On Fri, 28 Oct 2005, Erick Baum wrote:
We have 50 of these phones in one location and a couple remote phones. The
problem seems to be caused by the volume settings on the phone. We have
noticed that the echo seems to be worse when the volume is very high on the
phone (not using speakerphone).
On Thu, 29 Sep 2005, Jens [iso-8859-15] Kübler wrote:
Have I to use also prilocaldialplan ?
Can be left unknown.
Explains what you expect as the incoming number to look like
This is incorrect. It sets the TON/NPI pair for ougoing calling number
presentation, i.e. the format of the caller
On Wed, 14 Sep 2005, Carlos Arnt wrote:
Everything is perfect, but i have in point B now a C Network that comes over
Router.
Point B com see and interact with Point C , but point A can´t
In number :
Point A = 192.168.2.0/24
Point B = 192.168.1.0/24
Point C = 192.168.3.0/24
On Mon, 5 Sep 2005, Ben Brown wrote:
Any Particular recommendations on PRI protocol? I can chose from 4ESS, 5ESS,
and NI1
This is not a direct answer to your question since I am mostly familiar
with EuroISDN. Most PSTN providers in America seem to charge extra for
every single feature on a
On Mon, 5 Sep 2005, Ben Brown wrote:
So the only difference with PRI is caller ID? What I am trying to
determine is if the PRI has enough advantages to give up the voice
channel used by the D channel. For what I am doing, caller ID is not
necessarily that important for my application.
The
On Thu, 1 Sep 2005, Jesus Mogollon wrote:
We have all those problems and then some... after a while, the phone starts
degrading: The ringing becomes lower and lower and there is a lot of
stuttering in the conversation. Also, if I stop/start asterisk, half of the
phones reconnect while the
On Wed, 31 Aug 2005, Hadar Pedhazur wrote:
My only real problem with my current setup is that because I use Call
Files to contact the Agents, I have no direct way to cancel ringing
phones when the call has been bridged to another channel.
You can use the Manager interface with the Originate
On Tue, 30 Aug 2005, Anton Krall wrote:
Speaking of GS..
I know polycom phones can eb rebooted with some script using sip_notify.
Can GS phones do this also?
You can reset the phones by requesting the right page from their built in
web server as long as you know the admin password.
On Tue, 30 Aug 2005, Hadar Pedhazur wrote:
Following up on a thread that I started about Agents/Queue and
acknowledging calls before bridging them...
Greg Boehnlein said that he was putting his efforts into ICD.
I downloaded and installed ICD, and I can get simple queue and agent
stuff
On Tue, 23 Aug 2005, Gulzar Hussain wrote:
yeah i am using chan_zap and i have tried all
combinations of pridialplan and nationalprefix etc.
What does a pri intense debug span XX show?
Peter
___
Asterisk-Users mailing list
On Mon, 22 Aug 2005, Guy C. Guckenberger wrote:
Im using a TE110P as a trunk to a Panasonic KD-500 everything works
well.but Im having this problem where one of the channels becomes
blocked with a partial phone number after about two days. So if the
channel that becomes blocked is
once the signaling indicated that a B-channel was required.
I would be interested in how the commercial SS7 implementation for Asterisk
works. SS7 would normally allow the audio paths to change in mid-call to
potentially follow an altogether different route.
Peter
Peter Svensson [EMAIL
On Sat, 20 Aug 2005, Gulzar Hussain wrote:
I am having another strnage problem :)
When I dialout on any number from asterisk, it use to
add a leading zero in dialed number
for e.g
I dial a number 5832876
and when I check the tracer's result of PSTN switch
that shows me call request for
/asterisk-users
Peter
--
Peter Svensson ! Pgp key available by finger, fingerprint:
[EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
Remember, Luke, your source will be with you... always
On Sat, 13 Aug 2005, Jamin W. Collins wrote:
Is there a way to initiate a transfer using an analog handset? For
instance I'm looking for a way to do something like the following:
External call comes in and is answered by user A. After talking to the
caller they determine that the caller
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone is
disconnected.
To repeat the question from Matt Riddell:
Does he have Stereo Mix selected as a recording source?
We have found the most common cause of a strong echo to
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related.
Yes, this would be the logical conclusion, although it is hard to beleive
given what
On Thu, 11 Aug 2005, Joseph wrote:
In this case could somebody explain to me why run asterisk with ''-p
switch?
