Tag: Google Voice

Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk

 

Notice
(May, 2018): FreePBX and Asterisk users that wish to continue using Google Voice after Google drops XMPP support should go here: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware. The information in this article is VERY outdated and probably will not work.

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

If you have been less than thrilled with the Google Voice support in another software PBX, such as Asterisk or FreeSWITCH, you could try using YATE as a Google Voice Gateway.  It can be installed on either a separate server, or on the same server as your FreeSWITCH or Asterisk installation, however if you are running virtual machines then I recommend the separate server approach.  In fact, that may be the only way to do it with FreeSWITCH if you installed FreeSWITCH under Debian or Ubuntu, since the YATE install requires CentOS.  If you are a Linux expert you may be able to get around this, but don’t ask me how.

To install YATE, see this article from Nerd Vittles:

YATE in a Flash: Rolling Your Own SIP to Google Voice Gateway for Asterisk

EDIT: You may want to upgrade YATE to the latest version.

Just follow the instructions there, and the ones that you see after running the script to add a Google Voice user, and you should be fine, if you are using Asterisk.  The only things I would suggest that are not shown in those instructions are that you set your Trunk “Maximum Channels” to 2, because a Google Voice account will only permit two simultaneous channels of usage maximum, and that if YATE is on a separate server with a static IP address then I’d suggest adding permit/deny lines to the Asterisk Trunk PEER details to enhance security, like so:

permit=xx.xx.xx.xx/255.255.255.255
deny=0.0.0.0/0.0.0.0

Make sure the lines appear in that order, and replace xx.xx.xx.xx with the static IP address of the YATE server.  This may not help much because Asterisk is registering with the YATE server, but it can’t hurt either.

Also, you might want to consider changing the context statement to

context=from-pstn-e164-us

to remove the +1 from the start of the Caller ID number on incoming calls.

The instructions don’t tell you to add a Dialed Number Manipulation Rule to your trunk configuration, but if you want to allow ten digit calls from any of your endpoints then you should add one rule that prepends 1 to 10 digit calls:

1+NXXNXXXXXX (The 1 goes in the first field, the NXXNXXXXXX in the third field)

If you are using the CallerID Superfecta module, and you use “Trunk Provided” as one of your data source, then after adding a Google Voice account to YATE I suggest editing /usr/local/etc/yate/regexroute.conf on the YATE server. You may need to install an editor first. For example, to install nano and then edit the file:

yum install nano
nano /usr/local/etc/yate/regexroute.conf

Look for the [contexts] section and there you will see a line for each of your Google Voice accounts that looks like this:

${in_line}GV1234567890=;called=GV1234567890;jingle_version=0;jingle_flags=noping;dtmfmethod=rfc2833

Just add ;callername to the end of each such line:

${in_line}GV1234567890=;called=GV1234567890;jingle_version=0;jingle_flags=noping;dtmfmethod=rfc2833;callername

This will make sure that nothing is sent for a Caller ID name, so that Caller ID Superfecta will recognize that there is no “Trunk Provided” name and attempt to do a name lookup (note that you could also use ;callername=something to set the Caller ID name to a specific value). If you want to have ;callername
automatically appended whenever you create a new account, just use an editor to edit the script you use to add users, and find the line that looks like this (it should be near the bottom of the script):

${in_line}GV’$acctphone’=;called=GV’$acctphone’;jingle_version=0;jingle_flags=noping;dtmfmethod=rfc2833

Add ;callername to the end of the line, like so:

${in_line}GV’$acctphone’=;called=GV’$acctphone’;jingle_version=0;jingle_flags=noping;dtmfmethod=rfc2833;callername

Save the modified file, and any time you add a new user it will automatically write that line with ;callername appended.

Thanks to Bill Simon for telling me about this method of sending the blank Caller ID name. Alternately, if you don’t want to mess with the YATE configuration, you could add a new Caller ID Scheme in Caller ID Superfecta that is only used with your Google Voice DID’s and that doesn’t include “Trunk Provided” as a data source.

Whether you are connecting from Asterisk or FreeSWITCH, if YATE is running on a separate server and the other system can’t register with YATE, it may be a firewall issue on the YATE server.  After I did the install I found that iptables was configured to only allow incoming ssh connections.  I modified that rule to only allow incoming ssh from a particular IP address (the one I’d be coming in from) and then added rules to permit traffic from the two servers allowed to talk to that YATE server.

EDIT: Hopefully this will not affect you if you have upgraded YATE to the latest version, but if you have a moderate number of Google Voice accounts, you may experience another issue.  If you start seeing messages like this when you telnet to YATE and then use debug on to see what is happening:

<sip:MILD> Flood detected: 20 handled events

And if every so often, the server appears to go into a semi-catatonic state, where calls come in but they don’t go out (this happened to me at least twice before I figured out what was happening), then you may have this issue.  It occurs when you have the same Asterisk server using multiple trunks to connect to YATE.  It turns out that whenever you reload Asterisk (as you might after making a configuration change, for example the “orange bar reload” in one particular GUI), it resends all of the registrations at once, and gives them all a default timeout of 120 seconds, so they all attempt to re-register at the exact same intervals.  And if you have several trunks, there are a LOT of SIP packets sent.  Plus, with qualifyfreq value set to 240, that means that every other time the registrations are taking place, qualifies are also taking place at the same time.  It appears that this is sufficient to cause that warning to appear once in a while.

The method I found that seems to fix this may not be the best way (feel free to comment if you know a better way), but it’s one way to deal with it.  What you need to do is change the registration expiration on each individual trunk so they are not all the same.  In Asterisk this can be accomplished by adding both of these settings to the trunk configuration (susbtitute nn with some random number of seconds, say between 90 and 120, and make it the same for both settings in each trunk, but different for different trunks)

In the trunk PEER details, add:

defaultexpiry=nn

In the Register String, add  ~nn  to the end of the line, replacing nn with the same value used in the defaultexpiry setting, like so:
GV1234567890:password@exampleaddress.com:5060/1234567890~nn

You might also need to vary the qualifyfreq value a bit in each trunk, so that it’s a bit under the specified 240 seconds and different for each trunk.  If doing those things doesn’t fix the issue, and you still get the <sip:MILD> Flood detected: 20 handled events message frequently, that could mean you are being subjected to an actual SIP attack.  The YATE installation includes a script with the filename /usr/src/yate/share/scripts/banbrutes.php that can be used to deal with that issue, but it’s not enabled by default.  View the banbrutes.php script in a text editor, and you’ll find instructions at the beginning of the script.  Or, you could tighten up the iptables firewall to only allow traffic from systems that are supposed to be talking to your YATE server.

