François Delawarde wrote:
Stephen Bosch wrote:
François Delawarde wrote:
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems
Olivier:
Olivier wrote:
Our last trial was so conclusive (every call was affected), we step back
to previous situation without HPEC.
We will do our best to help to solve this (gathering audio captures for
instance) though it will be very hard for me to convince our customer to
try.
Hi:
Jerry Geis wrote:
I recently got a polycom 501.
I was trying to get the phone to accept the TFTP boot files.
I was REALLY confused when I finally figured out that
the phone does FTP by default and you have to go change it to TFTP using
the
keyboard menus to switch it to TFTP.
You
Per Jessen wrote:
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off
Alex Balashov wrote:
Zeeshan,
On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect:
MoH volume is uncomfortably high and I want to bring it down. Its
mpg123. How can I do it?
There are some settings in musiconhold.conf that may yield the desired
effect:
[default]
Steven wrote:
The Citel Handset Gateways were the best option for our scenario.
The cost per port for the number of buttons on our NEC DTerm/E phones
was about half.
Also, no network reengineering.
I've noticed that all the people who have good things to say about them
are using East Asian
Hi, Francois:
François Delawarde wrote:
Hello,
I had noticed strange crackling sound on my phone calls going through my
zaptel device (TDM400P), so i decided to check on possible timer issue,
and found lots of issues on forums concerning the sensibility of zaptel
with IRQs, and tried about
François Delawarde wrote:
Thanks Michael,
I've already been through all that unfortunately, and I have a SATA
drive, so no UDMA mode 2 as far as I know. I'm currently trying
everything again anyway, but i doubt it will work if nothing worked the
first time.
Anyone would know of issues
Alex Balashov wrote:
On Mon, 14 May 2007, Stephen Bosch said something to this effect:
Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.
From a cursory glance at the voicemail settings, I can't see a way
Jon Pounder wrote:
Who said I wanted to run DSL over it :)
no one - I'm sure you really just want to run 110baud modem over it :)
and I'm sure you probably don't want a handful of them between the same 2
locations either.
btw - here is an interesting strategy to get fibre or something
Alex Balashov wrote:
On Fri, 11 May 2007, John Treble said something to this effect:
Can you still do “homebrew” PTP T1 in the U.S. this way? I thought
this was nixed by the ILEC/CLECs years ago.
It's logically possible. But if you're trying to do T1 over a single
pair, you'd have to
Andrew Kohlsmith wrote:
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote:
again, I'm interested to know anyone whose actually done this, and what
the results were, since I have been thinking of the same thing for a
while.
I'd run about two dozen of these things using a variety of equipment.
Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but
Brian Capouch wrote:
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted
[EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted
[EMAIL PROTECTED] wrote:
In todays socio/political climate, telco infrastructure is seen as
foundational, and an essential service that is vital in times of
emergency. Any unauthorised modification can present an unacceptable risk
exposure to the telco, the emergency services, and to the
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a big place, and I suppose there's room for all kinds. In
these
Francesco Peeters (Asterisk) wrote:
On Fri, May 11, 2007 07:34, Armin Schindler wrote:
On Thu, 10 May 2007, Crazy Boy wrote:
Hi Friends,
Can anybody tell me other softPBX softwares like Asterisk?
- OpenPBX
- Freeswitch
Or try Googling for something like 'open source pbx'... Sheesh! :-o
Gavin Spurgeon wrote:
Ps. Please start new messages from scratch rather then reply to
existing ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
Actually, what he's referring to is that posters should start a NEW
thread for a new
Hi, Vitaly:
Vitaly Oborsky wrote:
Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes the message of a type to
Drew Gibson wrote:
Stephen Bosch wrote:
Gavin Spurgeon wrote:
Ps. Please start new messages from scratch rather then reply to
existing ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
Actually, what he's referring
Tom Lynn wrote:
Bilal,
I don't think anyone is telling you that digital phones don't need cards.
I do think they are telling you that NOBODY makes a card that drives
digital phones for use with Asterisk.
bilal ghayyad wrote:
Hi List;
As I know from AVAYA (I am AVAYA certified) that
Hi, folks:
I just took delivery of new PoE cables (PN 2200-11077-002) for our
Polycom phones here in the office.
