sip.conf is confusing me

[computergroove]

I have 4 agent machines and 1 vicidialnow predictive outbound server. I am using soft phones with X-Lite. I can register the X-Lite clients with the data that my VOIP provider sent me and all goes well.

1. Do I need to setup my VOIP service in X-Lite on my client machines with the info I got from my VOIP provider or do I need to modify the sip.conf users and register the X-Lite clients with the sip.conf user names?

2. What tags do I need to add to the client accounts in sip.conf so I can connect to the asterisk server via X-Lite on the agent machines?

3. After I make changes to the sip.conf and extensions.conf do I need to restart asterisk? If so what is the command from the console or should I just restart the server?

Thanks

[devafree]

Hello

This is in response to this post and your previous one too.

1. You need to first register with the VOIP provider thru the settings in sip.conf like this

register => <your VOIP username>:<your VOIP secret>@<your VOIP providers SIP server domain or IP address>

There are other option too, you can read about them in the sample sip.conf (if you did a "make samples" while installing asterisk.

Re-connect to Asterisk with the command "asterisk -r", and in the Asterisk CLI, do "sip reload". Then in the Asterisk CLI "sip show registry" should tell you you have registered with the VOIP provider. Type exit to quit the Asterisk CLI to go the Linux shell prompt again.

2. Then , configure your sip users, you can follow the scratch install from here and then point your x-lite phones to register with asterisk. You see the users registered realtime in the Asterisk CLI, or use "sip show peers"
For Eg I have used a user and phone 100(username and phone number arethe same in my case) like below in sip.conf, in a scenario where there is no Nat. Also, for a full example set please see, the example conf files in the astguiclient src directory.

[authentication]
[your-voip-provider]
type=peer
host=your-voip-provider-IP-addr-or-domain
username=VOIP-provider-given-username
secret=VOIP-provider-secret
fromuser=mostly-VOIP-provider-given-username
fromdomain=your-voip-provider-IP-addr-or-domain
context=default
insecure=very
canreinvite=no
disallow=all
;allow=g723.1
allow=g729 ;;
dtmfmode=rfc2833 ;; for codecs and dtmf mode, as per your VOIP
;; provider -
;; the above lines generally works with many
;; VOIP providers - you may need to change
;;these settings to ensure call go out

[100]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1002
secret=1234
host=dynamic
dtmfmode=rfc2833 ;; put what your VOIP provider tells you to here
canreinvite=no

In x-lite , I give the display name, user name etc as 100 and secret as 1234, domain as <my asterisk server IP addr> and enable "register with domain and receive incoming calls from target domain"

And I am ready to dial ,out manually from x-lite, registered as 100 with the Asterisk box, provided I have the valid extension for calls in extensions.conf.

in [default] in extensions.conf I have
; dial a long distance outbound number to USA thro the VOIP Provider
;; the @your-voip-provider is the authentication head you have under [authentication] in sip.conf
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:2}@your-voip-provider,55,tTo)
exten => _91NXXNXXXXXX,3,Hangup


3. You need to add the user name and phone number , in my case both were 100 , in the vicidial Admin. The the agent can login at the agc/vicidial.php page and start calls.

4. Any changes to sip.conf and extensions.conf - no need to re-start asterisk or the server , though these will work too. You simple type "sip reload" or "extensions reload" in the Asterisk CLI to re-load.

Good luck , hope its of use to you.

Regards

devafree