UCMen33260
Technical User
Hi team,
I deployed 2 IPO Select with the latest release (SP2) (primary and secondary) and an SBCE cluster used only for remote workers.
All is work fine for my remote workers.
But, when the primary IPO fails, remote worker (registered correctly on the secondary IPO) cannot call the voicemail (*17).
I have 200 vmpro ports one the primary IPO, and the voicemail configuration is on "centralized mode ).
The error on tracesbc is:
│SIP/2.0 480 Temporarily Unavailable │
18:39:00.302 │ │Via: SIP/2.0/TLS 10.219.6.93:5061;branch=z9hG4bK-s1632-000273005535-1--s1632- │
18:39:00.302 │ │Record-Route: <sip:10.219.6.93:5061;ipcs-line=4346;lr;transport=tls;subid_ipcs=3174493599> │
18:39:00.302 │ │From: <sip:100@mydomain.com>;tag=d152d106-984f-4fef-902f-d8e9fc15c6c2 │
18:39:00.302 │ │Call-ID: e74169fe-aec6-4643-95dd-0575ec61b458 │
18:39:00.302 │ │CSeq: 3 INVITE │
18:39:00.302 │◄─Te│Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE │
18:39:00.302 │◄──2│Supported: timer,100rel │
18:39:00.302 │────│Server: IP Office 11.0.4.2.0 build 58 │
18:39:00.302 │──IN│Reason: Q.850;cause=34;text="No circuit/channel available" │
18:39:00.302 │◄──T│Content-Length: 0 │
18:39:00.302 │ │To: <sip:*17@mydomain.com>;tag=7d7eb501ac0740c2
And on Monitor logs, it's the same issue, and error code:
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) INVITE Received ep 0adb065b0000041a 10004.1050.0 23 test1.0(f62b3c10), dialog f62bace8
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) LR is On and route is Route: <sip:10.219.6.93:5061;ipcs-line=4016;lr;transport=tls;subid_ipcs=3174493599>
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_RCVD(2)
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SdpClone
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::BuildFastStartFromSDPOffer sdpmsg f7284d10
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::FindConnectionParameters: found media proto <RTP/AVP>
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) BuildFastStartFromSDPOffer: known_stream 1 media_type 0 stream_type 5 allow_stream 1
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::BuildFastStartFromAudioMediaSDPOffer reinvite 0 mRTP_PType 255 prefer_existing_codec 0
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) HandleSDPContext: found_rfc2833 with payload 101
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SetRfc2833TxPayload: use RFC2833 for dtmf
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SetRfc2833TxPayload: payload 101 set_rx 1
7319528mS Sip: FindContactParameters mFarDisplayString <> mFarDisplayNumber <100> restricted 0
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) Process SIP request dialog f62bace8, method INVITE in state SIPDialog::INVITE_RCVD(9)
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateClone mMesg f62af430 smsg f62a7560
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) CMSetup forwarded to call model owner_ep 0adb065b0000041a 10004.1050.0 23 test1.0(f62b3c10), dialog f62bace8 has sdp 1
7319529mS Sip: 0adb065b0000041a 10004.1050.0 -1 test1.0(f62bace8) Terminating dialog f62bace8, state SIPDialog::INVITE_RCVD(9) for cause CMCauseNoChannel
7319529mS Sip: 0adb065b0000041a 10004.1050.0 -1 test1.0(f62bace8) SendSIPResponse: INVITE code 480 SENT TO 10.219.6.93 5061
7319529mS SIP Call Tx: phone
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TLS 10.219.6.93:5061;branch=z9hG4bK-s1632-000040841991-1--s1632-
Record-Route: <sip:10.219.6.93:5061;ipcs-line=4016;lr;transport=tls;subid_ipcs=3174493599>
From: <sip:100@mydomain.com>;tag=54333172-2ede-4df8-aa90-a3e04c1e64ac
Call-ID: 52ba2706-beb1-4b8d-851d-2d05bb71beae
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 11.0.4.2.0 build 58
Reason: Q.850;cause=34;text="No circuit/channel available"
Content-Length: 0
To: <sip:*17@mydomain.com>;tag=44bd9df81a001b01
someone have the same issue on this IPO release?
Regards
I deployed 2 IPO Select with the latest release (SP2) (primary and secondary) and an SBCE cluster used only for remote workers.
All is work fine for my remote workers.
But, when the primary IPO fails, remote worker (registered correctly on the secondary IPO) cannot call the voicemail (*17).
I have 200 vmpro ports one the primary IPO, and the voicemail configuration is on "centralized mode ).
The error on tracesbc is:
│SIP/2.0 480 Temporarily Unavailable │
18:39:00.302 │ │Via: SIP/2.0/TLS 10.219.6.93:5061;branch=z9hG4bK-s1632-000273005535-1--s1632- │
18:39:00.302 │ │Record-Route: <sip:10.219.6.93:5061;ipcs-line=4346;lr;transport=tls;subid_ipcs=3174493599> │
18:39:00.302 │ │From: <sip:100@mydomain.com>;tag=d152d106-984f-4fef-902f-d8e9fc15c6c2 │
18:39:00.302 │ │Call-ID: e74169fe-aec6-4643-95dd-0575ec61b458 │
18:39:00.302 │ │CSeq: 3 INVITE │
18:39:00.302 │◄─Te│Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE │
18:39:00.302 │◄──2│Supported: timer,100rel │
18:39:00.302 │────│Server: IP Office 11.0.4.2.0 build 58 │
18:39:00.302 │──IN│Reason: Q.850;cause=34;text="No circuit/channel available" │
18:39:00.302 │◄──T│Content-Length: 0 │
18:39:00.302 │ │To: <sip:*17@mydomain.com>;tag=7d7eb501ac0740c2
And on Monitor logs, it's the same issue, and error code:
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) INVITE Received ep 0adb065b0000041a 10004.1050.0 23 test1.0(f62b3c10), dialog f62bace8
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) LR is On and route is Route: <sip:10.219.6.93:5061;ipcs-line=4016;lr;transport=tls;subid_ipcs=3174493599>
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_RCVD(2)
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SdpClone
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::BuildFastStartFromSDPOffer sdpmsg f7284d10
7319527mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::FindConnectionParameters: found media proto <RTP/AVP>
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) BuildFastStartFromSDPOffer: known_stream 1 media_type 0 stream_type 5 allow_stream 1
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SIPDialog::BuildFastStartFromAudioMediaSDPOffer reinvite 0 mRTP_PType 255 prefer_existing_codec 0
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) HandleSDPContext: found_rfc2833 with payload 101
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SetRfc2833TxPayload: use RFC2833 for dtmf
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) SetRfc2833TxPayload: payload 101 set_rx 1
7319528mS Sip: FindContactParameters mFarDisplayString <> mFarDisplayNumber <100> restricted 0
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) Process SIP request dialog f62bace8, method INVITE in state SIPDialog::INVITE_RCVD(9)
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) UpdateClone mMesg f62af430 smsg f62a7560
7319528mS Sip: 0adb065b0000041a 10004.1050.0 23 test1.0(f62bace8) CMSetup forwarded to call model owner_ep 0adb065b0000041a 10004.1050.0 23 test1.0(f62b3c10), dialog f62bace8 has sdp 1
7319529mS Sip: 0adb065b0000041a 10004.1050.0 -1 test1.0(f62bace8) Terminating dialog f62bace8, state SIPDialog::INVITE_RCVD(9) for cause CMCauseNoChannel
7319529mS Sip: 0adb065b0000041a 10004.1050.0 -1 test1.0(f62bace8) SendSIPResponse: INVITE code 480 SENT TO 10.219.6.93 5061
7319529mS SIP Call Tx: phone
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TLS 10.219.6.93:5061;branch=z9hG4bK-s1632-000040841991-1--s1632-
Record-Route: <sip:10.219.6.93:5061;ipcs-line=4016;lr;transport=tls;subid_ipcs=3174493599>
From: <sip:100@mydomain.com>;tag=54333172-2ede-4df8-aa90-a3e04c1e64ac
Call-ID: 52ba2706-beb1-4b8d-851d-2d05bb71beae
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 11.0.4.2.0 build 58
Reason: Q.850;cause=34;text="No circuit/channel available"
Content-Length: 0
To: <sip:*17@mydomain.com>;tag=44bd9df81a001b01
someone have the same issue on this IPO release?
Regards