Title: Message
Hi,
Does
anyone know the solution to this issue?
Regards,Stojan
Sljivic
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan
Sljivic - GDSSent: Friday, June 10, 2005 13:21To:
'Asterisk Users Mailing List -
Is there any way to accomplish the following? (searched and searched and
can not find any examples)
In extensions.conf (text file) define a macro that accepts a handful of
arguments
From realtime mysql (extensions) - call the macro with arguments (where
the macro is static in the text file)
If
Hi!
I have a problem with one box running asterisk, one pots line and an
X100P. Almost every night the phones give 2-3 rings and then stop. There
are no actual incoming calls, I verified by putting a device that lists
the incoming telephone numbers parallell to the X100p and it doesn't
list
Hi,
when i use the *8 for the call pickup the call i fetch is directly
connected and i can't see the callers number.
What i want is that the call in the first only rings at my phone and in the
second i can see the callers number before i am connected.
I am using a polycom 500 ip phone. Is this
On Fri, 10 Jun 2005, Peter Svensson wrote:
On Fri, 10 Jun 2005, James Bean wrote:
Peter seems to be on the ball more then me about these phones as
grandstream gave me the standard replies, Peter do you know for sure if
grandstream have a timetable for the function led's cause I need to
Hi-
I want to learn asterisk code and its archetecture where can i get help.
--- Kib Eki [EMAIL PROTECTED] wrote:
Hi,
when i use the *8 for the call pickup the call i fetch is directly
connected and i can't see the callers number.
What i want is that the call in the first only rings at my
Good morning!
Asterisk 1.0.7 runs fine on my machine with Suse 9.3 when using a
downloaded tarball. But as I wanted to have a look at Realtime I
decided to download everything again via CVS with
# cvs checkout zaptel libpri asterisk
and install it.
Unfortunately though the Asterisk installation
When the inbound leg of the all is SIP and the outbound leg is Oh323
(Voip-to-Voip only here), the DTMF relay (either RFC2833 or SIP Info), fails
to go through, while it works perfectly when both legs of the call are SIP.
Is this a shortcoming of the Asterisk core or the Oh323 channel? Is this
Title: Message
Hi
Chee,
We are
experiencing the same issue.
Did
you find a solution for this and can you please share it with
us?
Regards,
Stojan
Sljivic
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I have the same on calls originating from a sip phone and going into a ZAP channel.Andre- Oorspronkelijk Bericht -Onderwerp:
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Title: 407 Proxy Authentication Required
I am getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network using sjphone snom.
How to overcome this..!!
Pls advice..!
Shahan
This e-mail may contain confidential and/or
Ibrar Ahmed wrote:
Hi-
I want to learn asterisk code and its archetecture where can i get help.
:)
You could try the psychiatrist. Or maybe just a local support group.
:)
Jokes aside, some good resources are:
www.voip-info.org
www.asteriskdocs.org
my news (www.sineapps.com/news.php)
IRC
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
I've this infrastructure:
|voip services| -- |*| -- |cme| -- |isdn|
the voip services are logged on my *, then forwarded to number 601 on
cme. The isdn calls too are forwarded to 601. On cme I've a timeout X
for call-forward noan (no
Hi,
I'm using *CVS Head
version and read the dialplan from MySQL.
I'm making A-Z
termination to over 4000 different country and city codes.I have 3
different dialing rules depending on the price level of the dialed
number.
Should my extensions
table contain 4000 lines? Is this realistic?
Hi,
I am using a number of snom190 phones, and an asterisk gateway
server, and recently started experimenting with call transfers. The
snom phones provide support for attended and un-attended call
transfer, so I would rather use that than call-parking.
I have found that un-attended transfer
Hi
I am planning to try to use Asterisk for testing
purpose , for PBX systems , I have one PC with RHEL
4
I want to buy digium cards for this
purpose , but I am not sure about which card to buy ,
this field is totally new to me , requesting guidance
for selecting the card
On Sun, Jun 12, 2005 at 06:10:53AM -0400, Michael Di Martino wrote:
However i still get the same error. Please help we cannot connect call
form my norstar to asterisk w/ it dropping in 10 seconds.
Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of
span 2
Jun 12
I just read about the partnership but was wondering what is actually going
to happen? Is asterisk going to be bundled with cepstral voices for free :)?
Or whats the deal?
