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


Joined: 19 Jul 2009
Posts: 85

PostPosted: Wed Aug 05, 2009 6:31 pm    Post subject:

az1324 wrote:
Ok new version that will prevent it from detecting the server as the client. Sorry about that. This should let people use the dialplan method if they want.

Please try not to post quoted links or mirrored files so that the latest version can be always in the first post and no confusion.



Just tested Version 1.1.2.0 and it does not work for me, the links mirroing have been deleted.

Thank you again
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JFMuggs
MagicJack Contributor


Joined: 03 Aug 2009
Posts: 74

PostPosted: Wed Aug 05, 2009 6:31 pm    Post subject: Re: dial plan not using outbound proxy server

anant wrote:
I have tested your dial plan. It is working with or without 1 in the front.

You should also be able to dial local numbers (i.e ones in the area code you used in place of aaa in the dial plan) by simply dialing the 7-digit number. The PAP2 will add the 1 + area code for you.
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Wed Aug 05, 2009 6:34 pm    Post subject:

Josemiami wrote:
Just tested Version 1.1.2.0 and it does not work for me

Try using the examples/suggestions I just posted.
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synchron
Dan isn't smart enough to hire me


Joined: 15 May 2008
Posts: 230

PostPosted: Wed Aug 05, 2009 6:36 pm    Post subject:

screenname wrote:


to cut down on confusions, maybe you can have some kind of revision info as part of the filename. At least the zip file?
Thank you again.


I think always getting the latest GVout works best for now since most of the mods are bugfixes and new features.

BTW, I can finally get it to work with X-lite and G5/GV but I have to stop/start the proxy for every call and sometimes force the X-lite to Rediscover/Re-register itself. I haven't tried AZ1324's latest yet.

I suspect my ATA problems have to do with the fact that there is no field for 'Outbound Proxy address'. Just SIP Proxy and Sip Domain. There is a checkbox for use outbound proxy which I never ever had to use for any Sip service including MJ and I don't think it has any effect. I just don't work.

Synchron Cool <sigh>
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Wed Aug 05, 2009 6:56 pm    Post subject:

JFMuggs wrote:
Josemiami wrote:
Just tested Version 1.1.2.0 and it does not work for me

Try using the examples/suggestions I just posted.


I have been using the same setup you just posted for the longest time, but the dial plan will not fit my purpose because I used to dial 10 digits numbers all the time, here in Miami we have two different area call that I call all the time, I don't make long distances calls.

I am happy with my dial plan, I will continue to encourage people to drop the G5 dependency, so I will continue to use Sipgate or Voxox.

I went back to GVOut version 1.1.1.0 and is working fine for me.

Thank you for the suggestion.
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Wed Aug 05, 2009 7:34 pm    Post subject:

Josemiami wrote:
the dial plan will not fit my purpose because I used to dial 10 digits numbers all the time, here in Miami we have two different area call that I call all the time, I don't make long distances calls.
The dial plan I posted handles 10-digit numbers (or 11-digits with a leading 1) just fine.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Wed Aug 05, 2009 7:39 pm    Post subject:

I power cycle my ATA, and now GVOut version 1.1.2.0 is working fine, just test it.

Thanks
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anant
Dan isn't smart enough to hire me


Joined: 12 Nov 2007
Posts: 114

PostPosted: Wed Aug 05, 2009 8:40 pm    Post subject: Questions

Hi Jose:

You said you will continue to use sipgate or voxox. Did u set up these in PAP2T for Line-1 and Line -2 . Do u run GVOUT in 2 different computers with 2 different GVOUT screens to fill data ? Or are you using sipsorcery by any chance ?.

One question pestering me is: Can I setup and run GVOUT in a laptop/PC not connected to the network to which PAP2T is connected ? I recall to have read something on this. Right now I am wrting this on a laptop not connected to the LAN which has PAP2T. Can I run GVOUT here ?


Thanks
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Wed Aug 05, 2009 9:14 pm    Post subject: Re: Questions

anant wrote:
Hi Jose:

You said you will continue to use sipgate or voxox. Did u set up these in PAP2T for Line-1 and Line -2 . Do u run GVOUT in 2 different computers with 2 different GVOUT screens to fill data ? Or are you using sipsorcery by any chance ?.

