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AuthorTopic: How to add linksys pap2t to callmanager  (Read 1228 times)

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« on: 17, 09, 2013, 08:54:41 »
Below is a quick guide I made for myself:


Setup CallManager
1. If not already existing, set up a Phone Security Profile (System/Security Profile/Phone Security Profile) "third-party SIP basic - PAP2" and select Enable Digest Credential for this profile.
2. Create a new user ccm end user or create one on the LDAP
a. %Userid%: normal value of userid (example: linksys)
b. Password: what you want, it is not used in SIP
c. PIN: same
d. Digest credentials: a sequence of digits
 
3. Create New Device "third party basic" (WARNING: to configure two lines on a single module, you must declare a SIP Third Party ADVANCED)
a. Mac address: value that you want, it is not used. Preferably, to the value of the real mac address of the module
b. Device security profile: profile third-party basic PAP2 previously created
c. Owner User ID: %userid% of the user (optional I guess)
d. Digest User id: idem
4. Associate a DN on this profile, like on any phone.
a. Busy trigger and max call to 1
5. Associate a second DN on the same profile if necessary
6. Return to the end user
a. In association device, associate the sip device you just created
 
 
Setup PAP2
1. Connect the module on the network, the port configured with access vlan in the vlan voice, not as voice vlan! The Voice Vlan CDP is not recognized by the module, therefore it must be put in access mode on the switchport.
2. DHCP is enabled by default, so the module will receive an IP address.
3. Connect an analog phone on the port 1 of the module and call **** to get an interactive menu
4. Dial 110 # to listen the configured IP
5. Login to the web page of the module
6. Switch to admin mode
7. Line 1 tab
a. Line enable:  yes
b. Proxy: IP of call manager who will be its unique subscriber
c. Register: yes
d. Display Name: the name of the fax for instance
e. User ID: the number of the DN associated with the profile
f. Use Auth ID: Yes
g. Auth ID: the %userid% that is associated with the SIP device
h. Password: the digest credential
i. Use pref codec only: Yes
j. SAVE

8. Reproduce the same configuration in Line 2 if the same module has two different lines
At this point, the module must be registered in the CUCM: check this in the CUCM webpage
 
 
Fine tuning needed for FAX:
On the web page, switch to Advanced view
1. System tab: set a user and admin password for web access (userid is the user and admin respectively for each part)
2. In Regional tab, Interdigit Long Timer:  5 (T302 is the timer, in seconds), save
3. In SIP tab: RTP Packet Size: 0020 (= 20ms)
4. In the Line tab:
a. Three way call serv : no
b. Three way conf serv : no
c. Echo cancel enable : no
d. Echo cancel adapte enable : no
e. Echo supp enable : no
f. Call waiting serv : no
g. Network jitter : very high
h. Jitter buffer adjustment : disable
i. FAX passthru method : reINVITE
 
This should then operate in FAX passthrough only.



ref: https://supportforums.cisco.com/thread/2082101