If you've considered the Snom, you might also want to test a
Zultys 4x4 or 4x5. I picked a 4x5 up off of ebay recently and
have been pleasantly surprised by it. While I don't currently
have a Polycom to compare it with, I would rank the audio
quality equal to the Cisco's. It also just 'does
Not sure the rquirements for your receptionist. I have found that the
IP600 does have most everything required to function properly. If you
do have an office without DID and a lot of traffic then you may want to
look at the tools to display status on her computer. I do have a Snom
inhouse for
Hi Grant -
As you probably know, the tdm400p needs an ide power supply, but the
dell poweredge 2600 that this card is destined for eventually has all
the power supplied on the backplane with no ide cables.
The thing is, on the motherboard in the server, there is an ide ribbon
connector, and beside
We have bought PBXware GUI from Bicom systems and configured
extensions
with Polycom Phones as UAs.
The Polycom Phones can dial out and make calls but I cannot make
extension to extension calling.
Googling did not help much.
As you may be aware PBXware is a closed source software GUI from Bicom
Also, I'm sure you've probably checked on this one,
but are the phones registered with asterisk?
You can make outbound calls on them without them
actually being registered. I'm assuming you can
still get in and see the CLI. What does sip show peers
look like? What does sip show peer xxx show?
1.4.1 over here.
Just to rule out all the possibilities - it's not MWI is it?
http://www.voip-info.org/tiki-index.php?
page=Getting%20MWI%20on%20Polycom%20Phones%20to%20work%20with%20Asterisk
It shows up as a little half-ring. It should also be accompanied by
the top LED flashing, and a
Hmmm... I have this aweful feeling that I'm choosing the exact wrong
time to ask a newbie question :) Oh well, here it goes.
You're obviously literate, and you've put forth effort, so hopefully no
one here will step on your toes.
The quick question is : How do I dial an extension? (answer is
That's kind of what I thought, but I am trying to put together a phone
to multi-phone paging system. I all ready have and overhead paging
systems, but the powers-at-be want a phone paging system.
Fortunately for us Polycom people, somebody took the time to write a
Perl AGI script for just this
1) Get a 4-port TDM card and install it into your Asterisk box.
Connect the TDM ports to your modem ports. Then forward incoming
calls on fax DIDs to those TDM ports.
Digium TDM 4 fxs is not really a good choice for a faxing system. I've
tested it for a while.
You should read old messages here
Hi Rich -
Those type changes to iax.conf require a full stop of
asterisk followed by a cold asterisk startup. A restart
from the CLI won't cut it.
Ahh! That's a very important piece of information!
Were you previously doing the CLI restart?
I did lots a CLI reloads, and few cold restarts to
Hi All -
Well, after happily existing in a one office environment with asterisk
for a few months, I've now decided to start adding in our other offices
with their own * boxes and IAX connections (over VPN). Unfortunately,
I'm an idiot and I can't get it to work. I'm having some kind of
-- Executing Dial(SIP/68-4ab6,
IAX2/ast33:pass at 192.168.1.130/08 at from-sip) in new stack
Feb 11 14:16:58 WARNING[5653]: channel.c:1898 ast_request: No
translator path exists for channel type IAX2 (native 0) to 4
Feb 11 14:16:58 NOTICE[5653]: app_dial.c:800 dial_exec: Unable to
Hi Rich -
Personal opinion (and everyone has one) is the problem is a little
deeper then just a codec. Try 'iax2 debug', place a call, and look
through the early part of that trace. Are IP's right, etc?
I believe the native 0 is simply suggesting there is no match when
negotiating the codec. The 4
does not do instant messaging. You
can check on the feature list and with the developers, but I'm pretty
sure that's not possible. See the wiki here:
http://www.voip-info.org/wiki-Asterisk+presence+jtodd
harry
--- Noah Miller [EMAIL PROTECTED] a écrit :
Hi Harry -
Can you get SUBSCRIBE
IMO, your best defence is leaving ssh's default setting
which disallows root logins entirely. There's no reason
for a remote user to ever have to log in as root. Root
access should be obtained by a logged-in normal user
using sudo, or su.
I'm not sure what happens when you do a fresh compile
I'll call bullshit on that. I know for a fact that
Debian does NOT allow root logins except from
console. Hell Debian isn't allowing root logins
from X anymore due to the likely hood for you to
try and use root for more than administration.
I'm sure that's true nowadays. I haven't played with
I get the following error when trying to compile asterisk 1.05 on red
hat 9.
Is this the tarball available for download from the asterisk website?
You might try CVS instead - try the CVS HEAD release:
# cd /usr/src
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login
Hi Harry -
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
What version of the SIP firmware are you using? I've had success with
1.3.0, 1.3.1, 1.3.4, and 1.4.1.
My sip.conf entries for my Polycom phones look like this:
[12]
Hi Vince -
My next goal is to setup 1 SIP channel, and be able to call the
Asterisk PBX
from a softphone.
Then setup 2 SIP channes and be able to call one from another.
What is the best open source softphone software available for this?
And what is the best documentation source for finding out
Hi Walid -
I need to use Asterisk to call out PSTN numbers via an analogue line. I
understand Digium manufactures these kinds of cards, but can someone
tell me
which model number it is. I really only need a card with one or 2
analogue
ports max.
You'd be looking for the TDM400P. You can get it
We have a client that wants to bond 2 DSL circuits instead of getting
a T-1 (or similar) at their office to run their VoIP traffic on. We
came across this Multihomed Gateway (MH200):
http://www.cyberpathinc.com/mh200/details.htm
Does anybody think this would work if installed at the client
I installed a tdm400p into a old p2 machine.
I'm not able to see it under /proc/interupts or using lspci..
we removed all other cards. changed slots, forced irq to that slot..
etc etc.
what is the min specs needed to get one of these cards running?
PCI 2.2
I don't think the TDM cards will work
I think Polycom has added another feature that nobody wants.
With MWI configured, and a phonexxx.cfg that has this:
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=XXX msg.mwi.2.subscribe=...
/msg
Under 1.3.4 and earlier, the
In most configuration files i see that they comment lines instead of
adding spaces.
e.g. - correct way
;
;IAX configuration
;
[general]
blah blah blah..
;
register = ..
Is this incorrect:
;
;IAX configuration
[general]
blah blah blah...
register =
I guess what i'm trying to
I tried to reinstall all previous version (zapata, zaptel, libpri and
asterisk)
I reboot. And then... same thing :(
Ring your T1 supplier, and ask them what they see.
They may well have marked it as out-of-service, in which
case it won't come back 'till they re-enable it.
I don't know
I have my * and polycom system setup to do Auto-Answer
for internal SIP/Staff calls, and I am running into an
issue with this and the polycom call transfer feature.
* is seeing a new call come through from the polycom
and is then transferring the call over. I need to know
if there is some way
192.168.1.130:5060;branch=z9hG4bK304b1809
From: Noah Miller sip:[EMAIL PROTECTED];tag=as44d2096b
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 04 Feb 2005 17:39:53 GMT
Alert-info: Ring Answer
Allow: INVITE, ACK, CANCEL
So I guess the problem is in my config for the phone? Or maybe
asterisk
has to send alert-info more than just once? Does anybody have this
auto-answer config working reliably on a Polycom phone?
Thanks!
Noah
Noah,
Please see my Polycom config files at
Hi All -
I'm trying to implement the auto-answer config from the wiki, but the
result for me is that the phone just rings as normal. I'm running
firmware version 1.4.1 on an IP500. I've added the following to my
sip.cfg:
alertInfo voIpProt.SIP.alertInfo.2.value=Ring Answer
-- Executing SetVar(SIP/27-8b27, ALERT_INFO=Ring Answer) in
new
stack
-- Executing Dial(SIP/27-8b27, SIP/19|20) in new stack
-- Called 19
-- SIP/19-e562 is ringing
and the phone just rings. Any ideas? Is it the firmware version?
Am I
not setting ALERT_INFO correctly?
I faithfully followed the instructions from:
http://www.voip-info.org/wiki-
Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
but still the message waiting indicator doesn't flash when a message is
waiting. There is a brief intermittent chirp but nothing more.
Using latest firmware 1.4.1
My
I don't know if Polycom decided to change their policy or not, but I
just went to their website and when you click on the downlooad link, it
will now present you with an option to join their extranet. I did that
and I got the following message immediately:
snip
Access to certain Resource
I have the same dilemma with Polycom phones. Given their support
(actually complete lack of), I am quite loathe to giving them business.
On the other hand they are so darned cheap compared to other similar
phones, I sure get tempted to use them if I can find a workaround.
Compare Polycom IP-500
Hi Chris -
I am getting to my wits end with these phones (and so is my boss). I am
getting an random echo on these phones and I have an issue opened with
Polycom and its been in their research and development department for
almost a month with no results.
