Re: [OpenSIPS-Users] route handling of http protocol in opensips

Hi Anmol, May I ask why do you want to handle HTTP custom traffic in OpenSIPS, which is a SIP server? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 01.05.2024 07:12, ANMOL PRAKASH via Users wrote: Hi all,

[OpenSIPS-Users] route handling of http protocol in opensips

Hi all, Is there any module in opensips inline with xHTTP in kamailio. In kamailio, xHTTP module offers a generic way of handling the HTTP protocol, by calling event_route[xhttp:request] in your config. In opensips, httpd module is there to enable http server on opensips but I am looking for

Re: [OpenSIPS-Users] route parameters ($param) on opens ips 3.2.11

On 09.03.2023 11:32, Alain Bieuzent wrote: When, I my script I call the route : route(Q850enable,38); Opensips generate this error and sometime segfault . Hi Alain, Indeed, this is a known issue with 3.2.11, please see #3024 [1] on GitHub.  A fix should be available soon, I'll keep you

[OpenSIPS-Users] route parameters ($param) on opens ips 3.2.11

Hi all, I’m trying to upgrade from 3.2.9 to 3.2.11 and I’m facing issue with this new version. I have a route where i used a parameter as input : route[Q850enable] {     $var(Q850enableset) = $dlg_val(Q850enable); if ($var(Q850enableset) == "1" || $avp(Q850enable)

Re: [OpenSIPS-Users] Route Header removal

: [OpenSIPS-Users] Route Header removal WARNING: External Email: Exercise Caution Hey Louis, I thought I might jump in here about your point (2). If you are referring to preloaded Route header(s) in the initial request, then you just need to take care when calling the loose_route [1] method

Re: [OpenSIPS-Users] Route Header removal

From: Users On Behalf Of Razvan Crainea Sent: August 25, 2021 6:04 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Route Header removal WARNING: External Email: Exercise Caution Hi, Louis! Almost all changes (perhaps the only exception is the R-URI and parts of the Contact head

Re: [OpenSIPS-Users] Route Header removal

er. Louis -Original Message- From: Users On Behalf Of Razvan Crainea Sent: August 25, 2021 6:04 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Route Header removal WARNING: External Email: Exercise Caution Hi, Louis! Almost all changes (perhaps the only exception is the

Re: [OpenSIPS-Users] Route Header removal

Hi, Louis! Almost all changes (perhaps the only exception is the R-URI and parts of the Contact header) performed on the SIP message will only be visible in the outgoing message, but *NOT* reflected in the script. This means that whatever change you perform to the message will not be visible

[OpenSIPS-Users] Route Header removal

We are trying to remove the Route header from an incoming Invite. Even with remove_hf, Opensips still routes towards the content of the Route header. Opensips.cfg: -- route { if ( is_method("INVITE") ) { . . . xlog("removing Route Header: $(hdr(Route)[0])\n");

Re: [OpenSIPS-Users] Route per user agent instead of prefix

Hi, Put an extra fields in subscriber table in order to assign to each user an routing group - and after auth (see load_credentialas in auth_db module for loading extra columns from subscriber table), use the routing group value in order to trigger the corresponding routing logic. Regards,

[OpenSIPS-Users] Route per user agent instead of prefix

Dears at opensips users list, I need to route per authenticated user instead of prefix based routing What is the best way to do this ? Br, AShabana ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] Route header in CANCEL requests

Hi, i have opensips behind another SIP proxy. When I need to t_relay an INVITE from an uac to the proxy, my opensips.cfg uses append_hf to add a Route header with a username required by the proxy (some sort of validation). This works fine. However, if the uac cancels the INVITE, the

Re: [OpenSIPS-Users] Route header in CANCEL requests

Hello, i have found the solution in the docs: append_hf("Route: \r\n"); t_newtran(); t_add_hdrs("Route: \r\n"); Regards, Takeshi On Sat, Oct 10, 2015 at 9:03 AM, mayamatakeshi wrote:

