asterisk users - Feb 2005 - Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite....

Mateo,

Dialing the extension to your softphone is the same as any hardware
extension.

Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty  
exten => 1000,2,Macro(vmessage,1000)
exten => 1000,3,Hangup

Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the specifics you are using.

Update the settings in your softphone to register the name and number as
1000

Then any attempt to dial 1000 should come to that phone.

Wiley


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Christian
Stredicke
Sent: Monday, February 28, 2005 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone ?
tryedtoforwarda call to X-Lite....

Try the snom soft phone! http://snom.com CS
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave 
> Chase
> Sent: Saturday, February 26, 2005 12:31 PM
> To: ich@mateo.ch; Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed 
> toforwarda call to X-Lite....
> 
> XLite does not support transfer... You have to buy their XPro
> 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo 
> Meier
> Sent: Tuesday, February 22, 2005 3:50 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? 
> tryed to forwarda call to X-Lite....
> 
> Hey Guys
> 
> Im trying to forward a call from the asterisk mainmenue to my second 
> computer with X-Lite installed..
> 
> What I've done so far is this:
> 
> Installed X-lite @my win PC.. 
> 
> X-Lite configuration: 
> Menu | System Settings | SIP Proxy | default Display Name: 
> mateo01 User Name & Authorization User: mateo01
> Password: ****
> Domain/Realm: 192.168.1.**
> SIP Proxy: 192.168.1.**
> 
> 192.168.1.** = IP address of Asterisk
> 
> and the sip.conf file looks like that:
> 
> [mateo01]
> type=friend
> username=mateo01
> callerid="mateo01" <1234>
> host=dynamic
> secret=****
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> context=sip
> nat=no
> 
> 
> Now, Im unsure what to do ? whats next ? and what do I type in to 
> extensions.conf  instead of the following:
> 
> exten=>2,1,Dial(capi/720****:078*******)
> 
> Thx for the help
> Mateo
> 
> 
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> Change [mateo01] to [1000] in your sip and you will be saying that ext.
> 1000 is registered with the specifics you are using.
> 
> Update the settings in your softphone to register the name and number as
> 1000
> 
> Then any attempt to dial 1000 should come to that phone.
> 
> Wiley
After doing thoses changes, you can also simplify your dialplan with
something like this:

exten => _1XXX,1,Dial(SIP/${EXTEN},20,tr)
exten => _1XXX,2,Voicemail(u${EXTEN})
exten => _1XXX,3,Hangup
exten => _1XXX,102,Voicemail(b${EXTEN})
exten => _1XXX,103,Hangup

This will match any extension dialed in the 1000-1999 range so you
don't have to write new rules each time you add a phone

hth