On March 13, 2005 09:57 am, Nigel Burgess wrote:
[door]
exten = s,1,Dial (SIP31,15)
exten = s,2,Playtones(dtmf)
However the call hangsup before trying to play the DTMF tone.
Make sure you use the 'g' flag in the Dial command to go on in the context
after a hangup. Now whether the tone will
On March 14, 2005 06:50 am, pixer wrote:
3: 0 XT-PIC t4xxp
Without loading the module the LED glows in red colour, but the moment we
load module, it goes off. (No red or green).
We ran zttool and tried to run a loop test, but zttool simply hung with the
message 'Looping
On March 14, 2005 06:43 am, Brett, Gary wrote:
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of
installing an e1/ISDN30 in my lab. I have two questions really, first does
anybody know of an effective
On March 14, 2005 04:20 pm, Eric Wieling wrote:
Skype does not interface with Asterisk in any way whatsoever. You
could just as well have asked if someone knows what RNA sequence 42 in
the turnip genome is for. About as many people on this list would be
familiar with that as would Skype.
On March 15, 2005 01:00 pm, Giudice, Salvatore wrote:
MySQL: Speed, Power and Precision
Now *that* is funny. Thank you for bringing some humour to the list. Now
take the rest of this email and file it under FUD and exaggeration on MySQL's
capabilities, especially the benchmarks.
-A.
On March 15, 2005 02:21 pm, Giudice, Salvatore wrote:
Sticks and stone still break my bones, but PostgreSQL is still a dog.
Until you actually show some benchmarks where the tests are clearly documented
and Postgres is properly tuned, you're spreading FUD. Your testing should
also demonstrate
On March 15, 2005 06:04 pm, Giudice, Salvatore wrote:
commercial licensing AND has a real enterprise class support structure
behind it, or are you going to run with PostgreSQL (bow wow) distributed
under a BSD license with some mom and pop support shops and some mailing
It's time to put up or
On March 16, 2005 05:57 am, pixer wrote:
I have following your advice and I have put this into /etc/lilo.conf
append = pci=noacpi
20: 0 IO-APIC-level t4xxp
modules (COM port, serial ports, etc), and shuffling the card around to a
different PCI slot, but unfortunately he does
On March 16, 2005 07:12 am, pixer wrote:
Unfortunately I have already also tried this, without results.
I do not know what to do any more..
Was it an entirely different motherboard (different manufacturer)? If so,
it's time to call Digium and open a ticket. It sounds like the card is DOA.
On March 18, 2005 07:08 pm, Mike Sander wrote:
But Budwieser tastes like water to most Australian beer drinkers.
No, it tastes like piss to pretty much everyone. They just have a great
marketing budget.
-A.
___
Asterisk-Users mailing list
On March 20, 2005 07:52 pm, [EMAIL PROTECTED] wrote:
who has purchased a V400 card from Varion ?
I need some help .
please help me .
Does Varion not provide any support for their products? I'm interested to
know why you chose them over Digium...
-A.
On March 21, 2005 01:07 pm, Kevin P. Fleming wrote:
Well, let's see.. 99.99% of the available VOIP hardware only support
SIP, MGCP and H.323, but not IAX2. Is that a good reason?
No. 95% of the marketplaces uses Windows. Drive the marketplace to use
better protocols.
IAX2 calls between
On March 22, 2005 03:08 pm, [EMAIL PROTECTED] wrote:
The phone in question is what I would consider to be a good-quality GE
two-line cordless telephone. Digium's guess is that it is putting power
on the telephone line and the card doesn't like that. They have given me
zero solution other
On March 23, 2005 08:25 am, Matt Schulte wrote:
Has anyone ever heard of this so called Dynamic Impedance matching on
the ADIT 600? I called their support and they've never heard of it. We
That's odd, I have always had excellent support from CAC. And FWIW I've never
had echo problems with
On March 23, 2005 07:37 am, Michael George wrote:
I was told to try changing PCI slots (I haven't had a chance to do that
yet), but since the TDM cannot share IRQs with anything else, changing
slots might just put it into a conflict situation. This one could be
sticky...
As I am learning
On March 23, 2005 08:05 am, Ernest Stokes wrote:
We are a small shop that had one t100p card, when it came time to
expand to a second card we found the price had been raised to $599
from $499 for the single port.
The 4 port cards from varion are $699 on special.
I believe that I can get it
On March 23, 2005 09:59 am, Anton Krall wrote:
But I get that the chan is unavailable eventhough I can make calls to that
channel. Is there any chatch?
The channels is defined as peer and Ialso tried doing a register on
iax.conf for that channel. Everything is registering ok and I CAN make the
On March 23, 2005 08:53 pm, Josh Alberts wrote:
Hello, I'd like to make it so that after 5 invalid attempts of entering
an extension, the Hangup command will be issued. How would I go about
doing this?
My guess would be a combination of
SetVar($[${VAR} + 1])
and GotoIf($[${VAR} 5 })
Not
On March 24, 2005 08:08 am, Rich Adamson wrote:
Then try the following in zapata.conf:
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
as a starting point for each fxo channel.
Does echotraining *improve* echo cancellation at all? All I've ever found it
to do is help the
On March 28, 2005 08:15 am, Matt Schulte wrote:
Thanks for the response, it's a rather simple setup. What worries me is
we're going into an old PBX, the channelbank goes 25pair about 20 feet
to a punchdown block. Then from the block goes to another block
(standard telco room layout) then to
On March 29, 2005 08:40 am, Richard Reina wrote:
This goes on continuously and no phones are ringing.
I am using a digium T1 card and ADIT 600.
Do you have the Adit600 configured correctly? It's not stuck in a test mode
or anything?
-A.
___
On March 30, 2005 05:24 am, Obihuan wrote:
My calls, depending the hour of the day, have diferent quality.
Sometimes I felt cuts in the conversation or lost the sound on one of
the end point.
All of the providers I tested had any kind of trouble.
Sounds like the trouble is on your end then.
On March 30, 2005 10:26 pm, Kristian Kielhofner wrote:
It is obvious that Asterisk/TDM support from Sangoma is (and has been)
secondary. Their cards support data like no other. Excellent. Voice,
on the other hand, appears to be immature.
I respectfully disagree. Sangoma's voice
On March 31, 2005 08:53 am, David Hajek wrote:
how to use Asterisk where I need to have lets say 40 analog lines. Any
ideas?
A pair of TE110Ps or a TE405P and an Adit600. This will get you any
combination of up to 48 ports, in groups of 8.
-A.
___
On March 31, 2005 10:17 am, Rich Adamson wrote:
I'll second that one for sure. Maybe someone can talk Sangoma into
developing a competing TDM04b card? ;)
Actually I've found the TDM4XXP very good lately -- FXS and FXO.
-A.
___
Asterisk-Users mailing
On March 31, 2005 10:26 am, Chuck Bunn wrote:
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum and
essentially keeping my own
On March 31, 2005 01:19 pm, [EMAIL PROTECTED] wrote:
Any decent on-line forum would be much better than these digium email
lists. The lists are poorly formatted, there is no easy way to post code,
you cannot neatly quote anyone, the s earch function in the archive is
elementary at best, there
On March 31, 2005 02:34 pm, Tim Bass wrote:
Email does not build an on-line community. The search function is
primitive at best. Just like Mr. Bunn said in his original email, email
lists for support are a dinosaur, and people who have moved beyond
dinosaurs are considered intelligent in
On March 31, 2005 01:07 pm, Chuck Bunn wrote:
I am curious just what the advantages of an email forum over an online
one. Thanks for the search tip, but it is still an annoying way to
This has been hashed over time and time again.
In a nutshell:
- offline access
- threaded access
- ease of
On March 31, 2005 11:58 am, Eric Wieling aka ManxPower wrote:
Maybe I can use procmail to send an automated message to anyone that
posts a message in HTML. 8-)
Until you hit your first out-of-office autoreply that sends HTML... :-)
-A.
___
On March 31, 2005 05:11 pm, Tim Bass wrote:
The UNIX Forums have over 28 thousand registered users. I have many
years of experience in both email lists and on line forums and I can tell
you without a doubt that on-line forums are far superior to email lists.
There is no comparison.
Prove
On March 31, 2005 11:28 pm, Tim Bass wrote:
The discussion should not be laced with profanity, you should treat this
list and others like there are women on the list and try to be polite so
everyone is comfortable. Most professionals discuss matters in a way where
everyone is comfortable to
On April 1, 2005 09:14 am, Tim Bass wrote:
(2) When you registered (if you registered two years ago, for example, you
receive mail in a large list before someone, say, who registered a month
ago);
You really have very little understanding of mailing list technology. Please,
do some basic
On April 1, 2005 11:07 am, Tim Bass wrote:
one, that is not such a serious issue, vice having a bit of profanity laced
discussions with women and students in the community.
