Kamailio 3.3.0
I have a variable $var(s:dst). It can store either a number or IP.
How do I check to determine whether it is a number of IP?
I tried the following and it did not work:
If ($var(s:dst) =~ "^\d+\.\d+\.\d+\.\d+$"){
It is a IP.
}else{
It is a Number
}
Thanks
Gary
Hello
I am very new in opensip.
i am trying to install kamailio and freeswitch...kamailio is running
fine...but i donot know how kamailio will communicate to my local sip
server.
kamctl start and stop ...starting and stopping the kamailio.
i have installed kamalio from
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git
--Piyush
Or we can just "port" Opensips's "make menuconfig" :)
15.07.2012 2:05 пользователь "aft" <aftnix(a)gmail.com> написал:
> Hi,
>
> Kamailio is a modular stack. So inclusion or exclusion of a module is
> very important. At this moment editing modules.lst is very *primitive*
> way of doing things.
>
> I like menuconfig style ncurses based interface. Which gives a very
> clear view of "dependency" and "which and which" information about the
> modules.
>
> As …
[View More]number of modules increases, most of us just compile most of them, or
> take the default modules.lst as granted.
>
> If a menuconfig target is added, it should be easier to customize a
> kamailio build.
>
> Adding a menuconfig target seems a trivial task for anyone who know
> kamailio build system well plus some intro to ncurses programming.
>
> So i guess devs should look into it.
>
> cheers
> aft
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Hi,
Kamailio is a modular stack. So inclusion or exclusion of a module is
very important. At this moment editing modules.lst is very *primitive*
way of doing things.
I like menuconfig style ncurses based interface. Which gives a very
clear view of "dependency" and "which and which" information about the
modules.
As number of modules increases, most of us just compile most of them, or
take the default modules.lst as granted.
If a menuconfig target is added, it should be easier to customize a
…
[View More]kamailio build.
Adding a menuconfig target seems a trivial task for anyone who know
kamailio build system well plus some intro to ncurses programming.
So i guess devs should look into it.
cheers
aft
[View Less]
Hello All,
I am trying to get a failure route to work, I have got it working partially.
When a call comes in, first I check the db_alias, if that is positive I
do a lookup location and relay if the location is valid.
But sometimes the sip client is still registered in the location table,
but not connected anymore, mainly with wifi connected clients.
So, I have got the time out on 3 seconds (fr_timer). When that hits I
have configured the following failure route:
failure_route[NOTONLINE]…
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{
xlog("SCRIPT: Notonline failure route\n");
t_on_failure("STOP");
if (t_is_canceled())
{
exit;
}
if (t_check_status("408"))
{
xlog("SCRIPT: Status is time out");
$rU = $avp(orig_called); /( called number and alias id
not equal, so have to revert the rU back to the called number)/
prefix("9993"); /( needed to get the right manipulation
done within asterisk)/
xlog("SCRIPT: uri is $ru");
$ru = "sip:" + $rU + "@w.x.y.x:5060";
xlog("SCRIPT: uri is $ru"); (w.x.y.z
ip address of the asterisk box)
append_branch();
t_relay_to_udp("w.x.y.x","5060");
break;
}
}
I am not sure if the above is correct. I have based this on an old
"voicemail" failure route I could find.
It is working correct, the call is forwarded to an Asterisk box, where
some manipulation is done and then send to an pstn gateway.
The only problem I have is one way audio. RTP from the called number
reaches the callee but not vice versa.
Now I am wondering, can that be caused by the failure route, or should I
be looking in another direction?
Hope someone can give me a pointer.
Thanks.
Gertjan Wolzak
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