devices my kamailio systems can stay up for weeks after
power loss. With the SIP devices and routers I'm limited to hours of uptime.
Henning Westerholt wrote:
> On Wednesday 25 May 2011, Jeremya wrote:
>
>> These figures pale into insignificance compared to the power required
&g
embedded servers at very low power. My
constraint is the SIP devices and communications devices that suck up an
order of magnitude more power at each small site - of which I have many.
Jeremya wrote:
> These figures pale into insignificance compared to the power required
> for standard SIP d
These figures pale into insignificance compared to the power required
for standard SIP devices - typically 5-8 watts per device multiplied by
the number of devices.
When you factor in Gigabit Ethernet the power ups significantly.
Optimisation at the server level is not significant on any scale.
O
Hi,
I have a kamailio 3.1.0 system.
I am using mostly the kamailio.cfg script that came with that release
but I've made a couple of changes
- The pstn.gw_ip bit doesn't work - syntax error - so I've substituted
fixed text for now
- I've expanded the PSTN section to IN_PSTN and OUT_PSTN to handle
basically all you need is to re-write some media attributes in the
> sdp). The rtpproxy daemon will also be needed.
>
> Cheers,
>
> Marius
>> 2011/1/26 Jeremya:
>>
>>> Whoops! some SIP traffic IS peer-to-peer.
>>>
>>> Jeremya wrote:
>>
accounting for INVITE and BYE and a few others.
Danny Dias wrote:
> Media NEVER goes through a Proxy core...the question is, how should i
> record conversations when the calls are all passing through a sip
> proxy? some lights will be enough for me :)
>
> 2011/1/26 Jeremya :
>
Whoops! some SIP traffic IS peer-to-peer.
Jeremya wrote:
> Danny Dias wrote:
>
>> Hello my friends,
>>
>> I have a requeriment, which indicates that i have to record every SIP
>> conversation between peers (also for callings to the PSTN); the
>> recording
Danny Dias wrote:
> Hello my friends,
>
> I have a requeriment, which indicates that i have to record every SIP
> conversation between peers (also for callings to the PSTN); the
> recording server will be built for our company following this
> requeriments (also requested for the client):
>
> My do
A large bunch of recent email on this list got consigned to the spam bin
by my Thunderbird.
This hasn't happened before.
Is there some list change that triggered this?
Regards
Jeremy A
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users m
Klaus Darilion wrote:
> Jeremya wrote:
>> Hi,
>>
>> My situation is that I have a number of SIP end points (server) that
>> support SUBSCRIBE method on various resources they hold. Primarily
>> these are dialog resources, but include arbitrary other resources
&
Hi,
My situation is that I have a number of SIP end points (server) that
support SUBSCRIBE method on various resources they hold. Primarily these
are dialog resources, but include arbitrary other resources with custom
MIME types.
I also have a large number of SIP end-points (clients) that use
SUB
Hi
I have a problem which pops up occasionally.
My scenario is a kamailio 3.0.x server coupled with SEMS 1.2.1 Calls go
to SEMS and occasionally it will issue a refer to the caller (an SPA962
linksys)
The problem is that the refer seems to get screwed up somewhere and the
referred call changes t
Depending on what you downloaded and configured it may be called kamctl
The build system seems a bit dodgy to me (and I don't use Ubuntu). There
is the special script to do selective compiles/installs and there is
also simple 'make' and 'make install' available - perhaps after some
config settings
Jeremya wrote:
> Hi,
>
> I'm having some issues with recording in rtpproxy when I parallel fork
> an incoming call.
>
> The scenario is that I want to record incoming calls to a small number
> of operators, and the call is converted to an 'all-call' by the
Is the flag "f" of any actual use in force_rtp_proxy, rtp_proxy_offer
and rtpproxy_answer?
With an upstream proxy the lack of "f" causes problems. Does some
problem occur when it is used without an upstream proxy?
