Hi,
It does not presenting well if we are using cascaded sip/proxy servers.
Please correct me if it is wrong.
On Mon, Sep 5, 2011 at 14:56, Daniel-Constantin Mierla wrote:
>
>
> On 9/5/11 9:39 AM, Juha Heinanen wrote:
>
>> Daniel-Constantin Mierla writes:
>>
>> thank you for sharing it with u
Hello List,
Can i have UA in the same nat with the box and also UA in different nat at
the same time?
My box is currrently working behind nat with ip 192.168.2.3
and advertised_address is set to sip.mydomain.com
is it possible to append the via header replace the sip.mydomain.com to
192.168.2.3
On 09/05/2011 08:07 PM, Sarat C. Vemuri wrote:
How do I remove the "public IP" entry from the route set before
forwarding the reply to Internal UAC?
You don't -- at least, not in a protocol-compliant way. You can, of
course, do remove_hf("Route"); if you want to, but with effects that are
u
Again, I apologize for this clumsy way of replying.
Ovidiu,
Thanks for the pointer on set_advertised_address. I had to patch rtpproxy
module (and rr module for the two parameters to request_route_preset) since I
am running 3.1.
However, I still have a problem with ACKs after following what yo
Alex,
Sorry about this clumsy way of replying, but I am stuck with Outlook 2010 and
Exchange.
You stated the problem correctly. Carrier GWs (UAS) and internal clients (UAC)
cannot reach each other directly (TCP/UDP). They have to go through Kamailio
and rtpproxy. I am goint to reply to Ovidiu
Hi,
you should look at the following things:
- dispatcher module (for loadbalancing between destinations)
- rtpproxy module if you want to have NAT-relay for clients behind NAT
To what GUI do you refer? Siremis? Probably you should look at the webpage:
http://siremis.asipto.com/
Kind regards,
Ca
Hello Daniel,
Thank You, and many thanks to Alex too!
The fix you made is working fine for me.
Cheers,
Misi
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Hi,
I'm having some problems accounting missed serial forked calls to mysql
database.
I have following setup. Each user can have up to two contacts: telephone
number (routed to asterisk) and SIP URI. Users can specify which contact has
higher priority - which one should ring first. There is also
Hello,
a surprise part of 10 Years SER Conference was granting *special* awards
to the people attending the event that had a major contribution in the
management and development of SER/Kamailio to date. Most of them are
well known in the community, but not all of them.
Hope you like them, it
MÉSZÁROS Mihály writes:
> Second:
> I am experiencing that TLS connection is dropped by sip proxy very
> frequently!
> And i think the tcp connection shouldn't dropped! So my guess is that
> TLS communication is not restarting the timeout counters!
> So in every 2 minutes the sip-router is resta
Hi,
I am experiencing two problem:
First:
I am experiencing that tls read buffer overflow, if i use default value
of tcp_rd_buf_size
If i doubled the size of read buffer and i use
tcp_rd_buf_size=8192
then the problem seems to be solved.
Second:
I am experiencing that TLS connection is dr
Mino Haluz writes:
> I found LCR and carrierroute module, but it does not have peak/offpeak
> feature. Correct me if I am wrong..
run from cron a script that modifies rules based on your needs.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSE
Hi,
I need a module which could allow me to send traffic to various carriers and
it has to support some important features. So some basic ones:
- possibility to re-route the call in case the original route fails
- peak/offpeak conditions (time-based)
- route traffic according to prefix
I found L
5 sep 2011 kl. 10:57 skrev Daniel-Constantin Mierla:
> Many thanks again to our sponsors for free drinks and food at this event:
> Sipgate (http://www.sipgate.de), Amooma (http://www.amooma.de), NG-Voice
> (http://www.ng-voice.com), Asipto (http://www.asipto.com), Frafos
> (http://www.frafos.c
Hello,
first I want to thank to all participants at 10 Years SER event in
Berlin last Friday. We spent great time, at least I did!
We accepted 50 visiting participants to match the maximum capacity of
the conference room, being full booked about 2 weeks before. Just few
people didn't show up
On 09/05/2011 10:58 AM, Daniel-Constantin Mierla wrote:
Hello,
On 9/1/11 3:17 PM, Henning Westerholt wrote:
Hi all,
I'm happy to announce a new developer for the sip-router project:
Sven Knoblich.
Sven is an experienced C/C++ developer in our offices in Karlsruhe and since
about a year part o
Hello
2011/9/5 Stas Bakulin :
> Hello!
>
> What is it right way to pass all rtp traffic through RTPProxy?
> How this can be configured in Kamailio?
* Create RTP proxying session on each INVITE
rtpproxy_offer();
* On every positive reply extract modify answer:
rtpproxy_answer();
* On every BY
Hello,
On 9/1/11 3:17 PM, Henning Westerholt wrote:
Hi all,
I'm happy to announce a new developer for the sip-router project:
Sven Knoblich.
Sven is an experienced C/C++ developer in our offices in Karlsruhe and since
about a year part of the team that develops our VoIP backend services.
Past
On 9/5/11 9:39 AM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
thank you for sharing it with us. Are the diagrams generated as image,
or does also kind of html where can browse the content of SIP messages?
Maye uploading some screenshots on the wiki of github will make it more
appea
Hello,
On 9/2/11 8:40 PM, Anto wrote:
Hello
I want to store in a file access and disconnections to the system. To
access (registrar), I thought scheduling a perl script and executed
with the perl module, the problem I find with disconnections. Since
the user can be disconnected for various reas
Hello!
What is it right way to pass all rtp traffic through RTPProxy?
How this can be configured in Kamailio?
Stas Bakulin
--
og...@kvant-telecom.ru | www.kvant-telecom.ru
Tel: +7 473 233 0330 (128)
_
Daniel-Constantin Mierla writes:
> thank you for sharing it with us. Are the diagrams generated as image,
> or does also kind of html where can browse the content of SIP messages?
> Maye uploading some screenshots on the wiki of github will make it more
> appealing to try.
i tried it and got a
Hello,
On 9/2/11 5:03 PM, Vladimir Broz wrote:
Hi,
for sake of Vaclav's courage, let me introduce the "SIP Analysis &
Testing tools".
This is a set of very powerful utilities that may help you to do SIP
call flows analysis. It can do SIP dialogs and transactions matching
and analysis in la
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