Hi Henry -
Just started using an asterisk-based PBX with Polycom IP501 phones. Am
Fairly satisfied and am starting to get into FTP setup of the phones.
Have figured out most things except for how button remapping works.
In sip.cfg, I have this entry:
keys
Hi Andrew -
I have a need to be able to identify incoming calls based on some factor
(could be time of day, caller ID, dialed number, it doesn't matter.) --
Assuming Asterisk can differentiate between the calls I want, how do I inform
the IP501? There are only three line appearances -- I
Hi -
Has anybody been able to successfully remap SpeedDials on Polycom phones?
The manual seems to indicate that you can, and I followed the advice in this
list message:
http://lists.digium.com/pipermail/asterisk-users/2005-October/129142.html
The result I get is that the remapped buttons act
Hi -
That is correct, The SIP phones are all on our LAN. I changed the nat's to
say no, but I still get the same problem. Another thing, when I call out to
the pstn from our local sip phones. The same problem happens. The outid line
rings, the person picks p but no sounds.
Any
Hi Matt -
I have a polycomm IP600 that about 30-50% of the time continues to
ring after asterisk has signaled an answer. Has anyone else experienced
this?
Not I. Weird! What versions of things are you running? Anything on the
CLI or in the logs?
- Noah
Hi Gavin -
I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.
What are some good
Hi -
I'm running 1.6.2.0041 according to my phone.
Which firmware worked for you?
It was the old firmware from when we first got the phones actually.
1.4.x I think. Then I read that they fixed the CID issue and decided
we needed an upgrade. I tried it out on my phone, but didn't really
Hi Brent -
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour
Hi Dan -
Got this warning when upgrading to 1.2.3 even when using the most
current asterisk-addons and even svn asterisk-addons.
WARNING WARNING WARNING
Your Asterisk modules directory, located at
/usr/lib/asterisk/modules
contains modules that were not installed by this
version
For what it's worth, I've been messing around with my install all
night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk
version 1.2.1. Even set the date ahead, still no problems. Could be
a fluke, I'm interested if anyone else is using 1.2.1 and has these
issues, but for
Hi Bill -
Thank you to all who responded to my inquiry below. As explained by a
few people, Polycom has a policy of withholding current firmware
releases from users, thus forcing them to contact authorized resellers
for support should they need this code. Similar to another reported
tried using other
DTMF combinations like *9 and just number combinations, still nothing. BTW:
Asterisk is in the media path.
What am I doing wrong?
Thanks,
Noah Miller
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Hi BK -
The blind transfer does not work.
The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey- the display shows Blind transfer to: and
cursor is in the second line
4. enter the number- when we enter the second digit of the number
Hi Bob -
I am now running sip 1.6.2 with a 2.6.1 bootrom.
After moving from a 1.5 I now only see 2 softkeys
at the main window: New Call and Forward.
How do I get a Park softkey?
Just so you know, the park softkey will only show up when a call is active, and
so far it will only show up
and/or a greeting
during this time.
- Noah Miller
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= secret
PORT= 3306
DATABASE= asterisk
Anybody know what I've bungled?
Thanks!
Noah Miller
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parking).
I didn't even realize there was a park softkey, because I don't have that
feature enabled (yet). Yay!
Thank You! Thank You! Thank You!
Noah Miller
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Asterisk
Hi Brad -
Of course, what I really want is an asterisk VM system where user
accounts are transparent across many servers, and VM's can be
shuffled around between the servers as a configuration option in
voicemail.conf. I wish I could code and knew Objective-C! Maybe I
should submit a
Hi Alvaro -
Instead of asking a question, I thought I'd post an answer. I got the
Polycom IP501 'Park' softkey working with * by doing the following:
You are my favorite person today! This rocks, and solves an annoying
problem I've been trying to figure out for a while (single
want.
http://dev.mysql.com/doc/refman/5.0/en/replication-howto.html
Ahh. I would have been fumbling around trying to do this the wrong way for
a long time. Thanks Are!
Thanks,
Noah Miller
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Hi Anthony -
We have a user (our CEO) who has phones in two different offices, and
we'd like him to be able to get all his VM in either office, regardless of
which office was originally called.
My idea instead is to use externnotify to run some kind of script to
forward the vm to another
the VM files if there's a duplicate name. The one
problem there are that my coding skills are seriously bad.
Anybody have any ideas or done something similar?
Thanks!
Noah Miller
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parameter as an IP rather than as a string, and I can't
get this to work with Cisco DHCP. Maybe somebody else has, though?
