I have kamailio 4.0.2 installed on Ubuntu 12.04.1 and I already have
user/subscriber registered and PSTN routing already working fine
(outgoing call) i just need to add config on kamailio.cfg (modparam,
ifdefs, routing logic, etc) to make this carrierroute module work based
on the db entries i created, if someone can provide sample config
entries that needs to add it would be great, since i believe this is not
included on the default installation of kamailio. Sorry for disturbing
you.
kamctl cr show
cr carrier names
+----+------------+
| id | carrier |
+----+------------+
| 1 | 61.8.XX.XX |
+----+------------+
cr domain names
+----+------------+
| id | domain |
+----+------------+
| 1 | 61.8.XX.XX |
+----+------------+
cr routes
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+
| id | carrier | domain | scan_prefix | flags | mask | prob | strip |
rewrite_host | rewrite_prefix | rewrite_suffix | description |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+
| 1 | 1 | 1 | 03 | 0 | 0 | 1 | 0 |
61.8.XX.XX | 0 | 0 | CR to voip gw |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+
My current kamailio.cfg was based on this,
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=etc/kama…
with just few modification for PSTN route usage (IP addr of PSTN and
allowed prefix)
Thanks in Advance,
--
Lucky Santiago
M: +63.932.88.LUCKY (58259)
Skype: luckysantiago
Dear list,
Its just a little thing to ask but I somehow can't seem to figure out which
psuedo variable to use to find out where a particular 200 OK is destined to
go.
the scenario is I've multiple media servers in dispatcher and calls are
distributed. I need to capture a 200OK that is destined to go to a
particular ip!
I've tried using the dst_ip, $dd, $du but none of them are giving any
results for the 200 OK. I just need to know where do kamailio is going to
send this 200 since it is already a part of an on-going INVITE/call.
Thanks,
Sammy
Hello, I've tried to configure *Websockets *server in *Kamailio *(i've
installed kamailio with these steps:
http://www.kamailio.org/wiki/install/4.0.x/git), but I think that is *not
working*,
because when trying to connect to the *Websockets* server in *JsSip *
http://tryit.jssip.net/ says:
<http://gyazo.com/f37d9f42caf1915223611f6f1ad18b84.png>
That's the *Websockets server*: ws://184.75.243.217:10080
*
*
*
*
The *configuration file* that I use is that:
https://gist.github.com/jesusprubio/4066845
I've changed the basic things.. directory of the modules, port of the
wss:// and the mysql user and password.
That's a printscreen of the *Kamailio* server running: (there are more that
just 1 IP because my server have 4 IP address because of ICE Server.
The* Kamailio is running* pretty well over programs like *Jitsi*, but not
in the browser because of Websockets, that's why I want to run a Websockets
Server.
<http://gyazo.com/ea6635a8162513c150e35fe2f8e0e381.png>
Hello, can someone let me know how I can modify the "from user" when making an outbound PSTN call. For example, when I make a PSTN call now the caller ID shows up as my 4 digit extension but I would like my PSTN number displayed instead. Thank you.
Arun
I'll check the logs and most probably have to do some test myself, but
it's going to take a bit.
Daniel
On 8/23/13 2:50 PM, Steve Davies wrote:
>
>
>
> On 23 August 2013 14:24, Daniel-Constantin Mierla <miconda(a)gmail.com
> <mailto:miconda@gmail.com>> wrote:
>
> Oh, I forgot that the group of t_set_* functions have one
> parameter, to allow set/unset the flag. Use:
>
> t_set_disable_internal_reply(1);
>
>
> Hi Daniel,
>
> Doesn't seem to work...
>
>
> Kamailio sends back 500, 477, 477, 477
>
> Return code from t_relay was still -1:
>
> t_set_disable_internal_reply(1);
> $avp(senttoast) = 0;
> $var(rr) = t_relay();
> xlog("L_NOTICE","SLD: in RELAY, t_relay returned $var(rr)
> err.rcode is $err.rcode t_r_c is $T_reply_code sent = $avp(senttoast)\n");
> if ($var(rr) < 0) {
> sl_reply_error();
> }
>
>
> >> Aug 23 14:34:11 ubuntu /usr/local/sbin/kamailio[22785]: NOTICE:
> <script>: SLD: in RELAY, t_relay returned -1 err.rcode is <null> t_r_c
> is 100 sent = 0
>
>
> Thanks,
> Steve
>
>
>
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
I'm using kamailio 4.1 Im having problem to see witch user is online, I
have the module presence.so and presence_xml.so load into the config file
of Kamailio.cfg.
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
modparam("presence", "presentity_table", "presentity")
modparam("presence", "active_watchers_table", "active_watchers")
modparam("presence", "watchers_table", "watchers")
modparam("presence", "db_update_period", 20)
modparam("presence", "clean_period", 50)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
# handle presence related requests
route(PRESENCE);
Any help please?
--
Kethzer Docteur
Hi,
I see that if 4.0.3 has a network error in sending a request, it sends back
a 477 response, but does not execute the t_on_failure route block.
Which I want to use to route the call via a local alternative gateway.
I saw a bug that talked about the failure_exec_mode modparam, but that's
not listed in the 4.0.3 documentation.
Is there a way to get this to work?
Thanks,
Steve
--
Steve Davies: Technical Director, Connection Telecom (Pty) Ltd
Email is preferred, but: Phone: 0878200200
Have you listed the profile via kamctl or kamcmd? The commands are in
the readme:
http://kamailio.org/docs/modules/stable/modules/dialog.html#idp3737432
Is the afferent profile listing any dialog?
Cheers,
Daniel
On 8/23/13 2:50 PM, Gertjan Wolzak wrote:
>
> Hello Carlos,
>
> I have been looking at your module, can't wait to work with it,
> especially the prepaid part.
>
> But as this system is in production, can't just start experimenting,
> want to.... But cant.
>
> So just have to wait for some tips on the cli....
>
> But thanks.
>
> Rgds,
>
> Gertjan
>
> *From:*sr-users-bounces@lists.sip-router.org
> [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Carlos
> Ruiz Díaz
> *Sent:* vrijdag 23 augustus 2013 14:37
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] hanging active dialog...
>
> I don't know what exactly happened in your case but if you don't have
> time to investigate you can try cnxcc module [1] channel control to
> achieve the same goal.
>
> Take a look at the sample configuration file located in [2]
>
>
> xlog("L_INFO", "Setting up channel based credit control")/;/
>
>
> $var(max_chan) *=* 2;
>
> $var(retcode) = cnxcc_set_max_channels("$var(client)", "$var(max_chan)");
>
>
> if ($var(retcode) *=*= -1) {
>
> xlog("Error setting up credit control");
> return;
>
> }
>
>
> $var(count) *=* -1;
>
>
> if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) {
>
> xlog("Error getting customer's channel count")/;/
>
> }
>
>
> xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)")/;/
>
>
> if ($var(retcode) < -1) {
>
> xlog("Too many channels for customer")/;/
>
> sl_send_reply(403, "Forbidden")/;/
>
>
> if (!cnxcc_terminate_all("$var(client)")) {
>
> xlog("Error terminating customer's calls")/;/
>
> }
>
>
> exit/;/
> }
>
>
> [1]http://kamailio.org/docs/modules/devel/modules/cnxcc.html
> [2]https://github.com/caruizdiaz/cnxcc/blob/master/example/kamailio-cnxcc.cfg
>
> Regards,
> Carlos
>
>
> On Fri, Aug 23, 2013 at 5:16 AM, Gertjan Wolzak <g.wolzak(a)foize.com
> <mailto:g.wolzak@foize.com>> wrote:
>
> Goodmorning All,
>
> I use the following route to check for concurrent calls by the same
> user, if a concurrent call is tried it is not allowed.
>
> route[CONCURRENT]
>
> {
>
> xlog("SCRIPT: Conccurrent call check");
>
> if(!get_profile_size("caller","$fu","$avp(nrcalls)"))
>
> {
>
> sl_send_reply("403", "Call not matching profile");
>
> exit;
>
> }
>
> xlog("SCRIPT: caller value for $fu is $avp(nrcalls)");
>
> if($avp(nrcalls)>= 1)
>
> {
>
> sl_send_reply("403", "Active calls limit exceeded");
>
> exit;
>
> }
>
> dlg_manage();
>
> if(!set_dlg_profile("caller","$fu"))
>
> {
>
> sl_send_reply("500", "No new channels in this profile");
>
> exit;
>
> }
>
> xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)");
>
> }
>
> Now I had a situation where a user could not call because the
> get_profile_size for this user gave the value 1. So another call was
> not allowed.
>
> But the user did not have a call active. As no dialogs were active,
> checked that.
>
> I assume there can be a lot of reasons why this happens, also I want
> to use the "I don't want to know the cause" method to solve this.
>
> So I looked at the dialog module documentation, to see if I can just
> clear the profile size for this specific user.
>
> I wanted to use the profile_list_dlgs(8.6) to get the dialog details,
> then the dlg_terminate_dlg(8.4) to, you guessed it, terminate that dialog.
>
> But it's not clear to me how I can give those commands....
>
> Is there a way to give those commands on the cli, if so can someone
> please write down some examples...
>
> Would be really appreciated.
>
> Rgds,
>
> Gertjan
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Carlos
>
> http://caruizdiaz.com
>
> +595981146623
>
>
>
> _______________________________________________
> 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
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda