SIP – IT Blog

Setting up SIP Trunk in Asterisk from Kyivstar

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.

ABillS. Installing and configuring Callcenter

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.

Configuring SIP Trunk in Asterisk from Ukrtelecom

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).

Asterisk compile error solution “‘pjsip_tcp_transport_cfg’ has no member named ‘sockopt_params’”

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.

Adding SIP clients to Asterisk

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):

Adding a SIP client to FreePBX

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)
Secret: PASSWORD
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

Solution to the Asterisk problem – no sound when calling via NAT

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).