According to asterisk man explanation for -p is as follow:
If supported by the operating system (and executing as root), attempt
to
run with realtime priority for increased
On Thu, 11 Aug 2005, Geoff Manning wrote:
We are having line noise issues in our Asterisk to legacy PBX integration.
All SIP calls originating from IP phones sound crystal clear. All calls that
originate from the legacy PBX (Isoetec 228) and route through the Asterisk
and out SIP have a lot
On Fri, 12 Aug 2005, Geoff Manning wrote:
OK. So I changed it to:
span=1,0,0,d4,ami
And the Blue Alarms are still occurring but now in conjunction with Slip
errors. I feel like I am on the right track though.
Which side shows the slips?
I am not that familiar with T1, Are you sure the
On Fri, 12 Aug 2005, Bruce Ferrell wrote:
Hardware, possible. Unlikely to be cabling. It's usually a timing setting.
The blue alarm is really a very specific alarm condition normally. It
cannot quite see how it can be generated accidentally. Something along the
path from the TE110P
On Tue, 9 Aug 2005, Fredrik Lithén wrote:
Yes, I tried that but it sent me a bit offtrack as it reported blue
which I assumed was a clocksync problem, or at least, that was the info
I could find.
As far as I can tell zttool/zaptel uses the term BLue Alarm for the E1
term AIS (Alarm Indication
On Tue, 9 Aug 2005, Andrew Kohlsmith wrote:
On Tuesday 09 August 2005 04:32, Peter Svensson wrote:
A bitstream is present at the receiver, though it is unframed and invalid
(i.e. the receiver is seeing a transmitter that does not quite know what
to transmit). This is different from a red
On Tue, 9 Aug 2005, Eric Wieling aka ManxPower wrote:
Panitaxx wrote:
yes. overlapdial=yes.
You want it to be no.
What would the reasons to want overlapdial=no on a pstn pri be? Since the
pri will happily signal once the number is complete there should not be
any downside to allowing
On Tue, 9 Aug 2005, Fredrik Lithén wrote:
Perhaps everything isn't as spiffy as I thought
When running zttool the card still reports as internally clocked
Zaptel.conf:
# Global data
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
Zttool still shows the card as internally
On Mon, 8 Aug 2005, Kib Eki wrote:
Hi,
we successfull managed to bridge a PSTN (E1) switch over the TE405P card to
our
old PBX. So now we could migrate to the * server.
But, there are two things we can't live with:
1. A call from the outside to the old PBX is missing a leading 0
On Mon, 8 Aug 2005, Fredrik Lithén wrote:
I'm having difficulties getting up my TE110P (running as a E1) when I
connect it to the PRI. If I start the server with a loopback connector
everything seems fine and the led is green but when I connect it to the
PRI the flashing starts
The
On Mon, 8 Aug 2005, Eric Wieling aka ManxPower wrote:
Peter Svensson wrote:
See internationalprefix, nationalprefix etc in the file zapata.conf.
Those options are only available in BRIStuff.
They have been in HEAD for quite some time. The 1.0.x-releaes are note
really usable in a lot
On Mon, 8 Aug 2005, Kib Eki wrote:
2. A call made from a SIP client to the outside lacks the extension in the
number: Ex: PSTN number is 6789-0. The extension 234 is not added to the
PSTN number like 6789-234 when dialing out over the PSTN.
Again, trivial dialplan stuff. Your sip.conf
On Sat, 6 Aug 2005, Angus Comber wrote:
I have a Grandstream GXP2000 with latest firmware. When I use it holding
the handpiece I don't hear any echo - neither does other end. However,
if I use it handsfree, the other end notices echo when they speak - ie
their voice is echoy. I hear
On Sat, 6 Aug 2005, Robert Goodyear wrote:
Can you educate us all on the appropriate circumstances in which to
use 'r'?
Some devices (voip phones, softphones) do not generate in band progress
information when ringing. You will quickly find out if a particular
end device requires the 'r'
On Wed, 3 Aug 2005, Tim Connolly wrote:
I'm trying to pass a 65K DATA call in one channel on my Digium
TE411P to another channel on a different span. Any idea what could keep this
call from going through?
-- Accepting call from '' to '5444' on channel 0/1, span 1
-- Executing
On Wed, 3 Aug 2005, Sascha Ferley wrote:
http://www.digium.com/index.php?menu=compatibility
What servers does one recommend though using ? Our company hates using HP
junk, dell used to be a good choice for most of our stuff. IBM is way
overpriced. Anyone have any suggestions?
