provided by Star2Billing S.L.

Ok Vulcan, I modified the files as patches (attached). I've renamed txt to attach them.
To Class.A2Billing.txt, I applied before star194.txt and after star194b.txt (check if I did something wrong).
Unfortunately, the result does not change...
I tried with this particular situation:
The user B has a VoIP number 0863185XXXX, which also responds to the ported number 0863528XXX.
In FreePBX-Inbound Routes->DID Number 0863185XXXX->Custom Destination->a2billing-did; In A2Billing Inbound DID 0863185XXXX->Destination->sip/38209XXXXX.
If User A (with "In Use at 0"), calls user B (with "In Use at 0"), using the DID 0863185XXXX, the call is internal (Inbound DID 0863185XXXX->Destination->sip/38209XXXXX), and hangs up before answering, the "In Use" of the user A changes to 1, and that of the user B changes to -1
During the call (I refresh the page), "In Use" of the user A goes to 1, while "In Use" of the user B stays at 0.
At the end of the call (no answer), "In Use" of the user A stays 1, while "In Use" of the user B goes to -1
If instead the same user A, calls the same user B, using the DID 0863528XXX, the call goes to the outside and then go back to FreePBX (In FreePBX-Inbound Routes->DID Number 0863185XXXX->Custom Destination->a2billing-did; In A2Billing Inbound DID 0863185XXXX->Destination->sip/38209XXXXX), and hangs up before answering, the "In Use" of the user A and user B remain at 0.
During the call (I refresh the page), "In Use" of the user A goes to 1, and "In Use" of the user B go to 1.
At the end of the call (no answer), "In Use" of the user A goes to 0, and "In Use" of the user B goes to 0.
Of course the same thing happens if I call from an outside telephone regarding the "In Use" of the user B: During the call, goes to 1, and the term goes to 0.
I hope these my instructions, can ... "enlighten"
Very Thanks, especially for the patience...

Corrado.

Just to be sure, did you make the changes in all copies of these files? When you extract the a2billing tar ball, admin, billing and customer have soft links to the lib folder and more. Many people copy files including the lib folder to other locations creating multiple copies.

I would advise people to extract the tar ball and leave it where it was extracted and link to it. Look into this and post your findings.

Hey Vulcan, you're right !!! (as always )
In fact, I had admin, agent, common e customer dir of /usr/src/a2billing, copied to /var/www/a2billing !!
I replaced the original files with those patched, also in / usr/src/a2billing, and now it works perfectly!!!
At this point, however, for not having more problems, I followed your advice:
I renamed the dir admin, agent, common e customer of usr/src/a2billing, I copied what I had in / var/www/a2billing (I have made ​​many changes) to /usr/src/a2billing (correct permissions) , and linked them under /var/www/a2billing.
Now, all works perfetly: If User A (with "In Use at 0"), calls user B (with "In Use at 0"), using the DID 0863185XXXX, the call is internal (Inbound DID 0863185XXXX->Destination->sip/38209XXXXX), and hangs up before answering, the "In Use" of the user A and user B remain at 0.
Great Vulcan!
The only difference with an external call, is that during the internal call, the "In Use" of the user B stays at 0, but I think this is correct.
I would like to take advantage of your experience to ask you one thing: about DID, an Inbound DID can receive calls in Realtime mode?
I have tried, but only receives calls to start, but after a while, no more. I think this is because, in Realtime, the account is authenticated only when it is called, right? Or is there a way to use Inbound DID, also with Realtime activated?
Very Very Thanks, Vulcan!

Corrado.

Yes, inbound DID can be routed to realtime peer if the peer is registered with the asterisk. So then, inbound DID -> A2B -> internal call to registered peer.

I do not know if I explained my situation well, and anyway, I need a more detailed explanation...
You carry the example of the user B of the above examples.
This user has an account with geographic number to another voip provider: 0863185XXXX.
I registered the trunk voip of this account on my FreePBX, then, still in FreePBX, I set to Inbound Route "DID Number" (0863185XXXX)-> Custom Destination-> a2billing-did.
Then, in A2Billing, I created an inbound DID (0863185XXXX), with destination-> sip/38209XXXXX (the card of the user B in A2Billing).
If registered the card 38209XXXXX of user B on a voip adapter, He receives calls made to the geographic number voip 0863185XXXX also in realtime mode, but only just registered, but after a few minutes... no more
If instead imposed "No Realtime", he receives calls made to the geographic number voip, always.
What is wrong? What should I change?

Very Thanks.

Corrado.

in the general section of sip.conf try using these.

rtcachefriends = yes
rtupdate = yes

Set the asterisk log to full, restart asterisk and look carefully through the log for errors relating to realtime.

For not edit sip.conf (as recommended), I've edit sip_custom.conf. Is ok?



The only ones found relating to real time, are:
res_config_ldap.c: Cannot load LDAP RealTime driver.
res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1:

...but I use mysql. I do not know what is LDAP RealTime driver.

The full.log:

[2012-05-13 17:49:20] ERROR[19547] pbx.c: Did not remove this priority label (60/vmxopts) from the peer_label_table of context macro-vm, extension vmx!
[2012-05-13 17:49:20] ERROR[19547] pbx.c: Did not remove this priority label (4/lookup) from the peer_label_table of context app-speeddial-set, extension s!
[2012-05-13 17:49:20] ERROR[19547] res_curl.c: res_config_curl.so (dependent module) is still loaded. Cannot unload res_curl.so
[2012-05-13 17:49:22] ERROR[19627] res_config_ldap.c: No directory URL or host found.
[2012-05-13 17:49:22] ERROR[19627] res_config_ldap.c: Cannot load LDAP RealTime driver.
[2012-05-13 17:49:22] ERROR[19627] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1:
[2012-05-13 17:49:23] ERROR[19627] chan_vpb.cc: No Voicetronix cards detected
[2012-05-13 17:49:23] ERROR[19627] ais/clm.c: Could not initialize cluster membership service: Try Again

Very Thanks.

Corrado.

No.

Since you are using Freepbx, go to Tools->Asterisk SIP Settings->Other SIP Settings and add them.

What is the displayed result from asterisk CLI for "sip show peers" when client is registered?

Right. In fact, in FreePBX, it gave me error ...
Now is ok.



This:
03863XXXXX/03863XXXXX 79.39.XX.XXX D N 5060 Unmonitored

"Unmonitored", why not set qualify = yes?
Perhaps it should be set to "YES", to ensure that the client receives any incoming calls (so that the connection remains active ...) ?

Very Thanks

Yes, to keep connection open in NAT situation with host=dynamic, nat=yes for friend.

Ok Vulcan, I do not know where you live, but I did not pick you to offer a beer .... but ten beers!!!
The Inbound DID, with RealTime, now work very very well, thank to you !!!
To complete the overview, I can confirm that the really important changes were those that told me to start:
" In the general section of sip.conf try using these.
rtcachefriends = yes
rtupdate = yes ".
Because the settings as: host=dynamic, nat=yes for friend, I had always set the default.
Qualify=YES, does not change the result...
The only changes that make a difference are those described above.
Now, the result of "sip show peers" (in realtime mode) is: Realtime peer: Yes, cached
476349XXXX/476349XXXX 2.228.XX.XX D N 1029 OK (38 ms) Cached RT (qualify=yes)
and
476349XXXX/476349XXXX 2.228.XX.XX D N 1029 Unmonitored Cached RT (qualify=no)
But the result is the same: The Inbound DID, with RealTime, Works Great !!!

Very, Very Thanks Vulcan, for the effort you put into helping others.
Even I, in my limited opportunities with respect to you, when I can, I try to help someone:
viewtopic.php?f=34&t=9719&p=37349#p37349
I also did the translation into Italian of A2Billing, the User, the Agent, the Admin, web interface (a long work) :
viewtopic.php?f=19&t=9592
And some reports of bugs:
viewtopic.php?f=34&t=9593

Thank you very much again, Vulcan

Corrado.

Hello People,

I have the same problem of hang calls, but instead in sip user I have the problem in accounts. Im using a2billing as calling card solution.
Should I apply the same solution?

jhperezb