Is it possible to monitor via SNMPStats the amount of dialogs currently
open in a particular profile?
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
Alex, I understand your point and appreciate your input highly, but let's say that asterisk is needed as the media server for other reasons. It may not scale as well as the other solutions but we need them to be there to provide other services. I just wanted to know whether there was any difference aside from the scalabilty issue.
We were under the impression that it was imperative to use either media-proxy or rtp-proxy but after examining in more detail how the 2 solutions work we find no reason not to use asterisk as the media proxy as well. Unless of course we are missing the big picture here and surely will find out once we set out to test it. But given the deep knowledge of a lot of people in this forum we deemed necessary to ask about the feasibility of this workaround.
If somebody detects any flaw on this solution or if there is someting else critical or not that we are missing we would love to hear it .
thank you in advance and thank you Alex again,
jp
----- Original Message ----
From: Alex Balashov <abalashov(a)evaristesys.com>
To: Juan Perez <jperezsip2008(a)yahoo.com>
Sent: Monday, March 16, 2009 6:51:34 PM
Subject: Re: [Kamailio-Users] Using Asterisk as media server
No, because the amount of throughput you get per media/RTP proxy is much
higher than you would get through something whose entire core is structured
to behave as a PBX, with all the attendant features and latent event loop
characteristics. A media relay does one thing and one thing only, and it
does it well. Asterisk does many things, among them media relay.
So, while you could use Asterisk in this capacity, I don't see the point.
It'd be a waste of time. Also, you don't really want to get Asterisk
involved in the signaling layer (as a B2BUA) if you can help it. It
introduces unnecessary complexity and possible points of interop problems
and failure. With a media relay that is controlled by API hooks into
Kamailio, you don't have that problem.
On Mon, 16 Mar 2009 15:22:27 -0700 (PDT), Juan Perez
<jperezsip2008(a)yahoo.com> wrote:
>
> thanks alex, but actually there will be several asterisk servers, managed
> by the LCR module, that should be OK right?
> it should be the same as adding several media-proxies or rtp-proxies, am
I
> wrong?
>
>
>
>
>
>
>
> ----- Original Message ----
> From: Alex Balashov <abalashov(a)evaristesys.com>
> To: Juan Perez <jperezsip2008(a)yahoo.com>
> Cc: users(a)lists.kamailio.org
> Sent: Monday, March 16, 2009 3:41:46 PM
> Subject: Re: [Kamailio-Users] Using Asterisk as media server
>
> No, nothing needs to be modified.
>
> However, in this scenario you are limited to Asterisk's performance
> characteristics in terms of the number of concurrent calls that can be
> handled. Just about any media relay will be much more scalable and
> lightweight.
>
> Asterisk is a PBX and a nice feature server. You don't build SIP service
> delivery platforms out of Asterisk used as a general-purpose call router.
>
> Juan Perez wrote:
>
>> Hello
>>
>> In this scenario, customers behind NAT register in kamailio, records are
> stored in a db etc. Kamailio uses Asterisk boxes as gateways to send
calls
> to PSTN.
>> Is it possible to use Asterisk as the media server instead of
> media-proxy or rtp-proxy?
>> In this case the SDP that Asterisk sends in the 180 message needs to me
> modified?
>>
>> thank you
>> jp
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (OpenSER) - Users mailing list
>> Users(a)lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
> -- Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
I have a 0.9.6 SER proxy running and one particular UA that is creating some problems with a RECORD-ROUTE header I am not handling correctly. Here is how I have my record_route configured (XXXs are replaced by my actual proxy IP):
if (!(uri=~^"sip:XXX.XXX.XXX.XXX"))
{
record_route_preset("XXX.XXX.XXX.XXX:5060;nat=yes");
};
When inbound traffic arrives for this particular UA, the INVITE is passed along. when the 200 OK comes back, there is a Record-Route header that looks like this (YYYs are replaced with the UA's public IP):
RECORD-ROUTE: <sip:YYY.YYY.YYY.YYY:5060;lr;transport=udp>
Instead of adding a new Record-Route header with my IP before this one, the message is simply proxied back to the call origination. The ACK is then sent DIRECTLY to the UA, bipassing my proxy...as well as the BYE/200 OK that terminates the call....thus rendering my stateful proxy as stateless.
