Hello,
I want to ask that if thee are some ATA decives that i can use to
connect
mutliple analog phone lines to my VOIP system..
I mean for example an ATA device with 24 ports with 24 independent
SIP
accounts.
For example for some dormitories in my area, i want to put an ATA
On Mar 19, 2009, at 9:05 AM, Ken D'Ambrosio wrote:
Hey, all. I'm all over MWI, but I gotta say that I think the
Polycoms go
a bit over the top. The blinking LED is enough for me; how do I
disable
the stuttered dialtone and the audible warble? I've looked through
the
config files,
On Feb 17, 2009, at 1:20 PM, David Gibbons wrote:
snip
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our
tests will
tell!
/snip
We
Sorry about off-topic, but can you advise the mail client who is
able to organise the web mailing list topic as web interface does ?
(i mean by blocks/topics) I wold be glad to use something else with
the same usability, but dont see any alternative.
Thank you
Just turn on threading
You will have a hard time finding a 24 port POE without fans - too
high of a power density. Do you really need 24 ports? perhaps a 12x12
otherwise multiple 8 or 12 port models may work
Do let us know if you find a 24 port without fans.
On Jan 31, 2009, at 2:06 PM, Claus Herwig wrote:
Instead you could always get a SIP/IAX provider.
On Jan 27, 2009, at 11:56 AM, Jon Pounder wrote:
Michael Higgins wrote:
At least here in Canada - DSL just seems to have killed BRI - you
practically have to know the secret handshake to even be allowed to
provision one any more. It killed
In dahdi_tool, there are three more indicators of error:
IRQ misses
Bipolar violation
CRC error
As I understand it now, these should be error counters and they
provide
additional information in case of RED alarm state.
Actually you need not be in RED alarm to have these. Just know that
On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote:
Hi all,
Suposing that 2 SIP phone register at a remote (internet)
asterisk, what is the best way, if any, to make the RTP traffic go
phone to phone, whithout using the internet conection (asterisk)?
Allow reinvite? Assuming both
On Dec 22, 2008, at 10:38 PM, Martin Lima wrote:
On Thursday 18 December 2008, Justin Phelps wrote:
I've been struggling with an ongoing problem the last month.
Here is the layout of the wiring:
T1 from ISP DiTech Echo Cancel device Voice Channel-Bank
(same) T1 from ISP (same) DiTech
Simple. A PRI can easily have multiple trunk groups. They just assign
chan 1-22 to trunk group 1. Chan 23 to trunk group 2. D to chan 24. As
an example, adjust to suit your needs.
On Dec 16, 2008, at 9:27 AM, Andrew Thomas wrote:
I can only assume it's a T1 thing - as E1's tend not to have
http://www.networkworld.com/news/2008/120608-fbi-criminals-auto-dialing-with-hacked.html?Inform=nlnetht=rn_120808nladname=120808dailynewsamal
Criminals are taking advantage of a bug in the Asterisk Internet
telephony system that lets them pump out thousands of scam phone calls
in an hour,
On Oct 29, 2008, at 9:19 AM, Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating
voice and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
After spending a couple hours scanning for an open source (non-
commercial) billing package yesterday I am underwhelmed. Almost all of
the packages listed on the WIKI appear to be defunct, for several
years now. I will be happy to get a login and edit them out if that is
the proper method
On Oct 29, 2008, at 12:30 PM, David Gibbons wrote:
Fair enough, I guess I was concentrating on this line in Jerry's
message :)
The only reason I can think of not to is to eliminate the cost of
the second cable.
I believe you're mistaken about the QOS though.
QoS is not required on
handled carelessly as Jerry Jones
suggested. But this is not a compelling argument to me in any but
the most critical scenarios such as public-safety applications, etc.
or you wish to eliminate service runs - that is unless they are always
billable and your customers do not mind you
On Oct 27, 2008, at 3:01 PM, Andrew Kohlsmith (lists) wrote:
On October 27, 2008 02:01:43 pm Jeff LaCoursiere wrote:
Speaking of fring, I just got my brand new iphone 3G. Anyone have any
comments on how well fring or any other sip client (siphon?) works on
iphone?
I do not like fring.
On Oct 23, 2008, at 3:10 PM, John Cheng wrote:
Maybe I just haven't thought of the right google search terms -- but
is
there a website/guide out there that will help me understand the
output
from pri debug span?
