I have setup FreePBX 2.8.1.0 on CentOS 5. I have an OpenVOX 12-port card with three FXO and three FXS ports installed. I have X-Lite installed on a Windows 7 box (ext 1050). I have one Vitelity trunk installed.
One analog phone is plugged in, one POTS trunk is plugged in. (I can’t switch the other two lines from POTS to Asterisk until I can make the first one work.)
If I pick up the analog phone and dial 1050, I’m told “The person at extension one zero five zero is unavailable” and offers voicemail. If I leave voicemail, it is saved and e-mailed to the address associated with the extension, correctly identifying the analog extension.
If I dial from the X-Lite I get “The person you are calling is unavailable. Please try again.”, X-Lite reports “Call failed: Forbidden”. This happens with calling an extension, a local phone number, or a full 10-digit number, with and without the leading 1.
If I dial from the analog phone to any number other than an extension, I get “The number you have dialed is not in service. Please check the number and try again.” It then recites the number dialed and goes to a busy signal.
I think I have several parts of this actually working, but obviously some crucial steps not working, or perhaps not even tried.
My immediate goal is to have the analog phones simply work the same way they do now for local and toll-free calls, but use the VOIP trunk for long distance. I can’t find any How-tos that are basic enough to get me going. (There’s lots of stuff that shows me how to add sexy features later!)
I’ve been poking at this for what seems like ages, with no real progress. I can call from either extension to the other.
Local calls from either extension get a busy signal.
“Dial failed for some reason with DIAL STATUS = BUSY and HANGUPCAUSE = 17”
Long distance calls (1 area code and seven digits) also fail.
“Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0”
From the analog extension I get a long wait and then a busy signal, with the SIP extension I get the long wait and hear this:
“The person you are calling is unavailable, please try again.”
Asterisk -V reports 1.8.2.2
The admin screen reports FreePBX 2.8.1.3
I’ve searched this site and the internet as a whole with no success. I’ve encountered solutions that sounded like they would help, but they were accompanied by screen captures that don’t match my version or suggestions to edit configuration files that are clearly off limits. Can my simple requirements really be so outlandish that nobody has run into them before?
Van
Here are some queries to asterisk. Where it says “myaccount” it actually displayed the account name correctly.
vocalize*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 from-internal default In Service
2 from-internal default In Service
3 from-internal default In Service
5 from-zaptel default In Service
6 from-zaptel default In Service
7 from-zaptel default In Service
vocalize*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
66.241.96.96 myaccount 17e6034a6cb6afb 0x0 (nothing) No
66.241.96.96 myaccount 351e1de70f36d06 0x0 (nothing) No
2 active SIP dialogs
vocalize*CLI> dahdi show channels
Chan Extension Context Lang MOH Interpret Blocked State
pseudo from-internal default In Service
1 from-internal default In Service
2 from-pstn default In Service
3 from-internal default In Service
5 from-pstn default In Service
6 from-pstn default In Service
7 from-pstn default In Service
(I deleted a few spaces to make it fit.)
The extensions are on 1,2,3 and the trunks are 5,6,7. I have no idea why 2 is showing the reverse here. I went to “Extension 56” and the context and channel were both wrong so I fixed those and applied the changes, but “dahdi show channels” continues to show the above.