Step 1 :-
A server ------> Invite --->> Bserver ------->> Invite ------->> PSTN
A server <------ 100 trying <<--- Bserver <<------- 100 trying <<------- PSTN
A server <------ 183 session Progress <<--- Bserver <<------- 183 session Progress <<------- PSTN
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.
5 freelances font une offre moyenne de 960 $ pour ce travail
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
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