Unable to make successfull call

real_redman · **Joined:** Wed Oct 30, 2013 10:00 pm **Posts:** 6

Hi,

I am pretty new here, so apologies if some questions might seem newbish to you.

First of all, here is my setup:
PIAF(PBX in a Flash) 2.0.6.4.5 - 32bit
A2Billing - 2.0.6
Asterisk 11
FreePBX 2.11.0.11

The target is to make successful calls through PSTN line in my home lab. I have installed PIAF and after the initial setup, I was able to make a successful call with an extension created. Then I installed a2billing using the following method:
http://www.powerpbx.org/content/a2billi ... l-guide-v2

Now I am able to create a user, make a call, the system is telling me the current balance, then asking me to input the number which I am trying to call. When I do, it tells me the available minutes and instead of making the call, it start over by telling me my balance and asking me to put the number which I want to call and everything repeats. I tried with a wrong prefix and then it tells me that the "number is unavailable" or something. So definetely the prefix is set up properly. Just dont know why it doesn't make the call.

I checked the logs from asterisk console and here is what I get:

Quote:

[2013-10-31 00:53:28] WARNING[7636][C-0000000a]: sip/config_parser.c:812 sip_par se_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
[2013-10-31 00:53:28] WARNING[7636][C-0000000a]: chan_sip.c:30629 build_peer: no value given for outbound proxy on line 0 of sip.conf
[2013-10-31 00:53:28] WARNING[7636][C-0000000a]: sip/config_parser.c:812 sip_par se_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
[2013-10-31 00:53:28] WARNING[7636][C-0000000a]: chan_sip.c:30629 build_peer: no value given for outbound proxy on line 0 of sip.conf
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [00359883305309@a2billing:1] Answer("SIP/6822465790-0000000a", "") in new stack
-- Executing [00359883305309@a2billing:2] Wait("SIP/6822465790-0000000a", "1 ") in new stack
-- Executing [00359883305309@a2billing:3] AGI("SIP/6822465790-0000000a", "a2 billing.php,1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'digits/1.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/hundred.gsm' (language 'en')
-- Playing 'credit' (escape_digits=#) (sample_offset 0)
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'digits/1.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/thousand.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/4.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/hundred.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/20.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/7.gsm' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'digits/50.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/1.gsm' (language 'en')
-- Playing 'prepaid-seconds' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'prepaid-enter-dest.gsm' (language 'en' )
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'digits/1.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/thousand.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/4.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/hundred.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/20.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/7.gsm' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/6822465790-0000000a> Playing 'digits/50.gsm' (language 'en')
-- <SIP/6822465790-0000000a> Playing 'digits/1.gsm' (language 'en')
-- Playing 'prepaid-seconds' (escape_digits=#) (sample_offset 0)

Here are the logs from a2billing as well with level4 logging enabled:

Quote:

Any help to what might be causing those problems and how to make the configuration to make a call, would be greatly appreciated.

On a side note, after the initial installation and the configuration of the extension in FreePBX, I was able to make a successful call. After the installation of a2billing, now the behaviour of calls tried by users made by a2billing is the above and when I try to make a call with the initial user which was working before setting up a2billing, does not work anymore. It asks for a PIN.

Awaiting your responses.
Thanks!

linuxciscodaniel · **Joined:** Wed Aug 31, 2011 1:54 pm **Posts:** 33

Hello,

you tell="I try to make a call with the initial user which was working before setting up a2billing, does not work anymore. It asks for a PIN"

you are need to do your call with ask pin ??

regards

bucasia · **Joined:** Mon Mar 02, 2009 8:56 pm **Posts:** 271

Do you have a limit of the number of calls on a trunk. Looks like it's hitting that limit -

Failover trunk cannot be used because maximum number of connections on this trunk is already reached

real_redman · **Joined:** Wed Oct 30, 2013 10:00 pm **Posts:** 6

Thank you both for your answers.

Here is an update, while I was awaiting your responses:
- I noticed the failover message in the logs and did some changes
- I changed the Trunk settings in a2billing. Specifically I had A2B set as Provider IP in a2billing. I decided to change it to a different value. First I tried with the static IP of the server, then with 127.0.0.1. Both cases it now produces different type of error:

Quote:

WARNING,"Rejecting unknown SIP connection from 127.0.0.1""

Received incoming SIP connection from unknown peer to 00359877088360

I then went to FreePBX and changed the "Asterisk SIP Settings" to "Allow Anonymous Inbound SIP Calls". After making another call, it produced the following:

Quote:

Received incoming SIP connection from unknown peer to 00359877088360

And during a call it says "The number it has dialed, is not in service."

