Check the 'Save Settings' box above the Start/Stop button and a GVOut.ini file will be created so that your settings are remembered.anant wrote:How do I stp and start the GVOUT proxy server ? Is there any simpler way of reentering the data ?
New Toy: GVOut Google Voice Outgoing Proxy Dialer (Windows)
Moderators: Bill Smith, Pilot
I'm using an original PAP2 that also works perfectly in every other situation. So much for the thought of it being tomato related.anant wrote:I am using RTP300 router and PAP2T as ATA. Both have been working fine.
I've tried lots of combinations since this was posted on SlickDeals a couple of days ago and the PAP2 simply won't register through GVOut. I'd love to get it working also.
Anyone out here having success with a PAP2?
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cooldude929
- MagicJack User
- Posts: 37
- Joined: Thu Jun 11, 2009 8:47 pm
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jose.miami
- MagicJack User
- Posts: 45
- Joined: Sat Aug 02, 2008 3:27 am
[quote="anant"]Sto / start GVOUT and still it does not register PAP2T[/quote]
Some times you have to stop and start the proxy and then try making a call and see if the light is coming on on the ATA.
Also go to the Sipgate account and in the bottom of the page should be a link help_center, in the next page should be a list of devises select PAP2 and it should give you pics with you own PAP2 config double check every thing, I know yous reg withot the proxy but just to make sure everything is OK.
Also you may want to try it with xlite or other soft phone and see if the proxy is running OK in your computer.
Good luck.
Some times you have to stop and start the proxy and then try making a call and see if the light is coming on on the ATA.
Also go to the Sipgate account and in the bottom of the page should be a link help_center, in the next page should be a list of devises select PAP2 and it should give you pics with you own PAP2 config double check every thing, I know yous reg withot the proxy but just to make sure everything is OK.
Also you may want to try it with xlite or other soft phone and see if the proxy is running OK in your computer.
Good luck.
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jose.miami
- MagicJack User
- Posts: 45
- Joined: Sat Aug 02, 2008 3:27 am
[quote="JFMuggs"]Xlite works fine with a proxy of proxy01.sipphone.com (I can make a 3 minute G5/GV call). When I change the proxy to 127.0.0.1, registration fails with a 408 Request Timeout. At least things are consistent here.
[/quote]
Actually the IP address you listed is your Host IP address you shoul have put your computer own networking address it shoul be something like 192.168.168.xxx.
Actually the IP address you listed is your Host IP address you shoul have put your computer own networking address it shoul be something like 192.168.168.xxx.
I tried both with no difference. Since X-Lite is running on the same machine as GVOut, I believe 127.0.0.1 would be the correct way to get to it.jose.miami wrote:Actually the IP address you listed is your Host IP address you shoul have put your computer own networking address it shoul be something like 192.168.168.xxx.FMuggs wrote:Xlite works fine with a proxy of proxy01.sipphone.com (I can make a 3 minute G5/GV call). When I change the proxy to 127.0.0.1, registration fails with a 408 Request Timeout. At least things are consistent here.
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UncleRunkle
- magicJack Apprentice
- Posts: 27
- Joined: Tue Jun 09, 2009 11:20 am
I just ran GVOut on a different PC on the same LAN (X-Lite on 192.168.1.100 / GVOut on 192.168.1.101), but the result is the same: 408 Request Timeout. I even put 192.168.1.101 in the router's DMZ. Still 408 Request Timeout. But going back to proxy01.sipphone.com allows me to make 3 minute G5/GV calls (no incoming audio with DMZ set to 192.168.1.101 as the RTP packets are being sent to 192.168.1.101 instead of 192.168.1.100; removing DMZ corrects it).jose.miami wrote:Actually the IP address you listed is your Host IP address you shoul have put your computer own networking address it shoul be something like 192.168.168.xxx.JFMuggs wrote:Xlite works fine with a proxy of proxy01.sipphone.com (I can make a 3 minute G5/GV call). When I change the proxy to 127.0.0.1, registration fails with a 408 Request Timeout. At least things are consistent here.
This would seem to indicate the problem is not with the PAP2 (at least not unique to it).
Thank you! Results in a couple of minutes...az1324 wrote:Here is a debug version http://www.mediafire.com/?2hvjztmmkmj
Just run from command line as "gvout d"
There isn't a ton of debugging info because it's a simple program.
That was less than exciting.
When GVOut starts up, it reports:
Starting Proxy on port 5060
When X-Lite is started, GVOut reports:
Found Client on 192.168.1.101
or
Found Client on 127.0.0.1
That's the extent of what I can get out of it.
The results are the same whether GVOut is running on a different machine from X-Lite (192.168.1.101) or the same machine (192.168.1.100).
Does that give you any clues?
When GVOut starts up, it reports:
Starting Proxy on port 5060
When X-Lite is started, GVOut reports:
Found Client on 192.168.1.101
or
Found Client on 127.0.0.1
That's the extent of what I can get out of it.
The results are the same whether GVOut is running on a different machine from X-Lite (192.168.1.101) or the same machine (192.168.1.100).
Does that give you any clues?
I'm out of ideas again. Firewall is disabled. No port forwarding set in the router. DMZ in router set to 192.168.1.100 where both GVOut and X-Lite are located. Configured as your example (using proper uername, password, phone number, etc.).
Even weirder, I occasionally see:
Found Client on 198.65.166.131
which is proxy01.sipphone.com
By simply disabling the outbound proxy option in X-Lite I can call out through G5/GV perfectly (for 3 minutes).
Even weirder, I occasionally see:
Found Client on 198.65.166.131
which is proxy01.sipphone.com
By simply disabling the outbound proxy option in X-Lite I can call out through G5/GV perfectly (for 3 minutes).
Run wireshark and see what's happening.
Bugfix posted which might have affected a small number of users if you could get your ATA to register through the proxy and got busy signal but no incoming ring.
Bugfix posted which might have affected a small number of users if you could get your ATA to register through the proxy and got busy signal but no incoming ring.
Last edited by az1324 on Tue Aug 04, 2009 1:22 am, edited 1 time in total.
By using proxy01.sipphone.com as the SIP proxy and an outbound proxy of the PC where GVOut is running and an AuthID, I'm now able to get registered and place an outgoing call. The debug info from GVOut shows "GV Commands Sent" but never "GV Response ...".
Occasionally when I fire up GVOut, it reports that sipphone.com is the client found rather than my PAP2 and nothing works without restarting GVOut.
Maybe these are the result of the bug being worked on.
Occasionally when I fire up GVOut, it reports that sipphone.com is the client found rather than my PAP2 and nothing works without restarting GVOut.
Maybe these are the result of the bug being worked on.
I finally got GVOut working with Gizmo5.
On a PAP2, it requires the following:
Proxy Server : proxy01.sipphone.com
Outbound Proxy : <ip address of pc>:5060
Use Outbound Proxy : Yes
Use OB Proxy In Dialog : Yes
Use Auth ID : Yes
Auth ID: <same as User ID>
I've still had a little instabilty with registration, so the jury is still out on whether all the kinks are out and if this will prove to be reliable enough for daily use.
The enhanced debug output is a significant improvement. With it, I'm seeing the following during idle periods (phone on-hook):
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Each series of NOTIFY's come in fairly quick succession followed by the 408 Request Timout. It repeats every 10 - 15 seconds. Is this normal?
Thanks az1324 for all your efforts.
On a PAP2, it requires the following:
Proxy Server : proxy01.sipphone.com
Outbound Proxy : <ip address of pc>:5060
Use Outbound Proxy : Yes
Use OB Proxy In Dialog : Yes
Use Auth ID : Yes
Auth ID: <same as User ID>
I've still had a little instabilty with registration, so the jury is still out on whether all the kinks are out and if this will prove to be reliable enough for daily use.
The enhanced debug output is a significant improvement. With it, I'm seeing the following during idle periods (phone on-hook):
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Each series of NOTIFY's come in fairly quick succession followed by the 408 Request Timout. It repeats every 10 - 15 seconds. Is this normal?
