Thanks Virendra, That was a good start but not quite what I am looking for. I want to know the details of listening to a specific event rather than listening to all. My questions specifically is regarding the Manager command originate. I just used..
,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [22..
, I want to use text file to get password information with Authenticate Application. I am using asterisk 220.127.116.11. I made text file at /tmp/pass.txt with below information. *pass.txt* Virendra: 81dc9bdb52d04dc20036dbd8313ed055 Vijay : 9996535e07258a7bbfd8b132435c5..
Why php? Isnt vi the only way? On Fri, Sep 2, 2011 at 7:28 AM, virendra bhatiwrote: >, > > I want ot do basic work (add-edit-delete) into asterisk configuration files, > like sip.conf, manager.conf,musiconhold.conf etc. > > Please guide me how to config..
, Do you have any suggestion in DTMFcase ?? I have change my sangoma card to digiumbut still same…… On Mon, May 23, 2011 at 4:08 PM, virendra bhatiwrote: >, > > After changes relaxdtmf=no in chan_dadhi.conf. problem is not resolve > > On Mon, ..
What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex Da: email@example.com [mailto:firstname.lastname@example.org..
Virendra, It may be problem for rtp packet port forwarding if u can dial through DID number. You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port. please, write how can you dial call mobile or other devices. e.g. DID number, PRI num..
Virendra, Set DTMF option in the Makefile to 1 and then recompile/install the app_konference module. Thanks Krishna On Tue, Jun 7, 2011 at 1:31 AM, virendra bhatiwrote: >, > > I am trying to get DTMF into conference room. for conference I am usin..
On 2011-05-30 14:32, virendra bhati wrote: >, > > Asterisk s *ControlPlayback* will used for play any recorded file as > an audio player. Is it possible that we can use it for multiple > forward and rewind ? > > ex:- > original: ControlPlayback(filename,skipms,ff,rew,stop,pau..
Thanks for the input. Yes, I did call out many times, but the recording doesnt happen even after the call is bridged and there is two way audio. I also took out the b option and so it should recording the ringing right (even before call is bridged) ..