On Tue, 2023-11-07 at 08:42 +0100, Luca Bertoncello wrote:
> The best will be a free service, but if not, I don't want to pay too
> much...
> As said: I need a SIP Provider to have an italian number (better if I
> can choose the prefix) only to receive calls.
>
> Any suggestion?
Assuming that
Does asterisk follow HTTP redirects? If so can you use something like
tinyurl.com to produce an alternative URL?
Or, base64 encode the URL, and then set a variable with
Set(url=${BASE64_DECODE(${encodedURL})) ?
Cheers,
Kingsley.
On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> I tried
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
>
>
> Just use the something like the following in your script:
>
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs are here:
>
>
On Thu, 2021-11-04 at 08:52 -0300, Joshua C. Colp wrote:
> > Thanks, that looks perfect. What is the syntax? I have tried a few
> > things but none work:
> >
>
> ${CHANNEL(pjsip,remote_addr)}
Hmm, I can't get this to work. This dialplan code:
exten => s,n,NoOp(### state=${CHANNEL(state)} ##)
> Is the app_fax.so module still in /usr/lib/asterisk/modules? If so -
> if you remove it do things work.
> Is app_fax.so explicitly being loaded in modules.conf?
Thanks.
I was already waiting for it to finish recompiling after Doug's
suggestion but yes, app_fax.so was still in there and
On Fri, 2021-10-22 at 11:11 -0300, Joshua C. Colp wrote:
> I don't provide direct support like that. As there seems to be a bug
> and you have a case that reproduces it with logs, then you can file
> an issue[1] and the current individual doing bug triage will look. If
> it is accepted there is no
On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote:
> # asterisk -V
> > Asterisk GIT-master-cc127a999cM
> > #
>
> That's the master branch from around March or so, not 18.
Wow, all this time I thought I was running 18! What version would it
be? How can I tell?
Should I download and compile
Thanks.
I tried to find the precise version but I got stuck at this point:
# asterisk -V
Asterisk GIT-master-cc127a999cM
#
Within /usr/src/asterisk I tried this
# grep -riE 'version\b *18' .
#
but it didn't match any lines. So I'm not quite sure what actual
version this is. Any idea how I can
if some fail2ban magic could keep OpenSIPs response from hitting
> Asterisk after N attempts ?
>
> Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
> kingsley.t...@barritel.com> a écrit :
> > Hi,
> >
> > We're using Asterisk 13.17.0 with PJSIP 2.8
On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote:
> This is not yet fixed, but is being worked on. I have it as a
> security issue currently out of caution (although I don't think we'll
> treat it as one after further investigation).
Right OK, thanks.
Do you have any idea of the sort of
Hi,
We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
I've found an issue when Asterisk tries to make a SIP call out using
auth, but has the wrong credentials and keeps getting returned a SIP
407, in this example to an OpenSIPs server requiring user auth.
Basically this happens:
1.
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