Switching between 2 sip clients during sip trafic

I am using 1 Asterisk instance and 3 Sip phones standard clients.
For clearance, Sip phone 1 is SIP1, sip phone 2 is SIP2, sip phone 3 is SIP3.
SIP1 and SIP3 are the same user but different machines.
My purpose is to call SIP2 from SIP1, then SIP3 will login to Asterisk with same username as SIP2, and then continue the call session with SIP3.
The specific scenario is as follow:

  • SIP1 calls SIP2
  • SIP2 rings
  • SIP3 login into asterisk ( “sip show peers” shows that SIP3 logged in and SIP2 is gone)
  • SIP3 rings
  • SIP3 answers
  • SIP1 and SIP3 are in call

As far as i tried, after SIP3 login, it doesn’t rings even if i saw that Asterisk replaced SIP2 by SIP3. (If i then call SIP3 from SIP1, it rings and call is ok)
Is it possible to configure Asterisk to support such a scenerio ?

SIP 2 is actually a server.

This can only be done by third party control, typically AMI. SIP 3 would need to know the callid and both from and to tags for the session to SIP 1 in order for it to do this as a first party.

By login, I assume you mean register. Registration isn’t really a login, it’s only purpose to say where to send calls outbound to the address of record. The dialplan is not informed of registrations.

You could probably do this by having SIP 3 call some dialplan indicating it want to take over the call, but the ring SIP 3 stage would be unnecessary, in that case as a call from SIP 3 would already be up.

Although it won’t help you here, if SIP 3 represents the same user, it should really just be an alternative SIP 1 and register as SIP 1. It won’t help you, because this will have no effect on active calls.

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