Re: [OpenSIPS-Users] Opensips 2.4.x and CP 8

2019-11-05 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9

2019-11-05 Thread qasimak...@gmail.com
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 >

Re: [OpenSIPS-Users] Website Down.

2017-12-26 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Website Down.

2017-12-26 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

2017-04-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

2017-04-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

2017-04-13 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Opensips 2.2.3 crash on async.

2017-04-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] opensips performance

2017-04-07 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips Late reply.

2017-04-05 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Opensips Late reply.

2017-04-05 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] ACC Db Duration

2017-03-28 Thread qasimak...@gmail.com
> 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

[OpenSIPS-Users] ACC Db Duration

2017-03-28 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] ACC db duration

2017-03-28 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips switch from SQL to NoSQL

2016-07-17 Thread qasimak...@gmail.com
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: >

Re: [OpenSIPS-Users] horizontal scaling / dimensioning

2016-07-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Segfault in opensips 2.2

2016-06-02 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Segfault in opensips 2.2

2016-06-01 Thread qasimak...@gmail.com
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,

Re: [OpenSIPS-Users] Segfault using Loadbalancer Module.

2016-06-01 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Segfault using Loadbalancer Module.

2016-06-01 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Segfault using Loadbalancer Module.

2016-05-31 Thread qasimak...@gmail.com
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 ___

Re: [OpenSIPS-Users] Problem using radius_send_auth

2014-03-24 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Problem using radius_send_auth

2014-03-24 Thread qasimak...@gmail.com
: 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

Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

2014-03-24 Thread qasimak...@gmail.com
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!!

Re: [OpenSIPS-Users] installing opensips on Debian / Ubuntu

2014-02-25 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Dialog timeout_avp.

2013-06-24 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Dialog timeout_avp.

2013-06-24 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout

2013-06-23 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

2013-05-27 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

2013-05-27 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

2013-05-27 Thread qasimak...@gmail.com
. 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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-27 Thread qasimak...@gmail.com
, 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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-24 Thread qasimak...@gmail.com
. 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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-22 Thread qasimak...@gmail.com
-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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-17 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-16 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-16 Thread qasimak...@gmail.com
/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

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-16 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

2013-05-13 Thread qasimak...@gmail.com
-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

[OpenSIPS-Users] VIA relay error using mhomed=1

2013-05-13 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] RTPProxy for users behind NAT.

2013-05-09 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] OpenSIPS with public/private interface and RTPProxy

2013-05-09 Thread qasimak...@gmail.com
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.

Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

2013-05-09 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

2013-05-09 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

2013-05-02 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] OpenSIPs Radius Accounting.

2013-04-29 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

2013-04-29 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Need to transcoding

2013-04-24 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] radius keep alive (Accounting Interim Update)

2013-04-21 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] acc module flags.

2013-04-20 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips and Asterisk

2013-03-29 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] open sips 1.9 issue

2013-03-18 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] open sips 1.9 issue

2013-03-18 Thread qasimak...@gmail.com
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

[OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

2013-02-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

2013-02-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

2013-02-13 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Bulding Own Distro

2012-12-31 Thread qasimak...@gmail.com
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 ?

Re: [OpenSIPS-Users] Help to Understand Loop

2012-11-06 Thread qasimak...@gmail.com
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:

Re: [OpenSIPS-Users] Help: Understanding ACK loop

2012-11-06 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Recover running configuration of OpenSIPs

2012-11-06 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

2012-11-01 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Transfering a call by opensips

2012-10-18 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] attack from friendly-scanner

2012-10-09 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

2012-08-31 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

2012-08-30 Thread qasimak...@gmail.com
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

Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

2012-08-29 Thread qasimak...@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

Re: [OpenSIPS-Users] OpenSIPS 1.7 + NAT + rtpproxy

2012-08-03 Thread qasimak...@gmail.com
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 +

Re: [OpenSIPS-Users] ACC module with AAA dont send failed transaction

2012-08-01 Thread qasimak...@gmail.com
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