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I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I
set the caller id correctly in my perl AGI script
$AGI-set_callerid($ani); , the gateway does not see any caller id coming
from my Asterisk box. I use the very latest version of Oh323 as published in
the Inaccess web site,
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
I never noticed any problems.. so I can't comment :) hehe
On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
Finally got a response from voipjet support and they say they have
switched to a new provider for US termination. I have yet to test
this out as I have not had a chance to build them back into
[EMAIL PROTECTED] will
not be able to configure polycom500 phones.
You need to add this entry in sip.conf manually with
one additional line as under:
progressinband=no
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
MillerSent: Monday, June 13, 2005 9:44
hallo all,
could sombody please help me,
i dont know why nativ bridging is not working when i choose the ilbc codec,
with speex it is working,??
iaxcomm (ilbc) ---asterisk -- ( asterisk2 -- sip grandstream (alaw) )
\-native bridge--/
1. if i use on
Hello,
I would like to ask, if there's presence support in Asterisk and how
to make it work with
Xten's Eyebeam client. I tried searching all the possible
documentation, google, but I found only a note, that there's a module
in SER, that supports the feature. Is there also support in asterisk?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Monday, June 13, 2005 6:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)
Hi,
I am using a number
Title: Message
You may need to look to see if you are
using peer or friend in the sip config for this phone. We needed to change ours
to friend to make it work for us with still using secret.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS
Title: Message
Greetings,
You have stumbled on to
one of the most troublesome flag for newbies;
autocreatepeer.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer
in your sip.conf file add a
line in the [general] section autocreatepeer=no
Now people can
Hi Juraj,
I have been trying for some time to fund video conferencing support and
have offered a personal bounty of several thousands of dollars in order
to get it developed.
So far 5 people have contacted me but apart from one point to point
solution I'm still waiting.
In the interim I have
It seems that the wiki pages at www.voip-info.org are not responding,
and this has happened before. Responds to ping but not http requests.
Is there a session limit on the web site? Is it too low? Maybe another
explanantion?
Anyone else notice?
___
Also check out the CISCO GKTMP API, that is their gatekeeper api. There
might be some cool stuff you might like to know.
Race the tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simone
Sent: Sunday, June 12, 2005 2:13 PM
To: Asterisk
Also subscribe to the asterisk-dev mail list. Watch it for a couple of
days before you ask a question or they will eat your lunch.
Pick a single thing you want to change in the PBX, and then learn how to
code for that. Something really simple like adding a parameter to a conf
file is a good place
Hello,
What would be the simplest and
the cheapest solution to get an Asterisk server working with 9 to 17 POTS?
Because for 1-8 POTS we are
using 1 or 2 Digium TDM cards and past 17 POTS in our
area it is economic to use a PRI.
We are looking for a hardware
solution on our side
Are they running on a windows server? :)=)
Maybe it has the Monday Morning Blues. (I can't get it to talk either.)
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Monday, June 13, 2005 11:05 AM
To: asterisk-users@lists.digium.com
Andrew Kohlsmith wrote:
On Saturday 11 June 2005 19:51, Lee Howard wrote:
I don't think that lack of mindshare completely defines the reasons
behind Asterisk fork failures. It places all of the blame on the
forkers. I think the truth, though, is that they not only fail due to
lack of
Hi,
Please forgive my terminology, still a bit new to T1s and such.
I'm looking for a way to have 5 T1s from a carrier terminate into some type
of box (multiplexer?), then be able to plug 7 asterisk servers into that box
(each with single port T1 card) and be able to have 2 * servers go down at
Hello.
I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use
in an Asterisk PBX/call center environment.
One feature the SPA-841 has, which I can't figure out how to implement
on the snom 190, is the make/accept calls without registration
feature. Or more specifically, produce
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List -
Just use a cisco with 5 T1 ports and have everything over IP use ultra
monkey to load balance your asterisk boxes. I have found this works
very well.
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
An accurately CDR it depends, i think, the way you want to make the
bills. So its a combination of using NoCDR, ForkCDR commands and
billsec, disposition and other database fields manipulation. For
example, IAX calls may be recorded with very few time of duration if
you dont use the parameter
On Monday 13 June 2005 16:42, alan wrote:
Hello.
I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use
in an Asterisk PBX/call center environment.
How about tackling this with iptables and matching specific IP addresses on
specific MAC addresses?
