Hi,
I got a problem with ser when it's been idle for a while. Any request
sent to it is answered with
SIP/2.0 500 I'm terribly sorry, server error occured (7/TM)..Via
Syslog says this:
Sep 10 16:18:41 gurke ser[65803]: ERROR: get_out_socket: no socket found
Sep 10 16:18:41 gurke ser[65803]: ERROR: add_uac: can't fwd to af 2,
proto 1 (no corresponding listening socket)
Sep 10 16:18:41 gurke ser[65803]: ERROR: t_forward_nonack: failure to
add branches
Sep 10 16:18:58 gurke ser[65809]: BUG: tcp_main_loop: dead child 1
Ser runs on freebsd4.8 here. Any suggestions what could be the reason
for this? Why is it complaining about problems with tcp? I thought that
sip is udp-based (though I know that latest version of ser supports tcp
connections also)?
regards,
felix
--
PGP-Key: http://belugalounge.net/~felix/Felix%20Schmid.asc
I'm sure this is pretty straightforward to some of you, but I could use
some help with a few issues if someone has the time. Below is my config
file. This is running on RedHat 9.
I am having numerous problems.
1) Having issues with correctly increasing the number of children. If I
increase it, it fails because it can't connect with mysql. I have tried
increasing the number of connections in mysql in the my.cnf file, but it
seems to cause even more problems.
2) I only want the call to fail to route 2 for calls that terminate in
my network. I don't want calls leaving the network to try to go to
voicemail after the time hits.
3) I can't seem to figure out how to format the rewrite/append/etc to
trigger the voicemail call. Maybe I don't understand the routing loop.
Can someone take a look at the config and give me some ideas?
4) The append to the null/parallel destination that allows the trigger
to activate I don't understand at all. Why does it need to be there?
Where should it be pointed to?
5) I'm not seeing the activity/routing logs going anywhere. If I run
ser manually, I see information on the stderr, but nothing in the
database or logs in terms of the connections.
I would really appreciate any help. I have other questions that aren't
related to the configuration file, so I will save those to amuse people
later.
Thanks,
Stephen
debug = 7
log_stderror=yes
check_via=no
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
children=4
# ------------------ module loading ----------------------------------
# modules not listed here for brevity
modparam("tm", "fr_inv_timer", 15 )
modparam("tm", "fr_timer", 10 )
modparam("usrloc", "db_mode", 2)
#modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("voicemail", "db_url","sql://ser:heslo@localhost/ser")
modparam("rr", "enable_full_lr", 1)
# -- acc params --
modparam("acc", "report_ack", 1)
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1 )
modparam("acc", "db_flag", 1 )
modparam("acc", "log_missed_flag", 3 )
modparam("acc", "db_missed_flag", 3 )
# report to syslog: From, i-uri, status, digest id
modparam("acc", "log_fmt", "fisu" )
# ------------------------- request routing logic -------------------
alias="209.242.10.153"
alias="test.net"
route{
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Wow -- Message too large");
break;
};
if (loose_route()) { t_relay(); break; };
if (method=="INVITE") {record_route();};
# account completed transactions via syslog
setflag(1);
if (src_ip==66.155.138.5) {
log("gateway-originated request");
} else {
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("test.net",
"subscriber")) {
www_challenge("test.net", "0");
break;
};
save("location");
break;
};
};
};
if (uri=~"sip:voicemail\+@") {
t_newtran();
t_reply("100", "trying -- just a second");
if (!vm("/tmp/am_fifo","announcement")) {
t_reply("500", "SEMS error");
};
break;
};
if (uri=~"sip:2[0-9]+@.*") { # JUST FOR TESTING USING 2 AS LOCAL
PREFIX
if (!lookup("location")) {
if (method=="INVITE" || method=="ACK") {
t_newtran();
t_reply("100", "trying -- just a second");
if(!vm("/tmp/am_fifo","voicemail")){
t_reply("500", "SEMS error");
};
break;
};
break;
};
