Hi there,
Awesome Klaus! A lot of thanks!
You're absolutely right, I've made a mistake calculating the response
because of I've been using the wrong URI until now.
On the other hand, during a REGISTER dialog I've been using
sip:10.1.20.40 as URI parameter to calculate 401 response and it's
still
Hello,
I am working on Kamailio 3.2.2. There is no traffic, only functional test done.
Based on the SIP Client used until now (Xlite), the INVITE is systematically
authenticated by 407 as there is no Proxy-Authorization in the initial INVITE.
Expected/normal behavior.
But when using an interna
Dear List
I am trying to run the below command from a REMOTE server where i use Cacti
to graph all my data. I am trying to graph on a per trunk basis.
kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
This command yields an output on the local server.
Could someone perhaps point me i
daniel,
you wrote:
This feature should be useful in mobile networks, to avoid stacking lot
of invalid contacts which could result in lot of branches causing many
retransmissions.
why would a sip ua that runs in a mobile network device be configured to
use udp instead of tcp?
poor batter
On 22.08.2012 09:57, David Patiño wrote:
Hi there,
Awesome Klaus! A lot of thanks!
You're absolutely right, I've made a mistake calculating the response
because of I've been using the wrong URI until now.
On the other hand, during a REGISTER dialog I've been using
sip:10.1.20.40 as URI parame
Hello,
On 8/22/12 10:51 AM, patrice.bode...@orange.com wrote:
Hello,
I am working on Kamailio 3.2.2. There is no traffic, only functional
test done.
Based on the SIP Client used until now (Xlite), the INVITE is
systematically authenticated by 407 as there is no Proxy-Authorization
in the
Hello,
On 8/22/12 12:18 PM, Juha Heinanen wrote:
daniel,
you wrote:
This feature should be useful in mobile networks, to avoid stacking lot
of invalid contacts which could result in lot of branches causing many
retransmissions.
why would a sip ua that runs in a mobile network device
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server. All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-
Hi,
I upgraded our system to Kamailio 3.3 to take advantage of the new
db_cluster module. Currently we have two registration/location servers
each with their own sql-server. Authenticated REGISTER messages are
forwarded to the other server, which uses the 0x02 flag to save() in
order to store it w
On 2012-08-22 at 15:32, Øyvind Kolbu wrote:
[...]
Some more details:
# kamailio -V
version: kamailio 3.3.1 (i386/linux) aae4e4
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC,
USE_FUTEX, FAST
Hello,
can you send the output of 'bt full' from gdb?
Simple bt shows only the line where a define is executed, hopefully bt
full will get the values from the inside defines to spot the problem.
Also, can you give it a quick try with 9p instead of 7p for both
connections?
Cheers,
Daniel
Hi,
Kamailio crash using Cassandra 1.1.2 during SIP unregister (registering works
fine)
The version of Kamailio is the latest GIT version:
version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_
I'm just starting to evaluate Kamailio to be used as a front end to a couple
of Asterisk Servers.
Currently I have an Adobe Flex application that interfaces with Asterisk via
AMI using a socket connection. I searched on how to interface with Kamailio
but wasn't able to find anything that helped
22 aug 2012 kl. 18:01 skrev Carlos Cruz:
> I'm just starting to evaluate Kamailio to be used as a front end to a couple
> of Asterisk Servers.
>
> Currently I have an Adobe Flex application that interfaces with Asterisk via
> AMI using a socket connection. I searched on how to interface with
Hello,
I would like to use the module and create our own version of the Application
Server using the SEAS protocol but there are some pieces of information
missing from the web site (links to images that contains information about
headers etc ). The basic information on how it works is ther
On 22.08.2012 14:26, Vijay Thakur wrote:
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server.
Kamailio is a SIP proxy, not a media server. Maybe you mean that you are
using Kamailio with rtpproxy as media relay.
All things
are working fine. Now i want to
The nonce is globally valid until it expires (for all kind of requests,
IIRC you could also change the user).
Maybe your "internal" SIP clients calculates the response wrong. You
could test the response calculation with this website (do not use 'real'
passwords):
http://pernau.at/kd/sipdigest
Thanks for clearing the doubts. You are very right, i am using kamailio
as Media Relay.
Can you send me some specific document URL, from where i can configure
Asterisk as PSTN Gateway.
Can we set Kamailio and Asterisk in one server.
Thanks in advance.
Vijay
Thursday 23 August 2012 11:24 AM,
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