Problems using more than ONE phone line (SIP)

[pw44]

Ticket #867 (new defect)

Environment:
LinuxMCE. 4 sip trunks (3 voip providers, 1 pstn provider via spa3102). 2 MD (core/hybrid - 200 + diskless md - 201), 2 cisco sccp 7970 (202 and 203), spa3102 FXS (204) 1 nokia sip client (205) wifi.

Definition: security disarmed, ring extensions 200, 201, 202, 203, 204 and 205. Extensions 201 and 205 offline during the tests. Test done calling from the mobile to the 4 trunks (pstn - landline, voipcheap, voipbuster eand sipgate)

Problem:
1) incoming call 1: extensions 200, 202 and 203 rings
2) incoming call 2: extensions 202 and 203 rings, 200 shows up but does not ring
3) incoming call 3: extensions 200 and 202 rings, 203 does not rings
4) incoming call 4: extensions 200 and 202 rings, 203 does not rings
5) incoming call 5: extensions 200 and 203 rings, 202 does not rings.

If only one sip trunk exists, there is no problem, Adding one or more trunks shows the problem.

If needed, logs can be generated, and sent with an exact description of what was done and the result. Usually asterisk log does not show problem, but the freepbx report shows which extensions did ring.

Is this a known problem, a bug, a feature? I could not find any answer yet why this is happening.

I would appreciate any assistance in order solve this.

Thx in advance,

Paulo

[merkur2k]

It should be considered a feature request not a defect as it is a use case that falls outside of current design goals.

[pw44]

Ok, i discovered it the hard way.
I did think it was supposed to work, as the Phone lines page in webadmin shows "To learn how....".
One question: the custom-app (102,1), can all the inbound routes use the same custom app?
TIA