similar to: Is there a way to compile app_macro in 16.30.1

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the "cannot be completed as dialed" message.

Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 382 and voicetel if you want an intro. On Thu, Aug 17, 2023 at 11:50 PM Federico <federico at digitalipvoice.com> wrote: > I am looking for a decent provider of SIP Trunks but it has to pass the > Stir Shaken token to the next carrier. Does anybody know about any? > Sipstation from Sangoma, does not

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

I tested this issue with version 13 and version 18. In res_odbc.conf, if I add a second, new data source like [asterisk] enabled=yes dsn=asterisk sanitysql => select 1 isolation => read_committed username=root ;password= pre-connect => yes forcecommit => yes connect_timeout => 10 negative_connection_cache => 0 max_connections =>500 my odbc.ini [cdr]

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

It's a great idea but it doesn't work. Maybe this should be the way that works. -----Original Message----- From: Eric Wieling <ewieling at nyigc.com> Sent: Thursday, August 17, 2023 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; Federico <federico at digitalipvoice.com> Subject: Re: [asterisk-users] Alternative

You can't set the variable in globals? I don't know if functions work in globals, but it is worth a try. [globals] LSESSION=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} On 8/16/23 20:39, Federico wrote: > I used to use the local channel to create a global variable > > (dialplan) > > [default] > > exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d

I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / 0013000000G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) Although it's mandatory, somehow they think it's ok. Go figure. -------------- next part -------------- An

Hi everybody, I have a Fedora Core 4 x86 32 bit install, which I recently upgraded from asterisk 1.2 to the office 1.4.4 tarball. In the process of doing that I had to upgrade some autoconf/automake stuff, but it worked fine, and my new asterisk works fine. Except that anytime I receive a fax with spandsp and app_rxfax, asterisk seg faults. I have applied the spandsp

I have a trixbox 2.2 and Nortel santral that are speak each other. I use digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some problems when I want to make any call from extension of trixbox to extension of nortel. When I attend to log (/var/log/messages) I meet with these messages as you see below. When I try to make any call from trixbox extension the call seems established but

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

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 reported by the community and would have not been possible without your participation. Thank You!

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 reported by the community and would have not been possible without your participation. Thank You!

I create my sip users using a common macro in 1.4: [internal] exten => 200,1,Macro(phones|200|SIP/200) [macro-phones] exten => s,1,Dial(${ARG2}|45|Tt) etc... But now in 1.6 this fails: -- Executing [200 at handsets:1] Macro("SIP/201-0942b530", "phones|200|SIP/200") in new stack [Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context

Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro("SIP/1007-2165", "dialnumber_wvm,1004,SIP/1004") Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro