Hello,
On 5/9/13 3:14 PM, Vitaliy Aleksandrov wrote:
Hello.
As I understand t_load_contacts() / t_next_contacts() can help only in
case when branches have different q values.
The only way I see is to get all contacts via reg_fetch_contacts(),
save the list to an AVP and then iterate through
Hello,
how many active calls do you expect to have?
At this moment, kamailio does not create local sockets dynamically, they
have to be specified in the configuration.
But if the number of active calls is not big, then you can create as
many sockets as expected calls and then use force send
Hello,
On 5/9/13 11:00 AM, Priyaranjan Nayak wrote:
Hi All,
After successful install of kamailio-3.1.5 I am trying to start
kamailio,but it throws following error
loading modules under /usr/local/lib/kamailio/
Hello,
On 5/8/13 8:24 PM, Bruno Bresciani wrote:
Hi all,
currently I have been used kamailio and rtpproxy to listen on two
different network (bridge mode), and the register of sip users are
saved on different locations (aliases and location) according the
interface where register is
Hello,
On 5/9/13 8:53 AM, andpe wrote:
Hi
I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones (trying
to run :-)).
The phones are seen as being registered in the PRESENCE, REGINFO work
unless properly.
But I have a question: in the table active_watchers watcher_domain
field is
Hello,
On 5/8/13 2:41 PM, ??? wrote:
hi,
i am from china.i have install a kamailio server with wss support
,and i set 4443 as the port(set 8080 as the ws port);the result of
netstat -tnl is:
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign
Hello,
presented a possible solution in another reply:
- http://lists.sip-router.org/pipermail/sr-users/2013-May/077872.html
Cheers,
Daniel
On 5/8/13 12:51 PM, Kamal Palei wrote:
Dear All
I have a Kamalio proxy, and number of phones are connected to it.
When all the phones make call,
Hello
On 5/8/13 11:20 AM, zhengyw wrote:
Hello:
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration
between Kamailio and Asterisk. I have no problem with registration but
when I try a call
This is related to:
- http://lists.sip-router.org/pipermail/sr-users/2013-May/077878.html
Cheers,
Daniel
On 5/8/13 9:28 AM, zhengyw wrote:
Hello:
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime
Hello,
any more comments on time storage for acc records?
Obviously, it has to be done via something that is configurable and
extensible.
If no more comments, I will go ahead and develop soon a configurable
framework for seconds and seconds, microseconds, then others can come
and add more.
Daniel wrote:
please do the patch to store the new value in _tr_buffer and attach it
to mailing list or bug tracker. I haven't looked at code yet, but sounds
like there is indeed an issue. I will review the patch and apply it.
What was the conclusion about this? Was the patch ever
I missed that $branch(..) variable is r/w. I like this way more than
playing with AVPs.
According to documentation $branch(..) variable gives access to
additional branches.
Is it possible to change Q value for the first branch ?
___
SIP Express
Hello, Daniel-Constantin!
I think yes, with documentation, it took some days.
ok, attach the patch as a file on the mailing list or bug tracker,
ready, attached
don't paste it inline because it is not easy to apply.
I think need documentation patch too.
what we get:
modparam(misc_radius,
10 maj 2013 kl. 13:56 skrev Kamal Palei palei.ka...@gmail.com:
Hi Daniel
Lets say I have more number of calls, say 50,000 or 100,000 calls at a time.
I need to have separate tcp connection for each call.
Just curious - why do you need this?
/O
Is there any possibility I can achieve this
Hi Olle
I am simulating a particular setup. A simulator behind Kamailio initiates
lots of call. The proxy ahead Kamailio needs message should come in
separate connection.
I should have choosen b2bua initially, but so far Kamailio helped a lot to
simulate the required environment. I really do not
I haven't tried this with Kamailio, so please read Daniel's comments.
I did not succeed in using Asterisk to set up multiple TCP connections, but
managed to get SIPP to set up one conneciton per dialog. This was during
testing the SNMPstats module in Kamailio.
/O
10 maj 2013 kl. 14:07 skrev
Dear list
I want to remove all raws from the presentity table after restart of
kamailio.
how can i achieve this ? (preferably without deleting the table manualy)
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
אורן אברהם writes:
I want to remove all raws from the presentity table after restart of
kamailio.
if it is ok after stop, but before start, you can run some script from
kamailio init script that does it.
-- juha
___
SIP Express Router (SER) and
Hi All,
I am new to SER and SEMS . Can we run RTP proxy with SER ?
If yes, which version of SER will compatible with RTP proxy ?
--
Thanks
Priyaranjan
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
Hello Daniel,
thank's for your help... I was thinking to use $fs (forced socket) variable
in branch_route to know what interface packet will be sending and set the
correct rtpproxy parameters (I don't know if is a better solution). I will
test your solution too.
Best Regards
2013/5/10
Hi All,
I am new to SER and SEMS . Can we run RTP proxy with SER ?
If yes, which version of SER will compatible with RTP proxy ?
--
Thanks
Priyaranjan
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
Am Freitag, 10. Mai 2013, 19:49:15 schrieb Priyaranjan Nayak:
I am new to SER and SEMS . Can we run RTP proxy with SER ?
If yes, which version of SER will compatible with RTP proxy ?
Hello,
you can use Kamailio with rtpproxy. All recent version of Kamailio are
compatible with
Hello,
I would like to send a SIP NOTIFY message to my SIP client(s), initiated by an
event on the server side.
//OFF Asterisk has a similar functionality, for instance (sip notify phone
vendor-reboot), to request a reboot of the phone device from the server
side. There is no need for an
Yes you're right. When we use the name instead of an IP address, the
entries in the table are correct.
Regards
Andy
W dniu 10.05.2013 09:32, Daniel-Constantin Mierla pisze:
Hello,
On 5/9/13 8:53 AM, andpe wrote:
Hi
I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones
(trying to
debian wheezy multiarch support allows installation of i386 packages to
amd64 host provided that there is i386 version of libraries used by the
package.
is there any advantage of amd64 kamailio versus i386 one? how does
kamailio benefit from 64 bit architecture or does it not?
if not, the on
I managed to find a solution in the meantime: using the xhttp module + uac
gives a perfect solution.
Regards,
Attila
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Attila Megyeri
Sent: Friday, May 10, 2013 4:57 PM
To:
Hello,
On 5/10/13 5:47 PM, Juha Heinanen wrote:
debian wheezy multiarch support allows installation of i386 packages to
amd64 host provided that there is i386 version of libraries used by the
package.
is there any advantage of amd64 kamailio versus i386 one? how does
kamailio benefit from 64
Yes, you can run.
http://rtpproxy.org
2013/5/10 Priyaranjan Nayak priyaranjan4...@gmail.com
Hi All,
I am new to SER and SEMS . Can we run RTP proxy with SER ?
If yes, which version of SER will compatible with RTP proxy ?
--
Thanks
Priyaranjan
Hi Daniel
Thanks for reply.
Just curious to know, how many number of calls we can achieve using the
solution you have recommended.
Also the solution, you have recommended, I am not very clear on that how to
do it. Kindly can you explain in detail.
Best Regards
Kamal
On Fri, May 10,
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