END OF EDIT.

As for FusionPBX, when you create a new Google Voice account on the YATE server using the provided add-yate-user script, at the end it will give you a bunch of configuration information for Asterisk.  These translate to FusionPBX Gateway settings as follows (showing what the script prints and the equivalent FusionPBX Gateway settings):

Trunk Name: YIAF1 ; or increment 1 if more than one (in FusionPBX I suggest you don’t use this; instead use the same setting as the Username for the Gateway name, particularly if you plan on having more than one Google Voice account)

host=x.x.x.x (Proxy in FusionPBX)
username=GV1234567890 (Username in FusionPBX)
secret=password (Password in FusionPBX)
type=peer (Not needed in FusionPBX)
port=5060 (Not needed in FusionPBX)
qualify=yes (Not needed in FusionPBX)
qualifyfreq=240 (Not needed in FusionPBX)
insecure=port,invite (Not needed in FusionPBX)
context=from-trunk (Not needed in FusionPBX)

Register String: … (Not needed in FusionPBX)

In FusionPBX, set Register to True and Enabled to True, and leave other Gateway settings at the defaults (EDIT: however, if you have several gateways to YATE, you might want to use the Expire seconds setting in FusionPBX to vary the registration timeouts a bit so that all your accounts aren’t trying to re-register at exactly the same time — see the longer EDIT section above for details).  Note that after you save the settings, it may take a few seconds for the state to change to REGED, so refresh the Gateways page after a bit and it should be okay if everything is configured properly and there are no firewall issues.

For your Inbound Route in FusionPBX, just use the Trunk Name/Username as the Destination Number (including the leading “GV“, which you can also use it in the Inbound Route name field if you like) and then choose the appropriate Action. When you first create the Inbound Route it will complain if you try to save a Destination Number that is not completely numeric, so just use any number and save the settings, then go back and edit the Destination Number field and also the Data field for the destination_number condition (which should be something like ^GV1234567890$, substituting your Google Voice number for the digits, of course).

For your Outbound Route, select your Google Voice trunk as the Gateway, and then select “11 digits long distance” from the dropdown in the “Dialplan Expression” setting. Save that, and if you only have one Google Voice trunk for all users on the system, that is all you need to do.  However, if you want to have multiple Google Voice trunks and have certain extensions only have access to certain trunks, the edit the Outbound Route you just created, and in the “Conditions and Actions” section at the bottom of the page, edit the last action on the page (the “bridge” action).  You want to change the Data field – it will contain something like sofia/gateway/GV1234567890/$1 and you want to change that to sofia/gateway/${accountcode}/$1 — save that change, and then when the Outbound Route page reappears, you may want to change the name to ${accountcode}.11d and add a Description like “Google Voice: Extension Account Code = Gateway Name” so you understand what the route is doing.  This single Outbound Route will handle all your Google Voice calls from all your extensions, if the Account Code setting for each Extension is set to the name of the Gateway for the Google Voice account you want that extension to use.

Note that if you are running PBX in a Flash, you can use the “Caller ID Superfecta” module to try to get a Caller ID name.  IF YATE itself has any ability to do Caller ID name lookups, someone will have to tell me how to enable and configure it, because at this point I would have no clue.  If you know, please leave a comment!

EDIT: To keep the YATE log file from growing too large over time, copy the file /usr/src/yate/packing/yate.logrotate into /etc/logrotate.d as “yate” (get rid of the .logrotate extension).  That file instructs the system logrotate job to roll the yate log file when it gets to 100 MB.  Thanks to Bill Simon for that tip!

EDIT 2: If you have ignored the advice given almost everywhere to create a new, separate Gmail account, and then use that account when you create your Google Voice account, then you have probably run into the issue of not receiving your incoming calls when you are logged into that Google account and for some time thereafter.  That problem, and one possible fix (along with the drawbacks) were discussed in a post in the thread “YATE in a Flash 1.2 Ready” on the PBX in a Flash Forum, which unfortunately disappeared from that site due to a server crash.  The post, originally by user Marian on Aug 6, 2012, read as follows:

Gmail sets a greater resource priority when you connect and don’t advertise unavailable for a while after you disconnect.
So, if you connect to GMail using the same account as yate the calls will be sent there until GMail advertise resource unavailable.
You can set priority=10 in accfile.conf, gmail account section.
But, if you do that you might not see your chat in GMail or another jabber client connected to GMail for the same account (like GTalk or Yate Client).
Unfortunately, the jabber protocol don’t allow setting different priorities for the same resource for different services (e.g. you can’t set a priority for chat and another one for another capatibility, like jingle calls).
I didn’t found a workaround for this situation: having, for the same account, a resource for chat and another one for jingle calls.
This would require a custom jabber client or a custom jabber server.

That, coupled with information from other posts around the web, means the best advice is to add a line of the form:

priority=127

in each of your Google Voice accounts in the file accfile.conf (in the /usr/local/etc/yate directory).

If you want that line to be added by default when you add a new Google Voice account to your YATE server, open the add-yate-user script (which is probably in your /root directory) in a text editor such as nano, and find this line:

echo “options=allowplainauth” >> accfile.conf

and underneath it add this:

echo “priority=127″ >> accfile.conf

Then save the edited file.  I make no guarantees that this will actually work, but it’s worth a try. NOTE: The thread mentioned above suggested setting the priority to 10, however, the Asterisk developers are now using 25. As this wiki page explains:

More about Priorities

As many different connections to Google are possible simultaneously via different client mechanisms, it is important to understand the role of priorities in the routing of inbound calls. Proper usage of the priority setting can allow use of a Google account that is not otherwise entirely dedicated to voice services.