I have to ask -- you keep harping about women and students -- why are they any
different from any other person who
On April 1, 2005 01:44 pm, Brian Litzinger wrote:
Made the suggested changes. Called in via SIP and Cell Phone. Still
no response to DTMF.
It's time to get lowlevel.
iax2 debug and look for received DTMF digit '3' or something. tethereal
will also show you the IAX2 IEs for DTMF.
If you
On April 1, 2005 02:44 pm, Brandon Patterson wrote:
Level 3 does DTMF inband DTMF. Period.
Not on IAX2 it doesn't, and not on any kind of compressed codec with SIP it
doesn't.
-A.
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On April 1, 2005 12:19 pm, Paul wrote:
I'd like to setup my Asterisk box to receive a call on the incoming POTS
line and immediately redirect back out to connect to another phone number.
Im thinking I could use either the threeway feature of that POTS line, or a
second POTS connected to a
On April 2, 2005 07:30 pm, Tore Hansen wrote:
Having read a number of mailing list memos on this subject, there is
much to be said for having a proper support forum BBS, rather than
getting an awkward long memo with a string of messages every 3 to 5 hours.
awkward long memo with a string of
On April 3, 2005 08:13 am, Tim Pushor wrote:
To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes, although one you learn PF you will never go back!
I've never seen the great advantage to pf over ip and
On April 4, 2005 05:58 pm, Paul Belanger wrote:
I have recently purchased a TE405P from Digium and have noticed the board
seems to take ~5 mins on a fresh boot (Slackware) to start (IE: Until leds
on back start flashing). Is this normal? Can it help speed this up?
I have the exact same
On April 4, 2005 06:40 pm, Sean Kennedy wrote:
If I have 10 copper wires coming in from the phone company, and I want
to get a channel bank that will turn those into a t1 to feed into an *
box with appropriate hardware, do I want an FXS or FXO channel bank?
you want an FXO channel bank, or at
On April 4, 2005 08:01 pm, Derrick Knight wrote:
Are you viewing the output to the console as you are booting the system?
I suspect that it has nothing to do with the Digium drivers and more to
do with other features of Slackware such as attempting to autodetect USB
or 1394 devices. If you
On April 5, 2005 10:30 am, Race Vanderdecken wrote:
You have to set asterisk up to look like the Vonage switch.
You have to spoof the switch.
Sure, if you have their RSA private key. Go for it. If you tweak it out
before it ever contacts Vonage you've got a chance, just like you can do
On April 6, 2005 03:47 pm, Gilbert Abboud wrote:
I created a .call file as mentioned in the WiKi but when i place it in
/var/spool/asterisk/outgoing, the Asterisk console shows unknown keyword
for all the keywords used in the .call file (i.e channel, context,
extension,...). Any ideas why?
On April 7, 2005 05:50 am, Roy Sigurd Karlsbakk wrote:
Does anyone have any details on the actual differences of using Sangoma
PRI cards as compared to the TE410? How are CPU usage, interrupt load?
Are there other diffferences?
They are completely different beasts; the details on the actual
On April 7, 2005 09:01 am, Tony Mountifield wrote:
Do the Sangoma cards use zaptel-compatible drivers or something different?
Do they provide a timing source in the same way as Digium cards do?
Yes.
-A.
___
Asterisk-Users mailing list
On April 7, 2005 01:44 pm, Ian Pattison wrote:
Ok... I've done a bit of emperical testing but don't really know what the
results mean. I'm starting to think I need an oscilloscope to measure this
properly. All I have is a DMM, I'm measuring on both the AC and DC
scales...
AC
On April 7, 2005 01:53 pm, Craig Guy wrote:
the server they're going into (Dell poweredge 750's). When a GPL'd
hardware design costs more than an entire proprietary server (including
chassis, motherboard, dual hard disks and remote access card) then there is
something very wrong in the
On April 7, 2005 04:02 pm, Tony Mountifield wrote:
That's a pity and I'm not convinced the assertion is true.
Andrew, if you read this, is your hacksawed TE405P board still in a 3.3V
slot and still working?
