Re: [OpenSIPS-Users] EXEC not working

ath the "ls" command... > > $var(out)=""; > $var(err)=""; > $avp(env)="a"; > > exec("/usr/bin/ls -l", , $var(out), $var(err), $avp(env)); > xlog("Out: $var(out)"); > xlog("Err: $var(err)&

Re: [OpenSIPS-Users] EXEC not working

me in OpenSIPS 3.1 with command... > > exec("/root/scripts/script.sh $si", , $var(out), $var(err)); > > Is it the quotes around $var(err) that is causing the null output?? > > > > On Thu, 4 Feb 2021 at 10:37, Dragomir Haralambiev > wrote: > >> Hello

Re: [OpenSIPS-Users] EXEC not working

issions or similar - have you tested the example from the docs? > > exec("ls -l", , "$var(out)", "$var(err)", "$avp(env)"); > xlog("The output is $var(out)\n"); > xlog("Received the following error\n$var(err)"); > > > On Thu

[OpenSIPS-Users] EXEC not working

Hello, I try to run external program like this exec("/test/opensips.pl",,,"$var(err)"); xlog("Received the following error $var(err)"); Here is log record: Received the following error The program is not running. Where is the reason? Best regards: Dragomir

Re: [OpenSIPS-Users] install RTPEngine on Centos 8

to install this package? На чт, 4.06.2020 г. в 15:35 Denys Pozniak написа: > Hello! > Try to use my build (not tested, as is): > https://copr.fedorainfracloud.org/coprs/denysp/rtpengine-centos8/ > > > чт, 4 июн. 2020 г. в 12:20, Dragomir Haralambiev : > >> Hello, >>

[OpenSIPS-Users] install RTPEngine on Centos 8

Hello, I need to install RTPEngine on Centos 8? Any help Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] WRONG relay in Sequential requests

Hello, I have problem with setup incoming call with TLS support: UA with TLS supprt is registered. Here is information from AOR AOR:: 33341656 Contact:: sip: 33341656 @192.168.1.106:50443;transport=TLS;ob Q= Expires:: 260

[OpenSIPS-Users] Problem with registrar mcontact_avp

Hello, I try to store the real IP in the location table. I make the following: a. specify a mcontact_avp[1] in my script: modparam("registrar", "mcontact_avp", "$avp(register)") b. after fix_nated_contact() I store the changed contact in that AVP: fix_nated_contact(); $avp(register) =

Re: [OpenSIPS-Users] TLS - nated contact is not fixed

if (proto==TLS) { force_tcp_alias(); } > > > > На чт, 5.09.2019 г. в 10:37 ч. Dragomir Haralambiev > написа: > >> Hello, >> >> I have : >> if (is_method("REGISTER|SUBSCRIBE")) { >> fix_nated_register(); >> } >> fix_nated_conta

[OpenSIPS-Users] TLS - nated contact is not fixed

Hello, I have : if (is_method("REGISTER|SUBSCRIBE")) { fix_nated_register(); } fix_nated_contact(); When I have Transport UDP all is OK. Wen I have Transport TLS, contact record in location table is not fixed. What I do to fix this? Regards, Dragomir

Re: [OpenSIPS-Users] Double SDP

Thanks! Now all is OK! На ср, 4.09.2019 г. в 15:34 ч. Ben Newlin написа: > If you don't want to have both in the second INVITE, you can try putting > both rtpengine_offer calls in branch routes instead. I haven't worked with > rtpengine, but with other messages changes like this if you place

Re: [OpenSIPS-Users] Double SDP

Hello, Thanks for your replay. If that's OK. Why do I have one-way voice? Rerards, На ср, 4.09.2019 г. в 15:29 ч. Alexey Vasilyev написа: > This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This > is > Invite from snom phone, for example: > > Sent to tls:135.42.212.82:5061

[OpenSIPS-Users] Double SDP

Hello, I try setup Opensips to make local calls using device with encryption. In this example, it is not known UA 2 whether it supported (RTP/SAVP). UA 1 (RTP/AVP) > Opensips+Rtpengine -> UA 2 1. I try to make call with: rtpengine_offer("RTP/AVP replace-session-connection replace-origin

