Thanks Stiles
Trying as u asked to do
Regards
On Fri, May 6, 2016 at 6:10 PM, A J Stiles <asterisk_l...@earthshod.co.uk>
wrote:
> On Friday 06 May 2016, Alok Srivastava wrote:
> > Dear List
> > wanna configure click2call in such a manner that my asterisk box call t
Dear List
wanna configure click2call in such a manner that my asterisk box call two
mobile numbers and connect both numbers to talk. I have configured voip
gateway, my internal and external calls are working fine.
please help ,
abhi
--
*Dear List*
Plz help, i am not much experienced with asterisk. i configured it on
ubuntu 12.04. no problem when i call any mobile no(0091XX) but when
i call on my local asterisk no.(101,102 or 105) it is not connecting
giving error
Auto fallthrough, channel 'SIP/lucknow-006f' status
yes u can access form same phpmyadmin both database, depends, for which
database u entered userid and password on phpmyadmin login page.
On Thu, Sep 11, 2014 at 2:06 PM, rafa alfurqan rafa.alfur...@gmail.com
wrote:
Hi,
thank you for your repplied,
As you're on Ubuntu, you can begin with
dear lists
trying to integrate google calendar with asterisk 1.8.20.1.
but 'calendar show calendars' not showing anything.
when i run 'show modules' on asterisk prompt.
it is not showing res_calendar_caldav.so module, only showing
res_calendar.so module .
is there anything wrong with google API.
can i test my asterisk11 on a single machine on which asterisk is installed
that sounds are working from both end properly.
i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone.
regards
abhi
--
_
-- Bandwidth and
dear
is there any study material for implementing click to call in asterisk.
plz help.
thanks
regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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dear
please Help. I am continously getting this message after sip set debug
on. and not getting clear voice from both side.
--- Transmitting (NAT) to 122.163.193.94:1893 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.106:5060
Sammy
On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote:
dear
please Help. I am continously getting this message after sip set debug
on. and not getting clear voice from both side.
--- Transmitting (NAT) to 122.163.193.94:1893 ---
SIP/2.0 404 Not Found
Via: SIP/2.0
systems and set sip | rtp as filter and see where are the RTP streams
going on each end !
Take a complete capture on Asterisk server by executing the command sip
set debug on and make a call.
BR
Sammy
On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon
dig...@sanguinarius.co.ukwrote:
alok
dear
i have configured properly asterisk. At the one end i am using x-lite soft
ph and another end twinkle. call is going properly from both end but after
picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always showing
closed while i have
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