My fellow employees like their Polycom 600s even more.
Andrei (MPI)
Jonathan k. Creasy wrote:
I'll second that. I really like the provisioning features. My customers
prefer the 501 because they like the layout and speaker phone
functionality.
-Jonathan
-Original Message-
From
some
echo, but then it disappears. I just trained myself to ignore this first
seconds echo. =)
Please give the list information regarding the phone that you use with
*. This maybe a cheap phone problem, as well.
Andrei (MPI)
Carey O'Shea wrote:
I have a bad echo problem on my TDM400P with one
Erick,
Please see message: Paul Mahler: Asterisk Scalability at the following
link:
http://asteriskvoip.blogspot.com/2005_06_01_asteriskvoip_archive.html
Much slower machine than yours was involved in tests:
47 Simultaneous VoiceMail messages
333 Simultaneous SIP Calls
122 Pass through
Hi All,
Has anybody ever tried to use Sphinx and Festival from Ruby AGI scripts
(Ruby on Rails and AGI) ?
Please share your experience or even samples of code - that would be great.
Thank you,
Andrei (MPI)
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Gary,
I would check echo cancelling parameters first. I've seen this to happen
with one of the zaptel echo cancellers. Try to change the default echo
algorithm in zconfig.h, and recompile and install new zaptel. Also
zapata.conf echo parameters may need to be changed either way.
Andrei
Your server is more than enough for 24 SIP users. Depends a bit on usage
patterns, though, you should be fine.
Erick Perez wrote:
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP
Hello,
Please email me OFF THE LIST, if you had good experience with any of US
VOIP providers (bother termination and origination).
Please let me know. Thanks a lot.
Andrei
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Asterisk-Users
and
prevent anyone from touching the server
Please feel free to contact me off the list, but I dont know if I can
help more.
Andrei (MPI)
Enrico Pizzorno wrote:
We have an asterisk box with the following configuration:
- AMD Athlon XP 2400+
- 512 MB RAM
- SUSE Linux 10.1
- a Digium card TDM400P with 3
essential,
too.
Andrei (MPI)
Brian Swan wrote:
I've spent the last week or so troubleshooting echo problems at my
Wife's business, and I've been able to clear up about 99% of the echo,
but there is still a little residual echo that I can't seem to tweak
out. The users describe it as buzzing
to have a Windows PC, and you cannot have more than one call
on this PC
- no support for DTMF (Asterisk does not recognize digits you press no
matter what)
Skype-SIP latency is somewhat bearable.
Andrei (MPI)
Michael Graves wrote:
At least two such Skype gatewats already have been written. I
I had the same problem with my IAX terminaton provider, when tried to
use codecs other than ulaw (gsm and some others).
Changing back to ulaw fixed the problem. Hope this would help.
Andrei (MPI)
[EMAIL PROTECTED] wrote:
Hi,
I have an asterisk machine for which the calls reach it via
to a next available DID? And keep main DID number free all the time?
Please help!!
Andrei (MPI)
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Hi
If you have conference or 2-way calling (or whatever is that called by
telco), look for Flash application. Basically, you would need to flash
the line on incoming call, dial new external number with DTMF and
hangup. It will redirect the call:
exten = 52,1,Wait(1)
exten = 52,2,Flash
exten
to individual numbers.
In other words the first call may come in on channel 1, the second on
channel 2. They may or may not have dialed the same number.
Or perhaps I am misunderstanding something in your setup
On Jun 1, 2006, at 10:47 AM, Andrei (MPI) wrote:
Hello everyone,
I'm sure
izo wrote:
On Sat, 29 Jan 2005 11:13:36 -0500, Jim Van Meggelen wrote:
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
Their cards work. The only problem about govarion is their delivery
time. The cards are just not shipped as promised. And
Eric Wieling wrote:
Adam Robins wrote:
The TE410P is a T1/E1 card. I need the card for POTS lines. Is there
also a TDM410P that does not appear on the Digium web site?
The TDM400P only works in standard PCI 2.2. Not PCI-X, not
PCI-Extreme, not PCI-64bit.
For your information PCI-X is
this line appearance button is? I am just
pressing arrow down to select the new incoming call and then press
Answer softkey (which is rather annoying, one press would be more than
enough).
Andrei (MPI)
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Asterisk-Users
it. 103 isn't actually needed
either.
I disagree about hangup. I would recommend to leave Hangup there. As you
never know what could happen to an app (Voicemail in this case) -
it may be spelled wrong or just not configured right.
Andrei (MPI
. Usually
they would send it to you with no problem.
(You probably know that Polycom same as Cisco is not allowing direct
downloads from their website).
