So who out there is aware of the FCC or FTC laws concerning spoofing caller
ID for deceptive purposes? There's a collection agency out there who has my
wife's name crossed with someone else's, and they are picking numbers from
our area code to present themselves as when calling us (over and over
Yeah I'm thinking either homeland security or some other identity-critical
legislation might be on my side here.
On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED] wrote:
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear [EMAIL PROTECTED]
wrote:
So who out there is aware of the FCC
/voip/chatter_bug.pdf
PaulH
On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote:
Does anyone know anything about the Chatterbug product? I can't tell
if it's an ATA with a modem or some sort of LCR proxy or somesuch.
Anyone
Does anyone know anything about the Chatterbug product? I can't tell
if it's an ATA with a modem or some sort of LCR proxy or somesuch.
Anyone?
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To
Has anyone out there tried to mix the envelope metadata for
voicemails into the audio payload that's stored by Asterisk? I would
like to have the CID and Timestamp baked into the beginning of the
WAV file, not just as text in the email itself.
Thanks!
-Rob.
So in my ignorance I bought a Zoom 5806 ATA from Micro Center. It was
cheap, what can I say?
Anyhow, the docs are horrible, but the control panel is fairly
straightforward. I can get it to register against Asterisk but I
cannot get it to dial.
Does anyone have a working configuration
of reliability and great service.
Just my $.02
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Knight West
949.542.7001 DIRECT
949.542.7010 FAX
888.272.6387 x501
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[EMAIL PROTECTED]
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On Apr 25
Hello all,
Witch snmp system do you use to collect info about their asterisk
boxes, for example, uptime, downtime, max load, HD, free memory,
asterisk status, ,etc?
I use Nagios and the extension that logs in to the * manager interface.
Check for the existence of INBOX and OLD folders in the VM folders in /var/spool/asterisk/voicemailI messed around with something and found this out the hard way.Let me know if this works, as I am curious.-Rob.On Oct 10, 2006, at 12:58 PM, stan ford wrote:the last thing i was trying to do was
Couple suggestions:On Win32, try Serv-U FTP. It's very reliable and supports a variety of protocols like SFTP and the like. Commercial software.http://www.rhinosoft.comOn Linux, I've fallen in love with FreeNAS, and not just because it utilizes the m0n0wall GUI. You can boot it off a USB stick,
Whoops. I totally generalized under the realm of Non Windows didn't
I? Doh!
On Oct 8, 2006, at 2:27 PM, Tzafrir Cohen wrote:
On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote:
On Linux, I've fallen in love with FreeNAS, and not just because it
utilizes the m0n0wall GUI. You
the
firewall to the server that's serving the image.
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On Fri, October 6, 2006 20:13, Robert Goodyear wrote:
Anyone know why I get HTTP Connection Closed on the display of a
7960 running a SIP image?
Only seems to happen when registering against my Asterisk box from
the WAN. I have 1:1
Anyone know why I get HTTP Connection Closed on the display of a
7960 running a SIP image?
Only seems to happen when registering against my Asterisk box from
the WAN. I have 1:1 NAT happening on my firewall. Phones function
perfectly otherwise. TFTP working fine across the firewall as
and it works fine now, and
we haven't noticed a severe degradation in sound quality - most of my
operators were just happy the echo was gone :)
+1 here too:
Uncommenting AGGRESSIVE_SUPPRESSOR and recompiling took care of 99%
of my TE110P/PRI echo.
-Rob.
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On Aug 18, 2005, at 3:07 AM, Stephen wrote:
Hi All,
How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip.
I want to lock my extension during my absence.
Can it be done in Asterisk?
regards,
Stephen
You could write a little script to
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or … any other brand and model that is known to work well with the TE411P ? Will an old Proliant
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio
from
the called channel(s) until one answers. Use with care and don't
insert
this by default into all your dial statements as you are killing call
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On Aug 4, 2005, at 2:10 PM, Robbie Hughes wrote:
I have a 12 channel PRI with SNOM 190's and asterisk CVS from January.
Most calls are fine, all incoming calls are fine, but I am getting
echo on a significant number of outgoing calls
On Aug 4, 2005, at 10:37 PM, Martin Kronstad wrote: Hi!Problem:I can’t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound.My current setup is:Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet -
On Jul 18, 2005, at 3:13 PM, Michael D Schelin wrote:
Here is a letter I sent them for my $150 paper weight.
The forum is not a place to post ransom notes. You've added zero
benefit to any reader here, nor to yourself, since you didn't actually
ask a question in your email.
On Jul 20, 2005, at 12:22 AM, Brian Capouch wrote:
Michael D Schelin wrote:
Real scary who
You certainly have found an unusual way to promote your business.
B.
Kinda sounds like a schoolyard taunt, usually found near most lemonade
stands, doesn't it?
Is is possible to specify the MOH Class when defining a MeetMe
extension?
I tried
exten = 300,1,MeetMe(300|M(class))
But that did not work.
Thx,
-Rob.
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On Jul 14, 2005, at 11:17 AM, Robert Goodyear wrote:
Is is possible to specify the MOH Class when defining a MeetMe
extension?
I tried
exten = 300,1,MeetMe(300|M(class))
Replying to my own query, just in case anyone else is as dense as I
am...
exten = 300,1,SetMusicOnHold(confclass
Anyone know how to bypass the CONFIRMATION of the user announcement
recording in MeetMe?
While I like the please say your name to announce a user into a
conference, I find it confusing and time consuming to make the user to
press 1 to accept a recording they haven't even previewed.
I'm not
, but this is a hardware-neutral
question.
Thanks.
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On Jul 8, 2005, at 12:43 AM, Jay Milk wrote:
All,
I'm currently only setting CID as a ten-digit number. Has anyone on
this list tested caller-id delivery with various services? Is there
*one* usable format (i.e. 1+10, or +1+10), or does it vary from
provider to provider?
Jay, FWIW the US
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Thx,
-Rob.
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On Jul 4, 2005, at 2:43 PM, Jimmy Smith wrote:
you guys are so friggin funny..
We try. Meanwhile, you are SO illiterate; are you trying?
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and recompiled and all
was well.
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On Jul 4, 2005, at 12:17 AM, Bryce Chidester wrote:
Just a thought, but I seem to recall that in the dialplan, inlcude and
other similar statements are not prefixed by the hash character (#).
Try include = .
-Bryce
You're thinking of contextual includes, not filesystem includes --
which
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
On Jul 2, 2005, at 1:00 PM, Roland Zagler wrote:
sorry for the misunderstanding, robert!
the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote:
Robert Goodyear wrote:
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try
to dial an extension (from X-Lite), next
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
the point is: during the caller is listening to the soundfile played
to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should
be
connected
to the sip phone 100.
Ah, now that's a clearer picture of
))
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On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote:
try this one
exten = 999,1,Answer()
exten = 999,2,playback(~.mp3)
exten = 999,3,dial (sip/100)
exten = 999,4,playbackground(~.mp3)
exten = 999,h,Hangup()
not sure abt playbackground should be before the dial command or after
Mahmoud:
On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:
Thank you, Robert!
The announcementfile plays well, but at Dial-option m i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).
Noted, which is why I offered option two.
Background command waits for a user
the problem by registering a hard SIP client on that server?
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On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote:
Robert Goodyear wrote:
I'm sure you really only want to know about the absence of problems.
From watching this list for 6 months it seems the SuperMicro products
are most lauded and have exhibited no hardware conflicts. Various
hem that have never missed a beat -- Dimension boxes, not PowerEdge.
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On Jun 29, 2005, at 3:40 PM, harry gaillac wrote:
Hello,
If asterisk.org can't provide you documentations have
a look here :
http://www.digium.com/index.php?
menu=product_detailcategory=softwareproduct=ABE
I do hope some people understand my posts.
Regards
Harry
Yeah, loud and clear.
On Jun 28, 2005, at 2:06 PM, Chris Stinson wrote:
Were you guys able to figure this out?
Robert Goodyear [EMAIL PROTECTED] wrote :
On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:
Hi Robert,
Let me guess... mailbox 5103 or 5203 were the last in the list to
receive it?
Every trials(1-6
On Jun 27, 2005, at 9:38 AM, Wilson Pickett wrote:
Just so you know who you're dealing with:
-- Forwarded message --
From: harry gaillac [EMAIL PROTECTED]
Date: Jun 24, 2005 7:58 PM
Subject: Re: [Asterisk-Users] polycom soundpoint ip 300
To: Wilson Pickett
i piss on you Wilson
On Jun 27, 2005, at 1:04 PM, David John Walsh wrote:
Hello,
does anyone know how to get the say number (say.c) agi application
to work in french [assuming that I have the French voice files]
Maybe Harry knows but hasn't documented it yet.
Sorry, couldn't resist :-)
Look for option o.
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Brand Up LLC
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this at you so can make an informed decision.
Robert Goodyear
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, but I was just throwing this at you so can make an informed decision.
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
x-tad-biggerRobert,/x-tad-bigger
x-tad-bigger /x-tad-bigger
x-tad-biggerEssentually I want to be able to have Server B dial the extensions connected to server A as well
or directory
I would like to copy to 100-150 mailboxes for one CPU.
I also need someone's help.
Let me guess... mailbox 5103 or 5203 were the last in the list to
receive it?
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:
Hi Robert,
Let me guess... mailbox 5103 or 5203 were the last in the list to
receive it?
Every trials(1-6) I got only 51 mailboxes copied. My quick guess is
256/5(u0103 and xx03s)=51...1, so changing tmp[256] to tmp[4096]
does not work.
-users/2005-May/107245.html
-Rob.
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still be paying the same amount thus half the savings would be...?
Sorry, just had to inject some Friday afternoon humor onto the list.
Seriously though, I was never able to get a T1 for that price anywhere myself until I moved to Orange County, CA.
-Rob.
--
Robert Goodyear | Managing
On Jun 16, 2005, at 8:23 AM, Kumara Jayaweera wrote:
Starting simple switch on 'Zap/1-1'
-- Executing
Dial(Zap/1-1,IAX2/[EMAIL PROTECTED]/10094472239112|30|tr)
in new stack
-- Called [EMAIL PROTECTED]/10094472239112
What country code is that you're dialing?
Robert Goodyear
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096] in an earlier
what I'm paying here in Orange County. Also,
I had a business cable modem before, which was *allegedly* not shared
for business customers (suspicious) and the throughput was a roller
coaster, as was the latency. The DS-1 cleared all that up.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand
the issue.
Let me know... I'm very curious now!
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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would want this and couldn't even come close to ever making it cost
effective for them to make such a change to their code.
Right?
Robert Goodyear
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. Might lessen the load on Asterisk by not waiting on a second, remote connection.
/rg
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before mucking around with source
and recompiling.
/rg
Robert Goodyear
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On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec set
the tmp[256] to be tmp[4096] in an earlier replay so I did.
static int vm_exec(struct ast_channel *chan, void *data
5631 are your 32 users.
exten = 400,1,VoiceMail(u401402403)
exten = 401,1,VoiceMail(u560056015602...5619)
exten = 402,1,VoiceMail(u562056215622...5639)
Wonder if that would work?
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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not registering, you're connecting via the HOST, USERNAME and SECRET in
the context in IAX.conf, right?
Robert Goodyear
Brand Up LLC
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phones. Also, what happens when one of those phones is
busy? If it goes straight to VM then that'll blow the whole timeout
trick.
Robert Goodyear
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Robert Goodyear wrote:
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote:
So, anyone else have any ideas? I tried the below suggestion and
it's still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make install
On 5/2/05, Chris
On Jun 6, 2005, at 4:31 PM, Chris Mason (Lists) wrote:
Digium did acquiesce and allow me to relicense the codec today,
essentially
they asked me how any times I would like to be able to re-license,
I hope you answered, As many times as is necessary to ensure I'm
continually able to
I've seen this in the list archives; nobody had an answer.
Having dug through tons of OSI docs, I cannot figure out what a second
ROSE component of type 0x6 even is, much less debug its origin or
reason the libpri pri facility code hates it.
Anyone?
Ref.:
PRI NI2
TE110P
CVS HEAD as of
Christian Hiller wrote:
Hello,
we got a SNOM 360 here and this gota TRANSFER button.
With this i can transfer a call from my phone another one. But when i
push this Button and transfer the call to another phone, i get kicked
out.
Now, every secretary first asks the chief if he is available
Anyone know what ring requested on channel 0/23 already in use on
span means?
Happened this morning and locked up all inbound and outbound calls. Had
to shutdown -r to repair it.
PRI
TE110P
CVS HEAD as of 05/24/05
Thanks in advance.
/rg
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Unfortunately, the technology does not currently exist to make that
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On May 31, 2005, at 4:30 PM, Karl J. Vesterling wrote:
Garrett, evidently there is some verbage to that effect on the site.
But just to let you know, no other business that we've done business
with requires anything like that. Not a one.
Also worthy of note is that the purchase was
On May 31, 2005, at 8:05 PM, Andy Hamilton wrote:
On 5/31/05, Robert Goodyear [EMAIL PROTECTED] wrote:
Does anyone know how to suppress the Missed Calls indication --
perhaps on a per-line basis -- on the 7960 running SIP?
Reason: I've configured a group of extensions to ring for inbound
On Jun 1, 2005, at 9:38 AM, Ing CIP Alejandro Celi Mariátegui wrote:
What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
No, renaming won't work, as it's a signed binary. Plus S versus O
designates the
Does anyone know how to suppress the Missed Calls indication --
perhaps on a per-line basis -- on the 7960 running SIP?
Reason: I've configured a group of extensions to ring for inbound calls
and it seems pointless to accrue missed calls on those line
presentations.
/rg
It's not the licenses, that's like 10% of the problem. One can always buy licenses... But the 16 man hours I wasted waiting for the across-town overnight shipment vastly outweighed the cost of the licenses.
To mitigate risk, why didn't you ask to pick up the product in person?
The main gripe
On May 24, 2005, at 9:30 PM, SYED ADEEL ALI wrote:
Assalam Alaikum
I want to know how can i connect IP phone with asterisk... which config
files, i need to configure... plz tell me stepwise ... i m new to
asterisk n
i just used softphones with asterisk
This is a _very_ green question, but I am just beginning to explore and
learn Linux. Have to admit I avoided it for years due to other
obligations but discovering Asterisk has forced my hand.
So: knowing that the X11 window GUI is a resource hog, is it
appropriate to use the GUI to install
On May 21, 2005, at 10:29 AM, Johnathan Corgan wrote:
Robert Goodyear wrote:
So: knowing that the X11 window GUI is a resource hog, is it
appropriate to use the GUI to install and configure various
components, then set RUNLEVEL to 3 once all is nicely set up and
running cleanly? Would
On May 21, 2005, at 3:46 PM, Johnathan Corgan wrote:
Robert Goodyear wrote:
Noted. To clarify, will dropping back to runlevel 3 still ensure a
smaller set of processes that would be as non-intrusive as if I had
installed Linux with console/command line support only or would there
still
Has anyone out there scripted the rollthrough migration of the Cisco
firmware?
It would be fantastic if there was an app that would generate a set of
templated .CNF and XML files based on the MAC addys entered, then
control and present your .BIN images through TFTP.
It could then also send
On May 20, 2005, at 3:48 AM, Chris Coulthurst wrote:
Well I installed this script on to my system (a few hiccups with php 5
but its not erroring anymore).
Still not getting any callerid info to pass to my polycom 500 screen.
Could it have anything to do with the fact that the number is prepended
On May 20, 2005, at 8:08 AM, Mark Johnson wrote:
Ok, guys... Please be gentle with me. I have what is going to be the
strangest question you will have ever heard, but I have no idea what
to tell this person.
I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My
receptionist
On May 20, 2005, at 8:11 AM, chawki hammoud wrote:
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
You want Quality of Service. Google around, and
then look at
http://www.mixdown.ca/~andrew/dump/rc.tc. It's what
I use and it seems to
work very well.
