[Freeswitch-users] blocking 183 w/o sdp

Sam u2nsam at gmail.com
Mon Jan 31 07:00:04 MSK 2011


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 at 192.168.2.16 <sip%3A599261244747199 at 192.168.2.16>
>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101 at 192.168.2.16<sip%3A0280910101 at 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 at 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 at 192.168.2.158<sip%3A0280910101 at 192.168.2.158>
>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199 at 192.168.2.6 <sip%3A61244747199 at 192.168.2.6>
>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199 at 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 at 192.168.2.16<sip%3A599261244747199 at 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 at 192.168.2.98 <sip%3A599261244747199 at 192.168.2.98>
>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101 at 192.168.2.16<sip%3A0280910101 at 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 at 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 at 192.168.2.158<sip%3A0280910101 at 192.168.2.158>
>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199 at 192.168.2.6 <sip%3A61244747199 at 192.168.2.6>
>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199 at 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 at 192.168.2.16<sip%3A599261244747199 at 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 at 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@
> ${dialed_domain}"/>
>
> -Steve
>
>
>
> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> So you say i need to put
>> <action application="bridge"
>> data="{sip_contact_user=${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 at 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 at bar.com
>>>
>>>
>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at 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
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
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>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>
>>
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>>
>>
>
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