Hi Rameez,<br>
<br>
I would do it that way:<br><br><span class="e" id="q_117bada7a4bfb2b4_1">R1<br>dial-peer voice 200000 voip<br>
destination-patern 200000<br> session target ipv4:<a href="http://1.1.1.2/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">1.1.1.2</a><br><br>R2<br>dial-peer voice 200000 pots<br> destination-patern 2000T<br>
port 1/0:15<br><br>That will allow you to send all dialed digits to R2 in one block and then strip 2000 of before forwarding to the PBX.<br><br>Otherwise the PBX2 has to be able to handle DTMF for call setup and you will have to wait for the timeout or press # in order to setup the call to 200000 before you can dial the extra digits. Last but not least if the user dials the entire number before the setup to 20000 is done, the call will be set up to the PBX with only the last two digits dialled (e.g. 200000014085550123 will send 23 to the PBX).<br>
<br>Regards,<br>Patrick<br><br></span><br><div><span class="gmail_quote">2008/1/27, Ramiz Sardar <<a href="mailto:ramizchaudhary@gmail.com">ramizchaudhary@gmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Friends<br> I am still waiting for assistance...............<br><br>Regards<div><span class="e" id="q_117bada7a4bfb2b4_1"><br><br><div class="gmail_quote">On Jan 25, 2008 1:58 PM, Ramiz Sardar <<a href="mailto:ramizchaudhary@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">ramizchaudhary@gmail.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Sir,<br><br>I am working on voip setup. My topology you can see below:<br><br><br>User-------PBX1------R1----------------------R2-----PBX2---------PSTN<br>
<br>PBXs are connected with Routers and PSTN via E1 Pri whereas Routers R1 and R2 are connect with each other through E1 link. Lets suppose site-1 has site code 1000 and site-2 has code 2000 so when user dial 2000 and extension, it connect to user of site-2. In old topology without voip, when user dial 2000, then 00 and then pstn number, it able to call users on pstn. Old topology is :<br>
<br>User-------PBX1---------------------------------PBX2---------PSTN<br><br>But now through voip setup, can i do the same thing ? I think it should be possible with configuration sample below:<br><br>R1<br>dial-peer voice 200000 voip<br>
destination-patern 200000<br> session target ipv4:<a href="http://1.1.1.2" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">1.1.1.2</a><br><br>R2<br>dial-peer voice 200000 pots<br> destination-patern 200000<br>
port 1/0:15<br> forward-digit 2<br><br>I think using above configuration R2 will send 00 to PBX2 and an end-to-end connection will be establish between user of site-1 and PBX2 and after that user can dial whatever pstn number he wants. IF this will not work then i have to configure a number of dial-peers for pstn like 7 digits, 8 digits and with destination-patern "T" etc.<br>
<br>Please help me out regarding this issue. Am i thing in good way otherwise purpose me solution Waiting........<br><br>Best Regards<br>Rameez<br>
</blockquote></div><br>
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