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showing posts tagged with 'voip'
 
by lunarg on October 23rd 2017, at 11:28

As found on Voip-info.org, a list of bandwidth consumption per codec.

It's worth mentioning that the table also indicates the total consumption including IP overhead. When making assumptions about bandwidth requirements, it is useful to take this overhead into account.

CodecBRNEB
G.711 64 Kbps 87.2 Kbps
G.729 8 Kbps 31.2 Kbps
G.723.1 6.4 Kbps 21.9 Kbps
G.723.1 5.3 Kbps 20.8 Kbps
G.726 32 Kbps 55.2 Kbps
G.726 24 Kbps 47.2 Kbps
G.728 16 Kbps 31.5 Kbps
iLBC 15 Kbps 27.7 Kbps
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 February 20th 2017, at 14:53
When provisioning non-Cisco phones on a Cisco-switched network, you may notice that the configured voice VLAN is not correctly provisioned to the phone. The phone ends up in the data VLAN, rather than the voice VLAN even though the switch port has been correctly configured.

There are two protocols which can be used to provision the correct VLAN to your phones: Cisco Discovery Protocol (CDP) and Link-Layer Discovery Protocol (LLDP). CDP is a proprietary protocol and is only supported on Cisco-switches and Cisco-phones. Non-Cisco devices usually use LLDP, which is an open standard supported by most other vendors. Although Cisco switches also support LLDP, it is by default not enabled, resulti  ...
by lunarg on March 4th 2016, at 11:54
You can make your own logo to be placed on the idle screen. Follow these specifications to create it.

Format: save the logo as BMP in 256-colors (8-bit) and without colorspace info.

About colorspace info: by default, Gimp includes this info, resulting in a malformed logo on the phone. During export to BMP, deselect this option (under compatibility options). MS Paint never saves colorspace info so no specific settings are required.

Do not save in 16-color (4-bit), especially with MS Paint: you will lose the darker shade of gray (404040) because it is not part of the default 4-bit palette.

Dimensions: 90x56

If the dimensions are larger, the phone will automatically downsize, although it i  ...
by lunarg on January 29th 2016, at 16:13
You can factory reset a Cisco 7911(G) and 7941/7961(G) at boot time by following the procedure below. Performing a full factory reset will completely wipe the phone. This includes its firmware, leaving the phone in an unusable state if no TFTP server with the necessary files is available, along with a DHCP server that pushes the TFTP server (using DHCP option 150). If you have a Cisco Call Manager, this part should already be set up and the phone will simply download everything from the CCM.

Use with caution!
The factory reset will also wipe the firmware. You will no longer be able to use the phone until a new firmware is installed on the phone. This can only be done through a TFTP server w  ...
by lunarg on November 16th 2015, at 12:55

The specifications for the background on a Yealink T46(G) can be found in the Administrator Guide:

  • Supported formats: .jpg, .png, .bmp
  • Maximum filesize: 5 MB
  • Resolution: 480x272
by lunarg on October 8th 2012, at 16:04

You can reset the SPA3102 to factory default using the Interactive Voice Response (IVR) and following these steps:

  1. Plug an analog (PSTN) phone into the Phone 1 port
  2. Dial ****: you'll hear a voice prompt.
  3. When the IVR is done talking, enter the sequence 73738, followed by a #.
  4. Press 1 to confirm. The unit will reboot to factory defaults.
by lunarg on June 22nd 2012, at 16:14
Free (as in "beer") SIP clients are very scarse on the Mac OSX platform. Most clients are too far outdated and only work on old versions and on PowerPC. Other, more recent clients start up but they lack proper quality and functionality to be even remotely usable.

Luckily, there is a client available. Originally designed as meeting software using the H.323 protocol, it also implements SIP and can be used with any SIP provider, including Asterisk. The client is called XMeeting, and while it's no longer actively maintained, it still runs on Mac OSX up to 10.7 (Lion).

http://xmeeting.sourceforge.net/


I've tested it out at work on our Asterisk server and it works flawle  ...
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 'voip'
 
 
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