Topic 'Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10' in forum 'VOIP Tech Chat' - dslreports.com

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 16 May 2015 23:53:35 EDT

twinclouds posted : Bill:
Thanks. You are right, it used to work but no longer. GV18585551212 works anyway.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 16 May 2015 19:15:22 EDT

phonesimon posted :

I am not sure I misunderstood you or not last time when you said I should use GV1NXXNXXXXXX as the peer account. This time you said it should be GV18005551212 which is the same as my GVGW login number. (I literary entered NXXNXXXXXX, rather than my GV number last time.) It appears both will work, though.

You didn't have "insecure=port,invite" in your peer definition, did you? That's the only way I can see it working, and it would not work consistently. If you had that, it would work like a regular type=peer, matching on IP address, and would sometimes fail when the registration changes.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 16 May 2015 19:05:53 EDT

twinclouds posted : Hi, Bill:
I am not sure I misunderstood you or not last time when you said I should use GV1NXXNXXXXXX as the peer account. This time you said it should be GV18005551212 which is the same as my GVGW login number. (I literary entered NXXNXXXXXX, rather than my GV number last time.) It appears both will work, though.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 16 May 2015 00:51:48 EDT

phonesimon posted : How-tos showing the type=friend setup for Asterisk sip.conf and FreePBX are now posted on our knowledge base at »support.simonics.com/sup ··· 00005974 .

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Tue, 12 May 2015 00:58:47 EDT

twinclouds posted : Hi, Simon:
Yes, after changes the peer name and the part in the extensions.conf correspondingly it works now. Thank you for your good work.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 11 May 2015 22:15:39 EDT

phonesimon posted : If you are using type=friend and the peer will be authenticating back to you (as in this case), then the section name must be the name that will authenticate.

Your peer should look like:

[GV1NXXNXXXXXX]
type=friend
username=GV1...

I am fighting another little problem that is causing some incoming calls not to get routed. After fixing that I'll get to the tutorial. Let me know if you get it working using the information here.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 11 May 2015 21:56:32 EDT

twinclouds posted : Simon:
I tried it but didn't work. I got the error:

[2015-05-11 18:37:02] NOTICE[4246][C-0000003e]: chan_sip.c:25528 handle_request_invite: Failed to authenticate device "Poway CA" ;tag=813928647

where 1858xxxxxxx is my calling phones caller ID.
I tried using my cell phone to make the call and the results are the same.
I put "match_auth_username=yes" in sip.conf's [general] section and "type=friend" is in the [gvgw] section. Hope these are correct.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 11 May 2015 09:47:51 EDT

phonesimon posted :

said by twinclouds:

There's nothing wrong with my asterisk client and Simon's server. It just the limitation of Asterisk. If one DNS has more than one real ip addresses, Asterisk can received the call from one of them only if (1) using allowguest and specify the DNS address, in this case, gvgw.simonics.com, in the context of the general section; or (2) To specify each one in a separate peer section. This is why localphone does not need to specify its DNS (only one ip address) in the general section and CC need to (many ip addresses). I had also tried taking off CC from the context in the general section. Once it was done, I cannot receive call to the CC number either. Asterisk CLI showed the same message as I got from gvgw.

I don't see why Asterisk cannot handle multiple ip addresses properly. I think it is something Asterisk should fix.

I released a new feature into production on the Google Voice Gateway this morning. It provides a third option, and is a far better solution than specifying peers for each proxy, so I will be retracting that suggestion.

News posting at »simonics.com/news/

Now the gateway will authenticate back to you if your PBX challenges it, which is what Asterisk will do if you configure allowguest=no and set up your peer as a type=friend.

There is a sip.conf setting needed to make it work: match_auth_username=yes. The reason is that by default Asterisk uses the From line as the username. We are putting the caller's number in the From line (as is typical) so that can't be used for user matching. By enabling that option in sip.conf, Asterisk will check the digest username instead.

