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Hi
I'm running 3CX small business server v. 6.1.0 & I have set my voicemail number as 700 in the General -> General Settings -> Voicemail & Fax delivery Settings page.
Unfortunately, I cannot access the voicemail from any of our company phones (Linksys SPA921) which have been provisioned. Having set the logging level to verbose, when I dial 700 I get the following messages:
In addition to the above, I have set a forwarding rule for a call to be directed to the mailbox when a phone is not picked up after a certain time. This doesn't work either and gives the following error messages:
I've tried restarting the "3CX PhoneSystem Voicemail Manager" which takes place successfully but doesn't really change the outcome. Upon restart, the log message on the server is:
I'd appreciate any help with this as we really need the voicemail capability for our company.
Cheers
I'm running 3CX small business server v. 6.1.0 & I have set my voicemail number as 700 in the General -> General Settings -> Voicemail & Fax delivery Settings page.
Unfortunately, I cannot access the voicemail from any of our company phones (Linksys SPA921) which have been provisioned. Having set the logging level to verbose, when I dial 700 I get the following messages:
Code:
19:37:03.040 Call::Terminate [CM503008]: Call(5): Call is terminated
19:37:03.040 Call::RouteFailed [CM503015]: Call(5): Attempt to reach [sip:[email protected]] failed. Reason: Not Registered
19:37:03.009 Call::RouteFailed [CM503015]: Call(5): Attempt to reach [sip:[email protected]] failed. Reason: Not Registered
19:37:03.009 CallCtrl::onSelectRouteReq [CM503016]: Call(5): Target is not registered: Ext:Ext.700
19:37:03.009 CallCtrl::onSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
19:37:03.009 MediaServerReporting::SetRemoteParty [MS210000] C:5.1:Offer received. RTP connection: xxx.xxx.xxx.130:16452(16453)
19:37:03.009 CallLeg::setRemoteSdp Remote SDP is set for legC:5.1
19:37:03.009 Extension::printEndpointInfo [CM505001]: Ext.102: Device info: Device Identified: [Man: Linksys;Mod: SPA-921;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA921-5.1.8] Transport: [sip:xxx.xxx.xxx.1:5060]
19:37:02.993 CallCtrl::onIncomingCall [CM503001]: Call(5): Incoming call from Ext.102 to [sip:[email protected]]
19:37:02.993 CallLeg::onNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.130:5060;branch=z9hG4bK-14a22b7e
Max-Forwards: 70
Contact: "Aris Krikelis"[sip:[email protected]:5060]
To: [sip:[email protected]]
From: "Aris Krikelis"[sip:[email protected]];tag=2b56078a9bd4d321o0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="102",realm="3CXPhoneSystem",nonce="12874102622:a5cf76a068a99053e713c9cd6426c522",uri="sip:[email protected]",algorithm=MD5,response="b0979b67c1b5957782272f363789d385"
Supported: replaces
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0
In addition to the above, I have set a forwarding rule for a call to be directed to the mailbox when a phone is not picked up after a certain time. This doesn't work either and gives the following error messages:
Code:
19:41:56.991 StunClient::onInitTests [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 75.101.138.128:3478 over Transport xxx.xxx.xxx.107:5060
19:41:56.991 StunClient::onInitTests [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 75.101.138.128:3478 over Transport xxx.xxx.xxx.1:5060
19:39:41.788 Call::Terminate [CM503008]: Call(6): Call is terminated
19:39:41.772 Call::RouteFailed [CM503015]: Call(6): Attempt to reach [sip:[email protected]:5060] failed. Reason: Not Registered
19:39:41.756 Call::RouteFailed [CM503015]: Call(6): Attempt to reach [sip:[email protected]:5060] failed. Reason: Not Registered
19:39:41.756 CallCtrl::onRerouteReq [CM503016]: Call(6): Target is not registered: Ext:Ext.700
19:39:26.711 MediaServerReporting::SetRemoteParty [MS210006] C:6.2:Offer provided. Connection(by pass mode): xxx.xxx.xxx.140:16466(16467)
19:39:26.696 CallCtrl::onSelectRouteReq [CM503004]: Call(6): Calling: Ext:Ext.102@[Dev:sip:[email protected]:5060]
19:39:26.696 CallCtrl::onSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]:5060]
19:39:26.696 MediaServerReporting::SetRemoteParty [MS210000] C:6.1:Offer received. RTP connection: xxx.xxx.xxx.140:16466(16467)
19:39:26.696 CallLeg::setRemoteSdp Remote SDP is set for legC:6.1
19:39:26.696 Extension::printEndpointInfo [CM505001]: Ext.120: Device info: Device Identified: [Man: Linksys;Mod: SPA-921;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA921-5.1.8] Transport: [sip:xxx.xxx.xxx.1:5060]
19:39:26.680 CallCtrl::onIncomingCall [CM503001]: Call(6): Incoming call from Ext.120 to [sip:[email protected]:5060]
19:39:26.680 CallLeg::onNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.140:5060;branch=z9hG4bK-d0001c1f
Max-Forwards: 70
Contact: "120"[sip:[email protected]:5060]
To: [sip:[email protected]:5060]
From: "120"[sip:[email protected]:5060];tag=6c4b34e5bf2df947o0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="120",realm="3CXPhoneSystem",nonce="12874102766:947e0526b3ce81860609dfbebebc5bfe",uri="sip:[email protected]:5060",algorithm=MD5,response="8ebae3107ada82de61decc0a8cb05518"
Supported: replaces
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0
I've tried restarting the "3CX PhoneSystem Voicemail Manager" which takes place successfully but doesn't really change the outcome. Upon restart, the log message on the server is:
Code:
19:45:19.999 ListenConnect SL: connected xxxserve:0/VoiceBoxManagerService at [xxxserve]/VoiceBoxManagerService
19:45:13.031 ListenDisconnect SL: disconnected xxxserve:0/VoiceBoxManagerService at [xxxserve]/VoiceBoxManagerService
I'd appreciate any help with this as we really need the voicemail capability for our company.
Cheers