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IP5000 unable to make any external calls

September 7, 2018, 8:17 am
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Hello Polycom Support,

im unable to make external calls, but can accept them. Internal calls work both ways. This is the log i got while trying to make a call. (I replaced the number i actually called with:  01111111111. Im using 3CX.

Any suggestions?

0907170626|sip  |2|03|SipCallNew 0 local port 2222 call appearance -1 IsRtrv 0 dialog 0
0907170626|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config '280' <280@10.201.0.101:0>
0907170626|sip  |2|03|CStkDialog::CStkDialog AddressLocal set to Config
0907170626|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
0907170626|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
0907170626|sip  |2|03|CStkDialog::CStkDialog TAG '3A0A676F-60F696F0' generated
0907170626|sip  |2|03|CStkDialog::CStkDialog local addr '280' <280@10.201.0.101:0> Tag '3A0A676F-60F696F0'
0907170626|sip  |2|03|CStkDialog::CStkDialog exit 0x95d67eec local list size 1
0907170626|sip  |2|03|CStkDialogList::CreateDialogObject localTarg usr '280'
0907170626|sip  |2|03|CUser::CallNew 0x956d2b20 0x95d79c00 CallAppr 0 IsRetrieve 0 ThrdParty '' Dialog 0x0 isCentConf 0
0907170626|sip  |3|03|CStkCall::NewCallState reason 16 'Unknown'->'Dialtone' (0x95d79c00)
0907170626|sip  |2|03|SipOnEvCallNewState 95d79c00,956d2b20 0,Dialtone
0907170626|sip  |2|03|SipCallMake 01111111111
0907170626|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x95d8ed90)
0907170626|sip  |1|03|CreateFailOverProxyList : Reg to Domain '10.201.0.101' nPort 5060
0907170626|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5060
0907170626|sip  |1|03|doDnsListLookup(udp): doDnsSrvLookupForARecordList for '10.201.0.101' port 5060 returned 1 results
0907170626|sip  |1|03|doDnsListLookup(udp): result 0 '10.201.0.101' port 5060 isInBound 0
0907170626|sip  |1|03|CreateFailOverProxyList : 'UDP Only' for '10.201.0.101' port 5060 IP 0 is '10.201.0.101' on udp port 5060
0907170626|sip  |1|03|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0907170626|sip  |2|03|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0907170626|sip  |2|03|CreateFailOverProxyList : IP 1 is '10.201.0.101' on udp port 5060
0907170626|sip  |2|03|CUser::GetFailBackMode 'NewRequest'
0907170626|sip  |3|03|CStkCall::NewCallState reason 16 'Dialtone'->'Proceeding' (0x95d79c00)
0907170626|sip  |2|03|SipOnEvCallNewState 95d79c00,956d2b20 2,Proceeding
0907170626|sip  |1|03|SipOnCommand: response 100,INVITE
0907170626|sip  |1|03|SipOnCommand: response 100,INVITE matches user 1 of 1 '280'
0907170626|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 65 timeout(0x95d8ed90)
0907170626|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 0 code 100, timeout=65 (0x95d8ed90)
0907170626|sip  |3|03|GetRemotePartyAddress from 'To'
0907170626|sip  |3|03|CStkCall::OnEvNewDest (0x95d79c00) new display '' user '01111111111' old 'From' new 'To' source
0907170626|sip  |2|03|SipOnEvNewDest 95d79c00,956d2b20,01111111111,
0907170626|sip  |3|03|CStkCall::NewCallState reason 16 'Proceeding'->'Proceeding' (0x95d79c00)
0907170626|sip  |2|03|SipOnEvCallNewState 95d79c00,956d2b20 2,Proceeding
0907170626|sip  |1|03|SipOnCommand: response 407,INVITE
0907170626|sip  |1|03|SipOnCommand: response 407,INVITE matches user 1 of 1 '280'
0907170626|sip  |3|03|UA Client INVITE INVITE trans state 'proceeding'->'completed' by 407 resp 65 timeout(0x95d8ed90)
0907170626|sip  |3|03|407 challenge received
0907170626|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x95d8fa50)
0907170626|sip  |1|03|Digest authentication
0907170626|sip  |2|03|CUser::GetFailBackMode 'NewRequest'
0907170626|sip  |1|03|CreateFailOverProxyList : Reg to Domain '10.201.0.101' nPort 5060
0907170626|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5060
0907170626|sip  |1|03|doDnsListLookup(udp): doDnsSrvLookupForARecordList for '10.201.0.101' port 5060 returned 1 results
0907170626|sip  |1|03|doDnsListLookup(udp): result 0 '10.201.0.101' port 5060 isInBound 0
0907170626|sip  |1|03|CreateFailOverProxyList : 'UDP Only' for '10.201.0.101' port 5060 IP 0 is '10.201.0.101' on udp port 5060
0907170626|sip  |1|03|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0907170626|sip  |2|03|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0907170626|sip  |2|03|CreateFailOverProxyList : IP 1 is '10.201.0.101' on udp port 5060
0907170626|sip  |2|03|CUser::GetFailBackMode 'NewRequest'
0907170626|sip  |2|03|CTrans::InitRetransForNthServer for UA Client INVITE INVITE state 'callingTrying' Server 1 of 1 (0x95d8fa50)
0907170626|sip  |1|03|SipOnCommand: response 100,INVITE
0907170626|sip  |1|03|SipOnCommand: response 100,INVITE matches user 1 of 1 '280'
0907170626|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 65 timeout(0x95d8fa50)
0907170626|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 1 code 100, timeout=65 (0x95d8fa50)
0907170626|sip  |3|03|GetRemotePartyAddress from 'To'
0907170626|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '' user '01111111111'
0907170626|sip  |3|03|CStkCall::NewCallState reason 16 'Proceeding'->'Proceeding' (0x95d79c00)
0907170626|sip  |2|03|SipOnEvCallNewState 95d79c00,956d2b20 2,Proceeding
0907170626|sip  |1|03|SipOnCommand: response 404,INVITE
0907170626|sip  |1|03|SipOnCommand: response 404,INVITE matches user 1 of 1 '280'
0907170626|sip  |3|03|UA Client INVITE INVITE trans state 'proceeding'->'completed' by 404 resp 65 timeout(0x95d8fa50)
0907170626|sip  |2|03|CUser::GetFailBackMode 'NewRequest'
0907170626|sip  |3|03|GetRemotePartyAddress from 'To'
0907170626|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '' user '01111111111'
0907170626|sip  |1|03|CStkDialog::SetDialogState: Dialog 'ide3192d0a' State 'Trying'->'Terminated'
0907170626|sip  |3|03|CStkCall::NewCallState reason 16 'Proceeding'->'Idle' (0x95d79c00)
0907170626|sip  |2|03|SipOnEvCallNewState 95d79c00,956d2b20 10,Idle
0907170627|sip  |2|03|SipCallDrop 95d79c00,956d2b20 reason 6
0907170627|sip  |3|03|CStkCall::Drop(reason = 6) (0x95d79c00)
0907170627|sip  |3|03|CStkCall::NewCallState reason 16 'Idle'->'Idle' (0x95d79c00)
0907170627|sip  |1|03|CStkCall::NewCallState Already Idle returning (0x95d79c00)

