broadbandvoice@comcast.net
2006-Jul-22 13:00 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
Anyone know how to use dial plan to play messages as soon as a phone is picked up. Like when a user picks up a phone, get a message to contact administrator instead of a dial tone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060722/7d1aa012/attachment.htm
youll have to decide what context this goes in either [internal] or [incoming] but i hope you can figure this out yourself here is an idea [internal] exten => s,1,Answer() exten => s,n,Playback(pbx-invalid) exten => s,n,Hangup() so now when a user from [internal] picks up the phone the pbx-invalid.gsm is played from the asterisk sounds directory so now you are saying to yourself i want my own voice, to record check this out [internal] exten => s,1,Answer() exten => s,n,Playback(custom) exten => s,n,Hangup() (dont tell anyone this part, or just add it temporarily) exten => 999,1,Answer() exten => 999,n,Record(custom.gsm) exten => 999,n,Wait(1) exten => 999,n,Playback(custom) you will hear a beep after u dial 999, start recording then hangup or i think u can type # to stop and move to the next option>From: broadbandvoice@comcast.net >Reply-To: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users@lists.digium.com> >To: asterisk-users@lists.digium.com >Subject: [asterisk-users] Asterisk Dial Plan to Play Message >Date: Sat, 22 Jul 2006 20:00:37 +0000 >MIME-Version: 1.0 >Received: from lists.digium.com ([69.16.138.164]) by >bay0-mc12-f15.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, >22 Jul 2006 13:02:18 -0700 >Received: from digium-69-16-138-164.phx1.puregig.net (localhost >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 81F652FC9F5;Sat, 22 >Jul 2006 13:00:42 -0700 (MST) >Received: from psmtp.com (exprod8mx22.postini.com [64.18.3.122])by >lists.digium.com (Postfix) with SMTP id DD3DF2FC9BFfor ><asterisk-users@lists.digium.com>;Sat, 22 Jul 2006 13:00:37 -0700 (MST) >Received: from source ([216.148.227.153]) by >exprod8mx22.postini.com([64.18.7.10]) with SMTP; Sat, 22 Jul 2006 13:00:38 >PDT >Received: from rmailcenter18.comcast.net ([204.127.197.128])by comcast.net >(rwcrmhc13) with SMTPid <20060722200038m1300da2kae>; Sat, 22 Jul 2006 >20:00:38 +0000 >Received: from [208.17.34.25] by rmailcenter18.comcast.net;Sat, 22 Jul 2006 >20:00:37 +0000 >X-Message-Info: txF49lGdW41NVVYVTXpoQhqJpeo4y5ph0jUyL25Wnpk>X-Original-To: asterisk-users@lists.digium.com >Delivered-To: asterisk-users@lists.digium.com >X-Mailer: AT&T Message Center Version 1 (Apr 11 2006) >X-Authenticated-Sender: bnRhbmRvaEBjb21jYXN0Lm5ldA=>X-pstn-levels: (S:99.90000/99.90000 FC:95.5390 LC:95.5390 R:95.9108 >P:95.9108M:96.8350 C:98.4741 ) >X-pstn-settings: 3 (1.0000:1.0000) s fc lc gt3 gt2 gt1 r p m c >X-pstn-addresses: from <broadbandvoice@comcast.net> [db-null] X-BeenThere: >asterisk-users@lists.digium.com >X-Mailman-Version: 2.1.5 >Precedence: list >List-Id: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users.lists.digium.com> >List-Unsubscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=unsubscribe> >List-Archive: <http://lists.digium.com/pipermail/asterisk-users> >List-Post: <mailto:asterisk-users@lists.digium.com> >List-Help: <mailto:asterisk-users-request@lists.digium.com?subject=help> >List-Subscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=subscribe> >Errors-To: asterisk-users-bounces@lists.digium.com >Return-Path: asterisk-users-bounces@lists.digium.com >X-OriginalArrivalTime: 22 Jul 2006 20:02:19.0387 (UTC) >FILETIME=[BCC7A4B0:01C6ADC9] > >Anyone know how to use dial plan to play messages as soon as a phone is >picked up. Like when a user picks up a phone, get a message to contact >administrator instead of a dial tone?>_______________________________________________ >--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_________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
broadbandvoice@comcast.net
2006-Jul-22 18:24 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