I have no intention of hacksawing a board myself, but the findings of a
year ago
On April 7, 2005 04:42 pm, Tony Mountifield wrote:
OK, I'd been told this chip could support both 3.3V and 5V, but from what
you're saying, it sounds like it can be set up to support 3.3V OR 5V, but
not both at once officially. Of course, when selling product it is
prudent only to work
On April 7, 2005 09:38 pm, Ugur GUNCER wrote:
How can i set asterisk for when call came from pri ring once then answer
pri call.
In now call cames from pri then asterisk directly answering pri call
without ringing. Then my carries hangup call because they said your box is
answer without
On April 8, 2005 08:30 pm, Aaron O'Hara wrote:
My understanding is that a standard residential/business phone line
carries the signal over 2 wires. Your 4-wire RJ11 wiring supports 2
phone lines. Given that each line takes 2 wires, and there are 8 wires
in an FXO port, can I conceivably
On April 8, 2005 10:59 pm, Carlos Rojas wrote:
You have well formed your file zapata.conf?
Why would an ATA use anything in zapata.conf? An ATA typically takes an
analogue interface and converts it to an IAX or SIP device. I'd suggest
looking at his iax.conf or sip.conf, depending on the
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please God, if you can
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
lspci)?
TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev
01)
I imagine the TE410 and TE110 are both also similarly lspci'd.
-A.
On April 10, 2005 10:50 am, Rich Adamson wrote:
One way to do that is simply:
[ snippage of simply mv'ing to a backup ]
That is *precisely* how I do it for small changes, and for full-out upgrades,
I have the old slackware packages standing by.
-A.
On April 10, 2005 04:47 am, cmisip wrote:
I got this from the voip wiki but the original script didn't seem to
work right so I fiddled with it a little bit. I am no expert so maybe
someone can look at it for errors. This is for my cable connection. So
far asterisk seems to use 1:10 while
On April 10, 2005 05:03 pm, John Novack wrote:
As to that hold button. What idiot decided it should be in the middle
of a row of keys, the same size as the others, and not a bright color?
Maybe me; I have no desire for a bright 'hold' button. Give me the Norstar
system where 'Rls' (release,
On April 10, 2005 12:01 pm, Matthew Boehm wrote:
I have a TE405P and mine shows up as Xilinx but a lvl 2 tech a digium says
it still uses the TigerJet chipset. That's why it won't work in my Dell.
I'll paypal you US$100 if you can find a TJ320 chip on either the TE410P or
TE405P. It doesn't
On April 11, 2005 10:08 am, Sean Kennedy wrote:
Honestly, the best script I've ever found is the wondershaper script (
google it ). I tried the correct one posted in this thread, tried
modifying it, but in the end I just used wondershaper.
:-) I started out with wshaper and just didn't like
On April 11, 2005 03:17 pm, Andres wrote:
Can you confirm if there will be some sort of DSP daughther card add on
of some sort for the DS3000 so that we can run G729 transcoding? I
don't see how the DS3 interface would be usefull unless we could offload
transcoding stuff to onboard DSPs. Or
On April 11, 2005 06:43 pm, Bicom Systems wrote:
Come June/July an USB/PCI DSP cost effective solution should be available
to address this issues. It will transcode nearly all codec's.
I am not in position to reveal the company name
at this stage unless MN wants to speak up :)
secondary card
On April 12, 2005 11:59 am, Joel Jn-Francois wrote:
I get an echo only from the caller end when I am making calls. I only get
it for some VOIP providers. I am using asterisk Asterisk
CVS-v1-0-03/26/05-16:54:47 and Grandstream HandyTone 486 and 488. My
default codec is ulaw. Is there any way
This is not a development question; it's actually a -biz question but I'm not
on that list so this'll have to do.
On April 12, 2005 05:15 pm, Tom Dickenson wrote:
Anyone know a good IAX Long Distance Trunking service that is not monthly?
Kind of like a calling card charge up service?!
There
On April 13, 2005 02:40 am, Me wrote:
== Primary D-Channel on span 1 down
Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No
D-channels available! Using Primary on channel anyway 24!
The telco hasn't turned up your D channel yet.
