In this article, I will provide an example of setting up MP3 for Music on hold (moh) in Asterisk.Continue reading “Asterisk. MP3 for Music on hold (moh)”
Once in one company, in order to simplify the search for callers in ABillS, it was necessary to install the Callcenter module, which has different capabilities, but what we needed was pop-up notifications in the browser with a link to the caller’s page.Continue reading “ABillS. Installing and configuring Callcenter”
In this article, I will provide examples of changing music on hold, force music on hold, and more.Continue reading “Asterisk. Music on hold”
Let me give you an example of setting call forwarding in Asterisk.
For example, on Grandstream IP phones, you can enable redirection by the functions of the phone itself, but if the phone is far away and there is no way to do it on it, but it is possible to log in using the SIP number of this phone, then to activate call forwarding, you can make a voice menu when dialing a certain number, for example, *21 to activate call forwarding and enter the phone number to which calls will be forwarded, and *22 to cancel call forwarding.
I noticed once an error in the logs /var/log/syslog:Continue reading “Solution the error “rc_avpair_new: unknown attribute 1490026597””
Once on one Asterisk server I noticed the following error:Continue reading “Asterisk. Solution “Not accepting call completion offers from call-forward recipient Local””
I will describe an example of setting up recording telephone conversations in Asterisk, first of all, make sure that the necessary modules are loaded.Continue reading “Asterisk Call Recording”
In the test, I connected a Chinese TDM410P board with four FXO ports (red) to the PCI slot of the turned off Ubuntu server.
For convenience, immediately switch to the root user:
To determine the number from analog lines, you need to specify in the context for each Trunk channel Dahdi:Continue reading “Configuring FSK in Asterisk to determine phone numbers”
I will give an example of setting up SIP Trunk in Asterisk, that is, Asterisk will be in the role of a SIP client.
From the provider Ukrtelecom received data: number, password and address of the telephony server (sip.ukrtel.net).
SIP number was taken to make multi-channel not an ordinary city number, by forwarding in case of busy lines.
I will configure on a Linux server with a real IP without using NAT, as well as on another with NAT (in the second case, you need to change nat=no to nat=yes and comment out canreinvite).