Hey, all. I'd like to forward an incoming call (e.g., to an on-call
rotation number), out to multiple recipients, BUT only hand the call
over to whoever answers _and acknowledges_ (e.g., "Press any key..."),
'cause I don't want it just going to their mailbox. I've thought of a
number of ways
On 2020-02-16 16:48, Dovid Bender wrote:
Hi,
It's been forever since I dealt with POTS lines. We have a client that
needs FXS and FXO support. If memory serves correct we used the TDM400P
with fxs_gs/fxo_gs. What's the equivalent of that card today?
Wow. Lotta replies. I haven't done
On 2018-04-26 09:49, John Tuxies wrote:
Hi. i am looking for a way to have headers for each section of the
Master.csv
eg call duration, hangup cause, destination,...
is there a way to add it and be there permanently, even after log
roratation due to size or date, please?
That's easy: no. ;-)
Hey, all. I've got an office set up with Asterisk, and forwarding's got
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang
up. If the remote phone doesn't connect, it goes to the original
phone's VM. Is this Polycom's fault, or Asterisk's? I've been
Hey, all. Just added an analog card to our dual-T1 system... and
clearly I'm doing something wrong. Less interested in having the
specifics pointed out than in finding out how/why certain things work.
So, really, three things:
* What the bloody Hell is the difference between
Not sure why you'd say it's OT -- seems perfectly topical to me.
Anyway, I have used the SNOM OpenVPN feature for remote clients. I'll
be honest: it's a bit of a pain to set up and get working. This is
triply true if the remote phone is moving around -- to the point that
I'd strongly advise
Hi, all. I'm upgrading my company's old 1.2 box with a
new-and-improved one. But a fair bit's changed in the interim. To
start, at least, I just want the new box to act as a pass-through for
all calls -- PSTN calls go, unmodified, to the internal T1, and
vice-versa. (That way, I can begin
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4
Hi! I'm helping set up a new Asterisk box. However, since I can't
just take the T1 to play with (and I *will* be making many changes,
e.g., going to Adhearsion), in order to test my dialplan, I'll need to
route calls through the old, Asterisk 1.4 box.
I've never really done this.
What's
Hey, all. I'm interested in doing some simple, very specific web pages
for some of my users -- things like call groups, setting forwarding, and
for the receptionist to transfer calls and see calls. Probably do this
in Ruby or PHP, though I'm open-minded. Anyway, if someone could point
me to
Heh. Shortly after I sent my e-mail, I bumped into the Adhearsion you
mentioned, below. Boy, but that looks exactly like what I'm thinking
of! Thanks much...
-Ken
On 2012-11-15 13:08, David M. Lee wrote:
On Nov 15, 2012, at 10:54 AM, Ken D'Ambrosio wrote:
Hey, all. I'm interested
Hey, all. It seems that Polycom has a bunch of offerings for
conference phones, and I'm just wondering which are the less-expensive
alternatives; what with their marketing, etc., it's not always obvious
which is which.
Thanks,
-Ken
P.S. If anyone's had really good experience with another
Apologies for the multiple sends -- I'd been having some outbound SMTP issues,
and thought the first one had fallen into the ether. Turned out, it was the
upstream host that was the issue. Once kicked, lo!
-Ken
On Wed, 25 Jul 2012 14:24:50 -0400 Ken D'Ambrosio k...@jots.org wrote
Hi, all
I know that I could jerry-rig something that would get me caller announce from
my Asterisk box, itself, but what I'd really like is a phone that does it like
my Panasonics. Panasonic has a beautiful DECT/SIP series of handsets... but I
guess they're aimed at the office, and jeepers, nobody wants
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my Sangoma card initializes right... but there's
no dahdi command -- not from
10:38 am, Tzafrir Cohen wrote:
On Sat, Feb 19, 2011 at 04:15:15PM -0500, Ken D'Ambrosio wrote:
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my Sangoma card initializes right... but there's
no dahdi command -- not from the base, nor as a subset
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my Sangoma card initializes right... but there's no
dahdi command -- not from the base, nor as a subset of the core
commands. I've got my channels configured in my chan_dahdi.conf file.
What am I missing,
I've had phones before where, with the phone on-hook, it still implements
the local dialplan. E.g., if I dialed 0 (on-hook), after three seconds,
it would dial the operator, and have the call on speakerphone. Does
Polycom allow this functionality? Clearly, not a necessary feature... but
it
Hey, all. I'm working on making a script to auto-provision my Polycoms.
I wanted one that:
- Gets the MAC by itself
- Fills in the provisioning info you supplied on a web page
- Creates appropriate files
- Reboots the phone (which then gets provisioned)
The last part was the sticking one,
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued.
The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130
more/handset. AND it doesn't look as nice.
Ouch.
Does anyone have any
On Thu, October 7, 2010 1:02 pm, Danny Nicholas wrote:
FWIW, open source is only truly dead when you can't find anywhere to
download the source.
I *totally* agree... if you can find me the source. I have, at this
moment, at least, no reason to believe ADA is OSS -- indeed, looking at
it, I
Hey, all. While ADA can still be downloaded, that's about all that I see.
No development, no recent mention, and -- perhaps worst of all -- it
appears not to work properly under 64-bit systems. So, assuming Digium's
abandoned it, are there any suggestions of alternatives? Right now, I'm
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go straight to that phone's voicemail.
Asterisk 1.4
Want to thank everyone who mailed; a couple of your ideas got me going
down certain paths, and found the answer here:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
Again, thanks!
-Ken
original message -
I'd *really* like to be able to
Hey, all. I'm looking -- if possible -- for a decent, multi-platform
soft-phone. Specifically, Linux and Windows; that way, I'll go through
the same issues my end users do. I've noticed a couple (e.g., minisip,
which seems abandoned, and sip-communicator, which, honestly, is probably
a great IM
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom
phones, and one thing that would make life easier for me would be if I
could have a per-phone sip.conf file. If not, no biggie -- but if there's
a way to do an include (as per extensions.conf) or something, that would
be
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
Thanks!
-Ken
--
This message has been scanned for viruses and
dangerous
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
far as I could tell, was that their firmware just doesn't seem that
stable, and is sometimes hard to get to.
- I managed to corrupt the firmware twice;
Hey, all. I love having a PRI to play with -- lets me do all sorts of
things with DIDs, fax-to-e-mail, etc. But for a small shop, a T1 is
pretty pricey. Is there any reason that a BRI can't do exactly the same
stuff, but on 2B+D instead of 23B+D?
Enquiring minds, etc.
-Ken
--
This message
On Thu, February 18, 2010 3:56 pm, Alex Samad wrote:
On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest
problem, as
Thanks for the review, I
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something? Since
OpenVPN, in one swell foop, deals with both NAT issues and securing
communications, I'd be very interested in hearing if other phone vendors
were embracing
On Thu, February 4, 2010 3:19 pm, Olle E. Johansson wrote:
Anyway - is there someone out there that know the behaviour of OpenVPN in
regards of retransmits and such? A VPN that retransmits will at some
point hurt you if you transmit media over it, especially if you scale it
up.
OpenVPN
Just wondering if there are any Linux-based hard phones out there -- if
so, it'd be neat to see if I couldn't take advantage of the underlying OS.
Thanks,
-Ken
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
--
Hi, all. I've got an Asterisk box installed that I'd really like to
leverage -- and installing a GUI for hunt groups would be awesome. So
long as I can have a trial copy, I could even pay money. It would have to
be able to make use of both SIP and ZAP extensions.
Suggestions?
(Note: I
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get:
Rejected connect attempt from 64.2.142.19, who was trying to reach
'6031234567@'
This leads
Hey, all. I'm seriously thinking about doing the VoIP thing at home. The
perfect platform seemed to be the Sheeva wall wart
(http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp).
It's a cute little doohicky with USB, SD-card, Ethernet, and runs on an
ARM CPU.
Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
but I'll take what-have-you -- that
a) can run on an Ubuntu/Debian box, and
b) allows a receptionist to see what calls are in-process, and forward
calls from their phone to somewhere else.
Thanks!
-Ken
--
This message has
Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.
Thanks!
-Ken
--
This message has been scanned for
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking it
might be time to look for a replacement. I'd love to install Asterisk
with an FXO card or something, but I don't think it supports whatever
Make a call to VM (has to go out on the port you have the handset plugged
into), answer it and listen.
If you hear a bunch of DTMF then you are golden.
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
--
Wow. Thanks for all the replies! Something just occurred to me, though:
which side would be FXO, and which side would be FXS? The PBX? Or the
Asterisk/VM side?
Thanks again for all the info!
-Ken
On Wed, July 1, 2009 3:36 pm, Jared Smith wrote:
On Wed, 2009-07-01 at 13:05 -0400, Ken
Hi, all. My autoattendant looks like this:
exten = s,1,Answer()
exten = s,n,Background(corporate-greeting)
exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten = s,n,WaitExten(30)
When the call
Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go
a bit over the top. The blinking LED is enough for me; how do I disable
the stuttered dialtone and the audible warble? I've looked through the
config files, but there are a HELL of a lot of options, and I haven't been
able
Idle curiosity: I like the look and feel of the Grandstreams, but it's
been my experience that the speakerphones suck (esp. when compared to the
pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000;
have any of their newer models changed that?
Thanks!
-Ken
--
This message has
Hey, all. I was just wondering if there were any
tools/utilities/what-have-you out there that would allow a user to click
on a contact in Outlook, and have their phone dial it? (Or, I guess, have
Asterisk dial both their phone and the destination number, and put the two
into a conference.)
Hey, all. I was going through a make configure on my Asterisk 1.4.23
Ubuntu box, and noticed something I'd forgotten: Asterisk now supports
IMAP_STORAGE. However, when I highlight it, it tells me that there's an
unmet dependency, presumably for imap_tk. I've apt-get installed
everything I can
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
Thanks,
-Ken
--
This message has been scanned for viruses and
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
abilities? Failing that, a WiFi phone that runs Linux? I already know
one phone that does meet my requirements -- the iPhone. The new software
comes with a Cisco VPN client, and a SIP client can be had from
third-party
Hi, all. I'm getting a lot of
[Feb 3 13:56:36] WARNING[3721]
/usr/src/asterisk/spandsp/agx-ast-addons/app_rxfax.c: Channel T30 DONE
0.
in my log file, and incomplete fax reception. Any idea what might be
going on? I've googled a fair bit, but haven't seen anything leap out at
me.
Thanks,
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
exten = 1,1,Set(MyWorkingDir=System(/bin/pwd))
might work, but apparently not. I also looked into AGI stuff, but that
doesn't quite seem to be the right approach.
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the
For no reason other than it would be cool, I'd like to be able to dial an
extension and have it play a random MP3. Since, however, MP3s are
kinda-sorta weird due to patents, I'm not sure what the right approach for
this is. Any pointers on how to go about this?
Thanks!
-Ken
Hi, all. I just tried to fire up app_txfax and app_rxfax, only to find
that I can't seem to compile them. The problem appears to be that my
libtiff library is wrong. Only problem is that, according to the README,
I need libtiff =3.8 and 4.0, which is all well and good... except that
there is no
What version of spandsp do you use?
Based on the fact that you asked that question, I suddenly got suspicious
that, despite his warnings, it might have worked for you with libtiff-4.
So I went and re-tried (using spandsp 0.0.4-pre16), and it failed
*differently*. So then I got suspicious that
Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to
receive faxes was, well, a PITA, what with having to patch the Asterisk
install with various driver patches and this, that, and the other.
Is that still true? Is there a fax HOWTO out there that reflects Asterisk
1.4.x?
Hi, all. This e-mail is a follow-up to an exchange I had several weeks
ago. I've got an Asterisk box with a dual-span T1 card. I want to place
it between the PSTN and my company's legacy PBX. I actually did do that,
but international calls from the legacy PBX were having the 011 stripped
off
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment. My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get shot down by
finance. Any
the provider may be tagging it on. have you checked pridialplan, or
prilocaldialplan settings and playing around with that in zapata.conf ?