The result is that you can define a single peer and even when Asterisk re-registers to a different proxy in our system, incoming calls will match the peer not based on IP but on authentication. This allows me to freely deploy more proxies as needed without major announcements of such and Asterisk to be properly secured if you don't want to permit anonymous SIP.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Tue, 21 Apr 2015 14:56:41 EDT

brg posted :

said by OzarkEdge:

Yes, but... I wonder why Google won't fix their 'Chat has gone missing, again' nonsense.

Couldn't agree more.

I forward GV to non-GV DIDs, so it's never been an issue..

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Tue, 21 Apr 2015 13:02:46 EDT

OzarkEdge posted :

said by brg:

But, dood, having Google Chat available and active is "Receiving Calls to a Directly-Registered-with-GV-Device 101".

Yes, but... I wonder why Google won't fix their 'Chat has gone missing, again' nonsense.

OE
--
My VoIP Project Notes

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Tue, 21 Apr 2015 12:33:53 EDT

brg posted :

said by N9MD:

Google Schmutz

Eww! You've got some on your chin!

But, dood, having Google Chat available and active is "Receiving Calls to a Directly-Registered-with-GV-Device 101". :)

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Tue, 21 Apr 2015 11:42:53 EDT

N9MD posted : May I publically offer my sincere thanks and appreciation to Bill Simon for his patience (and multiple back & forth PMs) dealing with my inability to make outbound calls thru the GVGW via a PAP2 (and ExpressTalk softphone). Confusion reigned in my attempts to deal with the inter-relationships among Google Voice, Google Chat, Google Mail, Google Hangouts, and Google Schmutz(?).

My devices would show successful registration ... but outgoing dialing would yield one ring followed by a busy signal.

With Bill's continuing suggestions and advice (and hours of Google research on my part), I finally realized that the Google Chat option was missing from the Google Voice phone settings (Call Forwarding) page ... because I had opted to install Hangouts (bad move) at some point in the process. So, I 'reverted' to old Google Chat ... which brought back the Google Chat forwarding option under the Google Voice phone settings ... made an outgoing call from Google Chat (or was it GMail ???) ... then re-registered my GV DID on the PAP2 (and ET softphone).

Success! Thanks, again, Bill!

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 20 Apr 2015 00:04:08 EDT

twinclouds posted : O.K. In that case, I am fine also when using my softphone and ATA. The problem was asterisk. It just stupid if more primitive ones can work but Asterisk cannot.
I use asterisk so I can have a few voip services as backup. For example, when Simon's server was down, mine worked fine with gvsip and localphone as backup.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:57:03 EDT

Davesworld posted : I'm using a Snom IP phone, you sound like you are using Asterisk thus playing with the big boys in comparison. The values in my web gui look different than they do in the xml config file in my case. The xml file shows user_host and user_outbound.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:52:09 EDT

twinclouds posted : Dave:
Let me make sure about what you did. Can you receive calls fine by specifying gvgw.simonics.com in context of the peer section [gvgw] only, but not in the context of the [general] section?

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:47:02 EDT

Davesworld posted : I noticed that I can get to either using the default gvgw.simonics.com. Not sure if this info contributes anything here though.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:43:52 EDT

twinclouds posted : Simon:
Thank you very much for looking into this. I tried this before and it worked. Actually, I used gvgw10.simonics.com and gvgw11.... instead of the two ip address and that worked as well. To make things simple, I right now simply using the CC approach by specify gvgw.simonics.com in the [general] context. I think that should be o.k. I don't think it will be that unsafe.
Asterisk really should fix this, if a soft phone can work fine, how difficult it could be?

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:32:22 EDT

Davesworld posted : Thanks! On my Snom M325, the default is 3600 seconds and on my firewall I set the parameters for nf_conntrack_sip at over 600 seconds to keep my Future Nine account alive as they send a keep alive signal or something like it every 500 seconds whereas without nf_conntrack_sip and a parameter, the firewall would time out at 180 seconds. My old Snom M9r could not maintain a connection any other way. On the new phone, CallCentric is not liking it with the default re-registration time of 3600 seconds.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:17:42 EDT

phonesimon posted : You don't need to explicitly set the timer unless you are seeing registration problems or need to set it low to maintain state through your NAT.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 23:09:37 EDT

Davesworld posted : What should we set our re-register timer at?

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 19 Apr 2015 22:51:35 EDT

phonesimon posted : I watched some registration activity and observed clients switching from one gateway to the other as they re-register at the prescribed interval. The clients look up the DNS record each time. So I can see how this is happening. Like CallCentric in the thread you linked, GVGW sends calls to the client from the proxy to which it registered. It's the exact same situation.

A solution for Asterisk might look like the following. First do a lookup of the A record for gvgw.simonics.com (subject to change!) :

$ dig +short gvgw.simonics.com. a45.55.163.124104.236.102.59

sip.conf:

[gvgw]; this is the peer we use for outgoing callshost=gvgw.simonics.comtype=peeruser=...secret=......context=gvgw [gvgw-in1]; set one up for each A record returned; not used for outgoing callshost=45.55.163.124type=peerinsecure=port,invitecontext=gvgw [gvgw-in2]; set one up for each A record returnedhost=104.236.102.59type=peerinsecure=port,invitecontext=gvgw [general]register => username:pass@gvgw.simonics.com/something-in-gvgw-context

now in extensions.conf:

[gvgw]exten => something-in-gvgw-context,1,...

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 18 Apr 2015 17:16:22 EDT

twinclouds posted : It's documented here: »issues.asterisk.org/jira ··· SK-21752
and discussed here: »Another asterisk question
Didn't see a elegant solution unless I missed something here.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Wed, 15 Apr 2015 12:47:53 EDT

twinclouds posted : Simon:
Thanks. I have also checked the Incredible PBX implementation. It has not problem to receive the call to the GV number. I tried to see how the freepbx specify the context in the [general] section, but I am not knowledgeable about asterisk to understand. However, if I commend out the context in the [general] Section, it will show "cannot find ... in the default context" and drop the call.
If you don't mind, maybe you can send me your peer section and the context in the [general] section in the sip.conf so I can see what's the differences. Either on this board or PM me will be fine.
Thank you for your effort on this. You are doing an important service to the VOIP community!

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Wed, 15 Apr 2015 10:18:27 EDT

phonesimon posted : Thanks for your testing. Oddly, in my environment (Asterisk 11 w/FreePBX) the peer and registrar always matched up when I specified "gvgw.simonics.com" in the peer definition and register statement, thus calls coming in to Asterisk would come from the peer it knows. Your testing shows that you may have registered with one IP of gvgw.simonics.com and the SIP peer was looked up as another IP, causing a mismatch.

The "CallCentric method" (define a SIP peer for each proxy's IP) would solve the problem here, or allowing SIP guests. Meanwhile I am trying to think of a more seamless way to get Asterisk to do what I originally had in mind, which is to register to gvgw.simonics.com, picking one of the IPs out of the DNS round-robin or SRV list, and also recognizing that as the peer. As mentioned before, calls will always be sent to the SIP client from the proxy to which it establishes registration.

I agree that this is something Asterisk could do better at. Simpler devices, such as ATAs and IP phones, seem to work just fine in the way described.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Wed, 15 Apr 2015 02:28:27 EDT

twinclouds posted : After a few days of digging, experimenting, and talking with people on the Asterisk forum, I think I got to the bottom of the issue.

There's nothing wrong with my asterisk client and Simon's server. It just the limitation of Asterisk. If one DNS has more than one real ip addresses, Asterisk can received the call from one of them only if (1) using allowguest and specify the DNS address, in this case, gvgw.simonics.com, in the context of the general section; or (2) To specify each one in a separate peer section. This is why localphone does not need to specify its DNS (only one ip address) in the general section and CC need to (many ip addresses). I had also tried taking off CC from the context in the general section. Once it was done, I cannot receive call to the CC number either. Asterisk CLI showed the same message as I got from gvgw.

I don't see why Asterisk cannot handle multiple ip addresses properly. I think it is something Asterisk should fix.

The discussion on the Asterisk Technical Support Forum can be found at: »forums.asterisk.org/view ··· &t=94233

Please let me know if I am wrong.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 13 Apr 2015 15:08:25 EDT

Graycode posted : Woo Hoo! My old (unlocked) SunRocket ATA is working well with this. It makes and receives calls better than I had hoped. Excellent :D

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 13 Apr 2015 14:15:59 EDT

carlm posted :

said by bitseeker:

Yes, an IPKall DID forwarded to CC works fine.

Beats me! Hopefully Bill can help you.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 13 Apr 2015 14:01:46 EDT

bitseeker posted :

said by carlm:

If you leave everything the same with your IP phone and with your CC account settings, and forward a DID to the same CC account, does that work?

Yes, an IPKall DID forwarded to CC works fine.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Mon, 13 Apr 2015 12:38:38 EDT

AllThumbs posted : Special pricing and One Click Installer for Incredible PBX for Asterisk-GUI are available on Nerd Vittles today.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 12 Apr 2015 20:11:18 EDT

graysonf posted : That doesn't answer my question.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 12 Apr 2015 20:08:28 EDT

Davesworld posted :

said by graysonf:

Is anyone able to FAX thru the gateway/OBi200 combination?

I've done dozens of faxes through the built in GV gateway. I don't see any reason to use Bill Simon's gateway on an OBI200 unless incoming CNAM is desired. It's heaven sent on a SIP phone or non OBI ata.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 12 Apr 2015 09:08:47 EDT

graysonf posted : Is anyone able to FAX thru the gateway/OBi200 combination?

I can't, the calls are dropped rather than handshaked.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sun, 12 Apr 2015 09:07:49 EDT

carlm posted : I just tried SIP forwarding from GVGW to CC again, both with and without having a SIP client registered to GVGW (which shouldn't make any difference), and it worked in both cases.

I assume you're working with Bill now to troubleshoot this, but I'll just ask one question:
If you leave everything the same with your IP phone and with your CC account settings, and forward a DID to the same CC account, does that work?

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 22:18:37 EDT

bitseeker posted :

said by phonesimon:

If you want to troubleshoot the SIP URI problem, open a ticket and I can grab a SIP trace and try to figure out what's going wrong.

OK, will do. Thanks, Simon. I also registered another GV number today thanks to the accidental extended special. :)

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 21:42:29 EDT

phonesimon posted :

said by bitseeker:

Next, tried logging out of GVGW and back in. Called my GV number. Same thing. IP phone was disconnected when picking up the call.

So, GVGW forwarding to CC SIP URI (in.callcentric.com) still not working for me.

There's no magic behind logging in and out of the portal. Whatever you see there is "live" when you see it.

If you want to troubleshoot the SIP URI problem, open a ticket and I can grab a SIP trace and try to figure out what's going wrong. Obviously lots of SIP URIs work just fine. Curious that CallCentric's wouldn't. If it were a home PBX or something, I'd be inclined to say it's a misconfiguration there, but I think we should be able to send a call to a CallCentric URI without any problem.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 21:19:21 EDT

bitseeker posted :

said by carlm:

I just added SIP forwarding from GVGW to my CC account and registered a softphone (Zoiper 2.0 Free, Windows) with my CC account.
I then logged out of the GVGW user portal (taoman54 mentioned a problem before that seemed to go away after a user portal login/logout -- possible user portal bug in updating settings??).
Finally I called my GV number from yet another softphone, answered the CC-registered softphone and had a brief conversation with myself.

I tried cycling the GV Offline/Online in the GVGW interface. Then, called my GV number. Disconnected when picking up the IP phone registered at CC.

Next, tried logging out of GVGW and back in. Called my GV number. Same thing. IP phone was disconnected when picking up the call.

So, GVGW forwarding to CC SIP URI (in.callcentric.com) still not working for me.

-----

Ultimately, I intend to register my IP phones directly to GVGW since I have plenty of available line buttons. But I figured I'd try to get the URI forwarding working in case anyone else runs into the same issue.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 16:27:58 EDT

brg posted : I've experienced clicking with both the Simon Gateway and the GVSip Gateway. I've very-very rarely (perhaps once or twice a year) heard the same thing when using my Obi100 for outbound GV calling. I =never= hear it on GV>Callcentric inbound calls.

