Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

On Thursday 28 February 2019 at 18:00:54, Ivan Demkovitch wrote: > Antony, > Ok, I see what you are saying. Yes, than NAT occuring on our router. > Asterisk server is on internal IP (192.168..) > # Now that I read what you say I think there might be 2 issues. "Randomness" > is one, but I am not

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

Antony, Ok, I see what you are saying. Yes, than NAT occuring on our router. Asterisk server is on internal IP (192.168..) # Now that I read what you say I think there might be 2 issues. "Randomness" is one, but I am not even sure we have it (randomness). All recent complains were from specific

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

On Thursday 28 February 2019 at 17:40:28, Ivan Demkovitch wrote: > Noone connects to Asterisk box/server from outside. Callcentric SIP trunk > configured and Asterisk maintains connection to it itself. Okay, I didn't actually mean "does anyone connect *inbound* to your Asterisk server" - I was

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

out on Facebook<https://www.facebook.com/mcts> & Twitter <https://twitter.com/RideMCTS> From: asterisk-users On Behalf Of John Kiniston Sent: Tuesday, February 26, 2019 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ast

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

E. g. You can also then try in /etc/hosts to put 10.10.0.103 mcts.org e. g. if any local reverse lookup is done "10.10.0.103" resolves to "mcts.org" and the converse for normal DNS lookups. You can even try in /etc/asterisk/asterisk.conf to make the "systemname" line be systemname=10.10.0.103

[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

Hi Thomas What is your IVR box's domain name in Linux? With a hostname of, for example, "mcts.org" do you have a line like this in /etc/hosts: 127.0.0.1 mcts.org in your /etc/hosts? Additionally, in /etc/asterisk/asterisk.conf, is there a line systemname = that is -uncommented- and

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

On Thursday 28 February 2019 at 00:26:17, Ivan Demkovitch wrote: > Asterisk is NOT exposed to internet, noone connects to Asterisk > from internet. We use Callcentric for VOIP trunk. That's the point where you lost me. Callcentric is out on the Internet. How does it connect to your Asterisk

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

-users] Asterisk - can't hear other side. Or other side does not hear us Hello, This is not technical post, just looking for suggestions on what to check. I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

Antony, thanks for response! It wasn't technical, now it's getting there :) 1. It's asterisk 13.1-cert12. Network. I actually tried multiple things initially but now it's plain vanilla. No NAT. I have Asterisk on our network. All of our phones is IN our network as well, same subnet. All

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

On Wednesday 27 February 2019 at 23:10:33, Ivan Demkovitch wrote: > Hello, > This is not technical post, Hm, no? > just looking for suggestions on what to check.I have asterisk for long time, Which version? > no updates, just maintain OS updates. I use SPA504G phones. Tell us about your

[asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

On Behalf Of Thomas Peters Sent: Tuesday, February 26, 2019 4:11 PM To: Asterisk User List (asterisk-users@lists.digium.com) Subject: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM Hello all, I hope someone can help me with this old Asterisk version. I have to run this versi

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

mailto:tpet...@mcts.org  From: asterisk-users <mailto:asterisk-users-boun...@lists.digium.com> On Behalf Of John Kiniston Sent: Tuesday, February 26, 2019 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <mailto:asterisk-users@lists.digium.com> Subject: Re: [

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Thomas, Does the Asterisk box need to do anything other than handle calls for this one specific IVR? IE does it ever originate calls? If it's only recieving calls then I'd turn on guest access and not even bother with a peer. Just set [general] context=transit-ivr allowguest=yes On Tue, Feb

[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information. We

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' > ); Problem 1: The url is

[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

[asterisk-users] Asterisk 13.25.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.25.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.25.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.2.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] Asterisk Dahdi Issue

On Fri, Feb 8, 2019 at 3:55 PM Alexander Perkins < alexanderhenryperk...@gmail.com> wrote: > Hi All. I am trying to install dahdi, but I get the error below. I have > installed kernal devel from yum and also rebooted. I've googled, but > cannot seem to find an answer. Any help would be