I am using 1.3.4 with no issues for a couple of months now.
Andrei (MPI)
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Greg Boehnlein wrote:
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading through the administration manual to try and solve this problem,
but I do not seem to be able to find the answers to my question. I figured
I would ask here and see if anyone has some
Michael Greb wrote:
On Mon, Jan 10, 2005 at 03:26:04PM -, Paul Brock wrote:
On Mon, Jan 10, 2005 at 15:18, Paradise Dove said:
On Mon, 10 Jan 2005 06:45:54 -0800 (PST), Jason Goecke
[EMAIL PROTECTED] wrote:
Hello,
Ever since I started using Asterisk I always get this
error:
Jan
clean you'll end up
with an asterisk box that acts retarded. So please before reporting a bug
do a fresh checkout or make clean and try again.
Also, do not forget to:
rm -rf /usr/lib/asterisk/modules
--
Andrei (MPI)
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Jason,
This problem may be happening because asterisk server and PC or hardware
phone clock are out of sync .
You need to find a way (e.g. ntp with atomic clock etc) to sync time up
to a second on all the devices involved in the network communication.
Andrei
Jason Goecke wrote:
Hello,
I was
it the right way.
--
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Tim,
For what it's worth, from my working sip.conf for Polycoms:
[2010]
type=friend
username=usr2010
callerid=MyName 2010
secret=nobodyknowswhatitis
host=dynamic
dtmfmode=inband
context=admin
defaultip=192.168.1.10
progressinband=no
Notes:
dtmfmode=inband and progressinband=no - that seems to be
Tim Jackson wrote:
Copied your sip.conf and changed the settings and I'm getting the exact
same error. I'm also running 1.3.4 of the SIP app for the IP500.
Someone has already pointed out that you might have ran into a network
problem. What's the network setup between phone and the server?
Michael Graves wrote:
That might work out where you do your deployments. In Verizon
territory, you can get analog business lines with unlimited long
distance and no metered minutes for about $37 a month. A BRI costs you
about double that for the loop, with metered minutes and bring your own
LD.
Hello All,
Sorry if it is known problem.
I have tried to get Asterisk latest from CVS and found out that my
sip.conf is not good for that: registration from all SIP phones -
Polycoms - failed as if all passwords were not right.
Had to go back to earlier version (1_0_stable from early December).
You guys probably don't know what Digium did recently to address TDM400
problem:
- they've sent new FXO modules to all customers who were complaining
about TDM/FXO issues.
What I've heard from a Didigum reseller/supplier it might be a situation
with specific telco lines here in US.
The new
Richard Scobie wrote:
It is a simple one liner.
...
Index: wctdm.c
...
+ reset_spi(wc,card);
...
This is exact same patch that Digium support tried before sending me new
fxo modules. That wctdm.c patch did not help in my case.
Andrei
___
to re-qualify the phone
during the middle of a call. Also do you have Busydetect enabled? I had
to disable that setting in my Zapata.conf.
Jared Armstrong
-Original Message-
From: Andrei (MPI) [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 22, 2004 7:09 PM
To: Asterisk Users Mailing
Short answer: try new FXO modules or a new card.
I've struggled with this for about a month. I've returned one TDM400
card, got a new one. Had same problems, Digium support installed a patch
for zaptel, no difference. Then I diagnosed one FXO was dead. Got a
replacement for that FXO.
Jon,
Yes, I have tried that. The problem does moves with the particular phone.
Andrei
Jon Radon wrote:
Try swapping the working phone and the non working phone. See if the
problem moves with the phone.
On Thu, 23 Dec 2004 10:52:27 -0500, Andrei (MPI)
[EMAIL PROTECTED] wrote:
Hi Jared,
Thank
Andrew Kohlsmith wrote:
On December 23, 2004 02:31 pm, Steven Critchfield wrote:
While I agree, I also must point out
- TDM400P - this card seems to be the #1 source of problems. I believe
the FXO module issues are solved but the FXS issues are still around.
Hopefully the same fix works.
richard wrote:
Hi,
I have the following scenario:
We currently have 1 incoming line, that 2 POT phones plug into, and
when we have an incoming call, both phones ring. Is it possible to
have Asterisks in parallel, so that when the 2 POT phones ring, I can
have a Voip phone, which is plugged in
This is working in my phone:
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe=299 msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=8 /
/msg
299 is both * extension and mailbox. 8 is special extension to dial from
local phone to check voicemail.
