# Hangup

**URL:** <https://community.asterisk.org/t/hangup/39607>\
**Category:** Asterisk Support\
**Created:** [June 21, 2012, 9:31pm UTC](https://community.asterisk.org/t/hangup/39607 "2012-06-21T21:31:46Z")\
**Posts on this page:** 4\
**Page:** 1

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**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\
**Post date:** [June 21, 2012, 9:31pm UTC](https://community.asterisk.org/t/hangup/39607/1 "2012-06-21T21:31:46Z")

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New to Asterisk please don’t hate on me. The problem is that users are dialing into a conference (or Meetme if thats what its called) and their phone is disconnected without sending a hangup notification or request and the channel is staying open. For that matter if I make a call from one phone to another and just disconnect them both from the network without hanging up the channels stay up and I am having to do a soft hangup from the CLI. Do channels ever time out or reset if their is silence?

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**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\
**Post date:** [June 21, 2012, 9:44pm UTC](https://community.asterisk.org/t/hangup/39607/2 "2012-06-21T21:44:19Z")

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FXS analogue phones should disconnect as soon as the loop goes high impedance.

FXO analogue phones will only disconnect if disconnect supervision is provided and correctly configured. One disconnect mechanism is to remove battery temporarily. I would expect an FXO interface configured for that would release if the wire were pulled out.

SIP phones will only disconnect if RTP timeouts are enabled. They may also respond to SIP qualify timeouts (i.e. ceasing to respond to OPTIONS requests), but I’m not sure that that will break a call that is already up.

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**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\
**Post date:** [June 21, 2012, 11:37pm UTC](https://community.asterisk.org/t/hangup/39607/3 "2012-06-21T23:37:28Z")

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Yes these are SIP phones I am having the problem with. I went to the Asterisk CLI from the FreePBX web interface and ran a SIP SHOW SETTINGS and got the following.

## Global Settings:

SIP Port: 5060  
Bindaddress: 0.0.0.0  
Videosupport: Yes  
AutoCreatePeer: No  
Allow unknown access: Yes  
Allow subscriptions: Yes  
Allow overlap dialing: Yes  
Promsic. redir: No  
SIP domain support: No  
Call to non-local dom.: Yes  
URI user is phone no: No  
Our auth realm asterisk  
Realm. auth: No  
Always auth rejects: Yes  
Call limit peers only: Yes  
Direct RTP setup: No  
User Agent: FPBX-2.8.1(1.4.39.2)  
MWI checking interval: 10 secs  
Reg. context: (not set)  
Caller ID: Unknown  
From: Domain:  
Record SIP history: Off  
Call Events: Off  
IP ToS SIP: CS3  
IP ToS RTP audio: EF  
IP ToS RTP video: AF41  
T38 fax pt UDPTL: No  
RFC2833 Compensation: No  
SIP realtime: Disabled

## Global Signalling Settings:

Codecs: 0xe (gsm|ulaw|alaw)  
Codec Order: ulaw:20,alaw:20,gsm:20  
T1 minimum: 100  
No premature media: No  
Relax DTMF: No  
Compact SIP headers: No  
RTP Keepalive: 0 (Disabled)  
RTP Timeout: 0 (Disabled)  
RTP Hold Timeout: 0 (Disabled)  
MWI NOTIFY mime type: application/simple-message-summary  
DNS SRV lookup: Yes  
Pedantic SIP support: No  
Reg. min duration 60 secs  
Reg. max duration: 3600 secs  
Reg. default duration: 120 secs  
Outbound reg. timeout: 20 secs  
Outbound reg. attempts: 0  
Notify ringing state: Yes  
Notify hold state: Yes  
SIP Transfer mode: open  
Max Call Bitrate: 384 kbps  
Auto-Framing: No

## Default Settings:

Context: from-sip-external  
Nat: RFC3581  
DTMF: rfc2833  
Qualify: 0  
Use ClientCode: No  
Progress inband: Never  
Language: (Defaults to English)  
MOH Interpret: default  
MOH Suggest:  
Voice Mail Extension: \*97  
Forward Detected Loops: Yes

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So looks like RTP is disabled so I am looking now to enable it but when I go into the Asterisk SIP settings it seems that it is set?

 ![](http://i.imgur.com/olO8t.png)

Again total newbie to Asterisk thanks for your help.

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**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\
**Post date:** [June 22, 2012, 6:14am UTC](https://community.asterisk.org/t/hangup/39607/4 "2012-06-22T06:14:23Z")

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I would assume that is a problem with the GUi, rather than Asterisk itself.