Oooh. That makes sense. I've poked around, but don't really see much
documentation on this. 'Cause going outbound is easy, but how do I check
to see if
You have handsets connected to your proprietary PBX. Most domestic
things you dial on your proprietary PBX handsets get passed directly
through to your asterisk box without getting mangled by your
proprietary PBX. International calls that are prefixed by 011 are
getting mangled by your
Hi, all. We've got a PoS legacy PBX at my company that doesn't have call
accounting. I figured, Hey, why not stick a dual-span T1 Asterisk-based
system in the middle? Then, I just passively pass in-bound calls to the
PBX, and outbound calls to the PSTN. I can then have Asterisk do all the
call
Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm
afraid I've forgotten a fair bit. One big thing that I've forgotten is
the syntax, etc., for extensions.conf. Where do I find that? I'm looking
for stuff about commands, syntax for commands, variables, etc. Is there a
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX (which
I hope to eventually replace). That being said, until I do, I'd like to
kill echo cancellation for the passed-through calls -- I don't want to
mess with
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive! One major component of the eye candy was an end-user interface
that allowed
http://www.youtube.com/user/voiceroute
Ming
On 8/7/08, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy
Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system
to stock Asterisk 1.4. Everything's working great, except that all the
prompts (both stock system prompts on the new system and people's old
recorded VM prompts) sound HORRIBLE. Call quality is great, both internal
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old
Hi, all. I'm trying to do some rudimentary testing of an Asterisk system,
but, for various reasons, I have to do this covertly, which means I'm
paying out-of-pocket. So I'm looking for somewhere that will do *cheap*
SIP and/or IAX termination, preferably with at least two simultaneous
calls, and
Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I
don't quite see how that works. Do you point a non-VLAN'd segment at it
(akin to when you uplink a VLAN_enabled switch), and have the phone
implement the VLAN? Or...? *puzzled*
Thanks much,
-Ken
Hi, all. I've got a PoS Emagen VM system tied in with our Telrad PBX. I
hate 'em both, but I'm stuck with the Telrad for the time being. That
being said, does anyone know of a way to replace the VM solution with
Asterisk? I'd -love- to get an Asterisk box in the loop, here.
Thanks,
-Ken
Hi, all. I've used the perl/AGI interface, and... well, I found it kind
of hokey. Granted, this was in 1.2 days -- perhaps things have changed.
Regardless, I guess I have two questions:
1) Has the Perl/AGI binding improved since then?
2) Is there any chance of a real API for Perl?
Thanks much!
Hi, all. I've done some Asterisk recelling, but recently got roped into a
Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much
all circuit-based systems do, it sucks. It sucks to administer, moves
suck... you know the drill. So, I'd love change to an Asterisk system.
My
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway. However, I'd need a few
things:
- Reliability. Can't have my branch office's DID's just going down. A
company with a proven track record would be very, very good.
- English.
On Mon, September 17, 2007 7:21 pm, Matt Riddell wrote:
In the past, you could help someone sort a problem, only for the config
files to be overwritten the next time the user did something in the GUI.
Are there any Asterisk GUIs out there that actually parse the data files,
themselves, instead
I got dragged away from Asterisk (somebody made me an offer I couldn't
refuse for system administration), but I'm thinking about seeing if I
can't get it deployed at my new employer. Regardless, there are two
things about older voicemail that used to annoy me:
- Dial by name. Has anyone made it
Hi, all. I've got a customer who's complaining of low volume, especially
for conference calls. If this were a Zap system, I'd just bump up txgain
in their zaptel.conf file... but it isn't. Should I crank the volume of
the phones (they're Polycoms), or is there some other, more graceful,
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Thanks!
-Ken D'Ambrosio
--
This message has been scanned
Hi, all. Today, our receptionist got an e-mail saying she had a 55-second
voicemail... but the attachment was 0 bytes. Turns out, so was the
message when accessed via the phone. A quick purview of the logs turned
up this:
VERBOSE[14836] logger.c: -- Playing 'vm-savefolder' (language 'en')
Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt. ;-)
Thanks!
-Ken
___
--Bandwidth and Colocation provided by Easynews.com --
I've set up a bunch of plain-jane Asterisk systems, but had heard good
things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox.
So I installed it, and fired it up, and it works fine.
