On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro <stot...@asteriskhelpdesk.com> wrote:
[...] > Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT > open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not "the truth". - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. > IPTables is simple and there are tons of howtos. > > Thanks, > Steve T > > > On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro <stot...@asteriskhelpdesk.com> > wrote: >> >> They said the same thing in 2005, 2008, now.... Every release. >> >> You never answered the question as to why you don't want to use SIP. Is >> there a reason, or do you just want to torture yourself? >> >> Thanks, >> Steve T >> >> >> On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford <ttelford.gro...@gmail.com> >> wrote: >>> >>> On 2012-02-28 21:22:44 +0000, Kevin P. Fleming said: >>> >>>> >>>> A serious bug with IAX2 trunking in recent versions of Asterisk (you did >>>> not mention what version you are using) was just resolved last week. You >>>> should test with 'trunk=no' to see if that is the cause of your problem; >>>> it seems very likely. >>> >>> >>> For the record: 1.8.8.2~dfsg-1 (via Debian packages). >>> >>> I've tried "trunk=no", and it might have made a difference (I'll have a >>> better idea after some more testing.) >>> -- >>> Troy Telford >>> >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users