On 7/5/06, raviprakash sunkara sunkara.raviprakash.feb14@gmail.com wrote:
hello Bogdan,
Same problem occurred for me , but I'm Using X-lite 3.0 . And i'm put on DMZ in my router ( NETGEAR and BELkenn) .
I install the openser 1.0.1 and rtp proxy 0.3 in same Linux System openser server is located with public id xx.xxx.xxx.xx of 192.168.2.2 , And UAC are outside the NAT, When one UAC call to other UAC( are both in outside the NAT where openser server), after the INVITE method get request by server, after 32 second its hung up automatically, Voice is ok , and callee is hung upping, not caller, UAC ( inside the nAT , openser server ) in not hung uping and voice is not ok....
I think problem is not in NETWORK and it may in RTp , NAT , Can u help on this , Where is the problem, in NAt with rtp or networking, Here by Mime's openser.cfg
route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; };
# NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("xx.xxx.xxx.xxx", "subscriber")) { www_challenge("xx.xxx.xxx.xxx", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1);
}
route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); };
if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit;
}
route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; }
route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); }
failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } }
onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isflagset(6)) { fix_nated_contact(); } exit;
}
On 7/5/06, Bogdan-Andrei Iancu <bogdan@voice-system.ro > wrote:
Hi,
it might be a signalling problem. Most of the UA drops the calls if they
do not get the ACK for 200 OK. check on the network if this is the case.
regards, bogdan
Hamid Ali Asgari wrote:
The calls are between two UAs. The problem is that with a certain type of UA (type A), the calls are
ok if
the calls are between two type A UAs and don't get disconnected. I can
talk
as long as I want.
But if I try calling from that UA (type A) to Windows messenger, the
call
gets disconnects after less than a minute. In the 1 minute I can talk
(so I
assume it's not a CODEC problem, correct me if I am wrong)
I have also tried with a UA and a Cisco gateway. On the Cisco debugs I
see
Disconnet cause code 102 (Session-End-Callback ) which I don't think
would
be the case. There is no callback config on the gateway or the UA.
I guess the UA is tearing down the call for some reasn I don't know.
Any clues? Hamid
-----Original Message----- From: users-bounces@openser.org [mailto:users-bounces@openser.org] On
Behalf
Of Mike Williams Sent: Wednesday, July 05, 2006 8:04 PM To: users@openser.org Subject: Re: [Users] Disconnect Cause on OpenSER
On Wednesday 05 July 2006 12:31, Hamid Ali Asgari wrote:
Are the calls from one UA to another, or from a UA to a gateway? I know
for
instance that Asterisk has problems with G729b silence detection and
will
drop calls because it thinks the call has dropped. Perhaps other
equipment
or carriers has this problem too.
---Mike
Hi,
I am having a problem with OpenSER and certain types of CPEs. The
problem
is that the calls get established and the parties can talk, however
after
a
very short period the call gets disconnected. Any guidelines how I
could
troubleshoot this?
PS: Is there anyway to see the calls disconnect cause on OpenSER? I am
currently running OpenSER with radius.
Thanks in advance,
Hamid
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-- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535