I have a current 1.4 asterisk build with a TE122 card.
I am currently creating .call files with both numbers I need to dial as one full number (555000116184654545).
The 5550001 number goes out the PRI to a hardware pbx and goes to another device that supplies me a second dial tone.
The 16184654545 number is then dialed.
Using the current method I get random failures and I know I am never making it to the second device(nothing shows up in the logs).
I would like to dial the first number, go to my agi script, then send through the second number (DTMF tones?). What happens though (I think) is that after the first number is dialed Asterisk still sees a dial tone (from the second device) and never continues past that point.
Any suggestions?
I don’t like to bump my own threads but surely someone has some idea about this? Any way to make Asterisk not do dial tone detection or something?