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showing posts tagged with 'asterisk'
 
by lunarg on August 4th 2017, at 16:17

An excellent resources website on how to configure Cisco phones to work on Asterisk. This includes the provisioning of phones, configuring them in Asterisk and enabling advanced functionality for Cisco on Asterisk.

http://usecallmanager.nz/document-overview.html

by lunarg on September 30th 2013, at 09:51
You can change/set the From-addresses in FreePBX to something you want.

Change the address in Admin » System Admin. To the right, click on Notification Settings, then fill in the From Address.

These are set up from within Asterisk, and are located at Settings » Voicemail Admin. Click on the Settings link, then scroll down until you find serveremail. Change it to whatever you want.

Next, because it uses Postfix to e-mail out voicemail notifications, you'll also have to change a parameter in there:

Open a shell to your box (SSH).

Type nano /etc/postfix/main.cf.

Find and change, or add a line: mydomain your-domain-name.com. Fill in the domain you want. It should be the same as  ...
by lunarg on November 28th 2012, at 13:02

An Asterisk TAPI driver, which is actually usable on Terminal Server:

http://www.xtelsio.com/en/products/ast_tsp/tapi_for_asterisk.htm

It's not freeware, but is absolutely affordable, and is usable on Terminal Server (multi-user profiles).

by lunarg on November 4th 2011, at 14:51
You can remote-reboot your Polycom phones from the Asterisk CLI by sending a SIP NOTIFY command. This command instructs the phone to recheck its configuration, and therefore, will reboot when certain parameters have been set in your phone configuration provisioning.

Make sure the following directive is set in your provisioning config:

<specialEvent voIpProt.SIP.specialEvent.checkSync.alwaysReboot="1" />

You find this in sip.cfg, or you can set it in your own configuration (mine's called aaps-settings.cfg, but that's just an arbitrary name).

Notice
When setting this variable in your own configuration, be sure that it's set in the right context, like so:<localcfg> &l  ...
by lunarg on November 6th 2007, at 22:57
The trouble with PSTN lines is that the CID has to be transmitted in-band, and there are more than one standard in doing this. On top of that, there are small differences between the various Telcos on the planet. And finally, to make things worse, documentation about it, is sparse and scattered at best.

The settings displayed usually go in /etc/asterisk/zapata.conf (either directly, or by inclusion).

How CID is handled is defined by the following variables:

cidsignalling: tells how the CID signalling occurs, can be bellcore (mostly US hardware), v23, or dtmf.

cidstart: specifies how the start of the CID transmission is indicated. Can be polarity (the polarity on the wire is swit  ...
 
showing posts tagged with 'asterisk'