# WHAT SHOULD THIS APPEND BRANCH ACTUALLY POINT TO?
# WITHOUT IT, THE CALL GOES DIRECTLY TO VOICEMAIL
append_branch("sip:stall@209.242.10.153:9");
} else {
rewritehostport("66.155.138.5:5060");
};
t_on_failure("1");
if (!t_relay()) {
sl_reply_error();
break;
};
}
failure_route[1]{
log(1,"call sent to voicemail due to no answer\n");
if (uri=~"sip:2[0-9]+@.*") {
if (method=="INVITE" || method=="ACK") {
append_branch("sip:voicemail@209.242.10.153");
t_relay();
};
break;
};
}
At 10:26 PM 9/9/2003, Steve Dolloff wrote:
>This works great if the user is offline. Is there anyway for voicemail
>to pick up after a certain time period or after a certain number of
>rings?
Roughly yes. The closest example is at
http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING
There is no number of rings in SIP -- there is the invite_fr timeout.
-Jiri
hi,
I'm trying to use www_authorize/www_challenge (ser 0.8.11/solaris)
together with my cisco 7960 (auth backend is mysql and calculate_ha1=1).
when I reach www_authorize (with, as I believe, proper authorization) I
get the following error message:
pre_auth(): Credentials received are not filled properly
sent to client:
WWW-Authenticate: Digest realm="zettou.net",
nonce="3f5e52fd5db948e3b493525f89fe24bfd0f86d9e", qop="auth"
sent from client:
Authorization: Digest username="jakob(a)schlyter.net",
realm="zettou.net",uri="sip:schlyter.net",
response="8503ab9b93e73ef0033d0e265f31dd0f",
nonce="3f5e52fd5db948e3b493525f89fe24bfd0f86d9e",
cnonce="718436df",qop=auth,nc=00000001,algorithm=md5
what have I done wrong?
jakob
Wouter,
If you want to try with an opensource SIP-H.323 Gateway, there is Vovida's
siph323csgw. It is not the most stable software in the world and not well
maintained. However, it makes the trick to call from H.323 to SIP clients (we
tried Netmeeting and MSN) and viceversa (you will need to restart it several
times, though).
There are commercial implementations, like RADVISION, that might help you much
more on this.
Jaime
Bogdan-Andrei IANCU <iancu(a)fokus.fraunhofer.de> on 09/09/2003 14:44:22
Please respond to iancu(a)fokus.fraunhofer.de
To: Wouter Prins <wouter.prins(a)qi.nl>
cc: "'serusers(a)lists.iptel.org'" <serusers(a)lists.iptel.org> (bcc: Jaime GIL/EN/HTLUK)
Subject: Re: [Serusers] SER in combination with a H.323 GK
We do not support such features.
To have some H.323 <-> SIP interpretability, you would need a SIP-H.323
gateway, but I not aware of any implementation (excepting the hardware
version built by net.com)
Regards,
Bogdan
Wouter Prins wrote:
>Hello,
>
>Is it somehow possible to let SER communicate with a H.323 gatekeeper like
>openh323's gnugk or some other gatekeeper software?
>I have been searching for documentation but couldnt find much, perhaps
>someone can point me in the right direction?
>
>Thanks in advance!
>
>Kind regards,
>
>Wouter Prins
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
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http://lists.iptel.org/mailman/listinfo/serusers
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Hi,
Can someone help me with setting up a SIP server to forward the calls to a
SIP phone? I'm trying to install it on a Slackware Linux box. What is the
additional configuration that I need to do? I'm new to this. Can someone
plzz help me? Thank you very much.
Sesha
Hi,
We(xten-india),tried to configure RADIUS with SER-0.8.11.
We have freeradius-0.9.0 server and radiusclient.
In the SER.cfg, we have radius_www_authorize method.But,when we
connect the Kphone(UA) with SER none of the username-password get
authenticated with RADIUS.
Please explain us, From which database or file the username-password
will be taken for authentication in RADIUS.
Please help us.
Thanks in Advance
Regards
Xten_india
---------------------------------
Do you Yahoo!?
Yahoo! SiteBuilder - Free, easy-to-use web site design software
This works great if the user is offline. Is there anyway for voicemail
to pick up after a certain time period or after a certain number of
rings?
Thanks,
Stephen
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Tuesday, September 09, 2003 1:15 PM
To: Steve Dolloff; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Implementing Voicemail
The key fragments of SER script with voicemail support are as follows:
if (uri==myself) { # vm only for my domain ...
/* if this is an in-dialog request from voicemail, jump out of
proxy mode and pass the request to voicemail; you need to have
'user_prefix_separator=+' in sems.conf
*/
if (uri=~"sip:voicemail\+@") {
/* some transaction processing to absorb
retransmissions;
the key part is "vm", which passes requests to the
SEMS machine (we should better integrate all this
stuff
in a single command someday, sorry scriptwriters
*/
t_newtran();
t_reply("100", "trying -- just a second");
if (!vm("/tmp/am_fifo","announcement")) {
t_reply("500", "SEMS error");
};
break;
};
if (!lookup()) { # user off-line?
if (method=="INVITE" || method=="ACK") {
t_newtran();
t_reply("100", "trying -- just a second");
if (!vm("/tmp/am_fifo","announcement")) {
t_reply("500", "SEMS error");
};
break;
};
# non-VoIP methods for off-line users will just
# just return 404
sl_send_reply("404", "Not Found");
break;
};
# continue with online users.....
That's it, very briefly.
You can improving by handling users who are on-line but whose
requests fail downstream (no reply timeout,busy,etc.)
If all is set up properly, the recepient will receive the message
by email/
-Jiri
At 06:37 PM 9/9/2003, Steve Dolloff wrote:
>I have the sems installed and it works great. Does anyone have a
config
>that they could share that would illustrate how to use the vm in normal
>use? I'm sending all off-net calls to a sip gateway. I would want
>calls to go to vm after x rings.
>
>Thanks,
>
>Stephen
>
>-----Original Message-----
>From: Steve Dolloff
>Sent: Tuesday, September 09, 2003 9:48 AM
>To: serusers(a)lists.iptel.org
>Subject: [Serusers] Implementing Voicemail
>
>I was looking at an old thread regarding this.
>http://lists.iptel.org/pipermail/serusers/2003-July/002003.html
>
>1) How do I know when to direct it to this portion of the code? After
x
>rings?
>
>2) Am I mistaken in thinking that there is a voicemail system built in?
>Do I need to install a separate voicemail system? If not, how do I
>configure the voice mail system? If so, can anyone recommend one?
>
>Thanks,
>
>Stephen
>
>_______________________________________________
>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
--
Jiri Kuthan http://iptel.org/~jiri/
Hello,
Is it somehow possible to let SER communicate with a H.323 gatekeeper like
openh323's gnugk or some other gatekeeper software?
I have been searching for documentation but couldnt find much, perhaps
someone can point me in the right direction?
Thanks in advance!
Kind regards,
Wouter Prins
I have the sems installed and it works great. Does anyone have a config
that they could share that would illustrate how to use the vm in normal
use? I'm sending all off-net calls to a sip gateway. I would want
calls to go to vm after x rings.
Thanks,
Stephen
-----Original Message-----
From: Steve Dolloff
Sent: Tuesday, September 09, 2003 9:48 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Implementing Voicemail
I was looking at an old thread regarding this.
http://lists.iptel.org/pipermail/serusers/2003-July/002003.html
1) How do I know when to direct it to this portion of the code? After x
rings?
2) Am I mistaken in thinking that there is a voicemail system built in?
Do I need to install a separate voicemail system? If not, how do I
configure the voice mail system? If so, can anyone recommend one?
Thanks,
Stephen
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
