mahesh katta
2012-Jan-16 15:55 UTC
[asterisk-users] How Can I configure the between call oneside IVR
Hi list, how we can configure between call add the IVR. My scenarios is "A" get the incomming call from "C".In between them I need to one side IVR play for "C", "C" enter the some DTMF inputs and "A" should be on hold. after finish "C" input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call...., and how my asterisk will detect the DTMF input.... Best Regards, Mahesh Katta -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120116/c40cc0ea/attachment.htm>
Danny Nicholas
2012-Jan-16 16:02 UTC
[asterisk-users] How Can I configure the between call oneside IVR
A should transfer C to a local channel that plays the IVR then returns the call to A. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mahesh katta Sent: Monday, January 16, 2012 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How Can I configure the between call oneside IVR Hi list, how we can configure between call add the IVR. My scenarios is "A" get the incomming call from "C".In between them I need to one side IVR play for "C", "C" enter the some DTMF inputs and "A" should be on hold. after finish "C" input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call...., and how my asterisk will detect the DTMF input.... Best Regards, Mahesh Katta -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120116/00cf35aa/attachment.htm>
mahesh katta
2012-Jan-16 16:21 UTC
[asterisk-users] How Can I configure the between call oneside IVR
I was tried it but its not going.. with same Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas <danny at debsinc.com> wrote:> A should transfer C to a local channel that plays the IVR then returns the > call to A.**** > > ** ** > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *mahesh katta > *Sent:* Monday, January 16, 2012 9:56 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] How Can I configure the between call oneside > IVR**** > > ** ** > > Hi list, > > how we can configure between call add the IVR. > My scenarios is > "A" get the incomming call from "C".In between them I need to one side IVR > play for "C", "C" enter the some DTMF inputs and "A" should be on hold. > after finish "C" input will complete again they want talk each other .This > is the scenario. > > Can anybody help to me how can I add this IVR in between those call...., > and how my asterisk will detect the DTMF input.... > > > Best Regards, > > Mahesh Katta**** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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/20120116/46dec8de/attachment.htm>
Danny Nicholas
2012-Jan-16 16:27 UTC
[asterisk-users] How Can I configure the between call oneside IVR
I would do it something like this [ivrandreturn] Exten => s,1,playback(message) Exten => s,n,waitexten(5) Exten => 1,1,noop(stuff for press 1) Exten => 1,n,dial(SIP/A) Exten => 2,1,noop(stuff for press 2) Exten => 2,n,dial(SIP/A) In real life SIP/A would be something like SIP/${ARG1} where ARG1 is the extension for A. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mahesh katta Sent: Monday, January 16, 2012 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How Can I configure the between call oneside IVR I was tried it but its not going.. with same Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas <danny at debsinc.com> wrote: A should transfer C to a local channel that plays the IVR then returns the call to A. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mahesh katta Sent: Monday, January 16, 2012 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How Can I configure the between call oneside IVR Hi list, how we can configure between call add the IVR. My scenarios is "A" get the incomming call from "C".In between them I need to one side IVR play for "C", "C" enter the some DTMF inputs and "A" should be on hold. after finish "C" input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call...., and how my asterisk will detect the DTMF input.... Best Regards, Mahesh Katta -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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/20120116/652f312d/attachment.htm>
mahesh katta
2012-Jan-16 16:45 UTC
[asterisk-users] How Can I configure the between call oneside IVR
Best Regards, ahesh Katta On Mon, Jan 16, 2012 at 9:57 PM, Danny Nicholas <danny at debsinc.com> wrote:> I would do it something like this**** > > [ivrandreturn]**** > > Exten => s,1,playback(message)**** > > Exten => s,n,waitexten(5)**** > > Exten => 1,1,noop(stuff for press 1)**** > > Exten => 1,n,dial(SIP/A)**** > > Exten => 2,1,noop(stuff for press 2)**** > > Exten => 2,n,dial(SIP/A)**** > > ** ** > > In real life SIP/A would be something like SIP/${ARG1} where ARG1 is the > extension for A. **** > > ** >In this scenario "A" does not HOLD, its Disconnect, I need it should be hold. it should be in conference.> ** > > ** ** > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *mahesh katta > *Sent:* Monday, January 16, 2012 10:21 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] How Can I configure the between call > oneside IVR**** > > ** ** > > I was tried it but its not going.. with same > Best Regards, > > Mahesh Katta**** > > On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas <danny at debsinc.com> wrote: > **** > > A should transfer C to a local channel that plays the IVR then returns the > call to A.**** > > **** > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *mahesh katta > *Sent:* Monday, January 16, 2012 9:56 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] How Can I configure the between call oneside > IVR**** > > **** > > Hi list, > > how we can configure between call add the IVR. > My scenarios is > "A" get the incomming call from "C".In between them I need to one side IVR > play for "C", "C" enter the some DTMF inputs and "A" should be on hold. > after finish "C" input will complete again they want talk each other .This > is the scenario. > > Can anybody help to me how can I add this IVR in between those call...., > and how my asterisk will detect the DTMF input.... > > > Best Regards, > > Mahesh Katta**** > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users**** > > ** ** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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/20120116/602e4e81/attachment.htm>