[Asterisk-Users] SIP INVITEs borked with iconnecthere

[alex at pilosoft.com]

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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