search for: outgoingcall

Hi all..!! I'm developing a system to use festival,asterisk, digium card. This system makes a auto-dialing (using digium TE410P card), when the customer answer the call: asterisk play an audio (pre-built from festival), that is a easy part. our problem is how we can detect a busy line or error in the line but outside of asterisk (by means of a process to detect status of call-line) then we

>I read your postings about faxdetection with CAPI. Interesting feature >but I'm not sure how to implement it on incoming calls (for example fax >dialed wrong number). Can you please show me a dialplan example how you >use this feature (incoming fax calls into Asterisk server). I did'nt test incoming fax, so this is how I would set my conf files : capi.conf : [contr1]

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call an other user and put the both in conference,I...

...use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial plan? Thanks!! [outbound-oversea] exten => _9011604.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;penang exten => _9011605.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;perak exten => _90114411.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten => _90114421.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten => _9011.,1,Macro(outgoingcall,${OUTBOUNDTRUNK})

Hi folks, Just wondering if someone have already done something like that : SIP Client_A ---1)call---> ASTERISK ---2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference, it's definitely easy, but call an other user and put the both in conference,I still don't...

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Hi, I've been using Asterisk for several months now with great success. I'm working on a system that tries to deliver a recorded message to a user, as follows: 1. a call file is placed in /var/lib/asterisk/outgoingcalls 2. This triggers a call to be placed 3. When answered, the caller hears the recorded message 4. After the message, the caller must confirm by pressing 1 5. If they confirm, the process ends 6. If they just hang up, wait 7. After waiting, dial them again and go to step 3 above The idea is that the...