Jump to content

CS410 not detecting PSTN call termination


jasch
 Share

Recommended Posts

I just purchased the CS410, and while verything was a breeze to setup and get working, I am having problems with it properly detecting PSTN calls being terminated.

 

Maybe because of the different tones our country uses, so I would like to request help from anybody setting up the proper values.

 

Country          Tone                   Freq/Hz      Cadence/s
Costa Rica	 Busy tone              450          0.33 on 0.33 off 	
Costa Rica	 Call waiting tone      450          0.15 on 0.15 off 0.15 on 8.0 off 	
Costa Rica	 Congestion tone        450          0.33 on 0.33 off 	
Costa Rica	 Dial tone              450          continuous 	
Costa Rica	 Ringing tone           450          1.2 on 4.9 off 

 

Using 3.0.0.2998 (Linux)

 

Thanks in advance.

Link to comment
Share on other sites

I just purchased the CS410, and while verything was a breeze to setup and get working, I am having problems with it properly detecting PSTN calls being terminated.

 

Maybe because of the different tones our country uses, so I would like to request help from anybody setting up the proper values.

 

Country          Tone                   Freq/Hz      Cadence/s
Costa Rica	 Busy tone              450          0.33 on 0.33 off 	
Costa Rica	 Call waiting tone      450          0.15 on 0.15 off 0.15 on 8.0 off 	
Costa Rica	 Congestion tone        450          0.33 on 0.33 off 	
Costa Rica	 Dial tone              450          continuous 	
Costa Rica	 Ringing tone           450          1.2 on 4.9 off 

 

Using 3.0.0.2998 (Linux)

 

Thanks in advance.

 

Have you already checked this page http://wiki.pbxnsip.com/index.php/Installi...N_gateway_setup ?

Link to comment
Share on other sites

? If you are on an extension call, the call should be cleared when you hang up on the IP phone. Otherwise we have not a busy tone detection problem here.

 

I call the FXO number, my extension rings. I hang-up my phone. Then I try calling, and calling, and the phone number is busy, because the CS410 never closed the line. It obviously it's not detecting the busy tone, or any changes in impedance.

Link to comment
Share on other sites

I call the FXO number, my extension rings. I hang-up my phone. Then I try calling, and calling, and the phone number is busy, because the CS410 never closed the line. It obviously it's not detecting the busy tone, or any changes in impedance.

 

Okay, but then we are talking about a completely different problem than the busy tone detection. Is the call still "up" on the PBX in the web interface? Maybe there is a problem between the SIP phone and the PBX. Did you change the port setup for the FXO gateway? Or the IP address (1.1.1.1 and 1.1.1.2 by default)? Hanging up from the PBX side was never a problem...

Link to comment
Share on other sites

Okay, but then we are talking about a completely different problem than the busy tone detection. Is the call still "up" on the PBX in the web interface? Maybe there is a problem between the SIP phone and the PBX. Did you change the port setup for the FXO gateway? Or the IP address (1.1.1.1 and 1.1.1.2 by default)? Hanging up from the PBX side was never a problem...

 

Hmmm. I still think it's related. When the person that makes the call, hangs-up on the receiving end, the receiving end hears a busy signal, meaning the call was terminated/disconnected.

 

How is this a different problem?

 

As for the SIP phone, the call was never answered. It was sent to Voice mail, also I tried it redirecting it to the Conference server, and after hanging up, the line was left open...

 

The port setup on the FXO gateway has not been changed

 

As for the 1.1.1.1 and 1.1.1.2 I have no idea where to look for them, since the Trunk is configured as 127.0.0.1:5062 (It came like that from the factory)

Link to comment
Share on other sites

Hmmm. I still think it's related. When the person that makes the call, hangs-up on the receiving end, the receiving end hears a busy signal, meaning the call was terminated/disconnected.

 

How is this a different problem?

 

As for the SIP phone, the call was never answered. It was sent to Voice mail, also I tried it redirecting it to the Conference server, and after hanging up, the line was left open...

 

Aha... So if you answer the call and hang up on the SIP phone, it works fine? I would expect that.

 

Otherwise we do have the busy tone detection problem. The PSTN side sings the "please hang up song" (AKA busy tone), and the PBX has the job of detecting it. But in that case you should be able to take the call down by clicking in hte web interface on the delete icon. It does not solve the problem, I know, but it should work and you are saving a reboot cycle.

