<div dir="ltr">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<div>
<br></div><div><span style="font-family:arial,sans-serif;font-size:13px">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</span><div style="font-family:arial,sans-serif;font-size:13px">
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.</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">as for my issue i believe i have nailed it.</div>
<div style="font-family:arial,sans-serif;font-size:13px">Here is the initial invite to Asterisk which i believe is hosting the VoiceMail System</div><div style="font-family:arial,sans-serif;font-size:13px">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 ?<br>
</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px"><<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP ---------------->>>>><br>
</div><div style="font-family:arial,sans-serif;font-size:13px"><<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP ---------------->>>>><br></div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px"><br><div><div><--- SIP read from <a href="http://127.0.0.1:5062/" target="_blank">127.0.0.1:5062</a> ---></div><div>INVITE sip:abc14843351444@voicebox.local SIP/2.0</div>
<div>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--></div><div>Record-Route: <sip:127.0.0.1:5060;ngcplb=yes;r2=on;socket=udp:10.254.1.21:5060;ftag=a5Qt2vrHX4UyQ;lr=on></div>
<div>Record-Route: <sip:54.208.75.0:5060;ngcplb=yes;r2=on;socket=udp:10.254.1.21:5060;ftag=a5Qt2vrHX4UyQ;lr=on></div><div>Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKbc93.0292d4c08bc9511f8dd8401810e78db6.0</div><div>
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKbc93.0e37e28d55c25857b6f992520a023b00.0</div><div>Via: SIP/2.0/UDP 24.229.51.68;rport=5060;branch=z9hG4bK4H38a0e2HyNpS</div><div>Max-Forwards: 14</div><div>From: <<a href="mailto:sip%3A14843351444@24.229.51.68" target="_blank">sip:14843351444@24.229.51.68</a>>;tag=a5Qt2vrHX4UyQ</div>
<div>To: <<a href="mailto:sip%3A2000@sip1.blueuc.com" target="_blank">sip:2000@sip1.blueuc.com</a>></div><div>Call-ID: 143ff8c2-0177-1232-5ea4-005056a433a6</div><div>CSeq: 55050529 INVITE</div><div>Contact: <sip:gw+BlueUC-SIP1@24.229.51.68:5060;transport=udp;gw=BlueUC-SIP1></div>
<div>User-Agent: NetBorder Session Controller</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div>Supported: precondition, path, replaces</div><div>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div>
<div>Content-Length: 201</div><div>X-Sipx-Handled: X192.168.10.1-24.229.51.68</div><div>X-FS-Support: update_display</div><div>P-Caller-UUID: 55d54a4f-6c14-44c0-81b5-7c8ead45f5e1</div><div>P-Callee-UUID: 55d54a4f-6c14-44c0-81b5-7c8ead45f5e1</div>
<div>P-NGCP-Caller-Info: <<a href="mailto:sip%3A4843351444@24.229.51.68" target="_blank">sip:4843351444@24.229.51.68</a>>;ip=24.229.51.68;port=5060</div><div>P-NGCP-Callee-Info: <sip:abc14843351444@voicebox.local>;ip=127.0.0.1;port=5070</div>
<div><br></div><div>v=0</div><div>o=nsc 1390759036 1390759037 IN IP4 54.208.75.0</div><div>s=nsc</div><div>c=IN IP4 54.208.75.0 <<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP ---------------->>>>></div>
<div>t=0 0</div><div>m=audio 30040 RTP/AVP 9 0 101 13</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=rtcp:30041</div><div><-------------></div></div><div><br></div>
<div><br></div></div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px"><<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP ---------------->>>>><br>
</div><div style="font-family:arial,sans-serif;font-size:13px"><<<<<------------------ NOTICE THE IP ADDRESS IN THE SDP ---------------->>>>><br></div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px">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 ?</div>
<div><br></div>-- <br><div dir="ltr"><p style="margin:0px">Thank You,</p><p style="margin:0px">Chris Rawlings</p><p style="margin:0px">BlueCloud Consultants – CEO</p><p style="margin:0px">Phone. 484-335-1444 x201</p><p style="margin:0px">
SIP URI / Lync / XMPP / Jabber / Google Talk - <a>chris@blueuc.com</a></p></div>
</div></div>