Hi, Please what is the purpose of settings this SIP_DOMAIN environment
variable. I tried setting mine to a different domain from the server
(export SIP_DOMAIN=anotherdomain)and it still works. I don't understand.
Thanks for your help.
regards,
Onyeka
Hi all,
Just installed serweb... But the authentication seems to be not working
for both the administrator as well as for users. When I try to recover
password with forgotton password option, there is an error message saying
that the user is not a registered user. When I look at the mysql database,
it's there. I think the there is no connection established with the ser
database. Any ideas about how to overcome this? For this to work is it
necessary to have the mysql.so module?
Regards,
Lakmal
Lankacom Services (Pvt) Ltd.
65C, Dharmapala Mawatha,
Colombo 07.
Sri Lanka.
Tel: +94-1-437545
www.lankacom.net
Hello everyone,
I have a question. I have set up SER with couple of grandstream phones
and i am using VEGA gateway. It works great. I have two dial plans for
outgoing calls:
# Destination PSTN or H323?
if( uri=~"^sip:9[0-9]*@voip.sip_proxy.com" )
{
route(1);
break;
};
if( uri=~"^sip:\*74@voip.sip_proxy.com" )
{
route(2);
break;
};
# ------------ Send it to our PSTN ----------------------
route[1] {
# Route to PSTN Gateways(s)
if (uri=~"^sip:9[0-9]*@voip.sip_proxy.com") { ## This
assumes that the caller is
log("Forwarding to PSTN\n"); ## registered
in our realm
strip(1);
#t_relay_to_udp( "voip.gateway.com", "5060" );
rewritehostport("voip.gateway.com:5060");
forward(uri:host, uri:port);
break;
};
}
route[2] {
if (uri=~"^sip:\*74@voip.sip_proxy.com") { ## This
assumes that the caller is
log("Picking up a Call on PSTN\n"); ##
registered in our realm
#t_relay_to_udp( "voip.gateway.com", "5060" );
rewritehostport("voip.gateway.com:5060");
forward(uri:host, uri:port);
break;
};
}
That is working fine. This *74 is for call pick up if someone is
wondering what the hell is that :)
However once I dial any number that has a menu options like: Press 1
for english, 2 for spanish and so on, i can't do anything. I keep on
pressing the numbers on my key pad but nothing is going on. I tried
with the 9 as well, ( to append 9 before the number ), but nothing. Is
it possible to have that with ip phones at all? Is there a dial plan
for something like that? I guess I am little bit confused 'cause it's
an established call, not a new one, so i am not sure where to start
looking. Any ideas?
thanks.
Srbo Cvetkovic | CityNet, Inc.
srbo(a)city-net.com | Pittsburgh, PA
voice: 412.481.5406 | fax: 412.431.1315
Hi,
I'm just trying SER for the first time ...
But I can't download serweb because I'm behind an HTTP proxy ... And
that ftp is imposible to reach for me ...
Please, some one could tell me an http url to download serweb ...
Thanks ...
Dear all,
We have encountered a problem when we try to use a remote mysql server
instead of localhost server.
Ser.cfg
modparam("usrloc", "db_url", "sql://ser:>heslo>@192.168.1.5/ser")
stdout:
mod_init(): Can't bind database module
We configure followed everything in admin document and database work
locally. Does anyone has any idea?
Regards
Hi,
maybe someone can help me with this Problem.
Actually we have 300 Users in our location table,
having nat_flag = 4;
Servermemory:
total used free shared buffers cached
Mem: 1029208 971552 57656 0 115460
572440
-/+ buffers/cache: 283652 745556
Swap: 996020 6300 989720
System: Debian 2.4.24
Ser: sip_router_0.8.12_stable
# -- nat Params
modparam("nathelper", "natping_interval" , 20)
modparam("registrar", "nat_flag", 4)
every 20 Seconds I have the Error in SYSLOG:
ERROR: nathelper::timer: out of memory
I found the Error message in the ser/modules/nathelper.c code
but i am not really a c programmer.
any ideas?
Best Regards,
Markus
Hey guys!
I'v got'n into a weird problem.. if you look at the 2 failure_route
statements I have added, There is not much difrent then the
failure_route[] that has been changed from 9 to 10.. Tho this makes my
config Stop working..
is there a limitation that does this?
failure_route[9] {
if (t_check_status("(408|486|501)")) {
rewritehostport("192.168.0.10:5060");
append_branch();
t_on_failure("11");
t_relay();
};
}
failure_route[10] {
if (t_check_status("(408|486|501)")) {
rewritehostport("192.168.0.10:5060");
append_branch();
t_on_failure("11");
t_relay();
};
}
Here's the output I get when I try start my ser with this..
ser: parse error (754,1-2): invalid reply routingtable number
- Atle
Hi Izzi,
As I told you, maybe the patch is outdated. Please switch to the
unstable version (CVS head) because you don't need anything special for
cpl-c. Meanwhile I will try to take a look to 0.8.12 patch.
Regards,
Bogdan
Izzi Fiorenzo wrote:
> Hi Bogdan
> I applied ptm patch as you said, but I had as result of compilation:
>
> [root@localhost ser-0.8.12-con_tm_patch]# make modules > errore1
> t_reply.c: In function `run_failure_handlers':
> t_reply.c:492: structure has no member named `body_lumps'
> t_reply.c:493: structure has no member named `body_lumps'
> t_reply.c:493: structure has no member named `body_lumps'
> t_reply.c:494: structure has no member named `body_lumps'
> t_reply.c:527: structure has no member named `body_lumps'
> t_reply.c:528: structure has no member named `body_lumps'
> t_reply.c: In function `t_should_relay_response':
> t_reply.c:654: warning: implicit declaration of function `failure_route'
> make[1]: *** [t_reply.o] Error 1
>
> If compile with old (of ser0.8.12) t_reply.c file, compilation is OK.
>
> I think there is something wrong or missing in t_reply.c patch.
>
> It seemd to me there was an "s" exceeding in "body_lumps" and
> eliminated it, but error remained.
>
> I attach my new t_replay.c with patch.
>
> Ciao.
>
> Enzo
>
>
Hi!
If ser relays an INVITE transaction stateful (t_relay), it automatically
sends a provisional response (100 trying) back to the caller. When
t_relay()ing non-INVITE requests (e.g. BYE), there will no (100 trying)
produced. Any special reason for this?
So if the request timeouts (no repsonse within 500ms), both - ser and
the callers UA - will retransmit the request.
My way to solve this would be:
...
if ((method!="INVITE") && (method!="ACK"))
sl_send_reply("100", "trying...");
t_relay()
...
Another possibility would be to relay non-INVITE requests stateless. Any
comments/suggestions on this?
regards,
Klaus
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
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