Can't Dial Out

Asterisk Asterisk Support
#1

I use SIP phone (X-lite) dail to PSTN number 18503
I have setup if dail a prefix “0” it will take the Zap trunk
I am sure it can pick the trunk but cannot dail out to pstn

here is the log

Sep 17 19:21:03 DEBUG[5074] pbx.c: Function result is 'ZAP/1'
Sep 17 19:21:03 DEBUG[5074] pbx.c: Expression result is '0'
Sep 17 19:21:03 DEBUG[5074] pbx.c: Not taking any branch
Sep 17 19:21:03 DEBUG[5074] chan_zap.c: Dialing '18503'
Sep 17 19:21:03 DEBUG[5074] chan_zap.c: Deferring dialing...
Sep 17 19:21:04 DEBUG[5074] chan_zap.c: Exception on 14, channel 1
Sep 17 19:21:04 DEBUG[5074] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0)
Sep 17 19:21:06 DEBUG[5074] chan_zap.c: Exception on 14, channel 1
Sep 17 19:21:06 DEBUG[5074] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0)
Sep 17 19:21:06 DEBUG[5074] chan_zap.c: Enabled echo cancellation on channel 1
Sep 17 19:21:06 WARNING[5074] chan_zap.c: Unable to request echo training on channel 1
Sep 17 19:21:13 DEBUG[5074] chan_zap.c: Exception on 14, channel 1
Sep 17 19:21:13 DEBUG[5074] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0)
Sep 17 19:21:13 DEBUG[5074] chan_zap.c: Echo cancellation already on

PLs help thank you

#2

It depends on how many pots lines you want, what kind of phones you want your epmpoyees to have etc.
For every POTS line that you have coming in you need an FXO port. There are several companies that make cards. Some include Digium (Digium.com) and Sangoma (Sangoma.com). You may also want to look at getting a card with echo cancelation. Although it is a bit more pricey it can make all the diffrence. Then you have to look at what kind of phones you want your employees to have. IMHO it ends up being a little more to go with a simple voip phone than to give everyone a pots line. As far as the features that you want asterisk can do it all. If you are a linux guru you should have no problem setting it up. Good luck.

#3

I got a X100p clone card and trixbox installed.

I found I can’t dial out to PSTN
I am sure the trunk and route seeting is alright, but don’t konw how to debug with asterisk