provided by Star2Billing S.L.

Hi,
I have asterisk and A2B 1.3.4 running with no problems, and I want this scenario:

Since Asterisk has the Conference features, I want to use it.... so the customers would call to a DID redirected to a SIP A2B client, but this SIP client in mention is a trunk in asterisk with an incoming configuration in order to redirect the calls to the conference room. Did I explain myself?

PSTN -> DID -> A2B -> ASTERISK SIP TRUNK -> CONFERENCE ROOM

More details:
In A2B:
- Customer with a DID destination working fine
- a SIP account e.g. 38347837834

In Asterisk
- a SIP trunk with the previous account 38347837834
- the incoming calls on this trunk will go to the conference room.

I think that this scenario would work.... but I have this error:

== Begin MixMonitor Recording SIP/195.20.xxx.xxx-b6917b28
> timelimit = 3600000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called 3102389946
-- Got SIP response 482 "Loop Detected" back from 75.127.xxx.xxx
-- Now forwarding SIP/195.20.xxx.xxx-b6917b28 to 'Local/3102389946@a2billing' (thanks to SIP/3102389946-08232708)
-- Forwarding SIP/195.20.xxx.xxx-b6917b28 to 'Local/3102389946@a2billing' prevented.
== Everyone is busy/congested at this time (1:1/0/0)
-- AGI Script Executing Application: (StopMixMonitor) Options: ((null))
-- Playing 'prepaid-isbusy' (escape_digits=#) (sample_offset 0)
== End MixMonitor Recording SIP/195.20.xxx.xxx-b6917b28



Am I missing something?
Kind regards
Christancho

Hi

Your scenario should read:-

PSTN -> DID -> A2B -> ASTERISK LOCAL TRUNK -> CONFERENCE ROOM

So set up a conference, if you are are using FreePBX, set it up there, and give it extension 8200, and then set a trunk, of type local, with a corresponding rate card, tech type, local, and %diallingnumber%@from-internal

Joe

re-think after a couple of minutes, bring the call as a DID, and set the DID destination as local/8200@from-internal

Joe

Hi Joe,
Thanks for your answer, I followed your instructions and the destination for the DID is local/3535@from-internal (3535 is my conference room) and the redirection seems to be working, but the caller listens the ring tone and the call does not connect. However, the conference room request the pin number in order to enter.

Am I missing something?

kind regards
Christancho

-- AGI Script Executing Application: (Dial) Options: (SIP/trk_provider/181865xxxxxx|60|HRgrL(70170000:61000:30000))
-- Limit Data for this call:
> timelimit = 70170000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called trk_provider/181865xxxxxx
-- Executing [181865xxxxxx@a2billing-did:1] NoOp("SIP/195.20.xxx.xxx-0825cc88", ""1576241098617135" <1576241098617135>") in new stack
-- Executing [181865xxxxx@a2billing-did:2] DeadAGI("SIP/195.20.xxx.xxx-0825cc88", "a2billing.php|1|did") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script Executing Application: (MixMonitor) Options: (1259599521.132.gsm|b)
-- AGI Script Executing Application: (DIAL) Options: (local/3535@from-internal|60|HRgirL(3600000:61000:30000))
-- Limit Data for this call:
> timelimit = 3600000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
== Begin MixMonitor Recording SIP/195.20.xxx.xxx-0825cc88
-- Executing [3535@from-internal:1] Macro("Local/3535@from-internal-4e3c,2", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("Local/3535@from-internal-4e3c,2", "user-callerid: 1576241098617135 1576241098617135") in new stack
-- Executing [s@macro-user-callerid:2] Set("Local/3535@from-internal-4e3c,2", "AMPUSER=1576241098617135") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("Local/3535@from-internal-4e3c,2", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("Local/3535@from-internal-4e3c,2", "TTL: ARG1: ") in new stack
-- Called 3535@from-internal
-- Executing [s@macro-user-callerid:14] GotoIf("Local/3535@from-internal-4e3c,2", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("Local/3535@from-internal-4e3c,2", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("Local/3535@from-internal-4e3c,2", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("Local/3535@from-internal-4e3c,2", "Using CallerID "1576241098617135" <1576241098617135>") in new stack
-- Executing [3535@from-internal:2] Wait("Local/3535@from-internal-4e3c,2", "2") in new stack
-- SIP/trk_provider-0827e050 answered SIP/1421382754-08283320
-- Executing [3535@from-internal:3] DeadAGI("Local/3535@from-internal-4e3c,2", "a2billing.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <Local/3535@from-internal-4e3c,2> Playing 'prepaid-enter-pin-number' (language 'es')
-- Playing 'prepaid-no-card-entered' (escape_digits=#) (sample_offset 0)
-- <Local/3535@from-internal-4e3c,2> Playing 'prepaid-enter-pin-number' (language 'es')
-- AGI Script Executing Application: (StopMixMonitor) Options: ((null))
-- AGI Script a2billing.php completed, returning 0

On this part:
-- <Local/3535@from-internal-4e3c,2> Playing 'prepaid-enter-pin-number' (language 'es')
-- Playing 'prepaid-no-card-entered' (escape_digits=#) (sample_offset 0)
-- <Local/3535@from-internal-4e3c,2> Playing 'prepaid-enter-pin-number' (language 'es')
-- AGI Script Executing Application: (StopMixMonitor) Options: ((null))

the conference is not aswering... is the same A2B requesting a prepaid pin number...

Am I missing something?
kind regards
Christancho