Re: [asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

Answering the below for search engine’s sake. > On Feb 18, 2019, at 11:23, Matt Riddell wrote: > > Hey, trying to use ARI with NodeJS - this doesn't work: > > play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav' > <http://www.nch.com.au/acm/8k16bitpcm.wav'&g

[asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

Hey, trying to use ARI with NodeJS - this doesn't work: play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav'); should it? https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation says: A sound file located on the Asterisk system. You can use the

Re: [asterisk-users] [asterisk-app-dev] ARI-client Node.js objects

> On Jan 11, 2019, at 10:46, Gilles VERRIEZ (SERENEO) > wrote: > > Hi, > > I would like to get the audio resource from a record in order to send it > threw AJAX request with my ARI-client Node JS source. I thought > Playback.media_uri could help me but it's value is undefined. Any ideas? >

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

> On Jan 11, 2019, at 11:14, Jean Aunis wrote: > > Le 11/01/2019 à 16:47, Matt Riddell a écrit : >> Hiya, >> >> When I hang up on a call to my stasis app I’m getting multiple >> channelDestroyed events for the same channel: > > It may happen if several

[asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

); Is this normal? I’m writing like a CDR on channel destroyed so don’t want to write it multiple times. Should I keep an array of channels and only write if I haven’t seen the event for that channel before? Cheers, Matt Riddell ___ asterisk-app-dev

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote: > > On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote: >> Hiya, >> >> I would have expected this to show the channels in the bridge inside >> the anonymous function - it shows the bridge is empty though?

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote: > > On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote: >> Hiya, >> >> I would have expected this to show the channels in the bridge inside >> the anonymous function - it shows the bridge is empty though?

[asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

ges depending on agent/customer status etc Cheers, Matt Riddell ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/li

[asterisk-users] Function CHANNELS

I appreciate it if someone can post an an example for function CHANNELS showing the usage of the regular expression filter. Basically I would like to get a count of active channels having a certain criteria. Is it possible to search for a channel having a custom variable set to specific value

Re: [asterisk-users] remove

Check the footer at the bottom of this message for instructions on how to unsubscribe :-) Matthew Fredrickson On Fri, Jun 1, 2018 at 12:11 PM, David Mutterer wrote: > > -- > _ > -- Bandwidth and Colocation Provided by

Re: [asterisk-users] shell dialplan application blocking

Use AGI Kind regards, Matt > On Jun 4, 2018, at 02:28, Benjamin Marty wrote: > > I'm calling a script which needs to wait a certain time and also hold the > call for this time. But the script dialplan application seems to work non > blocking. Is there a way to hold the cal

[asterisk-users] Testing...

Test from non-digium email. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] testing users list

testing users list -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:

[asterisk-users] Test from Digium address

Testing again. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] asterisk rules

mailman drools -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:

[asterisk-users] One more test

I need to send one more test. Here it is! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] More testing

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

[asterisk-users] Test

Testing again :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] AstriCon Approaching, Super Earlybird Pricing Expires In 3 Days

Hey All, So one of the jobs that I get to do as head of the Asterisk project is to help inform people about the yearly conference we have about Asterisk named Astricon. For those who are not familiar with it, AstriCon is a fantastic event for anyone that is serious about Asterisk. This year,

Re: [asterisk-users] Wanted: WebRTC tutorial

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell wrote: > A while back (last year maybe?), there was a Digium blog post on setting up > WebRTC. > > I was never able to get that working. > > I was working with Asterisk 15 on a RHEL derived distro and had no idea of > where to

[asterisk-users] Asterisk Community Services Outages

Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services. We apologize for any inconvenience that it has caused. Hopefully

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote: > In article >

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield wrote: > I have some more investigation to do on this, but I wanted to see if anyone > here had any insight into the issue I've run into. > > The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one > of

