Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.
On Mon, Jul 7, 2008 at 6:58 AM, FaberK <f.faberk at gmail.com> wrote:> Hi folks, > we use meetme application with pin so when a customer joins he's > prompted for his name. > Then the voice say:"press one to accept the recording..." > My question is, is it possible to cut off that request to"press one"? > > Thanks to all > > -- > .:FaberK:. >I suppose you could edit the source code to timeout after five seconds or something. Thanks, Steve T
FaberK schrieb:> we use meetme application with pin so when a customer joins he's > prompted for his name. > Then the voice say:"press one to accept the recording..." > My question is, is it possible to cut off that request to"press one"?Audacity. Edit the sound file. Gr??e, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Hello! We would like to receive a SIP call and keep the caller waiting listening to some music other sound. A secondary intelligence decides whom to connect to and creates an outbound SIP call and when it is ringing there, or after the recipient answered the call, and maybe after listening to some small IVR joins the waiting caller, thus cancelling the music. Although the DIAL command offers many many options and we can put all the intelligence of whom to connect to whom there (or in scripts) we have the problem that the music always starts from the beginning when a new DIAL is started. This isnt an elegant solution. So the idea we got was to keep the caller in a meetme conference of 2 people. But how then can we force asterisk to dial out (most likely a secondary asterisk invocation with a rx command), make it go through some minimal context/dialplan upon answering, and eventually connect the called person to the meetme conference of the incoming call? Naturally, all this without any pin-codes or such. Did anybody have this problem already and maybe even found a solution for it? Thank you Regards Philipp
Hi! FaberK schrieb:> My question is, is it possible to cut off that request to"press one"? >I think you want to get rid of the number-pressing. The only option to omit this seems to be option E - select an empty pinless conference. http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Regards
have you tried app_queue? On 7/7/08, Philipp Ott <philipp.ott at avalon.at> wrote:> Hello! > > We would like to receive a SIP call and keep the caller waiting > listening to some music other sound. A secondary intelligence decides > whom to connect to and creates an outbound SIP call and when it is > ringing there, or after the recipient answered the call, and maybe after > listening to some small IVR joins the waiting caller, thus cancelling > the music. > > Although the DIAL command offers many many options and we can put all > the intelligence of whom to connect to whom there (or in scripts) we > have the problem that the music always starts from the beginning when a > new DIAL is started. This isnt an elegant solution. So the idea we got > was to keep the caller in a meetme conference of 2 people. But how then > can we force asterisk to dial out (most likely a secondary asterisk > invocation with a rx command), make it go through some minimal > context/dialplan upon answering, and eventually connect the called > person to the meetme conference of the incoming call? Naturally, all > this without any pin-codes or such. > > Did anybody have this problem already and maybe even found a solution > for it? > > Thank you > Regards > Philipp > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >