I have been trying to configure QoS settings on our small fleet of polycom VVX 400 series phones and it appears that the phones are sending the "802.1Q User Priority" setting instead of the value that I am providing for the "IP DSCP (overrides IP ToS)" field. Is this a bug or am I misunderstanding how these settings work? I have attached an exported copy of an example phone's configuration and a screen capture of the wireshark trace that shows the incorrectly set DSCP values (DSCP is set to 40 and value should be 46).

Phone Model: VVX 400

Part Number: 3111-46157-002 Rev:A

MAC Address: 64:16:7F:A9:26:12

UC Software Version: 5.8.0.12848

Updater Version: 5.9.5.13643

Call Platform: SfB (Online)

Hello  ,

welcome to the Polycom Community.

Both the community's Must Read First and the community's FAQ reference the basic minimum information a new or follow up post should contain.

This ensures the questions having to be asked are limited and any new or follow up post contains the right amount of details to ensure any voluntary participant within the community does not spend additional time chasing basic information.

As a reminder the basic information asked for:

• Provide the exact Software Version of your Phone
• Provide the Phone Model
• Provide the Call Platform (aka openSIP,Skype for Business Online, Skype for Business on Premise, Lync)
• Additional Polycom Infrastructure (RPRM,PDMS or BToE)
• If applicable provide a backup of the phone in question

UC Software 4.0.0 or later via the Web Interface Utilities > Phone Backup & Restore > Phone Backup > Phone Backup. Please rename into .TXT or Zip the file to attach.
Since UC Software 5.9.0 simply provide this via the Web Interface Diagnostics > Download Support Information Package

• If possible provide a Log and either attach them or use the Code Tag.Consult the Troubleshooting Section found within the FAQ if applicable
• If possible provide the MAC Address or Serial of the device
• Provide details for example if the issue is a day 1 issue or only happened after an upgrade or any other relevant details
• For questions around Support please check here

Whilst providing some of these details may not directly impact any possible answer the community can provide, it does enable Polycom to have an overview of the current software used. In addition providing all details at the same time allow us to check logs or look up a potential support partners if an issue needs to come into support. It also enables us to verify the entitlement for using features.

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

Jun 17, 2015 Question:How can I add  a Speed Dial to the Phone?

Resolution: Please check => here <=

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

Hello  ,

welcome to the Polycom Community.

Our records show that VIDENDA DISTRIBUTION LIMITED sold the phone in 20/09/2013 so the phone may simply be defective.

Have you ensured the basics aka the cabling is fine ?

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

Hello  ,

welcome to the Polycom Community.

NETXUSA sold this phone back in 18/04/2013 so we assume this is a 2nd hand purchase as it must have worked at some point in its life.

The logs shows nothing but most likely this is a digitmap issue:

Oct 7, 2011 Question: Phone unable to Dial a number when Off Hook or on 2nd Call in a Conference or Digitmap issues

Resolution: Please check => here <=

and

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

Resolution: Please check => here <=

Next reply should at least contain a backup of the phone and valid logs.

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

@SteffenBaierUK, we actually purchased this phone new from Nextiva in 2013 when we were setting up service with them.  I have since moved our hosting to eTollFree and am trying to get a few of these phones to work with the new service.  I thought that would be a no-brainer, but apparently not!  Thanks for the tips, I will troubleshoot when I am back in front of the device and report back.

Yes, cables etc are all fine and I can see it on my network. I thought that it may need some settings adjusted so it could work as a standalone phone. Is there a complete guide to setup phone from scratch with all settings explained, network setting sip setting etc?

Hello  ,

The FAQ covers everything.

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

Resolution: Please check => here <=

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

@SteffenBaierUK, I changed the digitmap to match the one in the link you provided.  My starting digitmap was:

[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|**x.T|+x.T

Which I changed to:

[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT

This change did not solve my problem.  I also checked the second link you suggested regarding troubleshooting Polycom VoIP-related issues.  I confirmed that my Polycom log settings matched the recommendations.  I then rebooted the VVX 600 and attempted to dial an 11-digit number and a 10-digit number.  Both attempts timed out after about 30 seconds with a fast busy signal.  I then attempted to dial a 4-digit extension, which went through without issue.  I should also point out that a softphone running on a PC that is on the same LAN as the Polycom is able to send/receive calls using the same SIP account, so it seems unlikely that this is a LAN/router issue.

I have attached a zip archive containing the log file and phone backup.  Please advise of next steps.  Thanks again for your help - I am in way over my head on this one!