Thanks,
Noah
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Noah Miller
Sent: Thursday, 10 November 2005 9:08 AM
To: Asterisk
Hi Peter -
Hmmm, I tested this quite a bit as per below...
Sorry if this seems lame, but you are using FTP right? Because FTP is the
default, not TFTP (even though you use the DHCP TFTP option to set the FTP
server address).
Thanks Again! I haven't tried yet with the 3.x bootrom series.
Hi -
I've been trying to set up my Polycom phones to get the boot server info
(tftp-server-address) from DHCP on a Cisco router. I've previously just
specified it manually on the phone, and that works well enough, but I need
to change now (because of the number and geographic locations of the
Hi Matthew -
You could also take a look at features.conf, and use ** for blind
transfers, ## for attended transfers, *0 for recording, and *1 to
hangup.
I haven't tried mapping them to polycom buttons, but there was
recently a discussion about that, just this week you can search the
Hi Mojo -
The trouble Matthew and I were having was to stimulate presses of more
than one button in a sequence -- SpeedDial function was the only
one I
could find that was close, but this opens a new call appearance for
the
call rather than just playing the dtmf over the open one.
Yeah,
Hi James -
I am doing some experimenting with Asterisk 1.0.9 and Polycom
IP501's. I have the extensions setup, and everything is working
well up to this point. Now, I want to setup my system so that a
user at an extension can start a recording on demand. I have tried
various Google
Hi Alan -
Asterisk 1.0.7 (AAH really)
4 co lines from Bellsouth into a Diguim T400P.
Polycom 501 x 4 on the desktops.
My problem is on calls to or from the CO I hear a pinging (thing
sonar
ping in a submarine) every 12 seconds. You can set your watch to it.
COuld this be a call recording
Hi Andres -
I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP
over asterisk, and yes, sip are 100% tweakable,
how do you configure your system, all by hand?
Yeah, by hand. When I first started doing this there was no such
thing as AMP. Plus, I've got some
We have another box that is running 1.0.7 with H.323 to an H.323
gatekeeper and it is just acting as voicemail for a Cisco Call
Manager.
It crashes at least 1-2 times per week. Starting asterisk again
brings it back up. I don't know why it happens and I have been unable
to get anything
Hi Andres -
I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301
The Polycom misses 1 out of 2 dialout calls, this is the full log
from a
call which didn't go through.
303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 2: Found
Hi Jorge -
I got 3 Polycom 300 phones, and upgraded to the latest firmware
provided by the reseller.
This is my first experience with Polycom and I cannot make them
register in my Asterisk Box.
I followed every advice I found (including separating [user] and
[peer] in sip.conf.
Using
Is your Asterisk server listening on port 5061? If not, just change
the
entry to 5060.
Also, I'm not sure how your sip.conf is set up for asterisk, but if
you've set it up like:
[203]
type=friend
username=blah
secret=blah
etc...
Your Polycom config file will generally look like this.
Hi -
I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to
set up some redundancy on IAX connections between locations. I have
two IAX peers set up that work correctly by themselves: ast551-out
and ast551-out-backup:
[ast551-out]
type=peer
secret=secret
username=ast551
Hi Kurt -
I would like to know if any body is using the Polycom Soundstation IP
4000 SIP conference phone with Asterisk. I am thinking of purchasing
one.
Yes, we have one, and we have the add-on pod mics for it, too. The
setup works well, for the most part. The mics aren't quite as
Hi Eric -
I am having trouble with one of our IP600. Every five days or
so, the phone locks up. This is the third 600 I have put in place. I
am running asterisk 1.0.9. Has anyone had this problem with the
IP600?
What version of the bootrom and sip firmware are you using? Can we
Hi Doug -
The link I included had a link to the RFC document. I didn't see a
way to
prepend digits based on that information, which is what led me to
post the
question here.
Why not have the asterisk dialplan prepend the digit? It's easy
enough to do that way.
- Noah
With the 1.5.2 firmware, have you managed to get one-touch message
access when
pressing the Messages button? It worked for me with 1.4.1 but no
longer with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
msg msg.bypassInstantMessage=1
In the phone.cfg
Hi Mike -
I have configured my polycom ip600 and ip500.
The phone works well.
But the clock is wrong and flashes the whole time. Drives me nuts!
I have set the time offset on the DHCP / boot server. 36000 (I'm in
Australia!)
It doesn't seem to help...
Sounds like you're running the 1.5.2
it's fixed
russell at lists.digium.com russell at lists.digium.com
Tue Jul 19 11:23:06 CDT 2005
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c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in
front of me. Mahler's book was the first one to come up and it seems
like a good place to
Maybe it is not just me going crazy. I have garbled audio based on
July 17. And I thought it was me messing up the ztdummy setup.
