I'm having an echo related problem with Zap channels that are answered
before a dial takes place, such as for IVR menus or fax detection.
Basically, it sounds like the volume gets turned up to maximum while *
is ringing the internal extensions waiting for someone to pick up, so
when you pick
On Monday 19 July 2004 01:23 am, Brian K. West wrote:
Dont have to.. just add it to the voicemail.conf and it will auto do
everything for you.
bkw
Well, after having restarted * a few times, and rebooted once, I can say that
no mailboxes were created automatically. I'm running a week old
Not sure if it works for you, but the simplest way is:
[capi-in]
exten = DIDNUM1,1,DoSomething
exten = DIDNUM2,1,DoSomething
exten = DIDNUM3,1,DoSomething
where DIDNUMX is the direct indial number. Much nicer than goto
statements with complicated splits.
AFAIK you have only a DIDNUM if
Hi,
Thanks a lot for the configs Fabe.
I tried your zaptel.conf but I still get yellow and
red alarms in zttool and * is unable to create any Zap
channels (as expected with yellow and red alarms).
I realise I will now have to start talking to Colt (in
Ireland) to try and get the line up and
it never gets past the blue screen
Ahhh, now I know: MICROSOFT is making the software for the Grandstream
BT102. That explains something ... :-)
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From the quote bits below:
zaptel.conf
span=1,0,0,ccs,hdb3,crc4
Assuming that it is the only E1 present, or the only one connected with the
outside world, you should have the timing source configured:
span=1,1,0,ccs,hdb3,crc4
Also, it might be that Colt are not using crc4 on your link, so
Marty Mastera [EMAIL PROTECTED] wrote:
When I call the pstn number, the zaptel picks up the line on
the first ring and then forwards it to the sip phone and
rings it. Is there anyway to prevent the zaptel from picking
up the line until the sip phone actully answers the call.
This way I
Hi all,
Is there any updates on ast_data from svn.asteriskdocs.org/res_data to work
with Asterisk cvs Asterisk CVS-HEAD-07/14/04?
regards,
Glynn
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On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
They only get created as they are used and voicemail left, try leaving a
message and you should see that the structure etc is created.
Steve
-Original Message-
From: Steve [mailto:[EMAIL PROTECTED]
Sent: 19 July 2004 08:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Adding
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
There's a script to create the mailbox with the asterisk source code:
contrib/scripts/addmailbox
On 19/07/2004, at 8:17 PM, Steve Hanselman wrote:
They only get created as they are used and voicemail left, try leaving
a
message and you should see that
Does anyone have a recommendation for a 48 port LAN switch for a new *
system? I'm not happy with NetGear's reliability.
You can get Cisco 2950s for about $600/24 ports.
And 48 ports from Dell for about the same price. I haven't used any of
their latest round of switches, but their
Try picking up a 3com 3c17205 (4400 PWR) 24 port switch...it'll cost you a couple of
bucks more but has POE built in and if you use the NJ200 (there's a guy selling boxes
of 20 NJ200's on Ebay for $750 but he's got a lot so you can get him down to about
$575) or NJ220 wall jacks (DONT GET THE
On Sun, 2004-07-18 at 23:52, Bruce Komito wrote:
Following a(n apparently) failed attempt to upgrade a BT102, the phone is
now brain-dead. Although it still has enough smarts to get a dhcp address
and try to download the firmware and config, it never gets past the blue
screen, nor will it
Wiley E. Siler wrote:
I have a solution that allows me to assign a soft key with no problems.
However, it seems like a waste the the hard button labeled Voice Mail is
not dialing right into voice mail. Is there a known way yo do this? I
have tried everything in the manual but it doesn't seem to
This week starts with the exciting news: We're getting close to
Asterisk 1.0 again. After the failed attempt earlier this year,
we've been able to remove a lot of the MAJOR/CRASH bugs from the
bug tracker and Mark feel's it's time to target 1.0 again.
At this point, the community needs to work as
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main problem
I have is not being able to get the
Forgot to mention, both modules are show in ztcfg fine, see below:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
2 channels configured.
and zap show channel 2 give the following:
Channel: 2
same problem here. the display shows vulcan.
- Original Message -
From: Mark Elkins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Monday, July 19, 2004 7:39 AM
Subject: Re: [Asterisk-Users] Brain-dead Grandstream BT102?
On Sun, 2004-07-18 at 23:52, Bruce Komito
Hi all,
We're trying to hook up our Asterisk config (Card: TE410P) with a
Siemens EWSD switch. The link is ok on both ends (green), with no errors.
