provided by Star2Billing S.L.

Asterisk 1.8.7.1
A2B: 1.9.4
Two servers using one database in realtime mode.

I have calls coming into a2B from pstn. I get oneway audio. Caller A cannot hear Called Party B. If called party B sends DTMF after the call is answered, we have two audio.

If I don't pass the call through A2B the audio is fine.

This is what my dialpeer looks like. I check for caller ID and decide to bill the call or not. Calls that hits roamer1 are not billed and don't have any issues

exten => _1.,1,GotoIf($["${CALLERID(num):0:7}" = "1473520"]?localsub1:roamer1)
; local subscriber
exten => _1.,n(localsub1),AGI(a2billing.php,5)
exten => _1.,n,Hangup
;roamer
exten => _1.,n(roamer1),Dial(SIP/9198888${EXTEN}@TEL21461,60)
exten => _1.,n,Hangup()

Hi

The only difference between the two are the dial command parameters. Play with the dial_command_param in A2Billing's agi-conf.

Joe

Hi Joe,

I tried all the following with the same results:

(DIAL) Options: (SIP/ARB41/3071115616743800,60,HL(9893000:61000:30000))
(DIAL) Options: (SIP/ARB41/3071115616743800,60,HLr(9893000:61000:30000))
(DIAL) Options: (SIP/ARB41/3071115616743838,60,HRL(9732000:61000:30000))
(DIAL) Options: (SIP/ARB41/3071115616743838,60,(9409000:61000:30000))


I play the balance in the account before the call is placed. The caller hears the balance. I tell
a2b to answer the call and to play audio.

answer_call Yes
play_audio Yes

Dave

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

Try setting the H or h parameter in the dial command...