Hello,
I'm trying to create a DNSSRV record that resolves to multiple Kamailio servers for my clients to authenticate against. Before using Kamailio, we had a DNSSRV record setup that resolved to our multiple Asterisk servers via their WAN Address.
I've tried to do the same with Kamailio, but not having much luck.
The DNSSRV record is setup as follows, as you can see from the output of a `dig` query:
;_sip._udp.kam.some-domain.com. IN SRV
;; ANSWER SECTION:
_sip._udp.kam.some-domain.com. 14399 IN SRV 20 100 5060 kam1.some-domain.com.
_sip._udp.kam.some-domain.com. 14399 IN SRV 10 100 5060 kam2.some-domain.com.
I am not able to register any devices to this SRV record. However I /CAN/ register directly to kam1.some-domain.com and kam2.some-domain.com
Registration attempts to the DNSSRV Record (kam.some-domain.com) do not even appear in /var/log/messages.
We have the domain name for kam (SRV Record), kam1 and kam2 setup in kamailio.domain and kamailio.dispatcher.
What am I missing here? Thank you for taking the time to read this!
Derek Bolichowski
hi ,
we configured homer with kamailio sip capture (server-A)
then i try configured another kamailio with siptrace (server-B)
mysql database configured with "homer_data" DB
i try to call to the second kamailio server using sip phone but didn't get
data on homer ui
then I find an error on kamailio server configured with homer (server-B) as
follows,
5(18128) ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver
error on query: Table 'homer_data.sip_capture_call_20160223' doesn't exist
(1146)
5(18128) ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while
submitting query
kindly help me to solve this
--
Thanking you
Achintha
Im fancing some issues due to kamailio behind nat and thought double record
route could help me .
But enabling that :
modparam("rr", "enable_double_rr", 1)
And later using
record_route_advertised_address("X.X.X.X:5060");
is giving me this error:
w_record_route_advertised_address(): Double attempt to record-route
Off course, only a single Record Route is present..
What am i doing wrong?
Using Kamailio 4.3.4
when a ua that has registered over tcp has lost its connection to sip
proxy, t_relay() succeeds and branch failure route is executed:
Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** entering branch_route [CONTACT_BRANCH]
Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** activating t_on_branch_failure("contact)"
Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: Routing INVITE <sip:jh-0x16737a0@192.98.102.10:44186;transport=tcp> to contact via <sip:192.98.102.10:51297;transport=tcp>
Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: WARNING: tm [t_fwd.c:1543]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed
Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********* t_relay success
Feb 21 05:07:31 lohi /usr/bin/sip-proxy[20480]: INFO: ********** entering tm:branch-failure:contact
but when a ua that has registered over websocket over tcp has lost is
connection to sip proxy, t_relay() fails and branch failure route is not
executed:
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** entering branch_route [CONTACT_BRANCH]
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** activating t_on_branch_failure("contact)"
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: Routing INVITE <sip:an41pa6p@ggjdptrk9jm0.invalid;transport=ws> to contact via <sip:192.98.102.10:39057;transport=ws>
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: WARNING: <core> [msg_translator.c:2756]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: <core> [msg_translator.c:1974]: build_req_buf_from_sip_req(): could not create Via header
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches
Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********* t_relay failure
in the latter case, is there some means to get access to the failing
branch in order to be able to unregister it, delete its rtpengine
session, etc?
why can't kamailio behave the same way no matter if pure tcp connection
or one using websocket protocol over tcp connection fails?
-- juha
Hi all,
I am new to Kamailio but have previous experience with Asterisk, Freeswitch
and SIP.
I am using a fairly default config file with some sections removed (E.G
NAT) to simplify things, and some extra logging so I can see the way
packets traverse the configuration.
Config here - http://pastebin.com/raw/CCgT0C78
I have noticed that when a UAC sends a PUBLISH request, Kamailio
immediately responds '404 Not Found' and more worryingly the logs and a
packet capture running on the lo interface indicate it's sending the
PUBLISH request to itself in a loop until it hits the Max-Forwards limit.
Publish URI is PUBLISH sip:201@kamailio.marrold.co.uk;transport=UDP SIP/2.0
I have two questions,
1) Why is this request being sent a 404, when the UAC is in the USRLOC
table?
2) Why is the request looping until Max Forwards is exhausted?
Any help appreciated,
Marrold
I tried installing the pre-compiled version of 4.3 under Debian and had all sorts of problems. I found there were 2 sets of paths involved in documentation/procedures plus a reference to a 3rd which caused a complete failure when the directory referred to was missing. I ran into more problems when I tried to access my server from outside my network (static IP). So I thought I’d bite the bullet and install from source. I get to the bit where I am creating the database and even more errors:
a.. cannot connect to local mysql server
b.. kamdbctl.mysql 112: unary operator expected
In one part of the procedure it refers to db_mysql and another to mysql. Is there a simple installation guide around anywhere which will help me get around all these problems? All I want is a Debian with KAMAILIO running on it!
before route[relay] i set $avp(x)="test"
on the managed onreply_route i can use $avp(x).
on the default onreply_route it is not available. meaning <null>.
is that how it suppose to work?
it makes it difficult to block a certain 200ok, like i meant to, on the
issue - "how to drop 200ok and survive?"
thanks,
uri