Question from a true Dahdi dummy

Asterisk Asterisk Support
1

I have done at least 25 Asterisk builds on Centos 5.4, Fedora, Debian Lenny and Unbuntu Karmic. But I am a Dahdi dummy (no pun intended). Only 1 of those successful builds ever used dahdi for anything but a timing source.

I am trying to install a new Xorcom Astribank with 1 t1 port and 8 fxs ports. Fxs is working. I am in the US and this is a pretty standard PRI/T1 from Cincinnati Bell.

The one dahdi hardware build I did had 1 fxs and 1 fxo from Openvox on a PCI card (Ubuntu 9.10). I remember it being a bit of a challenge but nothing like this Xorcom Astribank. And,… Xorcom has pretty much snubbed me on tech support.

So after 3 days I finally got all 8 of the FXS ports to take and make calls and that was only after I looked up my notes from the Openvox install 8 months ago.

I hit a wall when it came time to plug in the T1 and cut over the rest of the system. Inbound calls work fine but if I try to call out I get all circuits are busy.

I have Zap/Dadhi trunk called g0 and I made it the only trunk on my outbound, to-pstn route.

When I plug in, “dahdi show status” gives me an OK on the T1 interface and the FXS unit. “Dahdi show channels” shows me channels 1 to 23 with ISDN-PRI signalling and 25 to 32 with Fxo kewl start signalling.

All my dahdi configuration files are listed below. Can anyone spot any mistakes and give me a little Dahdi for dummies lesson. I have to complete this cutover in a week and I am getting desperate with no help from Xorcom.

System Summary:
Ubuntu 9.10 - Updated with apt-get today
Asterisk 1.4.29 - from source
Dahdi Complete 2.2.1 - from source
FreePBX 2.7.0 - updated today
libusb-dev was installed before compile
xpploader is installed and seems to work
I get perl errors from usb.pm when I run dahdi_hardware but dahdi_genconf seemed to work anyway and build the fxs channels properly.
All FXS channels have matching extensions built by FreePBX
amportal…conf has ZAP2DAHDICOMPAT=true

/etc/dahdi/system.conf

Autogenerated by /usr/sbin/dahdi_genconf on Fri Apr 9 23:59:58 2010

If you edit this file and execute /usr/sbin/dahdi_genconf again,

your manual changes will be LOST.

Dahdi Configuration File

This file is parsed by the Dahdi Configurator, dahdi_cfg

Span 1: XBUS-00/XPD-00 “Xorcom XPD #00/00: T1” (MASTER) RED

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
echocanceller=mg2,1-23

Span 2: XBUS-00/XPD-10 “Xorcom XPD #00/10: FXS”

fxoks=25
echocanceller=mg2,25
fxoks=26
echocanceller=mg2,26
fxoks=27
echocanceller=mg2,27
fxoks=28
echocanceller=mg2,28
fxoks=29
echocanceller=mg2,29
fxoks=30
echocanceller=mg2,30
fxoks=31
echocanceller=mg2,31
fxoks=32
echocanceller=mg2,32

Global data

loadzone = us
defaultzone = us

/etc/dahdi/xpp.conf

pri_protocol T1

/etc/asterisk/chan_dahdi.conf

[channels]
#include chan_dahdi_custom.conf
#include chan_dahdi_additional.conf

/etc/asterisk/chan_dahdi_custom.conf

signalling=pri_cpe
switchtype=national
group=0
context=from-pstn
channel => 1-23
;faxdetect=incoming
usecallerid=yes
echocancel=yes
;echocancelwhenbridged=no
;echotraining=800
;context=from-internal
context = default

/etc/asterisk/chan_dahdi_additional.conf

This is just an excerpt their are 6 more channels defined and they are all the same

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

;;;;;;[6325]
signalling=fxo_ks
pickupgroup=
mailbox=6325@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device <6325>
busydetect=no
busycount=7
accountcode=
channel=>25

;;;;;;[6326]
signalling=fxo_ks
pickupgroup=
mailbox=6326@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device <6326>
busydetect=no
busycount=7
accountcode=
channel=>26

2

Hello,

what shows cat /proc/dahdi/* ?

could you post a log for an outgoing call?

have you tried to configure the trunk with 1 instead of g0? You will use (just for test purposes) the channel #1 for that call.

Very Important: have you set the specific callerID needed for the PRI line? Usually you have to ask your provider what callerID you have to use before dialing on the PRI otherwise they will reject your call.
(This is the big difference between FXO and PRI lines when you make outgoing calls._

HTH,
Ioan

3

do you really want two different context settings here?

4

There were 2 problems that caused my “all circuits are busy”. Xorcom finally stepped in and did some diagnostics to find out why the Telco was rejecting my calls and their level 2 tech noticed there was a problem with the sequence of my chan_dahdi_custom.conf

  1. I didn’t understand some of the syntax and the sequence of the .conf files The usecallerid=yes wasn’t aknowledged because it was after the channel => 1-23 line

  2. The above had to be done because my Telco’s (Cincinnati Bell), Adtran CSU required authentication via Caller ID with one of my DIDs and they didn’t bother to tell me

Once I made the 1st change so I was able to send the CID,… I put in my outbound route definition in FreePBX, got through the CSU and out to the PSTN.

Problem solved.

Thanks for your input mudslide. I put the 2nd context in there because I had seen it in several examples.

5

Glad to hear that you have solved your problem.

I will like to clarify the issue with usecallerid. In your specific case (PRI trunk) this parameter specify how to set the callerid on the incoming calls. As your problem was with outgoing calls, this parameter do not “participate” in solving your issue but for presenting the callerid information on the incoming call.

[quote]from: voip-info.org/wiki/view/Aste … apata.conf
usecallerid: For handsets, this option will cause Asterisk to send Caller ID data to the handset when ringing it. For trunk lines, this option causes Asterisk to look for Caller ID on incoming calls. Default: yes. [/quote]

My 2 cents,
Ioan.