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webrtcasterisksipjssip

Blind Transfer in JSSIP


I have been trying to make a blind transfer on a ongoing call.

Below is the code i have implemented:

transfersession(ext) {
    this.rtcSession.refer('sip:' + ext + '@' + serveraddress);
}

Can someone tell is there something more I have to write?

The above code disconnects the ongoing call and adds the call to he queue.

What mistake am I doing? Thanks in advance.


Solution

  • Other side should support refer. However, that can be insecure, so disabled by most of providers.

    Anyway, you should learn how to use tcpdump/wireshark and check sip trace.