I've been using a Polycom VoxBox speakerphone with my VVX 600 phone (connected via bluetooth) for several months now, but just in the last couple of days I haven't been able to connect the VoxBox to the phone.  I've reset the VVX 600, updated the firmware (5.9.3.2489 is installed on the phone), reset the VoxBox, updated the firmware (1.0.03.0002 is installed, and the VoxBox Companion says that's the current version), factory reset both devices, and still can't get them to pair or even get the VVX 600 to show the VoxBox as an available bluetooth device.

I can pair the VoxBox with my iPhone XS Max without any issues, and I can pair the VVX 600 with a Plantronics W740 headset and a Plantronics Voyager Legend, but can't get the VVX 600 to recognize the VoxBox for some reason.  Any help is appreciated.  Thanks!

Software version 6.3.1.1 on both phones. Google tells me it's a Polycom problem, I believe it's there problem.

Provisioned phpones display "Call Error The requested service is not available at the moment"

As Polycom is the only phone the phone they recomendwith Google Voice it would be nice to make sure they work together at all times.

Perhaps someone can reach out to Google to confirm all is well on both ends?

Anyone else experiencing this issue?

We have a large deployment of VVX 411 phones. Only one phone has issues so far. It will disconnect from the network if the computer attached to the PC port goes to sleep or is powered down. When that happens, the phone must be completely powered down and restarted before it will reconnect to the network. 

Any help or pointers on where to look would be appreciated 

Hi,

What is the maximum power consumption of VVX311 and VVX401 when connected to PoE switch?

Thank you.

Sinjo

I am using a deplyoing trios 8800 at a new site and registering to the CUCM

Phone lines are being configured as Line2 - Line1 is Skype

I am having issues where by it registers to the CUCM and immediately unregisters.

there are other trio 8800 at other sites that are getting registered to the same CUCM - All the working trios at other sites and non working trios have the same firmware

UC Software Version

5.7.2.3205

The error on the logs i seem to get it 

----------------------

0717175343|cfg |*|00|Prov|Finished updating configuration
0717175343|cfg |*|00|Prov|Starting to provision
0717175343|cfg |*|00|Prov|Starting to update
0717175343|copy |*|00|Server '10.112.100.57' said 'phoneservice/configfiles/64167f7697bd/contacts/64167f7697bd-directory.xml' is not present
0717175343|clist|4|00|dbSet::srv2mem:'64167f7697bd/contacts/64167f7697bd-directory.xml' File transfer failed, trying to get local file
0717175344|app1 |*|00|SoRegistrationEventLast - new AppRegLineC, szUser = 7326962
0717175344|sip |*|00|Sip Register Usr:7326962 Dsp:Phone Line Auth:'7326962' Inx:1
0717175344|app1 |4|00|AppPhoneC::OnEvReg - Ctx [1] Registered [0]
0717175344|app1 |4|00|AppPhoneC::OnEvReg - Ctx [1] Registered [0]
0717175344|copy |*|00|Server '10.112.100.57' said 'phoneservice/configfiles/64167f7697bd/contacts/000000000000-directory.xml' is not present
0717175344|clist|4|00|dbSet::srv2mem:'64167f7697bd/contacts/000000000000-directory.xml' File transfer failed, trying to get local file
0717175344|copy |*|00|Server '10.112.100.57' said 'phoneservice/configfiles/64167f7697bd/overridesProfile/64167f7697bd-phone.cfg' is not present
0717175344|cfg |4|00|Prov|cfgProvStatusSet: error code '22' respCode '404' desc '64167f7697bd/overridesProfile/64167f7697bd-phone.cfg'
0717175344|cfg |4|00|Prov|File transfer failed due to curl error code:22 and respCode:404
0717175344|cfg |4|00|Prov|provStatusCbSynced CfgProvTransferringFileResult_CurlError: error code '22' respCode '404' desc '64167f7697bd/overridesProfile/64167f7697bd-phone.cfg' provErrorPopupDisplayed 'F'
0717175344|app1 |4|00|AppPhoneC::OnEvReg - Ctx [1] Registered [1]
0717175344|wmgr |4|00|[findProxy] Invalid proxy mode '8' specified to find the proxy
0717175344|wmgr |4|00|[sendXMLObject] Proxy connect result '0'
0717175344|copy |*|00|Server '10.112.100.57' said 'phoneservice/configfiles/64167f7697bd/license/000000000000-license.cfg' is not present
0717175345|copy |*|00|Server '10.112.100.57' said 'phoneservice/configfiles/64167f7697bd/license/64167f7697bd-license.cfg' is not present
0717175345|cfg |*|00|Prov|Setting device parameters from configuration files.
0717175345|sip |*|00|Sip UnRegister Usr:7326962 Dsp:Phone Line Auth:'7326962' Inx:1
0717175345|sip |*|00|SipUserRemove: user 1 being removed.
0717175345|clist|4|00|dbSet::load:cleared load error state m_sync=<0x40>
0717175345|clist|4|00|dbSet::ioRequest:dir=<0> m_status=<true> src=<64167f7697bd/contacts/64167f7697bd-directory.xml> dst=</data/polycom/ffs0/user/64167f7697bd/64167f7697bd-directory.xml> m_sync=<0x10>
0717175345|cfg |*|00|Prov|Finished updating configuration

----------------

can anyone advise what the issue could be here please?

I didn't notic the previous information. After upgrading the software via Utilities > Software Upgrade >  Polycom Hosted Server, "UC Software Version 4.07.4180" is showed now. Does it mean the previous software was UCS x.x.x as well?

Can I set it for a cloud PBX SIP account on this software?

Can you use an FTP or other protocol client to view or modify the phone's file structure?

<?xml version="1.0" encoding="UTF-8" standalone="yes"?><User_CONFIG><OVERRIDES
            prov.login.localPassword="111111"
            reg.2.displayName="testkjj testkj"
            reg.2.address="1789/64167f959591@abhilash.sip.teledge.com"
            reg.2.label="testkjj testkj"
            reg.2.auth.domain="************"
            reg.2.auth.userId="1789"
            reg.2.auth.password="********"
            reg.2.outboundProxy.address="***********"
            reg.2.outboundProxy.port="0"
            reg.2.outboundProxy.transport="UDPOnly"
            reg.2.server.1.address="abhilash.sip.teledge.com"
            reg.2.server.1.port="0"
            reg.2.server.1.transport="UDPOnly"
            msg.mwi.2.subscribe="1789"
            msg.mwi.2.callBackMode="contact"
            msg.mwi.2.callBack="101"
            attendant.uri=""
            pres.reg="1"
            pres.idleSoftKeys="0"
            feature.presence.enabled="1"
            feature.callPark.enabled="1"
            attendant.reg="1"
            attendant.behaviors.display.spontaneousCallAppearances.normal="0"
            attendant.resourceList.1.address="54321"
            attendant.resourceList.1.label="yes"
            attendant.resourceList.1.type="normal" /></User_CONFIG>

i have created above config file and another directory file  for user profile ,
config file successfully get by phone but its attendants not showing, even after reboot phone 
I'm using 

Hello, I have a Trio 8800 fw 5.9.0.11398, generic profile, integrated with OTD. I am using OTD integration with service account and passtrough authentication. My exchange server on the Trio is https://otd.plcm.vc/EWS/Exchange.asmx and on the Trio I configured the username and password of the O365 room mailbox that I want to connect. Everything works fine, on the OTD panel I see the Trio connected to OTD and OTD connected to O365, calendar events are visible on the Trio and I am able to join meetings.

All works fine until an event is cancelled. As an example, if from my personal mailbox I create an event and invite the room, it is correctly displayed on the Trio.... then from my Outlook I delete the event (and send the cancellation) and a few seconds later the Trio raises the error 

"Failed to connect to exchange server: re-enter credentials to access exchange services"

Logs on the Trio say:

0718120448|pgui |4|00|onNetworkError: error occurred 204 
0718120448|pgui |4|00|New credentials needed for accessing Exchange Web Services, deactivating to wait for new credentials
0718120448|pgui |4|00|Stopped access token service
0718120448|app1 |4|00|EC has failed to authenticate with server
0718120448|pgui |4|00|Exiting EC for sign out.

The OTD panel shows Trio connected to OTD but OTD not connected to O365 with the error "calendar access not permitted". If I go on the trio, settings, basic, login credentials and I press submit without changing anything, it works again.

I tried this https://knowledgebase-iframe.polycom.com/kb/viewContent.do;jsessionid=D80E88F3E9B2AFC3AD3E732B659ECDD9?externalId=33896&sliceId=1

with the following settings

exchange.ADerror.retryCount="2"
exchange.ADerror.retryPeriod="10"
exchange.error.retryCount="2"
exchange.error.retryPeriod="10"

but did not solve the issue. Honestly I do not know if these setting are correct.

Thank you for your help.

Hi there, our organisation has a meeting room configured with a Trio 8800 set in Skype USB Optimized Mode, paired with a Group 500, which is connected to a Ceiling Microphone Array. 

We're using this configuration because it allows us to use the ceiling mic array on a Windows PC, which the Trio is connected to via USB.

However in this configuration, we no longer require the built-in microphone of the Trio to be active since the ceiling mics are doing the job. I was wondering if anyone knows of a way to disable, or turn down the sensitivity of the built-in mic on a Trio? Muting the trio also mutes the ceiling mics so that is not an option.

Thanks in advance,

-George P.

Hi,

I'm currently evaluating the VVX 601 before deploying a bigger number of phones. One issue that I found is this:

When the pbx hangs up a call because the called number is busy or invalid, I do not see any information about the hangup reason on the screen of the phone, I just hear a busy tone that is generated locally in the phone.

I tried submitting several different reasons from the pbx and I also tried adding an additional Q.850 reason header. But all this made no difference.

Here are sip traces of the replys I tried:

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK4d88fb59CA8F1CEE
Call-ID: d8b133c4bd2b2303274da8a5250a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=8B0CF535-464A0C6A
To: <sip:41@voip-pbx.m.i2n;user=phone>;tag=8692ef7f-6320-4265-8f90-68dafbeec99f
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Content-Length:  0

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK5c3cf6b9AD13D14E
Call-ID: 62039ea3098b4c1d7448efee080a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=3E49FC95-A5EDECCA
To: <sip:41@voip-pbx.m.i2n;user=phone>;tag=69dc0ab7-71aa-4540-af42-a6e36af3abbd
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Reason: Q.850;cause=17
Content-Length:  0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK4a849f4c1F54A2E9
Call-ID: 62e210e4cb552952ad4067f58e0a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=58808C88-44B97EC5
To: <sip:88@voip-pbx.m.i2n;user=phone>;tag=1536c460-8795-4de9-81bf-8a66d7914a76
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Content-Length:  0

Phones from other vendors show a message like "User busy" or "Number unavailable" when receiving these packets.

How can I show such hangup reasons or causes to the user with the VVX 601?

Firmware version is 6.0.0.4796.

Thanks.

Hi,

We have a few VVX411 phones out in the wild, they are getting ghost calls so I've had a poke about & found the 2 lines:

voIpProt.SIP.requestValidation.1.method="source"
voIpProt.SIP.requestValidation.1.request="INVITE"

When I put these into the config the phones stop accepting calls....

In the log, i've turned on debug for SIP & can see this:

0722091449|sip |1|00|Try to do source validation
0722091449|sip |1|00|CCallBase::IsTrusted : NAPTR lookup for '<CHANGED>.nebulaip.com' OK
0722091449|sip |1|00|CCallBase::IsTrusted : NAPTR lookup for '<CHANGED>.nebulaip.com' OK
0722091449|sip |3|00|Challenge failed !!

It then appears to send:

0722091449|sipp |0|00| SIP/2.0 400 Bad Request

If I take those 2 lines out it starts to work again.... (but so do the ghost calls.)

the part where it says <CHANGED>.nebulaip.com is the server that it connects to so i'm not sure why it's not working...

I have come across an extra line voIpProt.SIP.requestValidation.digest.realm="PolycomSPIP"

I'm not sure what that does though?

Can anyone shed any light on the matter?

Thanks!

Garry

Hi

Can I disbale http for web access and use only https???

Hi,

I have a Polycom Trio endpoint registered to Cisco UCM and I have the latest firmware 5.9 installed on Trio. My customer wants to integrate the Trio conference phone to webex so to connect easy to a webex meeting using the Trio. I can see in the settings menu that the "Skype for Business Sign In" is greyed out. I haven't found any useful document what explains in this case what to do and what are the integration opportunities for Webex. I would imagine to receive a Calendar invite on Trio then jump in the Webex session with one touch option.

Our Polycom Trio 8500 is receiving consistent calls (multiple at same time) to the Polycom.

Up until 12.21 the calls were coming from 200@IP_Polycom.  These have know stopped and calls are coming in from 201@IP_Polycom.

When I came in this morning I reduced the volume of the ringer but this is a persistent problem - Polycom shows over 1,100 missed calls.

Hi,

I know that SoundPoint is basicalluy dead, but we have a lot of these out there and want to improve security.  So far we are unable to get mutual TLS authentication to work with self-signed certificates.  (We've gotten mutual TLS to work flawlessly with other major vendor IP phones).  Polycom is the last hold-out.

We're trying to get the following setup to work:

Firmware 4.0.14.0987

Self-Signed Certificate with Self-Signed CA

Mutual TLS Authentication (Only accept signed certificates from polycom phones)

We first provision the polycom (IP-335/550/650/etc/etc) with HTTP and use <device> to set up for HTTPS

<device
device.set="1"
device.net.etherVlanFilter.set="1" device.net.etherVlanFilter="1"
device.dhcp.bootSrvUseOpt.set="1" device.dhcp.bootSrvUseOpt="2"
device.dhcp.bootSrvOpt.set="1" device.dhcp.bootSrvOpt="166"
device.dhcp.bootSrvOptType.set="1" device.dhcp.bootSrvOptType="String"
device.prov.serverType.set="1" device.prov.serverType="HTTPS"
device.prov.serverName.set="1" device.prov.serverName="provisioning-server-here"><device.sec
device.sec.TLS.customCaCert1.set="1" device.sec.TLS.customCaCert1="-----BEGIN CERTIFICATE-----
certificate-goes-here
-----END CERTIFICATE-----"
/></device>

So far, this acomplishes turning off future usage of option 66 (and use option 166 if we want to re-program in the future).  So now the polycom phone successfully will download the above xml via HTTP, and then it will TRY to connect to HTTPS but not succeed.

Flow:

The Polycom downloads the above XML via HTTP and does initial provisioning.  The polycom then reboots itself and then continually tries https conneection, but handshake always fails.

Here's the log from the polycom:

000015.462|so |*|03|Network initialized. Starting network tasks.
000015.464|log |*|03|Install file upload callback for 'so'

0723102804|cfg |5|03|Prm|Parameter acd.reg requested type 0 but is of type 2
0723102804|sip |*|03|Fast Boot Measurement Point: Ready for Call, uptime: 15.616 sec.
0723102804|app1 |*|03|Ctx [0] Registered [false]
0723102804|app1 |5|03|Corporate directory instance does not exists.
0723102806|copy |4|03|SSL_connect error SSL connect error.error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca
0723102808|copy |4|03|SSL_connect error SSL connect error.error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca
0723102810|copy |4|03|SSL_connect error SSL connect error.error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca

Why would the phone not recognize the CA?  When viewing the fingerprint of the CA in the polycom phone web portal, it matches the server.

We have VERIFIED that the polycom DOES trust the CA, because If we turn OFF requred client certificates at the HTTPS server (Apache 2.x) , the polycom SUCCESSFULLY downloads configuration from HTTPS (so obviously the phone can accept the self-signed certificate and connect to a self-signed server).

When we turn on debugging on apache, we clearly see the polycom is not sending the device certificate when connecting to the https server, and the phone is giving up the connection PRIOR to any attempt at sending the device certificate for client-cert-authentication.