I'm amazed it got that far with
Have you tried adding SetGroup(), and CheckGroup() functions
to the dialplan that rings the phone? It maybe something to try.
I think the problem is that these functions only work from the dialplan. In this case, Sean is trying to get calls from a Queue (and not the dialplan) to the correct
Noah, I could really use 1.3.4... however, a better question might be
how are *you* getting 1.3.4? I can't seem to get this from anyone,
including my reseller.
I know how you feel! Polycom has made a product that sells like
hotcakes, but they don't seem to want to support it at all, even
I got a problem with asterisk 1.0.2 - it drops the calls,
both sip--sip, and zap--sip.
The conntions can stay for seconds to several minuttes,
and then the connection just cut off.
I can't see anything in the logfiles. (or dont know what
to look at.)
It drops several connections at a time,
I purchased some 20 Polycom phones (brand new) for a very good price of
around $165 each. Now I am having a nightmare in configuring them. I
pulled the bootrom, SIP and config files from freedomphones.com,
modified them for my need and and started configuring the phones. First
couple of phones
Hi Mike -
I hope this isn't a double-post...but here goes. I have setup an * box
using WBEL, and I have * up and running. The problem I have is that
when I dial an extension I cannot hear anything. It's not my sound
card either. I can see the call going through on the CLI and I see
where it
Hi Mike -
I am using Firefly for the softphone (IAX option).
IAX.CONF
bindport=4569
bindaddr=0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
#include iax_additional.conf
Well, I don't see any definitions for devices. I'm assuming that
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?)
ATA. Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
2) use spandsp on my asterisk system and add suitable print
Has anyone been able to find a way to disable call-waiting on Polycom
phones?
I've not yet found any solution to this, and I haven't seen anyone else
who has. Definitely please let us all know if you do find the answer...
I wrote to Polycom about this a couple of weeks back, but I haven't
heard
Well that didn't workI now get this error
Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to
create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569/5, b) in
new
stackJan 12 16:56:21 WARNING[4989]:
snip
Well, thank you to Tor for the SetGroup/Checkgroup config. It works
well! (Thanks to John for the contexts/dialplan version, too).
/snip
I use also SetGroup and Checkgroup to roll 4 lines on a Polycom 600
for a receptionist. Just to let you know, you can press the
line-appearance button
Hi Mike -
its my first time to post here, im in the process of building asterisk
based telephone system (just small). i already installed asterisk
server, i just wanted to test 2 sip softphones to get working before i
move on, is it possible to have 2 softphones talk to each other
without
any
Hi Steve -
Is * a proper tool to provide a SIP-MGCP gateway? Am I even asking
for something that makes sense?
Sure. I think * is perfect for what you describe. In fact, I don't
really know of another tool that could bridge SIP and MGCP.
If so, where's the best from the ground up, assume I
Hi John, Kevin, Tor and Wiley (and everyone else) -
I guess the phone just doesn't register as busy when there is only one
call on a line. It has to have two calls on a line appearance to
register as busy. Has anyone figured out how to disable this hold
feature and just have the second call go
Sorry, wrong subject on last message.
Hi John, Kevin, Tor and Wiley (and everyone else) -
I guess the phone just doesn't register as busy when there is only one
call on a line. It has to have two calls on a line appearance to
register as busy. Has anyone figured out how to disable this hold
there is only one
call on a line. It has to have two calls on a line appearance to
register as busy. Has anyone figured out how to disable this hold
feature and just have the second call go to the second line, the third
call to the third line, etc?
Thanks,
Noah Miller
I have these very phones and took me a while to figure this out myself.
The phone considers each line registration to be a line with a second
line. So, call line while someone is on a call and another instance
will appear below. That means you only need one registered instance
for the phones to
Hi All -
I'm running version SIP version 1.3.4 on various IP300, IP500, and
IP600 Polycom phones. I'm having a tough time with MWI. I thought I
remembered somebody on the list saying that they had it working, but I
can't find it in the archives now. I have all the phones configured
for
I wonder if there is an application available, what would
allow me to have a conference call (meetme) with video.
Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this
functionality:
http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+video+conferencing
You can
Hi Varun -
What are the basic packages required to have
a basic asterisk PBX up and running with all functionality.
I am using fedora 3.
I have downloaded asterisk-1.0.3.
Do I need any other package ?
You'll also need zaptel and libpri. You can download these from the
Asterisk site
Hi -
We have a couple of large spaces that we'd like to
cover with dedicated conference units like the Polycom Soundstation
IP3000. We're concerned about adequately covering the spaces, though, one
of which is very long and narrow. I wanted to get external add-on
microphones for the
pri show span 1 pri show span 1 Primary
D-channel: 24 Status: Provisioned, Down, Active The
"Down" indicates the d-chan is down. Call Verizon and have them activate
it. (Its not uncommon for telco's to deactivate the d-chan when a
circuit hasn't been fully turned up. It reduces the
Hi -
We're moving up in the world to a PRI (Verizon), and I'm having some
problems with it. I'm new to this PRI thing, though, so maybe I've
just screwed up a simple config detail. I've got a TE410P on a Dell
PE1600SC (ServerWorks Chipset). The card itself has a green light for
the PRI, and
: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
The Status has me concerned - Provisioned, Down, Active. Is that
Down normal?
Thanks,
Noah
Date: Tue, 14 Dec 2004 21:27:11 -0500
From: Noah Miller [EMAIL
Slow busy tells me that the telco has busied all his channels out --
likely
waiting for a call from him to finish provisioning... Unless his
dialplan has
_.,1,Busy or something. :-)
His pri show span said Down, that's likely the cause.
I guess I'll have to get Verizon on the case. Thanks for
Hi Dean -
Noah, what client were you using on your treo for this 600ms voip call?
Oh, I wasn't using a SIP client (is there one for palm?). Sorry if
that was misleading - this is just web browsing and email. Once the
connection gets going, it is able to do the 2.2 KB/s that standard GPRS
Hi Shoval -
I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible pci
slots?
I ended up getting a Dell PE1600SC. It is a tower computer with 6 PCI
slots (2 32-bit, 2 PCI-X, 2 64-bit), but it can be rack-mounted with a
special kit
Why ???
if your wanting the pci 4 port fxo/fxs boards you should be looking at
a t1 card channel bank, its less expensive far superior quality
Simply because I'm afraid of buying a used channel-bank on ebay.
Never seen one up close.
I'm quite certain I can handle installing 4 PCI cards in linux
I'd have to agree with the channel bank plan. You don't even need to
buy a used channel bank. You can get a new Rhino for $1500:
http://www.channelbanks.com/
I gather you've had experience with Rhino.
How does it work with Asterisk?
Does it provide all features? Caller-id, echo cancellation,
Sorry, wrong subject on previous post. D'Oh!
I'd have to agree with the channel bank plan. You don't even need to
buy a used channel bank. You can get a new Rhino for $1500:
http://www.channelbanks.com/
I gather you've had experience with Rhino.
How does it work with Asterisk?
Does it provide
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution
, but have had success with both SJPhone and
Xlite.
Thanks,
Noah
-Original Message-From: Noah Miller
[mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004
14:56To: Asterisk Users Mailing List - Non-Commercial DiscussionCc:
[EMAIL PROTECTED]Subject: Re:
[Asterisk-Users] Sip
Hi -
I've told lots of people about the Flash Operator Panel before, but
I've never actually used it myself. I've got it all set up nicely, but
I can't seem to authenticate to the asterisk manager (which is running
on the same box). When I set the op_server.pl to give debug messages,
it
Help.
Why is it that I can call out from my GSBudgetone SIP phone but the
audio is one-way'?
Why is it that when I call my asterisk phone number, I get a fast busy?
Can we have a looksie at your config files? sip.conf, extensions.conf,
zapata.conf, zaptel.conf.
If you start asterisk with enough
Hi Michael -
Thanks very much. See below. I do not have a zaptel.conf
I made the assumption you were using Digium hardware, sorry. What
device are you using for your incoming lines?
For the fast busy:
[incoming]
exten = 321XXX,1,Goto(incoming,s,1)
exten = s,1,Answer
exten =
Tim Jackson wrote:
They aren't dumb hubs, they are dot1q capable switches. I do not
understand why people are saying they need special POE cables for the
IP500. Mine came with a cable that injected power into the cable, and
from what I read, its Cisco and 802.3af compatible out of the box. My
So the only issue left I have is with this skinny not found when
0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
what's this skinny anyway?
