Help configuring a TE110P E1 Card

davepearson1628 · October 19, 2006, 2:51pm

I really need some help configuring this card, since I cannot seem to successfully make a call through it - I always get the “All circuits are busy” message. I haven’t yet tried an incoming call. To help diagnose my problem, here are the config files:

zaptel.conf

Span 1: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0”

span=1,0,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16

Global data

loadzone = uk
defaultzone = uk

zapata.conf

;# Flash Operator Panel will parse this file for zap trunk buttons
;# AMPLABEL will be used for the display labels on the buttons
[trunkgroups]

[channels]
language=en
usecallerid=yes
hidecallerid=no
callerid=asreceived
restrictid=no
usecallingpress=yes
switchtype = euroisdn
signalling = pri_cpe
pridialplan=unknown
prilocaldialplan=unknown
priindication=outofband
overlapdial=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-3.0
txgain=-6.0
musiconhold=default
group=0
channel=>1-15
channel=>17-31

The ISDN Channels are recognised in Asterisk, as can be seen when I do a zap show channels command:

asterisk-slo*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default en default
1 default en default
2 default en default
3 default en default
4 default en default
5 default en default
6 default en default
7 default en default
8 default en default
9 default en default
10 default en default
11 default en default
12 default en default
13 default en default
14 default en default
15 default en default
17 default en default
18 default en default
19 default en default
20 default en default
21 default en default
22 default en default
23 default en default
24 default en default
25 default en default
26 default en default
27 default en default
28 default en default
29 default en default
30 default en default
31 default en default

Any help to get this working will be very gratefully received as I have search google to absolutely no avail (hardly anyone seems to be using the E1 configuration of the card)

Regards,
Dave.

davepearson1628 · October 20, 2006, 1:55pm

Success

Having set up the zapata.conf file as per degsod’s example, I looked a bit closer at the outbound route configuration - I had a typing error in there which meant that the number being passed to the isdn line was invalid (I wasn’t stripping out the dial prefix :shock: )

Once I was able to dial out, it was but a short step to being able to dial in.

Now if I can just get my TDM22B card working I’ll be really happy.

Regards,
Dave.

degsod · October 20, 2006, 10:27pm

Hi Dave,

Once i moved over to A@H I stopped using dialing prefixes for outbound calls, prefixing a 9 was really just a hangover from old pbx systems. I just dial the number like you do at home.

I have ISDN30 on my system with 8 lines enabled, DEMON VOIP and 12 hard phones and 5 softphones.

I am now looking at the possability of setting up a Freepbx system at a datacentre with 76 locations connecting to it with 2 phones at each location.

davepearson1628 · October 24, 2006, 10:17am

Yes, I think I shall remove the dialing prefix and set it up to detect the length of the number - the only problem I can see is when someone pauses midway through dialing.

Have you had any experience of setting up a TDM22B (TDM400) card alongside the ISDN card - we need a couple of analog phones to work and I also want to have a couple of emergency analog lines (in case the ISDN fails). Having introduced the TDM22B, it seems to be messing about with the ISDN Channel numbering (which doesn’t help), but I cannot seem to get the extra channels to show up in the Asterisk CLI (zap show channels only displays the ISDN channels).

Regards,
Dave.

Good luck with the data centre.

degsod · October 19, 2006, 9:09pm

Hi,

I am in the uk, and am using A@home 2.7 here are my settings:-

Hope it helps

ZAPTEL.CONF

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

loadzone = uk
defaultzone = uk

ZAPATA.CONF

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
rxwink=300 ; Atlas seems to use long (250ms) winks
;usedistinctiveringdetection=yes
callerid=asreceived
pridialplan=unknown
nationalprefix=0
internationalprefix=00
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

;Include AMP configs
#include zapata_additional.conf
channel => 1-15,17-31

davepearson1628 · October 20, 2006, 9:11am

Hi degsod,

thanks for the conf files, unfortunately they don’t make any difference.

I have tried making a call with “pri debug span 1” enabled and this is what happens

-- Executing Macro("SIP/4002-081e38f8", "dialout-trunk|1|905555111111||") in new stack
-- Executing Set("SIP/4002-081e38f8", "DIAL_TRUNK=1") in new stack
-- Executing Set("SIP/4002-081e38f8", "DIAL_NUMBER=905555111111") in new stack
-- Executing Set("SIP/4002-081e38f8", "ROUTE_PASSWD=") in new stack
-- Executing GotoIf("SIP/4002-081e38f8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing Set("SIP/4002-081e38f8", "GROUP()=OUT_1") in new stack
-- Executing Macro("SIP/4002-081e38f8", "user-callerid") in new stack
-- Executing GotoIf("SIP/4002-081e38f8", "0?report") in new stack
-- Executing GotoIf("SIP/4002-081e38f8", "0?start") in new stack
-- Executing Set("SIP/4002-081e38f8", "REALCALLERIDNUM=4002") in new stack
-- Executing NoOp("SIP/4002-081e38f8", "REALCALLERIDNUM is 4002") in new stack
-- Executing Set("SIP/4002-081e38f8", "AMPUSER=4002") in new stack
-- Executing Set("SIP/4002-081e38f8", "AMPUSERCIDNAME=Test User") in new stack
-- Executing GotoIf("SIP/4002-081e38f8", "0?report") in new stack
-- Executing Set("SIP/4002-081e38f8", "CALLERID(all)=Test User <4002>") in new stack
-- Executing NoOp("SIP/4002-081e38f8", "Using CallerID "Test User" <4002>") in new stack
-- Executing Macro("SIP/4002-081e38f8", "record-enable|4002|OUT") in new stack
-- Executing GotoIf("SIP/4002-081e38f8", "0 > 0?24") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/4002-081e38f8", "recordingcheck|20061020-095853|1161334732.4") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

recordingcheck|20061020-095853|1161334732.4 Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/4002-081e38f8”, “No recording needed”) in new stack
– Executing Macro(“SIP/4002-081e38f8”, “outbound-callerid|1”) in new stack
– Executing GotoIf(“SIP/4002-081e38f8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing NoOp(“SIP/4002-081e38f8”, “REALCALLERIDNUM is 4002”) in new stack
– Executing Set(“SIP/4002-081e38f8”, “USEROUTCID=”) in new stack
– Executing Set(“SIP/4002-081e38f8”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/4002-081e38f8”, “TRUNKOUTCID=05555222222”) in new stack
– Executing GotoIf(“SIP/4002-081e38f8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing GotoIf(“SIP/4002-081e38f8”, “0?usercid”) in new stack
– Executing Set(“SIP/4002-081e38f8”, “CALLERID(all)=05555222222”) in new stack
– Executing GotoIf(“SIP/4002-081e38f8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,17)
– Executing NoOp(“SIP/4002-081e38f8”, “CallerID set to “” <05555222222>”) in new stack
– Executing GotoIf(“SIP/4002-081e38f8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,12)
– Executing AGI(“SIP/4002-081e38f8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix Could not parse /etc/asterisk/localprefixes.conf
– AGI Script fixlocalprefix completed, returning 0
– Executing Set(“SIP/4002-081e38f8”, “OUTNUM=905555111111”) in new stack
– Executing Set(“SIP/4002-081e38f8”, “custom=ZAP/g0”) in new stack
– Executing GotoIf(“SIP/4002-081e38f8”, “0?customtrunk”) in new stack
– Executing Dial(“SIP/4002-081e38f8”, “ZAP/g0/905555111111|120|r”) in new stack
– Making new call for cr 32772
– Requested transfer capability 0x00 - SPEECH