Hello,

We have a two Trio 8800 paired and connected via SIP to a Shoretel VOIP system. Trio is in default mode and Noiseblock is enabled. Trio is using the latest firmware (5.8.0.15024). Each Trio was given its own sip username and number before pairing.

When making normal phone calls, users on the other end say they can hear minute sounds in the room including people typing, people setting laptops on table, or tapping on table. It interrupts what they hear from us so conversation is lost. 

I've checked on Noiseblock and tried it enabled and disabled with small differences. Is there anything else I can check on? 

Thanks.

Hello  ,

this landed in the SPAM so I moved this for you.

You have not followed the FAQ's so I got limited information I can provide.

Please also remember that this is a open Forum with volunteers answering.

You called:

0208142626|sip  |1|00|[CInvite]: szDest  - 13129445852

And we tried to send this but the server never responded:

	Line 1735: 0208142626|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send   500 of max 31500
	Line 1739: 0208142627|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  1500 of max 31500
	Line 1743: 0208142629|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  3500 of max 31500
	Line 1749: 0208142633|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  7500 of max 31500
	Line 1753: 0208142641|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 15500 of max 31500
	Line 1757: 0208142657|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 31500 of max 31500

You then called

0208142707|sip  |1|00|[CInvite]: szDest  - 3129445852

And we tried to send this but the server never responded:

	Line 1863: 0208142708|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send   500 of max 31500
	Line 1867: 0208142709|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  1500 of max 31500
	Line 1871: 0208142711|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  3500 of max 31500
	Line 1876: 0208142715|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  7500 of max 31500
	Line 1880: 0208142723|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 15500 of max 31500
	Line 1884: 0208142739|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 31500 of max 31500

You then called

0208142753|sip  |1|00|[CInvite]: szDest  - 8203

I suggest you take this up with your service provider once you re-visited the FAQ's as the logging levels are not set to the recommended level.

If you still struggle please work with above named Provider to 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.

End Customers are unable to open a ticket directly with Polycom support.

As the unit is no longer within warranty please be prepared to Pay Per Incident / PPI. This is all outlined in detail here

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

Hello  ,

welcome to the Polycom Community.

Both the community's Must Read First and the community's FAQ reference the basic minimum information a new or follow up post should contain.

This ensures the questions having to be asked are limited and any new or follow up post contains the right amount of details to ensure any voluntary participant within the community does not spend additional time chasing basic information.

As a reminder the basic information asked for:

• Provide the exact Software Version of your Phone
• Provide the Phone Model
• Provide the Call Platform (aka openSIP,Skype for Business Online, Skype for Business on Premise, Lync)
• Additional Polycom Infrastructure (RPRM,PDMS or BToE)
• If applicable provide a backup of the phone in question

UC Software 4.0.0 or later via the Web Interface Utilities > Phone Backup & Restore > Phone Backup > Phone Backup. Please rename into .TXT or Zip the file to attach.
Since UC Software 5.9.0 simply provide this via the Web Interface Diagnostics > Download Support Information Package

• If possible provide a Log and either attach them or use the Code Tag.Consult the Troubleshooting Section found within the FAQ if applicable
• If possible provide the MAC Address or Serial of the device
• Provide details for example if the issue is a day 1 issue or only happened after an upgrade or any other relevant details
• For questions around Support please check here

Whilst providing some of these details may not directly impact any possible answer the community can provide, it does enable Polycom to have an overview of the current software used. In addition providing all details at the same time allow us to check logs or look up a potential support partners if an issue needs to come into support. It also enables us to verify the entitlement for using features.

I am unsure hwat you mean by pairing but are you Daisy-Chaing these ?

If yes they would not need seperate accounts. The 2nd Device should not have any configuration on it.

If this is not your issue either please provide more details or 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.

End Customers are unable to open a ticket directly with Polycom support.

If this is some sort of an Internet discounter providing your MAC address or your Polycom devices serial will enable us to 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

Thanks @SteffenBaierUK for rescuing me from spam!  I set my Global Log Level Limit to Debug, and my SIP Module Log Level Limit to Level 2.  I then tried to make an 11-digit call (which failed), make a 4-digit call (which succeeded), and receive an inbound call from the same 11-digit number I had previously tried to call (which succeeded).  I copied the portion of the log file related to these calls, which is attached.  I also tried the same calls with the SIP Module Log Level Limit set to Debug, and attached that log excerpt as well.  Anything jump out at you as needing attention?

