Displaying 20 results from an estimated 1000 matches similar to: "Performance Issues Degradation After 6 Calls"
2007 Feb 18
3
chan_sip.c:1968 create_addr: No such host:
I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application.
Added the following
[callingcard]
; CallingCard application
exten => 777,1,Answer
exten => 777,2,Wait,2
exten => 777,3,DeadAGI,a2billing.php
exten => 777,4,Wait,2
exten => 777,5,Hangup
I am using 777 as the calling card
2004 Dec 22
4
how to add burstable rate?
Hi,
I managed to fix 128kbit for an IP address of 192.168.200.3 as below: -
tc qdisc add dev eth3 root handle 1: cbq avpkt 1000 bandwidth 100mbit
tc class add dev eth3 parent 1: classid 1:1 cbq rate 128kbit allot 1500
prio 5 bounded isolated
tc filter add dev eth3 parent 1: protocol ip prio 16 u32 match ip dst
192.168.200.3 flowid 1:1
tc qdisc add dev eth3 parent 1:1 sfq perturb 10
How can I
2003 Jul 03
9
HTB burstable for 2 interface , how ?
Dear folks,
Here goes my bandwidth manager:
INTERNET
|
|eth0 202.14.41.1
BW.Manager
| |
| +----eth1----192.168.1.0/24
|
+------eth2----192.168.2.0/24
Total incoming bandwidth to eth0 is 1024kbps
should be shared to eth1 and eth2, which mean each get 512Kbps and
burstable to 1024Kbps if other host is idle.
My question is how do i apply HTB to these situation ?
As far as i know eth1 and eth2
2006 May 26
3
using a billing system
Hello to all,
Im trying to use DeadAGI to implement billing with Asterisk2Billing.
Before the billing, I had something like:
exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider)
Now, with Asterisk2Billing would be something like this?
exten => _2XXXXXXXX,1,Answer
exten => _2XXXXXXXX,2,Wait,2
exten => _2XXXXXXXX,3,DeadAGI,a2billing.php
exten => _2XXXXXXXX,4,Wait,2
exten =>
2005 May 19
1
(no subject)
BJ,
>BJ Weschke <bweschke@gmail.com>
>Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
>SIP termination vs. DS3
>To: Asterisk Users Mailing List - Non-Commercial
>Discussion <asterisk-users@lists.digium.com>
>Message-ID:
<79cf63305051908056c284cc9@mail.gmail.com>
>Content-Type: text/plain; charset=ISO-8859-1
>Did I miss pricing/availability
2010 Sep 16
5
a2billing
Hey there,
I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at
Att,
Flavio Roberto
2011 Apr 05
2
Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2
OK Dears;
Is the exten => _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent?
I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me:
Executing [9615806234 at a2billing:1] Answer("SIP/gwsshihabuddinkw-00000014", "") in new stack
[Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application
2007 Jun 14
4
Que on A2Billing
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able to login using XLite Dialer and was
able to call out to my SIP Trunk also.
Now how can I remove the
2010 Oct 18
1
a2billing
Not sure if a2billing can be shared here, but ill give a shot
If the credit < min_credit the IVR play: sorry you have 0 credit and hangup,
I want it to FW me to the IVR to add voucher, please let me know: here is
log:
[18/10/2010 07:01:12]:[file:a2billing.php -
line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
[18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE
: standard]
2011 May 19
1
SIP 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has it's own context, in which I set the following:
;=====in extensions.conf======
2011 May 19
1
Getting 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has it's own context, in which I set the following:
;=====in extensions.conf======
2010 Jan 01
1
PBX Extension Help
hi all,
I have a little problem. I'm trying to configure a2billing
(asterisk2billing) with asterisk. Everything done successfully but when I
try to call following error occur
"WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application
'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
and it hang ups the call. Can someone please tell me why this error
occuring. My
2006 Nov 25
5
DID Provider
I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels.
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2010 Jun 15
2
a2billing for residential voip usage
Hello List.
I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,
2003 Dec 29
2
bandwidth requirement
Hi Folks,
have a question, on bandwidth.
I want to run an asterisk server SIP to H323, g729. Calls arrive on sip/iax
go to IVR get authenticated and egress through h323. So G729 license is only
used during IVR and then it is pass through.
I am collocating this server. Colo offer a monthly bandwidth quota. Lets say
I want to do 100K minutes per month of VoIP calling at the beginning. What
would
2011 Apr 09
1
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic. This
is the dialplan:
[a2billing]
exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten => _X.,n,AGI(a2billing.php,1)
exten => _X.,n,Hangup()
*exten => h,1,Wait(5)*
*exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})*
As you can see above, I even
2012 Jun 22
2
a2billing
hello,
I just installed a2billing, I did all the config, at least I guess ..
but I still can not integrate sip accounts that I had created (with sip.conf
) in a2billing to apply their billing ..
could someone tell me how to make the junction between asterisk and
a2billing??
I also noticed that the file
additional_a2billing_sip.conf : was always empty ...
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2010 Jun 29
1
Hot to configure trunk in asterisk with a2billing.
Hi All,
I am newbie in this asterisk and a2billing technology . i had successfully
installed asterisk in my server fedora -8 [server behind NAT/STUN]
i after installation i can able to create users and tested the call
features with X-Lite . the was working fine .
after i installed the A2Billing in my same server with follow the steps
from a2billing installation guide.
but u cant access the
2014 Aug 21
1
Billing software: Other than A2Billing because of the problem with the analogue channels
Hello;
I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the
2007 Oct 28
6
MFC requests for 6.3
I would like to request that some useful work on networking be MFCed from
-CURRENT to -STABLE in time for the release of FreeBSD 6.3. In particular,
I'd like to see some of the Netgraph nodes which are new or which have seen
extensive development brought in -- ng_nat and ng_car in particular. Bringing
in the latest version of ng_nat would allow more flexible in-kernel NAT,
while ng_car (which