Hi everybody,
I'im in charge of installing a sip server for my university. At the moment, I'm
testing SER from my stand alone computer.
But I've a problem : when I want to add a user with serctl, here is the answer
:
ERROR 2005: Unknown MySQL Server Host 'dbhost' (1)
mysqld is running, ser is running, what is this "dbhost" ??? Any idea ?
Cheers,
Nicolas
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Hello,
after setting fr_timer and fr_inv_timer to lower values than default:
modparam("tm", "fr_timer", 12)
modparam("tm", "fr_inv_timer", 24)
noticed that some SIP phones continues to ring even if a caller hangs up
the phone. After a callee receives CANCEL message it responds either with
481 ir 487 (transaction does not exists). If fr_inv_timer is not expired,
callee responds with 481 and cancels INVITE request. If timer, by the time
caller sends CANCEL, is expired, callee returns 487 and does not stop
ringing.
After comparing CANCEL messages in both cases (using algorithm
described in 3261, 17.2.3) i didn't find any difference between them
(Branch, sent-by, to tagm from tag, call-id, cseq - all matches). How UAS
distinguishes these CANCEL messages and how SER changes CANCEL message
after fr_inv_timer is exeeded?
Antanas
HI all,
Anyone who's deployed SER and Asterisk together, just wanted
to
know that what are difference in functionalities between SER and
Asterisk. I mean people generally use SER as front end to do NAT and
stuff. Asterisk is generally used behind SER to maintain the dialplan
and appropriate routing. I was wondering if Asterisk can to Natting and
SER can be used as a registrar to store information, why cant we use
them interchangeably? Is there any specific killer function these two
have which are different?
Thanks,
Hitesh.
Greetings:
With some tweaks (no pthreads, missing includes, reordering includes,
etc) I was able to build Ver 0.8.14 on OpenBSD/sparc 3.2, gcc 2.95.3.
When attempting to run the executable for the first few times, I got
the unresolved symbol messages other OpenBSD users have posted to this
list while 'ser' is loading modules; however this behavior vanished
spontaneously and now ser simply segfaults and dumps core after about
five seconds (Sun sparc IPX host).
OpenBSD 3.2 doesn't have 'pthreads' but instead has a similar 'pth'
package; 'ser' was built using SysV semaphores instead.
`ser -V`:
version: 0.8.14 (sparc/openbsd)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MM AP, PKG_MALLOC, F_MALLOC MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $
main.c compiled on 13:32:17 Nov 17 2004 with gcc 2.95
'ser -c' also segfaults but first reports:
0(26788) WARNING: could not read from /dev/random (5)
0(26788) ERROR: shm_mem_init: could not initialize lock
Backtrace of the core file from running 'ser' with an empty 'ser.cfg':
This GDB was configured as "sparc-unknown-openbsd3.2"...
(no debugging symbols found)...
Core was generated by `ser'.
Program terminated with signal 11, Segmentation fault.
Reading symbols from /usr/lib/libc.so.28.5...(no debugging symbols found)...
done.
Reading symbols from /usr/libexec/ld.so...(no debugging symbols
found)...done.
#0 0x9ee88 in lock_destroy ()
(gdb) bt
#0 0x9ee88 in lock_destroy ()
#1 0x9ede3 in shm_mem_destroy ()
#2 0x9e80b in shm_mem_init_mallocs ()
#3 0x9ea13 in shm_mem_init ()
#4 0x9d783 in init_shm_mallocs ()
#5 0x3acb7 in main ()
#6 0x118df in ___start ()
Anyone care to help?
All replies much appreciated.
Michael Grigoni
Cybertheque Museum
Hi,
I have installed the library and solved the first portion error but I still
encountered this:~
Please help........................
> Ivr.cpp: In member function `virtual void
> IvrDialog::onSessionStart(AmRequest*)':
> Ivr.cpp:243: no matching function for call to `IvrPython::cancel()'
> make[4]: *** [Ivr.o] Error 1
> make[4]: Leaving directory `/root/answer_machine/plug-in/ivr'
> make[3]: [all] Error 2 (ignored)
> make[3]: Leaving directory `/root/answer_machine/plug-in/ivr'
>
best regards,
shirley
> -----Original Message-----
> From: Soren Davidsen [SMTP:soren@tanesha.net]
> Sent: Tuesday, November 16, 2004 7:50 PM
> To: Shirley Toh
> Subject: Re: [Serusers] IVR Module Problem
>
> << File: ATT00000.txt; charset = utf-8 >>
--- "Greger V. Teigre" <greger(a)teigre.com> wrote:
> Hi,
> I've been following this thread as I have experienced the same problems
> myself. When I get incoming calls (both from Cisco IP-PSTN gateway and from
> other SIP phones) to a Grandstream behind symmetric NAT, the messages you
> have noted can be seen in the log when hanging up.
>
> I was not certain as to the conclusion you ended with. Do you use the
> filter:
> if (!(search("^Content-Length:\ 0")) {
> force_rtp_proxy();
> };
>
> to avoid the errors? I have been thinking about testing on method and not
> call force on BYE and ACK. Have you tried this?
>
> I also saw your question on RFC compliance and the Sonus equipment: In
> order to make Grandstream phones register properly when using STUN behind
> symmetric NAT, I had to patch nathelper with the rport != port of received
> address check. (I use 0.8.14 and I guess you already have the patch with
> the development version). The reason is that Grandstream attempts to
> rewrite the address using STUN even though it correctly detects a symmetric
> NAT. I have seen that this was introduced in a new firmware not long ago
> (release notes). This pussles me as sources I have seen claims this to be
> invalid behavior (which seems correct to me).
>
> Best regards,
> Greger
>
>
> Java Rockx wrote:
> > Hi All.
> >
> > I've hacked my ser.cfg but can someone comment on why I would be
> > recieving a "200 OK" with a
> >
> > The change I made to my onreply_route is below. The only thing I can
> > see about these messages versus others is that the CSeq says "CSeq:
> > {some digits} BYE" with "Content-Length: 0".
> >
> > So for these messages I'm just not calling force_rtp_proxy().
> >
> > I don't know if this is a symptom of my Grandstream BT100 only of if
> > other ATAs or IP phones do this.
> >
> > Regards,
> > Paul
> >
> > onreply_route[1] {
> >
> >
> > if (isflagset(2) && status =~ "(183)|2[0-9][0-9]") {
> >
> >
> > fix_nated_contact();
> >
> >
> > if (!(search("^Content-Length:\ 0")) {
> > force_rtp_proxy();
> > };
> >
> >
> > } else if (nat_uac_test("1")) {
> >
> >
> > fix_nated_contact();
> > };
> > }
> >
> >
> > --- Java Rockx <javarockx(a)yahoo.com> wrote:
> >
> >> Hi all.
> >>
> >> I've got nathelper and rtpproxy working very well with my firewall.
> >> However I do still recieve these messages in my syslog. I am only
> >> catching 183 and 2xx errors in my onreply_route so I'm very
> >> confused how to prevent these errors.
> >>
> >> I'm using ser-0.8.99-dev12. Can anyone give me some advise?
> >> Cheers,
> >> Paul
> >>
> >> NOTE: The SIP message that caused these errors is at the bottom of
> >> this message.
> >>
> >> 0(27011) ERROR: extract_body: message body has length zero
> >> 0(27011) ERROR: force_rtp_proxy2: can't extract body from the
> >> message 0(27011) ERROR: on_reply processing failed
> >>
> >> My onreply_route is here:
> >>
> >> onreply_route[1] {
> >>
> >>
> >>
> >> if (isflagset(2) && status =~ "(183)|2[0-9][0-9]") {
> >>
> >>
> >>
> >> fix_nated_contact();
> >>
> >>
> >>
> >> xlog("L_ERR", "%mb");
> >> force_rtp_proxy();
> >>
> >>
> >>
> >> } else if (nat_uac_test("1")) {
> >>
> >>
> >>
> >> fix_nated_contact();
> >> };
> >> }
> >>
> >>
> >>
> >> 0(27011) SIP/2.0 200 OK
> >> Via: SIP/2.0/UDP 67.184.42.101;branch=z9hG4bKe7ac.628388e2.0
> >> Via: SIP/2.0/UDP
> >> 192.168.0.83;rport=5060;received=68.184.42.171;branch=z9hG4bKb4bfc6084f306712
> >> From:
> >> <sip:9990010001@sip.mycompany.com;user=phone>;tag=0d5452e4bee210e3
> >> To: "Andrew"
> >> <sip:9990010002@sip.mycompany.com;user=phone>;tag=b73c75ac247b63db
> >> Call-ID: 0d5bc1fb337cd7eb(a)68.184.40.199 CSeq: 26976 BYE User-Agent:
> >> Grandstream BT100 1.0.5.11 Contact:
> >> <sip:9990010002@68.184.40.199;user=phone> Allow:
> >> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
> >> Content-Length: 0
> >>
> >>
> >>
> >>
> >>
> >> __________________________________
> >> Do you Yahoo!?
> >> The all-new My Yahoo! - Get yours free!
> >> http://my.yahoo.com
> >>
> >>
> >> _______________________________________________
> >> Serusers mailing list
> >> serusers(a)lists.iptel.org
> >> http://lists.iptel.org/mailman/listinfo/serusers
> >>
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > The all-new My Yahoo! - Get yours free!
> > http://my.yahoo.com
>
>
__________________________________
Do you Yahoo!?
Meet the all-new My Yahoo! - Try it today!
http://my.yahoo.com
This is not really the best way to handle unavailable redirection to VM
in asterisk. Also, for SIP only it's not a good idea to use the 'r'
parameter as this causes asterisk to generate a ringing signal even if
the endpoint is not responding. Please check the asterisk mailing list
or check the voip-info.org wiki on asterisk sip.conf.
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Ahmed Boreau
Sent: 18 November 2004 14:23
To: yair(a)hakak.com; blairs(a)isc.upenn.edu
Cc: serusers(a)fox.iptel.org
Subject: Re: [Serusers] Ser + Asterisk
Thanks for you help.
below are what I did in * extensions.conf
[globals]
SERADDRESS=@IP ser server:5060
[serserver]
exten => _3XXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)
exten => _3XXX,2,WaitMusicOnHold,15
exten => _3XXX,3,VoicemailMain(u${EXTEN:2})
In sip.conf, I just add
[general]
autocreatepeer=yes
Phones are ringing but I could not get voicemail.
Thanks in advance
At 13:36 18/11/2004, Yair Hakak wrote:
>hello,
> set autocreatepeer=yes in sip.conf and you should be fine.
> obviously you need to make sure that things are set up properly in
>extensions.conf to connect calls properly.
>
>
>On Thu, 18 Nov 2004 13:32:26 +0000, Ahmed Boreau <ahmed.boreau(a)esmt.sn>
wrote:
> > Hi,
> >
> > I need help. I'm actually trying to set up ser+asterisk which are
actually
> > working separately.
> > By now, I want to let asterisk receiving ser calls.
> >
> > I add these commands into ser.cfg
> >
> > if(method=="INVITE"){
> > if (uri=~"^sip:1[0-9]{10}@*") {
> > log(1,"Forwarding to Asterisk\n");
> > rewritehostport("10.0.0.13: 5061");
> > t_relay();
> > break;
> > }
> > }
> >
> > What could I need to do into sip.conf at * side.
> >
> > Thanks in advance
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi guys,
I have some problems with ser. When trying to call from internal ip (nat) I can hear
user that is behind nat but he can't hear me. Same problem exist when to users behind
nat call each other. The main problem comes when I forward calls - all calls starting
with 1-9 are forwarded to pstn provider (quintum gw). Whatever - internal or external
ip is used I cannot hear user at the other side but he hears me!
Here is my configuration:
debug=9 # debug level (cmd line: -dddddddddd)
#fork=yes
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=yes # (cmd. line: -r)
rev_dns=yes # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
fifo_mode=0777
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
#loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.15/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
modparam("acc", "db_url", "mysql://ser:heslo@192.168.2.15/ser")
modparam("acc", "log_level", 2)
modparam("acc", "log_flag", 3)
modparam("acc", "log_level", 1)
# number of flag, which will be used for accounting; if a message is
# labeled with this flag, its completion status will be reported
modparam("acc", "log_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "db_url", "mysql://ser:heslo@192.168.2.15/ser")
modparam("acc", "db_flag", 1)
modparam("acc", "log_missed_flag", 1)
modparam("msilo", "db_url", "mysql://ser:heslo@80.72.68.187/ser")
modparam("msilo", "db_table", "silo")
#modparam("msilo", "registrar", "sip:registrar@iptel.org")
modparam("msilo", "expire_time", 259200)
modparam("msilo", "check_time", 10)
#modparam("msilo", "clean_period", "3")
modparam("msilo", "use_contact", 1)
#modparam("rtpproxy", "rtpproxy_socket", "/var/run/rtpproxy.sock")
#modparam("nathelper", "rtpproxy_socket", "/var/run/rtpproxy.sock")
modparam("nathelper","rtpproxy_sock", "/var/run/rtpproxy.sock")
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1)
listen=83.74.45.87
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (!mf_process_maxfwd_header("70")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (nat_uac_test("3")) {
# Allow RR-ed requests, as there may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || ! search("^Record-Route:")) {
log("LOG: Someone trying to register from private IP, rewriting\n");
fix_nated_contact(); # Rewrite contact with source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("83.74.45.87", "subscriber")) {
www_challenge("83.74.45.87", "0");
break;
};
force_rtp_proxy();
save("aliases");
save("location");
if (m_dump())
{
log("MSILO: offline messages dumped - if they were\n");
}else{
log("MSILO: no offline messages dumped\n");
};
break;
};
};
if (uri=~"^sip:[1-9]*@83.74.45.87") {
rewritehost("122.44.75.176");
forward( 122.44.75.176, 5060 );
break;
}
setflag(1);
lookup("aliases");
if (uri==myself) {
if (method=="INVITE") {
record_route();
if (isflagset(6)) {
force_rtp_proxy();
};
};
};
if(!lookup("location"))
{
if (! t_newtran())
{
sl_reply_error();
break;
};
if (!method=="MESSAGE")
{
if (!t_reply("404", "Not found"))
{
sl_reply_error();
};
break;
};
log("MESSAGE received -> storing using MSILO\n");
if (m_store("0"))
{
log("MSILO: offline message stored\n");
if (!t_reply("202", "Accepted"))
{
sl_reply_error();
};
}else{
log("MSILO: offline message NOT stored\n");
if (!t_reply("503", "Service Unavailable"))
{
sl_reply_error();
};
};
break;
};
if (!t_relay()) {
sl_reply_error();
};
}
#route[1] {
# if (!t_relay()) {
# sl_reply_error();
# };
#}
route[1] {
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
break;
};
if (isflagset(6)) {
force_rtp_proxy();
t_on_reply("1");
append_hf("P-Behind-NAT: Yes\r\n");
};
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
break;
};
}
onreply_route[1] {
# NATed transaction ?
if (status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time or request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method=="MESSAGE")
{
break;
};
log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n");
# we have changed the R-URI with the contact address, ignore it now
if (m_store("1"))
{
log("MSILO: offline message stored\n");
t_reply("202", "Accepted");
}else{
log("MSILO: offline message NOT stored\n");
t_reply("503", "Service Unavailable");
};
}
Thanks in advance :)
Pavel
