Category: YATE

Stop SOME SipVicious attacks from reaching your Asterisk, FreeSwitch, YATE, etc. PBX server

This tip was posted by user “infotek” on the FreePBX site but applies to all software PBX systems that use the iptables firewall. “infotek” wrote:

By default the SipVicious scanner uses the ua : “friendly-scanner”. To block this ua, you can have iptables search the packet for that text.

add the following line to /etc/sysconfig/iptables

-A INPUT -p udp -m udp –dport 5060 -m string –string “friendly-scanner” –algo bm –to 500 -j DROP

Now the thing to keep in mind about this is that it only works if you know the string that will be sent as the user agent, and some hackers using SipVicious may take the trouble to change that default string, but some protection is better than none.  However this same technique can be used to block any attack that constantly sends the same string as the user agent, if you know what that string contains.

For those that use Webmin to manage iptables, here are the settings to use.  This should come BEFORE any other rules applicable to port 5060 – I made it the very first rule on the page “Incoming packets (INPUT) – Only applies to packets addressed to this host“:

Rule comment: Stop SipVicious
Action to take: Drop
Network protocol: Equals UDP
Destination TCP or UDP port: Equals Port(s) 5060
Additional IPtables modules: string
Additional parameters: –string “friendly-scanner” –algo bm –to 500

All other settings on the Webmin “Add Rule” page should be left at the default value (usually <ignored>).

To stop the hackers clever enough to change the default user agent string, consider also using this technique.

How to set up an alternate SIP port (other than 5060) using Webmin

One problem that some VoIP users are experiencing these days is that they have trouble connecting to their home Asterisk, FreeSWITCH, YATE, or other software PBX server, but only when using certain ISP’s.  One suspicion is that certain ISP’s that offer their own VoIP or traditional landline service attempt to mess with packets using the common SIP port 5060, hoping customers will think that VoIP is unreliable and will subscribe to the company’s overpriced offering instead (see this thread at BroadbandReports.com).

There are various ways to enable an alternate SIP port on the server (in addition to the usual port 5060) but if you are using Webmin to manage your firewall, here’s an easy way, in just three steps:

Step 1:

If you use Webmin to manage your firewall, then you already know how to get to the Linux Firewall page.  So go there and select the Network Address Translation (nat) table in the dropdown at the top of the page.  Then when the page changes, click the topmost Add Rule button (in the Packets before routing (PREROUTING) section):

SIP Port Forwarding 1

Step 2:

Now you should see this page. The items you need to change are indicated by the red ovals.  The Destination TCP or UDP port is set to 7654 in this example, but don’t use that.  Pick your own unique port; just make sure that it’s not used by anything else on the server already.  And yes, you really do put the alternate SIP port you want to use in the Destination setting; it may not make intuitive sense but that’s just how it is.  Avoid using ports in the range 10000 through 20000 because those are used for RTP traffic, and avoid ports below 1024 because those are protected ports that are reserved by the system.  There are also other ports you should avoid (those used by other software on your system) but if you don’t know how to find ports in use on your system, a bit of time with a search engine will lead you to several pages that show you how to detect already active ports.  It can vary a bit depending on your operating system, and it’s beyond the scope of this article.

SIP traffic is UDP only, not TCP (there may be rare exceptions but most software PBX’s use UDP by default).  If by some very odd chance you are using TCP for SIP traffic (why?!?) then you will need to specify that under the Network Protocol item.  Also, note that the incoming interface is set to eth0 in this example – Webmin will usually show the correct one by default, but you want to select the interface used by incoming SIP traffic if you happen to have more than one.  Don’t forget to click Save when you’re finished.

SIP Port Forwarding 2

Step 3:

After you have saved the page you should be back at the page you came from, and it should now show your new forwarding rule.  If it does, just click Apply Configuration at the bottom of the page to make it active:

SIP Port Forwarding 3One other thing you might need to to is add a rule on the main Linux Firewall page allowing incoming traffic on your selected alternate port, if you have a restrictive firewall that blocks most traffic by default.  Try it first without doing that, but if your clients can’t connect on the alternate port, just be aware that you may need to do that before they will be able to connect.  For more security, you can enable access to your SIP ports only from specific IP addresses, if your remote clients are at fixed IP addresses.

Now you should be able to change the port number on your SIP endpoints from the default 5060 to your alternate port.  Of course I can’t guarantee it will always work, but if you try it and suddenly find that your SIP connections become far more reliable, you might want to leave a comment, and mention the ISP that you suspect might be messing with SIP traffic.

