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 announce
reams will take a bit more cpu
than just proxying the rtp
-JimC
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> "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 address
> "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 pro
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 fo
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: *St4-
hat may be as difficult to get from your engineer as gsm.)
-JimC
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t supports them. They should be much more accurate
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
--
James H. Cloos, Jr. <[EMAIL PROTECTED]> <http://jhcloos.com>
___
> "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 a
...
You'll need to look in the variable ${SIPDOMAIN} to differentiate
eg [EMAIL PROTECTED] from [EMAIL PROTECTED]
-JimC
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>>>>> "Duane" == Duane <[EMAIL PROTECTED]> writes:
Duane> erm it already does, but it's labelled PROC=
Ack. Yes it is.
But it isn't used everwhere it could be
-JimC
--
James H. Cloos, Jr.
> "brian" == brian k west <[EMAIL PROTECTED]> writes:
brian> I toyed with -msse and -mmmx and others too but couldn't put
brian> any of those in. :P
The options -msse, -msse2, -mmmx et al are all implied by the
relevant -march options. uname only reports i686, so you have to use
some other c
be in the next release, and is now available in
ethereal's anon cvs tree.
-JimC
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3 2 2 2 2 1 6 - -19
ILBC - 5 4 4 4 4 3 8 -18 -
-JimC
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James H. Cloos, Jr. <[EMAIL PROTECTED]> <http://jhcloos.com/voip>
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This is from nanog; I presume there is significant interest from
readers here not also on nanog
I've edited it to only the interesting part...
-JimC
--- Begin Message ---
...
Randy Bush made the first offer of space, so the new list address is:
[EMAIL PROTECTED] The new list address is [EMA
long the lines of:
exten => .,1,Wait,1
exten => .,2,Dial(iax2/whereever/${EXTEN})
-JimC
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also presume we are all listed in the wiki:
http://voip-info.org/tiki-index.php?page=VOIP+Service+Providers
-JimC
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>>>>> "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
> "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
___
Asterisk-Use
Steven> My guess is they are providing the bandwidth and phone service
Steven> on that one T1. For that matter, it looks like they might be
Steven> pulling a data only line to your potential customer and then
Steven> running VoIP back to their end.
I took a look thru their website. They are using
> "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
> "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 t
se64Â of the sha hash of some
entropy and the call's ${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
--
James H. Clo
link, 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
--
James H. Cloos, Jr. <[E
,
firebird, oracle or whatever.
I'd extimate the code would take just a few hours to write
and debug. Surely less than a coder-week.
-JimC
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James H. Cloos, Jr. <[EMAIL PROTECTED]> <http://jhcloos.com>
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[
> "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:
nc tiffcrle
Eric> tiffg3 tiffg32d tiffg4 tifflzw tiffpack
Most of the fax solutions, including spandsp, will prefer
ghostscript's tiffg3 device.
-JimC
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James H. Cloos, Jr. <[EMAIL PROTECTED]> <http://jhcloos.com>
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> "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/
This was posted to the hylafax-devel list; I presume it
is also relevant to spandsp:
--- Begin Message ---
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 r
> "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 tha
> "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 leas
> "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 licens
Netlabz> Anyone got dialing pstn 800#'s from asterisk behind nat with
Netlabz> fwd.pulver.com or iaxtel working ?
I'd suggest using the service provided by/for freenum.org.
See the archives for how to setup * to do an enum lookup on
freenum.org, or do a manual lookup and use that data statically.
> "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
> "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:
ht
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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> "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
> "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 inte
> "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 o
> "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; U
> "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
extensio
> "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 i
> "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
> "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 arc
> "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
> "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.
Ma
> "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 s
> "D_JV" == Deepakumar JV <[EMAIL PROTECTED]> writes:
D_JV> when i dial a toll free no using sipphone i get this error message.
D_JV> channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW
sipphone is preferring to use the G729A codec; if they only support
that you will
|> 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
shoul
> "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
_
> "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 k
> "Darren" == Darren Martz <[EMAIL PROTECTED]> writes:
Darren> Is there another way that is more cost effective?
Get a quote on an 8 trunk voice T1 and on an 8B+D pri from your
telco -- and any clecs in the area.
If it is competative you may not need the channel bank
-JimC
__
It looks like digium is once again in the com. zone, although only
one nameserver is listed:
com.zone.gz: DIGIUM NS LINUX-SUPPORT.NET.
Odd that it fell out exactly 30 days before it was set to expire
Now two different registrars have fubared two different companies
here. One more and the co
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
> "T" == T Chan <[EMAIL PROTECTED]> writes:
T> So are you saying that I should see 1 PID for
T> "safe_asterisk" and many PIDs for "asterisk -vvvg -c"
On old distributions one would expect this.
T> or [1 PID for "safe_asterisk" and] just 1 PID for "asterisk -vvvg -c"
The Newest distribution
> "TC" == TC <[EMAIL PROTECTED]> writes:
TC> usual issue here is one unnamed distro's patched 'ps' cmd thinks
TC> you only want to see the parent PID of all running threads on the
TC> box, dont that just turn your red hat over ?
Get used to it. With NPTL all the treads share the same pid, s
> "John" == John Todd <[EMAIL PROTECTED]> writes:
John> For this iteration, I'll probably submit the files to my website
John> in both .aif and in .gsm
Is that aiff format (as defined by Apple, et al)?
It would be good to have uncompressed copies of all of the publicly
available prompts. Ha
> "Walt" == Walt Reed <[EMAIL PROTECTED]> writes:
Walt> Anyone know how to "turn off" the RFC3389 support on the ata 186?
I beleive that is the AudioMode setting. It should look something like:
AudioMode:0x00150015
(that is the default).
Make the 4th and 8th digits even to turn off g.711
> "Darren" == Darren Nickerson <[EMAIL PROTECTED]> writes:
Darren> Except that these boards are often sold by VARs like ourselves
Darren> who don't advertise prices in a way that makes it into
Darren> froogle, but are often cheaper than other channels.
Even better!
(The last time I took a
> "Darren" == Darren Nickerson <[EMAIL PROTECTED]> writes:
Darren> you might consider ... a digital fax board, like a Brooktrout
Darren> TR1034 of an Eicon Diva server.
Just be sure to shop around for the Brooktrout or Eicon boards;
froogle.google.com is your friend.
-JimC
_
> "JR" == JR Richardson <[EMAIL PROTECTED]> writes:
JR> I ran that export command you
JR> suggested, then launched *, everything worked fine. I'm still
JR> looking for info on what that command actually does. Can you shed
JR> some light please?
Exporting LD_ASSUME_KERNEL=2.4.1 tells libc to
> "Steve" == Steve Underwood <[EMAIL PROTECTED]> writes:
Steve> SRTP has been through the process of trying to deal with this
Steve> in the most effective manner, but doesn't seem to be widely
Steve> used right now. Free implementations exists - see
Steve> srtp.sourceforge.net. I guess it shou
James Golovich <[EMAIL PROTECTED]> wrote:
>Change your sprintf to:
>$hirestime = sprintf("%d%06d", $seconds, $microseconds);
>This will make it so that microseconds will always be 6 characters long
>or change it to something like:
>$hirestime = sprintf("%d.%d", $seconds, $microseconds);
>So there
> "Ryan" == Ryan Tucker <[EMAIL PROTECTED]> writes:
Ryan> They are the same key. I'm not sure how the # came to be associated
Ryan> with the word "pound", but in American English at least, they're the
Ryan> same key.
The weight measurement pound is abbreviated lb. # looks similar in
some ha
> "Matteo" == Matteo Brancaleoni <[EMAIL PROTECTED]> writes:
Matteo> I was wondering what are the values for sending dmtf via sip
Matteo> info. I mean, when I use dtmf relay via sip info, the sip/sdp
Matteo> message contains a Signal=X where X is the dmtf. That's ok
Matteo> for dtmf 0-9 . bu
> "Jamie" == Jamie Neil <[EMAIL PROTECTED]> writes:
Jamie> I had pretty good results recording from a standard headset, on a basic
Jamie> sound card at 44Khz mono.
