I mean to say that I have assigned this task from my management to configure it, I am familiar with Asterisk but first time I am using Cisco 5350.<br><br><br><br><div class="gmail_quote">On Tue, Nov 25, 2008 at 4:19 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;">This is not the homework list.<br>
<div class="Ih2E3d"><br>
A T I F wrote:<br>
<br>
</div><div class="Ih2E3d">> Alex,<br>
><br>
> I am new to _*5350*_ my senerio is this;<br>
><br>
> *1. ASTERISK ---outgoing------>CISCO5350 (both have live IP configured)<br>
><br>
> 2. ASTERISK <-----incoming----CISCO5350*<br>
><br>
> I need only configurations for Cisco for both in coming n outgoing to<br>
> asterisk. IF you need configuration of my Cisco Gateway I will provide<br>
> you. Sorry to bother you again. I have to make up assignment on it hope<br>
> you help me out.<br>
><br>
> Atif Shahzad.<br>
><br>
> On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov<br>
</div><div><div></div><div class="Wj3C7c">> <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a> <mailto:<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>>> wrote:<br>
><br>
> A T I F wrote:<br>
> > 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<br>
> 5350 right?<br>
><br>
> 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>
> voice-class codec 1<br>
> session protocol sipv2<br>
> session target ipv4:ip.addr.of.asterisk<br>
> session transport udp<br>
> incoming called-number .T<br>
> dtmf-relay rtp-nte<br>
> no vad<br>
><br>
><br>
> --<br>
> 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>
> _______________________________________________<br>
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><br>
><br>
><br>
</div></div>> ------------------------------------------------------------------------<br>
<div class="Ih2E3d">><br>
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<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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</div></div></blockquote></div><br>