[asterisk-users] Asterisk 18.1.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.1.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.1.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 17.9.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.9.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.15.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.15.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.15.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.38.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.38.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.38.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5. These releases are available for immediate download at

[asterisk-users] Asterisk 13 and Asterisk 17 Security Fix Only

Greetings all, This is a reminder that as of October 24th Asterisk 13 has gone security fix only. This means that it will receive no new bug fixes, and any outstanding bug fixes which have been merged (or are up for review) will be released in an upcoming final bug fix release. Asterisk 13 has

Re: [asterisk-users] Asterisk 18.0.0 Now Available

On 21.10.20 at 12:49 Joshua C. Colp wrote: > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > >> Hello! >> >> On 20.10.20 at 14:00 Asterisk Development Team wrote: >>> The Asterisk Development Team would like to announce the release of >> Asterisk 18.0.0. >>> This release is available for

Re: [asterisk-users] Asterisk 18.0.0 Now Available

Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk > 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec

Re: [asterisk-users] Asterisk 18.0.0 Now Available

On Wed, Oct 21, 2020 at 8:49 AM Michael Maier wrote: > On 21.10.20 at 12:49 Joshua C. Colp wrote: > > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier > wrote: > > > >> Hello! > >> > >> On 20.10.20 at 14:00 Asterisk Development Team wrote: > >>> The Asterisk Development Team would like to announce

Re: [asterisk-users] Asterisk 18.0.0 Now Available

On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > >

[asterisk-users] Asterisk 18.0.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 17.8.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.8.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.8.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.14.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.14.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.14.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.37.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.37.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.37.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] asterisk-users Digest, Vol 193, Issue 15

memory vs disk cache > This is an issue that has plagued Asterisk since day one. Basically there >> is no solution available because there is no way to set aside memory to be >> kept from a growing disk cache. I did some research and this looks like a >> bad design from the Kernel people.

Re: [asterisk-users] Asterisk Drop call

Thanks Luciano. But there is no active ALG on the modem. Attached the call flow, including the ACK. Em 22/09/2020 14:41, Luciano Moreira escreveu: Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if both parties are getting ACK correctly

Re: [asterisk-users] Asterisk Drop call

The problem has been detected. FXS equipment is causing the fall. Most likely from some bad contact. Thank you all for your help. Roberto. Em 22/09/2020 14:41, Luciano Moreira escreveu: Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if

Re: [asterisk-users] Asterisk Drop call

Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if both parties are getting ACK correctly after RTP starts. *--* *Atenciosamente,* *Luciano Moreira**(85)99974-2750* *__Logic Telecom* *0800-085-7799 | (85)4042-7799 | **(11)4210-7799*

Re: [asterisk-users] Asterisk Drop call

Hello. Thanks for the reply. Yes. In the traffic analyzed, the BYE is sent by the originator of the call, but there is no "human" hangup, but the asterisk one. BYE is sent, received and confirmed. I don't know how I could investigate the reason for this BYE. Em 21/09/2020 17:12, Dovid

Re: [asterisk-users] Asterisk Drop call

Is there anything in the Asterisk logs? Which side sends the BYE? Were you able to capture the traffic with sngrep/wireshark to see if any side stopped sending/getting RTP? What did the other side see? On Mon, Sep 21, 2020 at 3:22 PM Roberto < roberto.med...@gasparimsantos.com.br> wrote: >

[asterisk-users] Asterisk Drop call

Hello I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr =

[asterisk-users] Asterisk 17.7.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.7.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.13.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.13.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.13.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.36.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.36.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.36.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction

I think I found how to do it... menuselect/menuselect --disable BUILD_NATIVE Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction

> Under Compiler Flags uncheck >BUILD_NATIVE hi Doug - thanks! Is there a way to do that from the "./configure" command line ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction

On 8/8/20 8:35 AM, Jerry Geis wrote: The VM is Intel box (host) and the physical box is a celeron. So something is not right there. What would be a good ./configure option that asterisk can compile with on the VM image so this illegal instruction does on occur ? Jerry, Under Compiler Flags

[asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction

HI All I have a need to create a VM - install asterisk from source - then move the VM to a physical box. When I do this - I get illegal instruction when starting asterisk. So I recompile on the physical box and then everything is fine. Problem is there are a number of these physical boxes. The

Re: [asterisk-users] asterisk-users Confbridge

Sorry about the trouble. Unsubscribed that user from the mailing lists. Matthew Fredrickson On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote: > > I'm online on this site! > So contact me in my profile: > here > -- > _ > --