I've encountered the same problem (same negative timestamp log
messages, and garbled audio). The last version of CVS HEAD that
worked
on my system without any
Hi -
I've just been testing out the latest CVS HEAD (as of about 10:00a
EDT today). I'm getting some weird errors. Calls from one sip phone
to another have OK audio in one direction and highly scrambled audio
in the other direction. The console shows this error repeated ad
nauseum
Hi Jyran -
Has anyone attempted or have had any issues getting Asterisk and
Digium
cards working on a Dell SC420 server?
Yes. One of our offices has an SC420 running Tao Linux and Asterisk
(CVS HEAD), using a Digium TDM400 (TDM04B). It works pretty well,
for the most part.
Just so
Hi Patrick -
We are about to buy several Snom phones.
Does anyone have warnings or advices against these phones ?
Our finalists were Cisco, Polycom and Snom.
We will be using only the SIP protocol.
One particularly weird behavior on the Snom phones - When the phone
is either 1)
Hi Randy -
Hmm, quick question.
Does the issue occur on all current models of SNOM phones, or just the
320 and 360 mentioned below?
From what I can tell, the issue occurs an all existing models and
firmwares. I own some 190's and the issue has been there through all
versions of the
Hi Andrew -
Is this likely to be the problem? Are there known issues with this
configuration? Does anybody have any advice on how to configure the
kernel not to share those IRQs?
Depending on your BIOS and motherboard, you may be able to use
another IRQ if you move the card to a different
Hi Craig -
I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've
configured DHCP and TFTP and successfully updated both the BootRom and
SIP application. I've also created a custom cfg file for this phone's
MAC address and the settings seem to be taking just fine. I can see
Hi Matt -Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP
Hi Justin - Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I've been testing 1.5.2 for a few weeks now, and I'd have to say
Are you using inband DTMF? There are other options but I don't know much about the polycom phones. I have noticed that sometimes when accessing voicemail, it will 'miss' some dtmf tones if they are too short. This doesn't explain the number changing, unless your dial plan is putting in the
Also, you may want to check the digit map (the pattern recognition when you dial). It is in sip.cfg. I know that by default it treats numbers that start in "11" specially, but it shouldn't really transpose numbers. That's weird. Its not that it transposes numbers,it induces a delay that
Hi All -
I noticed that the Polycom IP501's are now shipping. Has anyone
gotten one yet, and if so, what's different about the phone? Any UI
improvements, or is it just better hardware?
Thanks,
Noah
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1. How do you set the music on hold to work with asterisk. Right now
when I place a call on hold the caller hears nothing. MOH works
with all
my other IP phones.
I don't know the answer to this one (it just works for me), but I
remember it coming up on this list before. You can probably
Hi Chris -
When I change a setting via the web interface on a polycom 500, it
takes
minutes to allow access through the web interface again. Any idea
why it is
so slow?
I've always had the same thing happen across all versions of the
firmware since 1.3.0. The phone will boot and be able to
Hi Wiley -
BTW - Anyone who gets these. Note that the IPMID and SIP config files
are now combined.
Just to save any confusion...
The link from teh Wiki shoudl be updated soon I think...
Thanks for getting these, and being good enough to share them with
the rest of us! BTW, what conscientious
Hi -
I've noticed that I'm getting audio delays when asterisk is playing
back a file from disk and new zap channels are being created or
destroyed. Audio streams are generally fine (person to person calls
do not experience this issue). Sometimes the drops are very short -
barely
Hi Chris -
I have the Polycom 500 and 600 phones. Rather than put an entry in for
each
line appearance, I would like to use the feature that shares one
extension
for the lines, so that I will get the call on the enxt available
button. How
do I configure that?
You can register each line on a
Hi Dan -
I would also like to figure out how to make the phone *ring* when
you're already on another line, but haven't had a chance to seriously
explore it yet.
Is this still a problem in the latest firmware? This could sink my
hopes of going with a Polycom phone if there isn't a way to have them
The phone's context is cytel-internal.
This allows us to hit 3XXX to get someone on the inside.
If you hit 9 at the beginning, you Goto() the cytel-outgoing
context.
So lets make a call..I'll dial 918005551212 (toll free directory).
