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 Post subject: A2billing doesn't see busy signal 486
PostPosted: Tue May 01, 2018 6:24 pm 
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Joined: Tue May 01, 2018 5:34 pm
Posts: 1
Hi everyone,

I've been using a2billing for quite a while. I have an installation of AsteriskNow and I manually installed a2billing.

I seem to be facing some issues. One of them which is the most crucial for me is the busy signal.

Customers can successfully call every destination. When the user that is being called declines the incoming call, I don't get a busy signal.
The customer may hear a recording of the provider (letting them know that the subscriber is unavailable) but as soon as the recording finishes another attempt is made and the callee is called again until the timeout of the dialplan is used.

The a2billing is the latest installation.

This is my dialplan:

[a2billing-private]
exten => _X.,1,AGI(a2billing.php,1)
exten => _X.,n,Wait(1)
exten => _X.,n,Hangup(17)

Here is the log from a call that I made to show you what I mean:
Code:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7fafdc6b04c0 -- Strict RTP learning after remote address set to: 192.168.1.6:61518
    -- Executing [[email protected]:1] AGI("SIP/999999999-00000b65", "a2billing.php,1") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
    -- AGI Script Executing Application: (MixMonitor) Options: (1525198618.4361.gsm,b)
  == Begin MixMonitor Recording SIP/999999999-00000b65
    -- AGI Script Executing Application: (DIAL) Options: (SIP/Provider/Dialed_Number,60,iL(4431000:61000:30000))
       > Limit Data for this call:
       > timelimit      = 4431000 ms (4431.000 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/Provider/Dialed_Number
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
       > 0x7fafe49c8a10 -- Strict RTP learning after remote address set to: 80.139.231.6:10748
    -- SIP/Provider-00000b66 is making progress passing it to SIP/999999999-00000b65
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
       > 0x7fafdc6b04c0 -- Strict RTP switching source address to 85.73.555.888:61518
       > 0x7fafdc6b04c0 -- Strict RTP learning complete - Locking on source address 85.73.555.888:61518
       > 0x7fafe49c8a10 -- Strict RTP switching to RTP target address 80.139.231.6:10748 as source
       > 0x7fafe49c8a10 -- Strict RTP learning complete - Locking on source address 80.139.231.6:10748
    -- Nobody picked up in 60000 ms
    -- AGI Script Executing Application: (StopMixMonitor) Options: ()
  == MixMonitor close filestream (mixed)
    -- <SIP/999999999-00000b65> Playing 'prepaid-noanswer.gsm' (escape_digits=#) (sample_offset 0) (language 'en')
  == End MixMonitor Recording SIP/999999999-00000b65
    -- <SIP/999999999-00000b65>AGI Script a2billing.php completed, returning 4
  == Spawn extension (a2billing-private, Dialed_Number, 1) exited non-zero on 'SIP/999999999-00000b65'


The destination number and the IPs have been equally altered.
When I call a DID that is on my network or a local sip account the call returns the busy signal successfully.

Any ideas?

Thank you


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