[Spce-user] Voicemail FAIL 3.5.1

[Daniel Grotti]

Hi, 
can you provide the kamailio-proxy.log related to the Voicemail calls and conference ? 
Maybe you have some Rewrite rules that fetch 2000 and rewrite it in something else. 
For the conference, you just need to configure a CF to the "conference". 

For the RTP, you have to change the Domain preferences use_rtpproxy to "always". 

Please paste here all the rate-o-mat error log related to that only line you have pasted. 

Daniel 


----- Original Message -----

From: "Aziz Palmer" <Aziz.Palmer at AirFIBER.com> 
To: spce-user at lists.sipwise.com 
Sent: Sunday, November 2, 2014 12:07:50 AM 
Subject: [Spce-user] Voicemail FAIL 3.5.1 



Good Morning SPCE Team, 

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 

1. 
Voicemail 
---- dialing voicebox or 2000, there is no audio - typically on my older sipwise installation I hear "comedian mail, mailbox" 
---- When forwarding call into voicemail – call disconnects as soon as it connects with sipwise 

Conference Calling 
----Same as Voicemail (no audio) 

RTP 
---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. 

Inbound call billing cost does not appear in the GUI call history but outbound works perfectly 
--- It does appear in the CDR (inbound and outbound cost) 

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. 
Thank you for your help – Warmest Regards 
Aziz 




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