[OpenSIPS-Users] need help -

Hello, What I do to check flag after has_totag() ? Here is example: route { if (has_totag()) { . if ( isflagset(BLA_BLA) ) { xlog("L_ERR", "Flag is active \n"); }; } if ( is_method("INVITE") ) {

Re: [OpenSIPS-Users] SRTP to RTP

r SDPs, verify all going to srtp is indeed SRTP SDP. And > all going to UAC is not SRTP > > On Thu, 1 Aug 2019 at 11:59, Dragomir Haralambiev > wrote: > >> Hi, >> >> 1. tpengine_offer >> >> UAC1 SRTP ---INVITE > Opensips+rtpengine >> $var(

Re: [OpenSIPS-Users] SRTP to RTP

> > You need to do this for every leg of the call. This means: > > Call from SRTP client TO non-SRTP: > Remove the ICE, etc. > > When the REPLY with the 200 SDP comes back FROM the non-SRTP, you need to > ADD ICE, etc. > > Hope that makes sense > > David > > O

Re: [OpenSIPS-Users] SRTP to RTP

re Engineer* > *3CLogic , ph:07827611765* > > > On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq wrote: > >> Use rtp/savp >> >> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, >> wrote: >> >>> Hi, >>> >>> Thanks for your rep

Re: [OpenSIPS-Users] SRTP to RTP

gt; $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force"; > rtpengine_answer("$var(rtpengine_flags)"); > > You can modify this according to your requirement . > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing

[OpenSIPS-Users] SRTP to RTP

Hello, I have 2 applications connected to Opensips+rtpengine: UAC1 -use encryption always. SRTP (RTP/SAVP) UAC2 - never use encryption . RTP (RTP/AVP) How to setup Opensips to make follow call: UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP Thanks, Dragomir

[OpenSIPS-Users] TLS to UDP

Hello, How to setup Opensps and rtpengine to translate TLS connection to UDP. Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] rtpengine_offer error

Hi Alexej, Yes. You are right if I know if the client has encryption (RTP / SAVP) or encryption (RTP / AVP) turned off. Example: MicroSIP1 --> Opensips -->MicroSIP2 *Media encryption* disableRtpEngine *Media encryption*

Re: [OpenSIPS-Users] rtpengine_offer error

When use comma Opensips get second one. На нд, 30.06.2019 г. в 10:53 ч. Olle Frimanson написа: > Try replace comma with space > rtp/avp rtp/save > I’m not sure of you can offer both at the same time > > Br Olle > > Skickat från min iPhone > > > 30 juni 2019 kl. 09

[OpenSIPS-Users] rtpengine_offer error

Hello, >From rtpengine_offer manual I see: RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF - these serve as an alternative, more explicit way to select between the different RTP protocols and profiles supported by the RTP proxy. For example, giving the flag “RTP/SAVPF” has the same effect as giving the

Re: [OpenSIPS-Users] How to change media protocol?

>> } >> >> rtpengine_offer("$var(rtpengine_flags)"); >> >> На пт, 28.06.2019 г. в 16:38 ч. Johan De Clercq >> написа: >> >>> If goes to provider: rtp/avp >>> Else rtp/savp >>> >>> So put y

Re: [OpenSIPS-Users] How to change media protocol?

ar(rtpengine_flags) = "RTP/AVP/SAVP replace-session-connection replace-origin ICE=remove"; } rtpengine_offer("$var(rtpengine_flags)"); На пт, 28.06.2019 г. в 16:38 ч. Johan De Clercq написа: > If goes to provider: rtp/avp > Else rtp/savp > > So put your pa

Re: [OpenSIPS-Users] How to change media protocol?

the params that you give to rtpengine_offer or answer with RTP/AVP > or RTP/SAVP accordingly. > > > > *From:* Users *On Behalf Of *Dragomir > Haralambiev > *Sent:* Thursday, June 27, 2019 10:36 PM > *To:* OpenSIPS users mailling list > *Subject:* [OpenSIPS-Users] How

[OpenSIPS-Users] How to change media protocol?