By the way, if you are NOT using Webmin to manage your firewall, and have iptables installed, then all you should need to do is enter these two lines from a command prompt (replace 7654 with your chosen alternate port).  But I strongly caution you NOT to do this if you are using Webmin to manage your firewall:

iptables -t nat -A PREROUTING -i eth0 -p udp --dport 7654 -j REDIRECT --to-ports 5060
/etc/init.d/iptables save

A possible way to thwart SIP hack attempts on your Asterisk (or other) PBX server

If you’ve had the problem of hackers trying to break into your Asterisk server, you probably know that you can use tools like Fail2ban to at least slow them down.  But why let them know you even have an Asterisk server in the first place?  Maybe you need to leave port 5060 open so that remote users (not on your local network) can connect to the server, but that doesn’t mean that you have to advertise to the bad guys that you might have something of interest.  With that in mind, we direct your attention to this post in the DSLReports VoIP forum:

The Linux netfilter/iptables firewall is capable of stopping these attacks before they even start.

At a bare minimum, this stops 99% of the attacks when added to your iptables ruleset:

-A INPUT -p udp --dport 5060 -m state --state ESTABLISHED,RELATED -j ACCEPT
-A INPUT -p udp --dport 5060 -m string --string "REGISTER sip:your.pbx.dns.name" --algo bm -j ACCEPT
-A INPUT -p udp --dport 5060 -m string --string "REGISTER sip:" --algo bm -j DROP
-A INPUT -p udp --dport 5060 -m string --string "OPTIONS sip:" --algo bm -j DROP
-A INPUT -p udp --dport 5060 -j ACCEPT

Warning
IMPORTANT: Be sure to have a separate iptables rule (higher on the list than those above) that allows connections to port 5060 from devices on your local network. Otherwise, you may find that new extensions that you are adding for the first time will not register with your Asterisk server, or that after a system reboot, none of your local extensions will register!

To understand how this works, read the original post by DSLReports user espaeth.

For another line of defense against such attacks, see the article Stop SOME SipVicious attacks from reaching your Asterisk, FreeSwitch, YATE, etc. PBX server.

Not receiving some incoming Google Voice calls? Try increasing the priority

A page on the Asterisk Wiki entitled Calling using Google contains this bit of information about priorities:

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.

Therefore, if you are not receiving some incoming Google Voice calls, check to make sure that your priority= statement in the configuration is set to at least 25.

Read more: Calling using Google

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.

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
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(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.

Linksys and Sipura adapter users – check your RTP Packet Size and Network Jitter Level

 

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: Reader Christopher Woods notes in a comment that the following is also applicable to at least some models of Linksys phones, e.g. SPA942 and SPA962.

Do you use a Linksys or Sipura VoIP adapter? Do the people you are talking to ever complain about your voice breaking up, or missing or dropped syllables, or unexplained clicks or noise?

There is an obscure setting in Linksys/Sipura VoIP adapters that is usually set incorrectly for most applications, at least on a factory-fresh adapter. Go to the SIP tab and check the RTP Packet Size – for most users, it should be set to 0.020 rather than the factory preset of 0.030. If you are running a connection where latency is critical (say you have a cable or satellite box that requires a phone connection to “phone home”, or you are trying to use a FAX machine) then you may even wish to set this to 0.010, which further reduces latency, at the expense of using a bit more bandwidth. In any case, the default 0.030 is not the correct setting when using the most commonly-used codecs. For more discussion of this issue, see this thread at DSLreports.com, which discusses how the RTP Packet Size and Network Jitter Level settings can be tweaked to achieve lower latency, along with the tradeoffs.

Be aware that the RTP Packet Size setting is found under the SIP tab, and that setting is applied to all lines served through that adapter. However, the Network Jitter Level can be set individually for each line, under the Line tabs. One interesting comment in the above-mentioned thread is that if a provider forces you to use a low-bandwidth codec, decreasing the RTP Packet Size may increase the quality of your calls, but again at the expense of increasing bandwidth used.

Changing the RTP Packet Size on one VoIP adapter resolved a few strange issues with audio quality. In this case the adapter was being used to connect to an Asterisk box on the same local network, so bandwidth usage wasn’t an issue. We set the RTP Packet Size to 0.020 and the Network Jitter Level to low, and it made a noticeable difference in the reduction of strange noises and breakups heard by the party on the other end of the conversation. However, changing the Network Jitter Level isn’t as critical as changing the RTP Packet Size, and in fact, changing the Network Jitter Level may be entirely the wrong thing to do on certain types of connections (probably not a good idea if your adapter is connected through a Wireless ISP, for example).

I must thank Paul Timmins for being the first to point out that the Linksys PAP2 has a default packet size of 0.030, which is incompatible with the uLaw (G711u) codec (or at least in violation of the standard). With that lead, I then discovered other articles (including the discussion thread linked above) that said essentially the same thing. So check those adapter settings, folks!

(And by the way, this advice probably does apply to some other makes of VoIP adapters, and even some IP telephones, but since I don’t have any readily available to look at, I can’t say for sure. If you know of any others that need to have a similar setting tweaked, please feel free to add a comment to this post).

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