Re: [asterisk-users] More testing - sorry guys

Thanks :-) On Wed, Mar 28, 2018 at 3:52 PM, Markus Weiler <markus_wei...@mailworks.org> wrote: > I received it :-) > > > Am 28.03.2018 um 22:44 schrieb Matt Fredrickson: >> >> Just a test. >> > > > -- >

[asterisk-users] Sorry for interruption of service

Hey All, Just as a public service announcement, we had a 12-16 hour window with mailing list service interruption. Last night we scheduled a time to update the mailing list server but today found some problems impacting mailing service after the updates. Due to this discovery, we quickly

[asterisk-users] More testing - sorry guys

Just a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] AMI potential memory leak

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp wrote: > We are communicating with Asterisk via AMI. Running Asterisk version > 13.18.5 on an Ubuntu box. > > > > If you look at the event response, the Result field is filled with random > characters. I’m not sure what to do because

[asterisk-users] Test

Testing, 1, 2, 3. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] Test

This is a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] FOSDEM

Hey All, For any interested in potentially meeting up to talk about Asterisk and other fun things, Ben Ford from Digium's Asterisk development team and myself will be in Brussels for FOSDEM Feb 3-4. I hope to see many of you there! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445

[asterisk-users] Recent Video Interview and Upcoming Webinar about Asterisk 15

, there's a fun 15 minute video interview featuring Matt Jordan and myself discussing Asterisk 15 and what's new with it. It can be found at: https://www.youtube.com/watch?v=0XSDOPftNpM=youtu.be For those who enjoyed the video or would like to get a bit of a deeper dive into Asterisk 15

[asterisk-users] Asterisk EOL Announcement

Dearly Beloved, We have gathered here today to mourn the passing of a deeply regarded branch of Asterisk - Asterisk 11. As of today, it has officially reached its end of life. It was a good branch, having served 5 years faithfully in the service of its users. As far as history goes, 11.0.0 was

[asterisk-users] Asterisk 14 Security Fix Only Mode

Hey all, For those who may not be aware Asterisk 14 transitioned from bug fix mode to security-fix-only mode a few weeks ago (Sept 26th). For those of you that are still on this release, it's a good time to consider building an upgrade plan for moving to 15.x.x. I sincerely apologize for the

Re: [asterisk-users] Asterisk chan_sip registration attempts

Maybe the provider has added an extra gateway and it is not processing accounts correctly. If they had one before and now two then 40-60% registration fails would show that. Kind regards, Matt > On Oct 10, 2017, at 06:27, Dmitriy Ermakov <demoni...@gmail.com> wrote: > > He

Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

nds of silence and caller > annoyance. At least in older versions you can use EAGI to get a handle to the audio stream. You can then pipe that stream to something like bluemix using Node.js and have a handle to the incoming recognition in realtime too. -- Cheers, Matt Riddell _

[asterisk-users] Asterisk 15 Beta Released

It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. It's a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or so years. There has been a lot of work done in the

Re: [asterisk-users] Moving call DAHDI from channel X to Y.

On Sun, Jul 30, 2017 at 8:34 PM, Daniel Harper wrote: > I am seeing the in the asterisk logs that channels (PRI ISDN) are > being moved .. > > [Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call > (DAHDI/57-1) from channel 57 to 58. > > I then see the moved

Re: [asterisk-users] OT: Want to capture all SIP messages

; >> Solvable by by writing a cleanup script that deletes files over a >> specific age, just a basic find in the daily crontab: >> find /path/to/captures -type f -name 'pattern*' -mtime +X -exec rm {} \; > > Been there, done that. Just 1 more thing for

Re: [asterisk-users] softphone instead of desktop phones

I use Bria on all of the above. Kind regards, Matt > On Apr 29, 2017, at 10:35 AM, Thomas <thomasit...@gmail.com> wrote: > > Hello, > Iam lookong for an Softphone for iPhor oder Android smartphone using togehter > with an headset. > I tried Zoiper and CSipSimple but

[asterisk-users] Asterisk 11 EOL 6 Month Notice

Greetings, This is your friendly 6 month warning that Asterisk 11 will be reaching an official end of life state on October 25, 2017. As many of you know, for the past 6 months Asterisk 11 has been in security fix only mode. This means it currently does not receive bug fixes, but it does

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins wrote: > > On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote: > >> Hello, >> >> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only >> problem until now which remained was that if dtls_rekey

Re: [asterisk-users] 100% CPU after upgrade.

