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The HP Community is where owners of HP products, like you, volunteer to help each other find solutions.
HP Recommended

hi, i am running FreePBX and have many phones already working fine.  I am setting up a new ip5000 and I am using sip-tls and srtp.  the polycom seems to register fine:

 

5353/sip:5353@172.30.2.71:52771;transport=TLS  29bd90a285 Avail         8.346

I am able to make calls from the polycom with no issue, however, I cannot receive calls.  when someone trys to call the polycom, the line just hangs up

 

on asterisk console, this comes up wihen someone tries to call in

 

== Setting global variable 'SIPDOMAIN' to '172.30.2.1'

uepbx1*CLI> 
[2017-05-24 14:23:13] WARNING[15695][C-00000005]: chan_sip.c:22927 func_header_read: This function can only be used on SIP channels.
[2017-05-24 14:23:13] WARNING[15695][C-00000005]: chan_sip.c:22927 func_header_read: This function can only be used on SIP channels.
[2017-05-24 14:23:13] WARNING[15695][C-00000005]: chan_sip.c:22927 func_header_read: This function can only be used on SIP channels.
[2017-05-24 14:23:13] WARNING[15695][C-00000005]: chan_sip.c:22927 func_header_read: This function can only be used on SIP channels.

uepbx1*CLI> 
[2017-05-24 14:23:13] WARNING[15695][C-00000005]: taskprocessor.c:888 taskprocessor_push: The 'subp:PJSIP/5323-00000021' task processor queue reached 500 scheduled tasks.

uepbx1*CLI> 
  == Spawn extension (from-internal, 5353, 1) exited non-zero on 'PJSIP/5353-0000000b'

uepbx1*CLI> 
  == Everyone is busy/congested at this time (1:0/0/1)

uepbx1*CLI> 
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'PJSIP/5323-0000000a' in macro 'exten-vm'
  == Spawn extension (ext-local, 5353, 2) exited non-zero on 'PJSIP/5323-0000000a'

uepbx1*CLI> 
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/5323-0000000a' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/5323-0000000a'

i am assuming i screwed something up in the polycom config, but i am not sure what....can someone point me in the right direction?

 

thanks

 

6 REPLIES 6
HP Recommended

Hello Flightline,

welcome back to the Polycom Community.

 

Sharing your configuration would make things easier rather than letting us guess.

The community's VoIP FAQ contains this post here:

Oct 7, 2011 Question: Can I register or is my Polycom Phone compatible with a “XYZ” SIP Server?

Resolution: Please check => here <=

 

or 

 

Jan 19, 2012 Question: How to troubleshoot Polycom VoIP related Issues?

Resolution: Please check => here <=


Please ensure you always check the community FAQ and/or utilize the community search before posting any new topics or follow up post’s.

 

The next step would be a support ticket.


In order to raise a support ticket you need to work with your Polycom reseller as they need to do this for you.

If this is some sort of an Internet discounter please post either your phone's MAC address or your Polycom devices serial so I can look up who would be able to support you. This may not be who you purchased the Polycom device from.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
HP Recommended

hi, thanks for the feedback, i have attached my configs.  in addition, i turned up the logging on the phone and see this in the log. 

 

