[Help] sip.conf parameter

I am using asterisk 1.2.0 with the Xlite softphone over linux system
I am able to build the connection and the phone are able to hear and talk

I have certain issues with some of the sip.conf file parameters
shown below is one of my context in sip.conf
callerid=“rama2” <2347>
canreinvite=no ; always go through Asterisk
allow=gsm ; add whatever other codecs we fancy

Q1)for this context i can only dial to this phone using digits 1231
so what is the use of the parameter “regexten=2345”

If you read the sample sip.conf that comes with 1.2, you’ll find out! It’s explained in that file.

Dang! I am not going to switch to 1.2 for a while, but I was curious too.

I understand why you sent him RTFM, but consider the poor lurkers reading the forum for random bits of wisdom :wink:

Oh, ok. I was just trying to discourage laziness, but seeing as you’ve got a reasonable (but not very convincing :wink: ) excuse, here you are. These are the relevant bits:

; If regcontext is specified, Asterisk will dynamically
; create and destroy a NoOp priority 1 extension for a given
; peer who registers or unregisters with us. The actual extension
; is the ‘regexten’ parameter of the registering peer or its
; name if ‘regexten’ is not provided. More than one regexten may be supplied
; if they are separated by ‘&’. Patterns may be used in regexten.

; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
[color=red];regexten=1234 ; When they register, create extension 1234[/color]
;callerid=“Jane Smith” <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;allow=gsm ; GSM consumes far less bandwidth than ulaw