Ya i recommend you DIY because sometime is not fully function on your
country (my country is under E1 711a)
My equipment is M1k but i think is same configure with you.
1. import the ini file from Sipxecs
2. if you want to call out configure is under
- Protocol
Definition (sipxecs ini file include)
-
Manipulation
Tables
- Dest
Number IP->Tel (this is translate the number from sipxecs address to
audiocode)
-
Routing Tables (sipxecs ini file include)
-
Trunk/IP Group (sipxecs ini file include)
hope can help you
ronald teng wrote:
hmmm haven't traced via wireshark yet. For my dial plan, i
only used the default settings for local. I also put in under digit
map, 9xxxxxxx (7-digit dialling). Im only trying to implement local
calls. Do i have to put in my country code as well? For your case, you
did configure some settings in the audiocodes gateway manually then?
Coz I thought sipXconfig is supposed to do that automatically. hmmm...i
don't really know how to configure the manipulation table for
audiocodes but i guess i'll just google that if it's really necessary. .
On Wed, Mar 24, 2010 at 11:43 AM, Winson
(Elabram) <[email protected]>
wrote:
do you try to use wireshark
to trace you sipx to audiocodes? and what
is the number to call out (means receive call work?)
because u have to know you number is correct.
for example my dial plan on sipxecs is 6012XXXXXXX(8 digit)
my country is only under 012XXXXXXX
so your audiocode have to reduce the number ( Protocol
Configuration ->Manipulation
Tables->Destination
Phone Number Manipulation Table for IP -> Tel Calls)
hope can help you :-)
ronald teng wrote:
hi everyone,
Im new to voip tech so please bear with me. Im currently trying to
setup a system using SipXecs (centos), AudioCodes MP118 FXO firmware
ver 5.4 and Polycom phones. I am already able to make calls internally
but im having problems calling out to the PSTN. Based on my current
read (an ebook published by packt for sipX), all configuration on the
gateway will be done using sipXconfig and i dont need to configure any
settings on the Audiocodes gateway itself. Already added the gateway in
sipX, added the local dial plan and enabled it but everytime i call
out, i just get a busy tone. Don't i need to configure some form of
dial plan like in Cisco (dial peers)? Any suggestions? Please help :(
Regards,
Ron
_______________________________________________
sipx-users mailing list [email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
sipXecs IP PBX -- http://www.sipfoundry.org/
|
_______________________________________________
sipx-users mailing list [email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
sipXecs IP PBX -- http://www.sipfoundry.org/