Cheers,
Gavin.
Hi Carlos. I have never used H323. But im interested in your problem.
Have you tried to use de 'sip debug' 'iax2 debug' commands? and check
the console with a high verbosity level? could you post any warning or
relevant output when the call is made?
best regards
On 6/11/05, Carlos Alberto Lara
I'm considering switching my incoming phones lines from standard analog
to a T-1 service from XO communications. They propose to bring in an
IAD which has 12 lines of voice and 768k of internet bandwidth as part
of a package deal. Since I want to keep the voice traffic in the digital
domain
It sounds like your looking for a t1 protection switch with will do what you
want It can switch t1's for failover or loss of signal on a t1. These are
usualy rather expensive and Might work 100% in your example because they
will only switch over if th actuall t1 goes down. So If your server
Twas an issue with the card. I tried a different TDM20B and it worked
perfectly. RMA time.
Whenever I load wcfxs I get ProSLIC 3210 version 2 is too old.
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Race Vanderdecken wrote:
Also subscribe to the asterisk-dev mail list. Watch it for a couple of
days before you ask a question or they will eat your lunch.
Or even more likely, eat you for lunch!
:D
--
Cheers,
Matt Riddell
___
Hello all,
After much googling I have come to the conclusion that in asterisk land
DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are
used rather interchangeably. If this is an incorrect assumption Please
correct me. Based on this assumption if I have everthing set up
I'm trying to setup SIP trunking between 2 asterisk servers. Eventually
there may be up to 5 servers linked together depending on the growth
needed. I have IAX2 trunking working, but I want both.
For simplicity, I have named the two servers, alpha and beta. Extension
7100 is a Polycom
Title: 407 Proxy Authentication Required
We
also have the same problem over long latency networks ATA also gives
Call Rejected: 407. We have tried a lot of different phones and soft
phones and the only one working is Xten.
In
any case this is apparently only problem with newer
I have had experience with both the Vina and with XO. If you ask for
it, you should be able to get an Adtran 600 series on the circuit. I
never had any success with the Vina and it really is not a piece of
equipment I would bet the farm on. They may have improved but I would
still just as fo
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes:
Protecting freedoms by putting limits on (thus restricting
freedoms). Interesting concept.
I need to repeat here. The gpl's purpose is to protect the freedoms
that comes with free software. So, you have only the freedoms that
comes
This is a very interesting converation, but it seems like the BIZ forum
might be more appropriate...
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Monday, June
DID number is the number commonly assigned to a PSTN trunk.
DNIS and DID may be the same. DNIS refers to the Dialed Number that is
passed as signaling with the call (or on ss7). Most calls have ANI and
DNIS.
Your extensions look ok, assuming that the carrier sends the digits that
match.
What
On Monday 13 June 2005 12:06, Matt Riddell wrote:
Race Vanderdecken wrote:
Also subscribe to the asterisk-dev mail list. Watch it for a couple of
days before you ask a question or they will eat your lunch.
Or even more likely, eat you for lunch!
:D
Phew! I thought lunches was going to
On Monday 13 June 2005 12:38, The VoIP Connection wrote:
This is a very interesting converation, but it seems like the BIZ forum
might be more appropriate...
How on earth is this a business-related discussion? -dev would have been my
guess. :-)
-A.
The box that you are talking about sounds a lot like a DACS. You might
google around on that term to see if any might have automatic failover. A
DACS can be reconfigured to cross-connect various DS0s on the fly -
although, no matter how fast the switchover, the carrier will always see
that
Gavin Hamill [EMAIL PROTECTED] wrote:
On Monday 13 June 2005 16:42, alan wrote:
I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use
in an Asterisk PBX/call center environment.
snipped
enforce SIP channel IP restrictions in Asterisk without
host=ipaddr, or get the
As a follow-up to my previous post where I stated the IAD would be a
Vina model, after some more prodding to XO, they have told me it will
either be an Adtran TA-600 or a CAC Adit 600. These products are covered
pretty well on the web and I have manuals on both. So, if those
knowledgeable
You will be able to purchase Cepstral voices from Digium just like you
dor for G729 already. I would guess it's 1 way to show the power of
asterisk by putting all the TTS orders thru a company such as Digium.
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AdTran can come in either flavor depending on the modules they install.
It can dump analog lines or it can be fully digital and split off voice
T.
I would recommend the digital domain for sure.
Get yourself a Digium T1 card and keep everything digital.
Get a block of DIDs (20 is the norm for
Andrew Kohlsmith [EMAIL PROTECTED] writes:
ABE is a VERY SPECIFIC version of HEAD (or is it STABLE?) with
features CUT OUT and nothing added that isn't in HEAD already.
This is what I mean with a custom set of features. I never claimed
anything was added.
I totally fail to see the problem
I can't load module wcfxs or wcfxo with modprobe command, I don't have
any error message, and when i try to start zaptel I've the error below
when :
ZT_CHANCONFIG failed on channel 4: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
as the error says did you forget how to configure FXO and FXS
interfaces signalling?
post your configs if in doubt...
best regards
On 6/13/05, Bouchra Benyelloul [EMAIL PROTECTED] wrote:
I can't load module wcfxs or wcfxo with modprobe command, I don't have
any error message, and when i
On Mon, 2005-06-13 at 18:20 +0200, Esben Stien wrote:
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes:
Protecting freedoms by putting limits on (thus restricting
freedoms). Interesting concept.
I need to repeat here. The gpl's purpose is to protect the freedoms
that comes
Hi,
I though some of you on this list might be interested in what Inveneo
is doing in Uganda. We are a San Francisco based non-profit
organization that builds rugged, low-cost, highly reliable and open-
source communications systems for under-served communities around the
world. We have
Use Adtran Atlas 800.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of qrss
Sent: June 13, 2005 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 multiplexer (or ?) for failover in large
installation
In bug 0002863 a patch is mentioned that sends hiss every 20 seconds, does anyone know who wrote this or where it is available at?
Scott England
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Hello,
I know that I can have DID on a single line, but will AMP support 2+ lines
with DID?
Has anyone tried this? Straight forward?
Thank you,
Tomas
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quote who=trixter http://www.0xdecafbad.com;
Protecting freedoms by putting limits on (thus restricting freedoms).
Interesting concept.
It maybe an interesting concept, but it is absolutely true.
True anarchy (no rules what so ever) cannot exist.
Your freedom to kill me would impose on my
Mark,
This is a wonderful thing to do for underserved societies like Uganda.
The datasheet you have provided and the layout could be the model for
many other developing societies both In Africa as well as central and
South America.
Kudos to Inveneo.org under your able leadership. Keep up the
When I start Asterisk, I receive these errors:
Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on
user 'gv_trunk' without zaptel timing
Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on
peer 'gv_trunk' without zaptel timing
Jun 13 16:26:05
-Original Message-
From: Iassen Hristov [mailto:[EMAIL PROTECTED]
Does this matter? All we are saying is that Exchange supports
IMAP and we
would use IMAP as the protocol to delete the message from the user's
mailbox. How does the user access his mailbox is his choice.
I think two
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail and MS Exchange
Synchronization
On 6/10/05, Dean Collins [EMAIL PROTECTED] wrote:
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
On Monday 13 June 2005 12:38, The VoIP Connection wrote:
This is a very interesting converation, but it seems like
the BIZ forum
might be more appropriate...
How on earth is this a business-related discussion?
-Original Message-
From: Esben Stien [mailto:[EMAIL PROTECTED]
The other problem is the issue that free software developers are
mostly (in my experience) not happy with the fact that their code
would be used in proprietary software. It conflicts with the whole
religion of free
A small simple non Dell system and T1/E1 card. If you can even think
about 17 lines then start with a PRI. Most PRIs can be ordered as
small as 4 lines voice and 768 data that leaves ~8 voice channels
open.
I can give you system specs off list as the change often.
On 6/13/05, Ken Dresdell [EMAIL
When I try to fing group/call pickup command in www.voip-info.org and
made a search like
*8
I got an error message.
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Tuesday, 14 June 2005 3:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Digium Website Update:
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096] in an earlier
Yeah, if you get the Microsoft Partners Newsletter emails they reported
the 75 GB expansion today.
Increased Storage Limit in Exchange Server Standard Edition
Get more out of mission-critical email. In the fall of 2005 the storage
limit for Exchange Server 2003 Standard Edition will increase
Title: MCI vs. XO/Allegiance
Hello All,
Anyone out there using ISDN PRI from either MCI or XO/Allegiance?
Gotta make the choice today and the difference per month is only about $25 in favor of MCI.
