PS this is starting to become a hobby.

I’ve install iso 2.8 of AAH, done yum -y update. Now, to setup updates, do the freePBX lines of code still apply:, or is there any update? since?

[WM: TrixBox has now replaced AAH. If you want to use freePBX, I’d recommend you install TrixBox which comes with everything already set up. It also has all of the Asterisk security updates.]

OH: Thank God (and Ward) for this article… w/o it I’d be up the creek w/o a paddle, boat or lifejacket!

Roger

The steps for Webmin seem to be broke. I eventually got the download to work by using wget http://prdownloads.sourceforge.net/webadmin/webmin-1.280-1.noarch.rpm . However, when I try to install (with your command line or the Webmin folks) I get a ‘not an rpm package (or package manifest)’ error. Still trying to work around that.

Roger

Where to I take out this extra quote mark for callerid on these incoming sip calls? I am running AAH 2.7

Thanks!

[WM: Duh! Why didn’t we think of that?? Thanks. As for Midnight Commander and Nano, we’ve loved MC since the good ol’ days, but it wasn’t part of Asterisk@Home until recently. It unfortunately has some function key quirks with some SSH implementations so we’ll probably stick with Nano for the tutorials, but I share your love of the product, and mcedit will let you call up the editor without the front end file manager.]

[WM: There still are some major problems with the 2.8 release. It may be 2.9 before everyone gets sufficient kinks worked out to make it useful in a production environment. You can follow the threads on SourceForge to see a list of the problems.]

externhost=name.dyndns.org
externrefresh=10

where the externrefresh specifies how often the host name should be checked to be resolved!

Thanks for the great play by play! This is awesome!
tphank

exten => fax,1,Goto(ext-fax,in_fax,1)

;next extension (99) is to handle incoming PSTN calls
exten => 99,1,GotoIf($["${CALLERIDNUM:0:2}" = "00″]?2:3)
exten => 99,2,SetCIDNum(${CALLERIDNUM:2})
exten => 99,3,Zapateller(answer|nocallerid)
exten => 99,4,Wait(1)
exten => 99,5,SetMusicOnHold(default)

exten => 99,6,GotoIf($["${CALLERIDNUM}" = ""]?who-r-u,s,1)
exten => 99,7,GotoIf($["foo${CALLERIDNUM}" = "foo"]?who-r-u,s,1)
exten => 99,8,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?who-r-u,s,1)
exten => 99,9,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?who-r-u,s,1)
exten => 99,10,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => 99,11,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)
exten => 99,12,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?who-r-u,s,1)
exten => 99,13,GotoIf($["${CALLERIDNAME:0:11}" = "OUT OF AREA"]?who-r-u,s,1)
exten => 99,14,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => 99,15,GotoIf($["${CALLERIDNAME:0:4}" = "PSTN"]?who-r-u,s,1)
exten => 99,16,DigitTimeout,3
exten => 99,17,ResponseTimeout,3
exten => 99,18,Background(custom/welcome)

exten => 0,1,Background(pls-hold-while-try)
exten => 0,2,Dial(local/200@from-internal,20,m)
exten => 0,3,VoiceMail(200@default)
exten => 0,4,Hangup
exten => 1,1,Background(pls-hold-while-try)
exten => 1,2,Dial(local/204@from-internal,20,m)
exten => 1,3,VoiceMail(204@default)
exten => 1,4,Hangup
exten => 4,1,Authenticate(123456)
exten => 4,2,Background(pls-wait-connect-call)
exten => 4,3,DISA(no-password|from-internal)
exten => 2XX,1,Background(pls-hold-while-try)
exten => 2XX,2,Dial(local/${EXTEN}@from-internal,20,m)
exten => 2XX,3,VoiceMail(${EXTEN}@default)
exten => 2XX,4,Hangup
exten => 2XX,103,Voicemail(${EXTEN}@default)
exten => 2XX,104,Hangup
exten => t,1,Background(pls-hold-while-try)
exten => t,2,NVFaxDetect(10) ; detect faxes while playing ring sound – goes to ?¢‚Ǩ?ìfax?¢‚Ǩ¬ù extension if detected
exten => t,3,Dial(local/204@from-internal,20,m)
exten => t,4,VoiceMail(204@default)
exten => t,5,Hangup
exten => i,1,Playback(wrong-try-again-smarty)
exten => i,2,Goto(99,18)

[WM: See #55 above.]

