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@bayou.com>
> Date: Mon, 28 Mar 2005 19:46:01 -0600
> To: <jwm@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@interlinc.net>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@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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>>
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>>
>