 

 

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backgrounds through provisioning Polycom Trio + Group Trio Mode

September 7, 2018, 8:44 am
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Hello, I need to change the background, when I have the group in Trio Mode

 

Trio: 5.7.1.AB

Group 310: 6.1.8-430297

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Sync or Import Contacts for Trio 8500 from IP office

September 7, 2018, 10:43 am
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Hello.

We purchased few Trio 8500 for our Ayaya IP office 500 and using it in Genric Mode for confernce room phone without Skype integration. The phone is working fine but we are not seeing any contacts - how do we import or sync contacts in SIP setup. 

Thank you

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Adding a Gamma SIP line to a Trio 8800

September 7, 2018, 11:40 am
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Hi,

 

I have just purchased a Trio 8800 Collaboration kit and and 2 sip lines from Gamma. However i can't get my head around adding the SIP info into the Polycom. My info doesnt seam to match what im offered and now confused.

i have for each SIP:

IP address - this is my WAN

Channels 2 - i assume this is as i have two lines so doesn't matter or need added?

Signalling UDP port 5060 egress/ingress - i am given an IP address for this but where does it go on 8800?

Media UDP ports 6000 - 40000 egress/ingress - again have an IP address but where does it go?

 

I am told the router is all done correctly and the ports are open.

 

Tried many options and still stuck can anyone help as its all install but no line!

Thanks

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VVX 411 Unable to enter "Unavailable" status from "Wrap-Up" status

September 7, 2018, 12:17 pm
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I'm looking for a way to keep a softkey on the phone 100% of the time.

 

I have softkeys for "new call", "directory", "DND", "Available", "aSignOut" and "Disp Code" when agents are in "Wrap Up" state, they have to enter the "available" state in order for that softkey to be replaced by "unavailable". 