Protocol Discriminator Q.931 (8) len=46
Call Ref len= 2 (reference 4/0x4) (Originator)
Message type SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext 1 Q.931 Std 0 Info transfer capability Speech (0)
Ext 1 Trans mode/rate 64kbps, circuit-mode (16)
Ext 1 User information layer 1 A-Law (35)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext 1 IntID Implicit, PRI Spare 0, Exclusive Dchan 0
ChanSel Reserved
Ext 1 Coding 0 Number Specified Channel Type 3
Ext 1 Channel 1 ]
[6c 0d 21 80 30 31 37 35 33 38 32 38 39 30 31]
Calling Number (len=15) [ Ext 0 TON National Number (2) NPI ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation Presentation permitted, user number not screened (0) ‘05555222222’ ]
[70 0d 80 39 30 31 37 35 33 38 32 38 39 31 30]
Called Number (len=15) [ Ext 1 TON Unknown Number Type (0) NPI Unknown Number Plan (0) ‘905555111111’ ]
[a1]sk-slo*CLI>
Sending Complete (len= 1)
– Called g0/905555111111
< Protocol Discriminator Q.931 (8) len=10
< Call Ref len= 2 (reference 4/0x4) (Terminator)
< Message type CALL PROCEEDING (2)
< [18 03 a9 83 81]
< Channel ID (len= 5) [ Ext 1 IntID Implicit, PRI Spare 0, Exclusive Dchan 0
< ChanSel Reserved
< Ext 1 Coding 0 Number Specified Channel Type 3
< Ext 1 Channel 1 ]
– Processing IE 24 (cs0, Channel Identification)
– Zap/1-1 is proceeding passing it to SIP/4002-081e38f8
< Protocol Discriminator Q.931 (8) len=13
< Call Ref len= 2 (reference 4/0x4) (Terminator)
< Message type DISCONNECT (69)
< [08 02 80 9c]
< Cause (len= 4) [ Ext 1 Coding CCITT (ITU) standard (0) 0 0 Location User (0)
< Ext 1 Cause Invalid number format (28), class = Normal Event (1) ]
< [1e 02 80 88]
< Progress Indicator (len= 4) [ Ext 1 Coding CCITT (ITU) standard (0) 0 0 Location User (0)
< Ext 1 Progress Description Inband information or appropriate pattern now available. (8) ]
– Processing IE 8 (cs0, Cause)
– Processing IE 30 (cs0, Progress Indicator)
– Channel 0/1, span 1 got hangup request
NEW_HANGUP DEBUG Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
Protocol Discriminator Q.931 (8) len=9
Call Ref len= 2 (reference 4/0x4) (Originator)
Message type RELEASE (77)
[08 02 81 9c]I>
Cause (len= 4) [ Ext 1 Coding CCITT (ITU) standard (0) 0 0 Location Private network serving the local user (1)
Ext 1 Cause Invalid number format (28), class = Normal Event (1) ]
– Hungup ‘Zap/1-1’
== Everyone is busy/congested at this time (10/0/1)
– Executing Goto(“SIP/4002-081e38f8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing NoOp(“SIP/4002-081e38f8”, “Dial failed due to CHANUNAVAIL”) in new stack
– Executing Macro(“SIP/4002-081e38f8”, “outisbusy|”) in new stack
– Executing Playback(“SIP/4002-081e38f8”, “all-circuits-busy-now|noanswer”) in new stack
– Playing ‘all-circuits-busy-now’ (language ‘en’)
< Protocol Discriminator Q.931 (8) len=5
< Call Ref len= 2 (reference 4/0x4) (Terminator)
< Message type RELEASE COMPLETE (90)
NEW_HANGUP DEBUG Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG Destroying the call, ourstate Null, peerstate Null
– Executing Playback(“SIP/4002-081e38f8”, “pls-try-call-later|noanswer”) in new stack
– Playing ‘pls-try-call-later’ (language ‘en’)
– Executing Macro(“SIP/4002-081e38f8”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/4002-081e38f8”, “w”) in new stack
– Executing NoCDR(“SIP/4002-081e38f8”, “”) in new stack
– Executing Wait(“SIP/4002-081e38f8”, “5”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/4002-081e38f8’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/4002-081e38f8’ in macro ‘outisbusy’
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/4002-081e38f8’

Any more ideas?

Regards,
Dave.