Hello  ,

as previsoely outlined I would ask you to take this up with the service provider you are using.

0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    INVITE sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Supported: replaces,100rel
0209121515|sip  |0|00|    Allow-Events: conference,talk,hold
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Type: application/sdp
0209121515|sip  |0|00|    Content-Length: 534
0209121515|sip  |0|00|    
0209121515|sip  |0|00|    v=0
0209121515|sip  |0|00|    o=- 1549736115 1549736115 IN IP4 192.168.123.163
0209121515|sip  |0|00|    s=Polycom IP Phone
0209121515|sip  |0|00|    c=IN IP4 192.168.123.163
0209121515|sip  |0|00|    b=AS:512
0209121515|sip  |0|00|    t=0 0
0209121515|sip  |0|00|    a=sendrecv
0209121515|sip  |0|00|    m=audio 2222 RTP/AVP 9 102 0 8 18 127
0209121515|sip  |0|00|    a=rtpmap:9 G722/8000
0209121515|sip  |0|00|    a=rtpmap:102 G7221/16000
0209121515|sip  |0|00|    a=fmtp:102 bitrate=32000
0209121515|sip  |0|00|    a=rtpmap:0 PCMU/8000
0209121515|sip  |0|00|    a=rtpmap:8 PCMA/8000
0209121515|sip  |0|00|    a=rtpmap:18 G729/8000
0209121515|sip  |0|00|    a=fmtp:18 annexb=no
0209121515|sip  |0|00|    a=rtpmap:127 telephone-event/8000
0209121515|sip  |0|00|    m=video 2224 RTP/AVP 109 34
0209121515|sip  |0|00|    a=rtpmap:109 H264/90000
0209121515|sip  |0|00|    a=fmtp:109 profile-level-id=42800d; packetization-mode=0
0209121515|sip  |0|00|    a=rtpmap:34 H263/90000
0209121515|sip  |0|00|    a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
0209121515|sip  |0|00|<<< End of data send
0209121515|sip  |2|00|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x40f02468)
0209121515|sip  |3|00|CStkCall::NewCallState 'Dialtone'->'Proceeding' (0x1bccc38) m_hUI(0x1d7c4e8),Control Channel(0), ResponseCode(-1)
0209121515|sip  |2|00|SipOnEvCallNewState 0x1bccc38,0x1d7c4e8 2,Proceeding, ResponseCode:-1
0209121515|sip  |0|00|listener: Received packet from 23.253.126.46:5060
0209121515|sip  |0|00|listener: Received packet from 23.253.126.46:5060
0209121515|sip  |0|00|<<<Data Received UDP
0209121515|sip  |0|00|    SIP/2.0 100 Trying
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    User-Agent: FreeSWITCH-mod_sofia/1.5.13b+git~20140519T124739Z~ea78f4d0e8~64bit
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |1|00|SipOnCommand: response 100,INVITE fromtag :7C0AF40B-18A4362C toTag :(null)
0209121515|sip  |1|00|SipOnCommand: response 100,INVITE matches user 1 of 1 '8201'
0209121515|sip  |3|00|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 65 timeout(0x40f02468)
0209121515|sip  |2|00|[CTrans::ResponseProcess] INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 0 code 100, timeout=65 (0x40f02468)
0209121515|sip  |3|00|Use common source preference for incoming and outgoing calls
0209121515|sip  |0|00|[CCommand::NeedToProcessCID] cmdType = 1 cmdMessage = 100 g_csSipRequestSourceMessage = -1 g_csSipResponseSourceMessage = -1---
0209121515|sip  |3|00|GetRemotePartyAddress from 'To'
0209121515|sip  |3|00|CStkCall::OnEvNewDest (0x1bccc38) new display '' user '13129445852' old 'From' new 'To' source
0209121515|sip  |2|00|CStkCall::OnEvSubmitDest CallIdType(1)
0209121515|sip  |0|00|SipOnEvNewDest 0x1bccc38,0x1d7c4e8,13129445852,
0209121515|sip  |3|00|CStkCall::NewCallState 'Proceeding'->'Proceeding' (0x1bccc38) m_hUI(0x1d7c4e8),Control Channel(0), ResponseCode(-1)
0209121515|sip  |2|00|SipOnEvCallNewState 0x1bccc38,0x1d7c4e8 2,Proceeding, ResponseCode:-1
0209121515|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0209121515|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0209121515|sip  |0|00|<<<Data Received UDP
0209121515|sip  |0|00|    SIP/2.0 407 Proxy Authentication Required
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>;tag=e2Xjc9F8c228p
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    User-Agent: FreeSWITCH-mod_sofia/1.5.13b+git~20140519T124739Z~ea78f4d0e8~64bit
0209121515|sip  |0|00|    Accept: application/sdp
0209121515|sip  |0|00|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
0209121515|sip  |0|00|    Supported: timer, path, replaces
0209121515|sip  |0|00|    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
0209121515|sip  |0|00|    Proxy-Authenticate: Digest realm="17683.etollfree-cloud.net", nonce="52489691-af1e-4c7f-ba4a-38ae12a9b625", algorithm=MD5, qop="auth"
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |1|00|SipOnCommand: response 407,INVITE fromtag :7C0AF40B-18A4362C toTag :e2Xjc9F8c228p
0209121515|sip  |1|00|SipOnCommand: response 407,INVITE matches user 1 of 1 '8201'
0209121515|sip  |3|00|UA Client INVITE INVITE trans state 'proceeding'->'completed' by 407 resp 65 timeout(0x40f02468)
0209121515|sip  |3|00|407 challenge received
0209121515|sip  |2|00|SipCallState is not Idle, So send Re-INVITE

0209121515|sip  |2|00|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x40f03868)
0209121515|sip  |1|00|Digest authentication
0209121515|sip  |2|00|CTrans:: SendCommand | ProxyList NOT empty.
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CTrans:: SendCommand | this=0x40f02468, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0209121515|sip  |0|00|Trying to send data to Destination 23.253.126.46 on socket 225
0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    ACK sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>;tag=e2Xjc9F8c228p
0209121515|sip  |0|00|    CSeq: 1 ACK
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |0|00|<<< End of data send
0209121515|sip  |2|00|adjustRetransWhenTimerCreated UA Client INVITE ACK state 'completed' timeout=65 (0x40f02468)
0209121515|sip  |2|00|SendCommand: reqDest '17683.etollfree-cloud.net' isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0209121515|sip  |1|00|SendCommand: isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0209121515|sip  |1|00|CreateFailOverProxyList : Reg to Domain '17683.etollfree-cloud.net' nPort 5060, lkup 3
0209121515|sip  |1|00|CreateFailOverProxyList : For INVITE Request nPort 5060
0209121515|sip  |1|00|doDnsListLookup(udp): doDnsSrvLookupForARecordList for '17683.etollfree-cloud.net' port 5060 returned 1 results
0209121515|sip  |1|00|doDnsListLookup(udp): result 0 '23.253.126.46' port 5060 isInBound 0
0209121515|sip  |1|00|CreateFailOverProxyList : 'UDP Only' for '17683.etollfree-cloud.net' port 5060 IP 0 is '23.253.126.46' on udp port 5060
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0209121515|sip  |2|00|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0209121515|sip  |2|00|CreateFailOverProxyList : IP 1 is '23.253.126.46' on udp port 5060
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CTrans:: SendCommand | this=0x40f03868, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0209121515|sip  |0|00|Trying to send data to Destination 23.253.126.46 on socket 225
0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    INVITE sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bKdfed03bfBB468780
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    CSeq: 2 INVITE
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Supported: replaces,100rel
0209121515|sip  |0|00|    Allow-Events: conference,talk,hold
0209121515|sip  |0|00|    Proxy-Authorization: Digest username="8201", realm="17683.etollfree-cloud.net", nonce="52489691-af1e-4c7f-ba4a-38ae12a9b625", qop=auth, cnonce="1fym0U6hKHgNyuA", nc=00000001, uri="sip:13129445852@17683.etollfree-cloud.net:5060;user=phone", response="257e387cc678c68a81416de15b5eedae", algorithm=MD5
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Type: application/sdp
0209121515|sip  |0|00|    Content-Length: 534
0209121515|sip  |0|00|    
0209121515|sip  |0|00|    v=0
0209121515|sip  |0|00|    o=- 1549736115 1549736115 IN IP4 192.168.123.163
0209121515|sip  |0|00|    s=Polycom IP Phone
0209121515|sip  |0|00|    c=IN IP4 192.168.123.163
0209121515|sip  |0|00|    b=AS:512
0209121515|sip  |0|00|    t=0 0
0209121515|sip  |0|00|    a=sendrecv
0209121515|sip  |0|00|    m=audio 2222 RTP/AVP 9 102 0 8 18 127
0209121515|sip  |0|00|    a=rtpmap:9 G722/8000
0209121515|sip  |0|00|    a=rtpmap:102 G7221/16000
0209121515|sip  |0|00|    a=fmtp:102 bitrate=32000
0209121515|sip  |0|00|    a=rtpmap:0 PCMU/8000
0209121515|sip  |0|00|    a=rtpmap:8 PCMA/8000
0209121515|sip  |0|00|    a=rtpmap:18 G729/8000
0209121515|sip  |0|00|    a=fmtp:18 annexb=no
0209121515|sip  |0|00|    a=rtpmap:127 telephone-event/8000
0209121515|sip  |0|00|    m=video 2224 RTP/AVP 109 34
0209121515|sip  |0|00|    a=rtpmap:109 H264/90000
0209121515|sip  |0|00|    a=fmtp:109 profile-level-id=42800d; packetization-mode=0
0209121515|sip  |0|00|    a=rtpmap:34 H263/90000
0209121515|sip  |0|00|    a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
0209121515|sip  |0|00|<<< End of data send

