-
Type:
Bug
-
Status: Closed
-
Priority:
Minor
-
Resolution: Support
-
Affects Version/s: None
-
Fix Version/s: None
-
Component/s: Asterisk SIP Settings
-
Labels:None
-
Bug Tracker:Customer Issue
-
ToDo:
-
Asterisk Version:asterisk 16
Hello,
I am facing issue with all asterisk versions I have tried all option of configurations but still not getting any solution on this issue could anyone please help me.
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.