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

We've got a trio 8800 which is registered with an Office 365 account with a Skype for Business Online (Plan 2) license plus Skype for Business PSTN Conferencing and Consumption. When logged into the Windows SfB client the user can dial out using PSTN. However, I can't dial out from the trio 8800, when I try to add someone via PSTN it fails with an error.

 

Should I be able to dial out to a PSTN number from the trio 8800?

 

If so, any idea why this isn't working?

 

Firmware is 5.4.4.7776.

 

Cheers,

Dan

14 REPLIES 14
HP Recommended

Hello dnichols,

welcome to the Polycom Community.

 

What error do you get ?

The community's Skype for Business FAQ contains this post here:

Jan 17, 2017 Question:How can I troubleshoot simple Skype for Bussines, LYNC or Office365 issues?

Resolution: Have a look => 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

------------------------------------------------
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 Dan,

 

Have you found a solution to this problem yet? I am experiencing the exact same issue - when trying to place a call to a PSTN number via Skype for Business from the Trio 8800, the call will not go through. I am also able to place the call successfully via Skype for Business on Windows when testing with the same user account. This account is an Office 365 user account with Skype Conference and Skype Consumption licenses.

 

I'm also running firmware version Version 5.4.4 Rev AD from the website. I've tested entering the PSTN number in various formats, but all unsuccessful. From the device logs, it seems that Skype for Business is treating the PSTN number entered as an SfB account name, and thus giving a not found error.

 

I'm pasting two sections of the Trio 8800 log where I could find related information. I did change the phone number and domain/account name for privacy:

 

0326180448|sip |3|00|GetRemotePartyAddress from 'To' 
0326180448|sip |3|00|CStkCall::OnEvNewDest (0x0x2b26630) new display '' user '+27721234567' old 'From' new 'To' source 
0326180448|sip |2|00|SipOnEvNewDest 2b26630,324ca00,sip:+27721234567@mydomain.com, 
0326180448|sip |3|00|SipOnEvNewDest: Updating Original Destination 
0326180449|sip |3|00|UA Client INVITE INVITE trans state 'proceeding'->'completed' by 404 resp 65 timeout(0x412a2088) 
0326180449|sip |3|00|getMSDiagnostics- Cause(1003), reason(User does not exist) 
0326180449|sip |2|00|CTrans:: SendCommand | ProxyList NOT empty. 
0326180449|sip |3|00|GetRemotePartyAddress from 'To'

0326173151|sip |2|00|Found ICE m-line "m=audio 50014 RTP/AVP 0" for 1 lines 
0326173151|sip |2|00|ByPassEnable is:[1],BypassId:[e584181b-a0c7-43d0-a0a5-00aa34dce451],RemoteUser:[1] 
0326173151|sip |2|00|getMsConvId : Generating convID(AdKmPbWSOTM3YmE3YjU0MDZkZjBlNg==) for CStkCall(0x2b25668) 
0326173151|sip |2|00|CreateFailOverProxyList : Exit 'TLS' lookup with 1 IP Addresses 
0326173151|sip |2|00|CreateFailOverProxyList : IP 1 is '52.112.194.22' on tls port 443 
0326173151|sip |3|00|CStkCall::NewCallState 'Dialtone'->'Proceeding' (0x0x2b25668) m_hUI(0x0x3290d58),Control Channel(0) 
0326173151|sip |2|00|SipOnEvCallNewState 2b25668,3290d58 2,Proceeding 
0326173151|sip |3|00|UA Client INVITE INVITE trans state 'callingTrying'->'completed' by 404 resp 65 timeout(0x412a5c08) 
0326173151|sip |2|00|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'completed' state at retryCount 0 code 404, timeout=65 (0x412a5c08) 
0326173151|sip |3|00|getMSDiagnostics- Cause(14010), reason(Unable to find an exact match in the rules set) 
0326173151|sip |2|00|CTrans:: SendCommand | ProxyList NOT empty. 
0326173151|sip |3|00|GetRemotePartyAddress from 'To' 
0326173151|sip |3|00|CStkCall::parseDiagnostics Call(0x0x2b25668), FailureReason = 14010;reason="Unable to find an exact match in the rules set";source="DB41E05FES14.INFRA.LYNC.COM";CalledNumber="#27721234567";ProfileName="za";appName="TranslationService" 
0326173151|sip |3|00|CStkCall::NewCallState 'Proceeding'->'Idle' (0x0x2b25668) m_hUI(0x0x3290d58),Control Channel(0) 
0326173151|sip |2|00|SipOnEvCallNewState 2b25668,3290d58 10,Idle 
0326173151|sip |3|00|GetRemotePartyAddress from 'To' 
0326173151|sip |3|00|CStkCall::OnEvNewDest (0x0x2b25668) new display '' user '%2327721234567;phone-context=za' old 'From' new 'To' source 
0326173151|sip |2|00|SipOnEvNewDest 2b25668,3290d58,#27721234567, 
0326173151|sip |3|00|SipOnEvNewDest: Updating Original Destination 
0326173151|sip |2|00|[CBAStkIntf::publishDialogUpdate()] publish dialog for call(2b25668) 
0326173151|sip |2|00|CBossUserOper::publishDialog - bossUri(myuser@mydomain.com) ; m_bisBossCall(0); pCall(2b25668)