Until I try to do an asterisk -r. I get the does /var/run/asterisk.ctl
exist? question,
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since
I'm not an authorized dealer -- I'm kind of wondering if anyone knows
where I can get it. freedomphones.net/polycom/files/ only goes up to
1.6.7. If anyone can either mail it to me, or mail me a link, I'd
certainly be
I'm trying to set up a dialplan that dials via PSTN for:
All eight-digit calls that start with 9
All 911 calls
All calls that start with 424 (the local exchange)
I haven't tested 911 -- for obvious reasons. I may do so after I feel
more confident. I've got the starts-with-9 working fine. But
I've seen analog-to-VoIP gateways such as the Audiocodes one -- which,
truthfully, looks very, very nice -- but I've got several hundreds of
analog phones to deal with, and I was wondering if anyone has seen
something with even higher concentrations than the Audiocodes
24-ports-per-rack-unit.
Hi, all. I've just set up an Asterisk box -- to the best of my knowledge,
no differently than any of the others that I've set up. Only one minor
caveat: there's no zap command. Huh? Glancing at the startup, there's
no mention of chan_zap, which I assume is partially the reason. However,
I'm
I have to bid on a hotel contract, but there are some things I don't know
how to do -- but clearly Asterisk has been used by hotels before, so I
figure someone on here must have some answers:
1) While the majority of the phones will be SIP, there will be a couple
hundred analogs (due to wiring
with timestamp, it mentioned either a line number, or -- more likely -- a
context/extension/priority triplet.
Is there anything like that?
Thanks,
Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
Hey, all. I've got a client who's interested in possibly using a
softphone for his receptionists. While I've certainly used some
softphones for single extensions, I'm not sure which one I'd suggest for a
receptionist.
Any favorites?
Thanks,
-Ken
screws stuff up).
Any ideas?
Thanks,
-Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, all. Every now and then, some of my users get Error on their
phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm
running Asterisk 1.2.4, and have the following firmware, etc.:
Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041
Any ideas as to why
Hi, all. I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension. Is there any
way to do that? I've tried to RTFM, but I'm coming up empty.
Thanks,
-Ken D'Ambrosio
___
--Bandwidth
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten =
wed
that the FTP transactions were being executed properly, but the phone
wasn't responding correctly. It was only when I went with ProFTPd that
things got better -- for me, at least. ;-) YMMV, etc.
Doug.
-Original Message-
From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, Ma
HTTP's nice, but FTP does the job. Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below. I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.
-Ken
On Tue, March 7, 2006 12:37
I'd like to set up a sort-of follow-me: on a call to a given extension,
I'd like to simultaneously call several different numbers, play them all a
prompt upon answering, and monitor for DTMF digit 1. I know how to get
Dial() to dial multiple numbers, and I know how to play prompts and
monitor for
On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:
I have installed several hundred polycom's, and I have never seen a
500/501
with a power jack. All with the inline cable, as you mention.
Of course, if someone can provide photo evidence I will stand corrected.
I think the confusion
Hi, all. I've just had my T1 re-provisioned to ISDN. Everything comes up
and seems to work fine, with the minor detail that there is no audio
whatsoever.
So: voice prompts are played, caller ID and DID information is seen and
acted on, etc., etc., etc., but at no point is any audio heard on
Hi, all. I've got a T1 through Choice One Communications
(www.choiceonecom.com), a provider in the US northeast. I recently tried
to switch to ISDN on it -- and failed miserably. I've posted my config
files, and nobody's seen anything obviously wrong. Has anyone else used
their ISDN T1's? If
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty
badly. Seemingly everything worked -- Asterisk would see the incoming
call (including CID and DID info), try to route it, and fail -- giving
me a telco (not Asterisk) call failure message. My zapata.conf and
zaptel.conf
Michael Graves wrote:
Does anyone on-list have direct experience with the new analog cards
from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports
but don't want to go through another TDM400 style experience.
First impressions (of which one should always be wary):
1) I really,
]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
D'Ambrosio
Sent: Thursday, February 16, 2006 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No D-channels available!
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed
1 - 100 of 160 matches
Mail list logo