___
perhaps this might be helpful?
Q.931
On Oct 19, 2008, at 1:21 AM, Alex Balashov wrote:
Stephen Reese wrote:
Does the latency remain more or less the same regardless of the
bandwidth load on the pipe?
If so, TOS bits (what you refer to as QoS) won't help you. You've
either got network issues (very likely if you have an intra-
On Oct 17, 2008, at 5:14 PM, Paul Douglas Franklin wrote:
When off site, our IP phones lose contact after a few minutes of
inactivity. They no longer receive calls, though they can call out.
Asterisk acts as if it is ringing the phone, but the phone does not
ring.
The phones are behind a
The times they are a changing - or something like that.
while gb on phones is not the norm today, it s becoming more so on the
higher end flavors and will continue to do so
since the life span of your switches will be several years, thinking
ahead is a good thing
my only concern is having
On Oct 6, 2008, at 1:53 PM, Steve Anness wrote:
I know I have asked about this before, but I thought that I would
ask again with some more detail and maybe someone will have an
idea. This is my first time to be setting up an asterisk server and
I have a server running. I sent Linksys
Google works
enter this along with your search string
site:lists.digium.com your.search.string.here
dont type the
On Sep 29, 2008, at 2:42 PM, Brian Webster wrote:
What is the best-recommended resource for searching archives of this
mailing list?
Thanks for your time
On Sep 29, 2008, at 9:55 AM, Yehavi Bourvine wrote:
Try AudioCodes MP-124 which is 24 ports FXS. I have one but haven't
used it much yet, so I cannot comment about its quiality.
\
Sorry, cant agree with this, tried a couple and replaced with channel
banks.
When the cards hears the fax tone it should auto disable the ec.
On Sep 2, 2008, at 9:42 PM, Octavio Ruiz wrote:
On Tue, Sep 2, 2008 at 6:16 PM, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my
On Jul 28, 2008, at 5:50 PM, Jason Parker wrote:
Philipp Kempgen wrote:
I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ).
screen doesn't solve the security aspect of your question though.
Grüße,
Philipp Kempgen
Actually, it could. What I've done before, is give out an
On Jul 16, 2008, at 3:11 AM, Femi wrote:
If you can get a machine at the other end of the link you could use
the
Mikrotik bandwidth tester
You can find it here - http://www.mikrotik.com/download.html
Femi
or just run iperf on each end
http://sourceforge.net/projects/iperf
On Jun 22, 2008, at 6:51 PM, Kevin P. Fleming wrote:
Jim Duda wrote:
My Telco service is Verizon FIOS. I know that MWI is working,
because
if I pick up an analog phone set attached to the line, I can hear the
stutter tone.
The MWI detection in chan_zap/chan_dahdi is not for stutter
On Jun 9, 2008, at 2:29 PM, Lyndon Griffin wrote:
Apologies - I know this isn't either Polycom or ISC support, but if
anyone would have an answer to my problem, I'm certain they would
be on
this list.
I'm experiencing odd behavior with Polycom handsets obtaining DHCP
addresses. It
On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote:
On the Linksys side, we have a load of SRW-224P switches out in
the wild powering 24 Snom 370s (around 7W each) off each switch.
Likewise, we sell these things by the bucket load and have no problems
powering phones from all 24 ports.
On Jun 5, 2008, at 5:08 PM, Bill Michaelson wrote:
I'm considering using a PoE switch like this...
http://www.tigerdirect.com/applications/SearchTools/item-
details.asp?EdpNo=3023334CatId=2800
...to power as many as 24 Polycom phones of varied kinds.
The sales lit indicates 190 watts
And you are using g.711 so the sounds are passing correctly and not
being distorted? Try calling a person and pressing digits to verify
they are inband during call?
On May 5, 2008, at 4:31 PM, Jason Wolfe wrote:
Yes, and I verified watching the output that it was reading the
new .conf
On Apr 2, 2008, at 9:22 PM, Al lists wrote:
Bad memories from AudioCodec :)
Second this.
My favorite is Vega, but they have terrible support in US.
Have many Adit600 connected via Digium T1 - work great. Even FAX if
PSTN PRI connected to same card.
