[Freeswitch-users] blocking 183 w/o sdp

Anthony Minessale anthony.minessale at gmail.com
Mon Jan 31 19:34:11 MSK 2011 

Also your trace is incomplete,

you are best off with

sofia global siptrace on
console loglevel debug



On Mon, Jan 31, 2011 at 10:33 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If it does not work for you, your version of FreeSWITCH may be too old
> for this particular feature.
> Did you try with the latest release snapshot?
>
>
> On Sun, Jan 30, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:
>> 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>;tag=3505434022-138257.
>> From: "0280910101" <sip:0280910101 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>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
>> To: <sip:61244747199 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>;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>;tag=3505434022-138257.
>> From: "0280910101" <sip:0280910101 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>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
>> To: <sip:61244747199 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>;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
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>
>>>>
>>>> _______________________________________________
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>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

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