<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
</head>
<body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ; font-family: 'Verdana'; font-size: 12px;; ">
<span style="font-family: 'Verdana'; font-size: 12px;">I don't believe the annunciator is what plays ringback. That should be a function of the PSTN gateway.
<div><br>
</div>
<div>Have you tried a SIP trunk between the gateway and CUCM? It'd eliminate the interop between H.323 and SIP.<br>
<div>
<div><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Verdana; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-variant: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">
<div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">
<br>
Thanks!<br>
<br>
Matthew Berry, CCIE #26721 (Voice) <br>
Sr. Unified Communications Engineer, CDW<br>
+1.763.592.5987 | protocol.by/matthewberry</div>
</span></span></div>
<br>
<div>
<div>On Dec 16, 2011, at 6:37 AM, Roger Wiklund wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
<div>Ensure you have a MRGL with ANN on your H323 GW and phones.<br>
<br>
Try changing service parameter -> callmanager -> Send H225 User Info<br>
Message -> use ANN for ringback<br>
<br>
/Roger<br>
<br>
On Fri, Dec 16, 2011 at 1:22 PM, Robert Hass <<a href="mailto:robhass@gmail.com">robhass@gmail.com</a>> wrote:<br>
<blockquote type="cite">Hi<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">I have problem with ringback tone when doing calls from IP Phone<br>
</blockquote>
<blockquote type="cite">connected to CUCM to PSTN.<br>
</blockquote>
<blockquote type="cite">CUCM is configured to use our gateway (Cisco 2811) via H.323 and our<br>
</blockquote>
<blockquote type="cite">gateway is using SIP Trunk<br>
</blockquote>
<blockquote type="cite">do our carrier.<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">We don't hear ringback tone when calling to PSTN. But connectivity is<br>
</blockquote>
<blockquote type="cite">working fine.<br>
</blockquote>
<blockquote type="cite">We hear ringback tones when taking call from PSTN to CUCM.<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">Our gateway configuration:<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">voice service voip<br>
</blockquote>
<blockquote type="cite"> allow-connections h323 to h323<br>
</blockquote>
<blockquote type="cite"> allow-connections h323 to sip<br>
</blockquote>
<blockquote type="cite"> allow-connections sip to h323<br>
</blockquote>
<blockquote type="cite"> allow-connections sip to sip<br>
</blockquote>
<blockquote type="cite"> no supplementary-service sip moved-temporarily<br>
</blockquote>
<blockquote type="cite"> no supplementary-service sip refer<br>
</blockquote>
<blockquote type="cite"> fax protocol pass-through g711alaw<br>
</blockquote>
<blockquote type="cite"> h323<br>
</blockquote>
<blockquote type="cite"> modem passthrough nse codec g711alaw<br>
</blockquote>
<blockquote type="cite"> sip<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">voice class codec 1<br>
</blockquote>
<blockquote type="cite"> codec preference 1 g711alaw<br>
</blockquote>
<blockquote type="cite"> codec preference 2 g711ulaw<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">voice class h323 1<br>
</blockquote>
<blockquote type="cite"> h225 timeout tcp establish 3<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">dial-peer voice 1 voip<br>
</blockquote>
<blockquote type="cite"> translation-profile incoming 1<br>
</blockquote>
<blockquote type="cite"> incoming called-number .<br>
</blockquote>
<blockquote type="cite"> codec g711alaw<br>
</blockquote>
<blockquote type="cite"> no vad<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">dial-peer voice 2 voip<br>
</blockquote>
<blockquote type="cite"> description SIP Trunk to carrier (CUCM->PSTN)<br>
</blockquote>
<blockquote type="cite"> translation-profile outgoing 2<br>
</blockquote>
<blockquote type="cite"> huntstop<br>
</blockquote>
<blockquote type="cite"> destination-pattern 0T<br>
</blockquote>
<blockquote type="cite"> progress_ind setup enable 3<br>
</blockquote>
<blockquote type="cite"> session protocol sipv2<br>
</blockquote>
<blockquote type="cite"> session target ipv4:x.x.x.x:5060<br>
</blockquote>
<blockquote type="cite"> dtmf-relay rtp-nte<br>
</blockquote>
<blockquote type="cite"> codec g711alaw<br>
</blockquote>
<blockquote type="cite"> ip qos dscp cs5 media<br>
</blockquote>
<blockquote type="cite"> no vad<br>
</blockquote>
<blockquote type="cite">!<br>
</blockquote>
<blockquote type="cite">dial-peer voice 3 voip<br>
</blockquote>
<blockquote type="cite"> description PSTN->CUCM<br>
</blockquote>
<blockquote type="cite"> huntstop<br>
</blockquote>
<blockquote type="cite"> destination-pattern 49xxxxxx...$<br>
</blockquote>
<blockquote type="cite"> progress_ind setup enable 3<br>
</blockquote>
<blockquote type="cite"> session target ipv4:192.168.36.2<br>
</blockquote>
<blockquote type="cite"> voice-class h323 1<br>
</blockquote>
<blockquote type="cite"> codec g711alaw<br>
</blockquote>
<blockquote type="cite"> ip qos dscp cs5 media<br>
</blockquote>
<blockquote type="cite"> no vad<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version<br>
</blockquote>
<blockquote type="cite">15.0(1)M4, RELEASE SOFTWARE (fc1)<br>
</blockquote>
<blockquote type="cite">CUCM is 8.6<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">Any hints what is bad configured ?<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">Thanks<br>
</blockquote>
<blockquote type="cite">Robert<br>
</blockquote>
<blockquote type="cite">_______________________________________________<br>
</blockquote>
<blockquote type="cite">cisco-voip mailing list<br>
</blockquote>
<blockquote type="cite"><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
</blockquote>
<blockquote type="cite"><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
</blockquote>
<br>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
https://puck.nether.net/mailman/listinfo/cisco-voip<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</span>
</body>
</html>