, July 11, 2014 at 8:23 AM
*From:* Răzvan Crainea raz...@opensips.org
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] cachedb_redis :: cachedb_url = FQDN not
working
Hi, Gary!
I've just created a patch[1] for this. Can you please apply it on your
sources and then retest?
[1] http
Hi, Dan!
Yes, you are on the right path. You do have to specify the evi_flag,
otherwise the acc module does not know what kind of accounting to do (DB
based, log based, aaa based, events based). If you don't specify the
cdr_flag, then you will get an old-style accounting - for a single call
Hi, Gary!
I've just created a patch[1] for this. Can you please apply it on your
sources and then retest?
[1] http://pastebin.com/TuEZ26Qz
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 07/11/2014 01:49 PM, Bogdan-Andrei Iancu wrote:
Hi Gary,
I
Hi, Kurtis!
I've just run your test and the output seems ok:
Kurtis [ 1, 2 ]
Are you sure there is no other function between those two lines that
could delete the json? Also, what version of OpenSIPS are you using?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Gary!
There's no way to tune the add_body() function buffer in the script. You
have to modify the PV_PRINT_BUF_SIZE (from pvar.c) variable and then
recompile opensips to make it work.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 07/04/2014
Hi, Jing!
Make sure you have the statistics module loaded.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 07/03/2014 10:48 AM, jing wrote:
hi,Liviu
Thanks for your reply.
But it doesn't work.
I use $stat(registered_users), and the log shows unknown
using just GNU make.
Are you guys registered on github? I've been trying to add you to the
repository ACL list but could not find anyone. Jev is off for few days,
but he should be back tomorrow to finish migration of the website etc.
Thanks!
-Maxim
On Fri, Jun 27, 2014 at 7:51 AM, Răzvan Crainea
Hi, Maxim!
Not sure what rtpp_2_0 is. Is it the latest git code[2]? If not, can you
point me the url to get it?
[2] https://github.com/sippy/rtpproxy
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 07/03/2014 02:23 PM, Răzvan Crainea wrote:
Hi
Hi, all!
We are proud to announce you that we have just released three new minor
releases for OpenSIPS: 1.11.2, 1.10.2 and 1.8.5. All three versions are
stable and ready for production use. Since they contain the latest bug
fixes, we strongly recommend you to upgrade your current instances!
Hi Russ!
If you need to determine the call duration, you should use the CDR flag[1].
Also note that 1.7 is no longer supported, so you might miss important
fixes of the acc module. I strongly suggest you to upgrade to a newer
version.
[1]
Hi, Jing!
1] Where do you want to add those users? In the subscriber table?
2] You can use the ratelimit module[A] to limit the amount of messages
per second.
3] I am not sure I get the question. You want to have a limit for the
maximum registered users? Or the maximum registrations of a
Hi, Jing!
Hi, Razvan ,
thanks for your reply.
1] Where do you want to add those users? In the subscriber table?
I want to add those users to subscriber.
I don't think of any method of limiting the rows in a table. Probably
only on the provisioning system side (opensipsctl, CP?). Just out of
reject calls or do whatever you want.
There is no explicit way to tell OpenSIPS how many users are allowed to
register.
Best regards,
Răzvan
Best regards,
-- 原始邮件 --
*发件人:* Răzvan Crainea;raz...@opensips.org;
*发送时间:* 2014年6月27日(星期五) 下午3:56
*收件人:* usersusers
Hi, Maxim!
Good news, I am glad you are interested in these features. I will fork
the project in our organization and push the requests there so you can
revise them before merging them.
We currently have them implemented for 1.2 - shall we adapt the changes
for the master branch?
Best
Hi, Salmuel!
Actually the dialog timeout is set to 120, but afterwards it is
overwritten by the SST module. Can you remove the SST module from your
script and re-test?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/24/2014 04:09 PM, Samuel
Hi, Samuel!
