Dear All,
We would like to store SIP text Messages when the destination Subscriber is Offline.We have insert into kamailio cfg file the configuration lines below.Unfortunately storing messages is unsuccessful.Any ideas of what missing or what could be wrong?
Best regards.
************************************************************************************************************loadmodule "msilo.so" #!ifdef WITH_MSILOmodparam("msilo","db_url","mysql://[% kamailio.proxy.dbrwuser %]:[% kamailio.proxy.dbrwpw %]@[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar@xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU@xxxxxxx.local")modparam("msilo","contact_hdr","Contact: <sip:registrar@xx.xx.xx.xx:5062>;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” #########################################################################Store messages to offline Subs########################################################################route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { 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(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay();} failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; 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("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };}***********************************************************************************************
Hi
I did setup kamailio to handle websocket, and I am testing using pjsip, my
problem is that I can logon using the pjsip client but SIP does not
register, pjsip displays a SIP connection timeout error. My xlite client
can logon just fine though. The log from kamailio can be seen below
http://pastebin.com/XrsQM8Cf
Please advice
Hello,
following several requests during Kamailio World, but also in the
direction of federating open source and local communities via SIP (as a
result of FOSDEM panel and discussions), a SIP service was built up to
be offered for free on best effort manner - the hardware and bandwidth
are offered by asipto.com, the service is using the domain openrcs.com.
Main goal is to offer a free option for community members (but not
restricted to) to create and use SIP accounts. The plan is to enable
many (common) new features to allow people to play with, helping also to
test kamailio.
There is a rather basic (and alpha stage) web portal based on
wordpress+buddypress, good enough for the moment to create an account
and tune some options - it is available at:
- https://www.openrcs.com
Along with voice/video calling and instant messaging, the service has
enabled features such as simple presence, msrp relay, xcap server and
websocket (web portal embeds a bare sip app built using jssip library).
Other features will be enable over the time, but have in mind **it is a
not a telephony service**, thus no such specific services like
emergency, pstn interconnect, etc.
Hope it will be useful for many people - who ever wants to try it, head
to https://www.openrcs.com !
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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Hi Daniel,
I nearly forgot, can you post your openrcs config ?
Many thanks,
John
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Hi,
There is a big delay in our Kamailio server when repling "100 Trying" after
incoming INVITE messages (more than 500 ms). It seems my perl script (and
database query) is executed before "100 Trying" being send out. I was
wondering if sending "100 Trying" is configurable so I can send it right
after INVITE and before executing my perl script.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
if (is_method("INVITE"))
{
perl_exec("test");
}
t_check_trans();
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
..
route(RELAY);
} (morthen
Thanks
R
Hello,
it's probably the time to have a IRC development meeting to set the
targets for the next major release as well as discuss what is needed
these days around the project.
I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin
time). I created a page to collect the topics, only few from my list
being added for now:
* http://www.kamailio.org/wiki/devel/irc-meetings/2013a
Feel free to add topics there.
Also, I would be available on Tuesday at the same time if proves to be
more convenient for more people. If none of these dates suit for
majority, we can look for other alternatives -- just propose and we can
start a poll.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hi All,
in a call forking, after one branch answer the call (200 OK reply), a
CANCEL SIP message has been sending to other/another branch(es) and I need
to process this/these cancellations in configuration file. After reading
some documentations, I discovered there is event_route[tm:local-request]
block, which is executed when tm generates internally and sends a SIP
request, Such cases are:
SIP messages sent by msilo module
SIP messages sent by presence server
SIP messages sent by dialog module
SIP messages sent via MI or CTL interfaces
I didn't understand very well this cases, so I insert event_route block in
my kamailio.cfg but neither CANCEL SIP message or other requests generated
by tm module was handled by event_route. I must be using wrong concept to
handle this CANCEL SIP message, it's possible handle this messages in
configuration file?
Best Regards
Hello,
already suggested on the irc channels, I plan to redirect #sip-router to
#kamailio to group in a single place and have consistency with the name
of the project.
That means making #sip-router an invite-only channel and automatically
redirecting those that want to join to #kamailio.
Hope is ok with everyone, invite-only flag can be removed later.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Dear All,
using the following link:-
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
I have successfully integrated Asterisk and Kamailio on the same box for
testing, but am now facing the problem of getting Freepbx to use the same
MySQL database tables.
My Kamailio and Asterisk install uses the following tables:-
sipusers
sipregs
voicemail
voicemail_data
voicemail_messages
However Freepbx uses the following tables.
devices
users
Any idea on how I fully integrate Freepbx with my Asterisk and Kamailio
configuration using Mysql?
My goal is to have a Kamailio LB to multiple Asterisk/Freepbx servers
sharing the same extension and voicemail databases.
I will also probably integrate a2billing and share the a2billing database
between the Asterisk/Freepbx servers to create a fully scalable solution.
Any advice most gratefully received.
Hi All,
One of the most thorough documents I've found for an introduction to Kamailio is the SER-GettingStarted.pdf, but it is now quite dated.
Is there an updated version available, or in the works?
Cheers,
Dave.
