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 The information in this article is VERY outdated and probably will not work.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.

10 thoughts on “Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk

  1. Poor caller ID name lookups are the reason that I have not tried to use a local YATE gateway with the free PBX that you know and hate. I will be watching to see if Free Switch does better. For now, Bill Simon’s GV gateway gives excellent caller ID name info.

    Moderator edit: Corrected spelling of Bill Simon’s name and added link.

  2. Looks like they tried to fix the bug in FreeSWITCH and failed spectacularly – see my edit in the next to last paragraph. Meanwhile, YATE continues to work like a champ.

  3. Also Asterisk/Digium have restarted their efforts to make Google Talk work in Asterisk 11 with ‘chan_motif’. From my tests it worked fine behind two or three firewalls with no issues.

    1. That’s great to hear, however I wonder if there will ever be a decent web-based GUI that works with Asterisk 11, particularly one that does not operate under the assumption that the bulk of its users are rank beginners or have low I.Q.’s.

  4. I am looking for a way to get callerid lookup in Yate as well; If Bill Simon ‘s GVGW gives a good CID lookup (perhaps he uses superfecta or some such thing on a different server?), there probably is a way to get CID lookup working in Yate itself..

  5. Hi, We have no open issues involving google voice calls not working.
    I think we had some open jiras with some configuration issues a month or so back but they are all resolved…..

    1. Anthony, that’s good to hear. At some point, probably after the start of the new year (assuming that the end of the Mayan calendar does not mean the end of the world), I hope to take a look at FusionPBX and/or blue.box again. The issue you guys have is that sometimes the GUI installers don’t install the latest version of FreeSWITCH, and if you try to upgrade FreeSWITCH to the latest version then the GUI breaks (and there is no easy path to roll back to the previous version).

      I’ll actually be interested to see what the blue.box people do in their new version. I understand it’s going to be a significant rewrite so just maybe I can figure it out. Right at this moment I like the FusionPBX GUI better but I had enough frustration in actually trying to make it work as it should (and in the fact that they have no forum where you can view previously-resolved issues) that I reluctantly gave up on it for now.

  6. NOTICE: All comments above this one were imported from the original Michigan Telephone Blog and may or may not be relevant to the edited article above.

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