When i call from my extension out, i have issues where when calling AT&T for instance there pbx tones, i think are messing up my asterisk setup? does any one have any pointers on how to avoid this issue??
… and explain your problem better.
During my call to AT&T over sip trunk through asterisk call goes fine, i navigate through there prompts, but after a certain point the call becomes muted. Called back several times and same issue. When calling from my cell phone i get to the same point only this time it works and it seems that AT&T PBX system is transferring me in their system, at these transfer points, is where it goes silent on my sip trunk call through Asterisk?
Or both?
What kind of phones do you have?
What dtmfmode are you using?
Can you get your asterisk log of the problem?
Some other information that might provide a clue.
A copy of the output of the status program. This will help those in the know a lot and it will go a long way to narrowing down your particular version.
What processor/motherboard/amount of ram/and if you are using a solid state drive.
PBX in a Flash Version: (valid ones are 1.0, 1.1, and 1.2) Linux status command
Operating system: (valid are 32bit Centos 5.0, 32bit Centos 5.1, 64bit Centos 5.1)
Asterisk Version: (valid is Asterisk 1.4 old*, Asterisk 1.4, Asterisk 1.6 BETA)
Without knowing enough to say so, it sounds like an AT&T problem. Good luck with getting them to resolve that. I have had night-marish experiences with them.
I would suggest that when AT&T offers to transfer you, you get the toll-free number that they are transferring you to. From my experience, you start over whenever they transfer you anyway. They don’t pass information between each departments or use a tracking database to track calls.
What kind of phones do you have?