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 Post subject: Direct Dialling out on SIP/IAX Trunks
PostPosted: Sat Jan 10, 2009 10:09 pm 
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Joined: Mon Dec 29, 2008 1:32 pm
Posts: 2
Location: Rochdale
Hi, I've returned to A2B after a number of years and am supprised at the comprehensive features available. I completed the installation, without a hitch, following the installation guide.

However I have reached an enpass, after loading the database with suppliers, trunks, ratecards, rates and users I cannot get the system to dial out directly on a sip or iax trunk. I have shown below the relative contexts and .conf files. I am probabling being thick, but if anyone can point me in the right direction.

extensions.conf

[from-internal-custom]
;asterisk2billing application
exten => _X.,1,Answer
exten => _X.,2,Wait,2
exten => _X.,3,DeadAGI,a2billing.php
exten => _X.,4,Wait,2
exten => _X.,5,Hangup

[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|2)
exten => _X.,n,Hangup

a2billing.conf

[agi-conf1]
asterisk_version = 1_4
use_dnid = YES
number_try = 3
force_callplan_id = 1
say_balance_after_auth = NO
say_rateinitial = NO
say_timetocall = NO
auto_setcallerid = YES
force_callerid =
cid_sanitize = BOTH
cid_enable = YES
cid_askpincode_ifnot_callerid = NO
cid_auto_assign_card_to_cid = YES
cid_auto_create_card = YES
cid_auto_create_card_len = 10
cid_auto_create_card_typepaid = PREPAY
sip_iax_friends = NO
sip_iax_pstn_direct_call_prefix = 555
sip_iax_pstn_direct_call = YES
ivr_voucher = YES
dialcommand_param = "|60|HRgL(%timeout%:61000:30000)"
dialcommand_param_sipiax_friend = "|60|HRgiL(3600000:61000:30000)"
switchdialcommand = NO
failover_recursive_limit = 2
agi_force_currency = GBP


Please help.... what am I missing............


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 Post subject: Re: Direct Dialling out on SIP/IAX Trunks
PostPosted: Sat Jan 10, 2009 10:20 pm 
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Joined: Thu Jun 22, 2006 2:19 pm
Posts: 2890
Location: Devon, UK
g0itp1 wrote:
However I have reached an enpass
An impasse? You say you populated A2B's trunks, but with what? The simplest option is to fill the 'Provider IP' field with the name of a trunk defined in sip/iax.conf.

It looks like you're running FreePBX too; if so, any custom contexts' names need to start with "custom-".

If you still can't get any further, please attach a text file (please don't paste it into a message) with the debug output of a failed call. You should first set 'debug = 3' in a2billing.conf, and 'core set verbose 15' at Asterisk's CLI.


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