On 2 May 2011 17:01, Roman Yeryomin wrote:
> On 2 May 2011 15:47, Daniel-Constantin Mierla wrote:
>>
>>
>> On 5/1/11 11:51 PM, Roman Yeryomin wrote:
>>>
>>> On 27 April 2011 20:19, Roman Yeryomin wrote:
On 27 April 2011 17:26, Daniel-Constantin Mierla
wrote:
>
> Can you t
On 09.05.2011 22:44, JR Richardson wrote:
Would there be any usage examples of dialog module? I'm not sure if I
really need profiles and [values] and when to set and unset. A push
in the right direction would be helpful.
Hi,
here is an example I've used for incoming and outgoing calls trackin
> > The problem I am having specifically is for outbound calls flowing
> > through a dispatcher, I want to separate calls into a profile for each
> > SIP domain that I send calls to. I'm not sure how do accomplish this.
> > I see in kamctl fifo dlg_list, there is 'caller_bind_addr::
> > udp:10.1
> On 09.05.2011 22:44, JR Richardson wrote:
> > Would there be any usage examples of dialog module? I'm not sure if I
> > really need profiles and [values] and when to set and unset. A push
> > in the right direction would be helpful.
>
> Hi,
> here is an example I've used for incoming and outgo
Hi,
We need a programmer who can modify SIP and RTP packets passing through
Kamailio and RTP proxy. Interested programmer can contact with me.
Thanks
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sr-users@lists.sip-router.
Hello,
On 5/6/11 11:01 AM, Torge Szczepanek wrote:
Hi!
In the Documentation of the TMX Module for the t_reply function there is given
a list of Parameters as can be seen here:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/tmx.html#id2822345
The third parameter is a trans_id. I cannot
Hello,
On 5/3/11 5:49 PM, Claudio Furrer wrote:
Hello,
Is this still applicable?
might not be if there is not there. Maybe one of the SER flavour
maintainers can say more.
For the Kamailio flavour I know that the 'update database structure' was
not really updated, instead, we keep track on
Be sure you send the mail on the rtpproxy list as well. Here are mainly
sip devs, so you may end up having to hire two, one for sip and one for rtp.
Also, if you need to do some heavy processing on rtp (i.e.,
transcoding), you should check also existing sip media serviers such as
SEMS (http://
Thanks so much Daniel.
If you have any known person (both kamailio and RTP Proxy or seperately)
please suggest.
On Wed, May 11, 2011 at 12:24 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:
> Be sure you send the mail on the rtpproxy list as well. Here are mainly
> sip devs, so you may
Hi Daniel!
> What exactly do you need to do, maybe there are some other options around?
I am currently trying to implement some kind of protocol translation in a
external daemon process.
The idea is:
A SIP Requests comes in and is processed in the following way in a REQUEST_ROUTE
if ((is_meth
Hi All,
Been playing around with the dialog module to help me keep track of
SIP trunk utilization, trying to identify how many calls are active on
each SIP trunk. My dialog count for inbound calls is 10 (correct) but
my outbound calls are always double that and I'm not sure why. Here
is a snip o
On Tue, May 10, 2011 at 3:54 PM, JR Richardson wrote:
> Hi All,
>
> Been playing around with the dialog module to help me keep track of
> SIP trunk utilization, trying to identify how many calls are active on
> each SIP trunk. My dialog count for inbound calls is 10 (correct) but
> my outbound ca
On 05/10/2011 05:11 PM, JR Richardson wrote:
I think I answered my own question, the double dialogs for the
outbound calls is a result of loose route response from the outbound
SIP Trunk carrier upstream using route[relay]. I guess I need to be
a bit more careful where I put my dialog flags for
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