Thanks, I'll try that in a few hours and share the experience. -------------- Original message -------------- From: "brandon kruz" <highendpcs@hotmail.com>> youll have to decide what context this goes in > either > [internal] > or [incoming] > but i hope you can figure this out yourself > here is an idea > > [internal] > exten => s,1,Answer() > exten => s,n,Playback(pbx-invalid) > exten => s,n,Hangup() > > so now when a user from [internal] picks up the phone the pbx-invalid.gsm is > played from > the asterisk sounds directory > so now you are saying to yourself i want my own voice, to record > check this out > > [internal] > exten => s,1,Answer() > exten => s,n,Playback(custom) > exten => s,n,Hangup() > > (dont tell anyone this part, or just add it temporarily) > > exten => 999,1,Answer() > exten => 999,n,Record(custom.gsm) > exten => 999,n,Wait(1) > exten => 999,n,Playback(custom) > > you will hear a beep after u dial 999, start recording > then hangup or i think u can type # to stop and move to the next option > > > >From: broadbandvoice@comcast.net > >Reply-To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >To: asterisk-users@lists.digium.com > >Subject: [asterisk-users] Asterisk Dial Plan to Play Message > >Date: Sat, 22 Jul 2006 20:00:37 +0000 > >MIME-Version: 1.0 > >Received: from lists.digium.com ([69.16.138.164]) by > >bay0-mc12-f15.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, > >22 Jul 2006 13:02:18 -0700 > >Received: from digium-69-16-138-164.phx1.puregig.net (localhost > >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 81F652FC9F5;Sat, 22 > >Jul 2006 13:00:42 -0700 (MST) > >Received: from psmtp.com (exprod8mx22.postini.com [64.18.3.122])by > >lists.digium.com (Postfix) with SMTP id DD3DF2FC9BFfor > >;Sat, 22 Jul 2006 13:00:37 -0700 (MST) > >Received: from source ([216.148.227.153]) by > >exprod8mx22.postini.com([64.18.7.10]) with SMTP; Sat, 22 Jul 2006 13:00:38 > >PDT > >Received: from rmailcenter18.comcast.net ([204.127.197.128])by comcast.net > >(rwcrmhc13) with SMTPid <20060722200038m1300da2kae>; Sat, 22 Jul 2006 > >20:00:38 +0000 > >Received: from [208.17.34.25] by rmailcenter18.comcast.net;Sat, 22 Jul 2006 > >20:00:37 +0000 > >X-Message-Info: txF49lGdW41NVVYVTXpoQhqJpeo4y5ph0jUyL25Wnpk= > >X-Original-To: asterisk-users@lists.digium.com > >Delivered-To: asterisk-users@lists.digium.com > >X-Mailer: AT&T Message Center Version 1 (Apr 11 2006) > >X-Authenticated-Sender: bnRhbmRvaEBjb21jYXN0Lm5ldA== > >X-pstn-levels: (S:99.90000/99.90000 FC:95.5390 LC:95.5390 R:95.9108 > >P:95.9108M:96.8350 C:98.4741 ) > >X-pstn-settings: 3 (1.0000:1.0000) s fc lc gt3 gt2 gt1 r p m c > >X-pstn-addresses: from [db-null] X-BeenThere: > >asterisk-users@lists.digium.com > >X-Mailman-Version: 2.1.5 > >Precedence: list > >List-Id: Asterisk Users Mailing List - Non-Commercial > >Discussion > >List-Unsubscribe: > >,> s-request@lists.digium.com?subject=unsubscribe> > >List-Archive: > >List-Post: > >List-Help: > >List-Subscribe: > >,> s-request@lists.digium.com?subject=subscribe> > >Errors-To: asterisk-users-bounces@lists.digium.com > >Return-Path: asterisk-users-bounces@lists.digium.com > >X-OriginalArrivalTime: 22 Jul 2006 20:02:19.0387 (UTC) > >FILETIME=[BCC7A4B0:01C6ADC9] > > > >Anyone know how to use dial plan to play messages as soon as a phone is > >picked up. Like when a user picks up a phone, get a message to contact > >administrator instead of a dial tone? > > > >_______________________________________________ > >--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 > > _________________________________________________________________ > Express yourself instantly with MSN Messenger! Download today - it's FREE! > http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060722/64bf3cda/attachment.htm
broadbandvoice@comcast.net
2006-Jul-22 18:35 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
