UPDATE:
I recompiled the chan_ooh323 module and still receive the same error. I
also uncommented out the section in the oq931.c file but I could not get
it to compile afterwards.
Here my notes:
yum install -y gcc-g++ gcc-c* gnutls-devel kernel-devel
pwlib* openssl* expat* flex* ncurs*
cd /usr/src
wget
http://downloads.sourceforge.net/project/ooh323c/ooh323c/0.9.0/ooh323c-0
.9.0.tar.gz?use_mirror=softlayer
tar zxvf ooh323c-0.9.0.tar.gz
cd /usr/src/ooh323c-0.9.0
./configure --prefix=/usr/lib/asterisk/modules
make
make check
make install
make clean
cd /usr/lib/asterisk/modules/
mv chan_ooh323.so chan_ooh323.backup2
cd /usr/lib/asterisk/modules/modules
mv chan_ooh323.so ../
asterisk -rvvvv
reload
core show channeltypes
Verify:
OOH323 Objective Systems H323
Channel Driver no yes
LOGS:
13:42:52 denial event 1218: Invalid IE contents D1=0x830033 D2=0x264
From: Mark Best [mailto:markb...@co.nezperce.id.us]
Sent: Tuesday, September 08, 2009 11:59 AM
To: ooh323c-devel@lists.sourceforge.net
Subject: [ooh323c-devel] No Outbound PRI/ISDN (Invalid Information
Element)
SUPPLIMENTAL:
My Asterisk connects using H.323 (attempted asterisk-addons-1.4.7-2 and
asterisk16-addons-ooh323-1.6.0.2-3) Trunks to Avaya S8300
(S8300-015-01.2.416.4).
SIP Phone<---->Asterisk<---->Avaya<---->ISDN/PRI
BACKGROUND:
-I have 4 digit dialing and bi-directional communication between all my
Avaya (H.323) phones and my Asterisk (SIP) Phones.
SIP Phone<---->Asterisk<---->Avaya<---->H.323 Phone
-My Avaya phones successfully dial '9' and get out via the PRI/ISDN
line.
H.323 Phone<---->Avaya<---->ISDN/PRI<---->Cell Phone
PROBLEM:
The Asterisk box seems to get a generic 'Information Element' (IE)
error, when trying to dial-out to the PRI/ISDN line.
SIP Phone<---->Asterisk<---->Avaya ???
LOGS:
list trace tac 851
Page 1
LIST TRACE
time data
#H.323 TRUNK = Trunk 51
15:54:59 Calling party trunk-group 51 member 1 cid 0x76
15:54:59 Calling Number & Name 4909 4909
#H.323 TRUNK = Trunk 51
15:54:59 active trunk-group 51 member 1 cid 0x76
#Dial 9 to get out.
15:54:59 dial 9X route:HNPA|ARS
15:54:59 term trunk-group 20 cid 0x76
15:54:59 dial 9XXX2054 route:HNPA|ARS
15:54:59 route-pattern 2 preference 1 cid 0x76
#ISDN TRUNK = Trunk 20
15:54:59 seize trunk-group 20 member 14 cid 0x76
15:54:59 Setup digits XXX2054
15:54:59 Calling Number & Name NO-CPNumber NO-CPName
15:54:59 denial event 1218: Invalid IE contents D1=0x830033 D2=0x264
15:54:59 idle trunk-group 20 member 14 cid 0x76
Here is my Trunk Group on the Avaya (Possibly the important information
being Q-SIG)
--------------------------------------------------------------------
display trunk-group tac 851 Page 1
of 21
TRUNK GROUP
Group Number: 51 Group Type: isdn CDR
Reports: y
Group Name: Asterisk COR: 7 TN: 1
TAC: 851
Direction: two-way Outgoing Display? y Carrier Medium:
H.323
Dial Access? y Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method:
manual
--------------------------------------------------------------------
display trunk-group tac 851 Page 2
of 21
Group Type: isdn
TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs: 6
Charge Advice: none
Supplementary Service Protocol: b Digit Handling (in/out):
enbloc/enbloc
Digital Loss Group:
18
Incoming Calling Number - Delete: Insert: Format:
Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0
CONNECT Reliable When Call Leaves
ISDN? n
--------------------------------------------------------------------
