provided by Star2Billing S.L.

Hello I was wanderfing to see that my isnattation work properly..

my configuration:
- Centos 5.1 (last release)
asterisk 1.4.x
freepxb last (last release)
a2billing 1.3

all seem working good

i need to know how remove voice message about charge and
how to remove the pound £ to make a phone call
- becouse when i make a first call i must compose two casual internal number before able to make a right phone
call...

when im logged wit telnet i can't see
destination number but just

-- examples of phone call termination -a2billing -begin--------
user id termination number
Event: Dial
Privilege: call,all
Source: SIP/8543489215-0925eff8
Destination: SIP/myvoip-0925c960
CallerID: 916155189345179 <<<==this===
CallerIDName: <unknown>
SrcUniqueID: 1215436766.60
DestUniqueID: 1215436793.61

----- examples of phone call termination -----end----a2bill----

Then if i use instead normal account direct with freepbx i can see destinatin number
as follows
----- examples of phone call termination ----freepbx -- -beg--------
Event: Newexten
Privilege: call,all
Channel: SIP/200-09256f30
Context: macro-dialout-trunk
Extension: s
Priority: 20
Application: Dial
AppData: SIP/myvoip/00390112522445566|300| <<=this=
Uniqueid: 1215437380.64
----- examples of phone call termination ----freepbx -- -end--------

Any suggestion about ?

Tk's Bye

** sorry for my poor english

Read a2billing.conf and I'm sure the answer will be obvious.
You don't need to press # to dial. You can just wait a few seconds instead.

You included a couple of captures from AMI. You'll probably find the Asterisk console (with a high enough verbose level) is a lot more informative. It seems you're complaining that A2B isn't including the dialled number in the dialstring sent to your carrier. Check your Trunks settings in A2B to make sure you've got '%dialingnumber%' spelled correctly.

Hi Stavros!

tks for your information ...
i learnt that that is nont necessary press £.

is working fine regarding the other precius informatio.

never changed ...

---------this is the webtrunk config----beg-----

VOIP-PROVIDER =>"myvoyp..etc"
--
LABEL =>"medianet24trunk"
--
ADD PREFIX =>"00"
--
REMOVE PREFIX =>"is empty"
--
PROVIDER TECH =>"SIP"
--
PROVIDER IP =>"myvoip"
--
ADDITIONAL PARAMETER "your parameter there=>%dialingnumber%"
--
FAILOVER TRUNK =>"Default"
--
---------this is the webtrunk config----end-----

This is the new captured telnet
------------------------

Event: Dial
Privilege: call,all
Source: SIP/8543489215-0925eff8
Destination: SIP/myvoip-09256f30
CallerID: 916155189345179
CallerIDName: <unknown>
SrcUniqueID: 1215445823.122
DestUniqueID: 1215445855.123

Event: Newcallerid
Privilege: call,all
Channel: SIP/myvoip-09256f30
CallerID: 30
CallerIDName: <Unknown>
Uniqueid: 1215445855.123
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

Event: Newstate
Privilege: call,all
Channel: SIP/myvoip-09256f30
State: Up
CallerID: 30
CallerIDName: <unknown>
Uniqueid: 1215445855.123

Event: Link
Privilege: call,all
Channel1: SIP/8543489215-0925eff8
Channel2: SIP/myvoip-09256f30
Uniqueid1: 1215445823.122
Uniqueid2: 1215445855.123
CallerID1: 916155189345179
CallerID2: 30

Event: Unlink
Privilege: call,all
Channel1: SIP/8543489215-0925eff8
Channel2: SIP/myvoip-09256f30
Uniqueid1: 1215445823.122
Uniqueid2: 1215445855.123
CallerID1: 916155189345179
CallerID2: 30

Event: Hangup
Privilege: call,all
Channel: SIP/myvoip-09256f30
Uniqueid: 1215445855.123
Cause: 16
Cause-txt: Normal Clearing

-------

Has you can see for make a call i have to type for exs.number =>"30"

Otherwise I can't make any outside phone call

I believe that i did e little mistake some where ...

Other good suggestion?

Than'ks again 4 your time Bye Paul

I'm not sure what your problem is, or how what you're seeing differs from what you are hoping to see. Please state your problem clearly.
Also giving us AMI captures is very little use. As I suggested previously, try watching the Asterisk CLI console instead.

Added after 7 minutes:

OK, now I've noticed your Subject: line which makes it clear to me that you're trying to eliminate the double dialling and the prompt for the number.
You need to start searching the forum before asking questions that have been answered countless times. For instance that question was dealt with most recently only 12 hours ago.

Thak's for your interest. You give me precius information
I will' let inform you if i will asap

The problem look like this one
I will follow your links tks again

Bye
Paul

Added after 47 minutes:

Hi Stavros.

Thank's so muck i solved the problem

I simply follow carefully the instrunction in the other links ..

The first information gave to me was essential!!!

I set "yes" also the parameter that shuld not working .. before
-------------

then i tune 'use_dnid = YES' in a2billing.conf

after I simply wait and then all was working perfectly

---------------

Thk's have a great day Bye Paul!!