On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl wrote: > Those are all rational questions, so here we go: > > We upgraded from 11.x, though the system was a backup server, so it was never > actually used. > > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should

Re: [asterisk-users] 100% CPU after upgrade.

One thing you didn't mention was what version you previously upgraded from... Also, more information about the system in general would help. (Endpoints, is it realtime or flat file configured, if realtime, what type of database, what channel drivers (SIP or PJSIP, and others). Matthew

Re: [asterisk-users] Bounty on Google Voice

On Wed, Mar 29, 2017 at 11:45 AM, Saint Michael wrote: > The channel motif and res_xmpp do not work. But there is one company that > does make it work and charges $US 6 for a lifetime connection to your own > free Google Voice number, from SIP. I wonder if anybody would be able

Re: [asterisk-users] UniMRCP and Asterisk 14

On Thu, Mar 23, 2017 at 8:27 PM, Richard Kenner wrote: > When I look at the lastest UniMRCP manual, they only mention as high as > Asterisk 13. Does anybody know if I need to do anything to allow it > to work on Asterisk 14 and, if so, what that is? I can't speak for the MRCP

Re: [asterisk-users] Manager events showing in CLI

Try doing a `core set debug 0` at the Asterisk CLI. That should disable it. Or remove debug from your console output in logger.conf. Best wishes, Matthew Fredrickson On Sun, Mar 26, 2017 at 5:35 PM, Telium Technical Support wrote: > I did that too – no debug related

Re: [asterisk-users] Call List Campaign to an IVR

one of them goes to voicemail the other will stop ringing. Typing calls this happens in a few ms (after post dial delay). Because they are both going out at the same time with the same provider this is super quick. -- Cheers, Matt Riddell ___ ht

Re: [asterisk-users] Call List Campaign to an IVR

. If your phone is ringing and another comes in it will go to voicemail. Call waiting allows you to get notified of a call if you're already on one, not multiple incoming at the same time. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/ne

Re: [asterisk-users] Call List Campaign to an IVR

Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt

Re: [asterisk-users] Call List Campaign to an IVR

>> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote: >> Love the idea. How? > > On Mon, 6 Feb 2017, Matt Riddell wrote: > >> exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3) > > Amazing. Who knew? > &g

Re: [asterisk-users] Call List Campaign to an IVR

X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full I

Re: [asterisk-users] Issue with handling of 480 DND

Response inline. On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler wrote: > Nobody any idea? > > It would be really helpful, > > Markus > > > > > Am 06.01.2017 um 12:07 schrieb Markus Weiler: > >> Hi List, >> >> we're calling a sip phone from our Asterisk Server, and

[asterisk-users] Asterisk 14 web broadcast

project now that Matt Jordan has been moved into the CTO role at Digium. You can get info about it at: http://bit.ly/2gDkFrh It will be live today at 8AM, 2PM, and 9PM CDT. Hope to see many of you there! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

> On 27/11/2016, at 6:44 PM, Joshua Colp <jc...@digium.com> wrote: > > On Wed, Nov 23, 2016, at 06:41 PM, Matt Riddell wrote: >> >> There doesn't appear to be a way to monitor general Asterisk events like >> you can in the Asterisk manager without polling f

[asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

. Is this a correct assumption? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] Force hangup not working on stuck channel

Also, it looks like in https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be a workaround (see the last comment at the bottom). Matthew Fredrickson On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson <cres...@digium.com> wrote: > On Thu, Nov 3, 2016 at 11:16 AM, Carlos Ch

Re: [asterisk-users] Force hangup not working on stuck channel

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez wrote: > I am unable to force a hangup on a channel that has been stuck for over two > days: > > IAX2/from-CD-11006 oficina 27701 Up Dial > IAX2/to-CD/2883 3467130007

Re: [asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?

