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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 support Stir Shaken. ( Case #01466843 / > 0013000000G8PLG / MAIN / Open [ ref:_00D306mPe._5...

...39;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 to Local channel 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 th...

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]

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

...calls hang on odbc calls to the local database. In my case the remote server is 5' away and in the same switch that Asterisk is plugged into, so it hasn't been critical to fix this - but it may provide some clues to your situation. On Thu, Aug 17, 2023 at 11:04 AM Federico <federico at digitalipvoice.com> wrote: > 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 > &...

What can I do to troubleshoot this error? The machine is fine. I can do isql 'anydatasource' and it works module load res_odbc.so Unable to load module res_odbc.so Command 'module load res_odbc.so' failed. [Apr 23 20:25:59] ERROR[28838]: loader.c:283 module_load_error: res_odbc has one or more unknown dependencies.

I am banging my head. Stock asterisk on Ubuntu 22.04 (Jammy) installs and works fine, but I want to update the source code. I use this configure line ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib/x86_64-linux-gnu --with-unixodbc=$(odbc_config --include-prefix)/ --with-pjproject-bundled --disable-dev-mode --with-user=asterisk And it works, but the when I do systemctl start

I am using Asterisk 16.30 inside Freepbx, with commercial modules, purchased from Sangoma and Symphony. After a few hours my memory usage reaches 900 GB, no kidding, in a box with 1 TB of RAM. The question is: how can I determine what is causing the memory leak? Can somebody send me instructions to find out what module is killing my box? FreePBX is 100% updated. Before I contact a vendor I need

I need to use app_macro, but it seems to be absent from asterisk 16.30.1 Is there a workaround? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230710/8b3814ad/attachment.html>

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 %H:%M:%S)}) to that end, I modified cli.conf [startup_commands] originate local/s extension s at default = yes But now I upgraded to Asterisk18 and there is no longer a local channels Does anybody have any idea of a workaround?

Yes that are, but how do I use them to execute a part of the dialplan, once, when Asterisk starts up. module show like originate Module Description Use Count Status Support Level app_originate.so Originate call 0 Running core res_clioriginate.so Call origination and

I cannot follow your instructions, because asterisk segfaults on start. It never starts Can you give me instruction to trap this segfault on starting asterisk? Like gdb …..asterist –gvvvvvvc From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Mark Murawski Sent: Saturday, August 19, 2023 11:04 PM To: asterisk-users at lists.digium.com Subject: Re:

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