Hello, I have OpenSips + rtpengine. Opensips receive INVITE with Media Protocol: RTP/SAVP How to setup Opensips to forward INVITE to ITSP with Media Protocol: RTP/AVP. Like this scheme: INVITE with Media Protocol: RTP/SAVP --> (OpenSips +rtpengine) >INVITE with Media Protocol: RTP/AVP --->

Re: [OpenSIPS-Users] more INVITEs

t; > *From:* Users *On Behalf Of *Dragomir > Haralambiev > *Sent:* Tuesday, January 29, 2019 6:40 PM > *To:* OpenSIPS users mailling list > *Subject:* [OpenSIPS-Users] more INVITEs > > > > Hello Opensips comunity, > > > > I must to solve the following cas

[OpenSIPS-Users] more INVITEs

Hello Opensips comunity, I must to solve the following case: One SIP device send to OpenSips not one, but three identical INVITEs (with same Call-ID). How do I set the OpenSips to only handle the first one and ignore others? Best regards, Dragomir ___

[OpenSIPS-Users] system metrics

Hello, For system metrics i need to get number active dialogs. How to do that easy. Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] rtpengine

Thanks a lot!!! На вт, 2.10.2018 г. в 10:11 ч. Răzvan Crainea написа: > Yes. You should also use the -x 1 parameter, to trigger the command only > using the callid. > > Best regards, > Răzvan > > On 10/1/18 7:22 PM, Dragomir Haralambiev wrote: > > Hello, &g

[OpenSIPS-Users] rtpengine

Hello, If I use -b parameter of rtpengine it should have active MI_XMLRPC_NG module in Opensips? Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] Async Radius support in RPM

Hello Team! In opensips 2.2.x I had to recompile opensips manually in order to apply the asynchronous radius request support. I am now planing on migrating the new 2.4 LTS. Do I still need to patch the code or I can simply use the pre-compiled binaries available trough the opensips repos? Best

Re: [OpenSIPS-Users] close Dialog setings

e (such as 7200 seconds). This way, if the ACK is not received, > the dialog will timeout after 5 seconds. > > Best regards, > Răzvan > > > On 06/02/2018 01:25 PM, Dragomir Haralambiev wrote: > >> Hello, >> >> I need help for the following situation. >>

[OpenSIPS-Users] Opensips Crashed

Hello, I use stable LTS: version: opensips 2.2.6 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll,

Re: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay

Hi, Opensips is signaling proxy. You need media proxy to make this. Regards, Dragomir 2018-02-08 17:10 GMT+02:00 Brian Southworth : > Hi All, > > > > How would I go about getting a opensips as a proxy to recognise the DTMF > tones or forward them onto asterisk ? > >

[OpenSIPS-Users] (no subject)

Hello, In my log I see : ERROR: non-ASCII escaped character in '%09004915213874522' @ 0 What I can do to remove all not digit simvols from dialing number? Thanks, Dragomir ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] LoadBalancer destination managment

Hello, I'm using Opensips 2.2 LTS latest version from git. I am trying to understand how exactly the load balancer module senses when a destination is out of order. As per documentation if the destination 'probe_mode' is set to 2 to always probe the destination on the defined interval and

Re: [OpenSIPS-Users] rtpp_notify_socke in rtpengine

nSIPS Developerwww.opensips-solutions.com > > On 12/14/2017 11:52 AM, Dragomir Haralambiev wrote: > > Hello, > > To using WebRTC I must to change rtpproxy with rtpengine module. > In rtpproxy have rtpp_notify_socket to interupt call when lose RTP. > > When setup rtpengine

[OpenSIPS-Users] rtpp_notify_socke in rtpengine

Hello, To using WebRTC I must to change rtpproxy with rtpengine module. In rtpproxy have rtpp_notify_socket to interupt call when lose RTP. When setup rtpengine I not see rtpp_notify_socket. Hot to set Opensips to work with rtpengine module when lose rtp sream to close the call? Best regards,

