Re: [asterisk-users] click2call for conferencing two mobile numbers

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

[asterisk-users] click2call for conferencing two mobile numbers

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 --

[asterisk-users] NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)

*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

Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?

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

[asterisk-users] res_calendar.so and res_calendar_caldav.so

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.

[asterisk-users] testing asterisk11 on single machine

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

[asterisk-users] click to call

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 -- New to Asterisk? Join us for a live introductory

[asterisk-users] sip set debug on always showing error

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

Re: [asterisk-users] sip set debug on always showing error

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

Re: [asterisk-users] port 5060 is blocked by ISP

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

[asterisk-users] port 5060 is blocked by ISP

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