Hello,
I need your advises
Here is my problem :
I setup asterisk ser :
Asterisk 5050--
|--Firewall---|----- SIP User agents
SER 5060---
My client use SER as outbound proxy and asterisk as
sip registrar.
SER is able to forward SIP register request to
asterisk however .
How SER can inform SIP User agents to send register
requests to asterisk.
May I have to configure a redirect server to advise
SIP User agents of the sip register server (asterisk)?
can you help me for ser.cfg if possible ?
Regards
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hi
I get Process Request Timedout ever whenever i try to register with SER. I get following messages on starting ser.
0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184
0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184
0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232
0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184
0(0) DEBUG: setting SO_RCVBUF has no effect
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) DBG: open_uac_fifo: opening fifo...
0(0) DEBUG: FIFO created @ /tmp/ser_fifo
0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432
0(0) init_unixsock_socket: No unix domain socket will be opened
1(0) DEBUG: init_mod_child (-1): tm
1(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57'
1(0) DEBUG: init_mod_child (-1): usrloc
1(0) get_connection(): Connection not found in the pool
2(0) INFO: fifo process starting: 10317
2(0) DEBUG: init_mod_child (-2): tm
2(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57'
2(0) DEBUG: init_mod_child (-2): usrloc
2(0) get_connection(): Connection not found in the pool
0(10313) DEBUG: init_mod_child (1): tm
0(10313) DEBUG: callid: '3fbfd771-10313(a)69.229.23.57'
0(10313) DEBUG: init_mod_child (1): usrloc
0(10313) get_connection(): Connection not found in the pool
1(10316) DEBUG: init_mod_child (-1): auth_db
1(10316) get_connection(): Connection not found in the pool
0(10313) DEBUG: init_mod_child (1): auth_db
0(10313) get_connection(): Connection not found in the pool
1(10316) DEBUG: init_mod_child (-1): domain
1(10316) DEBUG: init_mod_child (-1): nathelper
1(10316) rtpp_test: RTP proxy found, support for it enabled
0(10313) DEBUG: init_mod_child (1): domain
0(10313) get_connection(): Connection found in the pool
0(10313) DEBUG: init_mod_child (1): nathelper
0(10313) rtpp_test: RTP proxy found, support for it enabled
2(10317) DEBUG: init_mod_child (-2): auth_db
2(10317) get_connection(): Connection not found in the pool
2(10317) DEBUG: init_mod_child (-2): domain
2(10317) DEBUG: init_mod_child (-2): nathelper
2(10317) rtpp_test: RTP proxy found, support for it enabled
2(10317) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
2(10317) DEBUG: register_fifo_cmd: new command (print) registered
2(10317) DEBUG: register_fifo_cmd: new command (uptime) registered
2(10317) DEBUG: register_fifo_cmd: new command (version) registered
2(10317) DEBUG: register_fifo_cmd: new command (pwd) registered
2(10317) DEBUG: register_fifo_cmd: new command (arg) registered
2(10317) DEBUG: register_fifo_cmd: new command (which) registered
2(10317) DEBUG: register_fifo_cmd: new command (ps) registered
2(10317) DEBUG: register_fifo_cmd: new command (kill) registered
2(10317) WARNING: no fifo_db_url given - fifo DB commands disabled!
Can anyone please have a look at above information and let me know what is wrong with ser?
Thanks
Rajesh
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The scenario are ser 0.9.3 + sems 0.9.0 + mediaproxy + mysql, al instances
are in the same machine
The code in my ser.cfg
if (!lookup("location")) {
if ((method=="INVITE" || method=="ACK") && t_newtran() ) {
acc_db_request("Llamada perdida","missed_calls");
if(!t_write_req("/tmp/am_fifo","voicemail")){
t_reply("500","Error contacting sems
voicemail");
};
#t_reply("404", "Usuario no conectado");
break;
};
} else if(method=="BYE" || method == "CANCEL"){
if(!t_write_req("/tmp/am_fifo","bye")){
t_reply("500","Error contacting sems");
};
};
sl_send_reply("404", "Usuario no encontrado");
break;
setflag(3);
when i call a offline user, the softphone make a call but I don't ear any
message or received any notification, anyone can help me?
this is /var/log/messages
Nov 3 11:04:56 dns2 ser[23838]: Maxfwd module- initializing
Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially
110592
Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142
Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially
110592
Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142
Nov 3 11:04:56 dns2 ser[23847]: INFO: fifo process starting: 23847
Nov 3 11:04:56 dns2 ser[23847]: SER: open_uac_fifo: fifo server up at
/tmp/ser_fifo...
Nov 3 11:05:07 dns2 ser[23848]: WARNING: t_reply: ACKs are not replied
Nov 3 11:05:08 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied
Nov 3 11:05:10 dns2 ser[23851]: WARNING: t_reply: ACKs are not replied
Nov 3 11:05:14 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied
Nov 3 11:05:23 dns2 mediaproxy[3158]: command delete 3924182bdc181a37 info=
Nov 3 11:05:23 dns2 mediaproxy[3158]: command execution time: 0.20 ms
Hello,
Does acc module allow these requests to a radius
server according to rfc2866 ?
A Talk Start information is sent when SIP Server
receives 200 OK for a INVITE request.
A TAlK Stop information when SIP Server receives
BYE.
How may i configure it ? if acc module can't calculate
acct-session-time why does this module provide radius
support ?
Regards
Harry
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Hi,
How many simultaneous connection can SER + SEMS can handle? i
deployed SER + SEMS in Pentium 2 300MHz with 128MB memory, how many
connections can this handle?
Thanks,
--
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Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
I have installed SER 0.9.3.2 and media proxy 1.2.1 on the same machine.
The machine is on public network.
Calls work ok except when I have a UA on a LAN behind NAT calling to a UA that is in the same LAN.
In this case I get :
+----------+-------------------------------+--------------------------+| username | contact | received |+----------+-------------------------------+--------------------------+| 22000001 | sip:22000001@192.168.1.7:5060 | sip:xxx.xxx.xxx.xxx:5060 || 22000002 | sip:22000002@192.168.1.6:5060 | sip:xxx.xxx.xxx.xxx:2557 |+----------+-------------------------------+--------------------------+The number that takes up port 5060 on the NAT server,which is 22000001 can call the other UA but not the other way round.22000002 can only receive calls.Please help me,KM
Hello everybody,
I have a question related with conference servers. Do you now how is done
the mixing in a Conference? Specifically, how input RTP streams are mixed
into output string? Should I need to mix in a linear way, interleaving
samples of different users?
Thanks in advance!!
Hi,
I am trying to figure out how to interpret response code 408 in a
failure_route block. It seem that there is a couple of different cases. If
the code means that a PSTN gateway is not responding (the tm's fr_timer has
expired), I want to try another PSTN gateway. On the other hand, if the
code means that the user is not responding (the tm's fr_inv_timer has
expired), I want to stop trying and forward the code to the UAC.
How can I tell what response code 408 actually means?
Thanks!
Dmitry