Re: [asterisk-users] asterisk-users Confbridge

Im online on this site! So contact me in my profile: galleries.daswanitailors.com here -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] asterisk 13.33 and polycom

On 8/6/20 8:09 AM, Jerry Geis wrote: I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I

[asterisk-users] asterisk 13.33 and polycom

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just

[asterisk-users] Asterisk 17.6.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.6.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.6.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.12.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.12.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.12.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.35.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.35.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.35.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk Getting Crashed

On Thu, Jun 25, 2020 at 4:12 PM Ahmed Chohan wrote: > Hi, > > Currently I'm experiencing crashes on Asterisk more recently, see messages > below (crashed reason: segfault signal 6). > > abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT > - dumping core > > asterisk:

[asterisk-users] Asterisk Getting Crashed

Hi, Currently I'm experiencing crashes on Asterisk more recently, see messages below (crashed reason: segfault signal 6). abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT - dumping core asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!, Failed

[asterisk-users] Asterisk 17.5.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.5.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 16.11.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.11.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk conference manager

Hi, I'm looking for some software to allow users to login and manage Asterisk conferences (define conference user lists, call users to join a conference, etc.). I'm aware of astconfman (https://github.com/litnimax/astconfman). Are there any others? Thanks in advance. Regards, IanG --

[asterisk-users] Asterisk 17.5.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.11.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.11.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.34.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.34.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.34.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state

On 10/06/2020 15:40, Joshua C. Colp wrote: You wouldn't be able to access such information from ast_sip_presence_exten_state_to_str, that function is strictly for taking in instructions/data and producing the output. The user of it would need to pass in a value to turn on this new behavior.

Re: [asterisk-users] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state

On Wed, Jun 10, 2020 at 10:27 AM John Hughes wrote: > Asterisk can know that one of the attached phones is both "ringing" and > "on the phone". > > However the sip NOTIFY it sends out to interested parties can only > communicate one state, for example with pidf+xml it can either send > "Ringing"

[asterisk-users] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state

Asterisk can know that one of the attached phones is both "ringing" and "on the phone". However the sip NOTIFY it sends out to interested parties can only communicate one state, for example with pidf+xml it can either send "Ringing" or "On the phone" and so it sends "Ringing". This makes

Re: [asterisk-users] asterisk 16.9 function HINT with option n does not return anything

Thanks Joshua ! I was not aware that extension's hint in dialplan can have additional data in the parentheses - it might be worth adding that info to wiki. On Wed, Jun 3, 2020 at 10:08 AM Nenad Radosavljevic wrote: > > > Hello ! > > > > Is there something wrong with the function HINT(,n) ? > >

Re: [asterisk-users] asterisk 16.9 function HINT with option n does not return anything

On Wed, Jun 3, 2020 at 10:08 AM Nenad Radosavljevic wrote: > Hello ! > > Is there something wrong with the function HINT(,n) ? > Note the param n - it is supposed to get (as far as I understand the > documentation) CALLERID(name) of the extension (via extension's hint). > It does not get the

[asterisk-users] asterisk 16.9 function HINT with option n does not return anything

Hello ! Is there something wrong with the function HINT(,n) ? Note the param n - it is supposed to get (as far as I understand the documentation) CALLERID(name) of the extension (via extension's hint). Example configuration: pjsip.conf: [10] type=endopoint . . callerid = Test extension <10> .

Re: [asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls

It seems that if there is a pause in the auto attendant longer than a second this problem occurs.  I have this for an extension in my extensions.conf file: exten => 2799,1,GotoIf($["${CALLERID(num)}" = "${EXTEN}"]?500) exten => 2799,2,Dial(PJSIP/${EXTEN},14,tr) exten =>

[asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while

[asterisk-users] Asterisk and Prometheus

Hello, I've seen that Asterisk 17 supports Prometheus but beside [1], I've not much about how to use this. Can someone shed some light on this ? 1. If I'm not mistaken, Prometheus favors "a pull model over HTTP". So basically, a Prometheus instance should be able to query Asterisk "core

Re: [asterisk-users] Asterisk : CDR Analyzer Updated

On 5/25/20 5:56 AM, Mitul Limbani wrote: Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Anybody that would like to do this would be most welcome. I have no plans on supporting it. Basic instructions and attachment will

Re: [asterisk-users] Asterisk : CDR Analyzer Updated

Hello Doug, Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India

[asterisk-users] Asterisk : CDR Analyzer Updated

Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7

[asterisk-users] Asterisk and SIP Proxy on same host = media problem

Hi there I have a pbx (v16.10) on AWS (Ubuntu 18.04) with Freepbx (14) that I am trying to set up the proxy reSIProcate on the same host as pbx. I can make it all work when the proxy is on a different host but when the proxy is on the same host asterisk sends the media address as 127.0.0.1

Re: [asterisk-users] Asterisk versions?

