Congestion message for external calls and false rings for internal calls

I am a newbie to asterisk. i have just installed freepbx 2.7. I have a PRI line from my university Nortel PBX to my asterisk server. I have currently set up a single sip device (softphone) and one outgoing and incoming route as well as a single ZAP trunk associated with a single DID.

Whenever I try to dial a number within the university it takes a long time to connect and in the end gives a false ring. I know that as the number is of my lab itself. If I dial and external local number, like my mobile, it says all circuits are busy now, try after sometime. I cant figure out where did I err in my configuration. Please help me out. Here are the config files.


Span 1: TE2/0/1 “T2XXP (PCI) Card 0 Span 1” (MASTER) B8ZS/D4 ClockSource


termtype: te



; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf ####(this file is empty)######

; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n

; Span 1: TE2/0/1 “T2XXP (PCI) Card 0 Span 1” (MASTER) B8ZS/D4 ClockSource
switchtype = national
signalling = pri_cpe
channel => 1-23
context = default
group = 63

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callerid=device <1000>

First off, you are not using Zap, you are using dahdi drivers in zap compatibility mode. Are you using a T1 crossover cable between the Nortel PBX and your Asterisk box? Is the light on the T1 card green? What type of Nortel PBX is it? Some use switchtype=dms-100

Your /etc/dahdi/system.conf looks okay. But your /etc/asterisk/ configs are a little bit wonky. You have the T1 channel configured in both /etc/asterisk/chan_dahdi.conf and /etc/asterisk/dahdi_chan.conf. The T1 should only be configured in one.

dahdi_genconf generates /etc/asterisk/dahdi-channels.conf but I find that it is better used as a guide for configuring as it sometimes adds default channel configs that don’t match what FreePBX creates. You can end up with conflicts. asterisk is VERY picky about dahdi, if there is one error, nothing works! I delete the include from /etc/asterisk/chan_dahdi.conf if it exists.

My preference is for /etc/asterisk/chan_dahdi.conf to just include the following configs:

#include chan_dahdi_additional.conf <= FreePBX will create FXS extensions in here, don’t touch as anything manually edited will be written over by FreePBX

#include chan_dahdi_custom.conf <= create this one for your T1 and FXO ports, FreePBX won’t touch custom configs. I would put the following config in /etc/asterisk/chan_dahdi_custom.conf:

; Span 1: TE2/0/1 “T2XXP (PCI) Card 0 Span 1” (MASTER) B8ZS/D4 ClockSource
context=from-pstn <= this is not internal, it comes from outside
switchtype = national
signalling = pri_cpe
channel => 1-23
context = default
group=0,24 <= group 0 using channel 24 for D channel

Create a trunk in FreePBX, use g0 as that is the group that you created (the example uses g2):

Create an outbound route to use the trunk, this example takes any 7 digit number starting with 284 and outputs the last 4 digits out the trunk zap/g2:

Then create an inbound route based on the DID or CID coming from the Nortel PBX and direct it to your phone extension.

Here’s a website that I put together that details how to troubleshoot and to create a PRI T1 between two Asterisk PBXs that may further help you.

Here is my configuration from Nortel PBX. It is a CS 1000 server.

(Nortel PBX)-----T1 line ----(RAD IPMux-24 at remote location) ----- ethernet/fiber ------(RAD IPMux-24 in lab)------T1 line ----(Digium TE220B interface card in asterisk server in lab)

I have the one IPMux equipment in my lab and it is not showing any alarms. The T1 line at my end at both the interfaces, asterisk and IpMUX is showing green alarm. I am not sure about the remote IPMux showing any alarms but will check it out. I am just guessing here but it might be possible that alarms/issues with T1 line b/w nortel and remote IPMux might not be visible to me here.

i tried with the configuration above but still getting congestion message. I have set up and incoming route tying the DID with the extension, but whenever I try to call on DID no I get a busy tone. I find it strange as the voicemail is configured for the SIP extensions and it should go to the voicemail and not give busy tone. Calling b/w 2 extensions is working smoothly. So is possible there might be issues with the remote devices that is not visible to me. I cant go there and check it out immediately as the equipment is not managed by me but my university but hope they dont take their own time to verify it.