If you need
On Mon, 1 Aug 2005, Phoneguy wrote:
There are 2 methods blind and announced here you go:
Blind:Call someone, or receive a call. Hit 'Trnf'
The screen displays TRANSFER TO? and you hear a dial tone.
The other end can still hear you, so don't say anything nasty.
Dial the number and hit
On Tue, 2 Aug 2005, Frank Sautter wrote:
Maik Schmitt schrieb:
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx
On Wed, 27 Jul 2005, Paul Dracevich wrote:
I have just got a TE110P card, and I need the cable pin out.
The TE110P cards use the standard T1/E1 modular pinout. See
http://www.samhassan.com/isdn60.gif.
1 Receive from pstn (tip2)
2 Receive from pstn (ring2)
4 Transmit to pstn (ring1)
5
On Tue, 26 Jul 2005, Alex Ongena wrote:
I'am new to * and googled/read a lot, but did not find (yet)
a lot of info to do the above.
Some months ago, I did find a 'story' from somebody having
put * between his PRI and current PBX as IVR, but I can not
find it back :-(
We have an Asterisk
On Wed, 20 Jul 2005, Ed Greenberg wrote:
I'd like to write a snippet of dialtone that plays dialtone and collects a
specific number of digits into a variable.
Sort of like READ but with a generated dialtone.
Naturally, I want the dialtone to stop playing after the first digit.
I can't
On Wed, 20 Jul 2005 [EMAIL PROTECTED] wrote:
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
anyone else
On Wed, 20 Jul 2005, Paul Belanger wrote:
Any to back my clams that asterisk is fine, I'm using the TE405P, with a
different telco in my second span and it operates fine!!
What span is your clock source? A TE405P card can only operate in one
clock domain at a time. I.e. the same clock will be
On Mon, 18 Jul 2005, Guy C. Guckenberger wrote:
Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500.
I have Asterisk connected via crossover to the TE110P. We are able to
make internal calls into the Asterisk Box but the PBX vendor (I know
nothing about the KX-TD500) tells
On Fri, 15 Jul 2005, David Wilson wrote:
Thanks for your reply.
Would srx show ccmsgs 1 help ?
I am not familiar with the Sirrix line of BRI cards. However, someone else
on the list may be, or you may be able to diagnose the problem yourself.
Peter
On Thu, 14 Jul 2005, David Wilson wrote:
I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running
in TE (ptp) mode in a Asterisk box - this then links through Internet to
another Asterisk box via IAX2.
When a user on the Panasonic PBX system dials the extension of my Sirrix
On Thu, 14 Jul 2005, David Wilson wrote:
Yes, as far as I know ? In that context I have the following:
[pabx2ip]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,ResponseTimeout(3)
exten = s,4,Background,enter-ext-of-person
exten = _X.,1,Dial,IAX2/pmb/${EXTEN}
exten = t,1,Hangup
exten
On Sun, 3 Jul 2005, Josef Seger wrote:
I have one other Dect phone connected to Digiums Card(TDM400P), an
Ericsson DT 260. The Ericsson phone only supports true swedish standard
CallerID (DTMF signalling before the first ring), and CallerID does not
work for this phone:(
I have measured
On Sun, 26 Jun 2005, qrss wrote:
It seems
that the voip clock is slightly faster than the hardware clock that zaptel
is timing from. The extra samples/second must be being buffered. Of
course, this buffering would add up over time until the point that a VOIP
sample is played back several
On Thu, 23 Jun 2005, Robert Rozman wrote:
I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...
It
On Thu, 23 Jun 2005, Andrew Latham wrote:
On 6/23/05, Seamus Abshere [EMAIL PROTECTED] wrote:
That's what I'm confused about:
* two 4 port FXS cards
* one 24 port FXS channel bank
both, neither, and if both -- why do you need the dual digium cards?
shouldn't your channel bank just take
On Wed, 22 Jun 2005, Pavel Jezek wrote:
I had gxp-2000 for testing some days, but features are (in current
firmware) _very_ limited!
phone does not have missed, dialed numbers, phone book, speakerphone is
useless...
Some of these features are in the 1.0.1.9 version that was released last
On Tue, 21 Jun 2005, Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
We
On Mon, 20 Jun 2005, Dan Morin wrote:
Can you let me know what hardware you are using and how the two systems
are configured to work together? Thanks in advance.
We have an E1 PRI card in the KX-TD1232 and a TE405P in the Asterisk box.
The Asterisk box sits between the pstn and the
On Sun, 19 Jun 2005, Dan Morin wrote:
If anyone has any experience with a Panasonic KX-TD1232 phone system, I
would really like to talk to you for a few minutes.
I have asterisk connected to a Panasonic system via FXS - CO ports.
I'm trying to get the Panasonic configured so that if someone
On Fri, 17 Jun 2005, Paul Redstone wrote:
We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
softphones and one hardphone.
Working well except that we sometimes get dropped connections between IAX and
the server with a max retries exceed message, which comes from
On Tue, 14 Jun 2005, Amund Nygaard wrote:
We have around 50 phones in our company, and I am playing with the
thought to gradually go over to using sip services and ip-phones
internally. However at first I would liked the Asterisk just to sit
between the phone line and the Panaosnic, so I can
On Fri, 10 Jun 2005, Peter Svensson wrote:
On Fri, 10 Jun 2005, James Bean wrote:
Peter seems to be on the ball more then me about these phones as
grandstream gave me the standard replies, Peter do you know for sure if
grandstream have a timetable for the function led's cause I need
On Fri, 10 Jun 2005, The VoIP Connection wrote:
Have you received an updated tftp config template as well? We
asked for and received one with a 1.0.1.9 early beta version.
That is the entire package as it was submitted to us from Grandstream.
We requested and received the template
On Thu, 9 Jun 2005, Andrew Kohlsmith wrote:
I also check if I'm loosing interrupts and everything seems ok. Also I
pull out the TDM400 from the box.
This tells me it's got nothing to do with the TDM400 or lost interrupts.
It could be that the user-land side (i.e. Asterisk as opposed to
On Thu, 9 Jun 2005, Julian J. M. wrote:
I've just checked the download page, and the latest firmware available
is 1.0.1.8. Where did you find 1.0.1.9?
This phone has some nasty bugs, one of them being that the other end
HEARS you after you press the Transfer button and you hear a dialtone.
On Thu, 9 Jun 2005, Michiel van Baak wrote:
Did that pre-release version fix that bug where the other
party can hear you when you pressed the transfer button ?
That bug is not present in the testing version. Pressing the transfer
button gives music on hold from the server to the other party.
On Thu, 9 Jun 2005, The VoIP Connection wrote:
This is supposed to be the final version:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Release_1
.0.1.9.zip
From the changelog they seem to have corrected all bugs/misfeatures we
reported during our testing of 1.0.1.9.
On Thu, 9 Jun 2005, Michiel van Baak wrote:
I really like the way the gxp2000 looks.
I even prefer them above the snoms when it comes to looks.
The bugs and lacking functions prevent me from rolling them
out @ customers tho.
The leds would be great, but the bug with the transfer
button not
On Fri, 10 Jun 2005, James Bean wrote:
Unfortunately not, Grandstream didn't admit to me that they were going
to program the LED's like the snom SUBCRIBE/NOTIFY, they told me the
LED's were additional incoming line indicators, not LED's for the
function keys to be programmed. Which is a
On Thu, 9 Jun 2005, The VoIP Connection wrote:
This is supposed to be the final version:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Release_1
.0.1.9.zip
Have you received an updated tftp config template as well? We asked for
and received one with a 1.0.1.9 early
On Wed, 8 Jun 2005, David Phelan wrote:
If you download the configuration tool which I couldn't get working on my
systemthere is a cfg template in there for 1.0.1.8
Oh, then they have added it, or we missed it the first time around. We
have it running. We had to tweak the paths in the
On Thu, 9 Jun 2005, James Bean wrote:
Has anyone got the hint function working, and maybe with the GXP2000.
I don't think the current firmware release for the GXP-2000 supports
SUBSCRIBE/NOTIFY. That functionality is to be released at a later date.
Peter
On Tue, 7 Jun 2005, marek cervenka wrote:
can you someone post tftp template for gxp-2000?
like
http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt
I think it will be released with the 1.0.1.9 firmware. You may be able to
get it
On Mon, 6 Jun 2005, Robert Rozman wrote:
I'd like to use DISA properly for my case - I'd like to handle it right, if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
The file app_disa.c is hardwired to hang up the call if too many incorrect
passwords are
On Mon, 6 Jun 2005, Peter Nixon wrote:
On Monday 30 May 2005 13:28, Matteo Brancaleoni wrote:
and , what is more interesting,
they've omitted any reference to digium resellers
and specified only distributors :(
Yes. Our reseller info was removed. And some of our customers have been sold
On Sat, 4 Jun 2005, Tom Fanning wrote:
What's so special about Digium cards that makes them this expensive? $4000
for a PCB is extortion IMO!
I'd say low volume and high development and certification costs. A
contributing factor is what the market is willing to pay.
Peter
On Fri, 3 Jun 2005, Remco Barende wrote:
I am thinking of another solution for fax. I have an * box with one PRI
card and I'm thinking of adding a quad BRI card in the same box.
A separate box running fasx server software will also be equipped with a
BRI card for faxing (I cannot use
On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote:
I was trying to make call confrence available but all the asterisk
documents use the meeting room concept, where those who wanna meet have
to dial an extension corresponding to the meeting room, while call
conference actually means that I am on
On Mon, 30 May 2005, Remco Barende wrote:
What exactly is the meaning / function of the pridialplan
prilocaldialplan?
Both set the two fields Type Of Number (TON) and Numbering Plan (NPI)
markers on an outgoing isdn call. These two tell a receiving isdn switch
how to interpret the
On Fri, 27 May 2005, Mark Elkins wrote:
I tried to do an HTTP update from the Grand Stream web site...
You upgraded the firmware over the Internet? You are braver than I am. I
would have used a local http server.
Is there a magic re-incarnation routine ?
(Power on whilst holding down some
On Fri, 27 May 2005, Colin Anderson wrote:
It will be about 100 phones at about 20 locations all within
about 4 miles of each other.
Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres
On Fri, 27 May 2005, Mike Clark wrote:
brandt Milczewski wrote:
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be
On Wed, 25 May 2005, Shane Burrell wrote:
Anyone with any comments on DSS buttons and general phone features?
The BLF (Busy Light Field) part of the DSS buttons are not active in the
latest firmware.
The microphone part of the speaker phone needs some work, possibly just
software (too low
On Mon, 23 May 2005, Kanuri, Seshu (Company IT) wrote:
FireFly is the best of the IAX softphones. Other softphones do not work
as good as FireFly. DIAX has many bugs still. DIAX Softphone disconnects
with Windows DLL errors everytime there is a problem in the call like
Asterisk Channel Not
On Tue, 24 May 2005, Remco Barende wrote:
I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands.
I get a Red Alarm on the line.
Is there any way of debugging this? I've tried some configs that should
work but without success. Is there any way of telling if the cabling is
On Tue, 24 May 2005, Remco Barende wrote:
On Tue, 24 May 2005, Huddleston, Robert wrote:
OK, but being from Europe I haven't got a clue what an American SmartJack
is for :)
Would that mean that I would have to hook up the TE110P to the HDSL
device? If so, what sort of cable would be
On Sat, 21 May 2005, Companity wrote:
The sip phones and the internal phones on the PBX see the number of the
calling party correctly (e.g. 040-987654321). Cause we can´t set a
callerid to the public phone network (to show the calling party number),
we want to show an extension of our numbers
On Thu, 19 May 2005, Dean Collins wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Thursday, 19 May 2005 7:55 PM
Another and perhaps easier option for wireless konference phones may
be
http
On Thu, 19 May 2005, Marshall, Ed wrote:
Can anyone point me in the right direction of info regarding ACD methods
available in Asterisk.
As far as I can see there are time based ring strategies available but I
cannot find any info regarding skills based routing or queue priorities.
I don't
On Thu, 19 May 2005, Matthew Boehm wrote:
Hugh L. Johnson wrote:
Does ^ work as a NOT in an expression for extensions?
Are the following equivalent?
exten = _58[^389],1,dial(${${EXTEN}},${RINGLONG},tr)
exten = _58[0124567],1,dial(${${EXTEN}},${RINGLONG},tr)
Not sure which RegEx
On Thu, 19 May 2005, Michael B. Murdock wrote:
Is there anywhere (or anyone) who has compiled some recommendations on rack
mount servers for Asterisk?
