HI.
I wonder, if there is some log (like messages or another) which permit
to know the history of one user ( his last calls, when he was
called ...).
I want to know that in order to add, an history of each user in the
webcare.
Therefore, if someone can say me, if this log exist and where it is.
Thank you.
Here is what I want to do. Domestic routing defining interlata and intralata based off of ani and dnis (from uri and uri). Now let's not dive too much into the whole domestic routing thing or we'll be here for the next century. Bottom line is, this what I want to do with openser.
1. Look at ani (from_uri) (NPA - first 3 digits)
2. Look at dnis (uri) - (NPA - first 3 digits)
3. If the NPA (first 3 digits) in ani is in same as the DNIS NPA then redirect 555dnis@ip_address
4. If ANI NPA is in same as DNIS NPA then redirect 999dnbis@ip_address
I am redirecting so I can be and stay stateless
I am able to redirect but unable to figure out correct syntax for it to look at ani and dnis then redirect appropriately.
Thanks all
Hi,
in my 1.1.0 version I syddenly got a core dump. Does any one know why?
(gdb) bt
#0 0x000000304448660b in strftime_l () from /lib64/tls/libc.so.6
#1 0x0000002a95b29de4 in time2mysql (_time=Variable "_time" is not
available.
) at utils.c:54
#2 0x0000002a95b2a731 in val2str (_c=Variable "_c" is not available.
) at val.c:147
#3 0x0000002a95b26b13 in print_values (_c=0x643648,
_b=0x2a95db5269 "'SIP/2.0 100 Trying\\r\\nVia: SIP/2.0/UDP
217.17.222.4;branch=z9hG4bK1957.c42a1af6.0\\r\\nVia: SIP/2.0/UDP
10.1.93.186:10001;rport=44323;received=217.17.222.6;branch=z9hG4bK2eb74daf34599095\\r\\nFrom: \\\"Har"..., _l=65431, _v=0x7fbfffe990, _n=10) at dbase.c:144
#4 0x0000002a95b27aa4 in db_insert (_h=0x643e08, _k=Variable "_k" is
not available.
) at dbase.c:495
#5 0x0000002a97c26bdc in trace_sl_onreply_out (req=Variable "req" is
not available.
) at siptrace.c:1078
#6 0x0000002a956b30ad in run_sl_callbacks (req=0x643f28,
buffer=Variable "buffer" is not available.
) at sl_cb.c:88
#7 0x0000002a956b3608 in sl_send_reply (msg=0x643f28, code=100,
text=0x608ee8 "Trying") at sl_funcs.c:149
#8 0x0000000000409c38 in do_action (a=dwarf2_read_address: Corrupted
DWARF expression.
) at action.c:701
#9 0x000000000040a87e in do_action (a=dwarf2_read_address: Corrupted
DWARF expression.
) at action.c:89
#10 0x000000000040b43d in do_action (a=dwarf2_read_address: Corrupted
DWARF expression.
) at action.c:89
#11 0x000000000040b496 in do_action (a=dwarf2_read_address: Corrupted
DWARF expression.
) at action.c:89
#12 0x000000000040b762 in run_top_route (a=Variable "a" is not
available.
) at action.c:89
#13 0x000000000043188c in receive_msg (buf=0x2a97e4d2f8 "\026", len=822,
rcv_info=0x7fbffff810) at receive.c:155
#14 0x000000000044fc74 in udp_rcv_loop () at udp_server.c:465
#15 0x000000000041f7f4 in main_loop () at main.c:925
#16 0x0000000000420446 in main (argc=Variable "argc" is not available.
) at main.c:1477
--
Helge Waastad
Senior Engineer
Systemavdelingen
Smartnet
I'm using multi-domain and I can only call users by their aliases if I'm
dialling from a phone registered with the same domain of the called user
or if I'm dialling alias@domain_of_the_callee from a phone registered
with other domain.
How can I make Ser allow that every user can contact other users even if
they are from other domain by just dialling their alias (without
dialling also their domain when the callee is from other domain than the
caller)?
Regards,
Ricardo.
Hi to all!
I've a big problem with SER & Asterisk
I've a network where i put them working together.
Asterisk is the media gateway with PSTN as well as being the server which is
making all the services available (IVR, voicemail, conference calls, etc).
SER just routes calls either to the softphones, VoIP phones or to Asterisk
who will just forward the calls to PSTN (via Asterisk).
Till now everything works well except the service IVR which is working on
Asterisk.
Details:
192.168.226.13 - IP SER
192.168.226.45 - IP Asterisk
The problem is only after the user (roberto(a)192.168.226.13 - authenticated
and authorized in RADIUS) calls IVR.
