Hello,
On 5/6/12 9:40 PM, Fred Flintsone wrote:
I am attempting to route local registered users to local registered
users without going to the media server. I have a media server [*]
for PSTN. But it doesn't support all the CODECS i wan't to use.
Signalling seems fine, but i get NO AUDIO
Hello,
I see no reason why it crashes at line 104 when realm_prefix.len==0.
Might be a memory problem (hardware). Can you try with use_domain
parameter set to 0 for usrloc module?
Cheers,
Daniel
On 5/5/12 5:29 PM, Akan wrote:
I made the change and still have the problem. I have included the
hi Daniel,
your trick workednow I have mysql library files in the library folder...
Thanks.I have added `mysql` and `db_mysql` in include_module variable
in module.lst...
Regards,
Vineet Menon
On 8 May 2012 12:28, Daniel-Constantin Mierla mico...@gmail.com wrote:
Hello,
one
On 08.05.2012 12:19, Henning Westerholt wrote:
Am Dienstag, 8. Mai 2012, 12:06:46 schrieb Klaus Darilion:
I wonder what is the status of kamdbctl tool for 'migrate' command -
should it work? The description says:
kamdbctl migrateold_db new_db
.(migrates DB from
Hi,
How do we detect a message as retransmitted message, in a transaction
oriented proxy?
Regards,
Vineet Menon
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
t_check_trans() generally does the trick.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Vineet Menon mvineetme...@gmail.com wrote:
Hi,
How do we
See t_newtran() and t_check_trans() to create/check transaction state
for certain requests:
http://www.kamailio.org/docs/modules/3.2.x/modules/tm.html#t_newtran
http://www.kamailio.org/docs/modules/3.2.x/modules/tm.html#t_check_trans
There is no method to check if responses are retransmissions
Hello list,
I'm trying to have my Siremis interface send MI commands to multiple
kamailio servers i.e reload dispatcher of all the kamailio servers when I
reload from Siremis interface.
The issue Im facing is that the commands dont get executed on any other
server except localhost/
as
Hello,
We are running Kamailio 3.1.5 and using the acc module to generate CDRs and
provide reporting information. We have a requirement to report ring time and
we do this by accounting on early media for 183 provisional responses.
Unfortunately we have a couple of customer PBX's that cannot
Hello SR-users,
I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP
servers as users and let the Kamailio work as a Trunk/ Proxy connected to a
PSTN gateway. We have multiple domains and aliases per SIP servers.
When I register a SIP server to Kamailio everything
Hi,
i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up
on the same computer.
When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in
owner/creator session and connection information field with an error, i.e:
*Owner/Connection Information (o)*: doubango 1983
Hi,
On 05/09/2012 02:40 PM, Openser Kamailio wrote:
*Owner/Connection Information (o)*: doubango 1983 678901 IN IP4
*172.27.170.984* 172.27.170.98
*Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98
Could it be possible that you're calling rtpproxy_offer() twice?
Andreas
Hello,
I don't remember by hart all the parameter options for acc module, but
for accounting an event is always a backup route - use
acc_db_request(...) when that event happens -- in this case should be an
onreply_route with a condition on status code 180.
Cheers,
Daniel
On 5/9/12 1:32 PM,
Hello,
can you send here on the mailing lust the error log messages you
get? They should give some hints about what goes wrong.
Cheers,
Daniel
On 5/9/12 2:01 PM, Arjan Kuiken wrote:
Hello SR-users,
Hi Daniel,
Here are the error messages:
INVITE sip:0235630155@a1 SIP/2.0
Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK63637d58;rport
From: 0235630111 sip:0235630111@a1;tag=as0c12287c
To: sip:0235630155@a1
Contact: sip:0235630111@172.20.30.45
Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1
Greetings,
I'm having trouble getting parallel forking to work with aliasdb. I'm
running kamailio 3.2 with the standard kamailio.cfg script.
I have found that if an alias points to a set of addresses that all
reference local devices that are registered with the server, kamailio
sends an
Hi folks,
I have a strange problem when Kamailio ignores ACKs in a specific
scenario. The call flow is as follows:
A - INVITE - kamailio - INVITE - B
[omitting 100 and 180]
A - 200 OK - kamailio - 200 OK - B
A - ACK - kamailio
There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in
I call rtpproxy_offer() once, but i use also rtpproxy_manage().
When i disable rttproxy_mange(), it works well.
Thanks!
