by default, call shouldn't go voip to voip dialpeer unless you have:<br><br>voice service voip<br>allow-connections h323 to h323 (or sip to sip...whatever flavor you're using)<br><br>another more complicated option would be to use cor lists and apply a restricted inbound corlist to the inbound VOIP dialpeer that is receicving an inbound call from the 2nd SRST gateway. <br>
<br><br><br><div class="gmail_quote">On Wed, Nov 26, 2008 at 3:12 PM, Lelio Fulgenzi <span dir="ltr"><<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</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><div style="font-family: Verdana; font-size: 10pt; color: rgb(0, 0, 0);">The important thing is to ensure that when a call goes from one router to the other, you don't send the call back in a loop. <br><br>Not sure how to accomplish that. It's something that we might need as I'm recommending two SRST/HSRP routers for a business continuity solution when we get rid of our emergency PBX.<br>
<br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>
"Bad grammar makes me [sic]" - Tshirt<div><div></div><div class="Wj3C7c"><br><br><br>----- Original Message -----<br>From: "James Buchanan" <<a href="mailto:jbuchanan@ctiusa.com" target="_blank">jbuchanan@ctiusa.com</a>><br>
To: "Nick" <<a href="mailto:csvoip@googlemail.com" target="_blank">csvoip@googlemail.com</a>>, "<a href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
Sent: Wednesday, November 26, 2008 2:29:07 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] SRST inbound calls issue<br><br>
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Hello,</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Use a VoIP dial peer with a wildcard pattern and point it to the
other gateway on each gateway.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Thanks,</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">James</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On
Behalf Of </b>Nick<br>
<b>Sent:</b> Wednesday, November 26, 2008 1:20 PM<br>
<b>To:</b> <a href="mailto:cisco-voip@puck-nether.net" target="_blank">cisco-voip@puck-nether.net</a><br>
<b>Subject:</b> [cisco-voip] SRST inbound calls issue</span></p>
</div>
<p> </p>
<div>
<p>I have just come across a situation, I have 2 SRST
gateways with phones split across the 2, each gateway has a PRI which have a
range of numbers that round robin between the 2 PRI's </p>
</div>
<div>
<p> </p>
</div>
<div>
<p>If phone A is registered to SRST gateway A then because the
PRI round robins between the 2 gateways the call will only complete if it goes
to the gateway that number is registered, is there any way round this or to
send the call to the other gateway?</p>
</div>
<div>
<p> </p>
</div>
<div>
<p> </p>
</div>
</div>
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