search for: opensips

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...

...span style="color: #000000;"><span style="color: #000000;">&nbsp;Hi,</span></span></span></span></span></p> <p>&nbsp;</p> <p>sorry for cross-posting on both mailing lists, but I think a setup of Asterisk with OpenSIPS as frontend isn't unusual. So maybe both parties would be interested in this.</p> <p>&nbsp;</p> <p><span style="font-size: 9pt;"><span style="font-family: verdana,geneva;"><span style="background-color: transparent;">&l...

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 :) Which do you use and why? -- Thanks in advance, ----...

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to aste...

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP address of my host running Asterisk 11.17.1. T...

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" se...

Hello Everyone! We wanted to let everyone coming to Astricon know that we will be holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast Casino & Spa. Suncoast is about 10 minutes away from Red Rock and we will be provide shuttle service to and from the Summit. For those of you that had to book at Suncoast it should be really easy to find us! Here are some things you can exp...

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a Proxy- Authenticate header 3. Asterisk re-sends INVITE to OpenSIPs with Proxy-Authorization header, but has the wrong password...

...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. https://www.papercall.io/opensips-summit-2020 We welcome anyone ready to share knowledge! Best regards, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensi...

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 --------------...

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 t...

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) Suppo...

Hi list i setup successfull asterisk version 1.4 + opensips, Opensips is the Registrar Server, Asterisk is the IVR server the call flow IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE 5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP phone (5000) my case is: 1/ IP phone(5000) --->Opens...

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. > &gt...

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"...

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 PST...

Hi, What if some fail2ban magic could keep OpenSIPs response from hitting Asterisk after N attempts ? Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd < kingsley.tart at barritel.com> a écrit : > Hi, > > We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. > > I've found an issue when Asterisk tries to make a...

...-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? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline...

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSI...