11.2. Resolving hangup detection problems with fxo cards
When installing zaptel pstn cards, such as the x100p or a digium tdm400p card with fxo module(s), very often problems occur with hangups of the zaptel end not being detected by asterisk.
For example:
You are using zaptel to dial in into your asterisk server, noone is there, you hit voicemail, you leave a message and hangup the phone.
When you listen to the recordings of the answering machine, you notice a very large recording, asterisk kept the line open for hours and hours.
A common approach is to go have a look at voicemail and see what is going on, force it to hangup after some seconds of silence etc etc.
This is not a good approach as in most cases the problem is not only with voicemail but with any call with at least 1 zap leg. (although it doesnt show up as often on a ZAP to SIP call as normally at the end of a conversation both ends will hangup, and the sip bye will go through, ending both call legs).
Possible Solutions: ordered by recommendation - higher is better
1) Try reverse polarity (cvs-head only, dates after 17nov '04 - bug here )
If your asterisk version is recent enough (and you did a make samples) you should see the following in zapata.conf:
; In some countries, a polarity reversal is used to signal the disconnect
; of a phone line. If the hanguponpolarityswitch option is selected, the
; call will be considered "hung up" on a polarity reversal
;
;hanguponpolarityswitch
;
If you now uncomment this hanguponpolarityswitch line, and your pstn line uses polarity reversal, asterisk should detect it and hangup the line.
This solution is by far the best, as it will not get you false hangups and will hangup in 100% of the calls.
If you want to see if this might work, you might want to try to load the wctdm kernel module with the debug=1 parameter. (insmod wctdm debug=1) and look for reverse polarity messages on the console).
You might want to try this option if you enabled the fxs Kewlstart signalling in zapata.conf
2) Go fix your location in zapata.conf
By default, asterisk listens for a busy tone or a a dial tone on the line to detect if the line got hung up.
These tones are different in different countries, duration and tones might differ a little between carriers.
The definitions of these indications can be found in /etc/asterisk/indications.conf
Try to find your country in the list, if its there, go look at /etc/asterisk/zapata.conf
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies
;
;busydetect=yes
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate. Also, it is ONLY configured for
; standard U.S. tones
;
;busycount=4
;
;callprogress=no
Lets explain what those 3 options mean:
busydetect:
With this option enabled, asterisk will try to 'listen' for busy signals on the line, if your carrier sends you a busy tone on a hangup, this should detect it. (Note that this is not exact science, asterisk might think your cat screaming is a busy tone and hangup. In case of loopstart in zapata.conf, you will certainly need this.
To avoid this, they invented the next option,
Busycount:
Busycount defines the amount of busy tones asterisk needs to recognize before actually considering the line to really really give a busy tone, to avoid false hangups on non hangup sounds.
The standard setting is 3, but you could change that to 6 or 10 if you would want to.
A side effect is, that it will take longer to detect a hangup and in case of a real hangup, the line will stay occupied for a little longer than with a lower value of busycount.
callprogress:
This option is similar to the busydetect option, but will also listen for a ringing tone, congestion tone and will try to detect if the line got answered.
It might work for you, but its highly unreliable and will probably give many false hangups.
Please note that none of these 3 options should be activated on digital zaptel channels, (BRI, PRI) as they will not help with anything, they will just cause false hangups).
When you get the reverse polarity to work, dont use these options either.
The correct choice for the signalling= line depends on your carrier.
When you call them with this question they probably wont have a clue what you are talking about so trying several might be a good idea.
Kewl start is the best option there is, but it might not work in your situation, try to call your telco and ask for "disconnect supervision"
In this case, the telco will cut the power on the 2 telephone lines for a small period of time or will reverse the polarity on the lines.
(You could check this by calling to that pstn line, and connecting a phone with a lighted keypad and without external power supply there).
If the lights go out when the other end disconnects, the telco has disconnect supervision.
Loop start does not send a hangup signal, you will need the busydetect parameter.
I dont know if you need extra options with Ground start signalling, i never tried it myself. Let us know if you know.
4) Sell your x100p or clone on ebay or feed it to the dogs, and buy a real TDM card, like the tdm400p.
The x100p's are not very good with the hangups or with callerids.
The CP-1 and CP-2 are call progress decoders that allow any automatic telephone coupling device to hang up properly on PBX or other telephone lines that don't provide the customary end of call d.c. signaling, commonly known as CPC or COD. Many PBXs and a few telco COs do not provide the end-of-call drop in line current that makes telephone couplers hang up. So, the CP-1 and CP-2 listen for dial tone and/or busy (reorder) signals and force the attached unit to hang-up if they are detected.
