On Thu, 24 Feb 2005, Sascha E. Pollok wrote:
I am currently taking a look at queues. What I am trying to
achieve is that, beside the MoH, when the caller gets put
into the queue, she should hear an announcement like welcome
to snakeoil - please wait or leave a message by pressing #
then the
On Wed, 23 Feb 2005, Eric Bishop wrote:
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit
On Wed, 23 Feb 2005 [EMAIL PROTECTED] wrote:
2. many of the list police are active in the development process
well, so your remarkably clever comments about the lack of help are
uncalled for and untrue. People will help you, but they won't hold
your hand. If you want your hand held, then
On Wed, 23 Feb 2005, Johan Bilien wrote:
I'm trying to connect a PC with a TE410P to an E1/IP equipment.
Unfortunately I keep getting a yellow alarm from zaptel (in zttool)
and a Loss of Framing alarm on the remote equipment.
The E1/IP is connected on the other side to a PRI interface on a
On Wed, 23 Feb 2005, Scott Stingel wrote:
Good writeup! Question regarding Q.SIG: Can it be used to solve the
problem of signaling a remote switch to take a call back and extend it
to another channel instead? This, as you know, is always a challenge
when using IVR in a call centre
On Wed, 23 Feb 2005, Matt Fredrickson wrote:
On Tue, Feb 22, 2005 at 04:31:39PM +0100, Jan Berggren wrote:
How do I configure CAPI to use QSIG? Is QSIG supported by Asterisk?
Just set your switchtype in zapata.conf to type qsig
Is the zapata.conf file used at all for CAPI? I though all
On Wed, 23 Feb 2005, Kevin P. Fleming wrote:
Peter Svensson wrote:
Yellow alarm is the same as remote alarm - i.e. the other side is saying
that it cannot hear you. Given the Loss Of Framing on the other end this
seems resonable.
Actually, yellow alarm is most frequently generated
On Wed, 23 Feb 2005, Johan Bilien wrote:
But why would I only have a LOF error on one side?
You could have one damaged pair in the cable. I.e. Asterisk can hear the
other box, but the other box does not hear Asterisk. Ususally you would
expect a red alarm (Loss of Signal) and not a LOF in
On Thu, 24 Feb 2005, Rod Bacon wrote:
My understanding is that this is only required when using it inside a
dialplan. Eg, the extension answers, then switches to fax originator mode.
No. To quote the mail from Steve Underwood:
Many people have it working that way. Very few people use it for
message and after
that to dial internal number(ex. 101). It is possible to dial directly
700101 and asterisk to dial PBX prefix, wait for PBX to answear and
after to dial internal number?
show application dial
pay attention to the option D.
Peter
--
Peter Svensson ! Pgp key available
On Tue, 22 Feb 2005, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one side,
and open cables on the other for mounting in our own patch panels.
In Europe, or Sweden
On Tue, 22 Feb 2005, Michael Welter wrote:
Are you aware of the type 66 punch-down block with an AMP-50 connector?
Also the harmonica--an AMP-50 on one side and 12 RJ11 jacks on the
other (two pair/jack).
We punched the cable directly to the jacks since that was what we needed.
None of
On Tue, 22 Feb 2005, Jon Gabrielson wrote:
On Tuesday 22 February 2005 07:17 am, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk.
;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one
side, and open cables on the other
On Wed, 23 Feb 2005, Rod Bacon wrote:
No matter which version of SpanDSP I use, with which version of libtiff,
Asterisk, ... I simply cannot send faxes.
Did you remember to add the caller option to txfax?
Peter
___
Asterisk-Users mailing list
On Sun, 20 Feb 2005, Anton Krall wrote:
Im new to asterisk but is it possible to simulate a dialtone for example, in
other PBX when you pick up the phone you can hear a certain dialup, which is
the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is
this possible?
I'm not
On Sun, 20 Feb 2005, Duane wrote:
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get
On Sun, 20 Feb 2005, Anton Krall wrote:
That app_disa is new to me... Is there a list of available apps? Im still
quite new to asterisk but I guess you can find out which apps you have by
using a show applications but my question would be more of how to make new
apps or download/get new ones,
On Sat, 19 Feb 2005, Joel Vandal wrote:
I've just get a 3COM 3102 but is not configured to use SIP protocol. I've
read that I need an NCP PBX from 3com to upgrade to SIP firmware ? Does it's
true ? I must try to upgrade this =)
On an earlier thread on asterisk-users it sounded like the 3com
On Sat, 19 Feb 2005, Alistair Cunningham wrote:
The issue is with the provider of the PRI. They may or may not accept
arbitrary callerids on outbound calls. For example, in the UK many
providers will only accept callerids that are assigned to the trunk on
their end. You should check with
On Fri, 18 Feb 2005, Jeremy SALMON wrote:
I want to install an Asterisk Box in my Network and work with some IP
phones and ISDN phones.
Is this configuration is possible :
-E1AsteriskE1 or T1---channel bankISDN phones
Wich type of channel bank can I use to do this
On Fri, 18 Feb 2005, Robert Rozman wrote:
I wonder which PRI interface card is most stable and supported for EuroISDN
and Asterisk ? Are they stable enough ? Any tips ?
Digium TE410P and TE405P are well supported.
Peter
___
Asterisk-Users mailing
On Thu, 17 Feb 2005, Rich Adamson wrote:
In the post that I was responding to, the writer hinted his understanding
was that T1 to T1 channel connections didn't involve any asterisk code.
His impression seemed to suggest that codec selection, etc, wasn't a
factor since the analog fax modem
On Thu, 17 Feb 2005, Deti Fliegl wrote:
Peter Svensson wrote:
Asterisk only expects INFORMATION elements when expecting overlap digits
(i.e. before CONNECT, PROCEEDING etc). After that it expects digits as
inline dtmf.
Yep - but ISDN phones normally do not encode inline DTMF. Therefor
On Thu, 17 Feb 2005, Deti Fliegl wrote:
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 2 (reference 7/0x7) (Originator)
Message type: INFORMATION (123)
[2c 01 31]
Keypad Facility (len= 3) [ 1 ]
Feb 16 11:42:25 VERBOSE[2975]:
[ 02 01 e8 fc 08 02 00 07 7b 2c 01 31 ]
see
On Wed, 16 Feb 2005, Steve Underwood wrote:
If you really are using ulaw, and you do not have extreme packet loss or
jitter, DTMF detection should be very reliable. It is no better in CVS
HEAD because it wasn't broken in the first place.
We have some problems with dtmf detection on our
On Wed, 16 Feb 2005, Steve Underwood wrote:
If that is true, someone must have broken something. Not only does the
DTMF detector I wrote not care about small imperfections, it even
tolerates a dropped packet with the DTMF passes over a VoIP path (this
kind of tolerance was added a couple
On Wed, 16 Feb 2005, Rob Scott wrote:
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Asterisk clocks outgoing rtp data to a device from the
On Wed, 16 Feb 2005, David Ishmael wrote:
Is there a way to increase the volume for the voicemail? Whenever someone
leaves a message, the volume is so low it's hard to hear.
This is a known bug - see bug number 2023:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002023
Peter
On Wed, 16 Feb 2005, Deti Fliegl wrote:
I tried to use Voicemail from a PRI interface but it didn't work because
pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY
messages which are normally handled by a bri-stuffed libpri.
Unfortunately a wrong if condition stops
On Thu, 17 Feb 2005, Jim Van Meggelen wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid filename
character.
In what way
On Tue, 15 Feb 2005, tim panton wrote:
My best advice is to call your PTT and ask them how many digits
they expect you to send, I am guessing they only expect the
last 2, but only they know for sure.
Also ask them if they require a specific Type Of Number for the outgoing
callerid.
On Tue, 15 Feb 2005, Daniel Nyström wrote:
Where can I get E1 and/or Euro-ISDN specifications/data sheets?
Are there specs for other E./G./Q./etc. protocols as well?
The specifications are built one on top of another. Each just lists the
changes and clarifications relative to the underlaying
On Tue, 15 Feb 2005, Sylvain Gagnon wrote:
I'm using Asterisk (latest CVS head) to perform outbound call as
robot/testing tool for an IVR platform, with a Wildcard T100P configure as
ISDN Pri.
For develop the exten context script I was using a real PSTN ISDN Megalink
(DMS100) to reach the
On Tue, 15 Feb 2005, Daniel Nyström wrote:
Peter Svensson wrote:
| On Tue, 15 Feb 2005, Daniel Nyström wrote:
|
| Where can I get E1 and/or Euro-ISDN specifications/data sheets?
| Are there specs for other E./G./Q./etc. protocols as well?
|
|
| The specifications are built one on top
On Tue, 15 Feb 2005, Peter Svensson wrote:
On Tue, 15 Feb 2005, Daniel Nyström wrote:
What's exactly Euro-ISDN? Is it G.931? I don't really get this.
Is there a Q/G/E document for Euro-ISDN?
I've downloaded two out of three fron ITU, so I would like to know for
sure! :)
Ericsson has
On Mon, 14 Feb 2005, Frank Sautter wrote:
since my latest libpri update i get these messages:
!! Unable to handle ROSE operation 36
!! Unable to handle ROSE operation 30
i searched through ITU X.219 and X.229 but can't find any values for the
Remote Operations Service Elements.
are
On Fri, 11 Feb 2005, Edin Kozo wrote:
--- Altus Snyman [EMAIL PROTECTED] escribió:
I've installed a few systems with quad/octo bri
cards
On these systems incoming numbers are ether the full
number,example
12345657 or ether the last 4 digits,example 7654
But for some reason the latest
On Fri, 11 Feb 2005, Sverrir Valgeirsson wrote:
I'm looking at ordering a 30-channel ISDN connection from telia (a swedish
operator) and then using a Wildcard TE110P card with that and asterisk to do
IVR.
Can I be certain that the TE110P card will work with that ISDN connection? A
30
On Fri, 11 Feb 2005, Alistair Cunningham wrote:
30 channel ISDN (generally known as primary rate ISDN, PRI) is a layer
that runs on top of E1, just as Internet Protocol can run over Ethernet.
IP is run over just about anything that passes data, serial lines, atm,
ethernet, you name it, it
On Thu, 10 Feb 2005, Marco Castillo wrote:
Hi, I'm having a little problem when trying to make a call from asterisk. I
connect a SIP phone to asterisk, and in the asterisk box I have a TE110P
card connected to a E1. When a SIP client makes a call through the E1, I
received no dialtone in the
On Wed, 9 Feb 2005, Stefan Gofferje wrote:
I wanted to try out app_icd but...
[EMAIL PROTECTED]:/opt/app_icd make
=== Compile: /opt/app_icd/app_icd.c (app_icd.o)
app_icd.c: In function `app_icd__log_events':
app_icd.c:2104: error: structure has no member named `cid'
app_icd.c:2104:
On Wed, 9 Feb 2005, Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
Aw crap. I completly forgot about call pickup. Good point. If
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
Oh right.. I remember seeing that.. yeah that looked a whole lot more
elegant than *8. Why isn't it in HEAD?
I'm not sure. Once it started getting some testing BKW closed it. If
someone is interested in testing the patch I'm sure the bug could be
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
Peter,
Do you know what the current status of app_intercept is?
No, not really. See below for the errors listed.
I got it working on 1.0.2, but can't get it to complie on
CVS-HEAD-01/26/05-02:14:44
I get:
app_intercept.c: In function
On Tue, 8 Feb 2005, Daniel Nyström wrote:
Are this currently working with CVS-HEAD?
I've got an X100P-clone, and I've patched the zaptel drivers.
But the Asterisk patches seems to be there.
But I can't make it receive Caller-ID!
The X100P is unsuited for use with the Swedish PSTN for several
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams.
On Tue, 8 Feb 2005, Sven Lohmann wrote:
Yes, that would work - but I have no Zap and therefor no meetme - or is
there
a way to start meetme with SIP interfaces only ?
Use ztdummy or zaprtc. All that is needed is the zaptel timing. Another
option may be to use app_conference (use google).
On Tue, 8 Feb 2005, Frank Sautter wrote:
is there any usable documentation on the HLC or LLC octets (bytes)?
i searched etsi and was overwhelmed with the searchresults (1531). what
i need to modify libpri would be a table of possible values and where to
find the HLC and LLC fields in the
On Tue, 8 Feb 2005, Chris Wade wrote:
Exactly, but if you look, * queues don't quite measure up to these
requirements. The asterisk-ICD project is getting there, but still too
immature I think.
After a bit of man-handling icd works really well. We are doing weird and
wonderful things with
On Tue, 8 Feb 2005, Chris Wade wrote:
Kevin P. Fleming wrote:
I've been considering doing this as well... something like a dial
list, with a delay before dialing and a timeout for each entry.
Might also help to implement this type of 'strategy' for the 'Queue'?
Just another little idea,
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP
On Tue, 8 Feb 2005, Kevin P. Fleming wrote:
I believe this is correct, the call must be answered before app_queue
can handle it. However, how many customers do you think would sit there
for 3 or 4 minutes of ringing with no announcement messages or anything?
I doubt very many would last
On Mon, 7 Feb 2005, Juan Jose Comellas wrote:
I'm currently writing some code to support conferencing in Asterisk without
using the Meetme application. The conference runs in its own thread and every
new inbound or outbound channel that is created is passed to it. This thread
runs the
On Fri, 4 Feb 2005, Chamberland-Larose, Guillaume wrote:
- use one of the many patches for native MOH without mpg123
Well, doesn't that mean, I have to convert all the mp3s to
another bitrate/format? *sigh*
I'm using a whole bunch of my mp3s and I didn't have to convert any of
On Sat, 5 Feb 2005, Stefan Gofferje wrote:
As far as I know, you just need a second ISDN card and a X-cable. No
mods to the NT1 are needed. To build such a cable, just swap the outer
pair with the inner pair.
Termination and power are needed. Most isdn cards provide neither, but
some do.
On Sat, 5 Feb 2005, Frank Sautter wrote:
What network do you receive this from?
the calling party has an E1-PRI from the Deutsche Telekom (germany's
former monopolist) and our E1-PRI is from Arcor which is on of the new
telco companies founded after the liberation of the telco market in
On Sat, 5 Feb 2005, Peter Svensson wrote:
The behaviour you are seeing is described by ETSI ETS 300 092-1 Annex B.
According to 1TR67 this option *is* in use in Germany.
ETS 300 092-1 by default requires a strict checking of the calling number
(paragraph 9.3). An alternate method
On Sat, 5 Feb 2005, MvB wrote:
I have tested the asterisk and hardware in the same set up at a
geographical different location where it does work. I am pretty much
stuck now.
It seems that the progress indicator 1e 02 82 81 is the problem saying
the call is not end-to-end ISDN.
No, that
On Fri, 4 Feb 2005, Michael Bielicki wrote:
It's even easier. If you are taking an E1 with euroisdn, the D-channel
will allways be nr. 16. In neary all cases the rest of the settings
will be: ccs,hdb3 and depending on the carrier crc4.
Other things you will need to agree on / know:
* Number
On Fri, 4 Feb 2005, Frank Sautter wrote:
a feature of euroisdn is, that you dail a number e.g. 0732194490 (where
0 is the extension of the call dispatcher) and the phone is forwarded to
someone with an extension of 26.
our ericsson showed after the call was picked up 07321944926 and no
On Fri, 4 Feb 2005, Frank Sautter wrote:
how can the charge info from a E1-PRI be received and be forwarded to a
classic PBX?
Advice Of Charge, see the standard q.956 clause 2. There are several
flavours: AOC-S gives the expected per time unit charge at the start of
the call and any time
On Fri, 4 Feb 2005, Frank Sautter wrote:
no, they are not deflecting the call.
they are answering the call and making a new one to the mobile phone.
RDNIS is empty.
the main problem are not the redirected calls, but 'normal' calls from
there showing the trunk CLID instead of the trunk
On Fri, 4 Feb 2005, Remco Barende wrote:
Very regularly asterisk seems to lose connectivity with the ISDN line. If
you try to call in you get the information tone that the number is not in
use. Outbound calls do stil work however. Unloading the modules and
reloading them and start/stop
On Fri, 4 Feb 2005, Frank Sautter wrote:
Message type: SETUP (5)
[snip]
[6c 0c 21 80 31 37 32 39 38 37 36 35 34 33]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation
On Fri, 4 Feb 2005, Jeremy Kitchen wrote:
so many phone companies are offering free long distance anymore to compete
with cell phones (and of course, most, if not all cell phones are free long
distance) that I think we'll get to a point where anyone in the US will be
able to call anyone in
On Fri, 4 Feb 2005, Paul Brock wrote:
This is rather weird? What network do you receive this from? Neither
ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the Calling
Number IE among those that may be repeated.
q.931 (and q.931e) traces include both called number and calling
On Thu, 3 Feb 2005, Martin List-Petersen wrote:
On tor, 2005-02-03 at 20:02 +0100, Stefan Gofferje wrote:
Actually, that is wrong. Any company that uses a 0180x is just single
minded and purely focused on the german marked. Funny as hell, these
numbers are simply blocked by nearly any
On Wed, 2 Feb 2005, Robert Rozman wrote:
I wonder how SayNumber can handle international numbers (I can translate
numbers - but would also need different order...).
