Displaying 20 results from an estimated 1000 matches similar to: "Re-INVITE and bridge breakage"
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
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
2015 May 04
0
Asterisk proxying a REFER
--
Luca Pradovera
luca.pradovera at gmail.com
Hello,
sorry, I managed to lose the reply amidst the traffic.
What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer.
Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone
2007 Mar 28
1
SIP OPTIONS dialog not understood
I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm
getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch
which my Asterisk box replies to. The exchange looks like:
<-- SIP read from 172.b.c.d:5060:
OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
2009 Jun 05
5
How run AsyncAGI commands in background
Hi all,
I have an external application commanding asterisk by AMI and AsyncAGI. I
also have a dialplan like this:
; AsyncAGI extensions
exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN});
exten => _8.,n,AGI(agi:async);
exten => _8.,n,Hangup();
; Meetme extensions
exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT});
exten =>
2007 Mar 21
1
Metaswitch help needed
I'm attempting to connect to a Metaswitch, inbound only (at this time).
The Metaswitch is the only "connection" (at this time).
All I'm getting so far is a bunch of "OPTION" messages which my Asterisk
box replies to but I don't get inbound calls.
Here's my sip.conf. As you can see I've been trying a bunch of different
options without success :(
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
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2011 Mar 10
1
Metaswitch to Asterisk problems
I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please.
I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my
2010 May 28
2
Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue
Hi folks,
I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD
4.7 -release. Everything seems to work fine. I have a macro which
answers, receives the fax to a tiff, and then runs a script (mailfax) to
convert that to pdf and email it. It all works perfectly except for some
errors I am seeing in the console. After it hangs up I get a dozen or so
messages in the cli
2012 Aug 02
1
DTMF transmission problem
I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset).
My system set up as follows:
PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
Asterisk is running Asterisk 10.4.0 on a
2016 Sep 21
3
AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected
It means, AMI application is no more running or crashed or lost network connection with asterisk server.
In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls
Regards
Amit Patkar
On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote:
>On Sat, Sep 17,
2016 Sep 17
2
AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected
Hi
Is there any way to detect inactivity on channel when AsyncAGI is used?
I want to detect whether application handling calls using AMI & AGI has
stopped responding.
Alternatively, how can dialplan check if there is any AMI user connected
and decide dial plan execution?
Thanks & Regards,
Amit Patkar
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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
2006 Mar 03
1
SIP Problem - Asterisk to Provider Gateway
Hi All,
I'm stumped on a weird problem. I have an * server working fine for local
SIP phones and IAX2 connections. We just provisioned a second Ethernet
port to attach to a local SIP provider.
PSTN calls incoming work fine:
PSTN -> SIP Provider -> SIP -> *
but outgoing calls are not. Call setup takes place and the caller can hear
about 1-2 seconds of audio before the SIP provider
2008 Oct 21
1
hex b1 in CallerID sent by Asterisk On PRI
I'm trying to send CallerID info to a MetaSwitch system over a PRI. The
MetaSwitch gets the info exactly as it is sent by Asterisk, but I think
it might be having trouble with the Hexadecimal b1 that is being sent
just before the first character of the CallerID Name.
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1
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
2008 Dec 05
2
async agi question
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
2019 Nov 01
2
Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio.
Some background..
We are using asterisk 16.6.1
We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality.
My one thought is to
2009 Mar 30
1
The Redirect hangups the call while playing a file
Hi,
I'm bringing this discussion here from
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
about how to manage stopping a playback on a extension previously launched
with AsyncAGI and redirecting the call to another exension.
If I make the Redirect without a playback, the Redirect works:
http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd
But if I make the
2016 Oct 19
2
Streaming for ASR
Hello,
(sorry for not continuing the thread, I had set the list to digest).
Would UnicastRTP be able to output u-law frames directly? If so, I think
that is all I need.
Does anyone know what the EAGI output is? Raw RTP?
Best regards,
Luca
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