Re: [asterisk-users] set framing on dynamic interface DAHDI

You can do it like this: dynamic=eth,eth3/04:74:a1:00:05:8e/1,31,0 bchan=32-46,48-62 dchan=47 2016-01-22 23:22 GMT-02:00 Rafael dos Santos Saraiva : > Hi > > I working with DAHDI Dynamic Interfaces using ethernet boards. I need set > the framing to CCS, but the

Re: [asterisk-users] CEL entries over ODBC several hours late (Vinicius Fontes)

Since the issue seems to be table locking, why not take a shot with PostgreSQL? It's way better and more robust than MySQL/MariaDB. You should be able to create an additional DSN and output CEL to both PostgreSQL and MariaDB. 2015-12-11 8:59 GMT-02:00 Stefan Viljoen :

Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

Sorry for the probably obvious question, but it's better to cover all bases. The DBMS is running on the same box as Asterisk is? If that's the case then maybe the DBMS is using too much CPU and starving Asterisk? 2015-12-10 12:57 GMT-02:00 Stefan Viljoen : > Hi Matthew

Re: [asterisk-users] Call forwarding in Asterisk

You might want to use the Originate() application instead. Check its usage by issuing the command 'core show application originate' on Asterisk CLI. 2015-09-03 9:09 GMT-03:00 Kantharuban Ruban : > Hello Group, > > I have a requirement to dialout some external

Re: [asterisk-users] webrtc no audio

: are you sure you dont have this problem? https://issues.asterisk.org/jira/browse/ASTERISK-24146 i'm now fighting with https://issues.asterisk.org/jira/browse/ASTERISK-24602 Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a): I have it working now! *I had to install Asterisk 13 with PJSIP

Re: [asterisk-users] webrtc no audio

=stun.l.google.com:19302 *res_stun_monitor.conf:* stunaddr = stun.l.google.com:19302; Address of the STUN server to query. stunrefresh = 30 2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz: Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): Vinicius Fontes wrote: I'm having the same issue

Re: [asterisk-users] webrtc no audio

: Vinicius Fontes vinic...@aittelecom.com.br Date: 2015-07-27 13:54 GMT-03:00 Subject: No audio on SIP over WebRTC To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I'm following this tutorial ( https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial

[asterisk-users] No audio on SIP over WebRTC

I'm following this tutorial ( https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to deploy WebRTC support but I'm having an issue with RTP when the WebRTC softphone is behind NAT. In my scenario, the Asterisk server is running a public IPv4, and the softphone is behind NAT.

Re: [asterisk-users] How to make asterisk work with remote mysql database?

To install start script: make config To install samples (this will overwrite all files in /etc/asterisk!): make samples I usually do this when I need to compile and install Asterisk: ./configure make menuselect make make install make config make samples 2015-05-07 14:44 GMT-03:00 Manish

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villací­s Lasso a_villa...@palosanto.com: El 07/04/15 a las 17:38, Alex Villací­s

Re: [asterisk-users] OpenVZ with asterisk 13

I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani mi...@enterux.in: Show him this freaking thread, or else ask him to prove it otherwise. We all here have decades of exp dealing with

[asterisk-users] CDR dst value null after attended transfer

I'm having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm