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
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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
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Check out the new Asterisk community
applications. They can be configured (in res_fax.conf) to use
t38 when available.
-JimC
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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
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n some places (including here) static ip is not affordable.
-JimC
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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
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may also want to look into getting an ISN number, check out
http://freenum.org/ for the details.
-JimC
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the tls key.
The config name is tlsprivatekey; set that to the filename of your tls
key, akin to how tlscertfile is set.
-JimC
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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
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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
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http
. :)
JColp Yes.
Thanks!
-JimC
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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
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New
, but nothing quite worked.
-JimC
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New to Asterisk? Join us for a live introductory
, but will doing so also
block secure media?
-JimC
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New to Asterisk? Join us for a live
for things like sub-account, peer and/or trunk configs.)
-JimC
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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
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-iax.conf
before editing the password lines it will be easier to read them.
-JimC
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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
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.
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
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--
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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
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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
experimenting to determine the optimal amplitude to avoid both
clipping and too-little use of the available u/a-law bandwidth.
-JimC
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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
Wolfgang == Wolfgang S Rupprecht [EMAIL PROTECTED] writes:
Wolfgang One of the other posts mentioned their ATA that simply
Wolfgang registered with all the addresses. I don't think it would
Wolfgang be a big or difficult change to have asterisk register with
Wolfgang all the addresses also.
proxying the rtp
-JimC
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Has anyone does this with *?
Ie, ask for the caller's name and provide that to the callee before
bridging?
For calls to an extension, it should be doable via the dialplan. For
calls to queues, some changes would be required to app_queue.c to
allow an addional file to be played after the
Steven == Steven Critchfield [EMAIL PROTECTED] writes:
Steven oddly enough, there isn't much if any difference these days at
Steven the physical level. It is just the interface and the set of
Steven specs on the interface. SCSI drives usually will give you
Steven warning of their problems.
As I
Chris In this case, you want to not pay the T1 fee but still
Chris pay low per number rates.
That is not what he wrote.
And there is a definite market out there for exactly what he
specified: a fixed number of simultaneous calls for a fixed
MRC, plus some (typically larger) block(s) of DIDs for
I use DISA on the asterisk box and have the dialplan on the ata set
so that calls starting with 9 or 8 have only two digits.
disa extensions 90 - 99 are for pstn calls via various providers.
Those in 80 - 89 are for fwd and other similar services.
The ata's dialplan looks like:
DialPlan:
.)
-JimC
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than the older hw.
On a 2.8 GHz dual p4 (not xeon) I'm seeing a jitter of only about
2 ms as reported by ntpq with the (strat 2) remote clock 60 ms away.
-JimC
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Randy == Randy Bush [EMAIL PROTECTED] writes:
Randy i try to place a call
Randy exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
Randy where sip.conf has an entry
Randy [foo]
Randy type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user
} to differentiate
eg [EMAIL PROTECTED] from [EMAIL PROTECTED]
-JimC
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2 1 6 - -19
ILBC - 5 4 4 4 4 3 8 -18 -
-JimC
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it will be in the next release, and is now available in
ethereal's anon cvs tree.
-JimC
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brian == brian k west [EMAIL PROTECTED] writes:
brian I toyed with -msse and -mmmx and others too but couldn't put
brian any of those in. :P
The options -msse, -msse2, -mmmx et al are all implied by the
relevant -march options. uname only reports i686, so you have to use
some other construct
Duane == Duane [EMAIL PROTECTED] writes:
Duane erm it already does, but it's labelled PROC=
Ack. Yes it is.
But it isn't used everwhere it could be
-JimC
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/whereever/${EXTEN})
-JimC
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This is from nanog; I presume there is significant interest from
readers here not also on nanog
I've edited it to only the interesting part...
-JimC
---BeginMessage---
...
Randy Bush made the first offer of space, so the new list address is:
[EMAIL PROTECTED] The new list address is [EMAIL
Kyle == Kyle Hagan [EMAIL PROTECTED] writes:
Kyle In coming works fine from FreeWorld via IAX. But when
Kyle Dialing out i get [an error] ...
Does iax2.fwdnet.com even support iax2=fwd? I thought it was just
for registering an iax2 endpoint for fwd=iax2 calls.
