I will give an example of setting up a SIP Trunk in Asterisk from Kyivstar. Once it was necessary to increase the number of incoming lines, at first I used Goip4 gateways and 6 Kyivstar SIM cards, but when all the lines were busy, the clients could not get through, even hear the answering machine / voice menu, so I signed a contract for SIP Trunk with the Kyivstar manager.Continue reading “Setting up SIP Trunk in Asterisk from Kyivstar”
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”
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).
Once I compiled Asterisk version 13.13.1 and when running make I noticed the following error:
‘pjsip_tcp_transport_cfg’ has no member named ‘sockopt_params’
pjproject-2.2.1 has already been compiled.
Solved the problem by compiling a newer version of pjproject-2.4.5
cd /usr/src wget http://www.pjsip.org/release/2.4.5/pjproject-2.4.5.tar.bz2 tar -xjvf pjproject-2.4.5.tar.bz2 cd pjproject-2.4.5 CFLAGS='-DPJ_HAS_IPV6=1' ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr make dep make make install
After that, the error disappeared.
SIP clients in Asterisk are specified in the sip.conf file, so open it for example in the nano text editor (Ctrl+X to exit the editor, y or n to save or discard changes):Continue reading “Adding SIP clients to Asterisk”
To add a SIP client to FreePBX, open the menu “Applications” – “Extensions“, choose for example “Generic CHAN SIP Device” and we indicate the main parameters:
User Extension: 6000 (SIP number)
Display Name: Operator (any name to display)
and click “Submit“.
Done, SIP is added, it can already be registered at the specified number and password.
How to add SIP in the configuration file I described in this article – Adding SIP clients to Asterisk
I noticed recently that there is no sound when calling from IP-phone to another IP-phone which were both behind the same NAT (router).Continue reading “Solution to the Asterisk problem – no sound when calling via NAT”