 

Because of this they sometimes get calls when they're trying to go to break as they have to go to "available" before being able to flip to "unavailable".

 

How can i change the config to show the "unavailable" softkey 100% of the time?

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VVX Screensaver

September 10, 2018, 2:34 am
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Hi,

 

Can anyone please advise me on setting a screensaver on a VVX601? 

 

Since I do not want to use USB sticks I am using a page on our website.  The page is just a PHP script to select a 480*272px image from the folder at random.

 

I came up with the configuration below, but managed to get it wrong.  What I have ended up with is my images displayed as a ‘background’ (with the Voicemail / Time header and New Call / Forward /etc. footer), and default Polycom images displayed full screen as a screensaver.

 

mb.main.idleTimeout="120"
mb.idleDisplay.refresh="120"
mb.idleDisplay.home="myurl"
up.screenSaver.waitTime="30"
up.screenSaver.type="2"
up.screenSaver.enabled="1"

 

 

Thanks.

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Cannot setup speed dial extensions on Expansion modules

September 11, 2018, 1:32 pm
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The receptionist has a Polycom VVX601 with two VVX expansion modules. Currently, the expansion modules show a blank screen. I understand I need to log into the WEB GUI to add the lines but I am not able to as I receive the attached error. How do I update the features.cfg so I can add the lines to show up on the expansion module? 

Am i on the right page?

 

We do not use a provisioning server and the phones are basically setup on a per phone basis. 

 

Appreciate any support in advance!

 

Thanks

HB

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Call display

September 11, 2018, 6:32 pm
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We have an Administrative Assistant who needs to answer calls that come in to her boss' phone. I have the service provider (RingCentral.com) configured to have calls to her boss' phone ring on his phone and her phone simultaneously. What she needs to know is if the call is for her boss as opposed to a call coming in for her. How can I configure her phone to display something like "Call for Steve" on the phone's display?

Below is the phone info if that helps.

Thanks!
Shane

---------------
Phone Information
Phone Model VVX 300
Part Number 3111-46135-002 Rev:A
MAC Address 00:04:F2:C4:E1:A8
IP Mode IPv4
UC Software Version 5.6.0.20009
Updater Version 5.8.0.24992
---------------

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Polycom Trio 8800 Call / Mute Status Lights not working

September 12, 2018, 5:52 am
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We are evaluating the Trio8800 with our in house system, and while the status lights work during boot up, they do not light during calls, when muted, or ringing.  Touching them does toggle the mute/unmute function, but you would have to look at the display to know if mute is on.  We are running:

 

  • Phone Model     Trio 8800
  • Part Number     3111-65290-001 Rev:A
  • UC Software Version     5.5.4.2255
  • Updater Signature     Release

Thanks,
Larry

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Pair Trio 8800 with Group 500

September 12, 2018, 5:56 am
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Dear,

 

we got today our test environment a Group 500 + Trio 8800. We discussed with the distributor that we want to pair the Trio with the Group 500, how is this technical possible? Regarding the sales chat and a datasheet it should be possible within Q3 now?

 

Kind regards

 

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Local Admin Password on VVX450 UC 5.8.0.12848

September 12, 2018, 1:17 pm
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We recently got in a few of the new VVX450s and when working through the config for UC 5.8.0.12848, I've run across an issue where the admin password is not being changed by my config file.  That section of the config is below:

 

  <device>
    <device.prov device.prov.serverName="redacted" />
    <device.auth device.auth.localAdminPassword="redacted" />
    <device.auth.localAdminPassword device.auth.localAdminPassword.set="1" />
  </device>

 

The phone is getting the "device.prov.serverName" and "device.auth.localAdminPassword.set" values because I can see them when I export the config out of the phone:

 

        call.shared.exposeAutoHolds="1"
        call.shared.oneTouchResume="1"
        call.shared.reject="1"
        device.auth.localAdminPassword.set="1"
        device.prov.serverName="redacted"
        device.wifi.enabled="0"

This config has worked previously on older firmware on other VVX models, and it looks like the parameters are still the same according to the Admin Guide:

 

https://documents.polycom.com/bundle/ucs-ag-5-8/page/r_ucs_ag_administrator_and_user_password_parameters.html

 

Any help would be appreciated!

 

Thanks,

John

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Polycom IP 6000 won't boot, cant factory reset or change settings

September 4, 2018, 1:36 pm
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Hey all, 

our pollycom sounndstation ip 6000 has suddenly stopped working. it wont boot, and when interrupting the normal boot process, factory reseting and changing settings hangs. any help would be appreciated!

