My system:
PBX in a Flash PURPLE Status Program
┌────────────────────────SYSTEM INFORMATION───────────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PIAF Installed Version = 2.0.6.2 under HARDWARE │
│ FreePBX Version = 2.8.1.5 │
│ Running Asterisk Version = 1.8.8.0 │
│ Asterisk Source Version = 1.8.8.0 │
│ Dahdi Source Version = 2.6.0+2.6.0 │
│ Libpri Source Version = 1.4.12 │
│ IP Address = 192.168.1.100 on eth0 │
│ Operating System = CentOS release 6.2 (Final) │
│ Kernel Version = 2.6.32-220.7.1.el6.i686 - 32 Bit │
Testing the Les.net and making inbound route as described using the numbers after the / , advanced pattern matching strings, the phone number, heck even tried my birthdate! They all get caught by the “Any” DID routing entry. When I delete the “any” catch all entry all calls goto “number not in service” message from my PBX I assume.
The incoming rings through to the “catch all” but once that is deleted I get the “number you dialed is not in service” with the following log activity:
VERBOSE[13311] pbx.c: – Executing [2181112222@from-trunk:1] Set(“SIP/123456789-00000325”, “__FROM_DID=2181112222”) in new stack
VERBOSE[13311] pbx.c: – Executing [2181112222@from-trunk:2] NoOp(“SIP/123456789-00000325”, “Received an unknown call with DID set to 2181112222”) in new stack
VERBOSE[13311] pbx.c: – Executing [2181112222@from-trunk:3] Goto(“SIP/123456789-00000325”, “s,a2”) in new stack
VERBOSE[13311] pbx.c: – Goto (from-trunk,s,2)
VERBOSE[13311] pbx.c: – Executing [s@from-trunk:2] Answer(“SIP/123456789-00000325”, “”) in new stack
VERBOSE[13311] pbx.c: – Executing [s@from-trunk:3] Wait(“SIP/123456789-00000325”, “2”) in new stack
VERBOSE[13308] app_dial.c: – SIP/future9-00000324 is making progress passing it to SIP/122-00000323
VERBOSE[13308] app_dial.c: – SIP/future9-00000324 answered SIP/122-00000323
VERBOSE[13308] rtp_engine.c: – Locally bridging SIP/122-00000323 and SIP/future9-00000324
VERBOSE[13311] pbx.c: – Executing [s@from-trunk:4] Playback(“SIP/123456789-00000325”, “ss-noservice”) in new stack
VERBOSE[13311] file.c: – <SIP/123456789-00000325> Playing ‘ss-noservice.gsm’ (language ‘en’)
For my incoming route here I used the 123456789 in the DID number field and at the bottom of the page the extention it was to go to.
My outgoing trunk settings: (many revisions have been tried here too)
host=did.voip.les.net
context=from-trunk
type=peer
insecure=very
nat=yes
canreinvite=no
username=123456789
secret=xxxxxxxxx
qualify=no
Does any of this look out of sorts?