[asterisk-users] [asterisk-app-dev] Supported URIs for Playback

I was wondering what filetypes are supported to be played as sounds from the Playback application. Only raw audio, or can mp3 be used? Many thanks. ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com

Re: [asterisk-users] [asterisk-app-dev] IAX2 protocol documentation

Hi Wojciech, The IAX2 RFC is available here: https://tools.ietf.org/html/rfc5456 I once developped an IAX2 softphone based on libiax: http://downloads.asterisk.org/pub/telephony/libiax/ Maybe it would be easyer to start from here. Best regards, -- Jean-Denis Girard SysNux

Re: [asterisk-users] Asterisk on dynamich IP

Le 01/02/2019 à 09:59, basti a écrit : Hello, Hi my Asterisk is installed on my router. From my ISP I only get an dynamic IP. In sip.conf I have try: externhost=host1.mydns.unix-solution.de externrefresh=300 but after reconnect I cant call from "outside". asterisk*CLI> sip show registry

[asterisk-users] Asterisk on dynamich IP

Hello, my Asterisk is installed on my router. From my ISP I only get an dynamic IP. In sip.conf I have try: externhost=host1.mydns.unix-solution.de externrefresh=300 but after reconnect I cant call from "outside". asterisk*CLI> sip show registry Hostdnsmgr

Re: [asterisk-users] [Asterisk-video] asterisk playing video call to a local display

Le 30/01/2019 à 05:17, Jose Tavares a écrit : Hi guys .. I have some experience with asterisk and sip since I have been using it for over 10 years. But in the last years I have been just maintaining the installations we have without updating myself on the new features of it. Now I have a

Re: [asterisk-users] [asterisk-app-dev] Who uses the ari/sounds resource?

On Mon, Jan 21, 2019 at 2:45 PM sdut...@wazo.io wrote: > On 2019-01-21 10:42 a.m., George Joseph wrote: > > and what do you use it for? > > I assume that you're talking about the /sounds resource available in ARI > [1]. > > At Wazo [2], we use the /sounds resource to list the sound files that >

Re: [asterisk-users] [asterisk-app-dev] [asterisk-dev] Who uses the ari/sounds resource?

On Mon, Jan 21, 2019 at 11:55 AM Andrew Latham wrote: > Are you discussing > https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation#ARIandChannels:SimpleMediaManipulation-Example:Playingbackasoundfile > or something else? > I'm talking about the "sounds"

[asterisk-users] [asterisk-app-dev] Who uses the ari/sounds resource?

and what do you use it for? -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com · https://sangoma.com

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 applications subscribed to the channel.

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

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: app.js:985:13) Channel was destroyed: 1547220509.77 app.js:1029:17) This was a customer app.js:1030:17) Checking if this was a

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

Hiya, When I hang up on a call to my stasis app I’m getting multiple channelDestroyed events for the same channel: app.js:985:13) Channel was destroyed: 1547220509.77 app.js:1029:17) This was a customer app.js:1030:17) Checking if this was a customer talking to an agent app.js:1043:21) Customer

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? >> >>var

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? >> >> var

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

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? > > var bridge = ari.Bridge(); > bridge.create({

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

Hiya, I would have expected this to show the channels in the bridge inside the anonymous function - it shows the bridge is empty though? var bridge = ari.Bridge(); bridge.create({ type: 'holding',

[asterisk-users] Asterisk 16.1.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 15.7.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.7.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.7.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 13.24.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.24.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.24.1 resolves an issue reported by the community and would have not been

[asterisk-users] asterisk libsrtp 2.x status

hi, what's your experience with asterisk compiled with libsrtp 2.x and WebRTC(pjsip)? issues/crashes/speed/cpu usage? Marek official status https://wiki.asterisk.org/wiki/display/AST/libsrtp -- _ -- Bandwidth and

