NAT endpoint calling local endpount - switching to native_rtp then no audio,
both of them have direct_media=no, Verbose log:
-- Executing [99@dialmap:1] AGI(PJSIP/304-0022, /pbx/agi.php) in
new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial)
because of problems you are facing i decided to go way with second table
CREATE TABLE `cdr_extended` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`uniqueid` varchar(32) NOT NULL DEFAULT '',
`callid` varchar(256) NOT NULL DEFAULT '' COMMENT 'sip call-id',
`hangupcause` varchar(10)
On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome jl...@me.com wrote:
NAT endpoint calling local endpount - switching to native_rtp then no audio,
both of them have direct_media=no, Verbose log:
-- Executing [99@dialmap:1] AGI(PJSIP/304-0022, /pbx/agi.php) in
new stack
-- Launched
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC
endpoints registered on asterisk 13 it is a feature that definitely asterisk 13
should support .
the problems that i faced with this is the following and i hope i could get
2015-03-18 12:54 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
I'm confused this is not a patch, it's just garbage ;), I'm making a
connection xmpp with asterisk and not connected, at the cli shows me
the message every second:
RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP
On Wed, Oct 29, 2014 at 7:10 PM, sean darcy seandar...@gmail.com wrote:
On 10/29/2014 08:06 PM, Matthew Jordan wrote:
On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote:
Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
codec_silk for Asterisk 12
Hello.
I have plain text password for endpoint with outbound registration
(someone else's server).
My aim is to write it in pjsip.conf.
md5 means that I know realm. I do not always know it.
Is where any way?
Dmitriy Serov.
--
All;
I'm running Asterisk 11.6-cert9 and am trying to play a pre-recorded
audio file to extensions using the Page() command. The dial plan looks like
this:
exten = s,n,Page(${AVAILCHANS},A(${AUDIOMSG})) and the paging by itself
works great. However, when I try it with the audio file, it
On Thu, Mar 19, 2015 at 1:22 PM, Steve Murphy m...@parsetree.com wrote:
On Wed, Oct 29, 2014 at 7:10 PM, sean darcy seandar...@gmail.com wrote:
On 10/29/2014 08:06 PM, Matthew Jordan wrote:
On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote:
Can we expect a SILK codec
