New Toy: GVOut Google Voice Outgoing Proxy Dialer (Windows)

Josemiami · Post by **Josemiami** » Sat Aug 08, 2009 12:20 am

Mini PAP2T dial plan tutorial:

(<:1>xxx xxx xxxxS0|<:011>[2-9]x.S0|*xx|1900xxxxxxx!)

(*xx|[3-9]11S0|[2-9]xx[2-9]xxxxxxS0)

(*xx|[1235-9]11!|411S0|[2-9]xx[2-9]xxxxxxS0)

([49]11S0|[2-9]xx[2-9]xxxxxxS0|011!)

(411S0|911!|[2-9]xx[2-9]xxxxxxS0|011!)

(: To begin with, the entire dial plan must be enclosed within a pair of brackets '()'. So, go ahead and jot down an open bracket

<:1>

This part of the dial plan is telling the PAP2 to replace any characters within the <> brackets before the :,with what's written after the :. So in this case, there isn't anything written before the :, yet 1 is written after the :. So it's telling the PAP2 to prefix all numbers that match the remainder of the dial plan.

Lets see the next part xxxx xxx xxxxSO|. In PAP2 dial plans, an 'x' represents any single number between 0 and 9. In the case of the above, any 10 individual numerical digits would match the dial plan.

S0|

S0 (S followed by the number 0) represents 'Straight Out'. So this part of the dial plan is saying to your PAP2 that should a person dial a sequence of keys that 'fit' the above portion of the dial plan, process the call immediately (i.e., without waiting for more digits to be pressed on the keypad).

Above mentioned dial plan allows me to dial any US and Canada number without dialing 1 in the begning.

|: The '|' in a dial plan is for starting new plan <:011>[2-9]x.S0.

011 is for international dialing pre fix. Let see how it is going to work.

Anything enclosed within '[]' brackets represents 1 number. [2-9]: Any single number from 2 to 9 inclusive (i.e., a 2,3,4,5,6,7,8 or a 9). In this case, the first digit in the country code we're calling

x: Any single digit from 0-9 inclusive

.: the Period at the end of the above sequence represents that the preceding digit can be repeated one or more times. In the above example, the preceding digit is an 'x' representing any single number from 0-9 inclusive. Therefore, by placing a period '.' after the 'x', the dial plan is allowing for any number to be processed one or more times.

So as an example, dialing 011441235567895 would satisfy the dial plan since it allows for '011' followed by any single digit between 2 and 9 inclusive (in this case, a '4'), followed by any combination of numbers 'x.'. Note: Even though the plan allows for an unlimited number of digits to be dialed given the period after the 'x', your VoIP service provider is expecting you to dial a certain number of digits maximum and may not know hot to handle your call if you keep pressing numbers beyond what they're expecting.

*xx: this next part of the dial plan allows me to use calling features on my phone such as *69 etc. The * represents the * key on your telephone while the x represents any number from 0 to 9 (as stated previously). So, this tells the PAP2 to allow me to dial the Star key followed by any two sequence of numbers. Note: Even though my dial plan allows for call feature management, my VoIP service provider must also allow for these features in order for them to work.

|: The '|' in a dial plan is for starting new plan 1900xxxxxxx!

1900xxxxxxx!: This last part of the dial plan is one that blocks access to certain numbers (The '!' denotes 'block access' to the preceding sequence of dialed numbers). So in this case, 1900 followed by any 7 digit numerical sequence is not allowed to be dialed using my PAP2 and is hence, blocked.

You can add some thing like this for the 411 or 911 calls

|<911:17804213333>S0

This component of the dial plan deals with handling of calls to emergency services (911). In this example, dialing '911' tells the PAP2 to dial '14161234567' transparently and send the call Straight Out (without delay).
Note: Call 911 and ask them about the 7 digit help line number from them.

|<411:14161234567>|

This one is for 411 access. Find the 411 local number for your area, as i know it is a paid service.

Josemiami · Post by **Josemiami** » Sat Aug 08, 2009 12:45 am

rmp25 wrote:Posted: Tue Aug 04, 2009 by tj
--------------------------------------------------------------------------------
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.

The only reason I originally specified the #3 is so it doesn't interfere with the existing dialplan. If you don't have any other entries in the dialplan above you can't use your existing provider if needed. For some people this is fine but I just wanted to give an example to make it 100% independent and make it a choice(by dialing #3) to use the GVOut app on demand basically. So if the computer running GVOut is off the Linksys ATA still can place outbound calls if needed through the existing provider.
--------------------------------------------------------------------------------------

Hi all, great forum!

Regard to the above qoutes: in my understanding in PAP2T on one line you can use one proxy only. So the idea to use GVOut on demand is great, but after GVOut stop runing you can place free 3 min call thru G5 only, not to any other provider.

I was not able to implement this (GVOut on - callback, GVOut off - call thru G5) with any dialplan I tried using <#3:>, anybody?

In order for you to make your 3 min free calling using G5, you have to register your GV with G5 and give them your user account and your GV password which I will never recommend for the 3 min free calling, that will give them the right to get your money from your GV account if you got any and if you own them any.

Just my advise.

JFMuggs · Post by **JFMuggs** » Sat Aug 08, 2009 1:32 am

rmp25 wrote:Regard to the above qoutes: in my understanding in PAP2T on one line you can use one proxy only. So the idea to use GVOut on demand is great, but after GVOut stop runing you can place free 3 min call thru G5 only, not to any other provider.

I was not able to implement this (GVOut on - callback, GVOut off - call thru G5) with any dialplan I tried using <#3:>, anybody?

There can be only one default proxy per PAP2 line, but each dial plan term can specify an alternate proxy (using an '@' argument) to use instead of the default proxy when that term matches.

