Dear SR-Users,
I am using kamailio for decoding of SIP Packets with Message Body as
a GSM PDU. But I am not sure of the modules which i need to use for the
same. Some help on this would be greatly appreciated.
Thanks,
~Ramya.
Hi everybody,
I have been trying to setup a new installation of kamailio, and I need the
user location table to have a specific name. This will allow two kamailio
servers to use the same database for their registry information.
I cannot find any modparam in the documentation that allows me to set the
table name for the usrloc. Is there any way to do this? It seems like an
oversight, every other module I have been able to change the table name
without a problem.
Thank you,
Alex
Hi, all!
I had in config:
#!define KAMDB "cluster://postcash"
loadmodule "db_cluster.so"
modparam("db_cluster", "connection","con1=>postgres://user1:password1@host1/kamailio")
modparam("db_cluster", "connection","con2=>postgres://user2:password2@host2/kamailio")
modparam("db_cluster", "cluster", "postcash=>con1=9s8p;con2=9s8p")
modparam("usrloc", "db_url", KAMDB)
and save registered user's locations in cluster
after some server's restarts second base has empty set.
How can I see server's list with status?
How can I set second server worked? (I check - all OK with it)
--
WBR, Victor
JID: coyote(a)bks.tv
JID: coyote(a)bryansktel.ru
I use FREE operation system: 3.12.9-calculate GNU/Linux
Just trying to get to grips with Kamailio and thought I'd say hello to
you all...
Not sure if this is of any interest, but I've found it quite hard to get
into Kamailio compared to (say) Freeswitch. This is not meant as
criticism at all, but if you're looking to make it easier, I may be able
to help!
Installation was more or less OK, although a couple of slightly blind
alleys and lack of concrete examples of config files.
I believe I have a working server, but where to go next has left me a
little stumped!
What I'm trying to do is have a SIP load-balancer such that I can have a
set of Freeswitch servers that can have traffic routed to them on a
controlled basis, allowing me to point certain accounts at certain
servers and take individual servers out of service for upgrade etc.
I think this is SIP server redirection. If there was an example of how
to do it - eg. what files need to be changed, any database entries, etc.
that would be most useful. Unfortunately, all my Googling/list searching
so far has only come up with questions that were either badly worded or
not answered. Happy to turn anything I come up with into a
beginner-friendly case-study.
Hello all,
So I have three machines, we don't care about audio for this problem, so
everything I mention here is SIP related.
Freeswitch <--> Kamailio 3.3.2 <--> Asterisk
1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy.
2. Kamailio replies 100 Trying and forwards to Freeswitch
3. Freeswitch replies 100 Trying
4. Freeswitch replies 180 Ringing to Kamailio
5. Kamailio routes the answer to Asterisk
6. Freeswitch replies 200 OK to Kamailio
7. Kamailio replies 200 OK to Asterisk
8. Asterisk replies ACK to Kamailio
9. Asterisk sends a re-INVITE to Freeswitch through Kamailio
10. Kamailio routes the re-INVITE to freeswitch
11. Kamailio routes the ACK to freeswitch.
12. Freeswitch replies 500 Server error because it got a re-INVITE
before the ACK.
So, my problem is that Kamailio seems to process my re-INVITE more
quickly than the ACK. So Freeswitch replies an error because it got the
re-INVITE before the ACK.
So my "solution" is to add a usleep(20); for re-INVITEs on Kamailio, but
I think this is a lousy solution.
Has anyone here had to deal with problems where Kamailio routes a
re-INVITE faster than an ACK causing endpoints to return error
messages? Has anyone had to deal with a similar issue?
Thanks,
David
I'm having trouble divining the proper way to authenticate calls to
remote destinations that require it. I'm hoping someone can clear it up.
Freeswitch client server A is authenticated to Kamailio.
Freeswitch test server B is authenticated to Kamailio.
Calls from A -> Kamailio -> B fail with rejection of the authentication
information that A is using to Kamailio.
As far as I understand, I need to rewrite the authentication information
with UAC? However, UAC's uacreg table seems to tie remote users to
remote destinations, when all I want to do is authenticate all calls
routed to the destination regardless of what the user or the source is.
Am I using the wrong module?
Hi,
Can someone clarify here please?
Regards,
Shankar
From: Shankar [mailto:shankar.rk@plintron.com]
Sent: Monday, February 03, 2014 10:53 AM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Regd. MAX_LDG_LOCKS
Hello,
Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c?
Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash
size is increased?
Regards,
Shankar
Dear All,
Greetings,
I have configured Kamailio (4.0.3) with mediaproxy server, with this set-up
audio/video calls are going good and hence NAT issue has solved.
I have used two soft clients: 1) IMSDroid 2) Bria.
But i have a query about media-proxy statistics i.e in the syslog, the info
about media-proxy activity is shown as below:
*Case 1*: when Call between two IMSDroid clients.
caller_codec : 'Unknown(73)'
callee_codec : 'Unknown(73)'
callee_ua : 'unknown agent'.
*Case 2* : When call between two Bria clients.
'callee_ua': 'unknown agent'.
(Calls using Bria clients media-proxy's syslog shows Audio codec names, but
using IMSdroid clients it is showing 'Unknown' for Audio codec names ).
Find the attachment below for the full syslog about the mentioned unknown
behaviour. And also find my kamailio.cfg file.
And also logging in the syslog as shown like RTP: Unknown , RTCP: Unknown.
In this context, there is a jitter in audio calls and pixelled video
sometimes, So is there any wrong with the above mentioned "unknown' factor
in this problem of jitter/pixelled audio/video calls ?
(AFAIK these payload type 73 (i.e 72-76) is reserved for RTCP conflict
avoidance. right? )
What could be the problem for this 'Unknown' issue?
How can i solve this issue ? Anything can be done in kamailio script?
And how these RTCP payload types (72-76) plays a role in audio/video calls?
Any suggestions will greatly help. please help me in clarifying these
issues.
Thanks in advance.
Regards,
Nandini
Dear Kamailio'ns
I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on
Ubuntu 12.04. I am experiencing Latency, jitter, pixel led, choppy video
errors in my set-up. So i wanted increase buffer sizes in my kamailio
server , Mediaproxy server and Ubuntu also.
In that regards i have increased Shared memory value to 512 MB and Package
memory value 16 MB.
BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the
value for 'MAX_RECV_BUFFER_SIZE' ?
And are there any settings to increase Buffer Size of Mediaproxy server ?
Is this changes will affect on my issues ?
PS: I even changed ubuntu settings also :
net.core.rmem_max= 8388608
net.core.wmem_max= 8388608
(but these values reset to default values again when i restarted system.)
Please help me in resolving this issues.
Any help will appreciate.
Regards,
Ravi