Marco Cordeiro
2009-Jun-02 14:20 UTC
[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?
Thanks Philipp, Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could I find info about it? Thanks again, Marco -----Mensagem original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Em nome de Philipp Kempgen Enviada em: ter?a-feira, 2 de junho de 2009 11:02 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk? Marco Cordeiro schrieb:> I have being trying to replicate the following call scenario with my > Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html > <http://www.tech-invite.com/Ti-sip-service-8.html> > > I have a situation that I have to notify the calling party that the callis> being forwarded to another number. So far, in the tests that I made,calling> from a SIP extension to another SIP extension with the forwardingactivated,> I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP > Response 181 CALL_IS_BEING_FORWARDED). > > The forwarding of the SIP extensions is being set with AstDB. > > My doubt is if, only a SIP Proxy would be able to trigger SIP Response181,> or if it would be possible with an Asterisk Server.IIRC Asterisk trunk can send and handle 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.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/20090602/92bb5fc0/attachment.htm
Philipp Kempgen
2009-Jun-02 15:56 UTC
[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?
Marco Cordeiro schrieb:> Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could > I find info about it?IIRC = "If I remember correctly". Asterisk trunk is the bleeding-edge development version of Asterisk. See "How source code is organized" at http://www.asterisk.org/developers/getting-started and "Get the source" at http://www.asterisk.org/developers/get-source> -----Mensagem original----- > [mailto:asterisk-users-bounces at lists.digium.com] Em nome de Philipp Kempgen> Marco Cordeiro schrieb: > >> I have being trying to replicate the following call scenario with my >> Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html >> <http://www.tech-invite.com/Ti-sip-service-8.html> >> >> I have a situation that I have to notify the calling party that the call > is >> being forwarded to another number. So far, in the tests that I made, > calling >> from a SIP extension to another SIP extension with the forwarding > activated, >> I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP >> Response 181 CALL_IS_BEING_FORWARDED). >> >> The forwarding of the SIP extensions is being set with AstDB. >> >> My doubt is if, only a SIP Proxy would be able to trigger SIP Response > 181, >> or if it would be possible with an Asterisk Server. > > IIRC Asterisk trunk can send and handle 181 Call is being forwarded.Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de --
Marco Cordeiro
2009-Jun-02 16:29 UTC
[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?
Hi Philipp, So, what you are saying is that SIP trunks between 2 Asteriks might be able to handle SIP Response 181 ? Marco -----Mensagem original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Em nome de Philipp Kempgen Enviada em: ter?a-feira, 2 de junho de 2009 13:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk? Philipp Kempgen schrieb:> Marco Cordeiro schrieb:>> I have a situation that I have to notify the calling party that the callis>> being forwarded to another number. So far, in the tests that I made,calling>> from a SIP extension to another SIP extension with the forwardingactivated,>> I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP >> Response 181 CALL_IS_BEING_FORWARDED). >> >> The forwarding of the SIP extensions is being set with AstDB. >> >> My doubt is if, only a SIP Proxy would be able to trigger SIP Response181,>> or if it would be possible with an Asterisk Server. > > IIRC Asterisk trunk can send and handle 181 Call is being forwarded.However as a rule of thumb you could probably say that SIP B2BUAs send "302 Moved temporarily" whereas SIP proxies send "181 Call is being forwarded". Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.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/20090602/784776fe/attachment.htm
Danny Nicholas
2009-Jun-04 13:26 UTC
[asterisk-users] Play a file while transfering a call
You could call an AGI to read a database based on the caller/extension and play any number of MOH files. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Philipp Kempgen Sent: Thursday, June 04, 2009 1:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play a file while transfering a call Julien Chavanton schrieb:> I need to be able to play different files when calling different numbersdynamicaly from the dial-plan, this solution will still require the creation of multiple static entry in "musiconhold.conf"> > For example, if it was possible to specify one file from the "Dial CMD" inthe dial plan, it would be easy to acheive. How about multiple music classes in musiconhold.conf and Set(CHANNEL(musicclass)=foobar) in the dialplan then? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Julien Chavanton
2009-Jun-26 16:05 UTC
[asterisk-users] Play a file while transfering a call
ok, but the classes in "musiconhold.conf" are static and require a reload to be modified. ________________________________ From: asterisk-users-bounces at lists.digium.com on behalf of Philipp Kempgen Sent: Thu 04/06/2009 7:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play a file while transfering a call Julien Chavanton schrieb:> I need to be able to play different files when calling different numbers dynamicaly from the dial-plan, this solution will still require the creation of multiple static entry in "musiconhold.conf" > > For example, if it was possible to specify one file from the "Dial CMD" in the dial plan, it would be easy to acheive.How about multiple music classes in musiconhold.conf and Set(CHANNEL(musicclass)=foobar) in the dialplan then? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de <http://www.amooma.de/> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com <http://the-asterisk-book.com/> - http://das-asterisk-buch.de <http://das-asterisk-buch.de/> Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de <http://www.amoocon.de/> -- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.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/20090626/cfe8d2e5/attachment.htm