0525152400|sip  |1|03|Server State finished OPTIONS (0x94e90f38)
0525152400|sip  |1|03|Server State finished NOTIFY (0x94e90c44)
0525152400|sip  |3|03|CUser::TimeOut500ms: Destroying NoCall object 'Unknown' of type 'CCallNoCall' (0x94e83588)
0525152400|sip  |3|03|CCallNoCall::NewCallState 'Unknown'->'Idle' (0x94e83588)
0525152401|sip  |3|03|CUser::TimeOut500ms: Destroying NoCall object 'Unknown' of type 'CCallNoCall' (0x94e831e4)
0525152401|sip  |3|03|CCallNoCall::NewCallState 'Unknown'->'Idle' (0x94e831e4)
0525152411|copy |*|03|Server '172.30.2.1' said 'languages/Website_dictionary_language_en-us.xml' is not present
0525152411|cfg  |4|03|Web|[cfgSaProcessRequestC] Failed to download language file from provisioning server, request path Website_dictionary_language_en-us.xml
0525152411|utilm|4|03|uBLFCompressed: File /ffs0/languages/Website_dictionary_language_en-us.xml.zzz does not exist or is empty
0525152411|cfg  |4|03|Web|[cfgSaProcessRequestC] Website_dictionary_language_en-us.xml Language file doesn't exist in cache
0525152411|cfg  |4|03|Web|[cfgSaProcessRequestC] Website_dictionary_language_en-us.xml Language file doesn't exist in phone flash
0525152414|sip  |1|03|MsgSipTcpPacket
0525152414|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0525152414|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
0525152414|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
0525152414|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal from pRequest To: 'Phoenix Conference Room' <5353@172.30.2.71:0>
0525152414|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config 'Phoenix Conference Room' <5353@172.30.2.1:5061>
0525152414|sip  |2|03|CStkDialog::CStkDialog TAG '564B1607-2EC5FF5E' generated
0525152414|sip  |2|03|CStkDialog::CStkDialog local addr 'Phoenix Conference Room' <5353@172.30.2.71:0> Tag '564B1607-2EC5FF5E'
0525152414|sip  |2|03|CStkDialog::CStkDialog exit 0x94e90c44 local list size 1
0525152414|sip  |2|03|CCallBase::IsChallenged COPIED Dialog Tag to pRequest '564B1607-2EC5FF5E'
0525152414|sip  |2|03|CCallBase::IsChallenged 'INVITE' Dialog Tag '564B1607-2EC5FF5E' pRequest Tag '564B1607-2EC5FF5E' state 'Trying'
0525152414|sip  |2|03|new UA Server INVITE trans state 'proceeding', timeout=0 (0x94e2b870)
0525152414|sip  |3|03|CStateInviteServer::CStateInviteServer central conf user user '' found in contact user 'asterisk' for cent conf URI ''. Set is focus
0525152414|sip  |2|03|CStkCall::VoiceFmtpReadLine SRTP mismatch.
0525152414|sip  |3|03|UA Server INVITE INVITE trans state 'proceeding'->'completed' by 488 resp 65 timeout(0x94e2b870)
0525152414|sip  |1|03|Dialog 'idd59d3900' State 'Trying'->'Terminated'
0525152414|sip  |1|03|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '172.30.2.1' port 5061 returned 1 results
0525152414|sip  |1|03|doDnsListLookup(tls): result 0 host '' IP '172.30.2.1' port 5061 isInBound 0
0525152414|sip  |1|03|CTcp::Send(TLS) entry for address 172.30.2.1 port 5061 can Connect 1 canFailOver 1
0525152414|sip  |1|03|CTcpSocket::SendData TLS queuedTxData = 0 TotalLen 553 loop count 1 maxQueueDepth 20000
0525152414|sip  |1|03|CTcpSocket::SendData TLS Sent 553 loop count 1
0525152414|sip  |2|03|adjustRetransWhenTimerCreated UA Server INVITE INVITE state 'completed' timeout=65 (0x94e2b870)
0525152414|sip  |1|03|MsgSipTcpPacket
0525152414|sip  |3|03|CStkCall::NewCallState 'Unknown'->'Idle' (0x94e7fa10)

i am not sure how to read this, but is this the issue:  and what problem does that point to?

 

0525152414|sip  |2|03|CStkCall::VoiceFmtpReadLine SRTP mismatch.

btw, i am able to make calls on the polycom, but not receive them.

thanks

 

HP Recommended

Hello Flightline,

I am unsure who provided or setup your configuration or your phone but they do not know how to do this.

 

As an example the site.cfg contains Device.X Parameters but it is missing the device.set="1" Parameter to activate these.

 

There are plenty of examples with all the other files that is outside of the scope for a Polycom employee to go over but I suggest the following:

 

  • Factory default the Phone
  • Follow this => here <=
  • If you need to use SRTP follow this => here <=

If this still fails open a ticket.


In order to raise a support ticket you need to work with your Polycom reseller as they need to do this for you.

