Asterisk Expert Required
$100-2000 USD
Dibayar semasa penghantaran
Step 1 :-
A server ------> Invite --->> Bserver ------->> Invite ------->> PSTN
Step 2:-
A server <------ 100 trying <<--- Bserver <<------- 100 trying <<------- PSTN
Step 3:-
A server <------ 183 session Progress <<--- Bserver <<------- 183 session Progress <<------- PSTN
Step 4:-
A server <------ 183 session Progress <<--- Bserver <<------- 180 Ringing <<------- PSTN
A Server >>> Installed Asterisk with Freepbx
B Server >>> Installed Asterisk with Freepbx
PSTN (SIP) attached with Server B
An extension 123 created on Server B
On Server A created a SIP Trunk with configuration of extension 123 that was created on Server B
Also connected server B with the server A via chain_SIP Trunking.
Now created a Extension 321 created on server A and configured in soft-phone like zoiper and dialling call on XXXXXXXXXX number.
Now the problem is in Step 4 where I am not getting same 180 Ringing that I am getting from PSTN on Server B. How I can get same signal on Server A that are getting on Server B from PSTN.
ID Projek: #32236583
Tentang projek
4 pekerja bebas membida secara purata $1150 untuk pekerjaan ini
Hi Team, We will help you to configure two Server and PSTN and call routing also. We have integrated like this many servers. Regards KABIS
CERTIFIED PHD HOLDER IN SOFTWARE ENGINEERING AND DEVELOPMENT. DEGREE IN COMPUTER SCIENCE EXPERT IN CYBER SECURITY, PYTHON, JAVA, C#, C++, JAVASCRIPT, AND DATA MINING. HELLO DEAR CLIENT. I have thoroughly gone through y Lagi
Hi, I have over 10 years of experience in Asterisk development, deployment and administration. I have worked on AMI AGI ARI Dialplan IVR etc. I can debug the issue and will resolve it Thanks! Rehan Khan
