How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server?
Great regards,
Victor Olvera
Hello!
I am new to kamailio and trying to use it vanilla config.
Now main question is how to use ip based auth.
I found recent post of Daniel-Constantin Mierla:
http://lists.sip-router.org/pipermail/sr-users/2011-December/071147.html
Here he recommends to use 'address' table from permissions module,
I try yo use advice and add this lines at config begin:
#!define WITH_MYSQL
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!define WITH_AUTH
#!define WITH_IPAUTH
#!define WITH_USRLOCDB
Database is created and kamailio can access it.
I am add user 1000 to kamailio via kamctl and successful register it with
soft-phone.
I create trunk without registration to kamailio on asterisk server. And
trying to call from asterisk to user 1000. Call is successful. I try to
create file /etc/kamailio/permissions.deny with content 'ALL : ALL'. And
retry previous call. It still sucessful. I try to add record with asterisk
address to 'address' table with group 1. And retry previous call. It still
sucessful.
I am confused. I do not now how to disable any address for ip_auth except
if it in the
'address' table. And allow any address with if it request kamailio with
registration.
--
Best Regards,
Ivan Dudko
Hello,
I have a doubt related with DMQ dns behavior, I noticed that when kamailio
starts, it tries to resolve DMQ name configured on parameter
notification_address as the following sequence:
1. SRV
2. A
3. AAAA
Isn't supposed kamailio try first resolve the NAPTR DMQ name, and then
SRV?
I'm asking this because kamailio is trying resolve the SRV record without
any transport protocol specified on query, as my dns server only accepts
queries on the format "_Service._Proto.Name" the SRV will never be resolved.
Thank you.
BR
--
José Seabra
Hello,
Someone has an idea?
Again, Kamailio stop processing SIP during more than 30s. Nothing print in
the logs.
The processing back on when a script detect no answer and restart the
service.
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com]
Envoyé : vendredi 23 septembre 2016 11:05
À : sr-users(a)lists.sip-router.org
Objet : Dimension number of child
Hello,
I'm wondering if there's a best practise to dimension the number of child to
setup?
I see, very often, Kamailio that stops to reply to SIP traffic for couple a
seconds. This cause some calls drop if RE-INVITE are not processed for
example.
So, I'm wondering if the issue is not due to a lack of child for processing
the SIP traffic.
Thank you for your inputs.
Regards,
Igor.
Hello group,
I know this may be hard to believe but I'm in the process of upgrading
an old, but stable, OpenSER 1.0 group of servers to Kamilio. I'm
basically going to sort of start from scratch with the database and
export my old MySQL database and write some scripts to reimport the data
using kamctl scripts since the database structure has changed
substantially since then it looks like.
I've mostly migrated the config file over fairly easily, except for
AVPops. I've got a section of config file that writes into the
usr_preferences database for call forwarding, but I seem to get the same
error anywhere I try to write to the database using AVPops. Here's the
output I get when checking the config:
0(1378) DEBUG: <core> [pvapi.c:268]: pv_cache_add(): PV cache not initialized, doing it now
0(1378) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "avp"
0(1378) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:callfwd)] at [5 (5)]
0(1378) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 145, column 4-18: Can't get from cache: $avp(s:callfwd)
ERROR: bad config file (1 errors)
Here's what I think are the revelant portions of the config file:
modparam("avpops", "db_url", "mysql://user:pass@localhost/kamailio")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
<snip>
if (avp_db_load("$from/username", "$avp(s:callfwd)"))
{
if(!avp_db_delete("$from/username", "$avp(s:callfwd)/usr_preferences"))
{
sl_send_reply("500", "Server DB error"); }
}
# avp_write("$ruri","$avp(s:callfwd)"); # Old OpenSER 1.0 syntax...
$avp(s:callfwd) = $ruri; # New Kamailio syntax ??
avp_db_store("$from/username","$avp(s:callfwd)/usr_preferences");
t_relay();
return;
Basically I just want to write the ruri into the usr_preferences
database with an attribute of callfwd but not sure what I'm missing. I
see the error "Can't get from cache" but not sure how that pertains to
what I'm trying to do. Anyone have any ideas?