Re: [OpenSIPS-Users] get expires from REGISTER

quot;". Try > $(hdr(Contact){uri.param,expires}). > > Regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 27.11.2017 13:46, Dragomir Haralambiev wrote: > > Hello, > > I try to get expires from REGISTER. > xlog("Expires=$hdr

[OpenSIPS-Users] get expires from REGISTER

Hello, I try to get expires from REGISTER. xlog("Expires=$hdr(expires)\n"); I receive this: Expires= Here is REGISTER Session Initiation Protocol (SIP as raw text) REGISTER sip:sip-server.com:5060 SIP/2.0\r\n Accept: application/reginfo+xml, application/sdp,

Re: [OpenSIPS-Users] call_center module in Opensips 2.3

Hi, I try your idea to loop the INVITE back to Opensips and do a lookup(). I pit into "location" field from cc_agents table this one "sip:localnumber@Opensips_IP:Opensips_port". Call is connected but I receive follow log message: ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid=

[OpenSIPS-Users] outbound call center services

Hello all, The Call Center module is develop for inbound call center service. Does anyone have any idea how to develop (auto dialer) for outbound call center services with call distribution to agents. Best regards, Dragomir ___ Users mailing list

Re: [OpenSIPS-Users] Wrong Contact in location table

d_avp", "$avp(received)") 2017-11-07 16:54 GMT+02:00 Răzvan Crainea <raz...@opensips.org>: > Hi, Dragomir! > > So you did not have the parameter provisioned at all? Or what was the > initial issue? > > Best regards, > > Răzvan Crainea > OpenSIPS Develo

Re: [OpenSIPS-Users] Wrong Contact in location table

something that I noticed from the first email you have sent - the >> fix_nated_register() function is not called, or does not work properly. Can >> you also print the avp you are setting in the received_avp[1]. Also, call >> script trace for the reply too. >> >> [1] http://

Re: [OpenSIPS-Users] Opensips 2.3.2 chashed

e the following > snippet and attach the output in pastebin. > > > frame 3 > > p *call_it > > Thanks! > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/04/2017 02:35 PM, Dragomir Haralambiev wrote: > > Hello, > > I use latest updat

Re: [OpenSIPS-Users] Opensips 2.3.2 chashed

n Crainea <raz...@opensips.org>: > Hi, Dragomir! > > Can you to go in gdb top frame 3 and print call_it. Use the following > snippet and attach the output in pastebin. > > > frame 3 > > p *call_it > > Thanks! > > Răzvan Crainea > OpenSIPS Developerwww

Re: [OpenSIPS-Users] Wrong Contact in location table

derstand why you are trying to change >> it - if you need the real IP and port, you can take them from the received >> field. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/02/2017 11:43 PM, Dragomir Ha

[OpenSIPS-Users] Opensips 2.3.2 chashed

Hello, I use latest update from git at Opensips 2.3.2. When try to use call center module Opensips crashed: 1. Log message: Nov 4 14:09:52opensips[2658]: CRITICAL:call_center:cc_queue_rmv_call: QUEUE - call not in queue l=(nil), h=(nil) Nov 4 14:09:52opensips[2666]: CRITICAL:core:receive_fd:

Re: [OpenSIPS-Users] Wrong Contact in location table

pensips-solutions.com > > On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: > > Hi, > > Here is part of my script: > > modparam("registrar", "mcontact_avp", "$avp(register)") > . > > if (t_check_status("2[0-9][0-9]"))

Re: [OpenSIPS-Users] Wrong Contact in location table

you to? > Can you print the value of the register just before save()? Also, can you > send over some logs for the reply? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/02/2017 12:26 AM, Dragomir Haralambiev wrote: > >

