Linking SPA3102 with CUCM for using SPA3102 as PSTN Gateway for IP Phones

[faswar.mon]

Dear All,

I need your help in setting up SPA3102 with CUCM so SPA3102 can be used to make calls from IP phones (registered with CUCM) to PSTN numbers. I was struggling to make it work for many days but still not finding the clue where i am going wrong in configs. 

Let me explain what i have done,

I have configured SPA in CUCM as Third Party SIP Device and associated an extension 513, an End User "513" is also created on CUCM and entered as digest user.

Below fields are configured on SPA in Line 1 tab,

Proxy and Registration
Proxy: 10.255.255.40
Register: yes
Make call without Reg: No
Ans call without Reg: No

Subscriber Information
User ID: 513
Password: <password>
Use Auth ID: Yes
Auth ID: 513

(Above can be seen in attached Screenshot)

After configuring above, SPA successfully register itself with CUCM and i can make call to SPA extension 513 from my IP Phone (Ext 512) upon which attached analog phone rings.

Now the actual issue start here, how i can register my PSTN Line also, i have tried to use same settings as in Line 1 tab, Proxy IP,  User ID, Password, Auth ID (except port for PSTN Line is 5061), but not only PSTN Line does not register, Line 1 also becomes unregistered. and i am unable to find a way how to register PSTN Line or there is no need to register PSTN Line? and i am going in wrong direction.

Please help in setting up SPA3102 as PSTN gateway which is very necessary for me, but if someone also can help in configuration for VOIP Gateway (call from outside PSTN number to IP phones) that will be a plus for me.

Software Version of SPA is 5.1.7 (GW)

Any help will be highly appreciated in this regard.

[Howard Wittenberg]

I am not knowledgeable about configuring Cisco Unified Communications Manager (CUCM) but here are some SPA3102 fundamentals.

Think of the PSTN Line Tab on the SPA3102 as a separate adapter with its own userid and password. In your usage you already know the ip address and sip port number.  

If you send a call to the PSTN Line tab and the userid in the call matches the userid configured on the PSTN Line tab then the SPA will return a dial tone to the caller who then dials the number he wishes to call on the attached PSTN Line.

If you send a call to the PSTN Line tab and the userid is the number you wish to dial on the PSTN Line then the SPA will automatically dial that number without the caller entering the number. For this operation you need to have set "One Stage Dialing" to "Yes" in the VoIP-To-PSTN Gateway Setup on the PSTN Line Tab.

So the more sophisticated operation would be the latter and you would configure the SPA3102 ip_address:port as an outgoing trunk on the PBX. You can set the security on the SPA3102 ("VoIP Caller Auth Method") to "HTTP Digest" and set a password in your PBX trunk configuration matching a "VoIP User 1 Auth ID" and "VoIP User 1 Password" configured on the SPA3102 PSTN Line Tab under VoIP-To-PSTN Gateway Setup.

A Google search turns up this posting which may help. It shows a SPA3102 PSTN Line Tab configuration.
https://supportforums.cisco.com/discussion/11674651/generic-voip-pbx-spa3102-remote-pstn-trunk-not-subscriber

[faswar.mon]

Dear Howard Wittenberg,

Thanks for your help and detailed explanation, I am going to setup a SIP trunk on CUCM for SPA3102 PSTN Line with port 5065 and using digest for authentication. I'll get back again with results.

Regards,

[faswar.mon]

Dear Howard Wittenberg,

Need your help again.

I got busy in last few days on my job, now i got time to test according to your suggestions but still facing some issues.

Below are the steps which i have taken to configure SIP trunk but don't know why SIP trunk not working with SPA3102.

ON CUCM

1. An application user is created with name 520 and with password and digest set to 12345
2. SIP Realm is created with same name 520 and same digest 12345
3. A SIP trunk is created with name 520 and destination address 10.255.255.31 with port 5065 (SPA ip and port of PSTN Line). A SIP Trunk Security Profile is also attached with this trunk in which "Digest Authentication" is enabled and "Incoming Port" 5065 is used.
4. A Route pattern is created (e.g., 0811xxxxxxx) and linked with Gateway/SIP Trunk 520

ON SPA3102 PSTN Line

SIP Transport : TCP

SIP Port : 5065

Proxy : 10.255.255.40
Register : No
Make Call Without Reg: Yes
Ans Call Without Reg: Yes

UserID : 520
Password : 12345
User Auth ID: Yes (Testted with both Yes and No)
Auth ID: 520

VoIP Caller Auth Method: HTTP Digest (Also tested with this setting None and on SIP trunk security profile by disabling digest authentication)

VoIP Caller 1 PIN: 12345

VoIP User 1 Auth ID: 520
VoIP User 1 Password: 12345

(Attached are the settings on SPA)

After above configurations when i try to dial a number (e.g., 08112323333) from IP phone (10.255.255.22), i only hear reorder tone, i also captured SIP packets between IP Phone and CUCM, and CUCM sends "503 service unavailable" which i think issue is on SIP trunk between CUCM and SPA3102 due to which CUCM sends back message to IP Phone.

Can you or anyother person here please suggest to troubleshoot and rectify the issue.

Regards,

[Howard Wittenberg]

The Sip Transport setting must match the server's, normally you would use UDP. I am not sure TCP will work, I would try UDP.

Your dial plan of (xx.:@gw0) appears to work in my test call, however normally you would use (xx.) or none.

If the SPA3102 is sending the 503 Service unavailable it could be due to the attached PSTN Line Voltage. The PSTN Line voltage must be higher than the PSTN Line Tab setting of "Line-In-Use Voltage" which you have set to the Default 30v. This setting is used to determine if the line is already being used. You can read the PSTN Line voltage level on the SPA3102 INFO Tab. Normally an on-hook PSTN Line will have a voltage level of about 48v with an off-hook voltage level of about 6v, hence the default setting mentioned about half way between. If the PSTN Line is not connected the voltage level will be 0. If your PSTN Line voltage is lower then you will need to adjust your "Line-In-Use Voltage" setting to about half way between your on-hook and off-hook voltage levels.

The VoIP Caller 1 PIN: is not used with HTTP Digest. You have the correct security entry. I edit corrected my original statement.

I can't really comment on your CUCM settings, they sound reasonable. If you continue to have trouble it would be useful to see the sip packet exchange on a test call, either WireShark or a sip debug trace from the SPA3102.