Can I remove the RECORD-ROUTE that my terminating UA is attaching? If not, why am I not adding my own Record-Route header like normal?
What am I missing?
Thanks,
Steve
_________________________________________________________________
Windows Live™ Groups: Create an online spot for your favorite groups to meet.
http://windowslive.com/online/groups?ocid=TXT_TAGLM_WL_groups_032009
Hi,
I am facing problem with handling CANCEL/487 Request Terminated in a
case of a call hunting. The problem is that the call terminating carrier
doesn't process the ACK properly and keeps sending 487 messages.
Here is an example.
Let's say UAC1 has the following rules of call hunting :
1. call +33123456 for 10s
2. call +33789456 for 10s
Let's say that an incoming call to UAC1 triggers the call hunting.
Here is a diagram :
openSER -> Terminating Carrier (TC) : INVITE +33123456
... (time out : +33123456 didn't answer in 10s, so openSER sends an
INVITE to +33789456 - and let's say to an other Terminating carrier TC2
- and CANCEL the current INVITE to TC)
openSER -> TC2 : INVITE : +33789456
openSER -> TC : CANCEL
TC -> openSER : 487 Request Terminated
openSER -> TC : ACK
The problem is now TC doesn't process the ACK correctly and keeps
sending 487. So, in the case of +33789456 answering the call (a 200 OK
is sent to openSER), openSER will keep relaying the 487 to TC2 and TC2
will then send a BYE a terminate the call :
TC -> openSER : 487
openSER -> TC : ACK
TC -> openSER : 487
openSER -> TC : ACK
TC2 -> openSER : OK
openSER -> TC2 : ACK
<-- The call is taking place -->
TC -> openSER : 487
openSER -> TC : ACK
openser -> TC2 : 487 (openSER relays the 487 once the call has been
established to TC2)
TC2 -> openSER : ACK
TC2 -> openSER : BYE
<-- Call is ended but should not -->
According to the RFC, once a call has been OKed and a 487 is received,
TC2 may go on with the call or send a BYE (up to it). So it behaves the
right way (chapter 15. end of 3rd paragraph : "If the INVITE results in
2xx final response(s) to the INVITE, this means that a UAS accepted the
invitation while the CANCEL was in progress. The UAC MAY continue with
the sessions established by any 2xx responses, or MAY terminate them
with BYE.").
My question is then : is there a way to prevent this behavior when a
terminating carrier doesn't behave correctly, either by preventing
relaying of the 487 once it has been ACKed or once the call has been
OKed (but I guess we are not RFC compliant then) ?
Thanks
Bye,
Guillaume
Hi all,
i'll present in the next week the Kamailio project at the FFG 2009 conference.
Topics of my talk are:
- overview about Kamailio
- applications and usage scenarios
- new and interesting stuff in the 1.5.0 release
- future developments, the sip-router project
Date: Thursday, 12.03.2009
Place: Karlsruhe, Germany
Link to the abstract (in german as the talk will be also held in german):
http://www.guug.de/veranstaltungen/ffg2009/abstracts.html#3_2_1
More information about the event:
http://www.guug.de/veranstaltungen/ffg2009/
I'll provide the slides and eventual some notes after the talk as usual.
Cheers,
Henning
Hello,
sometimes when I'm investigating problems in my cfg files i think it
could be easier to understand them if i could put some breakpoints on
a gdb session.
I don't grok makefiles yet. But I suppose there would be a command
line option for the kamailio Makefile to instruct it to build the core
and modules with debug symbols.
So, how do you guys usually compile kamailio with debug symbols?
regards,
takeshi.
Please cc all the time to mailing lists.
If you reported an issue and solved it, then it is nice to close the
discussion thread by saying what was wrong and how was solved. Others
coming later and reading mailing list archive can learn from your
experience.
Thanks,
Daniel
On 03/11/2009 10:47 AM, Sara EL KABIRI wrote:
> Hello,
>
> Thanks a lot for you help.
> I've already solved the problem :-)
>
> Regards,
> Sara
>
> On Wed, Mar 11, 2009 at 8:09 AM, Daniel-Constantin Mierla
> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
>
> Hello,
>
>
> On 03/09/2009 08:48 PM, Sara EL KABIRI wrote:
>
>
> Hello,
>
> I've installed kamailio 1.4.3 (without tls) on ubuntu by
> following the provided INSTALL guide. I want to use it as a
> presence server but i've got some problems starting it.
