[Asterisk-Dev] Newbie - Transfer Problem

[ARNOLD S LIPTON]

Hello

   I have asterisk connected to a cisco gateway and a broadsoft softswitch. My SIP phones are custom software on a PC. I make a call via the gateway into asterisk and a 2-way talk path is setup from the
gateway to the sip PC. Next the PC (PC1) uses a SIP refer to transfer the call to the other PC (PC2)gateway hears ringing and then when PC2 answers, only a 1-way talk path remains between PC2 and the gateway. The following behavior is noted in asterisk starting from the call transfer:

1) Dial the number from PC2 and click mute transfer (initiates the transfer)

2) An invite is reeived at asterisk for Call Id (1) which causes 1 side of audio rtp stream to move to medea server in order to hear ringing.

3) Asterisk in turn sends an invite to Call Id (2) in order to move other side of the rtp stream to media server. PC2 is hearing ringing.

4) PC2 answers and an invite is sent to asterisk with content-len = 0 (no rtp stream specified).

5) Asterisk receives an ack to above invite and the rtp steam port is specified. A 1-way talk path is now setup.

No other messages come in or out untill hangup occurs. Seems like after the ack asterisk should move the other side in order to setup the other side of the talk path.

Is there a bug in asterisk or is there a different way the transfer should be done.

Thanks In advance For any help
Arnie
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