Hi,<br><br>After using , <action application="bridge" data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/><br><br>the 183 without udp is not blocked/ignored .<br>
<br>Below are the traces to visualize:<br>192.168.2.98 is provider<br>192.168.2.16 is FS<br><br><br>U <a href="http://192.168.2.98:5060">192.168.2.98:5060</a> -> <a href="http://192.168.2.16:5060">192.168.2.16:5060</a><br>
SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.<br>To: <<a href="mailto:sip%3A599261244747199@192.168.2.16">sip:599261244747199@192.168.2.16</a>>;tag=3505434022-138257.<br>
From: "0280910101" <<a href="mailto:sip%3A0280910101@192.168.2.16">sip:0280910101@192.168.2.16</a>>;tag=51SjQQQUX14QF.<br>Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.<br>CSeq: 7886492 INVITE.<br>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.<br>
Contact: <<a href="http://sip:599261244747199@192.168.2.98:5060">sip:599261244747199@192.168.2.98:5060</a>>.<br>Call-Info: <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".<br>Content-Length: 0.<br>
.<br><br><br>U <a href="http://192.168.2.16:5060">192.168.2.16:5060</a> -> <a href="http://192.168.2.6:5060">192.168.2.6:5060</a><br>SIP/2.0 180 Ringing.<br>Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.<br>
Via: SIP/2.0/UDP 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.<br>Record-Route: <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.<br>
From: "0280910101" <<a href="mailto:sip%3A0280910101@192.168.2.158">sip:0280910101@192.168.2.158</a>>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.<br>To: <<a href="mailto:sip%3A61244747199@192.168.2.6">sip:61244747199@192.168.2.6</a>>;tag=3F70K1Nm3Frjr.<br>
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.<br>CSeq: 1 INVITE.<br>Contact: <sip:61244747199@192.168.2.16:5060;transport=udp>.<br>User-Agent: SBC.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.<br>
Supported: timer, precondition, path, replaces.<br>Allow-Events: talk, refer.<br>Content-Length: 0.<br>Remote-Party-ID: "599261244747199" <<a href="mailto:sip%3A599261244747199@192.168.2.16">sip:599261244747199@192.168.2.16</a>>;party=calling;privacy=off;screen=no.<br>
.<br><br><br>U <a href="http://192.168.2.98:5060">192.168.2.98:5060</a> -> <a href="http://192.168.2.16:5060">192.168.2.16:5060</a><br>SIP/2.0 180 Ringing.<br>Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.<br>
To: <<a href="mailto:sip%3A599261244747199@192.168.2.98">sip:599261244747199@192.168.2.98</a>>;tag=3505434022-138257.<br>From: "0280910101" <<a href="mailto:sip%3A0280910101@192.168.2.16">sip:0280910101@192.168.2.16</a>>;tag=51SjQQQUX14QF.<br>
Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.<br>CSeq: 7886492 INVITE.<br>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.<br>Contact: <<a href="http://sip:599261244747199@192.168.2.98:5060">sip:599261244747199@192.168.2.98:5060</a>>.<br>
Call-Info: <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".<br>Content-Type: application/sdp.<br>Content-Length: 209.<br>.<br>v=0.<br>o=vsnl2 770 13521 IN IP4 192.168.2.98.<br>s=sip call.<br>
c=IN IP4 115.113.121.99.<br>t=0 0.<br>m=audio 49034 RTP/AVP 18 101.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-11.<br>a=ptime:20.<br>a=rtpmap:18 G729/8000/1.<br><br><br>U <a href="http://192.168.2.16:5060">192.168.2.16:5060</a> -> <a href="http://192.168.2.6:5060">192.168.2.6:5060</a><br>
SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.<br>Via: SIP/2.0/UDP 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.<br>Record-Route: <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.<br>
From: "0280910101" <<a href="mailto:sip%3A0280910101@192.168.2.158">sip:0280910101@192.168.2.158</a>>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.<br>To: <<a href="mailto:sip%3A61244747199@192.168.2.6">sip:61244747199@192.168.2.6</a>>;tag=3F70K1Nm3Frjr.<br>
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.<br>CSeq: 1 INVITE.<br>Contact: <sip:61244747199@192.168.2.16:5060;transport=udp>.<br>User-Agent: SBC.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.<br>
Supported: timer, precondition, path, replaces.<br>Allow-Events: talk, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 212.<br>Remote-Party-ID: "599261244747199" <<a href="mailto:sip%3A599261244747199@192.168.2.16">sip:599261244747199@192.168.2.16</a>>;party=calling;privacy=off;screen=no.<br>
.<br>v=0.<br>o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.<br>s=SBC.<br>c=IN IP4 192.168.2.16.<br>t=0 0.<br>m=audio 16922 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000/1.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-11.<br>
a=ptime:20.<br><br><br><br><br>Regds<br>Sam<br><br><br><br><br><br><div class="gmail_quote">On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Close. You can only have one set of {} brackets. You can separate multiple variables with a comma.<div class="im">
<br><br><action application="bridge" data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/><br>
<br></div>-Steve<div><div></div><div class="h5"><br><br><br><div class="gmail_quote">On 29 January 2011 04:29, Sam <span dir="ltr"><<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi,<br><br>So you say i need to put <br><action application="bridge" data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/><br><br>Regds<br><font color="#888888">Sam</font><div>
<div></div><div><br><br><br><br>
<br><div class="gmail_quote">On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
you need sip_ignore_183nosdp=true set on the b leg not the a leg.<br>
Put it in the dial string in {}<br>
<br>
{sip_ignore_183nosdp=true}sofia/foo/<a href="mailto:foo@bar.com" target="_blank">foo@bar.com</a><br>
<div><div></div><div><br>
<br>
On Fri, Jan 28, 2011 at 12:41 AM, Sam <<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>> wrote:<br>
> Hi,<br>
><br>
> how can i ignore 183 without sdp,<br>
> what happens is the provider sends 183 without sdp and by applying "<action<br>
> application="set" data="sip_ignore_183nosdp=true"/>" the FS sends 180 to<br>
> the leg a.<br>
> Here i want to block the 183 with SDP just like router as b2bua and send<br>
> nothing to leg a, and when actual 183 with sdp comes it should send .<br>
><br>
> Its because, providers are sending false signaling by sending 183 without<br>
> sdp,and it hampers while @ production,<br>
> Although by cisco sbc i have done this but i want to do it by FS,<br>
> Take a scenario, when call is send 183 without sdp for 10 secs and then<br>
> followed by 183 with sdp ( actual signal),<br>
> but when some one dials invalid number it rings for 10 secs and then gives<br>
> SIP cause 404, which is bad from the providers.<br>
> So this is the reason i want to block it.<br>
><br>
> Most of the providers do this, the way out is blocking.<br>
><br>
> I have got an advice from Tihomir to do "execute_on_ring and parse your 180<br>
> / 183 messages in search of SDP,<br>
> once you get 183 without SDP do not send it back to leg a and send signal<br>
> only when you got 183 with sdp or 180 "<br>
> And this could be valid call flow.<br>
><br>
> This happens in many cases where the provider is having nextone as a sbc and<br>
> that too tier 1 !<br>
><br>
> Regards<br>
> Sam<br>
><br>
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