On Tue, Nov 1, 2016 at 6:00 PM, Jonathan H wrote: > All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions. > > I don't need any other DB layer, I have no hardware, and I was wondering > what the smallest build possible was. > > I experimented, but

Re: [asterisk-users] RS485 Audio device

On Fri, Oct 28, 2016 at 7:09 PM, Jerry Geis wrote: > Hi All, > > Is there any devices or pair of devices that do audio over RS485 > and then convert to SIP for us in asterisk? > Of course a speaker and push button at the other end. > > Is there anything like that out there?

[asterisk-users] Asterisk 11 - Security Fix Only Notice

Hey All, This is a friendly notice that as of today Asterisk 11 has entered security fix only mode. From this point onward additional releases of Asterisk 11 will no longer be made unless there is a security fix being applied to the branch. Users of Asterisk 11 are encouraged to move to one of

Re: [asterisk-users] Got bitten by the 255 char variable limit - how best to work around it?

On Sat, Oct 22, 2016 at 8:05 PM, Jonathan H wrote: > I loop through a list in Asterisk which is generated by a Python AGI > and I've just been bitten by a variable limit I didn't realise existed > before. > > The only way I can think of working around this is to get Python

Re: [asterisk-users] Audio when card is in condition yellow

Usually a card is supposed to send yellow alarm (so it's transmitted) when it detects LOS (loss of signal) on the T1, or essentially a red alarm condition is detected. So if yellow is being sent, it means that at least one end is not able to sync up the line, which means you'll have junk/garbled

Re: [asterisk-users] queue_log/cel sqlite

On Thu, Oct 20, 2016 at 9:45 AM, marek cervenka <cerva...@gmail.com> wrote: > > Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a): >> >> On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com> >> wrote: >>> >>> i tested this &g

Re: [asterisk-users] queue_log/cel sqlite

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3

Re: [asterisk-users] Streaming for ASR

adcasting-asterisk-conferences/ <http://www.joshua-colp.com/broadcasting-asterisk-conferences/> I'm still working on mine :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_

Re: [asterisk-users] Streaming for ASR

> On 17/10/2016, at 4:07 PM, Joshua Colp <jc...@digium.com> wrote: > > Matt Riddell wrote: >> >>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com >>> <mailto:luca.pradov...@gmail.com>> wrote: >>> >>>

Re: [asterisk-users] Streaming for ASR

nd then sending that file once recording has finished. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange

Re: [asterisk-users] Surfing the web via Asterisk.

I'm guessing you're going to be wanting something closer to this: https://www.npmjs.com/package/speech-rule-engine <https://www.npmjs.com/package/speech-rule-engine> -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily As

Re: [asterisk-users] Surfing the web via Asterisk.

You'd really want to see if you can get ChromeVox as a library rather than as a browser though - otherwise you're going to be limiting yourself to using one concurrent channel and hacks like jack audio to move the audio from the browser to the channel. -- Cheers, Matt Riddell __

Re: [asterisk-users] Surfing the web via Asterisk.

. You'd use agi or something to navigate the object you create and tts to describe current position. The hard part will be parsing the HTML even though most HTML is broken :-) Kind regards, Matt Riddell > On Oct 17, 2016, at 9:00 AM, Jonathan H <lardconce...@gmail.com> wrote: >

Re: [asterisk-users] Installing Asterisk on MAC native

of the time I just use virtualization and spin up Debian in parallels. Kind regards, Matt > On Sep 20, 2016, at 4:07 PM, Glenn Geller (VDOPh) <ggel...@vdo-ph.com> wrote: > > If you're looking for installing on a MAC, best to start searching for MAC > OSX install > > See