Re: [OpenSIPS-Users] Route header in CANCEL requests

For completeness, it will be also necessary to call t_reply to send '100 Trying' as we are using t_newtran(): append_hf("Route: \r\n"); t_newtran(); t_reply("100", "Trying"); t_add_hdrs("Route:

Re: [OpenSIPS-Users] Route RTP packets through

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'

[OpenSIPS-Users] Route RTP packets through

Hello , I am using opensips and rtpproxy for sip and rtp proxy server. Here problem is rtpproxy is sending rtp packets on private ip address and so packet lost. How to use nat so 66.33.66.256 can transfer rtp packets on nated ip instead of private ip network that available in SDP. *Current

Re: [OpenSIPS-Users] Route RTP packets through

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

Re: [OpenSIPS-Users] Route RTP packets through

Hello Razvan Yes i am using r as rtpproxy_offer(r,xx.xx.xx.xx); -- Kind regards, Kaushik Parmar ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Route RTP packets through

Kaushik, 'r' means to trust the IP in SDP, so if it is private IP it will take private IP for media. Try changing connection string with 'c' flag also owner 'o' as per your requirement based on source public IP. http://www.opensips.org/html/docs/modules/devel/rtpproxy.html *r* - flags that IP

Re: [OpenSIPS-Users] Route to media-server, but reply negative

Hi, you can use SEMS with early announce module for this. http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_early_announce.html It does exactly what you are searching for. Best Regards Marc Langsdorf -- View this message in context:

Re: [OpenSIPS-Users] Route to media-server, but reply negative

** ** Max M. ** ** *Von:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *Im Auftrag von *Remco . *Gesendet:* Samstag, 13. Oktober 2012 22:52 *An:* OpenSIPS users mailling list *Betreff:* [OpenSIPS-Users] Route to media-server, but reply negative

Re: [OpenSIPS-Users] Route to media-server, but reply negative

...@lists.opensips.org] *Im Auftrag von *Remco . *Gesendet:* Samstag, 13. Oktober 2012 22:52 *An:* OpenSIPS users mailling list *Betreff:* [OpenSIPS-Users] Route to media-server, but reply negative ** ** Hi all, I have the following in the failure_route, for invalid destinations

Re: [OpenSIPS-Users] Route to media-server, but reply negative

, 13. Oktober 2012 22:52 *An:* OpenSIPS users mailling list *Betreff:* [OpenSIPS-Users] Route to media-server, but reply negative ** ** Hi all, I have the following in the failure_route, for invalid destinations: if(t_check_status(404

Re: [OpenSIPS-Users] Route to media-server, but reply negative

) the call. ** ** Best Regards ** ** Max M. ** ** *Von:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *Im Auftrag von *Remco . *Gesendet:* Samstag, 13. Oktober 2012 22:52 *An:* OpenSIPS users mailling list *Betreff:* [OpenSIPS-Users] Route to media

[OpenSIPS-Users] Route to media-server, but reply negative

Hi all, I have the following in the failure_route, for invalid destinations: if(t_check_status(404)) { # Dialed phone number does not exist # Cancel call billing resetflag(1); #

Re: [OpenSIPS-Users] Route to media-server, but reply negative

Auftrag von Remco . Gesendet: Samstag, 13. Oktober 2012 22:52 An: OpenSIPS users mailling list Betreff: [OpenSIPS-Users] Route to media-server, but reply negative Hi all, I have the following in the failure_route, for invalid destinations: if(t_check_status(404

[OpenSIPS-Users] route all parking related calls to one FS server

Hi, I have opensips running nicely, followed this tutorial: http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs Problems happen when using call parking. Because a call can be parked on any FS server, you can't know where to pick up the parked call. A simple

