Incoming calls disconnecting when xfer to digital receptionist

[[email protected]]

Ok, so I'm using V9, and I had to reinstall the software and restore a backup due to a server crash. Things are fine, as far as outbound calls, and internal calls. But if I try inbound calls, it seems to pick up and get disconnected right away. I don't know if this is because I am in free mode (I have an email in to support to get that fixed) or if there is a further issue. Here's the logging from when I tried calling in from the outside.

05:15:39.449 [CM503008]: Call(1): Call is terminated
05:15:36.834 Session 77 of leg C:1.1 is confirmed
05:15:36.740 [CM503007]: Call(1): Device joined: sip:[email protected]:40600;rinstance=8fc9e150ecb1fab0
05:15:36.725 [CM503007]: Call(1): Device joined: sip:[email protected]:5060
05:15:36.725 [MS210003] C:1.1:Answer provided. Connection(transcoding mode[unsecure]):10.0.0.30:7000(7001)
05:15:36.725 [MS210001] C:1.3:Answer received. RTP connection[unsecure]: 127.0.0.1:40610(40611)
05:15:36.725 Remote SDP is set for legC:1.3
05:15:36.725 [CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
05:15:36.725 [CM503002]: Call(1): Alerting sip:[email protected]:40600;rinstance=8fc9e150ecb1fab0
05:15:36.631 Currently active calls - 1: [1]
05:15:36.507 [CM503025]: Call(1): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=8fc9e150ecb1fab0]
05:15:36.507 [MS210002] C:1.3:Offer provided. Connection(transcoding mode): 127.0.0.1:7004(7005)
05:15:36.476 [CM503005]: Call(1): Forwarding: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=8fc9e150ecb1fab0]
05:15:36.476 [CM503003]: Call(1): Call to sip:[email protected] has failed; Cause: 486 Busy Here; from IP:10.0.0.193:5060
05:15:36.382 [CM503025]: Call(1): Calling Ext:Ext.1000@[Dev:sip:[email protected]:5060]
05:15:36.382 [MS210002] C:1.2:Offer provided. Connection(transcoding mode): 10.0.0.30:7002(7003)
05:15:36.351 [CM503004]: Call(1): Route 1: Ext:Ext.1000@[Dev:sip:[email protected]:5060]
05:15:36.336 [CM503010]: Making route(s) to <sip:[email protected]:5060>
05:15:36.320 [MS210000] C:1.1:Offer received. RTP connection: 10.0.0.228:4866(4867)
05:15:36.320 Remote SDP is set for legC:1.1
05:15:36.320 [CM503001]: Call(1): Incoming call from 2244655732@(Ln.10000@PattonSN4960) to <sip:[email protected]:5060>
05:15:36.180 [CM503012]: Inbound any hours rule (Fabric Main) for 10000 forwards to DN:1000
05:15:36.133 Looking for inbound target: called=8474889877; caller=2244655732
05:15:36.133 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.228:5060;branch=z9hG4bK24dc107d199b4be92
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=6d9314053c
Call-ID: fc0cb87866018373
CSeq: 11739 INVITE
Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d535c106d1e5883:0cf90683288a704bbec42ab6094e586d",uri="sip:[email protected]:5060",response="da95d84d4c2d2fda9dfc16073681d3f9",algorithm=MD5
Supported: replaces
User-Agent: Patton SN4960 1E24V 00A0BA0538F0 R5.4 2010-05-14 H323 RBS SIP M5T SIP Stack/4.0.28.28
Content-Length: 0

 

[leejor]

I see ext 1000 busy and the call forwarding to 8000. (transcoding going on too, so I assume that there not is Codec uniformity being deployed at the site).

What is at 8000?

What happens when another extension calls direct to 1000, and forwarded?

How about an extension call direct to 8000?

What about and outside call sent direct to 8000?

Do any of those work as expected?

 

[[email protected]]

I suspect this is because of v9 being in free mode right now (I can't get the license activated at the moment) but 8000 is our digital receptionist. Internally, if you call 1000, it forwards to 8000 and then immediately disconnects. A outside call sent straight to 8000 disconnects as well.