And no the Adit600 is not a switch,
What does your digitmap on your phone look like? This is what
controls sending the call to * when it recognizes a complete dial
pattern. The phone does not send digit by digit. If it is waiting for
you to press send, then it does not recognize your pattern.
On Mar 26, 2008, at 8:18 AM,
The Polycom will display a different icon if DND
On Mar 12, 2008, at 2:04 AM, Lee, John (Sydney) wrote:
Special dialplans for reception are entirely up to you. The only
reason
reception phones have different dialplans to normal extensions is
that
often people want the receptionist's
On Jan 7, 2008, at 6:19 PM, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Norman Franke wrote:
Greetings!
We have a somewhat noisy background in our call center, and I'd
like to
reduce this. Obviously, we could plaster the walls with sound
absorbing
material,
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:
I need the digit map to call China. Example number:
011-86-10-6887-
011-International (obvious)
86 is country code (China)
10 is city code (Beijing)
Last 8 digits are the number.
I tried using 011xxx.T but it always asks
[incoming]
exten = 2125551211,1,GoTo(companyA,1)
exten = 2125551212,1,GoTo(companyB,1)
exten = 2125551213,1,GoTo(companyC,1)
[companyA]
exten = 2000,1,Dial()
[companyB]
exten = 2000,1,Dial()
[companyC]
exten = 2000,1,Dial()
On Dec 13, 2007, at 5:53 PM, Diego Andrés Asenjo González wrote:
On Dec 10, 2007, at 7:45 AM, Michael Melia Jr. wrote:
I haven't found outcall that confusing though I do agree that a TAPI
Driver that makes use of the available outlook call functions will
make
for the easiest, most streamlined user experience.
I also agree that these convenience and
I will miss them. It was nice having a local company with a few
Polycoms in stock most of the time. A month or so ago we needed some
quick and were unable to contact them, either through their toll free
or local numbers. I swung by their office last week and nocticed it
was vacant.
On
How about 20+ on a Qwest DSL modem hitting our server? Works great.
On Sep 12, 2007, at 7:23 AM, Dovid B wrote:
Eric,
Try 5 polycoms behind the same NAT router. Let me know when you
grab a drink
;)
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To:
On Sep 11, 2007, at 7:29 AM, Eric ManxPower Wieling wrote:
Rizwan Hisham wrote:
well he does not have access to hi sip settings, so he cant edit the
host=differentIP every time he moves or registers from anyother
place.
Actually he should be able to register from anywhere in the world
On Aug 9, 2007, at 9:37 AM, Erik Anderson wrote:
On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote:
I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up. The bchannels
are all up and the T1 is not in alarm status. The
On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote:
All,
It's been some time since this thread was alive but we are now seeing
some progress in this project. Which I will document.
We have ordered a T1 for the new building which we are moving (We are
getting 14 channels of the T1.) and
On Jun 28, 2007, at 8:00 AM, pixiesfr wrote:
hello,
We looking for a channel bank to connect 120 analogs phones, in SIP to
an Asterisk ..
Did someone have some product in mind.
A channel bank must connect via a T1 by definition, which would then
give you 24 phone lines per T1. This
You do not need an L3 switch for this, just any managed switch which
does vlans
Unless there is something else?
On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to
You can add their gateway blade to convert to voip via ethernet, but
it only does mgcp.
How about doing GR303 to an access navigator with channel banks
hanging off that? Pricey but carrier class gear and scales WAY up.
Could also do Adtran total Access concentrator (4303?) feeding their
When turning of dhcp, dont forget to set all other attributes
manually. Ones that would effect this are
IP Address
Subnet mask
Gateway
boot method tftp/ftp
Server Address
username/password if ftp
vlan
Assuming you are setting a hard IP for the server, if using a url
then donot forget to add
A simple glance at their website will tell you this about the 501
G.711 μ/A and G.729A (Annex B) configuration
On May 2, 2007, at 12:22 PM, Jaswinder Singh wrote:
Try ilbc if the phone supports (free) or g729 ( better but your
asterisk will need license if you want to transcode calls
The only reboot issue I have with 1 sidecar is the side car deciding
to randonly rebbot, not the phone itself
Perhaps upgrading to 2.1 will help?