Re: [OpenSIPS-Users] Route group inside of drouting

My understanding of the sort_order parameter is that it will apply to sort to the entire group_id. So if my LCR should be CarrierA, CarrierB, CarrierC... and i create gateway_ids (3 thru 9 for CarrierA), 10 for CarrierB, and 1 for CarrierC.. I can have gw_list as :3;9,10,1 but it will

Re: [OpenSIPS-Users] Route group inside of drouting

Hi Kent, The DR module operates with GW/destinations, not directly with carries (how you define, understand and use the term of carrier is your own business). so, if you have the case you described (A with 7 Gw[1,2,3,4,5,6,7], B with one [8] and C with one [9]) and you want for A to have

Re: [OpenSIPS-Users] Route group inside of drouting

Thank you so much! I understand now. On Fri, Jun 24, 2011 at 9:33 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hi Kent, The DR module operates with GW/destinations, not directly with carries (how you define, understand and use the term of carrier is your own business). so, if

[OpenSIPS-Users] Route group inside of drouting

Okay, so using the drouting module I can handle a basic LCR by organizing the gwlist for each prefix. Say i have something like this: 212555, gwlist = 3,5,1 now.. lets say gw 3 actually needs to try multiple ips for that carrier before going on to gw 5, is this possible while using drouting or

Re: [OpenSIPS-Users] Route group inside of drouting

On 13.06.2011 16:40, Kent Pirlo wrote: 212555, gwlist = 3,5,1 now.. lets say gw 3 actually needs to try multiple ips for that carrier before going on to gw 5, is this possible while using drouting or do i need to scrap the drouting module to do something complex like this.. It is possible

Re: [OpenSIPS-Users] route based on previous load_balance dialog

Hi Stefano, A better approach (without the need of DB) is to use the get_dialog_info() function. See: http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id293888 The idea is that after the LB function, add the selected destination (stored in $dd variable) and the agent name

Re: [OpenSIPS-Users] route based on previous load_balance dialog

2011/2/27 Bogdan-Andrei Iancu bog...@opensips.org: And when you have a 123 call for agent, search the to get the destination of the ongoing call to that agent:   $var(x) = agent ??   get_dialog_info(dst,$var(dst),agent,$var(x)); works well, thanks! (i improved with if(get_dialog_info(...)))

Re: [OpenSIPS-Users] route based on previous load_balance dialog

Hi Dave, You might look at the dialog module. Check if what you need is stored in the db for a dialog. You would need to have the db mode set to realtime. As I said, dialog module stores what I need. My problem is how to fetch that information to reach the right asterisk server. Either way

[OpenSIPS-Users] route based on previous load_balance dialog

Hello folks, I have this load_balance situation: opensips in front of 3 asterisk servers in a outbound only call-center. When a agent places a call, opensips load balance through the three asterisks. Now, we need to use the ExtensSpy command from asterisk, dialing something like ***123 (where

Re: [OpenSIPS-Users] Route timeout

@lists.opensips.org mailto:users@lists.opensips.org *Sent:* Friday, August 13, 2010 1:17:43 *Subject:* [OpenSIPS-Users] Route timeout Hello All, Is there any way to configure a timeout per route? What I mean is: If my first route doesn't send a reply(100 Trying or any other

Re: [OpenSIPS-Users] Route timeout

@lists.opensips.org *Sent:* Friday, August 13, 2010 1:17:43 *Subject:* [OpenSIPS-Users] Route timeout Hello All, Is there any way to configure a timeout per route? What I mean is: If my first route doesn't send a reply(100 Trying or any other) in some seconds, the opensips

Re: [OpenSIPS-Users] Route timeout

On Fri, Aug 27, 2010 at 8:33 AM, Wesley Volcov wesleyvol...@gmail.comwrote: Follow my tests: The INVITE was sent by opensiups to route at 09:03:42 The 100 TRY came from route to opensips at 09:03:42 The 183 came from route to opensips at 09:03:43 But, at 09:03:47 the was sent to Failure

