Trickey Call Files

Hi All,

Looking for some input.

I have a php script that writes a call file and then dumps it to asterisk. That is good was working fine until yesterday. Now it looks like it is making the call and it even says the channel had been answered. However the call never makes it to my phone. I am at a loss as to what is going on here. If you look at line 195 in the pastebin below it looks as though it is droping the number from the sip string. Not sure where it is going wrong any input would be grate.

Note : The only change I made to the server between now and the time it worked was that I added a FTP server to it. Not sure if that makes any diffrence

Click here to see the sip debug output of the asterisk CLI

Sip trace looks ok to me. Looks like the carrier is sending the ringing, then the connect so according to them the number answered. Is it possible the number you are calling is your cell phone or another number and it went directly to voicemail instead of ringing? (Lines 195-199 in that pastebin are asterisk cli output, not part of the sip transaction)

i concur, everything looks fine to me. if your phone didn’t ring, perhaps your VM answered. i would try another number before you head too far down the ‘asterisk is broken’ path.


I should not have led down the asterisk is broken path in the first place. I am not bright and found out that the bill for the provider was not paid this month. It is all good after i paid the bill. Thanks to all who looked. :blush: