Category Archive: YATE

Apr 16 2015

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 …

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Feb 25 2015

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 …

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Jan 13 2014

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 …

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Apr 06 2013

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 …

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Oct 18 2012

Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk

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 …

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Sep 09 2012

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

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 …

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Jan 26 2012

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

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 …

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Nov 16 2007

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

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 …

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