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 Post subject: A2billing Dial options g , d , h, H hangs up channel
PostPosted: Tue Apr 28, 2015 5:44 pm 
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Joined: Sat Oct 01, 2011 4:41 am
Posts: 50
A2billing Dial options g , d , h, H hangs up channel instead going to dialplan..

i am trying to give a caller the option to break out a dialing channel and go back to the dialplan allowing them to leave a voice-mail.

all my attempts has disconnected the channel before leaving agi...
Is there a way for me to reach the dialplan without disconnecting the channel from agi?


Code:
Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
    -- AGI Script Executing Application: (DIAL) Options: [b](SIP/Voip-Innovations-out/15599340968,13,dHRrL(5400000:61000:30000))[/b]
       > Limit Data for this call:
       > timelimit      = 5400000 ms (5400.000 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 TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/Voip-Innovations-out/15599340968
    -- Timeout on Motif/[email protected], going to 't'
    -- Executing [[email protected]:1] AGI("Motif/[email protected]", "googletts.agi,"Request timed out.",en") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
    -- <Motif/[email protected]> Playing '/tmp/fd587fc46b7ccd4d2f262deb1b2cad78.slin' (escape_digits=) (sample_offset 0) (language 'en')
    -- SIP/Voip-Innovations-out-0000025a is making progress passing it to SIP/Voip-Innovations-IN-00000259
    -- <Motif/[email protected]>AGI Script googletts.agi completed, returning 0
    -- Executing [[email protected]:2] Goto("Motif/[email protected]", "18775743418,start") in new stack
    -- Goto (a2billing-callingcard,18775743418,5)
    -- Executing [[email protected]:5] SIPDtmfMode("Motif/[email protected]", "RFC2833") in new stack
[2015-04-28 13:43:16] WARNING[5689][C-00000baf]: chan_sip.c:33042 sip_dtmfmode: Call this application only on SIP incoming calls
[2015-04-28 13:43:16] WARNING[5689][C-00000baf]: chan_sip.c:33042 sip_dtmfmode: Call this application only on SIP incoming calls
    -- Executing [[email protected]:6] AGI("Motif/[email protected]", "googletts.agi,"Hi welcome back \,  If you are new \,  or want a free trial please press 8 !  \, Press 1 to make a call \, Press 2 to check your Voicemail \,  Press 3 to enter our Party Line \,  Press 4 to get information about your account \, 5 for Free 411 Calls \, Press 6 Frequent Asked Questions and Information about additional services \, Press 7 if you paid already and  have a voucher to add to your Calling Card \,   Press 9 To contact Support \,  ",en,any") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
    -- <Motif/[email protected]> Playing '/tmp/f372e994d0a0b9111aa1aec47073ad47.slin' (escape_digits=0123456789#*) (sample_offset 0) (language 'en')
   [b] -- User hit 9 to disconnect call.[/b]
    -- <SIP/Voip-Innovations-IN-00000259>AGI Script a2billing.php completed, returning 4
  == Spawn extension (a2billing-did-pass, 9, 0) exited non-zero on 'SIP/Voip-Innovations-IN-00000259'
 


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