Voxilla VoIP Forum

MillsapsPE · #1 (**permalink**) April 27th, 2005, 10:54 PM

I have 2 SPA3000's acting as FXO ports to Asterisk@Home 0.6. A SBC POTS line is connected to one 3000 (Box A), and my Vonage Cisco ATA is connected to the other (Box B). I was unable to get CID info through * and to the phones.

So... I factory reset the 3000's, and went through the Wizard here. After following the wizard, I could make and receive calls, but still no CID. Then I found that by default, the Wizard doesn't set the PSTN CID For VoIP CID to "yes". So, I set the value to yes on both 3000's. This change fixed Box A, but it did not fix the CID for Box B. Not only did it not fix it, Asterisk no longer answers the call. When I call Box B via my cell phone, I can see that Box B "sees" the call by using the GetSipura program from digiblur. It rings 2-3 times, then the call goes blank. No info is shown on the CLI (verbosity=6). When I switch the PSTN CID For VoIP CID back to No, the calls will go through, but still no CID. Actually, the CID on the phone is the Display and User ID from the PSTN tab of the 3000.

I then followed the instructions here on setting up Box B to not answer the call and let it pass through directly to *. This resulted in the same, no CID and if I turn on PSTN CID For VoIP CID, Asterisk doesn't "see" the incoming call.

Why would one box work and the other not? It would seem that all things are equal, with the exception of what's plugged into the PSTN port. After typing this last sentence, it dawned on me to try switching the 2 boxes. I plugged in the Vonage Cisco ATA186 into Box A and the SBC POTS line into Box B. Wouldn't you know, Box A stopped sending CID info. Therefore, the problem is following the Vonage Cisco ATA186 line.

What could be the problem with the interaction between the Cisco ATA186, the SPA3000 and Asterisk?

Side note: I have a livevoip account too. CID works for this trunk.

I'm using Asterisk@Home with an IP500 and SPA841's.

MillsapsPE · #2 (**permalink**) April 27th, 2005, 11:26 PM

Sorry for the confusion, Box B is a SPA3000 that I purchased. I have the Vonage supplied Cisco ATA186 line plugged into the PSTN port of the 3000. I have full control over the SPA3000.

ATA186 --> SPA3000 --> Asterisk

mberlant · #3 (**permalink**) April 28th, 2005, 12:15 AM

What happens when you connect an ordinary Caller ID phone (or CID display box) directly to the Vonage ATA? Do you get a clean CID there? How many seconds elapse between the time you receive the first ring and the time you see the CID display appear? Is this elapsed time a couple of seconds longer than when you run the same experiment with your POTS Line?

I am betting that the ATA186 delivers the CID information a couple of seconds longer than does your POTS service. The SPA3000 is answering the phone before receiving the CID info. You may need to lengthen the PSTN Answer Delay (I can't remember where the parameter is or it's exact name.) by a couple of seconds to give the ATA186 more time to deliver the CID to Asterisk.

MillsapsPE · #4 (**permalink**) April 28th, 2005, 05:17 AM

I know the Cisco ATA186 delivers CID. I used to have an analog phone hooked directly to it. I'm not sure about the delay though. I'll have to try that tomorrow when I get back to work. Thanks for the idea.

But... would that cause Asterisk not to even "answer" the call. When I turn on PTSN CID to VoIP CID, Asterisk doesn't even answer the call. The call will ring the correct number of times, then the call (from the caller side) goes silent. Almost like someone answered the phone but no talking. Asterisk CLI doesn't show any activity.

mberlant · #5 (**permalink**) April 28th, 2005, 05:48 AM

I think we're getting closer. First, I think you need to pith Box B again and restore it to the point where it can make and take calls, just without Caller ID.

Then, look at the Info page (or, better, Samurize) you receive a call via Vonage and see if the SPA sees the CID from the Cisco and fails to pass it on to Asterisk or if it never gets the CID from Vonage.

Good luck.

MillsapsPE · #6 (**permalink**) April 28th, 2005, 06:00 AM

Quote:

Originally Posted by mberlant

I think we're getting closer. First, I think you need to pith Box B again and restore it to the point where it can make and take calls, just without Caller ID.

Then, look at the Info page (or, better, Samurize) you receive a call via Vonage and see if the SPA sees the CID from the Cisco and fails to pass it on to Asterisk or if it never gets the CID from Vonage.

Good luck.

I use GetSipura already. I know the CID info shows up on the GetSipura screen. :idea: Now that I think about it, when I'm watching the CLI and the GetSipura screen side by side, I think the CLI starts moving about the same time the CID info pops up on GetSipura (with PSTN->VoIP CD off). So maybe I do need to increase the time before the SPA forwards the call to *. That will be the first thing I try tomorrow. I almost can't wait to go to work now. :wink:

BTW, GetSipura is very cool. Anyone with any SPA unit (except 841) should use it. While I'm at it, the backup copy utility posted here is also very handy. That has helped play with different configurations for the 3000. I can always go back to a good version.

MillsapsPE · #7 (**permalink**) April 28th, 2005, 04:14 PM

I have the following set:
VoIP Answer Delay: 4
PSTN Answer Delay: 10
PSTN Ring Thru Delay: 10

When I turn PSTN CID For VoIP CID on, now I get the following from the CLI. Note, this pattern continues until I hang up the cell phone I'm calling in from.

Verbosity is at least 6
-- Remote UNIX connection
-- Executing AbsoluteTimeout("SIP/192.168.1.25-08fc5e00", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/192.168.1.25-08fc5e00", "") in new stack