I have made changes on loose-route processing, so this part of script are this:
if (loose_route()) { log(1, "loose_route processing\n");
log(1, "-------------------------------------------\n"); log(1, "entering loose_route - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]\n"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7)\n"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; };
t_relay(); break; };
Anyway, I can hear anything on ATA side (SIP client), when call is comming from PSTN and forwarded to ser. The test I have made, include, comment this two lines:
>> if ((isflagset(5)) || (isflagset(6))) { >> if (method=="INVITE") {
The script pass but can't hear too.
fabio.
Klaus Darilion wrote:
If the cisco router sends in-dialog requests (e.g. re-INVITEs) which are handled by the loose_rote block, you also have to do NAT-traversal in this block.
klaus
Fábio Silvestri wrote:
SIPHello
I have a cisco PSTN forwarding specific call to my SER proxy, and this call (has a number phone) are right aliased (alias table) to a subscriber on my system.
Everything are ok, but the subscriber are inside nat, and the phone rings ok, but I can't hear anything!
I have make some debug log on ser.cfg, and seems to be after the call are answered the script hans on:
# loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; };And dont follow route[1], simply stops here!
Regards.
-- |o |o Drive defensively. Buy a tank. |o |o Fabio Silvestri |o fabio@informatec.com.br |o ICQ: 1667351 |o
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