Hi,
I was trying to debugg if I hit a memory leak issue. I delete "-DF_MALLOC"
line and insert a "-DDBG_QM_MALLOC" line of DEFS string in Makefile.def. I
was able to compile openser with new Makefile.def. But when running openser,
I get error, fm_free() is not defined, in many modules. And there is a flag
mismatch between openser core and modules compilation. Core compilation has
DDBG_QM_MALLOC flag, but module compilation still uses F_MALLOC instead of
DDBG_QM_MALLOC. Is there anything else I need to change in Makefile.defs?
Thanks,
-Joy
On Tuesday 16 December 2008, Juan Asencio wrote:
> [..]
> I need to test the carrierroute module in Kamailio and I been reading the
> documentation about it.
>
> http://www.kamailio.org/docs/modules/devel/carrierroute.html#id2506610
>
> But this something entirely new for me. So I have a couple of questions
> that I hope you can answer or guide me to where I could found an answer.
>
> I need to test it with two Asterisk gateways. So I guess I need to edit my
> kamailio.cfg file as the example on "Example 1.19. Configuration example -
> module configuration" My question is: On prefix 49 and prefix NULL, are
> these the two routes available? If the prefix is 49, so the call go
> through proxy1.localdomain or proxy2.localdomain and if the prefix is NULL
> it would go through register1.localdomain or register2.localdomain?
>
> It is proxy and register the gateways?
Hi Juan,
please write this questions to the user list.
The config in question is probably:
domain proxy {
prefix 49 {
...
}
}
domain register {
prefix NULL {
...
}
}
This defines two "domains" in the default carrier tree. The 'proxy' domain
will be used if you specify 'proxy' in the cr_route calls, the same applies
to the 'register' case. So if you want to distribute traffic between two
asterisk, just use one domain, and a empty (NULL) prefix. See the following
example config:
domain foobar {
prefix 49 {
max_targets = 2
target box1.localdomain {
prob = 0.500000
hash_index = 1
status = 1
comment = "test target 1"
}
target box2.localdomain {
prob = 0.500000
hash_index = 2
status = 1
comment = "test target 2"
}
}
prefix NULL {
max_targets = 1
target box3.localdomain {
prob = 1.000000
hash_index = 1
status = 1
comment = "test target 3"
}
}
}
This way all traffic with the prefix '49' will be equally distributed between
box1 and box2, and all other prefixes will be routed to box3. Hope this is
now more clear.
> Could you suggest me what to read in order to get a better understanding
> of this module?
You probably need to experiment a bit. If you increase the log level to INFO,
then the module will report what it does, increase to DEBUG to get even more
informations, e.g. about domain searching. Take a look to the mailing list
archives, if you get stuck just ask at the user list.
Cheers,
Henning
Hello,
I have uploaded xlog and (stripped) pv Kamailio/Openser modules to
sip-router.org site at:
http://sip-router.org/pub/tmp/miconda/
If you are willing to test them, then just get the branch "master" from
git and merge with "daniel/pv". Details about GIT repository:
http://sip-router.org/2008/11/18/git-repository-online/
The pv module does not include yet the pseudo-variables and
transformations dependent of K/O extensions (e.g., branch, script flags,
...). Probably one important to decide upon is the AVP (to start a new
discussion about it separately). The ones missing now will be gradually
activated as dependent code is sorted out. Testing and feedback is welcome.
Andrei is doing heavy work to improve the config script interpreter for
string and arithmetic operations, you can track the commits at:
http://lists.sip-router.org/pipermail/sr-dev/
Once work is finished and the developers' branches are merged, we can
mark an important milestone - K/O pseudo-variables working in sip-router
config file, opening full functionality for large set of K/O modules and
clean compatibility of config files for many typical deployments.
SIP-Router is able to load now both types of modules: ser and k/o. DB
APIs from both projects were tested as library, from K/O point of view
mi and statistics API are the extensions yet to be made libraries for
sip-router.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello,
I could install CDRTool.
I have in CDRs signallings informations but dont have Rating information.
Have you an idea?
