Alex,<br><br>I am new to <u><b>5350</b></u> my senerio is this;<br><br><b>1. ASTERISK ---outgoing------>CISCO5350 (both have live IP configured)<br><br>2. ASTERISK <-----incoming----CISCO5350</b><br><br>I need only configurations for Cisco for both in coming n outgoing to asterisk. IF you need configuration of my Cisco Gateway I will provide you. Sorry to bother you again. I have to make up assignment on it hope you help me out.<br>
<br>Atif Shahzad.<br><br><div class="gmail_quote">On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="Ih2E3d">A T I F wrote:<br>
</div><div class="Ih2E3d">> 1. dial-peer voice 500 voip<br>
><br>
> I use this configuration for inbound to asterisk.<br>
><br>
> 2. dial-peer voice 510 pots<br>
> description Fancy PRI - Outgoing<br>
> huntstop<br>
> destination-pattern .T<br>
> direct-inward-dial<br>
> forward-digits 10<br>
><br>
> And use this configuration for outbound from asterisk to Cisco 5350 right?<br>
<br>
</div>Yep.<br>
<br>
You may wish to have an incoming peer on the VoIP side to match first to<br>
do various translations in the future. It's generally considered better<br>
form. Then the call will enter in this dial peer and exit in 510.<br>
<br>
dial-peer voice 801 voip<br>
description Asterisk - inbound<br>
<div class="Ih2E3d"> voice-class codec 1<br>
session protocol sipv2<br>
session target ipv4:ip.addr.of.asterisk<br>
session transport udp<br>
</div> incoming called-number .T<br>
<div class="Ih2E3d"> dtmf-relay rtp-nte<br>
no vad<br>
<br>
<br>
</div>--<br>
<div><div></div><div class="Wj3C7c">Alex Balashov<br>
Evariste Systems<br>
Web : <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
Tel : (+1) (678) 954-0670<br>
Direct : (+1) (678) 954-0671<br>
Mobile : (+1) (706) 338-8599<br>
<br>
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