The 9 sends it to cytel-outgoing. Call is made. Bridged. I then
Hi Marty -
The complaint from the users is that calls cut out, kinda like when
you have spotty cell coverage. Doesn't seem to matter whether the call
is incoming or outgoing, although it might be true that my users hear
the remote party cut out, while the remote party doesn't notice the
same
from
Hi Marty -
Are you using vlans at any of these locations? Just to clarify, my two
locations don't interface with each other. They are two different
clients of ours each connecting only to the PSTN (one via IAX, the
other
via SIP).
Nope. No VLANs for us.
I should also mention that the
Hi Wiley -
I was afraid you would say that.
Does anyone out there have the latest firmware for the Soundpoint IP
4000?
You just need 1.4.1.0400. It comes on the phone. I tested ours first
just by using the web config, and it worked. I have it getting the
config from FTP now. If you want to
Hi -
We've been using IAX forwards between sites for a little while now
(with centralized VM). For the most part, it is fine, but I have some
very minor, yet persistent QoS issues on calls over the IAX forwards.
For most normal calls, there are very occasional minor glitches, just
an
On April 22, 2005 05:35 pm, Mark Phillips wrote:
Ext: 1 Cause: Invalid information element contents
(100), class = Protocol Error (6) ]
-- Processing IE 8 (cs0, Cause)
Your zapata.conf does not match your telco's provisioning of the PRI.
Contact
your switch tech and verify
I have two Asterisk boxes installed but am not sure how to setup the
configuration for what I want to do.
One box has two FXO cards in it that will connect two PSTN lines. I
want to
have Asterisk transfer the incoming calls to the other box which has
an FXS
card in it. That box should ring
Do the SetGroup and CheckGroup functions behavior differently in
CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to
work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Do you not need a
exten =
Citrix is bad for anything for that matter. First and foremost it is
not
secure as you have to open your PC ports for Citrix to access the
remote
client. And same is the case for remote client as it Needs ActiveX
installation.
I have been flooded with Spyware once I did this for a client of
Hello wonderful asterisk users list.
I have some energy traders that are currently using 2 wire
hoot-n-hollers (squawk box, always open direct line) to different
trading floors throughout the country. Each box has one
hoot-n-holler
line. I would like to make these boxes IP based by connecting
Hi Jorge -
I am using Asterisk and I activated the presence (status monitoring)
feature in the IP500 phones, but it is not working. It necessary to do
something in Asterisk addionally,
This is something that has been discussed on this list a few times, but
to the best of my knowledge the Polycom
Does anyone know how Polycom 500s will be able to update their time.
My
setup for a time sync with Public domain Time servers is not
successful.
I had the same problem. I would set up a time server, and it would
work for a while, and then just stop working. I switched to various
different
On Tuesday 12 April 2005 10:18 am, MobilPete wrote:
can anyone help ??
trying to get Polycom IP300 to utilize both lines, would like calls
to roll
to open line when incoming call arrives while user is on line 1.
Looked
everywhere and tried many things with no luck.
Do you have your lines
Does anyone know how setting the TOS bits in iax.conf corresponds to
the Cisco TOS types?
For example, if I set:
tos=0x04
in iax.conf, and on the Cisco, I use:
access-list 110 permit ip any any tos 4
I can't get the Cisco to match any packets. I've tried various
combinations of numbers on both
Also, what happens if for example, the user is accessing his VMB
on server 1 and changes his password, then travel to where server
2 is and tries to access his VMB? the config on server2 would
still have the old one so you need to sync voicemail.conf on
all servers too ...
If you use the
exten = s,1,answer
exten = s,2,SetCIDName('PMG')
In a lot of config files I see exten = s,snip ..
Is s just an extension or system variable for all extensions ? or
something else ?
The start extension. It is a default. When you put a call into a
context and it doesn't know where else to go, it
This this may sound ridiculous, but we've had problems with this when
the
users did not plug the handset cord in completely. 8 out of our 12
employees
made the mistake, as the plug on the IPX00's appears to be all the way
in
when it is actually not.
Not ridiculous at all. We had the same
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB),
and what I want to get done is that if I dial 1X on SrvB the call must
be routed to extension X on SrvA and if I dial 2X on SrvA the call
must be routed to extension X on SrvB. I've read the www.voip-info.org
wiki abouta
A word of caution, we ran that same setup for a while and then bagged
the TDM400P in favor of 2 Sipura SPA2000 ATAs. The TDM400P kept locking
up and the SPA2000 never has. No problems getting fax from * to the
SPA2000 via g.711 over a FastE LAN.
I am not sure if the TDM400P has gotten any better
Just pulled out our brand new $1,500 TE405P and put it into a Dell
Poweredge 6450. Nothing. Card not recognized nor listed under
lspci.