You may want to check
Hi there,
We are using 10+ Polycom SP IP 600 phones with Asterisk and TMD400P with
4 FXO lines.
So far we have 3 phones with following problem: more or less frequently,
for every call or ever other call, user of the phone would hear brief
interruptions on the line when dialing out via PSTN,
Gary wrote:
On Sun, 19 Dec 2004 12:52:40 +, w fm3 wrote:
Hi
I am struggling with hardware choices to get started with. My options are
narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
of importance is:
- functionality / integration with asterisk
- headset functionality
Carey Pillar wrote:
I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with
what seems to be correct settings (according to digium and asterisk
wiki).
As soon as I plug in my POTS line into FXO mod the line goes into
offhook
state (whether I have power to the card or not).
Eric Wieling aka ManxPower wrote:
Andrei (MPI) wrote:
I have same problem with TDM400P FXO. Had to reload wctdm 3 times in
a row tonight, in order to get rid of this annoying problem.
There is a hope that developers will fix the zaptel drivers sometime.
I have experienced this before
Peter Svensson wrote:
...a scenario when person calls in via PSTN via a Zap channel and
listens to IVR menu of Asterisk. Then (s)he presses an extension # and
then this call gets redirected to an extenal telephone number outside of
Asterisk. And the call to Asterisk is ended. Or I am dreaming
Ryan Courtnage wrote:
On Fri, 2004-10-12 at 10:29 -0500, Andrei (MPI) wrote:
Eric Wieling aka ManxPower wrote:
Andrei (MPI) wrote:
I have same problem with TDM400P FXO. Had to reload wctdm 3 times in
a row tonight, in order to get rid of this annoying problem.
We too have
Jayson Vantuyl wrote:
On Fri, Dec 10, 2004 at 10:18:17AM -0500, David Cook wrote:
With regards to the IRQ sharing situation on 400P/X100P cards how would
I know if I can use IO-APIC?
I am running RHEL 3 on a Dell PowerEdge 1400SC. RHEL installs without
IO-APIC support. Is this because RH is
Adam,
I can send you 1.3.4 firmware. Please let me know if you can accept zip
archive of 10Mb to your email address.
But please consider that you may have to upgrade to 1.3.1 first (with
2.5.0 boot rom), if you have not done so. The 1.3.1 files are available
here:
James,
Could you please advise me about any not so expensive brand/model of
headsets compatible with Polycom SP IP500-600 ??? I am looking to buy 5
headsets right now.
Thank you,
Andrei
James Milne wrote:
The Polycom Soundpoint IP500, is a great phone. We supply these to our
resellers and have
Hello,
Sorry if it is a different subject for this thread, but...
Is it possible to re-route incoming call on Zap channel of TDM400 FXO
card to completely different and external telelephone number via some
magic telephone command or signal? So, the Asterisk Zap channel would be
cleared off of
I have same problem with TDM400P FXO. Had to reload wctdm 3 times in a
row tonight, in order to get rid of this annoying problem.
There is a hope that developers will fix the zaptel drivers sometime.
Andrei
ps. I predict you will continue to do that, because the whole platform
is great. :)
m.
Leif Madsen wrote:
On Wed, 08 Dec 2004 13:44:10 -0700, Michael Welter [EMAIL PROTECTED] wrote:
I went on a service call yesterday snip
I fear the day when going to fix an Asterisk system is much like going
to fix a residential computer for $15/hr :)
[OT on]
Well, try $60-80$/hour..
[OT
My problem with TDM400 was bad FXO module. Now got a replacement from
Digium. So far it works fine.
I continue to have occasional loud noise when dialing out, even after
Didigum support installed a patch for wctdm driver on my server.
I have scheduled restarts of Asterisk and reload of
Hello,
I would be grateful if anybody could tell me what I should tell Verizon
in NJ so they would enable disconnect supervision for my lines.
Apparently remote hangup notification or disconnect supervision or
calling party control is NOT the magic phrase for them. Although
disconnect
Peter Osborne wrote:
On Tuesday 07 December 2004 12:34, Steven Critchfield wrote:
On Tue, 2004-12-07 at 11:00 -0500, Peter Osborne wrote:
Hello all,
We've been using our Asterisk system live for about a month now and I'm
looking to tuning a few things. First, is echo, I receive a fair
Jon Radon wrote:
Sorry this doesn't answer your question. Any reason to not leave them as wav's?
On Tue, 7 Dec 2004 10:42:58 +0100, Matthew Oulton
[EMAIL PROTECTED] wrote:
Currently I am creating .wav files and then converting them via SOX to .au
file format, then running them through a gsm
Richard Scobie wrote:
Rich Adamson wrote:
The tdm card does have some unusual issues that appear to be driver
oriented, but there are lots of folks using the card in production.