We are currently using Dell 2650 and Dell 2850 but are seeing some problems
with the raid controllers failing and are now shopping for a
On Thu, 19 May 2005, Dean Collins wrote:
Anyone seen these before?
http://www.ascomnira.com.au/servlet/Display?p=100
wondering if there is a use with asterisk.
Another and perhaps easier option for wireless konference phones may be
On Wed, 18 May 2005, Steve Underwood wrote:
The header is always in the received image. The TIFF file contains
exactly the same image that a receiving FAX machine would print out.
I think he is refering to the remote fax id to be presented, not the
header. I.e. the 20 digit user selectable
On Wed, 18 May 2005, Steve Underwood wrote:
Jean-Yves Avenard wrote:
On my Brother's fax machine (MFC-8820D) today, I've received 3 faxes:
all of them at the top showed the caller Fax identity.
I received 2 faxes on Asterisk with spandsp, one from the same sender
as earlier on the
We recently purchased a Grandstream GXP-2000 phone and I would like to
share our experiences with it, especially out very good support
experience.
The phone was easy enough to set up. The phone was configured using a
configuration file served via tftp. Creating the configuration file was a
On Wed, 18 May 2005, Anton Krall wrote:
Peter.. I just bought a gxp 2000 and I wanted to know, how are you
configuring them using templates?
There is a template-binary config file compiler available from the
download page at the Grandstream web site. Fill in the template and serve
it via
On Wed, 18 May 2005, John Mensel wrote:
Hi all. I'm in the process of putting together a new Asterisk system as a
proof-of-concept, and wanted to see which SIP phones all of you had the best
luck using with Asterisk. I've just come off a very trying experience with
some Cisco 7960s, and
On Wed, 18 May 2005, Erik Sundberg wrote:
Wonder if there was away to run a script/marco when the person who
originates the call hangs up.
I have use the g option in the dial application to continue running
applications in the dial plan, but that only works if the person who is
called
On Tue, 17 May 2005, tim panton wrote:
The 'if possible' thing relates to filesystem design.
Almost all of the native UNIX filesystems support mv as an atomic action
- but only within the same filesystem.
(Imagine you create the file on one physical disk then 'move' it
onto a different disk
On Tue, 17 May 2005, Steve Underwood wrote:
In most hardware the clock you use is not provided by a crystal. Rather
the crystal provides a reference for a pll. The conversion factor between
the crystan and the derived clock is usually tunable.
Nope. Its always a crystal. Its either a
On Tue, 17 May 2005, Seb Auriol wrote:
In fact, this is what I'm doing at the moment on the production system, but
we've had a complaint because it doesn't start at the beginning for each
caller. This is pretty important because the file starts with something like
Thank you for calling X. We
On Tue, 17 May 2005, Lenwood S. Sawyer, III wrote:
I have a PRI from Bellsouth going to my asterisk box with a Digium
Wildcard TE110P. I would like to be able to use call forwarding without
having to use two channels. Is it possible to use call redirect with a
PRI. Does the BRIstuff
On Mon, 16 May 2005, Michael Welter wrote:
Where is the clock source that the T1/E1 board, with 0 for timing,
uses to generate the tx data stream? Is there a PLL on each board? Or
is some central source used?
For example, I have one system with two separate T100P cards--one for a
On Mon, 16 May 2005, Rich Adamson wrote:
It doesn't make any difference. The pcm data that arrives from the telco
is buffered in the zaptel and/or asterisk code, and sent out the second
T1 card as soon as it can. That buffering reduces (or eliminates) the
need to sync one T1 card to another.
On Mon, 16 May 2005, Steve Underwood wrote:
It is possible, though complicated, to synchronize the 2Mbit clocks on two
unrelated cards by measuring the accumulated phase shift (difference in
interrupt rate) over time and compensating, thus implementing a PLL in
software. Digium has not
On Fri, 13 May 2005, jltaylor wrote:
Does the TDMoE only allow one T1 per segment?
You can add an index to have several TDMoE links and thus several
virtual T1/E1 links between two computers.
TMDoE is mostly used to provide an interconnect with a low latency over
ethernet.
Peter
On Thu, 12 May 2005, Chris Coulthurst wrote:
Does someone have a link to step-by-step instructions to making the
Line-In on the console sound card a MOH source?
You can probably use the Remote MoH patch from
http://bugs.digium.com/view.php?id=3565
Peter
1 - 100 of 598 matches
Mail list logo
