Hello all,
UC Software 5.7.2.1277 maintenance release and release notes are now published
on the Polycom Support web page. http://support.polycom.com
What is included in this release?
Customer escalation fixes and bug fixes. Please have a look at theRelease Notes.
Upgrade key required? No
What is available on the Support Website now?
Phone: VVX 501
Software: UC 5.7.2.1277
VoIP Server: SIP (Asterisk)
Provisioning: Config files, HTTPS
Does the msg.bypassInstantMessage parameter work with the VVX 501 (Messages button on home screen)?
I am trying to enable both oneTouchVoicemail and bypassInstantMessage so that the Messages button goes straight to voicemail for my users. oneTouchVoicemail works fine, but I can't get bypassInstantMessage to work - still have to press "Message Center" before the phone dials voicemail. I verified that it applied successfully to the phone by exporting and inspecting the config.
I'm aware that this parameter does not work for Soundpoint phones without a hardware messages button, but haven't seen any info on how it applies to the VVX line.
Config:
<msg msg.bypassInstantMessage="1">
<msg.mwi msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="1234" msg.mwi.2.callBackMode="contact" msg.mwi.2.callBack="1234">
</msg.mwi></msg>
<up up.transparentLines="1" up.oneTouchVoiceMail="1">
</up>
Hi
Can anyone provide the right configuration syntax to disable the on-hook dialing feature in the VVX handsets. Just to be clear I want to disable the functionality where user can dial number from key pad (while handset is onhook) then pressing "dial" soft key to send off call. I am testing with VVX400 running firmware 5.4.7. I tried to used the below configurations to no avail. Please advise.
<ALL feature.urlDialing.enabled="0"/>
or
<up up.onHookDialingEnabled="0"/>
Our Polycom is running firmware version 5.5.4.2255 and it is a trio 8800. We have line 1 configured for Skype for Business and line 2 configured to connect to our Mitel phone system. When we make a call via the dial a number button (on hook dialing) or via the line key that is configured to the Mitel phone system (off hook dialing) the phone starts dialing automatically after 3 seconds. I have tried extending the digitmap timeout, but this then breaks the digitmap I have created for the Mitel line. Any ideas?
Hi and welcome, as this is my first post here.
I'm having problems with VVX310 phones. Both have been upgraded to the latest version 5.7.2.1277 and been restored to factory settings.
Now, the thing is - I have two phones:
a) phone A, LAN port connected to unmanaged 1Gb/s switch port, negotiated speed is 1Gb/s
b) phone B, LAN port connected to unmanaged 1Gb/s switch port, negotiated speed is 100Mb/s
Both phones, if PC port is connected to either switch or PC (both devices can go up to 1Gb/s), they negotiate only 10Mb/s.
So there are two problems:
- LAN port speed for phone B
- PC port speed for both phones
The only difference is that phone A has "Rev K" written on the bottom sticker, and phone B - "Rev J".
Any suggestions?
Could those be a firmware issues?
Regards
Lukasz
Is there a way to reset the volume settings on the Trio's back to 50%?
My Trio's are connected to RPRM and I have a provisioning window set for 3-5am. I would like to be able to have the volume set back to 50% before the start of business the next day.
We were just informed that one of the units didn't have dialtone, however, this was due to someone turning the volume down to 0 from a previous meeting.
Hello, I am in dire need of assistance. Thank you in advance if you're able to help.
I'm setting up a new phone system with Jive using this unit. I can make outbound and inbound calls to other handsets outside of this office, but between two units on the same network here, they're not registering that they're dialing when dialing the extension or full number plus extension.