If this is some sort of an Internet discounter please post either your phone's MAC address or your Polycom devices serial so I can look up who would be able to support you. This may not be who you purchased the Polycom device from.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
HP Recommended

Steffen,

  thanks for your helpful advice.  I have factory reset the phone and re-red the faq's you  pointed out.  I have attached my new configs.

in addition, here are some logs.  in short, i can receive calls without issue.  but when i make a call, there is no audio.

 

your help is greatly appreciated.

 

thanks

 

0601135225|sip  |2|03|SipCallNew 0 local port 2226 SDP 0 call appearance -1 IsRtrv 0 dialog 0
0601135225|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config 'Phoenix Conference Room' <5353@172.30.2.1:5061>
0601135225|sip  |2|03|CStkDialog::CStkDialog AddressLocal set to Config
0601135225|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
0601135225|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
0601135225|sip  |2|03|CStkDialog::CStkDialog TAG '5569B6F2-57588A63' generated
0601135225|sip  |2|03|CStkDialog::CStkDialog local addr 'Phoenix Conference Room' <5353@172.30.2.1:5061> Tag '5569B6F2-57588A63'
0601135225|sip  |2|03|CStkDialog::CStkDialog exit 0x94e90f38 local list size 1
0601135225|sip  |2|03|CStkDialogList::CreateDialogObject localTarg usr '5353' 
0601135225|sip  |2|03|CUser::CallNew 0x9570f8e0 0x94e7fa10 CallAppr 0 IsRetrieve 0 ThrdParty '' Dialog 0x0 isCentConf 0
0601135225|sip  |3|03|CStkCall::NewCallState 'Unknown'->'Dialtone' (0x94e7fa10)
0601135225|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 0,Dialtone
0601135225|sip  |2|03|SipCallMake 5323
0601135225|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x94e2c4d0)
0601135225|sip  |3|03|AddIceDescription: No SDP to add
0601135225|sip  |1|03|CreateFailOverProxyList : Reg to Domain '172.30.2.1' nPort 5061
0601135225|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5061
0601135225|sip  |1|03|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '172.30.2.1' port 5061 returned 1 results
0601135225|sip  |1|03|doDnsListLookup(tls): result 0 host '' IP '172.30.2.1' port 5061 isInBound 0
0601135225|sip  |1|03|CreateFailOverProxyList : 'TLS' for '172.30.2.1' port 5061 IP 0 is '172.30.2.1' on tls port 5061
0601135225|sip  |1|03|CreateFailOverProxyList : 'TLS' Add rest Total to Try 1
0601135225|sip  |2|03|CreateFailOverProxyList : Exit 'TLS' lookup with 1 IP Addresses
0601135225|sip  |2|03|CreateFailOverProxyList : IP 1 is '172.30.2.1' on tls port 5061
0601135225|sip  |1|03|CTcp::Send(TLS) entry for address 172.30.2.1 port 5061 can Connect 1 canFailOver 0
0601135225|sip  |1|03|CTcpSocket::SendData TLS queuedTxData = 0 TotalLen 1241 loop count 1 maxQueueDepth 20000
0601135225|sip  |1|03|CTcpSocket::SendData TLS Sent 1241 loop count 1
0601135225|sip  |3|03|CStkCall::NewCallState 'Dialtone'->'Proceeding' (0x94e7fa10)
0601135225|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 2,Proceeding
0601135225|sip  |1|03|MsgSipTcpPacket
0601135225|sip  |1|03|SipOnCommand: response 401,INVITE
0601135225|sip  |1|03|SipOnCommand: response 401,INVITE matches user 1 of 1 '5353'
0601135225|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'completed' by 401 resp 65 timeout(0x94e2c4d0)
0601135225|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'completed' state at retryCount 0 code 401, timeout=65 (0x94e2c4d0)
0601135225|sip  |3|03|401 challenge received
0601135225|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x94e2edf0)
0601135225|sip  |1|03|Digest authentication
0601135225|sip  |1|03|CTcp::Send(TLS) entry for address 172.30.2.1 port 5061 can Connect 1 canFailOver 1
0601135225|sip  |1|03|CTcpSocket::SendData TLS queuedTxData = 0 TotalLen 604 loop count 1 maxQueueDepth 20000
0601135225|sip  |1|03|CTcpSocket::SendData TLS Sent 604 loop count 1
0601135225|sip  |1|03|CreateFailOverProxyList : Reg to Domain '172.30.2.1' nPort 5061
0601135225|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5061
0601135225|sip  |1|03|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '172.30.2.1' port 5061 returned 1 results
0601135225|sip  |1|03|doDnsListLookup(tls): result 0 host '' IP '172.30.2.1' port 5061 isInBound 0
0601135225|sip  |1|03|CreateFailOverProxyList : 'TLS' for '172.30.2.1' port 5061 IP 0 is '172.30.2.1' on tls port 5061
0601135225|sip  |1|03|CreateFailOverProxyList : 'TLS' Add rest Total to Try 1
0601135225|sip  |2|03|CreateFailOverProxyList : Exit 'TLS' lookup with 1 IP Addresses
0601135225|sip  |2|03|CreateFailOverProxyList : IP 1 is '172.30.2.1' on tls port 5061