I have the following to report, for the erudition of existing Polycom
users and those considering purchasing Polycom sets from the set of
models noted in the subject:
1. The cables
Hi:
I have a user saying that the volume of voice mails is too low.
Is there a way to tweak the recording level for voice mail?
-Stephen-
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Robert Augustyn wrote:
Can you connect existing Nortel system to Asterisk through fxs/fxo?
That way one could use existing infrastructure for few old phones and
Asterisk for new phones and all good things which come with it?
No. They are digital phones and use proprietary Nortel signalling.
Rizwan Hisham wrote:
Hi all,
There is a problem with my dialplan. here is the dialplan:
exten= 123,1,Dial(SIP/U1,,Ttg)
exten= 123,2,Hangup
exten= h,1,AGI(onhangup.pl)
The problem is whenever U1 is called or calls someone, if U1 hangsup
the call then the h extension is NOT executed.
Alvin Austin wrote:
Hi everyone, I have a user that needs a little extra volume on his
Polycom IP 601 phone set for all calls (beyond what the volume control
currently offers). Is there a provisioning setting for this anywhere?
(I'd like to avoid a separate amplifier between the phone and
Robert Augustyn wrote:
Stephen,
I understand that these sets are digital but what about connecting Asterisk
fxs to Nortel fxo and keep sets connected to existing Nortel?
If you leave the Nortel PBX in the picture, I see no reason why that
wouldn't work.
-Stephen-
Andrew Kohlsmith wrote:
On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote:
I understand that these sets are digital but what about connecting
Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel?
Yes you can do that; I have. No you don't want to; it doesn't work
SIP wrote:
Joshua Colp wrote:
Handling of OPTIONS in Asterisk has changed a little bit through
chan_sip versions... but for the most part the other side usually just
wants you to respond with something/anything. Is the other side
unhappy with the 404 Not Found?
Joshua Colp
Software
Hi:
It's been a few weeks since the great voip-info.org crash.
Around that time there was some lofty talk about a set of mirrors being
set up for it.
Has anything happened with that, or are we just going back to business
as usual?
-Stephen-
___
Tom Rymes wrote:
On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
Wow,
The guy makes a useful application and provides
Jay Austad wrote:
I configured my sla.conf to use with a Polycom phone. I have no idea if
I did it right, however, none of the console sla commands exist. Do I
have to something special to compile in this support, or should it just
work out of the box?
It sounds like you're missing a module
Gordon Henderson wrote:
On Thu, 3 May 2007, Per Jessen wrote:
(aren't you guys getting rid of ISDN anyway? :-)
H... Some people would like to think so, but it's going to be here
for a long time yet! BT have/are dumping the consumer versions of
ISDN2 - home highway which went a while
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
-Stephen-
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William Moore wrote:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
Uh... This is Daniel Bernstein's 'daemontools' -- and he's not going to
rename it, especially since his
Vicente Aguilar wrote:
El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
daemontools is not the name of my scripts,
Jason Fuermann wrote:
I've used these gateways and never experienced any of these problems. I
could imagine me missing the popping noise but I do know that MWI did
work just fine.
What he said was that he couldn't turn stutter dialtone off, not that
the MWI didn't work.
Not hearing the DTMF
Whoa. Calm down.
Jim Suber wrote:
PBX:
Asterisk 1.4
Phones:
PSTN phone connected to TDM400
X-Ten Lite
Polycom 430
Scenario
Polycom 430 = User1
User2 calls User1(Polycom 430) asks to be transfered to User3
User1 does an attended transfer using the
Steve Totaro wrote:
Citel makes SIP to Digital gateways. I have had poor experience with
them and doubt I would try it again without seeing many improvements
listed in their firmware releases.
Just to clarify, I had loud bursts of static when first picking up or
originating a call, phones
Ken D'Ambrosio wrote:
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
Per Jessen wrote:
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a
little dim if they believe they can openly go about borrowing
email-addresses like this.
Ha -- I was just about to post something myself!
Yes - I got this too, and immediately suspected a cull of
J. Oquendo wrote:
So I have whose autoattendant is colliding with their extensions...
Quick fix anyone?
Second someone presses say a person's extension (101) ... Autoattendant
sends them to the first context...
Two things:
1 -- your include statement is missing. Asterisk doesn't even know
CSB wrote:
I want to capture all my Asterisk traffic (including RTP) and then
analyse it.
My plan was to use tcpdump and then analyse with Wireshark. The
following works:
tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1
But I want to be a bit more selective:
tcpdump -C 100 -W 10 -w /tmp/tcpdump
Eric ManxPower Wieling wrote:
Steve Finkelstein wrote:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile
Dean Collins wrote:
Dude yes we know Estara is so cheap you said this 2 months ago, you said
this last month and you are saying this today.
Yet every customer that comes to us to buy a license says their quote
was around $50,000 for the first year for Estara click to talk.
I’m prepared to
Salvatore Giudice wrote:
Nortel digital Meridian phones are like $400/each. At least that was the
price of the phones at a hotel I did a job for recently.
Still?
(Is Nortel even making these phones anymore? I thought they spun off
their telephone set division -- anybody heard of Aastra? ;) )
Noah Miller wrote:
At the time I set this up, MySQL replication was really designed for
one-way replication. Two way replication was possible, but required
somewhat unorthodox methods. (Maybe this has changed, I don't know).
Configuration is also a little tricky. It's not too bad to set it
Hi:
I can try and answer some of your questions.
[EMAIL PROTECTED] wrote:
Hello All,
We have been doing Asterisk and CME implementations recently but we
almost always exlusively bring in analog lines and or PRI for PSTN
access to our systems. I have known about providers providing SIP
Justin Hamade wrote:
The 501 is more weird then that. The cat5 cable with the built in
power injector is cool but to use it with a PoE (802.3af) switch you
need a special cable (the pairs are just different you can probably
look it up and make your own).
Is this true? I read earlier on the
Klaverstyn, David C wrote:
All,
I have a Polycom 650 phone, when turned on displays “Checking application”.
Can any give me some information as to what is wrong? I have copied the
CFG files from a 601 phone to work with this 650.
1. You need at least SIP 2.0.1 (2.1.0 recommended minimum,
Paul wrote:
Third - I have enough exposure to Visual FoxPro to quickly rule it out
as a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
Anthony Rodgers wrote:
That's the way we want to go, but have been unable to divine the correct
settings for getting it working with MS Exchange.
Just for laughs...
what sort of problem do you have?
(Stinky, stinky MS Exchange... worst IMAP support -- but hell, maybe we
can find a solution)
Jeff Davis wrote:
The IP 501 supports both Cisco and 802.11af with different cables. While
there are pin assignments differences, there are also electrical
differences in the discovery protocols. The special cable is an artifact
of this.
I don't know of anyone who was able to make the phone
JR Richardson wrote:
Your experience with database replication is not unique. I have seen
this happen with many flavours of database, not just MySQL. At the
critical sites where I've worked, database replication is not even on
the table as an option for precisely the reasons you state above: I
Jim Freeze wrote:
In inspecting the MAC-boot.log files, the phones that fail have CDP
enabled,
while the phones that succeed have CDP disabled. I think this is
Continual Data Protection,
but don't see where to disable it on the phone interface. Is this a
cause of the failure?
CDP is Cisco
SIP wrote:
It can and it has, unfortunately.
PRS fraud is becoming more and more popular.
What does PRS stand for?
-Stephen-
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Diego Iastrubni wrote:
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install, vs 10-20 minutes compilation.
Then, how do you
uninstall? How do
John C. Wolosuk Jr. wrote:
Also I'm somewhat annoyed that I have to compile zaptel drivers that I
don't use in order to compile the app_meetme.so module so I can have the
SLA functions available to the dialplan...
If you're using SLA, you're using zaptel drivers, yes -- without the
timing
Hi again:
Michael Graves wrote:
On Mon, 23 Apr 2007 14:05:55 +0100, Chris Bagnall wrote:
Greetings list,
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home,
Eric ManxPower Wieling wrote:
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home, and
are given a SIP phone to take with them and hook up to their
broadband. For
Tzafrir Cohen wrote:
On Mon, Apr 23, 2007 at 06:36:25PM -0600, Stephen Bosch wrote:
He is better off installing from sources, and more likely to get
something that performs as it should.
Source installs are not complicated -- even when you are using zaptel.
But why do all the extra work
Ed Nuñez wrote:
Hello all
I would like to know if anyone here has had any experience trying to set
up SIP or IAX over VPN. I am testing with Cisco VPN client and when I
call the Asterisk server in my office I get one way audio.
Lest anyone think I am harping, I'll just quote Tzafrir on
John Novack wrote:
The list police are out in force today!
Yes, and with good reason. If we don't respond to this kind of crap with
strong negative reinforcement, it only gets worse. I do not want to see
the list fill with spam, thanks.
More archive space is used up in these kinds of
Hi, Tzafrir:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
attached file.
I tried this link, but it's
Hermann Wecke wrote:
Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized
showroom in Hyderabad (Andhrapradesh, India)?
Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
How was that a troll? Lazy, perhaps --
Hi:
Chris Bagnall wrote:
One of our clients has a number of employees that work from home, and
are given a SIP phone to take with them and hook up to their
broadband. For the most part, this works fine, but there are an
increasing number where sound quality is poor (chops in and out,
Diego Iastrubni wrote:
you need to use apt-get install asterisk.
If you MUST HAVE 1.217 or your cats die, there are repositories available.
For
example, read this: http://www.buildserver.net/
If you still MUST build asterisk yourself, I wish you good luck.
This kind of commentary isn't
Carlos Chavez wrote:
On Mon, 2007-04-23 at 18:18 +1000, Daniel Pittman wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a
Hi, folks:
Yesterday I added a second TDM400P card to a working, echo-free server
running HPEC.
Today, I'm getting these messages:
Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable
echo cancellation on channel 3
along with complaints of severe echo. The channels
Stephen Bosch wrote:
Hi, folks:
Yesterday I added a second TDM400P card to a working, echo-free server
running HPEC.
Today, I'm getting these messages:
Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to
enable echo cancellation on channel 3
along with complaints
Andrew Latham wrote:
I make a habit of just buying hamburgers and stopping by the CO or hut
where I see the vans. I tell them that I am buying favors with food
and they like it.. Its a lot of work but it helps...
This is by far the most effective way of getting something done with a
telco.
Hi, everybody:
Stephen Bosch wrote:
Kevin P. Fleming wrote:
Eric ManxPower Wieling wrote:
Any updates on this?
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
Under what circumstances would this clipping be present
Per Jessen wrote:
Remco Post wrote:
Hans Witvliet wrote:
The only obstacles currently, are the ISP's.
Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well
as an ipv4 address.
Not around here (Zurich, Switzerland) they won't. I think there is one
single provider with
Tim Panton wrote:
Ah, back in the old days our government privatized the state monopoly
(BT) intact (attitudes and all).
As one of the conditions they had to deliver within 6 weeks of order.
So I ordered a data line to my house (ok a bit obscure in those days, but
I needed it). 6 weeks
Arturo Ochoa wrote:
Hi List...
I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
and it also has the echo canceller...
I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
2.6.9-34.0.2.EL
I'm using Polycom's 501 with the SIP 1.6.2.0041
The problem is when
Kevin P. Fleming wrote:
Eric ManxPower Wieling wrote:
Any updates on this?
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
Under what circumstances would this clipping be present? Is this patch
going to be recommended for
the deadline? What's the article size? Who is the target
audience? Where is it going to be printed?
Cheers,
Stephen Bosch
Vodacomm Voice Data Corporation
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Brian McEntire wrote:
A follow-up with the solution in case anyone else is looking for this
answer:
I created two contexts in my zapata.conf file, since each VOIP line is
terminated by a VOIP adapter and then just comes in hardwired to the
TDM400 via RJ11 line, I know which VOIP number is
Steve Totaro wrote:
You can use the web interface and set it to -5 gmt. Google for free NTP
servers. I used to use time.nist.gov and got mixed results. I found
another one that works almost all of the time.