I've tried upgrading and downgrading the software (currently set to
Here is the full error log when trying to dial by extension OR by dialing full number and extension:
0523110131|cap |4|00|[SoVidVideoCaptureManagerC::captureFrameAndProcessLoop][ERROR] No active camera channel found! noActiveCameraChanCt=17 0523110200|clist|4|00|dbIO::processResult:no response 0523110403|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523110623|cap |4|00|[SoVidVideoCaptureManagerC::captureFrameAndProcessLoop][ERROR] No active camera channel found! noActiveCameraChanCt=33 0523110700|clist|4|00|dbIO::processResult:no response 0523110713|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523111036|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523111359|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523111731|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523112035|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523112414|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523112812|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523113155|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523113547|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523113854|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523114223|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523114526|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523114845|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523115219|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523115552|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523115920|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523120039|hw |5|00|Task ended without calling rtosTaskEnd() name=tTCPListen176 0523120242|hw |5|00|Task ended without calling rtosTaskEnd() name=tTCPListen154 0523120257|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523120629|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523121023|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523121356|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523121701|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523122101|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523122410|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523122810|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523123205|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523123603|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523124001|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523124317|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523124648|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523125026|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523125423|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523125808|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523130115|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523130504|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523130845|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523131153|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523131522|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523131843|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523132201|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523132515|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523132820|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523133148|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523133455|pps |4|00|ppsOnEvLocalProfileProvision(): ProvDoc is NULL 0523133459|clist|4|00|dbIO::processResult:no response 0523133736|cap |4|00|[SoVidVideoCaptureManagerC::captureFrameAndProcessLoop][ERROR] No active camera channel found! noActiveCameraChanCt=65
I am using VVX300 phones and running on UC 5.2.7. I am trying to hide the conference call soft key. I have looked through the device.cfg, feature.cfg, and the site.cfg. I have even tried to add softkey.feature.conference 0. Has anyone else been able to figure this out.
Are the VVX 411 VoIP phones able to separate number sources? In other words, accept incoming/outgoing calls from provider A (ringcentral) and also provider B (skype for business) - 2 phone numbers from 2 separate sources. All the information I've read defaults to having one provider issue multiple phone numbers (up to 12 separate phone numbers and users). Other answered posts seem to indicate multiple phone sources are not possible, but sighted firmware as the reason.
I have 200+ Polycom IP 5, 6 and 7k phones, plus a few Trio 8500 and 8800s. I have them all on the newest firmware. Unfortunately they're all bought by other departments on campus that don't necessarily buy any support for them so I'm at a loss on how to get support... Regardless, there appears to be a bug in the SIP RFC compliance of these phones and I'm wondering if anybody has a work-around or has also experienced this bug.
The issue is that the phones don't support the "SIPS:" URI. This wouldn't be a big deal, because the phones are all registered in my environment with TCP signaling and thus use "SIP:". However, my proxy server has some internal communications it does and includes VIA and record-route headers about midway down the list with the SIPS: URI. Here's where the non-compliance is exposed: The phone re-writes the entire list when it responds and changes all of the record-route and via headers that have "SIPS:" to "SIP:SIPS:". The proxy doesn't know what to do with that and it never forwards the phone's replies (e.g. 200 OK).
Here's an example:
INVITE message's record-routes:
Record-Route: <sip:SM1@192.168.0.73;transport=tcp;lr;av-asset-uid=794513d5> Record-Route <sip:127.0.0.2:15060;transport=tcp;ibmsid=local.1504124737047_2762749_2993075;lr;ibmdrr> Record-Route: <sips:127.0.0.2:15061;ibmsid=local.1504124737047_2762749_2993075;lr;ibmdrr> Record-Route: <sip:SM1@192.168.0.73;transport=tls;lr;av-asset-uid=794513d5> Record-Route: <sip:192.168.0.200:5061;transport=tls;lr>
Phone's 200 OK (new-lines added for readability):
Record-Route: <sip:192.168.0.222;transport=tcp;lr>,
<sip:rw479dcd66@192.168.0.207;lr;transport=TCP>,
<sip:192.168.0.206:15060;transport=tcp;lr;ibmsid=local.1503513256525_4565480_5029726>,
<sip:rw-479dcd66@192.168.0.207;lr;transport=TCP>,
<sip:SM1@192.168.0.73;transport=tcp;lr;av-assetuid=794513d5>,
<sip:127.0.0.2:15060;transport=tcp;lr;ibmsid=local.1504124737047_2762749_2993075;ibmdrr>;
<sip:sips:127;lr;ibmsid=local.1504124737047_2762749_2993075;ibmdrr>;
<sip:SM1@192.168.0.73;transport=tls;lr;av-assetuid=794513d5>,
<sip:192.168.0.200:5061;transport=tls;lr>
I've temporarily worked around this by forcing my Polycom phones to use my SBC so I can hide the internal topology from the phones but that's eating SBC licenses.