With priorities, the higher the setting value, the more any client using that value is preferred as a destination for inbound calls, in deference to any other client with a lower priority value. Known values of commonly used clients include the Gmail chat client, which maintains a priority of 20, and the Windows GTalk client, which uses a priority of 24. The maximum allowable value is 127. Thus, setting one’s priority option for the XMPP peer in res_xmpp.conf to a value higher than 24 will cause inbound calls to flow to Asterisk, even while one is logged into either Gmail or the Windows GTalk client.

Outbound calls are unaffected by the priority setting.

This would be true in Asterisk OR YATE, therefore the recommendation is to now use at least 25 as the priority value, up to the maximum of 127 as suggested above.

Link: Google Voice Customers Cry Out For Help, No One At Google Hears Them

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

The Consumerist is just waking up to a fact that many of us Google Voice users realized a long time ago:  There is virtually no such thing as customer support at Google Voice.  For example, they still haven’t fixed the bug that even if you disable call screening, it’s still turned on if the calls are delivered via Google Chat, and that’s been a problem for at least three or four years now.  Nor have they come up with a way to change the amount of time the call rings at the destination before Google snatches it back and sends it to Google Voice’s voicemail (approximately 25 seconds is just too short in some situations).

The Consumerist article doesn’t touch on either of those specific issues, but at least they’re beginning to understand that the complete lack of effective support at Google Voice can really be a problem:

Google Voice Customers Cry Out For Help, No One At Google Hears Them (The Consumerist via the Wayback Machine)

Yes, I know it’s a free service and some will say you get what you pay for, and I guess that will fly as long as the service remains free, but when they charge for a service (such as the number port mentioned in the article) then they should at least have an effective way to address issues and complaints about the services people have paid for (and perhaps not received)!

Related articles

 

Two things I wish you could do in Asterisk or FreePBX, or ANY free software PBX

 

Important
This is a heavily edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

I want to explain a problem that apparently exists in current implementations of Asterisk and FreePBX (and by extension, all distributions based on those pieces of software).

Let’s say you have several extensions on your system and many, if not all of them, have a specific “trunk” associated with that extension.  It may be a provider account or a Google Voice account that’s used exclusively by that extension.  Routing INCOMING calls is usually not difficult at all, you simply use the trunk’s DID in an Inbound Route and then route the calls from that DID directly to the desired extension.  However, OUTBOUND is another matter.  You have to create an Outbound Route, and in that route you have to put your dial patterns and use the /extension suffix. It can still be difficult to set up the dial patterns the way you need them.  In 2.8 and later it is much harder because of the individual boxes for each segment of each pattern.

Let’s say you want certain extensions to only be able to call numbers in U.S. area codes, but each of those extensions has its own trunk. And let’s say your extensions are numbered 1000 through 1099. Oh, and you want to support both 10 and 11 digit dialing. So in your outbound route for extension 1000, you might have a list of patterns like this (please scroll down to the end of this long list — it’s only about 600 lines!):