-JimC
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Tim == Tim Petlock [EMAIL PROTECTED] writes:
Tim Be very careful about them. Search the archives of
Tim comp.dcom.telecom for details - focus on the last twelve months.
Ah, yes. I knew the name sounded familiar.
-JimC
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Nick == Nick Knight [EMAIL PROTECTED] writes:
Nick What is the expected uptime for asterisk - assuming the
Nick box has all the resources it needs.
Months. You should only have to reboot for kernel updates and
restart * when updating it or (some parts of) its configuration.
Nick I ask this
just a few hours to write
and debug. Surely less than a coder-week.
-JimC
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, but the other
side seems to be ignoring those packets. So I tried inband on that
link; nothing was able to recognize my dtmf there either.
I have to presume that either oob-inband is broken for - rtp
or it is broken w/o a zap timing source
-JimC
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${UNIQUEID} for the filename.
The message order can be kept in the db table with the rest of the
meta data.
-JimC
(don't you love neologisms)
be sure to use a filesystem-friendly version of base64
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James == James H Thompson [EMAIL PROTECTED] writes:
James Would it make any sense to store the voice mail formatted as a
James email msg in a Maildir directory structure. Then you could
James also retreive them with an email client.
That is not a bad idea. * would have to convert the mime
Brian == Brian Cuthie [EMAIL PROTECTED] writes:
Brian Is anyone else having trouble placing toll-free calls though
Brian IAXTel lately? Mine just stopped working yesterday, yet I
Brian seem to be able to make 1-700 calls.
I'd suggest using enum lookups on freenum.org instead.
Cf:
Juan == Juan J Sierralta P [EMAIL PROTECTED] writes:
Juan I been playing with RxFax ... I received a FAX and it seems
Juan that the aspect ratio of the image is different, ... The image
Juan resolution is 1728x1092.
Traditional fax has two resolutions: 98 lines/inch and 196 lines/inch.
of the fax solutions, including spandsp, will prefer
ghostscript's tiffg3 device.
-JimC
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This was posted to the hylafax-devel list; I presume it
is also relevant to spandsp:
---BeginMessage---
On 2004.03.22 05:16 David Brownlee wrote:
Has anyone tried using Hylafax with libtiff 3.6.1?
On a NetBSD/i386 box libtiff 3.6.0 works flawlessly
but 3.6.1 gives corrupted tif files on
Steven == Steven Critchfield [EMAIL PROTECTED] writes:
Steven As I understand the PCI spec, there are 4 interrupt lines
Steven called A,B,C, and D. In slot 1, They appear in that order. In
Steven slot 2 they shift, in slot 3 they shift and again in slot 4.
That is correct, except that all
Greg == Greg Kedrovsky [EMAIL PROTECTED] writes:
Greg I started it with asterisk ... Then ... I did asterisk -r
Greg to ... get a console. The manual says ... type quit to
Greg disconnect ... But, [it didn't work] ...
What version of *? With recent cvs it works. Or at least exit
works.
Don == Don Pobanz [EMAIL PROTECTED] writes:
Don I do not know what 'Linux-style subscription license' means.
That one stalled me for a bit, too. Based on their ad copy they
are offering annual support contracts for the system, but releasing
the code itself under some free/open license. (I
Rana == Rana Dutt [EMAIL PROTECTED] writes:
Rana My attempts to use voice mail from my Grandstream Budgetone 101
Rana phone always fail because Asterisk is seeing either double
Rana digits or dropped digits, no matter what dtmfmode setting I try.
Try the patch in bug number 1034:
Dan == Dan [EMAIL PROTECTED] writes:
Perhaps its that Dan's box is trying to register with IAX1?
Dan Great!!! This must be my problem. I have IAX(1) still active (in
Dan order to test my DIAX). I will disable it.
The iax1 module will first try to load iax1.conf before trying to load
You'll probably want to re-quant them to 8kHz, but there are quite a
few classical tracks available at:
http://hebb.mit.edu/FreeMusic/
-JimC
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To
Mickey == Mickey Binder [EMAIL PROTECTED] writes:
Mickey I want to completely hide my outgoing CallerId when dialing
Mickey out on my Zap interface.
What kind of zap interface?
If it is an fxo card on a standard pots line, treat it as such and
prefix the dialed number with the right
bam == bam [EMAIL PROTECTED] writes:
bam Is there a way to allow a caller to enter an extension
bam number that is more than one digit long in a voice menu?