The problem is when we try to make a call from our side (via call
files), we get the pri/E1 error
Ext: 1 Cause: Temporary failure (41), class =
Hi,
The version of astgui is 1.0.2.
I am using PHP version 4.3.4-4 installed on a debian 3.0 system (testing) from apt-get.
I do not have any GLOBAL_VARS set in my environment. What should it be? I am not very
familiar with PHP.
I had installed this on an existing system but made sure to
On Mon, 2004-07-19 at 02:19, Steve wrote:
On Monday 19 July 2004 01:23 am, Brian K. West wrote:
Dont have to.. just add it to the voicemail.conf and it will auto do
everything for you.
bkw
Well, after having restarted * a few times, and rebooted once, I can say that
no mailboxes
Hi,
I'm in the process of switching over to Asterisk from Alchemy kit and have
hit a stumbling block.
We're in the UK and use ISDN. At the moment we don't accept calls from
withheld numbers (we just play them a message), but do accept calls from
unavailable numbers. There doesn't seem to be any
Nick
We are using QuadBRI cards from www.junghanns.net and also at home I
have a couple of cheap HFC cards - in by Asterisk box sandwiched between
my BT line and my Alchemy Cybergear Gold.
Using this there is the possiblility to differentiate between Withheld
and 'unavailable'. The latest
Quoting hskim [EMAIL PROTECTED]:
I have two questions.
- Is TE410P is same as TE405P, or did I received different card?
- zaptel.conf is configured CCS/HDB3. But It's configured as ESF/B8ZS.
Hong
TE4xx card are really cool they allow you to change type of interface
either E1 or T1. By
OK, I think I have my problem narrowed down on my Avaya 4602SW SIP hardphone.
When I reset the phone the phone works perfect up to the point until I
get the following error in the CLI:
-- Got SIP response 481 Call Does Not Exist back from my.home.external.ip
This is how I have the SIP extension
At the moment I'm prototyping an advanced ENUM application with PHP
fetched from LDAP. When a user enters a full hostname as SIP adress I get
loop problems from the AGI EXECUTE DIAL and from a Dial in the
extension.conf.
-- Executing AGI(SIP/1000-c3c3, enum.php) in new stack
-- Launched
Dear Sirs,
I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not
found. . The same uccurs when i type modprobe wcfxo
May you help me.
Thank
This is how I have the SIP extension setup:
[2002]
type=friend
username=2002
secret=mypassword
host=dynamic
context=from-sip
mailbox=2002
nat=yes
qualify=yes
dtmfmode=info
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callwaiting=1
Not sure how to
Dear Sirs,
I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not
found. . The same uccurs when i type modprobe wcfxo
Perform an
On Fri, 2004-07-16 at 18:45, Andrew Yager wrote:
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds
Second that. Using stacked HP 2650 switches to support ~120 users and with
QoS on, don't even have to worry about VLAN'ing the network. In HP/Compaq We
Trust...
I have been quite happy with our HP 2848 GigE switches that we put in
for our desktops a few months ago. I have also used the 2650 48
I cant do SIP -
CHAN_H323 transmit audio!!! I can hear rings, but when connected,
NOTHING
It happened in both:
SIP - CHAN_H323 and CHAN_H323 - SIP...
when it will be
solved?
On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote:
Hi,
I'm am currently in the process of trying to integrate an * box with an
NEC Electra Elite IPK.
Currently, we have 7 POTS lines coming into our building. These lines
are plugged into our NEC using the appropriate analog line
Speaking without experience of the exact combination you mention, but I'd
expect that BT will send these to you using the combinations as follows:
CLI: present Screening: available - Released number
CLI: absent Screening: withheld - Withheld number
CLI: absent Screening: available - Unavailable
On Mon, Jul 19, 2004 at 11:01:48AM -0300, PBX Portela wrote:
Dear Sirs,
I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not
found. .
On Mon, 2004-07-19 at 07:13, Nick Cobley wrote:
If I dial the extension I just get a 404 error on the phone
(Grandstream), but no errors at all on the console. I am using
CVS-HEAD-07/14/04. Here is a snippet of what I have in the various
config files.
Welcome to SIP. Dialtone is local to
Happen to have any NAT in the mix?
bkw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti
Sent: Monday, July 19, 2004 9:25
AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] STILL NO
AUDIO
I cant do SIP - CHAN_H323
I'm trying to spec out hardware for a new office, and I'd like to
include power over Ethernet as an option. I've seen a handful of PoE
injectors around $1000 for 24 ports and a couple switches up around
$2500 for 24 ports. Are there any cheaper options, short of buying a
boatload of 1-port
I suspect it will be solved when you put disallow=all and allow=ulaw in
sip.conf and h323.conf (and NO OTHER ALLOW= LINES)
On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote:
I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when
connected, NOTHING
It happened in
I'm using * connected to a EWSD too through an older zaptel (card T400P)
and there is no problem with PRI calls.
Did you put euroisdn?
[EMAIL PROTECTED] wrote:
Hi all,
We're trying to hook up our Asterisk config (Card: TE410P) with a
Siemens EWSD switch. The link is ok on both ends (green), with
No NAT, no FW, no nothing...
from cisco 5300 with public ip without FW, to * with public
ip without FW using SIP, and then from * to cisco 5300 without FW using
chan_h323
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
brianEnviado el: Lunes, 19 de Julio de 2004 11:39
I WANT TO USE G729, I HAVE TO USE IT...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Eric Wieling
Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] STILL NO AUDIO
I suspect it will be solved when you
I would be interested.
Tenorio, Leandro wrote:
Seshu, I'm interested could u provide more info...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu
Sent: Wednesday, July 14, 2004 11:02 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
For those on a low budget compex (http://cpx.com) has some very low cost
switches that support QoS.
http://www.cpx.com/proddetail.asp?c=Switchese=109
Bought a few of these myself, seem to work well. They are only
manageable through an rs-232 console though, and don't have some other
features
I was even considering going further and writing a crossplatform or a
webapp for configuring. However I was thinking if someone has written
some notes on the config file specification that could save a lot of
time. I have no intention of competing with gsconfigure since I think
it's an
Anyone else having problems with inbound Broadvoice this morning?
--
Chris Tooley / Network and Development Services
Networking Technologies Resource Center, LLC (NTRC)
8650 Spicewood Springs Road, Suite 105
Austin TX 78759
512-250-8985 / Fax 512-250-5909
www.ntrc.net / www.ntrcstore.com
Hello All,
I am having the problem with Inter-Tel configuration... and really
can't find source of it.
I was able to configure the phone, it picks up the IP from our DHCP
server and properly configured
After that it registered to IPC card and I can make a call. but
just for a few
Why don't you use asterisk-oh323?
Michael.
Sebastian Nocetti wrote:
I WANT TO USE G729, I HAVE TO USE IT...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Eric Wieling
Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m.
Para: [EMAIL PROTECTED]
Asunto: Re:
Thanks Steve,
The SIP handsets are working find as I can make calls to other handsets
as well as receive incoming calls via the FXO module. So all is good there.
Cheers
Nick
Steven Critchfield wrote:
On Mon, 2004-07-19 at 07:13, Nick Cobley wrote:
If I dial the extension I just get a 404
On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote:
Does anyone have the call hold feature working? If you do... how did
you make it work? The instructions say to press the left button to
place the call on hold, and the right button to take it off - except
when I am in a
When using a channel bank for analog handsets, you have a couple options in
the way you handle transactions involving the analog handsets and origination.
With immediate set to no, it appears to me that soon as a digit is pressed
after going off-hook, the single digit is taken and processed
On Mon, 2004-07-19 at 10:00, Sebastian Nocetti wrote:
I WANT TO USE G729, I HAVE TO USE IT...
Not while testing you don't. Once you get it working with ULAW ONLY
then see if you can get it working with G729.
--
Useful Asterisk Docs (BOOKMARK THEM!):
I WANT TO USE G729, I HAVE TO USE IT...
When you have no FW and no NAT, then you seem to be inside your local
network. In this case you shouldn't really care ?!?!
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Benjamin,
Is this package intended to mirror the directory structure of the linux
builds? If so, I may have an issue: While /var/lib/asterisk is properly in
place after running the installer, /usr/sbin/asterisk is not. I'm running on
OS X 10.3.4 and downloaded the package on Sunday afternoon, if
On Sun, Jul 18, 2004 at 10:12:20AM -0500, Chris Howard wrote:
I say on slashdot that the Linspire guys have released PhoneGaim.
PhoneGaim is Gaim with SIP added on. Anyone want to add IAX2 as
well...
I'm writing a plugin for gaim right now that does iax2 on my off time.
I haven't had much
Now that you mention it, yes... it seems that SIP isn't being passed from
their PSTN gateway to the rest of their network... It's ringing, but there's
no acknowledgement in * that anything's going on...
- Original Message -
From: Chris Tooley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Testing both...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Michael Manousos
Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] STILL NO AUDIO
Why don't you use asterisk-oh323?
Michael.
Sebastian
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
Hi,
I am looking for some affordable IP Phones. Any experiences with the
SipToneII by ipDialog?
What about soft phones? Any recommendations there (for Windoze and Linux)?
Thanks,
Yiannis
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Asterisk-Users mailing list
[EMAIL
Look out for 3c17205 switches from 3com and read the QOS thread posting here at the
moment.
P
-Original Message-
From: Scott Laird [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004, 7:58 AM
To: '[EMAIL PROTECTED]' [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cheap PoE
What kind of problem?
All works OK except that config
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Holger Schurig
Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] STILL NO AUDIO
I WANT TO USE
I'm using a channel bank with a T1 card on the Asterisk server and have
defined the FXS channels (user phones) to the context of [internal] and
don't have any problems using the dial plans with the full digits. I
haven't had any of them try to go to the i extension after the first digit.
Not sure
I restarted asterisk and tried calling and it works now so either they fixed
the problem and it's just a HUGE coincidence that I restarted * at the same
time, or restarting * did the trick...
P.S. What is the deal with the MailMan? When I send replies to the list,
I've had it take up to 4 hours
On Mon, 2004-07-19 at 09:04, Yiannis Costopoulos wrote:
Hi,
I am looking for some affordable IP Phones. Any experiences with the
SipToneII by ipDialog?
So far our experience with the IP Dialog SipToneII is not good. It
locks up after hang up on us, and just does not play nice. If
Actually very straight forward. After calling Digium and getting some
amazing tech support, I simply had to modify my dial string. The change
was simply to make * pick up the line, and wait before dialing the
requested extension. This was accomplished using several 'w' characters
in the
Hi
Can anyone with distinctive ring on their 7960's possibly post how they've got it to
work?
I understand that the ALERT_INFO variable is involved but using the examples for the
variable value from the WiKi I'm just getting an error message from the Asterisk
concole.
Thanks in advance.
P
3com have some goods POE kit and some very nice managed wall jacks that supply POE and
are fully managed.
Here's an auction that the seller just closed:-
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemcategory=40990item=5708757277rd=1ssPageName=WDVW
Last time I spoke to him he had 5 boxes of
I've actually just thought about using that solution. I realized over
the weekend that my current solution has one VERY serious flaw in it.
I forgot to mention that we, currently, have 24 phones/extensions in our
office, we quite a few agents, many which are in multiple groups, we
must be
Jason your a legend!!! I swear I tried include = internal in the sip
context, guess I managed to stuff it up somehow!!
Thanks so much for your help, sanity now saved :)
Regards
Nick
Jason Williams wrote:
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully
I'm trying to be able to access the call waiting and three-way calls
features on a line attached to my X100P. For example, a party calls, the
X100P/Asterisk ring the 7960 on my desk, and all is fine. If I want to three
way call another individual in, I need to send a Flash to the X100P, and the
Hey man,
I have a bunch of used power injectors that I would actually like to sell.
If you are interested I will gather them all up and count them, but I
know I have at least 24.
I'd be glad to send you one to test.
They are all Avaya/Lucent brand, they should work for any type of phone.
Just wondering if anyone has tried MAC OS X and panther.
I will like to do SIP to H323, not sure if this will be possible on the MAC
because of the Libraries PWlib and OPenh32 for Linux..
Just curious..
Anyway, anyone has an easy guide (step by step) to setup oh323 with
asterisk. I saw a guide
Hello,
In my quest to create several proof of concepts for what can be done
with Asterisk, I've run into a bit of a problem. I have a pair of
SPA-2000's acting as off premise extensions for an analog line. When a
call waiting call comes in, the caller id information makes it though
the ULAW
I've paraphrased the OS X installer developer's comments: there's a bug in
Installer that is preventing the archive from working right. Below is the
fix for the problem.
First (obviously) run the installer. Since the executables are in the
archive.pax.gz file in the installer package, first do a
On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote:
Look out for 3c17205 switches from 3com and read the QOS thread
posting here at the moment.
So $1600 for 24 ports. That's not *too* bad. HP seems to have a
similar model (2626-PWR) for a similar price. 3com also seems to have
a 24-port
Hi All,
Please checkout the following GUI web panels, which have
been created and installed from the source code available
in this forum.
http://67.109.153.236/*web/
It edits extensions.conf after some customization.However unable to
update sip.conf.
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
Tony Nichols wrote:
On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote:
Hi,
I'm am currently in the process of trying to integrate an * box with
Hi,
I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec
such as G.729 can be very CPU demanding. What are the real advantages of using a codec
such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the
scalability of my asterisk PBX ? This is
Thank you!
