Why would you use anything other than what's in the sip.conf file? You can now configure (though I have not tried it myself) usernames with "@" symbols in them. I am having a lot of difficulty parsing your question. Just give people usernames of "[EMAIL PROTECTED]" then and map them to some meta-extension, and set it up that way in sip.conf, like this:

[1234]
type=friend
[EMAIL PROTECTED]
secret=jtoddspassword
host=dynamic


Can asterisk act as a SIP registrar or location server?
I would like to be able for a user agent(client) to register with
whatever client they are using as "[EMAIL PROTECTED]". Rather
than the entry/username/password that is setup in the sip.conf file.
That way a user could log into any SIP enable client and their calls
would follow them around.
I have read the sip.conf man pages and have noticed the following
commented lines in sip.conf:

;
;register => [EMAIL PROTECTED]     ; Register with a SIP provider
;register => [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider
as 1234 here.

This is for registering to remote servers for calls inbound to Asterisk.


To explain: I have an account on my company's SIP proxy/gateway system at work. They gave me a username of "[EMAIL PROTECTED]" and password of "deadrabbit". I want to use Asterisk to route my calls from Bigcompany's SIP server into one of my SIP phones. I have created an extension in context "from-sip" called 2203 which takes inbound calls and routes them to my SIP phone.

So, in sip.conf I put the following config in the [general] context:

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = from-sip              ; Default for incoming calls
[EMAIL PROTECTED]:[EMAIL PROTECTED]/2203


; here is the definition for my ATA-186 SIP phone
; (note: this config is abbreviated; does not include outbound config for ATA-186)
[2203]
type=friend
username=2203
secret=passwordforata186
host=dynamic
canreinvite=no



Now, in extensions.conf I put something like this:


[from-sip]
exten => 2203,1,Dial(SIP/2203)


That's it! When I fire up Asterisk, it sends a REGISTER to sipserver.bigcompany.com and notifies that server that my Asterisk server is accepting calls for my account. Then, when calls come in from that account. Then, when calls come in from that account, Asterisk points them at extension 2203, which in turn dials my ATA-186.


For a more exhaustive example, see http://www.loligo.com/asterisk/

JT




But I am a little confused how this should be implemented.


--
Steve Woolley
ADS Telecom, Inc.
59 Skyline Drive #1250
Lake Mary, FL  32746
Phone: (407) 682-6226 x1110
Fax: (407) 682-3455
[EMAIL PROTECTED]


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