Matthews,<br>Thanks a bunch, I am most grateful. I am having a clearer view now, but have issues with the following to wrap up.<br>- About the DID, are you saying I should remove it from the following config, because that the only place I included it.<br>
<br>dial-peer voice 1 pots<br>
destination-pattern 12345<br> direct-inward-dial<br> port 0/0/0<br><br>- How do I know the protocol I am using by default, since there is no call manager.<br><br>- Once I place a call to the PSTN number to be routed to the extension, how do I monitor the progress of the call? debug ???<br>
<br>- How do I add bound ports, voice service voip config, and what do they add in this situation.<br><br>Thanks as I await your reply.<br><br clear="all"><b><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Cheers,</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);">
<span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Idowu Kehinde</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);"><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">TEL: 2348024017748</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);">
<span style="font-family: georgia,serif; color: rgb(0, 0, 102);">BB: 30E8E5F9</span><br></b><br> <br>
<br><br><div class="gmail_quote">On Tue, Feb 1, 2011 at 6:06 AM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
You don't need direct-inward-dial on FXO ports, only PRI ports. Other than that, the config looks alright.<br><br>If you've got other config such as bound ports, voice service voip config, etc it would be helpful to see that.<br>
<br>And from that config, you're not using SIP. 'session protocol sip' would have to be added to the voip dial peers.<br><font color="#888888"><br>-nick</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">
On Mon, Jan 31, 2011 at 11:31 PM, Just Kennie <span dir="ltr"><<a href="mailto:justkennie@gmail.com" target="_blank">justkennie@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">I am using SIP.<div><br><br clear="all"><b><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Cheers,</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);">
<span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Idowu Kehinde</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);">
<span style="font-family: georgia,serif; color: rgb(0, 0, 102);">TEL: 2348024017748</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);"><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">BB: 30E8E5F9</span><br>
</b><br> <br>
<br><br></div><div><div></div><div><div class="gmail_quote">On Tue, Feb 1, 2011 at 5:08 AM, neil dsilva <span dir="ltr"><<a href="mailto:neild25@gmail.com" target="_blank">neild25@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Which routing protocol you are using ?<div><br></div><div>Regards,</div><div>Neil Dsilva<font color="#888888"><br><br></font><div class="gmail_quote"><div>On Mon, Jan 31, 2011 at 5:48 PM, Just Kennie <span dir="ltr"><<a href="mailto:justkennie@gmail.com" target="_blank">justkennie@gmail.com</a>></span> wrote:<br>
</div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div><div></div><div>Hi guys,<br>I am in need of swift help, I am a cisco network freak, and found my self handling voice in my new company. I have been doing lots of reading on voice, but I now I have a major huddle to scale. I will explain the design in details and concisely.<br>
<br> PSTN(12345) - - - - - -(voice-port 0/0/0) CE1 - - - - PE1- - - - - - - - PE2 - - - - -CE2(voice-port 0/0/0)- - - -PHONE(1000)<br><br>That is the design above, Layer 2 and Layer 3 are fine. <br>The requirement is, when external call called 12345, the call should be routed and picked at extension 1000 (PHONE)<br>
I have built my config (below), please help me look at it, as I don't have the full equipment to test across.<br><br>CE1<br>voice-port 0/0/0<br> connection plar 1000<br> description connected to 12345<br><br>dial-peer voice 1 pots<br>
destination-pattern 12345<br> direct-inward-dial<br> port 0/0/0<br><br>dial-peer voice 2 voip<br>destination-pattern 1000<br> session target ipv4:20.0.0.2 (CE2)<br><br>CE2.........................<br><br>voice-port 0/0/0<br>
description connected to 1000<br><br>dial-peer voice 1 pots<br> destination-pattern 1000<br> port 0/0/0<br><br>dial-peer voice 2 voip<br> destination-pattern 12345<br> session target ipv4:10.0.0.2 (CE1)<br><br>Please help out in details, I will understand. Thanks!<br>
<br><br clear="all"><b><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Cheers,</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);"><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">Idowu Kehinde</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);">
<span style="font-family: georgia,serif; color: rgb(0, 0, 102);">TEL: 2348024017748</span><br style="font-family: georgia,serif; color: rgb(0, 0, 102);"><span style="font-family: georgia,serif; color: rgb(0, 0, 102);">BB: 30E8E5F9</span><br>
</b><br> <br>
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