see doc http://www.iptel.org/ser/doc/seruser/seruser.html#FIFOSERVER and
some development info is at http://iptel.org/~faqomatic/fom-serve/cache/23.html.
As Klaus responded, the primary issue is that your SEMS part was not running at all.
-jiri
At 05:28 PM 11/12/2003, Harry Behrens wrote:
>Hi Klaus,
>
>is there any documenation - save of course the source - with regards to this inter-FIFO communication protocol/mechanism?
>
>All I´ve understood so far is, that it seems one pushes SIP/SDP "messages" into the FIFO (s.th. like a named pipe/UNIX socket?).
>
>Also, I have seen scripts which seem to indicate that there is something like "application escapes", where SER functionality can be addressed via s.th like /^:name_of_function/)
>
>Can anybody point to some technical spec/description of this mechanism and how to configure/script it?
>
>Thanx,
>
> Harry
>
>Dr. Harry Behrens
>Projektleitung BIB3R
>DAI Labor - Technische Universität Berlin
>Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin
>
>Fon: +49 30 314 23383
>Fax: +49 30 314 21799
>
>Email: harry.behrens(a)dai-labor.de
>
>http://www.dai-labor.de
>
>
>
>> -----Ursprüngliche Nachricht-----
>> Von: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
>> Gesendet: Mittwoch, 12. November 2003 16:13
>> An: sesha(a)iic.com; serusers(a)lists.iptel.org
>> Betreff: RE: [Serusers] Voicemail
>>
>>
>> Have you installed sems? Have you started sems?
>> (http://developer.berlios.de/projects/sems)
>>
>> This is the voicemail programm for ser. You have to install
>> it on the same machine where ser is running and they will
>> communicate using a FIFO.
>>
>> regards,
>> Klaus
>>
>> > -----Original Message-----
>> > From: Sesha B [mailto:sesha@iic.com]
>> > Sent: Wednesday, November 12, 2003 4:11 PM
>> > To: Klaus Darilion; serusers(a)lists.iptel.org
>> > Subject: RE: [Serusers] Voicemail
>> >
>> >
>> > I'm getting this error from debugs:
>> >
>> > 500 couldn not contact announcement server.
>> >
>> > Please let me know what it means? Is it possible to put the
>> > voicemail on the
>> > PBX? I have the voicemail configured for this number on the
>> > PBX. Thank you.
>> >
>> >
>> >
>> > -----Original Message-----
>> > From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
>> > Sent: Wednesday, November 12, 2003 9:42 AM
>> > To: sesha(a)iic.com; serusers(a)lists.iptel.org
>> > Subject: RE: [Serusers] Voicemail
>> >
>> >
>> > Usually, if the phone (user) is registered, the call will
>> be forwarded
>> > to the SIP phone instead of forwarding it to the voicemail.
>> If a want
>> > to redirect the call to the voicemail after several seconds
>> > (e.g. if no one
>> > picks up the phone), you have to use the fr_inv_timer timer
>> and set up
>> > script as shown in:
>> > http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING
>> >
>> > regards,
>> > Klaus
>> >
>> > > -----Original Message-----
>> > > From: Sesha B [mailto:sesha@iic.com]
>> > > Sent: Wednesday, November 12, 2003 2:44 PM
>> > > To: Jiri Kuthan; serusers(a)lists.iptel.org
>> > > Subject: RE: [Serusers] Voicemail
>> > >
>> > >
>> > > Hi,
>> > >
>> > > I'm trying to implement voicemail. When a call is made to a SIP
>> > > phone, and if the SIP phone is ringing, the "ngrep port
>> 5060" on the
>> > > SER shows that it
>> > > is connected. So, it is looking it as a successful connection
>> > > and hence it
>> > > is not going to the answering machine. Is there a roundabout
>> > > solution for
>> > > this? Please let me know. Thank you.
>> > >
>> > > _______________________________________________
>> > > 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
>>
>>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
After some hassles I succesfully installed pdt module.
Now when I compose 22220 before a number the uri is rewritten with
"iptel.org", here is a transcript:
11(14926) SIP Request:
11(14926) method: <INVITE>
11(14926) uri: <sip:222220555555@sferica.net>
11(14926) version: <SIP/2.0>
11(14926) parse_headers: flags=1
11(14926) Found param type 235, <rport> = <n/a>; state=6
11(14926) Found param type 232, <branch> = <z9hG4bK6D151D75538C4C19987ADC66F1B9505F>; state=16
11(14926) end of header reached, state=5
11(14926) parse_headers: Via found, flags=1
11(14926) parse_headers: this is the first via
11(14926) After parse_msg...
11(14926) preparing to run routing scripts...
11(14926) DEBUG : is_maxfwd_present: searching for max_forwards header
11(14926) parse_headers: flags=128
11(14926) end of header reached, state=9
11(14926) DEBUG: get_hdr_field: <To> [32]; uri=[sip:222220555555@sferica.net]
11(14926) DEBUG: to body [<sip:222220555555@sferica.net>
]
11(14926) get_hdr_field: cseq <CSeq>: <34431> <INVITE>
11(14926) DEBUG: is_maxfwd_present: value = 70
11(14926) PDT: update_new_uri: sip:555555@iptel.org
but, in the software client what I get is a "403 No relay" error.
I'cant understand if the error is coming from my server, or from
iptel.org ....
Here is my config, any help will be appreciated, tnx !
debug=7
fork=yes
log_stderror=yes
alias=sferica.net
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/pdt.so"
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
modparam("pdt", "db_url", "sql://ser:heslo@localhost/pdt")
modparam("pdt", "db_table", "domains")
modparam("pdt", "prefix", "2")
modparam("pdt", "start_range", 2000)
modparam("pdt", "hsize_2pow", 2)
# ------------------------- 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 (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
prefix2domain();
# 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
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
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("sferica.net", "subscriber")) {
www_challenge("sferica.net", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("402", "Offline");
break;
# voicemail forward
#rewritehostport("interconnessioni.it:5090");
#t_relay_to_udp("interconnessioni.it", "5090");
#break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
--
Best regards,
Alessio mailto:alessiof@interconnessioni.it
Hi,
can someone send me a pdt.so module binary (I use RedHat 9)?
I'm not so familiar with compiling ... I fear for my installation that
by now is running properly.
Tnx
--
Best regards,
Alessio mailto:alessiof@interconnessioni.it
Hi
I have compiled and configured SER with FreeRadius
I can see digest auth working with radclient test but when I m trying to use any SIPUA , I m not getting thru with the registration of the user
I have attached all the relevant files for your ready reference..no changes made in the file..I m using the same files
Please let me know what is wrong and where, I ll appreciate your help in fixing my problem.
TIA,
Madan
No, it's only to upload a wav file as personal greeting message for the
voicemail system. And as the name says ("voicemail"), the recorded
messages will be sent to you by email. The messages aren't stored at the
voicemail server.
regards,
Klaus
> -----Original Message-----
> From: Samy Touati [mailto:samy@tunix.com]
> Sent: Wednesday, November 12, 2003 6:47 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] voicemail working finally
>
>
> Hi,
>
> I finally got the vm working, thanks for the help.
>
> In serweb there's a dial voicemail, what doesn it do ? Doest it
> call the vm and let you retrieve your messages ?
>
>
> Thanks.
>
> Samy.
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Have you installed sems? Have you started sems?
(http://developer.berlios.de/projects/sems)
This is the voicemail programm for ser. You have to install it on the
same machine where ser is running and they will communicate using a
FIFO.
regards,
Klaus
> -----Original Message-----
> From: Sesha B [mailto:sesha@iic.com]
> Sent: Wednesday, November 12, 2003 4:11 PM
> To: Klaus Darilion; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Voicemail
>
>
> I'm getting this error from debugs:
>
> 500 couldn not contact announcement server.
>
> Please let me know what it means? Is it possible to put the
> voicemail on the
> PBX? I have the voicemail configured for this number on the
> PBX. Thank you.
>
>
>
> -----Original Message-----
> From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
> Sent: Wednesday, November 12, 2003 9:42 AM
> To: sesha(a)iic.com; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Voicemail
>
>
> Usually, if the phone (user) is registered, the call will be forwarded
> to the SIP phone instead of forwarding it to the voicemail.
