Is Without Zaptel interface Installed, conference Bridge is worked
or not.
Why it need, For SIP conferences through OpenSER Zaptel interfaces provide timing that is necessary for meetme conferences. When you start a conference, on the cli you can see that asterisk opens a ZAP/pseudo channel.
What soundfile format, is the one that uses least transcoding
during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
When you get down to it, the asterisk native format is slinear.
Hi Carl -
Setup:
Asterisk server in NY.
Cisco 7960 IP Phones in NY and London.
Dedicated T1 from NY to Ldn.
T1:
Latency - 100ms
Qos applied
No errors
Default codec on Ldn IP Phones = g711alaw
Default codec on NY IP Phones = g711ulaw
Both codecs allowed on each phone.
I'm guessing you're
No need for a religious argument!
But my OS has been ordained by GOD! j/k
I have, on occasion, had to reboot Linux CentOS 3.x with Asterisk
1.2.10, but certainly not daily, once a week or longer. Only when
something is obviously insane.
I run Tao Linux on a number of the asterisk boxes I
Hi -
We are looking at migrating our office from a Samsung PBX to an Asterisk
PBX. I am looking at ordering a PRI with 12 Channels for now (we currently
have 8 analog lines) and need to know what PRI card you guys would recommend
that we use. I have seen some with Echo Cancellation and so on,
Hi Ady -
Imagine i want to create application like SMS Alert, however it's a call alert
when something happened, for example server is crashed, i want
to call 100 of my staff (administrator, manager, and others) using
asterix, when they pick up
their phone, my asterix will play an audio file
I am getting ready to image a production system. Right now I am
planning on using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3. I
will be using a Sangoma A200D card.
I read of some people having problems with Asterisk 1.2.12.1
crashing. Is this across the board or is there anyone out there
Hi Mark -
PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a
SIP ATA. When an incoming call comes in, I would like to ring both
phones, but if phoneA is answered first, I would like phoneB to be
answered as well and left in a off hook state so that when someone
So I'm sure many of you are using or have tried to use TrixBox. Thus far, I'm in
love with it. I haven't had a single snag. Then again, I don't need to get into
anything overly nitty gritty with my Asterisk box.
What are your views?
I've played with AAH/trixbox, but I'd have to say, in all
Hi Mike -
I've been noticing that my group of Polycom IP 501 phones seems to
randomly reset themselves nearly every night (I guess it usually
happens at night, since I've never seen it happen while I've been at
work during the day)..
When I say reset, I mean, the hands free volume and ring
Hi Dean -
I actually just use the SIP notify command on the Asterisk console to
remotely reboot my Polycom phones. It requires a pre-configured
sip_notify.conf file and the Polycom option to reboot on config
check. You can then call it from a script using:
To concur with Avi, I used the
Hi Avi -
username and password is PlcmSpIp. vsftpd cannot handle capitalized
usernames, so if you want to use vsftpd, you have to manually
re-configure the username on each phone.
I use vsftpd and I'm using the default PlcmSpIp username just
fine. :) Essentially, I configured PlcmSpIp as a
Hi Dean -
I'm using vsftpd quite successfully on several Asterisk boxes with
Polycom IP501 phones.
Just an addition: one requirement I had in deploying Polycom phones -
I wanted the user (and me) to be able to plug in a new phone and go
with no configuration needed on the Polycom end. The
Hi Ram -
so i want to forward some of the DID from my asterisks to other Server
how can i do that, and i need to give them access to calling out also
You need to connect your asterisk machine together. The most common
ways to do this are either with IAX or SIP. To do this with IAX, you
Hi Ed -
5. Digium TDM22 (TDM400P)
6. Analog phone plugged in port 3
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Zaptel.conf :
loadzone=us
Hi Forrest -
I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that
Any more suggestions,
Call Digium. They will get you to the point where the hardware will
work. If it won't work (and there's nothing wrong with your system),
they should exchange for a unit that will work.
___
--Bandwidth and Colocation provided by
You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..
This is the most important thing here - what does your zapata.conf look
like?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and
Hi Bilal -
We need to apply Video conference, can asterisk
support this?
No. Asterisk supports video calls between two end points, but not
video conferences with three or more participants.
There is a bounty for someone to add this feature, but nobody has
successfully implemented it yet.
Well I am using GSM as my main codec which seems to be very nice…
Polycom phones do not support GSM (GSM would not be necessary here
anyway, since all these phones are on a local LAN, so bandwidth does
not need to be conserved).
You can also change some settings in the zapta and zaptel
Hi Dean -
Just for clarification this is no longer a bounty for video conferencing.
I ended up purchasing an off the shelf system.
Oh, woops! Thanks for the clarification.
I might however restart it with a lower commitment for the benefit of the
community if someone showed an interest.
I haven't but how about adjusting the offset by 11 secs to compensate?
Lame I know but then you can go back to counting bricks on the side walk
again! j/k :)
From what I know, the Polycom phones use SNTP, and the margin of error
on SNTP is +-20 seconds. It is a little weird, though, that
Hi Paul -
It'd be great if I didn't have to enter the
digits and press the Park button again.
If you're interested in easier parking you might want to check out the patch at:
http://bugs.digium.com/bug_view_page.php?bug_id=7090
You can do one-touch parking with it.
When my users had to
. I actually don't use that patch. I wrote
a different one that's designed for the 1.2.x tarball releases.
- Noah
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
Hi Zeeshan -
Is there any better and receptionist friendly IP phone, with just one
button parking option, and maybe somebetter option for paging as well.
You might play with the ParkAndAnnounce() application which parks a call
and then plays the resultant parking slot number to a channel of
Hi Marek -
what is mean by partially incompatible in
http://www.digium.com/en/docs/misc/compatibility_notes.php
i have server with E7221+te110p mobo and i think i dont have any problems
You might want to ask Digium directly, but this generally means that
there's something on the
Hi Jordan -
On Wed, 2006-10-04 at 06:45 -0500, Jordan Novak wrote:
How is the best way to add,clear mailboxes and change passwords for
voicemail. I am guessing you need to remove the conf entries for the
mailbox restart asterisk and then add them back in and restart
asterisk. Is there a
with the information in the 7090 patch file.
- Noah
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Call Parking
Hi Kevin
Hi -
I believe that a lot of the Sound Point Success story here. Can some one
kindly let me know how to set up the Polycom Phones Sound Point or any
reference to refer from...
Your kindly help are appreciated.
You can find just about everything you need here:
Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP
application software?
My experiences with 2.01 have been mostly good. Definitely haven't
seen any rebooting or error messages. I am, however, having problems
getting the speeddial-to-key remapping to work. I waited for this
So the press announcement said that the new Digium GUI will be available in
v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some
other branch of development that the general public cannot access?
Do you mean this?
http://svn.digium.com/view/asterisk/trunk/static-http/
Hi Ron -
Is there a way to program one of the buttons on the 501 (Like the services
button) to do on the fly call recording? So in the middle of the phonecall
you can record the call without have to do a transfer type of setup. Ive
looked at the manual but cant seem how to do that, I only see
Hi Kevin -
Has anyone used the Polycom expansion module with multiple lines?
My application is for 20 lines and read there was a limit of 7 at one point.
I heard rumors that the newest version of the polycom sip firmware
(2.01) would lift the limit of 7. It just came out, and I haven't had
Hi Again Ron -
Yeah i was messing around last night and saw that! Now if I can only get the
other caller to not hear the DTMF digits id be set! I didnt know you could
remap the keys to DTMF digits, but since I can do that this will work
perfect for the most part!
Let me qualify by saying that
Hi Shawn -
Unfortunately, on a Polycom, you can no longer remap a speed dial to a
key. You can set extra line appearances to be speed dials (I can show
you that, if you want), but none of the other keys. This feature used
to be available, but was quietly removed as of 1.5.x. If you want to
I note that the SC420 is listed as incompatible but the SC430
appears to be a slightly different beast in terms of chipset, the 430
has the newer E7230 as opposed to the E7221 - does this make a
difference to compatibility?
I have an SC420 in one office that works quite well. I think the
Hi Doug -
AFAIK, you will need to tell it save a speed dial for *8,
and then map
the key to dial the speed dial number that you saved it as.
The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it
yesterday on 2.0.1
I just noticed that version 2.01 came out. I'm really glad
Hi Doug -
Let me start by saying when I first plugged it in, I didn't have the
files set up on my ftp server yet, and the phone used it's default
settings and it completed bootup. Now...
I started with sip v1.6.6b and bootrom 3.1.3 on the ftp server. Phone
boots, d/l's files, reaches Welcome
Hi Nathan -
The problem occurs during transfer and hold retrieval, answering the
call is fine, the call is put on hold then either a transfer is
attempted or the call is retrieved from hold. When this is attempted
the remote party (i.e. the caller in the case of a hold retrieval)
cannot hear
Hi Vincent -
Sorry for the long delay in responding. I didn't see you message
until now due to the postfix problems on the mailing list. Anyway, I
see some clues here:
exten = s,1,Answer
exten = s,2,Waitexten(10)
exten = 100,Dial(Zap/2/014XX)
Then call in and after you're connected,
Hi Jonathan -
I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead of over the voiceT at each location
Hi Brian -
I'm just trying to get my Polycom 601 to have multiple extensions on it. For
example, on
line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want
extension
23. Ideally I'd actually have each extension appear on 2 lines and therefore
filling up all
6. I
Hi Vincent -
I'm a little lost on how to set things up with the two FXO cards I have: I
want card #2 to dial a number when a call comes in on card #1. Using the
following configuration, card #1 picks up the line and remains silent,
instead of dialing out through card #2. Anybody knows what's
Hi Warren -
Questions:
(1) Good 2xT1 card with hard echo cancellation?
I'm not sure if it has onboard echo can, but Sangoma has a two port
model. I've never used them (I have a TE410P), but I've always read
very positive things about them, especially on the quality of the echo
can they use.
Why? It seems clear to me that if the person dialed through Zap/2 does not
answer, that he does not want to answer the ringing line...
I think there may be a gotcha here for analogue zaptel interfaces, wherein the
ring will keep getting detected an keep (re)trying to dial, but having to
answer
Hi Erick -
Now This thread tells me that my dual core pentium d (a 700$ computer)
will do the work. (the other equipment costs about 3500.00$). I do
realize that i must minimize transcoding (ulaw all the way) but you're
telling me it will work for 24 users (let's say 30 for round numbers)
all
Hi Erick -
So at the end, i cannot provide prices (without being overkill) to a
potential customer without spending money on a system prior to quote
and test it.
You can always do what you just did and ask on the list.
I think the real reason there is no table or chart for hardware
specs is
I ran into the strange problem that a totally non-asterisk log went
over the 2GB limit (yes, still using 32-bit OS) so the system sent a
SIGXFSZ signal. Even though it wasn't an asterisk log, asterisk
responded to the signal and went into the log rotation loop anyway.
I killed logging on the
Hi Domenico -
Check out the sound files page on the WIKI:
http://www.voip-info.org/wiki/view/Asterisk+sound+files
Here's a good tutorial, too:
http://www.linux.org/lessons/short/sox/index.html
- Noah
On 5/31/06, Mimmus [EMAIL PROTECTED] wrote:
Hi,
how can I convert .wav files to .WAV:
#
Hi Attilla -
So, this left me only one conclusion. The application with the memory
leak is Asterisk.
I know every situation is different, but I just thought that I'd point
out that I have machines running 1.2.7.1 that I haven't restarted in
months. Of course, 1.2.7.1 hasn't been out that
Hi Ken -
Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041
In any case, I'd suggest updating to a later firmware version. SIP
firmware 1.6.6 has been officially released. If you are unable to get
it, just send me a personal email (offlist).
