[OpenSIPS-Users] n00b SIP question
Hello, I have read this mail list for months, everything is way above me – but, I am ready to jump in. I have a twilio account, I have static COMCAST business class (Deluxe), not in bridged mode. What are the steps/products I should install on a Linux server to do my own VoIP? Design goal, 1 SIP phone in the same 10.1.0.x network, 1 SIP phone in Georgia, and 1 SIP phone in Philadelphia. Goal is to get an understanding, and migrate my 8 or 9 DIDs to inhouse, have a nice DELL (16GB RAM, 3TB of disk space), dual FAST-E NICs. * I gave I a whirl a couple weeks ago, with one of those “all-in-one” projects, like “Sip on a stick”. Migrated 2 DIDs without issue adding 3rd, and 4th brought the whole environment down where none of the DIDs worked. I am interested in just doing it module by module, product by product until I have this understood (like when do I need STUN or will I ever, etc). Thanks guys! Ozz ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS work behind COMCAST Business Class?
We are looking to switch everything from one of our datacenters to our corporate which runs COMCAST Business Class. Googling people have all kinds of problems with other solutions on COMCAST (But I think they are all residential setups). We have a Cisco DPC3939B COMCAST Modem running in non-bridge mode, with a large group of static IPs - 1 is earmarked for this install. The design is: Possibly OpenSIPS in VA NAT as 10.1.x.y <-> PUBLIC IP via DPC3939B Has two SIP phones here (Cisco 7940) - all 10.1.x.x addresses internally In GA I have one SIP phone (7940) In FL I have three locations with one SIP phone each (7940) In PA I have one SIP phone (7940) I have enough bandwidth if I need OpenSIPS to literally be the proxy for all points (e.g. no REDIRECT of the RTP). Total of 8 DIDs. I think that covers config questions. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Product Question....
I am wanting to see if OpenSIPS is what I need, or is there something else I should be using, thanks: Q. I want to have multiple asterisk/freePBX boxes send/receive calls... i. The uplink's SIP Trunk is pushing all INVITEs to my server. ii. However, all Asterisk/PBX systems reside on other networks. So, I would like to check my "Is Online" type DB, if it is only - redirect the SIP/RTP to the PBX. If not online, redirect to a CATCH-ALL PBX. iii.On outbound, if in my "Is Online" then redirect to the uplink's SIP Trunk. My goal to support thousands of calls, without being in the middle for the RTP streams - just for the SIP. Is that possible, or is there a better product for this??? Thanks again! O. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NEWBIE question.
We have installed OpenSIPs - we have two PBX's setup - they communicate to one another perfectly... now, we are ready to point OpenSIPs to our up-stream SIPTrunk... could someone explain how to do this in just a couple steps? Everything we are reading and trying, seem to be overly complex. Thanks in advance! O. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