I think you misread the previous poster's remarks. His dialplan doesn't implement an automatic fallback if GVOut is not running. He only showed a fragment of his dialplan which normally uses some provicer other than Gizmo5 and/or Google Voice. The fragment he listed allows him to override his normal provider by prefixing the call with '#3'.

Here's the dialplan I'm currently using:

Code: Select all

(<411:18004664411>S0 |<:1>747xxxxxxxS0 |1747xxxxxxxS0 |<:1>800[2-9]xxxxxxS0 |1800[2-9]xxxxxxS0 |<:1>888[2-9]xxxxxxS0 |1888[2-9]xxxxxxS0 |<:1>877[2-9]xxxxxxS0 |1877[2-9]xxxxxxS0 |<:1>866[2-9]xxxxxxS0 |1866[2-9]xxxxxxS0 |<:1aaa>[2-9]xxxxxx<:@192.168.1.100>|<:1>[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |1[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |011xx.<:@192.168.1.100>|<#:1aaa>[2-9]xxxxxx|<#:1>[2-9]xx[2-9]xxxxxxS0 |<#:>1[2-9]xx[2-9]xxxxxxS0 |*xx.<:@sipbroker.com>)

Replace aaa (2 places) with your local area code and replace 192.168.1.100 with GVOut's IP address.

This dialplan supports:

A. 7, 10, and 11 digit calling to US destinations using Google Voice callback (free)

B. 011 + country code + number calling to international destinations using Google Voice callback (billable)

C. 7, 10, and 11 digit calling to US destinations using Gizmo5 by prefixing a # (free / limited to 3 minutes)

D. Toll-free number calling using Gizmo5 (free)

E. Gizmo5 number calling (1747xxxxxxx) (with or without the leading 1) (free)

F. 411 directory assistance using Gizmo5 to call Google 411 (free)

G. *Sip-Code + number calling to other networks via SipBroker : http://www.sipbroker.com/ (free)

JFMuggs · Post by **JFMuggs** » Sat Aug 08, 2009 1:37 am

Josemiami wrote:In order for you to make your 3 min free calling using G5, you have to register your GV with G5 and give them your user account and your GV password which I will never recommend for the 3 min free calling, that will give them the right to get your money from your GV account if you got any and if you own them any.

You no longer have to give them your Google Voice username and password. The calls no longer go through Google Voice but are actually handled by Gizmo5, spoofing your Google Voice CallerID. If you have no Gizmo5 credit, you get free 3 minute calls. If you have Gizmo5 credit, all minutes are deducted from your credit.

Josemiami · Post by **Josemiami** » Sat Aug 08, 2009 12:42 pm

Voxox Proxy settings.

user: US phone number (xxxxxxxxxx)
password : signup password
server: sip01-west.voxox.com

aakimenko · Post by **aakimenko** » Sat Aug 08, 2009 3:08 pm

Does anybody know if you can make a dial plan like that with Handytone? (I use HT502)

Code:
(1747xxxxxxxS0 |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!)

Thanks!

rmp25 · Post by **rmp25** » Sat Aug 08, 2009 6:09 pm

@JFMuggs

Thanks for the reply.

I've inserted tj's fragment in different dialplans and it did not work for me.
I tried your dialplan, made the necessary changes, and it does not work either, complete silence after placing a call (GVOut works fine with most plans).

In my understanding this plan allows you to override sipbroker with <#:> and call out using GVOut/G5 callback for free? Am I right?
If I'm not using sipbroker, do I have just to replace "@sipbroker.com>)[/code]" with "@anyprovider.com)?
Would you please show an example?

Thanks.

JFMuggs · Post by **JFMuggs** » Sat Aug 08, 2009 7:15 pm

rmp25 wrote:I tried your dialplan, made the necessary changes, and it does not work either, complete silence after placing a call (GVOut works fine with most plans).

In my understanding this plan allows you to override sipbroker with <#:> and call out using GVOut/G5 callback for free? Am I right?
If I'm not using sipbroker, do I have just to replace "@sipbroker.com>)[/code]" with "@anyprovider.com)?

Make sure you don't have an Outbound Proxy set/enabled in your PAP2. Make sure your Proxy is set to proxy01.sipphone.com and you have the correct username (1747xxxxxxx) and password set. Make sure the PAP2 has successfully connected/registered with Gizmo5 (the appropriate Line LED should be lit). You can't proceed without being connected/registered with Gizmo5.

Now, make sure you can place calls through Gizmo5 using this simple dialplan:

Code: Select all

(xx.)

Dial: 1 + area code + number

This should give you free three-minute calls using Gizmo5. Google Voice is not involved.

You can't proceed until calling with this dialplan works.

Next, make sure you can place calls using Google Voice callback using this simple dialplan:

Code: Select all

(xx.<:@192.168.1.100>)

Replace 192.168.1.100 with GVOut's IP address.

Dial: 1 + area code + number

You should get an immediate busy, you hang up, Google Voice calls you back, you answer, and the number you dialed rings and answers.

You can't proceed until calling with this dialplan works.

At this point, you should be able to use the dialplan I previously posted. It works as follows:

Normal dialing (7 digits for a local number / area code + number or 1 + area code + number for any number) will go via Google Voice callback (busy, hang up, receive callback).

To call direct using Gizmo5 instead (free, but three minute limit), dial a # before the number. Be patient as Gizmo5's setup time is often lengthy.

Toll-free numbers (800/888/877/866) will always go via Gizmo5.

You can also call other Gizmo5 users (1747xxxxxxx).

You can also dial 411 for directory assistance.

