Hi,
Is it true that MSILO module only can handle text message?
So what is the usage of voice_silo table that written on message_store.php on serweb then?
Thanx
Regards,
Meidiana
---------------------------------
Access over 1 million songs - Yahoo! Music Unlimited.
Can the number of times that an INVITE is sent with UDP be configured
with a modparam type setting in the cfg file?
For example, a max invite retry count setting?
I saw modparam "tm" settings for timers, but nothing for the retry
count.
Thank you,
Paul
---------- Forwarded Message ----------
Subject: [Sems] About field of subscriber
Date: Monday 04 December 2006 09:09
From: "chungyu" <chungyu(a)ms11.voip.edu.tw>
To: sems(a)lists.iptel.org
Hi:
I add a field in subscriber and using the field routing process
example:
field name:mode
format :ture or false
if mode is true
route[2]
else
route[3]
how can write ser'configuration that call field of subscriber?
AVP can do?
Thank you!
changyu
2006/12/04
-------------------------------------------------------
--
Dipl. Inf. Raphael Coeffic
iptelorg GmbH
Am Borsigturm 11
13507 Berlin
Germany
rco(a)iptel.org
www.iptelorg.com
T +49-30-3251-3218
F +49-30-6908-8248
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Hi all,
I have some question about call transfer.
When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408).
Inside REFER message, i can see the refer-to and referred-by values are correct.
Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer.
Thanks for any help.
Regards,
jorain
Hi,
Regarding the SRV query addition in (ser-0.9.7-pre7), when the call
fails on the first priority destination, it never seems to attempt to
send the INVITE to the second priority destination in the SRV response
list. Is there anything else to script in the ser.cfg file to get this
to work?
Thanks for help,
Paul
[root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP
Capturing on eth0
10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE
sip:77710107773365@10.86.129.17:5060;transport=udp
10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your
call is important to us
14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout
14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK
sip:77710107773365@10.86.129.17:5060;transport=udp
2139 packets captured
[root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS
Capturing on eth0
4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV
_sip._udp.vxml.pats.cisco.com
4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV
1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3
1 5060 vxml-3.pats.cisco.com
4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A
vxml-1.pats.cisco.com
4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A
10.86.129.2 <--this is unplugged from the network
debug=3
fork=yes
log_stderror=yes
rev_dns=no
port=5060
children=4
check_via=no
sip_warning=yes
fifo="/tmp/ser_fifo"
uid="nobody"
gid="nobody"
#aliases for this proxy server
#ie hostnames/domains that it routes for
alias=sox.cisco.comalias=pats.cisco.com
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
modparam("tm","fr_timer",5)
modparam("tm","fr_inv_timer",5)
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (uri=~"^sip:1") {
log(1, "SER found 1*\n");
rewritehostport("ccm.pats.cisco.com");route(1);break;
}
if (uri=~"^sip:2") {
log(1, "SER Found 2*\n");
rewritehostport("ccm.pats.cisco.com");route(1);break;
}
if (uri=~"^sip:7") {
log(1, "SER Found 7*\n");
rewritehostport("vxml.pats.cisco.com");route(1);break;
}
if (uri=~"^sip:8") {
log(1, "SER Found 8*\n");
rewritehostport("SER.pats.cisco.com");route(1);break;
}
if (uri=~"^sip:9") {
log(1, "SER Found 9*\n");
rewritehostport("ringtone.pats.cisco.com");route(1);break;
}
log(1,"Could Not Match DN to Route\n");
route(1);
}
route[1]
{
if (!t_relay()) {
sl_reply_error();
};
}
Hi all,
I read on the docs of the tm module that I can use t_replicate to
send, for example, REGISTER to multiple destinations.
Can a I replicate to more than one server, such as
t_replicate("sip:1.2.3.4")
t_replicate("sip:1.2.3.5")
t_replicate("sip:1.2.3.6")
and so on ?
>From the docs it seems possible, but I don't know how to set branches.
TIA,
--
"Work and play are words used to
describe the same thing under
differing conditions." Mark Twain
Hi All
I need to know the usage of 'phplib_id, domn & uudi' columns into ser
database table subscriber. Please give me your valuable suggestion for
the same.
Regards
Kamal Mann
Dear All
i will setup openser behind NAT like these
OPENSER ---> ADSL ROUTER(A) <---> INTERNET <---> ADSL ROUTER(B)<-- UAC(1)
<-- UAC(2)
for UAC to talk each others in the same network with OPENSER server and
try to REGISTER AND INVITE IT OK .
but if i use UACs that located behind ADSL ROUTER(B) for REGISTER it OK
when UAC(1) INVITE UAC(2) it RING and session established can heard sound
but after about 30 second the CALLEE UAC(2) is HANG UP while CALLER UAC(1)
still show established
i saw some mail said that the problem may occour from ADSL ROUTER ITSELF
(AGL problem)or some say that ACK problem. if i connect UAC(1) and UAC(2)
through ADSL ROUTER(B) to iptel.org and i use UAC(1) INVITE UAC(2) it work
no problem for 30 second disconnection . also if i use UAC(1) and UAC(2)
conect to iptel.org through ADSL ROUTER(A) no problem . AT this point
can I CONCLUDE THAT BOTH ADSL ROUTER NO PROBLEM FOR SIP CONNECTION. is it
TRUE ??
Next i saw from LOG file it seem no ACK MESSAGE and error like these
-- LOG FILE -------------
ERROR: tcp_blocking_connect: poll error: flags 18
ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused
ERROR: tcpconn_connect: tcp_blocking_connect failed
ERROR: tcp_send: connect failed
---
for simply NAT testing i use openser.cfg like these
---------- OPENSER.CFG ---------
if (!method=="REGISTER")
record_route();
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("infowavenet.com", "subscriber")) {
www_challenge("infowavenet.com", "0");
exit;
};
fix_nated_register();
save("location");
exec_msg("cat >> /tmp/test");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
fix_nated_contact();
if(method == "INVITE")
{ fix_nated_sdp("3"); }
if (!t_relay()) {
sl_reply_error();
} else { exec_msg("cat >> /tmp/test");}
exit;
}
------------
MY QUESTION:
1) can openser server is BEHIND NAT ADSL ROUTER and receive Call from
UAC from other NAT ROUTER .
2) if so what the problem that CALLEE HANGUP every 30 seconds
( i use XLITE v.3 for UAC )
3) if it possible can you fix openser.cfg that work for this situation
Best
Somsak Vattanavakin
Not sure why that's happening. Probably setting canreinvite=no on the
asterisk side will eliminate the re-INVITEs as a temporary solution, but
still would like to know what is happening...
wrote:
> Sometimes, a calls b and b hears a, and a hears b for a second but a
second
> INVITE comes to phone B that causes it to redirect rtp to be point to
point.
> Sometimes there is no audio.
> Sometimes, everything works fine.
> At one point, rtp from a was going to asterisk, but asterisk was not
sending
> the rtp on to b, and b was trying to send traffic point to point.
Hi all.
After UA makes a PSTN call, if remote PSTN peer hangs up first, server
correctly receives BYE from gateway, hit BYE section in config and CDR
records accordingly through radius.
But if UA hangs up first, BYE section in config never gets hit, instead
server logs "ERROR: forward_reply: no 2nd via found in reply" right after
" ACC: transaction answered:...", which was logged when the call was
answered by called PSTN. CDR accounting only records without STOP time
and duration being zero.
Any comments will be appreciated.
