I have setup my system to give a company announcement if
somebody calls, ...
I would like to avoid these announcements, if the caller is
known by the system.
Each caller I would like to put into a database with name.
Now we know them!
If we know them, we do not announcement.
Is
Carlos,
Change one line to prepend es to the filename, (ie es/cerrado). See if
that works. It may be a simple fix where the language support is broken as far
as paths go.
If is works please report it as a bug on the Bug Tracker. http:/bugs.digium.com
Please include as much info as
When I place a call on my softphone to a external number the call is
placed, when I click transfer, dial internal extrention (e.g. 202)
then hit transfer again, the call is transfered to the 202 extention
fine.
However, when the other way Internal call comes in, extension 201
answers, and
-Original Message-
I have a Uniden UIP200 behind a NAT and an * server behind another NAT.
I am able to register with * and place calls. However, once the call is
established, I cannot hear anything from either end (UIP200 as well as
the called destination). Then, I did the exact same
Doesn't work that way, you have to know the exact channel that you want to
Monitor on SIP. What you can do is a Command Show Channels in the manager
and parse through the output to find the first channel then do your Monitor
command.
MATT---
-Original Message-
From: Dana Olson
Do you have the FXO version of the channel bank or the FXS?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Goscomb
Sent: Wednesday, April 20, 2005 8:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Rhino Channel Bank
Hi
I have
Looks like you have sip.conf set up to expect registrations for tycisco
since it has a D for dynamic.
You can either set up the 7960 to register with asterisk and use something
like this in sip.conf:
[tycisco]
type=friend
username= someusername
secret= somesecret
insecure=no
mailbox=757
Anyone know any asterisk compatible ADSI phones available for sale in
the UK?
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You will need to make sure that Transfer option in the Dialplan
i.e.
exten = 301,1,Macro(stdexten,301,${NATE})|Ttr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Goodyear
Sent: Wednesday, April 20, 2005 9:26 AM
To:
On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote:
On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com
said:
as a whole. I enjoy cheap computers, if it were not for microsoft
creating windows, making computers easier to use for everyone, the mass
production and
It is actually a different animal because you're not using a softphone
etc at all, give it a try on the site to see what I mean.
http://signate.com/callme.php
It actually calls you on a pstn number the proceeds to connect you to
a staff member. This is why I mentioned the potential for abuse. It
When I started with Asterisk, I registered with iaxtel. As I remember, I
got a number, but forgot it!
I also cannot find a listing of all others who are registered in iaxtel.com.
Can anybody enlighten me, please?
bye
Ronald
___
Asterisk-Users mailing
I am having an issue with the phone system
recognizing keys in an active call. examples being when i call the extension of
VoiceMailMain() it does not recognize the numbers that i put in (eg. mailbox and
password) if i call from an internal line. However, if i call from an outside
line
Alright, thanks you guys. I was hoping to not have to do that, but I
guess it's time to get my PHP on. I find myself re-learning it every
time I start a new project. I love the language, I'm just forgetful.
:-)
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There are a several people providing that service. The first time the
user invokes the service (clicks on the web link), he will have to download
the corresponding sip (or other) phone component. Here is where it gets
difficult because it would have to be either a public domain component or
The SPA-841 doesn't seem to have conference call feature. This is
extremely important.
- Daniel
On Apr 20, 2005, at 11:12 AM, Kerry Garrison wrote:
I currently use an SPA-841 on my desk and don't have any problems with
it
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
Hi,
I solved my last problem that was about receive calls.
Now I have another one, that's after a end a phone the zap extension
stay unavailable, until a restart on 1 minute.
Does anybody know what could be it?
Tkz,
Reiner
___
Asterisk-Users
This is an update to the message title LINE Noise HELP! Regarding the static
that floods the line and makes continuing a conversation impossible. This
was happening on my SIPura-841, but Damian on the list was experience this
same problem with POTS phones as well through his TDM400. Below is an
Hi
We had the same problem with the Invalid Data error. We solved it by turning
silence suppression off on the handset (Swissvoice). Try looking for similar
settings on your equipment.
Cheers
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
From your descriptions of your needs, you would be better served with an
AAH installation. Easier to understand than hand coding your contexts.
That aside, here are few answers...
Look here for more...
www.voip-info.org
Routing to the VoIP is just a matter of dial plan matching (see dial
I've done that...I think. :^)
Here's the excerpt from sip.conf:
[tycisco]
type=friend
username=cisco1
secret=***
qualify=200 ; Qualify peer is no more than 200ms away
nat=yes
;insecure=no
host=dynamic; This device registers with us
It is a kludge but should work:
Action: Command
command: show channels
Then sort based upon the result and you should have the two final
variables you need.
SIP/8000-? And Zap/??
You can then Monitor the SIP channel or just grab the Zap. I would go
with the former as you could then
The 841 is lacking in programmable buttons, it is an entry level phone. All
additional features have to be accessed via access codes. For example, to
transfer a call, dial #, voicemail dial *98, etc. The Zultys phones have
programmable buttons for those features.
-Kerry
-Original
On Mon, 18 Apr 2005, Joe Dennick wrote:
You can use the same six lines for both inbound and outbound calling
just like you do now. The 'roll-over' will start on line 1 and move up.
You'll have to configure your outbound calls to start on line 6 and move
down. If you ever get to the point
We'd appreciate it if people don't try it just to try it, since we have to
answer the calls. Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moody
Sent: Wednesday, April 20, 2005 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re:
There are a several people providing that service. The first time the
user invokes the service (clicks on the web link), he will have to download
the corresponding sip (or other) phone component. Here is where it gets
difficult because it would have to be either a public domain component or
make sure that you have a hangup command eg.
.
exten = s,2,Dial(SIP/101,30,t)
exten = s,3,Hangup
it would help if you put your extension.conf and zapata.conf file on the
email so that someone can tell you more conclusivly what needs to be done
- Original Message -
From: Roberto Reiner
Inband DTMF is not supposed to work on compressed voice. And
you can get a X86_64 optimised g729 from digium.
On 4/20/05, Marcin Kwiatkowski [EMAIL PROTECTED] wrote:
Ronald Wiplinger napisa(a):
I would like to install G723.1 and G729 on an Athlon 64.
I looked at
On Sun, 17 Apr 2005, Dave Weis wrote:
On Sun, 17 Apr 2005, Greg Boehnlein wrote:
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have
found
the best solution for all of them lies in a cylindrical receptacle that
sits
I run a system with 2 cards with no issues related that I can tell..
Per the whole interrupt conver, 5 sounds like way too many.
Wouldn't something like this work though?
http://www.voicetronix.com.au/vpb4_v4pci.htm
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I recently
updated my sip.conf and extensions.conf files and after
shutting
down asterisk and restarting it (asterisk -cvvv)
it shows and
empty dialplan (show dialplan)
*CLI
show dialplan-= 0 extensions (0 priorities) in 0 contexts.
=-
What could
cause somthing like this
below is a
Hi All,
Does anyone have experience using any of the fxs -- fxo converter?
Any configuration on asterisk side ?
I bought a fxs -- fxo converter and try to use it but failed.
Any one please help ?
thanks ,
Stephen
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Hi,
I just installed the asterisk and the X100P card. I can receive calls from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following
On Wed, Apr 20, 2005 at 09:01:56AM -0700, trixter http://www.0xdecafbad.com
said:
On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote:
On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com
said:
as a whole. I enjoy cheap computers, if it were not for microsoft
Hello,
I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't
get them to register. The annoying thing is that I've taken the phones to 3
other locations with non-Linksys NAT routers and the phones work immediately
without any problems.
I've tried STUN, outgoing proxy .
Are there any BYOD providers out that that people have had positive
experiences with? I have broadvoice and they suck lately. Anyony have
anyone with a good amount of peers and not a lot of downtime?
--
Michael Lyszczek
New York, NY, 10282
NEW EMAIL : [EMAIL PROTECTED]
Does anyone have any experience with this phone? I'm considering
purchasing it but wish to hear if anyone has any experience with it.
Thanks,
Daniel
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Hi,
I notice when you call digium and choose the option for tech support,
it asks you to enter
your reference number, it then looks up the job?
Does anybody know what they are using to do this?
I'd like to do this with asterisk and Request Tracker, maybe digium
are already doing somehting
I have a error when try to compile de chan_unicall.c with Asterisk. Others
modules like spandsp, libsupertone, libunicall libmfc2 is sucessfully
compiled (using --prefix=/usr) with the last version avaliable at
ftp.soft-switch.org
I am using the stable_version from CVS, with zaptel libpri...
You have 2 problems.
Zap/g2 is one. You are trying to dial out a non-existent Zaptel group.
Change your TRUNK variable to Zap/1-1 (or just Zap/1 will do).
