Displaying 7 results from an estimated 7 matches for "outgoingcalls".
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outgoingcall3
2004 Dec 03
0
Digium+asterisk+festival+outgoingcall: How detect a busy line..?
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
2007 Sep 17
0
[SOLVED] fax machine detection for outgoingcall on DIVAcard
>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]
2003 Nov 25
2
Outgoing-call and enter user in Conference - repost
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
2006 Feb 12
1
help on dial plan
The following is my dialplan for outgoing international call. What I want are:
- when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx,
90114421xxxxxxx, 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
2003 Nov 21
1
Outgoing-call and enter user in Conference
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
2006 Mar 23
1
wellgate 38XX FX & FXS voip gateways with outgoing call files
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
2006 May 02
0
Commands possible in the h extension, message delivery with confirmation
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...