Doing call transfer on VoIP line ...?

I am wondering how to do a call transfer (or 3 way conference) on my VoIP line for an incoming call to be transfered to an external phone number.

The transfer is normally initiated by issuing a “hook flash” then waiting for a dialtone and calling the external number and issuing another hook flash after answered (or hangup for call transfer)

I just can’t figure out how to do this with *. There is a Flash application but it seems to be for Zap channels only … How do i do this on a SIP (or IAX2) voip line ?

Tnx for any help


SIP [soft]phones normally have transfer button.
SIP adapters are recognizing the hook flash initiated by the attached ortodox telephone.

Tnx Andrew for your reply.

Yes i know about this … but i want my * to do the transfer.

i.e. * receive an inbound call from SIP VoIP “line”.
I wan’t to transfer this incomming call using the call xfer function of my VoIP line … so * need to send a hook flash, wait a little and the dial the number transfered to either hangup (xfer) or send hook flash again for 3-way conf.

Now how do i send a hook flash on a SIP channel.

I tried:

Flash() ; Only for Zaptel channels
Dial(fw … number) ; doesn’t work …
SendDTMF doesn’t seems to support hook flash …

So is there any way for * to issue a hook flash on a SIP channel ?


That’s impossible by the nature.

Regular analog line has no separation between the voice and signallig, so different tricks are used - hook flash (== drop the line for a short period of time), caller id presentation (sending modulated signal over the voice path), etc. This is normally referred as inband signaling.

In case of SIP there is no need for such tricks as soon as separate signalling path exists - SIP signalling is separated from the RTP voice path. Signalling is out-of-band.

If you need to transfer the call Asterisk will basically dial another number and bridge the call legs.