provided by Star2Billing S.L.

I installed a Trixbox (last release) and A2Billing (1.3.3). I had this problem:

a device configured with freepbx can connect to a sip trunk placing the call and having correct audio

a device created by a2billing can place the call correctly (sip proto is ok) but hears no audio at all.

Sniffing the net, I saw this: when the "freepbx configured" device places the call, the sip provider (betamax) gives a 183 code with a "Connection Address" that's different from the sip address itself. Asterisk correctly sends all RTP packet to this new IP.

When, instead, a a2billing device places the call and gets the same 183 message back with the Connection Ip address, asterisk starts sending rtp packets to the same "sip server" ip address, so missing all audio frames.

Any hints?

Alfonso

Hi

With the command sip show peer <<exten number>> do a comparison between the two, and see what is diferent.

Joe

Hi Jroper, thanks for answering

I made the compare, found some differences, modified the device description, but the problem persists.. Seems not to be a peer problem.

Alfonso

Added after 41 minutes:

To be more precise, I made some tests and it seems that the whole bunch of preconfigured stuff included in additional_a2billing_sip.conf works correctly only if I change "canreinvite" from yes to no AND i change the context from a2billing to from-internal. Of course this last removes all the code made to bill the call..
I think that the problem could really be inside something than the DeadAGI a2billing inserts in the call, if any

A good and bad news at the same time.

It seems that the problem is given ONLY in conjunction with shorewall (firewall setting). The configuration I used is full udp port open, and only 80 and 22 in tcp.
When the shorewall is started from the boot, the described problem is present, but is completely not present when shorewall is disabled.

I remind that's not a firewall problem (as it works with normal freebx account) but something related. In case someone wants, I can generate some sniffed ethernet traffic.

Alfonso

If A2billing sends rtp to SIP server address instead of advertised RTP port then apparently a2billing assumes that the sip provider is behind NAT.

I guess your general sip.conf settings or the conf settings of the provider have somewhere NAT enabled. Disable this

Cheers

Geejay

Thanks geejay

The only nat=yes are just under friends descritpion, together with account data. Do I have to disable these? I thought they were meant as "friend could be natted". No "nat0=" is under trunk description, do I have to explicitly say nat=no?

Alfonso

Alfonso,

yes. set nat=never so that Asterisk doesnt assume NAT.

I understand that your a2billing/asterisk is not behind NAT and that shorewall doesnt provide a private ip address to a2billing.

Though, If a2billing/asterisk is behind NAT then you must configure "externip" to your extrenal ip in sip.conf and you must forward all SIP and RTP related ports from the firewall to a2billing/asterisk. That way asterisk behaves as if it is not behind NAT.

Cheers

Geejay