You can also call people on many other VoIP networks for free using SipBroker. For example, if you have a friend who is a Callcentric subscriber, you can call him for free without using Gizmo5 or Google Voice by dialing *462 + his_number.

fixup · Post by **fixup** » Sat Aug 08, 2009 11:33 pm

First thank az1324 for this great program!

I registered to this forum just to report a bug to az1324:

If I use a port other than 5060 (e.g. 5061), then GVOut freezes after a call or two. If I don't use busy signal, then it is fine but difficult to use this utility.

I cannot use port 5060 because another program is using it.

Thank you very much in adavance for fixing this bug at your convenience.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 12:00 am

fixup wrote:If I use a port other than 5060 (e.g. 5061), then GVOut freezes after a call or two. If I don't use busy signal, then it is fine but difficult to use this utility.

I can reproduce the problem here also. Using port 5062, GVOut works one time only and must be restarted.

az1324 · Post by **az1324** » Sun Aug 09, 2009 12:49 am

Are you trying to use another port with the dialplan method or as a proxy? Because you can't choose another port and then use it as a proxy unless the SIP server will allow you to connect on that port. With the dialplan method it should work on a different port.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 1:17 am

az1324 wrote:Are you trying to use another port with the dialplan method or as a proxy? Because you can't choose another port and then use it as a proxy unless the SIP server will allow you to connect on that port. With the dialplan method it should work on a different port.

With GVOut out of the loop, proxy01.sipphone.com works on port 5062 without any problems. Then using @192.168.1.100:5062 in the dialplan, GVOut (also configured for port 5062) works once and that's it. With debug enabled, everything looks normal on the first call which works. After that, there's nothing else displayed.

fixup · Post by **fixup** » Sun Aug 09, 2009 3:34 am

az1324 wrote:Are you trying to use another port with the dialplan method or as a proxy? Because you can't choose another port and then use it as a proxy unless the SIP server will allow you to connect on that port. With the dialplan method it should work on a different port.

I meant GVOut was set to !5060.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 6:07 am

az1324,

The problem with GVOut failing on ports other than 5060 appears to be related to it receiving an unexpected SIP cancel.

When using port 5060, SIP cancel's only show up in the debug display when 'Use Busy Signal' is NOT checked:

Starting Proxy on port 5060
Client connection from 192.168.1.145
Client Sent: INVITE sip:[email protected] SIP/2.0

Outgoing Call Detected
Client Sent: CANCEL sip:[email protected] SIP/2.0
Initiating Call to 1xxxxxxxxxx
_rnr_se={data deleted}=
GV Commands Sent
GV Response: {"ok":true,"data":{"code":0}}

Client Sent: CANCEL sip:[email protected] SIP/2.0
Client Sent: CANCEL sip:[email protected] SIP/2.0
Client Sent: CANCEL sip:[email protected] SIP/2.0
Client Sent: CANCEL sip:[email protected] SIP/2.0
Client Sent: CANCEL sip:[email protected] SIP/2.0
Server Sent: SIP/2.0 408 Request Timeout

It appears you wait for the SIP cancel to know when the caller has hung up, but you don't send an acknowledgement back to keep the PAP2 happy that it was received, so the PAP2 resends it 5 times and finally times out. When the port number is 5060, this isn't fatal (but it certainly isn't nice).

When the port number being used is other than 5060, SIP cancel's show up in the debug display whether 'Use Busy Signal' is checked or not. When it's checked, it's at end of call as you'd expect, but for some reason these don't display when the port is 5060 (maybe you have an inconsistency in your debug display between the two modes?). When the port number being used is other than 5060, the receipt of an unexpected SIP cancel kills GVOut until it's terminated and restarted:

Starting Proxy on port 5062
Client connection from 192.168.1.145
Client Sent: CANCEL sip:[email protected]:5062 SIP/2.0

This particular example was when GVOut was restarted after having died, but the PAP2 was still resending SIP cancel's from the previous call. As soon as GVOut gets an unexpected SIP cancel under any condition when the port number is other than 5060, GVOut becomes non-responsive. In the case where GVOut starts up with no lingering SIP cancels, it works for one call, but the SIP cancel at the end of that call causes it to become non-responsive until it's terminated and restarted.

j-rad · Post by **j-rad** » Sun Aug 09, 2009 10:02 am

just set this up with eyeBeam and GVOut, but I'm having a problem.. it seems that i can only place one call, then any other calls i try placing/receiving dont go through. i have to shut down eyeBeam + GVOut, open them again, then i can place another call.

any idea what im doing wrong?

gm_w · Post by **gm_w** » Sun Aug 09, 2009 10:34 am

tillithz · Post by **tillithz** » Sun Aug 09, 2009 1:19 pm

Ok, I have been tinkering for some time with this. I can get things to work with dialplan including the proxy to local computer. I can also get it to work briefly if I use a dialplan without the proxy in it and then use outbound proxy as the local pc with GVOut running.

Something weird is that after getting registered It is registered for 1 second then seems to register again. after about 30 seconds of this 1 second registering it fails and wont work until I shut down GVOut and reboot the ata device. I tried changing the delay on GVOut to say 10000 and I get failed registration. So I cant see if the 1 second is due to GVOut or what. Also, with the outbound proxy disabled and dialplan containing the local pc with GVOut running, it works but some time later I lose the ability to hear when making a call out, it performs the google voice call and ring back but no audio coming or going and no ringing sound when picking up the return call from google voice.

Here is some screen shots, please give feedback as I am lost now at how to get things working continuously. Thanks in advance for any replys

---
[url]---

rmp25 · Post by **rmp25** » Sun Aug 09, 2009 2:25 pm

@tillithz

May be because your Line 1 is disabled?

Josemiami · Post by **Josemiami** » Sun Aug 09, 2009 2:48 pm

tillithz wrote:Ok, I have been tinkering for some time with this. I can get things to work with dialplan including the proxy to local computer. I can also get it to work briefly if I use a dialplan without the proxy in it and then use outbound proxy as the local pc with GVOut running.

Something weird is that after getting registered It is registered for 1 second then seems to register again. after about 30 seconds of this 1 second registering it fails and wont work until I shut down GVOut and reboot the ata device. I tried changing the delay on GVOut to say 10000 and I get failed registration. So I cant see if the 1 second is due to GVOut or what. Also, with the outbound proxy disabled and dialplan containing the local pc with GVOut running, it works but some time later I lose the ability to hear when making a call out, it performs the google voice call and ring back but no audio coming or going and no ringing sound when picking up the return call from google voice.

Here is some screen shots, please give feedback as I am lost now at how to get things working continuously. Thanks in advance for any replys

There are some things wrong in your setup.

I suggest you go to page 6 of this post and check the pics setup for sipgate in PAP2T.

1. In yout GVout setup just put the user name not @email, of your GV account.
2.In your ATA set Use Auth ID to no.
3. Set Use Outbound Proxy to NO.
4.Use OB Proxy: No
5.Include in your dial plan This:

Code: Select all

[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060

Where the IP address will be the address of the computer running the Pro
xy.

Correcting your dial plan will be very important, for the proxy to work.

And is very obvious you have to enable your line 1

JFMuggs have and excellent example of a dial plan on page 14 of this post.

Good luck

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 4:13 pm

j-rad wrote:just set this up with eyeBeam and GVOut, but I'm having a problem.. it seems that i can only place one call, then any other calls i try placing/receiving dont go through. i have to shut down eyeBeam + GVOut, open them again, then i can place another call.

any idea what im doing wrong?

Are you using a SIP port other than 5060? There appears to be a bug in GVOut that causes the problem you describe unless you use port 5060. Please see my post that's just before yours.

rmp25 · Post by **rmp25** » Sun Aug 09, 2009 4:20 pm

@JFMuggs

Your dialplan, which did not work yesterday, works just fine today without any correction, weird. I can call out using GVOut/G5, or dialing "#" and a phone number, I can call out using G5's 3 free minutes.

The next thing I'd like to accomplish is to get an ability to call out on the same line using any other provider then G5, based on this qoute from your early post:

"There can be only one default proxy per PAP2 line, but each dial plan term can specify an alternate proxy (using an '@' argument) to use instead of the default proxy when that term matches".

I tried to replace sipbroker.com with myuserid@12voip (I'm using 12voip.com on line 2 of my PAP2T), call goes thru, but same G5's 3 free min.

Would you please to follow this further.

Thanks.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 4:29 pm

gm_w wrote:Did gizmo5 terminate even the 3 min dialing support? I am using the dial plan below and when I dial # + 10 digit number, I get gizmo's sorry message asking to buy gizmo credit

(<411:18004664411>S0 |<:1>747xxxxxxxS0 |1747xxxxxxxS0 |<:1>800[2-9]xxxxxxS0 |1800[2-9]xxxxxxS0 |<:1>888[2-9]xxxxxxS0 |1888[2-9]xxxxxxS0 |<:1>877[2-9]xxxxxxS0 |1877[2-9]xxxxxxS0 |<:1>866[2-9]xxxxxxS0 |1866[2-9]xxxxxxS0 |<:1aaa>[2-9]xxxxxx<:@192.168.1.100>|<:1>[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |1[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |011xx.<:@192.168.1.100>|<#:1aaa>[2-9]xxxxxx|<#:1>[2-9]xx[2-9]xxxxxxS0 |<#:>1[2-9]xx[2-9]xxxxxxS0 |*xx.<:@sipbroker.com>)

I just checked and Gizmo5 is still allowing free 3 minute calls. Nothing apprears to have changed. I'm able to call using both Google Voice callback (without a leading #) and Gizmo5 direct (with a leading #). I have no Gizmo5 credit.

I would recommend you read the message I posted on Sat Aug 08, 2009 4:15 pm and follow the recommendations there to try to isolate where your problem is.

Also, don't forget to replace aaa and 192.168.1.100 in the above dialplan as necessary (neither of these are the cause of your current problem if you're dialing # + 10-digit numbers).

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 4:45 pm

Josemiami wrote:1. In yout GVout setup just put the user name not @email, of your GV account.

My Google Voice account username is an email address. If I don't use that whole email address as the username in GVOut, I don't get a Google Voice callback. You MAY get away with leaving the @domain off if your username email address is @gmail.com, but you really should use the exact same username that you log into your Google Voice account with.

tillithz · Post by **tillithz** » Sun Aug 09, 2009 5:05 pm

yes the line is enabled normally. thats how it actually works, sorry the screen shot shows it as no but that was just at a time I disabled it. Also, I set the time correctly. Email address IS the username for googlevoice. I edited my actual email address in the pic but again, it is an email address.

In addition I have tried both ways, putting the proxy in the dialplan and setting outbound proxy to no, as well as not including it in the dialplan and setting outbound proxy and setting it to yes. Whats even weirder is at the time of this posting, I was able to make a call and it worked properly. then after about 30 seconds it stopped, though i can make the outgoing call its just when google voice calls me back theres no audio coming or going. BUT I just picked up the phone right now after leaving it for sometime (like 2 hours) and it is working again properly. I am stumped. I did however notice that when it doesnt work, there is a much faster busy signal then when it does work.

I will try the:

2.In your ATA set Use Auth ID to no.

and the dialplans that I had used were from that page, thats why you can see where I posted it in the pictures. I set my computers to static so I am certain of there being no issue with that regard, maybe its 64bit os? id doubt it though.

thanks for the input

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 5:16 pm

tillithz wrote:Here is some screen shots, please give feedback as I am lost now at how to get things working continuously.

