Marek Greško writes:
> But I am not sure why this is happening. I have sip providers hostname
> in /etc/hosts file to prevent such situations. Should I reconfigure it
> not to use hosts file but rather some RPZ on DNS server? Does asterisk
> ignore hosts file? Or does it try to do some srv lookup
Łukasz Grzywański writes:
> I think it's a problem with DNS server availability
I have tried to and mostly succeeded at making things work when the WAN
is down. Elements needed:
run a local named, vs configuring resolver to your ISP
for names needed in the LAN, ensure they are answered l
Alexander Perkins writes:
> They ended up creating an AGI script for us that handles everything. At
> the end of the day, all we needed to do was pull down the script, and add
> the exten => s,n,AGI(TILTX-SHAKEN.agi) command and it handles
> everything else.
I wonder if you could step back and
D'Arcy Cain writes:
> Here is the first four lines from "pfctl -sr":
>
> pass in quick on bge0 from to any flags S/SA keep state
> block drop in log quick on bge0 from to any
> block drop in log quick on bge0 from to any
> block drop out log quick on bge0 from any to
agreed that I can't see
D'Arcy Cain writes:
> I have a script that checks for things like this and adds them to my
> packet filter (pf). Everything seems to work up to a point. The IP
> address gets added to my AUTOBLOCK table. The second rule, right after
> the friends whitelist, blocks any IP in that table. If I t
"Joshua C. Colp" writes:
>> I am curious if the "reuse registration TCP connection" is required by
>> standards or if it is merely obviously good practice.
>>
>> I have had this problem too with asterisk 16.5.0
>>
>> This is not the first recommendation I have seen to use kamailio as a
>> proxy f
Dovid Bender writes:
> So long as the tcp socket is open your SBC should send the call back over
> the same socket. Now it can be that your SBC is seeing the socket as
> timing out. If you are using Kamailio you can have it send tcp keep alives
> every so often so that the socket stays up.
SBC?
Doug Lytle writes:
> For a while now, I've had a small home Asterisk setup to connect to my
> Zimbra mail server's calendar. Making an entry on the calendar would
> cause Asterisk to schedule a wakeup call at the time of the calendar
> entry.
>
> The Zimbra mail server uses LetsEncrypt for the S
sean darcy writes:
> But the dreaded FXO/FXS issue. It's like trying to remember linear algebra.
>
> If I'm plugging the line from analog phone system into the 202, which
> then routes it to asterisk, I'm plugging the line into an FXS port ,
> correct?
>
> The line from the phone company is plugg
sean darcy writes:
>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO
>> port remoted over SIP. (I am not sure if this is discontinued.)
>
> "FXO port remoted over SIP"?
>
> I have an analog phone system. I can use the obi202 to connect the
> system to asterisk ? That is,
sean darcy writes:
> I'm moving asterisk to a laptop, so can't use the dahdi board. Is
> there any supported USB dahdi device ? I see the Sangoma USBfxo
> device, but the dahdi driver no longer supports it. Anything else ?
There is also the ObiHai OBi202 with an OBiLine, which provides an FXO
po
D'Arcy Cain writes:
> Not bad. I was toying with another idea. I find that if I don't answer
> a robot fast enough it just hangs up. How about ring two or three times
> before passing to the actual extension?
You could try that and let us know, but I suspect:
some robocallers don't hang up
"C.Maj" writes:
> Another option for a patch would be to extend the PJSIP_DIAL_CONTACTS
> function with an argument such as 'please' to minimally return the
> endpoint name in a Dialable format when no reachable contacts are found
> eg. "PJSIP/bar" -- instead of the current empty string, which is
>> So which option is preferred?
>>
>> A) Have a softphone aor/auth_user/password for a particular human, and
>> expect them to configure it on multiple devices. Do not worry that 1)
>> multiple are registered at once (because that's normal in SIP) and 2)
>> asterisk has no idea which is w
(I'm new to Asterisk, after having started VOIP with vat on the mbone in
the 90s.)
I am setting up my first Asterisk system, and trying to read
docs/guidance and follow best practices. I have read the 5th Edition of
"Asterisk: The Definitive Guide" and like the 3rd Edition on the web it
recommend
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