Is there a way to configure SIP extensions in FreePBX that are located in a gateway with multiple fxs ports, WITHOUT needing registration and WITHOUT assigning different UDP ports for each of them ?
What I see in SIP trace is the following:
-
for calls towards the FXS GW, Asterisk ommits to put the extensions number in the request URI:
The original Request-Line: INVITE sip:[email protected];user=phone SIP/2.0 coming for the calling SIP extension is changed by Asterisk in the new INVITE to the FXS GW . It looses the sip:86@ in the Request-Line and to-header: INVITE sip:10.134.63.130:5061 SIP/2.0 -
for calls from the FXS GW, something similar happens:
The From: sip:[email protected]:5061 line in the INVITE coming from the FXS GW contains the correct extension number 87, but this line is changed to ANOTHER extension number when forwarded to the destination: sip:[email protected]:5061 . In fact this is the first extension number which is configured in Asterisk for this FXS GW. These multiple FXS extensions are all defined with IP address 10.134.63.130 and UDP port 5061.
86 10.134.63.130 No No A 5061 Unmonitored
87 10.134.63.130 No No A 5061 Unmonitored
88 10.134.63.130 No No A 5061 Unmonitored
89 10.134.63.130 No No A 5061 Unmonitored
…
etc (32 in total per FXS GW)
The reason why I don’t like to register and use different UDP ports for each extension in this gateway is to avoid complex and unnecessary configuration in the FXS GW. I don’t care about security issues as this is for an internal FXS GW protected by access rules (only Asterisk may directly contact it).