-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jimmy Riley
Sent: Tuesday, 13 January 2004 13:02
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] GUI client for windows for live
monitoring/barge
I've seen a few but can't get them to work. I
I got SJPhone to work at home but not at work. It was the first time I
ever ran it, so I don't know if what I found is a new error or it has
always been there. It turns out that if your computer does not have a
gateway (mine at work does not), then SJPhone does not send the correct
VIA
Hi,
Yes Telesym, xten and one more I can't remember the name of it, they are all
for PPC-only. :(
/HHA
From: Ray Burkholder [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC
Date: Mon, 12 Jan 2004 14:33:39 -0500
What
Dear Sir / Madam,
I am a newbie in using Asterisk. I am interested in its
SIP.
Before I start to use it, I would like to know whether the
system can work between two Linuxbox without any FXO and FXS card and just
using microphone which connect tothe regularsound card? I am looking
into
knot yet. :)
cameron.
- I like puns.
On Mon, 12 Jan 2004, Sean Cheesman wrote:
knot n. A unit of speed, one nautical mile per hour
thanks to our good friends at reference.com. Are we done yet?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday,
As Robert's colleague that owns 7960s I can go on about the superiority of
the Cisco phone. The most immediate difference is the look and feel.
Everyone that has seen or held my phone says that it is nice. Everyone
that picks up a Grandstream phone or looks at one says they are cheap.
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
GotoIfTime application. I'll own up
I can't send mail to any addresses in nufone.net; they all get rejected
by a spam blocker.
And their website is gone, too!! The URL leads to a parking site.
My accounts still seem to work, but I wonder WTH is going on?
Thx.
B.
___
Asterisk-Users
Hi,
Pat Boyle wrote:
I have no problems lauching asterisk from the command line . . .
asterisk -c
However, I'm trying to autostart on boot up, so I'm trying safe_asterisk
When I do this, I get: Asterisk ended with exit status 127. Asterisk died
with code 127. Aborting. I've
On Tue, 2004-01-13 at 01:26, Brian Capouch wrote:
I can't send mail to any addresses in nufone.net; they all get rejected
by a spam blocker.
And their website is gone, too!! The URL leads to a parking site.
My accounts still seem to work, but I wonder WTH is going on?
looks like Jeremy
Scott Stingel wrote:
Hi-
I have posted a photo of the TE410P Digium card on my site,
so that those wishing to purchase a compatible motherboard
can see physically what the PCI slot requirement is:
http://www.evtmedia.com/TE410P.htm
I believe the required slot is a 64-bit, 3.3 Volt PCI, most
Walt Reed wrote:
I'm getting some warnings:
NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
Asterisk Version: CVS-01/06/04-13:50:26
Cisco ATA 186 version: v3.0.0 atasip (Build 031210A)
Is this something I should be
No he renewed it... but Gododaddy did the transaction over the phone
manually and they never posted the payment so they shut it down. It
should be back by now.
Switch-1 ip is 66.225.202.72
bkw
On Tue, 13 Jan 2004, Steven Critchfield wrote:
On Tue, 2004-01-13 at 01:26, Brian Capouch wrote:
Looks like they went off the air just after my PayPal
payment was processed. I gues we wait a couple days
to see if Nufone has gone belly up/bankrupt/gone or
if this is just a domain name screw up.
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Tue, 2004-01-13 at 01:26, Brian Capouch
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
Hi
I am attending the tutorial day, i am looking forward to it.
See you there.
Craig.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter
Junghanns
Sent: 13 January 2004 10:31
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Hi All
I have just a quick question regarding app_txfax for Asterisk.
When I send a fax from asterisk to a traditional fax machine connected to
asterisk via the digium analog card everything works perfectly. However the
same fax machine on the public telephoine network results in errors (looks
Hi,
Ing Isianto Istiadi wrote:
Thanks for the Info, and It worked.
But I have a couple of questions:
1. There's an echo. How to get rid of the echo?
2. Is there any way to call from x-lite just the extention number? (say that
in my extention.conf, I have extention 32 to connect to my fxs
KH Chow wrote:
Dear Sir / Madam,
I am a newbie in using Asterisk. I am interested in its SIP.
Before I start to use it, I would like to know whether the system can
work between two Linux box without any FXO and FXS card and just using
microphone which connect to the regular sound card? I am
Hi,
If anyone else had a problem I got kphone to work with Asterisk.
--
Steve
__
You actually need to constantly be alert
and willing to handle things, or life
will find a way to get you good!
___
On Wednesday 17 December 2003 09:48 am, Senad Jordanovic wrote:
Hi all,
Anyone has experience using * through
128 kbs (or bigger) satelite link?
In particular I am interested to hear how many calls could be put
through 128Kbs satelite link simultaneously?
There's only 500ms lag over
Though slightly off-topic, I was wondering if anyone would have any
ideas to the following regarding our Cisco 7960's. To keep this short -
the plan facts:
- With phone configured for NAT, works fine with Pulver FWD service
from any location (home, various peoples offices etc...) BUT
- ...
Steve wrote:
Hi,
If anyone else had a problem I got kphone to work with Asterisk.
I have problems with kphone + Asterisk. KPhone does not seem to ACK
invites, ie.
KPhone --- sends INVITE -- Asterisk
KPhone -- sends 101 Trying --- Asterisk
KPhone -- sends 202 OK --- Asterisk
Hi,
does anyone know if its feasible to run asterisk with a PRI card but not run
any q931 signalling.. basically push calls down the PRI and tell asterisk in
some other way to pickup a particular Zap channel?
Steve
___
Asterisk-Users mailing list
Hello,
Everything I've read says that 3.3v 32bit cards will work in 64 bit slots,
and the cards do fit, they just have some extra space left on the slot.
MATT---
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 3:15 AM
To: [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: January 12, 2004 11:25 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] GUI client for windows for live
monitoring/barge
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi
my question is:
which is the best distribution to work with asterisk?
thanks
mark
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
mattf wrote:
Hello,
Everything I've read says that 3.3v 32bit cards will work in 64 bit slots,
and the cards do fit, they just have some extra space left on the slot.
MATT---
Yes you are 100% correct.. A 3.3v 32bit card will just have a shorter
connector on the bottom that will not extend
[EMAIL PROTECTED] wrote:
Hi
my question is:
which is the best distribution to work with asterisk?
Hi Mark,
I am working on a distro called SAX built to optimize * and routing. It
works with RPMs and its HFS is RedHat like. I built all packages by
hand and created RPMs packages. It is in
[EMAIL PROTECTED] wrote:
Hi
my question is:
which is the best distribution to work with asterisk?
thanks
mark
You better dusck down cos here comes the war about who's distro is
better.. :)
Use the one you are most comforatable with is the easiest and most
logical answer.. IMO thats all that
the one you feel most confortable with.
as far as I know, asterisk is developed under RedHat,
but really, I run it with RH, debian, slack.
Many with suse and so on... so is up to you.
matteo.
Il mar, 2004-01-13 alle 12:48, [EMAIL PROTECTED] ha scritto:
Hi
my question is:
which is the best
cool idea :)
Il mar, 2004-01-13 alle 13:10, Daniel Bichara ha scritto:
[EMAIL PROTECTED] wrote:
Hi
my question is:
which is the best distribution to work with asterisk?
Hi Mark,
I am working on a distro called SAX built to optimize * and routing. It
works with RPMs and its HFS
only domain name screwed up.
mmh.. my registrar allows me an autorenew for all
domain names... pretty useful :)
matteo.
Il mar, 2004-01-13 alle 09:24, Chris Albertson ha scritto:
Looks like they went off the air just after my PayPal
payment was processed. I gues we wait a couple days
to see
Hi!
I am looking for a way to Forward to a external or internal number and
require a digit(s) in order to complete forward.
Consider using a queue and agents. Read more on the Wiki.
Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi!
