provided by Star2Billing S.L.

Hi,

I installed as per the procedures. However, the inbound DID takes me to the Asterisk demo voice sound. It is not redirected to a2billing. Any help is appreciated.

I am using Aterisk 1.6, the extension settings as follow:

[a2billing]
;CallingCard 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

Thanks

vi /var/www/html/admin/modules/core/etc/extensions.conf

for incoming dids

[from-sip-external]
include => a2billing-did

and in "extensions_a2billing.conf" make shuwer u have

vi /etc/asterisk/extensions_a2billing.conf
-------------------------------------------------
[a2billing]
exten => _.,1,Answer(1)
exten => _.,n,DeadAgi(a2billing.php|1)
exten => _.,n,Hangup

[a2billing-did]
exten => _.,1,DeadAgi(a2billing.php|1|did)
exten => _.,2,Hangup

Thanks for your reply.

I assume that there is not "extensions_a2billing.conf" in the newer version 1.4 a2billing. Can you please clarify this.

Thanks
joe

Hi

Where you put the contexts is entirely optional in Asterisk and A2Billing. We usually put them in extensions_a2billing.conf and #include this file from extensions.conf just for ease of management.

Joe

Hi,

Thanks for your reply. I used the context in the sip.conf as 'context=a2billing' and used the [a2billing] context in the extensions.conf.
The number did not passthrough a2billing. it stops at the asterix welcome message "congratulations! you have sucessfully installed Aster..........bla..." with the following is the asterisk log... the message

The message starts at the end of log "a=sendrecv" ..
Any help is appreciated.. Thanks.

------------------------------------------
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 1870025487 1870025487 IN IP4 192.168.1.22
s=Asterisk PBX 1.6.0.17
c=IN IP4 192.168.1.22
t=0 0
m=audio 13926 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------

for your DID provider in sip configuration you should put context=a2billing-did

Hi,

I appreciate your response. But, it does not make any sense to me..because I think the context name is immeterial. May be Iam wrong? is 'a2billing' is a reserve word?

please see below my config

sip.conf (please note the provider name and ip are examples)

[test]
context=a2billing-did
allow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=1.1.1.1
insecure=very
type=peer

externsions.conf

[a2billing-did]
;CallingCard application
exten => _X.,1,Answer
exten => _X.,n,Wait,2
exten => _X.,n,DeadAGI,a2billing.php
exten => _X.,n,Wait,2
exten => _X.,n,Hangup


-------------------------------
am I doing anything wrong here. I still not able to extablish connection between Asterisx and a2billing.

my 'asteriskinfo' under the maintenance menu does not show the correct information regards to the sip connections?

any help here is appreciated..

Thanks in advance
joe

felixjoe,

LOL, context is Clinically Material in asterisk otherwise the "connection will be dead".

what are you doing with sip.conf
No offence, you need to read the post and digest and maybe read asterisk cookbook.

you are sending sip user test to a2billing-did Hmmmmmmm!

Rgds
Eamonn