Yes, using that snippet should allow you to add information to the BYE
messages sent.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/25/2014 01:03 PM, Samuel Muller wrote:
Hello Razvan,
It's working when disabling SST module ...
Hi, Samuel!
Have you tested with the latest changes? Is the timeout properly printed
now?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/20/2014 12:28 PM, Samuel Muller wrote:
Hello,
Following all the ways, it did not work, even if using the
database?
On Thu, Jun 19, 2014 at 5:30 PM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Miguel!
There is no way you can change the contact header - it is something
taken from the SIP message. What you are changing there is the AOR
username and domain
Hi, Kneeoh!
Is this happening only from time to time or for all pipes? Are you
explicitly removing the keys from couchbase?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/19/2014 02:27 AM, Kneeoh wrote:
Any idea why a lab system with no traffic
Hi, Miguel!
Do you only want to change the Contact info that is stored in OpenSIPS
database or change the Contact header in the REGISTER message?
The code you pasted changes only the info in the location table. If you
want to change the header, you can remove it and add a new one.
Something
Hi, Gary!
What you are looking for is called parallel forking of a call and it
is the default behavior of OpenSIPS. All you have to do is to use the
default residential configuration (of course you have to tune it up with
your IP addresses and other custom settings) and register multiple
)=sip:$tU@$si:5092 '
but the variables $tU@$si doesn't like to be expanded in save().
I must be dynamic (not hard-code)
Any idea how to do it?
On Thu, Jun 19, 2014 at 4:39 PM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Miguel!
Do you only want to change
=wss
Contact is still the original. Any idea?
On Thu, Jun 19, 2014 at 5:04 PM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Miguel!
The save() function[1] only accepts a pseudovariable, not a format
as parameter. And the way you build the $avp(aor
Hi, Samuel!
The $DLG_timeout variable was indeed printing a bogus value. I fixed
that on trunk and 1.11, thanks for reporting it.
However, setting the value should work anyway. I think on your setup,
the timeout is taken into account and the dialog is deleted, but the BYE
messages are not
among the rabbit cluster nodes and see if that tricks it.
On Tuesday, June 10, 2014 9:01 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Kneeoh!
I think that the only solution that would work properly was your first
approach. However, since this is not yet implemented, all I can think
Hi, Kaushik!
Indeed, as Salman pointed out, you should not use the r flag unless
you really want to use the private IP in the SDP for RTP.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/12/2014 08:44 AM, Salman Zafar wrote:
Kaushik,
'r'
Hi, Rick!
I was asking for RTPProxy debugs, not OpenSIPS. Can you provide it?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/06/2014 03:42 PM, Rik Broers wrote:
I've sent you some Debug logs on a separate mail.
Kind Regards,
Met vriendelijke
Hi, Kaushik!
Are you using the r flag for rtpproxy_offer/answer() functions?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/11/2014 02:51 PM, kaushik parmar wrote:
Hello ,
I am using opensips and rtpproxy for sip and rtp proxy server. Here
Hi, Kneeoh!
I think that the only solution that would work properly was your first
approach. However, since this is not yet implemented, all I can think of
is an external process that periodically test if the node is up. If it
is not, unsubscribe it and re-subscribe the second node.
PS: I
Hi, Kneeoh!
Currently the event_rabbitmq module does not need expiration. Since it
is a connection oriented protocol, if the external application does not
want to receive messages any more, it should close the connection. If
you really need the expire feature in the near future, please open a
Hi, Rik!
The R flag indicates that the IP advertised in SDP should be trusted.
Can you increase the debugging level of rtpproxy to DEBUG? This way we
can see exactly what commands are sent by OpenSIPS and check what's
wrong with them.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Kneeoh!
I've never used haproxy, but I will give it a shot when I will find some
spare time.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/28/2014 05:08 PM, Kneeoh wrote:
Thanks Razvan, I'm definitely interested in what you may have in the
Hi, Kneeoh!
Currently failover between different nodes is not entirely supported. We
do have some work on progress towards this, but it is not finalized.
Stay tuned :).
