Greetings..
I'm fairly new to Kamailio, and am having fun so far... I'm using the
latest 4.3 rpm version on Centos 7.
I'm using Kamailio to front end a pair of FreeSwitch SBC boxes in an
active/active config.. My plan is to use dispatcher to load balance calls
between them. This is working fine.. the problem that I'm bumping into is
that the gateway probing options in dispatcher seem not to fit my use case..
In addition to the two SBC gateways, I also have a few external gateways
that I want to load balance between also using dispatcher. The problem is
that I can't send options packets to these gateways -- they don't respond
to them. This means I can't just configure dispatcher to probe all
gateways (probing_method=1) because it will mark these gateways as
inactive..
It looks like every other configuration of probing_method only probes a
gateway until the state is determined, and then it stops probing.
probing_method=2 does not appear to be used at all in the code..
Am I doing this wrong? Or is this just an area of dispatcher that needs
some improvement? Looking at the code, I can probably add the
functionality that I need, but I don't want to go that route if I'm just
missing how this is usually done..
Thanks,
--
*Joseph Dickson*
E: jdickson(a)evolvetsi.com
Hello everybody,
I'm using kamailio as a dispatcher in front of asterisk boxes and i use a
failure route if asterisk box does not respond or send 500error.
failure_route[RTF_DISPATCH]{
if(t_is_canceled()){
exit();
}
if(t_check_status("500") || (t_branch_timeout() &&
!t_branch_replied())){
xlog("L_WARN","[$fU@$si:$sp]{$rm} Asterisk Box $du is down\n");
ds_mark_dst("ip");
if(ds_next_dst()){
xlog("L_WARN","[$fU@$si:$sp]{$rm} Sending to Asterisk Box - $du\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
}
}
As you can see, i check if "t_branch_timeout() && !t_branch_replied()", but
where has been stored timeout value ? How can i change it ? For exemple, i
need to set 1s, and then, request expired.
Thanks,
Regards.
Hey All,
Im just after someone else thoughts on if this is a bug.
After reading :
http://kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#idp148408
Im lead to understand that setting *modparam("dialog", "db_fetch_rows", 0)*
should instruct the dialog module to write dialogs to the DB, but not
bother loading them on startup. This sounds like what I'm after.
However ever time I set the value to 0, dialogs are still loaded from the
DB.
a quick dig in the source takes me to the function
select_entire_dialog_table in dlg_db_handler.c
this chunk of code does not appear to properly implement " if
fetch_num_rows == 0 then do nothing "
Im after a couple of things from you guys.
a) validation that my understanding of what SHOULD happen is correct
b) possibly validating that I'm not crazy, and this is actually a bug :)
--
Sincerely
Jay
Hello everybody,
I have installed an IMS core using Kamailio and FHoSS.
In this environment, I am able to successfully register Bob and Alice with the same S-CSCF.
However, when I try to make a voice call between Bob and Alice it is not established.
The
code received is "403 Forbidden - You must register first with a
S-CSCF". This code is generated by the P-CSCF because the Contact Header
that is checked against the database ("location" table) it is not
found.
When I look up manually the "location" MySQL table, there are two things that surprise me:
-
The Contact header (AOR column content) that is included in the INVITE
Requet URI when entering in the P-CSCF is present in the table. So why
the pcscf_is_registered ("location") returns false?
- When Bob
registers two Contacts are stored, the first one is the one included in
the REGISTER message generated from Bob IMS Client. However, the second
Contact is the one present in the 200 OK message (coming from the
S-CSCF) which is different from the one present in the REGISTER (the
only thing different is the UDP port). Why the S-CSCF (with the default
kamailio.cfg) doesn't the Contact header and changes the UDP port? Is
this the standard behaviour? I don't think so...
Thanks for your attention
Regards,
Hello,
This is question on PBX behavior, what is the right thing to do, and how do
PBX's generally behave.
If a user on a phone, dials a number, which happens to be configured on the
same phone system (for example another tenant), there are two options:
1. The PBX notices this, and directly connects the phone to the DID on
that system (breaking separation of tenants)
2. The PBX sends the call out on the SIP trunk, and the provider-routing
sends the call back as an incoming call.
What are the pros and cons of each option from the SIP provider point of
view?
How do PBX's generally behave?
Thanks,
Antonio
PS: I reposted, because the original question was apparently not phrased in
a clear way.