Thanks az1324 for all your efforts.
GVOut passes everything on except when it intercepts an outgoing call. The NOTIFYs are probably not supported/expected by the server. So I think 408 is not a big deal unless it is affecting your functionality in some way.
section 3.2 of rfc 3265 says -
"If any non-SUBSCRIBE mechanisms are defined to create subscriptions,it is the responsibility of the parties defining those mechanisms to ensure that correlation of a NOTIFY message to the corresponding subscription is possible.
Designers of such mechanisms are also warned to make a distinction between sending a NOTIFY message to a
subscriber who is aware of the subscription, and sending a NOTIFY message to an unsuspecting node.
The latter behavior is invalid, and MUST receive a "481 Subscription does not exist" response (unless some other 400- or 500-class error code is more applicable), as described in section 3.2.4.
OK, I won't worry about it for now then.az1324 wrote:GVOut passes everything on except when it intercepts an outgoing call. The NOTIFYs are probably not supported/expected by the server. So I think 408 is not a big deal unless it is affecting your functionality in some way.
On another note...
Once a number is dialed, it appears GVOut immediately sends the call request to GV and at the same time issues a busy response to the SIP client. The caller must be very quick to hang up or the incoming callback from GV sees an off-hook (i.e. busy) condition and the GV call is abandoned. Can GVOut detect when the SIP client has gone back on-hook after receiving the busy generated by GVOut and defer sending the call request to GV until that time? If so, it would guarantee this race condition would never occur. If not, then maybe there should be a user settable parameter for a delay after the busy is issued before the call request request is sent to GV (default = 1000ms) to allow for hanging up. Then the user could at least tailor the delay to his own liking.
It is funny that you just arrived to the same set up I have been posting from the second page of this thread.JFMuggs wrote:I finally got GVOut working with Gizmo5.
On a PAP2, it requires the following:
Proxy Server : proxy01.sipphone.com
Outbound Proxy : <ip address of pc>:5060
Use Outbound Proxy : Yes
Use OB Proxy In Dialog : Yes
Use Auth ID : Yes
Auth ID: <same as User ID>
I've still had a little instabilty with registration, so the jury is still out on whether all the kinks are out and if this will prove to be reliable enough for daily use.
The enhanced debug output is a significant improvement. With it, I'm seeing the following during idle periods (phone on-hook):
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Client Sent: NOTIFY sip:192.168.1.100:5060 SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout
Each series of NOTIFY's come in fairly quick succession followed by the 408 Request Timout. It repeats every 10 - 15 seconds. Is this normal?
Thanks az1324 for all your efforts.
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screenname
- MagicJack Newbie
- Posts: 8
- Joined: Mon Aug 03, 2009 8:08 pm
I also see the same thing you are seeing. If I have the call ring my landline phone, the landline phone will ring and complete the call. For my Sunrocket, I have not been able to have it ring back no matter how fast I end the ringing.JFMuggs wrote:Once a number is dialed, it appears GVOut immediately sends the call request to GV and at the same time issues a busy response to the SIP client. The caller must be very quick to hang up or the incoming callback from GV sees an off-hook (i.e. busy) condition and the GV call is abandoned. Can GVOut detect when the SIP client has gone back on-hook after receiving the busy generated by GVOut and defer sending the call request to GV until that time? If so, it would guarantee this race condition would never occur. If not, then maybe there should be a user settable parameter for a delay after the busy is issued before the call request request is sent to GV (default = 1000ms) to allow for hanging up. Then the user could at least tailor the delay to his own liking.
As somone suggested, a version for DD-WRT would make this a perfect tool.
Thank you AZ1324 for writing and posting this tool.
Here are the setups pics for the PAP2T using SipGate, running GVOut Proxy in your computer I hope it help someone.
Note: I am using The IP address of the sipgate.com as the SIP Proxy for sipgate.com.










If you use GV just to make and receive calls and you don't want to get charge for any calls or you don't want to make any long distances calls. Use this dial plan, otherwise make your own.
