PJ Santos
2005-Sep-13 10:47 UTC
[Asterisk-Users] How to create IVR menu and transfer to another sip extensions.
Hi All, I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos --------------------------------- Yahoo! Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/cce6e847/attachment.htm
Moises Silva
2005-Sep-14 08:53 UTC
[Asterisk-Users] How to create IVR menu and transfer to another sip extensions.
mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos <pjsantoss@yahoo.com.br> wrote:> > Hi All, > I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , > press 1 to Sales, press 2 to Help Desk or wait to operator. > What function should I use for call transfer exten SIP to exten SIP. eg I > call to extension 190 and after answer, I do one transfer to another exten > SIP. > Regards. > Paulo Santos > > ------------------------------ > Yahoo! Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA CONVERSA. > Participe!<http://us.rd.yahoo.com/mail/br/taglines/*http://br.yahoo.com/messenger/promocao/> > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050914/0b49bb36/attachment.htm
PJ Santos
2005-Sep-14 09:15 UTC
[Asterisk-Users] How to create IVR menu and transfer to another sip extensions.
I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJ Moises Silva <moises.silva@gmail.com> escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos <pjsantoss@yahoo.com.br> wrote: Hi All, I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos --------------------------------- Yahoo! Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org " _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Yahoo! Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050914/294f48a2/attachment.htm
Anthony Rodgers
2005-Sep-14 09:28 UTC
[Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.
This is a sample that I built as part of our * pilot here - it demonstrates the various things you can do with an auto-attendant type of system. Is this the kind of thing you are looking for? [info-line] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,Background(demo-enterkeywords) exten => 1,1,Goto(library-info,s,1) exten => 2,1,Goto(lawn-sprinkling-info,s,1) exten => 3,1,Goto(closed-trails-info,s,1) exten => 4,1,Voicemail(u2348@default) [library-info] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,Background(demo-enterkeywords) exten => 1,1,Voicemail(u2471@default) exten => 2,1,Goto(internal,96045551212,1) exten => 3,1,Playback(demo-congrats) exten => *,1,Goto(library-info,s,5) [lawn-sprinkling-info] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,Background(demo-enterkeywords) exten => 1,1,Goto(internal,2348,1) exten => 2,1,Goto(closed-trails-info,s,1) exten => 3,1,Voicemail(u2348@default) exten => *,1,Goto(lawn-sprinkling-info,s,5) [closed-trails-info] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,Playback(demo-congrats) exten => s,6,Goto(info-line,s,5) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Sep 14, 2005, at 9:15 AM, PJ Santos wrote:> I need create one configuration to provide one Interactive Voice > Response. > ? > I read any docs about this. > ? > So, if you have one sample, please post. > ? > Thanks. > ? > Paulo Santos. > Brasil-RJ > > Moises Silva <moises.silva@gmail.com> escreveu: >> mmm actually i think that is a functionality most VoIP phones >> provide, you dont need do anything, just press transfer in your VoIP >> phone and the dial the extension you want to transfer to. >> >> On 9/13/05, PJ Santos <pjsantoss@yahoo.com.br> wrote: Hi All, >>> ? >>> I need help to create one IVR Menu, when a say "Welcome to PBX >>> Corp..." , press 1 to Sales, press 2 to Help Desk or wait to >>> operator. >>> ? >>> What function should I use for call transfer exten SIP to exten SIP. >>> eg I call to extension 190 and after answer, I do one transfer to >>> another exten SIP. >>> ? >>> Regards. >>> ? >>> Paulo Santos >>> >>> ? >>> ? >>> ? >>> >>> Yahoo! Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA >>> CONVERSA. Participe! >>> >>> >>> _______________________________________________ >>> --Bandwidth and Colocation sponsored by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> ? http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> "Su nombre es GNU/Linux, no solamente Linux, mas info en >> http://www.gnu.org " _______________________________________________ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! >> Messenger com voz: PROMO??O VOC? PODE LEVAR UMA VIAGEM NA CONVERSA. >> Participe!_______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider, all in g729 will require to purchase codecs from Digium? also, in this scenario the transcoding is almost non-existent right? I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can sustain more than 100 calls or up to a 100? I just looking at hardware capacity, since the machine will be located at an ISP with more than needed bandwith. There is no need for voicemail, web interfaces or anything else, since the * box will only function as a gateway to a US-based VOIP provider. The machine in question runs Centos4 Linux (Redhat enterprise 4) and CDR logging only. Thanks,
Erick Perez
2005-Sep-16 13:01 UTC
[Asterisk-Users] Re: g729 to asterisk to g729 voip provider
anyone with some info on this? thanks again. On 9/14/05, Erick Perez <eaperezh@gmail.com> wrote:> Using sipura sip/g729 to connect to an asterisk server that will > server as a gateway to a VOIP provider, all in g729 will require to > purchase codecs from Digium? > > also, in this scenario the transcoding is almost non-existent right? > I have read many documents about the type of codecs, and g729 seems to > be a good trade between almost-toll quality and low bandwith usage > right? > > > A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can > sustain more than 100 calls or up to a 100? > I just looking at hardware capacity, since the machine will be located > at an ISP with more than needed bandwith. > > There is no need for voicemail, web interfaces or anything else, since > the * box will only function as a gateway to a US-based VOIP provider. > > The machine in question runs Centos4 Linux (Redhat enterprise 4) and > CDR logging only. > > Thanks, >-- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama