I just bought a Pingtel Xpressa from VoipSupply for use with
Asterisk. I know that Pingtel has sold off their hardphone line and
discontinued support for their phones, but I'd like to track down a
few of the Java applications that they distributed before they went
away, specifically their
On Sep 21, 2005, at 1:46 AM, Zoa wrote:
True... But i tried several brands of cards, and several drivers, the
dual nic gigabit intel card was a lot better than all the other
combinations i tried.
Fun fact: short of buying a 10gigE card, your best performance will
probably be from a cheap
On Apr 19, 2005, at 12:30 PM, Damian Funnell wrote:
Thanks for your post, it's most insightful. It certainly puts a
pretty large dent in my confidence in the TDM for commercial use -
imagine if there was more than one TDM in a system (especially with a
RAID adapter).
Running a PABX without
On Apr 8, 2005, at 6:28 AM, Steve Mann wrote:
I was quoted about $700/month if I was within my downtown area for
ISDN PRI.
So your price is in the right ball park.
XO quoted me $500 for a PRI in downtown Seattle about 6 months ago. I
suspect that you could beat that with a bit of shopping
On Apr 8, 2005, at 9:12 AM, Maik Hassel wrote:
Hello everybody!
Does anybody have any recommondations for single port PRI cards that
work tohether with Asterisk? I have a system running using orinary
analog lines right now and we are thinking of switching over. I have
no idea what a good price
On Mar 23, 2005, at 9:54 AM, Parker, Blake (MIS) wrote:
If I get DID inbound and LD outbound service from a VoIP provider and
route all calls to them, would all calls even calls that would normally
be local, be considered ld and thus charged the per minute fee?
Generally, although it depends on
On Mar 18, 2005, at 1:41 PM, Reuben Grech wrote:
I am using an ASUS P4P800 Deluxe which is a motherboard that has been
tested with Asterisk. I am unsure about the PCI slot and IRQ settings
in the BIOS and am having problems with a WCFXO card sharing with
other devices. Any suggestions on the
On Mar 17, 2005, at 1:36 PM, Scott Nelson wrote:
I groked the answer and it is more than pseudo code, it actually
works. I'm using them in my extensions.conf now!
Priority n means 1+last defined priority. Putting a name inside
parens defines that name.
So, in the example you sent, checkavail
On Mar 16, 2005, at 3:31 PM, Richard J. Sears wrote:
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows Toll
Free
Call and will not give me the calling party's caller ID info.
Is this just something
On Mar 16, 2005, at 3:49 PM, Richard J. Sears wrote:
So how do I capture the caller number..? All I see is the 866 number
that NuFone has assigned to me.
What's in ${CALLERIDNUM}? It should also show up in the CDR. For me,
it works exactly like caller ID information generated by my X100P,
On Mar 11, 2005, at 7:20 PM, Scott Laird wrote:
On Mar 11, 2005, at 5:31 PM, TC wrote:
I was just reading the release notes for the latest SPA-841 firmware,
and noticed that Sipura added support for SIP-B to this release.
This apparently adds support for bridged line appearances, parking
softkeys
On Mar 14, 2005, at 5:20 AM, Doug Lytle wrote:
For those that are interested, I was just out on the Cisco site and
noticed that they had released firmware 7.4 as of March 11th for the
7940/7960 phones.
I don't see any major changes in the release notes--mostly small bug
fixes. They fixed some
I was just reading the release notes for the latest SPA-841 firmware,
and noticed that Sipura added support for SIP-B to this release.
This apparently adds support for bridged line appearances, parking
softkeys, called party ID, external missed call summary support, and a
handful of other
On Mar 11, 2005, at 5:31 PM, TC wrote:
I was just reading the release notes for the latest SPA-841 firmware,
and noticed that Sipura added support for SIP-B to this release.
This apparently adds support for bridged line appearances, parking
softkeys, called party ID, external missed call summary
On Mar 10, 2005, at 7:06 AM, Mark Halverson wrote:
Anyone know how many simultaneous calls you can receive on a NuFone
DID?
If you're talking about an 800-number DID, then I don't think there's a
limit, at least until you start saturating their capacity. There have
been reports of users with
It might be more interesting to see how well it works with Xen--it's
supposed to be quite a bit faster then UML, *and* you can delegate
specific PCI devices to individual virtual systems. That means that
you could probably use Zap cards, not just SIP. I have no idea how
well it'd actually
On Feb 10, 2005, at 2:39 PM, jurgen wrote:
This is interesting - it's something that I've been considering doing
for the Asterisk rollout at my company. We don't have enough Ethernet
ports and I'm not thrilled about the expense of re-wiring the place.
Have you tried D-Link's dual-channel gear for
On Feb 10, 2005, at 5:57 PM, Eric Bishop wrote:
Hi all,
Can someone give me a simple rational explanation why a $5 analog
handset gives me no echo whatsoever on an analog PSTN line, but
PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require
software-based echo cancellation. Surely a
On Feb 4, 2005, at 11:21 AM, Spencer Nassar wrote:
I've been doing a lot of background reading/searching of this list,
voip-info.org, and Google, looking to define a good candidate for a
server platform. I'm very interested in thoughts from others! So
here goes...
Axiom 1: if you are not
On Jan 25, 2005, at 1:22 PM, Eric Wieling wrote:
Adam Robins wrote:
The TE410P is a T1/E1 card. I need the card for POTS lines. Is there
also a TDM410P that does not appear on the Digium web site?
The TDM400P only works in standard PCI 2.2. Not PCI-X, not
PCI-Extreme, not PCI-64bit.
Huh? It
On Jan 23, 2005, at 7:30 PM, Nick Bachmann wrote:
As I understand, if HD activity is minimal, the probability of HD
failure is significantly reduced.
HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD
manufacturers have done head
On Jan 17, 2005, at 10:20 PM, [EMAIL PROTECTED] wrote:
- What size server will I need? Assume for now a pair of quad-T1
cards, 2 T1s
incoming, and 5 channel banks. Shouldn't require much horsepower
since it's T1
- T1 switching
You might want to consider breaking this up a bit, just for ease of
On Dec 24, 2004, at 5:44 AM, Brian Wilkins wrote:
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
Don't use _., it matches s, h, i, and all of the other 1-letter
extensions. Use _X. instead.
Scott
I have a problem with voicemail recording dialtone on hangups. From
time to time, callers hang up without leaving a massage, so Asterisk
records around 100 seconds worth of dialtone instead. I've seen this
off and on for almost 9 months now, it's currently happening with 1.0.2
on Linux
On Dec 16, 2004, at 10:34 AM, Randy MacKay wrote:
I have a Cisco 7960 phone. I cannot seem to use the settings button
to get
into the phone to change the TFTP server. I've set up a DHCP Server,
TFTP
Server with the same address, and the phone requests the address of
0.0.0.0,
the server offers
On Nov 24, 2004, at 4:18 PM, [EMAIL PROTECTED] wrote:
Is anyone succesfully running Asterisk behind verizon residential DSL?
I seem to
be having some problems with my Asterisk server switching to Verizon.
I'm
attempting to do some troubleshooting, but I'm really interested in
knowing of
On Nov 24, 2004, at 6:31 PM, [EMAIL PROTECTED] wrote:
Quoting Scott Laird [EMAIL PROTECTED]:
On Nov 24, 2004, at 4:18 PM, [EMAIL PROTECTED] wrote:
Is anyone succesfully running Asterisk behind verizon residential
DSL?
I seem to
be having some problems with my Asterisk server switching to Verizon
On Nov 18, 2004, at 8:25 PM, Steven Critchfield wrote:
Could someone get their hands on the driver to give it a good look and
inform of licensing. IT mentions linux, and it mentions that it is
channelized down to 672 DS0s. Sounds like the perfect card.
Also, since you can get PCI-PMC carrier
On Nov 4, 2004, at 9:23 AM, Richard Reina wrote:
Thank you very much for your thoghtful and thorough
response.
I guess I don't wan't to set up * to behave like a key
system, thank godness, I just want to be able to
juggle calls which it sould like Asterisk can do fine.
Just to clarify though, can
On Nov 4, 2004, at 10:28 AM, Chris Goodwin wrote:
Hi everyone,
I have a question regarding the use of callerID and call forwarding.
When I forward any of my Zap extensions in the office to an outside
line, such as a cell phone, the callerID info shows up as originating
from that office phone,
On Nov 3, 2004, at 4:16 PM, Ben Greear wrote:
Hello!
I have a Grandstream and a Cisco SIP phone, and I'm trying to make
a call between them. I added this to my sip.conf:
; Grandstream
[1001]
type=friend
host=dynamic
; cisco phone
[1002]
type=friend
host=dynamic
First, what's in your
On Oct 25, 2004, at 8:09 AM, Kevin P. Fleming wrote:
Shawn Dillon wrote:
The issue is this: How can I have a phone number in a city over 1000
miles connect to the Asterisk box in an economical way? I have only
tested the Asterisk box with a TDM11B and have no real experience with
T1's . Would they
On Oct 25, 2004, at 8:35 AM, Jay Milk wrote:
Thanks for the reply -- how well is this documented? Is information
available from Cisco to endusers, or is this a big-money affair only?
Publically documented but hard to find. Try these:
http://www.cisco.com/global/FR/documents/pdfs/ciscotheque/
Does anyone have any experience with running Asterisk on dedicated
servers from any of the cheap hosting providers, like 11?
I'd like to get my asterisk/mail/web server out of my house. There
isn't a whole lot of traffic involved, but I'd rather not end up with
someplace that *utterly*
On Oct 20, 2004, at 5:37 AM, Theo Zourzouvillys wrote:
after a couple of days work banging my head against the wall (bloody
standards
my arse), i've got chan_bluetooth to a point where it's starting to
function
- certianly more than just proof of concept now.
Cool, I've been looking forward to
On Oct 20, 2004, at 1:47 PM, Matt Hess wrote:
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell
switches only to give up on them because the dell switches just didn't
perform. Yes, price-wise they look good.. but
On Oct 20, 2004, at 3:38 PM, Michael Welter wrote:
Kristian Kielhofner wrote:
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
The reason I asked is because 802.1q isn't mentioned in the product
literature.
.1q is VLAN
On Oct 20, 2004, at 4:17 PM, Jay Wilton wrote:
What is the most important feature for VoIP quality:
latency, qos, vlan? I'm leaning towards least latency with
qos and/or vlans at the linux router. Might be my best
shot for an inexpensive gig switch ($100).
I have only seen the qos (802.1p) in
On Oct 18, 2004, at 11:58 AM, Martin McCormick wrote:
What kind of user interface is most common when managing an
Asterisk PBX? If this question sounds odd, here is what I am after.
Text config files and a command-line interface or two. There are a
couple GUIs around, but they're
On Sep 24, 2004, at 10:12 AM, Cirelle Enterprises wrote:
Has anybody been able to get in touch with anybody at digium today?
I suspect that they're all at Astricon.
Scott
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On Sep 20, 2004, at 2:14 PM, Wiley E. Siler wrote:
Here you go... No extension required
From extensions.conf
;--
; VOICEMAIL ENTRY INTO SYSTEM
;--
exten = 8,1,Answer
exten = 8,2,Wait(1)
exten =
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote:
we have a community of people on an * box that use SIP softphones to
talk each
other. Can you suggest me the quickest and simple way to let someone
know who
is online without have to call one by one the persons to look if they
are
present or
On Sep 7, 2004, at 7:15 PM, Chris wrote:
Asterisk never ever uses TCP for IAX or IAX2. It's ALWAYS UDP. I
don't
believe Asterisk supports SIP over TCP either. Heck, the manager port
is the only thing that uses TCP that I know of with Asterisk.
Hmmm I wonder why I had the impression that it
On Sep 7, 2004, at 4:43 PM, Chris Shaw wrote:
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Tuesday, September 07, 2004 4:39 PM
Subject: RE: [Asterisk-Users] MeetMe without ZAP?
Matthew
On Sep 3, 2004, at 4:47 PM, Chris Shaw wrote:
Ok Way OT, I didn't mean to get into a religious debate, I like the
Intel
cards, I have several of them and recommend them to my friends, etc...
Be that as it may... This was using these cards in a software bridge...
significantly more traffic than
On Sep 3, 2004, at 10:12 AM, Kevin Walsh wrote:
Competition is a good thing, in my view.
I didn't find out about the non-Digium X100P cards until after I'd
bought mine (for use at home). If I'd known then I probably would have
avoided the massive markup and bought one of the clones. These days,
On Sep 3, 2004, at 3:17 PM, Chris Shaw wrote:
The drivers have gotten much better, but yes, up until about 2.4.22 it
used
to hard lock my server every 24 hours or so under heavy packet loads...
Remember what I said about it being a software bridge... Intel cards in
general are not known for being
On Sep 1, 2004, at 11:35 AM, Eric Wieling wrote:
On Wed, 2004-09-01 at 12:26, Glen Johnson wrote:
Does anyone have disconnect supervision working on their TDM400P or
X101P
cards (to/from the telco)?
Yes on all 4 my X100P/TDM400PFXO ports across 3 servers in two states.
It Just Works. Of course
On Aug 29, 2004, at 7:51 AM, [EMAIL PROTECTED] wrote:
Has anybody tried to integrate a mobile phone via blutooth in asterisk
PBX?
I believe the most things needed are just existing in open source. I
found a kbthandfree
(http://docs.kde.org/en/HEAD/kdeextragear-3/kdebluetooth/
On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network
performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces.
About
50% of the time, systems
On Aug 25, 2004, at 9:57 AM, spectro wrote:
IMHO, If you plan to use analog phones the cheapest is to buy a bunch
of sipuras instead of TDM40B. (TDM40B = 4 FXS for $300, $75 each;
sipura SPA2000 = 2 FXS for $100, $50 each)
It might be worth looking at the new Linksys PAP2 -- it's only $50, and
On Aug 19, 2004, at 10:00 AM, Ryan Courtnage wrote:
Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 19:31, Ryan Courtnage wrote:
Theoretically, I know it's possible, but is any using multiple
tdm400ps
(fxo) in single * box? In a production environment? Any gotchas
aside
form irq sharing?
On Aug 19, 2004, at 11:30 AM, Muiz Motani wrote:
Does anybody know what happened to the opencall.org website? I can't
get
into the home page or the ftp site.
His DNS servers seem to be down. opencall.org is served by
name[12].coppice.org, which are 202.76.92.17[23]. Neither one responds
to
On Aug 18, 2004, at 2:00 PM, Jorge Verastegui wrote:
Hi
I'm buying a new box and it brings the new PCI-e standard and not the
old PCI slots.
I would like to know if the Digium Wildcard TDM400P and
http://www.digium.com/index.php?menu=wildcard_tdm400p2Wildcard
TE405P will work with this PCI
On Aug 13, 2004, at 9:31 AM, Shawn Parker wrote:
I got a call from our Cisco rep today saying that they couldn't sell
just phones to anyone because if my ethernet isn't to exact spec...
then they won't work at all. I've read over the Wiki documentation
and it seems that the 79xx series phones
On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like businesses.
Imagine a telemarketeer script that dials lists of VoIP addresses.
On Aug 10, 2004, at 4:34 PM, Walt Reed wrote:
On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said:
Why stop there--you can beam pre-recorded messages to phones without a
person or phone line ever being involved. You could send hundreds of
calls per minute without paying for more
On Jul 27, 2004, at 10:14 PM, John Baker wrote:
Um, these phones are less than $300 a piece.
http://www.google.com/froogle?q=polycom+600scoring=psa=Nstart=10
Hard to find a leasing company for that small an amount, but I'm sure
they're out there.
John
They're that low? I've hard a hard time
On Jul 28, 2004, at 2:07 PM, Peter Svensson wrote:
On Wed, 28 Jul 2004, Chris Johnson wrote:
1) Convert one of our PRIs into a two way voice/data bundle.
2) Plug this PRI into a piece of equipment (AS5300?) which can
terminate both PPP, *AND* feed our Asterisk system. We'd transfer
We're interested in leasing roughly 15 Polycom IP-600 phones. Does
anyone have a vendor that they can recommend for this?
Scott
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On Jul 22, 2004, at 9:41 AM, Deon Rodden wrote:
I'm confused. In the end, overall, which is best for a T100P (or even a
TE405P) card? IDE or SCSI? Raid or No Raid?
I was anticpating putting a single Quad-Port TE405P inside a Dell
PowerEdge,
Dual 1.3ghz Processors, SCSI Hard Drive (No Raid). Was
On Jul 21, 2004, at 1:33 PM, Steven Critchfield wrote:
Software raid is bad. IDE hardware raid isn't much better. Software
raid
is always going to eat your system alive since the CPU has to be busy
with 2 or more writes as opposed to it's normal 1.
That hasn't been my experience at all. Frankly,
On Jul 21, 2004, at 4:53 PM, Joshua McClintock wrote:
Our production environment is using a 4 port 3ware 8500 series card
with
2 drives (mirrored) on the pstn (2 t1 cards) machine and an 8 port
3ware
8500 series with 8 drives (raid5) on the pbx/vm machine.
Flawless so far.
Why so many drives
On Jul 21, 2004, at 4:25 PM, Kevin P. Fleming wrote:
Scott Laird wrote:
That hasn't been my experience at all. Frankly, I've never seen a
cheap ($3k) hardware RAID controller that can touch software RAID's
performance on Linux, especially in challenging setups, like
RAID-5. Sure, software
On Jul 21, 2004, at 7:01 PM, Steven Critchfield wrote:
BTW, my raid card on my Dell 2450 had this output
nash5:/home/critch# hdparm -tT /dev/sda
/dev/sda:
Timing buffer-cache reads: 128 MB in 0.61 seconds =209.84 MB/sec
Timing buffered disk reads: 64 MB in 2.52 seconds = 25.40 MB/sec
and I
On Jul 20, 2004, at 11:31 AM, Marty Mastera wrote:
No need for that!...thanks for the info everyone...I'm going to start
keeping my eyes open for a WRT54G for a good deal somewhere!
CompUSA in Seattle had a sale on WRT54G boxes for $59 yesterday.
Scott
On Jul 20, 2004, at 1:58 PM, Carmi Weinzweig wrote:
They assign me a phone number (a value of $0.01 and $0.10) and let me
receive as many simultaneous calls as my bandwidth allows (using these
numbers every call absorbs a channel that costs between $4.35 and
$43.48).
What I would like is to be
On Jul 20, 2004, at 5:27 PM, Jay Milk wrote:
I wouldn't mind buying an entire exchange for $0.01/number. What's
that, like $10/month? Heck, I'd even pay $30-$40/month for this.
Where
can I sign up?
Since I'll probably be PRI and DID shopping in the next month or two,
roughly what *do* DIDs
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
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
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
On Jul 18, 2004, at 7:14 PM, [EMAIL PROTECTED] wrote:
On Sun, 18 Jul 2004, Michael Welter wrote:
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
On Jul 16, 2004, at 11:07 AM, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but...
What's common and not so
I've been following the list for months, and I have a working Asterisk
setup, but it'd be *really* useful to me at this point if someone could
summarize when Asterisk has echo problems and when it doesn't. For
instance, I usually hear a far-end echo when talking on my 7940, but
not when using
On Jul 15, 2004, at 4:47 PM, Joshua McClintock wrote:
We're using asterisk in production at my place. We have about 45 users
(33 snom 200 sip phones and the rest on soft phones). We're connected
to the telco via a pri (23 useable channels) using the digium t1 card.
We have another t1 card hooked
On Jul 13, 2004, at 5:13 AM, Andrew Kohlsmith wrote:
On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote:
exten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
That's why you would stick the members into a global
variable
You global variable is still unwieldy. All you did was move the
problem.
It
On Jul 9, 2004, at 7:29 AM, asterisk wrote:
I've attached the /proc/pci below, but I think it's hardware related,
not os
- the dell does not seem to recognise that there is a card in the
slot. Or
any slot I put it in :(
Thanks for the help, though.
It sounds like some sort of hot-plug PCI
I need to do some modem testing at work, just to make sure that I can
dial into an emergency serial console modem before I put it into a box
and ship it. Unfortunately, we don't have easy access to a pair of
POTS lines in the office right now. Looking around the test lab, we
have a Cisco
On Jul 7, 2004, at 7:00 AM, Kevin Walsh wrote:
Perhaps service providers who allow the Caller*ID to be set should
insist that customers provide evidence that they own the phone numbers
that they want to publish, and then limit the customers' choices to
only the numbers in their approved list.
On Jun 30, 2004, at 2:05 PM, Brian Wilkins wrote:
Hi,
We are having an issue here. It seems that whenever we initialize
Asterisk
on our network, the router that the Asterisk server is connected to
(Cisco
7200) crashes and loses it configuration. This has happended five
times and
each time we
On Jun 30, 2004, at 2:31 PM, Brian Wilkins wrote:
IOS version 12.xx
As far as a traceback, that's going to be difficult now since we've
removed it
from our switch and brought it back here to the office for testing.
When we
test it tomorrow or later in the week, I'll see if it crashes again in
a
On Jun 22, 2004, at 7:44 AM, Matt wrote:
I've got a number (10) Cisco 7960's connected to my network. All the
phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887
sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22
On Jun 21, 2004, at 1:16 PM, Brian Weaver wrote:
Question is this, I know the ATA with packet8 is locked down, but is
there any reason I can't use it just like a regular POTS line with
Asterisk if I buy a X100P card? That way I could pick up a SIP device
to talk to Asterisk, and configure the PBX
On Jun 19, 2004, at 6:47 AM, Randy Bush wrote:
[ btw, search function in wiki is not real great, to be polite;
but that issue is not local to this wiki ]
Yeah. Google:
site:voip-info.org waitexten
Works much better.
Scott
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Asterisk-Users mailing
On Jun 18, 2004, at 8:56 AM, Randy Bush wrote:
Err, it works for me, with a 7940 and 6.3. I've never bothered with
'NewCall' or 'Dial'; you can get around them if you can set up a
decent
dialplan.xml.
aha. ok. thanks. on to sorting out a dialplan.xml. any simple
one that sez just give it all
On Jun 18, 2004, at 3:02 PM, Randy Bush wrote:
the sipura seems to act one way and the cisco another. the
sipura, x141, is happily served by
[in-206-sipura]
exten = s,1,SetVar(areacode=206)
exten = 141,1,GoTo(in-int,s,1)
[in-int]
Ahh. Okay, I think I see. The Cisco isn't doing
On Jun 18, 2004, at 4:10 PM, Randy Bush wrote:
Unless Asterisk does something weird with it that I haven't seen
before,
then you'll only get 's' in this context if you get the cisco to dial
without specifying a number.
oops! then how do i get a per-incoming-context SetVar?
I've done something
On Jun 17, 2004, at 4:48 AM, Stefan de Konink wrote:
It is that simple?
Probably you want something that actually boots the system too. I don't
know if the ISOLINUX pakage supports a LILO kind of thing, but I guess
it
does. That should be in the MBR of your flash disk and you could
probably
boot
On Jun 17, 2004, at 5:42 PM, Randy Bush wrote:
if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.
using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
go off hook
hit NewCall
punch 142 (or
On Jun 16, 2004, at 11:18 AM, Jay Milk wrote:
Cisco, for example, has different models such as the 7940 and 7960
which
seem to only differ in the software.
IIRC, the 7940 and 7960 run the same software, but differ slightly in
hardware. The 60 has 6 line appearance buttons, while the 40 has 2.
On Jun 15, 2004, at 7:27 AM, mattf wrote:
Polycoms do have programmable softkeys as well as several other
programmable
features. They don't however have expandable button modules like the
Ciscos
do to add extra physical buttons to push.
While Cisco does make an expansion module for the 7960, it
On Jun 15, 2004, at 1:14 PM, Bill Reid wrote:
For me the big issue of html is in the message digests. Since the html
is mixed in with plain text browsers do not detect the html. For
individual messages HTML is generally not a problem.
Good HTML isn't a problem. HTML that produces blue flyspeck
On Jun 14, 2004, at 9:06 AM, Jay Milk wrote:
I'm using zoneld numbers which I can terminate on any US number --
http://ld.net/mu has various options. You basically get your incoming
voicepulse, broadvoice, etc line, then get an 800# to terminate on
those
lines and you're in asterisk. Through
On Jun 14, 2004, at 10:40 AM, Jay Milk wrote:
Good deal, I didn't know about nufone's service -- but then, how could
I, it's not on their site. I've been to www.nufone.net a few times,
and
looking at their site, you'd think you're dealing with a front-business
for the mafia. If they'd publish
On Jun 11, 2004, at 1:56 PM, oi geli wrote:
I want to buy a 7940 to use with Asterisk. Does all
the features (i.e. Transfer, Hold, call waiting, MWI,
etc)work?
Transfer: yes
Hold: yes
Call Waiting: yes, although it works better if you configure two line
appearances for the same Asterisk
On Apr 22, 2004, at 11:09 AM, John Fraizer wrote:
Nick Knight wrote:
I havent used cisco phones as yet do they work with asterisk, are
they good which models are the best?
I use a 7960 with Asterisk and absolutely love it. It blows the snot
out of the Nortel phone I used to use.
I have a
On Apr 21, 2004, at 12:20 PM, David Carter wrote:
Hello,
I'm considering using Asterisk with some type of Cisco phone, and
currently
considering either the 7940 or 7960 because of its more-complete
functionality
(compared to the 7905).
I'm currently wondering:
Do all the expected functions
On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote:
When we receive or make a call to the outside they can hear us, but
we cant hear them.
With SIP, missing audio is *usually* either a firewall or NAT issue.
Check firewall logs and make sure that you aren't seeing packets being
lost. Do
On Apr 14, 2004, at 3:50 PM, Jeremy Bogan wrote:
Hi All,
I'm trying to get Asterisk auto fax detection working so that when a
fax tone is detected it will ring my fax server that is plugged into
my TDM400P. Does anyone know how to get this working successfully? I
created a fax extension:
On Apr 14, 2004, at 4:01 PM, Michael Shuler wrote:
Anyone know what protocols support a fax machine i.e. g.729, g.711,
etc?
Only G.711, and even that won't always work.
Scott
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