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/27/2014 05:28 PM, Kneeoh
Hi, Kneeoh!
RabbitMQ runs over TCP and you cannot simply switch an ongoing TCP
connection. The only thing you can do is close the initial TCP
connection and reconnect it.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/28/2014 12:04 AM, Kneeoh
Hi, Mike!
Pay attentions that you can not simply write the $fU variable. Instead,
you should re-construct the From URI with the correct username and
replace the whole URI using uac_replace_from() function[3].
[3] http://www.opensips.org/html/docs/modules/1.12.x/uac#id293710
Best regards,
Hi, Mike!
If you want to change the RURI, you can use:
$ru = sip: + $avp(2) + $avp(1);
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/20/2014 10:03 AM, Mike Claudi Pedersen wrote:
Hi guys.
i have 2 variables $avp(1) $avp(2)
$avp(1) =
It depends what URI you want to change. You can change the R-URI by
setting the $ru variable. However, for From/To URI, you have to use
special functions[1]. For other headers, you can simply remove the
header and add a new one, with the correct URI[2],[3]
[1]
Hi, Gordon!
Actually this CDR is not encrypted, the data is taken plain text from
the message. Here is the interpretation of a few fields:
26A86646-D8FC11E3-9507C13F-D7186497@10.0.119.253 - callid
4BD8F748-1DEE - from tag
as09912ec8 - to tag
Best regards,
Razvan Crainea
OpenSIPS Core
to modify
what is sent over to the MySQL database?
Thanks again,
Gordon
On 5/15/14, 4:20 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Gordon!
Actually this CDR is not encrypted, the data is taken plain text from
the message. Here is the interpretation of a few fields:
26A86646-D8FC11E3
Hi, Garmin!
First of all, please truncate the cc_calls table. This should prevent
the mysql duplicate error.
Next make the call. According to your previous mail, you are hearing the
queue message provided by the Asterisk box, right? If so, please run the
following opensipsctl commands:
Hi, Takeshi!
I am not sure you patch was successfully installed, since I don't see
the logs I put in the patch.
Is there any chance you could give me access to a development machine
where this is happening?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips
, May 7, 2014 at 5:10 PM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Takeshi!
I am not sure you patch was successfully installed, since I don't
see the logs I put in the patch.
Sorry. I was loading the modules from the wrong path so it was using
unpatched
about patch file formats. Please instruct me on how to
apply the patch.
Regards,
Takeshi
On Mon, May 5, 2014 at 7:06 PM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Takeshi!
Can you please apply this patch [1] and run again in debugging mode?
Thanks,
[1
on the initial INVITE. This works
with 1.10.
Then with 1.11, I tested setting the CDR_FLAG when processing BYE
requests but it didn't work.
Then I set DB_FLAG for BYE and after that, it worked.
So this behavior changed.
Thanks a lot.
Regards,
Takeshi.
On Wed, Apr 30, 2014 at 11:05 PM, Răzvan
Hi, Takeshi!
I just tested and there are no issues on my setup. Are you setting the
CDR_FLAG for all the requests? Have you also tried with different db modes?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/23/2014 03:05 PM, mayamatakeshi wrote:
Hi, Khaled!
After your loose_route() call, you should do something like this:
if ($DLG_status == NULL)
xlog(ERROR: $rm not matched\n);
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/24/2014 11:25 AM, M.Khaled W Chehab wrote:
HI Razvan,
Yes
#id293555
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/18/2014 05:33 PM, leo wrote:
Hello Razvan:
I've done one more test. I'm testing with my opensips.org accounts. So, same
clients, same connections. Just different server... and it is working
Hi, Rick!
Indeed, there was a typo in the configuration file. The
unforce_rtp_proxy() function has been replaced by rtpproxy_unforce()
function[1].
