I've been playing with OpenSIPS and am willing to change over if it means
I'll get a more informed answer from this community..
I need to set up a proxy forwarding server that can listen on a port other
than 5060 I.e. 8080 and forward the registration requests to another Voip
platform we have
Express Router (SER) - Users
Mailing List sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Proxy Registrations offset port
Hi Robert,
I had the same feeling about O***sips community.
I have a similar setup. A proxy that listens on port other than 5060,
forwarding to other servers on 5060
by looking at a wireshark/tcpdump dump on your PortaOne server
to know where the rtp stream is going, and then troubleshoot accordingly..
On your PortaOne server, what is the location of registered users? Is it
their IP@, or that of OpenSips server?
Reda
On 9 mars 2012, at 16:28, Robert rob
. NAT handling is delicate woth proxies.
Start by looking at some wireshark dump to see what's wrong.
Reda
On 9 mars 2012, at 16:57, Robert rob...@inteli-core.com wrote:
Two different tests I completed:
Test #1 it showed the opensips server
Test #2 it showed the client's public ip
:23:47 +0500
To: mico...@gmail.com, SIP Router - Kamailio (OpenSER) and SIP Express
Router (SER) - Users Mailing List sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Proxy Registrations offset port
Hi Robert,
There can't be any simpler start then using the configurations(kamailio.cfg
file
Hi,
I'm using outbound with gruu enabled and with edge proxy, registrar and auth
proxy all running as separate entities. My problem is once a dialog is
formed between two UAs, if one side re-registers and gets assigned a new
flow token, all subsequent in-dialog messages cannot be routed because
d socket connection works. Any help will be
appreciated!
Thanks,
Robert
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/send-all-blocks-indefinitely-tp156948.html
Sent from the Users mailing list archive at Nabble.com.
___
Hi,
I am trying to run multiple different instances of openser on the same
server using different ports.
Regardless of receiving INVITE on any of the ports (5061, 5062, ...) I want
openser always to proxy the message to port 5060.
invite:5061 --- openser -- invite:5060
invite:5062 --- openser
/kamailio.log
Don't forget to restart syslog
Regards,
Robert Verspuy
--
*Exa-Omicron*
Patroonsweg 10
3892 DB Zeewolde
Tel.: 088-OMICRON (66 427 66)
http://www.exa-omicron.nl
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sr-users
Hello,
I'm interested in using the htable and/or mtree modules to keep my
aliases and did lookups in memory to minimize the amount of SQL lookups
that are being performed. It's easy enough to use htable to cache a
single value after it is retrieved but what I'm not clear on is how I
can do it
Op 04-03-11 15:43, Henning Westerholt schreef:
On Friday 04 March 2011, Henning Westerholt wrote:
By searching for Robert you should be able to find the initial discussion
of this problem, i guess.
[sr-dev] invalid type (3) or nul (0) version columns for trusted (kamailio 3.1
latest from git
Op 04-03-11 16:05, Klaus Darilion schreef:
Hi Robert!
See the thread here:
http://lists.sip-router.org/pipermail/sr-dev/2010-October/009545.html
So, when it is fixed, why printing a WARNING? Is it something I have to
be aware of? I use mysql with 3 modules doing DB lookups. I get
== 1.2.3.1) {
force_send_socket(1.2.3.1:5060);
} else if ($Ri == 1.2.3.2) {
force_send_socket(1.2.3.2:5060);
}
With kind regards,
Robert Verspuy
--
*Exa-Omicron*
Patroonsweg 10
3892 DB Zeewolde
Tel.: 088-OMICRON (66 427 66)
http://www.exa
that mentioned this.
Unfortunately, Kamailio does not do this. So we could not even use this as a
discriminator.
Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.
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send an empty NOTiFY? If so, what would trigger it?
Robert Jongbloed
Vox Lucida Pty. Ltd.
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Yes, correct. I don't think this is anything to do with Kamailio, I will detach
you from the thread,
Thanks for all your assistance!
Robert Jongbloed
Vox Lucida Pty. Ltd.
On 12/06/2011, at 8:11 PM, Iñaki Baz Castillo wrote:
The id of both tuples are different so the server must
Hi,
How can I set a filter for receiving 5xx messages, i.e. how can I parse 503
messages received by the proxy?
I have tried the following and none works:
t_check_status(503)
is_method(503)
Thanks,
R
___
SIP Express Router (SER) and Kamailio
Thank a lot. $T_reply_code works.
