We use dialog module in our configuration.
uac_replace_from/to is called after dlg_manage().
>From UAC documentation: "If you create a dialog ( with dlg_manage() )
before calling uac_replace_from(), this avp will not be needed. The values
of the uris will be stored as dialog variables."
And we h
Hello,
I see it crashed in dialog, it was a fix for it after 4.1.6 for the
cleanup timer procedure, affecting some particular cases with dialog
states (e.g., dialogs not answered even after many minutes since
creation). It was there before 4.1.6 (most probably from 4.1.0), not
introduced by t
On 01/10/14 15:23, Julia Boudniatsky wrote:
I understand my mistake.
It's working when message is broken.
I wanted to use it for bad SDP.
We have a problem with SDP from some users.
In error of parsing SDP on log printed only wrong part of SDP
and correlation to call is very difficult.
I see
Storing the values in dialog module can be useful, you need to enable it:
- http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_dlg
But I am not sure if that is making dialog to use them automatically for
BYE requests.
What you can try is to enable event_route[tm:local-requ
Hi,
Is it possible to use kamailio to manage a very simple queue? All calls inbound
go directly in queue and then the system to to extract calls and route to
agent. A very simple call center.
I've seen that there's a mohqueue module...is there any conf example?
Many thanks
Кamailio is a proxy server. Call is simple sip seesion for it. It does not
have briges or thomething else. Use asteirsk or freeswitch as backend
application server.
2014-10-02 11:23 GMT+04:00 Marino Mileti :
> Hi,
>
> Is it possible to use kamailio to manage a very simple queue? All calls
> inbou
Thats true if you need to answer and do some IVR, otherwise you can hold
the INVITE or other request (until it is cancelled or times out).
Specific on a call queue, may worth checking the module:
- http://kamailio.org/docs/modules/stable/modules/mohqueue.html
Holding the invite in config file i
Hello,
apart of shared memory size (still to be considered), I made the other
changes.
Based on last days of testing at SIPit31 and reviewing my configs, few
other values look a bit small considering what we can get on the wire:
- tcp_rd_buf_size - it is 4096b, but with audio/video and ice,
I don't need any IVR so I would like to try this module and following doc i've
setup dbtable name parameters:
modparam ("mohqueue", "db_qtable", "mohqueues")
modparam ("mohqueue", "db_ctable", "mohcalls")
but when start Kamailio says:
ERROR: [modparam.c:156]: set_mod_param_regex(): set_mod_par
Hello Daniel,
Have you had the opportunity to look at this?
Thank you for your feedback.
Regards,
Igor.
De : Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Envoyé : mercredi 24 septembre 2014 09:00
À : Igor Potjevlesch
Cc : 'Kamailio \(SER\) - Users Mailing List'
Objet : Re: [SR
There was a typo in the parameter name. If you use default values, just
remove the modparam.
Otherwise, I pushed the patches to master and 4.1 branches. So you would
need to upgrade to latest git version.
Cheers,
Daniel
On 02/10/14 10:15, Marino Mileti wrote:
I don't need any IVR so I would
Daniel-Constantin Mierla writes:
> - tcp_rd_buf_size - it is 4096b, but with audio/video and ice, the size
> is to small. I propose either 8k or even 16k
i have used 16k in my setups.
> - pv_buffer_size - it is 1024b, given the above and that this is used
> for variables, many don't fit. The s
One more came in my mind:
- failure_reply_mode in tm set to 3 (now is 0)
Daniel
On 02/10/14 10:30, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
- tcp_rd_buf_size - it is 4096b, but with audio/video and ice, the size
is to small. I propose either 8k or even 16k
i have used 16k in my
Daniel-Constantin Mierla writes:
> One more came in my mind:
>
> - failure_reply_mode in tm set to 3 (now is 0)
i have been using 3 there.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.o
Storing the values in dialog module doesn't help.
I updated headers in event_route[tm:local-request] by subst_hf with saved
dialog variables (uac_replace_to/from has been used in branch route).
Now it's working.
Great! Thank you,
Julia
On Thu, Oct 2, 2014 at 9:11 AM, Daniel-Constantin Mierla
On 02.10.2014 10:48, Juha Heinanen wrote:
> Daniel-Constantin Mierla writes:
>
>> One more came in my mind:
>>
>> - failure_reply_mode in tm set to 3 (now is 0)
>
> i have been using 3 there.
IMO this makes sense
___
SIP Express Router (SER) and Kam
I'm using Kamailio 4.15 with Siremis 4.1.0
both are great BTW!
I'd like to add management of the Kamailio Dynamic Routing Module to
Siremis web manager
Any pointers on how to do this; I'm not familiar with OpenBiz
Thx in advance
___
SIP Express Router
Sorry. Now fixed.
I was missing the obvious.
route section works as expected.
Apologies for the noise.
Regards
Phil
On 1 Oct 2014, at 08:22, Daniel-Constantin Mierla wrote:
> Hello,
>
> can you check if there are log messages in syslog file? have you checked that
> kamailio is running?
>
>
Hello!
I just got some kamailio servers working with websockets using jssip in my
clients and now I want to get the SIP Messages to better understand how it
flows between my servers.
I have enabled the "siptrace" module, and I can now save information on my
database, but the problem is that I can
Hi Paul
Thank you very much for the quick response and the probably spot on
information. I really bet that is my issue.
I will be testing this out tonight and I will post back with what I find.
Thanks again and all the best
Will Ferrer
> On Oct 1, 2014, at 11:53 PM, Paul Smith wrote:
>
> H
Hi Paul
Just wanted to give you an update.
This looks like it has worked. Now I am dealing with my own natting issues
on my home network to get the call but the invites are being sent right now.
Thanks again for the assistance.
All the best.
Will Ferrer
On Wed, Oct 1, 2014 at 11:53 PM, Paul S
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