1. Line Enable must be set to yes.

2. Outbound Proxy must be left blank.

3. Use Outbound Proxy and Use OB Proxy In Dialog must be set to no.

4. Use Auth ID and Auth ID probably aren't required, but they shouldn't do any harm.

5. Your GVOut settings look reasonable.

You must get your ATA successfully connected/registered with your SipGate account and able to receive calls on your SipGate DID. You cannot proceed until this is working.

Next, make sure you can place calls using Google Voice callback using this simple dialplan:

Code: Select all

(xx.<:@192.168.1.100>)

Replace 192.168.1.100 with GVOut's IP address.

Dial: 1 + area code + number

You should get an immediate busy, you hang up, Google Voice calls you back, you answer, and the number you dialed rings and answers.

You can't proceed until calling with this dialplan works.

At this point, you can use a fancier dialplan if you wish.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 5:42 pm

rmp25 wrote:@JFMuggs

Your dialplan, which did not work yesterday, works just fine today without any correction, weird. I can call out using GVOut/G5, or dialing "#" and a phone number, I can call out using G5's 3 free minutes.

The next thing I'd like to accomplish is to get an ability to call out on the same line using any other provider then G5, based on this qoute from your early post:

"There can be only one default proxy per PAP2 line, but each dial plan term can specify an alternate proxy (using an '@' argument) to use instead of the default proxy when that term matches".

I tried to replace sipbroker.com with myuserid@12voip (I'm using 12voip.com on line 2 of my PAP2T), call goes thru, but same G5's 3 free min.

Would you please to follow this further.

You will need to get very familiar with PAP2 dialplan syntax before you can tackle this. I would suggest you do some Googling and find a PAP2 Administrator's Guide and study its section on dialplans as a starter. Dialplans sre not all that complicated, but they're probably not all that obvious at first glance.

The simple change you tried above won't work because I'm sure sure 12voip.com requires a password in addition to a username to place calls (otherwise, anyone could use your 12voip account to place calls at your expense). I don't recall if the PAP2 supports passwords in the dialplan. Also, you need to enter * + number to have what was the SipBroker term match.

Bottom line is you need to educate yourself on PAP2 dialplan syntax in order to attempt the change you desire.

Josemiami · Post by **Josemiami** » Sun Aug 09, 2009 7:48 pm

Read This:

A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).

tillithz · Post by **tillithz** » Sun Aug 09, 2009 8:15 pm

ok sorry for the confusion. ONLY in THAT picture was line set to no. I have it WORKING for everything else EXCEPT googlevoice about 50% of the time. theres NO way I would get calls or make calls if it was set to no. And it randomly seems to work with NO changes initiated by me.

Like I said I have tried it both ways

1. With the outbound proxy/dialog set to no and dialplan of:

Code: Select all

(<411:[email protected]>S0 |<:1510>[2-9]xxxxxx<:@192.168.2.101>|<:1>[2-9]xx[2-9]xxxxxx<:@192.168.2.101>S0 |1[2-9]xx[2-9]xxxxxx<:@192.168.2.101>S0 |011xx.<:@192.168.2.101>)

which I got from page 14 I believe.

-OR the other way-

2. With outbound proxy/dialog set to yes and local pc with GVOut running set in the outbound proxy box with/without the port. Dialplan used at thta time was:

Code: Select all

(<411:18004664411>S0 |<:1510>[2-9]xxxxxx|<:1>[2-9]xx[2-9]xxxxxxS0 |1[2-9]xx[2-9]xxxxxxS0 |011xx.)

which I got from page 8 of this thread.

--------------
I suspect that there may be a setting on another page like regional or sip that is causing the problem. Or that my router may be blocking the traffic some how. Or better yet theres a problem with googlevoice. I think the test to determine this is when I make a call, get busy signal, hang up and get return call from google voice, and then no audio either way, to try initiating the call from google voice and my web browser. Also, I will try putting the spa2102 directly to the cable modem instead of my router/switch.

A side question probably would be, there are some VIA settings under my SIP tab, any idea what those are specifically?

I do proceed because it does work set up either of these ways. BUT I DONT GET AUDIO IN/OUT SOMETIMES when ONLY googlevoice calls me back.

tillithz · Post by **tillithz** » Sun Aug 09, 2009 8:25 pm

Ok, I just tried using phone to make outbound call. I get dialtone, I dial, I get busy signal, I hang up. Few seconds later I get the phone ringing, I pick up and NO audio, no ringing, but my cell phone rings and I pick it up, BUT theres no audio on either phone. so I hang up both phones.

I then immediately log into my google voice via web browser and initiate a call from the call button option, calling the same number and my phone rings, I pick up and then I hear the ringing of my cell phone. I pick up and the call works as it should, audio in both directions.

So now Im left to believe theres a configuration issue with the ata, router blocking traffic, or issue with GVOut setup. Does this shed any light on things for anyone? thanks for the help.

I dmz'd the ata device and got audio back, will see if this lasts

Josemiami · Post by **Josemiami** » Sun Aug 09, 2009 8:36 pm

JFMuggs wrote:
Josemiami wrote:1. In yout GVout setup just put the user name not @email, of your GV account.
My Google Voice account username is an email address. If I don't use that whole email address as the username in GVOut, I don't get a Google Voice callback. You MAY get away with leaving the @domain off if your username email address is @gmail.com, but you really should use the exact same username that you log into your Google Voice account with.