I guess that solution for German language will also work in our native
language.
I think SayNumber already handles the
On Tue, 1 Feb 2005, Vladyslav wrote:
There is option automon = *1 in features.conf
As I understand when *1 pressed during conversation = recording should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that feature working?
You have to set the w or
On Tue, 1 Feb 2005, Andrew Kohlsmith wrote:
I am seeing this too, only I'm using 100% PRI. I'm not sure how it would be
a
Zap problem since the Zap driver doesn't see any digits when you're dealing
with PRI.
Several tries later and it suddenly works It's very sporadic.
Do you
On Mon, 31 Jan 2005, Eric Bishop wrote:
Do you have a config sample on how to handle digital PPP sessions in Asterisk?
No, but there may be examples in the wiki:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zapraspreview=3
http://www.digium.com/downloads/ppp.txt
On Mon, 31 Jan 2005, Remco Barende wrote:
What you want is impossible!
How can you expect Asterisk to play a message to the caller without
answering the phone?
It can be done on isdn connection and over VoIP links as well. The reverse
audio path is (can be) opened before the answer. The
On Mon, 31 Jan 2005, Alex Barnes wrote:
Also I was under the impression that in Europe calls are charged as soon
as you start ringing and not on pickup (this may be out of date as its
been a while since my school skiing trip ;-P )
Not in all countries at least. Sweden has always had the
On Mon, 31 Jan 2005, Frank Sautter wrote:
on our incoming E1-PRI from german telco Arcor the leading 0 for the
(area access code in europe) and the 00 (country accescode in europe)
are missing on incoming callerids.
only prepending a single 0 is not the solution as suggested by some
On Mon, 31 Jan 2005, Jeb Campbell wrote:
We are using Asterisk on a T1 (pri) to Bellsouth.
Calls are getting dropped calls every 60 minutes and we asked Bellsouth
to debug the line.
(Note, this was told to me and I have no idea what it means)
They said that we are sending an Audit and
On Sat, 29 Jan 2005, David Norton wrote:
Currently I only have 1 PRI which I am using for dial-in customers. The line
is connected to a Portmaster3. I have never used more than 10 concurrent
channels. The calls can be both analog or ISDN. It would be a waste to order
another PRI for my
On Sat, 29 Jan 2005, Matt Schulte wrote:
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we think?)
On Sat, 29 Jan 2005, Matt Schulte wrote:
Agh, what I meant was the echo is heard from the PSTN side. It seems
echo canceling on the T1 (going to channelbank) does nothing, I'm
assuming because the T1 is digital and the channelbank is the
traversal from digital to analog.
Still, echo
On Fri, 28 Jan 2005, Jeff Lists wrote:
I have a test install set up as follows
Grandstream 102 Asterisk X100P Adtran Express 3000 ---
ISDN line to PSTN
Most things I want to do work fine except I do have some intermittent
problems with an echo. I am assuming that this is
On Thu, 27 Jan 2005, Frank Sautter wrote:
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* from time to time (sometime within
On Thu, 27 Jan 2005, Frank Sautter wrote:
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten = _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
On Thu, 27 Jan 2005, Frank Sautter wrote:
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* some faxes from our analog fax
On Wed, 26 Jan 2005, Tobias Jönsson wrote:
On Tue, 25 Jan 2005, Peter Svensson wrote:
On Tue, 25 Jan 2005, Tobias Jönsson wrote:
No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier
1.0 releases too. Busy() may play a busy tone to the caller instead of
signalling busy
On Tue, 25 Jan 2005, Eric Wieling wrote:
Florian Overkamp wrote:
You can set a PRI_CAUSE variable. See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20PRI_CAUSE
This only works in CVS-HEAD. For production use just run Busy() in
the dialplan.
