I have a polycom registered to Callmanager 7.1.2. Does anyone know where I can find
documentation on the 3rd Party SIP basic calling features. I have a Polycom confernce phone and would
love to do meet-me or ah hoc conferencing.
Right now I only have limited to no features. Just New Call and Forward
The polycoms use internal conferencing resources, don't use the CUCM conferencing resources and will only support a 3 party conference - here are some possible workarounds:
* Initiate conference from Cisco IP phone * Use meet-me conference * Trade-in to CP-7937G conference station with SCCP firmware
Does anyone can setup Polycom soundstation IP 6000 in Callmanager server??
Please kindly share your experience. I haver setup basic SIP or advance SIP phone with digest user configuration, the phone is still display reject status.
Thanks in advance!!
I am also facing an issue for setting up Polycom IP 6000 with CUCM 8.0
If any one has setup this plz share the config process of the same.
Configuration on the Callmanager:
1 - Add a new End User
The User ID can reflect the SIP Extension that will be assigned to the Polycom SoundPointAR IP or SoundStationAR IP.
If the Customer uses Active Directory or does not want to use numerical User ID's a alpha numerical entry (name etc) can be created.
A minimum 5 Digit Password must be used. This Password is only used CCMAc internally Example: 12345
The same Value is used for the PIN.
The Pin is the authentication Password used on the SoundStationAR IP or SoundPointAR IP Phone.
The Last Name Entry is used to
identify this User and is mandatory
2 - On the Cisco Call Manager click on Device and select Phone.
3 - Click on Add New and select a Third Party Sip Device (Basic) and click on Next to proceed
4 - Add the Mac Address of the Soundstation Phone and ensure that you select the Phone Button Template as highlighted and the Owner User ID should reflect the User Id (case sensitive) that you have assigned to the User.
5 - Then choose the Device Security Profile, SIP Profile and the Digest User is the User Id (case sensitive) that you have assigned to the User and press save to store this Information within the Cisco Call ManagerAc.
6 - Click on Add a new DN Button in order to assign the Selected SIP Extension
7 - Add the desired SIP Extension and add the Alerting Name
8 - The Display (Caller ID) and ASCII Display (internal Caller ID) is a feature that displays the added entry on CiscoAc Phones when they receive a call from a Polycom SoundStationAR IP or SoundPointAR IP.
9 - After this finalize this process via clicking on the Save Button.
SIP 6000 :
Configuration on the PolycomAR Phone Menu:
Default login name Polycom password 456
1 Press the Menu Button
2 Choose (3) Settings
3 Choose (2) Advanced
4 Password is 456
5 Choose (1) Admin Settings
6 Choose (2) SIP Configuration
7 Select Server and add the IP Address of the Cisco Call Manager on the Address
Select port:5060, Transport: "UDP Only" and Register:"Yes"
8 Press Back
9 Select Outbound Proxy and add the IP Address of the Cisco Call Manager on the Address field. Select port:5060, Transport: "UDP Only".
10. Press Back.
1 Confirm this selection and scroll down to "Line"...:
Enter a Display Name 1316
Add the Extension that was added within the Call Manager Menu as the "Address". Example: if the extension number (dn) is 3553, the "address" is 3553.
Add the "Third Party Name": extension@ip of callmanager. Example: firstname.lastname@example.org
On the Auth User ID add the user that was selected as the Username within the Call Manager End User Menu
Add the 5 digit Auth Password that was selected as the PIN Entry within the Call Manager End User Menu.
Optional add a Label that will be displayed next to the Line Key
Press exit until the Prompt comes to save the Settings and the Phone will reboot after this
No need to modify admin setting>network setting
Except SNTP server ( NTP server)
No need to modify admin setting>server menu
Thanks for your detailed config steps. I am following the same steps.
I have Polycom SoundStation IP 6000 phones with SIP version 3.0.2.0917.
After doing SIP configuration on polycom phone, i am saving the same. But as soon as it reboots, configuration is wipe out. Configuration is not getting save. Server & Line configuration is disappeared.
Any idea why is this behavior ?
During the boot up, phone is showing error "Could not contact boot server, using existing configuration".
In DHCP pool in router, option 150 & option 66 (which is used by polycom) has configured to provide TFTP server.
Following are DHCP Menu configuration in phone:
Do i miss any config parameter ?
I'm testing an ip7000 device similar to yours. Basically I think you need a FTP server to do any sw upgrades hence the tftp issue-it does not recognise the tftp. I've found that the proxy is the only ip address you need to enter to make and receive calls but that ip address must be one of the servers you have in the dp/cmg you assigned to the device in ccm. The server option in the handset settings is the FTP server- you can blank this out and the phone registers ok as long as you add a proxy . The other one I've seen is that the end user Id does not have to be the dn -this normally needs to be the dn if you configure say a ipblue, Bria sip phone. With the ip6000/7000 range the user id and dn does not need to match, As the phone compares its auth user and auth pin configured in the phone against the user Id and digest pin when the security setting in the phone gui page states use digest
Hope it helps
Sent from Cisco Technical Support iPad App
Thanks for your reply.
Do we need any cnf file or any other file to upload on phone prior or polycom will fetch its cnf file from cucm directly ?
The DHCP scope defined in Callmanager server use TFTP only. The Polycom need around three to four minutes to bootup.
I don't know why the Polycom need longer bootup time than IP phone sets.