[Freeswitch-users] Hide the Caller ID

How to Hide the Caller id while I make outbound calls? I found that one way is "privacy" application using: But It is not working. Is there any better way to perform this? Thanks msp ___ Freeswitch-users mailing list Freesw

[Freeswitch-users] Test Proxy Media

Hi all, I have enabled proxy media from dialplan. After that, I can make calls same as it done before without enabling proxy media. So, how can i test that my calls are in proxy media mode after enabling "proxy-media" mode ? Thanks, MShehzad ___ Freesw

[Freeswitch-users] Hangup cause return false

Hi all, I bridge the call using bridge(session, newsession); After hangup i use the following line to get start time of the call: starttime=session.getVariable("start_stamp") ; This works fine in all case, except one, When called party hangup, "start_stamp" becomes false. is there any other met

[Freeswitch-users] Hangup detection

y query is that how to detect this sip response at our freeswitch side. And What if there is no response at all. fyi: I am using javascript for processing the calls. Thanks MSP ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitc

[Freeswitch-users] detecting unresponsive gateways in javascript

when i call outbound gateway, the gateway which i call is down, so there is no response, I am running the javascript for the routing outbound calls, so how can i detect that server is down after originate, so that i route can call to another gateway Thanks msp

Re: [Freeswitch-users] Fetching SIP Header Variable

y systems, thats all. Thanks you all for kind replys, msp ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/f

[Freeswitch-users] Fetching SIP Header Variable

Hi all, I have to fetch the variable from SIP header of incoming calls which comes from SIP proxy, The SIP proxy adds custom variable into sip header. When call comes at freeswitch, it is routed to Javascript, I need to fetch that sip variable in script. I tried mysipvar=session.getVariable("V

Re: [Freeswitch-users] Voice not work in bgapi

Hi all, It was the problem of softphone. FS is ok. My version of Zoiper was having problem with sound/voice while using bgapi. I changed it and its working fine. Thsanks msp ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http

[Freeswitch-users] Voice not work in bgapi

Hi all, I need to make calls using bgapi command, When I make call using bgapi, ring comes on my both softphones. After Accepting the ring, call is bridged but there is no voice on both ends of phones. Another problem is with Playback, Instead of dialing exten I put playback. When I run bgapi co