[Serusers] Re: configuring voice port....

[Steve Blair]

gunawan wrote:

>Hi, Steve.....
>
>I want to ask about dial-peers u provide in ur
>website(http://mit.edu/sip/sip.edu/ciscoGW.html)...
>
>  
>
Just for clarification this isn't my site. This is where some of the 
documents associated
with the Internet2 working group of which I am a member.

>1. dial-peer voice 680010 voip
>                description Only peer for inbound to
>SIP Proxy 215-746-8001:8009 extensions
>                huntstop
>                preference 2 
>                destination-pattern 6800[1-9]
>                progress_ind setup enable 3
>                voice-class codec 1
>                session protocol sipv2
>                session target sip-server
>                dtmf-relay rtp-nte
>                no vad
>
>what is 680010?????? is that ur phone number???? 
>something related to phone number???
>  
>
No it is not my phon enumber. This is an example only. The value following
dial-peer voice is just a label to uniquely identify that dial-peer. The 
value
specified in the destination-pattern parameter is what is matched 
against the
dialed digits. In this example the 6800[1-9]  specifies a range of numbers
from 68001 through 68009. These are real Centrex extensions within our
numbering plan bu tthey are test numbers reserved for use by our VoIP 
project.
We use 5 digits because that is how many digits our Centrex provider hands
us in the call setup message.

>2. dial-peer voice 61 pots
>   description Only peer for outbound 5-digit 746
>campus calls
>            translation-profile outgoing Prefix
>                 preference 3
>                 destination-pattern 6....
>                 direct-inward-dial
>                 port 1/0:23                  
>                 prefix 215746          
>
>why do u use 61???? something related to ur pone
>number????
>
>  
>
Same as above. This is an illustration. The 61 is just a label . In this 
example
61 is the first dial-peer defined to handle outbound calls to Centrex 
extensions
6xxxx.

>3. could u give me some other example configuration,
>bcoz I dun use PABX or analog router here.. I plug in
>my Telephone line direct to VIC2FXO card...
>so I Wish that my SIP client can call to PSTN
>client...
>my Telephone number that I plug to cisco router is
>62(761) 53808 , 62 is country code, 761 is area kode,
>53808 is my phone number...
>
>  
>
The FXS and FXO peers are similar to those mentioned in the document. You
may need to fiddle with parameters for your specific type of connection.
You may need to know if you are using wink-start or loop-start signaling,
what port to use, etc. A very basic setup would include something like:

voice-port 1/0/0
 input gain 10
 connection plar opx 32766
 description FXO 1/0/0 5732767 for VoIP

dial-peer voice 91 pots
 destination-pattern 9
 port 1/0/0
 prefix 9
!

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-- 
  
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  


voice: 215-573-8396 

       215-746-8001

fax: 215-898-9348    

sip:blairs at upenn.edu