provided by Star2Billing S.L.

Hi,

Any other suggestion how to fix this issue. I removed the H parameter from thee dialcommand but asterisk 1.8 still answering the call.

My dial command is this

|60|rL(%timeout%:61000:30000) and I have

Answer Call answer_call No

and

[a2billing]
exten => _X.,1,Agi(a2billing.php,1)
exten => _X.,n,Hangup

Here is from the log file.


[2012-11-25 13:14:47] VERBOSE[6625] res_agi.c: -- AGI Script Executing Application: (DIAL) Options: (SIP/000015745810090@Myvoipservice,60,rL(18720000:61000:30000))
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > Limit Data for this call:
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > timelimit = 18720000 ms (18720.000 s)
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > play_warning = 61000 ms (61.000 s)
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > play_to_caller = yes
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > play_to_callee = no
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > warning_freq = 30000 ms (30.000 s)
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > start_sound =
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > warning_sound = timeleft
[2012-11-25 13:14:47] VERBOSE[6625] features.c: > end_sound =
[2012-11-25 13:14:47] VERBOSE[6625] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-25 13:14:47] VERBOSE[6625] app_dial.c: -- Called SIP/XXXXXXXXXX@MyvoipService
[2012-11-25 13:14:48] VERBOSE[6625] app_dial.c: -- SIP/Myvoipservice-00000007 is making progress passing it to SIP/0349620744-00000005
[2012-11-25 13:14:49] VERBOSE[6625] app_dial.c: -- SIP/Myvoipservice-00000007 answered SIP/0349620744-00000005
[2012-11-25 13:14:49] VERBOSE[6625] rtp_engine.c: -- Locally bridging SIP/0349620744-00000005 and SIP/Myvoipservice-00000007
[2012-11-25 13:15:05] VERBOSE[6625] res_agi.c: -- <SIP/0349620744-00000005>AGI Script a2billing.php completed, returning 4
[2012-11-25 13:15:05] VERBOSE[6625] pbx.c: == Spawn extension (a2billing, XXXXXXXXXX, 1) exited non-zero on 'SIP/0349620744-00000005'



A2biliing 2.01 and asterisk 1.8.11.1


Any other suggestion?

Regards,

Al

Hi punt H in the dial again and option answer_call = no to fix problem

Hi

The dialplan, agi-conf settings, and correct for your system and you are not answering the call. However your carrier appears to be:-



I would have a conversation about this with them, because you are doing nothing wrong.

Yours

Joe

Thx Joe,

That is what though I already contacted them and tey argue that they are not. Customer service at Betamax is not good.

I am arguing with them for almost two month. I also had another server with asterisk 1.4 and have the same problem lately.

Regards,

Al

Hi

They, or one of their upstream carriers is answering the call, try a call with another carrier to confirm it to yourself.

Yours
Joe