Hi All,
I need one IPSec supported opensource SIP server for testing my sip client
endpoint.Could you please tell me which SIP server supporting IPSec ?
Thanks in advance.
Thanks
Priyaranjan
I realize this last email has no useful information, so let me elaborate.
I have a Kamailio server on AWS with an internal IP of 10.20.30.40 and it has an external elastic IP of 55.66.77.88.
When a SIP client subscribes it sends the initial subscription to user@realm (where realm resolves to 55.66.77.88) , then it gets a notify with a new address <sip:10.20.30.40:5060;transport=udp> . The client attempts to refresh the subscription, but we are now sending the subscribes to this new contact which will time out. 300 seconds later, we subscribe again to user@realm and contacts show up.
Other stuff like rtp, inbound/outbound calling all work properly.
Pramod Venugopal
pramod(a)dvnull.org
From: sr-users on behalf of Pramod Venugopal
Reply-To: "Kamailio (SER) - Users Mailing List"
Date: Thursday, August 20, 2015 at 10:38 PM
To: "Kamailio (SER) - Users Mailing List"
Subject: [SR-Users] Presence and NAT
Hello everyone,
I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ).
Where would I be able to change the IP sent in the Contact header ?
Pramod Venugopal
pramod(a)perseus-tech.com
_______________________________________________ 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
Hello everyone,
I'm new to Kamailio and are trying this tutorial:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
I've installed everything and are trying to launch Kamailio, but it fails
with the following error:
-----
systemctl status kamailio.service
● kamailio.service - LSB: Start the Kamailio SIP proxy server
Loaded: loaded (/etc/init.d/kamailio)
Active: failed (Result: exit-code) since Fri 2015-08-21 06:26:26 CDT; 8s
ago
Process: 1474 ExecStart=/etc/init.d/kamailio start (code=exited,
status=1/FAILURE)
Aug 21 06:26:26 sip kamailio[1474]: error in config file
/etc/kamailio/kamailio.cfg, line 234, column 12-25: failed to load module
0(1483) ERROR: <core> [sr_module.c:563]: load_module(): could not find
module <rtpproxy> in </usr/lib64...3432]: yyerror_at():
Aug 21 06:26:26 sip kamailio[1474]: r in config file
/etc/kamailio/kamailio.cfg, line 270, column 35: Can't set module parameter
0(1483) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module
matching <registrar> found 0(148...ailio/kamailio.cfg,
Aug 21 06:26:26 sip kamailio[1474]: : <core> [modparam.c:150]:
set_mod_param_regex(): No module matching <acc> found 0(1483) : <core>
[cfg.y:3435]: yyerror_at(): parse error in config file
/etc/kamailio/kamailio.cfg, line 295, column ...mod_param_regex(): N
Aug 21 06:26:26 sip kamailio[1474]: 1483) : <core> [cfg.y:3435]:
yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line
321, column 36: Can't set module parameter 0(1483) ERROR: <core>
[modparam.c:150]: set_mod_para...]: yyerror_at(): par
Aug 21 06:26:26 sip kamailio[1474]: file /etc/kamailio/kamailio.cfg, line
378, column 50: Can't set module parameter 0(1483) ERROR: <core>
[cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method
(params 1) 0(1483) : <co...mailio.cfg, line 417
Aug 21 06:26:26 sip kamailio[1474]: fg.y:3435]: yyerror_at(): parse error
in config file /etc/kamailio/kamailio.cfg, line 436, column 24: unknown
command, missing loadmodule? 0(1483) ERROR: <core> [pvapi.c:810]:
pv_parse_spec2(): erro...): wrong char [U/85]
Aug 21 06:26:26 sip kamailio[1474]: failed!
Aug 21 06:26:26 sip systemd[1]: kamailio.service: control process exited,
code=exited status=1
Aug 21 06:26:26 sip systemd[1]: Failed to start LSB: Start the Kamailio SIP
proxy server.
Aug 21 06:26:26 sip systemd[1]: Unit kamailio.service entered failed state.
Hint: Some lines were ellipsized, use -l to show in full.
-----
It seems that if it cannot find the module RTPPROXY. I have installed it
via apt-get install rtpproxy and it seems to be running.
What to do?
Hello everyone,
I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ).
Where would I be able to change the IP sent in the Contact header ?
Pramod Venugopal
pramod(a)perseus-tech.com