Yes I realize you can have a GV number register with with a non Gmail account, but you can always log in into your GV account setup with a gmail without using the @gmail, I have tried with all my GV voice Gmail accounts and thy will all work fine no need to enter @gmail. Just the user name will do.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 8:44 pm

tillithz wrote:I dmz'd the ata device and got audio back, will see if this lasts

Audio problems are almost always the result of router/routing issues (GVOut doesn't get involved in the audio side of things). The fact that using a DMZ fixed your problem pretty much confirms it in your case.

DMZ is overkill and precludes using other applications that also need proper routing. A much better option is to simply forward RTP ports to your ATA. In the PAP2 you can find the range of RTP ports used on the SIP tab:

RTP Port Min: 16384 RTP Port Max: 16482

Set up a port forwarding entry in your router to forward this range of ports to your ATA's IP address and you should still work without the need for DMZ. Keep the STUN server enabled also so the ATA can accurately determine its public IP address.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 8:53 pm

Josemiami wrote:Yes I realize you can have a GV number register with with a non Gmail account, but you can always log in into your GV account setup with a gmail without using the @gmail, I have tried with all my GV voice Gmail accounts and thy will all work fine no need to enter @gmail. Just the user name will do.

There's a lot of us who don't have a GMail account and don't want one. All of my dealings with various Google accounts is with a non-GMail email address. So telling people to unconditionally leave the @domain off of their username will cause those who do NOT use GMail to fail.

You have to be careful about giving instructions to people when those instructions only apply to people in your particular situation.

tillithz · Post by **tillithz** » Sun Aug 09, 2009 8:57 pm

right on, thank you. Yup, it seems to be working more constantly now. I will take your advice and set up the port forwarding in the router.

I think the real benefit here with GVOut is that though you have to have this running on a computer you can be able to dial 7 digit local numbers where as magicjack forces (through the use of the dongle) to dial the full number with a 1 I think. That is just horrible. afterall, we are still having to run something on the computer but least this way the user has way more control over the line when using an ata device. Great app by the way, I know I havent mentioned it as I was concentrating on setting it up.

thanks again

Josemiami · Post by **Josemiami** » Sun Aug 09, 2009 8:59 pm

tillithz wrote:Ok, I just tried using phone to make outbound call. I get dialtone, I dial, I get busy signal, I hang up. Few seconds later I get the phone ringing, I pick up and NO audio, no ringing, but my cell phone rings and I pick it up, BUT theres no audio on either phone. so I hang up both phones.

I then immediately log into my google voice via web browser and initiate a call from the call button option, calling the same number and my phone rings, I pick up and then I hear the ringing of my cell phone. I pick up and the call works as it should, audio in both directions.

So now Im left to believe theres a configuration issue with the ata, router blocking traffic, or issue with GVOut setup. Does this shed any light on things for anyone? thanks for the help.

I dmz'd the ata device and got audio back, will see if this lasts

stun01.sipphone.com:3478
stun.sipgate.net:10000

You can try either one of this stun services,

Also make sure to include the port 5060 in the SIP call address.

Code: Select all

[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060>|

And check your Nat Support Parameters.

JFMuggs · Post by **JFMuggs** » Sun Aug 09, 2009 9:04 pm

Josemiami wrote:Also make sure to include the port 506 in the SIP call address.
Code: Select all
[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060>|

You only need to explicitly include a SIP port if you're using something other than the default 5060.

Josemiami · Post by **Josemiami** » Sun Aug 09, 2009 10:54 pm

JFMuggs wrote:
Josemiami wrote:Also make sure to include the port 506 in the SIP call address.
Code: Select all
[2-9]xx[2-9]xxxxxx<:@192.168.169.105:5060>|
You only need to explicitly include a SIP port if you're using something other than the default 5060.

Since I was only trying to help but according to you I am giving wrong information, this is also true, yes I have make a lot of mistakes, my ata have been working great since the beginning and some time is very hard to duplicate other people problems, base on my own experiences some times and just guessing another. I want to thank az1324 for the great job well done, and I know he will be back next week with a Proxy running in a router.
I do apologize for all the wrong info but it was not intentional.

Ok maybe we have to get real picky.

Lest analyze your dial plan that I have been recommending and part of that I have contributed.

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(<411:18004664411>S0 |<:1>747xxxxxxxS0 |1747xxxxxxxS0 |<:1>800[2-9]xxxxxxS0 |1800[2-9]xxxxxxS0 |<:1>888[2-9]xxxxxxS0 |1888[2-9]xxxxxxS0 |<:1>877[2-9]xxxxxxS0 |1877[2-9]xxxxxxS0 |<:1>866[2-9]xxxxxxS0 |1866[2-9]xxxxxxS0 |<:1aaa>[2-9]xxxxxx<:@192.168.1.100>|<:1>[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |1[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |011xx.<:@192.168.1.100>|<#:1aaa>[2-9]xxxxxx|<#:1>[2-9]xx[2-9]xxxxxxS0 |<#:>1[2-9]xx[2-9]xxxxxxS0 |*xx.<:@sipbroker.com>)

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<411:18004664411>S0 |

You said the space after 'Straight Out' S0 dialing its necessary its wrong you don't need the space.

Code: Select all

1[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |

S0 after a SIP dialing its not necessary neither The ATA will dial Straight Out after a sip address.

Code: Select all

<:1>747xxxxxxxS0

As I started originally suggesting this dial plan for calling G5 numbers I realized than some town in California as I have Posted before has a real 747 area call!! so you may never know if you are making a free call or getting charge if you call using this dial plan.

So this is my suggestion:

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1747xxxxxxx<:@proxy01.sipphone.com

Anyway I want to Thank the admins for putting out with this post that have noting to do with Magic Jack.

I I apologize again if I misled anyone.

Thank you all.

My friend is opening a Forum Forr Google Voice.

