dan at amtelco.com
Guest
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 Posted: Thu Aug 20, 2015 2:44 pm Post subject: [asterisk-users] Transfer |
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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@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@192.168.xxx.xxx>;tag=as44000cf4
To: <sip:3400@192.168.yyy.yyy>;tag=7Iy0JkwDC
Contact: <sip:3344@192.168.xxx.xxx:5060>
Call-ID: jdEuqpAK-0002-@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@192.168.yyy.yyy>
Referred-By: <sip:3344@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@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@192.168.xxx.xxx>;tag=3c10f423-e468-42ea-87a1-658ae106581c
To: <sip:3400@192.168.yyy.yyy>;tag=WITKDakt
Contact: <sip:192.168.xxx.xxx:5060>
Call-ID: s6Wk6l6Q-0001-@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@192.168.yyy.yyy>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.5.0
Content-Length: 0 |
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