Hello all,
The nathelper module in modules_k was split in two:
- nathelper (dealing with signaling);
- rtpproxy (dealing with rtpproxy protocol).
I would like to move the rtpproxy module from "modules_k" into
"modules" and remove rtpproxy functionality from nathelper (s).
This will give to (s) and (k) users:
- rtpproxy: a single module for dealing with rtpproxy servers;
- nathelper: two variants for dealing with NAT signaling.
Next step, will be to merge the two nathelper modules into a single one.
Thoughts?
Regards,
Ovidiu Sas
Hi,
We recently had a problem with the nathelper module and rtpproxy in a
scenario where the SDP offer is sent only in the 200 OK. We use
sip-router 3.1 and rtp-proxy from git master. The sip-router
configuration uses the rtpproxy_offer() and rtpproxy_answer() functions
in appropriate places. The problem is, that the arguments sent to
the rtpproxy, when the ACK with the sdp answer arrives, seems to be not
in the order, that rtpproxy expects.
On the 200 OK, the nathelper module sends callid, to-tag, from-tag to
rtpproxy. On the ACK, the nathelper module sends callid, from-tag,
to-tag (with different command prefixes, of course, but I cannot
remember them atm). The version of rtpproxy that we are using seems to
expect, that the order of arguments sent on the ACK request is the same
as on the 200 OK.
My question: are there any module parameters, to correct this behaviour?
Regards,
Emil
--
Emil Kroymann
VoIP Services Engineer
Email: emil.kroymann(a)isaco.de
Tel: +49-30-203899885
ISACO GmbH
Kurfürstenstraße 79
10787 Berlin
Germany
Amtsgericht Charlottenburg, HRB 112464B
Geschäftsführer: Daniel Frommherz
Hi all,
I would like to use kamailio 3.1 with TLS and verified also a client
certificate.
My tls.cfg file is as follow:
--- tls.cfg ----
.....
.....
[server:MY_IP:5061]
method = TLSv1
verify_certificate = yes
require_certificate = yes
private_key = default_key.pem
certificate = default_cert.pem
ca_list = default_ca.pem
[client:default]
verify_certificate = no
require_certificate = no
....
....
So I want to verify the client certificate, to do that I use
"is_peer_verified()" function in kamailio.cfg.
As tls.cfg shows, I have to send to my clients the CA certificate and
the client certificate (default_cert.pem + default_key.pem - signing
by the CAcert).
This client certificate is unique for all clients.
Everything works fine.
But suppose I wanted to create a client certificate for client 1
(cert_1.crt), and a different client certificate for client 2
(cert_2.crt) and I want to configure kamailio to be able to verified
this different certificates.
Does it possible ? How can I configure the tls.cfg file to do that ?
I try to do something like this:
[server:MY_IP:5061]
method = TLSv1
verify_certificate = yes
require_certificate = yes
private_key = default_key_1.pem
certificate = default_cert_1.pem
private_key = default_key_2.pem
certificate = default_cert_2.pem
ca_list = default_ca.pem
But when kamailio restart it seems that it read only the last couple
of row certifcate/private_key.
Regards,
Daniel G
high.all!
i'm wondering if there is any support of uaCSTA in openser (planned)?
i'm just working on the integration of asterisk (*) environment to OCS 2007
environment, having openSER in the middle (mainly for TCP/UDP translation
and smoothing out the protocol deficienes on both sides). in this setup the
* having the openSER in front is talking to the OCS (and vice versa) via the
OCS mediation server, which is moreorless sending standard SIP messages,
which enables normal softphone (integration to *) of the office
communicator. this configuration is already working...
now i'm planning to go for the CTI integration, where there is no OCS
mediation server in between OCS and openSER, doing the translation of
SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why
i'm looking for a CSTA module or perl programm, which is capable of this
functionality.
afaik for the CTI communication there isn't the full complexity of CSTA
needed, just a subset mainly for call setup and call clearing.