[asterisk-users] asterisk -rx "cmd" truncates cmd's output

Hello, I've edited my diaplan to print some data on screen with statements like: [foobar] exten = foo,1,Verbose(0,Whatever I need to display) exten = bar,1,Verbose(0,Some more text) When using rasterisk and entering "channel originate Local/foo@foobar application Noop", I can read lines such

[asterisk-users] Asterisk / FreePBX Anlaog Fax behind audiocodes MP112

Hello, At the moment we have a Swyx phone server. We would like to switch to a free asterisk PBX. All phone extensions work as expected except the fax. This is a analog Fax behind a Audiocode MP-112 FXS. I have create a extension for the MP112. The MP112 can register to asterisk. When I send a

Re: [asterisk-users] Asterisk 16.1.0 Now Available

On Tue, Dec 11, 2018, at 7:32 PM, Mitch Claborn wrote: > When building a new release, is it possible to copy the output of "make > menuselect" from a previous build directory? If so, what files need to > be copied? That would save some time in the upgrade process. The result of menuselect is

Re: [asterisk-users] Asterisk 16.1.0 Now Available

When building a new release, is it possible to copy the output of "make menuselect" from a previous build directory? If so, what files need to be copied? That would save some time in the upgrade process. Mitch On 12/11/18 4:11 PM, Asterisk Development Team wrote: The Asterisk Development

[asterisk-users] Asterisk 16.1.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 15.7.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.7.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.24.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.24.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] asterisk is not seeing my queues in database

yep here is the result: mysql> SELECT * FROM queues WHERE name = 'cou0002-test'\G *** 1. row *** name: cou0002-test musicclass: NULL announce: NULL context: NULL

Re: [asterisk-users] asterisk is not seeing my queues in database

Maybe post the result from that query here? Mitch On 12/4/18 10:46 AM, Dominic wrote: I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z  

Re: [asterisk-users] asterisk is not seeing my queues in database

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED

Re: [asterisk-users] asterisk is not seeing my queues in database

Maybe try capturing the queries that are executed on the mysql server? That might point you in the right direction. -- show the log file name SHOW VARIABLES LIKE 'general_log%'; -- turn logging on and off SET GLOBAL general_log='ON'; SET GLOBAL general_log='OFF'; Mitch On 12/4/18 7:50 AM,

[asterisk-users] asterisk is not seeing my queues in database

Hi I am facing an issue where asterisk cannot see the queues that exist in my database through realtime. I am using res_odbc and a local mysql database. If I run: realtime load queues name myqueue I get "No rows found matching search criteria.", however if I do the same for a peer: realtime

Re: [asterisk-users] Asterisk non-root - selinux - astdb

Hi Jean Thanks, you've solved my problem. Reference: https://bugzilla.redhat.com/show_bug.cgi?id=1342733 Solution: semanage fcontext -a -t asterisk_var_lib_t /var/lib/asterisk/ restorecon -v /var/lib/asterisk/ Regards Rafael S. Saraiva Porto Alegre - RS | Mobile: (51) 981-747-956

[asterisk-users] Asterisk PJSIP useragent in Dialplan

Found a way to solve it with the following Snippet: `same => n,NoOp(${PJSIP_CONTACT(${PJSIP_AOR(${EXTEN},contact)},user_agent)})` -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] Asterisk PJSIP useragent in Dialplan

Hello, I have an Asterisk 16.0.1 installation with PJSIP SIP Driver. I like to get the useragent in the Dialplan in the form of an Variable to check if it is allowed to place a Call. Is there anything available to achieve that in Asterisk? With the old chan_sip driver this was possible with

Re: [asterisk-users] Asterisk non-root - selinux - astdb

Hello, I haven't tried but this post probably gives a solution : https://bugzilla.redhat.com/show_bug.cgi?id=1342733 Regards Jean Aunis Le 30/11/2018 à 19:24, Rafael dos Santos Saraiva a écrit : Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk is

