jasch Posted September 19, 2008 Report Share Posted September 19, 2008 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. Quote Link to comment Share on other sites More sharing options...
Pradeep Posted September 19, 2008 Report Share Posted September 19, 2008 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 ? Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 Have you already checked this page http://wiki.pbxnsip.com/index.php/Installi...N_gateway_setup ? Yes I did. If you notice, there's no explanation of the settings at all. Just the PORTs and GAINS. No explanation atm all for the bottom part of the configuration page (detenting tones, polarity, cadence, etc). Thanks for replying. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2008 Report Share Posted September 19, 2008 Did you turn the busy tone detection on? In reg_ip.htm, setting is called "Detect Busy Tone". Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 Did you turn the busy tone detection on? In reg_ip.htm, setting is called "Detect Busy Tone". Yes, I already did. I have to restart the unit, everytime I receive a call so it get's closed. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2008 Report Share Posted September 19, 2008 Yes, I already did. I have to restart the unit, everytime I receive a call so it get's closed. ? 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 ? 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. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2008 Report Share Posted September 19, 2008 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... Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 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) Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2008 Report Share Posted September 19, 2008 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 19, 2008 Author Report Share Posted September 19, 2008 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 23, 2008 Author Report Share Posted September 23, 2008 So, no solution for my problem? I still cannot redirect any incoming calls. Quote Link to comment Share on other sites More sharing options...
kshan22 Posted September 30, 2008 Report Share Posted September 30, 2008 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 30, 2008 Report Share Posted September 30, 2008 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". Quote Link to comment Share on other sites More sharing options...
jasch Posted September 30, 2008 Author Report Share Posted September 30, 2008 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. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 30, 2008 Report Share Posted September 30, 2008 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 30, 2008 Author Report Share Posted September 30, 2008 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. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 30, 2008 Report Share Posted September 30, 2008 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. Quote Link to comment Share on other sites More sharing options...
jasch Posted September 30, 2008 Author Report Share Posted September 30, 2008 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. Quote Link to comment Share on other sites More sharing options...
kshan22 Posted October 2, 2008 Report Share Posted October 2, 2008 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 Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted October 3, 2008 Report Share Posted October 3, 2008 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 Quote Link to comment Share on other sites More sharing options...
jasch Posted October 3, 2008 Author Report Share Posted October 3, 2008 I just set the Type 1 and Type 2 Busy Tone Cadence (ms) to 375 on all 4 fields. Since I made that change, I haven't had any calls stuck at all. Quote Link to comment Share on other sites More sharing options...
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