It was added to Asterisk
On Tue, 25 Jan 2005, Tobias Jönsson wrote:
No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0
releases too. Busy() may play a busy tone to the caller instead of
signalling busy so using PRI_CAUSE is much better in PRI or BRI
environment.
The behaviour of Busy() and
On Tue, 25 Jan 2005, David Brodbeck wrote:
From: Peter Svensson [mailto:[EMAIL PROTECTED]
The SmartUPS ups's from APC that are = 1kVA seem to be of a
lot better
quality then their smaller siblings. We have lost none of the 1kVA or
larger ups:es while several of the smaller ones have
On Wed, 26 Jan 2005, Klaus-Peter Junghanns wrote:
Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter:
the setup before:
Arcor TelCo PRI(E1) Ericsson BP250 PRI(E1)
the setup desired with asterisk spliced in:
Arcor TelCo PRI(E1) P1 asterisk
On Mon, 24 Jan 2005, Daniel Nyström wrote:
Daniel Nyström wrote:
And another question at the same time; what's really E1?
How is E1 devices connected? Seems like regular Cat5 cables, but it
problably ian't?
If anyone's using Adit 600, did they send all cables required for
On Mon, 24 Jan 2005, Andrew Kohlsmith wrote:
As far as integrating with a website or database -- that is a piece of cake.
Your backend logic just determines when a call is needed and gerates the
approprate .call file. Just remember to create it in /tmp or something,
close it and then
On Mon, 24 Jan 2005, steve szmidt wrote:
The days of shoddy UPS's are long gone, unless you always buy the cheapest
stuff you can find all the time. In which case you might be able to find
something crappy. APC gives good support and make decent UPS's at a decent
price.
The SmartUPS ups's
On Sun, 23 Jan 2005, Andrew Kohlsmith wrote:
Why would the heads come in contact with the platters on a powerfail? The
arms are very rigid -- the heads only float a few thousandths of an inch over
the platters -- something that I don't believe has anything to do with the
platters spinning
On Sat, 22 Jan 2005, Mike Dent wrote:
On Fri, 21 Jan 2005 19:25:06 -0500, Glenn Powers [EMAIL PROTECTED] wrote:
Mike Dent wrote:
What do you mean by provisioning?
loading the config files, with proxy servers, usernames, passwords, etc.
So basically its just a silly word for
On Fri, 21 Jan 2005, Daniel Nyström wrote:
Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router
using MGCP IP protocol, instead of controlling it through an E1.
Have anyone tried this configuration? How does MGCP works? I've tried to
search for it on Google, but I only
On Fri, 21 Jan 2005, Daniel Nyström wrote:
Do you think it's hearable? All communication will be on a dedicated
Fast Ethernet link (just a cross-over cable). And it will still use aLaw
codec (same as Euro ISDN afaik).
Since E1 = 2048Kbps and FastEth = 100Mbps, I concidered it fast enough
to
On Fri, 21 Jan 2005, Daniel Nyström wrote:
As CarrierAccess states, there can be potential mismatch regarding the
TDM signaling required to terminate the voice channels onto the FXS
cards.
I'm not sure I understand this fully.
You can run a number of signalling protocols over a channelized
On Thu, 20 Jan 2005, taf taffey wrote:
Does anyone know of a way to dial two different outbound numbers and
bridge them together using the Asterisk API?
Which api do you mean? There are at least two ways:
- Using a call file in the spool directory
- Using the originate command in the
On Thu, 20 Jan 2005, Steve Clark wrote:
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
On Tue, 18 Jan 2005, Daniel Nyström wrote:
What server hardware would you recommend for an Asterisk system which
are really critical? The additional hardware will probably be two digium
TE110P cards, and an Adit 600 platform.
If it's possible to run on -48VDC, It would be great!
Are
On Tue, 18 Jan 2005, Daniel Nyström wrote:
Are you using dual PSU's or something? What redundance do you have on
your system? Do you use any channel banks? Is it only telco connections
on the TE405P? I will use one E1 from the telco, and one from Asterisk
to the Adit. Is it enough with one
On Tue, 18 Jan 2005, Christopher L. Wade wrote:
Scott Stingel wrote:
The reason for an asterisk system to be able to work from -48v is really
only if the end-user customer requires it as part of a spec. - I would
think only a minority of sites would require this though.These days,
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