PSU VoIP: Comments

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Fri, 25 May 2012 15:06:50 -0500

Strike that last comment. I was able to get this going fine with "Make Current." The glitch probably had to do with my being signed into the Google account with my Android tablet. One thing I can say is that trying to use Google Voice this way can be pretty squirrelly.

- NYZack

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Wed, 23 May 2012 19:51:07 -0500

I used this solution for a while and gave up on it for a few months because of intermittent unreliability - mainly the problem was that FreeSwitch would occasionally (once every few days) disconnect from Google Voice (perhaps due to a router hiccup?), and, when I'd make an outgoing call, I would just hear ringing and no indication that the call wasn't being connected. I'd either have to restart FreeSwitch, or the act of trying to make a call would reconnect it (for the next call).

I came back to this installation out of curiosity, and did "Make Current" on my FreeSwitch installation to see if there were any changes, and I now find that I can make outgoing calls OK (and I see FreeSwitch showing up on Google Chat), but FreeSwitch doesn't register incoming calls. I restored to the old version (from late last year, I believe), and things worked fine again, but another "Make Current" gave me the same incoming call failure.

- NYZack

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Mon, 30 Apr 2012 12:38:49 -0500

First, to answer Jrey's comment, I suspect most of us are not using "raw" Asterisk as you appear to be. When you write your own dialplans, as you must when you don't use any sort of configuration tool, then you are pretty much on your own. FreeSWTCH appears to Asterisk just as any other trunk would appear (assuming you use the trunk configuration shown above) and it's pretty much up to you to route calls to and from that trunk.

(And, I might add, if you are going to go through all the trouble of writing your own dialplans, why not just use FreeSWITCH all the way? I don't know offhand how hard it would be to get it to work on a Debian dockstar but it might be worth a try, since FreeSWITCH seems to be a more stable product, whereas the Asterisk developers have a nasty habit of breaking existing functionality with each new release.)

Now, I also have a question. I'm not real good with regular expressions, otherwise I might be able to figure this out myself. But I have one user that's connecting to our server using Sipdroid, which is a VoIP application for the Android operating system. He has the ability to cut and paste numbers into Sipdroid and apparently it will send whatever is pasted. So he pasted in a number of the form xxx-xxx-xxxx and it sent it, hyphens and all, and of course the call failed.

In the configuration there is a line like this:

The page at explains the .* pattern like this: "The dot matches any character, and the star allows the dot to be repeated any number of times, including zero." The problem is that if the user pastes in a number with the pattern xxx-xxx-xxxx or (xxx) xxx xxxx or anything with non-numeric characters, it will still (if I understand correctly) get passed to Google Voice and Google Voice will reject it.

What I'd like to know is whether that regex could be modified to strip out all non-numeric characters before passing the call to Google Voice. Note I'm not trying to block a call that looks like xxx-xxx-xxxx or (xxx) xxx-xxxx, but I want to strip out anything that's not a number (hyphens, parens, spaces, and any other stray non-numeric character), so that either of those would get passed to Google Voice as xxxxxxxxxx (actually +1xxxxxxxxxx after we add the +1 in the line that bridges the call). Seems like there ought to be an easy way to do that in the regex but if there is, I don't know what it is.

I also find it strange that Asterisk apparently totally ignores hyphens in pattern matches, but that's another issue.

- Michigan Telephone

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Wed, 04 Apr 2012 13:36:34 -0500

Can someone show an example of sip.conf and extensions.conf in Asterisk? I am a newbie and don't quite understand what to do with the comment:
"3. Set up appropriate inbound and outbound routes in FreePBX or in your extensions.conf dialplan. This is outside the scope of this how-to."

I have install Asterisk1.8 and Freeswitch on a debian dockstar and use an unlocked sunlocket for ATA.

Any help would be very appreciated. Thanks.