[asterisk-users] Asterisk 11 - Security Fix Mode

Hello Everyone, As many of you are already aware, we are rapidly approaching the time when the Asterisk 11 branch will go into what is known as security fix only mode. Up to this point, bug fixes have been included and merged into the 11 branch. For Asterisk 11, this new phase of life shall

Re: [asterisk-users] Farewell

Best of wishes to you in your retirement! It's been a great 10 years, and I'm personally looking forward to the great things coming in the next 10. Matthew Fredrickson On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina wrote: > I just wanted to wish all of you good luck I'm

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

urrent versions of Chrome or Firefox. > That said, LetsEncrypt certs work fine for this, so no need to spend out on > one. > > Switch to Asterisk 13.10 and save yourself a whole lotta headache. > > On 11 August 2016 at 15:09, Jonas Kellens <jonas.kell...@telenet.be> wr

Re: [asterisk-users] Asterisk & Vitelity Invite issues

PM, Tammy Firefly <tammy-li...@wiztech.biz> wrote: > > > On 8/9/16 12:40 PM, Matt Fredrickson wrote: >> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> >> wrote: >>> Hi All, >>> >>> We have asterisk 11.23 run

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

How are you attempting to view the original CallerId? Matthew Fredrickson On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb wrote: > Hi > Is there any configuration change in asterisk 13.9.1 to show original > callerid on a transfer > In asterisk 11.21 it works as expected > >

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

now > after my personal experience with Asterisk 11 and webRTC. > > You also say Asterisk 13. How about Asterisk 12 then ?? > > > > Kind regards. > > > > On 10-08-16 21:53, Matt Fredrickson wrote: > > I don't see an ice-ufrag or ice-pwd line in the response from &

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

I don't see an ice-ufrag or ice-pwd line in the response from Asterisk, correlating with your suspicion that there is no ICE. Are you sure that the stun server you're using (the google one) still works? I haven't tried that server in a while, but I distantly seem to recall that maybe they shut

Re: [asterisk-users] EAGI script with missing audio on /dev/fd/3

On Tue, Aug 2, 2016 at 11:42 AM, nik600 wrote: > Dear all > > i'm trying to access to the input audio raw stream with a very basic EAGI > script: > > > #!/bin/sh > echo "EXEC Queue 2001" > cat /dev/fd/3 > /tmp/pippo > > This is my dialplan: > > exten => 001,NoOp(test) > exten

Re: [asterisk-users] Asterisk & Vitelity Invite issues

On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote: > Hi All, > > We have asterisk 11.23 running sip to vitelity and from there IAX trunks > split off to where they need to go. We are having a problem getting > chan_sip to quit ignoring re-invites from Vitelity. Our

Re: [asterisk-users] AstriCon 2016 - XMPP and Asterisk

Looking forward to seeing you there, and hopefully to seeing your talk! Matthew Fredrickson On Tue, Aug 2, 2016 at 11:02 AM, Marcelo Terres wrote: > Going to AstriCon 2016 ? > > Don't miss my talk about how to use XMPP and Asterisk to improve the > user experience. > >

Re: [asterisk-users] Force out-bond call to specific CIC

Yes, as far as I remember, in your dial string, simply use a Dial(DAHDI/X/1234567) where X is the dahdi device channel number. Hope that helps. Matthew Fredrickson On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi wrote: > Hi > > How is it possible to use Dial application

Re: [asterisk-users] Unable to create channel DAHDI

Looks like the hookstate is listed as offhook. I don't think chan_dahdi will attempt to make a call out a device that is offhook. Hope that helps, Matthew Fredrickson On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson wrote: > In trying to troubleshoot the Delay after

Re: [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?