Everything else is working, I can dial/add other Skype for Business participants to a meeting on the Trio 8800 etc, I just cannot dial any PSTN number via Skype for Business on the device.

 

 

Thanks,

John

 

HP Recommended

Hello John,

welcome to the Polycom Community.

 

As already replied to the original poster:

The community's Skype for Business FAQ contains this post here:

Jan 17, 2017 Question:How can I troubleshoot simple Skype for Bussines, LYNC or Office365 issues?

Resolution: Have a look => here <=

 

Can you get us some logs ?

 

In addition the VoIP Community FAQ contains:

 

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 <=

 

The above shows example logging for Digitmap issues.

 

If all of the above fails please contact 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

Hi Steffen Baier,

 

I did have a look at the Skype for Business FAQ, but it did not help resolving the issue we're expericing.

 

The "Phone unable to Dial a number when Off Hook or on 2nd Call in a Conference or Digitmap issues" does not seem to be applicable:

- Phone is not off hook

- The problem occurs when not on a call (so 2nd Call in a Conference is not applicable).

- The Digitmap is empty in the SIP settings, is a Digitmap required for Skype for Business to be able to phone out via PSTN?

 

Also, I did provide the log extract that you've requested. Can I please get someone from Polycom technical support to have a look at the log extracts and provide feedback?

HP Recommended

Hello John,

I am from technical support and again the logs are not in debug mode as shown and suggested in the referred FAQ post.

 

In addition the referred digitmap example contains a log section which can help us to pin point the issue.

 

Please open a ticket with your reseller.


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.

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 Steffen,

 

Ok I've now set the logging as per FAQ:

Settings > Logging > Global Settings > Global Log Level Limit > Set to DEBUG
Settings > Logging > Log File Size (Kbytes) > 512
Settings > Logging > Module Log Level Limits > SIP > Set to DEBUG
Settings > Logging > Module Log Level Limits > Support Object Log > Event 1

 

Log exported and attached.

 

My Trio 8800 MAC address: 00:04:F2:FD:BA:40

 

Thanks,

John

HP Recommended

Hello John,

I can see the Phone sends 27721234567 to Office 365.

 

0327115523|so |3|00|[SoDigitMapElementC]: Checking 27721234567 (11) against x.T (0)
0327115523|so |3|00|[SoDigitMapElementC]: x.T (0) (27721234567) - Delay match until time-out
0327115523|so |3|00|[SoDigitMapC]: dial(char*) Map Element 0 (0x41fe190 - x.T - 0) declared match (4)0

This is send directly and un-manipulated to the Skype for Business Server.

 

The server then decides that this is not a valid number or he cannot find it:

 

