[asterisk-users] Transfer

[Dan Cropp]

I am running Asterisk 13.5.0.

I have the Transfer working when using the chan_sip support.
However, when I try to perform a Transfer using pjsip, it is failing.

The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By.  PJSIP does not.
The switch provider I am working with has never seen a REFER without the "Referred-By" line

In both cases, I am performing the Transfer via AMI
EXEC Transfer ....

Does Asterisk 13.5.0 PJSIP support require a flag or something to force the Referred-By line to automatically be passed when a Transfer is performed?

chan_sip (succeeds)
19:27:32.512123 IP (tos 0x0, ttl 64, id 11492, offset 0, flags [none], proto UDP (17), length 630)
    192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 602
        REFER sip:3400 at 192.168.yyy.yyy:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK58f4bd1d
        Max-Forwards: 70
        From: <sip:3344 at 192.168.xxx.xxx>;tag=as44000cf4
        To: <sip:3400 at 192.168.yyy.yyy>;tag=7Iy0JkwDC
        Contact: <sip:3344 at 192.168.xxx.xxx:5060>
        Call-ID: jdEuqpAK-0002- at 192.168.yyy.yyy
        CSeq: 102 REFER
        User-Agent: Asterisk PBX 13.5.0
        Date: Thu, 20 Aug 2015 19:27:32 GMT
        Refer-To: <sip:370 at 192.168.yyy.yyy>
        Referred-By: <sip:3344 at 192.168.xxx.xxx:5060>
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Content-Length: 0

Pjsip
18:46:58.386372 IP (tos 0x0, ttl 64, id 38690, offset 0, flags [DF], proto UDP (17), length 654)
    192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 626
        REFER sip:3400 at 192.168.yyy.yyy:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjec41c3b9-d734-482d-82c1-2a6f8d9452a3
        From: <sip:3344 at 192.168.xxx.xxx>;tag=3c10f423-e468-42ea-87a1-658ae106581c
        To: <sip:3400 at 192.168.yyy.yyy>;tag=WITKDakt
        Contact: <sip:192.168.xxx.xxx:5060>
        Call-ID: s6Wk6l6Q-0001- at 192.168.yyy.yyy
        CSeq: 981 REFER
        Event: refer
        Expires: 600
        Supported: 100rel, timer, replaces, norefersub
        Accept: message/sipfrag;version=2.0
        Allow-Events: message-summary, presence, dialog, refer
        Refer-To: <sip:370 at 192.168.yyy.yyy>
        Max-Forwards: 70
        User-Agent: Asterisk PBX 13.5.0
        Content-Length:  0
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