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SPA3102 No incoming SIP calls even directly connected to Internet

SPA3102 FW 5.2.13(GW002)    Hardware Version:    1.4.5(a)

- ATA DOESN'T RECEIVE CALLS WITH ANY OF THE 3 SIP SERVICE PROVIDERS I TRIED. I'm able to receive calls with any of those accounts registering with Android Csipsimple App, connected via wifi to my router without port forwarding.

- Can generate calls with either of SIP providers, via Line1 or any programmed  gateway.

- Not connected to PSTN, just an analog phone to the FXS Port

- Network topology described below (ATA in front o behind router) doesn't seem to impact performance/results. No port forwarding needed as the router seems to handle traffic. Currently SPA3102 directly behind cable modem to minimize variables, such as any potential NAT issues (Setup A below)

Physical setup

A) Current:

   Internet - Cable Modem - SPA3102 (As Bridge) - WRT54GL (Tomato Firmware)

B) Previous setup:

    Internet - Cable Modem - WRT54GL (Tomato Firmware) - SPA3102

I've tried multiple configuration changes but I'm running out of ideas and I'd welcome any suggestions you may have. I'm attaching my "config.xml" file (with some personal info removed).

Thanks in advance for any suggestions you may have.

S

2 REPLIES
New Member

SPA3102 No incoming SIP calls even directly connected to Interne

Now I'm even more baffled than before:

-I'm still unable to receive sip incoming calls.

-BUT I've managed to receive incoming calls using Sipbroke PSTN Numbers. I dial the access number using PSTN, upon automated answering I dial my sip provider code + UserId and the call gets through to my SPA3102 with no issues.

I'm pulling my hairs on this one. Any ideas?

Re: SPA3102 No incoming SIP calls even directly connected to Int

-BUT I've managed to receive incoming calls using Sipbroke PSTN Numbers. I dial the access number using PSTN, upon automated answering I dial my sip provider code + UserId and the call gets through to my SPA3102 with no issues.

In both cases the call goes from your voip provider's server to the SPA3102 which implies that the SPA3102 is satisfactorily registered to your voip provider's server.

The difference would seem to be how the call gets to your voip provider.  In the case of SipBroker, sipbroker is sending a sip uri call to your account at your voip provider.  I am not sure how you were doing the test call testing, however, you would either be calling a DID that you had with the voip provider where an incoming call would be going at some point thru the pstn network to your DID or you would be sending a sip uri call directly to your account in a manner similiar to SipBroker but bypassing SipBroker. 

If you were testing by calling a DID that you have setup, what about the routing for the DID setup on your account at your voip provider?

You didn't mention anything about setting NAT Mapping Enable: Yes, which is a setting that I would have, however the fact that you can communicate thru SipBroker probably means that the NAT setting doesn't make any difference.

I would run a sip debug trace to see if the failing call reaches the SPA3102 or not.  You run a sip debug trace by downloading and installing a syslog program on a pc, putting the pc's local network address under Debug Server on the SPA3102 System Tab, setting the Debug Level to 3 on the SPA3102 System Tab, and setting the Sip Debug Option to FULL on the SPA3102 Line 1 Tab.  You can download a simple syslog program from Cisco here:

https://supportforums.cisco.com/docs/DOC-9862

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