Hi Stefano,
I suppose that you have the nat traversal handled also on opensips,
right? ( since the host part of the clients registration in asterisk
points to opensips). Then in this case it is normal that the flow of the
invite is the following:
sip client -> opensips -> asterisk -> opensips (again)
So you must take care when the Invite arrives the second time at
opensips not to send it again to asterisk. For this do a filter after
the source ip and port, so you should have:
if (is_method("INVITE")) {
if (!ds_is_in_list("$si", "$sp")) ) /* if it's not from asterisk */
if (!ds_select_dst("1", "5")) {
....
} else
route(1); /* send it out */
Regards,
--
Anca Vamanu
www.voice-system.ro
On 09/27/2010 12:01 PM, Stefano Sasso wrote:
> Hello,
> I started testing OpenSIPS with my asterisk cluster.
> For this testing configuration I'm using only one asterisk server of
> the pool. (the registration is done on asterisk server)
>
> The OpenSIPS configuration routing part is:
>
> route{
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> xlog("SIP route info: [$fu/$tu/$ru/$ci]");
> xlog("SIP method: $rm");
>
> if (!has_totag()) {
> # initial request
> record_route();
> }
> else {
> # sequential request - obey the indicated route
> loose_route();
> t_relay();
> exit;
> }
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> if (is_method("OPTIONS")) {
> t_relay();
> xlog("OPTIONS for: [$fu/$tu/$ru/$ci]");
> exit;
> }
>
> # from now on we have only the initial request
> # select the node that'll handle the call
> # different method is used for INVITE/REGISTER requests
> # unknown methods are rejected here
>
> xlog("First route:
> [USER:$au/CALL-ID:$ci/FROM-TAG:$ft/FROM-URI:$fu/FROM-URI-USER:$fU/METHOD:$rm]");
>
> if (is_method("INVITE")) {
> if (!ds_select_dst("1", "5")) {
> xlog("Service unavailable for ds_select_dst[INVITE]");
> send_reply("503","Service Unavailable");
> exit;
> }
> }
> else if (is_method("REGISTER")) {
> if (!ds_select_dst("1", "5")) {
> xlog("Service unavailable for ds_select_dst[REGISTER]");
> send_reply("503","Service Unavailable");
> exit;
> }
> }
> else {
> send_reply("405","Method Not Allowed");
> exit;
> }
>
> # route the request
> if (!t_relay()) {
> sl_reply_error();
> }
> }
>
>
> the dispatcher.list is simple:
> 1 sip:192.168.6.131:5060
>
>
> the registration part, on asterisk seems OK (I think), but for every
> SIP phone in 'sip show peers' I saw only the openSIPS IP address:
> Name/username Host Dyn Nat ACL Port Status
> 2001/2001 192.168.6.130 D N 5060 OK (1342 ms)
> 2002/2002 192.168.6.130 D N 5060 OK (1403 ms)
>
> My asterisk dialplan is simple:
> exten => 101,1,Answer()
> exten => 101,n,Playback(demo-congrats)
> exten => 101,n,Hangup()
>
> exten => _XXXX,2,NoOp(internal context called on Server A to ${EXTEN},
> CALLERID_num ${CALLERID(num)}, CALLERID_num_f2 ${CALLERID(num):0:2})
> exten => _XXXX,3,Dial(SIP/${EXTEN},15)
>
>
> when, from phone 2001 or 2002 I place a call to 101 everything works
> fine, but when I call 2001 from 2002 (and vice-versa) I can't place
> the call, and in asterisk log I found:
>
> -- Executing [2...@internal:1] NoOp("SIP/2001-0000000f", "2002")
> in new stack
> -- Executing [2...@internal:2] NoOp("SIP/2001-0000000f", "internal
> context called on Server A to 2002, CALLERID_num 2001, CALLERID_num_f2
> 20") in new stack
> -- Executing [2...@internal:3] Dial("SIP/2001-0000000f",
> "SIP/2002,15") in new stack
> == Using SIP RTP CoS mark 5
> -- Called 2002
> -- Got SIP response 482 "Loop Detected" back from 192.168.6.130
> -- Now forwarding SIP/2001-0000000f to 'Local/2...@internal'
> (thanks to SIP/2002-00000010)
> -- Executing [2...@internal:1] NoOp("Local/2...@internal-187c;2",
> "2002") in new stack
> -- Executing [2...@internal:2] NoOp("Local/2...@internal-187c;2",
> "internal context called on Server A to 2002, CALLERID_num 2001,
> CALLERID_num_f2 20") in new stack
> -- Executing [2...@internal:3] Dial("Local/2...@internal-187c;2",
> "SIP/2002,15") in new stack
> == Spawn extension (internal, 2002, 3) exited non-zero on
> 'Local/2...@internal-187c;2'
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [2...@internal:4] Hangup("SIP/2001-0000000f", "") in new
> stack
> == Spawn extension (internal, 2002, 4) exited non-zero on
> 'SIP/2001-0000000f'
>
> Could you please give me some hints on where I am wrong?
> registering directly on asterisk box, bypassing OpenSIPS, the call
> works correctly.
>
> thanks so much
>
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