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...http://downloads.digium.com/pub/telephony/codec_opus/ It took me a while to figure this out, because initially I tried downloading via selecting the Opus codec in make menuselect and realizing that it isn’t there after make install step. Can someone from Digium/Sangoma please confirm? FLORIAN FLOIMAIR Software Development - IMS Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria Phone: +43 662 85 62 25 Mail: f.floimair at commend.com commend.com LG Salzburg / FN 178618z -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/p...

Well, that sounds pretty straight forward. I can do this and push it to gerrit. Do I need to create a ticket for this? With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com <http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg Am 22.08.19, 11:55 schrieb "asterisk-users im Auftrag von Joshua C. Colp&quo...

...situation. So far Alcatel just tells us that this is not SIP-compliant and that we have to change things on the Asterisk side, but I'm not quite sure that this is really the case and having arguments could help me clarifying this situation. Thanks in advance. With best regards Florian Floimair Technical Support COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 312 Sip: f.floimair at commend.com <file:///T:\KAT\Signaturen\=%22sip:f.floimair at commend.com%22> Fax: +43-662-85 62 26 f.floimair at commend.com <mailto:f.floimair at commend.com>...

...uation when using Alembic in Debian 9 (could happens in other Distros too). Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 12 July 2017 at 13:11, Floimair Florian <f.floimair at commend.com> wrote: > Nevermind guys! > > I just found out the solution myself: > > MariaDB in Debian uses utf8mb4 as default character set (see here: > https://mariadb.com/kb/en/mariadb/differences-in- > mariadb-in-debian-and-ubuntu/). > > I...

All, I'm using asterisk 16.4.0 with h264 and opus quite well using linphone 4.1 client on android and baresip on linux. I'm exploring use of h265 for improved video quality/lower network bandwidth. I do not see pass through support on asterisk for h265/hvec. All my SIP clients and underlying hardware have hvec/h265 encoding and decoding available. I would have liked vp9 however, vp9

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

...notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fissues.asterisk.org%2Fjira%2Fbrowse%2FASTERISK-27946&amp;data=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259&amp;sdata=JdT9Yvi7ml%2FqzIYMO39ks68rdMKY2P2DFIAGKCCh6a8%3D&amp;reserved=0> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't...

...ls this operation with error "Specified key was too long; max key length is 767 bytes" when it tries to increase some fields to varchar(255). Any idea how to solve this? Do I maybe have to switch to a different encoding for this to work? Thanks in advance ? ? With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag von...

...conf configuration and Asterisk 13 > with sip.conf (chan_sip). In both cases I just put the both case > AST_FRAME_VIDEO: statements before the two voice cases, like in your diff > and recompiled/reinstalled. > > Regards > > Benjamin > > > > 2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floimair at commend.com>: > >> Hi Benjamin! >> >> You're obviously using a similar scenario that I have in place for >> testing. >> I initially had issues with early media (not only video also audio) as >> well in that scenario. What I had...

...arly media (not only video also audio) as well in that scenario. What I had to do was to additionally set external_media_address=<your external IP> in pjsip.conf Also, as I wrote the patch for early-media video I'd be interested in any feedback from it. ? ? With best regards Florian Floimair Innovation - Software-Development -? VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg -----Urspr?ngliche Nachricht----- Von: asterisk-u...

I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg Von: asterisk-users-bounces at lists.digium.com [...

...ile? So far I have only been able to turn on ARI debugging in the console which results in the debug output being logged in /var/log/asterisk/messages I would love to have ARI debug log messages in /var/log/asterisk/debug or even better in it's own ari-debug file. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg -------------- next part -------------- An HTML attachment was scrubbed... URL: &lt...

On 02/13/2018 at 08:41 AM Floimair Florian wrote: > No you're reading it wrong. > > There are 188K received with no loss, and 16441K transmitted. This doesn't make any sense to me, either. There can't be more packages transmitted than received. It's the same codec in and out and it's been running exact...

...low? [2018-02-13 12:57:43] WARNING[4917][C-0004c2cb] codec_sangoma.c: [526559][g722toulaw] Got Seq 15944 but expecting 10106 (time since last read = 0ms), dropped 5838 packets On 02/13/2018 01:24 PM, Andres wrote: > On 2/13/18 11:55 AM, Michael Maier wrote: >> On 02/13/2018 at 08:41 AM Floimair Florian wrote: >>> No you're reading it wrong. >>> >>> There are 188K received with no loss, and 16441K transmitted. >> This doesn't make any sense to me, either. There can't be more packages >> transmitted than received. It's the same codec in...

...endpoint A to Asterisk there is no inband DTMF signal in the RTP audio stream. Can someone confirm this behavior? If yes than this is clearly a bug. I had a look in the code which introduced this feature and couldn't find anything obvious why this is happening. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com <http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg

Hello, I've discovered homer-api-postgresql and homer-api-mysql packages in Stretch repo. I'm not sure I understand how Homer-API relates to Homer. My questions are: 1. What is the simplest available installation option to install Homer on a dedicated box, this dedicated box gathering data from one or several Asterisk systems on the same LAN ? 2. Is it possible to centralize data on a

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: > Le 07/12/2018 à 14:32, hw a écrit : > > [...] >> >> Queues seem to be the only way to have several phones ring at once, or >> are there other ways? > > Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,) > Good to know, thanks! What are the entries needed in the queue_members table when using

Hi Carsten, On 02/11/2018 at 07:46 PM Carsten Bock wrote: > Hi, > > Lost percent (%).... Are you sure? I'm seeing here: ...........Receive......... .........Transmit.......... Count Lost Pct Jitter Count Lost Pct Jitter RTT.... 188K 0 0 0.000 188K 16641K 8809 0.000 0.026 => This doesn't sound reliable to me: there are 188K packets and 16641K

...de in this release: ----------------------------------- * ASTERISK-27278 - [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE (Reported by David J. Pryke) * ASTERISK-27255 - alembic: Add support for Microsoft SQL server (Reported by Florian Floimair) * ASTERISK-27253 - [patch] libsrtp-2.1.x support (Reported by Alexander Traud) * ASTERISK-27220 - Enable CHANNEL function to get from and to tag from SIP Headers (Reported by Andre Nazario) * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif (Reported b...