Hi Users,
Does Asterisk provide any way to monitor the SIP call setup time between the
clients ??
I understand that there is a way to monitor the RTP data flow for jitter and
packet losses
using *ship show channelstats*. I am looking something on similar lines to
monitor call
setup time during SIP
The cellphone can be presented to Asterisk as SIP device using OpenBTS (GSM
to SIP conversion).
On Tue, Feb 15, 2011 at 10:40 PM, Faisal Hanif wrote:
> Hi,
>
>
>
> Your question is not clear but below are possible answers to your question,
>
>
>
> If you want to attach you cell-phone to asterisk
aved
from extconfig.conf thing :)
Thanks,
Abhinav
On Wed, Jan 19, 2011 at 4:43 PM, Carlos Chavez wrote:
> On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote:
> > Un-top-posting...
> >
> > On Wed, 19 Jan 2011, abhinav anand wrote:
> >
> > > I am using Ast
[pbx_ael]
[ Context 'ael-international' created by 'pbx_ael' ]
Include => 'ael-longdistance'[pbx_ael]
Include =>'ael-trunkint'[pbx_ael]
Ignore pattern =
information:
- My asterisk version is *Asterisk 1.6.2.5-0ubuntu1.1 built by buildd @
palmer on a i686 running Linux on 2010-07-16 13:24:33 UTC*
- I am not able to verify the symlink between the two extensions.conf files
Thanks,
Abhinav
On Wed, Jan 19, 2011 at 2:38 PM, Steve Edwards wrote:
> On
ailed.
*
I have already created a dialplan in my extensions.conf, I am not sure what
is happening here ??
Badly need help in this.
Thanks,
Abhinav
On Tue, Jan 18, 2011 at 8:37 PM, Steve Edwards wrote:
> On Tue, 18 Jan 2011, abhinav anand wrote:
>
> The exact error thrown on Asterisk C
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of "extension not found" when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI
