Sirs,
I a new user for sip server, I've install the sip server but I start
service I've a error for load database module
I need help for this problem!.
regards,
Ivan
Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??
Thnaksm
Aisling.
---- Original Message ----
From: ashling.odriscoll(a)cit.ie
To: asterisk-users(a)lists.digium.com
Subject: FW: SER Asterisk Voicemail
Date: Thu, 10 Feb 2005 16:45:53 -0000
Hi all,
I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message.
Now, this may seem a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java Rockx using
sipsak for sending mwi sip notify messages to the phone but is there
a simpler way which I am blindly ignoring??
Thank you in advance,
Aisling.
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Someone has any idea about this one?
The device making the call is a Linksys PAP2 behind a firewall.
Ricardo Javier Martinez Ogalde
> -----Mensaje original-----
> De: Ricardo Martinez [mailto:rmartinez@redvoiss.net]
> Enviado el: Miércoles, 09 de Febrero de 2005 16:18
> Para: 'serusers(a)lists.iptel.org'
> Asunto: [Serusers] Hold Status in Mediaproxy session.
>
>
> Hello List.
> A quick question for mediaproxy users: What does it
> means the HOLD
> status?
> I see this when i execute the ./sessions command :
> Caller Via Called Status
> Duration Codec Type Traffic
> --------------------------------------------------------------
> --------------
> --------------------------------------------
> 200.xx.xx.x:50179 - xx.xx.xx.35:35050 - xx.xx.xx.66:23240 hold (558)
> 9'48" G723 Audio 49.50k/99.94k/51.34k
>
> The call is already over, but the mediaproxy still shows the
> hold status.
> Any ideas?
>
> Thanks in advance,
>
> Regards
>
> Ricardo Martinez
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
I'm trying to fake a BYE message to a UA to make it drop a call. I input
the following:
serctl fifo t_uac_dlg BYE "sip:116@xxx.xxx.55.252:5060"
"From:82616925@xxx.xxx.131.216" "To:116@xxx.xxx.55.252" "CSEQ: 2 BYE"
but it returns with :
400 fifo_uac: next hop uri invalid
can anyone tell me what's wrong, or give me a good example of how to
properly do this?
Thank you.
I'm considering implementing SER in a production environment where at
least 80 people are using SIP phones to register to and dial through
SER to FXO gateways to analog lines. These 80 people will be on the
phone for the full work day.
I was wondering if anyone could give me any first-hand accounts of
using SER in production, and if possible, could you tell me how many
calls you are making in a day, how many concurrent users, how are you
backing up (I assume just the ser.cfg file and MySQL database is
sufficient), what is the uptime, stability, and reliability like?
I've got a test box up, but I would like to hear from others using it
before I roll it out into my production environment.
Thanks in advance!
--
Dana
Hello,
I have this problem :
>service ser status
>ser dead but subsys locked
Does, anyone know, how can I resolve this problem ??
Best Regards
Nicolas RUIZ
FRANCE, Paris
Hello,
I have a litte problem with the SER service's.
The service SER die !!!!!
I restart it by : service ser start on fedora system, and it's good.
But, in one day or 4 hours, the services is another time die. Why it die ???
Best regards
Nicolas RUIZ
FRANCE, Paris
Vos Solutions Voix-Data !
STC
Service Technique Clients
Téléphone : 0811 02 60 61
Fax :+ 33 (0) 1 47 24 74 77
stc(a)vivaction.com
Immeuble Plein Ouest
177 av. Georges Clemenceau
92024 Nanterre - France
Tel : 0 811 02 6000
www.vivaction.com
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Unclassified.
On Oct 26, 2004 at 06:02, Java Rockx <javarockx(a)yahoo.com> wrote:
> Hi All.
>
> Below is a snapshot of "serctl moni" and as you can see it shows 1
waiting
> message. Can anyone tell me what would keep these messages in a
waiting state?
Each transaction will live a few more seconds after finishing (see the
rfc). By default it will live for 5 more seconds. If you want to
change
that use the wt_timer tm param.
> I think I must have an error in my ser.cfg file because call waiting
doesn't
> work properly.
>
> How can I prevent messages from accumulating in a waiting status? Is
there a
> timeout to /dev/null them?
Yes, they will be deleted automatically after "wt_timer" seconds.
It doesn't have anything to do with call waiting.
Andrei
>>> "Chris" <ser(a)cannes.f9.co.uk> 02/11/05 08:15PM >>>
I had it once (350 waiting) when I was running on a very old slow PII
machine
and I had a UA trying to re-register continuously after failure.
use ethereal to see what is coming into to your ser.
Mine was fixed by stopping continuous retries without gap
and making register succeed :))
Chris
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
Behalf Of Ricardo Martinez
Sent: 11 February 2005 20:06
To: 'serusers(a)lists.iptel.org'
Subject: RE: [Serusers] serctl moni question.
Hello again.
Have anyone idea about this issue?
Thanks!
Ricardo Martinez.-
> -----Mensaje original-----
> De: Ricardo Martinez [mailto:rmartinez@redvoiss.net]
> Enviado el: Jueves, 10 de Febrero de 2005 16:29
> Para: 'serusers(a)lists.iptel.org'
> Asunto: [Serusers] serctl moni question.
>
>
> Hello list.
> I used the "serctl moni" command and i have found these
> statistics :
>
>
> Transaction Statistics
> Current: 0 (40 waiting) Total: 584 (0 local)
> Replied localy: 1082
> Completion status 6xx: 27, 5xx: 7, 4xx: 212, 3xx: 0,2xx: 356
>
> I'm particular interested in the Transactions Statics -
> Current and the 40
> Waiting.
> Does this means something?
>
> Thanks in advance.
>
> Ricardo Martinez
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
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Using the Jabber gateway: what happens when you SUSBCRIBE to a conference
room? Can anybody tell me how does the jabber server then?
Thank you
Sandra García Espada
DMR Consulting
Hi,
After studying some ser.cfg files, I could route a SIP call to PSTN gateway successfully.
If ser receives "866+any PSTN num", ser will forward to T1 trunk gateway. T1 trunkgateway will drop the prefix 866 and route the call out to PSTN line.
i.e. SIP client (866+PSTN num) ---> T1 Trunk Gateway (drop prefix 866) ---> PSTN call (PSTN num)
SER version: ser-0.8.14 or ser-0.9.
rtpproxy version: latest cvs version from berlios
I have posted a ser.cfg in this email.
Best Regards,
Thomas
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=""
port=5060
#children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
# ------------------ 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/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- Nathelper params --
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# Nathelper
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these 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");
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority is
# smart enough to be symmetric. In some phones it takes a configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is
# called "symmetric media" and "symmetric signalling".
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
};
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
if (uri=~"^sip:866*") {
log(1, "going to PSTN route\n");
route(2);
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
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 client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
break;
};
}
route[2] {
force_rtp_proxy();
record_route();
t_on_reply("1");
t_relay_to_udp("T1 gateway IP","T1 Gateway UDP port");
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
# Not all 2xx messages have a content body so here we make sure
# out Content-Length > 0 to avoid a parse error
if (!search("^Content-Length:\0")) {
force_rtp_proxy();
};
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