I get to voicemail either way. It just doesn't playback the unavail on
the IAX call. Plays back fine on the SIP call. Both calls show up as
playing voicemail/company/6711/unavail on the console.
Sounds like a codec problem - check which codecs are being used during
the IAX connection
No need to go Xeon, I have one of these:
http://www.tyan.com/products/html/thunderk7x.html
Dual AMD Athlon MP with one 3.3v 64bit PCI slot
MATT---
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 7:07 AM
To: [EMAIL PROTECTED]
Subject: Re:
Will cisco 7910 ip phone compatible with Asterisk? I know
that 7960 are
fine.
David Kwok
Cisco's site shows SIP drivers for 7960, 7940, 7912, 7905 only. If you want
to run 7910 in Skinny mode, that may work. I'll leave that up to the
chan_sccp and chan_skinny people.
Ray Burkholder
I use Fedora FC1. Best is a matter of opinion. Whatever you know is
best for you.
Michael
On Tue, 13 Jan 2004 12:48:09 +0100, [EMAIL PROTECTED] wrote:
Hi
my question is:
which is the best distribution to work with asterisk?
thanks
mark
___
Hi!
| firstperiod | int(10) unsigned | | | 0 ||
(Explain?)
How long is the first billing interval. The first 60 seconds might be
billed at $.04 per minute which then changes...
| startcost| int(10) unsigned | | | 0 ||
Hi,
Sorry Chris, actually, I cannot help you regarding your problem!
But I would like to know how allow an user to change of conferences (go
to an other room) !?!
Regards, Aresk
On Tue, 2004-01-13 at 02:47, Christopher Arnold wrote:
Hi all,
i have a setup with chatrooms, several MeetMe
On 13-01 12:17, Maciek Kaminski wrote:
Steve wrote:
Hi,
If anyone else had a problem I got kphone to work with Asterisk.
I have problems with kphone + Asterisk. KPhone does not seem to ACK
invites, ie.
KPhone --- sends INVITE -- Asterisk
KPhone -- sends 101 Trying ---
On 13/01/04 11:48, [EMAIL PROTECTED] wrote:
which is the best distribution to work with asterisk?
They're all just Linux. There is no best. This question is asked so
frequently it almost looks like a troll to me. :)
I've therefore updated the FAQ on the wiki:
-
Hello,
I have Asterisk running on a RH9 box; Everything seems to be working as it
should, except for Festival. Every time that Festival is called from
Asterisk, Asterisk silently shuts down. Festival doesn't give any error
indication and Asterisk just plain dies without a peep.
Festival was
mattf wrote:
No need to go Xeon, I have one of these:
http://www.tyan.com/products/html/thunderk7x.html
Dual AMD Athlon MP with one 3.3v 64bit PCI slot
MATT---
Athon MP or Xeon IMO are the same thing.. They are just the high end
version of either the AMD or Intel proc respectively..
If you've got spans from different providers...you're in for an
adventure. You'll be able to do one of the following (which one is telco
and luck dependant):
So what you're saying is that the TE410P is not capable of *independently*
clocking each of the T1s. Hell even the venerable old
HiList !
I received an unit of the Symbol
NetVision Phone and i will test it with asteriskusing H.323 or Skinny , somebody tested thisphone
with asterisk and can share experience?
Miklos
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
Mark
p.s. there was no 0.6.0 release.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Here are a list of mirrors for the 0.7.0 tarball.
http://66.225.202.82/downloads/asterisk-0.7.0.tar.gz
http://parc.styx.org/asterisk/asterisk-0.7.0.tar.gz
http://www.bkw.org/asterisk-0.7.0.tar.gz
http://www.moctel.com/asterisk/asterisk-0.7.0.tar.gz
http://matrix.gs/asterisk-0.7.0.tar.gz
Hello * world,
i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen
from the 16th til 20th january. Together with Christian Richter i will
be speaking about * on monday. And we will give an * tutorial on
tuesday. I will be presenting some ISDN stuff there, including the
quadBRI
I'd be happy to give my docs to the project. I just noticed that it was
in progress after I posted but I'd be happy to help.