1201NXXXXXX/1000
1202NXXXXXX/1000
1203NXXXXXX/1000
1205NXXXXXX/1000
1206NXXXXXX/1000
1207NXXXXXX/1000
1208NXXXXXX/1000
1209NXXXXXX/1000
1210NXXXXXX/1000
1212NXXXXXX/1000
1213NXXXXXX/1000
1214NXXXXXX/1000
1215NXXXXXX/1000
1216NXXXXXX/1000
1217NXXXXXX/1000
1218NXXXXXX/1000
1219NXXXXXX/1000
1224NXXXXXX/1000
1225NXXXXXX/1000
1228NXXXXXX/1000
1229NXXXXXX/1000
1231NXXXXXX/1000
1234NXXXXXX/1000
1239NXXXXXX/1000
1240NXXXXXX/1000
1248NXXXXXX/1000
1251NXXXXXX/1000
1252NXXXXXX/1000
1253NXXXXXX/1000
1254NXXXXXX/1000
1256NXXXXXX/1000
1260NXXXXXX/1000
1262NXXXXXX/1000
1267NXXXXXX/1000
1269NXXXXXX/1000
1270NXXXXXX/1000
1274NXXXXXX/1000
1276NXXXXXX/1000
1281NXXXXXX/1000
1301NXXXXXX/1000
1302NXXXXXX/1000
1303NXXXXXX/1000
1304NXXXXXX/1000
1305NXXXXXX/1000
1307NXXXXXX/1000
1308NXXXXXX/1000
1309NXXXXXX/1000
1310NXXXXXX/1000
1312NXXXXXX/1000
1313NXXXXXX/1000
1314NXXXXXX/1000
1315NXXXXXX/1000
1316NXXXXXX/1000
1317NXXXXXX/1000
1318NXXXXXX/1000
1319NXXXXXX/1000
1320NXXXXXX/1000
1321NXXXXXX/1000
1323NXXXXXX/1000
1325NXXXXXX/1000
1327NXXXXXX/1000
1330NXXXXXX/1000
1331NXXXXXX/1000
1334NXXXXXX/1000
1336NXXXXXX/1000
1337NXXXXXX/1000
1339NXXXXXX/1000
1347NXXXXXX/1000
1351NXXXXXX/1000
1352NXXXXXX/1000
1360NXXXXXX/1000
1361NXXXXXX/1000
1364NXXXXXX/1000
1385NXXXXXX/1000
1386NXXXXXX/1000
1401NXXXXXX/1000
1402NXXXXXX/1000
1404NXXXXXX/1000
1405NXXXXXX/1000
1406NXXXXXX/1000
1407NXXXXXX/1000
1408NXXXXXX/1000
1409NXXXXXX/1000
1410NXXXXXX/1000
1412NXXXXXX/1000
1413NXXXXXX/1000
1414NXXXXXX/1000
1415NXXXXXX/1000
1417NXXXXXX/1000
1419NXXXXXX/1000
1423NXXXXXX/1000
1424NXXXXXX/1000
1425NXXXXXX/1000
1430NXXXXXX/1000
1432NXXXXXX/1000
1434NXXXXXX/1000
1435NXXXXXX/1000
1440NXXXXXX/1000
1442NXXXXXX/1000
1443NXXXXXX/1000
1458NXXXXXX/1000
1469NXXXXXX/1000
1470NXXXXXX/1000
1475NXXXXXX/1000
1478NXXXXXX/1000
1479NXXXXXX/1000
1480NXXXXXX/1000
1484NXXXXXX/1000
1501NXXXXXX/1000
1502NXXXXXX/1000
1503NXXXXXX/1000
1504NXXXXXX/1000
1505NXXXXXX/1000
1507NXXXXXX/1000
1508NXXXXXX/1000
1509NXXXXXX/1000
1510NXXXXXX/1000
1512NXXXXXX/1000
1513NXXXXXX/1000
1515NXXXXXX/1000
1516NXXXXXX/1000
1517NXXXXXX/1000
1518NXXXXXX/1000
1520NXXXXXX/1000
1530NXXXXXX/1000
1534NXXXXXX/1000
1539NXXXXXX/1000
1540NXXXXXX/1000
1541NXXXXXX/1000
1551NXXXXXX/1000
1559NXXXXXX/1000
1561NXXXXXX/1000
1562NXXXXXX/1000
1563NXXXXXX/1000
1567NXXXXXX/1000
1570NXXXXXX/1000
1571NXXXXXX/1000
1573NXXXXXX/1000
1574NXXXXXX/1000
1575NXXXXXX/1000
1580NXXXXXX/1000
1585NXXXXXX/1000
1586NXXXXXX/1000
1601NXXXXXX/1000
1602NXXXXXX/1000
1603NXXXXXX/1000
1605NXXXXXX/1000
1606NXXXXXX/1000
1607NXXXXXX/1000
1608NXXXXXX/1000
1609NXXXXXX/1000
1610NXXXXXX/1000
1612NXXXXXX/1000
1614NXXXXXX/1000
1615NXXXXXX/1000
1616NXXXXXX/1000
1617NXXXXXX/1000
1618NXXXXXX/1000
1619NXXXXXX/1000
1620NXXXXXX/1000
1623NXXXXXX/1000
1626NXXXXXX/1000
1630NXXXXXX/1000
1631NXXXXXX/1000
1636NXXXXXX/1000
1641NXXXXXX/1000
1646NXXXXXX/1000
1650NXXXXXX/1000
1651NXXXXXX/1000
1657NXXXXXX/1000
1660NXXXXXX/1000
1661NXXXXXX/1000
1662NXXXXXX/1000
1667NXXXXXX/1000
1669NXXXXXX/1000
1678NXXXXXX/1000
1681NXXXXXX/1000
1682NXXXXXX/1000
1701NXXXXXX/1000
1702NXXXXXX/1000
1703NXXXXXX/1000
1704NXXXXXX/1000
1706NXXXXXX/1000
1707NXXXXXX/1000
1708NXXXXXX/1000
1712NXXXXXX/1000
1713NXXXXXX/1000
1714NXXXXXX/1000
1715NXXXXXX/1000
1716NXXXXXX/1000
1717NXXXXXX/1000
1718NXXXXXX/1000
1719NXXXXXX/1000
1720NXXXXXX/1000
1724NXXXXXX/1000
1727NXXXXXX/1000
1731NXXXXXX/1000
1732NXXXXXX/1000
1734NXXXXXX/1000
1737NXXXXXX/1000
1740NXXXXXX/1000
1747NXXXXXX/1000
1754NXXXXXX/1000
1757NXXXXXX/1000
1760NXXXXXX/1000
1762NXXXXXX/1000
1763NXXXXXX/1000
1765NXXXXXX/1000
1769NXXXXXX/1000
1770NXXXXXX/1000
1772NXXXXXX/1000
1773NXXXXXX/1000
1774NXXXXXX/1000
1775NXXXXXX/1000
1779NXXXXXX/1000
1781NXXXXXX/1000
1785NXXXXXX/1000
1786NXXXXXX/1000
1801NXXXXXX/1000
1802NXXXXXX/1000
1803NXXXXXX/1000
1804NXXXXXX/1000
1805NXXXXXX/1000
1806NXXXXXX/1000
1808NXXXXXX/1000
1810NXXXXXX/1000
1812NXXXXXX/1000
1813NXXXXXX/1000
1814NXXXXXX/1000
1815NXXXXXX/1000
1816NXXXXXX/1000
1817NXXXXXX/1000
1818NXXXXXX/1000
1828NXXXXXX/1000
1830NXXXXXX/1000
1831NXXXXXX/1000
1832NXXXXXX/1000
1843NXXXXXX/1000
1845NXXXXXX/1000
1847NXXXXXX/1000
1848NXXXXXX/1000
1850NXXXXXX/1000
1856NXXXXXX/1000
1857NXXXXXX/1000
1858NXXXXXX/1000
1859NXXXXXX/1000
1860NXXXXXX/1000
1862NXXXXXX/1000
1863NXXXXXX/1000
1864NXXXXXX/1000
1865NXXXXXX/1000
1870NXXXXXX/1000
1872NXXXXXX/1000
1878NXXXXXX/1000
1901NXXXXXX/1000
1903NXXXXXX/1000
1904NXXXXXX/1000
1906NXXXXXX/1000
1907NXXXXXX/1000
1908NXXXXXX/1000
1909NXXXXXX/1000
1910NXXXXXX/1000
1912NXXXXXX/1000
1913NXXXXXX/1000
1914NXXXXXX/1000
1915NXXXXXX/1000
1916NXXXXXX/1000
1917NXXXXXX/1000
1918NXXXXXX/1000
1919NXXXXXX/1000
1920NXXXXXX/1000
1925NXXXXXX/1000
1928NXXXXXX/1000
1929NXXXXXX/1000
1931NXXXXXX/1000
1936NXXXXXX/1000
1937NXXXXXX/1000
1938NXXXXXX/1000
1940NXXXXXX/1000
1941NXXXXXX/1000
1947NXXXXXX/1000
1949NXXXXXX/1000
1951NXXXXXX/1000
1952NXXXXXX/1000
1954NXXXXXX/1000
1956NXXXXXX/1000
1970NXXXXXX/1000
