[Spce-user] Voicemail FAIL 3.5.1

[Raúl Alexis Betancor Santana]

What I have tested from 3.5.1 ... this is an issue with the IP putted on the SDP of the 200 Ok reply from the appserver, as the sip_int IP appears there ... where there should be the sip_ext IP 

I have also another issue with 3.5.1 ... and it's that for one, and only ONE subscriber, we get the Route header on the 200 Ok reply from the proxy to the lb ... to apears without IP and port ... and loking like: 

Record-Route: <sip::;r2=on;lr=on;ftag=18EA83DE-5454FFED00004422-68413700;ngcplb=yes>' 

That results on the SIP UA refusing the reply, because of truncated header found. 

----- Mensaje original -----

De: "Aziz Palmer" <Aziz.Palmer at AirFIBER.com> 
Para: spce-user at lists.sipwise.com 
Enviados: Sábado, 1 de Noviembre 2014 23:07:50 
Asunto: [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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