sip.conf

[general
port = 5060
bindaddr = xxx.xxx.xx.xx
fromdomain=xxx.xxx.xx.xx
callerid = yyy
userid = yyy
fromuser = yyy
insecure = very
qualify = no
regexten = yyy
canreinvite=no

extensions.conf
exten => s,2,Dial(SIP/{EXTEN}@196.41.5.74:5060) ...very wierd...but it works! If the port is not included it does not work & if there is no text before the @ it does not work.

Contact me if you need help with this. Michael Toop: michael{at}bizcall.co.za

Some hints and a sample. There are 3 files we need to edit on the Asterisk,

/etc/asterisk/sip.conf
/etc/asterisk/extensions.conf
/etc/asterisk/rtp.conf



The C5 setup is a typical SIP client Configure the endpoint and phone as you normally would


or typical SIP Gateway Configure the Gateway as any other gateway would be configured in the Clarent Assist


Sip Client / SIP gateway C5 – Asterisk Example


In this example a “sip UA client”

a phone number of 6386782250 an
ID of 6386782250@65.166.180.132 with
a password of 6386782250

The C5 in this example is located at 65.166.180.132

The C5 address and the endpoint ID Host does not have to be the same, but in this case are.


/etc/asterisk/sip.config
register = 6386782250:6386782250:6386782250@65.166.180.132


; with a SBA installed and a SIP Gateway 
;register = 6386782250:6386782250:6386782250@65.166.180.132:55061
; Hint
;register = phonenumber:username:password@hostC5:port  



"8578"
type=friend
        ;Incoming  Outgoing call requirements ....
        ;friend 2 way..... 
        ;peer incoming
        ;user outgoing
        ;

context=sip ; context in extensions.conf to use, may be default or any other extensions context
secret=6386782250
username=6386782250        ;The C5 Network ID
        ;Auth User

fromuser=6386782250        ;The Phone number associated with the C5 ID
insecure=very              ;relax Asterisk authorization for incoming calls
host=65.166.180.132        ;the address of the C5
fromdomain=65.166.180.132  ;The Address of the same as The ???@domain in C5 ID Setup
canreinvite=no             ;use asterisk as the Media Gateway Keep Asterisk in the call
                           ;codecs may be stated as the following
                           ;any endpoint codec would be acceptable here
disallow=all               ;start to define Codecs
allow=ulaw                 ;C5 Default codec
allow=alaw                 ;Another good one to add






/etc/asterisk/extension.conf
;Sample entry
exten = 8578,1,Dial(SIP/8578/16783669)

so in the above example we are saying when someone

dial's extension 8578 the first thing I want to do is dial using the

sip Context "8578" (see above the 8578 context) and the digits I want
to send are 16783669, could also be a variable or any digits that are routed by the C5.

8578 is the Extension number to dial and

8578 happens to also be the sip context (See sip.conf)




/etc/asterisk/rtp.conf
general
;default configuration is
;
; RTP start and RTP end configure start and end addresses
;
;rtpstart=10000
;rtpend=20000
;
;we want to change it to
;
rtpstart=30000 ;C5 RTP Ports
rtpend=65535

Verso Technologies Inc will support the connection of Asterisk to the Verso Softswitch for Customers

This sounds great but how can I obtain the Verso Clarent Class 5 Call Manager? Is it open source?