1971NXXXXXX/1000
1972NXXXXXX/1000
1973NXXXXXX/1000
1978NXXXXXX/1000
1979NXXXXXX/1000
1980NXXXXXX/1000
1984NXXXXXX/1000
1985NXXXXXX/1000
1989NXXXXXX/1000
201NXXXXXX/1000
202NXXXXXX/1000
203NXXXXXX/1000
205NXXXXXX/1000
206NXXXXXX/1000
207NXXXXXX/1000
208NXXXXXX/1000
209NXXXXXX/1000
210NXXXXXX/1000
212NXXXXXX/1000
213NXXXXXX/1000
214NXXXXXX/1000
215NXXXXXX/1000
216NXXXXXX/1000
217NXXXXXX/1000
218NXXXXXX/1000
219NXXXXXX/1000
224NXXXXXX/1000
225NXXXXXX/1000
228NXXXXXX/1000
229NXXXXXX/1000
231NXXXXXX/1000
234NXXXXXX/1000
239NXXXXXX/1000
240NXXXXXX/1000
248NXXXXXX/1000
251NXXXXXX/1000
252NXXXXXX/1000
253NXXXXXX/1000
254NXXXXXX/1000
256NXXXXXX/1000
260NXXXXXX/1000
262NXXXXXX/1000
267NXXXXXX/1000
269NXXXXXX/1000
270NXXXXXX/1000
274NXXXXXX/1000
276NXXXXXX/1000
281NXXXXXX/1000
301NXXXXXX/1000
302NXXXXXX/1000
303NXXXXXX/1000
304NXXXXXX/1000
305NXXXXXX/1000
307NXXXXXX/1000
308NXXXXXX/1000
309NXXXXXX/1000
310NXXXXXX/1000
312NXXXXXX/1000
313NXXXXXX/1000
314NXXXXXX/1000
315NXXXXXX/1000
316NXXXXXX/1000
317NXXXXXX/1000
318NXXXXXX/1000
319NXXXXXX/1000
320NXXXXXX/1000
321NXXXXXX/1000
323NXXXXXX/1000
325NXXXXXX/1000
327NXXXXXX/1000
330NXXXXXX/1000
331NXXXXXX/1000
334NXXXXXX/1000
336NXXXXXX/1000
337NXXXXXX/1000
339NXXXXXX/1000
347NXXXXXX/1000
351NXXXXXX/1000
352NXXXXXX/1000
360NXXXXXX/1000
361NXXXXXX/1000
364NXXXXXX/1000
385NXXXXXX/1000
386NXXXXXX/1000
401NXXXXXX/1000
402NXXXXXX/1000
404NXXXXXX/1000
405NXXXXXX/1000
406NXXXXXX/1000
407NXXXXXX/1000
408NXXXXXX/1000
409NXXXXXX/1000
410NXXXXXX/1000
412NXXXXXX/1000
413NXXXXXX/1000
414NXXXXXX/1000
415NXXXXXX/1000
417NXXXXXX/1000
419NXXXXXX/1000
423NXXXXXX/1000
424NXXXXXX/1000
425NXXXXXX/1000
430NXXXXXX/1000
432NXXXXXX/1000
434NXXXXXX/1000
435NXXXXXX/1000
440NXXXXXX/1000
442NXXXXXX/1000
443NXXXXXX/1000
458NXXXXXX/1000
469NXXXXXX/1000
470NXXXXXX/1000
475NXXXXXX/1000
478NXXXXXX/1000
479NXXXXXX/1000
480NXXXXXX/1000
484NXXXXXX/1000
501NXXXXXX/1000
502NXXXXXX/1000
503NXXXXXX/1000
504NXXXXXX/1000
505NXXXXXX/1000
507NXXXXXX/1000
508NXXXXXX/1000
509NXXXXXX/1000
510NXXXXXX/1000
512NXXXXXX/1000
513NXXXXXX/1000
515NXXXXXX/1000
516NXXXXXX/1000
517NXXXXXX/1000
518NXXXXXX/1000
520NXXXXXX/1000
530NXXXXXX/1000
534NXXXXXX/1000
539NXXXXXX/1000
540NXXXXXX/1000
541NXXXXXX/1000
551NXXXXXX/1000
559NXXXXXX/1000
561NXXXXXX/1000
562NXXXXXX/1000
563NXXXXXX/1000
567NXXXXXX/1000
570NXXXXXX/1000
571NXXXXXX/1000
573NXXXXXX/1000
574NXXXXXX/1000
575NXXXXXX/1000
580NXXXXXX/1000
585NXXXXXX/1000
586NXXXXXX/1000
601NXXXXXX/1000
602NXXXXXX/1000
603NXXXXXX/1000
605NXXXXXX/1000
606NXXXXXX/1000
607NXXXXXX/1000
608NXXXXXX/1000
609NXXXXXX/1000
610NXXXXXX/1000
612NXXXXXX/1000
614NXXXXXX/1000
615NXXXXXX/1000
616NXXXXXX/1000
617NXXXXXX/1000
618NXXXXXX/1000
619NXXXXXX/1000
620NXXXXXX/1000
623NXXXXXX/1000
626NXXXXXX/1000
630NXXXXXX/1000
631NXXXXXX/1000
636NXXXXXX/1000
641NXXXXXX/1000
646NXXXXXX/1000
650NXXXXXX/1000
651NXXXXXX/1000
657NXXXXXX/1000
660NXXXXXX/1000
661NXXXXXX/1000
662NXXXXXX/1000
667NXXXXXX/1000
669NXXXXXX/1000
678NXXXXXX/1000
681NXXXXXX/1000
682NXXXXXX/1000
701NXXXXXX/1000
702NXXXXXX/1000
703NXXXXXX/1000
704NXXXXXX/1000
706NXXXXXX/1000
707NXXXXXX/1000
708NXXXXXX/1000
712NXXXXXX/1000
713NXXXXXX/1000
714NXXXXXX/1000
715NXXXXXX/1000
716NXXXXXX/1000
717NXXXXXX/1000
718NXXXXXX/1000
719NXXXXXX/1000
720NXXXXXX/1000
724NXXXXXX/1000
727NXXXXXX/1000
731NXXXXXX/1000
732NXXXXXX/1000
734NXXXXXX/1000
737NXXXXXX/1000
740NXXXXXX/1000
747NXXXXXX/1000
754NXXXXXX/1000
757NXXXXXX/1000
760NXXXXXX/1000
762NXXXXXX/1000
763NXXXXXX/1000
765NXXXXXX/1000
769NXXXXXX/1000
770NXXXXXX/1000
772NXXXXXX/1000
773NXXXXXX/1000
774NXXXXXX/1000
775NXXXXXX/1000
779NXXXXXX/1000
781NXXXXXX/1000
785NXXXXXX/1000
786NXXXXXX/1000
801NXXXXXX/1000
802NXXXXXX/1000
803NXXXXXX/1000
804NXXXXXX/1000
805NXXXXXX/1000
806NXXXXXX/1000
808NXXXXXX/1000
810NXXXXXX/1000
812NXXXXXX/1000
813NXXXXXX/1000
814NXXXXXX/1000
815NXXXXXX/1000
816NXXXXXX/1000
817NXXXXXX/1000
818NXXXXXX/1000
828NXXXXXX/1000
830NXXXXXX/1000
831NXXXXXX/1000
832NXXXXXX/1000
843NXXXXXX/1000
845NXXXXXX/1000
847NXXXXXX/1000
848NXXXXXX/1000
850NXXXXXX/1000
856NXXXXXX/1000
857NXXXXXX/1000
858NXXXXXX/1000
859NXXXXXX/1000
860NXXXXXX/1000
862NXXXXXX/1000
863NXXXXXX/1000
864NXXXXXX/1000
865NXXXXXX/1000
870NXXXXXX/1000
872NXXXXXX/1000
878NXXXXXX/1000
901NXXXXXX/1000
903NXXXXXX/1000
904NXXXXXX/1000
906NXXXXXX/1000
907NXXXXXX/1000
908NXXXXXX/1000
909NXXXXXX/1000
910NXXXXXX/1000
912NXXXXXX/1000
913NXXXXXX/1000
914NXXXXXX/1000
915NXXXXXX/1000
916NXXXXXX/1000
917NXXXXXX/1000
918NXXXXXX/1000
919NXXXXXX/1000
920NXXXXXX/1000
925NXXXXXX/1000
928NXXXXXX/1000
929NXXXXXX/1000
931NXXXXXX/1000
936NXXXXXX/1000
937NXXXXXX/1000
938NXXXXXX/1000
940NXXXXXX/1000
941NXXXXXX/1000
947NXXXXXX/1000
949NXXXXXX/1000
951NXXXXXX/1000
952NXXXXXX/1000
954NXXXXXX/1000
956NXXXXXX/1000
970NXXXXXX/1000
971NXXXXXX/1000
972NXXXXXX/1000
973NXXXXXX/1000
978NXXXXXX/1000
979NXXXXXX/1000
980NXXXXXX/1000
984NXXXXXX/1000
985NXXXXXX/1000
989NXXXXXX/1000

