[Spce-user] differences between standar calls and VSC call?

[Sergio Serrano]

Hi again,

	here I am with other extrange case(probably for my limited
knowledge).

I have SIPWISE CE confgiured and all standar calls works. But now we
want to setup VSC and We haven't audio when activate or deactivate a
service. I launch sngrep and I see next:

INVITE sip:*72*666666666 at A.A.A.A:59060 SIP/2.0
.........
v=0
o=ATCOM 3686380481 3686380481 IN IP4 87.221.105.155
s=ATCOM Audio Call
c=IN IP4 87.221.105.155
t=0 0
m=audio 57464 RTP/AVP 8 0 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20This invite is passed from LB to Proxy as

INVITE sip:*72*666666666 at A.A.A.A:59060 SIP/2.0
.........
v=0
o=ATCOM 3686380481 3686380481 IN IP4 87.221.105.155
s=ATCOM Audio Call
c=IN IP4 87.221.105.155
t=0 0
m=audio 57464 RTP/AVP 8 0 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

And this invite is passed from Proxy to sems as

INVITE sip:*72*666666666 at vsc.local:59060 SIP/2.0
v=0
o=ATCOM 3686380481 3686380481 IN IP4 A.A.A.A
s=ATCOM Audio Call
c=IN IP4 A.A.A.A
t=0 0
m=audio 31756 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:31757

This SDP make to rtpengine send audio to A.A.A.A which is the Public IP
of SIPWise.

Where Can I see to find error?

Best Regards and thanks for your help,

Sergio
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