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).

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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 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