No cisco phone with voice port?

Hi
could any one advice me about "switchport voice vlan"?
I have a configuration like:
interface GigabitEthernet 1/-/1
description test phone
switch port access vlan 100
switchport mode access
switchport voice vlan 200
when I plug a no-cisco phone into this port, the phone try to get dhcp from vlan100, not vlan 200.
do I have any chance to force it go to vlan 200? o
Any comments will be appreciate
Thanks in advance
julxu

Hi
Most IP phones have some mechanism for setting the VLAN - Cisco phones use CDP, but other phones do not. Check with the manufacturer, or post up the type of phone you have.
For example, some phones require you to set custom DHCP options to set the VLAN.
Aaron HarrisonPrincipal Engineer at Logicalis UK
Please rate helpful posts...

Similar Messages

Cisco 2960x - auto qos voip cisco-phone on access ports

After configuring "auto qos voip cisco-phone" on a stack of 2960x switches, we later noticed that after a power cycle (config was saved first) it went missing when doing a "show run" from the interface configs, on one of the switches (g1/0/1-48 had it applied, then after power cycle it didn't show). However, doing a "show auto qos interface" shows it is still applied to all interfaces g10/1-48.
I try to do "auto qos voip cisco-phone" again on g1/0/1-48, and it shows this:
SwitchStack01(config-if-range)#auto qos voip cisco-phone
AutoQoS Error: AutoQoS already configured
% Range command terminated because it failed on GigabitEthernet1/0/1
If I attempt to remove "auto qos voip cisco-phone", it shows this:
SwitchStack01(config-if-range)#no auto qos voip cisco-phone
AutoQoS Error: AutoQoS not configured
% Range command terminated because it failed on GigabitEthernet1/0/1
When I do a "show run", it is no longer displayed for the first switch of the stack.So is it still applied to the interfaces or not? Is this a bug? Has anyone else experienced this? This has happened to at least a couple stacks of 2960x's we have.
Thanks!

It appears to be a bug in the code. Open a TAC ticket with Cisco and send them the output and the IOS version you are using so they can help you resolve the issue. Most likely, they recommend an upgrade.
HTH

HT3577 How to unlock my phone with voice over

I turned on voice over & now I can't unlock my phone. Please help. Thank you.

ONLY the carrier to which it is locked can unlock it. No one else at all.
You would have to contact AT&T and see if they offer this service now. They have never unlocked iphones, but there have been articles saying that they may start unlocking them soon,
Contact them and ask.

Best way to go for Cisco UC with Lync 2010 Integration

I have requirements to integrate Cisco UC 9.1 (single cluster) with Lync 2010. The decision is to use Lync for IM/presence and use Cisco phone for VOICE features.
Now I have two options which are Microsoft RCC and CUCI-Lync in my mind.My questions are:
Are CUCI-Lync and MS. RCC conflicting features? If not, can we enable both of these features and what could be the strong points to do this way?
Cisco Application dial rule and directory lookup features will allow Lync client to see the calling name displayed when making calls to the other Lync users. However, the inter site calls are sent through PSTN which does not accept the calling name IE in ISDN so the called party will NOT see the calling name displayed on the Cisco IP phone. In this scenario, would the Lync client still be able to see the name displayed on this call?
Can we have presence status updated to Lync clients when Cisco handset if offhook/ busy?
Appreciate for any feedbacks on the above. Thanks

Hi
Firstly, Lync + Cisco often isn't the best solution - yes, Lync is good, and yes, Cisco voice is good. That doesn't mean that by having two systems you are getting the best solution.
CUCI-Lync works when implemented properly but it's not perfect.
Anyway.... I'm sure you'll properly pilot the solution!
1. Yes, they conflict. CUCI-Lync performs more advanced call control than RCC and there's no real reason to implement RCC with it. The only case I can think of for that is maybe for Macs, as there is no CUCI-Lync for MAC. As I recall there's nothing to stop both being configured....
2. As long as each endpoint has it's own DDI presented when dialing over the PSTN, the reverse lookup rules should still allow a correct lookup against AD (so if each AD account has both the 'internal' extension number and the DDI in attributes that CUCI-Lync uses for reverse lookup then it should display the name).
3. Yes - you need to carry out the steps in the documentation that describe setting up 'custom presence states' for this to work though. People miss that a lot...
Regards
Aaron

How to Configuration Cisco 3725 with NEC ASPILA EX

Dear all;
Now i have Cisco 3725 with 1-Port Channelized E1/T1/ISDN-PRI, i am connect to NEC ASPILA EX with PRI I/F (1PRIU-A1.
The controller link state up, but when clients dialin to RAS not have ring back or not connect to RAS.
anyone can help me?