Re: [OpenSIPS-Users] Wrong Contact in location table

s public IP and uses that in Via: and Contact: > headers unless you set ‘Firewall traversal method’ to ‘None’ in the account > settings. > > > > > On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev <goup2...@gmail.com> > wrote: > > When REGISTER received from X-li

Re: [OpenSIPS-Users] Wrong Contact in location table

eloperwww.opensips-solutions.com > > On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: > > Hi, > > Thanks for your replay. > > I execute fix_nated_contact() after fix_nated_register(). > > Here part of my script: > if (isflagset(NAT)) { > i

Re: [OpenSIPS-Users] Wrong Contact in location table

inea > OpenSIPS Developerwww.opensips-solutions.com > > On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: > > Hi, > > Thanks for your replay. > > I execute fix_nated_contact() after fix_nated_register(). > > Here part of my script: > if (isflagse

Re: [OpenSIPS-Users] Wrong Contact in location table

. > Please print the Contact header as I instructed earlier and send back the > script trace. Also, please post the logs on pastebin.com or something, > don't do it directly in the email. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com &g

Re: [OpenSIPS-Users] how to handle nat with mid registrar

Hi Bogdan, *fix_nated_regis**trar not working. * *please see my latest emails. * *Best regards, * *Dragomir* On Oct 31, 2017 13:42, "Bogdan-Andrei Iancu" wrote: Sorry, there is a type in the prev email, it should be "So, have you actually tried the

Re: [OpenSIPS-Users] Wrong Contact in location table

What will I do to solve this problem? 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev <goup2...@gmail.com>: > Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: 57995010: REGISTER > before fix_nated_register <sip:57995010@192.168.2.196:5060>[1] > Oct 27 11:47:32 web /us

Re: [OpenSIPS-Users] Wrong Contact in location table

ERED (200 OK) > Oct 26 22:14:30 /opensips[17718]: 57995010: REGISTERED. Save in location. > > > 2017-10-26 10:36 GMT+03:00 Răzvan Crainea <raz...@opensips.org>: > >> Hi, Dragomir! >> >> Again, I really doubt fix_nated_register() is executed , since I am not >

Re: [OpenSIPS-Users] Wrong Contact in location table

g/2017/09/20/troubleshooting-opensips-script/ > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 10/24/2017 12:41 AM, Dragomir Haralambiev wrote: > > Hi, > > I made setup like this help: > > https://blog.opensips.org/2016

Re: [OpenSIPS-Users] Wrong Contact in location table

> OpenSIPS Developerwww.opensips-solutions.com > > On 10/21/2017 01:40 AM, Dragomir Haralambiev wrote: > > Hello, > > > I have follow line in my script: > if (is_method("REGISTER")) { > fix_nated_register(); > fix_nated_contact(); > > save("

[OpenSIPS-Users] Wrong Contact in location table

Hello, I have follow line in my script: if (is_method("REGISTER")) { fix_nated_register(); fix_nated_contact(); save("location","c1", "$fu"); } Opensips receive REGISTER with "Contact: " I try to see location table opensipsctl ul show AOR::

[OpenSIPS-Users] Problem with t_relay()

Hello, I see problem when Opensips relay BYE. Opensips 2.3 is latest from GIT. Opensips listen in diffrent port from 5060: loadmodule "proto_udp.so" modparam("proto_udp", "udp_port", 2010) Opensips relay BYE if ( t_relay() ) { xlog("L_ERR", "Send relay ($rm) to $rd:$rp "); }; In log all is

[OpenSIPS-Users] Problem with 200 OK after BYE

Hello, I use Opensips 2.3 in follow scheme: SBC ---Opensips --Zoiper When Opensips Send BYE to SBC, SBC not return 200 OK. Zoiper send BYE to Opensips. BYE sip:Opensips_IP:Opensips_port;did=363.27c9faa4 SIP/2.0 Via: SIP/2.0/UDP

Re: [OpenSIPS-Users] RTPEngine and monitoring MOS

, and then parse them manually and fetch the > interesting information. > > [1] http://www.opensips.org/html/docs/modules/2.3.x/rtpengine. > html#idp6085728 > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 10/03/2017 12:56 AM