Final Questions:

Does mutual TLS only work with officially signed certificates?  Ie. verisign, starfield, etc?

Does mutual TLS work at all on a Polycom Sountpoint?

Phones: VVX501 (Skype for Business) (BToE)

UC Version: 5.9.0.9373

Provisioning distribution: FTP server

I have a strange problem where phones do apply a config file which is not destined for their MAC address.

Scenario: The 000000000000.cfg has 3 references to config files. The lines look like:
CONFIG_FILES: shared.cfg, 12345678ABCD-custom.cfg, 12345678ABCE-custom.cfg

The Shared.cfg has some general config for background pics and so on for ALL phones.

The 2X MACADDRESS-custom.cfg files have a soft button config line just targeting 2 handsets.

However all handsets in the network are reading the MACADDRESS-custom.cfg and are applying this soft key, not only the 2 phones which MAC ADDRESS has been specified in the config files.

I am wondering if this is a bug as my understanding is that phones are not supposed to apply configuration files which are named "MACADDRESS-something.cfg"

Hi,

Trio and Visual+ Firmware: 5.9.0 AB

I'm using a Trio registered (SIP) to A Video Border Proxy.

Everythings working fine, except for making SIP calls with a specific uri including an IP address.

The customer wants the trio to be able to dial string that look like this:

"xxx@ipaddress"

Calling actual sip uris, like: "xxx@domain.com" works fine, but the trio doesn't seem to know what to do with the string including an IP address.

Looks like the trio just drops the the "@ipaddress" and tries dialing thepart in front of the @ directly.

My Question is:

Is there a way to allow the trio to dial strings like the above one.

If not, how would I implement the Trio digit map dial plan to translate a specifc Ip to a specific domain? I've been playing around with the later, but didn't get anywhere.

For example, could I translate "example@10.10.10.10" to "example@domain.com" using the digit map on the line?

Thank you very much in advance,

Martin

i have problem with Hoteling user registration .

when i register user at second line,phone registers it successfully, but when i do to first line it fails somehow.
this is my file format  i send to phone when it request for
REQUEST

GET /hoteling/64167f94f8b7/123.cfg HTTP/1.0.
Host: abhilash.teledge.com.
X-Real-IP: 202.131.119.122.
Connection: close.
Accept: */*.
User-Agent: FileTransport PolycomVVX-VVX_410-UA/5.9.1.0615 Type/Application.

RESPONSE

<?xml version="1.0" encoding="UTF-8" standalone="yes"?><User_CONFIG><OVERRIDES
prov.login.localPassword="11111"
reg.1.displayName="ronak patel0012"
reg.1.address="123/64167f94f8b7@abhilash.sip.teledge.com"
reg.1.label="ronak patel0012"
reg.1.auth.domain="abhilash.sip.teledge.com"
reg.1.auth.userId="123"
reg.1.auth.password="889mP3OIy5lGcTH"
reg.1.outboundProxy.address="abhilash.sip.teledge.com"
reg.1.outboundProxy.port="0"
reg.1.outboundProxy.transport="UDPOnly"
reg.1.server.1.address="abhilash.sip.teledge.com"
reg.1.server.1.port="0"
reg.1.server.1.transport="UDPOnly"
/></User_CONFIG>

then phone requests for 

T 10.50.7.18:53060 -> 10.50.8.7:80 [AP] #9297
GET /123-phone.cfg HTTP/1.0.
Host: abhilash.teledge.com.
X-Real-IP: 202.131.119.122.
Connection: close.
Accept: */*.
User-Agent: FileTransport PolycomVVX-VVX_410-UA/5.9.1.0615 Type/Application.
.

but i have not given it .
then phone request for directory file 

GET /media/tenant/909/provisioning/polycom/vvx/64167f94f8b7/123-directory.xml HTTP/1.0.
Host: abhilash.teledge.com.
X-Real-IP: 202.131.119.122.
Connection: close.
Accept: */*.
User-Agent: FileTransport PolycomVVX-VVX_410-UA/5.9.1.0615 Type/Application.

which gets uploaded successfully.