It is possible, kind of. The only way I was able to get this to work
for me (and only for two email addresses) was to specify one email
address as normal, and the second email address as the pager email
address. Possibly not what you're looking for, but the best solution
(short of email
But as mentioned in the same email, the feature set just isn't the same
and it's a /lot/ more difficult to implement.
I'll be extremely disappointed if it /does/ get removed from Asterisk
without a suitable /easy/ and /equivalent/ function being made readily
available.
Matt Riddell wrote:
Not using IMAP storage here, although that was one of the primary
drivers for upgrading to Asterisk 1.4. Why?
The short answer is that it's too confining. There are too many caveats
that don't fit into our existing IMAP structure and make the entire
project rather iffy. Security is a
If anyone has managed to compile and run Asterisk on a server from this
particular era, I'd /love/ to know about it. :)
What's the performance like? For that matter, what phones were
available at the time?
randulo wrote:
On Thursday 28 February 2008 05:13:06 randulo wrote:
Will your
For your own sanity's sake, steer as far away from Grandstream as
possible. The firmware is appalling and isn't improving a great deal.
They make great steps in one area while another gets worse and worse.
randulo wrote:
On Fri, Feb 29, 2008 at 1:12 PM, Agnello George
[EMAIL PROTECTED]
wrote:
On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote:
For your own sanity's sake, steer as far away from Grandstream as possible.
The firmware is appalling and isn't improving a great deal. They make great
steps in one area while another gets worse and worse.
I
cost models. Although the lesser models are still over $100.
Michael
--Original Message Text---
*From:* Rob Hillis
*Date:* Sat, 01 Mar 2008 11:07:58 +1100
For your own sanity's sake, steer as far away from Grandstream as
possible. The firmware is appalling and isn't improving a great deal
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put
an actual Power Supply on the phone - thinking the voltage is dropping
and causing the reboot. I don't buy that, but we are putting one on
next Monday. We'll see.
That's almost certainly your problem. When you
I'll admit in my case, the main reason the boot time of the Polycom
drives me nuts is that I don't see the phones unless I'm installing them
for the first time or supporting them when something isn't quite right.
It's for this reason that I very much appreciate a phone that either (a)
boots
Interesting. Which type of SIP packet did you send?
The method Paul is referring to below is a page you get GET with certain
parameters to reconfigure the phone via HTTP...
Matt Florell wrote:
I have done integrations with sipsak to do this with Snom phones as well:
http://sipsak.org/
Polycoms need to reboot if you do much more than pick up the handset and
dial a number. A change of config of /any/ scale certainly qualifies here.
If you alter the sip.cfg file on your TFTP/FTP/HTTP server, the Polycoms
/should/ pick up the fact that config file has changed and reboot when
it
this is just an example, in the application messages are
posted on the phone display dynamically by the server depending on
agent status.
MATT---
On 3/5/08, Rob Hillis [EMAIL PROTECTED] wrote:
Interesting. Which type of SIP packet did you send?
The method Paul is referring to below is a page
Unfortunately the Polycom's propensity to reboot at the drop of the hat
is one the things I really dislike about the phone - especially when
coupled with the fact that they take so /long/ to reboot.
I must admit, I'm surprised that they don't handle the config file
changing for them on the server
.
Matt Florell wrote:
That sounds great, could you post an example?
MATT---
On 3/5/08, Rob Hillis [EMAIL PROTECTED] wrote:
Ahh... our method actually set an alias on the phone for the first identity
rather than simply sending a message to the phone. This means that whenever
the phone would
...and that's saying something! ;)
Then again, we just need to remember the old saying... to err is human,
but to really foul things up requires a computer.
Paul Hales wrote:
It was one of those moments in life where I felt a lot less smart than I
usually do...
PaulH
On Fri, 2008-03-07
That's because A is the joining point between B and C. If either B or
C hung up, the remaining party would still be left.
This is a phone function, not an Asterisk one. From Asterisk's
perspective, phone A is simply on two simultaneous calls to B and C - it
has no idea that A is bridging the
To be perfectly honest, the REALTIME function is absolutely hideous when
it comes to reading data from the RealTime database. What on earth the
Asterisk developers were thinking when they replaced the perfectly
usable RealTime (which sets a channel variable for each field in the
database) with
Special dialplans for reception are entirely up to you. The only reason
reception phones have different dialplans to normal extensions is that
often people want the receptionist's phone to behave a little differently.
The Polycom 601 (nor any of the other common IP phonse designed for
Yes it is.
The reason you get more entries in queue_log is that there are several
queue_log events per call - most commonly you get an ENTERQUEUE,
CONNECT and COMPLETECALLER/AGENT for each call.
Vieri wrote:
Hi,
Surely, I must be overlooking something. If I run the
following SQL queries I
Not ideal if you've got people who speak multiple languages using the
phone system.
You may want to review
http://www.voip-info.org/wiki/view/Asterisk+multi-language - I suspect
this is going to do what you want it to.
Alex Balashov wrote:
Replace the vm-* recordings in
Maybe the Alison voice is better, but I've found Cepstral to be a bit
too mechanical. I'm using Ceptral Millie (since a UK accent is more
acceptable in Australia than an American one)
The best TTS engine I've /ever/ run in to was Nuance Realspeak (see
http://www.nuance.com/realspeak/) though
Bill Hackensack wrote:
On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=5014
The response on that issue from Russell is the kind of response that
really ticks me off. No, no, no, we don't
The only method I'm familiar with for an analogue line to signal which
number was called is a very old service that loops the line first and
then dials the number. The only way to capture this would be to handle
the incoming line as a standard extension with a different context.
I've only
) in
new stack
It still does not give me the dialed number. Could you explain how
to match it again the zap channel to extract the dialed number?
Will I be able to get the dialed number if I am using a E1 line?
Thanks,
Mark
On Mon, Mar 24, 2008 at 2:29 PM, Rob Hillis [EMAIL PROTECTED
Distinctive ring is still not going to provide the line that was called
in the ${EXTEN} variable, so you're still stuck with dialplan trickery
to figure out which number was rung.
Mojo with Horan Company, LLC wrote:
Distinctive Ringing might be available from your telecom provider.
mark
Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has
the menuselect configuration, though for most applications you don't
really need to fiddle with it.
James M Kupernik wrote:
There actually is no menuselect, its just a simple
./configure
make
make install
in that order
Most likely, you don't have any hangup detection available or
configured. If these are analogue lines, you will almost certainly need
to configure busy detection in order to figure out that the call has
been terminated.
Do some Googling for asterisk busy detection
mark morreny wrote:
Hi,
We have BLF buttons working fine on the SPA932 side-car. What does
show hints tell you under Asterisk, and what syntax did you use when
configuring the side-car buttons?
John Meksavan wrote:
Asterisk Users,
I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B
wildcard. I
All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
Have you set a call limit for each SIP peer? This is now required as of
version 1.4. It took me a while to figure out all the issues when
migrating to 1.4.
John Meksavan wrote:
Thanks for you guys help. The status LED on the sidecar takes an
awfully look time to change from GREEN to RED
You may be able to achieve the desired result using queues rather than
Dial statements.
Overkill perhaps, but it's the only way I can think to implement it at
the moment.
John Millican wrote:
Tilghman Lesher wrote:
On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
I am hoping
For a receptionist, you generally want to go with a quality phone since
they're going to be the heaviest user of the phone system in the
building. (Inbound/outbound call agents may take/make more calls, but
their requirements are far more simple than the complex call juggling a
receptionist
I'd find that very strange considering that the 57i itself has facility
for at least 20 BLF buttons and /each/ attendant console has facility
for another 60!
Matt Watson wrote:
We are using 57i + 560M combination as well... though we are not using the 57i
ct... but the idea of giving them a
Google is your friend. I discovered very quickly what they were talking
about by googling.
bilal ghayyad wrote:
Dear Steve Doug;
Sorry I did not understand any thing from your reply.
---
Bogen Rulez
On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi;
Anyone
That would at least be long enough to cover the entire boot process. ;)
Lee, John (Sydney) wrote:
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
Thanks Erik. I can probably replace it with my beloved Mozart Symphony
no 40
:
Guys thanks a lot. I should be going with a Polycom 650 for all such
jobs.
If grandstream receives such bad reviews- how are they selling anything?
Phones hanging or voice cut-outs are simply unacceptable!!
On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote:
I'd find that very strange
Alex Balashov wrote:
Anisha Kumar wrote:
Please provide the required Setup / comfiguration details or redirect
to appropriate to resource.
You are decreasing your chances of getting a favourable response with an
imperative tone like that, which also suggests that you are
Configure the extension as a softphone using the format
extension@asterisk.ip.address.
Works fine for me - and works even better for agents!
Simon wrote:
Hi There,
We have some users using x-lite as their SIP phone... but im wondering
how to get the Calls Contacts to show as being
X-Lite. Of course, Asterisk will need a hint configured for that
extension as well...
Simon wrote:
Thanks for the reply.. Sorry for the lame question.. Do i do that in
X-Lite or Asterisk?
On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Configure the extension
their app. But not free/busy type changes.. Any idea why here?
Simon
On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote:
X-Lite. Of course, Asterisk will need a hint configured for that extension
as well...
Simon wrote:
Thanks for the reply.. Sorry for the lame question
Using _. is going to result in warnings. A much better practice is to
use _X.
Ali Jawad wrote:
Thx again patrick it worked, I used
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
while it should have been
[google-in]
exten =
Try TEST=${X-CALLID}; and see how you go.
Eric Dantie wrote:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've already got a variable X-CALLID with the
content ctprueba-123456789.12
How can I copy the content X-CALLID to the new variable TEST?
something like
Every CPU core shows up as a separate CPU under Linux. For those that
have hyperthreaded processors, a single core processor will show up as
two processors - assuming you have hyperthreading enabled.
linuxian iandsd wrote:
top says asterisk 1.2.25 is using multiple cores:
Cpu0 :
To the best of my knowledge, multi-core processors are not hyperthreaded
- certainly my Core 2 Quad processor isn't. I would expect a Core 2 Duo
to be the same.
Steve Totaro wrote:
On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis [EMAIL PROTECTED] wrote:
Every CPU core shows up
Steve Totaro wrote:
On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
That is interesting. I have an intel C2D and I can only see two
procs, not four, is that normal? Are you sure what you are saying is
I believe Intel removed HyperThreading
As is just about always the case, posting twice to the list within three
hours is not only unlikely to get a faster response, I would in fact
imagine it would /reduce/ your chances of getting a response at all.
lotusscript wrote:
A good while back when installing 1.2 there were major issues
No, a dual core processor has two cores. :)
My Quad core shows four processors.
Steve Totaro wrote:
On Sat, Apr 26, 2008 at 2:37 PM, Benny Amorsen [EMAIL PROTECTED] wrote:
Steve Totaro [EMAIL PROTECTED] writes:
My dual proc, dual core AMD boxen show as four procs. I guess the AMD
Lee, John (Sydney) wrote:
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} =
Steve Totaro wrote:
Man that is an ugly hack, but I guess it may be required in some rare
situation, I guess if the power supply of the server is already
nearing it's rating.
I would still go with a sata to molex connector and even a Y molex
splitter if at all possible.
Actually I can see
SIP channels can't be grouped. What you need when you're dialling is
the following if you want to use all three SIP channels:-
Dial(SIP/troncal-1/${number})
Dial(SIP/troncal-2/${number})
Dial(SIP/troncal-3/${number})
That way if the first Dial fails, it will try the second one and so on.
Steve Totaro wrote:
Actually I can see exactly where this may be useful. I can think of at
least one customer off the top of my head who has insisted on having a
TDM2400 installed in a Dell server that we know can't provide enough power
to the card where it's being used as a collection of
Probably for the best - you'd look mighty silly otherwise.
(spoken by someone who's done his own share of jumping around, yellng
YES!)
Steve Totaro wrote:
I never kick myself on issues like this. I enjoy the challenge and
the eventual success by jumping around and yelling YES, YES, YES!
There are krone blocks designed for CAT5, and I've seen them in use as well.
However, there's no way I'd be using them for today's networks.
/Especially/ having seen one of these krone blocks used to double-punch
two network ports together.
Bill Andersen wrote:
Oh, yes. I saw an entire
SPA942s do not currently support BLF keys. The four lit buttons are
line keys only with the current firmware, although our Linksys rep has
assured us that it's a feature to be supported soon.
John Signorello wrote:
We use the linksys 942's and they work flawlessly and are easy to setup
The
At this stage, I've only seen one machine that didn't come with the old
style power connectors. SATA power connectors may be a standard, but
they haven't (yet?) supplanted the older power connectors.
In fact, most power supplies I've bought recently have had more molex
style connectors than
mark morreny wrote:
Hi,
I am looking for a very low cost way of receiving and sending T38 fax
reliably. Is there any possible solution using Asterisk as the PSTN
SIP gateay and Digium E1/T1 card? Is there other open source package
that can help to accomplish this purpose?
Asterisk is
Some. Apparently not complete - and not capable of acting as a gateway.
Olivier wrote:
I thought that 1.6 carried along T.38 origination-termination
capabilities.
Is it true ?
2008/5/22 Rob Hillis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
mark morreny wrote:
Hi,
I am
Remind me to pick on your poor Spanish next time I see you for a
mid-morning meal. :)
Steve Totaro wrote:
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote:
When and where is the 1.6 brunch? ;-)
___
-- Bandwidth and
Steve Totaro wrote:
Darn, it was 87% off just yesterday!
But with all that wonderful value-adding spam, it's worth paying more
for isn't it?
(then again, I guess that very much depends on /what/ you're paying for!)
___
-- Bandwidth and
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions where having Oslec and hardware echo
I believe Ubuntu is in the process of migrating from sysvinit to
Upstart. Upstart is supposed to be capable of monitoring services to
ensure they don't fail, so I suspect this is likely to be the reason
behind the safe_asterisk script not being used.
Paul Belanger wrote:
I noticed
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
OSLEC in this case) will not be used.
That's not always been my
In my experience, the ringback you get over a zap channel (be it
analogue or digital) is generated by the remote end, /not/ Zaptel.
The ringback you get over a SIP or IAX2 channel is often generated by
either Asterisk or the SIP/IAX2 device you're calling from.
James Lamanna wrote:
Hi,
I've
Chris Bagnall wrote:
have used many fsm7326p to power 24 phones or 726tp to power 12
phones and they work great
On the Linksys side, we have a load of SRW-224P switches out in the wild
powering 24 Snom 370s (around 7W each) off each switch.
Likewise, we sell these things by the
for /any/ switched network.
Jerry Jones wrote:
On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote:
On the Linksys side, we have a load of SRW-224P switches out in
the wild powering 24 Snom 370s (around 7W each) off each switch.
Likewise, we sell these things by the bucket load and have
.
James Lamanna wrote:
Hmm ok.
This was a call from a SIP phone registered with Asterisk outbound on
a Zap trunk.
So would Asterisk or the phone be generating the ringback tone in that case?
It also happens very intermittently (maybe 1 in 10 calls at most...)
-- James
Rob Hillis wrote
Steve Totaro wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Answer is the /cause/? Or do you mean it's the solution?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
core show function SPRINTF does work on my 1.4.20 system.
Eric ManxPower Wieling wrote:
Oddly core show function SPRINTF works on my 1.6. SPRINTF function
does not seem to be in 1.2 and I don't have any 1.4 systems.
Venefax wrote:
Believe it or not, I cannot find online a single piece
Doug wrote:
There is a bug in these units that won't let
you put punctuation in the extension name.
A Grandstream product with a bug... what an unusual concept. cough
Seriously, with all the grief I've had with GXP-2000s, BT-200s and
GXV-3000s, I wouldn't touch Grandstream gear with a
like a seperate
line tied to the same DID in a hunt group.
Eric
On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably
looking at the Linksys PAP2 - the functional
and after a week their answer is . use udp.
Rob Hillis wrote:
Doug wrote:
There is a bug in these units that won't let
you put punctuation in the extension name.
A Grandstream product with a bug... what an unusual concept. cough
Seriously, with all the grief I've had
80 rooms? I guess you and I have slightly differing opinions as to what
a small motel is. :)
If you have 80 analogue channels, then you'd need 4 TDM2400P cards.
Unless your server is powered by a small nuclear reactor, you'll be
better off with either 3 E1 or 4 T1 channels banks or 2 32
Can I assume you're using Asterisk 1.4 and that you've configured your
phones as peers?
If this is the case, then you need to set limitonpeers to yes and
call-limit to some value in sip.conf. Once this has been done, you
should find that BLF behaves as you expect.
Vazquez David wrote:
Hi
Vazquez David wrote:
Yes, I'm using 1.4. And I don't really use sip.conf, but have all my
phones on users.conf. Should I put limitonpeers and call-limit on the
general section of sip.conf? or on each entry in users.conf
[general] should be sufficient, so long as having them set as default
for
Yeah, it's a real pain in the proverbial.
Short version is that if you're not planning on using the LAN interface
for phones then you need to enable provisioning on the WAN interface,
allow access to the web interface from the WAN interface and configure
the provisioning URL with your WAN
.
but as Rob hillis said.
It will work via WAN which ive now got. SO I can access the asterisk
appliance via 192.168.1.15
The problem is now…How do I connect the phone.
Ive got the phone (Ethernet) connected from the LAN port on the phone
to a LAN port in the asterisk appliance.
im
RoLaNd RoLaNd wrote:
hi all,
/is there any way of removing this line from showing on the console?
my verbosity level is 3.
and this is the following output on cli 24/7 unless its interrupted by
the msgs tht really counts like connected sip and so on../
[...]
[Jul 4 10:32:38]
voip crazy wrote:
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
Is this correct? 60 channels (assuming an E1 connection, not a T1)
between 900 extensions means only 1 in 15 people can be on the phone at
once - which is a pretty low ratio.