Usually in situations where the client knows how to and tolerates
having to reload drivers and/or reboot his PBX
Hi Samudra,
It is known behaviour for TDM400P card Rev E/F with FXO module.
Apparently it is a driver issue, which is yet to be fixed (confirmed
with Digium support).
I had this happening in about 6-8 hours after normal work, all channels
would become unusable. No re-wiring of physical lines
Rich Adamson wrote:
Inline...
snip
Rich,
Thank you for your answer. Now I've figured that one of the FXO modules
on the card may be defective. Whenever I plug in telco line in it - that
line will be like shortened (if you pick up parallel telephone, the dial
tone will be heard weaker than
I got 1.3.4 already. No major changes, works smoothly so far.
Matthew Marlowe wrote:
Im working on getting 1.3.4.. Will post.
On Thu, 02 Dec 2004 13:15:29 -0500, Andrei (MPI)
[EMAIL PROTECTED] wrote:
I would sniff UDP packets with tcpdump and see what is going on:
separately in 10.24.102
Hi,
I am preparing to roll out Asterisk setup with TDM400P, 4 FXO modules in
a small office. Asterisk will replace legacy system (4 telco lines, 8
extensions PBX), but before the new system and ip phones would be
installed, the legacy system is still in use. The four telco lines are
now
Rich Adamson wrote:
...
Thought that was really funny since the 500's were ordered from a reseller
with SIP image, and the reseller never even bothered to include a
CD or url.
...
Yes, that's a fact. You were lucky he knew what exactly he was selling.
I could tell more horror stories as I was
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrei
(MPI)
Sent: Wednesday, December 01, 2004 11:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
Tim,
You may see description of new 1.3.4 firmware
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrei
(MPI)
Sent: Thursday, December 02, 2004 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
Hello Tim,
You are saying that: phone is on 10.24.102.0/24
Rich Adamson wrote:
I've successfuly converted 7940 from call manager firmware version 3 to
SIP 7.3. just last week. You need to upgrade to firmware version 6
first, then upgrade to 7.
Also once you've upgraded the phone, you should remove firmware config
file from tftp server, otherwise the
Hi Chris,
First of all, you need to configure ftp or tftp and watch syslog
closely - what the phone is looking for at boot time. You would need to
put config files into (t)ftp directory, named according to MAC address
of you phone. XML and Web is really weird - they do not even share same
Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Polycom IP500
Any idea if 1.34 makes Daylight Savings work for us people in Australia?
PaulH
-Original Message-
From: Andrei (MPI) [mailto:[EMAIL PROTECTED]
Sent: Thursday, 2 December 2004 9:30 AM
To: Asterisk Users Mailing
Mike,
I've successfuly converted 7940 from call manager firmware version 3 to
SIP 7.3. just last week. You need to upgrade to firmware version 6
first, then upgrade to 7.
Also once you've upgraded the phone, you should remove firmware config
file from tftp server, otherwise the phone would be
David Boyd wrote:
On Wed, 2004-11-24 at 04:14, Mike Dent wrote:
Hi,
I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and
1 SIP phone.
I've noticed some horrible buzz/rasping type of sounds! These seem to occur when
* is trying to play back some audio or sound to me?
E.g. If I
Guys,
If anybody interested, removing USB support from Debian kernel (2.4.27)
solved the problem.
Thank you for your help.
Andrei
Andrei (MPI) wrote:
...In about 6-8 hours, when I would try to dial out from any
extension through Zap/1 ... Zap/4 (I have group #1 defined) I would
hear
Matt,
If you have PCI-X slots - those are backward compatible (like on my
1750). You should be able to use Digium cards in any of them. Please let
us know about your experience though after you try them.
Andrei
Matthew Boehm wrote:
I have a Dell Poweredge 6450, 4 proc Xenon with 1Gb ram and the
Hi Patrick,
Patrick wrote:
On Wed, 2004-11-17 at 14:44 -0500, Andrei (MPI) wrote:
Hello All,
Please let me know if you have any comments or suggestions how to
resolve the following situation. My configuration is:
- Dell PowerEdge 1750 (standard), one CPU Xeon 3.0GHz
- TDM400P with 4 FXO
Digium
cards, you would have to try and see.
Sincerely,
Andrei
Matthew Boehm wrote:
What is PCI-X?
Matthew
- Original Message -
From: Andrei (MPI) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 10:55 AM
Hello All,
Please let me know if you have any comments or suggestions how to
resolve the following situation. My configuration is:
- Dell PowerEdge 1750 (standard), one CPU Xeon 3.0GHz
- TDM400P with 4 FXO, connected to 4 regular phone lines (PSTN)
- Asterisk 1.0.0 from cvs
- Zaptel and Zapata
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