I didn't get the time for it yet, due to traveling and conducting a
kamailio advance training, the spare time was quite limited. Hoping to
be able to address it in the near future.
Cheers,
Daniel
On 28/11/2016 18:38, Pranathi Venkatayogi wrote:
>
> Any luck? Any help from other users who used -
Can you try with siremis 4.3.0 -- it was just recently released with
fixes for these kind of issues.
Cheers,
Daniel
On 28/11/2016 16:06, Diogenes Aviles wrote:
> I have Kamailio 4.4 installed in ubuntu 16.04.03, with Siremis 4.2.0,
> PHP7 and apache2.
>
> All is OK, but when I try to install the
Hello,
a new release of Siremis, the web management interface for Kamailio, has
been released recently. The main work was towards making it compatible
with PHP 7 and the new constraints from MySQL 5.7+. The announcement is
available at:
- http://siremis.asipto.com/2016/11/26/siremis-v4-3-0-rele
Laura writes:
> The LCR modules make the new INVITE using the original To field and
> don't rewrite it like on the first request..
> How is that possibile ?
LCR module never does anything to To header. Look at your config
elsewhere.
-- Juha
___
SIP
Ouch... It didn't work for me as expected. I forgot that I have configured
FreeSWITCH to work with TLS.
When I reverted sofia profile to work only over UDP originating call fails
again with message:
[ERR] sofia_glue.c:943 TLS not supported by profile
2016-11-29 0:21 GMT+02:00 Vladyslav Zakhozha
Daniel, Alex, thank you for your answers.
FreeSWITCH works with path as expected and it is my solution. add_path and
add_path_received works fine in kamailio's config.
2016-11-28 19:02 GMT+02:00 Vladyslav Zakhozhai :
> That is very interesting.
>
> I've added add_path_received in Kamailio conf
Hi Rafael,
There should not be a problem with the VMware mage.
Regarding your question: you have to check kamailio-pcscf.cfg to recognize
when and it how forward the SIP message towards the I-CSCF.
I guess it is the statement "t_relay();" in line 896.
You may provide a core tracefile (gen
I found my problem, because my custom "dispatcher" table is missing some
fields, the loading failed.
No issue :)
On Mon, Nov 28, 2016 at 7:56 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> I got
>
> Nov 28 13:38:25 testSVR kamailio[25455]: DEBUG: [db.c:403]:
> db_table_version():
I got
Nov 28 13:38:25 testSVR kamailio[25455]: DEBUG: [db.c:403]:
db_table_version(): no row for table alias found
Nov 28 13:38:25 testSVR kamailio[25455]: ERROR: dispatcher
[dispatch.c:839]: init_ds_db(): invalid table version (found 0 , required
1, 2, 3 or 4)#012(use kamdbctl reinit)
Nov 28 13:
David,
There should be some additional log messages and/or some more context around
this message, although not necessarily immediately around it (due to the
parallel nature of Kamailio child processes):
On Mon, Nov 28, 2016 at 07:33:47PM +0100, David Villasmil wrote:
> mod_init(): could not in
Hello all,
I have a databse where i created a "version" table only with a version, a
dispatcher records. I only want to use dispatcher on a db.
I don't know if kamailio is looking for the version tables somewhere else,
or it does find it in my database (not named kamailio) but the format is
wrong
Hello Linux,
Each xavp has a name and name values. So if you first define the name of
the variable then you assign the name value it will work for you:
$xavp(caller_peer=>accountname) = "Account Variable"; */*This is an
example*/*
$xavp(caller_peer[0]=>accountname) = "AAA";
xlog("L_WARN", "XAVP A
Any luck? Any help from other users who used - pv_www_authenticate?
From: Pranathi Venkatayogi
Sent: Friday, November 25, 2016 12:37 PM
To: 'mico...@gmail.com'; Kamailio (SER) - Users Mailing List
Subject: RE: [SR-Users] MSRP auth does not work with blink
Hi Daniel,
Please find attached Kamaili
That is very interesting.
I've added add_path_received in Kamailio config. And I can see that
FreeSWITCH received it and reflected in registration info.
With SIP/UDP there is no problems. FreeSWITCH gets path and respnses and
INVITEs goes through Kamailio.
But in case of TLS INVITES goes to Kama
Hi,
I am sure i am doing something very basic wrongly.