Id Start time Sip Proxy SIP caller In SIP destination Out Dur Price KBIn
KBOut Status Codec
<http://192.168.1.150/CDRTool/callsearch.phtml> 1N 2008-12-16 11:23:24
2000(a)192.168.1.155 33170725015@192.168.1.15100:19 Ok (200)
Signalling information
<http://192.168.1.150/CDRTool/callsearch.phtml?cdr_source=ser_radius&cdr_tab
le=radacct&order_by=RadAcctId&order_type=DESC&begin_datetime=1229420460&end_
datetime=1229468100&maxrowsperpage=15&action=search&call_id=YjUwMzg1OGVlYzU4
M2QwNzNkZjQ4ZjNmYzUyZTczNGM.> Click here to show only this call id
Call id:
YjUwMzg1OGVlYzU4M2QwNzNkZjQ4ZjNmYzUyZTczNGM.
<javascript:void(null);> Click here to see the SIP trace for this call
From/to tags:
636e0825/as5d7d6d2a
Start time:
2008-12-16 11:23:24 Europe/Amsterdam
Stop time:
2008-12-16 11:23:43
Method:
Invite from 192.168.1.60:39366
From:
2000(a)192.168.1.155
Domain:
192.168.1.155
To (dialed URI):
33170725015(a)192.168.1.155
Canonical URI:
33170725015(a)192.168.1.151
Next hop URI:
33170725015(a)192.168.1.151
Billing Party:
2000(a)192.168.1.155
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Hello,
today the PV&T are documented in several places:
- core PV&T were on dokuwiki page
- PV exported by modules in the README
As of now, all PV&T from core were moved in pv module, but they are sill
documented on dokuwiki page.
However, we should get to a consistent way. My suggestion is to document
PV&T in docuviki, mentioning which module to load to become available.
We keep a single page a reference, being easier to find all the PV&T,
rather than going over all modules' README. I think user experience is
more straightforward.
The module readme should maintain a small section, listing the names of
PV&T plus a link to dokuwiki page. Any other opinions?
I am just about to document the new $T_rpl(pv) and $T_req(pv)
pseudo-variables from TM, so I am waiting to do it in the proper place:
http://lists.kamailio.org/pipermail/devel/2008-December/017108.html
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello, I'm a newbie with OpenSER/Kamailio so please bear with me.
I'm trying to use OpenSER 1.3.4 and rtpproxy to connect two IP PBXs.
The two PBXs (and the phones they serve) cannot talk directly to each
other, they need to go through my OpenSER system. One system is a
Cisco CallManager cluster (v5.1.3) which I administer and the other is
a Mitel system run by another organization. I have a CentOS box
running OpenSER and rtpproxy with two interfaces one on each network.
Routing on the CentOS box appears correct as I can ping everything I
expect to. I've gotten OpenSER configured so that SIP messages are
sent between the PBXs, but the SDPs aren't getting rewritten properly
and the RTP isn't getting passed through the rtpproxy. I've attached
my config so far, can anyone point out where I've gone wrong or point
me to some example configs that might help me out? All of the example
configs that I've found so far seem to be oriented towards SIP
endpoints.
--
Jeff Ollie
"You know, I used to think it was awful that life was so unfair. Then
I thought, wouldn't it be much worse if life were fair, and all the
terrible things that happen to us come because we actually deserve
them? So, now I take great comfort in the general hostility and
unfairness of the universe."
-- Marcus to Franklin in Babylon 5: "A Late Delivery from Avalon"
Hello,
while adapting the PV API, I got into the AVP script flags? Is anyone
still using them? If yes, for what purpose, maybe we can figure out a
nicer way and drop these flags.
We have to find the solution to make AVPs from kamailio/openser and ser
compatible in the new sip-router.org project.
Just for those not being sure what I am talking about, avp names can
have flags after the name type specifier, like:
$avp(i3:23) - meaning that flags 1 and 2 (bitmask ...0011) are set for
that avp.
As I haven't seen any related discussions for quite some time, the
functionality might not be updated all over the code.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello,
as we have a big number of modules and we did updates like renaming,
merging, ..., I have created a dokuwiki page to track a module.
Now it is a table that should show for each module the date (release)
when it was introduced, current status and description about that. Feel
free to enhance the format, the page is located at:
http://www.kamailio.org/dokuwiki/doku.php/modules:status
Now it is just a skeleton. I hope all of you will contribute there. You
can add to description anything you believe is relevant for that module.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi all, I'm trying to configure my kamailio sip server with two asterisk
gateways.
I have look for some tutorials, but the one in Kamailio website have me a
little confuse, because the files content is different. I guess it must be
the .cfg file from Openser the one that is describe on the website.
http://kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
Could any one suggest me some tutorials, to see how the kamailio.cfg file
should be edited?
Best regards,
Juan.-