If this is indeed
Klaus Darilion wrote:
Hi!
I want to test Asterisk--Siemens HiCom integration using Q.SIG instead
of ISDN. I did not find any documentation about Asterisk und Q.SIG.
Thus, I wonder is it sufficient to set switchtype from euroisdn to
qsig or are there any other things which I have to take
Jared Smith wrote:
On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote:
I use them before some patch. But this example work :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,7,NoOp(Nopnopnopnopnop)
exten = s,10,NoOp(Matthieu)
and this not :
exten =
Steve Underwood wrote:
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report bugs
for people from
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
if you still want donate t.38 development please contact
Alexander Olekhnovich wrote:
I just think because of the Asterisk design it can not be implemented.
On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
I'm interested if it's possible to configure Asterisk the
following
Matt Watson wrote:
That being said... i;m also quite pleased to see T.38 support being
worked on for Asterisk... its a pretty important area to further
develop IMHO.
I absolutely agree. It's been a notable omission for some time.
Unfortunately getting it written isn't the major part of
Brian J. Murrell wrote:
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18
(Ring Begin)...
And
Brian J. Murrell wrote:
Unless you want to invest in a
better card, you may just have to live with the problem.
Which means what, a multiport and multi-hundreds of dollar card? I'm
just a home user. I don't have hundreds of dollars to spend on a single
piece of phone hardware.
Loic Didelot wrote:
Hello,
I would like to double check what Echo Cancellation my Digium Card uses.
I thought I bought the little more expensive card that integrates
EchoCancellation. How can I check?
Easiest way is to eyeball the card. If your card has a daughter board
on it, then it's
Chris Bagnall wrote:
Greetings list,
Is ARI by Littlejohn Consulting still being maintained? Their website seems
to have disappeared, and the only download links I can find through googling
are back in 2005.
If it has disappeared, what are people using as alternatives these days?
A
Paul Hales wrote:
I've just had half a day of wierd SIP stuff on 1.4.21.1 - most of it
related to realtime.
Sip peers showing up as UNKNOWN, but if you reboot the phone the problem
goes away. For a while...
Interestingly enough, I've had my Grandstream suffering from the same
problem since
Florian Hackenberger wrote:
Hi!
I'm trying to build an HA system using heartbeat for failover.
Everything works fine with SIP, but I cannot connect my IAX phone to
the asterisk server using the managed IP address.
I've had a similar issue with HA, although in my case SIP wouldn't
register
Philipp Kempgen wrote:
Come on. People want simple answers. So:
Can Asterisk duplicate CallManager? [y/n]
*scnr*
I think for questions like this, we should always consider the m
(maybe) option. :)
___
-- Bandwidth and Colocation Provided by
Alex Balashov wrote:
It is known, as a matter of established fact, that it is possible to
disable call waiting on the eyeBeam phone. How to do it is not
something in which I can instruct you, but I gather it's a fairly
straightforward process, especially if you are autoprovisioning via the
Rob Hillis wrote:
Since when does eyeBeam have any kind of autoprovisioning? I've not
seen any reference to it in the manual or on their web site and I /have/
gone looking for it. If I've missed something, I'd be extremely
grateful if you could point it out - this is a feature I've wanted
Alex Balashov wrote:
Although, oddly enough, a lot of them can do VLAN trunking, etc.
Not odd at all as far as I'm concerned - I know a number of places that
segregate LAN traffic from VoIP traffic using multiple VLANs over the
one physical link. VLANs would be the best solution (short of
ballamudi madhulika wrote:
Can I use Asterisk as an announcement server. We want to build
announcement server with ISDN PRI card terminating on our server and
announcement being fed on the incoming calls.
With the right dialplan and scripts or AGIs, I don't see why not.
Shouldn't be a
Patrick wrote:
Andrew Latham wrote:
Read my hacks on the Cisco phones in Oreilly's VoIP Hacks book
Why the sales pitch for a 3 year old book? Can't you just give some
information?
It's not a sales pitch - the book that he refers to is freely
downloadable from the web.
He
J.M. wrote:
I've followed the instructions here
(http://www.voip-info.org/wiki-Asterisk+RealTime) and other places,
however, Asterisk still reads information from the .conf files. How
can I get Asterisk to read from the database and not from the .conf files?
I realize the information
Jerry Geis wrote:
Jerry Geis wrote:
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
Dan Peters wrote:
We have had Asterisk up and running for a while now and it works very
well. Recently we tried to integrate a Linsys SPA962 with the
associated SPA932 console. We can get the BLF lights to blink when a
phone is ringing and we can get the BLF lights to go solid when that
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