ATA 186 to connect to an SPA3102

I have things hung as follows:

PBX<==>SPA3102<==><==>ATA186 analog telephone

IP addresses:

SPA3102 - 10.1.1.144

ATA186 - 10.1.1.45

SPA3102 Config:

5.2.13 firmware (GW002)

Line1 is off.
Ethernet cable is connected to the Internet port

On the PSTN line tab:

SIP port: 5060

POST SIP Port: 5060
Proxy: 10.1.1.45

Register: No.

Call without Reg: Yes

Call for years without Reg: Yes
Full name: 423
User ID: 423

Password: 123456

Numbering plan 1: (S0<:gw0>)

Dial Plan 2: (S0<:10.1.1.45>)

PSTN ring across the line 1: No.

ATA186 Config:

Firmware: 3.02.01

Under SIP settings:

UID0: 423

Password0: 123456

SIPRegOn: 0

Proxy: 10.1.1.144

Anyway, I can't get these two to work in this configuration. When I try to make a call from a PBX to the ATA186, I see from the SPA3102 will receive an invitation to the ATA186. The ATA186 responds with a "404 Not Found".

When I try to make a call from the ATA186 to the PBX, I see an invitation of the ATA for the SPA. Inside this prompt, there "sip:@10.1.1.144. Then I see the answer of the SPA with a "status: 100 Trying" followed by a "status: 503 Service unavailable '.

No indication on how to connect the two devices together will be very appreciated! Thank you!

Assuming that your Dial Plan 1 is the VoIP calling by default DP under the Voip to PSTN gateway, I think the dial plan should be (xx.).  Maybe the (S0<:gw0>) works, I don't know, try (xx.) and see if that makes a difference.  You told me that was planted in the line PSTN tab UserID to 423.  If the ATA call 423 ([email protected]/ * /: 5060) the SPA should return a tone and wait the digits for numbers.  If the ATA call the PBX extension number ([email protected]/ * /) where xxx is the number of post, then you have a phone on a floor and the SPA3102 should dial this number on the FXO port after there he won.

The INVITATION contains the number sent to the SPA3102.

If you are running WireShark he reformatted the sip signaling so that you can see exactly what is happening.  After you capture a test call click you on the phone, then on "Voip calls".  WireShark then puts up a screen display detected than VoIP calls.  You click your test call to highlight, and then click 'stream '.  Wireshark is implementing another magnificent screen that shows the SIP signaling between the ATA and the SPA3102.  My version of WireShark is Version 1.6.4.  There may be future versions.

Another debugging tool is the Trace of the Sip Debug.  You install a program on a local computer syslog, put the ip address of the local computer under debug the server on the SPA3102 System tab, set the level of debugging on 3 in the System tab and set the Option Debug Sip to FULL-on the tab line PSTN (then on the tab line too if you use).  You can download a simple Windows pc syslog program here:

https://supportforums.Cisco.com/docs/doc-9862

Traces of Debug Sip will show internal things going on inside the SPA3102, which do not appear in the Trace of WireShark, but WireShark trace shows exactly what is happening inside and outside the SPA3102 on the ethernet link.

The duration of CPC Min setting you mentioned is for the detection of a signal coming into the attached FXO analog line port CPC.  As you know a signal CPC is a complete the voltage drop on the line of signalling the far part disconnected from the call.  You would increase the minimum to stop detecting voltage drops false that disconnect an ongoing call.  I'm surprised that increase the duration would solve your problem of disconnection.

Tags: Cisco Support

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