On Apr 24, 2007, at 10:51 AM, J French wrote:
I have a Polycom 601 with 3 expansion modules running 2.0.3. We
have Buddywatch set up on around
On Apr 12, 2007, at 1:49 PM, Kevin P. Fleming wrote:
Got off the phone with Polycom on this I have the same
problem with
my new 601 phone here (haven't seen the problem on the 650).
I am using an IP650 with the latest firmware (and the corresponding
sip.cfg file) and I have seen this
Hmm - just received an email from these guys last week. I know
nothing about them.
On Apr 15, 2007, at 9:23 PM, cb wrote:
On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote:
When a call comes in I want to ring an extension that happens to
be loud speaker. The users can the press *8
I have actually noticed it on my personal 601 after upgrading past
1.6.7 to 2.0 and 2.1
Yes it is still doing this and is very annoying. Hopefully Polycom
will fix by next release.
On Apr 11, 2007, at 4:33 PM, Noah Miller wrote:
Hi Mike -
Somebody was helpful enough to give me the
It has nothing to do with actually dialing. Even trying to press end
call or the speakerphone button does not work at times.
Have tried removing side cars etc, but definately seems to be a bug
in the 2.x code stream.
On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote:
Jim King
The reasoning behind all of this is that I want to ring desk phones
and then if they don't answer, I want to ring cell phones. If I
ring the cell phones too long, someone's voicemail will pick up,
which I don't want. So if I set it up where they have to ack it, I
can ring the cell
On Mar 6, 2007, at 1:55 AM, Tomislav Parcina wrote:
Kristian Kielhofner wrote:
Hey everyone,
I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.
Hi Kristian!
Thank you for your work. I'm not
We had an issue, and I know others had posted the same on the list.
Scenario:
Polycom phone user sets call forward to a toll free number(in our case)
Call arrives for the phone, the phone notifys asterisk, asterisk
dials new number.
Telco drops call. But if you dial direct to the number it
Not sure about others, but on Polycoms a blind transfer sends
original callerid, screened sends operators callerid
On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator
that.
Rob
Jerry Jones wrote:
Not sure about others, but on Polycoms a blind transfer sends
original
callerid, screened sends operators callerid
On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:
I'm sure this was asked before, but I can't seem to make this
work...
If a customer dials one
This is a common issue with large inbound call center operations.
They like to cheat. They actually start sending prompts to the caller
without actually signalling their carrier that they have answered the
line. Typically they do not answer until a phone is ringing or you
are in a queue. I
OK I need some help. Looking for comparisons for a large customer
wishing to provide voip service over a region. We are up against
Metaswitch who is claiming they can do anything Asterisk can do. I do
not have too much information on Metaswitch so am looking for any
information, preferably
On Jan 24, 2007, at 10:20 AM, Thinselin, Vincent wrote:
Hello,
I'm trying to make my asterisk box to act as a telco, in order to
reproduce a US environment in europe.
Our telco provider is giving us those settings:
ESF
B8ZF
Inbound = EM Immediate
Outbound sig =Wink Start
Yield to Glare =
analog station ports = fxs
analog line ports = fxo, assuming 2 wire loop start
On Jan 18, 2007, at 8:26 PM, Erick Perez wrote:
Thanks Jerry. Are the avaya station ports a special type ?
On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote:
Connect to the avaya line ports, not station ports
Connect to the avaya line ports, not station ports.
On Jan 18, 2007, at 10:46 AM, Erick Perez wrote:
Hi, this is a signalling question:
I have a 4port fxs-to-sip where i connect standard analog phones. I
want to connect this device to an avaya PBX and then the device talks
to asterisk via
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives
between the first and second ring, if you have * dial after the first
ring it will not be there yet to pass along
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem
I suspect any 24port will have a fan. The Netgear FSM7326P are not
too bad and we have had good luck with them.
ps - I also load their open source software.
On Jan 3, 2007, at 4:51 PM, John French wrote:
I have an upcoming install which places the switch close to some
employees in a quiet
add a wait before you dial the sip phone, keep in mind the callerid
information arrives later than the call setup info
On Dec 31, 2006, at 4:38 PM, David Sampson wrote:
For some reason something that seems like it should be simple is
leaving me a bit perplexed. I am receiving incoming
Or web into the phone and click any submit button - not a great idea
though if you remotely provision, just make sure you do not change
any settings as they will then over ride the remote file settings
On Dec 19, 2006, at 1:09 AM, Douglas Garstang wrote:
From the Asterisk console:
sip
OK with the remote server on one side doing G729, what will you be
connecting to on the other side? If it does G729 then no license, if
not then one license per active call. Also if * will be doing any
voicemail etc then you will also need the license.