Hi;
i'am config cisco as you recommended is "isdn protocol-emulate network" and "clock source should be internal". After the remote computer call to RAS it have modem signal and then connected, next time it disconnect. can i change some parameter for this problem or what i'am wrong?. I post config, status, and debug message for you. Help me..
===== show isdn status ===========
#show isdn status
Global ISDN Switchtype = primary-net5
ISDN Serial2/0:15 interface
******* Network side configuration *******
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0xFFFF7FFF
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
=============== sh controllers e1 2/0 brief ========
#sh controllers e1 2/0 brief
E1 2/0 is up.
Applique type is Channelized E1 - unbalanced
No alarms detected.
alarm-trigger is not set
Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
Module type is Channelized E1/T1 PRI
Version info Firmware: 0000001D, FPGA: 0
Hardware revision is 0.0 , Software revision is 29
Protocol revision is 1
number of CLI resets is 0
receive remote alarm : 0,
transmit remote alarm : 0,
receive AIS alarm : 0,
transmit AIS alarm : 0,
loss of frame : 1,
loss of signal : 1,
Loopback test : 0,
transmit AIS in TS 16 : 0,
receive LOMF alarm : 0,
transmit LOMF alarm : 0,
========== Interface config.=============
controller E1 2/0
clock source internal
line-termination 75-ohm
pri-group timeslots 1-31
interface Serial2/0:15
no ip address
ip nat inside
encapsulation ppp
ip policy route-map nachi-worm
dialer rotary-group 1
dialer-group 1
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice modem
no fair-queue
no cdp enable
=================Debug Message when call to RAS ===========================
Mar 6 22:40:29 BANGKOK: ISDN Se2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x000B
Bearer Capability i = 0x8090A3
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838B
Exclusive, Channel 11
Calling Party Number i = 0x0081, N/A
Plan:Unknown, Type:Unknown
Called Party Number i = 0x81, '075205600'
Plan:ISDN, Type:Unknown
Low Layer Compat i = 0x8090A3
High Layer Compat i = 0x9181
Mar 6 22:40:29 BANGKOK: ISDN Se2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x800B
Channel ID i = 0xA9838B
Exclusive, Channel 11
Mar 6 22:40:29 BANGKOK: ISDN Se2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0x800B
Mar 6 22:40:29 BANGKOK: ISDN Se2/0:15 Q931: TX -> CONNECT pd = 8 callref = 0x800B
Mar 6 22:40:35 BANGKOK: %ISDN-6-CONNECT: Interface Serial2/0:10 is now connected to unknown unknown
Mar 6 22:40:46 BANGKOK: %ISDN-6-DISCONNECT: Interface Serial2/0:10 disconnected from unknown , call lasted 17 seconds
Mar 6 22:40:46 BANGKOK: ISDN Se2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x800B
Cause i = 0x8290 - Normal call clearing
Mar 6 22:40:47 BANGKOK: ISDN Se2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x000B
Mar 6 22:40:47 BANGKOK: ISDN Se2/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x800B
==============================================

Cisco Phone 7960 and SIP provider

Hi,
i have an account with a Sip provider.
I have all information for make a connection with xlite sip client but if i try to configure a Cisco Phone with SIP Firmware (7.5), phone not work.
My provider is messagenet.it.
Can you help me?
Thanks

Hello,
have a look at the configuration guide "Getting Started with Your Cisco SIP IP Phone" at
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080080edf.html
This should pretty much answer your questions and allow you to succeed with your task.
Hope this helps! Please rate all posts.
Regards, Martin

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

How to provide voice to more than 2 phones with WRTU54G-TM?