0601135225|sip  |1|03|CTcp::Send(TLS) entry for address 172.30.2.1 port 5061 can Connect 1 canFailOver 0
0601135225|sip  |1|03|CTcpSocket::SendData TLS queuedTxData = 645 TotalLen 1544 loop count 1 maxQueueDepth 20000
0601135225|sip  |1|03|CTcpSocket::SendData TLS Sent 1544 loop count 1
0601135225|sip  |1|03|MsgSipTcpPacket
0601135225|sip  |1|03|SipOnCommand: response 100,INVITE
0601135225|sip  |1|03|SipOnCommand: response 100,INVITE matches user 1 of 1 '5353'
0601135225|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 64 timeout(0x94e2edf0)
0601135225|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 0 code 100, timeout=64 (0x94e2edf0)
0601135225|sip  |3|03|GetRemotePartyAddress from 'To'
0601135225|sip  |3|03|CStkCall::OnEvNewDest (0x94e7fa10) new display '' user '5323' old 'From' new 'To' source
0601135225|sip  |2|03|SipOnEvNewDest 94e7fa10,9570f8e0,5323,
0601135225|sip  |3|03|CStkCall::NewCallState 'Proceeding'->'Proceeding' (0x94e7fa10)
0601135225|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 2,Proceeding
0601135225|sip  |1|03|MsgSipTcpPacket
0601135225|sip  |1|03|SipOnCommand: response 180,INVITE
0601135225|sip  |1|03|SipOnCommand: response 180,INVITE matches user 1 of 1 '5353'
0601135225|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0601135225|sip  |3|03|GetRemotePartyAddress from 'To'
0601135225|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '' user '5323'
0601135225|sip  |1|03|Dialog 'id9b4c8575' State 'Trying'->'Early'
0601135225|sip  |3|03|CStkCall::NewCallState 'Proceeding'->'RingBack' (0x94e7fa10)
0601135225|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 3,RingBack
0601135225|sip  |1|03|MsgSipTcpPacket
0601135225|sip  |1|03|SipOnCommand: response 180,INVITE
0601135225|sip  |1|03|SipOnCommand: response 180,INVITE matches user 1 of 1 '5353'
0601135225|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0601135225|sip  |3|03|GetRemotePartyAddress from 'To'
0601135225|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '' user '5323'
0601135225|sip  |3|03|CStkCall::NewCallState 'RingBack'->'RingBack' (0x94e7fa10)
0601135225|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 3,RingBack
0601135226|sip  |1|03|MsgSipTcpPacket
0601135226|sip  |1|03|SipOnCommand: response 200,INVITE
0601135226|sip  |1|03|SipOnCommand: response 200,INVITE matches user 1 of 1 '5353'
0601135226|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0601135226|sip  |3|03|UA Client INVITE INVITE trans state 'proceeding'->'terminated' by 200 resp 0 timeout(0x94e2edf0)
0601135226|sip  |2|03|SendCommand: reqDest '172.30.2.1' isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0601135226|sip  |1|03|SendCommand: isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0601135226|sip  |1|03|CreateFailOverProxyList : Reg to Domain '172.30.2.1' nPort 5061
0601135226|sip  |1|03|CreateFailOverProxyList : For ACK Request nPort 5061
0601135226|sip  |1|03|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '172.30.2.1' port 5061 returned 1 results
0601135226|sip  |1|03|doDnsListLookup(tls): result 0 host '' IP '172.30.2.1' port 5061 isInBound 0
0601135226|sip  |1|03|CreateFailOverProxyList : 'TLS' for '172.30.2.1' port 5061 IP 0 is '172.30.2.1' on tls port 5061
0601135226|sip  |2|03|CreateFailOverProxyList : Exit 'TLS' lookup with 1 IP Addresses
0601135226|sip  |2|03|CreateFailOverProxyList : IP 1 is '172.30.2.1' on tls port 5061
0601135226|sip  |1|03|CTcp::Send(TLS) entry for address 172.30.2.1 port 5061 can Connect 1 canFailOver 1
0601135226|sip  |1|03|CTcpSocket::SendData TLS queuedTxData = 0 TotalLen 600 loop count 1 maxQueueDepth 20000
0601135226|sip  |1|03|CTcpSocket::SendData TLS Sent 600 loop count 1
0601135226|sip  |3|03|GetRemotePartyAddress from 'To'
0601135226|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '' user '5323'
0601135226|sip  |2|03|CStateInviteClient::OnEvResponse Normal case
0601135226|sip  |3|03|CStkCall::RemoteSdpAnswer(1) -> ReportCodec( 1)
0601135226|sip  |2|03|CStkCall::ReportCodec: held set to false 
0601135226|sip  |2|03|SipOnEvNewCodec ac1e0201,0 0 PCMU/8000 12522,2226 ptime=20,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
0601135226|sip  |2|03|SipOnEvNewCodec ac1e0201,127 127 telephone-event/8000 12522,2226 ptime=0,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
0601135226|sip  |3|03|CStkCall::ReportCodec: call state 'RingBack' exit with held 0 (0x94e7fa10)
0601135226|sip  |1|03|Dialog 'id9b4c8575' State 'Early'->'Confirmed'
0601135226|sip  |3|03|CStkCall::NewCallState 'RingBack'->'Connected' (0x94e7fa10)
0601135226|sip  |2|03|SipOnEvCallNewState 94e7fa10,9570f8e0 4,Connected