[asterisk-users] Asterisk non-root - selinux - astdb

Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk is running ok, but astdb not works. When i try to put in astdb, console shows this message: WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic error or missing database CentOS 7.5.1804

Re: [asterisk-users] Asterisk PJSIP enforce Transport

On Tue, Nov 27, 2018, at 3:15 AM, Benjamin Marty wrote: > Hello, > > I have an Asterisk 15.6.0 installation with PJSIP SIP Driver and Sorcery > for Realtime. My Goal is to enforce endpoints to UDP, TCP or TLS. For that > I set the 'transport' column in the endpoint to the corresponding transport

[asterisk-users] Asterisk PJSIP enforce Transport

Hello, I have an Asterisk 15.6.0 installation with PJSIP SIP Driver and Sorcery for Realtime. My Goal is to enforce endpoints to UDP, TCP or TLS. For that I set the 'transport' column in the endpoint to the corresponding transport in pjsip.conf. But if I e.g. set the transport to my

Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 9

Sebastian, Well, this can't be problem with trunk because:1. Call coming from outside, so trunk works2. sip show registry shows it registered. Trunk allows for 2 channels which is not a problem here either It's just weird that out of 4 queue member only 2 being called and log doesn't show

Re: [asterisk-users] Asterisk 16 PJSIP and set_var

Le 20/11/2018 à 19:50, Administrator TOOTAI a écrit : Hi, I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a problem with user variable defined in sip.conf using setvar. It work like a charm -even on asterisk 13 version- but can't get it work in 16. The variables are defined

[asterisk-users] Asterisk 16 PJSIP and set_var

Hi, I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a problem with user variable defined in sip.conf using setvar. It work like a charm -even on asterisk 13 version- but can't get it work in 16. The variables are defined in pjsip with set_var and a pjsip show endpoint does

Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 1

Expose some logs (full file in /etc/asterisk/). Also you can active CDR debug in cli to see exactly what is going on there. Asterisk 13 brings many changes in CDR procedures, but I am using it for a while and I don’t have any problem. > On 2 Nov 2018, at 18:00,

[asterisk-users] Asterisk 15 and Cepstral

    It seems that app_swift does not work with Asterisk 15 or 16.  I just get errors when trying to compile: [root@pbxoficina app_swift]# ./configure checking gcc... checking swift... checking asterisk... creating Makefile     *  Now run

Re: [asterisk-users] Asterisk 16.0.0 Now Available

On Tue, Oct 16, 2018 at 8:08 AM Marcelo Terres wrote: > Guys, > > just a small thing: > > the link on "thanks for download" webpage is still pointing to Asterisk 15. > > Here: > > https://www.asterisk.org/download-asterisk-thank-you > > Your download should begin in a few seconds. If not,

Re: [asterisk-users] asterisk 16 manager --END COMMAND--

On 2018-10-12 12:22, Dmitry Melekhov wrote: >> AMI: >>   - The Command action now sends the output from the CLI command as a >> series >>     of Output headers for each line instead of as a block of text with >> the >>     --END COMMAND-- delimiter to match the output from other actions. >> >>    

Re: [asterisk-users] asterisk 16 manager --END COMMAND--

On Fri, Oct 12, 2018, at 7:22 AM, Dmitry Melekhov wrote: > 12.10.2018 14:10, Joshua C. Colp пишет: > > On Fri, Oct 12, 2018, at 3:35 AM, Dmitry Melekhov wrote: > >> Hello! > >> > >> Just upgraded asterisk from 13 to 16 and found that php-agi library is > >> not compatible. > >> > >> It waits for

Re: [asterisk-users] asterisk 16 manager --END COMMAND--

12.10.2018 14:10, Joshua C. Colp пишет: On Fri, Oct 12, 2018, at 3:35 AM, Dmitry Melekhov wrote: Hello! Just upgraded asterisk from 13 to 16 and found that php-agi library is not compatible. It waits for --END COMMAND-- after command is completed, but, as I see from tcpdump, now asterisk

