similar to: Asterisk, OCS and Caller-ID

Dear OCS fellows, I've been thinking about working on adding power devices knowledge to OCS for years. Following the last Ubuntu Developer Summit, I know have an "excuse" to do so: https://blueprints.launchpad.net/ubuntu/+spec/servercloud-p-cloud-power-management My below proposition is related to the above blueprint. So please keep in mind that the target is also to be able to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, everyone. I added a new guide for installing and configuring the OCS Inventory NG server/client system on CentOS 5.x. Also, there's some brief explanations for integrating it into GLPI. I added it under non CentOS applications, since I think it fits in best there. Regards, Max - -- # find . "*imbecile" -exec sed -ie

Since OCS seems to be a popular host, I''m wondering if anyone has gotten attachment_fu working on their shared enviroment using s3 as storage. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to

I am attempting to run Poole's OC package in R, using Windows NT 5.1. Everytime I try to estimate a two-dimensional model, the estimation begins and then I am informed "R for Windows GUI front-end has encountered a problem and needs to close". There is no problem estimating a one-dimensional model. I have no idea why the estimation will not contine. I have included the error

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 asterisk to play the auto attendant messages like Welcome and Dial the

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

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

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) --->Opensips 2/ IVR number : 1001 3/ IP

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.

When a client send me an INVITE with this type of caller ID From: "eurus" <sip:<null>@XX.XX.XX.XXX>;tag=3430296121-3809549020-352327076-1077499159 Asterisk 14 sends back SIP/2.0 500 Server error occurred (1/SL) My client says "Yes, I know the null is there but this not illegal and perfectly acceptable according to rfc 3261. " Should I open a bug ticket?

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

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 SIP call out using > auth, but has the wrong

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

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, > >

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

Hello i have nvidia gtx 650 ti boots oc 2gb with 2 monitors and gt 240 for xen emulation .My screen displays random pixels and freeze in random times . I don't found errors in /var/log/syslog and /var/log/kern.log its nouveau error on nvidia-driver works fine . Sometimes its freeze. sometimes there are font errors. uname -a Linux debian 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u2 (2017-06-26)

Hello i have nvidia gtx 650 ti boots oc 2gb with 2 monitors and gt 240 for xen emulation .My screen displays random pixels and freeze in random times . I don't found errors in /var/log/syslog and /var/log/kern.log its nouveau error on nvidia-driver works fine . Sometimes its freeze. sometimes there are font errors. uname -a Linux debian 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u2 (2017-06-26)

Hello folks, my company has experience in setting up single asterisk setup, but recently one of our customers asked us to set up an asterisk cluster, that must be High Availability and Load Balanced. So I wrote here to have some hint or advice about the configuration we thought. First of all I'll explain you the scenario: The asterisk cluster must serve as Call Center

Hello i have nvidia gtx 650 ti boots oc 2gb with 2 monitors and gt 240 for xen emulation .My screen displays random pixels and freeze in random times . I don't found errors in /var/log/syslog and /var/log/kern.log its nouveau error on nvidia-driver works fine . sometimes there are font errors. uname -a Linux debian 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u2 (2017-06-26) x86_64 GNU/Linux

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");