Re: [OpenSIPS-Users] Route timeout

to trigger next failure route. failure_route[1] { #alternate route } -- *From:* Wesley Volcov wesleyvol...@gmail.com *To:* OpenSIPS users mailling list users@lists.opensips.org *Sent:* Friday, August 13, 2010 1:17:43 *Subject:* [OpenSIPS-Users] Route timeout

Re: [OpenSIPS-Users] Route timeout

Wesley, Yes, it's possible to control the timeout per route, this could be made by the fr_timer_avp and fr_inv_timer_avp, you need just set a valeu in seconds in those avps to control the timeout per branch. Have a look in the documentation of TM module [1]. [1] -

Re: [OpenSIPS-Users] Route timeout

Hey Antonio, It seems to work great in my test environment! Thank you very much! Cheers On 12 August 2010 17:09, Antonio Anderson Souza anto...@voicetechnology.com.br wrote: Wesley, Yes, it's possible to control the timeout per route, this could be made by the fr_timer_avp and

Re: [OpenSIPS-Users] Route timeout

this in your route to trigger next failure route. failure_route[1] { #alternate route } From: Wesley Volcov wesleyvol...@gmail.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Friday, August 13, 2010 1:17:43 Subject: [OpenSIPS-Users] Route

Re: [OpenSIPS-Users] Route calls based on CPS rate

Thanks to all. I have doubt again, what is the use of timer_interval and how we can utilize it with the CPS limit for pipes. Regards Rajib On Mon, Feb 15, 2010 at 10:45 PM, Ovidiu Sas o...@voipembedded.com wrote: Yes, you can use the ratelimit module to control the cps. You will need to

Re: [OpenSIPS-Users] Route calls based on CPS rate

http://www.opensips.org/html/docs/modules/1.6.x/ratelimit.html#id228149 When you specify a limit in the definition of a pipe, that limit will be divided by the timer interval in order to get the cps. If you want accurate cps control, then you need to set the timer_interval to a low value. For

Re: [OpenSIPS-Users] Route calls based on CPS rate

Hi Rajib, LB module is doing routing based on the load as current ongoing calls per destination. To compute the CPS for a destination can be a bit tricky - there is no module for doing it, Probably you can try to count the call using some shared mem variable (directly in script) and to try

Re: [OpenSIPS-Users] Route calls based on CPS rate

Hi Bogdan, I agree with you. But I have seen that RATELIMIT module is doing something like that. Can we use that module for each gateway by identifying the gateway at run-time, like after LB selected the destination, we can have something like if($du == sip:some gateway) { if

Re: [OpenSIPS-Users] Route calls based on CPS rate

Yes, you can use the ratelimit module to control the cps. You will need to assign a pipe for each outbound destination. Regards, Ovidiu Sas On Mon, Feb 15, 2010 at 11:50 AM, rajib deka raji...@gmail.com wrote: Hi Bogdan, I agree with you. But I have seen that RATELIMIT module is doing

[OpenSIPS-Users] Route calls based on CPS rate

Hello all, Is it possible to route calls based on cps rate using OpenSIPS load balancer module. We have an enterprise implementation here using OpsnSIPS load_balancer, which is handling 100 cps using our different trunks. So we want to place calls according to trunks cps capacity. Is there any

Re: [OpenSIPS-Users] Route selection based on SDP codecs

: [OpenSIPS-Users] Route selection based on SDP codecs John, That's a great idea... Regarding (a), have you looked the codec_* functions available in 1.6's textops? The codec_exists() function will return back true if a particular codec is available in the existing SDP. Perhaps this functionality

[OpenSIPS-Users] Route selection based on SDP codecs

If a call is has G.729 set as the preferred codec in the SDP, I want to send it to an Asterisk server with transcoding capabilities. If the preferred codec is G.711 I will send it directly to the media server, gateway or other destination. I can use TEXTOPS to search for entries like this in

Re: [OpenSIPS-Users] Route selection based on SDP codecs

John, That's a great idea... Regarding (a), have you looked the codec_* functions available in 1.6's textops? The codec_exists() function will return back true if a particular codec is available in the existing SDP. Perhaps this functionality could be extended to return an AVP to give the

