[Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI

Sébastien Mortier smortier at absystech.fr
Mon Mar 20 03:51:48 MST 2006


Hello,

I recently bought a Junghanns Octobri Card. I have some problems with 
this card to make outbound calls but I can receive calls.

I have 3 lines to PSTN and 3 lines to my existing PBX

   FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h 
<-- OctoBRI --> PABX e-Generis <----> ISDN Phones
                                           |
                                           |
                                      SIP Phones


France Telecom --> SIP Phones : Works
France Telecom --> ISDN Phones : Works
SIP Phones ------> ISDN Phones : Works
ISDN Phones -----> SIP Phones : Works
SIP Phones ------> France Telecom : DOESN'T WORK
ISDN Phones -----> France Telecom : DOESN'T WORK


Here are some characteristics of my Asterisk Setup

OS Linux Gentoo 2.6.15-r1

zaptel 1.2.3
libpri 1.2.2
asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h
ISDN Lines : EuroISDN not EuroISDN+

Junghanns OctoBRI PCI ISDN Card
S/T 1+8 - S/T 2+7 : TE Mode
S/T 3+6 - S/T 4+5 : NT Mode
modprobe qozap ports=60


zaptel.conf
-----------


loadzone=fr
defaultzone=fr
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,1,3,ccs,ami
span=6,0,3,ccs,ami
span=7,0,3,ccs,ami
span=8,0,3,ccs,ami

bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
bchan=13,14
dchan=15
bchan=16,17
dchan=18
bchan=19,20
dchan=21
bchan=22,23
dchan=24



-----------
zapata.conf
-----------

switchtype = euroisdn
pridialplan = dynamic
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
usecallingpres = yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = 100
callprogress=yes


context=isdn-incoming
group = 1

; S/T port 1,2,7,8
channel => 1-2
channel => 4-5
;channel => 19-20
channel => 22-23

context=pbx-incoming
group = 2

channel => 7-8
channel => 10-11
;channel => 13-14
channel => 16-17


---------
Here's the output BRI debug when I try to make outbound calls from a SIP 
phone :



-- Executing Dial("SIP/400-c8dc", "Zap/1/1013")
1 -- Making new call for cr 137
-- Requested transfer capability: 0x00 - SPEECH
1 > Protocol Discriminator: Q.931 (Cool len=26
1 > Call Ref: len= 1 (reference 9/0x9) (Originator)
1 > Message type: SETUP (5)
1 > [1 041 031 801 901 a31 ]
1 > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer 
capability: Speech (0)
1 > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
1 > Ext: 1 User information layer 1: A-Law (35)
1 > [1 181 011 811 ]
1 > Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, 
Preferred Dchan: 0
1 > ChanSel: B1 channel
1 ]
1 > [1 6c1 051 411 801 341 301 301 ]
1 > Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1 > Presentation: Presentation permitted, user number not screened (0) 
'400' ]
1 > [1 701 051 c11 311 301 311 331 ]
1 > Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ]
-- Called 1/1013
1 < Protocol Discriminator: Q.931 (Cool len=8
1 < Call Ref: len= 1 (reference 137/0x89) (Terminator)
1 < Message type: RELEASE COMPLETE (90)
1 < [1 081 021 871 e41 ]
1 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 
Location: International network (7)
1 < Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ]
1 -- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup, cause 100
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/400-c8dc", "")
== Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc'
1 received TEI check request for TEI = 127


I've already tested several configurations for zapata.conf especially 
with the pridialplan and switchtype lines but without success.

Could you help me to analyse and solve this odd problem ?
Thank you in advance,


-- 
Sébastien Mortier
AbsysTech
Tel : +33 3 20 50 99 02
Fax : +33 3 20 74 50 05
Gsm : +33 6 20 79 24 29

http://www.absystech.fr









More information about the asterisk-users mailing list