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
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 :
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
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
:
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
=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
: 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
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.
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
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
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
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
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