On Mon, May 11, 2020 at 7:29 PM Dave Woodfall wrote: > Thanks for that info, Ben. I do like to test out the latest and most > up-to-date versions of things when I can, so I'll check those files > and see how it goes. > I'll also add that different versions have different lifetimes. A great

Re: [asterisk-users] Asterisk versions?

Thanks for that info, Ben. I do like to test out the latest and most up-to-date versions of things when I can, so I'll check those files and see how it goes. On 2020-05-11 17:20, Ben Ford put forth the proposition: > Hey Dave, > > In the case of 13 and 16, these are LTS versions which means

Re: [asterisk-users] Asterisk versions?

Hey Dave, In the case of 13 and 16, these are LTS versions which means that they get long term service. 17 is a standard release. The benefit of an LTS is that you can expect it to get bug fixes and improvements for an extended period of time without anything major being changed. If you find an

[asterisk-users] Asterisk versions?

Hi all, I'm a fairly long time user of Asterisk, but I'm new to this list. I used to use the old forums some few years ago. I wanted to ask why there are different Asterisk versions, as shown by the announcements in the past week or 2: Asterisk 13.33.0 Asterisk 16.10.0 Asterisk 17.4.0 I'm

[asterisk-users] Asterisk 13.22.0 unstable on Azure Centos 7 & cannot encode .gsm files

Hi all, I'm running a Centos 7 instance in Azure with Asterisk 13. The Centos VM has 24GB of RAM and identifies the CPU as Intel(R) Xeon(R) CPU E5-2660 0 @ 2.20GHz. This is a virtual copy of a physical Centos 7 machine which has 16GB of RAM and the physical CPU identifies itself as the

Re: [asterisk-users] Asterisk and CentOS 8

On Sun, May 3, 2020 at 6:07 PM Patrick Wakano wrote: > Hello George, > Hope this finds you well! > I wonder if there has been any progress on this matter? > Sorry, I missed this last week. We have an internal issue open to revisit CentOS 8 and it's at the top of our backlog right now so we

Re: [asterisk-users] Asterisk and CentOS 8

Hello George, Hope this finds you well! I wonder if there has been any progress on this matter? I've been working to have Asterisk running on CentOS 8 and our jump from CentOS 6 to 8 doesn't look too bad The missing packages found are: gmime-devel, iksemel-devel, corosynclib-devel,

[asterisk-users] Asterisk 17.4.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.4.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.10.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.10.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.10.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.33.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

Hello, Hard to tell but from [1], " Exceptionally long voice queue length queuing" should not happen. 1. Are you seeing WARNING or ERROR instances in log before this "Exceptionally long voice" WARNING ? 2. If possible, I would try to load test a similar setup with SIPp and see if I can reach or

Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls Are you using NFS? Any ODBC connections? On Tue, Apr 21, 2020 at 10:23 AM Stefan Viljoen <mailto:vi

Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

Are you using NFS? Any ODBC connections? On Tue, Apr 21, 2020 at 10:23 AM Stefan Viljoen wrote: > Hi all > > I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - > 32GB RAM. > When I approach about 320 channels, I -sometimes- get thousands of these > messages suddenly

[asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

Hi all I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - 32GB RAM. When I approach about 320 channels, I -sometimes- get thousands of these messages suddenly streamed in the CLI / Asterisk log: WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length

Re: [asterisk-users] [Asterisk-video] gstreamer integration

This also seems to not be viable: "Asterisk 1.6 and 11 (help us to port it to Asterisk 13/14)" I am using asterisk 13 - RTMP does not seem to be available for that. Jerry -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] [Asterisk-video] gstreamer integration

Thanks - not familiar with chan_rtmp. Will look at it. I was also wondering about directfb ? Currently I am missing a dependency to try it. Trying to find that also for baresip. I dont see an example for "directfb" in the config file for baresip. Thanks Jerry --

Re: [asterisk-users] [Asterisk-video] gstreamer integration

: Re: [asterisk-users] [Asterisk-video] gstreamer integration Thanks Jeremy. I am seeing questions similar to mine when I search - but I am not finding a solution. Was wondering if you know of something I have not found yet. Jerry

