Alcatel 4400 via PRI
Created by: mdawson,Last modification on Thu 14 of Sep, 2006 [08:34 UTC]
This page explains how to connect Asterisk to an Alcatel PCX 4400 via a 30 channel ISDN E1 trunk. It might not be the only way or the best way of doing it as I have been figuring this out as I go along but could be a useful guide. Please update or comment if you know a better way of doing a particular thing.
This guide assumes you have already setup an Asterisk server and have an Alcatel 4400 connected to the PSTN.
A more upto date version of this document is at the forum here:
You'll need the following:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
trunkgroups
trunkgroup => 1,16
spanmap => 1,1,1
channels
language=uk
context=default
switchtype=euroisdn
signalling=pri_cpe
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel => 1-15,17-31
; this needs improving as it only allows fixed length of extension numbers at present
exten => _0-24-8XX,n,Dial(zap/g1/${EXTEN}) ; internal calls
exten => _9XXXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}) ; 9 for external calls
1. Install the PRA2 card
2. Create a trunk group for Alcatel<->Asterisk
3. Set the physical link parameters
4. Set a route for calls from Alcatel->Asterisk
My Asterisk extensions begin with 3, so under Translator/Prefix plan
5. Enable trunk to trunk dialling
This enables Asterisk->Alcatel->PSTN (thanks again Frank :)
I have also tried setting both ends to QSIG instead of isdn all countries/euroisdn but haven't had any luck with this.
This guide assumes you have already setup an Asterisk server and have an Alcatel 4400 connected to the PSTN.
Resources and current progress
A more upto date version of this document is at the forum here:
- The Alcatel Unleashed Asterisk forum
Hardware
You'll need the following:
- A PRA2 isdn card for the PCX 4400
- A T1/E1 card for Asterisk, I've used a Sangoma a102u, an equivalent Digium card would do too
- a cable to connect the two - I used a TY2 96PT at the Alcatel end which I patched into the back of a UTP socket and then used a normal UTP cable to connect to the Asterisk box. The connections are like so
| Alcatel | Asterisk |
|---|---|
| 1 blue txb (tip) | 1 |
| 1 red txa (ring) | 2 |
| 2 blue rxb (tip) | 4 |
| 2 red rxa (ring) | 5 |
Asterisk Configuration
zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
zapata.conf
trunkgroups
trunkgroup => 1,16
spanmap => 1,1,1
channels
language=uk
context=default
switchtype=euroisdn
signalling=pri_cpe
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel => 1-15,17-31
extensions.conf
; this needs improving as it only allows fixed length of extension numbers at present
exten => _0-24-8XX,n,Dial(zap/g1/${EXTEN}) ; internal calls
exten => _9XXXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}) ; 9 for external calls
Alcatel Configuration
1. Install the PRA2 card
- Under Shelf/Board create a PRA2 object at the appropriate address
- Plug the board in
2. Create a trunk group for Alcatel<->Asterisk
- Under Trunk Groups, name it something like Asterisk
- Q931 signal variant=ISDN all countries
- Under Trunk Group:
- B channel choice=yes
- Under Access create an object:
- If your card is in shelf 0, slot 4 the physical address will be 0-4-0
- access type=T2
3. Set the physical link parameters
- Under Shelf/Board/PRA2/Digital Access set
- Network mode=yes
- CRC4=yes
4. Set a route for calls from Alcatel->Asterisk
My Asterisk extensions begin with 3, so under Translator/Prefix plan
- 3 - Open routing no.
- Node number=the ID number of the Asterisk trunk group
5. Enable trunk to trunk dialling
This enables Asterisk->Alcatel->PSTN (thanks again Frank :)
- Under Classes of Service/Connection COS/5 set Rights COS 5 to 1
Different Connection Types?
I have also tried setting both ends to QSIG instead of isdn all countries/euroisdn but haven't had any luck with this.
Comments
333cannot call alcatel extension from SIP
I have a TE110P card fitted in my linux box running :
Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005
I followed the installation steps on digium website...no errors reported.
The modules seem to have loaded...here's what lsmod shows:
Module Size Used by
wcte11xp 30496 31
zaptel 196740 67 wcte11xp
still the light on my card is off....does that mean the card has not initialised properly?
On loading Asterisk, I do not get any errors, but I do see these warnings:
Parsing '/etc/asterisk/zapata.conf': Found
Nov 1 11:57:21 WARNING3454: chan_zap.c:10874 setup_zap: Ignoring switchtype
Nov 1 11:57:21 WARNING3454: chan_zap.c:10874 setup_zap: Ignoring signalling
on running asterisk -cvvv . I do see Aterisk Ready at the end ...then What do these warnings mean?
Also, I DO NOT get these lines on asterisk startup:-
channel 0/1 successfully restarted on span 1
— B-channel 0/2 successfully restarted on span 1
— B-channel 0/3 successfully restarted on span 1
— B-channel 0/4 successfully restarted on span 1
— B-channel 0/5 successfully restarted on span 1
— B-channel 0/6 successfully restarted on span 1
does that mean my channels are not available?
- CLI> zap show status
Description Alarms IRQ bpviol CRC4Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0
- CLI> pri show span 1
Primary D-channel: 16Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
here's my extensions.conf:
general
static=yes
writeprotect=no
autofallthrough=yes
sip
exten => 9820,1,Dial(SIP/iyer)
exten => 9821,1,Dial(SIP/shweta)
exten => 9810,1,Dial(SIP/shashi)
exten => 9851,1,Dial(Zap/g1/851,20)
incoming
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()
exten => 9821,1,Dial(SIP/shashi)
exten => 9851,n,Dial(Zap/g1/851)
here's zapata.conf
trunkgroups
trunkgroup => 1,16
spanmap =>1,1,1
channels
switchtype=euroisdn
signalling=pri_cpe
context=incoming
language=uk
group=1
callgroup=1
pickupgroup=1
echocancel=yes
immediate=no
channel => 1-15,17-31
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancelwhenbridged=yes
musiconhold=default
here's zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone=us
Now the problem
I can call and talk SIP to SIP...here's what I see on asterisk CLI
-- Executing Dial("SIP/iyer-09326480", "SIP/shweta") in new stack
— Called shweta
— SIP/shweta-0932b9c0 is ringing
But when I call zap extension, here's what I get:
Executing Dial("SIP/iyer-09326480", "Zap/g1/851|20") in new stack
Nov 1 12:07:55 NOTICE3513: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/iyer-09326480' status is 'CONGESTION'
I have connected the PBX to digium card with the specified cable and done the settings in PBX specified at:
http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI
What am I doing wrong?
Any help would be greatly appreciated.
Thanks in advance....
Kind Regards
Shweta