On Tue, 11 Jan 2005, Alejandro G wrote:
> The call is received from the PSTN by an NMS E1 with ISDN Pri (EuroISDN) and
> is switched to the NMS T1 using NMS MVIP internal bus switch that is similar
> to a H.100 bus with few streams/timeslots.
>
> Once in the T1 (which is running National ISDN 2 P
Hi, I found it and got it working excellently fixing a few annoying
issues I was having with the Polycom IP300's.
Peer Oliver Schmidt wrote:
Andrew Thrift wrote:
Hi Remco,
just wondering how you got Asterisk 1.0.3 BRI-Stuffed.
On Junghanns.net I can only see 0.1.0-rc4 of bri-stuff and it uses
s
Andrew Thrift wrote:
Hi Remco,
just wondering how you got Asterisk 1.0.3 BRI-Stuffed.
On Junghanns.net I can only see 0.1.0-rc4 of bri-stuff and it uses
something like asterisk 0.8
I got 1.0.2 using the bristuff v0.in the download section. It is not
linked from the main page, but available in the
Hello,
The below worked, and the 'Request to schedule in the
past?!?!?' messages are gone! Thanks for the
assistance.
Jason
--- Tom Ivar Helbekkmo <[EMAIL PROTECTED]> wrote:
> Jason Goecke <[EMAIL PROTECTED]> writes:
>
> > I have compiled and installed the format_mp3 and
> > ensured the modul
Hello all!
If I place a call to our number, the call is routed to our Asterisk box
from teliax --> IAX2 --> firewall w/ port forwarding --> *
If that caller dials an extension that rings an outside line, where our *
box makes an outbound connection to teliax to terminate the call, we get
chop
I am new to asterisk but learn a lot about it to this mailing list and
> wiki currently i am facing problem about sip phone i have "PA 1688"
> chipset ip-phone and i have iptel.org sip account i registered locally
> and through iptel.org comfortably my problem is that when i called
> from my sip ph
I've put in a request to get the board replaced. Hopefully this will work
out. I too drove myself crazy because the Digium diagnostics don't really
indicate anything is wrong with the hardware.
-Original Message-
From: Steve Prior [mailto:[EMAIL PROTECTED]
Sent: Monday, January 10, 2005
Hi All,
We are currently running Asterisk 1.0.3-BRISTUFFED on Gentoo Linux with
a 2.6 Kernel and this is connected to a Telecom ISDN-BRI connection with
an HFC-S card, this works quite well but as you can imagine, 2 lines is
not enough for a company with 40 VoIP phones.
We are about to have Tel
test
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This message has been scanned for viruses and
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Asterisk-Users ma
The call is received from the PSTN by an NMS E1 with ISDN Pri (EuroISDN) and
is switched to the NMS T1 using NMS MVIP internal bus switch that is similar
to a H.100 bus with few streams/timeslots.
Once in the T1 (which is running National ISDN 2 Pri) this board "calls" the
TE110P in asterisk whi
http://www.digitnetworks.com/store/
$30
Its the card on the bottom left, I have one, which I have been
testing with, works without any problems so far.
On Mon, 10 Jan 2005 20:09:54 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> > Does asterisk support the intel 537/md3200 chipset? I don'
That's data or fax CNG, not dtmf. And yes, it's disabled for the duration
of the fax or data session.
-d
At 11:36 PM 1/10/2005, you wrote:
Hello all,
I am getting console debug messages about "tone detected on channel XX,
disabling echo cancelation on channel XX" when using echocancel=yes with a
Hello all,
I am getting console debug messages about "tone detected on channel XX,
disabling echo cancelation on channel XX" when using echocancel=yes with a
Digium T1 card.
does this mean that DTMF breaks the echo can? Does Asterisk permanently
disable the echo can or is it for that channel i
Hello list,
I have about 20 Digium TE-405Ps out in the field, and I started having
trouble with one just recently. The card had worked fine for a month with 4
PRIs in NFAS configuration, and then all of a sudden I started getting a
disappearing D channel. A restart of asterisk / ztcfg /module
How can found dynamic dialplan?
in extensions.conf
[default]
exten => 111,1,DBget(aaa=111/forwarding);It
can be 2 to 9 begins.
exten => 111,2,
exten => _[2].,1,Dial(SIP/[EMAIL PROTECTED])
;AA
exten => _[3].,1,Dial(SIP/[EMAIL PROTECTED])
;BB
...