Cisco's VoIP protocol, like SIP, or MGCP, but Cisco developed it
themselves, and it is the default
Hi,
What can it be when I can establish a connection between two
Softphones but no voice is transfered ?
thnx
Hugo,
It could be a codec problem, or many other things - can you provide
more detail? What softphone is it? What codec(s) are you trying to
use? If it's a SIP softphone, what's your
I am looking at placing a system in an office with a central
receptionist,
and phones for each individual employee thereafter. Could I use a
Snom 220
with additional keypads to view if the lines are in use by the other
employees?
Fred is in sales... A call comes into the receptionist and they
is not longer working for PLIVA
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 3:59 PM
To: Administrator
Subject: Re: MARIO SPOLJAR is not longer working for PLIVA
Please adjust your autoreply settings. Every time we post
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though,
that
the IP 300's don't support PoE even though their brochures say they
do.
Has anybody
What is PCI-X?
I usually use google to look up things like this. 6th search result:
http://www.webopedia.com/TERM/P/PCI_X.html
In practical terms, they are a longer PCI slot on the motherboard, and
I've always known them to be green in color.
___
Hi everyone,
Could someone tell me if I could make a Asterisk PBX + Digium
hardware with more than 20 FXS? If I have a 5 PCI PC, I could only
plug 5 TDM40B (4-port FXS) maximun. Is there a solution, a hardware in
Digium or something that let me have a PBX with about 50 or 100 FXS
and internal
We're considering using Asterisk in our small (8 user)
office. There is one feature that we have on our
current phone system that I haven't seen in the
documentation that I've read that I'd like to be able
to replicate with Asterisk.
On our current phones (Iwatsu) we have a button on the
phones
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though, that
the IP 300's don't support PoE even though their brochures say they do.
Has anybody
o I've tried this card in all three PCI slots but no luck
o I've tried two other TDM31Bs in a similar manner with no luck
o I've tried the same with a TDM22B and get similar behaviour
Could all my PCI slots be dead or is it likely that all 3 TDM31B
cards are
dead + the TDM22B? Any clues are
I bought a Wildcard TDM400P earlier this week.
I compiled the software from CVS and installed it. When ztcfg runs I
get
the error:
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
After checking /proc/pci I don't see the board. Why wouldn't the
board be
showing up? Its in a
I bought a Wildcard TDM400P earlier this week.
I compiled the software from CVS and installed it. When ztcfg runs I
get
the error:
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
After checking /proc/pci I don't see the board. Why wouldn't the
board be
showing up? Its in a
Message: 11
Date: Mon, 8 Nov 2004 11:17:03 +1000
From: JB Hewit [EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom 220 (or other phones) - line
apperances?
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII
Hi,
I've googled and searched the wiki
Essentially I wish to have buttons on a panel (like the Snom 220's
extension board) that show when people are on the phone or off the
phone for a receptionist.
As far as I know, you can't do this with asterisk, at least not
easily.
From what I've read, most people call this shared lines or
Message: 1
Date: Fri, 5 Nov 2004 09:31:27 -0500
From: Matthew Marlowe [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
work in CVS 11/03
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
What is a good multiline sip phone for an operator? Model and and
manufacturer.
The list that I came up with for multiple line/presence SIP phones is:
Snom 190, 200 (5 lines)
Snom 220 (Expandable number of lines)
Cisco 7940 (2 lines)
Cisco 7960 (6 lines)
Polycom IP 500 (3 lines)
Polycom IP 600 (6
Hi -
I think I might have seen this problem on the list before, so I'm sorry
if this is a duplicate, but I couldn't find it when searching through
the archive
I'm just setting up a new machine with asterisk. It's a RH9 box, and
I've tried the RC2 tarball, the 1.0 CVS and the 1.0 RPM's
Hi -
I'm wondering if any has experience with the Uniden UIP-200 phones.
The product info says that the 8 led buttons at the top are all
programmable. Can they be programmed as separate line appearances (ala
Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the
phone capable
Another promising candidate is Apple's dual G5 (PPC970) Xserve (a 1U
server).
http://www.apple.com/xserve
this one looks as if it might beat the price/performance ratio of a
high end Intel server.
The Apple G5 Xserv system has a PCI-X interface. Does anyone know
what that is and will a T405P or
, the Zultys 4x4's support 4, etc. I'm really
looking for a sub-$200 handsets that supports more than one
simultaneous call, and preferably 3 or more. Maybe they all do, but
I'm just unclear on it.
2. Is there a sub-$200 phone that also supports automated call parking?
Thanks,
Noah Miller
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