If I change signalling to pri_net the
On April 13, 2005 12:35 am, amna saleem wrote:
I was wondering if the i extension works ,i mean i have included
this in my extensions.conf ie
exten = i,1,Answer
exten = i,2,Playback(pbx-invalid)
exten = i,3,Hangup
You've already answered the call; no need to answer again, although it won't
On April 12, 2005 11:36 pm, Kevin P. Fleming wrote:
Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI
bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per
Yes, but then what are you doing with it? You're shuttling the new data
to/from a network card in a
On April 13, 2005 10:57 am, Kevin P. Fleming wrote:
Very true; realistically, modern PC hardware has more than enough
bandwidth to do what is required. The real issue is timing, based on
contention for resources, and how that impacts latency. The existing
boxes out there (not PCs) that handle
On April 13, 2005 11:20 am, Ezabi wrote:
Recently I've been having strange behaviour on my calls to PSTN, when
dialing from any extension to the PSTN through ZAP the line hangs up
after exactly 3:03 mins., tried to look everywhere for a string defining
this timing but of no use, I even set the
On April 12, 2005 07:21 pm, Matt Fredrickson wrote:
If you're using a TDM card, you might see if the fxotune program will help.
It does impedance tuning of the card and finds the line impedance that has
the lowest mean power (i.e. least echo). I've been working on it for a
while and some
On April 13, 2005 01:18 pm, chawki hammoud wrote:
I placed a call through a voip provider from my
console CLI. Then i ran a test iax2 show netast and
here what i get when at the beginning of the call
EMOTE
ChannelRTT Jit Del Lost %
On April 13, 2005 03:42 pm, Trent Tuggle wrote:
The symptom is a loud, brief buzz, almost exactly every 6 seconds, on
the dot. It is only audible to remote parties, when I use an analog
phone connected to my Digium TDM card. All other audio through my
Asterisk box is fine, including SIP
On April 13, 2005 03:45 pm, Mystery Glitch wrote:
In my [incoming] context I have something like this:
exten = 8885861575,1,Macro(vrforward,${EXTEN},8136361451)
Make sure you have a 'h' extension defined that just hangs up.
-A.
___
Asterisk-Users
On April 14, 2005 08:31 am, Eric Wieling wrote:
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure
the dialplan
On April 14, 2005 09:42 am, Eric Wieling wrote:
exten = h will not be called unless the channel has ALREADY hung up.
I understand that, which is why I'm still suggesting a WARNING and not an
error.
Something like No need to execute Hangup from the h exten, line is already
hung up
-A.
On April 14, 2005 10:44 am, Rich Adamson wrote:
Sounds like its time to swap motherboards. :(
I just wish that the PCI bridges on the TDM and TExxx cards would allow you to
utilize INTA,INTB,INTC or INTD... if the mobo's fucked up at least let the
card route around the damage. I'm not sure
On April 14, 2005 12:51 pm, Rich Adamson wrote:
Maybe because some motherboard designs have copper trace from
the interrupt controller to the individual pci slots?
There are four INT# lines on every PCI slot. INTA of slot1 is supposed to be
routed to INTB of slot2, INTC of 3, INTD of 4. INTB
PLEASE!! trim these replies!!!
-A.
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To UNSUBSCRIBE or update options visit:
PLEASE TRIM YOUR POSTS, it takes less than 30 seconds!
On April 14, 2005 04:27 pm, Damon Estep wrote:
The user stated that the line is PRI ISDN, not likely to be a physical
short as that would take the digital line out, not produce crosstalk,
had to be a switching issues with the telco or *,
On April 14, 2005 05:48 pm, Michael Di Martino wrote:
Call come in over the pots lines however Outbound goes out thru the VOIP
provider.
However looking at the configs I cannot figure out what controls how
call are sent out.
In other words where in the config files does it determine that all
On April 14, 2005 06:34 pm, chawki hammoud wrote:
I previously posted about the huge latency introduced
by iax2. It is a problem introduced by the codec. in
iax2.conf, i disllowed=all and allow=gsm and the RTT
is the same as I do ping shell command. When i change
from gsm to ulaw or alaw,
I'm Andrew.
On April 14, 2005 10:01 pm, Qiao Yuansong wrote:
My asterisk box and sip phone are not behind a nat, the sip phone and
asterisk box are connected by LAN, so the delay is not caused by network
congestion, and furthermore, there is no delay from sip to pstn.
[sip
Please don't post HTML to the list, and PLEASE TRIM your posts! Maybe I'm
getting oversensitive to this lately but the sheer volume of bandwidth wasted
due to people not taking 30 seconds to trim replies is staggering! My reply
is an example of proper reply trimming; only the essential bits
On December 24, 2004 08:48 am, Patrick wrote:
I read somewhere that to be able to hear the fax tones you need to give
Asterisk 1 or 2 seconds to be able to pick them up. So put a Wait(1) or
Wait(2) in your dialplan (directly after Answer would make sense to me)
so Asterisk can figure out it's
On December 24, 2004 09:17 pm, Michael Giagnocavo wrote:
MS SQL 2005 Express is probably the best free DB out there? And I run lots
of Mono code just fine...