i see a lot of these, is this normal?

 

0601135356|so   |4|03|soStream: RTCP PacketLength extends past end of buffer. Type=85 PLen-4=81424 Pos=32 BLen=54
0601135401|so   |4|03|soStream: RTCP PacketLength extends past end of buffer. Type=76 PLen-4=242924 Pos=32 BLen=54
0601135406|so   |4|03|soStream: RTCP PacketLength extends past end of buffer. Type=159 PLen-4=244968 Pos=32 BLen=54
0601135411|so   |4|03|soStream: RTCP PacketLength extends past end of buffer. Type=51 PLen-4=152992 Pos=32 BLen=54
0601135416|so   |4|03|soStream: RTCP PacketLength extends past end of buffer. Type=160 PLen-4=93560 Pos=32 BLen=54

 

HP Recommended

Hello Flightline,

I already asked you to open a ticket.

In addition it is always useful to include the currently used SIP or UC Software version as issues experienced or a question asked may already be addressed in a newer release.

lf and others to check against current software release notes, Administrator Guides or FAQ post’s.

The above is also stated in the "Must Read First"

Therefore the Polycom VoIP FAQ contains this post here:

Question: How can I find out my SIP or UC Software Version of my Phone?
Resolution: Please check here

 

You can keep posting information but from my perspective you need to work with your Polycom reseller.


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
HP Recommended

Hello Flightline,

looking briefly at your Configuration the community's VoIP FAQ contains this post here:

Jun 30, 2015 Question:What is the difference using the UC Software 4.0.0 for compatible SoundPoint or SoundStation IP Phones?

Resolution: Please always check the release Notes or

Software Version  Call Server
4.0.X SIP Only
4.1.X LYNC Only

 

If this still fails after upgrading to the correct software please open a ticket!


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
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Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
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