In addition to what the other replies say, I'd note that it
is usually a good idea to not use the initial digit of the
extensions as one
Steve == Steve Rodgers [EMAIL PROTECTED] writes:
Steve BTW: If you are a low volume user, it seems to make more sense
Steve to go with one of per-minute plans offering IAX connectivity.
Low volume in this case is quite large. USD 20 per month will
net you around 675 to 690 minutes; USD 30
Costa == Costa Tsaousis [EMAIL PROTECTED] writes:
Costa Are there any well known good H/W configurations for high
Costa density E1 setups supporting * and FAX?
To do fax well still requires something on the board itself handling
the (de-)modulation.
Unfortunately, the current state of the art
Darren == Darren Nickerson [EMAIL PROTECTED] writes:
JimC Hylafax.org has pointers to a couple of good boards for fax.
Darren The HylaFAX.org website is a little lacking in terms of
Darren describing high-density (T1/E1) fax with HylaFAX
Darren We recommend Brooktrout or EICON intelligent fax
Kris == Kris Stark [EMAIL PROTECTED] writes:
Kris On a different note - is something up with the freenum.org enum
Kris lookups? ... I've had them fail on all US numbers...
The nameservers for freenum.org. have glue records for 1.freenum.org.
that point to garthim.fox-den.com. (which is at
Brian == Brian Capouch [EMAIL PROTECTED] writes:
Brian On a broader note, I would love to try to play with the
Brian very-low-bandwidth versions of Speex. I could have sworn I saw
Brian things on the bugtracker some weeks back on that topic, but I
Brian can't find them anymore.
It is bug
Marc == Marc Fargas [EMAIL PROTECTED] writes:
Marc I've seen its possible to use the System applications, but what
Marc about passing arguments to the command ?
A quick look at app_system.c shows that it just passes the string
unaltered to system(3). So, running man 3 system will show exactly
Marc == Marc Fargas [EMAIL PROTECTED] writes:
Marc It drives me to a new question... how can I concatenate three
Marc strings on extensions.org ?
Marc That is, the command, and the two args; The arguments are the
Marc source e164 and destination e164 numbers of the current call.
Marc Something
Scott == Scott Russ [EMAIL PROTECTED] writes:
Scott Does anyone know if/how well Asterisk will run under User Mode
Scott Linux? Will the ztdummy or zaprtc modules work with it?
Haven't tried the modules, but an all-voip setup works well, provided
there is enough ram set aside for the instance,
T == T Chan [EMAIL PROTECTED] writes:
T if I configure that way, even 01163 calls will all go to the second
T IP address as per 011.,1,Application(). If I take out the 011.,
T then calls WILL go to 01163., if I put the two together it will
T always go to 011. extension.
The list archives have a
Steve == Steve Foy [EMAIL PROTECTED] writes:
Steve Is there a way of logging all SIP debuging info to a file
Steve somewhere?
Use tethereal or tcpdump to log sip (and/or rtp/rtcp) packets to a
pcap file, then use ethereal (presumably on a different box) to view
them.
-JimC
| I'm still getting a sementation fault with mpg123.
Isn't it time to get mg3 out of the equation?
Sox can convert just about anything to 16 bit signed mono pcm in
just about any container that support that. It looks like *'s
format_wav.c is for exactly that format, so for local files we
T == T Chan [EMAIL PROTECTED] writes:
T Whenever I get to 10 calls or more, I would start to get
T choppy sound. I tried to ping other IP addresses from the Asterisk
T and noticed a big packet loss in the vincinity of 7% to 10%, ...
What does your /proc/interrupts look like?
Which kernel are
With a bit of coding it can be done w/o mp3.
asterisk/res/res_musiconhold.c needs to me modified to know how to
handle some other format. I'd suggest 16-bit mono pcm files with
either wav or au headers. If you are only dumping your MoH to zap
ports, g.711 with wav or au headers is also a good
T == T Chan [EMAIL PROTECTED] writes:
T Thanks alot for your explanation. Can you tell me if there is a way
T to confirm if I have the nptl in the boxes ?
grep for nptl in the installed pthread libs:
grep -i nptl /lib/libpthread.so.0 /usr/lib/libpthread.a
does it on my box.
-JimC
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