- Jrey

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sun, 01 Apr 2012 18:18:57 -0500

I've been working with this setup for over a year now, GREAT work guys! I'm using Asterisk 1.4 and this freeswitch + google voice integration is seamless and lightweight. I do however, have a couple of issues that perhaps the community is already aware of, but I haven't seen any threads on it. First thing, when making many calls, google servers stop taking your call requests for about 5-10 minutes. I'm guessing google has a limit on how many outgoing calls you can place. I have 6 gvoice accounts authorized in dingaling. Second, as of late, the dingaling mod shows my accounts as "unconnected" for a long while after i start the freeswitch server. I have to wait, perhaps attempt to make a call, before I see that 5 of 6 accounts have been "authorized." What may be causing the delay?

- JP V

Comment on "Skype for Asterisk using FreeSWITCH, for hackers"

Fri, 30 Mar 2012 12:29:11 -0500

It looks like a small patch was released. The new link is http://mirrors.kernel.org/archlinux/community/os/i686/skype-oss-2.0.0.72-3-i686.pkg.tar.xz . I am updating the article. Thanks!

- Bill

Comment on "Skype for Asterisk using FreeSWITCH, for hackers"

Fri, 30 Mar 2012 12:25:57 -0500

hi,Bill - great creator.

Bill and Jason and Ranga.

could not get download of - wget "http://mirrors.kernel.org/archlinux/community/os/i686/skype-oss-2.0.0.72-2-i686.pkg.tar.gz"
how do for install Skype for Asterisk using FreeSWITCH, need of skype-oss-2.0.0.72-2-i686 rpm or other.

grateful for helping us.

- kessius

Comment on "Guarding your productivity sweet spot"

Tue, 20 Mar 2012 17:08:54 -0500

For me, multi-tasking N things = doing each thing only 1/N as fast at 1/N the quality. But your point is well-taken. At least some transitional time would be saved by using the phone.

- Bill

Comment on "Guarding your productivity sweet spot"

Tue, 20 Mar 2012 15:26:44 -0500

I'm the same way, my sweet spot is from 10 AM to 2PM. That's not to say that the other times are completely off, but I need to watch that up time to make sure I'm doing my most important things for the day when my energy is maximized.

Being a VOIP / IP Telephony expert... have you considered doing your meetings by voice so you can, um, multi-task? :-)

- Matthew Turner

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Wed, 07 Mar 2012 13:51:04 -0500

Matt, did that fix from the PBX in a Flash forum resolve the problem with inbound calls, or does it still keep coming back? By the way, the link to that thread changed this morning, it is now:
http://pbxinaflash.com/community/index.php?threads/gvoice-stopped-working.12377/

- Michigan Telephone

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sun, 26 Feb 2012 16:01:56 -0500

/usr/local/freeswitch/conf

- Bill

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Fri, 24 Feb 2012 23:03:36 -0500

where is the freeswitch.xml normally located?
i am having trouble finding it.

- nsdemon

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sun, 19 Feb 2012 00:26:15 -0500

Outbound calls have all work 100%. The same was true for Asterisk when I used that. Only inbound calls have an issue.

It seem if I introduce a delay in answering, as mentioned above on Feb 1st, does seem to make things work for a *short* while. Ultimately though the same problem comes back. I've tried this:
http://pbxinaflash.com/forum/showthread.php?t=12377. It too soon to say for sure since I just install freeswitch today but it appears to help.

- Matt

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sat, 18 Feb 2012 22:06:12 -0500

Can you make calls outbound?

- Bill

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sat, 18 Feb 2012 19:22:43 -0500

Does this still work? Trying this myself I am seeing exactly the same sort issues I had with Asterisk's built in functionality. Incoming calls will ring my phones for a few seconds only. If I answer them I get silence. Regardless what happens the caller hears ringing until the call ultimately goes to google voicemail. At least with Asterisk direct a few calls would make it through. With freeswitch this happens 100% of the time. Outgoing calls work fine with either.

If it makes any difference I'm using Asterisk 1.8.7 (rpmfusion) with FreePBX. My asterisk system is behind a firewall.

- Matt

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sat, 18 Feb 2012 15:03:19 -0500

Greg, I haven't seen the 5-second disconnect issue, but suspect it has to do with NAT and Asterisk believing FreeSWITCH is not on your network. Check your Asterisk SIP/NAT settings and make sure that 127.0.0.1 is configured as a "localnet". (See the Asterisk section above.)

