Hello,
I would like to ask, whether Kamailio does connect WebSocket user with SIP
user. Because, I'm trying to do it regarding this code:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex…
and when i'using X-lite i've got
1, Trying
2, 415 Unsupported Media Type
and at Blink client I've got:
1, Trying
2, 488 Not Acceptable Here
as WebSocket client i'm using JsSIP.
Do you think I need to use media server(like Asterisk) to connect it?
Thank you for help!
Patrik
Hi all,
So I moved away from LCR and dispatcher to use drouting but I can't see how
to use probing. In OpenSIPS I can see this is built into the module but
it's not in Kamailio.
Any ideas how I can get round this? I want to use drouting for all my
routing needs.
Cheers
Keith
Hi all,
First time poster here.
I had a freeswitch installation configured to allow users to register with
different domains e.g.
tenant1.mydomain.comtenant2.mydomain.com
..
tenantN.my domain.com
This worked fine. Users were able to call extensions in their own domain
and access their voicemail.
All tenants/domains shared the same diaplan (default).
I followed the great tutorial at
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc
I now have my users (using different domains as above) registering to
Kamailio.
It was a little tricky but I can now place calls between users on the same
domain.
For some reason the variable ${domain_name} (from the tutorial) seems to be
empty so I had to replace it with a channel variable (I'm not in front of
my dialplan now but it was something like ${sip_from_domain}.
Was this the right thing to do? Why does the domain_name variable not
populate with the domain of the user making the call?
Another issue I have is to do with usernames.
I want my users to have alphanumeric usernames e.g.
user1(a)tenant1.example.com
user2(a)tenant1.example.com
But I'd also like them to be able to have an associated extension and
mailbox.
E.g. user1 may be extension 1001 and mailbox 1001
user2 may be extension 1002 and mailbox 1002.
In Kamailio I've handled this by using aliasdb.
So user1.tenant1.example.com has an alias of 1001.tenant1.example.com
Scenario:
User2 dials 1001.
Kamailio evaluates this and determines we want user1. It passes the call to
FS with a $rU of user1. However, even though user1 is registered and
online...the call fails and diverts to voicemail - any ideas?
Also I had to hack the above so that after the alias db lookup Kamailio
sends an $rU of 1001@tenant1...
If anyone has got FS and Kamailio running with multiple domains I would
really love some feedback on routing and dialplan configuration.
Thanks so much in advance,
CD
Hello. I started configure Kamailio 3.3 server with mysql database, that running on the remote server. I customise settings at kamailio.cfg (for BDURL) and kamctlc file (for my db - changed user, password, dbhost and other settings, such as dbengine and etc.) Now, when I tried add new subscriber from kamctl util (kamctl add newuser@kamailioIP passwd). I see "access denied for <myuser>@<interface of kamailio server>", But as I aleady says I configured filew with other settings for db server. Where I must configure ip of dbhost to connect kamctl to my dbserver?
Hi,
I'm using modparam("dialog", "db_mode", 1), but only confirmed dialogs are
shown in the database.
How I can enable to show unconfirmed dialogs in the database? Now are only
in-memory.
Best regards
Hi All,
I'm playing with instant messaging with a PJSIP client, but as soon as I send body with payload to about 100k I'm
getting the error below:
Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: <core> [tcp_main.c:715]: _wbufq_add(): ERROR:
wbufq_add(40552 bytes): write queue full or timeout (0, total 0, last write 51946505 s ago)
Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: tm [../../forward.h:240]: msg_send(): msg_send: ERROR:
tcp_send failed
Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2910]: ERROR: <core> [tcp_main.c:3638]: handle_ser_child():
handle_ser_child: ERROR: received CON_ERROR for 0xb30ddcd8 (id 352), refcnt 2, flags 0x5092
Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR:
t_send_branch: sending request on branch 0 failed
Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR:
sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL)
And the app get
SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL)
Via: SIP/2.0/TLS XX.XX.XX.XX:55859;rport=55859;branch=z9hG4bKPjkkazEy.XcjJkYDCEzwbwUVAONDzcCN8t;alias
From: <sip:aaaaa@test.net>;tag=9gcb.Ay3wSCZNFgs8PNCbdkRq7jfxbDx
To: <sip:bbbbb@test.net>;tag=9a285338e10af1c086406d08dc7f5d79.e457
Call-ID: v9JlsK1AEH4So8AIEfqiGuUOEx5Fub1C
CSeq: 28450 MESSAGE
Server: kamailio (4.1.3 (i386/linux))
Content-Length: 0
--end msg--
Kamailio is already set with tcp_rd_buf_size=204800
Does anyone can suggest how to fix this?
Thanks in advance,
Roberto Fichera.
Hi,
I have a complex setup consisting of two sip server, lets call them main
server and presence server.
The main server manages SIP register, calls, messages and so, however it
does not support presence at all. It returns SIP response "405 Method Not
Allowed" for any SIP PUBLISH, SUBSCRIBE or NOTIFY.
The presence server is a Kamailio server, which unfortunately has no access
to SIP subscriber database (so it can not do authentication), however it
has access to User Location database.
Now at presence server i want to put some kind of security for presence
requests, since i can't do authentication, so i was thinking if i can
verify if request has come from a location registered in location db. I
found method "registered" in registrar module but unfortunately it checks
if R-URI or any given URI is registered, while i want to do check against
"received" address or contact address.
Can you guys suggest any easy way to do it? I am thinking about using
"reg_fetch_contacts" but not sure if it can serve the purpose.
Thank you.
Hi all,
i am testing a very basic IMS core Kamailio-based using as HSS FHoSS.
However, while using SIPp to stress the system, i noticed that the VM
hosting the FHoSS is overloaded (even with 2 registration scenarios per
second), in particular using top i was able to see that the java process
used almost 200% of the CPU (i run the VM using qemu with options -m 1024
-smp 4). Is it normal or have i misconfigured something? Someone else has
the same problem?
Andrea