And the Signaling-group. (Again, tried Q-SIG and AT&T Service Protocols)
display signaling-group 51
SIGNALING GROUP
Group Number: 51 Group Type: h.323
Remote Office? n Max number of NCA
TSC: 0
SBS? n Max number of CA
TSC: 0
IP Video? n Trunk Group for NCA
TSC:
Trunk Group for Channel Selection: 51
TSC Supplementary Service Protocol: b
T303 Timer(sec): 10
Near-end Node Name: procr Far-end Node Name: asterisk
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 4
LRQ Required? n Calls Share IP Signaling
Connection? n
RRQ Required? n
Bypass If IP Threshold
Exceeded? n
H.235 Annex H
Required? n
DTMF over IP: in-band Direct IP-IP Audio
Connections? n
Link Loss Delay Timer(sec): 90 IP Audio
Hairpinning? n
Enable Layer 3 Test? n Interworking Message:
PROGress
DCP/Analog Bearer Capability:
3.1kHz
Here is Asterisk view of the error
;/var/log/asterisk/h323
15:31:27:686 H.225 Call Proceeding message received (outgoing,
ooh323c_o_1)
15:31:27:756 Receiving H.2250 message (outgoing, ooh323c_o_1)
15:31:27:756 Received Q.931 message: (outgoing, ooh323c_o_1)
15:31:27:756 Received H.2250 Message = {
15:31:27:756 protocolDiscriminator = 8
15:31:27:756 callReference = 11
15:31:27:756 from = destination
15:31:27:756 messageType = 5a
15:31:27:756 Cause IE = {
15:31:27:756 Q931NormalUnspecified
15:31:27:756 }
15:31:27:756 h323_uu_pdu = {
15:31:27:756 h323_message_body = {
15:31:27:757 releaseComplete = {
15:31:27:757 protocolIdentifier = {
15:31:27:757 {
15:31:27:757 0 0 8 2250 0 5 }
15:31:27:757 }
15:31:27:757 callIdentifier = {
15:31:27:757 guid = {
15:31:27:757 '6f6f68333233632d029fa3ffffffff6a'H
15:31:27:757 }
15:31:27:757 }
15:31:27:757 }
15:31:27:757 }
15:31:27:757 h245Tunneling = {
15:31:27:757 FALSE
15:31:27:758 }
15:31:27:758 }
And Asterisk's config file (again tried several settings).
;/etc/asterisk/ooh323.conf
;START OF FILE
[general]
port=1720
bindaddr=192.XXX.XXX.252 ; Trixbox's IP
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
disallow=all
allow=ulaw
;dtmfmode=rfc2833
dtmfmode=inband
context=from-internal
e164=100
h323id=ObjSysAsterisk
callerid="Asterisk PBX"
; Add PROGRESS information element to SETUP message sent on outbound
calls
; to notify about required backward voice path. Valid values are:
; 0 - don't add PROGRESS information element (default);
; 1 - call is not end-end ISDN, further call progress information can
; possibly be available in-band;
; 3 - origination address is non-ISDN (Cisco accepts this value only);
; 8 - in-band information or an appropriate pattern is now available;
progress_setup = 8
; Add PROGRESS information element (IE) to ALERT message sent on
incoming
; calls to notify about required backwared voice path. Valid values are:
; 0 - don't add PROGRESS IE (default);
; 8 - in-band information or an appropriate pattern is now available;
progress_alert = 8
[Avaya]
type=friend
context=from-internal
host=10.XXX.XXX.200 ; IP Address of AVAYA
port=1720
disallow=all
allow=ulaw
canreinvite=no
; END OF FILE
(Some people suggest changing the ooq931.c file to permit
'Q931TransferSpeech'. But didn't explain how they got that logical
leap.)
http://www.trixbox.org/forums/trixbox-forums/h-323/avaya-ipoffice-pri-ou
tbound-and-ooh323-q931
I've asked several times in the IRC channels - and on a few forums to
very limited responses.
Can any provide further insight? Perhaps help me get more information
about the 'Q931NormalUnspecified'?
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