fan on wildcards.
then le came along, and then added dns01 support.
now i prefer a separate cert each plus a 3/1/1 tlsa for each port.
but at the time it was anoying.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
--
_
advise as set in stone, and so
asterisk refuses such certs. i doubt that stance is different
under sangoma.
the only workaround is to remind twil of the rfc and get them to
replace the wildcard with an rfc-copliant cert. at least for the
sip ports.
-JimC
--
James Cloos
>>>>> D'Arcy Cain writes:
>> Ie after both sides select t38, until they agree on the t38 terms.
> OK, so does that mean that setting it to 25000 should leave time for the
> re-invite or does the timeout start after that.
As I wrote above, after that. After the sip/
ce it starts.
Ie after both sides select t38, until they agree on the t38 terms.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the
sterisk, run this as root:
su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk
If it fails, then the problem is permissions.
You may need to alter the permissions on /etc/letsencrypt to allow
non-root uids to access the symlinks and their targets.
-JimC
--
James Cloos <cl...@jh
o->sip gateway will
cancel the sip call just like it would if the caller hung up.
(There is a possibility that any given gateway may not cancel the sip
call until the analog call is completed; you need to test.)
-JimC
--
James Cloos <
exec()), including a log at the start of what is in *data and args.
Looking at it, it only plays vm-whichbox when ast_strlen_zero(data),
which implies that the args to Voicemail are not making it through.
-JimC
--
James Cloos <cl...@jhcloos.
I enable full log and run 'core set debug 9' before doing a pair of
tests.
(The full log is easier to grep than the console output.)
Then compare a working vs stocktrans2 side by side.
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED
in the SIP From: header.)
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community
applications. They can be configured (in res_fax.conf) to use
t38 when available.
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api
the rounding mode), fmod(3) is
defined to trunc(3)ate the quotient.
So the result of x%y will always be in the range [0,x] and the results
of remainder(x,y) will be in the range (-y/2,y/2].
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED
n some places (including here) static ip is not affordable.
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Ast
at announce you'd have received -- I
expect -- quite a few complaints.
This flies in the face of all of the (very welcome) work which went into
supporting reload rather than restart.
Getting pjsip to support changes on a reload would be an acceptable
first step.
-JimC
--
James Clo
ck of full support for traversing nat makes pjsip worthless for a
large number of users. And the whole point of realtime is to have all
of the rt config fully dymanic.
If ari cannot avoid that limitation, chan_sip should get full ongoing
maintainance until pjsip is fixed.
-JimC
--
Jame
to use wscat with such a sub to get a better idea of what the
various events look like.
Thanks,
-JimC
--
James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://w
may also want to look into getting an ISN number, check out
http://freenum.org/ for the details.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http
the tls key.
The config name is tlsprivatekey; set that to the filename of your tls
key, akin to how tlscertfile is set.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation
like this:
register = tls://username:xxx...@sip-tls-proxy.example.org
(copied from the example sip.conf).
Set tlsbindaddr to the address to which to bind(2) the tls socket.
tlsbindaddr=0.0.0.0 is typical in ipv4-only configs.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP
. Those show the path of the SIP. In your
example, look for a Via which mentions 65.211.180.237.
Note that the From does not necessary have to be a sip address reachable
from the outside.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
far have
failed.
At the moment I'm on 12.7, but still using chan_sip. Converting the
chan_pjsip will be the next project for this box.
What is the proper way to set this up?
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
that.
The option space for espeak has large variability.
Flite also needs such tuning for nice output.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided
for both the incoming and outgoing legs,
it is a bit of a waste to proxy the rtp.
And even when the legs are associated with different remotes, I'd prefer
to proxy only when NATs a/or v4-v6 gatewaying are involved.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
. :)
JColp Yes.
Thanks!
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
on the matter of
what other endpoints are willing to do in such cases?
Thanks,
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
, but nothing quite worked.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
, but will doing so also
block secure media?
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
for things like sub-account, peer and/or trunk configs.)
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Expected:
RdSS Subject: 1504|12|Teste - Rafael 1570|16
The sent header decodes to this string:
Subect: 1504|12|Teste_-_Rafael_1570|0:16
Note the colon from $VM_DUR (minutes:seconds).
MUAs are supposed to decode that.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
/in the script session if the asterisk
box has limited storage. The debug output could get LARGE before
a modem stops.)
The script command is in the bsdutils package (apt-get install bsdutils).