- Noah
On 5/25/06,
Hi Michael -
Can someone please advise me about configuring my Polycom IP600? I have an
account with a SIP based IP Centrex provider. The basic SIP info and line 1
config
points to them. That's working fine.
I'd like to register line 2 with my own asterisk server. I've tried putting the
Hi Doug -
Office A routinely looses connection to Office B. When typing IAX2
Show Peers, it will show as Unreachable. I issue IAX2 Reload and it
will work again for 1-3 days (haven't narrowed the time down yet). My
theory is that the DSL at Office2 is changing IP addresses regularly
and this is
Hi Ken -
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.
There's a couple of ways I can think of that would get this done.
1) For
Hi Domenico -
First, you can remove the quotes aorund your variable
reference. I've seen examples with it, but you don't need
it.
I'm not sure: if variable is empty, you got an error.
In addition, double quotes around text that may contain spaces
will force the surrounded text to be
. Forward will send
subsequent calls to your specified destination, not existing calls.
On May 19, 2006, at 8:53 AM, Noah Miller wrote:
Hi Ken -
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
Hi Steven -
It's not in the general release yet and it doesn't do everything you
want, but check out the metermaid patch
(http://bugs.digium.com/view.php?id=5779).
If you're interested, I also wrote a patch to do one button parking
(http://bugs.digium.com/view.php?id=6340). You can use it now,
Hi Tom -
I have had nothing but problems receiving faxes over PRIs with spandsp. I
currently have 4 systems, 4 PRIs from 4 different providers... none of them
get better than 50% success rates receiving faxes in spandsp, I constantly
get cut off pages. No body seems to have a fix for it, and
Hi Damon -
The following dialplan should result in the voicemail message being
delivered two both mailboxes ([EMAIL PROTECTED] and
[EMAIL PROTECTED]);
The actual result is an error copying the message to the second mailbox as
follows;
I do the same thing (with version 1.2.7.1), and it's
Hi Domenico -
I'm trying to match a few of numbers in a GotoIf; numbers are not starting
with but contain some strings:
GotoIf($[${CALLERIDNUM} =~ 984836|984899|498993|644110]?8:11)
Expression result is always '0'.
First, you can remove the quotes aorund your variable reference. I've
seen
Hi Jordan -
I have two IAX trunked *, there are loud crackles and pops, they are dialing
out a T-1 and are sip devices, it also drops words, any help or Ideas?
Let's fill in a few blanks first:
1. What version of asterisk are you using?
2. What routing devices are you using?
3. Are you using
Hi bp -
Does anyone know how to limit the amount of registrations that a sip user
can have?
For example, I have 2 softphones that I use on my laptop desktop, both use
the same username password. If I have both softphones up at the same time,
I can make simultaneous calls with each of them.
Hi Gary -
I had to make asterisk use ODBC for everything when I converted to
ODBC message storage. res_odbc and res_mysql wouldn't co-exist.
Hmm. That's odd. I've got the voicemessages table accessed through
res_odbc (of course), and the voicemail_users table accessed directly
through
Hi -
We've got a number of offices, and they're all using ODBC message
storage using MySQL. I've been trying to get MySQL replication set up
so messages left in a voicemail box at one office will get copied to
the corresponding voicemail box at all the offices.
We're also using MySQL
/06, Noah Miller [EMAIL PROTECTED] wrote:
Does anybody have any ideas on what's causing this error? Why would
MySQL not have enough memory? What does it mean, points outside data
file? Is anybody else doing this successfully, or am I a lone freak
Hi Josh -
Another approach you may want to consider for data redundancy that
does not rely on MySQL's finicky replication stuff is DRBD. Think of it
as RAID-1 across Ethernet. I have used it in production on some VERY
busy ( 1200 qps) MySQL servers for a couple years with no problems.