0327115523|sip |0|00| SIP/2.0 101 Progress Report
0327115523|sip |0|00| ms-user-logon-data: RemoteUser
0327115523|sip |0|00| Authentication-Info: TLS-DSK qop="auth", opaque="215FBE6D", srand="B33FF64B",
snum="146", rspauth="ca4c6deb1e627b58845e6b46e49b05325eba70e6", targetname="DB41E05FES03.infra.lync.com",
realm="SIP Communications Service", version=4
0327115523|sip |0|00| Via: SIP/2.0/TLS
192.168.100.138:51176;branch=z9hG4bKf37ada422A05C923;received=13.100.5.254;ms-received-port=62512;msreceived-cid=2D22600
0327115523|sip |0|00| From: "IT _removed_privacy_" <sip:_removed_privacy_@_clientdomain_.com>;tag=8FCC5807-CE227F55;epid=0004f2fdba40
0327115523|sip |0|00| To: <sip:27721234567;phone-context=za@_client-domain_.com;user=phone>
0327115523|sip |0|00| Call-ID: 572865d2615d1a54a4590fbbdefdba40
0327115523|sip |0|00| CSeq: 1 INVITE
0327115523|sip |0|00| ms-telemetry-id: FD45BE66-E2DF-5755-B0F3-FD4F2E806CF3
0327115523|sip |0|00| ms-diagnostics: 14011;reason="Called Number
translated";source="DB41E05FES03.INFRA.LYNC.COM";RuleName="ZA
Default";CalledNumber="27721234567";TranslatedNumber="+2727721234567";appName="TranslationService"
0327115523|sip |0|00| Server: TranslationService/7.0.0.0
0327115523|sip |0|00| Content-Length: 0
0327115523|sip |0|00|
0327115523|sip |0|00|<<< Data received TLS
0327115523|sip |0|00| SIP/2.0 404 Not Found
0327115523|sip |0|00| ms-user-logon-data: RemoteUser
0327115523|sip |0|00| Authentication-Info: TLS-DSK qop="auth", opaque="215FBE6D", srand="57A04FD0",
snum="147", rspauth="b9c981fde7107550ae8275820090e6321817c9fe", targetname="DB41E05FES03.infra.lync.com",
realm="SIP Communications Service", version=4
0327115523|sip |0|00| Via: SIP/2.0/TLS
192.168.100.138:51176;branch=z9hG4bKf37ada422A05C923;received=13.100.5.254;ms-received-port=62512;msreceived-cid=2D22600
0327115523|sip |0|00| Content-Length: 0
0327115523|sip |0|00| From: "IT _removed_privacy_" <sip:_removed_privacy_@_clientdomain_.com>;tag=8FCC5807-CE227F55;epid=0004f2fdba40
0327115523|sip |0|00| To: <sip:27721234567;phone-context=za@_clientdomain_.com;user=phone>;tag=7CF54F5F0E86ACD7EED88533A0C28D76
0327115523|sip |0|00| Call-ID: 572865d2615d1a54a4590fbbdefdba40
0327115523|sip |0|00| CSeq: 1 INVITE
0327115523|sip |0|00| ms-diagnostics: 1003;reason="User does not exist";domain="_clientdomain_.com";source="sipfed1E.online.lync.com"
0327115523|sip |0|00| ms-telemetry-id: FD45BE66-E2DF-5755-B0F3-FD4F2E806CF3
0327115523|sip |0|00| Server: RTC/7.0

As you can see the number gets translated by Skype for Business from "27721234567" into "+2727721234567" basically adding an additional +27

 

That is not a Trio issue so I would look on your Skype for Business end via Microsoft support if needed.


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 Steffen,

 

Thank you for the investigation and suggestions. The funny thing is, if I dial the exact same number with the same Skype for Business user accout from the SfB Windows desktop application, the calls goes through to the PSTN destination successfully. Microsoft will most probalby just refer me back to Polycom, as I cannot replicate the issue in their Skype for Business desktop application. 

 

From the logs it seems that the Skype server thinks the PSTN number is an actual user account:

0327115523|sip |0|00| ms-diagnostics: 1003;reason="User does not exist";domain="_clientdomain_.com";source="sipfed1E.online.lync.com"

Is there no way to configure the Polycom unit to handle user accounts and PSTN numbers differently when sending the request to the Skype for Business sever?

 

I will log the issue with Microsoft Office 365 support also and see if they have any advice.

 

Thanks,

John

HP Recommended

Hello John,

again now knowing what you have removed from the log in order to hide your clients or your own Identity I am unable to comment.

 

Simply checking the matched rule being used aka ZA Default it seems it matches it to that.

 

Remember the LYNC / Skype for Business Client may do the matching differently but simply dialing this as a number with a 0 or anything else (not knowing what country you are in) I would simply try another number.

 

The reseller who sold the Trio is Hardware.com (Pty) Ltd back in 23/03/2016 so please work with them if you need any further support from our end.

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
† The opinions expressed above are the personal opinions of the authors, not of HP. By using this site, you accept the <a href="https://www8.hp.com/us/en/terms-of-use.html" class="udrlinesmall">Terms of Use</a> and <a href="/t5/custom/page/page-id/hp.rulespage" class="udrlinesmall"> Rules of Participation</a>.