[asterisk-users] Question regarding SIP 183 Session Progress handling in Asterisk

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

Re: [asterisk-users] pcapsipdump or general sip debug question - the solution

Or you may use sngrep if you prefer command line tools     With best regards -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Yves Gesendet: Dienstag, 17. Januar 2017 14:20 An:

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

files: 50-client.cnf (1 line) 50-mysql-clients.cnf (1 line) 50-server.cnf (2 lines)     With best regards Florian Floimair  -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet

[asterisk-users] 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

[asterisk-users] Asterisk realtime in combination with ARI - error while trying to prepare SQL statement for writing into database

Hey guys! I successfully got Asterisk realtime (14.6.0) with MariaDB (MySQL fork) running on Debian 9. I will document the steps to do so shortly (the main difference is default encoding and the odbc connector & its configuration). What I’m trying to do now is to use ARI to create PJSIP

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

/mhterres https://linkedin.com/in/marceloterres On 12 July 2017 at 13:11, Floimair Florian <f.floim...@commend.com<mailto:f.floim...@commend.com>> wrote: Nevermind guys! I just found out the solution myself: MariaDB in Debian uses utf8mb4 as default character set (see here: https://mariad

Re: [asterisk-users] PJSIP list of peers online/offline?

You can try: pjsip show endpoints However, there is no summary line in the end (only the total number of objects) so you will have to parse the status of each entry yourself to get these statistics.     With best regards Florian Floimair COMMEND INTERNATIONAL GMBH http://www.commend.com

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet: Donnerstag, 13. Juli 2017 11:52 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: Re: [asterisk-users] Asterisk re

Re: [asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.

I have seen this log message as well. No clue yet as to what the reason for this might be. Any hint is appreciated.     With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com Security and Communication by Commend FN 178618z | LG

Re: [asterisk-users] [OT] Overview of Homer installation on Debian Stretch

@1) Not on the host, as jessie is only used within the container but you may run it on a Stretch host of course. @2) The HOMER API contains modules for mysql and postgresql. The Debian maintainers simply split the resulting packages into three subpackages (homer-api general parts, db parts for

Re: [asterisk-users] [OT] Overview of Homer installation on Debian Stretch

1. The simplest option would be to run the Docker multicontainer-setup as described on the HOMER wiki here: https://github.com/sipcapture/homer/wiki/Docker-Install 2. Sure it is. Just edit your homer configuration to reflect the remote DB server 3. If you really want to learn and

[asterisk-users] Asterisk Realtime PJSIP - slow output on "pjsip show xxxxx" commands

Hi all! I have an Asterisk instance (15.4.1) with a MySQL DB in realtime configuration (using MySQL ODBC connector). I noticed, that when I issue “pjsip show endpoints” in the CLI this takes forever (on average about 1 minute with 1029 entries in ps_endpoints table). If I query the database

Re: [asterisk-users] Big leap - 1.8 to 15.4.0

You more or less have to follow the guidelines as if you were doing one step at a time (in terms of version) upgrades. So you should consider the following documents in the source tree: UPGRADE-10.txt UPGRADE-11.txt UPGRADE-12.txt UPGRADE-13.txt UPGRADE-14.txt UPGRADE.txt (which would be 15)

[asterisk-users] 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

Re: [asterisk-users] Logging ARI debug messages

messages On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote: > 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 o

Re: [asterisk-users] asterisk mysql contacts

Yes there is. You can follow the ODBC section in this document: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime     With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH http://www.commend.com Security and Communication by Commend

Re: [asterisk-users] What does pct mean?

No you're reading it wrong. There are 188K received with no loss, and 16441K transmitted. Still 8809 does not sound like a percentage to me  so there is something wrong with either the label or the value. From what's in the code, you can see it's clearly a lost Packet count not a percentage.

[asterisk-users] Encrypting passwords in config files

Hey there! I was wondering what the best practice is concerning passwords in Asterisk's config files. ari.conf has a neat feature where one can use a pre-encrypted password by using password_format=crypt for an ARI user However, I was wondering how to do similar things with e.g.

