similar to: OpenSIPS Summit Oct 21st before Astricon

A SipPulse acaba de liberar o TFPS (www.tfps.co), solu??o para combate a fraudes de fomento de tr?fego internacional em telefonia. O sistema ? capaz de detectar 99.99% das tentaivas de fraude em tempo real. Durante o Webinar, abordaremos como proteger servidores Asterisk e Elastix/FreePBX de fraudes, medidas basicas como configura??o de firewall e remo??o de servi?os desnecess?rios e em seguida

Hi! I've been using v.-4.4 since last Oct, and I'm quite satisfied, *except* for *one* thing: I can't scan! Both kooka and xsane report "no devices available." That doesn't affect printing, but every time I have to scan something, I've to go back to Mandriva or Ubuntu. I've had this HP psc-2410 for a couple of years. I scanned successfully with Fedora

I currently have OpenSIPS set up with users and most of my call handling. OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD, etc. So I want to send these types of requests to Asterisk. I also want to set Asterisk up as Multi Tenant. So my question is How can I send requests to Asterisk and have them funnel into the specific context for that specific Tenant? So if

Hi all, I've been trying to register a SIP user agent to an Asterisk server using OpenSIPS as SIP router. The functionality is working fine. However, Asterisk uses the IP address of the OpenSIPS server as the peer IP address. How can I use the original IP address of the peer without changing the peer's nat=yes? I appreciate any kind of help. Thanks! Regards, Ronald -------------- next

Hi, I'm using opensips as the registrar server for my users. I am redirecting calls going out to pstn to my asterisk server. call flow is basically: ua --> opensips server --> * server --> sip gateway provider if (uri=~"sip:00[0-9]*@sip\.myserver\.com") { xlog("L_INFO", "Call to PSTN\n"); #strip(2); #prefix("011");

Hello fellows VOIPer, If you want to share with the rest of the VoIP & RTC community some news, interesting or breaking through ideas, or even more, some experience you had in terms of designing, integrating or operating various solutions or platform based on Open Source Softwares, then you should consider submitting a paper for the OpenSIPS Summit 2020 in May, Amsterdam.

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

XDC 2023 A Corunha Spain Tuesday Oct 17 ? Thursday Oct 19, 2023 Join with Google Meet https://meet.google.com/azn-uwfp-pgw?hs=224 Hello! Registration & Call for Proposals are now open for XDC 2023, which will take place on October 17-19, 2023. https://xdc2023.x.org As usual, the conference is free of charge and open to the general public. If you plan on attending, please make sure to

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote: > ** > Hi Nick, > > The BYE is not properly formed and rejected by script - in the 200 OK of > the INVITE, you can see that your opensips is doing Record-Routing, but the > BYE does not contain the corresponding Route hdr, so SIP routing is > impossible. > > Regards, > >

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Hi everybody, CentOS will be present for the first time at the Technical Dutch Open Source Event (T-DOSE) in Eindhoven, NL. We have a booth at the entrance as well as a presentation about Virtualization, presented by Daniel de Kok (danieldk). No doubt the presentation will include information about the new CentOS Virtualization SIG ! (Other topics are still accepted for T-DOSE) If you're

Hello, As i still have issue i wonder if anyone using this version of openssl ? OpenSSL 0.9.7e-p1 I still cannot make puppet accept Certificate on freeBSD6 with this openssl version. I got the famous error 7 at 1 depth lookup:certificate signature failure and i havent find a way to get around that :( -- Cordialement, Ghislain _______________________________________________

Howdy, Not sure, but it appears that OpenSSH_4.3p1 on solaris creates bad wtmpx entries during login? mgoebel pts/5 Wed Dec 31 19:00 still logged in It is creating entries for Dec 31st 1969. Thanks, Matt -- Matthew Goebel : goebel at emunix.emich.edu : Unix Jockey @ EMU : Hail Eris Neo-Student, Net Lurker, Donut consumer, and procrastinating Furry Fan.

Hi list, quick question : With GoToIfTime, how to define a period of holiday that starts at the end of the month and ends at the beginning of the next month ?? Like September 25th till October 10th when incoming calls need to go to the voicemail... Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I wanted to make sure you were all aware of these upcoming events. There is a seminar in Predictive Analytics on Oct. 24-25 in DC, and in San Francisco Nov 6-7. This is intensive training for managers, marketers, and IT people who need to make sense of customer data to predict buying behavior, profit, etc. Past attendees have given rave reviews. You can find more info at