[asterisk-users] asterisk release 21.1.0

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 20.6.0

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 18.21.0

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 21.0.2

The Asterisk Development Team would like to announce the release of asterisk-21.0.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 20.5.2

The Asterisk Development Team would like to announce the release of asterisk-20.5.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 18.20.2

The Asterisk Development Team would like to announce the release of asterisk-18.20.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release certified-18.9-cert7

The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and

[asterisk-users] asterisk release certified-18.9-cert6

The Asterisk Development Team would like to announce security release Certified Asterisk 18.9-cert6. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert6 and

[asterisk-users] asterisk release 21.0.1

The Asterisk Development Team would like to announce security release Asterisk 21.0.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

[asterisk-users] asterisk release 20.5.1

The Asterisk Development Team would like to announce security release Asterisk 20.5.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

[asterisk-users] asterisk release 18.20.1

The Asterisk Development Team would like to announce security release Asterisk 18.20.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hello > > How do I achieve the same with chan_sip? > We run a cron script each 10min who will check the registration state > and send a register if not registered. Well it's a simple CPE which needs to be autoprovisioned via either a tftp config file or TR69. So that cronjob somehow would

Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon ) a écrit : Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure:

[asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? Mit freundlichen Grüssen

Re: [asterisk-users] Asterisk and Teams integration?

On Thursday 26 October 2023 at 19:11:45, Carlos Chavez wrote: > Does anyone know of a good solution to integrate Asterisk and MS > Teams? Something where you can use the MS Teams client as a regular > extension? Kamailio is the usual intermediary I have seen for doing this. Antony. --

[asterisk-users] Asterisk and Teams integration?

    Does anyone know of a good solution to integrate Asterisk and MS Teams?  Something where you can use the MS Teams client as a regular extension? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 --

[asterisk-users] asterisk release 21.0.0

The Asterisk Development Team would like to announce the release of asterisk-21.0.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 20.5.0

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] asterisk release 18.20.0

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Hello Jerry, when you run asterisk using su, ownership of audio device files does not get updated. When you login, you get the permissions. You can verify by ls -l and getfacl on the device file. Marek --- Original Message --- On Thursday, September 14th, 2023 at 14:33, Jerry Geis

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >An issue[1] was already created by asterisk at phreaknet.org and they > also put > >a fix up for review and inclusion[2]. > > >[1] https://github.com/asterisk/asterisk/issues/308 > >[2] https://github.com/asterisk/asterisk/pull/309 > > > The

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>An issue[1] was already created by asterisk at phreaknet.org and they also put >a fix up for review and inclusion[2]. >[1] https://github.com/asterisk/asterisk/issues/308 >[2] https://github.com/asterisk/asterisk/pull/309 The change "seems" to be working. Will test more tomorrow - had to

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

An issue[1] was already created by aster...@phreaknet.org and they also put a fix up for review and inclusion[2]. [1] https://github.com/asterisk/asterisk/issues/308 [2] https://github.com/asterisk/asterisk/pull/309 On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis wrote: > > I have found that I can

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I have found that I can add the restart of asterisk (killall -9 asterisk) to the h extension- BOY is that UGLY. chan_console is not a testing device - how can we get this nasty bug fixed ? Jerry -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

> After a hung call, can you run core restart now from the asterisk console? Doing a "killall -9 asterisk" is the only thing that works I tried killall asterisk - does not free up the channel the asterisk "core restart now" takes like a good 20 seconds to return but does work. The issue is I

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

It worked with my test. I'm on Asterisk 18.19.0 -- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk -rx 'core restart now'") in new stack -- Remote UNIX connection Asterisk uncleanly ending (0). Executing last minute cleanups == Destroying musiconhold processes

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I have noticed that once my message speaks - the server thinks its done and HUNGUP, the endpoint STILL thinks the channel is active - the last message says "Rx: BYE" on sip show channels I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial. Its NOT getting there to hangup. Jerry --

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>> Is there a dial plan call that can "exit asterisk" or completely reload >> everything - killall active calls and start over ? Using system() you could issue a asterisk -rx 'core restart now' Doug -- _ -- Bandwidth and

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Is there a dial plan call that can "exit asterisk" or completely reload everything - killall active calls and start over ? seems the console/dummy (chan_console) is holding some resource. How do I just "exit" and start over after call came in ? Thanks Jerry --

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I don't know if this will help you, but looking back through an old config I have for an older version of Asterisk, I had used chan_console with the old and now defunct app_rpt app to listen to audio on various nodes via the console for testing. Here is what I did: In console.conf, I defined

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>>> How do we get this working For the time being, go back to 18.14.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

> > > Not sure if this is the same thing you're seeing, but chan_console > currently has a known deadlock issue that has not been resolved: > https://issues-archive.asterisk.org/ASTERISK-30481 > It's quite easy to reproduce, so it may be the case that the module is > currently unusable. > Well