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Polycom 3xx

September 13, 2018, 6:48 am
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I have a few Polycom phones and we use four digit extentions. I am having a problem on a a few phones where if a transfer is attempeted to a 112x number it will dial after those three numbers, wher as on 110x or 113x it will dial ok, the digimaps are the same between phones, where am I going wrong?

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Retain volume settings after reboot - VVX201

September 13, 2018, 7:07 am
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Hello,

 

I have contacted the seller, Veracomp.  I have contacted the reseller ATHQ.  Neither have replied to my request for assistance.

 

I have 36 VVX201 phones on UC Version 5.8.0.12848 and I would like to configure them with the following settings:

-Set ringer to maximum volume

-Set speakerphone (handsfree) to maximum volume

-Set push-to-talk (PTT) to maximum volume

-Set ringer volume to nonvolatile (volume setting remains after a reboot)

-Set speaker (handsfree) volume to nonvolatile (volume setting remains after a reboot)

-Set push-to-talk (PTT) to nonvolatile (volume setting remains after a reboot)

 

I think I have the code for the ringer settings.  Is the following correct?

up.ringer.minimumVolume="0"

feature.nonVolatileRingerVolume.enabled="1"

 

If that is correct, can I use the same convention for speaker (handsfree) and push-to-talk (PTT)?  If so, what is the terminology I should use for speaker (handsfree) and push-to-talk (PTT)?

 

Thanks for your assistance.

 

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Polycom Trio 8500 - Audio Problems

September 13, 2018, 8:40 am
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Hello,

we're using a Polycom Trio 8500 confernce phone.

Software-release: 5.7.1.4133

We a a problem with the speaker voice quality during a call.

If the microphone is muted the voice quality of the integrated speakers is excellent.

But if I activate the microphone the voice from the speakers is interrupted/intermittet and hard to understand..sometimes louder sometimes quieter.

First I though a codec/network-problem or something like this, but the error only appears if the mute-mode of the microphone is disabled.

Has anyone an idea to solve the problem?

Thank you and regards from Germany, Nils

 

 

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Polycom Trio - logout user via API

September 13, 2018, 8:56 am
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I'm looking through the Trio API documentation, but I don't see any mention on how to programatically logout a user. Is there an API call available to do this?

 

Thank you!

 

   Richard

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Config Files by Phone Type

September 13, 2018, 1:05 pm
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I am provisioning phone through our own FTP Server.

I would like to have the phones automatically go to the provisioning server Using DHCP Scope Options and I have some of that figured out already.

 

My question is can I have the phones go to the server look at the default config file.  (What is the name of the default config file the phone looks for?)

The default config file then tells the phone if you = Real Presence Trio 8500 then goto

\Polycom\Conf\8500\8500.cfg

 

If anybody can let me take a look at a config file that does this or point me to the documentation I would appreciate it!

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VX411 Making Calls on its own

September 14, 2018, 5:59 am
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Model:VVX411
Codec:G711
Mac Address:64:16:7F:3B:EB:F3
Profile Type:Polycom VVX411

 

This device is making calls on its own.

 

Is there a fix for this please?

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VVX PHONE - Corporate Directory - LDAP Search Request customisation

September 14, 2018, 7:11 am
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Hello everyone,

 

I've recently set up a fresh VVX Phones installation with corporate Directory. (https://support.polycom.com/global/documents/support/technical/products/voice/Corporate_Directory_Best_Practices_TB41137.pdf)

 

I'm using polycom 5.5.1 UC Software version and openLDAP 2.4 on Debian 9.

 

Everything is working fine but i'd like to know if it's possible to add a star before the search string ?

 

Example :

Original ldap request made by the phone :


Filter: (&(&(sn>=0)(sn<=zz))(|(sn=Au*)(givenName=Au*)))


(here i'm using dir.corp.attribute.x.addstar="1" in order to search everything which begin with "Au" in the surname field or givenName field)

 

Is there an option on the phone side or the server side to prepend de search string with a star like this :

Filter: (&(&(sn>=0)(sn<=zz))(|(sn=*Au*)(givenName=*Au*)))

 

Best regards,

 

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Polycom IP7000 API

September 14, 2018, 1:58 pm
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Hi, 

I am writing a driver for the IP7000 in order to be able to control it from a third party control system. I searched through all Polycom public web repositories and still can't find a clear document that explains the protocol, details the API endpoints and stuff like that.

 

Anybody can help with this?

 

Thank you.

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