[asterisk-dev] SIP Session-timers and bug 15621

[Morten Isaksen]

Hi!

I think bug https://issues.asterisk.org/view.php?id=15621 has been
reintroduced in 1.6.1.9.

When I tested 1.6.1.9 it sends this:

SIP/2.0 200 OK.
Via: SIP/2.0/UDP 62.61.XX.XX;branch=z9hG4bK754f.78a98d95.1;received=62.61.XX.XX.
Via: SIP/2.0/UDP
10.253.253.2;rport=5060;received=62.61.XX.XX;branch=z9hG4bKac1440537976.
Record-Route: <sip:62.61.XX.XX;lr;ftag=1c1440530347;nat=yes>.
From: <sip:4347XXXX at pstn.uni-tel.dk>;tag=1c1440530347.
To: <sip:6910XXXX at sip.uni-tel.dk;user=phone>;tag=as084553a5.
Call-ID: 1440529481811200921545 at 10.253.253.2.
CSeq: 1 INVITE.
Server: Asterisk PBX 1.6.1.9.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Require: timer.
Session-Expires: -1;refresher=uas.
Contact: <sip:6910XXXX at 80.63.XX.XX>.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 39099150 39099150 IN IP4 80.63.XX.XX.
s=Asterisk PBX 1.6.1.9.
c=IN IP4 80.63.XX.XX.
t=0 0.
m=audio 12038 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

Which makes our Audiocodes choke. The fix was to add
Session-Timers=refuse in sip.conf


-- 
Morten Isaksen