Hi,
Here is your error:
ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST]
If you look up in documentation it says on first line:
JSON support via HTTP GET for Management Interface
Regards
Qasim
On Wed, 6 Nov 2019 at 6:58 AM, Jeff Wilkie wrote:
> Attempting to get CP8 and
You can use latest version of freeradius it has both client and server.
Regards,
Qasim
On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie wrote:
> Attempting to find current docs since radiusclient-ng is referenced in
> several old docs but is no longer available. Currently, what is the
>
I think there was a short flux for around 10-15 minutes.
Regards,
Qasim
On Tue, Dec 26, 2017 at 8:21 PM, Impala Tux <impala...@gmail.com> wrote:
> Hi
>
> In Brazil it's right, no problems
>
> Em 26 de dez de 2017 08:05, "qasimak...@gmail.com" <qasimak...@gmail
Hi,
I am getting 504 Gateway Timeout on opensips.org. Is anyone else facing the
same issue?
Regards,
Qasim
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
cu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/13/2017 02:00 PM, qasimak...@gmail.com wrote:
>
> Hi Bogdan,
>
> Yes i ha
gt; I guess you try to do some async stuff in the reply route, right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amster
he output ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/13/2017 12:09 PM, qasimak...@gma
Hi,
I have upgraded my script from 1.11 to 2.2.3 which works fine until i put
async function on a single rest_get query. When the async line is executed
i get following errors:
2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
> opensips[11475]: segfault at 10 ip 00427260
I think there is no such direct command that will calculate CPS for you,
However there are certainly ways you can calculate this CPS (One example
given by Aqs) but since you are crunching maximum CPS from opensips i would
recommend that you dont use opensips for its calculation, reason being that
Because i have a similar scenario (150cps) but i
> set children to 20 or 24, never 200. You don't need 1 children per request.
>
> On Wed, Apr 5, 2017 at 9:44 AM, qasimak...@gmail.com <qasimak...@gmail.com
> > wrote:
>
>> Hi,
>>
>> I have this scenario where i
Hi,
I have this scenario where i originate calls from mi_datagram and the calls
are cancelled as soon as it starts ringing. The problem i am facing is that
are running for a few minutes the response from opensips becomes slow i.e.
it send packets back to far end after a few seconds. Keeping it
> Regards
>
> On Tue, Mar 28, 2017 at 10:29 AM, qasimak...@gmail.com <
> qasimak...@gmail.com> wrote:
>
>> Hi,
>>
>> Sorry for the spam last email i miss-clicked on send amidst writing the
>> email.
>>
>> Anyways the problem i am facing i
Hi,
Sorry for the spam last email i miss-clicked on send amidst writing the
email.
Anyways the problem i am facing is that my ACC module is configured with
MySQL DB backend and the CDR's are being written. However the problem i am
facing is that it is not logging duration into DB or syslog. Here
Hi,
I have enabled acc module in my opensips installation with db, My CDR's are
being written in MySQL backend but for every call the duration remains 0, I
have checked but according to documentation duration is automatically
logged in ACC module. PLease note following debug filtered where query
Dear Feroz,
>From past experience DO NOT ATTEMPT this :). You will surely run into many
bottle necks from switching to NoSQL. NoSQL is good for caching during call
processing not for storing persistent data.
Regards,
Qasim
On Thu, Jul 14, 2016 at 11:38 PM, Jim DeVito wrote:
>
I guess you can reffer to this -> (
http://www.opensips.org/About/PerformanceTests-StressTests) document for
some estimates and +1 Eric.
Regards,
Qasim
On Wed, Jul 13, 2016 at 11:09 PM, Eric Tamme wrote:
> Short answer, no. There are many variables that will change what a
Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.06.2016 22:21, qasimak...@gmail.com wrote:
>
> Dear Team,
>
> There is another segfault when i try to run with my old configuration from
> opensips 1.1
Dear Team,
There is another segfault when i try to run with my old configuration from
opensips 1.11. as far as i can understand it dosent go beyond loading the
modules. Please find below logs and backtrace.
syslog: http://pastebin.com/EAqTKu1n
backtrace: http://pastebin.com/rP9JDeDW
Regards,
lease pull the latest sources and redo your tests!
