[cisco-voip] SIP Trunk

[Wes Sisk]

James,

It is quite possible.  Consider the SIP trunk like an 
interclustercluster trunk.

inbound calls arrive from the trunk, CM treats them with the CSS 
assigned to your SIP trunk definition.  You can configure Sig Digits to 
truncate the inbound digits back to the number of digits you use for 
enterprise dialing.

The only challenge might be MTP resources.  SIP normally uses DTMF 
piggybacked on the media stream (RTP stream) where as the rest of CM 
uses DTMF passed through the signaling session.  For this reason, you 
must check the box 'MTP Required' on the SIP trunk.  If your telco 
presents calls in G.711 the default software MTP resources with CM will 
work fine.  If your telco presents calls only using a low bit rate codec 
such as G.729, then you will need hardware DSP resources (6608 / CMM ACT 
card / IOS conferencing and transcoding using dspfarm).

/Wes

James O'Farrell wrote:
>
> We are currently looking at installing a Unified CalManager system. At 
> one of the sits we have to provision an E1, one option we have been 
> given is to use SIP Gateway instead of an e1, utilising our current 
> internet connection.
>
>  
>
> First of all is it possible to send and receive calls over a sip trunk 
> into a SCCP environment?
>
>  
>
> Second how would the DN be organised in this type of situation?
>
>  
>
> If anyone has any good links to guides/articles I would appreciate it.
>
>  
>
> Regards
>
> James
>
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