One question pestering me is: Can I setup and run GVOUT in a laptop/PC not connected to the network to which PAP2T is connected ? I recall to have read something on this. Right now I am wrting this on a laptop not connected to the LAN which has PAP2T. Can I run GVOUT here ?


Thanks


anant, you have to be in the same network for the GVOut Proxy to work.

I don't thing sipsorcery will work with this setup, but you never know, they run a proxy before for MJ and was working until MJ blocked theirs IP. they may do the same for GV.

Not I am not running Sipgate and voxox at the same time, just Sipgate at the moment but I did installed voxox in my PAP2T as a backup line just in case.
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VoipDude
Dan isn't smart enough to hire me


Joined: 06 Mar 2009
Posts: 129

PostPosted: Wed Aug 05, 2009 10:28 pm    Post subject:

How do I run GVOut as a service, so that it will start automatically every time the computer boots up?

Last edited by VoipDude on Wed Aug 05, 2009 11:23 pm; edited 1 time in total
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JFMuggs
MagicJack Contributor


Joined: 03 Aug 2009
Posts: 74

PostPosted: Wed Aug 05, 2009 10:46 pm    Post subject:

VoipDude wrote:
[b]How do I run GVOut as a service, so that it will start automatically every time the computer boots up?

There are ultilites that will allow virtually any program to be run as a service, but all you really need to do is make a shortcut to GVOut in your Start -> Programs -> Startup folder. Set its properties to Start Minimized and you're done.
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VoipDude
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Joined: 06 Mar 2009
Posts: 129

PostPosted: Wed Aug 05, 2009 11:23 pm    Post subject:

I added the program to the startup folder, and it starts when windows starts, however, all of the settings are blank, and need to be entered again. How can I avoid this? Also, how do I set it to start minimized?
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JFMuggs
MagicJack Contributor


Joined: 03 Aug 2009
Posts: 74

PostPosted: Wed Aug 05, 2009 11:43 pm    Post subject:

VoipDude wrote:
I added the program to the startup folder, and it starts when windows starts, however, all of the settings are blank, and need to be entered again. How can I avoid this? Also, how do I set it to start minimized?

There's a checkbox on GVOut called 'Save Settings' that, if checked, creates a GVOut.ini file that contains your settings and will be used each time it's started.

If you right-click on your GVOut shortcut in your Startup folder and select properties, there's a drop-down menu called 'Run'. Set it to minimized.
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mer
MagicJack Newbie


Joined: 15 Jun 2009
Posts: 1

PostPosted: Thu Aug 06, 2009 1:28 am    Post subject:

anyone try Grandstream Handy tone 486. What are the settings.

SIP SERVER:
Outbound proxy:
local sip port: (default 5060) --should this be the same port as gvout.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 11:58 am    Post subject:

We need help from 'Taken83oveR' and 'Pageman' to make a DDWRT Version of GVOut.

They are the top guns of this forum, they developed the DDWRT Version Of MJMD5 for MJ Proxy server.

Your names will be in the History book of the Wiki.

And that will be with az1324 permission of course.


Maybe they will if we ask nicely. Very Happy
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frenchfry
MagicJack Newbie


Joined: 01 Oct 2008
Posts: 4

PostPosted: Thu Aug 06, 2009 1:00 pm    Post subject:

Great job! The latest version is the best for me.

What is the source code written and compiled in?
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anant
Dan isn't smart enough to hire me


Joined: 12 Nov 2007
Posts: 114

PostPosted: Thu Aug 06, 2009 1:26 pm    Post subject: Pros & Cons of Google Voice - Excerpts from FierceVoip.

3. Pros and cons of Google Voice from around the Web
By Pete Wylie Comment | Forward

Google Voice has been all the rage lately, and this week was no exception. In addition to reports and analysis of Apple and AT&T's removal of the Google Voice app from the Apple iPhone App Store, there have been some interesting takes on the service in general.

Nerd Vittles has a really good primer on how to connect the Google Voice client to any Asterisk system, as well as reports on progress for a Python interface for Google Voice.

Basically, with a little tweaking you can use a Google Voice account for free U.S. calling and SMS from Asterisk phone systems by leveraging a DID with free inbound calling. Nerd Vittles provides an in-depth look at the benefits of this set-up, as well as all the code necessary to make it work.

VoIP opinion blog truvoipbuzz.com takes the opposite approach in an editorial about eight reasons why you should not use Google Voice as your primary phone number. The article lists limitations in carrier networks, spotty SMS reception, and overall reliability issues as evidence that Google Voice is not a good choice for your main line.

I'm impressed by Google voice's international rates, especially to mobile devices, but the voicemail transcription is really awful currently. I tested it several times and found about a 60 percent success rate, which left the messages largely indecipherable. I did not have problems, however, with excessive rings between connections, as some have noted. If you've got an account, what are your thoughts on Google Voice and its performance so far?

For more:
- see the Nerd Vittles GVoice-Asterisk primer here
- see the truvoipbuzz.com article here

Related articles
FCC probes Google Voice iPhone app removal
What's Google up to with Voice and other toys?

Read more about: Voicemail Transcription, Mobile Devices, iPhone, Google

---------------------------------------------------------------------
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anant
Dan isn't smart enough to hire me


Joined: 12 Nov 2007
Posts: 114

PostPosted: Thu Aug 06, 2009 1:44 pm    Post subject: FierceVoip.com link

Link for previous e-mail

http://www.fiercevoip.com/story/pros-and-cons-google-voice-around-web/2009-08-06
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 1:50 pm    Post subject:

frenchfry wrote:
Great job! The latest version is the best for me.

What is the source code written and compiled in?


This may help you


This is the public domain code for asterisk made by Paul Marks:

Code:
#!/usr/bin/env python

# google-voice-dialout.agi
# Paul Marks (www.pmarks.net)
# This code is Public Domain.
#
# This is an Asterisk 1.6 script to place outgoing calls through Google Voice.
# It will automatically sign into the web interface, and submit a click2call
# request through your registered Gizmo number.  Asterisk can then answer
# the incoming call, and Bridge() it into your original outgoing call.
#
# I deduced the click2call sequence by using the "Live HTTP Headers" Firefox
# plugin.  If the website changes too much, this script will probably stop
# working, so don't use it for anything too important.
#
# This assumes you've already configured Asterisk to receive Gizmo calls.
#
#
# This rule will redirect outbound calls to this script:
#   exten => _1NXXNXXXXXX,1,AGI(google-voice-dialout.agi)
#
# This rule will connect the inbound GV/Gizmo calls:
#   exten => s/6502650000,1,Bridge(${DB_DELETE(gv_dialout/channel)}, p)
#              ^-- Put your 10-digit Google Voice number here.
#
#
# To test this script from the command line without Asterisk, type the
# following.  Be sure to type a few linefeeds at the end:
#
#   $ ./google-voice-dialout.agi
#   agi_channel:
#   agi_dnid: 18004664411
#

# Put your Google login and Gizmo number here:
USERNAME = "bob"
PASSWORD = "hunter2"
GIZMO_NUMBER = "17470000000"

import httplib
import urllib
import re
import sys
import time

class Error(Exception):
    pass


def ReadAgiEnvironment():
    env = {}
    while 1:
        line = sys.stdin.readline().strip()
        if not line:
            break
        key, data = line.split(':')
        env[key.strip()] = data.strip()
    return env


def SendAgi(cmd):
    sys.stdout.write("%s\n" % cmd)
    sys.stdout.flush()
    sys.stdin.readline()


class SimpleCookieJar(object):
    cookie_re = re.compile(r"(?i)set-cookie: (\w+)=([^;]+).*")
    def __init__(self):
        self.cookies = {}
    def addCookies(self, response):
        for header in response.msg.headers:
            m = self.cookie_re.match(header)
            if not m:
                continue
            self.cookies[m.group(1)] = m.group(2)
    def get(self):
        return "; ".join("%s=%s" % kv for kv in self.cookies.iteritems())


class GVClickToCall(object):
    USER_AGENT = "google-voice-dialout.agi/1.1"

    def __init__(self, username, password, via, dial):
        self.username = username
        self.password = password
        self.via = via
        self.dial = dial
        self.cj = SimpleCookieJar()
        self.h = httplib.HTTPSConnection("www.google.com")
        self.login()
        self.placeCall()
        self.logout()

    def login(self):
        print >>sys.stderr, "Logging in."
        postdata = urllib.urlencode({ "Email": self.username,
                                      "Passwd": self.password })
        self.doRequest(
            method="POST", url="/accounts/ServiceLoginAuth",
            body=postdata,
            headers={ "Content-Type": "application/x-www-form-urlencoded" })

