Hey Daniel,
As said in the past, this module is good news, thanks for making it available.
As a big fan of delayed billing, sleep() does a great job of delaying
200 OK from the upstream for me (useful for example with analogue/CDMA
gateways sending 200 OK on ringback).
Cheers,
DanB
Hi
I was wondering what DTMF modes Kamailio supports? If i wanted to force a
DTMF mode on Kamailio how could i go about doing it?
thanking you in advance
Phillip
hey list,
i currently test on my asterisk 1.6 box on on pc with private ip
and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc
with public ip and private ip.
everything installed successfully. i follow this tutorial realtime
intergration with asterisk
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
my config.
rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user
#!define WITH_NAT
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("nathelper", "sipping_from", "sip:pinger@publicip")
nat_uac_test("19")
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
is my config correct?
may i know the best version work out from the box?
please advice..
thanks in adv. :)
--
Regards,
MingHon
Hello,
a new module is now in GIT master branch, named async.
Its purpose is to provide asynchronous SIP request processing using
t_suspend()/t_continue() mechanism from tm module. While the tm module
had it for quite some time, it was not available for usage in
configuration file.
There are two functions implemented by now:
- async_route(routename, interval) - execute asynchronously a route
block after a time interval
- aync_sleep(interval) - sleep asynchronously for a time interval and
then resume SIP request processing
See the readme for more details:
http://kamailio.org/docs/modules/devel/modules/async.html
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time:
[root@registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/
[root@registrar modules_k]# ls
acc.so db_text.so drouting.so mi_datagram.so permissions.so rtimer.so statistics.so usrloc.so
alias_db.so db_unixodbc.so exec.so mi_fifo.so pike.so rtpproxy.so textops.so xlog.so
auth_db.so dialog.so group.so msilo.so pua_mi.so siptrace.so tmx.so
auth_diameter.so dispatcher.so htable.so nathelper.so pv.so siputils.so uac_redirect.so
benchmark.so diversion.so imc.so nat_traversal.so qos.so speeddial.so uac.so
call_control.so domainpolicy.so kex.so path.so registrar.so sqlops.so uri_db.so
cfgutils.so domain.so maxfwd.so pdt.so rr.so sst.so userblacklist.so
[root@registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/
[root@registrar modules_s]# ls
acc_db.so avp.so diversion.so fifo.so nathelper.so pike.so speeddial.so uri_db.so
acc_syslog.so db_ops.so domain.so gflags.so options.so print.so textops.so uri.so
auth_db.so dialog.so eval.so maxfwd.so pdt.so registrar.so timer.so usrloc.so
avp_db.so dispatcher.so exec.so msilo.so permissions.so rr.so uac.so xlog.so
[root@registrar modules_s]#
I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something?
This is from the syslog
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume
and this is from the fifo interface:
[root@registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload
500 command 'lcr' not available
[root@registrar kamailio]#
Hi,
I'm working on a jsonrpc (client) module that we would like to make async,
using t_suspend and t_continue from the tm module. I have a couple questions
related to this.
The way I understand these functions, t_suspend will freeze the transaction
in shared memory and immediately return and t_continue will unfreeze the
transaction. What I'm not clear on here is whether t_continue will continue
processing of the transaction within the process that calls t_continue, or
will it delegate to a worker process?
I assume I should spawn an io process in child_init which will be
responsible for sending/receiving on the socket(s), but what is the best way
to communicate between processes? Is there a way to get a fd from
fork_process (that can be written to by any child and read from using
select() in the io thread)?
This is my first attempt at a real module, so any pointers are greatly
appreciated.
Regards,
Matthew WIlliams
Hello friends!
Having trouble to start newly installed Kamailio
This is what I got:
root@debian:/home/user# /etc/init.d/kamailio status
Status of kamailio: kamailio is running.
root@debian:/home/user# kamctl start
INFO: Starting Kamailio :
ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start
failed
root@debian:/home/user#
Any ideas will appreciated
Regards
Roman
Hello,
I just wanted to mark the 15000th development commit in our project
(counted only on GIT master branch). It happened several days ago and it
was done by the same person that did the first commit, Andrei:
git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1
512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000
Initial revision
The 15000th one is:
git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000
| tail -1
8a90dd3 Andrei Pelinescu-Onciul Sat Jun 11 11:24:05 2011 +0200
core: remove unused variables + coding style
I published the news about it on project sites as well:
* http://asipto.com/u/37
* http://asipto.com/u/38
Now it's the time to look forward to the 20000th commit!
Enjoy the summer, autumn comes with v3.2.0,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hello,
I want to present an idea and see if it is something people find
benefits with such approach.
Some time ago, due to existence of only SIP client applications (aka
softphones) that have static GUI, which were designed for generic
purposes, do not present what servces the currently used provider offers
nor are easy to customize, I played with the idea of a SIP application
with the GUI in HTML/CSS/Javascript.
Being busy with Kamailio, the whole thing was postponed, but during last
days I took the time and updated the code to latest developing tools
(QT/Webkit and pjsip) and made it public on github, see:
* http://asipto.com/u/3c
A screenshot and some details in the wiki at:
* http://asipto.com/u/3b
The UI can be pulled from a remote site via http/https for example. The
approach was to have a framework to render the UI, used QT/Webkit for
that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to
Javascript. I worked with pjsip, but the system is plugin based, so
pjsip plugin can be replaced, plugins for other things (not only
sip/voip) can be written easily.
The code is just for demo/prototyping, with the goal of showing
feasibility. At this moment can be used to make/receive calls and
send/receive instant messaging. In addition, content of UI can be
carried by SIP messages - the examples added so far is to request some
server statistics when clicking on a link of UI and the SIP server
pushing periodically some info to a section of UI.
Personally I don't have any plan to go into client side development,
still the idea was too appealing to try and I thought would give lot of
value and enable new features on SIP services to make user experience
more pleasant.
Therefore the target is to see if any of you think is something that
worth pursuing and if yes, get people on board to develop it further
from prototype phase.
In the wiki I listed a part of my wish list with current VoIP/SIP
services that would be trivial to implement with this approach. The
naming as SIP browser instead of softphone hopefully suggest also the
purpose of the application - to be able to browse the services offered
by provider at that time and the provider to be able to push content
back to application (on demand or periodically). Development was done on
Mac OS X, a tarball with binaries for this OS is available for download
on github - compiling for other OSes should go pretty straightforward.
At the moment the idea poped up there was no such approach that I could
find (at least not in open source space), did I miss something, are you
aware of any similar approach? What is your opinion overall?
Thanks,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda