Hi,
I am trying to make kamailio PCSCF working with rtpproxy (for NAT traversal).
I would like to ask, if there is any tutorial for this, or if you can
give me any advice.
I also have installed rtpproxy package; #!define WITH_NAT, and defined
rtpproxy port in config file...
But this is not working. I noticed, that in default config file of
pcscf is nothing in routing logic with rtpproxy (or it is unnecessary
?), so I try to copy route[RELAY] route[NAT] route[RTPPROXY] modules
from kamailio (no pcscf) config file... but also with no success.
Can you help me please ? What I am doing wrong ? Do you have any
configuration working with rtpproxy ?
Best wishes
Lubo
Hi,
I'm trying to make when each subscriber dials 7 digits add subscriber based
specific prefix to dialed number defined in subscriber or dialplan table. I
know that it's possible with avp but I'm not professional about avp. could
you send me a successful scenario and config example please?
Hello,
assuming that I want to use contact from 302 response as outbound proxy
but keep the original Request-URI, what should I do? Calling the
revert_uri() after get_redirects() in failure_route doesn't do the trick.
Hi guys.....
the header searching and saving the Q850 code didn't work.
here is what i wrote in the cfg, in the failure route:
*# manage failure routing cases
failure_route[MANAGE_FAILURE] {
xlog("L_CRIT", "failure \n");
if($avp(CAR)=="CALLOUT"){
xlog("L_CRIT", "failure in callout \n");
##if(is_present_hf("Reason")) {
xlog("L_CRIT", "failure in reason\n");
$avp(Q850)=$(hdr(Reason){param.value,cause}{s.int});
xlog("L_CRIT", "failure in reason $avp(Q850)\n");
##}*
here is the log print......
* 3(3433) CRITICAL: <script>: failure
3(3433) CRITICAL: <script>: failure in callout
3(3433) CRITICAL: <script>: failure in reason
3(3433) ERROR: <core> [lvalue.c:347]: non existing right pvar
3(3433) ERROR: <core> [lvalue.c:407]: assignment failed at pos:
(524,15-524,54)
3(3433) CRITICAL: <script>: failure in reason <null>*
the command *is_present_hf("Reason")) * is under ## because of tests...
when it was in use, the script didn't even went in the "if", the command
didn't recognize the header.
**
here is the sip trace for the 486, the header is there...
* Message Header
Call-ID: 6994099017399497785-1328029200-13379
From: <sip:36264529;ss7-cpc=10;cpc=ordinary@172.16.10.3
;user=phone>;tag=6994099017399497785
To: <sip:39955011@172.16.60.161;user=phone>;tag=9299942026613191738
CSeq: 1 INVITE
Via: SIP/2.0/UDP
172.16.60.161;received=172.16.60.161;branch=z9hG4bK08d6.c653f346.2,SIP/2.0/UDP
172.16.10.3:5060;branch=z9hG4bK-6110080000017439-ac100a03-1
Supported: timer,100rel
Reason: Q.850;cause=17;text="UserBusy"
Reason Protocols: Q.850
Cause: 17(0x11)[User busy]
Content-Length: 0*
Apologies if any of the questions below are a bit dumb - I don't pretend
to be an expert in SIP/VOIP - I am just an ordinary user looking for
answers.
Our current setup involves processing a small number of internal sip
accounts (up to 10, no more than that) and one "public" one (with a
separate registrar) in the following way:
On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our
entry point to the public internet, eth1 faces our internal network and
tun0 is a private vpn, which connects all our smartphones to the
internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility.
Up until now, we have been routing voip calls via a commercial (closed
source), very limited, terribly outdated (Pentium code base!) and rather
buggy sip proxy. I had to employ a lot of hacks on our server in order
to route calls as this proxy can only listen on a single interface. It
was also a nightmare to maintain. Unsurprisingly, I decided that enough
is enough and I am now determined to replace it.
We route calls in the following way: all machines (PCs are all Linux
based) & smartphones have their own sip/voip client installed on them
(also using bluetooth). Internal calls are routed via the proxy between
ourselves either on the internal net (eth1), or between the vpn and eth1
(eth1<->tun0).
External calls (going out, i.e. outbound) are routed externally to our
registrar, using a single separate voip account, via eth1<->eth0 or
tun0<->eth0.
As I am now looking to replace our proxy, I looked at Kamailio, but was
soon completely overwhelmed by it (no offence intended, it was just too
much to take at first). I would appreciate if any of you could give me a
hand, or at least point me in the right direction, with the following
issues:
I presume I could configure Kamailio to listen on more than one
interface and act as a proxy. How do I do that, so that it listens on
all 3 interfaces and proxies requests in the following way:
- calls made to <userX>@ourdomain.net to be routed internally via eth1
(internal net) or tun0 (private vpn);
- calls made to anybody else to be routed externally via eth0 (public)
using the separate "public" sip account with our external registrar;
- calls made to the public sip account (from outside - the "public")
need to be routed to a "nominated" internall account (say
<user0>@ourdomain.net);
- all other (internal) calls need to be routed depending on which
interface this account has been registered/logged in - either the
internal net (eth1) or the private vpn (tun0 - the smartphones).
Obviously, calls need to be received (and routed properly) from all 3
interfaces.
Is all of this possible with Kamailio?
I want to avoid unnecessary complexities of the setup (as I already
mentioned above - I am just a user and by no means an expert in
sip/voip) and do not want to deploy something I do not need - I need to
keep the memory footprint to a bare minimum, possibly without
sacrificing performance.
Once this is done, I would then move on to the next phase and use IM &
ENUM, but this is once the above works.
I looked at other alternatives, but I got very confused there as well -
I couldn't figure out what exactly is the difference between, say,
OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best
software to use in order to achieve the above setup?
One last thing - I am a developer by trade and I am not afraid of
"tweaking" things when needed. I was successful in compiling Kamailio
from source (I use Fedora on all our machines) and I was pleased that I
could exclude from the RPM .spec file the modules I think I did not need.
I also made some modification of my own to make the database modules
(mysql, postgresql and unixodbc) configurable in the same way the rest
of the modules are. I could submit patches, if needed, so that these are
incorporated into future releases - how do I do that?
I could not do the same with OpenSIPS, however (which I also tried - out
of curiosity!) - everything there seems to be lumped and compiled
together regardless of whether it is needed or not.
Any help as to helping me with the above issues is greatly appreciated,
many thanks in advance for taking the time!
Hello,
We have been using Asterisk for sometime and over the last year have
started hosting instances for our clients on a vmware platform. These
virtual pbx are located on public ip addresses and each customer has
their own SIP trunk arrangements with various providers. We have
decided to pool our resources and would like to start aggregating
traffic.
From reading online I have installed Kamailio 3.1.x with Asterisk
1.6.x. On top of this I have installed Siremis 3.2 in hopes of using
it as a graphical front end.
I am wanting to use Kamailio as a proxy to the SIP providers and allow
the Asterisk to register only with Kamailio. Is there a link to some
example configs with a tutorial pertaining to this type of deployment?
If this is a good solution, it is my intention to recommend a
consultant be hired to assist with a production server. I just need
to learn allot more, before I can recommend Kamailio for the job.
Do you think it's a good fit?
Thank you for your kind responses,
Greg
Hello all,
I'm trying to set up kamailio as a sip server with
presence support for Bravis.
So far I got the call processing
running on either openser-1.3.2-3build1 or 3.2.0+lucid1 running.
Problem is the notification is not using Content-Type:
application/pidf+xml
My openser config is at http://pastebin.ca/2106401
My kamailio config is pretty much a stock configuration.
It is at http://pastebin.ca/2106404 I enabled mysql and
presence.
I also tried using x-lite but there is no
presence information either.
modparam("presence_xml", "force_active", 1) did not seem to make
a difference. (I cleaned my watchers table)
I can see the database has the information but the notify packets
are just like this:
NOTIFY sip:2001@10.0.76.0:56239 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.23;branch=z9hG4bKcb76.b2dab2d4.0
To: sip:2001@sip.intern.domain;tag=90cd05fe2b0619108127da16a6e721e1
From: sip:2002@sip.intern.domain;tag=10.785.1327490724.1
CSeq: 22
NOTIFY
Call-ID: 90cd05fe-2b06-1910-9660-da16a6e721e1
Content-Length: 0
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Max-Forwards: 70
Event: presence
Contact: <sip:10.0.10.23:5060>
Subscription-State: active;expires=300
Where is my mistake. As I'm new to kamailio please bear with me,
if I just didn't find the right switch.
Thanks
Jonathan Vogt
Hello,
I would like to schedule an IRC meeting to discuss about the development
of the project in 2012.
The topics coming in my mind at this moment:
- roadmap to next major release v3.3.0
- outstanding issues at this time, if any
- packaging - ways to automate the process better
- documentation - suggestions to improve it and attract more
contributions in this field
- ideas for new features
- administration - web site, mailing lists, etc ... do we miss some good
resource out there that should be added in the eco-system?
Please add your suggestions for agenda to the wiki page:
* http://www.kamailio.org/wiki/devel/irc-meetings/2012a
I proposed two dates (Jan 26 and Jan 31, 15:00GMT) and made a Doodle
poll for it:
* http://www.doodle.com/qzteweb62sqax5r6
If you wish other options for dates, let me know. Based on results, we
will announce the final date.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Dear List
I have Kamailio v3.1.5 currently installed and working, as this is
currently in production, could someone please tell me what would be the
safest way to upgrade to v3.2 without losing any data?
Thanking you in advance for your support
Phillip