Till then the user calls, IVR gives some information to the user and
everything goes well.
Problem is when he presses a number which will have to redirect the call to
another user.
And why is there a problem.
Because the initially roberto(a)192.168.226.13 to be able to call IVR had to
rewrite host & port 192.168.226.45 so that the call would reach Asterisk who
would then call to the IVR.
That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45.
That is, I can say, that it became a "new user".
While it's on the IVR there's no problem, but when call is diverted it has
to give its credentials...
Credentials that don't exist in RADIUS cause it only accepts usernames that
end in @192.168.226.13 (IP of SER)
Here is the following code, both of extensions file in Asterisk and
configure file of SER
I hope somebody knows how to solve it or at least find a way how to.
extensions.conf - Asterisk
[sip]
exten => 74001,1,Answer
exten => 74001,2,SetMusicOnHold(default)
exten => 74001,3,Set(TIMEOUT(digit)=5)
exten => 74001,4,Set(TIMEOUT(response)=10)
exten => 74001,5,Background(menu)
exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r)
...
exten => 7,1,Goto(sip,74001,1)
exten => 8,1,Hangup
ser.cfg - SER
...
if ((uri=~"^sip:7[0-9]{4}@.*")) {
route(5);
break;
};
...
route[5] { # Redirecting to Asterisk
# MsgLog
xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n");
rewritehostport("192.168.226.45:5060");
route(1);
}
Thanks for your help,
Roberto Lopes
Hi Users,
I am using SER-0.9.6 on public domain . and I am using
another Sip-proxy behind NAT (ip-address=192.168.1.100:5060)
i have some clients registered to that nated sip-proxy and here that nated
sip-proxy forwards PSTN calls to SER on public domain.
when call make from client (registered to nated sip-proxy)
the session establishing suscessfully , but when the client hung the phone
the SER on public domain is taking the BYE message and is not sending the
BYE message to the other end and it is sending directly to the Sip-proxy
which is a private-ip . So, here SER is not getting ACK and its keep on
sending the BYE requests to the privateip instead of sending to the NAT
address .
In some docs which jiri wrote on record route he
mentioned the same problem below is the link ; but he dint mention how to
solve that record route problem, If SER send messages back as it recieving
path (from NAT)there is no problem , but its changing the route to send BYE
,
For the request "CANCEL","ACK","INVITE","200 OK" every
thing is happening in the same Route but it is not happening only for the
"BYE" request .
Hope some body can help me.
Thank You.
here is the link which i
told;-http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/…<http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…>
Regards,
Ravi.
I'm using multi-domain and I can only call users by their aliases if I'm
dialling from a phone registered with the same domain of the called user
or if I'm dialling alias@domain_of_the_callee from a phone registered
with other domain.
How can I make Ser allow that every user can contact other users even if
they are from other domain by just dialling their alias (without
dialling also their domain when the callee is from other domain than the
caller)?
Regards,
Ricardo.
Glenn Dalgliesh wrote:
>This is from previous list
>
>"i Glenn,
>
>For this feature to work, you need to be sure that you do record_route() on
>the INVITE and that "append_fromtag" RR module param is not disable.
>
>regards,
>bogdan "
>
>
Yes i have read that thread. i always record_route all messages before
do things.
if (!method=="REGISTER")
record_route();
by the sample openser.cfg
and for be sure i have also forced:
modparam("rr","append_fromtag",1)
>-----Original Message-----
>From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf
>Of tele
>Sent: Wednesday, June 28, 2006 11:05 AM
>To: users(a)openser.org
>Subject: [Users] more on accounting
>
>Hi all,
>
>I'm using OpenSER v1.0.1 and i have done an accounting compatible with
>my billing system.
>Now i have problem with detection of flow direction for billing reason.
>
>i'm talking about openser 1.0.x:
>
>the problem is when the BYE come from callee.
>i know the problem is solved in openser1.1.x with the
>
>modparam("acc", "detect_direction", 1)
>
>i have read in modules documentation of 1.0.x that i can use
>is_direction() of RR module for detect call flow direction but it says
>that this must be called after loose_route()
>
>so for the accounting i must call setflag before loose_route() but what
>can i do if i want detect call flow
>before of exec setflag?
>
>Hmmm i think that i must install OpenSER 1.1.x and test with it
>
>:tele
>
>
>
>_______________________________________________
>Users mailing list
>Users(a)openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>
Hello,
Is there a way to have a machine that hosts media proxy have its one
interface in a private network and the other on a public?
>From what I read rtpproxy supports bridging which is basically what I want
to do but I have not found anything for mediaproxy.
Thanks
KM
