Hello,
Can any one please provide the documentation for integrating PCSCF with
Kamailio.
The link for the documentation for ims_registrar_pcscf is not found. (
http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html)
Please tell me what are the modules are to be loaded before
Hello,
Can any one please provide the documentation for integrating PCSCF with
Kamailio.
The link for the documentation for ims_registrar_pcscf is not found. (
http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html)
Please tell me what are the modules are to be loaded bef
Hello, I have a question about the load balancer module of kamailio.
As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to
scale Asterisk.
Can I run a kamailio instance as the load balancer, and other several
instances as voice service replace of Asterisk?
If I can
Daniel,
Thank you for the examples.
Sorry, for the direct email, was not my intention.
Tried, and I get my results.
Thanks again.
Rgds,
Gertjan
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: vrijdag 23 augustus 2013 17:31
To: Gertjan Wolzak; 'Kamailio (
Hi,
I wrote a small Kamailio-IMS Installation howto in the Kamailio-Wiki:
=> http://www.kamailio.org/wiki/tutorials/ims/installation-howto
Maybe that will help you.
Kind regards,
Carsten
2013/8/27 Sai Krishna Kota :
> Hello,
>
> Can any one please provide the documentation for integrating P
Hello,
You can't replace Asterisk with Kamailio, well. depends on what you do with
Asterisk.
But remember Kamailio is a sip proxy server, Asterisk is a B2BUA.
As soon as Asterisk does your voicemail, transcoding, etc. Kamailio is not
an option.
Just google Kamailio vs Asterisk, you
Thanks Olle. I know it's like asking MySQL to implement a picture filter ..
:-) but I am still wondering in what scenario do we use the following two
functions from the rtpproxy module:
rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stop_stream2uac()
I am using the latest kamailio 4.0.3 and r
Hi,
a typical usecase for
rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stop_stream2uac()
is "MusicOnHold". In this case, Kamailio does not have to create a new
SDP, but it would only insert the RTPProxy to play the prompt.
Kind regards,
Carsten
2013/8/28 Adnan <112linuxstockh...@gmail.com
Hello,
if anyone here is considering to attend Astricon, the discount code
AC13DIGI gives 20% off.
- http://www.astricon.net
The offer won't last too long and the seats are filling up quickly, so
better hurry up. It's the biggest chance so far to learn about
Kamailio+Asterisk together as
I think I found my missing ACKs! Can anyone tell me why they work be
being sent to the loopback interface? The destination address is
still the external (eth0) IP.
--
Marc Soda, Sr. Systems Engineer
*CoreDial, LLC* | www.coredial.com
1787 Sentry Parkway West, Building 16, Suite 100, Blue Be
If it is eth0 of the same server, then is the kernel sending via
loopback interface at it detects the destination is itself.
You should paste here ngrep with the sip traffic from invite to bye in
order to give more hints about what is going wrong there.
Cheers,
Daniel
On 8/28/13 3:09 PM, Mar
Vitaliy Aleksandrov writes:
> If anybody else except me need this It would be great to fix known
> problems and add it to kamailio.
i don't know if this come already up, but why not use this in branch
failure route:
unregister("location", "", "$T_reply_rid");
-- juha
_
Thanks, I appreciate it.
In this setup the there are 2 endpoints (700 and 701) peered up to an
Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20). 700
(172.16.60.28) is calling 701 (172.16.3.65). When 701 answers the OK is
sent to the proxy and then to Asterisk. Asterisk is then
I need help to fixed my NAT configuration I have nat module load but nat
configuration is not working. ANY HELP PLease
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
loadmodule "nat_traversal.so"
#!endif
#!ifdef WITH_NAT
# - rtpproxy params -
modparam("rtpproxy", "rt
On 08/28/2013 06:45 PM, Juha Heinanen wrote:
Vitaliy Aleksandrov writes:
If anybody else except me need this It would be great to fix known
problems and add it to kamailio.
i don't know if this come already up, but why not use this in branch
failure route:
unregister("location", "", "$T_reply
Vitaliy Aleksandrov writes:
> Didn't know about $T_reply_rid variable and that unregister can remove
> only single contact.
> In my case the problem with unregister is that stale contact will be
> removed only if somebody tries to call to a disconnected phone.
so you get one call to unregistere
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