Any thoughts out there? Polycom want to fix this? The lack of support for the SIPS URI is understandable, it's newish (like 9 years ago). But to rewrite and hose the headers that don't apply to the phone directly is actually a basic SIP RFC violation.
Hi,
I have a Polycom IP 6000. Can somebody here please help??
I got several issues with the polycom:
- When we try to use the polycom to call to our customer conference birdge. The conference bridge asks for the passcode to join. I entered the passcode but It always failed. IF I try to use a cisco linksys 504G to join the conference bridge, it succeed. So the problem is only with polycom ( Our passcode is 7-digits passcode, xxxxxx#)
- We use a SIP provider, keyyo.fr for our SIP server. When we try to provision the polycom to keyyo. It failed and report unknown ca error. I followed some guides on the forum about installing Platform CA for the polycom, but it still failed.
For more information: provision server: https://puc-prov.keyyo.com/enroll.html, is using a GANDI Standard Certificate. And Below is the error from polycom's log
000033.160|sip |*|2815|Fast Boot Measurement Point: Ready for Call, uptime: 33.160 sec.
000033.163|app1 |*|2815|Ctx [0] Registered [false]
000033.165|app1 |5|2815|Corporate directory instance does not exists.
000033.175|copy |3|2815|ConfigureCurl : curl option usrpwd is not set
000033.176|so |4|2815|Could not find IP address for SNTP server. 86400
000033.215|copy |3|2815|'https://puc-prov.keyyo.com/enroll.html/0004f2e43145-directory.xml' from 'puc-prov.keyyo.com(83.136.160.85)'
000033.215|curl |3|2815|timeout on name lookup is not supported
000033.215|curl |3|2815|About to connect() to puc-prov.keyyo.com port 443 (#0)
000033.216|curl |3|2815| Trying 83.136.160.85...
000033.216|curl |3|2815|the local port callback returned 0
000033.216|curl |3|2815|Local port: 26225
000033.469|curl |3|2815|Connected to puc-prov.keyyo.com (83.136.160.85) port 443 (#0)
000033.666|curl |3|2815|successfully set certificate verify locations:
000033.667|curl |3|2815| CAfile: /ffs0/ca1.crt
CApath: none
000033.667|curl |3|2815|SSLv3, TLS handshake, Client hello (1):
000033.922|curl |3|2815|SSLv3, TLS handshake, Server hello (2):
000033.923|curl |3|2815|SSLv3, TLS handshake, CERT (11):
000033.959|curl |3|2815|SSLv3, TLS handshake, Request CERT (13):
000033.959|curl |3|2815|SSLv3, TLS handshake, Server finished (14):
000033.960|curl |3|2815|SSLv3, TLS handshake, CERT (11):
000033.966|curl |3|2815|SSLv3, TLS handshake, Client key exchange (16):
000034.006|curl |3|2815|SSLv3, TLS handshake, CERT verify (15):
000034.007|curl |3|2815|SSLv3, TLS change cipher, Client hello (1):
000034.008|curl |3|2815|SSLv3, TLS handshake, Finished (20):
000034.262|curl |3|2815|SSLv3, TLS alert, Server hello (2):
000034.262|curl |3|2815|error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
000034.262|curl |3|2815|Closing connection #0
000034.276|copy |3|2815|Download of 'enroll.html/0004f2e43145-directory.xml' FAILED on attempt 1 (addr 1 of 1)
000034.276|copy |4|2815|SSL_connect error SSL connect error.error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
000034.276|copy |3|2815|Making further download attempts for 'enroll.html/0004f2e43145-directory.xml'
000034.276|copy |3|2815|'https://puc-prov.keyyo.com/enroll.html/0004f2e43145-directory.xml' from 'puc-prov.keyyo.com(83.136.160.85)'
000034.276|curl |3|2815|timeout on name lookup is not supported
000034.277|curl |3|2815|About to connect() to puc-prov.keyyo.com port 443 (#0)
000034.277|curl |3|2815| Trying 83.136.160.85...
