Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Transfer


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
dan at amtelco.com
Guest





PostPosted: Thu Aug 20, 2015 2:44 pm    Post subject: [asterisk-users] Transfer Reply with quote

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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services