So in my extensions_custom.conf I have:
[from-internal-custom]
exten => fax,1,Goto(ext-fax,in_fax,1)
;next extension (99) is to handle incoming PSTN calls
exten => 99,1,GotoIf($["${CALLERIDNUM:0:2}" = "00″]?2:3)
exten => 99,2,SetCIDNum(${CALLERIDNUM:2})
exten => 99,3,SetMusicOnHold(default)
exten => 99,4,Answer
exten => 99,5,Wait(1)
exten => 99,6,Background(welcome)
exten => 99,7,DigitTimeout,2
exten => 99,8,ResponseTimeout,2
exten => 1,1,VoiceMail(203@default)
exten => 1,2,Hangup
exten => 2,1,VoiceMail(201@default)
exten => 2,2,Hangup
exten => t,1,Answer
exten => t,2,Wait(1)
exten => t,3,Background(custom/tom-sherry-kids)
exten => t,4,NVFaxDetect(4) ; detect faxes while playing ring sound – goes to "fax" extension if detected
exten => t,5,Dial(SIP/200&SIP/201,20,m)
exten => t,6,VoiceMail(201@default)
exten => t,7,Hangup
exten => i,1,Answer
exten => i,2,Wait(1)
exten => i,3,Playback(wrong-try-again-smarty)
exten => i,4,Goto(99,5)

Perhaps the SPA-3000 configs here are out of date, but I couldn’t get the built-in fax working following the NerdVittles instructions verbatim. Now to figure out why outgoing PSTN calls disconnect after a minute…
tg.

Your backup.sh will copy only the changes to the remote server is that right?

[WM: Correct.]

I’m using Ast@home 2.7 and following your tutorial for redundancy and backup, all the backup went ok but i get this error for /sys/ directory:
rsync: send_files failed to open "/sys/bus/pci/drivers/parport_pc/new_id": Permission denied (13)

Is this vital for future recover? I mean, i will always have to reinstall asterisk, could i make –exclude"/sys/" on rsync ?

[WM: I don’t think it’s vital. It would get replaced in a reinstall so you can either ignore the error or add the exclude as you suggested. Thanks.]

I found that, i think is macro-dial , not sure:

I call from Zapchannel then directory and choose a SIP extension to call.

I got the person @ extension XXX is on the phone —> vmail.

In fact this extensions was unregistred…

It shouldn’t give voicemail message unavailable instead of voicemail message for busy status? I hope you can help me to fix this…

Best regards,
Marco Mouta

Wouldn’t it be easier to number lines after "1″ in your extensions with "n", rather than real numbers? Then no renumbering is required when you add/delete lines.

Example:

[custom-recordme]
exten => 5678,1,Wait(2)
exten => 5678,n,Record(/tmp/asterisk-recording:gsm)
exten => 5678,n,Wait(2)
exten => 5678,n,Playback(/tmp/asterisk-recording)
exten => 5678,n,Wait(2)
exten => 5678,6,Hangup

Regardless, thanks for the articles, they’re great!

[WM: Several have written about this. And, yes, there is a reason. We’re used to the old way. Only kidding. The new way works fine in version 2 of AAH but not at all in version 1 so we try to make most articles generic enough for all. It also forces you to use labels or mix-and-match (as you did above) to jump around in contexts so we think the numbers are more straight-forward. Finally, the numbers make it easy (at least for us) to spot a problem or a change so we’ll probably stick with them a while longer so just bear with us on this one.]

But is there any way to secure this? I do need port 80 to administrate AAH remotely, however I would prefer not to be used for some phishing scam lol.

Thanks.

Again, awesome articles guys!

[WM: Samba with no password requirement is news to me. My default installation requires the root password. We never recommend putting an Asterisk server in the DMZ. You can accomplish the same thing by opening some ports on your firewall. Unless you’ve changed something on your default system, port 80 shouldn’t provide networking access to your system so far as I know. If someone knows otherwise, please post the steps and we’ll have a look.]