Also, you are stripping the 1st number off your outgoing call. If you don't
want to do this, then change TRUNKMSD to 0.
- Original
Try searching the list.
There's a thread from a few weeks back of exactly the same name.
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 21, 2005 4:23 AM
Subject:
Michael Lyszczek wrote:
I have broadvoice and they suck lately.
Can you elaborate?
- Dan
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On Monday 18 April 2005 11:23 pm, Joe Dennick wrote:
You probably can NOT use the existing Meridian phones because they are
digital phone sets, not standard analog ones. You can purchase 5
TDM400P cards (assuming you have 5 available PCI slots in your Asterisk
Server), and configure two with
Weiming,
At the Asterisk CLI the Show Version command will print a string
similar to the following:
Asterisk CVS-v1-0-04/14/05-13:17:05 built by [EMAIL PROTECTED] on an
i686 running Linux
At the Linux command line, asterisk -V will print a string similar to
the following:
Asterisk
Ok, well, much to my dismay I just had a call go south on me. It seems to
only happen when it's business. I can talk personal all night and not have
the problem. It must be the black cloud that lingers over my head. Either
way, I am so sick of this. I hope someone can figure out what the hell is
Hi,
I just installed the asterisk and the X100P card. I can receive calls from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following
Hi,
I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:
[PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188] - Siemens 8825 (Analog)
When we
You need to do this on intel chipset. You can not do it on AMD.
I guess digium has it.
Thanks
Quoting Ronald Wiplinger [EMAIL PROTECTED]:
I would like to install G723.1 and G729 on an Athlon 64.
I looked at http://readytechnology.co.uk but I could not get a clue how
to compile / get all
Does anyone know how to use the spa 3000 pstn with amp (as a trunk)?
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Hi,
I hope this will help. I will give you my configuration. I'm using nikotel
to make international calls thru
My SIP provider.
Sip.conf
[general]
port=5060
bindaddr=10.0.0.10
disallow=all
allow=g726
allow=alaw
reinvite=no
register = myid:[EMAIL PROTECTED]/myid
[2000]
type=friend
Wow! What a great fight!
Let me egg you guys on.
Furthermore, (if you knew your history) MS had been doing funny
things with DOS / and windows to make it difficult for other windowing
systems and DOS clones to work with MS-DOS / Windows, further cementing
their market dominance.
As someone who
Hey,
I just asked the same question a few hours ago :-)
I too am interested in the phone.. It looks nice.. I did send their tech
support staff a question about the phone, and received an answer already..
I looked through the manual on-line, and wasn't able to determine if it did
CallerID with
Jon Lewis wrote:
On Mon, 18 Apr 2005, Joe Dennick wrote:
You can use the same six lines for both inbound and outbound calling
just like you do now. The 'roll-over' will start on line 1 and move up.
You'll have to configure your outbound calls to start on line 6 and move
down.
Alright, so what does this (now mangled)thread have to do with
Asterisk again?
- Original Message -
From:
Walt
Reed
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, April 20, 2005 2:20
PM
Subject: Re: [Asterisk-Users] US$200
Well, I guess that I'm not as good as I once was... If anyone would
care to assist me in this, I would appreciate it. If you want to
contact me even on IM, IRC or off-list, anything's cool with me... I
would really appreciate the help.
--
Dana
___
List Members,
I am involved in the process of designing a large Asterisk setup for a
call center. A graphical overview of our tentative design can be found
here:
http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif
Originally, we planned to implement this design by purchasing one
Hi,
My users use sip phones (grandstream 286 / 486).
No echo between sip calls (g729 too).
Calling the 'world' though an h323
VoIP provider, I have a very high echo level.
(I do not have this problem calling
through sip)
The connectivity to this partner is rather
good:
No
On Wed, 20 Apr 2005, Daniel Salama wrote:
Every once in a while I read messages about people having problems with
certain models of SIP phones, some of them being well known models.