I haven't yet experimented with inbound calls with either gateway because I am unlikely to use that capability much. My plan is to stick with forwarding GV calls to my Callcentric DID and my Cell. Bullet-proof; no need to change it unless Google does something at its end...

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 14:32:14 EDT

david3 posted : Has anybody noticed any call quality issues? I sometimes notice significant packet loss with choppy audio and periodic loud clicks and pops. When I forward the google voice calls through my Callcentric DID, the audio is excellent.

I've also tried setting up my own gateway to google voice with asterisk, and audio is also a little choppy, but without the loud clicks and pops at least. Forwarding to Callcentric always sounds significantly better, though.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 13:21:48 EDT

twinclouds posted :

said by Trev:

said by phonesimon:

The round-robin A will not be a problem and you do not need to specify all the IP addresses as peers. Callls will come in from the specific gateway server to which the client is registered. Is Asterisk looking up its peers separately from its registrars?

In this case, Asterisk users would probably fare better if they enable the DNS Manager. Look in /etc/asterisk/dnsmgr.conf and make sure enable=yes.

It was disabled. However, after I enabled it, still not working without specified in the general section. Actually, I think it is easiest by simply put it there (in [general]/context). There might be some minor risk but people do that often anyway, e.g., CC suggested to include it in the general section.
Since it works if I include it in the [general]/context, does this mean Asterisk can find it but somehow it just didn't looking for it in the [gvgw] and why?

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 11:59:41 EDT

phonesimon posted : Last chance at the $3.99 price. I forgot to switch the price tag this morning. Some of you have realized that. :-) Get the discounted price through the end of the day today.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 11:37:09 EDT

carlm posted : Normally I have only one connection with Bill Simon's GVGW: my Obi is registered to it.
For test purposes I added a second connection to GVGW: I told GVGW (by editing settings in the GVGW web user portal) to forward via SIP URI to my CC SIP account. Then, when I called my GV number, both the phone plugged into my Obi and the softphone on my Windows PC rang.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 11:19:03 EDT

taoman posted :

said by carlm:

I then logged out of the GVGW user portal (taoman54 mentioned a problem before that seemed to go away after a user portal login/logout -- possible user portal bug in updating settings??

I think there might be a possible bug in updating the portal settings. The test I did yesterday (and had you try also) of turning registration off on your UA device and then calling your GV number should not have worked as it did. The reason it did, I think, is because the change in registration status was not recognized by the GV gateway yet and your account was still logged into Chat. Therefore calls did not go to your GV XMPP trunk. They should have.

Apparently, as long as your GV gateway account remains logged into GV Chat all incoming calls to your GV number will be routed to the GV Gateway. If your GV gateway account is logged out (goes offline) from Google Chat incoming calls will revert to being routed to your GV XMPP trunk (if configured).

By the way, there is a much easier way to test this. Just log into the GV gateway web portal and click on the "offline" button. Incoming calls to your GV DID should immediately start routing to your XMPP trunk.

said by Simonics FAQ :

Why is my Google Voice (Chat) account logged in? Why is it logged out?

The Google Voice (Chat) account login is typically triggered by the availability or inavailability of a SIP route. In other words, if you have no devices registered and no SIP URI defined, an incoming call can't be routed through the Gateway. Thus, during this time, your account will be logged out. When you register a device or define a SIP URI, then a path becomes available for a Google Voice call and your account is logged in.

»support.simonics.com/sup ··· ged-out-

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 10:34:27 EDT

flinchlock posted :

said by carlm:

I normally just register my Obi202 with GVGW -- no SIP forwarding.
I just added SIP forwarding from GVGW to my CC account and registered a softphone (Zoiper 2.0 Free, Windows) with my CC account.

Here is hoping there is no such thing as a stupid question...

I have had a VOIP setup for years, but really no messing with any settings... it just works.

I am confused with the quoted two sentences. The 1st sentence is what you normally do, but you did the 2nd sentence instead?