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
the maxregexpire setting in asterisk's iax.conf
(in the [general] section) topermit values at least as large as 300 seconds.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
--
_
-- Bandwidth and Colocation Provided
-iax.conf
before editing the password lines it will be easier to read them.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
relying on an MUA.
The default mailcmd for app_voicemail is '/usr/sbin/sendmail -t' You
might also want to use the -oi flag.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
--
_
-- Bandwidth and Colocation
.
Sip/rtp over private ptp ethernet is an option with at least some of the ILECs.
They may call it virtual-pri or some such.
Of course, if they are installing an actual sonet ring, and not just a
spur, that can have built-in redundancy, depedning on physical routing.
-JimC
--
James Cloos cl
t == troxlinux xserverli...@gmail.com writes:
t I try to install asterisk on vps server , but fails when I want to
t install dahdi
There is no hardware for dahdi to use; you shouldn't need to install it.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
like xen does not automatically
break stuff. Were the box doing ISDN, on the other hand, routing the
pci card to the right partition can be an issue. But for 100% sip it
should just work.)
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
you /need/ the asterisk in the middle?
And if you /do/ need something between the two, might a sip proxy work
better than a pbx?
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
--
_
-- Bandwidth
which needs one. Then add that CA cert
to the bundle. Recent versions of tls (claim to have) deprecated the
idea of using self-signed certs for anything other than root ca certs,
but you can always create your own CA.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
homepage is at:
http://xinetd.org/
Your distribution will have documented.
And as an aside, the agi and fastagi frameworks for other languages
document what one needs to do well enough to code a similar library
in c, c++ or any other language.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP
) to handle the tcp side of things;
it can call your AGI app whenever asterisk makes the tcp connection, and
keep it open for future calls.
Then, just use stdin and stdout as you would for a normal AGI app.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
() functions may be enough.
You want to skip the silence and tone creation steps.
Or perhaps #defining CALLWAITING_REPEAT_SAMPLES to 0 might work.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
can discover what the
illegal instruction is.
I suspect your compile may expect a more recent cpu than you have, and
may use sse instructions which it doesn't support. A disassembly around
the failing instruction will confirm whether that is true and which
instruction it is.
-JimC
--
James Cloos
Rob == Rob Scott [EMAIL PROTECTED] writes:
Rob For work environments where you only get HTTP or HTTPS access,
Rob what is the feasibility of doing IAX over HTTP?
Tunnelling tcp/ip over http/(tls/)?tcp/ip is viable, tunnelling
rtp/udb/ip or iax/udp/ip over http/(tls/)?tcp/ip however will only
Matthew == Matthew Boehm [EMAIL PROTECTED] writes:
Matthew There is no way to convert existing files to g729?
The reference codec has a cli to do that. It converts from raw
16-bit signed linear files (sox filetype sw) to g729 files that
should work with *'s format_g729.
I beleive it is even
Bill == Bill Seddon [EMAIL PROTECTED] writes:
Bill My use of sox for down sampling is limited to
Bill this kind of command:
Bill sox in.wav -r 8000 out.gsm
You really want to use the polyphase app in sox for resampling.
It is significantly slower than the other options, but that is
irrelevant
Larry == Larry Shields [EMAIL PROTECTED] writes:
Larry On most VM systems you can press the * key or # key to get a
Larry login prompt during your greeting. Is that not possible with
Larry this system?
If you hist * during the outgoing message you'll get sent to the a
extension, if that exists
Rich == Rich Adamson [EMAIL PROTECTED] writes:
Rich If they are suggesting the sip negotiation process is trying to
Rich negotiate something like silence-suppression=off, and their
Rich equipment won't handle _anything_ other then
Rich silence-suppression=on, then that sounds like a short-coming
Christoph == Christoph Rothe [EMAIL PROTECTED] writes:
Christoph Which Formats will * accept and what extensions may
Christoph be used? Is there a page in the wiki about that ?
Look in the formats dir in the asterisk src. Each of those formats
can be used.
They are well documented in terms of
David == David Gurr [EMAIL PROTECTED] writes:
David As a result, I'd like to ensure that the voice prompts I'm
David using have the best possible audio quality.
David My callers will be coming in over PSTN to a VoIP gateway and
David then to me by uLaw/aLaw ...
The optimal quality in the case
Given:
Queue(foo|tHnr||bar)
where queue foo includes something like IAX2/gw/18005551212
should # transfer work on the remote phone?
A read of app_queue.c looks like it ought to work, but all
I get is dtmf sent to the caller.
(Incidently, I'd really prefer to be able to hit eg * during
the
50 matches
Mail list logo