Thank you for the Proxy az1324.
Note: I am using The IP address of the sipgate.com as the SIP Proxy for sipgate.com.










If you use GV just to make and receive calls and you don't want to get charge for any calls or you don't want to make any long distances calls. Use this dial plan, otherwise make your own.
Code: Select all
(1866[2-9]xxxxxxS0 |1877[2-9]xxxxxxS0 |1888[2-9]xxxxxxS0 |1800[2-9]xxxxxxS0 |<411:18004664411>S0 |911!|[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060>|011!)Thank you for the Proxy az1324.
Last edited by MagicDump on Fri Aug 07, 2009 5:35 pm, edited 6 times in total.
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didaboston
- magicJack Apprentice
- Posts: 18
- Joined: Sat Jul 18, 2009 10:47 pm
Yes it did not work when I try to test with the the new GVOut proxy that az1324 poster, but I when bag to the original and now is working fine.JFMuggs wrote:You last reported that yours had stopped registering. Is that still the case?Josemiami wrote:It is funny that you just arrived to the same set up I have been posting from the second page of this thread.
But you are right, some times it reg directly with the sipgate.com proxy and I have to stop it and start it again until it register with my computer IP address
It is strange but it happen to me also.
Ones you get it to reg it will work until you turn off the computer or ATA.
I am sure the bug will be fix by az1324.
Just cant wait for the Linux version for the DD-WRT.
This new version is working Great it registered every time, I tried differents delays and it registered every time.JFMuggs wrote:Is it not possible to detect when the client has gone on-hook from the busy and automatically start the GV call at that point?az1324 wrote:New version adds configurable delay and fixes crash bug under extended debugging.


My laptop, running Vista32.
Thank you again az1324.
For the people interested in trying a different soft phone this is a free software version of a free soft phone with 6 lines. I have tested this guy and the sound and connections are excellent I am not promoting anything because it is free.

This is the link for the web page:
http://www.nch.com.au/talk/index.html

This is the link for the web page:
http://www.nch.com.au/talk/index.html
Last edited by Josemiami on Tue Aug 04, 2009 10:23 pm, edited 1 time in total.
I'm very sorry if you fell I was being picky. That was not my intent at all and I very much appreciate all your efforts.az1324 wrote:OK for all you picky people there's now two options... you can either get the busy signal and use the delay, or you get no busy signal and GVOut will wait for you to hang up.
I think that's about as far as GVOut will go in terms of functionality. If you want more, it's better to install Asterisk.
I confess I'm not up on SIP protocol. I've been meaning to study the RFC, but I just haven't found the time. I understand the options are limited given the nature of the way Google Voice operates as a callback system.
From a user standpoint, it would be nice if GVOut detected a call being initiated, returned an immediate busy to the SIP client, waited for the SIP client to go back on-hook, and then initiated the Google Voice callback request. If this is possible, then no delays are necessary and GVOut is pretty much fool-proof.
If the SIP protocol, Google Voice callback interface, or whatever precludes allowing GVOut to sequence things in the manner just described, then the current delay option is the next best thing.
Again, I greatly appreciate your work and did not mean to offend you. I hope you will continue to develop GVOut as it appears to be the only alternative for people using ATA's that can't justify or don't have the ability to set up a full-blown Asterisk system.
Hi, I've never used the program but thought I would pass along this tip for anyone using a Linksys ATA:
Insert the following in your dialplan and you can make your Linksys ATA 100% independent of this program so you no longer need the outbound proxy settings:
<#3:>1[2-9]xx[2-9]xxxxxx<:@192.168.0.101:5060>
Note: Just replace the 192.168.0.101 with the ip address of the computer running the GVOut app. Then to dial a number just dial #3 1 areacode and number. Obviously you can change the #3 to something else that works better in your dialplan.
This should make the app more practical\reliable for Linksys ATA users.
Hope this helps..
