Hello
i was install latest elastix 2.30 from source forge by ISO ,
i saw all training about a2bill from sysadminman.net/documentation.html but
when i add custemer -> voip setting and add setting
User name and Secret doesnt work and reg to elastix
elastix log ...
[Jul 20 15:50:23] NOTICE[3151] chan_sip.c: Sending fake auth rejection for device "7000"<sip:7000@0.10.10.8;transport=UDP>;tag=fd3e5354
[Jul 20 15:50:23] NOTICE[3151] chan_sip.c: Sending fake auth rejection for device "7000"<sip:7000@0.10.10.8;transport=UDP>;tag=fd3e5354
[Jul 20 15:50:23] NOTICE[3151] chan_sip.c: Sending fake auth rejection for device "7000"<sip:7000@10.10.10.8;transport=UDP>;tag=fd3e5354

i take pic from a2bill


please help me

Are you trying to say that when you register your SIP phone with the A2billing user name and secret that it won't register? If so, you need to generate SIP configs manually... Elastix is running an older version of FreePBX and real time A2billing setting does not work. Disable real time setting on your global config and then click the generate SIP config and reload asterisk button in the SIP customer page in A2billing.. That will generate the A2billing customers in Elastix... I had the same problem...

ALso,

Once you get that working you can make it even better.. You can create extensions in Elastix like your normally would and add the accountcode and a2billing context to the Elastix extension. Then your customers/users can register with extensions directly to FreePBX. This gives them a much better experience for they can call each other by 3 digit extensions and not deal with long card number.. But.. you need to follow my first instructions. That will get your SIP phones to register via the A2billing user name and secret.

HEllo
thanks dear for ur quick replay that problem is solved ,
now i have new problem !
at time useres can register with user and secret match in a2b and not problem yet,
i settup trunk - cart plan- cart rate - and rate ! in a2b
but usres cannot dial and make call ! i set manuel exten to extentions_a2billing.conf
this is correct or no ?
exten => _X.,1,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup

after dial call answer and say dial prepaid-enter-pin-number , what need to change ?

Your context and AGI config need to be change. Below is my context for SIP users. I also added a voicemail config to allow my SIP users to dial *97 to check their voicemails... I get a fast busy if I don't have these in my context. Now, You need your AGI config to have:

Asterisk_version 1_6
use_dnid_ Yes
cid_sanitie CID
cid_enable No



exten => _X.,1,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup
exten => *97,1,Answer
exten => *97,n,Wait(1)
exten => *97,n,VoiceMailMain(${CALLERID(num)}@default)
exten => *97,n,Hangup

Hi my friend
i check this itmes all items matech my config except cid_sanitie CID i havent items in Global list ,
i just need to remove give dial prepaid-enter-pin-number ! please help me

hi my friend
really thanks
last issue is done , so standard item for dialcommand_param is ,60,HL(%timeout%:61000:30000) ?

This seems to be a billing setting pertaining to calling a DID. I've always seen this not changed... I also leave it alone.

Hi again
Ok ,..
and what to need change for support T.38 and other fax protocols?
and
some times provider got 'prepaid-dest-unreachable or Busy how can automatic make again call ?

I haven't done too much with fax calls on Elastix.. The built in fax server is more for inhouse use and not for reselling to customers... AvantFax is something you probably need to look into. I use the Elastix fax server (Hylafax) and it seems to be okay over IP network.. But.. I'm on fiber internet in a data center.. Many IP networks don't do well on IAX virtual faxes... You really need a fax board to do it properly. Avantfax seems like it possible to send to a2billing and support multi tenant use.. But.. I don't know much about it yet and that would be a post for the Hylafax section of the forums for that is the fax server Elastix uses.