ok :)
I replied to your last mail, just read this one. You could've replied to
the already send mail instead of composing a new one. Anyways good to hear
that.
On Tue, May 15, 2012 at 11:51 PM, Magnus magnus.ke...@gmail.com wrote:
Just in case anyone comes up with same issue - post solving the
Hello all,
Following through the manual at :
http://kb.asipto.com/siremis:install32x:mi-commands I'm able to send MI
commands on external/remote server But next hurdle is how can I define
multiple remote servers i.e
Remote name=remote0 address=127.0.0.1 port=8033/
Remote name=remote1
, it will be committed and released with the next
version. In case you plan more consistent contributions, once you have
them
developed, I can grant write access to sourceforge.net git repository
for
siremis, so you will be able to commit yourself.
Regards,
Ramona
On 5/14/12 8:33 AM, SamyGo
Hi,
If you haven;t gotten any positive response try asking the question on
broader forums and also on CISCO forums. Ideally any SIP version which
registers on any SIP serevrs should work perfectly fine with kamailio as
well.
Regards,
Sammy
On Wed, May 9, 2012 at 8:21 PM, KA Veelenturf
Hi,
I can see you've tried calling route[NATMANAGE] just before the
route[TOVOICEMAIL] ! and that didn't work. Can you paste your configuration
as well as a SIP trace for a voicemail call ! some logs of the same calls
will help too.
Regards,
Sammy
On Wed, May 9, 2012 at 9:10 PM,
:
In that case it maybe a network or security problem.
Can you ping the remote server? Have you checked your firewall settings on
both servers?
Reda
On Thu, May 10, 2012 at 7:21 AM, SamyGo govoi...@gmail.com wrote:
Hello Reda,
I really appreciate that, yes I forgot to mention that I've already
?
Reda
On Thu, May 10, 2012 at 8:20 AM, SamyGo govoi...@gmail.com wrote:
Yes Reda, those are basic things, I can telnet to any other port on that
server - ping it but the only port which instantly drops my connection is
8033 !! even if netstat shows its listening to it on all interfaces
/a]: unable to
write to socket [22]: Invalid argument',
Any directions from here?
Regards,
Sammy
On Thu, May 10, 2012 at 11:53 AM, SamyGo govoi...@gmail.com wrote:
hmmm..the only thing that I've missed , my favourite, tcpdump/wireshark.
Now I try that and come back here.
On Thu, May 10, 2012 at 11:41
Hello list,
I'm trying to have my Siremis interface send MI commands to multiple
kamailio servers i.e reload dispatcher of all the kamailio servers when I
reload from Siremis interface.
The issue Im facing is that the commands dont get executed on any other
server except localhost/
as
I think its some path issue in your kamialio ctl config file !
On Tue, May 1, 2012 at 5:16 PM, Vineet Menon mvineetme...@gmail.com wrote:
Hi,
I have a kamailio installation running, but in kamctl the moni command
works, but stop, restart, start command doesn't work...
why is it so??
, Karsten Horsmann khorsm...@gmail.comwrote:
Hi SamyGo,
thanks for the hint! It works with rtpproxy -l internal.ip/external.ip
with the slash between it.
Strange - on older posts i read the opposite to get bridging working
--
Mit freundlichen Grüßen
*Karsten Horsmann
Hi,
These are not the sip-traces we are looking for. Please attach sipgrep /
ngrep / tcpdump traces so someone can help you better.
Regards,
Sammy.
On Thu, Apr 12, 2012 at 2:24 PM, Karsten Horsmann khorsm...@gmail.comwrote:
Hello!
I get step by step to my multihomed setup and have now
Hi,
I am using the 3.2.2 version with RTPproxy in bridging mode - the only time
I was having segmentation fault (with any version) was when I started
RTPproxy in bridging mode(in my mind) but it wasn't and I was missing a /
in-between the PublicIP and the Private IP - Which, when I read your
for the beginner questions, but I have found no place where this
questions are adressed.
On Mon, Apr 9, 2012 at 9:17 PM, SamyGo govoi...@gmail.com wrote:
:-|
http://www.kamailio.org/w/sip-router-releases/
On Mon, Apr 9, 2012 at 11:06 PM, Daniel Gonzalez
gonva...@gonvaled.comwrote:
Hello,
I am
:-|
http://www.kamailio.org/w/sip-router-releases/
On Mon, Apr 9, 2012 at 11:06 PM, Daniel Gonzalez gonva...@gonvaled.comwrote:
Hello,
I am starting to deploy a SIP router, and after reading the
documentation in http://sip-router.org I am a bit confused. I am
planning to integrate the
Hello Ryan,
connect the kamailio to the asterisk using a private ip and the
kamailio is connected using a public ip for the clients .
I assume you've enabled mhomed =1 in your configuration script .
N.B : i have tried RTPproxy but it didnt work i think because their
is no proper NATING .
Hi,
Yes that is the behaviour when the media isn't flowing through a regulatory
tool (in-terms it sees the media and know call is actually going on
rtpproxy/media-proxy) but in the absence of any such tool SIP server is
not aware that the call-media is still in progress or is dead ! so it
always
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