On Wed, May 9, 2012 at 2:57 PM, Andreas Granig agra...@sipwise.com wrote:
Hi,
On 05/09/2012 02:40 PM, Openser Kamailio wrote:
*Owner/Connection Information (o)*: doubango
I had the same problem when calling mediaproxy twice by mistake.
rtpproxy_manage( ) calls implicitely rtpproxy_offer( ). This is the problem.
Either you use only rtpproxy_manage once on the INVITE and let it start and
terminate the session, or you use rtpproxy_offer, rtpproxy_answer and
You can use something like wireshark on Kamailio server to see if ACK
packets go in right direction.
I had problem with ACK and BYE, and I saw that in some cases ACK and BYE
packets looped back in kamailio.
May be I used wrong client.
On Wed, May 9, 2012 at 5:15 PM, Efelin Novak
Seems like a loose routing issue. Are you loose routing in your config file?
On Wed, May 9, 2012 at 4:34 PM, Stoyan Mihaylov stoyan.v.mihay...@gmail.com
wrote:
You can use something like wireshark on Kamailio server to see if ACK
packets go in right direction.
I had problem with ACK and BYE,
Hi,
The Kamailio server you wish to control remotely should have the
mi_datagram module listening on the correct interface and not on the
loopback one (127.0.0.1).
loadmodule mi_datagram.so
*modparam(mi_datagram, socket_name, udp:192.168.2.156:8033)*
Reda
On Wed, May 9, 2012 at 1:02 PM,
Hi,
I would like (and a many people here I believe) to have a functional of
including a multiple config files like (foe example asterisk's
#include path/to/some/config.conf).
Is it possible to implement a such feature ?
Thanks!
___
SIP Express Router
Hello everyone,
I am trying to get a new SIP firmware version on our Cisco 7971G-GE. Now the
problem is that I don't really know which configuration files have to be
included to get everything working.
For now on I have the new SIP firmware and SEPmaccnf.xml and the
XMLDefault.cnf.xml files.
Greetings,
Here's another problem I'm having with kamailio 3.2 and the standard
kamailio.cfg script.
If the calling device is NATed, everything works fine if the original
call gets connected. That is, the INVITE sent to the called device has
the correct NAT fixups applied.
But if the called
Hello,
It is already there, see
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x
On 05/09/2012 06:04 PM, Konstantin M. wrote:
Hi,
I would like (and a many people here I believe) to have a functional of
including a multiple config files like (foe example asterisk's
#include
Hi Andrew,
I have missed that. Thank you very much!
2012/5/9 Andrew Pogrebennyk apogreben...@sipwise.com
Hello,
It is already there, see
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x
On 05/09/2012 06:04 PM, Konstantin M. wrote:
Hi,
I would like (and a many people here
After including a part of main config to included file -- I got a several
errors like:
0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command
is_method
0(1582) : core [cfg.y:3532]: parse error in config file
/opt/kamailio/etc/kamailio/debug.cfg, line 4, column 55: unknown
Kamilio 3.2.0
I'd like to use one gateway as primary gateway and the another gateway as
backup for failover.
I could not make it to work. Here is my table entries:
Lcr_gw:
Gary Chen writes:
When making call , it only uses the first gateway. If first gateway
failed, it could not find second gateway.
What is the correct table entry for this to work?
there is not necessarily anything wrong with your tables. put some xlog
statements to your script to find out
Konstantin,
You should put the include_file directive after loadmodule and modparam
directives. So it can be either before main route block or at the bottom
of your main kamailio.cfg.
On 05/09/2012 06:48 PM, Konstantin M. wrote:
After including a part of main config to included file -- I got a
Thank you, I found a logical error in order.
Also would be good if lex parser can understand a wildmasks, like:
include_file modules/*.cfg...
2012/5/9 Andrew Pogrebennyk apogreben...@sipwise.com
Konstantin,
You should put the include_file directive after loadmodule and modparam
directives.
I changed the use_domain parameter for usrloc to 0 and retried. Kamailio
did not crash and I was able to get registered. I have another server
with the same configurations as the one having the problem. I will apply
the patch there and see if the problem still occurs. Also, setting the
Thanks Juha.!
Regards,
-Mensaje original-
De: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Juha Heinanen
Enviado el: miércoles, 09 de mayo de 2012 0:39
Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing
Hello.
I’m using the dialog module to keep control of simultaneous calls. In some
cases, and I’m still trying to find why this happens, the dialog stays in
“STATE:: 1”, which according to the docs is a dialog which no provisional
response has been sent yet.
Is there a way to eliminate this kind
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