Thanks,
Brian
Hello,
I'm planning on doing some smart load balancing with Kamailio.
We have a distributed network, with multiple Kamailio boxes in different
locations serving as Ingress SBC,
these Kamailio boxes are the entry point for a SIP call and then they route
the call to a pre-configured Asterisk boxes.
I want to move away from this, I would like these Kamailios to be able to
distribute the traffic to Asterisk boxes based on the
actual load on these boxes, the goal is to be more dynamic?
Is there any Kamailio module which could do that? Do I need to integrate
some other tool(Homer etc) with Kamailio to achieve this?
Any suggestions are welcome.
--
*Nitesh Bansal*
Business Development Engineer
http://www.voxbone.com/
we have serial forking already setup properly but need to t_suspend().
found out that it completely breaks the destination set.
other than manually rebuilding the destination set after t_continue(), are
there any other solution out there?
Kelvin Chua
Hi all,
I'm trying to send a call to an AS via a kamailio S-CSCF (release 4.4.3). I'm addressing the AS by a distinct PSI: sip:04325432101@domain.net;user=gsmr.
At the first try the S-CSCF sends a SAR to HSS and receives a successful SAA (tshark traces attached), but the S-CSCF rejects the call with 555 AS Contacting Failed - iFC terminated dialog.
At the second try (a few seconds later), the S-CSCF doesn't exchange SAR/SAA with the HSS, and now the call is successfully forwarded to the AS.
This problem is permanent, it is not the result of a temporary loss of connectivity between S-CSCF and AS.
I checked the iFCs that I had configured at the HSS and I think, they are OK.
I attached
- tshark traces from the node, where the S-CSCF resides: 2016_09_19_error_555_17_scscf_filtered.pcap
- WITH_DEBUG logs from kamailio: 2016_09_19_error_555_17_kamailio_1st_try.log
- A tarball of /usr/local/etc/kamailio: 2016_09_19_error_555_17_kamailio.tar.gz
Can anybody help? Is it a bug in kamailio or is the problem on my side?
Thanks,
Christoph
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Hello,
Thanks to all to develop the such type Sip Proxy server.
Please suggest whether SIP proxy server is open source ,If yes than it can fulfil the my below requirement .
1) I having some sip clients which are registered on IMS .Can I use the kamailio Sip proxy sever between sip client and IMS.
Sip client ------------------Kamailio Sip Proxy ---------------IMS
2) Second things, can I use Kamailio Sip Proxy to interconnect two IMS nodes . Can I create the SIP trunk without registration mode (username and password) between IMS nodes using Kamailio. Because the my IMS node doesn't support sip trunk in registration mode. Its support one normal trunk (SIP-T)
My current IMS set up like below.
IMS-1 (IBCF)---------------IMS2(IBCF).
Regards
Surender Singh
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Hello list,
I have a proxy that manage fix calls and a proxy that manage mobile calls.
I want to implement call forking so when I call to a fix phone the call goes
to the fix proxy and it forks to the mobile proxy who manage the call and
viceversa. I want the fix to ring 2 or 3 times before the mobile.
In order to do so I relay first to the fixed proxy, which will bounce back
to mobile proxy the branch for mobile device.
But now if I want to be able to choose which phone (mobile or fixe) rings
before I have tried by doing the "async_route("RELAY", "7");" in the fixe
proxy like in the mobile proxy but it doesn't work.
My code is:
Fix proxy:
if($rU==123456789){
add_diversion("forking");
$var(priority)=1;
append_branch("sip:987654321@proxy_mobile");
}
route[INVITE]{
if($var(priority) == "1"){
async_route("RELAY", "7");
}else{
route(RELAY);
exit;
}
Mobile proxy:
if($tU==987654321){
$rU=123456789;
}
route[INVITE]{
if($dir == "forking"){
async_route("RELAY", "7");
}else{
route(RELAY);
exit;
}
}
Thanks.