Joe I had to set both answer_call and play_audio to NO inorder for the caller to hear the audio. Below shows call setup for a good call and calls with oneway audio
dialcommand_param |60|rL(%timeout%:61000:30000)
good call == Using SIP RTP CoS mark 5 -- Executing [15616743838@smobile:1] GotoIf("SIP/STARMG1-000001dc", "1?localsub1:roamer1") in new stack -- Goto (smobile,15616743838,2) -- Executing [15616743838@smobile:2] AGI("SIP/STARMG1-000001dc", "a2billing.php,5") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (DIAL) Options: (SIP/9195147315616743838@T21461,60,rL(3169000:61000:30000)) > Limit Data for this call: > timelimit = 3169000 ms (3169.000 s) > play_warning = 61000 ms (61.000 s) > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 ms (30.000 s) > start_sound = > warning_sound = timeleft > end_sound = == Using SIP RTP CoS mark 5 -- Called SIP/9195147315616743838@T21461 -- SIP/T21461-000001dd is making progress passing it to SIP/STARMG1-000001dc -- SIP/T21461-000001dd is making progress passing it to SIP/STARMG1-000001dc -- SIP/T21461-000001dd answered SIP/STARMG1-000001dc -- Locally bridging SIP/STARMG1-000001dc and SIP/T1461-000001dd
oneway audio after caller hears balance -- Executing [15616743838@smobile:1] GotoIf("SIP/STARMG1-000001c2", "1?localsub1:roamer1") in new stack -- Goto (smobile,15616743838,2) -- Executing [15616743838@smobile:2] AGI("SIP/STARMG1-000001c2", "a2billing.php,5") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0) -- <SIP/STARMG1-000001c2> Playing 'digits/6.gsm' (language 'en') -- Playing 'dollars' (escape_digits=#) (sample_offset 0) -- Playing 'vm-and' (escape_digits=#) (sample_offset 0) -- <SIP/STARMG1-000001c2> Playing 'digits/50.gsm' (language 'en') -- <SIP/STARMG1-000001c2> Playing 'digits/4.gsm' (language 'en') -- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0) -- AGI Script Executing Application: (DIAL) Options: (SIP/9195147315616743838@T21461,60,HRrL(5263000:61000:30000)) > Limit Data for this call: > timelimit = 5263000 ms (5263.000 s) > play_warning = 61000 ms (61.000 s) > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 ms (30.000 s) > start_sound = > warning_sound = timeleft > end_sound = == Using SIP RTP CoS mark 5 -- Called SIP/9195147315616743838@T21461 -- SIP/T21461-000001c3 is making progress passing it to SIP/STARMG1-000001c2 -- SIP/T21461-000001c3 is making progress passing it to SIP/STARMG1-000001c2 -- SIP/T21461-000001c3 answered SIP/STARMG1-000001c2
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