Hi All,
Can anyone please provide me the details that how to install kamailio on
Redhat 7.2.
I am following the below url guide for installation of Kamailio.
https://www.kamailio.org/wiki/install/4.4.x/git
when i compile kamailio code I am not able to find certain scripts in
/usr/local/sbin
ne as
media relay. Also I was able to do the TLS encryption.
But when I set the flags for rtpengine, I can just set the whole media as
SRTP or RTP. I am not able to do one side in SRTP and the other with RTP.
(I wanted Kamailio with rtpengine to encrypt calls).
Is that conversion able with
You can use rtpproxy-ng.so that is included (It does the conversion of SDP
packets) with rtpengine media relay from Sipwise , the rtpengine.so is
still a devel module.
Regards.
2014-07-03 10:38 GMT+02:00 Yuriy Gorlichenko :
> Hello. I install kamailio 4.1.3 and it works fine. But I neen pr
Hello,
I need to build a VoIP system who receives SIP and RTP traffic in a public
IP and encrypt both of them with TLS and SRTP respectively. The main point
is to have security inside of the local network (I know it may sound
unuseful).
So, I was trying to build the whole system in Kamailio but
I am having the exact same problems as issue as this user. There is
nothing in the siremis/log/ folder when this happens. I'm running
Siremis 4.0.0 with Kamailio 4.0.4. Has this issue been fixed?
Hello,
On 5/18/13 3:22 AM, Alexander Albert wrote:
> Hi there, i hope you can help
Why did 'transport=tls' 'r2=on' 'lr=on' and 'nat=yes' all disappear
after just adding 'record_route_preset("myurl.com");' (I had to edit my
routing logic to fix a problem with kamailio sending record-route to WAN
clients telling them t
Kamailio SIP server replies with a route to
192.168.1.173:5061 in the
INVITE causing callee to drop call. Both caller and callee are
always
connecting from WAN. rtpproxy is configured with
listen=192.168.1.173
and advertise=WANIP. How can
Below please find a filtered SIP packet capture showing a problem I'm
having with callee '408' responses in kamailio 4.0.4. The '408'
response occurs after both caller and callee successfully establish
2-way audio. In this setup kamailio is running behind a NAT and bot
@lists.sip-router.org
Both callee and caller successfully establish 2-way audio however callee
client disconnects with 408 response. Any troubleshooting ideas would
be appreciated.
Regards,
Aaron
Kamailio Debug Logs and complete SIP packet capture:
http
Both callee and caller successfully establish 2-way audio however callee
client disconnects with 408 response. Any troubleshooting ideas would
be appreciated.
Regards,
Aaron
Kamailio Debug Logs and complete SIP packet capture:
http://www.n00bunlimited.net/pastebin.php?show=64292
,
Aaron
Debug Logs:
http://pastebin.com/9Rw7zSQ0
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I'm pretty sure I'm just missing something very simple here like a port
forward. Let me know if you have any ideas! This setup completely works
when both csipsiimple clients are inside my LAN and kamailio is
configured with alias=192.168.1.LAN. The problem begins when I connect
bo
66.87.76.CALLEE is Callee running csipsimple
192.168.1.LAN is LAN ip running rtpproxy and kamailio
66.41.221.WAN is WAN ip with ports 5004-5082 forwarded to 192.168.1.LAN
And firewall turned off.
## kamailio -v
version: kamailio 4.0.4 (i386/linux) cabe58
flags: STATS: Off
le/modules_k/nathelper.html#id2533354
>
> Reda
>
>
>
> On Tue, May 15, 2012 at 10:54 AM, Openser Kamailio <
> kamailioopen...@gmail.com> wrote:
>
>> hi,
>>
>> I try to change the IP address in connection information field from a 200
>> OK request pas
Thanks Reda, it finally works with fix_nated_sdp("10","@IP Kamailio").
On Tue, May 15, 2012 at 11:00 AM, Reda Aouad wrote:
> Hi,
>
> Try this function from nathelper module: fix_nated_sdp
> http://kamailio.org/docs/modules/stable/modules_k/nathelper.html#id2533354
Thanks but it still doesn't work.
On Tue, May 15, 2012 at 11:00 AM, Reda Aouad wrote:
> Hi,
>
> Try this function from nathelper module: fix_nated_sdp
> http://kamailio.org/docs/modules/stable/modules_k/nathelper.html#id2533354
>
> Reda
>
>
>
> On Tue, May 15,
hi,
I try to change the IP address in connection information field from a 200
OK request passing through Kamailio.
I want the IP address of the Kamailio in connection information. So, I've
tried rtpproxy_answer("","@IP kamailio"), rtppproxy_offer("",&quo
Greetings,
Here's another problem I'm having with kamailio 3.2 and the standard
kamailio.cfg script.
If the calling device is NATed, everything works fine if the original
call gets connected. That is, the INVITE sent to the called device has
the correct NAT fixups applied.
But if
I call rtpproxy_offer() once, but i use also rtpproxy_manage().
When i disable rttproxy_mange(), it works well.
Thanks!
On Wed, May 9, 2012 at 2:57 PM, Andreas Granig wrote:
> Hi,
>
> On 05/09/2012 02:40 PM, Openser Kamailio wrote:
> > *Owner/Connection Information (o)*: douba
Greetings,
I'm having trouble getting parallel forking to work with aliasdb. I'm
running kamailio 3.2 with the standard kamailio.cfg script.
I have found that if an alias points to a set of addresses that all
reference local devices that are registered with the server, kamaili
Hi,
i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up
on the same computer.
When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in
owner/creator session and connection information field with an error, i.e:
*Owner/Connection Information (o)*: doubango
Hi,
I'm currently working with kamailio 3.2. I try to remove the Contact Header
field with remove_hf("Contact") through the file named kamailio.cfg.
When i apply remove_hf("Contact"), it only removes the word Contact, but
the contact address is still present. Let me sho
How do I compare $si to a particular IP address value? This doesn't
seem to work:
if( $si == "123.123.123.123")
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Hello,
I have attempted to ask questions on IRC with no response. I really
appreciate any help from anybody!
I am using Kamailio 3.2.2.
I am unable to get anything loaded in $avp(i:709).
I am attempting to route between 2 media handling asterisk boxes. I am
trying to load balance between
I need similar assistance. Drop me an email, too!
--
Mark Sidell
Partner
Forte, Inc.
919-942-7068
fax 919-969-2844
www.forteinc.com
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http
On Sun, 27 Feb 2011 13:29:10 -0500, Alex Balashov wrote:
> Try this:
>
> $ru = ... new URI ...
> append_branch();
> t_relay();
Thanks for the suggestion, but it didn't work!
The new INVITE goes out with the new URI, but kamailio sends the
INVITE to the IP addres
}
But, it doesn't work. For example, let's say the initial INVITE
resolves to a local device "me@1.1.1.1". This works fine, and the
phone rings. After a timeout, the failure_route executes. The branch
"foo@2.2.2.2" gets appended, and kamailio sends a new INVITE, but
inst
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
if (is_method("INVITE"))
{
setflag(FLT_ACCMISSED);
Sidell
Partner
Forte, Inc.
919-942-7068
fax 919-969-2844
www.forteinc.com
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uot;sip:f...@bar.com", a
target that is not necessarily local to the server?
Many thanks!
--
Mark Sidell
Partner
Forte, Inc.
919-942-7068
fax 919-969-2844
www.forteinc.com
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