I'm using SIP channel in Sip.conf and hand the calls over to a termination vendor. -------------- Original message -------------- From: Russell Bryant <russell@digium.com>> On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > [internal] > > exten => s,1,Answer() > > exten => s,n,Playback(custom) > > exten => s,n,Hangup() > > This, by itself, does not solve the problem where you want the message > to be played back when the phone is picked up without any user > intervention. If you're using zap phones, you can simply set this > option: > > immediate=yes > > Then, as soon as the phone goes off hook, the call will begin at the 's' > extension in the configured context instead of providing dialtone. > > -- > Russell Bryant > Software Developer > Digium, Inc. > > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060722/f2ea9cdf/attachment.htm
broadbandvoice@comcast.net
2006-Jul-23 04:30 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal. -------------- Original message -------------- From: "brandon kruz" <highendpcs@hotmail.com>> thank you russel > forgot to mention this. > > > >From: Russell Bryant > >Reply-To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >Subject: RE: [asterisk-users] Asterisk Dial Plan to Play Message > >Date: Sat, 22 Jul 2006 21:29:23 -0400 > >MIME-Version: 1.0 > >Received: from lists.digium.com ([69.16.138.164]) by > >bay0-mc9-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, > >22 Jul 2006 18:32:43 -0700 > >Received: from digium-69-16-138-164.phx1.puregig.net (localhost > >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 563422FC409;Sat, 22 > >Jul 2006 18:29:35 -0700 (MST) > >Received: from abita.digium.internal (gateway.digium.com [216.207.245.1])by > >lists.digium.com (Postfix) with ESMTP id 9CD5C2FC25Cfor > >;Sat, 22 Jul 2006 18:29:24 -0700 (MST) > >Received: from heineken.digium.com (heineken.digium.internal [10.16.1.2])by > >abita.digium.internal (Postfix) with ESMTP id C536AA94939for > >;Sat, 22 Jul 2006 20:29:25 -0500 (CDT) > >Received: from [172.17.99.18] ([172.17.99.18])by heineken.digium.com > >(8.13.6/69.69.69) with ESMTP id k6N1UIr9030086for > >; Sat, 22 Jul 2006 20:30:19 -0500 > >X-Message-Info: LsUYwwHHNt14xbUYi+9bCaWgpoxRQZbXIFwSWMVl+QA= > >X-Original-To: asterisk-users@lists.digium.com > >Delivered-To: asterisk-users@lists.digium.com > >References: > >Organization: Digium, Inc. > >X-Mailer: Evolution 2.6.1 X-BeenThere: asterisk-users@lists.digium.com > >X-Mailman-Version: 2.1.5 > >Precedence: list > >List-Id: Asterisk Users Mailing List - Non-Commercial > >Discussion > >List-Unsubscribe: > >,> s-request@lists.digium.com?subject=unsubscribe> > >List-Archive: > >List-Post: > >List-Help: > >List-Subscribe: > >,> s-request@lists.digium.com?subject=subscribe> > >Errors-To: asterisk-users-bounces@lists.digium.com > >Return-Path: asterisk-users-bounces@lists.digium.com > >X-OriginalArrivalTime: 23 Jul 2006 01:32:44.0557 (UTC) > >FILETIME=[E58083D0:01C6ADF7] > > > >On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > > [internal] > > > exten => s,1,Answer() > > > exten => s,n,Playback(custom) > > > exten => s,n,Hangup() > > > >This, by itself, does not solve the problem where you want the message > >to be played back when the phone is picked up without any user > >intervention. If you're using zap phones, you can simply set this > >option: > > > >immediate=yes > > > >Then, as soon as the phone goes off hook, the call will begin at the 's' > >extension in the configured context instead of providing dialtone. > > > >-- > >Russell Bryant > >Software Developer > >Digium, Inc. > > > >_______________________________________________ > >--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 > > _________________________________________________________________ > Don’t just search. Find. Check out the new MSN Search! > http://search.msn.click-url.com/go/onm00200636ave/direct/01/ > > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060723/94618472/attachment.htm
broadbandvoice@comcast.net
