Hi Guys,
I have just setup an FreePBX (with Asterisk-1.4.20) with 12 Ports Sangoma FXO Card. Incoming calls is fine, but when I tried calling out, I got the following logs,
[Apr 2 15:16:20] DEBUG[2773] dsp.c: dsp busy pattern set to 0,0
[Apr 2 15:16:20] DEBUG[2773] chan_zap.c: Dialing ‘08033189788’
[Apr 2 15:16:20] DEBUG[2773] chan_zap.c: Deferring dialing…
[Apr 2 15:16:20] VERBOSE[2773] logger.c: – Called 9/08033189788
[Apr 2 15:16:21] DEBUG[2773] chan_zap.c: Sent deferred digit string: T08033189788w
[Apr 2 15:16:24] VERBOSE[2773] logger.c: – Zap/9-1 answered SIP/1010-0945afe8
[Apr 2 15:16:31] VERBOSE[2776] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Apr 2 15:16:31] VERBOSE[2776] logger.c: Found
[Apr 2 15:16:31] VERBOSE[2776] logger.c: == Parsing ‘/etc/asterisk/manager_additional.conf’: [Apr 2 15:16:31] VERBOSE[2776] logger.c: Found
[Apr 2 15:16:31] VERBOSE[2776] logger.c: == Parsing ‘/etc/asterisk/manager_custom.conf’: [Apr 2 15:16:31] VERBOSE[2776] logger.c: Found
[Apr 2 15:16:31] VERBOSE[2776] logger.c: == Manager ‘admin’ logged on from 127.0.0.1
[Apr 2 15:16:31] VERBOSE[2776] logger.c: == Manager ‘admin’ logged off from 127.0.0.1
[Apr 2 15:17:25] DEBUG[2773] dsp.c: ast_dsp_busydetect detected busy, avgtone: 175, avgsilence 145
[Apr 2 15:17:25] VERBOSE[2773] logger.c: – Hungup ‘Zap/9-1’
What could be wrong. And why am I getting the deferring dialing line.
Cheers to All.
anyone on this please. I am running FreePBX 1.5.1.
FreePBX 1.5.1 might be a little dated, so I assume you mean 2.5.1. Are you using a pre-fab ISO like PBX in a Flash, Elastix or Trixbox?
Since inbound calls seem to work, one test you can perform is to make an outbound route and in the dial pattern put:
9|
Add one of your Zap Trunks as your trunk.
Place this route fairly high in your dial plan. Now dial 9 and wait. If you get dial tone, try dialing the number and see if that works. The 9| will strip off the 9 you dialed and pick up the Zap channel. The dial tone you hear is dial tone from the telco.
If this doesn’t work, post your zapata.conf, zapata-auto.conf, zapata-channels.conf and /etc/zaptel.conf and let us take a look at them.
sorry I mean to say FreePBX 2.5.1 not 1.5.1.
I am not using any of those pre-packaged distros. What I always do is install Asterisk and Port FreePBX onto it. And so far, I have 100% of it working perfectly except for this issue now.
What you explained above worked perfectly. Thank you so much for that insight. What I noticed is that, its a dual wireless DTMF phone that also have the capability of making phone calls directly from the box the provider supply along with it. I think the Box doesn’t understand when I dial all the digits from the IP Phones/Softphones.
I also noticed that using this method one cannot restrict users using the Trunk Password. Once the 9| give the tone, its out of FreePBX and any number dialed with pass through.
I will explore further to see if there is a way I can bypass its limitations.
Cheers to all.
I meant the dial 9 thing for testing. What may be happening then is the PSTN is not ready for your DTMF tones when Asterisk is ready to send them. In you trunk definition for the Zap channels put:
ww+X.
in the dial rules. Each “w” creates a few milliseconds of delay before dialing the number. Also you might add
relaxdtmf=yes
to zapata.conf