[asterisk-bugs] [JIRA] (ASTERISK-25666) Path header is ignored
Peter Baines (JIRA)
noreply at issues.asterisk.org
Thu Jan 7 10:20:33 CST 2016
Peter Baines created ASTERISK-25666:
---------------------------------------
Summary: Path header is ignored
Key: ASTERISK-25666
URL: https://issues.asterisk.org/jira/browse/ASTERISK-25666
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: pjproject/pjsip
Affects Versions: 13.6.0
Environment: Debian Jesse
Reporter: Peter Baines
I am adding a Path header before forwarding a REGISTER onto asterisk. The problem is when asterisk recieves an INVITE it does not use the value set in the Path header, instead it sends it directly to the device.
I can replicate this on asterisk 13.6.0 and 11.13.1, however in 1.8.32.3 it works as expected (i.e. the INVITE is sent to the value set in the Path header).
To replicate:
In sip.conf I have uncommented:
supportpath=yes
rtsavepath=yes
In users.conf I have got:
[6000]
secret =
host=dynamic
context = default
[6001]
secret =
host=dynamic
context = default
[6002]
secret =
host=dynamic
context = default
In extensions.conf I have made default look like:
[default]
;include => demo
exten => 6000,1,Dial(SIP/6000,18)
exten => 6000,n,Hangup()
exten => 6002,1,Dial(SIP/6002,18)
exten => 6002,n,Hangup()
exten => 6001,1,Dial(SIP/6001,18)
exten => 6001,n,Hangup()
Below is the 6000 user REGISTER going from opensips (10.15.20.137:5060) into asterisk (192.168.68.68:5070) with the Path header.
U 2016/01/06 10:04:23.399170 10.15.20.137:5060 -> 192.168.68.68:5070
REGISTER sip:10.15.20.137 SIP/2.0.
Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bKcc2c.b40fb511.0.
Via: SIP/2.0/UDP 10.15.20.53:52666;received=10.15.20.53;branch=z9hG4bK-d8754z-91422161f08a7943-1---d8754z-;rport=52666.
Max-Forwards: 69.
Contact: <sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786>.
To: <sip:6000 at 10.15.20.137>.
From: <sip:6000 at 10.15.20.137>;tag=9e95da50.
Call-ID: OTQ1ZTdmZmE3OTM1ZWVkYzMzYWZiMDMzMDgyODhmOTU.
CSeq: 2 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
User-Agent: Bria 3 release 3.5.5 stamp 71243.
Content-Length: 0.
Path: <sip:10.15.20.137;lr>.
Below is the INVITE going from opensips to asterisk for 6000
U 2016/01/06 10:11:13.668929 10.15.20.137:5060 -> 192.168.68.68:5070
INVITE sip:6000 at 10.15.20.137;transport=UDP SIP/2.0.
Record-Route: <sip:10.15.20.137;lr;nat=yes>.
Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bK8f77.9d6ef7e7.0.
Via: SIP/2.0/UDP 188.39.51.2:35631;rport=35631;received=10.15.20.53;branch=z9hG4bK-d8754z-d46f3a0333dc5d49-1---d8754z-.
Max-Forwards: 69.
Contact: <sip:6001 at 10.15.20.53:35631;transport=UDP>.
To: <sip:6000 at 10.15.20.137;transport=UDP>.
From: <sip:6001 at 10.15.20.137;transport=UDP>;tag=870fdf72.
Call-ID: YWVjN2VjMDZmYmZmNjg4MTE2MzJlZGU1ZDNjZGU2NDc..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
User-Agent: Z 3.3.21933 r21903.
Allow-Events: presence, kpml.
Content-Length: 237.
.
v=0.
o=Z 0 0 IN IP4 188.39.51.2.
s=Z.
c=IN IP4 188.39.51.2.
t=0 0.
m=audio 8000 RTP/AVP 3 110 8 0 98 101.
a=rtpmap:110 speex/8000.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
I would now expect asterisk to send the INVITE to the value of the Path header in the registration (10.15.20.137:5060) however it is sending the INVITE directly to the device (10.15.20.53:52666):
U 2016/01/06 10:11:13.671345 192.168.68.68:5070 -> 10.15.20.53:52666
INVITE sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786 SIP/2.0.
Via: SIP/2.0/UDP 192.168.68.68:5070;branch=z9hG4bK308a4ef5;rport.
Max-Forwards: 70.
Route: <sip:10.15.20.137;lr>.
From: "New User" <sip:6001 at 192.168.68.68:5070>;tag=as3daea415.
To: <sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786>.
Contact: <sip:6001 at 192.168.68.68:5070>.
Call-ID: 55202bc71f9e684d0b82c7cb2e8684ab at 192.168.68.68:5070.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 13.6.0.
Date: Wed, 06 Jan 2016 10:11:13 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer, path.
Content-Type: application/sdp.
Content-Length: 286.
.
v=0.
o=root 887525354 887525354 IN IP4 192.168.68.68.
s=Asterisk PBX 13.6.0.
c=IN IP4 192.168.68.68.
t=0 0.
m=audio 12356 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=maxptime:150.
a=sendrecv.
Let me know if you require any further information / traces.
Regards,
Peter
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