[Spce-user] No Audio from VoiceMail System

[Chris Rawlings]

to try and correct this issue i setup a loopback ip address as my WAN
address on the SIPWise system... this fixed the no audio to asterisk issue
but broke a few other things ... like call forwarding to Voicemail if no
answer... and rejecting a call was a whole other issue

i am having 0 issues with our setup for quality.. no crashing.. none what
so ever... this is not an Amazon EC2 or virtual machine issue but a NAT
issue
i am sorry that you are unable to get this working from a VM to a Cloud
Hosted System in Amazon EC2 but it was a breeze to get our audio and
stability working without issue. Honestly just don't use the RTP proxy if
you are going to have this setup in the cloud so that your Audio is not
anchored to a Virtual Machine. Also i have clients with thousands of phones
and we run 100% of their ACD / Phone System / SBC's / PBX / and other
resources in VMware / Virtual Environments without issue. We resell and
install 100% virtual solutions so my ranting here about your comment was
based on the same things i have heard over the years that is no longer true
if you configure your environment properly for a virtual infrastructure.


as for my issue i believe i have nailed it.
Here is the initial invite to Asterisk which i believe is hosting the
VoiceMail System
The IP address in the SDP header while talking to itself  is an external IP
address of the system when talking to the outside world... this is not
going to work when it needs to talk to itself. What should i do here ?

<<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP
---------------->>>>>
<<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP
---------------->>>>>


<--- SIP read from 127.0.0.1:5062 --->
INVITE sip:abc14843351444 at voicebox.local SIP/2.0
Record-Route: <sip:127.0.0.1:5062
;lr=on;ftag=a5Qt2vrHX4UyQ;did=827.47e2;mpd=ii;ice_caller=strip;ice_callee=strip;savp_callee=force_rtp;rtpprx=yes;vsf=VDl0THN2dW1FXjNnZk4CN19rUkpfLm9IaWJ3Ug-->
Record-Route: <sip:127.0.0.1:5060
;ngcplb=yes;r2=on;socket=udp:10.254.1.21:5060;ftag=a5Qt2vrHX4UyQ;lr=on>
Record-Route: <sip:54.208.75.0:5060
;ngcplb=yes;r2=on;socket=udp:10.254.1.21:5060;ftag=a5Qt2vrHX4UyQ;lr=on>
Via: SIP/2.0/UDP 127.0.0.1:5062
;branch=z9hG4bKbc93.0292d4c08bc9511f8dd8401810e78db6.0
Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKbc93.0e37e28d55c25857b6f992520a023b00.0
Via: SIP/2.0/UDP 24.229.51.68;rport=5060;branch=z9hG4bK4H38a0e2HyNpS
Max-Forwards: 14
From: <sip:14843351444 at 24.229.51.68>;tag=a5Qt2vrHX4UyQ
To: <sip:2000 at sip1.blueuc.com>
Call-ID: 143ff8c2-0177-1232-5ea4-005056a433a6
CSeq: 55050529 INVITE
Contact: <sip:gw+BlueUC-SIP1 at 24.229.51.68:5060;transport=udp;gw=BlueUC-SIP1>
User-Agent: NetBorder Session Controller
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201
X-Sipx-Handled: X192.168.10.1-24.229.51.68
X-FS-Support: update_display
P-Caller-UUID: 55d54a4f-6c14-44c0-81b5-7c8ead45f5e1
P-Callee-UUID: 55d54a4f-6c14-44c0-81b5-7c8ead45f5e1
P-NGCP-Caller-Info: <sip:4843351444 at 24.229.51.68>;ip=24.229.51.68;port=5060
P-NGCP-Callee-Info: <sip:abc14843351444 at voicebox.local
>;ip=127.0.0.1;port=5070

v=0
o=nsc 1390759036 1390759037 IN IP4 54.208.75.0
s=nsc
c=IN IP4 54.208.75.0   <<<<<------------------ NOTICE THE IP ADDRESS IN THE
SDP ---------------->>>>>
t=0 0
m=audio 30040 RTP/AVP 9 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp:30041
<------------->



<<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP
---------------->>>>>
<<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP
---------------->>>>>

The IP address in the SDP header while talking to itself on port 5070 is an
external IP address of the system when talking to the outside world... this
is not going to work when it needs to talk to itself. What should i do here
?

-- 

Thank You,

Chris Rawlings

BlueCloud Consultants - CEO

Phone. 484-335-1444 x201

SIP URI / Lync / XMPP / Jabber / Google Talk - chris at blueuc.com
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