Hi,
Mahi Haile wrote:
Hi all,
I am a grad student in CS, and has been recently interested in voice
solutions. I am a complete beginner in this area. I am starting out with
SEMS, and using the included SER module.
I have built sems, and thats where I am. Reading though the
documentations and presentations, I could not find an end to end
tutorial that covers everything. In most of the SEMS presentations, I
see a 'Proxy' box that sends requests to SEMS.
in SIP usually you have some registrar where the phones register to,
and a ("home") proxy through which the calls are placed. you can
easily set up this for a small to medium system with ser, sip-router
or kamailio or the like, and there are lots of documentation and
tutorials on how to do that.
with ser for example, if you are on debian, you can basically download
the source package (e.g. checkout from svn or download from
ftp.iptel.org/pub/ser), create the ser-oob package (cd ser-x.y.z; ln
-s pkg/debian debian; dpkg-buildpackage -rfakeroot -us -uc), install
it (dpkg -i ser_x.y.z ser-oob_x.y.z), configure it (dpkg-reconfigure
ser-oob) and adapt the ser-oob configuration file
(/etc/ser/ser-oob.cfg), which does implement all you need to start.
now, lets say you want to have some certain announcement played for
the case that the user dials a certain number, e.g. starting with 300.
In ser (kamailio, sip-router, opensips etc), you can catch calls to
that number and send them to SEMS, like this:
if (uri=~"sip:300.*") {
t_relay_to_udp("192.168.5.106","5080");
break;
}
where the IP address/port is the SIP listen address of your SEMS
server (set in sems.conf).
In the ser-oob config, the best place to do this is in SITE-SPECIFIC
route, and there is also already an example in there I think
(forwarding to iptel.org conference for 000777).
Now you have all INVITEs to 300 sent to your SEMS server. You tell
SEMS to load and run the application announcement, by setting in
/etc/sems/sems.conf:
load_plugins=wav;ilbc;announcement;sipctrl
application=announcement
and possibly set the announcement to play in announcement.conf (in
/etc/sems/etc/announcement.conf).
You should also set the SIP listen address and port in sems.conf.
Now all calls to 300 go to the SEMS server that runs the announcement
application, and you should hear that playing on your phone.
Now lets say that you wanted to have your SEMS server not do only
announcements, but also an echo service, if you call the number
'echo'. So you need to load both echo and announcement applications,
and somehow the proxy needs to tell SEMS which applicaiton should be
executed.
There is several methods how to select an application in SEMS (by RURI
user part, RURI parameter, regexp matching on RURI, or a separate
header), and the preferred way is to add some special header to the
INVITE, P-App-Name.
so, in sems.conf you set:
load_plugins=wav;ilbc;announcement;sipctrl;session_timer;echo
application=$(apphdr)
and in the ser-oob.cfg you do:
if (uri=~"sip:300.*") {
append_hf("P-App-Name: announcement\r\n");
t_relay_to_udp("192.168.5.106","5080");
break;
}
if (uri=~"sip:echo.*") {
append_hf("P-App-Name: echo\r\n");
t_relay_to_udp("192.168.5.106","5080");
break;
}
Now lets say there is an application in SEMS which needs some
parameters (e.g. the email address to send the voicemail to, or the
user ID for voicebox). Usually, the proxy has looked up the user
profile from some kind of DB anyway, so it makes sense to just add
that information to the INVITE and not have the voicemail or voicebox
server care for databases and stuff and look it up again. So there is
a header called P-App-Param which SEMS understands and gives as
parameters to the applications. For voicebox that is for example:
if (uri=~"^sip:1000") {
append_hf("P-App-Name: voicebox\r\n");
append_hf("P-App-Param:
[email protected]%|;[email protected]%|;lng=%$f.lang%|;uid=%$f.uid%|
;did=%$f.did%|;\r\n");
rewritehostport("mysemsserver.ip.addr:5080");
route(FORWARD);
}
or for leaving a voicemail (in failure_route, see the comment in
ser-oob.cfg)
append_hf("P-App-Name: voicemail\r\n");
append_hf("P-App-Param:
mod=%$t.voicemail%|;eml='%$t.email%|';[email protected]%|;snd='%[email protected]%|'
;[email protected]%|;uid=%$t.uid%|;did=%$t.did%|;");
rewritehostport("mysemsserver.ip.addr:5080");
route(FORWARD);
Of course, pretty much the same applies to
kamailio/sip-router/opensips config, sometimes the syntax is a little
different.
I hope that gets you somehow started... also note that each sems
application has got documentation; Readme files in doc/ and for the
examples under the source tree. The 1.1 documenation is also online at
iptel.org/sems/.
hth
Stefan
P.S. For testing, I often use a very simple ser configuration, have a
look at ser-basic.cfg in the ser source tree under etc/ser-basic.cfg .
I put the checks for r-uri between REGISTER handling and USRLOC lookup.
P.S. 2: for testing simple things like announcement or some service,
you can also do it completely without setting up a SIP proxy: register
two users A and B at iptel.org/service, load registrar_client;uac_auth
in sems.conf, set the A user's credential in registrar_client.conf,
register ekiga to your iptel.org account B, and call the number of A
- this way the iptel.org proxy sends the call to SEMS just like any
other SIP phone.
As a very basic process, here is what I would like to do: call from my
Ekiga softphone and hear an announcement served by Sems. My question is,
please point me to a tutorial on how to set up the complete system. What
pieces do I need besides sems, and how do they interact? After that, I
would like to find out how to add more users, so that they have separate
voiceboxes etc.
Thank you very much,
------------------------------------------------------------------------
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--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:[email protected]
email/xmpp:[email protected]
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