        # Start at https://www.google.com/voice, and collect cookies as we
        # follow all the redirects.
        PREFIX = "https://www.google.com/"
        location = "/voice"
        for i in xrange(5):
            response, html = self.doRequest(
                method="GET", url=location,
                headers={})

            location = response.getheader("location")
            if not location:
                # No more redirects, yay!
                break

            # All redirects should fall within the same domain.
            if not location.startswith(PREFIX):
                raise Error("Unexpected redirect: %s" % location)
            location = location[len(PREFIX)-1:]

        # Scrape magic _rnr_se value from the HTML.
        m = re.search(r'name="_rnr_se" type="hidden" value="([^"]+)"', html)
        if not m:
            raise Error("Can't find _rnr_se.  Not logged in?")
        self.magic_rnr_se = m.group(1)

    def placeCall(self):
        print >>sys.stderr, "Calling %s via %s" % (self.dial, self.via)
        postdata = urllib.urlencode({ "outgoingNumber": self.dial,
                                      "forwardingNumber": self.via,
                                      "_rnr_se": self.magic_rnr_se })
        response, http = self.doRequest(
            method="POST", url="/voice/call/connect",
            body=postdata,
            headers={ "Content-Type": "application/x-www-form-urlencoded" })
        print >>sys.stderr, "Dial response:", http

    def logout(self):
        self.doRequest(
            method="GET", url="/accounts/Logout",
            headers={ "Connection": "close" })
        print >>sys.stderr, "Logged out."

    def doRequest(self, headers, **kw):
        headers["User-agent"] = self.USER_AGENT
        headers["Cookie"] = self.cj.get()
        self.h.request(headers=headers, **kw)
        response = self.h.getresponse()
        self.cj.addCookies(response)
        return response, response.read()


def main():
    env = ReadAgiEnvironment()
    print >>sys.stderr, env

    agi_channel = env["agi_channel"]
    agi_dnid = env["agi_dnid"]

    # Write the channel ID to Asterisk's database, so it can be accessed
    # by the incoming call when it arrives.
    SendAgi("database put gv_dialout channel %s" % agi_channel)

    SendAgi("answer")
    try:
        GVClickToCall(username=USERNAME, password=PASSWORD,
                      dial=agi_dnid, via=GIZMO_NUMBER)
   
        # Asterisk should patch in the incoming call while we're asleep.
        time.sleep(10)
    finally:
        SendAgi("hangup")


if __name__ == '__main__':
    main()

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synchron
Dan isn't smart enough to hire me


Joined: 15 May 2008
Posts: 230

PostPosted: Thu Aug 06, 2009 2:31 pm    Post subject:

Well, with the latest GVout, I finally got things to work, (sort of) with G5 using the debug mode cmd window to help. What's weird is that I have to set the ATA up so that it doesn't register and no dialtone. This means that I leave the SIP domain empty (no proxy01.sipphone.com entry) and I just stick the PC IP address in SIP Proxy field.

So, with no dialtone, I can still dial out, disconnect, see the correct status in the cmd window and get the GV callback ring. With the ATA registered and a dialtone, I don't get the constant client messages from the ATA IP address and it won't work.

Weird, to dial out with no dialtone, but it works, nevertheless. This is with the Sunrocket/Vonics/Innomedia ATA. It works best if you set GVout to no delay/no busy tone.

Synchron Cool
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VoipDude
Dan isn't smart enough to hire me


Joined: 06 Mar 2009
Posts: 129

PostPosted: Thu Aug 06, 2009 3:13 pm    Post subject:

JFMuggs wrote:
VoipDude wrote:
I added the program to the startup folder, and it starts when windows starts, however, all of the settings are blank, and need to be entered again. How can I avoid this? Also, how do I set it to start minimized?

There's a checkbox on GVOut called 'Save Settings' that, if checked, creates a GVOut.ini file that contains your settings and will be used each time it's started.

If you right-click on your GVOut shortcut in your Startup folder and select properties, there's a drop-down menu called 'Run'. Set it to minimized.


JFMuggs, thanks for your help with this. Everything works now. The problem was I had not extracted the exe from the zip file.
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az1324
Dan isn't smart enough to hire me


Joined: 20 Feb 2008
Posts: 100

PostPosted: Thu Aug 06, 2009 5:56 pm    Post subject:

Josemiami wrote:
We need help from 'Taken83oveR' and 'Pageman' to make a DDWRT Version of GVOut.

