Help with phone config


(Vladimir HOUZANME) #1

Dear,

Please give me a hand to well-configure 2 SIP-Phones Planet VIP-256PT on FreePBX Server 14 (with axterisk 13).

I have created 2 extensions (phone numbers) on the FreePBX System, and have applied each extension on a different SIP-PHONE Planet VIP-256PT.

Below all the configurations on the Server and on each sip-phone PLANET.

SERVER CONFIG :

IP : 192.168.1.10

PORT / PJSIP : 5060

Codecs : ALaw, ULaw, G.729,G.723,G.722

NB: codecs are set in this order.

Two extensions created : 200, 201

PHONE 1 : Config

SERVER ADDRESS : 192.168.1.10

PORT SIP : 5060

ACCOUNT : 200

Codecs : (in this order)

G.711A

G.711U

G.729

G.723

G.722

PHONE 2 : Config

SERVER ADDRESS : 192.168.1.10

PORT SIP : 5060

ACCOUNT 201

Codecs : (in this order)

G.711A

G.711U

G.729

G.723

G.722

Both phones (Planet VIP-256PT) are registered on the FreePBX système, but none of them is giving sound (NO SOUND).

If on the first sip-phone (200) and try to call the second-one (201), the call cames and rings, but if taken and talk, i can not hear anything.

Please help me. Maybe something is wrong in the configurations.

[moved to new topic from unrelated thread - mod]


Sipura/Linksys 3102 + PJSIP as Device / Extension
(Shahin Nazir) #3

@vhouzanme
Check your RTP Range 10000-20000 UDP Ports.
Extension Register used 5060
Audio used 10000-20000 UDP ports.