,how transfer to correct router.
nam
On 10/01/2005 04:39 Leif Madsen said the following:
The Asterisk Documentation Project is proud to present, "The History
of the Zapata Telephony Project as it relates to the Asterisk PBX".
Written by Jim Dixon, the founding father of the Zapata telephony project
(http://www.zapatatelephony.org) whi
On 11/01/2005 04:21 Miguel Ruiz Velasco Sobrino said the following:
In a setup I've made i have a problem in the two way origination of the call.
Asterisk 1 <==> Public internet <==> NAT <==> Asterisk 2
I'm pretty sure it's a NAT loosing state too fast, and i can do nothing to fix
the NAT.
Is ther
Just a thought I had on this.. Why not setup a sip peer entry in
sip.conf with a qualify statement in it and send the call to the peer
entry? That way asterisk will know if the peer is alive or not and I
would think it would skip that particular peer accordingly.. that or
it's really late and I
Here's a strange one - when I run safe_asterisk on either of these
distros, words that are colored blue or violet (but not red) turn up in
Russian (and some other languages, I think). If I run asterisk with the
same arguments (-vvvg -c) as safe_asterisk does, from the console, it's
OK. If I r
Hello All,
I have 4 X100P cards. I was hoping to have card (line) go to separate ext.
i.e.
Card 1 (XXX)555-0001 My Ext
Card 2 (XXX)555-0002 Wife's Ext
Card 3 (XXX)555-0003 Fax Ext
Card 4 (XXX)555-0004 My and Wife Ext.
This is what I have now and all incoming line rings this one extension.
exten
AHBLWEB wrote:
Aha! Remove the Digium card and everything sounds fine. Leaves me with a
SIP-only server though.
Looks like I'd better RMA that sucker.
I had a similar problem with a TDM11B - even VOIP calls to the Digium
demo server were broken up when the card was in and the FXO and FXS
port
Brian Dingman wrote:
Anyone care to pass on a makefile that works. This is what my
makefile.rej looks like:
[SNIPPED]
Really it's not that hard. Open two console windows. In one open that
patch. In the other open the Makefile.
If you look at the patch you can see what lines need to go into the
Hi
I've been running asterisk for a year or so now, and recently upgraded
to the lastest CVS-HEAD version along with oh323 v0.7.1. (previously
oh323 version 0.6.3 used to crash about once per week during high call
volumes). Since upgrading, however, asterisk seems unable to
successfully receive h
Anyone care to pass on a makefile that works. This is what my
makefile.rej looks like:
***
*** 71,76
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ $< $(CURLLIBS)
Hi,
My * (latest stable CVS) is not sending the caller id on its zap channels
(digium TDM40B). The callerid is shown on call-waiting, but is hidden if
the ringing channel is not already in a call.
The same * and configuration was working before upgrading to the latest
stable CVS.
Of course I hav
On Mon, Jan 10, 2005 at 03:26:04PM -, Paul Brock wrote:
> On Mon, Jan 10, 2005 at 15:18, Paradise Dove said:
> > On Mon, 10 Jan 2005 06:45:54 -0800 (PST), Jason Goecke
> > <[EMAIL PROTECTED]> wrote:
> > > Hello,
> > >
> > > Ever since I started using Asterisk I always get this
> > > error:
> >
Adam, I think I got it worked out...
I changed disallow=723.1 to disallow=all and then accepted back in
ulaw,alaw,gsm and ilbc and
it started accepting the calls. I do not know why, but its working now.
