On Fri, 24 May 2013, Markus wrote:
Am 23.05.2013 16:04, schrieb Richard Kenner:
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)
www.mhspot.com/sts/
(site is down right now)
And that's related to the problem with it: it hasn't been maintained f
Hi.
simply i followed the instructions on
http://www.asteriskguru.com/tutorials/asterisk_gui.html and it works.
up to now i have it working in the following OS's:
-debian squeeze
-ubuntu server 12.04
-centos 5.9
all of them work fine.
:
Hi Alec,
I followed that both instructions . It
On 05/25/2013 11:42 AM, luke devon wrote:
> Hi Alec,
>
> I followed that both instructions . It was not success. got the same
> issue. I'm not sure the reason is the OS. I am Using CentOS 6.4.
>
> However, following method was helped me to resolve the problem.
>
> http://www.digip.org/jansson/r
Hello,
Trying to use g729 in pass-thru mode.
Call flow:
SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729)
When using G.729, call is not getting connected. Below is the extract from CLI.
== Using SIP RTP CoS mark 5
-- Executing [12127773456@default:1] AGI("SIP/100-", "call
Hi,
i am trying to install asterisk newer version on the Elastix machine, but
while installing the chan_sip,c module is not building while make. when i
see in make menuselect options it showing "XXX" -- extended , please let
me know how to enable it and make build chan_sip module.
--
Upendra
-