 

The port setup on the FXO gateway has not been changed

 

As for the 1.1.1.1 and 1.1.1.2 I have no idea where to look for them, since the Trunk is configured as 127.0.0.1:5062 (It came like that from the factory)

 

Okay, don't touch that part.

 

For the sake of testing things out, I would suggest you pick up the phone call on the SIP phone. Then you can first hang up on the PSTN side (hear busy tone), and if the PBX does not detect the busy tone, then you can still hang up on the SIP phone.

 

Maybe the tone sounds very silent, distorted, or maybe the carrier just says "Your cann has been disconnected. Please hang up." In that case it will be really difficult to perform an automatic hangup detection.

Link to comment
Share on other sites

I will try that right away. But, like I tried to say on my first post, maybe the problem is that the dial/busy tones on my Country, are different from the US, so that might be the problem of them not being completely recognized?

 

I used to do this perfectly on my Sipura 3102, I even managed to get the changed detected by checking the change of the voltage when the line was off-hook and on-hook.

Link to comment
Share on other sites

I will try that right away. But, like I tried to say on my first post, maybe the problem is that the dial/busy tones on my Country, are different from the US, so that might be the problem of them not being completely recognized?

 

I used to do this perfectly on my Sipura 3102, I even managed to get the changed detected by checking the change of the voltage when the line was off-hook and on-hook.

 

Calling the extension, if I pick up, and then hang up (either side), the call get's terminated fine.

 

Then I redirect incoming calls to the AutoAttendant. I dialed the PSTN number, waited for the attendant to pick up, and hanged the phone.

 

It's been 5 minutes, and the CALL still shows as 'live' on the status page.

 

Everything is OK, unless I get directed something other than the SIP phone...

 

Currently Active Calls

Start From To State Action

2008/09/19 15:08:05 88302595 [pstn] 22830598 connected

 

And if I call again, I get a busy signal.

Link to comment
Share on other sites

Aha... So if you answer the call and hang up on the SIP phone, it works fine? I would expect that.

 

Otherwise we do have the busy tone detection problem. The PSTN side sings the "please hang up song" (AKA busy tone), and the PBX has the job of detecting it. But in that case you should be able to take the call down by clicking in hte web interface on the delete icon. It does not solve the problem, I know, but it should work and you are saving a reboot cycle.

 

 

 

Okay, don't touch that part.

 

For the sake of testing things out, I would suggest you pick up the phone call on the SIP phone. Then you can first hang up on the PSTN side (hear busy tone), and if the PBX does not detect the busy tone, then you can still hang up on the SIP phone.

 

Maybe the tone sounds very silent, distorted, or maybe the carrier just says "Your cann has been disconnected. Please hang up." In that case it will be really difficult to perform an automatic hangup detection.

 

 

I have exactly the same problem and I have been trying to see if there is a fix for this. My box is running 3.0.0.0.2998 as well. I am located in Singapore. As long as I hang up the call on the incoming side after the auto attendant directed me to the extension and not pick up the destination sip phone. The line will remain open for a long period time and only a power cycle can clear the call. I hope someone have a fix for this.

 

Thanks

 

Kshan22

Link to comment
Share on other sites

I have exactly the same problem and I have been trying to see if there is a fix for this. My box is running 3.0.0.0.2998 as well. I am located in Singapore. As long as I hang up the call on the incoming side after the auto attendant directed me to the extension and not pick up the destination sip phone. The line will remain open for a long period time and only a power cycle can clear the call. I hope someone have a fix for this.

 

Short term workaround for the AA is to define a "Hangup Timeout".

Link to comment
Share on other sites

Short term workaround for the AA is to define a "Hangup Timeout".

 

I am kinda dissapointed I just spent $1200 on a piece of hardware, and I cannot even get proper technical support for the couple of problems I've experienced, except vague answers and no solutions, and if I wanted to open a support ticket, I have to pay $150 on top of that.

Link to comment
Share on other sites

I am kinda dissapointed I just spent $1200 on a piece of hardware, and I cannot even get proper technical support for the couple of problems I've experienced, except vague answers and no solutions, and if I wanted to open a support ticket, I have to pay $150 on top of that.

