provided by Star2Billing S.L.

Kind time of day.
Excuse for troubling, the answer please on 1 question:
How to register the provider in billing?
I do here that:
-registered on a site of free-of-charge calls internetcalls.com
-for connection with them it is necessary SIP to transfer a login and the password.
-th has registered a trunk on extension "internetcalls" and has got SIP the friend under the name "internetcalls" thus in a file additional_a2billing_sip.conf has appeared extension "internetcalls"
-at a call a2billing.php trying to call on SIP/${NUMBER}@internetcalls, but the answer such: " Everyone is busy/congrested at time (1:0/0/1) "
I go correct by?.
Prompt please that it is necessary to do.
Thankful in advance.

one of the things you should do is make sure the sip trunk works before trying it in a2billing.

so can you call over the SiP trunk without going thru a2billing?

If not then your problem will be in how your intercall context is setup in additional_a2billing_sip.conf . Some providers require that you specify the "fromuser=username" parameter. Since you are using a2billing to create your SIP friend you have to make sure the file created additional_a2billing_sip.conf is include in /etc/asterisk/sip.conf. if it is not inculded the sip trunk will not be found

if it does work then

1. make sure that additional_a2billing_sip.conf is included in /etc/asterisk/sip.conf
2. in the Trunk definition in a2billing in the IP/host field just reference the peer context name that is in additional-sip.conf , "internetcalls". just put internetcalls . that is the name of the context you are using.

lastly check the log file or the trace on the console to make sure that a2billing is dialing the right number to ensure that you have the prefix stuff right.

the sip context for internetcalls must look like this:

[internetcalls]
type=friend
username=yourusername
secret=
host=
fromuser= yourusername ; YOU MUST HAVE THIS variable in place
dtmfmode=inband
nat=yes

Make sure the additional_a2billing_sip.conf file is included in /etc/asterisk/sip.conf like so

#include additional_a2billing_sip.conf

Strange.
I have done all but......
Phone registered.
In parameters Trunk I specify Provider IP - "internetcalls". I create sip the friend " internet calls ".
Its parameters:

[internetcalls]
type=friend
username =-USR-
secret =-password-
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=yes
disallow=all
allow=g729
host=sip.internetcalls.com
fromuser =-USR-
regseconds=0
cancallforward=yes


a2billing.php the call submits on SIP/$(NUMBER)@internet calls.
sip.conf includes additional_a2billing_sip.conf
But the answer the same - terminatecause: "CHANUNAVAIL"
how to make??

This should be
SIP/${NUMBER}@internetcalls the parenthesis around the number should be a pair of "braces" or wing brackets.

I prefer to just put "internetcalls" in the IP/HOST field of the Trunk definition in a2billing since all the parameters are specified in the context

a. change type=friend to type=peer

b. depending on how well you know asterisk you should be able to set qualify amd monitor if you can reach the "host/client" and see if it successfully communicates with the service provider server [like ping].

qualify=yes

when you do "sip show peers" in the CLI you should see some time in the monitor column if the device is communicating with the host. if you don't see a time number you are not communicating with the host server

c. When you looked in the call trace did your see
"dial xxxxxx@internetcalls"

make sure you are dialing the right number with the right prefix

d. you can also try:
fromdomain=internetcalls.com

Thanks big. Business was at all in it. Asterisk was install by the local machine, without real IP. Top it is possible to remove to not mislead the others. Excuse for it.