Can you tell me more about the dial plan feature? How do you setup the
correct digitmap?
W
-Original Message-
From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004 4:56 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error
I'm having no end of trouble with some Siemens Gigaset phones and GS
HT286s.
Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once
then it goes off and then just flashes it's LEDs and displays incoming
call on the LCD with no further ringing. According to the manual it is
CTR37
Has anyone tried the new dlink powered switches? I remember seeing an online
voip store selling these as a good option for providing power in a voip
application. They were price at 1100 for a 24 port model.
The lowest cost solution I have seen are the individual 3com power injectors
which can
Hello,
I need help in creating a simple PSTN Gateway. This is the scenario:
-I have one client sending me VoIP traffic (they dont have
asterisk, so IAX is out of the picture for me) and I need to validate that
traffic (only accept calls coming from his IP). After that I would
Scott Laird wrote:
So $1600 for 24 ports. That's not *too* bad. HP seems to have a
similar model (2626-PWR) for a similar price. 3com also seems to have a
24-port injector for $800.
I still don't understand why I can buy single-port injectors for $20,
but multi-port models are $30 per port
Thank you!
Can you tell me more about the dial plan feature? How do you setup the
correct digitmap?
Check the Administrator's Document. You can find it on the Wiki, under IP
Phones.. Polycom. Did you try to look up the digitmap feature before
sending this post? If not, you should be
If you have the bandwidth then use ulaw :)
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, July 19, 2004 12:44 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Codecs - Advantages
Hi,
I'm
Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and
it doesn't have a button labelled Voicemail.
On the left side are the blue speaker, red mute, and blue headset buttons,
then next to them top to bottom are the three Line buttons (clear covers
for putting your own
I don't know how serious pulse dial is, but it is supported in asterisk.
You will not likely find any device that converts pulse to tone though.
Although it might be possible if it went through a channel that doesn't
use pulse dialing like sip.
Hp So a channel bank w/ FXS ports will pass
On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote:
Thank you!
Can you tell me more about the dial plan feature? How do you setup the
correct digitmap?
It is all in the Admin Guide you can download from the Polycom web site.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
From: Christopher L. Wade [EMAIL PROTECTED]
Organization: Unistar-Sparco Computers, Inc.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Reply-To: [EMAIL PROTECTED]
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't
You can pick up 3c17205's on ebay for usually around $500-$700 new in the box (non on
there at the moment).
They come up about 3-4 a month although it's summer at the moment so a bit quiet.
P
-Original Message-
From: Scott Laird [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004,
Hello. Here is what my extension which uses distinctive ring on a
Cisco 7960 running V6.2 firmware looks like. Note that the distinctive
ring tones are changes in cadence, rather than changes in ringing sounds on
the 7960. Also, if you adjust the ringer volume wile the distinctive ring
On Mon, 19 Jul 2004, Kevin P. Fleming wrote:
Scott Laird wrote:
So $1600 for 24 ports. That's not *too* bad. HP seems to have a
similar model (2626-PWR) for a similar price. 3com also seems to have a
24-port injector for $800.
I still don't understand why I can buy single-port
Hi,
Does anybody knows where can I change timing for collect calls?
tks
Oz
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Asterisk-Users
I think this has been an ongoing issue. When you figure out a solution
let me know.
The only solution I could come up with isnt feasible for everyone. I
have another pbx that provides the dialtone for my asterisk box. I put
two zap cards in my asterisk. On my other switch I set it so that if
line
I read the administrator document repeatedly. I have not been able to
find a wiki that applied to digitmap feature at all and I have searched
repeatedly and read several of the wikis regarding Polycoms. The
administrators guide doesn't have enough context explanation to make the
use of the
Wallingford, Ted wrote:
(B
(BI've paraphrased the OS X installer developer's comments:
(Bthere's a
(Bbug in Installer that is preventing the archive from
(Bworking right.
(BBelow is the fix for the problem.
(B
(BApple may have a reputation for attention to detail and
(Bperfectionism, but
And it is throughly convoluted in the admin guide. What is the T for?
Pipe obviously separates entries. X = any digit one would assume? I am
just luooking for a brief explanation. Thanks.
Here is the excerpt from the manual.
Attribute
dialplan.digitmap
Permitted
[Try this again...]
Hello. Here is what my extension which uses distinctive ring on a
Cisco 7960 running V6.2 firmware looks like. Note that the distinctive
ring tones are changes in cadence, rather than changes in ringing sounds on
the 7960. Also, if you adjust the ringer volume wile
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