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I use Asterisk 16.5. Also i use this option. 

rtptimeout = 10

This option work correct when call is not holded. Asterisk terminate call after 11 seconds if no RTP or RTCP activity on the audio channel.

But when sip client holds the call this option is not works correctly. And Asterisk dos not terminate call after 11 seconds if no RTP or RTCP activity on the audio channel.

Ramin Darvishov
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1 Answers1

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Asterisk chan_sip have different option for holded call

rtpholdtimeout=300

Please note, acordinly to SIP RFC endpoint(UA) may not send any rtp data during hold. So it WILL hangup some real calls and will work like max hold time in most cases.

arheops
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