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 class="gmail_quote">On Tue, Nov 25, 2008 at 3:42 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;">My attention to my dial peer. It has nothing about H.323 and much about<br>
SIP.<br>
<div class="Ih2E3d"><br>
A T I F wrote:<br>
<br>
</div><div class="Ih2E3d">> Alex,<br>
><br>
> 1 more thing my gateway is configured with H.323 so tell me how can I<br>
> configure it with SIP?<br>
><br>
> On Tue, Nov 25, 2008 at 3:10 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>
> Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind<br>
> that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so<br>
> you can use the same destination pattern matching for both in this<br>
> simple scenario, but if it gets any more complicated than that, some<br>
> degree of translation is almost certainly required.<br>
><br>
> The process can be fairly complex, but the general idea, if you have<br>
> your TDM side set up, is:<br>
><br>
> dial-peer voice 500 voip<br>
> description Asterisk<br>
> destination-pattern .T<br>
> progress_ind setup enable 3<br>
> voice-class codec 1<br>
> session protocol sipv2<br>
> session target ipv4:ip.addr.of.asterisk<br>
> session transport udp<br>
> dtmf-relay rtp-nte<br>
> no vad<br>
><br>
> 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>
><br>
> A T I F wrote:<br>
><br>
> > Hello, everybody!<br>
> ><br>
> > I need help connecting my Cisco AS5350 to Asterisk.<br>
> ><br>
> > What i want to do is forward all outgoing calls from Asterisk<br>
> server to<br>
> > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.<br>
> ><br>
> > How could this be done?<br>
> ><br>
> > Thanks in advance<br>
> ><br>
> > Atif Shahzad<br>
> ><br>
> ><br>
> ><br>
> ------------------------------------------------------------------------<br>
> ><br>
> > _______________________________________________<br>
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> ><br>
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><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>
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> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
><br>
><br>
> ------------------------------------------------------------------------<br>
><br>
> _______________________________________________<br>
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><br>
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> To UNSUBSCRIBE or update options visit:<br>
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<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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</div></div></blockquote></div><br>