anyone having experience on this topic?
thx & cheers
-hugo
Great Ideas for Small Devices
Hugo Koblmueller
Senior Staff Engineer Software Development COMNEON electronic
technology GmbH & Co. OHG
Freistaedter Strasse 400
4040 Linz
Austria
hugo.koblmueller(a)comneon.com
tel:
fax:
mobile:
Skype ID: +43 (5) 1777 - 15730
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+43 (5) 1777 - 15810
+43 (676) 82051280
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%29+82051280&Email=hugo(a)koblmueller.com>
drhookson
Want to always have my latest info?
<https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join
&invite=1&lang=en> Want a signature like
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this?
please keep the mailing list cc-ed, sending private messages is not in
the spirit of public mailing lists. Others may want to follow up the
discussion now or later.
Thanks,
Daniel
On 2/9/11 5:01 PM, Bruno Bresciani wrote:
> Daniel,
>
> thanks for your reply, really ser-0.8.1.4 is too old but i need to
> solve this problem on that version. My great doubt is Why the aug_free
> function corrupt the url of database after some attemps to reconnect.
> Well, I'll try to understand this question...
>
> Best Regards
>
>
> 2011/2/9 Daniel-Constantin Mierla <miconda(a)gmail.com
> <mailto:miconda@gmail.com>>
>
> Hello,
>
>
> On 2/9/11 2:19 PM, Bruno Bresciani wrote:
>
> Hi,
>
> I've seen the problem in a postgres module (SER-0.8.1.4), if
> the connection fails and module tries to reparse url it fails
> as CON_SQLURL(_h) is corrupted by the function aug_free()
> after some reconnect attempts . When the postgres database
> back to work, some modules doesn't get reconnect because the
> db_url is corrupted. Why this is happening? There are some
> solution for this problem?
>
> ser 0.8.1.4 is soooo old and I cannot fully remember, but I think
> postgres module had no reconnect functionality at all by that time.
>
> However, version 3.1.x of SER (as well as Kamailio flavour) has db
> reconnect functionality for postgres. You can try it and report if
> something is not working, it will be fixed in 3.x, but I think
> nobody is still developing on 0.8.x to backport anything there.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
--
Daniel-Constantin Mierla
http://www.asipto.com
I have been trying to figure this out While using kamailio and rtpproxy,
the caller is not receiving the bye when callee hangs up but audio is
two way and everything seems to be working fine, any one had this issue ?
--
Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
602-234-0917#112
http://www.phoenixinternet.net
Hi everyone,
I am sure someone have been working with this scenario, how about
accounting multiple calls from same account or ip address while using ip
auth ?
--
Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
602-234-0917#112
http://www.phoenixinternet.net
Hello,
this year the FOSDEM developer conference was a again a really nice event, the
first time with an own room dedicated completely to open source telephony
solutions! If you're interested in our presentation about the new p_usrloc
module and how to scale location services with Kamailio, you can find it at
the usual place on our webserver:
http://kamailio.org/events/2011-fosdem/p_usrloc.pdf
Best regards,
Henning
Hi,
I've seen the problem in a postgres module (SER-0.8.1.4), if the connection
fails and module tries to reparse url it fails as CON_SQLURL(_h) is
corrupted by the function aug_free() after some reconnect attempts . When
the postgres database back to work, some modules doesn't get reconnect
because the db_url is corrupted. Why this is happening? There are some
solution for this problem?
Best Regard
Hello,
the presentation I did during Fosdem in Brussels last weekend is
available at:
http://www.kamailio.org/events/2011-fosdem/dcm-sip-web-lua.pdf
The focus was to show how to interact with Kamailio via HTTP and how
Kamailio can interact with Web services via HTTP, using Lua to make it
easier. There are slides for a demo config of sending asynchronous
notifications to Twitter on missed calls (using modules app_lua, mqueue,
sqlops and rtimer).
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
