Hi Guys,
I’m curious if its possible at all to transfer a call externally, from an extension in a way it will free up channels, like blind transfer but even blind transfer “shwo channels” shows the bridged channels.
I haven’t tried it, but transferring the calling party to the Transfer application may work, as long as the target technology is the same as the source technology and supports transfers (e.g. both are SIP).
the problem is i want to transfer it without answering the call, so probably at the zap (isdn) level.
Transfer works before answer. In fact, SIP transfer works best before the call.
However, in the other thread, you were talking about analogue zaptel connections. Pre-answer redirects (redirect rather than transfer) must be programmed in the upstream exchange, as there is nothing in standard loop/DTMF signalling protocols to allow a redirect prior to answer.
I’m confused, though, in that your other thread on this is asking about whether both channels are released. Before answer there can only be one channel, and even after answer, there may be only one channel, if you explicitly answer or use some PABX service without bridging the call.
thank you for your paitence with my query …
i’m probably asking something fairly difficult here, basically im trying to find a method to transfer a ringing call to an external number without answering it, so it remains ringing so:
during ringing:
callerA ----> ringing --> ZAP->asterisk —> redirect —> still ringing --> external sourceA
once answer by external sourceA the call will be:
callerA—>external sourceA
so asterisk pretty much redirects the call without remaining in the loop.
Zaptel covers a large number of different signalling protocols. For analogue presented lines, this is not possible, pre-answer, because the network will not provide any such capability.
dahdi, which you should use for new systems, doesn’t have a dahdi_transfer function, so I don’t think that redirects are possible for other signalling protocols, pre-answer.
Post answer, you may be able to hack it, as others have described, but only for a subscriber type line, not for a normal PABX type line.
Note, the network will generally charge you as though the call had been relayed through asterisk, although you would save on line rental.
thank you for the reply, (on both threads) it’s an ISDN line, i was looking into ss7 but wanted to know other alternatives.
just as ive noticed asterisk responds with a hangup reason , i was hoping it cuold respond with a function that requests call to be transfered… then how do companies do that collect call stuff?