Its definitely a progress, but still I am not able to make a successful call.

bucasia · **Joined:** Mon Mar 02, 2009 8:56 pm **Posts:** 271

Not too sure what you're trying to do by changing the trunk IP to 127.0.0.1?

The IP should be the IP of your call provider. This assumes you call provider is set up to allow IP authentication?

I would not recommend setting allow anonymous SIP to YES.

Did you check what MAXUSE was set to on the trunk. -1 means no limit. I would suggest leaving at this whilst testing.

real_redman · **Joined:** Wed Oct 30, 2013 10:00 pm **Posts:** 6

bucasia wrote:

Not too sure what you're trying to do by changing the trunk IP to 127.0.0.1?

The IP should be the IP of your call provider. This assumes you call provider is set up to allow IP authentication?

I would not recommend setting allow anonymous SIP to YES.

Did you check what MAXUSE was set to on the trunk. -1 means no limit. I would suggest leaving at this whilst testing.

I am using my own PSTN line, not an external SIP provider. That's why I don't know what to put as Provider IP and tried with the local address.

The anonymous SIP was set to yes, just for the purpose of this test. It proved that the request is no more rejected. If it is no, the request is rejected as said in my previous response.

I changed the MAXUSE of the trunk to -1 and it no longer produces the error that the system will use the failover trunk.

Can you please advise what IP should use as Provider IP in my case where I want to use my own PSTN line?
Additionally here is the configuration of /etc/asterisk/manager_custom.conf

Quote:

[phpconfig]
secret = php[onfig
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

[phpagi]
secret = phpagi
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

[myasterisk]
secret=mycode
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read=system,call,log,verbose,command,agent,user
write=system,call,log,verbose,command,agent,user

bucasia · **Joined:** Mon Mar 02, 2009 8:56 pm **Posts:** 271

How is your PSTN line configured in Asterisk? Is it plugged in to an FXO card directly in to the PC?

If so that's not a SIP trunk.

real_redman · **Joined:** Wed Oct 30, 2013 10:00 pm **Posts:** 6

bucasia wrote:

How is your PSTN line configured in Asterisk? Is it plugged in to an FXO card directly in to the PC?

If so that's not a SIP trunk.

Yes, it is plugged in an FXO card. I didn't know this is not a SIP.
Can you please let me what would be a proper configuration? Maybe it would be more optimal if I grant you access to the server so that you can quickly see what went wrong. I am ready to pay if you would like me to and since most of it is configured, I suppose it wont take too long.

bucasia · **Joined:** Mon Mar 02, 2009 8:56 pm **Posts:** 271

I'm not currently doing any admin work but if you're looking for paid support you should check with star2billing - http://www.star2billing.com/consultancy/

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

I tend to ignore posts with long pastes of logs, they are difficult to read when you are in a hurry, compared to seeing them in say pastebin.

However, in respect of how your card should be configured in A2Billing, it's simply, techtype = DAHDI, and the IP will either be the group number, e.g. G1 or g1, or the channel number, e.g. 1

Joe

real_redman · **Joined:** Wed Oct 30, 2013 10:00 pm **Posts:** 6

Hi,

Thanks for all the responses till now. I apologies for the long log, but as I said, I am very new to this and wasn't aware how disturbing it is to go through such information. I am now getting more info each day, based on my experience, so with confidence I can conclude the following:

1. You can ignore the above case. It was caused by not properly set maximum connections per trunk. Even after I resolved it, the system still wasn't able to make a proper call.
2. If I am not wrong, after reading a log about this, my assumption is that as you suggested, the whole problem is the trunk which is not set up in A2Billing. Initially, when I installed FreePBX and the Trunk was recognized and worked like a charm with a test call from an extension created in FreePBX, I thought there is no need to create a trunk in A2Billing, as it is already existing in FreePBX. However, now I understand that I have to set up the trunk in A2Billing as well.
The problem here is however, that I am not sure what exactly I should put. Obviously, I am trying to set up a PSTN line as trunk. It was straight forward in FreePBX, but here I am not sure. Thanks for the instructions jroper, but they doesn't mean a lot for me (based on my knowledge currently).

I would really appreciate, if someone have 5 mins to explain to me as a newb how exactly I should set the trunks in order to have working PSTN line in A2Billing. As far as I understand, I should set it up in Providers > Trunks. A few questions regarding that:
- Should I set a provider beforehand?
- In Provider Tech, I don't see DAHDI. Does that mean I should not use it, or it simply means it is not included in the example, but can be used?
- As a Provider IP, let's say that the FXO port which is active in my card is 1, what should I put there? Just 1?

Thanks again for all that participated.

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

Type DAHDI into the Provider Tech field.

Type the channel number or Channel Group into the Provider IP field.

Joe

provided by Star2Billing S.L.

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