;# sip.conf

[gismotel-out]
type=friend
username=7XXXXX ;# 7XXXXX Number you received while registering.
fromuser=7XXXXX ;# Same number as above. Without fromuser field it will not work!
secret=******* ;# Put here the password you received while registering.
host=sip.gismotel.com
port=6060
insecure=very
disallow=all
allow=g723
allow=g729 ; Gismotel allows g723 and g729 over SIP protocol.

Now register your SIP endpoint with GismoTel like:

Format:
register => user[:secret[:authuser]]@host[:port][/extension]

In our case an example using this variables would be:
register => 70XXXX:password@sip.gismotel.com/1234 ; Register 70XXXX with sip.gismotel.com as 1234 here.

* user is the user id for this SIP server (ex 70XXXX)
* authuser is the optional authorization user for the SIP server
* secret is the user's password
* host is the domain or host name for the SIP server. This SIP server needs a definition in a section of its own in SIP.conf (mysipprovider.com).
* port send the register request to this port at host. Defaults to 5060
* /1234 is the Asterisk contact extension. 1234 is put into the contact header in the SIP Register message. The contact extension is used by remote SIP server when it needs to send a call to Asterisk. The default context extension is "s".

Now adjust your dial plan so outgoing PSTN calls follow the correct context.

;# example in extensions.conf

[gismotel]
exten => _0.,1,SetCIDNum(7XXXXX) ;# Put your registation number here
exten => _0.,2,Dial(SIP/0${EXTEN:1}@gismotel-out,60)
exten => _0.,3,Hangup()

Inbound Calling

With inbound calling your endpoints can be called from our SIP infrastructure.
In sip.conf in the general section you need to register your device.

register => 7XXXXX:secret@sip.gismotel.com/1234
Any inbound call to 7XXXXXX will be forwarded to 1234 at your Asterisk box.

Adjust your Dial Plan to route the incoming calls to their proper destination endpoint.

Please send an email to GismoTel to inform us about your Inbound plans and which DID's you would like to order if you want your Asterisk server to be reachable from the normal telephone network.

DID or Virtual Numbers

GismoTel can provide you with a phone number in your region that when people call to this number it will end up on your Asterisk box.