DR Drops Calls

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jeffwake

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Hi. I wonder if anyone can advise on this issue. All seems to be working well with the 3CX system except the DR function. The DR answers and forwards correctly, however if you pick up the targeted extension the call is dropped. Also if you do not answer the targeted extension it rings 4 times and then hangs up the call instead of going to voice mail as setup for an unanswered extension. Any clues?
 
Hi jeffwake,

There is "Sticky: Include server status log, and equipment in use in posts!"

Thanks
 
Call Log & Configuration

Using 3CX Phone System V5.0 - Release - build 3790
Broadvoice Voip
Cisco 7940 Phones
Windows 2003 Server

10:25:11.645 Call::Terminate [CM503008]: Call(8): Call is terminated
10:25:11.599 Call::Terminate [CM503008]: Call(8): Call is terminated
10:25:11.427 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:81
10:25:10.520 MediaServerReporting::SetRemoteParty [MS210002] C:8.1:Offer provided. Connection(transcoding mode): 68.162.219.181:9010(9011)
10:25:09.833 CallCtrl::onLegConnected [CM503007]: Call(8): Device joined: sip:
10:25:09.802 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:81
10:25:09.770 CallCtrl::onRerouteReq [CM503005]: Call(8): Forwarding: IVR:RecordMessage@[Dev]
10:24:50.020 Extension::printEndpointInfo [CM505001]: Ext.10: Device info: Device Identified: [Man: Cisco;Mod: 7940;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CP7940G/8.0] Transport: [sip:192.168.0.5:5060]
10:24:50.020 CallCtrl::onAnsweredCall [CM503002]: Call(8): Alerting sip:[email protected]:5060;transport=udp
10:24:49.473 CallCtrl::onSelectRouteReq [CM503004]: Call(8): Calling: Ext:10@[Dev:sip:[email protected]:5060;transport=udp]
10:24:49.473 CallCtrl::onSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
10:24:46.239 MediaServerReporting::DTMFhandler [MS211000] C:8.1: 147.135.12.250:6586 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
10:24:41.473 MediaServerReporting::SetRemoteParty [MS210003] C:8.1:Answer provided. Connection(transcoding mode):68.162.219.181:9010(9011)
10:24:41.114 MediaServerReporting::SetRemoteParty [MS210000] C:8.1:Offer received. RTP connection: 147.135.12.250:6586(6587)
10:24:41.098 CallLeg::setRemoteSdp Remote SDP is set for legC:8.1
10:24:40.895 CallLeg::onConfirmed Session 2260 of leg C:8.1 is confirmed
10:24:40.567 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:81
10:24:40.520 MediaServerReporting::SetRemoteParty [MS210003] C:8.1:Answer provided. Connection(transcoding mode):68.162.219.181:9010(9011)
10:24:40.520 CallCtrl::onLegConnected [CM503007]: Call(8): Device joined: sip:[email protected]:5060
10:24:39.911 CallCtrl::onLegConnected [CM503007]: Call(8): Device joined: sip:
10:24:39.880 MediaServerReporting::SetRemoteParty [MS210000] C:8.1:Offer received. RTP connection: 147.135.12.250:6586(6587)
10:24:39.864 CallCtrl::onSelectRouteReq [CM503004]: Call(8): Calling: IVR:81@[Dev]
10:24:39.848 CallCtrl::onSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]:5060]
10:24:39.848 CallLeg::setRemoteSdp Remote SDP is set for legC:8.1
10:24:39.848 Line::printEndpointInfo [CM505003]: Provider:[Broadvoice] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.5:5060]
10:24:39.848 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:81
10:24:39.661 CallCtrl::onIncomingCall [CM503001]: Call(8): Incoming call from 9787457878@(Ln.10000@Broadvoice) to [sip:[email protected]:5060]
10:24:39.536 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10000 forwards to DN:81
10:24:39.426 CallLeg::onNewCall [CM500002]: Info on incoming INVITE:
INVITE sip: @192.168.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 147.135.12.221:5060
Contact: [sip: @147.135.12.221:5060]
To: " "[sip: @192.168.0.5]
From: " "[sip: @147.135.12.221;user=phone];tag=hijk
Call-ID: [email protected]
CSeq: 1 INVITE
Supported: 100rel
Content-Length: 0
 
2 vs 3 digit extensions

Is it possible that the 2 digit extensions I have used are problematic and that 3 digit extensions are better tested and more stable?
 
Hi !

Can you please repeat both situations (call dropping after targeted extension has been picked up and call hanging up after 4 rings) and then send the log-files (from your '<3CX PhoneSystem folder>\Data\Logs' directory) directly to me ([email protected]).

Thanks in advance
Alexander
 
After removing both Supports Re-invite and Supports "Replaces" Header for Broadvoice this problem went away.

Thanks
 
jeffwake said:
After removing both Supports Re-invite and Supports "Replaces" Header for Broadvoice this problem went away.

Thanks

Jeffwake,

Did you configure provider using "generic" template, using BroadVoice template or migrate from previous version(3)?

Thanks
 
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