I've configured our PBX so that when a user dials 80 on the PBX extension, it goes out an ISDN TE interface on the PBX and into an NT interface on my asterisk machine, where it jumps into the 's' extension. Asterisk then does a DISA(no-password|sip_provider_out) which allows the call to go out via a sip provider, to give us cheaper calls. Unfortunately if the user doesn't wait for DISA to give dialtone, asterisk doesn't hear all of the digits. When they dial '0' on the PBX there is no need to wait for dialtone, so it is a bit confusing for the users. Any suggestions? I'm using misdn, and 'immediate' and 'always_immediate' in the config for that port, so maybe there is something I could do there to take whatever digits have been dialled so far... Thanks James
Daniel Cyt
2006-Oct-04 02:00 UTC
[asterisk-users] IVR for the called part (IVR inside out)
Hello, I'm trying to get it to work but I can't find the right way. I would be glad if the list could point me the right directions. What I want: My Asterisk dialing out to a number (my mobile phone) and playing an IVR to the called part saying "press one to accept this call". If the called part (my mobile) press 1 the call goes thru, otherwise it goes straight to asterisk voicemail. Reason (my scenario): I'm going to setup a follow me from my extension to my mobile phone and I don't want people to find out they are actually rining on my mobile. I don't have the option to disable voicemail feature on the mobile company. The problem happens if I don't pick the call or I'm, for instance inside a tunnel, where my mobile lose signal. Asterisk will think my mobile voicemail is somebody answering, and whoever called me will her the mobile voicemail. I've been searching for a while before emaiil the list but I could not find anything like it. Thank you very much _________________________________________________________________ MSN Messenger: instale grĂ¡tis e converse com seus amigos. http://messenger.msn.com.br
I've used the prompt pls-wait-connect-call to give my users a cue to cool their heels for a second or two in circumstances like this, and no one has complained. That's probably the most useful prompt in Asterisk! -----Original Message----- From: James Harper [mailto:james.harper@bendigoit.com.au] Sent: Wednesday, October 04, 2006 1:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DISA and legacy PBX I've configured our PBX so that when a user dials 80 on the PBX extension, it goes out an ISDN TE interface on the PBX and into an NT interface on my asterisk machine, where it jumps into the 's' extension. Asterisk then does a DISA(no-password|sip_provider_out) which allows the call to go out via a sip provider, to give us cheaper calls. Unfortunately if the user doesn't wait for DISA to give dialtone, asterisk doesn't hear all of the digits. When they dial '0' on the PBX there is no need to wait for dialtone, so it is a bit confusing for the users. Any suggestions? I'm using misdn, and 'immediate' and 'always_immediate' in the config for that port, so maybe there is something I could do there to take whatever digits have been dialled so far... Thanks James _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users