Transfer (SIP REFER) with 2 Asterisk Servers


#1

Incoming call to a Polycom phone comes from one asterisk server. Outgoing calls from Polycom phone go to a different asterisk server.

This creates a problem for the TRANSFER feature of the PHONE to work. The REFER SIP message is sent to the first asterisk server where the incoming call was and it has no information about the outbound leg created in the transfer.

I can use Asterisk to do the transfer but prefer to use the phone buttons then a key sequence.


See snippet trace:

REFER sip:asterisk@1.1.1.1 SIP/2.0
From: sip:1385@5.5.5.5;tag=691C3E69-E40425DA
To: “6885” sip:asterisk@1.1.1.1;tag=as68003bdf
CSeq: 2 REFER
Contact: sip:1385@5.5.5.5
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134
Refer-To: <sip:1381@2.2.2.2?Replaces=5fb3d711-ab2c1573-122b631c%4010.0.116.241%3Bto-tag%3Das33d2188b%3Bfrom-tag% 3DBF591C85-BED96BC6>

Referred-By: sip:1385@1.1.1.1


SIP/2.0 481 Call leg/transaction does not exist


#2

I haven’t tried it, but I believe there is a SIP option that causes Asterisk only to treat its own addresses as its own addresses in this context, and therefore generate a real INVITE/replaces in response to the REFER/replaces.

I would suspect it is not well exercised, and may well be buggy.