> installed gems into users paths).
> As I'm new to Ruby, I
2007 Apr 25
0
Asterisk Users Conference Friday 12:30 PM EDT
AUC is Friday at 12:30 PM EDT. See http://x2z.eu
Hi,
One of our guests this week will be Jay Phillips to tell us about
Adhearsion. Haven't heard about the open-source Adhearsion? Look here:
http://www.linuxjournal.com/article/9519
Be with us to ask Jay questions. If you can't be there, download the
recorded version (including last week's chat with Mark Spencer) here:
2010 Jul 31
0
Disconnect supervision tone detection working for india
Hi ,
Thanks danny nicholas. Finally we get the things done with following.
If i specify busypatten=500,500 then asterisk does not recognize hang up
signal. After removing it only all are working fine.
I choosed 2nd option as per your suggestions.
working chan-dahdi.conf:
====================
signalling = fxs_ks
busycount = 3
busydetect = yes
callprogress = yes
progzone=in
usecallerid=yes
2013 Aug 27
1
Introducing Sippy Cup: SIPp Load Testing Made Easy
Hello everyone,
Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR.
So to make our own lives
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
Hi All.
I'm running some tests with the latest Asterisk SVN-branch-12-r410493M
compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS
machine (2.6.32-358.18.1.el6.i686).
As a client I'm using the sipMLP WebRTC javascript softphone running on
Chrome 33.0.1750.146 m.
I have the softphone correctly registered on the Asterisk machine but as
soon as I try to start a new call
2012 Sep 05
6
Async AGI
Hi,
Is there a way to execute next priority in the dialplan if you have called
agi:async? I want to play warning message if adhearsion is down. Currently
I wasn't able to make it work. The dialplan execution ends after the first
priority.
[incomming]
exten => _X.,1,AGI(agi:async)
exten => _X.,2,Answer
exten => _X.,3,Playback(some-message)
exten => _X.,4,Hangup
Regards,
Pavel
2011 Nov 28
1
Queue-Tip/Adhearsion installation tip
Hi,
I'm giving Queue-Tip a try, following installation instructions in
http://queue-tip.rubyforge.org/install.html.
My setup is :
ruby 1.8.7
rubygems 1.3.7
rails 3.1.3
Adhearsion 1.2.3
I'm struck in step 7 in the above installation procedure :
# rake --trace db:create
(in /usr/local/src/queue-tip)
rake aborted!
no such file to load -- initializer