[1] http://www.opensips.org/html/docs/modules/1.11.x/rtpproxy#id248034
Best regards,
Razvan Crainea
OpenSIPS Core Developer
,
Samuel MULLER
Telecom Media Consultant
co-owner
L33 NETWORKS
+33 663 128 505
s...@l33.fr
www.l33.fr
On Mon, Jan 20, 2014 at 2:59 PM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Samuel!
Are you loading the 'rr' module in your script? Also, have you tried the
'manual' mode?
Bet regards,
Razvan
Hi, Leo!
From the logs, for me it is pretty clear that the connection does not
exist, therefore OpenSIPS cannot contact your client behind NAT.
After finally managing to call over TCP (after 10 seconds or so),
placing another call succeeds? Or you can never call that client over TCP?
Best
Hi, Michael!
There is no function that allows you to change the whole SIP message. In
order to change the From/To header, you have to use
uac_replace_from/to() functions from the uac module.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On
Telecom Media Consultant
co-owner
L33 NETWORKS
+33 663 128 505
s...@l33.fr
www.l33.fr
On Thu, Apr 17, 2014 at 2:33 PM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Samuel!
Sorry for getting back so late. Do you still have this issue with the latest
1.10 source?
Best regards,
Razvan Crainea
Can you trace the traffic between the server and the clients in those 10
minutes? Is the TCP connection kept alive by any sort of pinging?
Are you using the keep_alive parameter?
[1] http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc98
Best regards,
Razvan Crainea
OpenSIPS
Hi, Khaled!
You should first check what is the problem with that ACK that is ignored
by the Gateway. I presume you are using dialog support and most likely,
the ACK and the BYE are not matched by OpenSIPS.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Leo!
The first logs show that opensips is unable to ping your client over
TCP. Most likely because OpenSIPS was restarted and the TCP connection
was lost.
The next errors show that opensips is trying to connect to a TCP
connection, but it gets a timeout. That's because probably your
Hi, Maksim!
You should trace the traffic on the opensips machine and check whether
opensips sends the correct messages (SDP body) to each user agent.
Next, you should trace on the RTPProxy to see if you get media stream
from both ends.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Chen-Che!
No, OpenSIPS does not have any built-in mechanism for this. As far as I
understand, you sometimes need to rewrite the IP OpenSIPS advertises in
the SDP. You can specify the new IP in the second parameter of the
rtpproxy_offer/answer() function.
Best regards,
Razvan Crainea
Hi, Leo!
This doesn't seem to be related to OpenSIPS. My suggestion is to trace
this at the network layer and check if and why the TCP/TLS connection is
taking 15-20 seconds.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/14/2014 01:17 AM, leo
Hi, Duane!
I fixed the link, it was 1.11 indeed.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/20/2014 11:43 PM, Duane Larson wrote:
Good job. Looks like the Read More . . . link for New Call_Center
Module on page
Hi, Chen-Che!
Are you sure it is a parsing problem? Do you see any errors in the logs?
Because my assumption is that your problem is in RTPProxy since (if I
remember correctly) it does not know how to handle port 0 sessions.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Chris!
Can you provide on pastebin the system debug logs?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/19/2014 05:53 PM, Chris Maciejewski wrote:
Hi,
Just upgraded to 1.10.1 and when starting openSIPS getting error as
below in a log file:
Hi, Chris!
Can you try loading the db_mysql module first, above all modules?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/20/2014 01:08 PM, Chris Maciejewski wrote:
Hi,
Please find below link to a pastebin showing system debug logs:
Hi, Nick!
Migrating from 1.8 to 1.9 should not be very hard. You can start by
following the migration tutorial[1].
[1] http://www.opensips.org/Documentation/Migration-1-8-0-to-1-9-0
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/13/2014 02:01
Hi, Michael!
Let us check with the Debian maintainer. I'll let you know as soon as I
have further information.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/13/2014 10:12 AM, Michael Renzmann wrote:
Hi Razvan.
Just wanted to let you know
/
All versions are stable and include all the latest fixes. Enjoy!
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/10/2014 06:38 PM, Răzvan Crainea wrote:
Hello, all!
