[Asterisk-Users] CIC Code

[Jason Miller]

I guess I should have included more information to get assistance after
reading what I posted,

I am running MF not SS7, the local Telco does see me sending the appropriate
info but not in the correct protocol according to them. They say it looks
more like I am dialing the dial code and CIC code manually instead of
Asterisk sending it before the number dialed. So in essence asterisk needs
to send two packets, one with the 017CICCode and the second packet with the
phone number. Also I need to be sending ANI information as well. I am
stumped to what to do at this point and have tried everything I have found
on google and what Digium tech support suggested, if anyone could assist me
in this it would be greatly appreciated and if need be I would be willing to
pay for outsourced technical support if you have experience setting this up
since I need to get it up and going quickly instead of learning how to do
it. I can learn later... HEHE

Thanks again,
Jason

Red Hat Linux release 9 (Shrike)
Kernel \r on an \m
2 Each Digium T100X
(Snippets of my config files, some phone/server specific info changed for
post IE sip username/secret, CICCode and default IP)

-Extensions.conf-
[general]
static=yes
writeprotect=no

[globals]
CONSOLE => Console/dsp
IAXINFO => guest
TRUNK => Zap/g1
TRUNKMSD => 1
TRUNK2 => Zap/g2
TRUNKMSD => 2
TRUNK3 => Zap/g3
TRUNKMSD => 3


exten => _1NXXNXXXXXX,1,Dial(Zap/g2/017CICCode${EXTEN:1})
exten => _1NXXNXXXXXX,2,Hangup()

-zapata.conf-

context=default
usecallerid=yes
callwaiting=yes
immediate=no
group=1
echocancel=yes
signalling=em
channel => 1-24

context=twoway
usecallerid=yes
callwaiting=yes
immediate=no
group=2
echocancel=yes
signalling=featdmf
channel => 25-36

context=incoming
usecallerid=yes
immediate=no
group=3
echocancel=yes
signalling=featb
channel => 37-48

-Zaptel.conf-

# Zaptel Configuration File
span=1,1,0,esf,b8zs
e&m=1-24
defaultzone=us
span=2,1,0,esf,b8zs
e&m=25-48
loadzone=us
defaultzone=us


- Sip.conf -

[User]
username=User
secret=nothing
type=friend
host=dynamic
defaultip=1.1.1.1
dtmfmode=info
context=incoming ;twoway ;default
canreinvite=no
disallow=all
nat=yes
allow=ulaw
allow=alaw
mailbox=107

- Lsmod -
Module                  Size  Used by    Not tainted
soundcore               6404   0  (autoclean)
wct1xxp                13024  48
zaptel                179712  98  [wct1xxp]
autofs                 13268   0  (autoclean) (unused)
natsemi                19552   1
keybdev                 2944   0  (unused)
mousedev                5492   0  (unused)
hid                    22148   0  (unused)
input                   5856   0  [keybdev mousedev hid]
usb-uhci               26348   0  (unused)
usbcore                78784   1  [hid usb-uhci]
ext3                   70784   2
jbd                    51892   2  [ext3]





> From: Tom Chandler <tchandle at bayou.com>
> Date: Mon, 28 Mar 2005 19:46:01 -0600
> To: <jwm at interlinc.net>
> Subject: Fw: [Asterisk-Users] CIC Code
> 
> Jason,
> If you get any answers, I too would be interested.
> 
> I believe on terminating, the CIC is not sent, AMA recording uses the
> CIC assigned to the trunk group.  If in SS7, then the CIC is passed
> in the IAM message.
> 
> I have not worked on the originating side, so I can not help.
> 
> Thank You
> Tom Chandler
> 
> ----- Original Message -----
> From: "Jason Miller" <jwm at interlinc.net>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Monday, March 28, 2005 7:22 PM
> Subject: [Asterisk-Users] CIC Code
> 
> 
>> Has anyone ever setup Asterisk to pass Feature Group D access while using
> a
>> CIC code for outbound calls? If so can you please email the configuration
>> you have done? I have tried to get this up and running but with no luck. I
>> have also contacted support and I cant seem to get this going.
>> 
>> 
>> 
>> Thanks in Advance,
>> Jason Miller
>> 
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>