I never tried it myself, but it might work, and its a hardware solution instead of detecting the busytones in software. (as with the busydetect option).
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Obaid Khan (obaidkhan1313 at gmail dot com) 23 November 2015 08:47:15 Can anyone help me out the same hang up problem with Alcatel Lucent PABX.
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vishu (vishugaddi at gmail dot com) 30 April 2009 09:28:15 Hi All,
I am facing a problem with my asterisk server..calls hang up automatically (as asterisk restart automatically)..
i Couldn\'t find any error logs showing the reason for restart..
Anybody having idea, what is possible causes.
I M using asterisk 1.4.21 with digium cards
Thanks in advance
Mohammed Hasan (rupams2002 at yahoo dot com) 04 September 2008 22:22:00 Hi,
I am using TDM400 card and can call from SIP to SIP phone, pstn to SIP, transfer the call everything working fine in the local network with single router. If I setup network with 2 router still I can call from one network to another. But the problem is SIP does not send HANGUP signal to asterisk. So, the line shows always busy for FXO. What I have to do ?
ma2d_te (ma2d_te at yahoo dot com) 12 December 2007 16:44:47 I've problem when I want to active zaptel. Please see my problem explanation:
afauzi@laserjet5210i:~$ ztcfg -vv
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
somebody can help me to solve my problem??
*) for additional information, I use Ubuntu server 7.10. And I've detected my Wildcard TDM400p.
afauzi@laserjet5210i:~$ lspci | grep ISDN
03:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
Rafael Lobo (rafaelobo at gmail dot com) 05 December 2007 15:45:19 I think I found the way to fixed, at least it worked for me. If you have echotraining on you channel configuration on zapata.con just disable and the channel will hangup fine after that. I hope that it will work for you.
ciscoine (ciscoine at gmail dot com) 24 November 2007 13:28:13 Hangup problems still exist in Turkey, even "hanguponpolarityswitch" does not resolve the problem.
If someone unplugs POTS cable it hangups, but there is no hangup way to disconnect called number using voip software or ipphones.
Turkish indication support does not make asterisk to work clearly with X100P. Maybe a TDM400P card will work.
Elton (eltontech at telkomsa dot net) 09 September 2007 21:30:53 I Hope someone can help me. I'm from South Africa. I also had the hang-up detection problem, but the busydetect=yes solve the problem. A thousand thanks. I have another problem. I have more than one fxo ports. If there is no line in the first port in numerical order then I cant make any outbound calls. Is there any config option that can detect that there is no line in whatsoever port and then try another line. I dont have any problems if all the ports has a line. I only want to prepare my system, so if the line in the first port is down at service provider side, than I will still be able to make calls thru the other lines thanks. Im using AsteriskNow Beta6.
kevil (jason at toplinks dot hk) 09 May 2007 06:03:06 HI,all
the same problem ! and what to do ?
I use not cards.....
Buks Potgieter (buks at ites dot co dot za) 25 March 2007 12:04:27 Is there anybody willing to insert the 2 tone sets of South Africa into indications.conf? Please don't tell me to just insert it - it sissapears after a SIP reload or Genzaptelconf. It has to be put into functions.inc.php as far as I can tell to be included into the code. Does South Africa not count in any way to the FreePBX guys? :)
Vlad (vlad_ant at mail dot ru) 19 February 2007 13:08:19 Asterisk does not have a proper BUSY signal detection, I mean like DTMF detection. All your troubles from that. Big echo or noise will prevent asterisk from detection of the busy signal. Is it so hard to do things proper way ? :-(
Mark (mark__015 at yahoo dot com) 31 January 2007 22:22:40 My Asterisk using a x100p wouldn't hang up on my phone line. I called the phone company and bugged enough people to finally get a technician on the line that understood "disconnect supervision" they called it clear call disconnect. Try this first and see if it is setup on your line.
crcaicedo (crcaicedo at cantv dot net) 06 December 2006 23:33:10 please anyone know if someone fixed the callprogress code? i really need to detect if line is really answered. Thanks in advance!