[OpenSIPS-Users] RTPEngine and monitoring MOS

Hello, When read RTPEngine presentation ( https://www.slideshare.net/AndreasGranig/sipwise-rtpengine) I see follow information for Kamalio modparam(“rtpengine”, “mos_min_pv”, “$avp(mos_min)“) modparam(“rtpengine”, “mos_max_pv”, “$avp(mos_max)“) modparam(“rtpengine”, “mos_average_pv”,

Re: [OpenSIPS-Users] tcp_blocking_connect failed and webRTC error

Hi, What JS library use to make Webrtc connection? Best regards, Dragomir 2017-09-13 10:29 GMT+03:00 Bart Remans : > Hello, > > I'm trying to connect through WSS but I keep getting a webRTC error when I > answer the call. > It gives errors: tcp_blocking_connect failed

Re: [OpenSIPS-Users] [NEW] SIP Recording module

Hello, I did not understand this module can record sound or only SIP signaling. Best regards, Dragomir 2017-09-28 20:45 GMT+03:00 Răzvan Crainea : > Hello, everyone! > > I am happy to announce you we have just released a new module: the > SIPREC[1] module can be used to do

Re: [OpenSIPS-Users] Problem: Registration Proxy with WebRTC

7-02 22:20 GMT+03:00 Dragomir Haralambiev <goup2...@gmail.com>: > Hello, > > I try to setup Registration proxy (opensips 2.3) like this article: > https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ > > 1. When use Zoiper all is OK > Zoiper <>

[OpenSIPS-Users] Problem: Registration Proxy with WebRTC

Hello, I try to setup Registration proxy (opensips 2.3) like this article: https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ 1. When use Zoiper all is OK Zoiper <> Opensips REGISTRATION Proxy <-> SBC 2. I have problem when use WebRTC. (*Via sent-by in the response

Re: [OpenSIPS-Users] WEBRTC - Bad Request/Body

sipmsg_validate error: -13 2017-05-24 22:09 GMT+03:00 Liviu Chircu <li...@opensips.org>: > Can you do an xlog("sipmsg_validate error: $rc\n") in the error block, > please? > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 24.05.2

Re: [OpenSIPS-Users] WEBRTC - Bad Request/Body

error reason, which would be very helpful > if printed. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 24.05.2017 19:15, Dragomir Haralambiev wrote: > > In Opensips log I not see any errors. > > > On May 24, 2017 10:37, "Răzvan Cra

Re: [OpenSIPS-Users] WEBRTC - Bad Request/Body

penSIPS Solutionswww.opensips-solutions.com > > On 05/23/2017 09:39 PM, Dragomir Haralambiev wrote: > > Hello, > > After update from Opensips 2.2.3 to 2.2.4 I have follow problem with > WebRTC (sip.js): > When WebRTC send CANCEL Opensips return SIP/2.0 400 Bad Request/Body. &g

[OpenSIPS-Users] WEBRTC - Bad Request/Body

Hello, After update from Opensips 2.2.3 to 2.2.4 I have follow problem with WebRTC (sip.js): When WebRTC send CANCEL Opensips return SIP/2.0 400 Bad Request/Body. Here is WebSocket log: Tue May 23 2017 21:10:47 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message: CANCEL

Re: [OpenSIPS-Users] WSS and Opensips

Hi, If you like to work with WEBRTC you must to use WSS. In SIP.UA add follow lines: hackIpInContact: true, contactTransport: "wss", rtcpMuxPolicy: 'negotiate', Best regards, Dragomir ___ Users mailing list

Re: [OpenSIPS-Users] ERROR topology_hiding_match() and WSS

iling list too, so people can > benefit of it ;) > > 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 05/18/2017 04:20

Re: [OpenSIPS-Users] Set Acc Module in 2.3

gt; This will translate to $avp(system_name) being sent as radius avp > MV-System. > > > [0] http://www.opensips.org/html/docs/modules/2.3.x/acc#ACC-extra-id > > Ionut Ionita > OpenSIPS Developer > > On 04/29/2017 01:50 PM, Dragomir Haralambiev wrote: > > Hello, >