I would
Hi Dan -
Seems like other postings tend to think that saving recordings as
files and not as blobs in the database are a more reliable way to go.
Opinions on this? Looking at supporting it for ARI and judging
interest.
As far as integrity of the actual data, I think you're safe either
way.
Hi Kevin -
Okay, so calls going to and from office A have no problems at all.
Office B is having a bit of a delay (about 5 seconds before the CLI
shows the call is even started). The odd part is, it only happens when
they are making an outbound call. Incoming calls go directly to them
I am still looking for a solution and I am sure that I am not the only
one having that problem:
If provider A fails for any reason, the next provider should be taken.
exten = s,n,GotoIf($[${DIALSTATUS} : (CHANUNAVAIL|CONGESTION)]?
tryiax02:Hangup)
Yes, this is exactly how I've been doing
Can you post your iax.conf?
On 4/4/06, Marco Mouta [EMAIL PROTECTED] wrote:
Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server
Scratch that! Apparently asterisk just needed to be restarted..
wonder why a reload didn't work?
Reloads don't update music on hold settings.
moh reload seems to work just fine for me.
I don't remember how things worked with mpg123, and I've never used
madplay, but with the native
Hi Andy -
Anyone care to quote on 25 Linksys PAP2-NA units unlocked can email me
direct.
Straight forward sale best price new equip etc etc... I am a buyer located
in the U.S.
Need someone with stock that can ship right away. Will want 25 more in less
than a week.
You may get a better
Hi Marco
My asterisk for all my users, everything was fine for 3 days, but now
i can't access it.
But it is running...
Could any one help me on this?
Can you provide some specific information? At least the following:
Asterisk version
Operating System
Hardware
Technologies used (zap, sip,
Hi Tofik -
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
The wiki also lists GX::Transcoder which looks like it can do g729
Is there any way to get a polycom 601 to do overlap dialing?
I can't find anything on the subject, which confirms my initial hunch:
I really doubt it. You could probably work something up in asterisk,
though.
- Noah
___
--Bandwidth and Colocation
Hi Again Avi -
Sadly, that doesn't work -- the Polycoms store their
directories locally as well and re-upload them on reboot.
Another idea: Can you create the mac address-directory.xml files as
symlinks to the central file? Maybe if the phone sees a directory
file already there it will not
Hi Sam -
Thanks for your link. how to build asterisk into
this hardware?
As mentioned earlier, have a look at astlinux:
http://www.astlinux.org/
There are pre-built versions for soekris/wrap, and general x86
computers. Kristian (the astlinux developer) made this run on a
gumstix, too, but I
Hi Bjorn -
Everything you mentioned seems to point to the problem being a
hardware issue, or more specifically the way that FC and CentOS are
using your hardware.
Why not use different hardware and/or OS? Maybe FC and CentOS just
use faulty driver for your NIC?
- Noah
Hi Sebastian -
sorry for the long debug output below. I configured Asterisk with AMP to send
the whole number including the extensions of the callers to the called party.
Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
doesn't seem to work.
033811234451 is the
Hi Avi -
I know this is off-topic for Asterisk, but I don't know where else to
ask: I've setup a central directory.xml file for my Polycom IP501 phones
with a list of all the internal extensions. None of them have sd1/sd
as I don't want to enable any speed dials, just have a list in each
Doug this is in no way an offense to you but I think
we need to start the asterisk booze fund. This will be
for all of us that have ups and downs in working on
getting asterisk set up. I for one have my friend
Johny Walker right by my side when ever it gets to
me.
I'll second that. Maybe
Hi Domenico -
I have a IAX2 trunk between two sites (connected with an high bandwidth
link) but sometime/often I get:
chan_iax2.c: Auto-congesting call due to slow response
and call is dropped (and routed on a PSTN link).
In iax.conf, I have:
[iax-out]
username=iax-in
type=peer
I could ask why it can't authenticate against the key, but we've already been
there.
So, if I have 5 asterisk systems, and I want to have a different key on each,
and each system has a user and a peer section, and I have to use different
usernames... oh boy... this sounds like a horrible
Hi Erick -
Where can I do a keyword search of the posting in biz and users forums?
asterisk.org just
links to http://lists.digium.com/pipermail/ and that doesn't let me do a
string search across
all postings.
I'm guessing you mean the mailing lists rather than the forums. If
so, you can
Yeah, that's kinda what I've got set up in mine:
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=44198/phone.cfg,
sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=44198
OVERRIDES_DIRECTORY=44198 CONTACTS_DIRECTORY=44198/
It's pulling 44198/phone.cfg from the server fine, but for some reason
it's not
Hi JR -
I'm wondering has anyone made any further progress with getting MWI to work
from a stand-alone asterisk VM server to other asterisk servers where the sip
phones are actually registered to? MWI link accross IAX2?
To my knowledge there is no direct implementation to do this. The
Hi -
I am not sure what I did but blind transfers do not work. The Polycom does
not allow me to dial the extension of the person I want to transfer to after
I hit:
transfer - blind
I would strongly suggest getting the latest firmware, and using the sample
configuration files with
Hi Scott -
We're using Asterisk to develop a specialized IVR system for our
employees and someone is telling us there is some OSHA requirement that
you have to always be able to reach a live human on such systems. I've
never heard of that and google didn't turn up anything in my searches.
Hi Giorgio -
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
If this is possible, it would be quite
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
This is not just a function of the phone. The phone has no idea what the
caller id of the receiving end of the call will be.
Aren't you bothered by the fact that the sound file quality goes up and down
as different sound files are played? It's quite obvious to hear the difference
between a ulaw file and a gsm file.
Aside from a few company specific greetings, the slinear package has all the
sound files we need.
-
Hi Michael -
I may be way behind here, but I see that digium redesigned their site.
I cannot find the mailing list search screen.
I don't believe there has ever been a search screen.
Do I need to just rely on google and other generic search engines or is
there a search on the digium site?
Howdy -
The transer button on the polycom phone does not seem to transfer/park the
call properly. I have to use the # - 700 to park the call.
If I recall, using the Polycom transfer, you have to make sure it is done as
a blind transfer. The Polycom attended transfer (default) option
Hi Andrew -
On Monday 13 March 2006 10:20, Noah Miller wrote:
The transer button on the polycom phone does not seem to transfer/park
the call properly. I have to use the # - 700 to park the call.
If I recall, using the Polycom transfer, you have to make sure it is done
as a blind
Hi -
I was able to install Asterisk and configure many of it's features.
Currently I am using Extensions.conf for giving all my contexts and
extensions. Whenever I change my extensions or add a new context I have to
reload extensions.conf and practically it is not possible reloading many
I've contact Polycom via their web form and they emailed me back
around 2
days later. We've purchased all of our phones via authorized
resellers.
Dunno if that makes a difference or not.
My question was about the button programability and their response was
that the manual was incorrect.
Hi Damon -
Has anyone been able to get the IP501 to discover the FTP server IP
address (via dhcp or dns) and download 100% of the config from a
provisioning server?
Sure, works great! I'm not sure if you got the TFTP config from the
gentleman who suggested it, but this is really dependent
/proftpd.conf (or wherever
you choose to put your config file):
UserAlias PlcmSpIp plcmspip
- Noah
-- Forwarded Message
From: Noah Miller [EMAIL PROTECTED]
Date: Thu, 23 Feb 2006 11:34:31 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: [EMAIL
Hi -
After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday. Complaints
went from 10-20 per day to ZERO. Literally overnight.
I wonder if this is an ILBC frame size issue of some sort? Seems odd.
I've got to add my name to
Hi Andrew -
How do all of y'all out in asterisk-users land set these phones up, and why
did you choose to do it the way you did? Were there nifty features you
discovered through your particular configuration, are they set up to
specifically avoid problems, or is it a mix of the two?
We've
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