 

FXO is analog stuff. It is very difficult to troubleshoot this. We have had cases where the CO was reading out "the call has been disconnected. Please hang up." Now how do we support this? Having the auto attendant hang up the line is a reasonable solution to to clear the line.

Link to comment
Share on other sites

FXO is analog stuff. It is very difficult to troubleshoot this. We have had cases where the CO was reading out "the call has been disconnected. Please hang up." Now how do we support this? Having the auto attendant hang up the line is a reasonable solution to to clear the line.

 

I understand. But let's start with the PSTN Gateway TAB, and the lack of even documentation or an explanation of what all the options between "Select Region" and "CPC Duration" mean and what they are used for.

 

There are three different options under Select Region (Default, Australia and South Africa), so Obviously, there are different settings for different countries and dial tones. On my original message, I posted the hertz frecuency and tones my country uses, obviously the person that programmed that feature on pbxnsip, would surely know what settings need to be adjusted, or with what settings I need to run tests.

 

This is one of the sections of pbxnsip that comes with no documentation at all.

 

Another problem I've been experiencing, is that about 10% of the calls I make, I cannot hear and I cannot be heard. I hang up, redial, and everything works fine. I have no idea how to diagnose, since 90% of call work fine, and the problem gets fixed just by redialing, I assume the phones are configured correctly (if they werent, I would not hear anything on 100% of the calls).

 

I am sorry for venting, but it's been a hard start and set-up.

Link to comment
Share on other sites

We are also not happy with the situation with the hang-up detection. We want to avoid to get into this messy area or this country does it this way and the other country that way. This will be an endless, frustrating setup nightmare.

 

We'll probably remove the hangup tone detection from the PSTN gateway and pull it into the PBX (just like DTMF tone detection). Then we can check if the PBX is actually connected to an extension and if that is the case let the user decide when to hang up or not. Otherwise when connected to an auto attendant, ACD or a IVR node, the PBX will aggressively hang up as soon as there is a tone that could be a disconnect tone (or even no input for some time). I believe that will solve a lot of problems.

Link to comment
Share on other sites

We are also not happy with the situation with the hang-up detection. We want to avoid to get into this messy area or this country does it this way and the other country that way. This will be an endless, frustrating setup nightmare.

 

We'll probably remove the hangup tone detection from the PSTN gateway and pull it into the PBX (just like DTMF tone detection). Then we can check if the PBX is actually connected to an extension and if that is the case let the user decide when to hang up or not. Otherwise when connected to an auto attendant, ACD or a IVR node, the PBX will aggressively hang up as soon as there is a tone that could be a disconnect tone (or even no input for some time). I believe that will solve a lot of problems.

 

BTW, I did manage to get it working over the weekend (99% of the time) by adjusting the Type 1 and Type 2 Busy Cadence.

 

Like I said in my previous email. There is no documentation or explanation at all about these settings... I just had to spent 4 hours of my time playing with these values, while someone at pbxnsip probably knows how to 'play' with them accurately.

 

I just have my other pressing problem which drives users nuts.

Link to comment
Share on other sites

BTW, I did manage to get it working over the weekend (99% of the time) by adjusting the Type 1 and Type 2 Busy Cadence.

 

Like I said in my previous email. There is no documentation or explanation at all about these settings... I just had to spent 4 hours of my time playing with these values, while someone at pbxnsip probably knows how to 'play' with them accurately.

 

I just have my other pressing problem which drives users nuts.

 

 

Hi jasch,

 

I am experiencing the same problem as you. Can you let me know what is the value that you set in the Type 1 and Type 2 Busy Cadence. I am also puling my hair out on this one.

 

Thanks

 

Kshan

Link to comment
Share on other sites

Thanks in advance.

 

We have a staff member from the mountains in and around Limon, Costa Rica, He longs to visit his Mother Soon. The CS410 is a good choice when considering the other options. The Local Telecoms in all countries can advise on what the Cadences are, and having that information is a wise option.

http://nemesis.lonestar.org/reference/tele...g/ringring.html

 

Perhaps a Cross reference on setting the CS410 would be useful, Have you shared the settings that acheived the 99% success?

 

Cheers

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...