[OpenSIPS-Users] Set Acc Module in 2.3

Hello, I try to move from 2.2.3 to 2.3. In 2.3 modparam("acc", "aaa_extra",... is removed. In 2.2.3 I use extra data with: modparam("acc", "aaa_extra", "MV-System=$avp(system_name); MV-UserName=$fU; MV-Number=$rU; MV-Timestamp=$avp(timestamp)") For 2.3 I rewrite to: modparam("acc",

Re: [OpenSIPS-Users] Opensips 2.3 ERROR:core:sr_load_module

For asyn radius I apply the patch. 2017-04-28 15:24 GMT+03:00 Dragomir Haralambiev <goup2...@gmail.com>: > Hi, > > Opensips 2.2.3 working fine here is ldd info: > > [root@dev aaa_radius]# ldd aaa_radius.so > linux-vdso.so.1 => (0x7ffde55f2000) >

Re: [OpenSIPS-Users] Opensips 2.3 ERROR:core:sr_load_module

(0x7f790aec9000) libdl.so.2 => /lib64/libdl.so.2 (0x7f790acc4000) In Opensips 2.3 Manual (AAA Radius module) I see information for setup RADIUSCLIENT. In first line at Makefile I put: RADIUSCLIENT="FREERADIUS" For asyn radius I apply the patch. 2017-04-28 15:09 GMT+03:00 D

Re: [OpenSIPS-Users] Opensips 2.3 ERROR:core:sr_load_module

/lib64/ld-linux-x86-64.so.2 (0x7f7153c98000) 2017-04-28 15:02 GMT+03:00 Ionut Ionita <ionution...@opensips.org>: > what radius library is aaa_radius using? Please post the output of: > > ldd aaa_radius.so > > Ionut Ionita > OpenSIPS Developer > > On 04/26

[OpenSIPS-Users] Opensips 2.3 ERROR:core:sr_load_module

Hello, Opensips 2.2.3 working fine. When try to start new 2.3 I see follow ERROR: Apr 26 23:10:15 dev opensips: ERROR:core:sr_load_module: could not open module : /usr/local/lib64/opensips/modules/aaa_radius.so: undefined symbol: rc_dict_findvend Best regards, Dragomir

[OpenSIPS-Users] ERROR topology_hiding_match() and WSS

Hello, I have problem with "topology_hiding_match()" and WSS. Zoiper ---(send BYE)--> Opensips --(can not relay to)---> SIP.JS Here part ot script: if (has_totag()) { if (topology_hiding_match()) { t_relay(); exit; } ... Opensips receive BYE. When execute "t_relay()" give follow ERRORS:

[OpenSIPS-Users] ERROR when receive INVITE

Hello, I make test with WebRtc. SIP.js ---send call to ---> (Opensips A) -send call to ---> (Opensips B) - On Opensips A is working rtpengine. This is rtpengine_offer command: rtpengine_offer("RTP/AVP replace-session-connection replace-origin ICE=remove"); (Opensips A) send

Re: [OpenSIPS-Users] codec_delete_except_re() has no effect

Hi Razvan, How to make follow connection using rtpengine? Zoiper(g729) <-> Opensips(rtpengine) <> browser (SIP.JS with g711) 2017-04-18 19:10 GMT+03:00 Răzvan Crainea : > Hi, Jeff! > > Unfortunately you can't use both rtpengine and codec_delete_*, that's >

Re: [OpenSIPS-Users] WebRTC

in the only media relay that > supports translation. Also consider using a library that supports sip > headers. JSSIP or SIPJS > > Thanks, > Tito > > On Thu, Apr 13, 2017 at 3:00 PM, Dragomir Haralambiev <goup2...@gmail.com> > wrote: > >> Hello, >> &

[OpenSIPS-Users] WebRTC

Hello, For WebRTC I must to use rtpengine. In this case I need to stop rtpproxy? Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] tls_mgm module ERROR