(Note the above does not include the “toll free” area codes nor Canadian area codes; I have separate routes for those).

Now THAT is bad enough, but then imagine having to duplicate this list for each of your extensions (changing only the extension number after the / character), because each will need its own outbound route in order to select its own trunk. In pre-2.8 versions of Asterisk, you could simply copy this list into a text editor, do a search and replace on the /1000 (replacing it with the next extension number), and paste the changed list into a new outbound route. However, with the new way of entering dial plans, you have to enter each line in each field manually, OR (in 2.9 and later) mess with .CSV files, which although easier than manual entry are still a lot harder to deal with than simple cut-and-paste.

But that is actually not the subject of this article; it just sets the stage for what I’m thinking SHOULD be part of Asterisk (or any other soft PBX that requires entering patterns in a similar manner, that is, one line for each pattern). There are actually TWO ways this could be handled, but neither will work at present, as far as I know.

1) Stacking Routes

Let’s suppose you had an outbound route that had all the USA patterns, but did NOT include the extension field. You could have it near the top of your Outbound Route list. And let’s say that you could make the destination of that trunk another “group” of outbound routes rather than a trunk. In that second group, you could have routes with just two patterns per extension:

1XXXXXXXXXX/1000
XXXXXXXXXX/1000

So the call would be effectively pre-screened in the first (primary) group of outbound routes, then sent to the second group (NOT part of the primary group) which would route by extension. That way, you’d only need ONE route with a list of USA patterns, one route with a list of Canada patterns, one route with a list of “toll free” patterns, etc. Each could go directly to a trunk, or to a secondary group of outbound routes.

I think Asterisk might actually be capable of doing something like this (though I’m uncertain of that), but FreePBX definitely is not. So some FreePBX users literally have THOUSANDS of lines of dial patterns in their configuration. Does this slow things down? You betcha, at least when making a configuration change! It takes forever for that darn frog to stop eating flies (if a real frog ate that many flies in that short a time, its gut would probably explode!).

2) Macros

Now here we have a solution that would likely need to be implemented in Asterisk itself. The basic idea is to allow macros in dial patterns. For example, you create a list such as the one above (but without the /extension field – just the number patterns only) and call it [pattern-USA]. Then in your outbound routes, you do something like this:

[pattern-USA]/1000

Changing the extension as needed for each Outbound Route. As noted, this would require implementing this type of macro feature in Asterisk, but it would also necessitate a way to turn off the syntax checking in FreePBX, which is currently impossible.

EDIT: For another way to handle this that probably will work, see How to use the FreePBX [macro-dialout-trunk-predial-hook] macro and regular expressions to blacklist or whitelist outgoing calls on all trunks.

I no longer recommend using Asterisk’s Google Voice support — try these methods instead!

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

EDIT (May, 2018): FreePBX and Asterisk users that wish to continue using Google Voice after Google drops XMPP support should go here: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware.

This article was originally written in January of 2012, and has been heavily edited in an attempt to update it a bit.

Not that anyone probably cares what I think, but anyone who regularly reads this blog (or any of the other VoIP-related blog that cover Asterisk) may have noticed that prior to the release of Asterisk 11, Asterisk’s support for Google Voice had become less and less reliable over time, particularly for incoming calls. You have to do all sorts of “tricks” to make it work, and these usually involve adding delays that don’t always fix the problem, inconvenience your callers, and possibly cause more hangups as people get tired of waiting for you to answer the phone.

