6.1.2.1. AbsoluteTimeout (dialplan application)
1. AbsoluteTimeout - this application allows you to limit the duration of the conversation.
NOTE: This application is valid for Asterisk version 1.0.9 and 1.0.10.
If you are using version 1.2 or newer, then you have to know that this application is deprecated. However, if you want to use its functionality, there is a new syntax, which require the usage of the Set application. For more information click on the link - The new syntax - the AbsoluteTimeout function.
Syntax:
Purpose and usage
Prerequisites
Asterisk PBX configurations
2. Screenshots of what you can see on the CLI of the Asterisk PBX
3. Additional information
4. Similar dialplan applications
NOTE: This application is valid for Asterisk version 1.0.9 and 1.0.10.
If you are using version 1.2 or newer, then you have to know that this application is deprecated. However, if you want to use its functionality, there is a new syntax, which require the usage of the Set application. For more information click on the link - The new syntax - the AbsoluteTimeout function.
Syntax:
AbsoluteTimeout(delay) ; the delay is in seconds
Purpose and usage
With the help of this function you could limit you conversations. You could set a period of time after which expiration, the line will be hung up automatically.
Prerequisites
To use this application you need a working Asterisk PBX with registered users in iax.conf, sip.conf or mgcp.conf(It depends on which protocol you would like to use) and made extensions.
To see how the application works, you can use the Idefisk 2.0 softphone. You could download it from here and you could read our tutorial about how to configure it to work with Asterisk PBX
To see how the application works, you can use the Idefisk 2.0 softphone. You could download it from here and you could read our tutorial about how to configure it to work with Asterisk PBX
Asterisk PBX configurations
NOTE: This is only an example of one of the uses of this application. Of course you can use it and for other things.
We need two registered users in sip.conf file. This is because we are going to use the SIP protocol. If you want to use other protocol such as IAX2 or MGCP, you have to do the configurations below respectively in iax.conf or mgcp.conf.
We need two registered users in sip.conf file. This is because we are going to use the SIP protocol. If you want to use other protocol such as IAX2 or MGCP, you have to do the configurations below respectively in iax.conf or mgcp.conf.
So, we have registered the users anatoliy and user1
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
Now, when somebody dials 113, the first executed extension will be Answer. Then we have the AbsoluteTimeout application. This application can take argument in seconds. In our example this is 10. So after 10 seconds the call will be hung up. Please pay attention that the 10 seconds include the time before the another site to answer the call. So, if you dial somebody but he/she does not answer within 10 seconds the call will be hung up. If he/she answers after 5 seconds than you have only the rest 5 seconds for conversation.
The extension with the next priority contains the Dial application. This application will allow you to connect the user anatoliy through IAX2 channel.
We recommend you always to make an extension for hanging up. In this way you will be sure that the Asterisk PBX will hang up the line after the conversation is over.
The extension with the next priority contains the Dial application. This application will allow you to connect the user anatoliy through IAX2 channel.
We recommend you always to make an extension for hanging up. In this way you will be sure that the Asterisk PBX will hang up the line after the conversation is over.
2. Screenshots of what you can see on the CLI of the Asterisk PBX
3. Additional information
For more information about extensions.conf you can check here.
For more information about iax.conf you can check here.
This application is tested with idefisk 2.0. You can download it from here.
For more information about this softphone please read our tutorial.
For more information about iax.conf you can check here.
This application is tested with idefisk 2.0. You can download it from here.
For more information about this softphone please read our tutorial.
4. Similar dialplan applications
| Add Comment |
Latest Headlines:
-
T.38 faxing with Zoiper 2.15 is now easier than ever
section: voip software
-
Asterisk 1.4.21 Released
section: Asterisk
-
Asterisk 1.4.20 Released
section: Asterisk
-
Asterisk 1.4.20-rc2 Released
section: Asterisk
-
Asterisk 1.4.20-rc1 Now Available
section: Asterisk
- News Archives (older news)
Latest Tutorials:
-
Sending Fax from Zoiper to Zoiper using T.38
added 08/Dec/2008 18:16
-
VMAuthenticate (dialplan application)
added 01/Mar/2008 15:57
-
Siptronic ST-530
added 06/Nov/2007 17:57
-
Siemens C455 IP hardphone
added 05/Nov/2007 10:24
-
Zoiper
added 22/Oct/2007 17:53
Latest Comments:
-
nice...
tutorial: X-Lite - SIP softphone
-
Badly need your help. Things are only im...
tutorial: Grandstream HandyTone-486 - SIP Analog Telephone Adapter (ATA)
-
how do i rectify this problem? all incom...
tutorial: Troubleshooting
-
How are you. Summer afternoon - Summer a...
tutorial: Cisco 7960 IP Phone - SIP firmware version
-
where is the tar file you mention above ...
tutorial: Cisco 7960 IP Phone - SIP firmware version