Thanks again,
Charlie

I was wondering about the e164.org registration for my GV number. I followed the directions using the +1 in front of my 10D GV number but e164.org doesn’t like something.

Here is the error:
Wasn’t able to test your route, your system returned the following information: -1|Error 504 – No response was returned from the remote end, this might indicate a problem with a firewall connection

The enumplus.org went great. No troubles.

Any thoughts?

[WM: Probably the result of Google shutting down the SIP service again. 🙄 ]

Just curious.

Why Does Google Hate SIP?

[WM: True, but without the Pioneers, where would we all be?]

Has Google shut this down already?

[WM: Yep. It’s dead for us, too, but still works for a precious few. That was fun.]

On the other hand, on an Asterik 1.8/FreePBX 2.8 based system, if the call was not to a GV number it would not fall through to another trunk. The weird part was that I watched the CLI and it SAID it was falling through to another trunk, it just wouldn’t actually DO it. Calls to Google Voice going out the normal way (using the Asterisk 1.8 channel drivers) work fine, as do calls using the above method when made to a Google Voice number, but when I tried calling a non-GV number it just wouldn’t fall through to the next trunk (and I even tried adding the useless 1NXXNXXXXXX pattern, though I know it has no effect — I guess we’ll just have to agree to disagree on that one, Ward).

Anyway, I’d be a little careful about doing this on any production system. I wasn’t going to say anything about the problem on the second system but when the 30 second delay appeared out of nowhere (and disappeared after I disabled the custom trunk), I figured maybe I should let you guys know, so you can at least make sure it’s not happening on your systems.

[WM: It’s automagic on Google’s end.]

[WM: This depends upon how your Outbound Routes and other Trunks are set up. Calls start at the Outbound Route with a matching Dial String and then proceed down through the designated Trunks in sequence. To have a match on a Trunk, you have to have a Dial String entry for the Trunk that matches what was passed from the Outbound Route. Keep in mind that our articles are written to accommodate system designs of thousands of different Asterisk users. So what you or I would do is secondary to assuring that something "just works" for everybody. Without getting too deep in the weeds, it’s worth noting that it is not uncommon to see an Outbound Route with only two dial string entries: 1NXXNXXXXXX and NXXNXXXXXX. With such a setup, the 1 does need to be added as a prepend in the Google-SIP Trunk, or calls dialed as 10-digit numbers would always fail when the call hit the Google-SIP trunk without the 1+NXXNXXXXXX entry. And, for anyone that dials an 11-digit number, the 1NXXNXXXXXX entry catches that type of call. You are correct that the rule is redundant with FreePBX 2.8; however, not everyone is quite there yet. There is no single "right way" which, of course, is one of the beauties of Asterisk and FreePBX. Our design was intended to eliminate as many "wrong ways" as possible.]

[WM: Google Voice calling via Gtalk still works as advertised. The SIP addition simply provides another (free) way to make calls to Google Voice numbers. And it works worldwide. Finally, if Google does ever start charging for calls, this will provide a way to continue to reach Google Voice numbers at no cost. ]

[WM: It’s a little of both. SIP support adds a new way to receive incoming calls to your Google Voice numbers. And, for those using SIP phones or SIP-based PBXs such as Asterisk or FreeSwitch, it provides a new way to place outbound calls to Google Voice numbers at no cost.]

Also, SIPifying in one direction SIP=>GV; but not in the other – GV=>SIP.

I have heard rumors that GV will support SIP for callback on Android 3.0(?).