it keeps ringing. <br><br>
<div><span class="gmail_quote">On 10/4/07, <b class="gmail_sendername">Paul Choi</b> <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I don't think that AA parameter has anything to do<br>with this issue since that is more related to enabling
<br>whether callers can transfer to DNs outside of the<br>range specified within CCME/CUE.<br><br>Does the phone keep ringing after your CNFA timeout or<br>does it just go to fast busy?<br><br>Paul<br><br>--- "Andrew F. Wolfe"
<br><<a href="mailto:deadpool.slayer.telecomm@gmail.com">deadpool.slayer.telecomm@gmail.com</a>> wrote:<br><br>> CODEC is G.711. I will check that parameter.<br>> Thanks guys.<br>><br>> On 10/4/07, Rasim Duric <
<a href="mailto:rduric@uoguelph.ca">rduric@uoguelph.ca</a>> wrote:<br>> ><br>> > Check your system AA script parameter<br>> 'allowExternalTransfers'. Should<br>> > be 'TRUE'.<br>> >
<br>> ><br>> ><br>> > Rasim Duric<br>> > Network Analyst (CCS)<br>> > University of Guelph<br>> > Guelph, N1G 2W1, ON<br>> > 519-824-4120x53146<br>> > <a href="mailto:rduric@uoguelph.ca">
rduric@uoguelph.ca</a><br>> ><br>> > *From:* <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>> [mailto:<br>> > <a href="mailto:cisco-voip-bounces@puck.nether.net">
cisco-voip-bounces@puck.nether.net</a>] *On Behalf Of<br>> *Andrew F. Wolfe<br>> > *Sent:* Thursday, October 04, 2007 9:47 AM<br>> > *To:* <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net
</a><br>> > *Subject:* [cisco-voip] CUE not picking up on<br>> outside calls<br>> ><br>> ><br>> ><br>> > Good Morning,<br>> ><br>> > I have an issue that is odd. My first CUE and CME
<br>> implementation.<br>> > Internal calls will CFB and CFNA to voicemail but<br>> outside callers never get<br>> > transferred to voicemail. This is a native SIP<br>> trunk bringing in the calls<br>
> > and they are delivering 4 digits. I have the<br>> allow-connection sip to sip<br>> > command in the voice service voip section. From<br>> what I have found on CCO<br>> > that should allow it but looking for any insight
<br>> out there. All help is<br>> > appreciated. If you need to see any configs I can<br>> strip them down and send<br>> > you what you need. Thanks.<br>> ><br>> > Andrew<br>> ><br>> > _______________________________________________
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