T1 with TCIC's

Anyone know how to configure a T1 with TCIC codes? The T1 is AMI and I have no alarms but the framing is not right and there are TCIC codes to be configured from upstream.

Is Support even active? I’ve been waiting 3 days for a call back and heard nothing. They should at least acknowledge their tickets with a timeframe. Yes I did pay for support.

Yeah…That seems to be my experience with support as well. As far as I can tell, there is no SLA for response time, so they will get to you when they get to you…
You can always call in though once you created a ticket. That might help you get an answer quicker.

not many details here… Last one of these @lgaetz poked someone…
In any case here is the mile high, generic, canned overview if you are using something like a Digium T1 Card with dahdi…

Configuring Channelized and PRI T1 in Asterisk with dahdi.conf and system.conf

Here are comprehensive examples for configuring Channelized and PRI T1 in Asterisk with dahdi.conf and system.conf:

Important Note: These are examples and may need adjustments based on your specific hardware, provider settings, and Asterisk version. Refer to your hardware documentation and Asterisk documentation for detailed configuration options.

1. system.conf

This file defines the overall configuration for your DAHDI channels.

A. Channelized T1 Example:

; Define span configuration
span=1,0,0,ESF,B8ZS  ; Span 1, No Timing Source (slave), No Line Build Out, ESF Framing, B8ZS Coding

; Define channel types
e&m=1-24            ; Channels 1-24 as E&M (Analog trunks)

B. PRI T1 Example:

; Define span configuration
span=1,1,0,ESF,B8ZS  ; Span 1, Use Timing from far end (master), No Line Build Out, ESF Framing, B8ZS Coding

; Define PRI channel signaling
bchan=1-23            ; Channels 1-23 as B-Channels for voice


  • span: Defines a span configuration. The parameters specify timing source, line build out, framing, and coding.
  • e&m: Defines channels as E&M (analog trunks).
  • bchan: Defines channels as B-Channels for PRI.
  • dchan Defines the data (signalling) channel
  • loadzone and defaultzone: Load tone zone data for call signaling.

2. dahdi.conf

This file defines how Asterisk interacts with the DAHDI channels.

Common Configuration:

context=from-pstn        ; Incoming calls go to 'from-pstn' context
switchtype=national      ; Use national dialing rules

; Include channel specific configurations

dahdi-channels.conf (Generated by dahdi_genconf):

This file is automatically generated by the dahdi_genconf tool and should not be edited manually. It includes specific configurations for each channel based on the system.conf settings.

Additional Notes:

  • You might need to adjust contexts (e.g., from-pstn) based on your Asterisk dial plan.
  • Consider adding specific options for each channel type (e.g., call waiting, caller ID) in dahdi-channels.conf.
  • Remember to restart Asterisk after making configuration changes (service asterisk restart).


My T1’s are UP and no Alarms, but due to the CIC codes I cannot get calls. The whole dahdi_genconf and freepbx are indeed conflicting. When I used the dahdi_genconf dahdi would’t start. I went back to using FreePBX dahdi conf, save and restart DAHDi @ Asterisk

There is also a bug in the FreePBX that will create a syntax error in your system.conf file

[root@freepbx asterisk]# dahdi_cfg -v
DAHDI Tools Version - 2.11.1
Notice: Configuration file is /etc/dahdi/system.conf
line 11: Syntax error in range ‘1-2425-4849-7273-95’. Should be -.
line 12: Syntax error in range ‘1-2425-4849-7273-95’. Should be -.

These lines should have comma between them.

So 1-2425-4849-7273-95 should be 1-24,25-48,49-72,73-95

Suggest you don’t use FreePBX helper here

for non PRI T1’s in the US using DAHDI supported hardware , system.conf should likely have (as I do) something like

loadzone = us                                                                                                            

for two old fashioned 24 channel T1’s

My system.conf looks like this:


When I try to add the ss7 stuff DADHI won’t start. Specifically when I put in the CIC information. I am trying to do ss7 ISUP

ISUP is the signalling protocol used to establish, maintain, and release circuit-switched connections across an SS7 network. The most common ISUP messages include IAM, ACM, ANM, REL and RLC. Initial Address Message (IAM) is a message sent to establish a connection. The message body includes Circuit Identification Code (CIC) informing the partner switch about which voice circuit the call will be placed on, information identifying calling parties - called and calling party number and type of service (speech or data) and many more optional parameters. Address Complete Message (ACM) is a message sent by the terminating side of a connection after collecting enough information to reach the called party. Answer Message (ANM) is sent when called party answers the call and charging is started. Release (REL) clears the call and Release complete (RLC) message acknowledges the call release and the physical timeslot can be used again.

Sorry, no experience with ss7 in Asterisk/Dahdi


has a caveat at the very top that would make me go back to Cisco

Page 4 of https://ftp.sangoma.com/linux/custom/konrad/How-to-Libss7.pdf may help, even if most of it doesn’t relate to your hardware.

Although the actual question was never answered, libss7 CIC alignment configuration - Asterisk Support - Asterisk Community gives an example for multiple E1s.