thr3ads.net - asterisk users - [Asterisk-Users] SIP INVITEs borked with iconnecthere [Mar 2003]

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John Todd

2003-Mar-06 00:40 UTC

[Asterisk-Users] SIP INVITEs borked with iconnecthere

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

alex at pilosoft.com

2003-Mar-06 01:12 UTC

head link

[Asterisk-Users] SIP INVITEs borked with iconnecthere

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
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Brian Capouch

2003-Mar-06 01:22 UTC

head link

[Asterisk-Users] SIP INVITEs borked with iconnecthere

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.
> 
> http://lists.digium.com/mailman/listinfo/asterisk-users
I had the same problem; Mark helped me with it and IIRC he set a 
callerid field in one of the conf files and it started working like a charm.

I don't remember the details, and unfortunately I've got a lot of 
"callerid" fields in my confs :-(

B.

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