[Asterisk-Users] SIP INVITEs borked with iconnecthere

William X Walsh william at wxw.org
Thu Mar 6 03:28:23 MST 2003 

I don't think it is a filter (SIP uses UDP, so telneting to the port
(which uses TCP) doesn't mean much.

There is a problem right now with receiving incoming calls from
iconnecthere/d3.  Something on d3's end is messed up. It hits the
asterisk server, and then cancels the connection.  I'm stumped. 

On Wed, 2003-03-05 at 23:12, alex at pilosoft.com wrote:
> John,
> 
> A heads-up: iconnect has apparently put up a filter against my IP address, 
> for whichever reason (apparently they don't like people using asterisk?).
> 
> I've sent them an email and am pursuing this also through sales side (I'm 
> about to make a resale deal with them), so hopefully tomorrow I'll find 
> out just what they don't like about asterisk.
> 
> To check if they put up a filter, just do a telnet 213.137.73.141 5060
> 
> If you get connection refused, its something else. if it times out, you've 
> been filtered.
> 
> -alex
> 
> On Wed, 5 Mar 2003, John Todd wrote:
> 
> > 
> > Symptoms: when calling my iconnect phone number (13033913323 in my 
> > bogus example below) from my cell phone, I can see that the call 
> > makes it to my asterisk server, and my phones even ring once as * 
> > passes the call through during the "180 Ringing" period.  However, it 
> > seems that iconnecthere.com cannot see my "100 Trying" and "180 
> > Ringing" messages, as they continue to send INVITES to me.  After two 
> > seconds, they either give up or error out and send a CANCEL message.
> > 
> > To further increase my suspicions of something weird in the ability 
> > to "see" my replies, they send 11 CANCEL messages over the period of 
> > 30 seconds, despite my "200 OK" replies.
> > 
> > 
> > Notes: 204.31.11.32 resolves to asterisk.something.com.  204.31.11.35 
> > is my ATA-186.  Neither the domains nor the IP addresses are real, 
> > except when referencing iconnect servers.
> > 
> > 213.137.73.176 is the real IP address of the SIP proxy at 
> > iconnecthere.com (deltathree.com)
> > Note that I actually do my REGISTERs against 213.137.73.178, not .176 
> > - not a big deal, but who knows what clues will be helpful.
> > 
> > Unsuccessful Asterisk->iconnect->PSTN call:
> > 
> > tethereal port 5060 and host 213.137.73.176
> > Capturing on fxp0
> >    0.000000 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> > INVITE sip:13033913323 at 204.31.11.32:5060, with session description
> >    0.001293 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
> >    0.039058 asterisk.something.com -> 213.137.73.176 SIP Status: 180 Ringing
> >    0.490181 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> > INVITE sip:13033913323 at 204.31.11.32:5060, with session description
> >    0.490497 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
> >    1.530125 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> > INVITE sip:13033913323 at 204.31.11.32:5060, with session description
> >    1.530439 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
> >    2.070160 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >    2.070461 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >    2.594680 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >    2.595419 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >    3.634908 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >    3.635179 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >    5.674595 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >    5.674889 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >    9.664659 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >    9.664956 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   13.645471 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   13.645755 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   17.635194 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   17.635502 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   21.665856 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   21.666146 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   25.676487 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   25.676767 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   29.666942 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   29.667231 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> >   33.647019 213.137.73.176 -> asterisk.something.com SIP Request: 
> > CANCEL sip:13033913323 at 204.31.11.32:5060
> >   33.647305 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> > 
> > 
> > 
> > 
> > A successful ATA-186 to iconnect session, no Asterisk server involved:
> > 
> > 1338.126534 213.137.73.176 -> 204.31.11.35 SIP/SDP Request: INVITE 
> > sip:13033913323 at 204.31.11.35:5060;user=phone, with session description
> > 1338.143921 204.31.11.35 -> 213.137.73.176 SIP Status: 100 Trying
> > 1338.155993 204.31.11.35 -> 213.137.73.176 SIP Status: 180 Ringing
> > 1345.703299 204.31.11.35 -> 213.137.73.176 SIP/SDP Status: 200 OK, 
> > with session description
> > 1345.819676 213.137.73.176 -> 204.31.11.35 SIP Request: ACK 
> > sip:13033913323 at 204.31.11.35:5060;user=phone
> > 1362.560816 204.31.11.35 -> 213.137.73.176 SIP Request: BYE 
> > sip:13033913323 at 213.137.73.178;user=phone
> > 1362.681272 213.137.73.176 -> 204.31.11.35 SIP Status: 100 Trying
> > 1362.684056 213.137.73.176 -> 204.31.11.35 SIP Status: 200 OK
> > 
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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-- 
William Walsh <william at wxw.org>
Jabber: william at wxw.biz

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