i have to switch from Asterisk to FreePBX in two Steps (i also change the Provider)
Old Asterisk = 1.4.21
FreePBX = 18.104.22.168
What is working:
- The Trunk from FreePBX to the new Provider is up and running
- The Trunk from FreePX to old Asterisk seems to be up
- Doing a call from external over new Provider is working
- FreePBX routed Call to old Asterisk and the snoms rings
At the Moment this should be so!
What is not working
- Call an external Number on SNOM at old Asterisk and use FreePBX to route them to New Provider
What habve i try:
- First i tried to use an SIP-Trunk (the same as i use above) between old Asterisk and FreePBX
Error i get:
WARNING: chan_sip.c:12424 handle_response_invite: Received response: “Forbidden” from ‘“K.Schmidt” <sip: 37@ 10.53.2.230>;tag=as069c578a’
chan_sip.c: username mismatch, have <asteriskout>, digest has <s>
NOTICE[C-0000006b] chan_sip.c: Failed to authenticate device “K.Schmidt” <sip: 37@ 10.53.2.230>;tag=as069c578a
OK, that seems to be an Problem Auth Problem, but i can’t figure out, how to resolve that.
- Second i tried to do that with an IAX Trunk between Old Asterisk and Free PBX
Error i got:
Dial(“SIP/37-b56660a8”, “IAX2/ast1/THENUMBERICALLED”) in new stack
– Called ast1/THENUMBERICALLED
[Dec 13 07:05:18] WARNING: chan_iax2.c:7832 socket_process: Call rejected by 10.53.2.233: No such context/extension
– Hungup ‘IAX2/ast1-6063’
There is no Entry in Log!
That seems that i have not an Extension for that. I tried many Extension at In and Outbound Route in FreePBX the last Days, but nothing works at all :-/
I only want to get that:
SNOM -> Old Asterisk -> TRUNK (SIP or IAX) -> FreeBPX -> Change Outgoing Number from Two Digits (e.g 37) to something like 0401234537 -> New Provider -> Arrive at Destination
Is this possible?