Re: [asterisk-users] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))

Can an administrator please throw out i...@online4you.nl of the mailing list. It's a bit annoying when one third of the messages is always the one below. Thanks! With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße

Re: [asterisk-users] Asterisk behind NAT Early Media Video

/reinstalled. Regards Benjamin 2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floim...@commend.com<mailto:f.floim...@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 a

Re: [asterisk-users] 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= in pjsip.conf Also, as I wrote the patch for

Re: [asterisk-users] Early or Pre VIdeo

Hi John! There is no "clean" way yet that I am aware of, however I got it running by modifying the code in the Dial() application. By default Dial() blocks anything other than audio in the early-media stage (no idea if this is wanted behavior or just not yet implemented). I have a patch in the

[asterisk-users] 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

Re: [asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints

loimair 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 26.09.18, 15:47 schrieb "asterisk-users im Auftrag von Joshua Colp

Re: [asterisk-users] how to use a database

Alembic currently doesn't cover queue_logs. As of now it only covers configuration, voicemail and cdr. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com

Re: [asterisk-users] Detecting a fax

I would guess from your explanation that the "outgoing" call somehow ends up in your Asterisk machine again, either at the voicetest or fax extension. You don't answer it in either of the extensions. That's what TOOTAI meant. If this is done in another extension, than this part of the Dialplan

Re: [asterisk-users] Freepbx / Asterisk PJsip multipe devices

Or just do it right using the PJSIP_DIALPLAN_CONTACTS function: Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com

Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working (John Covici)

You need to run make uninstall_all while you still have 13.24.0-rc1 checked out. Then checkout the previous version, rebuild it and make install. 13.15.0 doesn't know anything about modules added by 13.24.0. You usually would get a warning when running make install that there are modules present

Re: [asterisk-users] Early Media Issue

Just a guess, but I suspect that this might be related with strictrtp setting in rtp.conf, which learns the correct source in doing so drops a few packets. I would try to disable strictrtp for testing purposes and if this works add some delay before playing back the media. With best regards

Re: [asterisk-users] 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

Re: [asterisk-users] Digium's Opus Codec download links broken?

Colp" Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion Datum: Donnerstag, 14. November 2019 um 14:51 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Digium's Opus Codec download links broken? On Thu, Nov 14, 2019 at 9:46 A

[asterisk-users] 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

Re: [asterisk-users] Length of dial string

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the

Re: [asterisk-users] [External] 180 Ringing missing

If you have 183 Session progress, there is no need to send 180 Ringing (especially not AFTER 183 Session progress), as you already have early media instead. Having both is actually a bit misleading IMHO. So this is actually correct. One should not rely on any of these 1xx "Provisional"

[asterisk-users] Multiple 183 Session Progress for a single call

Hi List! I am running into an „issue” that I cannot really explain. I have a call from Station A to Station B with both legs connected to Asterisk via Kamailio. Asterisk used is latest 18.1.0 with chan_pjsip. Station A -> Kamailio -> Asterisk -> Kamailio -> Station B Now when Station B gets

Re: [asterisk-users] [External] Re: Multiple 183 Session Progress for a single call

n Progress for a single call CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. On Thu, Dec 17, 2020 at 10:14 AM Floimair Florian mailto:f.floim...@commend.com>> wrote: Hi Lis

[asterisk-users] SIP INFO messages with Content-Type: application/media_control+xml

Hi all! We have a WebRTC user-agent (using sip.js) that is giving me headache. When a different user-agent calls this user-agent, we frequently see Asterisk generating SIP INFO messages with Content-Type: application/media_control+xml Content-Length: 178 In the payload,

Re: [asterisk-users] [External] Re: SIP INFO messages with Content-Type: application/media_control+xml

fe. On Wed, Oct 13, 2021 at 10:05 AM Floimair Florian mailto:f.floim...@commend.com>> wrote: Hi all! We have a WebRTC user-agent (using sip.js) that is giving me headache. When a different user-agent calls this user-agent, we frequently see Asterisk generating SIP INFO messages wi

Re: [asterisk-users] [External] Re: Asterisk 16.23.0 doesn't respond anymore

Still this is a PITA. Just use your name so we know who we are talking to from the headers without looking at the body. There should only ever be one "Administrator" on a mailing list, which is the admin of the list itself. If you are admin in your own domain, we simply shouldn't care about. So