Brian Griffin (bgriffin at ridgewayindustrial dot com) 11 October 2006 18:42:23 I have a Toshiba PBX and a TDM2400p in my Trixbox I have when I dial int o the Trixbox from the Toshiba PBX the zap channel rings and then the Trixbox is seeing a hangupcall, here is what is showing up on the CLI display:
-- Starting simple switch on 'Zap/43-1'
-- Starting simple switch on 'Zap/43-1'
-- Executing Macro("Zap/43-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/43-1", "w") in new stack
-- Executing NoCDR("Zap/43-1", "") in new stack
-- Executing Wait("Zap/43-1", "5") in new stack
-- Executing Macro("Zap/43-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/43-1", "w") in new stack
-- Executing NoCDR("Zap/43-1", "") in new stack
-- Executing Wait("Zap/43-1", "5") in new stack
-- Executing Hangup("Zap/43-1", "") in new stack
-- Hungup 'Zap/43-1'
-- Executing Hangup("Zap/43-1", "") in new stack
-- Hungup 'Zap/43-1'
Any ideas on how to fix this...Thanks Brian
John Gray (gray at agora-net dot com) 16 September 2006 17:28:40 Is there an easy to determine the dial tone params? Some way to listen, and say Yeah that's 300+50 (or whatever)?
We have a channel bank that goes to dial tone when it hangs up, we haven't been able to detect it correctly. I'm hoping it just a matter of figuring out the correct params to describe the tones.
rosivelt (rosivelt at gmail dot com) 12 September 2006 21:50:35 I've tried all configurations above but nothing run for me. Can somebody help me?
KG (kgtee at twinson dot com dot sg) 12 September 2006 03:44:06 Thank you! I am in Singapore. I set country=sg in indications.conf, and busydetect=yes in zapata.conf, and don't get any more hangup problem.
iCyberNet (icybernet at gmail dot com) 30 August 2006 20:13:14 How to update the patch here? Thank you. Quote
>>1) Try reverse polarity (cvs-head only, dates after 17nov '04 - bug here )
chukaman (jack dot large at gmail dot com) 24 August 2006 07:30:07 Living in South Africa and this worked perfectly for me. Thanks :)
Ryan (Ryan dot G dot Steele at gmail dot com) 11 August 2006 21:28:31 I'm using a tdm400p. I've tried setting callprogress, busydetect, busycount, and hanguponpolarity switches, but none seem to solve the problem. For any call where an inside line calls out to the Great Beyond, if the phone on the other end hangs up, the Asterisk box will sit there foerever listening to the busy signal. I'd love to hear any input...currently, still looking for anything that works.
Leopoldo Rodriguez (leopoldo at yts dot com dot mx) 31 July 2006 06:45:16 Great for me.
I'm using Trixbox 1.0.03 witj TDM422,
I just adjust my contry in indications.conf to mx (Mexico).
And use busydetect=yes and busycount=4 in my zapata.conf and IT'S ALIVE!!
Thanks
rudra narayan acharya (rudra_ach at yahoo dot com) 18 July 2006 22:18:39 Hi,
I've a TDM400p with 4 fxo module.I'm using it india.When I try to place call out side I get one ring then followed by echo of my voice.I'm not sure what could be the problem.
any help
regards
rudra
Parag Srivastava (parag7732 at yahoo dot com) 03 July 2006 16:30:57 I don't know why the above settings are not working for me....pls help me
Ammar Kassmia (info at quatrain-jo dot com) 22 April 2006 11:05:50 I have a norstar PABX. Is it possible to use Asterisk with it to detect the busy tone?
paolo (noaddress at nothanks dot com) 22 March 2006 15:43:33 Have another Problem:
Calls are being interrupted even if no one hangs up the phone, it sounds to me like a related issue. It is a problem that does happen sometimes. Call ending is detected normally, but as said sometimes the calls are being interrupted and reported as normal hangup
jason (noreply at thanks dot com) 09 March 2006 20:10:23 Just wanted to say thanks for this -- it resolved hangup issues for us with our digium tdm400p. I guess we have a funky signaling issue from our provider since most don't have problems with these cards. Anyway, your suggestion for busydetect=yes did the trick. This will allow us to put our asterisk machine into production. Can't thank you enough,
Jason
Ricky Yu (yu at clinicavictoria dot com) 22 February 2006 03:21:26 Thank you so much for all the info, I'm putting up another AAH box (PII460, 540MBRAM) with two X100P clones. At first I had to place two DIFFERENT X100P clones because with two identical cards the box did not identify them separately. Now I'm having the hangup issues so I'll try these options.
vinit (vinitsid at yahoo dot co dot in) 15 February 2006 14:10:16 I have Motorola 5068 PCI Fax/Voice modem. I want to use it with Asterisk server and make calls to PSTN phones. Is it possible? If so then please help me out.