>
> [1]: https://github.com/OpenSIPS/opensips.git
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 01.06.2016 13:17, qasimak...@gmail.com wrote:
>
> Dear Razvan,
>
> Please find below b
stebin for further
> investigation.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 05/31/2016 10:12 PM, qasimak...@gmail.com wrote:
>
> Hi,
>
> I was using default script generated by opensips menuconfig and it gives
> the
Hi,
I was using default script generated by opensips menuconfig and it gives
the following segfault
http://pastebin.com/6zuimn5N
I was evaluating opensips 2.2 latest release. Please let me know if core
dump is required
Regards,
Qasim Ayyaz Khan
___
Try using opensips dictionary.
Regards,
Qasim
On Mon, Mar 24, 2014 at 4:37 PM, John Quick john.qu...@smartvox.co.ukwrote:
I'm using OpenSIPS version 1.8.2 with radiusclient-ng.
I need to be able to make custom radius authentication requests using
radius_send_auth (a function in the
: 0x..
When I include dictionary.rfc2869, I get this error on startup:
rc_read_dictionary: invalid type on line 13 of dictionary
/usr/local/etc/radiusclient-ng/dictionary.rfc2869
John
From: qasimak...@gmail.com [mailto:qasimak...@gmail.com]
Sent: 24 March 2014 11:57
To: john.qu
I think a little more information than this would be required if you need
help :).
Regards,
Qasim
On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote:
Hello!
1. In log file I see many errors
CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!
Hi,
Please follow this tutorial line by line.
http://saevolgo.blogspot.com/2012/05/installing-opensips-on-ubuntu-server.html
Regards,
Qasim
On Tue, Feb 25, 2014 at 6:07 PM, Tomasz Chmielewski man...@wpkg.org wrote:
I'm trying to install opensips on Debian or Ubuntu.
However, the provided
Hi,
I am using radius accounting and during that accounting i calculate the
maximum amount to call duration. I am setting dialog timout_avp in my route
but the call doesn't hangup. From documentation i see that i should use
timeout_avp before loose_route().
My question is that how i can use
yes.
Regards,
Qasim
On Mon, Jun 24, 2013 at 6:51 PM, Laszlo las...@voipfreak.net wrote:
Do you use create_dialog(B); ?
-Laszlo
2013/6/24 qasimak...@gmail.com qasimak...@gmail.com
Hi,
I am using radius accounting and during that accounting i calculate the
maximum amount to call
Hi Dani,
You most probably don't have correct dictionary files placed. You can turn
debug=6 and it then see if you have any dictionary items missing. Every
time i install a new opensips with radius accounting i end up missing
dictionary file in one or more places and opensips does not show it to
Hi,
I wanted to know that there are two flags to force Symmetric RTP in
rtpproxy module i.e. s/w. Does these two flags have different working
principles or does it work exactly like the i/e flags which work in
bridging mode?
Regards,
Qasim
___
Users
Hi,
I wanted to know that there are two flags to force Symmetric RTP in
rtpproxy module i.e. s/w. Does these two flags have different working
principles or does it work exactly like the i/e flags which work in
bridging mode?
Regards,
Qasim
___
Users
. It is not similar to the i/e flags.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.**com http://www.opensips-solutions.com
On 05/27/2013 09:53 AM, qasimak...@gmail.com wrote:
Hi,
I wanted to know that there are two flags to force Symmetric RTP in
rtpproxy
,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 05/24/2013 04:58 PM, qasimak...@gmail.com wrote:
Dear Bodgan,
This is what i am doing for ACK path (This is minus all the other crap
like auth, redis stuff). I dont think that there would be any loop here
.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 05/22/2013 03:21 PM, qasimak...@gmail.com wrote:
I think that is retransmission of ACK packet because it didn't get its 200
ok back.
Regards,
Qasim
On Tue, May 21, 2013 at 10:08 PM
-solutions.com
On 05/20/2013 02:46 PM, qasimak...@gmail.com wrote:
Hi Bodgan,
Sorry for the late reply as i was traveling this weekend. Please find
attached call logs with debug mode 4.
Regards,
Qasim
On Fri, May 17, 2013 at 8:50 PM, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
Funny
to post a SIP capture of the full call, to see how the RR part
is done.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 05/16/2013 01:07 PM, qasimak...@gmail.com wrote:
On further investigation i see that i only face this issue when both
On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
Hi,
I am using OpenSIPs in Public-Private bridging mode and have enabled
mhomed=1. But the problem is that when we have a call in which both parties
are on Public interface the INVITE gets relayed properly but and ACK of
that invite gives
/16/2013 12:32 PM, qasimak...@gmail.com wrote:
Hi Bodgan,
Yes i see the following route header in my packet.
Route:
sip:622190004002@202.152.203.195:6000;lr;ftag=3b710c25;did=e55.a77ff685
And yes i am routing it through loose_route.
