similar to: Feature wishes for Asterisk 14

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 app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: > On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Where should core file be created when Asterisk is run as a daemon by > > > asterisk user and group ? > > > Is there a setting I

I think were getting closer: I did: - I edited /etc/default/asterisk to include : AST_USER="root" AST_GROUP="root" # systemctl daemon-reload # systemctl start asterisk # ps aux | grep asterisk root 3602 7.1 2.5 60332 26012 ? Ssl 16:00 0:03 /usr/sbin/asterisk -U root -G root -g # rasterisk # pkill -SEGV asterisk Then console showed: Segmentation error (core

Hello, I?m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean?

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

Hi, One of the key features of Asterisk is that we can install it on many hardware platforms. We've done our best to script this installation process, so that, in case of hardware failure, we can re-install Asterisk on another platform. The question I have is how can we adapt our process so that Digium's G729 licences (or other licenced software) could be installed without asking too

2014-08-13 19:06 GMT+02:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Aug 13, 2014 at 12:01 PM, Olivier <oza.4h07 at gmail.com> wrote: >> After installing various packages, here is what I did: >> >> TDIR=/usr/src >> cd $TDIR >> PJOPTIONS="--prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register

I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress

Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Some simple questions about codecs: What codec does the Zap channel use by default? Can this default be changed, and to what? (g729 too?) What codec does meetme use? (I think this is ulaw, but asking to be sure) Can you use another codec, or does everything have to be transcoded to ulaw? Finally ... if I have a 3way call going, between 1 g729 caller and two other callers, do I need one

Hi, I've installed a brand new Asterisk 12.0.0 system in which I can see, with make menuselect, in Test Modules tab, that each test entry such as test_acl can't be installed due a to missing TEST_FRAMEWORK(E) dependency. Where this TEST_FRAMEWORK(E) comes from ? How can it be installed ? I've installed testsuite according

Hello, I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. I followed this: cd /usr/src wget ... asterisk-13.13.1.tar.gz tar zxf asterisk-13.13.1.tar.gz cd asterisk-13.13.1 ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr" ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled make menuselect (shows res-srtp is available) make latest make command fails with

I am trying to install G729 licence on my Virtuozzo server running asterisk but I keep getting an error as it has no eth0. I get the following error when running register: [root@lvpsXXX-XXX-XXX-XXX root]# /root/register G729-XXXXXXXX Digium Product Registration Copyright (C) 2004, Digium, Inc. Analyzing key 'G729-XXXXXXXX' Connecting to Digium License Server (216.207.245.3:5646)...OK

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

With these planned changes, you mention they will be removed from 'core', so is the intention to allow more of these legacy plugins to more to their own repo's for those that might still want to use them? Eg, more in the the path of plug-in architecture? Just curious for clarity. Oh, and have to add.. still waiting on the variable capability patch, to allow plugins to modify

Hi, I'm trying to install spandsp from source in a Debian Lenny system. I did : cd /usr/src wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre5.tgz tar xvf spandsp-0.0.6pre5.tgz cd spandsp-0.0.6 ./configure make make install When doing this, spandsp warns me that librairies are installed in /usr/local/lib with this message : "Libraries have been installed in:

Amongst other plug-ins, yes.. On 2020-03-17 10:15 p.m., Aki Tuomi wrote: > Those plugins can be resurrected with simple `git revert`. > > Are you planning on open sourcing your client id authetication plugin? > > Aki >> On 18/03/2020 01:44 Michael Peddemors < michael at linuxmagic.com >> <mailto:michael at linuxmagic.com>> wrote: >> >>

Hi, just a question: Let say I have 2 phones with G729 onboard, but no 729 licence for Asterisk. Preferred codec set up in phones is G729, followed by ULAW, in Asterisk I have allow=ULAW deny=ALL. When call is reinvited by Asterisk will the two phones use G729 between each other or they will stick to ULAW they used for first part of the call ? A quick test showed that they will use ULAW ...

Hi! I understand that it is not trivial to move away from vpopmail and does require changing a working system. But then again, one should be able to configure MySQL passdb/userdb with vpopmail schema. I am not familiar with vpopmail but if someone comes with instructions we can polish them a bit (if necessary) and publish them as howto on doc.dovecot.org. Aki > On 18/03/2020 17:52 Remo