<html><body><div style="font-family: times new roman, new york, times, serif; font-size: 12pt; color: #000000"><div>Hi,<br></div><div>can you provide the kamailio-proxy.log related to the Voicemail calls and conference ?</div><div>Maybe you have some Rewrite rules that fetch 2000 and rewrite it in something else.</div><div>For the conference, you just need to configure a CF to the "conference".</div><div><br></div><div>For the RTP, you have to change the Domain preferences use_rtpproxy to "always".</div><div><br></div><div>Please paste here all the rate-o-mat error log related to that only line you have pasted.</div><div><br></div><div>Daniel</div><div> </div><div><br></div><div><br></div><hr id="zwchr"><div style="color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>From: </b>"Aziz Palmer" <Aziz.Palmer@AirFIBER.com><br><b>To: </b>spce-user@lists.sipwise.com<br><b>Sent: </b>Sunday, November 2, 2014 12:07:50 AM<br><b>Subject: </b>[Spce-user] Voicemail FAIL 3.5.1<br><div><br></div>
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<p class="MsoNormal">Good Morning SPCE Team,</p>
<p class="MsoNormal">I’m very impressed with the new features 3.5.1 has to offer but it seems that a few features aren't working properly</p>
<p class="MsoListParagraph" style="text-indent: -0.25in;">1.<span style="font-size: 7pt; font-family: 'Times New Roman';"> </span></p>
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<div> Voicemail</div>
<div>---- dialing voicebox or 2000, there is no audio - typically on my older sipwise installation I hear "comedian mail, mailbox"</div>
<div>---- When forwarding call into voicemail – call disconnects as soon as it connects with sipwise</div>
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<div>Conference Calling</div>
<div>----Same as Voicemail (no audio)</div>
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<div>RTP</div>
<div>---When looking at the RTP IP addresses from the subscriber's phone to sipwise all RTP data bypasses sipwise (RTP connects from my provider to subscriber's phone ) - I would like to turn that off at the domain level but be able to apply it to some subscribers.
I use to be able to do that on my older version of sipwise.</div>
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<div>Inbound call billing cost does not appear in the GUI call history but outbound works perfectly</div>
<div>--- It does appear in the CDR (inbound and outbound cost)</div>
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<div>5. [....] Restarting rate-o-mat: rate-o-matrate-o-mat: Can't call method "disconnect" on an undefined value at /usr/sbin/rate-o-mat line 1501. Active node or transition.</div>
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<div>Thank you for your help – Warmest Regards</div>
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<div>Aziz</div>
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