Using polarity reversal or not!?

Hello all of you,
i have freePBX 2.7.0, asterisk 1.6.2.13.
i have a problem with my analoge FXO gatway, when there is a call it go very good from the box till it come to the gateway then to callee but the bill calculation (A2Billing) is not correct it start from the ring time to calculate, i found that polarity reversal can solve my problem, so i changed the Gateway with polarity reversal and also configured the chan_dahdi.conf and added :
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
but now there is no calculation at all, and after 55sec i become the massege from Astrisk, that no body answer, altho the calle is talking, debug the call give:

Jul 20 21:24:07 WARNING [2796] chan_sip.c: Re-invite to non-existing call leg on other UA. SIP dialog ‘[email protected]’. Giving up.
Jul 20 21:24:07 VERBOSE [14155] app_dial.c:-- SIP/sudan_connection-00000169 is circuit-busy
Jul 20 21:24:07 VERBOSE [14155] app_dial.c:== Everyone is busy/congested at this time (1:0/1/0)
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:20] NoOp(“SIP/306-00000168”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 18”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:21] Goto(“SIP/306-00000168”, “s-CONGESTION,1”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Goto (macro-dialout-trunk,s-CONGESTION,1)
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:1] Set(“SIP/306-00000168”, “RC=18”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:2] Goto(“SIP/306-00000168”, “18,1”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Goto (macro-dialout-trunk,18,1)
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:1] Goto(“SIP/306-00000168”, “s-NOANSWER,1”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Goto (macro-dialout-trunk,s-NOANSWER,1)
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:1] NoOp(“SIP/306-00000168”, “Dial failed due to trunk reporting NOANSWER - giving up”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:2] Progress(“SIP/306-00000168”, “”) in new stack
Jul 20 21:24:07 VERBOSE [14155] pbx.c:-- Executing [[email protected]:3] Playback(“SIP/306-00000168”, “number-not-answering,noanswer”) in new stack
Jul 20 21:24:07 VERBOSE [14155] file.c:-- <SIP/306-00000168> Playing ‘number-not-answering.gsm’ (language ‘en’)
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:4] Congestion(“SIP/306-00000168”, “20”) in new stack
Jul 20 21:24:09 WARNING [14155] channel.c:Prodding channel ‘SIP/306-00000168’ failed
Jul 20 21:24:09 VERBOSE [14155] app_macro.c:== Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘SIP/306-00000168’ in macro 'dialout-trunk’
Jul 20 21:24:09 VERBOSE [14155] pbx.c:== Spawn extension (from-internal, 00249912290500, 4) exited non-zero on 'SIP/306-00000168’
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:1] Macro(“SIP/306-00000168”, “hangupcall”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:1] GotoIf(“SIP/306-00000168”, “1?noautomon”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Goto (macro-hangupcall,s,3)
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:3] NoOp(“SIP/306-00000168”, “TOUCH_MONITOR_OUTPUT=”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:4] GotoIf(“SIP/306-00000168”, “1?skiprg”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Goto (macro-hangupcall,s,7)
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:7] GotoIf(“SIP/306-00000168”, “1?skipblkvm”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Goto (macro-hangupcall,s,10)
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:10] GotoIf(“SIP/306-00000168”, “1?theend”) in new stack
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Goto (macro-hangupcall,s,12)
Jul 20 21:24:09 VERBOSE [14155] pbx.c:-- Executing [[email protected]:12] Hangup(“SIP/306-00000168”, “”) in new stack
Jul 20 21:24:09 VERBOSE [14155] app_macro.c:== Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/306-00000168’ in macro 'hangupcall’
Jul 20 21:24:09 VERBOSE [14155] pbx.c:== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-00000168’
Jul 20 21:25:01 VERBOSE [2610] asterisk.c:-- Remote UNIX connection

i hope i can finde some help here,

thanks in advance,

This is an a2billing question I would suggest you ask in their forum.

it is not an A2Billing, i am using A2Billing to generate calling card, the CDR in the PBX give me the same wrong time it start to calculate as soon as the call path to the Gateway.

The CDR is generated by Asterisk not by FreePBX.

I think it is better to try to give me some tip, as to send me every where else!!! even if it is Asterisk Prob. or A2Billing Problem or even Dahdi one, try to give some solutions, it is better -:slight_smile:

We are not here to support other packages, I have enough to do just keeping up with this project. I have no idea what your problem is or I would have said something.

Why would you think because you are running FreePBX that we will support every software package on your computer.

My dear friend it wasnt meannig as you understand it. I am sure you dont have to help me because i only have free pbx on my computer. All what i wanted. If any bood in this fourm have an idea what it could be then i will be greatfull. Once again sorry if it come wrongly by your side