search for: danishsamad

Displaying 7 results from an estimated 7 matches for "danishsamad".

2006 Jan 23
5
dial out and message playback
Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of
2006 Jun 08
6
how to identify agi crash cause
Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and
2005 Dec 11
0
[Asterisk-Dev] C++ AGI debuggin
Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
2006 Jan 16
0
Set(LANGUAGE()=language throwing warnings
Hi, I tried setting a particular language through the agi framework using the following command "EXEC Set(LANGUAGE()=%s)\n", language) // language contains EN or en On the asterisk command line I get the following warning: Jan 16 22:55:23 WARNING[4212]: res_agi.c:1085 handle_exec: Could not find application (Set(LANGUAGE()=EN)) strangely all IVR's are picked from the
2007 Jun 21
0
retreiving callid of call from the dial application
Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I
2007 May 02
2
delay in switching between contexts
Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)
2007 May 19
1
asterisk not sending ACK after reinvite
Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.