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12, 2004 1:56 PM
To: [EMAIL
Thanks to everyone that replied!
I'm getting some warnings:
NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
Change audiomode to 0x00140014
The above setting did it - the other info people provided gave me the
On Mon, 2004-01-12 at 19:20, Rich Adamson wrote:
That was my thought too... I sent him a few bucks, but then noticed that
everyone else seems to be sending him a lot more. Maybe if I offer to
write some Asterisk documentation (which I am doing, by the way) people
will send me money!
Can anyone help me with the term that SBC uses to refer to disconnect
supervision? I have an Adit 600 channel bank which has helped improve the
disconnect detection time down to about 8 seconds. This is still causing some
issues in particular with call progress enabled in * we are having a few
There's a BIG difference in price, depending upon what you consider the
equivalent, the Xeon's are about twice as expensive as the Athlon MP's
MATT---
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 8:19 AM
To: [EMAIL PROTECTED]
Subject: Re:
On Tuesday, January 13, 2004 7:36 AM, Andrew Kohlsmith
[SMTP:[EMAIL PROTECTED] wrote:
If you've got spans from different providers...you're in for an
adventure. You'll be able to do one of the following (which one is
telco
and luck dependant):
If all providers are referenced back to a
I am having probelms connecting to voicepulse this morning. Is anybody else
having issues..
burak
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
mattf wrote:
No need to go Xeon, I have one of these:
http://www.tyan.com/products/html/thunderk7x.html
Dual AMD Athlon MP with one 3.3v 64bit PCI slot
MATT---
Athon MP or Xeon IMO are the same thing.. They are just the high end
version of either the AMD or Intel proc respectively..
I have come to a stumbling block.
We have 8 lines coming into an ADTRAN channelbank that then
goes to the * server via a T100P card. I need to route lines 1 and 2 to
everyone when a call comes in on either of them. I also need lines 3 8 to
ring first at specific sip extensions (direct
same here... with nufone too... i was just getting everyone is busy at the
moment message in CLI... it was working fine before..
was it them or was something wrong with my network? will check tomm.
cm
- Original Message -
From: Burak Balasaygun [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Yes, the card works nicely in the 64-bit slots, it just doesn't use all of
the pins. Example, the Tyan S2723 works fine.
The 3.3v key helps to hold it snugly.
Cheers
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:
Hi,
I,m trying to pickup remote call using the SIP protocol and *8# from my
phone but with no success.
I just installed * 0.7.0 and my Phones are connected to one ATA 186 with
image 2.16.1.
I set in the sip.conf the follow parameter:
callgroup=1
pickupgroup=1
for each phone.
Someone can help me ?
Yes, they will - I've tried it. 64-bit, 3.3v slots
Regards
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:www.evtmedia.com http://www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
Hi,
does anyone know if its feasible to run asterisk with a PRI card but not run
any q931 signalling.. basically push calls down the PRI and tell asterisk in
some other way to pickup a particular Zap channel?
Steve
Well, not quite PRI nor quite what you're describing, but would SS7
be what
Hi all!
Is it possible to tell * to allow connecting an incoming (SIP-) call with the
G711 codec (a simple fax). I have not found any setting in sip.conf that
would refer to this problem.
I am using * and the spandsp library to receive faxes from a SIP gateway.
Everything works for now except
Does anyone know of companies or individuals who provide 24x7 asterisk
support options?
-j
--
Jeffrey Paul - [EMAIL PROTECTED] - (877) 748-3467
Senior Network Administrator, Diamond Financial Products
An expert is a man who has made all the mistakes which
can be made in a very narrow field.
Also, I have an error with make make install under
asterisk:
/bin/sh: line 1: ./mkdep: Permission denied
make: *** [.depend] Error 126
Any idea ?
Tnanks,
Marin Blu
--- C. Maj [EMAIL PROTECTED] wrote:
On Mon, 12 Jan 2004, marin blu waxed:
I'm trying to install * on Mandrake 9.2/P4, but
John Brown (CV) wrote:
It appears that zttool doesn't actually report T1 span
errors.
