i have entries in htable with key name lpi::20033::prefix:
# sip-proxy_ctl mi sht_dump htable | egrep 20033
lpi::20033::prefix[0]:: +3583796294
lpi::20033::prefix[1]:: +35837824030
lpi::20033::prefix::size:: 1
as you see in above, size of that key name is 1 although there clearly
is two entries in the table with that key name.
according to readme ::size is number of items:
3.7. array_size_suffix (str)
The suffix to be added to store the number of items in an array.
i verified with another key name where there was only one prefix, and
then ::size was again 1. so sometimes ::size is correct and sometimes
incorrect.
this looks like a serious bug to me. any comments?
-- juha
friend I am interested in learning more about VoIP or IP telephony, I read a
lot about you and I still have more to read, as it can not released through
theofficial documentation page errors Please let me send them am email,
thanks
I DEVELOP voip server to sip in Venezuela, I need all the help possible.
thanks
*Rommel Malave .M **
* *T.S.U Universitario Extracción y Producción de Gas *
* T.S.U en Informatica***
04148241688
Hello,
please do not write direct private emails for questions about kamailio,
use public mailing list sr-users(a)lists.sip-router.org (cc-ed) -- private
emails are ignored, after the first notification.
exit is not a function exported by a module, it is implemented in core.
On the other hand, you try to stop the kamailio cfg interpreter when lua
interpreter is executing the Lua script, so it won't work
straightforward. Set a variable in kamailio from Lua, then check it
after executing the lua script to decide to further exit or not.
Cheers,
Daniel
On 5/24/11 11:44 AM, Bret McDanel wrote:
> I am having a terrible time trying to get a conditional response from a
> lua script.
>
> Basically I want
>
> if error then
> sr.sl.send_reply(404,"Not Found");
> sr.modf("exit"); -- this does not work!
> end
>
>
> How can I have such a conditional? I have tried with return values from
> the script to see if that would change anything but that did not seem to
> work, sr.modf("exit") is not found.
>
> Any assistance would be greatly appreciated.
>
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi,
I'd like to know the functionality of the struct run_act_ctx. I need to
adapt my route module to use this struct, the function do_action require
this parameter. I'm migrating of the kamailio 1.5.0 to 3.1.2.
Regards,
We are happy to announce the availability of the
SIP Express Media Server version 1.4.0.
SEMS is a high performance media and application server
for SIP based VoIP networks.
This release features a powerful Session Border Controller (SBC)
module. From completely transparent B2BUA to customized URI/From/To,
strictly filtered (messages, headers, codecs) with RTP anchoring,
Session Timer enforcement, prepaid and call timer, the SBC facilitates
interconnect and core routing in a simple and secure way.
Thanks to the new multihoming support, SEMS can now be employed
at the border of the networks. This addition also allows to
overcome the bottleneck of one NIC - giving the possibility to
fully exploit SEMS' great performance.
In the app development area, the DSM language has matured to
become a viable candidate also for implementing complex
application logic, thanks to language constructs like for, if
and functions.
SEMS can be downloaded in source from its ftp site at:
ftp.iptel.org/pub/sems/1.4/1.4.0/
Packages for debian/ubuntu should appear shortly on the OBS page:
https://build.opensuse.org/project/show?project=home:team-sems
Contributors to this release:
Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Matthew Williams,
Peter Lemenkov, Anton Zagorskiy, Robert Szokovacs, Jon Bonilla,
Thomas Gelf, Erik Dekkers, Tobias Lindgren, David J
Many thanks to the companies sponsoring SEMS development, especially
FRAFOS GmbH and TelTech Systems Inc.
"E stands for Express."
* Apologies if you receive multiple copies of this message. *
according to core cookbook:
~ : bitwise NOT
however, looks like it has no effect. i have
$var(test) = ~2;
xlog("L_INFO", "test is <$var(test)>\n");
and i get to syslog:
May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2>
have i misunderstood the operator or is there a bug?
-- juha
On 5/20/11 6:33 PM, Jon Farmer wrote:
> On 20 May 2011 17:25, Daniel-Constantin Mierla<miconda(a)gmail.com> wrote:
>
>> the module is good, but not for this case -- the warning seems to be from
>> startup, saying that no reverse dns could be done to the ipv6 listen address
>> and it is harmless (done for the purpose of building aliases list).
>>
>> Do you get the warning at runtime?
>>
>> Normally, the ipv6 should be in between square brackets.
>>
> Adding the address in question to my hosts file temporarily takes the
> warning away :)
ok, let me know how is going with the testing of ipv6 -- the support is
in core since 2002, but not many entered the challenge so far :-)
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi all,
Need some newbie advice on how to write INVITE routing for my scenario. I
currently have kamailio+asterisk-realtime integration working in the
following way:
Outbound calls:
USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then
Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then
Asterisk(1.6.2.18) -> SIP/PSTN
This works, though I'd prefer to use Kamailio + LCR and gateway failover as
Asterisk doesn't do this well.
Here's what I want to do: (for now, small steps)
Outbound Calls:
USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then
Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then
Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN
I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's
and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to
write the routing logic so that kamailio sends calls from asterisk to PSTN.
route[LOCATION] {
...non-INVITE check ends here...
if(route(FROMASTERISK))
{
# coming from Asterisk --- Does this mean check kamailio
local users first before continuing?
if (!lookup("location")) {
switch ($rc) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
## -- If match 11 digits and starts with '1' or if match 10
digits and starts with [2-9]
if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") {
xlog("L_INFO", " FROMASTERISK: This will
become route(NORTHAMERICA); \n");
exit; ## to be removed
} else {
xlog("L_INFO", " FROMASTERISK: This will
become route(INTERNATIONAL); \n");
exit; ## to be removed
}
# new call - send to Asterisk
route(TOASTERISK);
}
Will this logic work? I haven't determined how to write route(NORTHAMERICA)
or route(INTERNATIONAL) yet so I'm just logging to debug the logic and
syntax.
Thanks in advance.
Hi
I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk
cluster. I am planning this for a call center and I spec 80 currents calls
for 30 agents sometimes, so I need to handle big queues and call and agent
states, is it possible for kamailio handle the call states and agent states,
and leave asterisk for the call center functions?
Thanks in advance
