Dear all, hope to get some information from you guys!
For a redundant service we have to build up SIP proxy/registrar server in a high available scenario. Therefore, we will have 2 SIP proxy/registrar on two completely independent servers. They should be SIP dialog stateful and replicate all the session/call states between each other. If one server fails, the other should have all the SIP dialog information for call handling and accounting. How can this done in a stable and reliable way? Is this feature mature enough to support enterprise requirements?
The only information I have is to build it up with the DISPATCHER module, but it seems more to be a stateless load-balancer instead of a HA module.
Who has experience with HA failover designs with Kamailio and can give me some hints?
Thank you very much in advance
Kind regards
Carel
Carel Reinhard
Security and VoIP Engineer
SRG SSR Switzerland
Telefon direkt +41 31 350 94 07
Carel.Reinhard(a)srgssr.ch<mailto:Carel.Reinhard@srgssr.ch>
Hi,
We have 2 Kamailio 3.0.3 servers that has been running with
carrierroute for about 2 years without any problems. They have 128 MB
shared memory and
modparam("carrierroute", "config_source", "db")
modparam("carrierroute", "db_url", "<DBURL>")
modparam("carrierroute", "fetch_rows", 500)
The carrierroute table is about 91K lines and have been growing slowly.
Suddenly we get this ERROR: carrierroute [cr_data.c:585]: could not
allocate shared memory from available pool after a few "kamctl cr
reload".
I increased the shared memory to 256 MB but with the same result. I
have now increased it to 512 MB and it seems to work better now.
I have noticed this. After a restart the shmem counters is like this:
shmem:total_size = 536870912
shmem:used_size = 28486752
shmem:real_used_size = 40147128
shmem:max_used_size = 41135424
shmem:free_size = 496723784
shmem:fragments = 555
And after the first "kamctl cr reload" it is like this:
shmem:total_size = 536870912
shmem:used_size = 28619016
shmem:real_used_size = 51842768
shmem:max_used_size = 76993616
shmem:free_size = 485028144
shmem:fragments = 722063
Notice the increase in fragments. Sequentials "kamctl cr reload" does
not change the fragments allot.
Any ideas?
--
Morten Isaksen
Hi,
I have this setup:
Mediation server with internal ip -- NAT firewall -- Kamailio proxy --
PSTN gateway
The mediation server sends a contact header with maddr=<internal IP>
fix_nated_contact only changes the host part and leaves the maddr intact.
When the PSTN gateway sends a PRACK or a BYE it puts the <internal IP>
in the RURI, and kamailio then try to forward the message directly to
the <internal IP> and not the NAT ip.
>From RFC 3261 19.1.1:
The maddr parameter indicates the server address to be
contacted for this user, overriding any address derived from
the host field. When an maddr parameter is present, the port
and transport components of the URI apply to the address
indicated in the maddr parameter value. [4] describes the
proper interpretation of the transport, maddr, and hostport in
order to obtain the destination address, port, and transport
for sending a request.
I can see that OpenSIPS has fixed this.
http://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3312423&group_…
Would it be possible to get this patch into Kamailio also?
--
Morten Isaksen
Hello,
GIT branch 3.3 was created to hold the release series 3.3.x -- version
3.3.0 will be out in max. 1 week, a decision after collecting eventual
issues still open to be reported today.
Here are few details about how to use it.
Download and use branch 3.3 from git:
git clone --depth 1 git://git.sip-router.org/sip-router kamailio
cd kamailio
git checkout -b 3.3 origin/3.3
Pushing patches:
- only bug fixes or documentation improvements have to be committed
to branch 3.3
- first commit to master branch (if the bug applies there as well),
cherry-pick the commit to branch 3.3 and push to remote repository
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
Hello,
short note to let everyone know that new features can be committed again
to the master branch -- they will be part of future version 3.4.0 (if no
other numbering release will be decided).
Testing phase of version 3.3.0 within master is ended, a dedicated
branch being created for it (details in a separate message).
Happy coding,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
Hi all!
Do you know any comfortable tools to filter out a certain users call?
e.g. searching in SIP packets for the user pattern, get the media port
out of SDP and capture also RTP+T.38
I found pcapsipdump but have not tried it yet.
Any suggestions or do I have to write my own tool?
thanks
Klaus
Hello,
I am going to create the GIT branch 3.3 in short time -- this branch
will be used for releasing 3.3.x versions.
After that master can take again commits of new features, to be part of
next major release 3.4.x. Branch 3.3 will take only fixes.
From my point of view, I have two issues that I am trying to solve and
the 3.3.0 is ready to go out, namely:
- a solaris sigbus
- pua_dialoginfo issue reported on mailing list
Otherwise, nothing else critcal is out there, to my knowledge. Report
them if you have other ones. Packaging specs were updated, so we are
pretty much ready to release when no relevant issues is open.
The wiki page with what is new in this release was updated a bit as
well, add if you find something missing at:
- http://www.kamailio.org/wiki/features/new-in-devel
Also, contribute to the upgrade guidelines:
- http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0
Cheer,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
Hello
I am not able to dial to pstn with kamailio, the call is routed to my
pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is
hanged up.
My outbound route is:
route[PSTN] {
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not
defined\n");
return;
}
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
route(TOASTERISK);
exit;
return;
}
route[TOASTERISK] {
sl_send_reply("100","Trying");
uac_replace_from("$fn","sip:$fn@$fd");
route(NATMANAGE);
ds_select_dst("1","4");
t_on_failure("1");
t_relay();
}
failure_route[1] {
ds_mark_dst("i");
if (!ds_next_dst()) {
t_reply("503", "Service unavailable: no more dst");
exit;
}
route(TOASTERISK);
}
With a traffic capture i can see the traffic returning to my kamailio
server.
Any suggestion will be appreciated.