On Dec 19, 2006, at 8:31 AM, Michel
Or any of a number of gateways that do this. Off the top of my head
you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix,
Adtran, and others.
Just try to be very careful as they all have their strengths and
weaknesses and you need to evaluate how they would fit your needs.
Best
Use an empty line key to monitor the other phone
On Dec 7, 2006, at 1:44 PM, Bill Gibbs wrote:
Figures I email this and realized I can hit
Menu
1 (Features)
4 (Presence)
2 (Buddy Status)
Wow that’s a lot of key strokes. Anyway to reduce that to a one
button touch? I don’t mind
Greetings,
I have a customer with an old PBX which cannot accept a PRI.
Has anyone tried/tested connecting a CAS T1 to provide 2way DID
trunks to a pbx? Either directly to an * server or a gateway?
Thanks
Jerry
___
--Bandwidth and Colocation
you can change the configs to have multiple beeps, and adjust the
timing of them, but when we tried the problem then is the beep is not
added to the incoming audio, but replaces it, so you lose the far end
speaking, went back to default.
On Nov 29, 2006, at 3:34 PM, Dovid B wrote:
Hi
Intertex
Not cheap, licensed per number of users
But seem to work great and have some nifty tools
very confusing picking models though
On Nov 5, 2006, at 3:54 PM, Erick Perez wrote:
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
--
zttool is your friend here
red is LOS or no signal coming in
On Oct 26, 2006, at 3:54 AM, Florian Hars wrote:
I have a TE205P, jumpered for E1, added the missing wct4xxp-line
to /etc/modprobe.d/zaptel, zaptel.conf is just
span=1,1,0,ccs,hdb3,crc4,yellow
span=2,2,0,ccs,hdb3,crc4,yellow
You will also perceive jitter as echo
If any links are getting busy and routers or switches have to buffer
you will hear what sounds like echo, not to mention if you have a
high packet loss also
Of course jitter would have to be above 100ms or so to be noticeable
as far as acoustic echo,
We use almost all Polycoms, several hundred
had one way audio with 1.6.4 or 5, forget which
1.66 and 2.01 seem to be ok
We did have a few phones (2-3) that had random one way for a long
time, replaced everything feeding them and it still happend. A month
ago I replaced the phones and have
Sounds like they are rebooting. Is power being interrupted at night?
On Oct 13, 2006, at 9:40 AM, Mike Garey wrote:
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
Resellers claim it will ship in December or there abouts
Uses g.722
About $30 more than 601
On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote:
Has anyone used the Polycom HDvoice phone yet? I am curious if it
uses a different codec. Does it actually sound any better?
Enable buddy watch in your poly config files
also set each speed dial to have this enabled also
On Oct 9, 2006, at 12:04 AM, Doug wrote:
Hey Folks,
Been wrestling with the 601 and the expansion module. Finally
figured out how to populate both with speed dial entries. Also
hints are
We are trying a couple of the Intertex - seems to work so far
On Sep 29, 2006, at 2:59 PM, Andrew Joakimsen wrote:
The VoIP version of DD_WRT runs Ser by default
On 9/24/06, David Gagnon [EMAIL PROTECTED] wrote:
You could take a WRTSL54gs, install openwrt then openser
David
-Message
There have been many threads regarding specific uses for digitmaps.
One of the most common is for the telephone to perform digit
substitution and prepend some digits. Never thought this was possible
until I found a reference in a Sipura tech note .
Anyway hope this helps someone.
Add
Had problems the first night I downloaded and installed, but tracked
to very poor net conditions. Reloaded this week and all has been
working fine. Nice to finally be able to use all the buttons on the
sidecar for blf:)
It may be my imagination, but it also seems that it is staying in
set group/check group
On Sep 21, 2006, at 8:22 AM, Benny Amorsen wrote:
ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes:
ZZ Why don't you simply give them separate extensions and put them in
ZZ a ring group.
I'm not quite sure what you mean by ring group. Perhaps you could
elaborate?
ZZ Or
: Jerry Jones [mailto:[EMAIL PROTECTED]
Sent: Mon 9/18/2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom Expansion Module
Poly 2.0.1 says it can do 48
On Sep 17
the digitmap only tells the phone when to send the digits it has
collected. They have no digit substitution feature. This would be
done within your * dialplan
On Sep 19, 2006, at 7:57 AM, Jordan Novak wrote:
This is really starting to get to me. I have deleted this field in
the phones
We tested a couple 9133i, dont remember the specifics right now but
we stopped as there was some inconsistency in provisioning. I was
very optimistic as I like the look and feel. We did deploy a couple
480iCT which worked very well - when they worked. But they keep
locking up and freezzing
Poly 2.0.1 says it can do 48
On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote:
As far as I know, it's 12.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Sun 9/17/2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial
Do not know of a card that does. But think a digium T1 to a channel
bank (ie Adit600) would.
On Sep 1, 2006, at 2:06 PM, Tim Sharp wrote:
I am looking at CTPX's VP2000 product. I haven't tried it yet.
Please let me know if you find a solution that works.
Tim
-Original Message-
We used some way back (a year ago) when they first came out. Had
several issues which they were very helpful in working with us on.
They resolved many, had to upgrade and load patches. Unfortunately
they were lacking a couple features we required so they have been
replaced.
Give tech
On Aug 30, 2006, at 2:58 PM, Mike wrote:
Hi,
I have a few questions on the Polycom 501. I am using latest
firmware.
1) When I press the Call List button (on the left row of
buttons), I get the call lists (as expected). When I press the
Directory button, I get the choice between
Such an objective question. Everyone, including different users will
have a different answer.
Is this within an enterprise? at home? with a paid service? what
codec? pure IP or TDM mix?
I would say anything over 200 is bad, now how close you get to that.
We try to engineer our on net
Manually config to point to your boot server, which should have a
good copy of the software and it should go get it. If not sniff the
traffic in/out and see what it IS doing.
I have had several firmware updates get interrupted in the past
corrupting the image and this has always worked.
rumor has it Sangoma will be releasing their ct3 card in a couple months
do no transcoding or EC and one server can handle a large quantity of
T1s
On Aug 16, 2006, at 8:52 PM, Matt Florell wrote:
Use multiple servers. What kind of calls are you handling that you can
have more than 3 quad
Been awhile but IF memory serves...
Manually enter the boot server IP on the phone. I do not think causes
a reboot - of course this was several versions back in sofware.
Then edit a contact and press save. Every time it updates the list on
the phone, it tries to copy to the boot server.
If you see no errors on your MX2800 for the ds3 then they are
probably not the issue.
What does the MX2800 show for T1 which do not work? If you loop
toward * does the card see itself? Loop toward GX do they see?
On Aug 4, 2006, at 11:15 AM, Steve Totaro wrote:
I have a DS3/T3 that was
up to speed on the MX2800 but I have gone to
loopback tests and loop T1 25-28 and selected every possible
selection while watching pri debug span 1 on the console, no output
at all.
Jerry Jones wrote:
If you see no errors on your MX2800 for the ds3 then they are
probably not the issue
probably need a crossed t1 cable
1-4
2-5
On Aug 4, 2006, at 4:20 PM, James Arscott wrote:
Hi, this is my first post, so go easy on me !
Sorry if this has been covered before, I could not find an answer that
helped me.
I am trying to achieve the following :
Telco ISDN30e PRI - Asterisk
You will get a call waiting beep. One. However you can change the
config file and have multiple beeps. You can also change the beep
'sound'. However you must also be aware that while the phone is
playing the beep(s), you are not hearing the far end of the call.
On Aug 3, 2006, at 9:21 AM,
It has been several years since I had to address similar situations,
but I used TUT Systems devices back then. worked great. There are
several DSL variants which should work ok.
On Jul 27, 2006, at 6:02 PM, Manrique Feoli wrote:
another thought, if you are in a bowl, all you need to find
I really like the IP60x phones. Have started using the IP430, so far
after 20 or so they are fine.
But the IP30x and 50x I refuse to use.
The aastra 480i is also good.
The 9133i has promise.
I do not like the snoms - any.
Grandstream are so so
Budgetone is not bad for the price, but not
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