rtpproxy_offer and answer functions have "r" flag described as follows:
r - flags that IP address in SDP should be trusted. Without this flag,
rtpproxy ignores address in the SDP and uses source address of the SIP
message as media address which is passed to the RTP proxy.
how does rtpproxy
Hi Daniel,
Oh I missed the reply for some reason. Thank you very much for both replying
and reminding :)
We do want to receive the replies so closing the connection isn't an option.
Listening on multiple tcp ports sound like a good temporary solution, but lack
of scalability?
Regards,
Allen
On 07/11/13 13:04, Juha Heinanen wrote:
> regarding "r" flag, if sip ua is behind nat, how can ip address in sdp
> be "trusted", because source address of rtp packets does not match the
> one in sdp?
Mediaproxy-ng pays attention to the source address of incoming packets
and adjusts the forwarding
Thanks a lot.
Regards, Volodya Ivanets.
2013/7/11 Charles Chance
> I'm not aware of anything, but others may be.
>
> However, before diving straight into the Asterisk/Kamailio integration,
> may I suggest you familiarise yourself first with Kamailio on its own. In
> my opinion, this will give
After installing Siremis as per the instructions, i get the following error.
Anyone know how to solve this?
"Not able to get the right data table of given form
help.form.HelpWidgetListForm"
After clicking OK I get this:
"Unable to create object from class Openbiz.TableForm. TypeError: table is
Richard Fuchs writes:
> There was a bug in using the "received from" address. We always have the
> "trust address" flag set, so we never noticed it. This is now fixed in
> git master.
thanks. it now worked fine without "r" flag.
regarding "r" flag, if sip ua is behind nat, how can ip address in
I'm not aware of anything, but others may be.
However, before diving straight into the Asterisk/Kamailio integration, may
I suggest you familiarise yourself first with Kamailio on its own. In my
opinion, this will give you a far better understanding of how it all fits
together, and the learning cu
Hello,
Yes I have. But I was hoping find some tutorial with experience sharing.
Because I'm new to Kamailio, official tutorial did not cower all my
questions.
Thanks.
Regards, Volodya Ivanets.
2013/7/11 Charles Chance
> Hello,
>
> Have you looked at
> http://kb.asipto.com/asterisk:realtime:k
Hello,
Have you looked at
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb?
Charles
On 11 July 2013 16:01, Володимир Іванець wrote:
> Hello everyone!
>
> Recently I became interested in concept of putting couple of Asterisks
> servers in cluster. So I tried to repli
Hello everyone!
Recently I became interested in concept of putting couple of Asterisks
servers in cluster. So I tried to replicate this scenario in testing
environment. I was using database back-end for Asterisks settings and
statistics and Kamailio as a load balancer.
Obviously it did not worked
Hi
I am fine with registered users sending and receiving calls from/to PSTN.
Now I route registered users calls to/from PSTN fine
Kamailailio<->freeswitch<>IVR (PSTN)
Best Practice for routing Trusted peers (inbound) to IVR/PSTN (SIP Trunking
Service)
Trusted peer1
On 08.07.2013 12:47, Klaus Feichtinger wrote:
P.S. the body of the original SIP message looks as follows:
[...]
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length:
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
Daniel, I got the answer for gdbthanks.
Arun
From: arun Jayaprakash
To: "mico...@gmail.com" ; Kamailio (SER) - Users Mailing
List
Sent: Thursday, July 11, 2013 6:03 AM
Subject: Re: [SR-Users] RTP proxy high CPU utilization...
Thank you Daniel. Can yo
Thank you. I will take a look at it.
From: Daniel-Constantin Mierla
To: arun Jayaprakash ; Kamailio (SER) - Users
Mailing List
Sent: Thursday, July 11, 2013 6:03 AM
Subject: Re: [SR-Users] configuring an incoming PSTN line...
Hello,
On 7/6/13 5:12 PM,
Hello,
On 7/6/13 5:12 PM, arun Jayaprakash wrote:
Hello,
can someone please give me some pointers as to how to configure the
proxy server to receive incoming calls from a PSTN line. I had not
problem setting up extensions and making outgoing PSTN calls. I am not
able to find any documents (
Thank you Daniel. Can you let me know what do you mean by "attach with gdb",
thanks again.
Regards,
Arun
From: Daniel-Constantin Mierla
To: arun Jayaprakash ; Kamailio (SER) - Users
Mailing List
Sent: Thursday, July 11, 2013 5:57 AM
Subject: Re: [SR-Users]
Hello,
On 7/10/13 1:48 PM, Shankar wrote:
Hello,
The 'type' of 'expires' column in location table is 'datetime' whereas
in table presentity the type is 'int'. Is there any reason behind this?
I hope for the sake of uniformity 'expires' can be changed to 'datetime'.
it was developer choic
On 7/10/13 5:52 PM, arun Jayaprakash wrote:
I am still having issues with this, can someone shed some light on
this so I will know where to start looking for the problem as I am
new to Kamailio. Thank you.
Have you seen the reply:
http://lists.sip-router.org/pipermail/sr-users/2013-July/0788
Hello,
have you seen the reply:
http://lists.sip-router.org/pipermail/sr-users/2013-July/078830.html
?
Cheers,
Daniel
On 7/10/13 11:04 PM, Allen Zhang wrote:
Hi list,
I have a loadbalancer before sipproxies. I call t_reley() on the
dispatcher to forward requests to destinations.
All goe
Hello,
just following the tutorial on the wiki should get you started quickly:
- http://www.kamailio.org/wiki/install/4.0.x/git
But knowing sip is also very important.
Cheers,
Daniel
On 7/11/13 4:55 AM, G. Sherman wrote:
Thanks. I don't see any new logs for REGISTER, but I'm going to just
go
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