Regards,
Qasim
On Wed, May 15, 2013 at 10:40 PM
On further investigation i see that i only face this issue when both caller
and callee are on the same network. If both are on separate network it
works fine.
Regards,
Qasim
On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com
qasimak...@gmail.comwrote:
yes.
Regards,
Qasim
On Thu, May
-solutions.com
On 05/10/2013 06:41 AM, qasimak...@gmail.com wrote:
Yes exactly that is being done perfectly but what i want to do is to
handle NAT on client's end. The IP of client that comes in the SDP's c=
param is his local IP address and rtpproxy swaps that IP with server's
local IP
Hi,
I am using OpenSIPs in Public-Private bridging mode and have enabled
mhomed=1. But the problem is that when we have a call in which both parties
are on Public interface the INVITE gets relayed properly but and ACK of
that invite gives the following error.
ERROR:core:get_out_socket: no socket
Hi,
I am facing a problem when a client connects to opensips from NATed
network. I am using rtpproxy in bridging mode i.e. from publicnetwork to
private network. When i use fix_nated_sdp function from nathelper the local
IP address of the caller is replaced by its public IP but the problem
starts
For engage_rtpproxy there are two flags that are used i.e. i for LAN
interface and E for WAN interface. you can use these two flags to specify
your direction of bridging. e.g. ie for LAN to WAN bridging and ei for WAN
to LAN bridging. Meanwhile look at this documentation for detailed flag
usage.
scripting please.
Nick.
On 5/9/13, qasimak...@gmail.com qasimak...@gmail.com wrote:
Hi,
I am facing a problem when a client connects to opensips from NATed
network. I am using rtpproxy in bridging mode i.e. from publicnetwork to
private network. When i use fix_nated_sdp function from
end nat related scripting please.
Nick.
On 5/9/13, qasimak...@gmail.com mailto:qasimak...@gmail.com
qasimak...@gmail.com mailto:qasimak...@gmail.com wrote:
Hi,
I am facing a problem when a client connects to opensips
from NATed
the event) - the generate event is the
same. See:
http://www.opensips.org/html/docs/modules/1.9.x/acc.html#id294346
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 04/30/2013 08:00 AM, qasimak...@gmail.com wrote:
I have tried
Hi,
I wanted to confirm if radius accounting requests are generated on a
successful transaction or it can be generated on a received BYE only. To
elaborate my question you can look at 2 diagrams below. Is first scenario
correct based on RFC's? My next question is that if scenario A is correct
and you should use stop time as accounting end time
rather then the time your receive account stop request on radius (they both
may differ, e.g. under high load scenarios).
Thank you.
On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com
qasimak...@gmail.com wrote:
Hi,
I wanted to confirm
Just forward your call to any Media Server capable of transcoding and let
it forward the call to destination. You can use Asterisk or Freeswitch.
This should be a simple scenerio.
Regards,
Qasim
On Thu, Apr 25, 2013 at 1:49 AM, Dragomir Haralambiev goup2...@gmail.comwrote:
I use follow
You can turn on in dialog ping using pP flag i.e. create_dialog(Pp);
and send acct packet to radius on its reply.
-Qasim
On Sun, Apr 21, 2013 at 9:18 AM, Ewgeny ev...@ukr.net wrote:
Hi!
I use Opensips 1.9 with RADIUS accounting functions - radius_send_auth
and radius_send_acct (AAA RADIUS
Hi,
Just wondering how to use new string flags in acc module in opensips
version 1.9. My script works fine on opensips 1.8 but in 1.9 i dont get
accounting packet in radius.
Regards,
Qasim
___
Users mailing list
Users@lists.opensips.org
If you are using Asterisk then you dont need media proxy as asterisk can
handle NAT and Media issues. You just need to forward SIP messages to
asterisk. Use OpenSIPs in LoadBalancer/Dispatcher scenerio.
Regards,
Qasim
On Thu, Mar 28, 2013 at 12:07 AM, Jagadish Thoutam
Have you confirmed that table/DB name is correct? You can verify your
version table from config.h in your sources.
-Qasim
On Sun, Mar 17, 2013 at 10:32 PM, Peter Zoltan Keresztes zozo6...@gmail.com
wrote:
Hello,
I have installed an opensips 1.9
I have the configuration copied from an 1.8
Also since you copied script from 1.8 you should consider going through
this document.
http://www.opensips.org/Resources/DocsMigration180to190
-Qasim
On Mon, Mar 18, 2013 at 2:34 PM, qasimak...@gmail.com
qasimak...@gmail.comwrote:
Set debug level to 6 and then send the logs again.
