Thanks for your hint, but unfortunately this does not result in the behaviour I am looking for. When I start "MusicOnHold" Asterisk streams the OnHold music itself, even if I specifiy an invalid MoH class or one without files.
What I was looking for is a way to send a re-INVITE to its upstream SIP provider to inform it that the call should be placed on hold, exactly as described in the example in Section 2.1 of RFC 5359 (http://tools.ietf.org/html/rfc5359#section-2.1). Does anyone know how to do this from Asterisk dialplan? Any ideas are appreciated! :-) Greetings from a snowy Vienna, John :-) 2012/1/16 Johannes Zweng <[email protected]>: > Ok, I will try this and let you know! > > Kind regards, > John > > > > 2012/1/16 Sammy Govind <[email protected]>: >> Hey, >> I have never worried about looking at the SIP re-invites or anything when we >> engage MoH() application in asterisk. You can do a quick test on your test >> machine for this. >> >> Regards, >> Sammy >> >> On Mon, Jan 16, 2012 at 2:57 PM, Johannes Zweng <[email protected]> wrote: >>> >>> Hi! >>> >>> Many thanks for this hint. I will try this! :-) >>> >>> A quick question: when doing this with "MusicOnHold()": will the SIP >>> server be aware that the call is placed onHold (i.e. will Asterisk >>> send the mentioned re-INVITE)? >>> >>> The point is - if possible - we want the caller to hear the OnHold >>> Music from the SIP server. If not we would have to copy the MoH to our >>> Asterisk (and change it on our side too, when it changes at the >>> SIP-server). >>> >>> >>> Kind regards, >>> John >>> >>> >>> >>> 2012/1/16 Sammy Govind <[email protected]> >>> > >>> > Hi, >>> > >>> > yes, please see MusicOnHold() Application. You can call this app in your >>> > dialplan. This however will use the default music class and the >>> > corresponding music files placed in the asterisk server. If you don't want >>> > to stream music from Asterisk server side, try creating a new MusiconHold >>> > Class without any proper directory. That way Asterisk would only complain >>> > that there is no file to be streamed. >>> > >>> > Regards, >>> > Sammy >>> > >>> > On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng <[email protected]> >>> > wrote: >>> >> >>> >> Hi! >>> >> >>> >> Maybe I am missing something or am a little blind at the moment, but I >>> >> didn't find out how asterisk can place a call on hold when acting as user >>> >> agent client to another SIP server. >>> >> >>> >> Scenario: >>> >> ---------- >>> >> Asterisk registers to another SIP server (provider) as user agent. >>> >> An inbound call from this other SIP server comes in and arrives at >>> >> asterisk. >>> >> Asterisk performs some actions in the dialplan and should place the >>> >> call on hold after some time, so that the caller only hears the on hold >>> >> music from my provider (not streamed by my Asterisk). >>> >> >>> >> Technically speaking I want asterisk to send a re-INVITE >>> >> message containing an updated SDP body with the attribute "a=sendonly" or >>> >> "a=inactive" added so that the SIP server of my provider (where Asterisk >>> >> is >>> >> registered to as user) will recognize that the call should be placed on >>> >> hold. >>> >> >>> >> >>> >> A good example of what I want to achieve is presented in Section 2.1 of >>> >> RFC 5359 (Session Initiation Protocol Service Examples) >>> >> (http://tools.ietf.org/html/rfc5359#section-2.1) where "Bob" would be my >>> >> Asterisk (as UAC), "Alice" is the external caller and "Proxy" is the >>> >> provider's SIP server. >>> >> >>> >> >>> >> Question: >>> >> ---------- >>> >> Is there any way to perform this from the dialplan or by means of the >>> >> manager API? Is there an application like "Hold"? >>> >> >>> >> >>> >> Kind regards and greetings from Austria, >>> >> John :-) >>> >> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 -- _____________________________________________________________________ -- 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
