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If I dial someone and put them on hold, asterisk will hang up after few minutes. I'm thinking there is a setting somewhere that I'm not finding. Ideas? i think change this setting

freepbx => tools => asterisk sip setting => MEDIA & RTP Settings

Log excerpt:

[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:1] Macro("SIP/100-000804aa", "user-callerid,SKIPTTL,") in new stack
[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:2] NoOp("SIP/100-000804aa", "Calling Out Route: to-outside") in new stack
[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:3] Set("SIP/100-000804aa", "MOHCLASS=ros-moh") in new stack
[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:4] Set("SIP/100-000804aa", "_NODEST=") in new stack
[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:5] Macro("SIP/100-000804aa", "record-enable,100,OUT,") in new stack
[Mar  9 09:49:16] VERBOSE[19807] pbx.c:     -- Executing [788787636@Local-route:6] Macro("SIP/100-000804aa", "dialout-trunk,1,88787636,") in new stack
[Mar  9 09:50:11] VERBOSE[19807] res_agi.c: <SIP/100-000804aa>AGI Tx >> agi_dnid: 788787636
[Mar  9 09:50:11] VERBOSE[19807] res_agi.c: <SIP/100-000804aa>AGI Tx >> 200 result=1 (788787636)
[Mar  9 09:50:11] VERBOSE[19807] pbx.c:   == Spawn extension (Local-route, 788787636, 6) exited non-zero on 'SIP/100-000804aa'
tripleee
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abbas
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2 Answers2

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Very likly you use sip. It have paramater in sip.conf

rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're on hold (must be > rtptimeout)
arheops
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  • same settings can be at provider end. You can do "sip set debug on" and check which side stop call. – arheops Mar 10 '14 at 15:29
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The parameter you are looking for is, as @arheops said, rtpholdtimeout.

By default it's configured on /etc/asterisk/sip.conf. But you shouldn't set the value on that file, instead it should be set through Elastix web GUI (actually, FreePBX web GUI). PBX -> Unembedded FreePBX -> Tools -> Asterisk SIP settings -> Media & RTP settings, or on /etc/asterisk/sip_general_custom.conf since sip.conf is auto-generated by FreePBX and should not be modified manually.

Doraemoncrew
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