i've seen that opening a socket on the asterisk server i can originate a call from one extension to another in a specific context. Is it possible to transfer an existing call from the extension ... SIP/xxx to another extension in a specific context? thanks
nik600 wrote:> Is it possible to transfer an existing call from the extension ... > SIP/xxx to another extension in a specific context?you can do this with the redirect action: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect =Stefan -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 271 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060322/8ee77d2d/signature.pgp
ok thanks, it works! but...how can i know the channel used in a specific period from the called? for examle...i know SIP/200 but how can i know that the channel is SIP/200sfhj3e ? thanks
It is tricky. Is this for an outbound call? I missed the original post. If someone can re-post it, I might be able to help as I just finish a commercial addon using all manager api calls into *. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of nik600 Sent: Thursday, March 23, 2006 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] transfer calls via Manager Api ok thanks, it works! but...how can i know the channel used in a specific period from the called? for examle...i know SIP/200 but how can i know that the channel is SIP/200sfhj3e ? thanks _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
nik600 wrote:> Is it possible to transfer an existing call from the extension ... > SIP/xxx to another extension in a specific context?you can do this with the redirect action: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect =Stefan ok thanks, it works! but...how can i know the channel used in a specific period from the called? for examle...i know SIP/200 but how can i know that the channel is SIP/200sfhj3e ?
nik600 wrote:> nik600 wrote: > but...how can i know the channel used in a specific period from the called? > > for examle...i know SIP/200 but how can i know that the channel is > SIP/200sfhj3e ?either by following the events (NewChannel, Rename, ...) or by issuing a StatusAction that will return a list of all active channels. =Stefan -- reuter network consulting Neusser Str. 110 50760 K?ln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: srt@reucon.net Jabber: srt@jabber.reucon.net -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 258 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060324/26073ec8/signature.pgp