Newbie Needs Help

After installing the system (CentOS 5.2 final latest ver and so for Asterisk 1.4.24 and freePBX 2.6.0.1),Sangoma A200 with 4 FXS and 2 FXOs ports. I know that the hardware is working, I can get the channel numbers by using chan-dahdi.conf file and all that. Then I proceed to create the SIp extension and later on my trunk and out/in routes, here i where my problem lays.

I create a Zap trunk, name it ATT1234, and then proceed to fill in the dial rules as follows:
NXXNXXXXXX
1NXXNXXXXXX

As far my understanding is this is the way that my card (trunk) will dial the number, and is actually the way I want it to dial it.

The I delete the 9_ trunk (that comes pre defined) even wanting the use of the number 9 to dial external lines.

Then I created an outbound route, Named it out and used the following dial rules:
9|1NXXNXXXXXX
9|NXXNXXXXXX

I used the 9| so users should dial a 9 in order to make the call
then I specify the trunk sequence to g1-29 which is the trunk I created for this.

Then I tried to make a call and I always get the all circuits busy message

What Im not setting right? is there something I’m missing to be able to make a call using one FXS port out of the 4 in my card?. Where do I locate the config file for this other stuff, meaning, where is the rest of the config related to dial plan? I can see the cards in chan-dahdi.conf but at the moment I’m not aware of what other config files I need to pay attention to try to find my errors.

Thanks, all help will be appreciated.

You just leave the dial rule blank for trunks. You create an outbound rule only. An inbound route for the FX0 Ports and point them to your operator or hunt group.

If you have 4 FXS and 2 FXO ports then the numbering would be something like this:

4 FXS ports - channel 1-4, group 0 (g0)
2 FXO ports - channel 5-6, group 1 (g1)

The trunk sequence g1-29 doesn’t make sense. On your outbound route, you would indicate trunk g1.

/etc/dahdi/system.conf holds the hardware configuration and it should match what is in chan-dahdi.conf.

/etc/asterisk/chan-dahdi.conf holds the asterisk config. Sometimes depending on your distribution, chan-dahdi.conf holds global settings and there is chan-dahdi-addition.conf which holds the FXS extension info. Then chan-dahdi-custom.conf holds the FXO configs.

A good place to look when things aren’t quite right is /var/log/asterisk/full file. It is the log file for asterisk (can be extremely large) and if you search for the term dahdi, you’ll see the registration between the hardware and Asterisk. It may show error messages on what is not registering properly.

Eugene,

Thanks for your suggestions, I will try them tomorrow. The fXS and FXO ports are shwinf in the chan-dahdi.conf as group 1 from 27 to 31, since group 0 belongs to the T1 card, tha tI think goes from 0 or 1 to 24 or something like that. I will post the chan-dahdi.conf as soon as I get to play with your suggestions on the server, and post any error that might show in the file you indicated.

I do appreciate your help

I review the info and tried on my system, but is still not letting me dial outside. I’m gong to post some of those files you told me to see if anyone can catch what is that I’m missing or doing as a mistake.

Thii is the system.conf file:

#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2010-01-27
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma A101 port 1 [slot:4 bus:4 span:1]
span=1,0,0,esf,b8zs
bchan=1-23
echocanceller=mg2,1-23
hardhdlc=24

#Sangoma AFT-A200 [slot:0 bus:4 span:2]
fxoks=27
echocanceller=mg2,27
fxoks=28
echocanceller=mg2,28
fxsks=29
echocanceller=mg2,29
fxsks=30
echocanceller=mg2,30
fxsks=31
echocanceller=mg2,31
fxsks=32
echocanceller=mg2,32
(END)

This is the dahdi-chan.conf file:

autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2010-01-27
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A101 port 1 [slot:4 bus:4 span:1]
switchtype=national
context=from-pstn
group=0
echocancel=yes
faxdetect=incoming
signalling=pri_cpe
channel =>1-23

;Sangoma AFT-A200 [slot:0 bus:4 span:2]
context=from-internal
group=1
echocancel=yes
faxdetect=incoming
signalling = fxo_ks
channel => 27

context=from-internal
group=1
echocancel=yes
faxdetect=incoming
signalling = fxo_ks
channel => 28

context=from-zaptel
group=0
echocancel=yes
faxdetect=incoming
signalling = fxs_ks
channel => 29

context=from-zaptel
group=0
echocancel=yes
faxdetect=incoming
signalling = fxs_ks
channel => 30

context=from-zaptel
group=0
echocancel=yes
faxdetect=incoming
signalling = fxs_ks
channel => 31

context=from-zaptel
group=0
echocancel=yes
faxdetect=incoming
signalling = fxs_ks
channel => 32

I’m still trying to figure out what is missing in my config to make and outbound call. Every time I try to make a call I get the “all circuits busy” message. Below is what shows under active channels at the moment of the call. Any help or guide to troubleshoot will be appreciated. I’m had this issue for over a week now and I have a deadline to develop this system.

Active Channel(s)

Channel Location State Application(Data)
SIP/2220-00000003 emergency@macro-outi Up Playback(all-circuits-busy-now
1 active channel
1 active call