.el7_3.x86_64 What I do to start Opensips with tls_mgm? Best regards: Dragomir 2017-04-06 23:56 GMT+03:00 Dragomir Haralambiev <goup2...@gmail.com>: > rpm -qa |grep ssl > > openssl-1.0.1e-60.el7_3.1.x86_64 > openssl-libs-1.0.1e-60.el7_3.1.x86_64 > mod_ssl-2.4.6-45.el7.centos.x

Re: [OpenSIPS-Users] tls_mgm module ERROR

against (libssl-dev version) and what is the > runtime library you are using? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 04/05/2017 03:19 PM, Dragomir Haralambiev wrote: > > Hello, > > I try to start opensips with tls_

[OpenSIPS-Users] tls_mgm module ERROR

Hello, I try to start opensips with tls_mgm module and receive follow errors: Apr 5 15:14:05 dev /usr/local/sbin/opensips[2628]: WARNING:tls_mgm:mod_init: disabling compression due ZLIB problems Apr 5 15:14:05 dev /usr/local/sbin/opensips[2628]: INFO:tls_mgm:check_for_krb: KRB5 cipher

[OpenSIPS-Users] OpenSips and WebRTC ERRORS

Hello, I try to test WebRTC. OS: CentOs7.2 Opensips: 2.2.3 Tutorial: http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 Zoiper ---> Opensips -> Crome(tryit.jssip.net) In Opensips log have many ERRORS: rtpengine[719]: INFO: [xgo3OaZkRrYqO3SQFH03JQ..]: Received command 'offer'

Re: [OpenSIPS-Users] Radius acc - wrong Sip-Call-Setuptime

Please do not post the same issue on different channels, because it is > hard to follow all of them. > > [1] https://github.com/OpenSIPS/opensips/issues/1032 > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 01/27/2017 11:10 PM,

[OpenSIPS-Users] Radius acc - wrong Sip-Call-Setuptime

Hello, I try to use OpenSips 2.2 with Radius ACC. Radius server receive wrong Sip-Call-Setuptime like this Sip-Call-Setuptime = 2938 When use OpenSips 1.10 all is OK. Where is problem? Best regards, Dragomir ___ Users mailing list

[OpenSIPS-Users] number validation

Hello, How Opensips 2.2 can validate number ? "491234567" is valid number , but "49a12b34567" is not valid Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Call per second limit

Hello, How to made Call Per Second limitation using Opensips 2.2.2 ? Regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] async radius problem

; Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 01/19/2017 10:40 AM, Dragomir Haralambiev wrote: > > Hi, > > Thanks for your quick replay. > > Please add this information in AAA_Radius module. > > Best regards, > Dragomir > > 2017-01-19 10

Re: [OpenSIPS-Users] async radius problem

the onreply_route you can only use synchronous operations for > now. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 01/18/2017 03:45 PM, Dragomir Haralambiev wrote: > > Hello, > > I have problem with async radius impl

[OpenSIPS-Users] async radius problem

Hello, I have problem with async radius implementation. OS - CEntOs 7 Opensips 2.2.2 - git hub from 15.01.2017 Here pasrt of srcipt: onreply_route[outgoing] { . if (t_check_status("200")) { async( radius_send_auth("prepayout","prepayin"), return_prepay ); } } route[return_prepay] {

Re: [OpenSIPS-Users] Problem with switch .. case

in a > variable and test that variable. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 01/16/2017 12:24 PM, Dragomir Haralambiev wrote: > > Hello, > > I see problem when using switch .. case. > Opensips version 2.2.

[OpenSIPS-Users] Problem with switch .. case

Hello, I see problem when using switch .. case. Opensips version 2.2.2. Here is part of my script: radius_send_auth("out","in"); xlog("L_ERR", "Radius return $rc"); switch ($rc) { case -1: xlog("L_ERR", "ERROR during authentication"); sl_reply_error(); exit; case -2: xlog("L_ERR",

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