They are the top guns of this forum, they developed the DDWRT Version Of MJMD5 for MJ Proxy server.

Your names will be in the History book of the Wiki.

And that will be with az1324 permission of course.


Maybe they will if we ask nicely. Very Happy


Actually, teddy_b wrote the code for those versions. I am going to try and play a little more with SDP and RTP streams when I get some time in the next week or so. After that if nobody has done it already I might build a router version.
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JFMuggs
MagicJack Contributor


Joined: 03 Aug 2009
Posts: 74

PostPosted: Thu Aug 06, 2009 6:33 pm    Post subject:

az1324 wrote:
I am going to try and play a little more with SDP and RTP streams when I get some time in the next week or so. After that if nobody has done it already I might build a router version.
*PLEASE* include a version that runs on Tomato firmware (current version is 1.25). Having GVOut do it's magic from inside the router would be great.
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doohickey
MagicJack User


Joined: 08 Jul 2009
Posts: 41

PostPosted: Thu Aug 06, 2009 7:18 pm    Post subject:

I just installed GVOut for the first time (version 1.1.2.0) and I'm getting a "The application failed to initialize properly" error.

I'm guessing I need to install .NET or something? Please tell me that's not the case because I don't think my little old workhorse will be able to manage to do that on 256MB of RAM.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 7:20 pm    Post subject:

Quote:
az1324 wrote:
I am going to try and play a little more with SDP and RTP streams when I get some time in the next week or so. After that if nobody has done it already I might build a router version.



Great news.

It will be a good addition to GVOut, if the program automatically detect your local IP address and ad it to the number you are dialing, so you don't need to put it in your dial plan, and you will be able to dial from any wifi connection using a SIP Phone or ATA without having to worry about your Local IP address.

EG.

You dial 3055551212 and the proxy will auto detect your local IP address and will dial 3055551212@192.168.1.1:5060 for you.

Just a thought.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 7:49 pm    Post subject:

Express Talk soft Phone setup for GVOut and Gizmo5.










Last edited by Josemiami on Thu Aug 06, 2009 8:06 pm; edited 1 time in total
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JFMuggs
MagicJack Contributor


Joined: 03 Aug 2009
Posts: 74

PostPosted: Thu Aug 06, 2009 7:59 pm    Post subject:

Josemiami wrote:
It will be a good addition to GVOut, if the program automatically detect your local IP address and ad it to the number you are dialing, so you don't need to put it in your dial plan, and you will be able to dial from any wifi connection using a SIP Phone or ATA without having to worry about your Local IP address.

EG.

You dial 3055551212 and the proxy will auto detect your local IP address and will dial 3055551212@192.168.1.1:5060 for you.

The '@GVOut's_ip_address:5060' in the PAP2's dial plan is telling the PAP2 where to send the call request to instead of what's in the Proxy field (proxy01.sipphone.com in the case of Gizmo5). If you don't put it in your dial plan, the call request will be sent to proxy01.sipphone.com and would never reach GVOut (and therefore no busy signal and no callback request to Google Voice). Gizmo5 would do the calling as GVOut wouldn't be in the picture. The '@GVOut's_ip_address:5060' isn't actually sent to anybody.

Once GVOut is running in your router, your dial plan will use '@router's_ip_address:5060', which doesn't change, so you don't have to worry about your PC's local IP address. In the meantime, you probably ought to set up a Static DHCP address for the PC that's hosting GVOut so its IP address can't possibly move around.
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Thu Aug 06, 2009 8:13 pm    Post subject:

Josemiami wrote:
Express Talk soft Phone setup for GVOut and Gizmo5.

Unless Express Talk (or any other softphone or ATA) has the ability to specify an outbound proxy server for INVITE requests in the dial plan like the PAP2 does, you're pretty much forced to set the Outbound Proxy to 'GVOut's_ip_address' unless you're willing to add it to every number you dial like your example shows.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 8:18 pm    Post subject:

JFMuggs wrote:
Josemiami wrote:
It will be a good addition to GVOut, if the program automatically detect your local IP address and ad it to the number you are dialing, so you don't need to put it in your dial plan, and you will be able to dial from any wifi connection using a SIP Phone or ATA without having to worry about your Local IP address.