I tried all variables in pv doc ($rc, $err.rcode, $rs ... ) and none of
them works,
On Thu, Dec 29, 2011 at 9:27 AM, Andrew Pogrebennyk
apogreben...@sipwise.com wrote:
On 12/29/2011 03:21 PM, Robert R wrote:
What is the Pseudo-variables for Status-Line
Hi,
What is the Pseudo-variables for Status-Line filed of SIP response
messages (2xx, 3xx,4xx,5xx,6xx)? i.e., is there a Pseudo-variables to
display the SIP response code?
Thanks,
R
___
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Hi,
The openser service died on my servers after memory allocation failure.
This issue has been occurred a couple of times on two different servers
within the last two months.
Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:do_action: memory
allocation failure
Feb 29 08:31:59
Hi,
When I am trying to modify FROM field in the failure_route block I get an
error message:
failure_route[1]
{
...
uac_replace_from($avp(i:204));
append_branch();
t_relay();
}
Not starting openser: invalid configuration file!
Mar 22 10:51:39 [6457] CRITICAL:core:yyerror: parse
Hi,
When I am trying to modify FROM field in the failure_route block I get an
error message:
failure_route[1]
{
...
uac_replace_from($avp(i:204));
append_branch();
t_relay();
}
Not starting openser: invalid configuration file!
Mar 22 10:51:39 [6457] CRITICAL:core:yyerror: parse
/03/2012 10:06, Robert R escreveu:
Hi,
Even though when I try to use uac_replace_from( ) in branch_route
block I get the same error message:
branch_route[2]
{
if ( $avp(i:204) )
{
uac_replace_from(hello);
}
}
error message:
Not starting openser
Hi,
*
*
*
*
We are facing a problem in our opener proxy receiving: *Message too big
error.*
Looking at the INVITE message sent by opener I notice the following in the
record-route header:
Record-Route:
sip:10.10.10.10;lr=on;ftag=5114A38-F80;vsf=AAA-
Is
settings.
With kind regards,
Robert Verspuy
--
*Exa-Omicron*
Eenspan 8-K
3897 AL Zeewolde
http://exa.nl
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settings.
With kind regards,
Robert Verspuy
--
*Exa-Omicron*
Eenspan 8-K
3897 AL Zeewolde
http://exa.nl
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Hi,
I am trying to add a log when receiving 1xx or 2xx reply message:
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg(incoming reply\n);
if(status=~[12][0-9][0-9])
{
xlog(L_INFO,reply message: $T_reply_code ci:$ci);
route(NATMANAGE);
}
}
Adding the above
AM, Robert R rob1...@gmail.com wrote:
Hi,
I am trying to add a log when receiving 1xx or 2xx reply message:
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg(incoming reply\n);
if(status=~[12][0-9][0-9])
{
xlog(L_INFO,reply message: $T_reply_code ci:$ci
Hi,
There is a big delay in our Kamailio server when repling 100 Trying after
incoming INVITE messages (more than 500 ms). It seems my perl script (and
database query) is executed before 100 Trying being send out. I was
wondering if sending 100 Trying is configurable so I can send it right
:
Hello,
you can use sl module for sending a reply as soon as you get a request:
sl_send_reply(100, Trying);
Cheers,
Daniel
On 5/15/13 9:06 PM, Robert R wrote:
Hi,
There is a big delay in our Kamailio server when repling 100 Trying
after incoming INVITE messages (more than 500 ms
Hi,
I was wondering how to force Kamailio to add port field in the via
header.
Thanks,
Rob,
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SCA Developers,
I ran into a problem using the SCA module and was looking for some
assistance for the following situation. Can you help? Here is the scenario:
1. When a conversation is occurring on a shared line with Polycom phones;
2. Another caller on the same shared line picks up the
Mark,
That part depends on you. There are so many ways to implement that
function I decided not to make it a part of mohqueue. mohq_retrieve ()
provides a way to move calls out, mohq_count () allows you to check for
queued calls within Kamailio scripts, and the database tables allow you to
Jorj,
I work on the same team with J. Arruda and we would like to get this bug
fixed. When amortensen was at Penn he was responsive to bug requests but
now he seems to have other responsibilities. It looks like you are
offering to step in for him. Can you officially inquire about picking up
Only with rtpproxy.