Insert the following in your dialplan and you can make your Linksys ATA 100% independent of this program so you no longer need the outbound proxy settings:
<#3:>1[2-9]xx[2-9]xxxxxx<:@192.168.0.101:5060>
Note: Just replace the 192.168.0.101 with the ip address of the computer running the GVOut app. Then to dial a number just dial #3 1 areacode and number. Obviously you can change the #3 to something else that works better in your dialplan.
This should make the app more practical\reliable for Linksys ATA users.
Hope this helps..
It's ok I was just being playful.
The busy signal is generated by the ATA so at that time the communication is already done and there is no additional notifications that the phone is back on the hook. It could play a ringing tone but that would be confusing. There is a chance it could play a custom ringing tone but this might be limited by client's support for ringback. If you want to research it and make some suggestions feel free.
The busy signal is generated by the ATA so at that time the communication is already done and there is no additional notifications that the phone is back on the hook. It could play a ringing tone but that would be confusing. There is a chance it could play a custom ringing tone but this might be limited by client's support for ringback. If you want to research it and make some suggestions feel free.
lj wrote:Hi, I've never used the program but thought I would pass along this tip for anyone using a Linksys ATA:
Insert the following in your dialplan and you can make your Linksys ATA 100% independent of this program so you no longer need the outbound proxy settings:
<#3:>1[2-9]xx[2-9]xxxxxx<:@192.168.0.101:5060>
Note: Just replace the 192.168.0.101 with the ip address of the computer running the GVOut app. Then to dial a number just dial #3 1 areacode and number. Obviously you can change the #3 to something else that works better in your dialplan.
This should make the app more practical\reliable for Linksys ATA users.
Hope this helps..
Yes it works and I don't have to put the Outbound Proxy: any more in my PAP2T.
Here is the pic

And here is my dial plan, I am dialing from my cordlessphone with this dial plan you dont have to dial the #3 or the 1
I works perfect just dial the 10 digits number.
Code: Select all
(<411:18004664411>S0 |911!|[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060|011!)Thank you for the tip
Last edited by Josemiami on Tue Aug 04, 2009 9:16 pm, edited 4 times in total.
looks interesting
Hi lj
This looks interesting. I had problem all along to register PAP2T with proxy: sipgate.com and outboundproxy 192.168.16.xxx:5060 . I always got a dial tone with proxy: sipgate.com and the outboundproxy fileld left blank.
Now with your dial plan, PAP2 registers and get dial tone since outboundproxy field left blank. . I have incorporated the above outbound proxy into your OUTBOUND dial plan as suggested.
Now we have to make this work.
I have dialed: #3 1 (area code) ( number) . After a long pause I get a busy tone. How could I proceed further to make my calling number ring and my call go thru ?
Need further help. Thanks in anticipation
anant
This looks interesting. I had problem all along to register PAP2T with proxy: sipgate.com and outboundproxy 192.168.16.xxx:5060 . I always got a dial tone with proxy: sipgate.com and the outboundproxy fileld left blank.
Now with your dial plan, PAP2 registers and get dial tone since outboundproxy field left blank. . I have incorporated the above outbound proxy into your OUTBOUND dial plan as suggested.
Now we have to make this work.
I have dialed: #3 1 (area code) ( number) . After a long pause I get a busy tone. How could I proceed further to make my calling number ring and my call go thru ?
Need further help. Thanks in anticipation
anant
I was afraid that might be the case. Given that, the 'Delay' and 'Use Busy Signal' options may be the only alternatives.az1324 wrote:The busy signal is generated by the ATA so at that time the communication is already done and there is no additional notifications that the phone is back on the hook.
Thanks to you, we now have a way to initiate free calls through Google Voice from our ATA's. The procedure may be a little different from what we're used to, but it's a whole lot better than having to initiate calls from a web browser. My ATA is wired to Line 2 on all the phones in my home, so I'm quite pleased to have two-way Google Voice access on all of them now.
Thanks!
Re: looks interesting
Can you post your entire dial plan?anant wrote: I have dialed: #3 1 (area code) ( number) . After a long pause I get a busy tone. How could I proceed further to make my calling number ring and my call go thru ?
anant