EG.

You dial 3055551212 and the proxy will auto detect your local IP address and will dial 3055551212@192.168.1.1:5060 for you.

The '@GVOut's_ip_address:5060' in the PAP2's dial plan is telling the PAP2 where to send the call request to instead of what's in the Proxy field (proxy01.sipphone.com in the case of Gizmo5). If you don't put it in your dial plan, the call request will be sent to proxy01.sipphone.com and would never reach GVOut (and therefore no busy signal and no callback request to Google Voice). Gizmo5 would do the calling as GVOut wouldn't be in the picture. The '@GVOut's_ip_address:5060' isn't actually sent to anybody.

Once GVOut is running in your router, your dial plan will use '@router's_ip_address:5060', which doesn't change, so you don't have to worry about your PC's local IP address. In the meantime, you probably ought to set up a Static DHCP address for the PC that's hosting GVOut so its IP address can't possibly move around.


Just read my post I have been doing that with the PAP2T dial plan, I have to travel a lot so it will be a good idea to incorporate the the local IP address with the dialer. I am pretty sure az1324 will know what I am talking about.

Auto detect, auto dialing for you, The local computer IP address is just a variable that can be detected and replace, its just a matter of programing.

Just another option so you can use your soft phone anyware.

Thanks for the tip
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Thu Aug 06, 2009 8:22 pm    Post subject:

JFMuggs wrote:
Josemiami wrote:
Express Talk soft Phone setup for GVOut and Gizmo5.

Unless Express Talk (or any other softphone or ATA) has the ability to specify an outbound proxy server for INVITE requests in the dial plan like the PAP2 does, you're pretty much forced to set the Outbound Proxy to 'GVOut's_ip_address' unless you're willing to add it to every number you dial like your example shows.


If you look at the pics you see That I don't have to put the local IP, in my outbound proxy. I do have to dial the number using a SIP. So Express Talk behave just like PAP2T. I that's what the GVUot dialer can do for you if its program to auto detect an ad your local IP address to the number you dial.

The program can do a search for your local IP address.

Local IP = "Z"
You dial 3055551212.

The program will dial 3055551212@'Z'

In this case 'Z' will be a variable.
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Thu Aug 06, 2009 9:09 pm    Post subject:

Josemiami wrote:
The program can do a search for your local IP address.

Local IP = "Z"
You dial 3055551212.

The program will dial 3055551212@'Z'

In this case 'Z' will be a variable.

If you don't put '@GVOut's_ip_address' in the PAP2's dial plan (or as part of the number dialled in Express Talk when you haven't set an Outbound Proxy in Express Talk), GVOut will never receive your call request (or anything else) to do anything with it or to it.

'@GVOut's_ip_address' is not for use by GVOut or Google Voice (it's not sent to either of them), it's to instruct the SIP client (the PAP2 or Express Talk) where to send the INVITE request (GVOut instead of proxy01.sipphone.com). Without it, the client would send the request directly to proxy01.sipphone.com and GVOut would be out of the picture and never know anything was going on.
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Josemiami
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Joined: 19 Jul 2009
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PostPosted: Fri Aug 07, 2009 12:47 am    Post subject:

JFMuggs

This have been a learning experience, I tried putting the Local Host into the outbound proxy of the Soft Phone like you suggested before and it does work nicely, this will allow me to change networks wile I'm traveling withot having to input my Computer IP address every time, thank you for the tip.

There is the pic for the express Talk Soft Phone.

I have tried before with XLite but it did not work.

Now I just have to dial the 10 digits number. I have to try it now in my PAP2T.



Thank you
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 1:34 am    Post subject:

Josemiami wrote:
I tried putting the Local Host into the outbound proxy of the Soft Phone like you suggested before and it does work nicely, this will allow me to change networks wile I'm traveling withot having to input my Computer IP address every time, thank you for the tip.

Yes, if you're going to use a softphone from the same machine that GVOut is running on, an Outbound Proxy of localhost is the way to go. The more traditional value to use for localhost is 127.0.0.1. Even better, you can simply use the word: localhost
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synchron
Dan isn't smart enough to hire me


Joined: 15 May 2008
Posts: 230

PostPosted: Fri Aug 07, 2009 10:53 am    Post subject:

Guys:

Is it necessary to specify the Gizmo 5 username (1747XXXXXXX) in the Authentication ID field of your ATA sip credentials? Do you need to be logged on to your G5 account online?

