transferring to voicemail v5 blank space

[nateb]

when someone external dials in, and i try to transfer them to voicemail, my phone shows it transfers successfully, but the person hears hold music until it just disconnects them. and then the voice mail generated is just blank space. here is the call log ;

14:59:53.490 MediaServerReporting::SetRemoteParty [MS210002] C:68.1:Offer provided. Connection(transcoding mode): 172.16.1.4:7324(7325)
14:59:53.381 CallCtrl:nLegConnected [CM503007]: Call(68): Device joined: sip:
14:59:53.381 LineCfg::getInboundTarget [CM503010]: Inbound office hours' rule for LN:10102 forwards to DN:800
14:59:53.287 CallCtrl:nRerouteReq [CM503005]: Call(68): Forwarding: IVR:RecordMessage@[Dev]
14:59:38.241 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.1.5.15] Transport: [sip:172.16.1.4:5060]
14:59:38.241 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:59:38.085 CallCtrl:nSelectRouteReq [CM503004]: Call(68): Calling: Ext:101@[Dev:sip:[email protected]:5060;transport=udp;user=phone]
14:59:38.085 CallCtrl:nSelectRouteReq Making route(s) to [sip:[email protected];user=phone]
14:59:38.069 CallLeg:nRefer Refer: from=[sip:[email protected];user=phone];tag=2ca635927072e7b2; to="BONDELID N "[sip:[email protected]:5060];tag=6231ae2c; RefTo=[sip:[email protected];user=phone]
14:59:36.726 MediaServerReporting::SetRemoteParty [MS210003] C:68.2:Answer provided. Connection(transcoding mode):172.16.1.4:7326(7327)
14:59:36.726 MediaServerReporting::SetRemoteParty [MS210000] C:68.2:Offer received. RTP connection: 172.16.1.40:5004(5005)
14:59:36.710 CallLeg::setRemoteSdp Remote SDP is set for legC:68.2
14:59:36.413 CallLeg:nConfirmed Session 2176 of leg C:68.1 is confirmed
14:59:36.054 CallCtrl:nLegConnected [CM503007]: Call(68): Device joined: sip:[email protected]:5060;transport=udp;user=phone
14:59:36.054 LineCfg::getInboundTarget [CM503010]: Inbound office hours' rule for LN:10102 forwards to DN:800
14:59:35.882 CallCtrl:nLegConnected [CM503007]: Call(68): Device joined: sip:[email protected]:5060
14:59:35.866 MediaServerReporting::SetRemoteParty [MS210003] C:68.1:Answer provided. Connection(transcoding mode):172.16.1.4:7324(7325)
14:59:35.632 MediaServerReporting::SetRemoteParty [MS210001] C:68.2:Answer received. RTP connection: 172.16.1.40:5004(5005)
14:59:35.616 CallLeg::setRemoteSdp Remote SDP is set for legC:68.2
14:59:34.273 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:59:34.132 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;user=phone
14:59:34.132 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:59:34.132 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:59:34.132 CallCtrl:nAnsweredCall [CM503002]: Call(68): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:59:34.116 MediaServerReporting::SetRemoteParty [MS210002] C:68.6:Offer provided. Connection(transcoding mode): 172.16.1.4:7334(7335)
14:59:33.929 MediaServerReporting::SetRemoteParty [MS210002] C:68.5:Offer provided. Connection(transcoding mode): 172.16.1.4:7332(7333)
14:59:33.788 MediaServerReporting::SetRemoteParty [MS210002] C:68.4:Offer provided. Connection(transcoding mode): 172.16.1.4:7330(7331)
14:59:33.554 MediaServerReporting::SetRemoteParty [MS210002] C:68.3:Offer provided. Connection(transcoding mode): 172.16.1.4:7328(7329)
14:59:33.335 MediaServerReporting::SetRemoteParty [MS210002] C:68.2:Offer provided. Connection(transcoding mode): 172.16.1.4:7326(7327)
14:59:33.257 CallCtrl:nSelectRouteReq [CM503004]: Call(68): Calling: RingAll:800@[Dev:sip:[email protected]:5060;transport=udp;user=phone, Dev:sip:[email protected]:5060;transport=udp;user=phone, Dev:sip:[email protected]:5060;transport=udp;user=phone, Dev:sip:[email protected]:5060;user=phone, Dev:sip:[email protected]:5060;transport=udp;user=phone]
14:59:33.179 CallCtrl:nSelectRouteReq Making route(s) to [sip:[email protected]:5060]

 

[nateb]

blind transfer

I forgot to mention I'm using a blind transfer on granstream phones.

 

[nateb]

update

inner office voice mail works fine. As does voice mail coming from a digital receiptionist. just not when you transfer someone to an extension and they don't answer. they just hear hold music and never get voice mail options.