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On 9/8/2023 8:18 AM, Jerry Geis wrote: But when I do a second test. Asterisk HANGS on ChanIsAvail() Then I thought lets SKIP that - and just let it do the Dial() - I stopped everything - got it running again. - and then the Dial() hangs on the second call. So both ChanIsAvail() or Dial()

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Some progress to report: I had to run asterisk as the user logged in - actually not even that. I could not "su user -c " to that user - I had to run it as that user. Then I did a test and got audio! Great... But when I do a second test. Asterisk HANGS on ChanIsAvail() Then I thought lets SKIP

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

In the old days when I was using console/dsp, I would have to use alsamix from the console to verify that the output wasn't muted.  Maybe it's still the same. Doug On 9/7/23 03:43 PM, Jerry Geis wrote: ok switching to "Console/default" does show the text  --- <("<) --- Call to device

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

ok switching to "Console/default" does show the text --- <("<) --- Call to device 'default' on console from 'default' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- However I don't hear any audio. Thanks Jerry --

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:27 PM Jerry Geis wrote: > > I found "console list available" > > === > === - > === Device Name: default > === ---> Default Input Device > === ---> Default Output Device > ===

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I found "console list available" === === - === Device Name: default === ---> Default Input Device === ---> Default Output Device === - === ===

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:20 PM Joshua C. Colp wrote: > On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > >> Joshua >> >> Asterisk 18.14.0 with chan_alsa and Console/dsp works. >> This does not work in 18.18.0 with chan_console enabled. >> I am on Ubuntu 20.04 LTS. >> >> Is there a howto for

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > Joshua > > Asterisk 18.14.0 with chan_alsa and Console/dsp works. > This does not work in 18.18.0 with chan_console enabled. > I am on Ubuntu 20.04 LTS. > > Is there a howto for the new chan_console ? > I'm not aware of one. The module itself

[asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Joshua Asterisk 18.14.0 with chan_alsa and Console/dsp works. This does not work in 18.18.0 with chan_console enabled. I am on Ubuntu 20.04 LTS. Is there a howto for the new chan_console ? how can I get this working again ? I am trying to just play audio on pulse audio. Thanks, Jerry --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

On Thu, Sep 7, 2023 at 3:07 PM Jerry Geis wrote: > > I am trying to get audio to play on Pulse - so just the monitor basically. > > I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others. > > The error is always the same "console_request: Console device 'dmix' not > found. >

[asterisk-users] Asterisk 16.23.0 strange issue where Answer request succeeds and able to perform actions but Asterisk never sent 200 OK to answer call

Some background... We use AMI and AsyncAGI to be able to receive events about calls (and other Asterisk details) and control it from our application. Works great and have about 100 sites (some newer, some older) without issues. I was notified this morning about a customer who had something

Re: [asterisk-users] asterisk 18.18.0 and chan_console

I am trying to get audio to play on Pulse - so just the monitor basically. I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others. The error is always the same "console_request: Console device 'dmix' not found. What is the correct "Console/" to play on pulse for UBuntu

Re: [asterisk-users] asterisk 18.18.0 and chan_console

On 9/6/23 03:23 PM, Jerry Geis wrote: I am trying to just play on PulseAudio actually. This used to work - I have just recently updated to 18.18.0, so I'm puzzled. All of my Asterisk installs are running in virtual machines, so I have no way to test. Doug --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

> What is the device that you're connecting to? I am trying to just play on PulseAudio actually. This used to work - I have just recently updated to 18.18.0, so I'm puzzled. Jerry -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] asterisk 18.18.0 and chan_console

What is the device that you're connecting to? Doug -- _ -- 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?

Re: [asterisk-users] asterisk 18.18.0 and chan_console

This is hte error I get for Console/dsp or console/dsp ERROR[230711][C-0001]: chan_console.c:477 console_request: Console device 'dsp' not found Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] asterisk 18.18.0 and chan_console

In a past work life, I did use console/dsp to connect to a sound card that hooked up to a bogan paging amp. I still have access to the programming and everything I have show as using a lower case c for console Doug -- _ --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

> I don't use it; just figured I'd try to help Thanks Doug... So then for the list - I have chan_console working now But I am trying Console/dsp and Console/ALSA and both give an error about not found. What have I missed ? Thanks Jerry --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

>>> hi Doug - so what device do you use? I am getting and error for Console/dsp I don't use it; just figured I'd try to help. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] asterisk 18.18.0 and chan_console

> > > > Oh that is a good one - I thought I did - but apparently not. menuconfig > now shows "*" > > So is chan_alsa going away ? What is it being replaced with? > > thank you! > > Jerry > hi Doug - so what device do you use? I am getting and error for Console/dsp exten =>