Here is my config file snippet
$xavp(caller_peer[0]=>accountname) = "AAA";
xlog("L_WARN", "XAVP ACCOUNT NAME $xavp(caller_peer[0]=>accountname) \n");
No matter what i set this XAVP at, it always prints null in log file as
below
WARNING:
Dear list,
i have a strange situation with module LCR on kamailio 4.3.6.
A customer send a call to my first kamailio and i forward it with some
tech prefix to my supplier.
Here the 2 correct invites.. customer is sending to me with prefix 9001
14:34:29.155719 CUSTOMER_IP.101:5060 -> EU_KMAILIO_
I have Kamailio 4.4 installed in ubuntu 16.04.03, with Siremis 4.2.0, PHP7
and apache2.
All is OK, but when I try to install the Siremis Wizard, the step 2 (Data
Base Configuration) doesn't work, and there is the message "Something Went
Wrong".
I tried all the option on Step 2, but the problem con
Request -> route[RELAY] -> branch_route[MANAGE_BRANCH] ->
route[RTPENGINE_DETECT] -> route[NATMANAGE]
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if(is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
On 28/11/2016 14:44, Anthony Messina wrote:
> I meant that before 4.4.4, I was building from the 4.4 branch will all
> commits in that branch up until 81df84b . git diff 81df84b 4.4 should
> show the difference, I just can't figure out what part isn't working.
>
>
> I am not using t_suspend()/t_co
I meant that before 4.4.4, I was building from the 4.4 branch will all
commits in that branch up until 81df84b . git diff 81df84b 4.4 should
show the difference, I just can't figure out what part isn't working.
I am not using t_suspend()/t_continue() or other async processing
functions and
On Mon, Nov 28, 2016 at 01:15:03PM +0100, Daniel Tryba wrote:
> > UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH
> >
> solution is to use Path on the frontend/loadbalancer.
According to this closed bug report it should work for
Kamailio/Freeswitch:
https://freeswitch.org/jira/si/jira.issue
Path is indeed the exact solution for this type of problem.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-ro
On Mon, Nov 28, 2016 at 01:00:37PM +0200, Vladyslav Zakhozhai wrote:
> UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH
>
> My main problem is in Contact header of SIP packet which passes through
> Kamailio SIP proxy and remains unmodified.
>
> For example, REGISTER request. There is FreeSWI
Hi,
I'm trying to understand the best (or reasonable) approach of offloading
SSL encryption from backend to Kamailio. Let me simplify a little bit:
UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH
My main problem is in Contact header of SIP packet which passes through
Kamailio SIP proxy and
On Fri, Nov 25, 2016 at 06:55:34PM +0200, Sergey Basov wrote:
> Is it safe to use remove_hf("User-Agent") without check if this header
> exist?
> or better use if(is_present_hf("User-Agent")) { remove_hf("User-Agent"); } ?
Just remove_hf is enough. is_present_hf/remove_hf might be more readable
th
That's done Daniel, thx for the elaboration !
2016-11-28 9:48 GMT+01:00 Daniel-Constantin Mierla :
> Hello,
>
> can you open a ticket on the bug tracker:
>
> - https://github.com/kamailio/kamailio/issues
>
> Besides the output of bt full, add also:
>
> info locals
> p dlginfo
> p *dlginfo
>
>
> C
The commit 81df84b is related to tls, so there should be no relation
with nat functions.
Are you using t_suspend()/t_continue() or other async processing functions?
Anyhow, whatever changes you want to be specific for each outgoing
request must be done in a branch_route. request_route changes are
Hello,
can you send the ngrep output for such case? Because there is no
response to ACK, so there should be no 404 for an ACK, maybe it is for
something else.
Cheers,
Daniel
On 26/11/2016 19:42, Rodrigo Moreira wrote:
> I need help.
>
> I'm building a load balancer using the dispatcher. It does
Hello,
can you open a ticket on the bug tracker:
- https://github.com/kamailio/kamailio/issues
Besides the output of bt full, add also:
info locals
p dlginfo
p *dlginfo
Cheers,
Daniel
On 27/11/2016 13:00, davy van de moere wrote:
> Gents,
>
> On a not really loaded kamailio (less than 50 us
Prior to the upgrade to Kamailio 4.4.4, I was using 4.4.3 with updates
through 81df84b from the 4.4 branch and the following fix_nated_sdp
call worked through the initial INVITE and when a voicemail branch was
appended from the failure_route.
route[NATMANAGE] {
...
# RTPEngine is not neede
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