Compleetly Confused

[Thomas sexton]

I have a V9 system. we switched isp and of course switched external ip addresses when that happened.

I altered the ip address in the config to reflect the change and set all my firewall forwarding rules as required. now i get an interesting problem.

when you call our number you get silence.. then after about 60 sec you get all phones are busy message. If you enter and known prompt the call will be put through to that extension or call group.

If i change the default incoming rule to a extension the extension will ring. as normal

you can make outbound calls perfectly.

If i put the stun server on the greeter will work and everything is fine but you can not make outbound calls. and after about 30 sec any incoming call will be dropped. and if you try to make outbound calls they fail with a FORBIDDEN error.

i don't know whats going on none of this makes any sense to me.. I'm new to 3cx and this is my first experience doing anything with the system other than changing names and numbers for extensions.

 

[leejor]

Are you seeing any error messages in the 3CX log for any of the call failures? I suspect that the old IP might still in there somewhere, but the logs should give you some information as to what's actually happening.

 

[Thomas sexton]

No errors except the Forbidden when i have the stun server on..

any suggestion as to what i should look for error wise? or where i should look to find the old ip that's hiding?

 

[Thomas sexton]

Excerpt from a single incoming call from my cell to the pbx.
**numbers have been changed to protect the inocent**


INVITE sip:[email protected]:5060;rinstance=238bed003926f464 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bKjsjh2820e0l1u5hkt1h0.1;received=24.123.15.155
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected]:5060>
From: "THOMAS SEXTON"<sip:[email protected]:5060>;tag=gK0b039322
Call-ID: [email protected]
CSeq: 1 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
In-Reply-To: [email protected]
Supported: timer
Content-Length: 0

<br>
12:49:06.571|.\Line.cpp(301)|Log5||LineCfg::getInboundTarget:Looking for inbound target: called=5137182494; caller=+15130000000<br>
12:49:06.571|.\Line.cpp(348)|Log2||LineCfg::getInboundTarget:[CM503012]: Inbound any hours rule (5137182494) for 10000 forwards to DN:8000<br>
12:49:06.602|.\CallCtrl.cpp(210)|Log2||CallCtrl:nIncomingCall:[CM503001]: Call(24): Incoming call from +15130000000@(Ln.10000@Nextiva) to <sip:[email protected].
233:5060><br>
12:49:06.618|.\Line.cpp(1336)|Log2||Line:rintEndpointInfo:[CM505003]: Provider:[Nextiva] Device info: Device Not Identified: User Agent not matched; Capabilit
ies:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]<br>
12:49:06.634|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:24.1<br>
12:49:06.634|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:24.1:Offer received. RTP connection: 208.73.146.95:27192(271
93)<br>
12:49:06.634|.\CallCtrl.cpp(398)|Log3||CallCtrl:nSelectRouteReq:[CM503010]: Making route(s) to <sip:[email protected]:5060><br>
12:49:06.649|.\CallCtrl.cpp(530)|Log2||CallCtrl:nSelectRouteReq:[CM503004]: Call(24): Route 1: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=196bcccd0a
f8705a]<br>
12:49:06.680|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210002] C:24.2:Offer provided. Connection(transcoding mode): 127.0.0.1
:7064(7065)<br>
12:49:06.696|.\Target.cpp(430)|Log2||Target::makeOneInvite:[CM503025]: Call(24): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=196bcccd0af8705a]<
br>
12:49:06.868|.\CallLeg.cpp(289)|Log3||CallLeg:nAnswer:[CM503002]: Call(24): Alerting sip:[email protected]:40600;rinstance=196bcccd0af8705a<br>
12:49:06.883|.\Extension.cpp(1153)|Log3||Extension:rintEndpointInfo:[CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: 1]
Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]<br>
12:49:06.914|.\CallLeg.cpp(1095)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:24.2<br>
12:49:06.930|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210001] C:24.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40
640(40641)<br>
12:49:06.946|.\SLServer.cpp(823)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210003] C:24.1:Answer provided. Connection(transcoding mode[unsecure])
:24.123.15.155:9040(9041)<br>
12:49:06.992|.\CallCtrl.cpp(699)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(24): Device joined: sip:[email protected]:5060<br>
12:49:07.008|.\CallCtrl.cpp(699)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(24): Device joined: sip:[email protected]:40600;rinstance=196bcccd0af8705a<br>
12:49:28.006|.\Call.cpp(42)|Log3||??:Currently active calls - 1: [24]<br>
12:49:39.066|.\InviteADS.cpp(545)|Log2||InviteADS:nAckNotReceived:[CM503021]: Call(24): ACK is not received<br>
12:49:39.097|.\Call.cpp(1226)|Log2||Call::Terminate:[CM503008]: Call(24): Call is terminated<br>
12:49:39.487|.\SLServer.cpp(791)|Error1|MediaServer|MediaServerReporting::RTPReceiver:[MS105000] C:24.1: No RTP packets were received:remoteAddr=208.73.146.95:2

 

[leejor]

12:49:39.066|.\InviteADS.cpp(545)|Log2||InviteADS:nAckNotReceived:[CM503021]: Call(24): ACK is not received<br>
12:49:39.097|.\Call.cpp(1226)|Log2||Call::Terminate:[CM503008]: Call(24): Call is terminated<br>
12:49:39.487|.\SLServer.cpp(791)|Error1|MediaServer|MediaServerReporting::RTPReceiver:[MS105000] C:24.1: No RTP packets were received:remoteAddr=208.73.146.95:2

Looks like 3CX dropped the call because of the no Acknowledgement received.

I don't doubt the provider sent one. However, it was either sent to the wrong IP (your public IP), and/or the wrong port... or, something in your network has blocked it.

If they are helpful, you might ask your provider what public IP and port 3CX is using to register with them, and, if that info is being used for the remainder of the call. The call is getting through to begin with, but then seems to be blocked. This might just be a voice packet issue. If that is the case then I would check over all setting on your router.

 

[Thomas sexton]

Actually that was probably from me hanging up the phone. I just wanted to get a example that shows the rule transfer to 8000 our greeter (digital reception) but we are getting no audio.