Re: [asterisk-users] asterisk 18.18.0 and chan_console

On Wed, Sep 6, 2023 at 12:01 PM Jerry Geis wrote: > >> Just to verify that you did rerun configure after installing the >> libraries? >> >> Doug >> > > Oh that is a good one - I thought I did - but apparently not. menuconfig > now shows "*" > > So is chan_alsa going away ? What is it being

Re: [asterisk-users] asterisk 18.18.0 and chan_console

> > > Just to verify that you did rerun configure after installing the libraries? > > Doug > Oh that is a good one - I thought I did - but apparently not. menuconfig now shows "*" So is chan_alsa going away ? What is it being replaced with? thank you! Jerry --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

>>> Thanks doug - I did that - still showing XXX for chan_console Just to verify that you did rerun configure after installing the libraries? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

> portaudio19-dev Thanks doug - I did that - still showing XXX for chan_console libportaudio2/focal,now 19.6.0-1build1 amd64 [installed] libportaudiocpp0/focal,now 19.6.0-1build1 amd64 [installed,automatic] portaudio19-dev/focal,now 19.6.0-1build1 amd64 [installed] Jerry --

Re: [asterisk-users] asterisk 18.18.0 and chan_console

On my debian 11 install I needed to install portaudio19-dev Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

[asterisk-users] asterisk 18.18.0 and chan_console

I am still using chan_console. I compiled 18.18.0 and chan_console is not built. I am using ubuntu 20.04.6 LTS make menuselect says XXX chan_consoel and it needs "portaudio" What do I do next ? Also menuconfig is saying XXX on Also - what alsa library is needed ? Thanks jerry --

[asterisk-users] Asterisk Release 18.19.0

The Asterisk Development Team would like to announce the release of Asterisk 18.19.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.19.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] Asterisk Release 20.4.0

The Asterisk Development Team would like to announce the release of Asterisk 20.4.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

Re: [asterisk-users] asterisk sees private IP address of a device behind NAT

On Tuesday 11 July 2023 at 10:00:22, Fourhundred Thecat wrote: > Hello, > > my asterisk is working fine, I am just confused why, on the server I see > private IP address of an endpoint SIP is rather like FTP in that it embeds IP addresses (layer 3 of the OSI network model) in the protocol

[asterisk-users] asterisk sees private IP address of a device behind NAT

Hello, my asterisk is working fine, I am just confused why, on the server I see private IP address of an endpoint: WARNING: Retransmission timeout reached on transmission 0_252301488@10.1.3.8 for seqno 2 (Critical Response) the IP 10.1.3.8 is a phone behind NAT. Does it mean something is

Re: [asterisk-users] Asterisk Release 20.3.1

At the moment, I can't see any differences here. sha512sum is identical. Regards Michael On 08.07.23 at 01:50 Jean-Denis Girard wrote: Le 07/07/2023 à 12:49, Joshua C. Colp a écrit : On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard > wrote:     There seems to be a

Re: [asterisk-users] Asterisk Release 20.3.1

Le 07/07/2023 à 12:49, Joshua C. Colp a écrit : On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard > wrote: There seems to be a problem with the tar.gz archive on github. It's correct on downloads.asterisk.org . Can you be

Re: [asterisk-users] Asterisk Release 20.3.1

On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard wrote: > There seems to be a problem with the tar.gz archive on github. It's > correct on downloads.asterisk.org. Can you be more specific? They are identical and the same tarball. I just downloaded both from each place and confirmed that, and

Re: [asterisk-users] Asterisk Release 20.3.1

There seems to be a problem with the tar.gz archive on github. It's correct on downloads.asterisk.org. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 07/07/2023 à

[asterisk-users] Asterisk Release 20.3.1

The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

[asterisk-users] Asterisk Release certified-18.9-cert5

The Asterisk Development Team would like to announce security release Certified Asterisk 18.9-cert5. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert5 and

[asterisk-users] Asterisk Release 19.8.1

The Asterisk Development Team would like to announce security release Asterisk 19.8.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/19.8.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

[asterisk-users] Asterisk Release 18.18.1

The Asterisk Development Team would like to announce security release Asterisk 18.18.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.18.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

[asterisk-users] Asterisk Release 16.30.1

The Asterisk Development Team would like to announce security release Asterisk 16.30.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/16.30.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

On Wed, Jun 21, 2023 at 4:07 PM TTT wrote: > Something perhaps noteworth, since this is a multihomed system I bound the > transport to 172.31.253.4:5060 > > I don't *think* that would cause Asterisk to use that IP in the FROM...at > least it shouldn't. > > Copy/paste from FreePBX forum: It

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

...@lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

Asterisk to do that. -Original Message- From: Eric Wieling [mailto:ewiel...@nyigc.com] Sent: Wednesday, June 21, 2023 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion ; TTT Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE type