If I inject BPV's, crc errors, framing errors, etc into
a T1 span, the counters on zttool don't change.
It works OK for me with Tormenta 2 and TE410P boards. Both zttool and
the /proc/zaptel/x files seem
Lane Hoskins wrote:
I have come to a stumbling block.
We have 8 lines coming into an ADTRAN channelbank that then goes to
the * server via a T100P card. I need to route lines 1 and 2 to
everyone when a call comes in on either of them. I also need lines 3
- 8 to ring first at specific sip
If you've got spans from different providers...you're in for an
adventure. You'll be able to do one of the following (which one is telco
and luck dependant):
So what you're saying is that the TE410P is not capable of *independently*
clocking each of the T1s. Hell even the venerable old
On Tue, 13 Jan 2004, Jonathan Moore wrote:
LSRB = Loop Start with Reverse Battery
I believe I currently have the lines set to LSCPD which improved the hangup
situation, but hasn't completely fixed it.
Try LSRB - it may work.
--
Joel
___
is just *8
see ya.
matteo.
Il mar, 2004-01-13 alle 16:03, massimo ha scritto:
Hi,
I,m trying to pickup remote call using the SIP protocol and *8# from my
phone but with no success.
I just installed * 0.7.0 and my Phones are connected to one ATA 186 with
image 2.16.1.
I set in the sip.conf
Hi,
Is there a problem with the cvs.digium.com ?
I can not download the asterisk repository.
Thanks,
Marin Blu
__
Do you Yahoo!?
Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes
http://hotjobs.sweepstakes.yahoo.com/signingbonus
Can anyone help me with the term that SBC uses to refer to disconnect
supervision? I have an Adit 600 channel bank which has helped improve the
disconnect detection time down to about 8 seconds. This is still causing
some
issues in particular with call progress enabled in * we are having a
It may not be you, I think the Festival driver is buggy. Specifically,
I've found that the the way in which you pass the text to Festival matters.
If I use the Festival () suntax then it won't work. If I use the wrong
sort of quotation mark instead of ' there are problems. Asterisk will
Il personally use Mandrake 9.2 and it works perfectly.
On Debian, we've never got the FritzCard USB2 ISDN card working, but
nothing to do directly with Asterisk.
The only performance issue I've got was while running X (many comments
around this issue).
JC
[EMAIL PROTECTED] wrote:
Hi
my
On Tue, 2004-01-13 at 07:52, Lane Hoskins wrote:
We have 8 lines coming into an ADTRAN channelbank that then goes to
the * server via a T100P card. I need to route lines 1 and 2 to
everyone when a call comes in on either of them. I also need lines 3
8 to ring first at specific sip extensions
Why not quickly patch the source an release 0.7.1 if the bug is critical?
Give it a few days and I bet we will. because chan_h323 is broken also in
0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.)
bkw
___
Asterisk-Users mailing list
On Tuesday 13 January 2004 03:42, Jason Penton wrote:
I have just a quick question regarding app_txfax for Asterisk.
When I send a fax from asterisk to a traditional fax machine
connected to asterisk via the digium analog card everything works
perfectly. However the same fax machine on the
On Tuesday 13 January 2004 02:27, WipeOut wrote:
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all
the people who worked so hard this weekend to make this
possible!
There is one bug so far and
On Tue, 13 Jan 2004, Jeffrey Paul wrote:
Does anyone know of companies or individuals who provide 24x7 asterisk
support options?
My company does, http://www.internetsolver.com/
dave
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the
Follow-up question, what does * use for fax? T38 or passthrough?
- Original Message -
From: Peter Bittner [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 10:12 AM
Subject: [Asterisk-Users] Specifying a codec to be used in /etc/sip.conf
Hi all!
Is it possible
On Tue, 13 Jan 2004, John Todd wrote:
does anyone know if its feasible to run asterisk with a PRI card but not run
any q931 signalling.. basically push calls down the PRI and tell asterisk in
some other way to pickup a particular Zap channel?
