Disable session timers wil probably fix this On Thursday, March 19, 2015, Денис <denis@ringme.ru> wrote:
<param name="disable-transfer" value="true"/> ?
On 17.03.2015 20:02, Patrick Shea wrote:
I am seeing 2 problems that I can't seem to find the answers to. I've scoured the Freeswitch Wiki (old), Confluence, and the users list.
I have 2 SIP calls bridge through Freeswitch - they are CONTROL channels, so no media is negotiated. A Leg is SIP over Websockets, B Leg is SIP.
SIPjs (WS) <---- A Leg ---> Freeswitch <---- B Leg ---> EP
The problem I have is in some cases I get a REFER from the EP to transfer the call somewhere else, and this results in a Re-INVITE from Freeswitch to the SIPjs client. (Ideally, I'd like to "skip" this re-Invite and just bridge the A-leg to the new AP).
Problem 1. Freeswitch sends media in the SDP of the Re-INVITE. This doesn't look right to me, but I assume using some local codecs specified somewhere. m=audio 28692 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 Can I disable these and just send the original m= line received for the initial call? m=application 28118 udp as
Problem 2. SIPjs doesn't support Re-INVITE, yet, so it responds with 488 Unacceptable Here. At this point, Freeswitch, drops the A leg and the transfer fails. I can't find a solution to work around this - any ideas?
Like I said, Ideally I'd like to skip the re-INVITE altogether. Currently debugging SOFIA to see how this can be done - is there a parameter already somewhere that I am overlooking?
Thanks Patrick
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