> If a want to
> redirect the call to the voicemail after several seconds
> (e.g. if no one
> picks up the phone), you have to use the fr_inv_timer timer and set up
> script as shown in:
> http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING
>
> regards,
> Klaus
>
> > -----Original Message-----
> > From: Sesha B [mailto:sesha@iic.com]
> > Sent: Wednesday, November 12, 2003 2:44 PM
> > To: Jiri Kuthan; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Voicemail
> >
> >
> > Hi,
> >
> > I'm trying to implement voicemail. When a call is made to a
> > SIP phone, and
> > if the SIP phone is ringing, the "ngrep port 5060" on the SER
> > shows that it
> > is connected. So, it is looking it as a successful connection
> > and hence it
> > is not going to the answering machine. Is there a roundabout
> > solution for
> > this? Please let me know. Thank you.
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
>
>
Hi Klaus,
is there any documenation - save of course the source - with regards to this inter-FIFO communication protocol/mechanism?
All I´ve understood so far is, that it seems one pushes SIP/SDP "messages" into the FIFO (s.th. like a named pipe/UNIX socket?).
Also, I have seen scripts which seem to indicate that there is something like "application escapes", where SER functionality can be addressed via s.th like /^:name_of_function/)
Can anybody point to some technical spec/description of this mechanism and how to configure/script it?
Thanx,
Harry
Dr. Harry Behrens
Projektleitung BIB3R
DAI Labor - Technische Universität Berlin
Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin
Fon: +49 30 314 23383
Fax: +49 30 314 21799
Email: harry.behrens(a)dai-labor.de
http://www.dai-labor.de
> -----Ursprüngliche Nachricht-----
> Von: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
> Gesendet: Mittwoch, 12. November 2003 16:13
> An: sesha(a)iic.com; serusers(a)lists.iptel.org
> Betreff: RE: [Serusers] Voicemail
>
>
> Have you installed sems? Have you started sems?
> (http://developer.berlios.de/projects/sems)
>
> This is the voicemail programm for ser. You have to install
> it on the same machine where ser is running and they will
> communicate using a FIFO.
>
> regards,
> Klaus
>
> > -----Original Message-----
> > From: Sesha B [mailto:sesha@iic.com]
> > Sent: Wednesday, November 12, 2003 4:11 PM
> > To: Klaus Darilion; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Voicemail
> >
> >
> > I'm getting this error from debugs:
> >
> > 500 couldn not contact announcement server.
> >
> > Please let me know what it means? Is it possible to put the
> > voicemail on the
> > PBX? I have the voicemail configured for this number on the
> > PBX. Thank you.
> >
> >
> >
> > -----Original Message-----
> > From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
> > Sent: Wednesday, November 12, 2003 9:42 AM
> > To: sesha(a)iic.com; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Voicemail
> >
> >
> > Usually, if the phone (user) is registered, the call will
> be forwarded
> > to the SIP phone instead of forwarding it to the voicemail.
> If a want
> > to redirect the call to the voicemail after several seconds
> > (e.g. if no one
> > picks up the phone), you have to use the fr_inv_timer timer
> and set up
> > script as shown in:
> > http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING
> >
> > regards,
> > Klaus
> >
> > > -----Original Message-----
> > > From: Sesha B [mailto:sesha@iic.com]
> > > Sent: Wednesday, November 12, 2003 2:44 PM
> > > To: Jiri Kuthan; serusers(a)lists.iptel.org
> > > Subject: RE: [Serusers] Voicemail
> > >
> > >
> > > Hi,
> > >
> > > I'm trying to implement voicemail. When a call is made to a SIP
> > > phone, and if the SIP phone is ringing, the "ngrep port
> 5060" on the
> > > SER shows that it
> > > is connected. So, it is looking it as a successful connection
> > > and hence it
> > > is not going to the answering machine. Is there a roundabout
> > > solution for
> > > this? Please let me know. Thank you.
> > >
> > > _______________________________________________
> > > 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
>
>
