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phonesimon
join:2014-10-08
Pennsylvania

phonesimon to twinclouds

Member

to twinclouds

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

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.
twinclouds
join:2010-06-12
San Diego, CA

1 edit

twinclouds

Member

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.
taoman
Premium Member
join:2013-09-13
Seattle, WA

taoman to bitseeker

Premium Member

to bitseeker
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.
david3
join:2000-03-21

david3 to phonesimon

Member

to phonesimon
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.

bitseeker
join:2014-03-05

bitseeker to taoman

Member

to taoman
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.

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

Trev
AcroVoice & DryVoIP Official Rep
Premium Member
join:2009-06-29
Victoria, BC

Trev to phonesimon

Premium Member

to phonesimon
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.
carlm
join:2014-09-29
united state

1 edit

carlm to taoman

Member

to taoman
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.

flinchlock
Premium Member
join:2003-04-25
Augusta, MI
ARRIS SB6121
Obihai OBi200

flinchlock

Premium Member

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
taoman
Premium Member
join:2013-09-13
Seattle, WA

taoman to carlm

Premium Member

to carlm
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-
carlm
join:2014-09-29
united state

carlm to flinchlock

Member

to flinchlock
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.
phonesimon
join:2014-10-08
Pennsylvania

phonesimon to WhyADuck

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to WhyADuck
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.
twinclouds
join:2010-06-12
San Diego, CA

twinclouds to Trev

Member

to Trev
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?
david3
join:2000-03-21

david3 to WhyADuck

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to WhyADuck
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.

brg
Premium Member
join:2001-01-03
Chicago, IL

brg

Premium Member

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...

bitseeker
join:2014-03-05

bitseeker to carlm

Member

to carlm
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.
phonesimon
join:2014-10-08
Pennsylvania

phonesimon

Member

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.

bitseeker
join:2014-03-05

bitseeker

Member

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.
carlm
join:2014-09-29
united state

carlm to bitseeker

Member

to bitseeker
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?

graysonf
MVM
join:1999-07-16
Fort Lauderdale, FL

graysonf to WhyADuck

MVM

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

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

Davesworld
join:2007-10-30
Thermal, CA

Davesworld

Member

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.

graysonf
MVM
join:1999-07-16
Fort Lauderdale, FL

graysonf

MVM

That doesn't answer my question.

AllThumbs
join:2006-02-07
Charleston, SC

AllThumbs to WhyADuck

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

bitseeker
join:2014-03-05

bitseeker to carlm

Member

to carlm
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.
carlm
join:2014-09-29
united state

carlm

Member

said by bitseeker:

Yes, an IPKall DID forwarded to CC works fine.

Beats me! Hopefully Bill can help you.
Graycode
join:2006-04-17

1 recommendation

Graycode to WhyADuck

Member

to WhyADuck
Woo Hoo! My old (unlocked) SunRocket ATA is working well with this. It makes and receives calls better than I had hoped. Excellent
twinclouds
join:2010-06-12
San Diego, CA

twinclouds to phonesimon

Member

to phonesimon
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.
phonesimon
join:2014-10-08
Pennsylvania

phonesimon

Member

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.
twinclouds
join:2010-06-12
San Diego, CA

twinclouds

Member

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!
twinclouds

twinclouds to phonesimon

Member

to phonesimon
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.
phonesimon
join:2014-10-08
Pennsylvania

phonesimon

Member

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. a
45.55.163.124
104.236.102.59
 

sip.conf:

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

now in extensions.conf:

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