[VoiceOps] SIP phone without using local dial plan

[Mark R Lindsey]

Normally, the SIP endpoint that device that collects digits is responsible for knowing when a correct and complete number pattern has been dialed. The same is true for MGCP devices when configured to  match a digitmap; they only send off the NTFY when a pattern of digits has been dialed, or else a timeout has occurred.

I'm almost certain that there is no way in the Cisco 7940/7960 SIP firmware to override this norm. You probably need to configure the digit map in the phone to match the digit map from your switch.

However, some platforms will let you do something similar: when you pick up the phone, you can have the CPE automatically INVITE to a specific SIP uri, like
	sip:switch at telco.com
Then the switch could accept that media stream, play back the local dialtone, and collect digits itself. A similar method is used on platforms like BroadWorks and Metaswitch to provide the "second dial tone" some users expect to hear after dialing 9 "for an outside line". I can't recommend for this approach, under normal circumstances. There are a lot of advantages in having the SIP endpoint construct a proper SIP INVITE using the digits dialed, with a sip...;user=phone  URI.

mark at ecg.co  |  +1-229-316-0013  |  http://ecg.co/lindsey



On Sep 23, 2011, at 09:21 , Feby Francis wrote:

> 
> Does anyone know any parameter available on Cisco 7940/7960  which start using the switch dial plan instead of phone specific dial plan ? The protocol used here is SIP.
> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/voiceops/attachments/20110923/bc58512e/attachment-0001.html>