Re: [asterisk-users] asterisk 16 manager --END COMMAND--

On Fri, Oct 12, 2018, at 3:35 AM, Dmitry Melekhov wrote: > Hello! > > Just upgraded asterisk from 13 to 16 and found that php-agi library is > not compatible. > > It waits for --END COMMAND-- > > after command is completed, > > but, as I see from tcpdump, now asterisk does not send such

[asterisk-users] asterisk 16 manager --END COMMAND--

Hello! Just upgraded asterisk from 13 to 16 and found that php-agi library is not compatible. It waits for --END COMMAND-- after command is completed, but, as I see from tcpdump, now asterisk does not send such string after command is completed. Could you tell me, is it possible to get

Re: [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found

./configure --with-crypto --with-ssl --with-srtp --with-pjproject-bundled The culprit of this behavior is option --with-ssl Version 15.5 does not have this problem. 26.09.2018 16:46, George Joseph пишет: On Tue, Sep 25, 2018 at 2:18 PM Dmitriy Serov > wrote:

Re: [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found

On Tue, Sep 25, 2018 at 2:18 PM Dmitriy Serov wrote: > Hello. > > After successful compilation 15.6.1 (bundled pjsip) and start asterisk i > has error Symbol pjsip_tls_transport_start2 not found. > > /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 > and

[asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found

Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has

Re: [asterisk-users] Asterisk 16 AMI changes

Usually yes. You'll need to read the UPGRADE.txt and CHANGES files to get a good idea of the specific changes though. Best wishes, Matthew Fredrickson On Thu, Sep 6, 2018, 7:44 PM Telium Support Group wrote: > Does anyone know if Asterisk 16 includes changes to the AMI? (syntax / > commands /

[asterisk-users] Asterisk 16 AMI changes

Does anyone know if Asterisk 16 includes changes to the AMI? (syntax / commands / etc) I see a release candidate is forthcoming. Just curious -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Asterisk 15.6.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.6.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.23.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.23.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] asterisk-users Digest, Vol 168, Issue 14

The limit 10 is just an assumption. For real number I've set in my conference server is 350 max participants to join. As per the load testing I've performed on the server, after 400+ participants voice is getting choppy so I've set the max limit to 350 globally for safe reason. If 5 other

Re: [asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings - solved

Hi Guys Found the solution for this...! https://wiki.asterisk.org/wiki/display/AST/New+in+12#Newin12-channels_chan_a gent and https://reviewboard.asterisk.org/r/2657/diff/1/ and https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agen t_pool/ clarifies the situation.

Re: [asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings

On Thu, Aug 2, 2018, at 5:54 AM, Stefan Viljoen wrote: > Hi All > > With the below config, I just keep gettings this in the Asterisk 13.22.0 > CLI: > > WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type > registered for 'Agent' chan_agent doesn't exist anymore[1] in

[asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings

Hi All With the below config, I just keep gettings this in the Asterisk 13.22.0 CLI: WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type registered for 'Agent' whenever a caller gets sent to that agent queue with logged in agents waiting for calls on Asterisk 13. 3997 and

[asterisk-users] Asterisk 11 with volume control for Confbridge

Hi all, I found the volume function. I am wondering if that works for ConfBridge ? The web page mentions works on a channel - but what about adding volume to a ConfBridge. How do I do that? Thanks, Jerry -- _ -- Bandwidth and

Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 21

Hi, I sent the requested information. I always get this responde: Response: Success Message: Timeout Set But keep the old timeout, interestingly, decreasing the timeout works perfectly. The problem is increasing. > Which verson? Version Asterisk 1.8.32. > > Show us what command you are

Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 17

Hi Daniel Thanks for the reply! Yes, turns out it was all my fault, I had a line feed character (0x0a a.k.a printf("\n")) in one of the Asterisk channel variables passed via system() / shell() to my target script. It seems 13.22.0 (I'm using the same version as you) reacts to a line feed in

Re: [asterisk-users] Asterisk 13 - system() dialplan app cannot call bash scripts

Hi Guys Just feedback on this particular thread, this issue is SOLVED. The reason why SYSTEM() and SHELL() was not working for me was that I was passing a linefeed character (\n, hex 0x0a) in one of the channel variables in Asterisk that was then parsed in the call to SYSTEM() and SHELL(). It

[asterisk-users] Asterisk 13.22.0 - "stat" dialplan function clears channel vars?