Well, not quite PRI nor quite what you're
Hi,
I'm trying to use the say-ani agi asterisk-perl script and am experiencing
crashes, I am also experienceing problems with the test-agi scripts shipped
with asterisk.
The clearest demonstration of the problem is that if I dial extension 125
configured as...
exten = 125,1,Ringing
exten =
- Original Message -
From: C. Maj [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 10:14 PM
Subject: Re: [Asterisk-Users] ADSI. used beyond own phone network?
What kind of security implications would this have?
Probably the same as using DTMF when you call the
.323/SIP loads with versions
earlier than 2.16.1
SOLUTION:
See patch matrices and workarounds in original advisory:
http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml#software
http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml#workarounds
PROVIDED
hi!
i had some good news regarding the cisco 7920 and the internetworking
with asterisk (and possibly SIP ?).
Status: chan_sccp.so not coredumping anymore :-)
Phone contantly in reboot loop [see below] :-(
Reboot Loop means:
--
Phone auth's with AP
Phone gets IP from
On Tue, 13 Jan 2004, Lane Hoskins waxed:
We have 8 lines coming into an ADTRAN channelbank that then goes to the
* server via a T100P card. I need to route lines 1 and 2 to everyone
when a call comes in on either of them. I also need lines 3 - 8 to ring
first at specific sip extensions
If you've got spans from different providers...you're in for an
adventure. You'll be able to do one of the following (which one is telco
and luck dependant):
So what you're saying is that the TE410P is not capable of *independently*
clocking each of the T1s. Hell even the venerable old
Hello,
I have been playing around with call queuing very cool.
So at the same time I also tried to implement the agent via the agent call back
routine.
This is causing problems, in the queue.conf if I have a
member as
Member = Sip/nick
It works
But if I set up an agent,
I have to agree with the below but only if it is an answer
to the limited question of Which is best to use for my
Astrisk server. For a server you are using such a small
percentage of the Linux distribution that they are effectivly
all the same.
A server will not make us of any of the
I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong?
This on the heels of switch-1.nufone.net being missing out of DNS.
We have customers that expect their VOIP to work. Is there anybody that's reliable?
I am having probelms connecting to
I have a little more info on this. Following the suggestion of another post on
this topic I tracked down an analog phone with lighted buttons powered by the
phone connection. I directly connected the phone to one of my inbound lines and
called it with my cell phone. Picked up the analog phone,
http://shtoom.sourceforge.net/
I haven't tried it yet, but it looks promising. Written in Python.
Supposedly works on Linux/FreeBSD, Windows, MacOS X. Written
specifically with Asterisk as a server testbed, I believe.
JT
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Asterisk-Users mailing
Thanks David,
That is exactly what we had to do. We got some help from Digium as well
and have it taken care of.
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 10:33 AM
To: [EMAIL
I think the exchange below shows us that before 0.8.0 comes
out, maybe there should be a 0.8.0-beta then after no problems are
reported in a few week period a 0.8.0-release candidate and
ten 0.8.0 itself.
It's hard to call a realease stable until a number of people outside
the developer's lab
Hi,
I just bought the E100P from digium. It has both
keys: 3.3V and 5V, so it would fit both, in a 5V-PCI
slot and in a 3.3V PCI slot.
Is it true, that I can plug it without destroying it in an
ordenary 5V PCI slot?
Roger.
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In our last exciting episode, Tilghman Lesher ([EMAIL PROTECTED]) said:
I want you to look at the headers of my reply and note that I'm running
my mail client on FreeBSD.
Now my advice: run your Asterisk server on Linux.
First, a disclaimer: this is not mean to be flame-bait nor is it an
If you don't have a voltmeter to look at this, try just listening on the
line (using an analog telephone) when the far end hangs up. You should hear
a distinct click-click on the line a second or two after they hang up. If
you hear this, it's likely you are getting the required disconnect
is just *8
I've tried but it does not pick up the call and don't show nothing in the
consolle
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To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source? The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each device) happen
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