Support A2Billing :

provided by Star2Billing S.L.

Support A2Billing :
It is currently Sun Jul 15, 2018 9:00 pm
Voice Broadcast System


All times are UTC




Post new topic Reply to topic  [ 3 posts ] 
Author Message
 Post subject: problem inboud DID a2billing 2.0.1+ Asterisk v11
PostPosted: Sun Dec 16, 2012 4:44 am 
Offline

Joined: Tue Oct 30, 2012 11:56 am
Posts: 23
Hi Experts

i try configure DID of DIDWW, the problem is when a2b recived call and foward to cardnumber, i test in asterisk creating sip account, context and dial plan and work fine. send info of my configuration. i belive the problem is authentication but I do not know how to fix it

configuration Sip.conf of DIDWW

[46.19.209.14]
host=46.19.209.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=a2billing-did
insecure=invite
nat=yes
disallow=all
;allow=ulaw
allow=gsm
qualify=yes


Configuration extensions.conf


[a2billing-did]

exten => _X.,1,AGI(a2billing.php,2,did)
exten => _x,1,Hangup()


Create INBOUND DID

DID CUSTOMER BILLING START DATE DIDGROUP COUNTRY ACT MU RES MR SR ACTION
18299548026 TEST ACCOUNT Fix 2012-12-13 14:34:37 DIDWW Dominican Repub Active 00:00 Yes 1

Create destination

DESTINATION CREATION DATE DID ACCOUNT NUMBER ACTIVE VOIP PRIOR MINUTES USED ACTION
SIP/6367970869 2012-12-13 14:36 18299548026 6367970869 Active Active 1 00:00 Validated


create new agiconf2

use_DNID=yes
use_dnid =yes
number_try=1
setcallerid=yes
cid_sanitize = NO
cid_enable= no
sip_iax_friends=yes


this is the message of CLI Aterisk when dailing DID number 18299548026 from celular phone


Spawn extension (a2billing-did, 18299548026, 1) exited non-zero on 'SIP/46.19.209.14-00003108'
== Using SIP RTP CoS mark 5
-- Executing [[email protected]:1] AGI("SIP/46.19.209.14-0000310a", "a2billing.php,2,did") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
a2billing.php,2,did: file:a2billing.php - line:117 - uniqueid: - IDCONFIG : 2
a2billing.php,2,did: file:a2billing.php - line:118 - uniqueid: - MODE : did
a2billing.php,2,did: file:Class.A2Billing.php - line:712 - uniqueid:1355591878.12556 - get_agi_request_parameter = 8096052800 ; SIP/46.19.209.14-0000310a ; 1355591878.12556 ; ; 18299548026
a2billing.php,2,did: file:a2billing.php - line:625 - uniqueid:1355591878.12556 - [NO ANSWER CALL]
a2billing.php,2,did: file:a2billing.php - line:635 - uniqueid:1355591878.12556 - [DID CALL - [CallerID=8096052800]:[DID=18299548026]
a2billing.php,2,did: file:Class.A2Billing.php - line:1252 - uniqueid:1355591878.12556 - [A2Billing] DID call friend: FOLLOWME=1 (cardnumber:6367970869|destination:SIP/6367970869|tariff:2)
a2billing.php,2,did:
-- AGI Script Executing Application: (DIAL) Options: (SIP/6367970869,60,HRgirL(3600000:61000:30000))
> Limit Data for this call:
> timelimit = 3600000 ms (3600.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
-- Called SIP/6367970869
[Dec 15 13:17:59] NOTICE[28322][C-000019fc]: chan_sip.c:24739 handle_request_invite: Sending fake auth rejection for device <sip:[email protected]>;tag=as4b32b6ef
-- Got SIP response 482 "Loop Detected" back from 127.0.0.1:5060
-- SIP/6367970869-0000310b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php,2,did: file:Class.A2Billing.php - line:1293 - uniqueid:1355591878.12556 - DIAL SIP/6367970869,60,HRgirL(3600000:61000:30000)
a2billing.php,2,did: file:Class.A2Billing.php - line:1305 - uniqueid:1355591878.12556 - [SIP/6367970869 Friend][followme=1]:[ANSWEREDTIME=-DIALSTATUS=CONGESTION]
-- <SIP/46.19.209.14-0000310a>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing-did, 18299548026, 1) exited non-zero on 'SIP/46.19.209.14-0000310a'
-- <SIP/5081880471-00003105>AGI Script a2billing.php completed, returning 4


Top
 Profile  
 
 Post subject: Re: problem inboud DID a2billing 2.0.1+ Asterisk v11
PostPosted: Thu Dec 20, 2012 10:58 pm 
Offline

Joined: Tue Oct 30, 2012 11:56 am
Posts: 23
Hi

if change INBOUND/destination in a2b admin webpages to call celular number it work ej. (SIP/TRUNK/PHONENUMBER), but if try to send call to sip account a2billing ej. (SIP/ACCOUNTNUMBER) not work.

this is CLI asterisk when i call to the number DIDWW 18299548026 and a2b try of send call to sip account a2billing SIP/6367970869, it possible auth account in INBOUND/Destination and how any suggestions please

Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [[email protected]_did:1] AGI("SIP/46.19.209.14-00000070", "a2billing.php,1,did") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script Executing Application: (DIAL) Options: (SIP/6367970869,60,HRiL(3600000:61000:30000))
> Limit Data for this call:
> timelimit = 3600000 ms (3600.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
-- Called SIP/6367970869
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [[email protected]_did:1] AGI("SIP/190.167.212.30-00000072", "a2billing.php,1,did") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/190.167.212.30-00000072>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing_did, 6367970869, 1) exited non-zero on 'SIP/190.167.212.30-00000072'
-- Got SIP response 603 "Declined" back from 0.0.0.0:5060
-- SIP/6367970869-00000071 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- AGI Script Executing Application: (Busy) Options: (1)
-- <SIP/46.19.209.14-00000070>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing_did, 18299548026, 1) exited non-zero on 'SIP/46.19.209.14-00000070'


Top
 Profile  
 
 Post subject: Re: problem inboud DID a2billing 2.0.1+ Asterisk v11
PostPosted: Thu Apr 03, 2014 6:25 pm 
Offline

Joined: Wed Apr 11, 2007 5:49 pm
Posts: 8
Hi there rgarcialvarez, did you found any solution? im having the same issue.


Top
 Profile  
 
Display posts from previous:  Sort by  
Post new topic Reply to topic  [ 3 posts ] 
Hosted Voice Broadcast


All times are UTC


Who is online

Users browsing this forum: No registered users and 1 guest


You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot post attachments in this forum

Search for:
Jump to:  
cron
Powered by phpBB © 2000, 2002, 2005, 2007 phpBB Group