Anthony it worked with FS 1.0.7 but FS crashes every 2 hours. Regds Sam On Mon, Jan 31, 2011 at 10:03 PM, Anthony Minessale < anthony.minessale@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@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@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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