I've been on hold at digium for a while now and I get the feeling
that Digium is gonna say too bad sucka..the card works..*click* but
if enough people have
Hi Courtney -
If I were to buy 20 did's how do I know within asterisk which number
was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your
Hi Paul -
I am very new in asterisk community. I just compiled installed
asterisk on a fedora core 3 machine and I want for test purpose to do
a small PBX that use X-lite windows sip clients and no trunk for the
begining.
Where can I find a good how-to to do this job. A small starting
how-to
With recent discussions in regards to a forum, I have set-up a
multi-faceted Asterisk and Open Source Discussion Board. The link is
www.voipnewbie.com/forum It is open and ready for use.
Hey Great! Thanks! Just make sure to get linked from the asterisk
website (probably in the Digium
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone. I considered that as a possibility originally,
and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same problem.
The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing
The page in the wiki used to say that the person would not recomed
Polycom phones to anyone.
I missed that part of the WIKI! Maybe it's more recent. I don't want
to invalidate anybody's experience, but in the process of picking
phones for my company, I evaluated a huge number of IP phones (and
Hi Eric -
I'm having a problem with my Polycom phones and hoping someone else
has experienced the same thing: Outbound calls are fine, and inbound
calls originating from another SIP phone are fine, but inbound calls
to the Polycom phone from an IAX channel sound like you're talking to
a robot.
Hi Matt -
how can I get all the phones to enable call waiting by default
instead
of having to dial *70 on each one to activate call waiting?
What phones? are you using Avaya, or Toshiba? Since you are posting to
this list I will guess you are using Asterisk, in which case I have no
clue why *70
Hi Josh -
I've got an issue on the snoms, and I'm wondering if anyone has some
recent experience with it; I've contacted the one specific reference I
found to it in the list archives, and the person in question didn't
seem to find an answer (and snom doesn't appear to be finished moving
their
Hi Josh -
I don't have a 220, and I haven't really tested the 360, but on our
190's I just register each line appearance to the same sip device,
and multiple simultaneous calls automatically roll from line 1 to
line 2 to line 3, etc. Are you using any CheckGroup/Setgroup
statements, or
How can TOS tagging on the IAX channel affect a phone that is
completely
SIP?
Quite right, Wiley. I think I need my head checked.
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To
P.S. Just to reiterate the point: Snom - I'd prefer a second ringing
call to ring to another line appearance rather than have the call get
sent back as busy. Maybe there can be an option?
Yeah, and we're probably not the only ones. The 360 is so far ahead
of anything else we've evaluated in
The vast majority of our handsets are Polycoms. I know that they do
this correctly (with a little help from CheckGroup/SetGroup and
multiple SIP registrations). Of course, you can't get the nifty
sidecar for the Polycoms like you can for the 220. You can get a
sidecar for the Cisco 7960,
Hi Matt -
I apologize for not providing enough info My question though is
not how to get my provider to provide it... how do I enable call
waiting from asterisk TO my sip device by default without having to
dial *70 and have asterisk put a CF mark in the database for my sip
device.
Sorry for this n00b type question but what is the use of a multi line
phone?
Occasionally I see discussions going on about this subject but with a
single phone you can only answer one call at a time (at least I can
with
only one phone).
What would you need the extra lines for?
Usually for a
Hi Noah -
I've managed to get my asterisk server up and running with a single
POTS
line and a polycom IP500.
It will happily answer the phone line, tranfer calls, voicemail, etc.
The problem comes when I pick up the polycom phone and want to place an
outside call.
If I dial 913237773456 it just
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have
| the following extensions:
| On the Polycom phones, when I want to dial from extension
100 to any
| extension 120 or above, or dial out, it dials just fine. If
I
with asterisk. I don't think it's
always right to direct all new people to [EMAIL PROTECTED] If they want
that, fine. If not, why push it?
Thanks,
Noah Miller
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Hi -
We've got multiple offices with their own asterisk boxes (CVS HEAD
11/03/04-14:59:37) connecting to each other using IAX forwards. All
users are on SIP phones. Voicemail is centralized to one location.
Everything is hunky dory except that the users in the remote offices
don't get
Hi Noah -
I got everything to load via ftp. The phone appears to correctly boot
from the config files. I also put the latest firmware there and the
phone sucessfully loaded it.
For some reason, the phone and * don't see each other. This is the
part
that confuses me. Any clues as to why the
You arent going to make this happen as you describe. Vonage is not a
good
service to use with Asterisk. To quote from the Wiki:
Vonage service is locked to the ATA they send you. It is not possible
to
connect Asterisk (or any other SIP UA) directly to your main Vonage
service.
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