Note that if your soundcard can do eg 48kHz or 32kHz linear, the
conversion down to 8kHz log will sound a bit better -- perhaps a
> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes:
Mark> The problem with this and MySQL is letting MySQL understand
Mark> Asterisk-style pattern matching,
Didn't someone post some code that converts * patterns to regexen?
That seems like it would be useful for this kind of stuff, yes?
-Ji
> "Dan" == Dan <[EMAIL PROTECTED]> writes:
>> Agreed. Jeremy McNamara of Nufone.net is the top dog in Asterisk
>> VOIP and long distance.
Dan> Hi, How can you subscribe to this service? There is no web page
Dan> available to do it.
I emailed them at [EMAIL PROTECTED], as per one of the pa
> "Marcus" == Marcus Adolfsson <[EMAIL PROTECTED]> writes:
Marcus> Nufone.net is the best VoIP provider for Asterisk
Marcus> integration. They offer IAX termination, 2.9 cents outgoing
Marcus> long-distance and incoming 800. We use them at our office for
Marcus> all phone calls.
I second this
> "Dan" == Dan <[EMAIL PROTECTED]> writes:
Dan> I have seen all of this on a Cisco 17xx router, including IVR
Dan> and sending faxes through e-mail, but it is far too expensive for
Dan> me... Theoretically is possible to have let's say an IAX or SIP
Dan> software phone, even on a separate co
> "Iain" == Iain Stevenson <[EMAIL PROTECTED]> writes:
Iain> I didn't used to have any trouble with FWD and * is registering
Iain> with FWD OK. Has FWD changed or * changed in a way that might
Iain> cause this error?
Jeff just announce an upgrade to fwd the other day.
One change is that cal
> "Jim" == Jim Flagg <[EMAIL PROTECTED]> writes:
Jim> Have you tried limiting your fax machines to a lower baud rate
Jim> like 9600. I know on Vonage this seems to help.
Speaking of which, IIRC the docs for the ata mention that fax at
greater than 9600 is b0rked up to a recent firmware relea
Jeff just posted on [EMAIL PROTECTED] about some changes to
fwd, and mentioned that at least some clients may need to register
@fwd.pulver.com rather than @192.246.69.223.
Given the recent thread either here or on -dev about the syntax of
the register command for sip peers, I suspect the register
> "Christopher" == Christopher Arnold <[EMAIL PROTECTED]> writes:
Christopher> But is asterisk -rx "show channels" really not meant to
Christopher> work from cron or an noninteractive script? If so i guess
Christopher> some documentaion about the limitations would help other
Christopher> folks
> "Moshe" == Moshe Yudkowsky <[EMAIL PROTECTED]> writes:
Moshe> The "sox" program will convert wav into gsm:
And can also be used for playback. The sox package usually comes
with a script called play; you can use that for easy playback of most
audio formats:
play foobar.gsm
-JimC
> "Martin" == Martin Dommermuth <[EMAIL PROTECTED]> writes:
Martin> looks like Germany is again laggin behind all others in the
Martin> communication field. Or I asked at the wrong place. There
Martin> might not be to many people from Germany in this list.
One possibility is Pulver's LibréT
> "flickds" == flickds <[EMAIL PROTECTED]> writes:
flickds> Is it possible for two PDA's to communicate like
flickds> telephones via SIP channels on a PC running Asterisk?
Certainly. There are several softphone apps available for the
various platforms. And you can of course talk to any othe
I tried putting this as the ata's dailplan:
*St4-|#St4-|9|^9t4>$.-
this is sip.conf
[ata2001]
type=friend
username=ata2001
secret=SoMeSeCrEt
host=dynamic
context=fromata
canreinvite=no
and this in extensions.conf
[fromata]
ignorepat => 9
exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTEC
> "Tim" == tmassey <[EMAIL PROTECTED]> writes:
TIm> My first thought on the higher end was a PRI line. However, the
TIm> cost seems *very* prohibitive. The average cost for a PRI line
TIm> was $550/month, just for "dial tone"! I've heard others say that
TIm> PRI becomes cost effective in t
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