similar to: Question regarding SIP 183 "Session Progress" handling in Asterisk

Displaying 20 results from an estimated 1100 matches similar to: "Question regarding SIP 183 "Session Progress" handling in Asterisk"

2018 Jan 11
2
Logging ARI debug messages
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
2018 Apr 11
2
Asterisk behind NAT Early Media Video
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
2017 Jul 12
2
Asterisk realtime - Error with index length in alembic script
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
2018 Apr 10
2
Asterisk behind NAT Early Media Video
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
2017 Jul 12
2
Asterisk realtime - Error with index length in alembic script
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
2018 Apr 10
2
Asterisk behind NAT Early Media Video
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
2019 Nov 14
3
Digium's Opus Codec download links broken?
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
2019 Aug 22
2
h265 codec pass through on asterisk
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
2018 Sep 26
2
chan_pjsip: DTMF mode "auto_info" on endpoints
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
2018 Feb 13
2
What does pct mean?
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.........
2018 Feb 13
3
What does pct mean?
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
2017 Dec 12
2
[OT] Overview of Homer installation on Debian Stretch
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
2019 Aug 22
2
h265 codec pass through on asterisk
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
2019 Jun 14
2
Early Media Issue
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
2018 Dec 07
4
how to use a database
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
2018 Feb 12
2
What does pct mean?
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
2018 Sep 09
2
Autoreply ( Autoreply (Re: getting invites to rtp ports ??))
Bedankt voor uw bericht. Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u geinformeerd over de omstandigheden en uw opties. Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of beantwoord. Indien uw abonnement is overgenomen door KovoKs, kijk dan voor contactgegevens op https://www.kovoks.nl/. Dank voor uw vertrouwen de afgelopen jaren! Met
2018 Apr 09
3
Asterisk behind NAT Early Media Video
wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark
2020 May 01
4
Length of dial string
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
2019 Jan 24
2
trying to upgrade asterisk and Debian -- not working (John Covici)
What procedure did you follow to revert back to the old version? It sounds like your binary has been revereted, but the modules it needs to load are still the 13.24.0-rc1 modules... --- Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian 8 to DEbian 9 to get a more recent operating system and