I'm interested in purchasing new SIP phones for my office and wanted to
know which brand/model is most
i have one of these phones at my office, and its set up and working with *.
very easy to set up.
i'm quite dissapointed though in the sound quality. people on the other end
can hear me with no problems, but my end the quality isn't so great. some
static, clicking, etc.
also to me the phone
On Sun, 17 Apr 2005, Dave Weis wrote:
On Sun, 17 Apr 2005, Greg Boehnlein wrote:
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have
found
the best solution for all of them lies in a cylindrical receptacle
that sits
beside
Thanks for your kind help, I understand ip precedence and that's ok. I also
found on Snom phones how to mark 802.1p ( which is what I need now ). On
the 3Com 3300 802.1p is enabled and correctly priorized . The only thing I
miss
is how to tell asterisk to originate rtp packet marked with 802.1p
I put 0.0.0.0 on the bind_addr line so all interfaces (private lan and
internet) are bound. Not sure if you are using a multihomed box, but
just thought I'd point it out in case you were :)
Chuck Smith wrote:
Its funny as soon as I sent this I looked at my sccp.conf file and saw that
the
On Wed, 20 Apr 2005 18:33:44 +
Jaime Blanco [EMAIL PROTECTED] wrote:
Hi,
I just installed the asterisk and the X100P card. I can
receive calls from PSTN and it can ring on a Grandstream
SIP Phone. From the SIP Phone I can dial the demo
extension on asterisk pbx. The issue is as soon as
Anyone know of a Lucent EMRS PRI Card? Know where to get one? Ours went
dead.
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I seem to recall an ocx applet that someone is working on that will
enable IE only (afaik nothing else will run ocx applets) make a SIP
call. Maybe this was on pulver.com but I thought it was on sf.net.
In effect this could be used to do basically what you want. A java
applet would be more
Hi,
I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:
[PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188] - Siemens 8825 (Analog)
When we
There are plenty on the wiki...
Are there any BYOD providers out that that people have had positive
experiences with? I have broadvoice and they suck lately. Anyony have
anyone with a good amount of peers and not a lot of downtime?
--
Michael Lyszczek
New York, NY, 10282
NEW EMAIL :
I have unlimited local calling on my cell phone provider but not long
distance; so I wanted to create authentication based on me calling in and
authenticating based on the callerid of my cell phone.
Here is what I tried based on the wiki:
exten = s,1,answer
exten =
That is very interesting stuff!!!
I've experienced, and still continue to experience, the line noise issue,
as well as the beeping issue.. The beeping issue took one of the people I
was talking to by surprise, he thought the conversation was being recorded..
I have 3 X101P cards for my 3 inbound
Chuck:
I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED]
If you post your config files (sccp.conf, SEPX.cnf, etc), I can
have a look at them for any suggestions.
-Andy
On 4/19/05, Chuck Smith [EMAIL PROTECTED] wrote:
Has anyone been able to get chan_sccp to work
Stephen,
It would be very kind if you provided the make and model of the fxs-fxo
converter. Also, what you mean by 'failed' (which symptoms - no ring
tone? No dial tone? etc.). Maybe also some more specific information
about your setup? This would help.
Thanks very much
Peter
Hi All,
Does
The Zultys Zip phone is crap
though.
As with the 841, no PoE
No speakerphone
No display
I am unable to get the message waiting indication to work
I am unable to get it to register with Asterisk, though I can place and
receive calls
There is no wall mounting bracket, and support doesn't have a
All,
I'd like to use the Voicemail to Email feature of asterisk, but I dont want
to use sendmail. We have a seperate email server that we would like to use
for this feature. How do and where do I specify this?
Thanks,
Jon
___
Asterisk-Users
I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a
built in DSL modem, and a single FXS port. Decent little router, now that
the latest firmware is out, but tcp and udp timeouts through NAT seem to
be set a little low, so I lose SSH sessions.
I bought a dozen
Hi All,
I'm looking for a single FXS port ATA capable of doing both SIP and IAX (not at
the same time of course). Can anyone make a recommendation?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]
Let me see if I understand this correctly:
I have an * box with a TE410 in it. If I install spandsp and all of its
requirements, does it mean that I could have my * box receive faxes and
put the tiff files in some organized location without the need of
having a fax or fax/modem and any
To help you out i will post my config files... The only problem that i have
is in an active call i can't get my phones to send responses to voice menus
such as dialling the voicemailmain cmd.
I am using a TDM04b card with four ports instead of one my zapata.conf file
looks like:
[trunkgroups]
From what I have heard it works but has still some issues.
It's on sale from VoipSupply for 114.95
http://www.voipsupply.com/product_info.php?cPath=95_111products_id=331
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Normandin
Sent: Wednesday,
Guys,
Thanks a mil. I'll try it out and see how!
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax :
Not a mailing list but VOIP forums on DSL Reports are large
and active:
www.dslreports.com
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Gerard
Marcel
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 3:47
AM
Hi All!
I already have this as a 'product' developed by Nicolas Gudino of Flash
Operator Panel especially for me as a fully functional system. You can
see this at http://www.eezeephone.com under callback services. Though it
may not be working now due to some misuse in the past.
Unfortunately
Ok you guys enough. The debate will go on forever. The only thing
that seperates the boys from the men in this world is marketing. Beta
vs VHS.
Is Unix is better then Windows - Yes, but it doesn't matter. We live
in a Windows world because Microsoft is the greatest marketing company
on the
Hiya everybody - I got myself a cisco wireless 7920, and have had no
trouble at all setting it up. I used the easter2005-testing version of
the chan_sccp driver from chan-sccp.sf.net. Calls in my LAN are crystal
clear and sparkling. I set up my firewall to allow connections from the
just wanted to let those out there having a similar issue know that ...
envir: normal phone - chanbank - asterisk - iax2 -pstn -normal phone
the chanbank side could hear the pstn side, but not vice-versa (this
happend everytime), and would happen with both ulaw or gsm codec's.
seems there was
I had this problem too and update to the new firmware and all OK !
Mike Roelofs
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Colin E. McDonald
Verzonden: woensdag 20 april 2005 12:29
Aan: asterisk-users@lists.digium.com
Onderwerp: [Asterisk-Users]
I know there was a posting regarding how to configure 5300 and asterisk so I
can dial pstn and get connected to asterisk. Can somebody share the sip.conf
and dial-peer config with me?
Thanks
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Hello,
I am looking for a solution to connect about 40 analog telephones to an
Asterisk pbx. Initially I wanted to use an Adit 600 channel bank, but
yesterday I talked to Carrier Access, and they recommended the Adit 3104
gateway.
All I am looking for is a device that multiplexes many analog
it is a silent suppression error. make sure its turned off on all the devices
being used to process that call
-Original Message-
From: Doug Reid - Stormcorp [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 20, 2005 6:02 AM
To: [EMAIL PROTECTED] Digium. Com; Asterisk Users Mailing
List -
I guess we are not thinking about the global extent of asterisk.
$200 in a third world would be great money. You can almost buy a Dell
computer for that much.
But this is more like a $200 bounty to design, build and replace your
Yugo engine with a Ferrari engine. And I only get the money if and
chase1*CLI realtime mysql status
No such command 'realtime mysql' (type 'help' for help)
chase1*CLI
This is your problem. You do not have res_config_mysql.so loaded.
You said that you have downloaded the newest asterisk-addons. Did you compile
them? Did you install them?
-Matthew
Hi All, I just installed a TE110P card and I'm trying to compile the
code. I followed to the letter the instructions. This is what happens.
[EMAIL PROTECTED] zaptel]# make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo
On Wed, 2005-04-20 at 14:20 -0400, Walt Reed wrote:
and hoiw many operating systems were so popular during the 80s and early
90s? What operating system shipped on almost every computer during that
period?
BTW, in the 80's, it wasn't windows - it was DOS (I know, well before
your time.)
Michael Lyszczek wrote:
Are there any BYOD providers out that that people have had positive
experiences with? I have broadvoice and they suck lately. Anyony have
anyone with a good amount of peers and not a lot of downtime?
I like voicepulse. They raised their rates recently, but they are
You have to do a flash on the Siemens which gives you * dialtone then Dial
*0 which flashes the line. So the steps are flash *0
- Original Message -
From: Sascha Ferley [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 6:32 PM
Subject:
Ok I [EMAIL PROTECTED] up. I didn't realize the card is 3.3 volts and my new computer
is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions?
Mike
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My call waiting beeps will blow an eardrum. Adtran 750 with Cortelco
analog sets. We have txgain set at 3.0 because speech volume is too
low. Is there a way to reduce the beep volume without impacting the
rest of the system?
Thanks
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The whole system is running on an older dual processor PII 450Mhz machine
with SCSI drives.. 512Mb ram.The system runs RH9 with asterisk version
SCSI drives cause beeping too do to the demand for interrupts!!! With
IDE drives you can give the processes a low priority but those SCSI drives
Greg,
Are you using AMP?
And If so, you have any tips and tricks on how to easily manage phones via a
amp plugin/fix?
- Original Message -
From: Greg Boehnlein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
Rich Adamson responded to an earlier reply (not from me)
Eric, those links have nothing to do with his stated problem. The
problem is 105v AC on the pstn line when on-hook and no ringing.
No, he says the issue is about ringing and strange voltages on his
Digium TDM400 FXS ports, not the PSTN
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