Hangup

[1sandos]

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?

[david55]

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.

[1sandos]

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

----

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?


Again total newbie to Asterisk thanks for your help.

[david55]

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