we get a 407 challenge for our initial INVITE and the Server never responds to our new INVITE

You can compare the logs to the working scenario or if all fails follow up as already advised or await any other volunteers to comment.

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

I have a VVX411 running 5.5.2, and I recently purchased a Plantronics CS540 with an APP-51 EHS.

After connecting everything, and changing the hookswitch mode from "Regular" to "Plantronics EHS", the phone reboots and after rebooting, the system reverts back to "Regular". 

Any help is appreicated. 

Hello  ,

welcome to the Polycom Community.

Both the community's Must Read First and the community's FAQ reference the basic minimum information a new or follow up post should contain.

As you have not attached any kind of log or a backup we can only assume that the issue is caused by a configuration being used.

The suggested configuration was already provided and I attached a file which you should be import using the Web Intereface:

Utilities > Import & Export Configuration > Import Configuration

In addition UC Software 5.5.2 is no longer a supported version so please upgrade to something more current like UC Software 5.9.1

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

Hi Polycom.

This is a strange issue, we're experiencing white squares on our Polycom VVX400 & VVX600 dsiplay. These phones are are scheduled each night to update it's configuration

UC 5.4.4.2473

I've attached a screenshot of the issue. This problem goes away about restarting the phone

Thanks for the feedback @SteffenBaierUK.  As you suggested, I asked the service provider to look into this, but their position is that since two different softphones (X-Lite and UltraSIP) are able to send/receive calls from a PC running on the same network as the Polycom VVX 600, the issue must be somewhere in the Polycom settings.  They are not Polycom experts.  They asked that I do a firmware update and factory-reset on the Polycom, which I have already done, and have confirmed all of the settings I have entered into the Polycom, but otherwise they are not sure how to proceed.

If it is helpful, I have attached the debug log from one of the softphones (MicroSIP) while completing the 11-digit outside call referenced above.  Does anything jump out at you?

Just to add to this we have it set on the Polycom provisioning file to restart each night

cat site.cfg | grep polling

    <license.polling license.polling.time="02:00">

    </license.polling>

    <prov.polling prov.polling.enabled="1" prov.polling.mode="rel" prov.polling.period="3600" prov.polling.time="01:00" prov.polling.timeRandomEnd="06:00">

    </prov.polling>

I am losing my mind, I have freedom voice for my cloud phone services - they gave me a vvx250 since I needed a wifi option for my phone - I cannot run ethernet cables where I need to be.  I've tried 2 different (same model) Obi5gwifi dongles, the first would flash blue then nothing.  I spent an hour with FV on the phone - they don't help with support or install of the phone or dongle - Obi doesn't help and no one at Polycom will speak with me over the phone - so here I am.

With my new dongle I was able to get to the point of searching for networks - entered my network and key - it seemed to 'connect' but never fully - now it seems anytime I reset/reboot the phone it just keeps going in and out of wifi dongle connected/disconnected.  I scan for networks, reboot, etc and continually does the same thing over and over.  I'm at wits end and am ready to toss the FV service, this phone and dongle out the window and go some place else... ANY HELP OUT THERE?

Hello  ,

I cannot provide free support and the possible escalation via a PPI has already been outlined.

Stating this the only brief difference I can see at present is that the VVX 600 INVITE contains Video and Voice where the Softphone only contains Voice codecs.

I suggest you disable video on the phone and test.

If this fails the next step has been outlined.

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