JFMuggs · Post by **JFMuggs** » Mon Aug 10, 2009 12:01 am

Josemiami wrote:I do apologize for all the wrong info but it was not intentional.

I hope you don't feel I was accusing you of intentionally posting wrong information. My only desire is to see correct information imparted so that others don't go down the wrong path when they're trying to resolve a problem.

Josemiami wrote:
Code: Select all
<411:18004664411>S0 |
You said the space after 'Straight Out' S0 dialing its necessary its wrong you don't need the space.
Code: Select all
1[2-9]xx[2-9]xxxxxx<:@192.168.1.100>S0 |
S0 after a SIP dialing its not necessary neither The ATA will dial Straight Out after a sip address.

S0 doesn't stand for 'Straight Out'. It's an 'S' and a zero. It sets the Short timer to 0 IF that particular term matches. Having an alternate proxy SIP address specified doesn't change anything. And the space IS required. From the Linksys/Sipura Administrators Guide:

=====
The long and short timeout values can be changed for a particular sequence starting at a particular point in the sequence. The syntax for long timer override is: ‘L’ delay-value ‘ ‘. Note the terminating space character. The specified delay-value is measured in seconds. Similarly, to change the short timer override, use: ‘S’ delay-value <space>.
=====

Without the S0, the long (interdigit) delay timer is still active and the PAP2 will wait a longer time to start the call. You and I know that there are no numbers such as 411-1234, but the PAP2 isn't programmed to know that, so it can't assume that when it sees 411, you're done dialing. With the S0<space> after 411, we're telling it to make that assumption.

Josemiami wrote:
Code: Select all
<:1>747xxxxxxxS0 
As I started originally suggesting this dial plan for calling G5 numbers I realized than some town in California as I have Posted before has a real 747 area call!! so you may never know if you are making a free call or getting charge if you call using this dial plan.

So this is my suggestion:
Code: Select all
1747xxxxxxx<:@proxy01.sipphone.com

There has been talk about overlaying the 818 area code with a new 747 area code for that past decade, but it still hasn't been implemented (I just called the Verizon Operator and verified there is no 747 area code at this time). That aside, with or without <:@proxy01.sipphone.com> appended, 1747xxxxxxxS0 will go to Gizmo5.

jay235 · Post by **jay235** » Mon Aug 10, 2009 11:24 am

I am using the x-lite softphone with sipgate and GVOut to test if this works for me before purchasing the ATA.

My softphone X-lite is configured as follows:

Display name: My name
User Name: My sipgate phone number
password: my sipgate password
Auth user name: my sipgate phone number
domain: sipgate.com

Proxy Address: 192.168.1.47:5060 ( IP address of my laptop)

My GVOut is configured as follows:

Port: 5060
Sip Server: sipgate.com
Phone: my sipgate phone number
GV Username: my google voice username
GV Password: My google voice password
Delay (ms): 2000 ; checked the "Use busy Signal box"

Now I have GVOut, sipgate and x-lite running on the same PC. I try to call using my x-lite softphone to an outside number (my cell) and I see the sipgate receiving the call. I don't see the GVOut intercepting it. I can pick up the sipgate and am able to answer the call. The call does not go through any outside number. Any number I call just rings my sipgate.

If someone has used this configuration, let me know what you did to get it working.

JFMuggs · Post by **JFMuggs** » Mon Aug 10, 2009 2:28 pm

jay235 wrote:I am using the x-lite softphone with sipgate and GVOut to test if this works for me before purchasing the ATA.

.
.
.

If someone has used this configuration, let me know what you did to get it working.

Try these changes:

X-lite:

Proxy Address: sipgate.com

Outbound Proxy: localhost

GVOut:

Delay (ms): 1000

rmp25 · Post by **rmp25** » Mon Aug 10, 2009 10:25 pm

JFMuggs wrote:
rmp25 wrote:@JFMuggs

Your dialplan, which did not work yesterday, works just fine today without any correction, weird. I can call out using GVOut/G5, or dialing "#" and a phone number, I can call out using G5's 3 free minutes.

The next thing I'd like to accomplish is to get an ability to call out on the same line using any other provider then G5, based on this qoute from your early post:

"There can be only one default proxy per PAP2 line, but each dial plan term can specify an alternate proxy (using an '@' argument) to use instead of the default proxy when that term matches".

I tried to replace sipbroker.com with myuserid@12voip (I'm using 12voip.com on line 2 of my PAP2T), call goes thru, but same G5's 3 free min.

Would you please to follow this further.
You will need to get very familiar with PAP2 dialplan syntax before you can tackle this. I would suggest you do some Googling and find a PAP2 Administrator's Guide and study its section on dialplans as a starter. Dialplans sre not all that complicated, but they're probably not all that obvious at first glance.

The simple change you tried above won't work because I'm sure sure 12voip.com requires a password in addition to a username to place calls (otherwise, anyone could use your 12voip account to place calls at your expense). I don't recall if the PAP2 supports passwords in the dialplan. Also, you need to enter * + number to have what was the SipBroker term match.

Bottom line is you need to educate yourself on PAP2 dialplan syntax in order to attempt the change you desire.

OK, I understand that it's not simple thing. I'm learning Voip last couple of months, and I'm having fun.
Thanks for your help, JFMuggs.

aaronkephart · Post by **aaronkephart** » Tue Aug 11, 2009 10:43 am

I'm a bit confused...how could I use my SIP phone with Gizmo5 and GV, with this program?

I tried inputting my number and proxy in, however, I never got a call, I tried to dial out and the call went through without the busy tone or anything.

Please help, thanks.

jay235 · Post by **jay235** » Tue Aug 11, 2009 2:16 pm

Good News,

I did get the GVOut to initialize and am able to place an outgoing call using x-lite softphone.

