Hello,
I experience the same issue in 3.1.5
best regards,
Antanas Masevicius
On 2012.03.01 08:07, Juha Heinanen wrote:
> Ben WIlliams writes:
>
>> here is the dump, prefix 8 should use gateway number 2. Its not a big
>> issue, I've managed to do it without LCR now.
>>
>> INSERT INTO `lcr_gw`
Hello. SIP over TCP would reduce server performance significantly when compared
with SIP Over UDP. Please read the following two papers. Combining RTP proxy
with SIP over TCP would degrade SIP server performance even worse.
---
Hi
I am wondering what are the major differences between kamailio 3.3 msrp
module and sylkserver (http://sylkserver.com/features.phtml ) ?
thanks.
min
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Hello
I am not able to dial to pstn with kamailio, the call is routed to my
pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is
hanged up.
My outbound route is:
route[PSTN] {
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled
I am sorry, that was not intentional.
Am 07.06.12 13:15, schrieb Daniel-Constantin Mierla:
Do not write private emails, keep the mailing list cc-ed for the topics started
there.
Cheers,
Daniel
On 6/7/12 12:22 PM, Dominik Mauritz wrote:
Daniel, Stoyna,
the solutions you described look very
Hi,
On 06/05/2012 08:21 PM, Saul Waizer wrote:
> I am running Kamailio and rtpprox on ec2
>
> I do have a DBUG log for it, one interesting thing to point out: On the
> above line notice how I have 127.0.0.1, I changed this from my external
> ip address because i was getting the following on the r
Anyone?
I've been also looking at http://www.voipmonitor.org/ which seems to be
recording something, at least the .pcap files are a lot larger in size but
no luck trying to extract sound with sox.
The end goal is to be able to record SIP calls (Audio and Video) for later
review. Has anyone done t
Hi,
Siremis v3.2.1 is out, the web management interface for Kamailio SIP
Server. It is the last release with new features planned to be
compatible with Kamailio v3.2.x. Next major releases for Siremis will
target upcoming Kamailio v3.3.x.
This time new features relate to web pages that allow
Hi again,
yes my scenario is quiet simple. I've lots of users and groups of those
users are defined by UIDs, one UID means 70 users of one client whereas
other UID could've 3 users of another client.
So what I am trying to implement here is that calls from one same UID are
always routed to exactl
Hello,
On 6/4/12 7:14 PM, Austin Einter wrote:
Hi All
Now I am using Kamailio 3.1.5 and RTP proxy 1.1.
Looks both are compatible and working fine.
The RTP Proxy basically sends/receives RTP packets over UDP.
Is there any RTP Proxy available that does send/receive of RTP packets
over TCP and als
Hi,
sure, I'll give that debug messages a try tonight, Also I was thinking of
alternative approach and like you said, you are using htable. If I've to
use that htable entry manually how do I make sure that any call data gets
only deleted/erased after the call hangsup !? i.e I don't want the htable
On Thu, Jun 7, 2012 at 5:24 PM, Daniel-Constantin Mierla
wrote:
> Hello,
>
>
> On 6/5/12 11:17 AM, Aft nix wrote:
>>
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Hi all,
>>
>> We are thinking about launching a different "usage policy". The usage
>> policy will be prepaid in some sense
Hello,
On 6/5/12 11:17 AM, Aft nix wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
We are thinking about launching a different "usage policy". The usage
policy will be prepaid in some sense.
but instead of limiting "number of minutes" we are thinking about
limiting "number of call
Hello,
that hash function is intended for hashing alpha-numeric usernames and
has a fair distribution for such cases.
I don't think is good for hashing numbers, at the end of a day, a number
is already like a hash code. You can use in the config file modulo
operation to select a particular d
Hello,
do you get any hints about what happens if you run with debug=3? When
dialog is deleted from database, is any other query sent to delete the
dialog vars? This features was not used by me so far, still relying on
htable for my needs of such cases.
You can send the logs here for trouble
Do not write private emails, keep the mailing list cc-ed for the topics
started there.
Cheers,
Daniel
On 6/7/12 12:22 PM, Dominik Mauritz wrote:
Daniel, Stoyna,
the solutions you described look very interesting to me. I will have a
look at both and pick the one that fits best.
Thanks guys.
Hello,
On 6/6/12 9:57 PM, Stoyan Mihaylov wrote:
We use Jitsi as SIP client, and openxcap along with camailio to handle
presence. Then jitsi know if account is online or offline.
Our Asterisk dont know nothing about accounts (it accepts all calls
from kamailio). There I run AGI scripts, which c
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