[Asterisk-Dev] SIP Authentication problem between Cisco router and Asterisk when calls are forwarded

John Lange john.lange at open-it.ca
Fri Jun 10 12:03:49 MST 2005

I've heard this mentioned before but I don't really understand how it
makes any sense in terms of building a reasonably efficient dial plan.

We have a large block of DIDs assigned to us so this is our dial plan:

exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail2(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Hangup()
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1) ; everything else is treated as no

exten => _20422233XX,1,Macro(stdexten,SIP/${EXTEN})

As you can see, we can handle hundreds (or even thousands) of DIDs in
only a few short lines of extensions.conf. If we started assigning
device names like "joesmith" we would have to define every single

Am I missing something?

John Lange
President OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Server Co-Location

On Fri, 2005-06-10 at 20:53 +0200, Olle E. Johansson wrote:
> The real solution is *not* to mix extensions in the dial plan (phone
> numbers) with device names. That will always lead to some sort of
> problem. In some cases, it also will confuse the manager (me or you).
> You need different name spaces for extensins and device names (peers/users).
> Extensions are what we place calls to or from (phone numbers).
> Peers/users are devices with certain permissions to use the pbx.
> Good luck!
> /Olle
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev

More information about the asterisk-dev mailing list