If you use pool.ntp.org (or a regional variant thereof, such as
ca.pool.ntp.org) it
Hi, Ed:
Ed W wrote:
Agreed. My experience is that quality is higher on Voip than it is via
a TDM400p. However, my experience hasn't been that VoiP is as reliable
as copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for
Chris Bagnall wrote:
However, my experience hasn't been that VoiP is as reliable as
copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for the main
carrier? Does this match with the experience from others?
Until
Kenneth Padgett wrote:
I have learned the hard way that using old configs with new firmware is
asking for trouble. It is much better to keep your custom configurations
in a MAC specific overrides file and replace the sip.cfg and phone1.cfg
files completely.
This doesn't guarantee that you
Steve Totaro wrote:
Stephen Bosch wrote:
Steve Totaro wrote:
You could try to get it working but it may never be 100%. If your needs
are 100% then I suggest using a standard fax and get an analog line and
do it the old fashioned way. If you need Hylafax type features then buy
a modem
Suity Zsolt wrote:
Hi everyone,
I'm in trouble with queue.
There are a little local radio station with one studio and we have to
switch queued callers to the live program. Everything works fine
(counting callers, periodic announcements), but while the announcement
is played for 'firs in
Steve Edwards wrote:
On Mon, 16 Apr 2007, David Cook wrote:
Remember, you don't need to activate all 23 lines so if you just need 15
then you can activate only that number. You also can have potentially
hundreds of numbers that terminate on this group of lines. This makes
some of your
Tzafrir Cohen wrote:
On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote:
Has anyone figured out the way of getting the caller id for BSNL on Asterisk
1.4.2
I have tried following link
http://bugs.digium.com/view.php?id=6683nbn=24
but was not able to get it, although did not ge
Mike wrote:
Here is what I had to change on the phone1.cfg file:
Which means we caught you red-handed! Remember when I asked you about
whether you were using the default configs from the new firmware package?
I had this value in my 1.6.7 file, put in there following suggestions
from the Wiki
Hi:
Salvatore Giudice wrote:
Product selection is not cut and dry. What are your business requirements?
So you need encryption? If so, what kind?
No.
Do they need support for outbound proxies?
No.
Are you going to use the same model for remote deployments?
Yes.
Do you need WAP
Salvatore Giudice wrote:
Roflol. The chance of that happening are slim to none.
And Slim just left town.
-Stephen-
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MAS! wrote:
My colleague left our company, then I have to manage all our phones
lines and asterisk: please, apologize me because I'm 'absolute beginner'
about voip/asterisk!!
Well... all seems work fine; we have some queues and some agents; the
music on hold works fine when the agent
Steve Totaro wrote:
You could try to get it working but it may never be 100%. If your needs
are 100% then I suggest using a standard fax and get an analog line and
do it the old fashioned way. If you need Hylafax type features then buy
a modem that is compatible with Hylafax and run it on a
Bob Smither wrote:
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it
Lee Jenkins wrote:
Stephen Bosch wrote:
Sidetone can be set in the phone configuration; before you do that,
though, I need to know what you mean by feedback.
Sorry, should have been more detailed. It's a sort of background
humming noise, almost like that if you placed the phone next
Drew Gibson wrote:
We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.
I only recommend the Cisco phones to people I don't like, overpriced and
far too much work.
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used
Salvatore Giudice wrote:
BTW, the main problem with these patents is that they tend to lower the rate
of adoption for new standards. Nothing kills a standard quicker than when
someone patents it.
For example, someone out there even has a patent on ENUM:
Anthony Rodgers wrote:
Hi there,
We're trying to get IMAP voicemail storage working on an MS Exchange
server - I would be grateful if anyone who has successfully done this
could post the magic soup here, as extensive Google searching has
yielded nothing other than tantalizing references to
Greg Siemon wrote:
Thanks for the helps Stephen. I was running non standard gains but setting
regain and txgain to zero (then reloading chan_zap.so) does not help. I
still get the broken audio, in fact sometimes I don't get any audio at all.
In testing the server just froze a number of times
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm trying to decide which phones to experiment with. I have these options:
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like
Stephen Bosch wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm trying to decide which phones to experiment with. I have these options:
- A combination of Polycom, Aastra and Snom
- Just
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