George Pajari
2005-Nov-12 03:25 UTC
[Asterisk-Users] How to let caller continue after Dial cmd
We have a need to allow the caller who is in the middle of a call (i.e. who is already bridged between a PRI channel and a SIP channel as the result of entering a Dial cmd in the current context) to type something like "##" to cause the called party to be disconnected and to return from the Dial command with a distinctive STATUS so they can proceed to do other things within the context. Sort of a combination of the H and g flags to the Dial cmd. Any thoughts about whether or not this is possible in 1.0 or 1.2? If not, is there sufficient interest in such a feature for us to submit it once complete (I don't want to go through the effort to properly document, post, and maintain such a patch if its an improvement no one wants). -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
This might help you: ${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority after a blind transfer (use ^ characters in place of | to separate context/extension/priority when setting this variable from the dialplan) Check /usr/src/asterisk/doc/README.variables On 11/12/05, George Pajari <George.Pajari@netvoice.ca> wrote:> We have a need to allow the caller who is in the middle of a call (i.e. > who is already bridged between a PRI channel and a SIP channel as the > result of entering a Dial cmd in the current context) to type something > like "##" to cause the called party to be disconnected and to return > from the Dial command with a distinctive STATUS so they can proceed to > do other things within the context. Sort of a combination of the H and g > flags to the Dial cmd. > > Any thoughts about whether or not this is possible in 1.0 or 1.2? > > If not, is there sufficient interest in such a feature for us to submit > it once complete (I don't want to go through the effort to properly > document, post, and maintain such a patch if its an improvement no one > wants). > > -- > George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) > Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) > www.netvoice.ca www.ip-centrex.ca > www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca > > _______________________________________________ > --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 >