elastix - asterisk hangup call when hold -


if dial , put them on hold, asterisk hang after few minutes. i'm thinking there setting somewhere i'm not finding. ideas? think change 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' 

very likly use sip. have paramater in sip.conf

rtpholdtimeout=300             ; terminate call if 300 seconds of no rtp or rtcp activity                                 ; on audio channel                                 ; when we're on hold (must > rtptimeout) 

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