Moderators: Bill Smith, Pilot
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.
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.anant wrote:I have tested your dial plan. It is working with or without 1 in the front.
I think always getting the latest GVout works best for now since most of the mods are bugfixes and new features.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.
<sigh>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.JFMuggs wrote:Try using the examples/suggestions I just posted.Josemiami wrote:Just tested Version 1.1.2.0 and it does not work for me
The dial plan I posted handles 10-digit numbers (or 11-digits with a leading 1) just fine.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.
anant, you have to be in the same network for the GVOut Proxy to work.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
VoipDude wrote:How do I run GVOut as a service, so that it will start automatically every time the computer boots up?
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.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?
This may help youfrenchfry wrote:Great job! The latest version is the best for me.
What is the source code written and compiled in?
Code: Select all
#!/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()
JFMuggs, thanks for your help with this. Everything works now. The problem was I had not extracted the exe from the zip file.JFMuggs wrote: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.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?
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.
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.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.
*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.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.
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.
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.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 [email protected]:5060 for you.
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.Josemiami wrote:Express Talk soft Phone setup for GVOut and Gizmo5.
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.JFMuggs wrote: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.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 [email protected]:5060 for you.
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.
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.JFMuggs wrote: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.Josemiami wrote:Express Talk soft Phone setup for GVOut and Gizmo5.
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.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.
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: localhostJosemiami 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.
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.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?
It does per my previous post on the matter:JFMuggs wrote:Why don't you just run GVOut and see if it complains?
The error is a bit vague.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.
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.doohickey wrote:After doing some googlin' it appears that Windows 7 does come with .NET preinstalled.
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.
Sounds like you're on the right track. Can you post some screen shots of your Line and SIP tab settings?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
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.gm_w wrote:I used the dial plan
(<411:[email protected]>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
Code: Select all
(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!)Gizmo5 numbers are allocated with 17470xxxxxx and 17471xxxxxx. You need to drop the [2-9] qualifier on this term.Josemiami wrote:Code: Select all
1747[2-9]xxxxxxS0
Since some Gizmo numbers are allocated like the that then.JFMuggs wrote:Gizmo5 numbers are allocated with 17470xxxxxx and 17471xxxxxx. You need to drop the [2-9] qualifier on this term.Josemiami wrote:Code: Select all
1747[2-9]xxxxxxS0
Code: Select all
(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!)So I think this will be a better option: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.
Code: Select all
(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!)