>
> "kamctl start" returns nothing.
>
> check the syslog file for errors. There you can get some hint.
>
> Alternatively, set log_stderror=yes in your config file and run
> "kamailio" (/usr/sbin/kamailio or /usr/local/sbin/kamailio if
> installed from sources) and see what you get on the screen.
>
> Cheers,
> Daniel
>
> and "kamctl moni" returns "Error opening kamailio's FIFO
> /tmp/kamailio_fifo"
> I've checked in my .cfg file and the 'loadmodule' and
> 'modparam' lines related to mi-fifo module are uncommented.
> I've checked also the mi-fifo module was installed indeed
> (mi_fifo.so appears in /usr/local/lib/kamailio/modules directory).
>
>
> Could anyone help please?
>
> --
> Sara
> ------------------------------------------------------------------------
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org>
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
>
>
> --
> S.
--
Daniel-Constantin Mierla
http://www.asipto.com
I'm using
CentOS 5.2 ( 2.6.18-92.1.22.el5)
I tried to install it with yum, but it says:
"Package expat - 1.95.8-8.2.1.i386 is already installed"
Could be some other package missing?
Thanks
Joao Pereira
TCB wrote:
> you need to install expat-dev , what distro are you using ?
>
> On Mon, Mar 16, 2009 at 8:54 PM, gomespereira <gomespereira(a)startel.pt
> <mailto:gomespereira@startel.pt>> wrote:
>
> Hello
> While compiling Kamailio's jabber module.... this error appears:
>
> In file included from xjab_jcon.c:43:
> xode.h:31:19: error: expat.h: No such file or directory
> In file included from xjab_jcon.c:43:
> xode.h:281: error: expected specifier-qualifier-list before
> âXML_Parserâ
> make[1]: *** [xjab_jcon.o] Error 1
> make[1]: Leaving directory
> `/usr/src/kamailio/kamailio-1.5.0-notls/modules/jabber'
> make: *** [modules] Error 2
>
> Its only works if I exclude the jabber module.
> What could be the problem? Are there any modules dependencies?
> Thanks
> Regards
> Joao Pereira
>
>
>
>
> --
> StarTel - A Rede Livre
> Joao Gomes Pereira
> www.startel.pt <http://www.startel.pt>
> +351 304500650
> sip: gomespereira(a)startel.pt <mailto:gomespereira@startel.pt>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org>
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> --
> TC
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespereira(a)startel.pt
Hello
In this scenario, customers behind NAT register in kamailio, records are stored in a db etc. Kamailio uses Asterisk boxes as gateways to send calls to PSTN.
Is it possible to use Asterisk as the media server instead of media-proxy or rtp-proxy?
In this case the SDP that Asterisk sends in the 180 message needs to me modified?
thank you
jp
Hello
We are implementing an architecture with Kamailio and Asterisk with
real time and are debating who should be the Registrar for the SIP
clients.
We are biased towards leaving that to Kamailio and use Asterisk to provide the media services: vmail, conference etc. It looks like this is the recommended way as per several tutorials on the net suggest that solution. As a down side we will have to use a combination of near-end [STUN, ALG] and far-end Nat solutions [rtp-proxy] to handle the media and also are kind of worried about how we should handle the Cass 5 features tipical of a residential line.
But a member of our team suggests to use Asterisk for that and have Kamailio forwarding the REGISTER request from the clients to the Asterisk. Asterisk can handle media and also users behind a NAT. We have never done it that way and we are not sure whether it is possible or not, whether it is acceptable SIP-wise or not. For example: if the proxy is load balancing the REGISTER messages then the clients should be given the IP of the proxy to send the REGISTERs and the NAT binding in the customer's NAT/Firewall will be associated to the IP of the proxy, not the IP of the Asterisk.
On then other hand, Asterisk as a PBX supports a lot of Class 5 features natively.
We would like to hear your input, suggestions and comments about this, what other people are doing. If we had only a few user then 1 Asterisk box should be enough but we are planning on having way more that what an Asterisk machine alone can handle, that is why we want to load balance using kamailio.
regards
jp
