Hi,

After using , <action application="bridge" data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>

the 183 without udp is not blocked/ignored .

Below are the traces to visualize:
192.168.2.98 is provider
192.168.2.16 is FS


U 192.168.2.98:5060 -> 192.168.2.16:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
To: <sip:599261244747199@192.168.2.16>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101@192.168.2.16>;tag=51SjQQQUX14QF.
Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
CSeq: 7886492 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:599261244747199@192.168.2.98:5060>.
Call-Info: <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
Content-Length: 0.
.


U 192.168.2.16:5060 -> 192.168.2.6:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
Via: SIP/2.0/UDP 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
Record-Route: <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
From: "0280910101" <sip:0280910101@192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199@192.168.2.6>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199@192.168.2.16:5060;transport=udp>.
User-Agent:  SBC.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Length: 0.
Remote-Party-ID: "599261244747199" <sip:599261244747199@192.168.2.16>;party=calling;privacy=off;screen=no.
.


U 192.168.2.98:5060 -> 192.168.2.16:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
To: <sip:599261244747199@192.168.2.98>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101@192.168.2.16>;tag=51SjQQQUX14QF.
Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
CSeq: 7886492 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:599261244747199@192.168.2.98:5060>.
Call-Info: <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
Content-Type: application/sdp.
Content-Length: 209.
.
v=0.
o=vsnl2 770 13521 IN IP4 192.168.2.98.
s=sip call.
c=IN IP4 115.113.121.99.
t=0 0.
m=audio 49034 RTP/AVP 18 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=ptime:20.
a=rtpmap:18 G729/8000/1.


U 192.168.2.16:5060 -> 192.168.2.6:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
Via: SIP/2.0/UDP 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
Record-Route: <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
From: "0280910101" <sip:0280910101@192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199@192.168.2.6>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199@192.168.2.16:5060;transport=udp>.
User-Agent:  SBC.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 212.
Remote-Party-ID: "599261244747199" <sip:599261244747199@192.168.2.16>;party=calling;privacy=off;screen=no.
.
v=0.
o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
s=SBC.
c=IN IP4 192.168.2.16.
t=0 0.
m=audio 16922 RTP/AVP 18 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=ptime:20.




Regds
Sam





On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre@gmail.com> wrote:
Close. You can only have one set of {} brackets. You can separate multiple variables with a comma.


<action application="bridge" data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>

-Steve



On 29 January 2011 04:29, Sam <u2nsam@gmail.com> wrote:
Hi,

So you say i need to put
<action application="bridge" data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>

Regds
Sam





On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you need sip_ignore_183nosdp=true set on the b leg not the a leg.
Put it in the dial string in {}

{sip_ignore_183nosdp=true}sofia/foo/foo@bar.com


On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam@gmail.com> wrote:
> Hi,
>
> how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the leg a.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to leg a, and when actual 183 with sdp comes it should send .
>
> Its because, providers are sending false signaling by sending 183 without
> sdp,and it hampers while @ production,
> Although by cisco sbc i have done this but i want to do it by FS,
> Take a scenario, when call is send 183 without sdp for 10 secs and then
> followed by 183 with sdp ( actual signal),
> but when some one dials invalid number it rings for 10 secs and then gives
> SIP cause 404, which is bad from the providers.
> So this is the reason i want to block it.
>
> Most of the providers do this, the way out is blocking.
>
> I have got an advice from Tihomir  to do "execute_on_ring and parse your 180
> / 183 messages in search of SDP,
> once you get 183 without SDP do not send it back to leg a and send signal
> only when you got 183 with sdp or 180 "
> And this could be valid call flow.
>
> This happens in many cases where the provider is having nextone as a sbc and
> that too tier 1 !
>
> Regards
> Sam
>
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>



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