search for: incomingcall

...t playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] play message to callee before connect to incomingcall Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) shou...

Good day all We have a voicetronix openline4 card.Asterisk is configured for sip with all the extensions and all&all. I can call out and internally,to dial out I have to dial 0... My problem is with incoming calls If I call my external PSTN number,asterisk answers with the default message and if I press the extension it goes to the right sip client. BUT As soon as I hangup this call all gets

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149 context=incomingcall disallow=all allow=gsm allow=alaw allow=ulaw in extensions.conf I have at the bottom of [incomingcal...

At time to time somebody is trying "their luck" and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those "beeps" (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a

...then decided to route it directly to voice mail to see if that would narrow the problem down, but it too doesn't make any sound. I can see asterisk display what it is saying when going through the voicemail sequence, it is just I cannot hear it on my mobile. here is my zapata.conf context=incomingcall ; from-1800 signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no group=1 channel=>1 context=incomingcall signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=>2-3 context=from-fa...

I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse? I appreciate any assistance. Phil

I'm having this problem where if the phone is ringing from IncomingCall #1, IC#2 will be immediately sent to VM. Is there somethign wrong with my dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer()...

...=1-4 zapata.conf [channels] ? busydetect=1 busycount=7 ? relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ? usecallerid=yes ? echocancel=yes echocancelwhenbridged=yes ? rxgain=0.0 txgain=0.0 ? group=1 pickupgroup=1-4 ? immediate=no ? context=incomingcall ? signalling=fxs_ks callerid=asreceived channel=1-4

...sers-bounces@lists.digium.com] On Behalf Of Min Hwan Chang Sent: Tuesday, August 09, 2005 6:57 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Incoming call #2 sent to VM immediately whenalready on phone with incoming. I'm having this problem where if the phone is ringing from IncomingCall #1, IC#2 will be immediately sent to VM. Is there somethign wrong with my dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer()...

...irm up if either of those statements is true. Any suggestions would be greatly appreciated! Thank You Paul M. Oster Here are the relevant portions of my sip.conf file... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = incomingcall ; Default for incoming calls tos=lowdelay disallow=all allow=alaw allow=gsm allow=ulaw [104] accountcode=vsllc type=friend context=employee username=104 secret=**redacted** host=dynamic qualify=yes reinvite=no canreinvite=no mailbox=104@internalextensions,750@internalextensions callgroup...

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hello everyone, Well here is my initial posting to the list, and I will admit Asterisk is new to me. I just got everything running here a couple days ago, so still learning the ropes for sure. OK, here is my problem. Currently I have it setup talking to a couple Cisco IP phones, and some Xten softphones, this works great. I also got an account with FreeWorld Dialup using IAX2 and that