Reason I ask is that I am primarily still using MJMD5/ATA until my pw no longer authenticates and will permanently switch to GVout once that happens. When I just did a switch in the ATA, the phone wouldn't ring (both from dialing out and using the G5 webpage) until I did both of the above and I need a consensus as to what I did to get it to work? (Logging onto G5 account or adding Auth ID in ATA or both??)

Thanks,

Synchron Cool
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 3:58 pm    Post subject:

synchron wrote:
Is it necessary to specify the Gizmo 5 username (1747XXXXXXX) in the Authentication ID field of your ATA sip credentials? Do you need to be logged on to your G5 account online?

My 'Auth ID' is blank and 'Use Auth ID' is set to no in my PAP2. I have 'Proxy' set to proxy01.sipphone.com and 'Register' set to yes. I don't have to be logged into my Gizmo 5 account on a web browser for things to work. Operation has been extremely reliable with the latest version of GVOut.
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doohickey
MagicJack User


Joined: 08 Jul 2009
Posts: 41

PostPosted: Fri Aug 07, 2009 4:47 pm    Post subject:

Before I spend the next 60 minutes or so listening to my old PIII's hard drive choking on a .NET 3.5 install, can someone please confirm that NET 3.5 is indeed required by GVOut. Thanks.
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JFMuggs
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Joined: 03 Aug 2009
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PostPosted: Fri Aug 07, 2009 4:53 pm    Post subject:

doohickey wrote:
Before I spend the next 60 minutes or so listening to my old PIII's hard drive choking on a .NET 3.5 install, can someone please confirm that NET 3.5 is indeed required by GVOut.
Why don't you just run GVOut and see if it complains?
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doohickey
MagicJack User


Joined: 08 Jul 2009
Posts: 41

PostPosted: Fri Aug 07, 2009 5:07 pm    Post subject:

JFMuggs wrote:
Why don't you just run GVOut and see if it complains?

It does per my previous post on the matter:
doohickey wrote:
I just installed GVOut for the first time (version 1.1.2.0) and I'm getting a "The application failed to initialize properly" error.

The error is a bit vague.
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 5:27 pm    Post subject:

I just ran GVOut on a freshly installed copy of Windows 7 Ultimate and it worked prefectly. Unless Microsoft is pre-installing .NET framework with Windows 7, it's not required.
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doohickey
MagicJack User


Joined: 08 Jul 2009
Posts: 41

PostPosted: Fri Aug 07, 2009 5:30 pm    Post subject:

Thanks, JFMuggs. That kinda leaves me in something of a limbo, but I appreciate that you checked that out.

Edit: After doing some googlin' it appears that Windows 7 does come with .NET preinstalled.
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JFMuggs
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Joined: 03 Aug 2009
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PostPosted: Fri Aug 07, 2009 6:01 pm    Post subject:

doohickey wrote:
After doing some googlin' it appears that Windows 7 does come with .NET preinstalled.
Well, if it does, it can't be removed as it doesn't show up in the installed programs area in order to be able to uninstall it.

Edit: Upon further investigation, it does appear that every version of .NET *is* pre-installed on Windows 7 at \Windows\Microsoft.NET\. Unless you have a system available with .NET already installed where you can verify the error message doesn't occur, it appears you're going to have to install it to know for sure.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Fri Aug 07, 2009 7:08 pm    Post subject:

There is some Hope read this:

pagemen wrote:
It pains to digg through RFC and mess with RTP stream, imho, would be easier to use Asterisk other than reinventing the wheel.

I wouldn't release my finished work atm, but just a hint -- find or write a replacement of Paul Marks' Python script, make it fit in our underpowered routers, then coupled with Asterisk dial plan as suggested in the PiAF guide, you'll get a nice no-pc gvout solution. I've been running this setup on my OpenWRT+Asterisk box for over a month with satisfactory result.



Thanks for the great news, yes the future is here. I hope you will release it soon after your testing its done I still got to wait for the code for the DD-WRT, or maybe I be better off re flashing my my Linsys with the OpenWRT image either way is good to know it was doable, and soon we will enjoy the finish product.

Thank you for all your work.