Information on the phone model, software version, SIP version:
000002.638|so |3|01|Platform: Model=SoundStation IP 6000, Assembly=3111-15600-001 Rev=G Region=
000002.640|so |3|01|Platform: BootBlock=3.0.2.0024 (15600-001) 30-Nov-10 15:05
000002.640|so |*|01|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56
000002.640|so |3|01|Application, main: Label=Updater, Version=Azurite 5.0.12.0033 05-May-17 12:30
000002.646|app1 |3|01|Application, load: Type=SIP, Version=4.0.13.1445 06-Oct-17 19:45
Hi Guys much smarter than me,
We've purchased three Polycom Trio's recently with the hope of moving to a Skype platform - in the interim however we are using these as SIP phones on an existing Mitel Platform. We have found in using the devices that we are unable to get the devices to recognise either the * or # key when pressed inside an ongoing call - as this is received by the SIP services as a 1 key press. Wiresharking the packets outbound, it seems to be sending as 1.
Where should we go to correct these?
Thanks in advance,
Adam,
Hi, i have a strange BLF issue
VVX600, Color keypannel, SIP/Broadsoft, UC 5.6.1.1740
In Broadsoft i asign the maximum of 50 BLF entries, the screen stays blank. I figured out, the maximum entries what works is 36. After 37, the screen go blank.
What i allready did is make an modification in the Cisco router, swith off virtual-reassembly on the LAN interface and the Dialer.
(Before that, the maximum BLF fields are 23)
Tried turn of storm filter but no success
Does anyone know what the problem is?
Thnx, regards, Arjan
Tests :
1 - ) Any consecutive calls toward IP5000 are normally received on Ip5000
2 - ) A call from Ip5000 to another phone is normally received.
3 - ) If a fisrt call is dialing from IP5000, we are obliged to waiting about 30s to make a second call
4 - ) Two or more consecutive immediate calls from IP5000 are not sent over ip network ( Trouble on invite authentication).
Joigned the pcap file
According to the BLA draft https://tools.ietf.org/html/draft-anil-sipping-bla-03#section-6.2, when a phone attempts to seize a bridged line appearance (e.g. NOTIFY with state=trying) and another user has just seized the line, a 500 with Retry-After header should be returned to the requester. The phone should then wait the amount of time in the header before retrying another trying NOTIFY. However, what I'm seeing in practice is very different. I have a scenario where a 500 is returned with Retry-After=2, and the phone does not respect the header and immediately sends a NOTIFY with state=terminated for the same line-id, before attempting to seize another line appearance (e.g. x-line-id=1). This diverges from the spec entirely. Any ideas why?
The BLA phone config:
<entryreg.1.callsPerLineKey="1"/><entryreg.1.lineKeys="1"/><entryreg.1.strictLineSeize="1"/>Firmware 5.6.0
I tried to upgrade the version 3.3.3 to 4 and in my phone dont have the option provisioning server.
Hello Community
Is there a way to monitor the above systems via SNMP? We have a central monitoring system in which I also want to monitor our Polycom systems. I haven't found anything suitable in the community and on the web.
We want to be notified when a device/system stops working or is no longer logged in for some reason.
Thanks in Advance
Kindly Regards
Jacek
I have a VVX601 and a color side car. I also have a VVX600 with the same color side car. The lighting on the VVX601 side car is quite a bit dimmer than that of the VVX600. Does anyone else see this on theirs. We are running 5.7.0.11768 firmware. It doesn't seem to matter with the backlight brightness settings. The console is quite a bit lighter for some reason. When the backlight timer kicks in it is very hard to even read.
I have several employees who turn their volume all the way down & often miss calls. Is there a way to prevent them from turning it down?
I have searched the community, and only find very little, and what I did find, seemed to require programming knowledge, which I don't possess.
Thanks in advance,
Michael
I have a vvx 410 phone that does not recognize an expansion module (2201-46300-01). I have tried this module in other phones and if it works. The phone I have installed the latest firmware version available and I tested with the same firmware that other phones have exactly the same. is currently operating with version 5.4.0.10182. I need this phone to work with this module but it does not matter anymore what to try. I await your response Thank you