Therefore, I suggest that if you are using a version of Asterisk earlier than Asterisk 11, you stop using Asterisk’s Google Voice support completely. Assuming that you feel you must keep using an older version of Asterisk, I suggest trying one or more of the following:

  1. Use YATE as a gateway between Asterisk and Google Voice. See Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk, this article and this forum thread on YATE in a Flash, and this thread on YATE Tips & Tricks). YATE is what powers Bill Simon’s gateway (mentioned below). See comments by Bill and pianoquintet under this article.
  2. Use Bill Simon’s Google Voice-SIP gateway to handle your Google Voice calls. Some people may not want to rely on an external service for this, while others may very much appreciate having the option. I mention it for those in the latter group. For more information see Bill Simon’s Free SIP-to-XMPP Gateway Easily Puts Google Voice on Your VoIP Phone (Voxilla). While the linked articles talk about using the gateway with a SIP device, it can be used as an Asterisk trunk.  EDIT: As of April 7, 2015 the Google Voice Gateway has been relaunched and there is now a one-time fee to sign up.
  3. If your only issue is with incoming calls, you could use a DID to bring the calls into your system.  But keep in mind that Google Voice does not like it when calls are answered the moment they connect, so in your FreePBX Inbound Route be sure to set the “Pause Before Answer” option to at least 1.  I have found that a 1 second pause is sufficient, but I’m not saying that is the correct value for everyone, or even that everyone will need to include such a pause (some DID providers may delay the call sufficiently before connecting through to your system that the pause isn’t needed).

At this point, any of those would likely produce better results than using the Google Voice support in any version of Asterisk prior to Asterisk 11.

EVERYTHING in this article is my personal opinion.  Nothing here should be taken as a statement of fact.

EDIT:  Ward Mundy reports that he just may have found a workaround for the incoming calls issue — see this thread in the PBX in a Flash forum.

Asterisk 1.8.x and FreePBX users: How to NOT answer Google Voice calls UNTIL the called extension answers

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

EDIT (May, 2018): FreePBX and Asterisk users that wish to continue using Google Voice after Google drops XMPP support should go here: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware.

This article was originally published in December, 2010 and may contain out-of-date information.

Many folks are experimenting with Asterisk 1.8.x and Google Voice.  In most cases the way it’s set up is that when a Google Voice call arrives, Asterisk answers the call, then sends a touch-tone digit “1” to Google Voice to answer the call, then proceeds to ring the destination extension.  This is necessary because when you configure Google Voice to use a Gtalk destination, they require you to press “1” to accept the call, even if you’ve configured Google Voice not to require that.  I don’t know if this is a bug in Google Voice or if they did it that way deliberately for some reason, but answering the call and accepting it upon arrival at the PBX has a few unintended side effects:

  • If your callers pay for long distance by the minute, they get charged from the moment the called extension begins ringing – even if you never answer the call.
  • You can’t use Google Voice’s Voicemail, nor their transcription service, because you’ve already answered the call.
  • Callers may hear a confusing double ringing tone at the start of ringing — one ring from Google Voice and the rest from Asterisk.

On the other hand, there are some advantages to doing it that way:

  • Because you’ve answered the call, you can let the extension ring as long as you like before sending it to voicemail, and Google Voice won’t snatch it away in 25 seconds and send it to their voicemail.
  • You can use Asterisk’s voicemail, if that’s what you prefer.

For those who’d prefer to let Google Voice handle their voicemail, or who object to making callers pay to listen to up to 25 seconds of ringing, there is a way to not answer the call and send the touch tone “1” until  after the destination extension has actually picked up the call.  If you are using plain vanilla Asterisk, all you have to do is make sure your Dial() command contains two additional options.  Consider this example line of Asterisk dialplan:

exten => gvoicein,n,Dial(SIP/1004,35,rTWtwaD(:1))

The important part here is the aD(:1) — the other options can be whatever you’d normally use, if any, but it’s the aD(:1) that does the magic. Now at this point, if you’re a FreePBX user you may be wondering how on earth you can modify the Dial() string, since the code that generates it is buried deep within the bowels of FreePBX. Fortunately, there is a way. Consider the following piece of code that might be used in extensions_custom.conf to bring in Google Voice calls:

[googlein]
exten => _[0-9a-z].,1,Noop(Incoming Google Voice call for ${EXTEN})
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten => _[0-9a-z].,n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim)
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CALLERID(name):2})
exten => _[0-9a-z].,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,Answer
exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,SendDTMF(1)
exten => _[0-9a-z].,n,Goto(from-trunk,gv-incoming-${CUT(EXTEN,@,1)},1)
exten => h,1,Macro(hangupcall,)

With this context you’d use gv-incoming-username (where username is the part of the associated gmail address before the @) as the DID in your inbound route — a DID doesn’t have to be numeric even if FreePBX whines about it, and the advantage is you only need one context to handle incoming calls for all your Google Voice accounts.  This particular context is slightly modified from one found in the PBX in a Flash forum, but note that it contains these four lines that wait ONE second, answer the call, wait ONE second (you do NOT have to wait two seconds, despite what any other article may say, and in fact the one second wait might be unnecessary), and then send the touch tone digit 1:

exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,Answer
exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,SendDTMF(1)

You will find those four lines, or some variation on them (sometimes just the last three), in just about every published method for using Google Voice with Asterisk and FreePBX.  But, in FreePBX at least, you can replace them with this:

exten => _[0-9a-z].,n,Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))

This slides the aD(:1) into the options that will be used with the Dial command, so when the extension answers, the call will be answered and then the touch tone “1” will be immediately sent to Google Voice, and then the audio between Google Voice and the called extension will be bridged as usual.

Unfortunately, or maybe fortunately depending on your point of view, it appears that if the call should go to Asterisk’s voicemail, the call will not be answered and the DTMF 1 will never be sent.  This means that if, for whatever reason, you don’t answer the incoming call, after 25 seconds it will go to Google’s voicemail.  There are doubtless ways around that (and if anyone’s truly interested, leave a comment and I’ll suggest a way that may work, that involves routing the incoming call to a ring group first) but I suspect that the majority of people who want to do this will be doing it because they want to use Gmail’s voicemail.

