On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass <a...@p2ee.org> wrote:
> On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro > <stot...@asteriskhelpdesk.com> wrote: > > > > > > [...] > > Yes, I have had no problems with Grandstream first gen ATAs, configured > with > > server and credentials and shipped off, they just work. > > We use the HT-286, the server is on a public IP the nat setting on > asterisk is set to yes and without port re-direction the ATAs have > never connected from a private network, so I honestly find this "SIP > plug and play" very hard to believe. But if it is true, then maybe you > can actually help us figure out all the NAT issues we've had with SIP > for the past 5 years. Perhaps, it is simply ignorance on our side and > we have something fundamentally wrong in our set-up somewhere that may > be have been causing these issues with NAT. > > Our set-up is fundamentally public and private Asterisk servers > running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and > Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD > 8.2 and Asterisk 1.8 but we are in that process right now. Some > Asterisk run in jails so I can understand the NAT issues there may be > caused by the server itself. I honestly *love* your OpenVPN idea but I > have to find a cheap ATA that could run as an OpenVPN client. > > Taking the simplest example a simple Asterisk 1.6 server on a public > IP running on the base system (not in a jail): > > We run an operation that spans several countries including Canada, the > US and the Latin American Andean region. As examples, with Canadian > ISPs such as Rogers and Bell we have always had to redirect the ports > and use STUN server for the HT-286 to register to the Asterisk server. > > In the US we have the same problem with Comcast networks, so I don't > understand how you say that you plug a Grandtream SIP ATA to a Comcast > router and it just works. However, in a couple of NOLA countries the > ISP's routers actually give public IPs, so if the SIP ATAs are > connected directly to the ISP router, or in the DMZ then it just works > as you say, BUT if the ATA is connected behind the firewall, or to a > WiFi router, then we've _allways_ had to redirect ports. In every > sigle customer we have had to send instructions on how to redirect > ports, and of course to configure firewall if present. > > I just don't understand how you and other here say that a SIP ATA can > "just work". On the contrarty, with IAX2 using cheap AG-188N from > Atcom they are just plug and play when shipped with a standard conf, > and we have none of the quality issues you are referring to. We do > have some call drops however, and some hangup problems but they don't > affect our clients as much as having to deal with NAT issues. > > We may not run 15K extensions like you but I think we have a pretty > good testing ground and have dealt with a fair share of NAT problems > with SIP, that you and others here apparently don't have, so I am as > amazed by your likeness of SIP as perhaps you are amazed as our > likeness of IAX. > > If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. If I cannot help you in a few emails, we can take this offline. Actually paste your entire sip.conf in pastebin or something, as well as sip debug. Also the configs of your ATAs. I think you have over-engineered to the point of creating problems. This is very common. My philosophy is "KISS" Thanks, Steve T
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