Re: [OpenSIPS-Users] Route selection based on SDP codecs

John Quick wrote: (a) Is there a better way than using regular expression searches of the message body? (b) How certain is it that G.729 will always use 18 to identify it in the SDP? http://www.opensips.org/html/docs/modules/devel/textops.html#id271999 Cheers, Thomas -- mail:

[OpenSIPS-Users] Route function errors

Hi all, I upgraded the opensips-1.5 to the opensips in the trunk. when i run the script after the upgrade i get the following error at route functions Sep 2 05:18:13 [13423] CRITICAL:core:yyerror: parse error in config file, line 214, column 10-11: syntax error Sep 2 05:18:13 [13423]

Re: [OpenSIPS-Users] Route function errors

Hi Ashwini, It might be because of the work I did yesterday to add script routes with names (and not numbers). So, are you sure you updated/downloaded the entire source tree? Also , please post the lines 214 and 300 to see what may be the problem. Regards, Bogdan ASHWINI NAIDU wrote: Hi

Re: [OpenSIPS-Users] Route function errors

Hi Bogdan, Ya i am sure that today evening itself i downloaded and updated the whole trunk Line No 214: route(1); Line No 300: route(1); On Tue, Sep 1, 2009 at 7:09 PM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Hi Ashwini, It might be because of the work I did yesterday to add

Re: [OpenSIPS-Users] Route function errors

Hi Ashwini, Should be fixed now. Please update from SVN. Thanks and Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, Ya i am sure that today evening itself i downloaded and updated the whole trunk Line No 214: route(1); Line No 300: route(1); On Tue, Sep 1, 2009 at 7:09 PM,

Re: [OpenSIPS-Users] Route function errors

Hi Bogdan, Can you tell me if any changes have been made for *t_check_trans* function and *sl_send_reply* functions. Should i pass any arguments. On Tue, Sep 1, 2009 at 9:28 PM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Hi Ashwini, Should be fixed now. Please update from SVN.

Re: [OpenSIPS-Users] Route function errors

No, there are no changes on that functions - use them as so far (check the online docs for params) Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, Can you tell me if any changes have been made for *t_check_trans* function and *sl_send_reply* functions. Should i pass any arguments.

Re: [OpenSIPS-Users] Route function errors

Hi Bogdan, I use to call them as t_check_trans() but i see parse errors over there also also for sl_send_reply. that is y i asked. On Tue, Sep 1, 2009 at 10:42 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: No, there are no changes on that functions - use them as so far (check

Re: [OpenSIPS-Users] Route function errors

Could you past the script part and the error logs... ? Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, I use to call them as t_check_trans() but i see parse errors over there also also for sl_send_reply. that is y i asked. On Tue, Sep 1, 2009 at 10:42 PM, Bogdan-Andrei Iancu

Re: [OpenSIPS-Users] route

El Jueves, 27 de Noviembre de 2008, Raghavendra D P escribió: I am using opensips1.4 I am trying B2BUA scenario with IMS, how to route back to the caller the same message by removing its own route address and changing Call-id Is this question really related to OpenSIPS (proxy SIP)? Of course

Re: [OpenSIPS-Users] route

textops module. Raghavendra D P wrote: Hi Route :sip:190.10.19.20, sip:45:128 I am using oopensips 1.4 How to remove fist route information *Thanks and Regards* *Raghavendra DP**|** Tech Mahindra* 9/7, Hosur Road, Bangalore – 560 029, India ( Office:

[OpenSIPS-Users] route

I am using opensips1.4 I am trying B2BUA scenario with IMS, how to route back to the caller the same message by removing its own route address and changing Call-id Thanks and Regards Raghavendra DP| Tech Mahindra 9/7, Hosur Road, Bangalore - 560 029, India * Office: +91 80 4024 3458 |