Josemiami
MagicJack Former User
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gm_w
MagicJack Newbie


Joined: 14 Jun 2009
Posts: 3

PostPosted: Fri Aug 07, 2009 10:18 pm    Post subject:

I used the dial plan

(<411:18004664411@192.168.1.100>S0 |<: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>)

I changed aaa to my area code and 192.168.1.100 to my PC's ip address

When I dial out (10 digit), I immediately get busy signal. I hang up, after a few seconds, phone rings, when I pick it up, after a few seconds I hear a fast batch of fast busy signals. In the meantime, the other side rings and answers. Both sides can't hear each other.

Any solutions to this? I am using GVout 1.1.2.0 and linksys PAP2 -NA
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 10:32 pm    Post subject:

gm_w wrote:

When I dial out (10 digit), I immediately get busy signal. I hang up, after a few seconds, phone rings, when I pick it up, after a few seconds I hear a fast batch of fast busy signals. In the meantime, the other side rings and answers. Both sides can't hear each other.

Any solutions to this? I am using GVout 1.1.2.0 and linksys PAP2 -NA
Sounds like you're on the right track. Can you post some screen shots of your Line and SIP tab settings?
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Fri Aug 07, 2009 10:57 pm    Post subject:

gm_w wrote:
I used the dial plan

(<411:18004664411@192.168.1.100>S0 |<: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>)

I changed aaa to my area code and 192.168.1.100 to my PC's ip address

When I dial out (10 digit), I immediately get busy signal. I hang up, after a few seconds, phone rings, when I pick it up, after a few seconds I hear a fast batch of fast busy signals. In the meantime, the other side rings and answers. Both sides can't hear each other.

Any solutions to this? I am using GVout 1.1.2.0 and linksys PAP2 -NA


That happen to me when I try to run a soft phone, and a PAP2T in the same computer "by mistake" I'm running the GVout Router. the soft Phone worked fine and then I got exactly what happen to you when I tried to place a cal in the PAP2T, then I exit the soft Phone program, reboot the computer and no more problems.

I hope it helps


Last edited by Josemiami on Sat Aug 08, 2009 1:16 am; edited 1 time in total
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 11:09 pm    Post subject:

Josemiami wrote:
That happen to me when I try to run a soft phone, and a PAP2T in the same computer Im running the GVout Router.
You can't have two SIP clients (softphone + PAP2) both trying to register with the same account on port 5060 at the same time.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Fri Aug 07, 2009 11:11 pm    Post subject:

Dial plan for free G5 calls and, toll free, no proxy running needed for the G5 and toll free calls.

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!)


You can add or take out to fit your needs, I just adding the toll free, you will always dial 10 digits, no long distances allow, no 911 allow, its op to you to keep it simple or add more since I live in an area with two different area calls I like to dial 10 digits, I like to keep my dial plan the way it is, if is any help use it, if not say nothing.


Last edited by Josemiami on Fri Aug 07, 2009 11:31 pm; edited 2 times in total
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JFMuggs
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Joined: 03 Aug 2009
Posts: 74

PostPosted: Fri Aug 07, 2009 11:17 pm    Post subject:

Josemiami wrote:
Code:
1747[2-9]xxxxxxS0
Gizmo5 numbers are allocated with 17470xxxxxx and 17471xxxxxx. You need to drop the [2-9] qualifier on this term.
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Josemiami
MagicJack Expert


Joined: 19 Jul 2009
Posts: 85

PostPosted: Fri Aug 07, 2009 11:29 pm    Post subject:

JFMuggs wrote:
Josemiami wrote:
Code:
1747[2-9]xxxxxxS0
Gizmo5 numbers are allocated with 17470xxxxxx and 17471xxxxxx. You need to drop the [2-9] qualifier on this term.


Since some Gizmo numbers are allocated like the that then.

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!)


But according to this you may get charge if you call a real PSTN 747 in California.

Starting Saturday, residents living in the San Fernando Valley will have to dial a little differently.

Quote:
The California Public Utilities Commission is reminding folks living in the 818 and 747 area codes that they will have to dial 1 plus the area code plus the telephone number to complete their calls.


So I think this will be a better option:


Code:
(1747xxxxxxx<:@proxy01.sipphone.com:5060> |866[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!)
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rmp25
MagicJack Newbie


Joined: 07 Aug 2009
Posts: 7

PostPosted: Sat Aug 08, 2009 12:08 am    Post subject:

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?
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