*cough* okay. Sure. Whatever you say...
-A.
___
Asterisk-Users mailing list
On December 25, 2004 07:29 am, Jean-Michel Hiver wrote:
To answer the real question which is on the back of your head, unless
you're lucky you'll probably have to do a lot of fiddling around no
matter which distro you choose to get * to work...
I have no idea what fiddling you're talking about
On December 25, 2004 02:07 pm, Lane wrote:
I can make asterisk run, and I can connect to it using a software SIP
phone. I can even hear the demo, but it is wa choppy. So I figure
that the choppiness will diminish once I can get the FXS module to load.
Remove the card entirely and run
On December 26, 2004 07:40 pm, Michael Di Martino wrote:
Regards,
Michael Di Martino
Director of MIS
The telx Group
Office: 212 480 3300 X.2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld
We're impressed. Really we are.
On December 28, 2004 11:44 am, Rich Adamson wrote:
I would seriously doubt that you can actually squeeze 12 channels through
that dsl and obtain anything reasonable for quality, regardless of which
asterisk codec you choose. But, it certainly would not be that hard to
test it and validate
On December 28, 2004 06:32 pm, mohammad wrote:
How can a user block his caller-id in Astersik?
show application SetCallerPres
-= Info about application 'SetCallerPres' =-
[Synopsis]:
Set CallerID Presentation
[Description]:
SetCallerPres(presentation): Set Caller*ID presentation on
a call
On December 29, 2004 06:25 pm, PHP Mechanic wrote:
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or
hold to work when using asterisk, which means I can't use three way calling
or the call waiting functions. I've tried using combinations of hook flash
button and *0
On December 29, 2004 07:05 pm, Richard Reina wrote:
For threeway calling (analog phone) I just hit the
flash button get a dial tone, dial the number and hit
the flash key again.
You're missing the point.
POTS - Asterisk - Analog phone
He's got call waiting/threeway calling on his POTS line
On December 29, 2004 07:52 pm, Andrew McRory wrote:
System is built on a SuperMicro motherboard with Serverworks chipset, IRQ
is not shared. Have a dialplan that worked for 8 months without errors,
tried reverting to older release then upgraded to 1.0.3 stable release,
currently running on
On December 29, 2004 09:42 pm, Andrew McRory wrote:
Well it is hard to go back to a specific configuration since I have used
the system to test the rpm packages I compile.
Yikes.
Nothing like using a production server for testing, eh? I have reverted to
a (actually several) pre 1.0 release
On December 29, 2004 11:03 pm, Sudhir Kumar wrote:
1. When it dials out, many times the digits are not properly recognized
by telco as I hear the announcement please check the number and dial
again although I see on the screen that the dialed number is correct.
I would try to stretch out the
On January 1, 2005 04:09 pm, Rich Adamson wrote:
b. don't ever post anything to the -dev list regarding a TDM card as
that is NOT the forum for digium cards or drivers,
Eh? If you're hacking on the code for wctdm, -dev is most certainly an
appropriate place to post. If you're just going
On January 1, 2005 06:24 pm, Steven Critchfield wrote:
1. Power alarms. WTF does that mean? Wish I had some support docs.
2. On bootup, Excessive leakage module x, ProSLIC failed Auto
Configuration. Again, WTF? Reboot and it's ok. But, just a reboot
after driving 100+ miles to the
On January 3, 2005 04:15 am, Nathan Alberti wrote:
Is it normal for the following to occur hourly on an E1 PRI ?
-- B-channel 0/1 successfully restarted on span 1
...
Yes, perfectly normal.
-A.
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On January 3, 2005 05:35 am, Bob Goddard wrote:
What matters is the volts, amps and the voltage drop when the rails
are put under load. You have to ask yourself how many amps does the
mb require on each rail and can the psu supply it? The total power
supplied by the psu means nothing if it
On January 3, 2005 07:53 am, Rich Adamson wrote:
Not sure why the restart code was added, but it was some time ago
(maybe up to a year ago). I'd have to guess that it was added to
address an issue back then and probably really isn't needed any
more (but that is a 'guess').
IIRC it was added
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