On Fri, Oct 3, 2014 at 5:25 AM, Marino Mileti marino.mil...@alice.it
wrote:
mohqueue module is also compatible with rtpproxy-ng or only with rtppoxy?
Because during startup i've noticed this error:
mohqueue : Unable to load rtpproxy_stream2uac
There was a typo in the
Ansuman,
The mohqueue module was designed to redirect waiting callers into a
Music-On-Hold (MOH) queue and when a line became available to pull them out
of queue and redirect them to an available line. I'm not sure if this is
what you have in mind. If so, I'm sorry but I don't understand your
...@lists.sip-router.org] *On
Behalf Of *Robert Boisvert
*Sent:* 04 January 2015 02:24
*To:* Kamailio (SER) - Users Mailing List
*Subject:* Re: [SR-Users] Queue implementation in kamailio
Ansuman,
The mohqueue module was designed to redirect waiting callers into a
Music-On-Hold (MOH
Hello all,
I would like to setup a DMQ bus where servers can come on and offline and
have DMQ continue working with the online servers. I'm having trouble
understanding how to do that with the notification address. Can someone
point me in the right direction?
More specifically, suppose I have
:-)
Cheers,
Charles
On 3 Mar 2015 23:39, Robert Boisvert rdboisv...@gmail.com wrote:
Hello all,
I would like to setup a DMQ bus where servers can come on and offline and
have DMQ continue working with the online servers. I'm having trouble
understanding how to do
You might want to consider this function:
http://kamailio.org/docs/modules/4.3.x/modules/dmq.html#dmq.p.multi_notify
On Wed, Jul 22, 2015 at 10:25 AM, Asgaroth 00asgarot...@gmail.com wrote:
sorry, I forgot to mention, this is with Kamailio v4.3.1:
# kamailio -V
version: kamailio 4.3.1
David,
mohqueue was designed to hold calls in a FIFO queue until retrieved (
http://kamailio.org/docs/modules/4.3.x/modules/mohqueue.html#overview). It
does not include an algorithm that decides when to remove it from the queue
since there are many ways to determine that. Whether you program
This error: db_check_table_version(): invalid version 0 for table mohqcalls
found, exp
Indicates you didn't create the tables or the user does not have access to
it.
On Thu, Jul 21, 2016 at 2:31 PM, Gaurav Bmotra <saigauravmehr...@gmail.com>
wrote:
> Hi Robert
>
> Now it gives
ocs/modules/devel/modules/mohqueue.html#table.parms
and
http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#mohqueues.dbase
On Thu, Jul 21, 2016 at 3:23 PM, Gaurav Bmotra <saigauravmehr...@gmail.com>
wrote:
> i have created two tables
>
> mqueues and mcalls
>
> is that ok
>
It looks like you haven't correctly setup and configured the rtpproxy
feature which mohqueue relies on. See
http://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp2409144
On Fri, Jul 22, 2016 at 3:06 PM, Gaurav Bmotra <saigauravmehr...@gmail.com>
wrote:
> hi Robert
>
; but in my kamailio configuration file rtpproxy is configure
>>
>>
>>
>> On Sat, Jul 23, 2016 at 2:36 AM, Robert Boisvert <rdboisv...@gmail.com>
>> wrote:
>>
>>> It looks like you haven't correctly setup and configured the rtpproxy
>>> fea
Did you use kamdbctl to create the tables?
https://www.kamailio.org/wiki/install/4.3.x/git#create_mysql_database
On Thu, Jul 21, 2016 at 4:05 PM, Gaurav Bmotra <saigauravmehr...@gmail.com>
wrote:
> hi Robert
>
> this is my kamailio config
...@gmail.com>
wrote:
> Hi Robert:
>
> 1. all the MOH uncommented
> 2. mohq_process() set now
> 3. rtpproxy module is loaded
> 4. tables created
> 5. Audio dir is created and Audio file is there
>
> But when srtart the kamailio it gives the error
>
> -
Gaurav,
Your request is a little vague but the following things may be helpful:
1. Your configuration has commented out the module statements which
disables it.
- #loadmodule "mohqueue.so"
- #modparam ("mohqueue", "db_url",
I seem to recall reading something stating that you shouldn't modify the
contact header - But, I can't find the text.
Might suggest adding a Record-Route header instead of re-writing the
contact header?
I'm curious as to what the list has to say about this question, I've
considered doing
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