- Bill

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sat, 18 Feb 2012 14:55:51 -0500

Doc,

Yes, that's my gateway running Yate. I haven't tried Yate and Asterisk on the same server yet. It would not be hard to do, and the procedure would be much the same as we are doing here with FreeSWITCH: set Yate to listen on a different SIP port, configure the Google Voice account on Yate and a "trunk" between Yate and Asterisk on localhost. Add some simple routing rules and you have a gateway.

Yate is actually simpler to configure and it starts up bare-bones. You add what you want. FreeSWITCH's default has a bunch of things to remove if all you want is a proxy/gateway. A big difference is that Yate uses several config files, whereas you can configure FreeSWITCH in a single file, as I've done here. So a blog post describing a Yate gateway setup might need more instructions even though I am claiming here that it is simpler.

I don't know whether Yate or FreeSWITCH is better. I have come to like Yate more, just because. As to how well it scales... just between you and me and the rest of the Internet, the Yate/GV gateway test server has over 250 GV accounts logged in at this time. It's basically idle, because XMPP connectivity doesn't cost much in terms of server resources. With any VoIP server your resource usage will be tied to media streams (call legs, multipoint conferencing/mixing).

PS: I am not sure whether our blog service does comment notification. I'll check on it. If you use a news reader, you can subscribe with RSS to this blog's comments feed using the link on the right side of the page.

- Bill

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Thu, 16 Feb 2012 00:12:15 -0500

Bill,

THANK you for responding. I don't know if I was supposed to be notified - but I wasn't. I just happen to come across your thread/article again continuing my search for this endeavor.

If I am correct - you are the same Bill that has the info online about the YATE/Google Voice Gateway? (and you even have a test server up and running for that, correct?)

With the experience that you have both using FreeSwtich and YATE (as gateways to Asterisk allowing use of the Google Voice service) - which one is better, and why?

If you are going to have multiple Google Accounts (say like anywhere from 3 or 4 or even up to 10) - then would you change your position on which Gateway method is better?

Thanks again for responding!

- Doc

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Mon, 13 Feb 2012 17:03:04 -0500

I can confirm that the FreeSwitch integration will work with Asterisk 1.4.

Try a pay-as-you-go SIP provider like CallCentric (www.callcentric.com) for more certain results with your call center testing. You can get a test number with a bunch of inbound and outbound minutes for under $10. Just my 2 cents.

- Bill

Comment on "Using FreeSWITCH to add Google Voice to Asterisk"

Sat, 11 Feb 2012 23:55:14 -0500

Thanks for all the work on this!

I don't have the time nor inclination to try and even think I'm going to learn this stuff. (I'm a computer born and raised kid, but chose to go financial services!)

I am looking for an arrangement for my small office 2 and 1/2 people - don't ask! :) using the newly (for me) found ViciDial open source call center solution.

I believe I have seen enough to know I want to go with said solution - but I'd really like to take it for a test drive on the shoestring budget I have. And that includes setting up carriers.

I'm at the point where I'm ready to hire a freelancer or something just to get the server I got up and running to have some valid SIP trunks working so I can actually experience live in and outbound calls from the ViciDial software.

So since ViciDial uses Asterisk 1.4 (older version) - and everything I've read on Asterisk supporting Google Voice (not going to happen!) - I wondered if I could use your set up here to get a couple Google Voice SIP trunks as carrier lines for my ViciDial solution?

Any help is appreciated. As stated - I'm burning myself out trying to learn about what's available for a shoe string budget (and am completely humbled with awe as to the power of the open source stuff available) - but with everything I'm trying to do to keep my kiddies from starving - I'm unable to commit much more brain power in trying to tinker with these things myself. So unless there was some straight forward guide - I'd really just like to know if I actually put a few dollars out of pocket for a professional to set this up - will it have success in actually getting a live working SIP carrier phone line working with ViciDial?

Your help is GREATLY appreciated!

- Doc