Hi,
I'm using kamailio from latest git-HEAD. The rtpproxy i'm using also
from latest git.
Our network topology is following :
sip-softphone->kamailio/rtpproxy>softswitch>gateway
Because of saving bandwidth we need to use the "re-packetization"
feature of rtpproxy.
When we d
On 07 May 2014, at 02:42, Teleport Engineer wrote:
> Hello,
>
> I'd setup a test environment where Kamailio accepts external invites and
> routes them to an internal Asterisk (actually working as an SBC).
> My dialplan works fine routing the INVITE to the internal Asterisk, but the
> SIP mess
I was able to resolve the issue, I had the wrong IP in the hosts file.
Regards,
Arun
On Tue, May 6, 2014 at 10:18 PM, 5060 wrote:
> hi:
> you can check by kamctl online, maybe some users are not there due to bind
> ..has expired.
>
>
> 2014-05-07 6:03 GMT+08:00 VOIP Tests :
>
>> Hello, I have
hi:
you can check by kamctl online, maybe some users are not there due to bind
..has expired.
2014-05-07 6:03 GMT+08:00 VOIP Tests :
> Hello, I have a set up with Asterisk-Kamailio as explained in
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
> .
>
> This set up ha
Hello,
I'd setup a test environment where Kamailio accepts external invites and
routes them to an internal Asterisk (actually working as an SBC).
My dialplan works fine routing the INVITE to the internal Asterisk, but the
SIP message goes into the defaul sip context (the one configured in sip
gene
Hello, I have a set up with Asterisk-Kamailio as explained in
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
This set up has been working well for us for sometime now. We are now
moving Asterisk to a stand alone server ( not installing it on the same
box as asterisk )
hi, all.
After study the source code of kamailio, I found that:
If a more higher openssl was adopted,
#if OPENSSL_VERSION_NUMBER >= 0x00908000L
then, within function init_tls_compression(), it (kamailio) will read conf
parameter :
cfg_get(tls, tls_cfg, disable_compression)
but, this met
The general recommendation from TLS security experts is to never ever use TLS
compression.
/O
On 06 May 2014, at 16:38, Daniel-Constantin Mierla wrote:
> Hello,
>
> what libssl version do you have? Some of them have bugs related to
> compression and the feature is disabled if such case is di
Hello,
what libssl version do you have? Some of them have bugs related to
compression and the feature is disabled if such case is discovered.
You can set debug=3 and send over the log messages from startup.
Cheers,
Daniel
On 06/05/14 11:36, 刘日新 wrote:
Hi, all.
I has configure kamailio.cf
Thank you Olle
I am looking into it now. I currently have presence presence_xml and
presence_dialoginfo enabled. I will look into pua today.
On Tue, May 6, 2014 at 3:12 AM, Olle E. Johansson wrote:
>
> On 05 May 2014, at 19:21, Joel White wrote:
>
> > I am working on enabling presence and b
Hi, all.
I has configure kamailio.cfg as below:
#!ifdef WITH_TLS
modparam("tls", "session_id", "vic22")
modparam("tls", "session_cache",1)
modparam("tls", "tls_disable_compression",0)
modparam("tls", "config", "/usr/cfg/tls.cfg")
#!endif
I was sure that this configure segment was i
Hi Volkan,
A possible solution is to have the FS responsible for one (or more) specific
domain(s) into a dedicated dispatcher set.
e.g.:
Dispatcher set 1 (generic):
- FS1
- FS2
Dispatcher set 2 (for domain1, domain2, etc):
- FS1
Dispatcher set 3 (for domain5, domain6, etc):
- FS2
Then inside y
Hello Daniel,
I´ve reproduced the scenario with Kamailio version 4.1.3. Behaviour is the same
and this is the output from kamailio-4.1.3 with debug=3:
May 6 09:08:54 sipsrvnode1 /usr/local/kamailio41/sbin/kamailio[20480]: INFO:
-<|XLOG|>-: handle_subscribe From: sip:117711@10.16.48.14 execute
On 05 May 2014, at 19:21, Joel White wrote:
> I am working on enabling presence and bla in a Kamailio + FreeSWITCH
> environment
>
> All handsets are Polycom with a handful of Grandstream ATA's
>
>
>
> I enabled presence and presence_xml
>
>
> I have not got it to work as I keep seeing th
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