[asterisk-users] SIP Extension Config Issue
Eric "ManxPower" Wieling
eric at fnords.org
Mon Sep 8 15:03:54 CDT 2008
You're joking, right?
exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr)
exten => _1xxxxxxxxxx,n,Voicemail(mailbox at vmcontext)
Use whatever voice mailbox and voicemail context you want.
Joseph L. Casale wrote:
> I have a setup with a SIP DID inbound, and several SIP phones inside.
> Obviously if the SIP phones are off/unplugged/otherwise not available,
> incoming calls ring busy. My extensions.conf looks like this for inbound
> calls:
>
> exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr)
>
> So what could I do to send the call to voicemail if none of the extensions
> are online?
>
> Thanks!
> jlc
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
More information about the asterisk-users
mailing list