I can also place an incoming call to my GV and receive it on the x-lite.

The problem is (and this may be with using x-lite) that after placing or receiving one call, the GVOut doesn't sync (or ack) of the termination of the call. I have to restart GVOut and x-lite to be able to place another call.

I did read others having this issue but did not see any resolution. If I missed it, please let me know. Otherwise, I would think this only exists using the softphone and I would not see this using an ATA.

Also, any advise on where to buy a good ATA and cost?

Thanks in advance.

JFMuggs · Post by **JFMuggs** » Tue Aug 11, 2009 3:49 pm

jay235 wrote:The problem is (and this may be with using x-lite) that after placing or receiving one call, the GVOut doesn't sync (or ack) of the termination of the call. I have to restart GVOut and x-lite to be able to place another call.

This is a bug in GVOut but the the author hasn't yet responded. Are you using a SIP port other than 5060? This bug usually isn't fatal if you're using port 5060. GVOut becomes non-responsive whether you're using a softphone or an ATA when you're using a port other than 5060.

JFMuggs · Post by **JFMuggs** » Tue Aug 11, 2009 4:06 pm

aaronkephart wrote:how could I use my SIP phone with Gizmo5 and GV, with this program?

I tried inputting my number and proxy in, however, I never got a call, I tried to dial out and the call went through without the busy tone or anything.

You MUST get your SIP phone working normally with Gizmo5 before attempting to use GVOut. You have to be able to receive incoming calls as well as place place outgoing calls (to other Gizmo5 users or free 3-minute PSTN calls with no Gizmo5 credit).

Once Gizmo5 incoming/outgoing is working properly, start and configure GVOut on your PC. Then configure your SIP phone for an outbound proxy of the IP address of the PC where GVOut is running. If your SIP phone is a softphone running on the same PC as GVOut, you can simply enter localhost.

Once GVOut is working properly, there may be a slightly more efficient configuration if you're using a Linksys/Sipura ATA or one with a similarly configurable dialplan, but first things first.

jay235 · Post by **jay235** » Tue Aug 11, 2009 4:12 pm

I am using port 5060. So not sure why I am encountering this issue.

When I try a port other than 5060, I cannot get the softphone to register.[/quote]

JFMuggs · Post by **JFMuggs** » Tue Aug 11, 2009 4:17 pm

jay235 wrote:I am using port 5060. So not sure why I am encountering this issue.

When I try a port other than 5060, I cannot get the softphone to register.

I believe you're the first to report having any problems related to this bug when using port 5060. I and many others have no problem using X-Lite with GVOut on port 5060. Maybe it's time to double-check the configuration of both X-Lite and GVOut.

jay235 · Post by **jay235** » Tue Aug 11, 2009 4:35 pm

JFMuggs wrote: I believe you're the first to report having any problems related to this bug when using port 5060. I and many others have no problem using X-Lite with GVOut on port 5060. Maybe it's time to double-check the configuration of both X-Lite and GVOut.

GVOut settings:

Port 5060
SIP server: sipgate.com
Phone: my sipgate phone
GV Username: my google voice userid
GV Password: my google voice pass
Delay(ms) 2000; checked for use busy signal
checked for save settings.

X-lite Settings:

Display name: my name
User name: sipgate id (1234567e0)
Password: my sipgate password
Authorization user name: sipgate id (1234567e0)
Domain: sipgate.com

checked for "Register with domain and receive incoming calls"

Send outbound via:
checked "Proxy" Address: localhost:5060

After placing a call, the GVOut in command view shows the following. It stays like this and does not show the hangup (disconnect) of the call until I restart the x-lite and GVOut.

Starting Proxy on port 5060
Client connection from 127.0.0.1
Client connection from 127.0.0.1
Client connection from 127.0.0.1
Client connection from 127.0.0.1

Outgoing Call Detected
Initiating Call to xxxxxxxxxx
_rnr_se=42663+G4IqTNtiRMXOqGan3f6o=
GV Commands Sent
GV Response: {"ok":true,"data":{"code":0}}

JFMuggs · Post by **JFMuggs** » Tue Aug 11, 2009 4:55 pm

jay235,

At first glance, your configuration looks reasonable.

If you remove the outbound proxy from X-Lite, does it work perfectly with SipGate (incoming and outgoing)?

jay235 · Post by **jay235** » Tue Aug 11, 2009 5:11 pm

JFMuggs wrote:jay235,

At first glance, your configuration looks reasonable.

If you remove the outbound proxy from X-Lite, does it work perfectly with SipGate (incoming and outgoing)?

I removed the Outbound Proxy from x-lite and stopped the GVOut executable. I am able to perform both incoming and outgoing call through x-lite and sipgate.

This leads me to believe something messed up in GVOut. I am using the latest GVOut executable dated 8/5/09.

JFMuggs · Post by **JFMuggs** » Tue Aug 11, 2009 5:35 pm

jay235 wrote:This leads me to believe something messed up in GVOut. I am using the latest GVOut executable dated 8/5/09.

GVOut properties should say version: 1.1.2.0

What you're experiencing is what we all see when running a SIP port other than 5060, but you're the first one that I'm aware of to see it on 5060. Hopefully, az1324 will be back soon to address it and provide us with a new version without this problem.

magicJack and magicJack Plus Support, Reviews, FAQs and Tricks

New Toy: GVOut Google Voice Outgoing Proxy Dialer (Windows)

Need help testing X-lite - Sipgate - GV - GVOut

Re: Need help testing X-lite - Sipgate - GV - GVOut

So...how do I use my SIP Phone with this program?

GV+SIPGATE+X-LITE+GVOut

Re: GV+SIPGATE+X-LITE+GVOut

Re: So...how do I use my SIP Phone with this program?