2006-Jul-23 06:37 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid) exten => 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xxxxxxxxxx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 ;incominglimit=1 ;mailbox=8407@default ;disallow=all ;allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723.1 I also tried changing type to peer instead of freind and does not work either. I am running Asterisk 1.2.3. Any help will be appreciated and thanks for all your inputs. -------------- Original message -------------- From: Steve Totaro <stotaro@asteriskhelpdesk.com>> If you are using phones attached to a ZAP FXS port the immediate=yes > will work. Otherwise, some SIP phones (Grandstream for instance) allows > you to enter an autodial number. It depends on what is providing the > dialtone to the handset. If your device does not support autodial, then > the next best thing is to do what has already been suggested. > > OR > > [somecontext] > exten=s,1,answer > exten=s,2,Authenticate(insertdigitshere) > exten=s,3,(continue with a real dialplan) > > Change the corresponding authenticate gsm file to say what you want > about contacting the boss. > > This gives the impression that phone is restricted for outbound calling > but if you enter the authenticate string, you can dialout for > emergencies or convenience. > > Thanks, > Steve > > broadbandvoice@comcast.net wrote: > > It did not work, how can I put in some user intervention so that any > > numbers they dial will send them to a message? Restrict their outbound > > calls and a get a message to contact administrator instead of a busy > > signal. > > > > -------------- Original message -------------- > > From: "brandon kruz" > > > > > thank you russel > > > forgot to mention this. > > > > > > > > > > >On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > > > > [internal] > > > > > exten => s,1,Answer() > > > > > exten => s,n,Playback(custom) > > > > > exten => s,n,Hangup() > > > > > > > >This, by itself, does not solve the problem where you want the > > message > > > >to be played back when the phone is picked up without any user > > > >intervention. If you're using zap phones, you can simply set this > > > >option: > > > > > > > >immediate=yes > > > > > > > >Then, as soon as the phone goes off hook, the call will begin > > at the 's' > > > >extension in the configured context instead of pro viding > > dialtone. > > > > > > > >-- > > > >Russell Bryant > > > >Software Developer > > > >Digium, Inc. > > > > > > > >_______________________________________________ > > > >--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 > > > > > > _________________________________________________________________ > > > Don’t just search. Find. Check out the new MSN Search! > > > http://search.msn.click-url.com/go/onm00200636ave/direct/01/ > > > > > > _______________________________________________ > > > --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 > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --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 > > > > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060723/6c2f24a2/attachment.htm
broadbandvoice@comcast.net
2006-Jul-23 07:15 UTC
[asterisk-users] Asterisk Dial Plan to Play Message
That was a typo its corrected to [8407] but problem still persist with original questions though. -------------- Original message -------------- From: "Eric "ManxPower" Wieling" <eric@fnords.org>> "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. > Do you really want them to be different? > > Dial() will stop processing of the dialplan until the call ends. Do you > really want this? > > "r" option to Dial will force a ringing sound to the caller, even if the > caller should be hearing a "all circuits are busy", or "your call cannot > be completed as dialed" or similar message. Do you really want that? > > broadbandvoice@comcast.net wrote: > > Thanks for the response, its looks logical, for some reason the authentication > is not working for me, I'm using a Linksys Phone adapter and here is a sample > dial plan in extensions.conf and also SIP channels. > > > > exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer > > exten => 8407,n,Authenticate(9461) > > exten => 8407,n,Playback(pbx-invalid) > > exten => 8407,n,Hangup() > > > > and in sip.conf > > > > [9507] > > type=friend > > user=8407 > > secret=xxxxxxxxxx > > ;context=from-sip > > callerid=8407 > > host=dynamic > > nat=yes > > qualify=yes > > canreinvite=no > > dtmfmode=rfc2833 > > -- > Now accepting new clients in Birmingham, Atlanta, Huntsville, > Chattanooga, and Montgomery. > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060723/fec0d64f/attachment.htm