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 Post subject: sip login
PostPosted: Mon Nov 02, 2009 7:18 pm 
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Joined: Wed Oct 24, 2007 7:02 pm
Posts: 39
Hi, I have a2billing 1.42, created customer with sip field checked, but
where can I see the sip customer created, how to login with a softphone and check calls.
thanks a lot


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 Post subject: missing link sip login
PostPosted: Thu Nov 05, 2009 6:21 pm 
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Joined: Wed Oct 24, 2007 7:02 pm
Posts: 39
Hello, there must be a missing link in my a2billing install for this to work.
I dont find enough literature for this issue.

I installed PIAF following intructions from
Asterisk on Steroids: PBX in a Flash Turns 21
then installed a2billing 1.4 from
viewtopic.php?f=21&t=5010
both installs are ok

after doing some config changes found on the web as
cid_enable=yes
adding this to /etc/asterisk/sip_additional.conf :

[2532857917]
type=friend
secret=0433672645
qualify=yes
port=5060
pickupgroup=
nat=yes
[email protected]
host=dynamic
dtmfmode=rfc2833
dial=SIP/2532857917
context=from-internal
canreinvite=no
callgroup=
callerid=device <2532857917>
accountcode=
call-limit=50

this to /etc/asterisk/extensions_custom.conf :


[from-internal-custom]
exten => 1234,1,Playback(demo-congrats) ; extensions can dial 1234
exten => 1234,2,Hangup()
exten => h,1,Hangup()
include => custom-recordme ; extensions can also dial 5678

;by me

exten => 225,1,Answer
exten => 225,2,Wait,2
exten => 225,3,DeadAGI,a2billing.php
exten => 225,4,Wait,2
exten => 225,5,Hangup

dont know why,

i have a asterisk extension 101 and a a2billing account 532857917.

I can dial from a2billing account to ext 101 making good conection, but don't see the call in the customer call history yet

this is my cli> output

-- Registered SIP '2532857917' at 192.168.1.50 port 9639 expires 1800
-- Saved useragent "PortSIP softphone 2.0" for peer 2532857917
-- Unregistered SIP '2532857917'
-- Executing [[email protected]:1] Macro("SIP/2532857917-0971b4a0", "exten-vm|novm|101") in new stack
-- Executing [[email protected]:1] Macro("SIP/2532857917-0971b4a0", "user-callerid") in new stack
-- Executing [[email protected]:1] Set("SIP/2532857917-0971b4a0", "AMPUSER=2532857917") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/2532857917-0971b4a0", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/2532857917-0971b4a0", "1|Set|REALCALLERIDNUM=2532857917") in new stack
-- Executing [[email protected]:4] Set("SIP/2532857917-0971b4a0", "AMPUSER=2532857917") in new stack
-- Executing [[email protected]:5] Set("SIP/2532857917-0971b4a0", "AMPUSERCIDNAME=2532857917") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/2532857917-0971b4a0", "0?report") in new stack
-- Executing [[email protected]:7] Set("SIP/2532857917-0971b4a0", "AMPUSERCID=2532857917") in new stack
-- Executing [[email protected]:8] Set("SIP/2532857917-0971b4a0", "CALLERID(all)="2532857917" <2532857917>") in new stack
-- Executing [[email protected]:9] Set("SIP/2532857917-0971b4a0", "REALCALLERIDNUM=2532857917") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/2532857917-0971b4a0", "0?continue") in new stack
-- Executing [[email protected]:11] Set("SIP/2532857917-0971b4a0", "__TTL=64") in new stack
-- Executing [[email protected]:12] GotoIf("SIP/2532857917-0971b4a0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [[email protected]:19] NoOp("SIP/2532857917-0971b4a0", "Using CallerID "2532857917" <2532857917>") in new stack
-- Executing [[email protected]:2] Set("SIP/2532857917-0971b4a0", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/2532857917-0971b4a0", "VMBOX=novm") in new stack
-- Executing [[email protected]:4] Set("SIP/2532857917-0971b4a0", "EXTTOCALL=101") in new stack
-- Executing [[email protected]:5] Set("SIP/2532857917-0971b4a0", "CFUEXT=") in new stack
-- Executing [[email protected]:6] Set("SIP/2532857917-0971b4a0", "CFBEXT=") in new stack
-- Executing [[email protected]:7] Set("SIP/2532857917-0971b4a0", "RT=""") in new stack
-- Executing [[email protected]:8] Macro("SIP/2532857917-0971b4a0", "record-enable|101|IN") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/2532857917-0971b4a0", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [[email protected]:4] AGI("SIP/2532857917-0971b4a0", "recordingcheck|20091105-062143|1257420102.28") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20091105-062143|1257420102.28: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [[email protected]:5] MacroExit("SIP/2532857917-0971b4a0", "") in new stack
-- Executing [[email protected]:9] Macro("SIP/2532857917-0971b4a0", "dial||tr|101") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/2532857917-0971b4a0", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [[email protected]:3] AGI("SIP/2532857917-0971b4a0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '2532857917' number is '2532857917'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 101 to extension map
-- dialparties.agi: Extension 101 cf is disabled
-- dialparties.agi: Extension 101 do not disturb is disabled
> dialparties.agi: extnum 101 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/101 to 2532857917
-- dialparties.agi: Filtered ARG3: 101
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [[email protected]:7] Dial("SIP/2532857917-0971b4a0", "SIP/101||tr") in new stack
-- Called 101
-- SIP/101-096ae6b8 is ringing
-- SIP/101-096ae6b8 answered SIP/2532857917-0971b4a0
-- Registered SIP '2532857917' at 192.168.1.50 port 9639 expires 1800
-- Saved useragent "PortSIP softphone 2.0" for peer 2532857917
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2532857917-0971b4a0' in macro 'dial'
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2532857917-0971b4a0' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/2532857917-0971b4a0'
-- Executing [[email protected]:1] Macro("SIP/2532857917-0971b4a0", "hangupcall") in new stack
-- Executing [[email protected]:1] ResetCDR("SIP/2532857917-0971b4a0", "w") in new stack
-- Executing [[email protected]:2] NoCDR("SIP/2532857917-0971b4a0", "") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/2532857917-0971b4a0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [[email protected]:6] GotoIf("SIP/2532857917-0971b4a0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/2532857917-0971b4a0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [[email protected]:11] Hangup("SIP/2532857917-0971b4a0", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2532857917-0971b4a0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2532857917-0971b4a0'
== Parsing '/etc/asterisk/manager.conf': Found