We are proud to announce you that during this week, we'll be releasing
three new
Hi, Mike!
You can find here[1] the core pseudo variables for OpenSIPS 1.8. Each
module has it's own variable, for example[2]. If you need a variable
from a module, you should search in that module's documentation page.
For example, if you need the number of onging calls, you should look in
Hello, all!
We are proud to announce you that during this week, we'll be releasing
three new OpenSIPS minor versions: 1.8.4, 1.9.2 and 1.10.1. All of them
will include the latest fixes we've done during the last 6 months. If
you have any other outstanding issues, please report them asap on
Hi, Gary!
The configuration file looks ok. Can you check if the /var/log/acc
directory exists? Do you see any errors in the logs?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/05/2014 11:40 PM, Gary Nyquist wrote:
Hi,
I am trying to configure
Hi, jzw!
For devel questions, please use the devel list[1].
The struct sip_msg structure is not a global variable, but for a
specific message, it is the same for the entire route execution.
You can use pseudo-variables to store information. Depending on the
scope you want to use them, you
Hi, Yavari!
Yes, I think you should call this method when you receive the INVITE.
Also, you should probably convert the 180 to 183. Also, try to do some
traces on the RTPProxy machine, to see if it tries to generate media.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Yavari!
That error indicates that RTPProxy couldn't find any available codecs
for your client. Please check the files in /var/rtpproxy/prompts/092.*
to make sure you have the proper files installed.
I find something weird in your examples: you're saying that you are
calling the function
Hi, Yavari!
In your scenario, you want to play media to UAC, not UAS, right? Does it
work properly now, can the caller hear the ringing?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/26/2014 01:34 PM, H Yavari wrote:
Hi,
Your hint solved the
Hi, Miha!
You can run 'opensipsctl fifo which'
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/26/2014 03:32 PM, Miha wrote:
Hi Razvan.
tnx for this:)
where can I find all commands that are avalible?
br
miha
Dne 2/26/2014 1:57 PM, piše Răzvan
Hi!
What do you mean the function does not work properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to
Hi, Wilmar!
Have you tried setting the $DLG_dir pvar in the db_extra? For example:
modparam(acc, db_extra, direction=$DLG_dir)
Don't forget you have to add the direction column to your database
table, both acc and missed_calls.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, John!
There's no way to automatically add the direction attribute. If you
want to add it, you have to manually change the SDP using a regular
expression[1,2,3].
[1] http://www.opensips.org/html/docs/modules/1.10.x/textops.html#id293689
[2]
Hi, Samuel!
Are you loading the 'rr' module in your script? Also, have you tried the
'manual' mode?
Bet regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/15/2014 04:08 PM, Samuel Muller wrote:
FYI,
I noticed that OpenSips does not crash when module
Hello!
You can try the $stat(active_dialogs) pseudo-variable. Make sure you are
loading the statistics module.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/10/2014 03:29 PM, M.Khaled W Chehab wrote:
Dears,
I there a function in the script
Hi, Gordon!
Are there any other errors? Can you check if the /etc/opensips directory
exists?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/24/2013 03:27 PM, Gordon Ross wrote:
I’m trying to use osipsconfig to generate an OpenSIPs script. But
,IP media works. I am able to traverse from eternal to internal
play media and then on failure do external to external with media
flowing between public interfaces. Just wondering if you know this
method or certify.
On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea raz...@opensips.org
mailto:raz
Hi, Miha!
remove_hf() is the function you should use. According to your scenario,
you should remove the RPID and PAI in the failure route. Are you sure
that code is reached for the second INVITE?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On
Hi, Eddie!
By default OpenSIPS uses parallel forking for all the branches, created
by the enum_query() function in your case. If you want to use serial
forking, you should call the serialize_branches() function[1] after
creating the branches.
[1]
Hi, Ahsan!
Can you please enable the the memory debugging (DBG_QM_MALLOC flag in
menuconfig)? You can follow this tutorial[1].