I'd like to use this router to serve more than one phone on each port (since I have more than one phone in the house that share the same line). Ideally, I could connect the router to the wall jack and allow it to serve all of the phones in the house that are connected to that line - something that the User Manual specifically warns against. Any suggestions? Allowing the router to serve more than one phone per port would significantly broaden the market for this router, as the current restriction excludes anyone with more than one phone (including cordless) per line (Directory Number). WORKAROUND The only workaround that I can think of to get more than one phone on a line is a multi-handset cordless solution, where one base station connects to one Phone port, but it connects to multiple, cordless phones. Example: Vtech i5871). Other than this restriction, I am pleased with the new, T-Mobile-specific router, WRTU54G-TM. Robert

Hi -
Unfortunately, I'm not aware of any changes you can make to the router or phone service to change the way the voice mail indicator work with various phones.
The stutter tone has been a standard for a long time - I'm not sure what the variables might be between phones, but I've read a number of accounts of folks with other phones that do work the T-Mobile@home service and the Linksys router and activate the VMWI (visual message waiting indicator) properly.
If you haven't yet, I'd suggest trying T-Mobile customer care, asking for the department that supports @home service, or the Linksys router. Unfortunately, with the popularity of the new service, it seems that T-Mobile underestimated the need for tech support on this, and the wait times (which I normally found to be short for T-Mobile over the last 7 or 8 years I've been a mobile customer) are now generally pretty long. But you should be able to leave a phone number, and have them return the call to you if the wait queue is too long.
Also might be worth checking with the phone manufacturer to see if there's anything you can do to modify the way the phone works.
I did a LOT of googling about vmwi's, stutter tones, FSK (another technology for controlling vmwi's), etc. and was amazed at how LITTLE is out there on the net for an industry standard like this.
I wish I had more info to share with you about this. Please post back your experience -- this is clearly an area where we need more information sharing in general.
- Jon

Cisco 3640 cant configure voice-ports

I've an cisco 3640 with a VWIC-2MFT-E1 network module. Can anybody tell me why i cant configure any voice-port? without able to configure voice-port, i cant assign any port for my "dial-peer voice xx pots". Anybody has any solution?
what im trying to do is sending voice data traffic to 192.168.10.1
controller E1 1/0
framing NO-CRC4
clock source line primary
channel-group 0 timeslots 1-31
interface Serial1/0:0
bandwidth 2048
ip address 192.168.10.2 255.255.255.252
ip mtu 300
encapsulation ppp
no ip route-cache
no fair-queue
no cdp enable
dial-peer voice 777 pots
destination-pattern 77T
no digit-strip
direct-inward-dial
forward-digits all

I've only got this from my controller E1
Vertigo(config-controller)#?
Controller configuration commands:
channel-group     Specify the timeslots to channel-group mapping for an
                    interface
clock             Specify the clock source for a DS1 link
default           Set a command to its defaults
description       Controller specific description
detect            Enable detection of loopback request
exit              Exit from controller configuration mode
framing           Specify the type of Framing on a E1 link
guard-timer       Guard timer (0-20) for xcsp calls - timer duration and
                    accept/reject on expiry
help              Description of the interactive help system
line-termination Specify the line termination for E1
linecode          Specify the line encoding method for a E1 link
loopback          Put the entire E1 line into loopback
mode              Configure the controller mode
no                Negate a command or set its defaults
shutdown          Shut down a E1 link (send Blue Alarm)
tdm-group         Configure DS0 group for TDM
Do i need DSP for the voice-ports? Please guide me on what more module i need to get. Thanks!!
https://supportforums.cisco.com/message/3022639;jsessionid=AF45FBC0DDB8636D70F9B526A367016F.node0

Catalyst Express 500 802.1q with non-Cisco Phones

This weekend we spent hours trying to get 802.1q tagging to work on a VLAN with ShoreTel phones. The user interface on this switch seems to only allow "Cisco-Voice" VLAN, without any specifics. This didn't work. The specs on this switch say that the .1q is supported, but we couldn't figure it out. The more expensive switches were easier to configure for Voip QoS.
Can anyone advise me on the tricks to getting this to work with the lower end Catalyst Express 500? Or does this switch only support 802.1q with Cisco phones?