Re: [asterisk-users] [Asterisk-video] gstreamer integration

Thanks Jeremy. I am seeing questions similar to mine when I search - but I am not finding a solution. Was wondering if you know of something I have not found yet. Jerry -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] [Asterisk-video] gstreamer integration

Hi - Is this list still alive ? How do I get audio/video into gstreamer from an asterisk call? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE

[asterisk-users] Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable

Re: [asterisk-users] Asterisk 16, 9.0 - res_rtp_asterisk compilation error

Le 13/03/2020 à 13:30, Joshua C. Colp a écrit : On Fri, Mar 13, 2020 at 9:27 AM Administrator > wrote: Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error     [CC]

Re: [asterisk-users] Asterisk 16, 9.0 - res_rtp_asterisk compilation error

On Fri, Mar 13, 2020 at 3:36 PM Administrator wrote: > > Le 13/03/2020 à 13:30, Joshua C. Colp a écrit : > > On Fri, Mar 13, 2020 at 9:27 AM Administrator wrote: > >> Hello, >> >> 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, >> I can't compile asterisk having error

[asterisk-users] Asterisk 16, 9.0 - res_rtp_asterisk compilation error

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’  

Re: [asterisk-users] Asterisk 16, 9.0 - res_rtp_asterisk compilation error

On Fri, Mar 13, 2020 at 9:27 AM Administrator wrote: > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, > I can't compile asterisk having error > > [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o > res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka

[asterisk-users] Asterisk 17.3.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.3.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.32.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.32.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.32.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 16.9.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.9.0 resolves several issues reported by the community and would have not been

[asterisk-users] [asterisk-app-dev] True suppression of DTMF from audio

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP

Re: [asterisk-users] [asterisk-dev] Need feedback on the use of AMI events generated by MESSAGE requests

On Thu, Feb 6, 2020 at 12:34 PM sdut...@wazo.io wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > For those of you who actually process SIP MESSAGE requests... Do you > > use any of the AMI events generated by the "Message/ast_msg_queue" > > channel? We want to change that channel to an

Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

On Tue, Feb 4, 2020 at 3:11 PM Administrator wrote: > Done, ASTERISK-28725 > but against > version 16,7.0 and 13.30.0 as the new ones aren't available in the list > > For those interested this has been resolved. The patch files were used

Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

Done, ASTERISK-28725 but against version 16,7.0 and 13.30.0 as the new ones aren't available in the list Le 04/02/2020 à 19:30, Joshua C. Colp a écrit : On Tue, Feb 4, 2020 at 2:28 PM Administrator > wrote:

Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

On Tue, Feb 4, 2020 at 2:28 PM Administrator wrote: > Hello, > > we just installed the latest 13 and 16 version of asterisk and face > problem on incoming calls: they are ended like in Asterisk 16 > > [2020-02-04 19:19:48] ERROR[3768][C-0001]: stasis_bridges.c:199 > bridge_topics_init:

[asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

Hello, we just installed the latest 13 and 16 version of asterisk and face problem on incoming calls: they are ended like in Asterisk 16 [2020-02-04 19:19:48] ERROR[3768][C-0001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization required [2020-02-04 19:19:48]

[asterisk-users] Asterisk 17.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.2.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.2.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.8.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.8.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.8.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.31.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.31.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.31.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] [asterisk-app-dev] ARI Get Channel Variable

On Wed, Jan 22, 2020 at 5:32 PM Phil Mickelson wrote: > I'm trying to get the Call-ID from the SIP HEADER using getChannelVar. > When I pass SIP_HEADER() and anything as the variable I get Unable to read > provided function. If use Call-ID I get Provided variable was not found. > > This is a

Re: [asterisk-users] Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?

On Fri, Jan 10, 2020 at 12:25 PM Benoit Panizzon wrote: > Hi List > > I have been pondering over a problem to use an asterisk server behind > an SBC unable to successfully handle registrations. > > Now I observed something strange which I suspect might be a bug on the > asterisk side. > > The

[asterisk-users] Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?

Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk.

[asterisk-users] Asterisk 17.1.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.1.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.1.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.7.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.7.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.30.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.30.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.30.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

https://issues.asterisk.org/jira/browse/ASTERISK-28656 Dne 12/12/2019 v 16:49 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 11:40 AM marek > wrote: examples of "interesting" information like ICE result and howto make "minimal" configuration of

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 11:40 AM marek wrote: > examples of "interesting" information like ICE result and howto make > "minimal" configuration of pjproject.conf > > i.e. > > for debugging app_queue.so > > core set debug 5 app_queue.so > > for debugging RTP > > core set debug 10 rtp_engine >

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