I’ve tested this and it works for me, though I would not use it on a regular basis because I prefer Asterisk’s voicemail.  If it doesn’t work for you for some reason, the only suggestion I can offer is adding a w before the :1, so the added options look like aD(w:1) – that will add a one-half second delay before the “1” is sent, and more than likely it won’t help one bit, but may cause callers to not hear your “hello” or other greeting.  But, you can try it and see — at least one user has reported it to be necessary.  If that doesn’t work, I probably won’t be able to help you but if you leave a comment, maybe someone else can.

And, should anyone from Google Voice read this, it would be really helpful if you’d do two things:

  1. Give us a way to disable Google Voice’s voicemail so we don’t have to resort to hacks like this to discourage callers from leaving a message there.
  2. Fix the bug (or “feature”) so that when we turn off call screening, it’s off for ALL destinations, including Gtalk!

How to stop people from leaving messages in your Google Voice voicemail box

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

One irritating thing about Google Voice is their voicemail – they must be really proud of it, because they give you no way to turn it off (at least not that I can find), and worse yet, they don’t even let you pick the timeout before your Google Voice calls go to voicemail.  If people actually had to pay for their service I’m sure they’d get boatloads of complaints, but since it’s free and since Google doesn’t seem to care much what users think, we are stuck with their voicemail whether we want it or not. However, there are some situations where you really don’t want to have to check Google’s voicemail, so how do you discourage people from leaving a message?

The answer is simple, and probably 99.9% effective: Change your greeting to a busy signal! Most people, and even most automated calling equipment, will hang up after receiving a busy signal, and will not leave a message (and if anyone does, it’s probably a stupid robo-caller that you can safely ignore).

So how do you change your greeting in Google Voice? Glad you asked…

To start with, download this audio file (right-click on the link and save it to your hard drive). It is 24 seconds of North American busy signal, followed by the DTMF # button (the latter is needed to signal the end of the greeting).

Now, what you have to do is go to your Google Voice settings, Voicemail & Text tab, and in the Voicemail Greeting section click on “Use phone to record a new greeting.” I suggest you try this once or twice just to get the hang of how it works (you can delete any recording you make on the same page). Note that once you’ve recorded your first greeting, the button will change to say just “Record new.”

Google Voice - Location of "Record New Greeting" button

Now, the obvious thing to do here is to hold the phone up to your computer’s speaker and play the audio file after Google calls you to record the greeting. If you’re careful about your volume levels it might work, but I don’t recommend it.

A better method is to temporarily redirect your Google Voice callback so that it comes to a softphone on your computer, preferably one that has the ability to select audio inputs and outputs.  How to do that is left as an exercise for the reader, but I can tell you that the free version of Zopier will allow you to select inputs. You’ll also need an audio program that can play back .wav files and allows you to select outputs (an example for the Mac would be Vox), and depending on your computer, you may need a third piece of software that allows you to redirect the output of one program to the input of another (for example, on a Mac you can use Soundflower). On a Mac you’d go into the audio player preferences, set the output to go to the redirection software, then set the softphone to get its audio input from the redirection software.

Vox audio player output settings
Zoiper softphone audio input settings

After doing this, get Google Voice to call your softphone, answer the call and immediately click “play” on the audio player software (into which you will have pre-loaded the audio file) and if all goes well it will play the file and at the end, after the DTMF # tone is played, Google Voice should play back the file, and ask you to accept it or try again. If you hear some ringing tone in the playback prior to the busy signal, just click “2” on Zoiper’s touch-tone keypad to try again (when you are given that option) and then as soon as you hear the beep, click “play” again. When it’s right, click the “1” on Zopier’s touch tone keypad when given the option. You can confirm that this is set as the default from the Google Voice page mentioned above. Also, note that by using the Google Voice “Groups” feature, you can play this “greeting” only to certain callers, if for some reason you want some people to actually be able to leave you voicemail in your Google Voice voicemail box.

I know that Zoiper is cross-platform and can be used under all major operating systems, and I’m pretty sure there are audio redirection programs available for Windows and Linux, though I don’t know what they are called offhand (if you do, please feel free to leave a comment, provided the software you suggest is free to use — I’m not going to promote any commercial software for this purpose because I’m pretty sure there is free software available, and in any case, it may even be possible to achieve this function from the Windows sound control panel, but it’s been so long since I’ve tried to do anything like this in Windows that I don’t remember how it’s done, I just know it’s possible).

Related Link:
Proof of concept: Automatically transfer Google Voice voicemail to Asterisk voicemail

Link: Using FreeSWITCH to add Google Voice to Asterisk

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

EDIT (2018): This article is extremely out-of-date and in no way useful today, and will probably be removed from this site at some point in the future. You might find this article more useful: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware.

For those of you using Asterisk, Bill over at the PSU VoIP blog has come up with a way to interface Asterisk with Google Voice, by co-installing FreeSWITCH (which also supports Google Voice).  Turns out that Asterisk and FreeSWITCH can co-exist on the same server, though you do have to change the configuration a bit so they don’t compete for the same ports.  Anyway, Bill has come up with a how-to on adding Google Voice integration to current versions of Asterisk, so if that interests you, head on over and have a look:

Using FreeSWITCH to add Google Voice to Asterisk

The bonus is that once you get FreeSWITCH installed you can play around with it and look at some of its other features, if you are so inclined. Of course, the Asterisk folks could backport the Google Voice support to previous versions and make it unnecessary to do things like this, but I’m not holding my breath.

EDIT (January 26, 2012): The Google Voice channel drivers in Asterisk 1.8 have become unreliable enough (in my personal opinion, anyway) that I just used the technique shown in this article, and I must say that it works a LOT better than Asterisk 1.8’s Google Voice support.  I also added some comments to that article (probably too many!) that among other things show how I got it working for multiple Google Voice accounts.  So I would now recommend using this method to bridge Asterisk to Google Voice in preference to using Asterisk 1.8’s native channel drivers (unless you are very short on memory and/or storage space) — it just works, and calls connect faster.  Read the article AND the comments under it first, so you’ll know what to expect, and do be aware that it takes a relatively LONG time to compile and install FreeSWITCH (compared to Asterisk).  At points during the installation it may look like it’s stuck in an endless loop, but it really isn’t. Just go away and take a walk outside or something, and come back in a while and it should be done.

Related articles