Hi
Does VitalPBX have integration with SIP Homer Capture?
We’ve instaled Heplify client in a VitalPBX, and it is sent HEP data to the Homer server.
In the homer server, we can saw the HEP protocol data arriving from that Heplify client installed on a VitalPBX:
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…foo…&0a1fad2f92d54faaa0f8b2511c6a4ce1…INVITE sip:44506369@10.253.10.19:50777 SIP/2.0.
Via: SIP/2.0/UDP 201.87.151.62:2387;rport;branch=z9hG4bKPj5284227dc6674924b57bdef8c84782b2.
Max-Forwards: 70.
From: “1000” sip:1000@10.253.10.19;tag=f02efa053fa442a69c09ec457fd7e611.
To: sip:44506369@10.253.10.19.
Contact: “1000” sip:1000@201.87.151.62:2387;ob.
Call-ID: 0a1fad2f92d54faaa0f8b2511c6a4ce1.
CSeq: 7940 INVITE.
Route: sip:10.253.10.19:50777;lr.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: MicroSIP/3.21.3.
Content-Type: application/sdp.
Content-Length: 344.
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v=0.
o=- 3938758466 3938758466 IN IP4 201.87.151.62.
s=pjmedia.
b=AS:84.
t=0 0.
a=X-nat:1.
m=audio 2398 RTP/AVP 8 18 101.
c=IN IP4 201.87.151.62.
b=TIAS:64000.
a=rtcp:2399 IN IP4 201.87.151.62.
a=sendrecv.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ssrc:198385589 cname:6ccb68fc099b03d5.
I 10.253.10.248 → 10.253.10.19 3:3 #2
…E…3d@.@…
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…#e…HEP3…Y…
g.Z…
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…
…
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…
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…foo…&0a1fad2f92d54faaa0f8b2511c6a4ce1…INVITE sip:44506369@10.253.10.19:50777 SIP/2.0.
I 10.253.10.248 → 10.253.10.19 3:3 #10
…E…3h@.@…u
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…#e…\fHEP3…Y…
g.Z…
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…
…
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…
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…foo…&0a1fad2f92d54faaa0f8b2511c6a4ce1…cSIP/2.0 100 Trying.
Via: SIP/2.0/UDP 201.87.151.62:2387;rport=49624;received=10.11.10.11;branch=z9hG4bKPj93d6cf52aa0049d78e72cc81c2e97abb.
Call-ID: 0a1fad2f92d54faaa0f8b2511c6a4ce1.
From: “1000” sip:1000@10.253.10.19;tag=f02efa053fa442a69c09ec457fd7e611.
To: sip:44506369@10.253.10.19.
CSeq: 7941 INVITE.
Server: VitalPBX.
Content-Length: 0.
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U 10.253.10.19:38080 → 10.253.10.248:9061 #11
HEP3…Y…
g.Z…
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…G…
…
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…
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…foo…&0a1fad2f92d54faaa0f8b2511c6a4ce1…%SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 10.253.10.19:50777;rport=50777;received=10.253.10.19;branch=z9hG4bKPjbb858056-3f68-4386-ba1c-b7c7acb7f153.
Call-ID: 2ec7e3e1-63b5-406d-9bd8-3e017d252b4e.
From: “551151990722” sip:551151990722@10.253.10.19;tag=067ee737-6f17-46d7-84de-8bd7b5cfe135.
To: sip:%2314551144506369@10.253.10.20;tag=89945857-d635-463c-8c65-3f83445bc1b5.
CSeq: 16086 INVITE.
Server: Attimo.
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER.
Contact: sip:10.253.10.20:50750.
Supported: 100rel, timer, replaces, norefersub.
Session-Expires: 1800;refresher=uac.
Require: timer.
Content-Type: application/sdp.
Content-Length: 306.
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v=0.
o=- 683342940 683342942 IN IP4 10.253.10.20.
s=Asterisk.
c=IN IP4 10.253.10.20.
t=0 0.
m=audio 17510 RTP/AVP 0 8 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:150.
a=sendrecv.
I 10.253.10.248 → 10.253.10.19 3:3 #18
…E…4e@.@…
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…#e…HEP3…>…Y…
g.Z…
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…
…
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…
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…foo…*2ec7e3e1-63b5-406d-9bd8-3e017d252b4e…SIP/2.0 200 OK.
But in Homer Web Gui the calls isn’t showed, same configuring the Alias correctly.
In the Homer server the postgres DB is connected and working.
Has anyone managed to do this integration?
Regards
Marco