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Summary:ASTERISK-08706: Asterisk doesn't see a bridged call even though the call goes through
Reporter:Edward Concilio (econcilio)Labels:
Date Opened:2007-01-31 15:41:30.000-0600Date Closed:2011-06-07 14:08:00
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/RTP
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) verbosedebug.txt
Description:When calling two specific 800 numbers, these are the only two that I know of, the call goes through and the two parties communicate with each other, but asterisk doesn't recognize the call, the call is not bridged, see upload file for details.

I have tested both of these numbers on three different phones, Linksys SPA942, Eyebeam 1.5.10.2, and vaxVOIP 5.0.7.6 over two different carriers, VOIPstreet and jnctn.net, all combinations and all with the same results.

While the call is still connected, if the calling party, after a few seconds, presses any of the tone/number keys, the call is finally bridged:

   -- Called jnctn/18002282144
   -- SIP/jnctn-08d61f88 is making progress passing it to SIP/102200-08d54a28
(Here is where I pressed the ?#? key on the phone)
   -- SIP/jnctn-08d61f88 answered SIP/102200-08d54a28
   -- Attempting native bridge of SIP/102200-08d54a28 and SIP/jnctn-08d61f88

Note: The above is not part of the upload file. I can provide a new debug text file if need be.



****** ADDITIONAL INFORMATION ******

Also, I have setup a 1.4 asterisk testing server with the same results. I can provide a verbose debugging text file if need be.
Comments:By: Jared Smith (jsmith) 2007-01-31 15:54:05.000-0600

I'm not sure why you don't think the calls are bridged... if Asterisk says:

-- Called jnctn/18002282144

it means just that -- that the current channel dialed to that channel and bridged the two.

(Also, as a side note, there was a bunch of stuff in your verbose log about SIP/2.0/TCP.  Asterisk doesn't yet support SIP signalling over TCP, so you may want to check your SIP devices and make sure they communicate with Asterisk over UDP.)
By: Joshua C. Colp (jcolp) 2007-01-31 15:55:31.000-0600

This is being caused from the telephone number you are dialing. It is setup to do early media until you are bridged to a live person. Nothing Asterisk can do since it was never told the call was answered.