On Wed, May 18, 2016 at 9:44 AM, Olivier wrote: > I've got the following setup: > > PSTN ITSP SDSL Modem-Router Gateway - > Asterisk with B410P --- SIP Phones Wow. > Both SDSL Modem-Router and Gateway are managed by my ITSP. > > Some calls coming from PSTN

Re: [asterisk-users] Avaya Phones and Asterisk

On Fri, May 20, 2016 at 8:54 AM, Diogo Cosito wrote: > Dear gentlemen, how are you? > I wonder if anyone has experience with Avaya devices, 9608G and 9641GS > models, running on SIP and using TCP transport. > The calls work well, but the callerid only "pass" number of

Re: [asterisk-users] pjsip segfault problem

Have you tried updating to pjproject version 2.5.x? It should have the patch that you listed in your other email, which I believe should be included in that branch. Hope that helps, and best of luck. Matthew Fredrickson On Thu, May 26, 2016 at 4:11 AM, Marek Červenka

Re: [asterisk-users] T.38 with Audiocodes gateway

On Fri, Apr 29, 2016 at 1:34 AM, Olivier wrote: > Hello, > > I'm helping a colleague (*) which has the following setup: > > ITSP --- --- Asterisk 13 --- -- > Audiocodes MP-112 --- --- Fax machine > > My issue is the following : > Audiocodes gateway reject INVITEs with 488

Re: [asterisk-users] PRI error "ROSE REJECT"

Fredrickson On Fri, Mar 25, 2016 at 9:15 PM, Carlos Chavez <cur...@telecomabmex.com> wrote: > On 2016-03-25 16:02, Matt Fredrickson wrote: >> >> PRI debug of the entire call would be great, also, switchtype would be >> awesome as well. >> >> Thanks! >> >

Re: [asterisk-users] PRI error "ROSE REJECT"

PRI debug of the entire call would be great, also, switchtype would be awesome as well. Thanks! Matthew Fredrickson On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas wrote: > Hi > > Did you activate the pri debug on the cli asterisk? > > On Thu, Mar 24, 2016 at 12:59 PM,

Re: [asterisk-users] OPUS support in Asterisk 13

On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai wrote: > Hi all, > > Sorry if this has been asked before. I searched a lot, but found conflicting > answers, so hoping for some clarification. > > My question is does Asterisk 13 support OPUS? If so which version exactly? Sort

Re: [asterisk-users] 1000 analogue lines with asterisk

There is definitely no way you should put 1000 lines on a single box. To be honest I do wonder what you want to do with 1000 lines as your description probably changes the recommendations. Kind regards, Matt > On Feb 17, 2016, at 5:09 PM, Goke Aruna <gok...@gmail.com> wrote: &g

Re: [asterisk-users] accept DMTF tone during ringing

For what channel driver, and what use case? It's my understanding that in the traditional telephone network (ISDN/SS7/analog), prior to a call being answered, you were not necessarily guaranteed a two way media path. Sometimes it was available (there are few stories of large companies who

Re: [asterisk-users] Looking for PRI Card with automatic fail over

There was a product called something like red box or similar that I saw around 5 years ago. Probably not entirely helpful but maybe Google will help. Kind regards, Matt On Aug 3, 2015, at 9:50 AM, Eric Klein eric.kl...@greenfieldtech.net wrote: Hi all, Strange request, I have

Re: [asterisk-users] Looking for PRI Card with automatic fail over

Yep it was red fone http://red-fone.com 7 years ago :-) http://www.venturevoip.com/detail.php?news_id=1927 -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

and reenable SELinux regardless of the outcome. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels

calls for just one originate: It looks like you’re originating with local channels. Try with /n at the end (not sure if that helps anymore) and also send us your originate line. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

. The console channel gets errors on opening. What errors are you getting? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com

Re: [asterisk-users] Branch based on call volume

On 27Jun, 2015, at 15:34, Michelle Dupuis mdup...@ocg.ca wrote: Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? Do you mean large number of calls or how loud the call is? -- Cheers, Matt Riddell

Re: [asterisk-users] Branch based on call volume

://wiki.asterisk.org/wiki/display/AST/Function_VOLUME https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk

Re: [asterisk-users] Product CDR/Queue/Meetme

I will post this to the Asterisk news for you. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP

  1   2   3   4   5   6   7   8   9   10   >