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

@lists.digium.com Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: I've split this thread off from another (PJSIP authentication) because I think the root cause is

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: I've split this thread off from another (PJSIP authentication) because I think the root cause is something different.I think the problem is the following FROM line in my SIP INVITE

[asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

I've split this thread off from another (PJSIP authentication) because I think the root cause is something different.I think the problem is the following FROM line in my SIP INVITE transaction: From: "MYNAME" ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an

[asterisk-users] Asterisk Release 20.3.0

The Asterisk Development Team would like to announce the release of Asterisk 20.3.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

[asterisk-users] Asterisk Release 18.18.0

The Asterisk Development Team would like to announce the release of Asterisk 18.18.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.18.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues

Re: [asterisk-users] asterisk 18.17.1 unreachable

On Thursday 11 May 2023 at 21:15:50, Jerry Geis wrote: > I have 4 devices that I connect here local and there is no issue. > I have those same 4 devices connecting from another location across the > internet. > > They all boot up, connect and register I can send audio to them and they > play. >

[asterisk-users] asterisk 18.17.1 unreachable

I have 4 devices that I connect here local and there is no issue. I have those same 4 devices connecting from another location across the internet. They all boot up, connect and register I can send audio to them and they play. - then at times they show UNREACHABLE and I can no longer send audio.

[asterisk-users] Asterisk issue reporting is now live on GitHub

All Asterisk issues should now be reported at https://github.com/asterisk/asterisk/issues The previous issue system at https://issues.asterisk.org remains in read-only mode for reference but will eventually be replaced with a searchable archive. --

Re: [asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.

On Fri, Apr 28, 2023 at 10:43 AM Benoît Panizzon wrote: > Hi List > > Asterisk 16.28.0 in use. > > PJSIP in use > Two endpoints > Both using IPv6 > > One Endpoint on UDP, the other via TLS. > > Both with: > > t38_udptl=yes > ;fax_detect=yes > ;fax_detect_timeout=30 > rtp_ipv6=yes > > Both sides

[asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice

[asterisk-users] Asterisk Infrastructure Move to GitHub

In order to reduce the amount of system maintenance and administration that needs to be done by the Asterisk team at Sangoma, we've decided to move capabilities such as issue tracking, code management/review and documentation/wiki to hosted solutions. Last year, we compared GitHub and GitLab and

[asterisk-users] Asterisk 20.2.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 20.2.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 20.2.1 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 18.17.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.17.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.17.1 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 20.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 20.2.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 20.2.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 18.17.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.17.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.17.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk simply stops call processing

I've been having a related problem. I have Asterisk with some call processing accessing Maria (hosted on the phone server, running Ubuntu) via func_odbc. That same odbc driver is used to write cdr records on a different server. I had never noticed a problem (and no threading attribute defined)

Re: [asterisk-users] Asterisk simply stops call processing

On Wednesday 22 February 2023 at 15:29:38, John Harragin wrote: > If there are multiple connections that the utilize the same driver, try > putting: > > Threading = 2 > > in the appropriate driver section of > /etc/odbcinst.ini I'll give that a go, however I doubt that it is the problem,

[asterisk-users] Asterisk PJSIP setting don't fragment bit on UDP

Hi Gang I noticed, that when I enable multiple codecs and rtp encrypting (generating a large SDP) invites with credentials do not get through anymore. So sniffed the connection and found that the IP packets have the don't fragment bit set, causing a VDSL router with 1472 MTU in the path to

Re: [asterisk-users] Asterisk simply stops call processing

If there are multiple connections that the utilize the same driver, try putting: Threading = 2 in the appropriate driver section of /etc/odbcinst.ini ...this would be a possibility if the problem is intermittent. Also can you successfully execute the same SQL from the cli? By the way,

[asterisk-users] Asterisk simply stops call processing

Hi. I have a strange problem and I'm looking for suggestions on how to investigate it. I have a dialplan which is processing a call, and Asterisk simply stops doing anything for that call. I have verbose and debug logging turned on. There are two steps at a particular point in the dialplan:

Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage

On Mon, Feb 6, 2023 at 6:05 PM Dan Cropp wrote: > A quick follow-up. > > > > Looking at other customers running 18.12.1 who reported problems at the > exact same time with AWS issue described below. > > > > We are seeing similar behavior. > > For these systems, the third STUN failure occurs. We

Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage

A quick follow-up. Looking at other customers running 18.12.1 who reported problems at the exact same time with AWS issue described below. We are seeing similar behavior. For these systems, the third STUN failure occurs. We were able to answer the call because the SIP provider didn't CANCEL

[asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage

Over the weekend, we had several customers running at AWS. AWS had an outage during this time. This customer is running Asterisk 16.23.0 (which has the STUN timeout crash fix). >From what I have been told, other customers are running newer Asterisk 18.12.1 >but encountered similar issues. (I

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