Re: [asterisk-users] VoiceMail and SMS

How many users are you thinking of supporting? For a large-scale setup you might want to take a look at Kamailio as a front-end - if you even think you're going to get a high user volume you may want to start out with a Kamailio front-end so that you don't have to start over from scratch when it

Re: [asterisk-users] Toll free pattern matching

Don't forget to handle the other special extension cases in a [context]. What I find is a good practice is to write a root "catch-all" (template), then I tag that onto any new [context](template). [Special-Extensions]

Re: [asterisk-users] Blacklist callers from file

Hi Kevin, Looks like your gmail did the same thing mine did, took the conversation off-list (unless you meant to do that). If as you mention you're running a mix of POTS and SIP, I'd recommend sticking with the transfer call method (set the hook/trigger in the features.conf file). That way if

Re: [asterisk-users] Blacklist callers from file

-Tim On Mon, Aug 29, 2016 at 2:31 PM, <kc6...@gmail.com> wrote: > Tim, I would like to see the code for this. I also am a home user and I > have been thinking of how I would do this same type of thing. Right now I > have a black list db and it is manual. When the unwanted caller call

Re: [asterisk-users] SIP trunk down. Wireshark shows ICMP Communication administratively filtered

Sounds like a firewall setting to me. If you can ping, then Internet Control Message Protocol (ICMP) packets are allowed, but if SIP traffic is returning the ICMP Type 3 (code 13) response, then your SIP ports are blocked (at least the firewall admin was nice enough to leave the reason code

[asterisk-users] Blacklist callers from file

I'm a home user (not business), but I implemented a blacklist function too after a harassing call to my wife. Using the Asterisk DB functions, I have a caller ID look-up function before my IVR-tree starts, a simple if then. Lookup caller ID in blocked-caller DB, if found then I kick them to a

Re: [asterisk-users] cloud solution?

I run Asterisk on a virtual Ubuntu machine. You can install Asterisk in Docker as well and make it portable across basically any platform that can run docker containers (it's hard to find a cloud provider that DOESN'T support docker now). I imagine that soon even Snappy containers will be an

Re: [asterisk-users] RS485 Audio device

Before walking down this path, take a moment to think critically: How far away is the AoR from the attendant station? Does there need to be local rescue/fire service access to the communications? How reliable does the link need to be? Will power always be available when the AoR pone is required

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

Is that IP in your network or outside (I can ping it so I'm guessing it's outside your network)? Do you have a firewall between your asterisk box and the internet? Is there a WHITELIST of IP addresses that only allow your provider's limited IP pool to connect to your asterisk box from outside?

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

ed services faced to public networks like > Internet. And configure these services properly, so they listen only > selected interfaces and IPs, and not from 0.0.0.0 > > 2017-04-21 13:47 GMT-03:00 Tim S <tim.strom...@gmail.com>: > >> Is that IP in your network or outside (I

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Rather than that, if you're looking for a phone solution - as part of the customer contract, install an IP phone that registers with your system (use a VPN tunnel to your phone system). Think of it like a "red-phone" hotline. You own the phone, and you physically install it and it only talks to

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

Another "me too" (also Gmail). I just received my 4th "account suspended, too many bounces" email, after having several days of lost mailing list content over a short vacation break the last time. When I notified the admin email account of the failure, it seemed the responder missed the

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I'd hazard to say it probably is Digium's "fault", this was a recent and now consistent problem, which started within the last month or so. I'm on 7 other Linux-related mailing lists which all use similar mailer daemons, and none have this issue. I have been subscribed to Asterisk

Re: [asterisk-users] Call does not go voicemail

The way you have the GotoIf is making it so that no matter what the busy condition of the line, it will execute the next line in the dial plan. What you'd need is an "if" or "then" which goes to a tagged line in the dial plan. How it reads now is: "If [busy] then line2, else execute next line".

Re: [asterisk-users] Call does not go voicemail

re why are you saying line 2 is FD_L2 needs to be fixed. > Do I need to removde "t", the call can not be transferred? > > Even when I put: > exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2) > exten => 4,n(line2),Dial(${FD_L2},20,trw) > exten

Re: [asterisk-users] Call does not go to voicemail

, the number they enter can feature a "Goto" with a text entry in the dial plan. This makes it harder for those at a phone to go places in your phone system they shouldn't. -Tim On Mon, May 8, 2017 at 4:51 PM, <the...@sys-concept.com> wrote: > On 05/08/2017 04:37 PM, Tim S wr

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

Just a quick and dirty thought, try the MONITOR application. Pseudo-code: Anchor-point PLAYBACK ("press or say") MONITOR (use the split audio files mode, not the mixed - this way you can roughly separate which side did the "talking") READ (audio file "1 to 5", try to grab one digit) STOPMONITOR

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

ion with Monitor, to make sure > you can record when there's no bridge between two channels. > >> I'll give this a go tomorrow and let you know what I come up with! > > Please do report back - this is a useful feature. > > > Antony. > >> On 20 January 2018 at 17:03, Tim

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

IMHO, manual IP-tables is probably better for those who have a single provider - whitelist only your SIP trunk provider's IP adress (or address pool). But... that leads onto a train of thought that might help. First, realize you don't have to manually read your security logs, you can script that

Re: [asterisk-users] Iridium integration / gateway

Hi, I use an Iridium 9555 handset and a "POTSdock". This takes a standard 9555 handset, gives it a fixed antenna mount, and a telco service jack. Interfacing to the POTSdock, is a matter of providing an analog POTS interface as if you were attaching to a standard POTS phone provider. The