# Asterisk wont hangup PSTN when finished **URL:** \ **Category:** Asterisk Support\ **Created:** [May 9, 2005, 9:24pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261 "2005-05-09T21:24:06Z")\ **Posts on this page:** 9\ **Page:** 1
**Author:** ![bill78uk](https://avatars.discourse-cdn.com/v4/letter/b/e8c25b/32.png) [@bill78uk](https://community.asterisk.org/u/bill78uk)\ **Post date:** [May 9, 2005, 9:24pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/1 "2005-05-09T21:24:06Z")
Hi Everyone, When I make a call from my SIP phone via Asterisk with a TDM400P card connecting to my Analogue phone line, when I end call, it doesn’t hangup the line. How can I get it to hang up? Thanks in advance, Bill (I am in the UK) ---
**Author:** ![phcollignon](https://avatars.discourse-cdn.com/v4/letter/p/a183cd/32.png) [@phcollignon](https://community.asterisk.org/u/phcollignon)\ **Post date:** [May 9, 2005, 9:31pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/2 "2005-05-09T21:31:12Z")
I have same issue but just with voicemail. Does this occurs also for call forward to a SIP Phone ? Try to use such an extension config exten =\> s,1,Dial(SIP/MySipPhone,2,tr) exten =\> s,2,Voicemail(MyUser) ;exten =\> t,3,Hangup exten =\> s,3,Hangup ---
**Author:** ![bill78uk](https://avatars.discourse-cdn.com/v4/letter/b/e8c25b/32.png) [@bill78uk](https://community.asterisk.org/u/bill78uk)\ **Post date:** [May 9, 2005, 9:37pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/3 "2005-05-09T21:37:11Z")
I’m not sure yet, I am working my way through a bit at a time. Just on the simple stuff at the moment!! I must admit though so far I have been more than impressed with how things have worked. This has been my first major brickwall. 😕 ---
**Author:** ![phcollignon](https://avatars.discourse-cdn.com/v4/letter/p/a183cd/32.png) [@phcollignon](https://community.asterisk.org/u/phcollignon)\ **Post date:** [May 9, 2005, 9:48pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/4 "2005-05-09T21:48:11Z")
I am new to Asterisk also (I installed it one month ago) Maybe we have the same issue because we missed something in our configuration. You did not say me what hardware you have. I would be sure hardware is not the issue before buying a digium one. Also in which country are you ? My phone line is on a Belgium VODSL network. ---
**Author:** ![bill78uk](https://avatars.discourse-cdn.com/v4/letter/b/e8c25b/32.png) [@bill78uk](https://community.asterisk.org/u/bill78uk)\ **Post date:** [May 9, 2005, 9:59pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/5 "2005-05-09T21:59:11Z")
Well, the SIP phone, is a Cisco 7960G and the hardware is a Digium TDM400P with 1 FXS module and 1 FXO module. I’m in the UK on a standard BT phone line. Thanks for your suggestion, I shall try that tomorrow. I’m also going to try a Software SIP phone to see if that makes any difference. I’ll let you know how I get on. I only installed asterisk yesterday, so it’s a steep learning curve !!! ---
**Author:** ![bill78uk](https://avatars.discourse-cdn.com/v4/letter/b/e8c25b/32.png) [@bill78uk](https://community.asterisk.org/u/bill78uk)\ **Post date:** [May 10, 2005, 9:19am UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/6 "2005-05-10T09:19:46Z")
No luck, I already had that line in my config, and I just tried the software SIP phone, and it does the same thing. It doesn’t translate the hangup signal from the phone to the phone line. Hmmm… ---
**Author:** ![phcollignon](https://avatars.discourse-cdn.com/v4/letter/p/a183cd/32.png) [@phcollignon](https://community.asterisk.org/u/phcollignon)\ **Post date:** [May 10, 2005, 10:02am UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/7 "2005-05-10T10:02:56Z")
I think it must be related to Asterisk or its configuration. When someone calls and I answer with my SIP phone, then if the caller hang up all works ok. When someone calls and I do not answer the call is forwarded to the voicemail but in that case Asterisk does not hang up ?! I let you know if I find any solution. Philippe ---
**Author:** ![phcollignon](https://avatars.discourse-cdn.com/v4/letter/p/a183cd/32.png) [@phcollignon](https://community.asterisk.org/u/phcollignon)\ **Post date:** [May 10, 2005, 10:20am UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/8 "2005-05-10T10:20:44Z")
From the FAQ I ll try that this evening and let you know … Q. I’m using X100P and I have random call hangups. What can I do ? A. If you have busydetect=yes in zapata.conf then this is propably causing the problem. You might want to try to add busycount=5 or experiment a little bit with a busycount value. As a last resort you may just change busydetect to no: busydetect=no. Then asterisk won’t try to detect the hangup on incoming calls. Notice that if “Loopstart with Remote Disconnection Supervision” (Kewlstart) is working for you than you don’t need busydetect=yes at all. You can know if your current config is set to Kewlstart if you have a line fxsks=channel\_no in zaptel.conf and signalling=fxs\_ks in zapata.conf. ---
**Author:** ![dferdi](https://avatars.discourse-cdn.com/v4/letter/d/58f4c7/32.png) [@dferdi](https://community.asterisk.org/u/dferdi)\ **Post date:** [May 28, 2005, 8:09pm UTC](https://community.asterisk.org/t/asterisk-wont-hangup-pstn-when-finished/261/9 "2005-05-28T20:09:23Z")
hello, i have also a problem with hangup by calls from sip-account, which i don’t answer. i check the log from asterisk and find a new call to my phone direct after asterisk hangup by some calls/providers. i insert the following line at first line in my context for incomming sip-call’s: **exten =\> h,1,Hangup** with this extension, no new call to my phone is maked from asterisk after the sip-call is ended. i don’t know why, but sometimes asterisk excute the same extensions from call again at hangup. goodby