Could you please tell me where and how to
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one voicemail account
to be
accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one
voicemail account
to be accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote:
The issue would be voicemail. He would want only one
voicemail account
to be accessed via the messages button.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Infinite SIP account(s) could roll over into one VM
On May 15, 2005, at 5:29 PM, Anton Krall wrote:
Guys.
This is a good one... Is anybody doing callerid on the PC? What are you
using besides yac or things like that? And are you using some CRM like
Goldmine with it?
Good huh?
Yes, I touch userrecords in my SugarCRM implementation when an in- or
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Robert Goodyear
|Sent: Domingo, 15 de Mayo de 2005 08:42 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Callerid on PC and more
|
|
|On May 15, 2005, at 5
On May 15, 2005, at 7:19 PM, Chris Mason wrote:
I need to connect up to sixteen phones per building, I can use a cheap
hub,
but POE would be useful. Is there a cheap POE hub available?
Everything I
have seen has been expensive.
Chris Mason
You could wire your own. It's simply pins 4,5,7 and 8.
On May 15, 2005, at 10:44 PM, Waldo Rubinstein wrote:
On May 16, 2005, at 1:32 AM, Robert Goodyear wrote:
A listener on each client that calls an internet UA on significant
events.
I suppose by this you mean some sort of client software installed on
the client PC that listens to events targeted
Does anyone know of a way to simulate the signaling of a PRI trunk for
testing/setup purposes? I realize this may be a rather naive question,
but I was wondering if you could take a TE110, for example, and using a
crossover cable (or not?) and some means of emulating the NI2 signaling
protocol
On May 13, 2005, at 12:39 PM, Chris A. Icide wrote:
You should note that while this works technically, it won't catch
any issues that you may experience when connecting to other PRI
switches.
In other words, two asterisk servers connected back to back with a T1
cross-over cable won't tell you
There is no way, with an asterisk box to emulate a particular vendor's
switch. You can always change your signalling type and present
different signalling protocols (asterisk supports quite a few), but
you will never be able to emulate a switch from a different
manufacturer, and then of
On May 12, 2005, at 9:51 AM, Timothy R. McKee wrote:
Those are SCCP based phones.
move the cursor to option 3, but do not press select. press **#, then
press
select. You should see the padlock icon with an unlocked appearance.
press
32 and see if you have a YES option (alternate TFTP). If
I use tree names, alphabetically, e.g.:
Ash, Birch, Cedar, Dogwood, Elm, Fir et al.
Never had anyone asking me how to pronounce any Sci-Fi arcana that way.
No offense meant to Sci Fi zealots, of course.
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On May 6, 2005, at 11:02 AM, Christopher Jacob wrote:
Hey All,
I had a user ask how to go in and listen to her current outgoing
messages. I
must confess, I can't figure out how to. Any ideas?
~c
Call herself?
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Other notes:
The clever integrator of this application will save themselves some
lookup $ by caching the responses from the database into their own
database, along with a datestamp. Perhaps if an entry is 90 days
old, the system will re-lookup the entry in the Accudata database but
On May 3, 2005, at 6:32 PM, snacktime wrote:
On 5/2/05, Robert Goodyear [EMAIL PROTECTED] wrote:
On May 1, 2005, at 11:39 AM, Gene Naden wrote:
When we call out from our Asterisk system we consistenly lose the
first
roughly 1500 milliseconds of the audio from the destination. This is
easiest
On May 3, 2005, at 6:46 PM, Matt Riddell wrote:
Chris Mason wrote:
Why not?
Because you have not licensed the file for broadcasting across your
telephone network.
How many other people are there here that write music? Would there be
any interest in creating a pool of music for Asterisk?
Would
On May 1, 2005, at 11:39 AM, Gene Naden wrote:
When we call out from our Asterisk system we consistenly lose the
first
roughly 1500 milliseconds of the audio from the destination. This is
easiest
to demonstrate with a recorded announcement. In other words, Hello
for
example is missing.
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before you torch a generous
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