Hi Guys I have the following dialplan code that I use to play back recordings, the filename being provided in an originate statement to the AMI (AJAM) interface: Action: Originate ActionID: test Channel: SIP/3015 Exten: Context: local Priority: 1 CallerID: 3015 Account: recordinglisten

Re: [asterisk-users] [asterisk-app-dev] how to use snoopChannel

On Tue, Jul 24, 2018, at 10:41 AM, Phil Mickelson wrote: > Ramesh, > > I'm also using ARI and Nodejs. I use the snoopChannelWithId command. I > always specify both directions so that the snooper can whisper to one of > the listeners and then just silence the snooper if I don't want anyone to >

Re: [asterisk-users] [asterisk-app-dev] how to use snoopChannel

On Tue, Jul 24, 2018, at 10:20 AM, Ramesh C wrote: > Hello , > > I want to use spy and whisper using snoopChannel , but i do not understand > how to use this snoopChannel() method. > > Actually my need is that, one user on a channel wants to whisper/spy a talk > running between two caller(means

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

So, that's not quite a debug log, but just the console log with Verbose+ output. A debug log will show a lot more information, including what the media cache modules are trying to do when they go to get the file. You can find information on getting debug information on the Asterisk here:

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

> On Jul 20, 2018, at 1:39 PM, Naftoli Gugenheim wrote: > > I've tried it with .wav. Same result. It doesn't even hit my server. > Can you provide a debug level 5 log (including all higher level verbose+ messages) from Asterisk that shows the playback operation? > > On Fri, Jul 20,

Re: [asterisk-users] Asterisk pjsip realtime extensions

On Wed, Jul 18, 2018, at 7:23 PM, Benjamin Marty wrote: > Hello > > I'm currently using Asterisk 13 with the chan_sip sip driver. The > extensions are offloaded via realtime module to a MySQL database (via > ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords > and the

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim wrote: > > Crickets... > > I've tried this now on 15.5.0. Still completely broken. > > I suspect you’re encountering behavior that is working as intended. Normally, when Asterisk plays back a file, it scans the file system for all files

[asterisk-users] Asterisk pjsip realtime extensions

Hello I'm currently using Asterisk 13 with the chan_sip sip driver. The extensions are offloaded via realtime module to a MySQL database (via ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords and the other stuff from the standard Asterisk database schema. Now I want to

[asterisk-users] Asterisk 15.5.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.5.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.22.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.22.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.22.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] [Asterisk-video] (no subject)

http://secret.loynin.com Pankaj Pandey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Asterisk crashing on AAAA lookup

On Wed, Jun 27, 2018 at 9:22 AM, George Joseph wrote: > > > On Wed, Jun 27, 2018 at 5:15 AM Dovid Bender wrote: > >> >> >> On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett >> wrote: >> >>> >>> >>> On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender >>> wrote: >>> I have Asterisk running on a

Re: [asterisk-users] Asterisk crashing on AAAA lookup

On Wed, Jun 27, 2018 at 5:15 AM Dovid Bender wrote: > > > On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett > wrote: > >> >> >> On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender >> wrote: >> >>> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so >>> often asterisk crashes and

Re: [asterisk-users] Asterisk crashing on AAAA lookup

On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett wrote: > > > On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender wrote: > >> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so >> often asterisk crashes and then restarts. I am not seeing any core dumps on >> the box. The only I

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