[1] http://www.opensips.org/Documentation/TroubleShooting-OutOfMem
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On
Hi, Vishnu!
The PEM pass phrase is like a password you should add to protect your
certificates. It does not have a default value, it is something that you
should specify.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/08/2014 10:11 AM, Vishnu
Hi, Salman!
The sockets used by RTPProxy are created when the session is started
(the first offer) and cannot be updated afterwards. Therefore the only
solution I can see is to configure a per branch scenario, as you mentioned.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Vishnu!
This means that mi_xmlrpc does not consider the port parameter. Have you
tried moving it to the begining of your script?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/20/2013 02:44 PM, Vishnu Vardhan wrote:
Hi,
Thanks for response
Hi, Dragomir!
You can take the parameter from the dlg_list MI command. Alternatively,
you can take the id from the dlg_id column - first 32 bits (most
significant) represent the h_entry and the last 32 bits (least
significant) represent the h_id.
Best regards,
Răzvan Crainea
OpenSIPS Core
received the INVITE request.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/21/2013 03:24 AM, mayamatakeshi wrote:
Hello,
I have a requirement to delay the relay of '180 Ringing' for 3 seconds.
So what I did was to intercept the '180 Ringing
/1.10.x/dialog.html#id295287
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
Hello Guys,
Im getting a strange situation here that i dont know how to deal
i have an enviroment where i have freeswitch receiving
Hello!
Can you attach with gdb to one of the processes that are using 100% CPU
and post the backtrace on pastebin? You should do something like:
gdb /path/to/opensips pid
bt full
And upload the entire output on pastebin.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Nick!
The t_check_trans() function should absorb BYE retransmissions too. Are
you sure your BYE messages reach that function call?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/16/2013 01:09 PM, Nick Altmann wrote:
Hello!
What is proper
and one in missed_calls.
Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com mailto:mustafin.aleksa...@gmail.com
02 дек. 2013 г., в 15:07, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org написал(а):
Hi, Alexander!
Are you sure you are populating all the multi_leg_info
:44, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org написал(а):
Hi, Alexander!
If you want to have multiple rows for each leg, then you should use
multi-leg acc support[1]. Note that you should not use the CDR flag,
since you are doing old two-steps accounting.
[1] http
, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org написал(а):
Hi, Alexander!
Have you tried setting the ACC_FAILED flag in the request route?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com http://www.opensips-solutions.com/
On 11/28/2013 10:35
Hi, Alexander!
Have you tried setting the ACC_FAILED flag in the request route?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/28/2013 10:35 AM, Alexander Mustafin wrote:
Hello!
I need to store all CDRs for all calls, but some failed calls are
Hi, Jeff!
Unfortunately it is not possible to use engage_rtp_proxy() if the
initial INVITE did not use late negociation. Also, using
engage_rtp_proxy() is not recommended when using bridging scenarios, so
you'd better check all the possible cases and make sure everything works
properly. But
Hi, Jeff!
Are you using a DB_ONLY mode for the usrloc persistence[1]?
[1] http://www.opensips.org/html/docs/modules/1.10.x/usrloc.html#id294121
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/22/13 20:19, Jeff Pyle wrote:
Hello,
On 1.10 I'm
the AOR is no longer visible in usrloc.
Where does the 36 seconds come from? Is it adjustable?
- Jeff
On Mon, Nov 25, 2013 at 4:09 AM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Jeff!
Are you using a DB_ONLY mode for the usrloc persistence[1]?
[1
to eliminate the deprecated warning:
...
$avp(my_port) = 5060;
if (ds_is_in_list($avp(my_ip),$avp(my_port),1,1)) {
...
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/20/2013 03:03 PM, Samuel Muller wrote:
Hey,
I would like to be sure that since
Hi, Aldo!
I need the command that generates the core dump. You should see it above
the line you pasted and should be a C instruction.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/11/2013 06:33 PM, Aldo Jose Spanghero Romao wrote:
Hi, Răzvan
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