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Posts Tagged ‘Asterisk’

Exchange 2010 UM with FreePBX 2.11 and Asterisk 11

October 14th, 2013

This article assumes that you have an Exchange UM server already configured. The directions cover the use of a module I created for FreePBX 2.11 and assume FreePBX is in device and user mode. The module removes the need to override macro-vm, which allows you to use Exchange UM for some users and FreePBX voicemail for others.

NOTE: Support for the MWI (message waiting indicator) and play on phone requires patching and compiling Asterisk from source. The patches were created against Certified Asterisk 11.2-cert2. Read more…

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Asterisk 1.8 Caller ID Wrong in CDR

October 23rd, 2011

After upgrading to Asterisk 1.8 from 1.6.1 I noticed the CALLERID() function was not updating the clid and src fields in the CDR. Previously on Asterisk 1.6.1 with FreePBX 2.7 the clid and src would be set to the outbound cid. With Asterisk 1.8 those fields were staying as the sip device id. Read more…

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Asterisk Called Party Name

July 2nd, 2010

With a traditional PBX when dialing an extension the name is displayed on the phone. With Asterisk 1.4 and 1.6.x the name is not displayed even though phones, like the Polycom SoundPoint series, have support for it with the Remote-Party-ID SIP header. Read more…

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Polycom Soft Keys and Exchange UM

March 6th, 2010

In Asterisk 1.6, FreePBX, and Exchange UM I mentioned how to setup an extension to transfer a caller to voicemail. Let’s take this one step further and add a soft key on the Polycom. Instead of having to dial ##407ext, you will be able to press the Xfer VM button and it will prompt for the extension. Read more…

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Polycom BLF with Astersisk

March 6th, 2010

They are two methods of setting up BLF on Polycom phones. The first gives you idle or inuse status only. A ringing phone will show up with a solid inuse light. The good thing about this method is users can create BLF for extensions themselves. If this is all you need start by enabling the presence feature in the phone provisioning file. Read more…

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Asterisk 1.6, FreePBX and Exchange UM

February 28th, 2010

Setting up Asterisk 1.6.1 and FreePBX 2.5 or 2.6 to work with Exchange 2007 UM is easier than Asterisk 1.4 thanks to Asterisk 1.6 including support for SIP over TCP. However only Asterisk 1.6.1.4 and lower work without modification. Read more…

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Asterisk and FreePBX Caller Initiated Blind Transfers

February 28th, 2010

If you place a call and perform a blind transfer Asterisk will continue dialplan execution with a result of ANSWER. This causes an issue with FreePBX as it thinks dialing on the trunk failed. The transfer will go through, but you receive the all circuits are busy message. Read more…

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