[Freeswitch-users] SIP 183 on a leg before bleg invite in bridge

[Barnaby Ritchley]

Hi Guys

I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg.

Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing.

The call goes:

Cisco UA (1.1.1.1)  -> FS (2.2.2.2) -> Gateway (3.3.3.3)

Sofia config has nothing special in it, relevent sections:

<param name="inbound-codec-negotiation" value="generous"/>
<param name="disable-transcoding" value="true"/>
<param name="manual-redirect" value="true"/>
<param name="disable-transfer" value="true"/>

and the dial plan is straight forward, just a bridge between the two endpoints.

Any ideas?




Here is the trace of the call:

INVITE sip:1234 at 2.2.2.2:5060 SIP/2.0.
Date: Fri, 20 Apr 2012 07:58:36 GMT.
Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500".
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
Allow-Events: presence, kpml.
Supported: 100rel,timer,resource-priority,replaces.
Supported: X-cisco-srtp-fallback.
Supported: Geolocation.
Min-SE:  1800.
Content-Length: 212.
To: <sip:1234 at 2.2.2.2>.
Expires: 180.
Content-Type: application/sdp.
Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
CSeq: 101 INVITE.
Session-Expires:  1800.
Max-Forwards: 69.
.
v=0.
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 31422 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
To: <sip:1234 at 2.2.2.2>.
CSeq: 101 INVITE.
User-Agent: FreeSWITCH.
Content-Length: 0.
.


U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
To: <sip:1234 at 2.2.2.2>;tag=vm45yDZjgmDKH.
Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
CSeq: 101 INVITE.
Contact: <sip:1234 at 2.2.2.2:5060;transport=udp>.
User-Agent: FreeSWITCH.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 247.
Remote-Party-ID: "1234" <sip:1234 at 2.2.2.2>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2.
s=FreeSWITCH.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 26820 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.


U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060
INVITE sip:05600660128 at 3.3.3.3:5060 SIP/2.0.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS.
Max-Forwards: 68.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
Contact: <sip:gw+C4DF4B2A088C91 at 2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>.
User-Agent: FreeSWITCH.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 201.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "00000000" <sip:00000000 at 2.2.2.2>;party=calling;screen=no;privacy=full.
.
v=0.
o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2.
s=FreeSWITCH.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 21466 RTP/AVP 8 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
User-Agent: SBC01.
Content-Length: 0.
.


U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>;tag=7NNjeSyyjtHSB.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
Contact: <sip:05600660128 at 3.3.3.3:5060;transport=udp>.
User-Agent: SBC01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Length: 0.