Cisco IP Phone uses CDP to let the ip phone know what vlan it's suppose to be (via voice-vlan). shore tel would definitely not use CDP since CDP is cisco proprietory, so it's voice vlan must be defined on it, I rememer Avaya being the same way. So, having said that, just make sure that the Shore tel Ip phone are in the right vlan. what does not work anyway? shore Tel IP Phone will not come up? Will not get it's configuration from it's software PBX? Use the smartport configuration on CE500.
Please rate all posts.

Using Cisco 7961 Phones with Linksys SRW208P POE Switch

I am having a problem getting our Cisco 7961 phones to work with a Linksys SRW208P POE switch. The internet works great , but when I connect a 7961 Cisco phone it hangs at the " configuring IP " screen. Are these model phones compatible with this switch? Any Ideas. Thank You!

I'm having a similar problem with the Cisco 7941G phones using a Linksys SLM224G4PS (with PoE). Did you ever get them to work?
The 7941Gs work with the plain Linksys SR224 and using power injectors, but not with the smart switch. It has nothing to do with the PoE, I've used the power injectors with the SLM224G4PS. It wasn't a current overload problem.
I've gotten the phones to work for somewhere between 10-40 seconds, which has given me enough time to find the IP address (DHCP) then pull up the web interface of the Cisco 7941G.
The phone on the SR244 had GARP enabled and Voice VLAN enabled and the operation, the phone on the SLM224G4PS had both disabled.
The working phone on the SR244 had a operational VLAN ID of 31, the non working phone on the SLM224G4PS didn't have any indicated.
I'm guessing that the root of the problem is the Voice VLAN, but tech support tells me that out of the box the switch should pass everything.
Apparently not so much...

SPA112 can not hear voice on my android phone with CSipSimple

I configure a SPA112 and between lines 1 and 2 the calls and answers is ok (I receive, attend calls and hear the voice). When my cell phone call for any of the rows, is also okay. The problem is when one of the lines dials to my cell phone. I answer calls and only hear noises in the cell and the line is silent. I've used iptables the, seep show peers (in asterisk 1.8) is ok, etc. ... my dial plan is: (* xx | [3469] 11 | 0 | 00 | [2-9] xxxxxx | 1xxx [2-9] xxxxxxS0 | xxxxxxxxxxxx. )
e.g.: I'm using dyndns ... Does the problem is with RTP port?? Or codec?
thank you

Hi Neuberfran,
It is sould like RTP issue part of RTP is codec. If it is codec issue you can go to:
ATA GUI --> select admin mode in top right site, Select Advanced mode in top right site
select Voice in menu bar,
Line 1:
go to Audio Configuration section
Preferred Codec G711u
Second Preferred Codec G711a
Thired preferred Codec G729a
click Save,
do same setting in your Line 2 because you useing two lines
example: if your ATA IP is 192.168.1.100
go to your router port forrwarding and make rules UDP port 10000 to Port 30000 to your ATA 192.168.1.100
another port forwarding for SIP UDP port 5060 to port 5080 to your ATA 192.168.1.100
Thanks,
Moh

IPhone 5s with hearing aids via Bluetooth. Need to use the phone for voice, what's the best distance that the mic will pick up your voice

I wear hearing aids that are made for Apple, the audio will go into the hearing aids via Bluetooth. I'll need to use my phone for voice only.. I drive for work and have a wok oh one too and both are synced to a bluetooh headset (LG). I'm trying to figure out how to talk on both phones ( use speakerphone on work phone?) without using a bluretooh headset. I don't know the range of the microphone on the IPHON, thinking about hanging it around my neck or put the phone into a stand/holder on the dash and use the speakerphone feature. Looking for ideas to make this simple

You can't do anything with that problem.
Take your iPhone to the closest Apple Store and they will give you new one for 200€/$ or if you bought your iPhone in less than 12 months you will get it for free.

I have a new iPhone 5S. While trying to learn about it, I accidentally recorded a voice memo with no content. I cannot now figure out how to get rid of it. There is a banner across the top of my phone with this memo which I don't want. Help!

I have a new iPhone 5S. While trying to learn about it, I accidentally recorded a voice memo with no content. I cannot now figure out how to get rid of it. There is a banner across the top of my phone with this memo which I don't want. I have deleted it from iTunes but cannot get it off the phone. Help!

The banner usually indicates that the memo is "Paused." If you go back into voice memos, touch the word "Done" beside the big red pause button, give it a name, then it will show in a list. Touch the memo in the list then touch the trash can icon that should appear.