Thanks
Jeremy
___
SIP Express Router
Hi,
I'm having some issues with recording in rtpproxy when I parallel fork
an incoming call.
The scenario is that I want to record incoming calls to a small number
of operators, and the call is converted to an 'all-call' by the local
kamailio instance - parallel forking.
It appears - though I co
I had this problem on one (dual xeon) system. The cause was a very slow
disk system - it was SAS but the OS decided to use it as IDE - and the
killer was bandwidth logging by iptables.
Checking with wireshark it could sometimes take over a second for a
packet to get out of an interface.
When I ch
Alex Balashov wrote:
> They should exist. Are you sure you aren't using the git HEAD, where
> they have been moved into the "rtpproxy" module?
>
> On Nov 30, 2010, at 3:14 AM, Jeremya wrote:
>
>> http://kamailio.org/docs/modules/3.0.x/modules_k/nath
http://kamailio.org/docs/modules/3.0.x/modules_k/nathelper.html
refers to functions rtpproxy_offer() and rtpproxy_answer() But I get an
error when using them in my script.
Are these not available in 3.0.x?
Regards
Jeremy
___
SIP Express Router (SER
Hi,
I've been following some examples using has_sdp() function but it is not
supplied by 3.0.x TEXTOPS
I've seen some discussion on mailing lists about has_sdp() being
included since 1.5. But checking the online documentation is doesn't
exist in any version.
I use has_body("application/sdp") as
Daniel-Constantin Mierla wrote:
>
>
> On 11/19/10 8:27 PM, Iñaki Baz Castillo wrote:
>> 2010/11/17 Daniel-Constantin Mierla:
>>> I made an easy-to-do tutorial where all the SIP routing logic is
>>> implemented
>>> in a Lua script (including authentication, accounting, registrar, user
>>> location).
These text lines parse
if ($sht(ipban=>$si)!=$null)
$sht(ipban=>$si)=1;
These don't parse
if($sht (ipban=>$si)!=$null)
$sht (ipban=>$si)=1;
In both cases the gap between $sht and the opening '(' is the problem.
Is this expected behaviour?
Jeremy
___
Elena-Ramona Modroiu wrote:
> On 11/02/2010 12:22 PM, Jeremya wrote:
>> Elena-Ramona Modroiu wrote:
>>
>>> On 11/02/2010 07:57 AM, Jeremya wrote:
>>>
>>>> Elena-Ramona Modroiu wrote:
>>>>
>>>>
>>>>&g
Elena-Ramona Modroiu wrote:
> On 11/02/2010 07:57 AM, Jeremya wrote:
>> Elena-Ramona Modroiu wrote:
>>
>>> Hi,
>>>
>>> Siremis v2.0.0 is out – the web management interface for Kamailio
>>> (Openser) v3.1.0 and SIP Express Router (SER).
>&g
Elena-Ramona Modroiu wrote:
> Hi,
>
> Siremis v2.0.0 is out – the web management interface for Kamailio
> (Openser) v3.1.0 and SIP Express Router (SER).
I downloaded and installed the latest source version of kamailio 3.1.0
(not git) and compiled an installed it. This works as expected.
I then do
While you are on the topic of documentation errors:
The module documentation for DB_TEXT is still wrong - after a long while.
The URL should be text:// rather that dbtext://
Sergey Okhapkin wrote:
> Documentation of auth_db module is wrong too, error codes returned by
> www_authorize and proxy_
Hi,
I have an application that uses dbtext. Starting with an empty location
table I have a client register with kamailio.
When kamailio is shut down the location records are written.
When kamailio starts again it fails when reading the location value.
The client does not provide an Allow: header.
I've discovered - after quite a while - that #!endif doesn't work if
there is extra text on the line
#!endif
works
#!endif # someblock
doesn't work.
Could this be addressed in code or at least noted in documentation?
I also have the case of SUBSCRIBE where it is a later (re)SUBSCRIBE
after
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