-Qasim
Hi,
My opensips configuration was running fine until I enabled dialog ping flag
in create_dialog. Now after enabling ping my opensips crashes randomly 3-4
times daily. I have collected opensips logs which are as follows:
P.S: Please let me know if anything else is needed. I am preserving my core
Hi,
Sorry for the spamming but my opensips version is 1.8 SVN Rev 9447.
Regards,
Qasim
On Wed, Feb 13, 2013 at 2:00 PM, qasimak...@gmail.com
qasimak...@gmail.comwrote:
Hi,
My opensips configuration was running fine until I enabled dialog ping
flag in create_dialog. Now after enabling ping
advice you to
update from SVN (1.8 branch) and give it another try.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 02/13/2013 11:04 AM, qasimak...@gmail.com wrote:
Hi,
Sorry for the spamming but my opensips version is 1.8 SVN Rev 9447
I would start from here...
https://help.ubuntu.com/community/LiveCDCustomization
Regards,
Qasim
On Thu, Dec 27, 2012 at 9:13 PM, M.Khaled W Chehab kche...@icucall.comwrote:
-How can I make an iso image in order to install this distro in other
servers to avoid working from scratch ?
Can you share your Config and SIP Dump? I don't seems to have it in this
email chain.
On Tue, Nov 6, 2012 at 3:05 PM, spady spad...@gmail.com wrote:
Hi, can someone help me understand this issue?
Thanks
--
View this message in context:
Try replacing your ACK block:
t_relay();
exit;
with:
if (src_ip == 10.9.6.40) {
route(1)
}
if (src_ip == 10.9.6.3) {
route(2)
}
Regards,
Qasim
___
Users mailing list
Users@lists.opensips.org
I think a better way than adding copy command to init script would be to
use incron http://inotify.aiken.cz/?section=incronpage=aboutlang=enutility.
This utility triggers cron jobs based on file system triggers so
you can backup a file when it is changed. You can do SVN/Git commit or even
do an
VIA Parser Patch for WS WSS:
http://sourceforge.net/tracker/?func=detailaid=3545859group_id=232389atid=1086412
Regards,
Qasim
On Fri, Nov 2, 2012 at 6:43 AM, Binan AL Halabi binanalhal...@yahoo.comwrote:
Hi All,
If oversip uses Path extension OpenSIPS must support it.
1- Sending Path
I haven’t tried doing this before but if i am not wrong you can write a
script in perl using opensips perl module.
Regards,
Qasim
On Thu, Oct 18, 2012 at 11:53 AM, Engineer Voip forvo...@gmail.com wrote:
Hello all,
I want to transfert the call to user C when user A calls user B in
interval
Advise: Read threads initiated by you thoroughly.
Read: http://blog.sipvicious.org/ to know more about the tool we all face
every once a while.
Regards,
Qasim
On Tue, Oct 9, 2012 at 2:27 PM, Engineer voip forvo...@gmail.com wrote:
Hi All,
thank you for your reply, Know i want to simulate
Help,
Nick.
On Fri, Aug 31, 2012 at 12:11 AM, qasimak...@gmail.com
qasimak...@gmail.com wrote:
Most probably your RTP stream is directly bein connected to * server.
RTP is
used for many purposes most commonly to hide your internal network
topology
i.e. in your case your * server also
Most probably your RTP stream is directly bein connected to * server. RTP
is used for many purposes most commonly to hide your internal network
topology i.e. in your case your * server also it helps in NAT traversal.
Regards,
Qasim
On Fri, Aug 31, 2012 at 8:05 AM, Nick Khamis sym...@gmail.com
My friend has this good walkthrough's for Opensips configuration and RTP.
and example is
http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html
You can also find other posts there. Just go through them and you will be
good to go.
PS: I also learned using rtpproxy using above mentioned
Have you installed and started rtpproxy? if not just scroll through this
website http://www.rtpproxy.org/.
Regards,
Qasim
On Fri, Aug 3, 2012 at 2:27 AM, Ashish Kundu kash...@gmail.com wrote:
Opensips is a great product, but I have been having problem in configuring
the nat traversal +
Maybe this would help
For failed SIP sessions a radius packet type FAILED is generated. A failed
SIP session is a session that has been rejected by the Proxy server or by
the destination. Unfortunately Freeradius server is not able to cope with
FAILED packets. The server must be patched and
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