Cisco Phone Control and Presence 8.6.1.1185 with IBM Lotus Notes 8.5.2 (Integrated Sametime Client 8.0.2) - No presence status visible

Hi community,
I am trying to integrate Cisco Unified Presence 8.6.1.10000-34 with IBM Lotus Notes 8.5.2 with the integrated Sametime Client version 8.0.2 via the Cisco Plugins 8.6.1.1185.
Phone control is working fine, whereas the presence status is not shown (= no handset symbol next to the Sametime user). When I look in the preferences of the plugin, I can see that the plugin has connected successfully to the CUCM (8.6.2.20000-2),whereas the connection to the CUPS has not been established.
The user id as well as the password are all the same on all systems. Here is a description of what I have configured via the ciscocfg.exe tool:
Feature Control:
- Enable Phone Status -> checked
- Enable Dial Using Cisco IP Communicator -> unchecked (not required)
- Enable Control Desk Phone -> checked
- Default Mode -> Control Desk Phone
Control Desk Phone Settings:
- Voicemail Pilot Number -> left blank (no voicemail)
- Cisco Unified Communications Manager
     - Servers -> IP address of CUCM
     - Read Only -> unchecked
     - Use as Default CUCM -> checked
     - Synchronize Credentials -> checked
          - Use Sametime Credentials -> checked
Use Secure Connection: -> not required
LDAP Phone Attributes: -> not required
Phone Status Settings:
- Cisco Unified Presence Servers -> IP address of CUPS
- Read Only -> unchecked
- Synchronize Credentials -> checked
     - Use Sametime Credentials -> checked
- Sametime User ID Mapping
     - Use Business Card Attribute -> MailAddress
     - Remove Domain -> checked
- Display Off-Hook Status Only -> unchecked
At the moment I don't see an error in the configuration, but maybe I am wrong. Could anyone please tell me what the error could be?
Thanks a lot in advance!
Kind regards,
Igor

Hi all,
here are some additions to my above post:
Servers and clients used:
1x CUCM 8.6.2.20000-2
1x CUPS 8.6.1.10000-34
1x IBM Lotus Domino Messaging Express Server 8.5.2
1x Sametime Entry Server 8.5.2 (on top of the Domino server)
2x IBM Lotus Notes 8.5.2 with integrated Sametime 8.0.2
2x Cisco Phone Control and Presence with Lotus Sametime (PCAP) 8.6.1.1185
2x Cisco Unified Personal Communicator 8.5.5.19839
Setup:
- CUCM, CUPS and CUPC are working fine, i.e. Desk Phone control via CUPC, as well as availability and presence status are working without issues
- IBM Lotus Domino server is the LDAP Directory, the Sametime Entry Server is installed on top of the Domino server and uses the Domino Directory
- User ID and password on CUCM/CUPS match the ShortName field and password in Domino
- The PCAP plug-in has been manually deployed to both Notes clients with the following configuration:
     - Enable Phone Status -> active
     - Desk Phone Control -> active
     - no credential synchronization for CUCM and CUPS, i.e. every user must fill the user details himself
     - Sametime User ID Mapping is implemented via the LDAP Attribute uid (which is equal to the user id in CUCM)
     - LDAP configuration filled in with details of the Domino server
Phone Control is working fine, also the connection to the LDAP server (Domino) is fine. However, when I type in the credentials for the CUPS server login, I can see (in Troubleshooting pane) that the user (pparker) is connected to the CUPS server for a short period of time and then gets disconnected. After that no connection is possible to the CUPS server, i.e. status is always disconnected.
I have collected the Tomcat (EPASSoap00010.log and security00010.log) logs via RTMT and compared them to the logs from the PCAP plugin. The relevant time period is from 15:14 to 15:17. In the Tomcat logs I can see that the authentication is successful (see attached files), however in the log of PCAP plugin I can see the following messages:
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
I don't understand why the connection is rejected although the Sametime Internal ID and CUPS User ID match. Does anyone know what the issue could be?
All posts are very much appreciated!
Thanks a lot in advance!
Kind regards,
Igor

No cisco phone with voice port?

Similar Messages

Maybe you are looking for