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I tried to download Digium’s Opus Codec via the following link, but the server is unavailable: 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

Hi, I am working on R 2.15.2 on Win. 7. I am trying to run some simple commends. >class(SWX.RET) # SWX.RET is a data file that has been loaded. But, I cannot see the expected output. I have deselected "buffered output". Still it does not work. Any help will be appreciated. Thanks [[alternative HTML version deleted]]

Dear Asterisk community! I am having trouble with a project concerning the 183 Session Progress SIP messages. Asterisk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is

Hi there! Is there any way I can turn on debug for ARI and sending the output to a separate log file? 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

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

Dear expert, I have two coefficients and covariance matrix. My objective is sampling 1000 times from the mean and covariance matrix. In order to get that, what kind of commend should I use? If you do not mind, could you tell me the comment in detail about parameter used in that commend also? Thank you. Sung. [[alternative HTML version deleted]]

Hello all ,  I have a data frame like this  X1X2X3 11815 22916 331017 441118 551219 661320 771421 now i want to randomly reorder the variable X2  but the row element should be same  as for example  X1X2X3 12916 251219 331017 471421 561320 61815 741118 how can i do that ?? Hint :  this could be helpful :  if X2 is only a vector like this  X2<-c(8,9,10,11,12,13,14) so i can easily reorder

Hi !  Facing problem with " getSequence" commend .  when only biomaRt package loaded the following example working well  >mart <- useMart("ensembl",dataset="hsapiens_gene_ensembl") >seq = getSequence(id="BRCA1", type="hgnc_symbol", seqType="peptide", mart = mart) show(seq) but when i have loaded the seqinr, i got problem

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

I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed? (On another P2P SIP Server the early media video works.) 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>: > Hi Florian > > I already have the external_media_address set in the

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 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

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

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 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 exactly the same time. > ...........Receive.........

Hi! I just tried setting up Asterisk realtime database following the wiki article https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to MariaDB). One has to install mariadb-plugin-connect, python-mysqldb and alembic packages (alembic does not work when installed via pip). Additionally - since MariaDB by default does not have a

Please open a Ticket (https://issues.asterisk.org), to let them know that they need to update the documentation in Wiki and also handle this situation 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

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

Could this gap in sequence numbers caused by a codec change generate errors like the one below? [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

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