Billing is pretty much the same between the two so I have pretty much no point of
Make it there if you can!
PaulH
On Fri, 2005-06-10 at 13:16 +1000, jurgen wrote:
Hi all,
If you're in Melbourne Australia and interested in Asterisk, you're
invited to join us for the second in an irregularly scheduled casual
evening to talk about Asterisk, VOIP, networks, and just
Wiley Siler wrote:
Anyone out there using ISDN PRI from either MCI or XO/Allegiance?
We have a DS-3 full of PRI from X/O. They work great, mostly, but their
tech support sucks. They screw up number ports all the time and about
every week there is some local number I can't dial to via XO which
I'm using an XO pri, and as long as you never change anything on the pri
XO's not bad. Our experience is that if you chance anything on the PRI
configuration they'll screw it up somehow (YMMV). One thing we have
learned is that XO doesn't monitor our voice circuits, so if one of our
PRI's
I prefer MCI since we use their pri and internet. MCI's support is very pro.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Coulson
Sent: June 13, 2005 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
You should use DHCP to enforce IP address to MAC binding when the phones
boot.
And then let the phones register and use host access (deny/permit)
permissions in peer section to restrict by IP address/mask.
alan wrote:
Gavin Hamill [EMAIL PROTECTED] wrote:
On Monday 13 June 2005
Well, the fact that two negatives for XO and a positive for MCI all came
at once says a lot to me.
Interestingly enough their SLA reads...
* 24/7/365 Network Monitoring and Service. If for some reason your
network is having problems, the chances are XO will know about it before
you do and
Let me throw another complaint against XO on the table. They actually shut
off the wrong T1 and they transferred all of the DIDs to the T1 they shut
off! how screwed up is that? We are now about 2 years later and their
billing department still calls us every month for nonpayment of the T1 that
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a shared
path on sdsl.
The s
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg from * to sip and vice versa.
#1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as
return
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
The only technical difference between a T1 and SDSL is how it's physically
delivered to the customer, what usually happens is that
Are you sure? Everything I have seen says SDSL = Full Duplex.
That being achieved by dropping the pair that provided voice and using
it for signalling.
Where ADSL utilizes unoccupied frequencies and averts conflict with
analog voice frequencies, SDSL takes over the whole line. SDSL
eliminates
Damon,
I have no idea where you are getting your information from, but what
you said makes no sense.
DSL based lines, be it ADSL or SDSL, are based upon a connection
technology in the ATM
family. As a result, the upstream and downstream of the connection can
be controlled seperately.
If
Marcelo Pacheco wrote:
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or
Title: MCI vs. XO/Allegiance
we have been using XO/Allegiance for over 3 years
and have had no problems. I can't compare to MCI but we also had a sprint
t1 that we had to get remove due to them being bad in billing and also not very
reliable for faxing.
- Original Message -
Nir Simionovich wrote:
Now, E1 and T1 lines are based upon a channel based connection, which
means you get a line
with X number of data lines and a single control/signalling line. On T1
it means that you have 23
lines dedicated for Voice/Data (each is 64kbps) and a single signaling
line
Title: MCI vs. XO/Allegiance
Sprint nevermore... I switched voer to Sprint a few
years ago and they literally dropped service form under us.
It was during that Sprint ION fiasco. They sold to
me, installed, and literally terminated the service10 days
later.
The whole time they were working
BTW - The Speakeasy SDSL connection I originally posted about is
delivered via Covad.
The SLA (some of it at least)
Average Network Delivery and Delay2 - Further proof that Covad has
confidence in the performance of our network.
Delivery - 99.9% successful delivery of all data packets sent from
Hi David,
You are correct, I always get those 2 confused. Thanks for the clearing.
Nir S
David Coulson wrote:
Nir Simionovich wrote:
Now, E1 and T1 lines are based upon a channel based connection, which
means you get a line
with X number of data lines and a single control/signalling
Not really true about T1 description. When you apply for T1, you need tell
vendor if it's channelized or non-ch. If you are going to use it for 1.5M
network, you need use unchannelized T1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Leon Sun wrote:
Not really true about T1 description. When you apply for T1, you need tell
vendor if it's channelized or non-ch. If you are going to use it for 1.5M
network, you need use unchannelized T1.
T1 is T1. How you use the DS0s delivered across it is up to you. You can
mux them out
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