On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel.However - what does Asterisk do when the network cable is unplugged from one of the SIP devices...?!
Jared already mentioned SIP session timers, which are supported starting with 1.6. Here's my experience. While I am running 1.6, the software stack that is used for agent softphone (PJSIP) does not support the session timers. If the softphone crashes in a call, the call would get stuck exactly as you describe.
I am working around this problem by setting rtp timeouts in sip.conf: [general] rtptimeout=10 rtpholdtimeout=300This means that if RTP flow stops while the agent is in the call, the call will be disconnected in 10 seconds. If the call was put on hold by the agent, it will be disconnected in 300 seconds. Your timeouts may vary.
The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state.
I am able to get away with the small timeout because I set the PJSIP client to always transmit RTP, by turning off voice activity detection feature (VAD). If you want to support that feature, set rtptimeout as high as for how long you allow absolute silence on the line without disconnecting it.
I do not know if these settings are available in 1.2 though. -kkm
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