Mike

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 08:54:09 EDT

carlm posted :

said by taoman:

said by bitseeker:

The IP phone registered to CC can't pick up the call. It just stops ringing as soon as the handset is picked up.

FYI: I switched my forwarding to my CC SIP URI address and also tried my CC iNum and got the same results you are experiencing using an OBi200 ATA. Quite odd.

This definitely does not happen when forwarding to VoIP.ms. It forwards as expected. When I pickup my OBi connected phone the call connects and my cell phone stops ringing.

I normally just register my Obi202 with GVGW -- no SIP forwarding.
I just added SIP forwarding from GVGW to my CC account and registered a softphone (Zoiper 2.0 Free, Windows) with my CC account.
I then logged out of the GVGW user portal (taoman54 mentioned a problem before that seemed to go away after a user portal login/logout -- possible user portal bug in updating settings??).
Finally I called my GV number from yet another softphone, answered the CC-registered softphone and had a brief conversation with myself. :-)

Edit: I used 'in.callcentric.com' in the SIP URI.

P.S. I have asked CC if they could pass on the caller name in the SIP INVITE that they are getting. That would be great! Free inbound, free CNAM and all of CC's call treatments.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 03:25:54 EDT

Trev posted :

said by phonesimon:

In this case, Asterisk users would probably fare better if they enable the DNS Manager. Look in /etc/asterisk/dnsmgr.conf and make sure enable=yes.
--
I represent AcroVoice, a full service Canadian VoIP Provider.
Buy your Obihai ATA or IP Phone shipped from within Canada.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 00:36:53 EDT

bitseeker posted :

said by taoman:

said by bitseeker:

The IP phone registered to CC can't pick up the call. It just stops ringing as soon as the handset is picked up.

Interesting. I hadn't tried VoIP.ms yet. Thanks for confirming the issue forwarding to CC. IPKall DID forwards to CC just fine, though.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 00:28:24 EDT

david3 posted :

said by phonesimon:

said by david3:

Thanks. I ran across that while I was searching for answers, too, and tried adding the "ignorecryptolifetime=yes" setting, but it didn't have any effect. I don't think that's the problem.

Update on this. It should help some other Asterisk users that may be having difficulty also. The incompatibility is interesting and difficult to solve and has to do with disagreements about how optional SRTP encryption should be handled. Asterisk does not accept it, and Yate, which runs our gateways, only offers it.

Since a number of GVGW users are on Asterisk, I've made the decision to disable SRTP until there is a clean solution. TLS transport is unaffected by the change and is still available.

That sounds about right. Thanks for looking into it.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 00:10:00 EDT

taoman posted :

said by bitseeker:

The IP phone registered to CC can't pick up the call. It just stops ringing as soon as the handset is picked up.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Sat, 11 Apr 2015 00:01:56 EDT

twinclouds posted : Simon:
I understand what you are saying. However, specify one server does not work for me. To make it work, I either have to specify it in the sip.conf/[general] section or provide multiple peer sections. I agree with you these are not good solutions but it is the problem that I encountered and cannot figure out how to make it work. Maybe you can give some suggestions that I can try?
The asterisk version that I am using is Asterisk 11, which should be a relative new one.
BTW, the only set up I encounter this problem is this asterisk. My softphone and ATA both work fine.

PS. I added srvlookup=yes in sip.conf. It still does not work. It should be enabled by default anyway.

Re: Biil Simon relaunches his Google Voice gateway; deal expires 4/10

Fri, 10 Apr 2015 23:19:18 EDT

phonesimon posted : The DNS settings for the gateway service provide both SRV records and round-robin A records. This should cover all standards-compliant software/equipment (that lookup SRV) and duds that only look up A records (old Asterisk).

The round-robin A will not be a problem and you do not need to specify all the IP addresses as peers. Callls will come in from the specific gateway server to which the client is registered. Is Asterisk looking up its peers separately from its registrars?

As the service grows, the SRV records and round-robin list will contain new hosts. I do not recommend pointing to an individual IP address from the list as they could change. Use gvgw.simonics.com.