thanks a lot again for your help


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 Post subject: Re: sip login
PostPosted: Sat Nov 07, 2009 3:38 am 
Offline

Joined: Wed Oct 24, 2007 7:02 pm
Posts: 39
Well, I´m here going ahead slowly but surely.
Now I can sip login my a2billing account, configured my trunk, etc with my voip provider
but when I dial i get this clï output :
-----------------------------------------
pbx*CLI>
-- Executing [[email protected]:1] Answer("SIP/3642148671-09350b88", "") in new stack
-- Executing [[email protected]:2] Wait("SIP/3642148671-09350b88", "2") in new stack
-- Executing [[email protected]:3] DeadAGI("SIP/3642148671-09350b88", "a2billing.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script a2billing.php completed, returning 0
-- Executing [[email protected]:4] Wait("SIP/3642148671-09350b88", "2") in new stack
-- Executing [[email protected]:5] Hangup("SIP/3642148671-09350b88", "") in new stack
== Spawn extension (a2billing, 0012123631111, 5) exited non-zero on 'SIP/3642148671-09350b88'

pbx*CLI>
-----------------------------------------
can anyone please tell me what this chinese message means or give me some links where to read ?

thanks a lot again


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 Post subject: Re: sip login
PostPosted: Mon Nov 09, 2009 6:00 pm 
Offline

Joined: Wed Oct 24, 2007 7:02 pm
Posts: 39
Hello,
when I run
# yes "" | php /var/lib/asterisk/agi-bin/a2billing.php 1
I get
PHP Fatal error: Class 'AGI' not found in /var/lib/asterisk/agi-bin/a2billing.php on line 62
what that it means?

thanks a lot


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 Post subject: Re: sip login
PostPosted: Tue Nov 10, 2009 4:58 am 
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Joined: Wed Oct 24, 2007 7:02 pm
Posts: 39
Hello,
my line 62 is:
$agi = new AGI();
thanks


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