FWIW, here is the full frame as it was before:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 T
> Does asterisk support the intel 537/md3200 chipset? I don't want to start
> any flames here, I know all about using generic crap in asterisk,[*] which
> I really don't approve of other than for testing, but I have a customer
> demanding a generic chipset for his one backup analog line. He wi
> I have a similar problem. I asked the same question in a message to the
> list a few days ago titles "IAX outgoing redundancy". It would seem
> app_dial would need to have some code added to it to have two different
> kind of timeouts, one an answer timeout (which is the current timeout in
> the
Can you paste the full NEW frame please. Could be Preference vs capability
thanks,
Adam
Ernie Ankele wrote:
On a sip to iax :
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 0ms SCall: 19170 DCall: 1 [
On Mon, 10 Jan 2005 12:51:49 -0500, Michael Lyszczek
<[EMAIL PROTECTED]> wrote:
> Anyone have any issues like thisI am fwding broadvoice to zaptel,1
> with my t100p and the t1 goes to a zhone zplex10b.. I can ring
> extension 1, which is pair 1 of the channel bank, but it doesnt
> recognize off
On a sip to iax :
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 0ms SCall: 19170 DCall: 1 [xx.xxx.xxx.xxx:20406]
FORMAT : 4
-- Call accepted by xx.xxx.xxx.xxx (format ulaw)
-- Fo
use ethereal or iax2 debug to see what capabilities are been set in your
NEW message
Ernie Ankele wrote:
Hello,
Could someone give me clues where to figure out this problem?
If I call from a Sip client to an Firefly client running IAX, the call
connects fine, no problems.
I can connect to asteri
Hello,
Could someone give me clues where to figure out this problem?
If I call from a Sip client to an Firefly client running IAX, the call
connects fine, no problems.
I can connect to asterisk using any codec (excepting g.729) on firefly
to voicemail and music-on-hold, other sip extensions and e
Henry Devito wrote:
I really don't approve of other than for testing, but I have a customer
demanding a generic chipset for his one backup analog line. He will not
spend the money for a Digium card and says he will find another
company if I can not provide a generic FXO port.
Is that really the ki
Hi All,
I'm aware of this statement by Mark circa July 2004:
-begin-
SIP is an IETF standard. While there is some fledgling documentation
courtesy Frank Miller, IAX is not a published standard at this time.
-end-
And the thread "How far is IAX to be a Standard" circa November 2004. It
is not my in
On Mon, 2005-01-10 at 18:50 -0600, Henry Devito wrote:
>
> Does asterisk support the intel 537/md3200 chipset? I don't want to start
> any flames here, I know all about using generic crap in asterisk,[*] which
> I really don't approve of other than for testing, but I have a customer
> demanding
Hi Remco,
just wondering how you got Asterisk 1.0.3 BRI-Stuffed.
On Junghanns.net I can only see 0.1.0-rc4 of bri-stuff and it uses
something like asterisk 0.8
Your help is much appreciated.
Regards,
Andrew Thrift
Remco Barende wrote:
On Sun, 9 Jan 2005, Remco Barende wrote:
I hva ean HFC-S car
Aha! Remove the Digium card and everything sounds fine. Leaves me with a
SIP-only server though.
Looks like I'd better RMA that sucker.
-Original Message-
From: AHBLWEB [mailto:[EMAIL PROTECTED]
Sent: Monday, January 10, 2005 5:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial D
Hi all,
I have taken my family in hostage and setup an asterisk environment at
home...
I am using a x100p fxo card, a Iaxy and X-ten softphones.
Works ok with local calls.
There is a strange behavior, when we do long distance calls, it keeps
ringing on our end, remote callee answers the call bu
Does anyone know Zultys ZIP 2 ip phone is fully compatible with asterisk?
Does the MWI and call transfer work correctly?
Henry Devito
Telephone Connection, Inc
Network Design / Implementation
Phone: 402.330.7510
Fax: 402.330.8586
Toshiba CTX/DK/Stratagy Certified
Cisco Certified Internetwork
Major difference is an Adaptec I2O SCSI RAID controller driving an external
RAID 5 array. I'm also wondering if I haven't gotten a bad Digium TDM11B
card (Dev Kit PCI). Although the zaptel drivers seem to load properly and
there are no errors when starting Asterisk I don't get a dial tone on the
FX
You'll probably find the IRC channel to be a lot of help. The folks
there generally are also here. And *most* of them like to help others.
It sounds like some sort of I/O bottleneck.
Have you opened up another terminal and run top at the same time you are
calling in to get an idea of what's go
Does asterisk support the intel 537/md3200 chipset? I don't want to start
any flames here, I know all about using generic crap in asterisk,[*] which
I really don't approve of other than for testing, but I have a customer
demanding a generic chipset for his one backup analog line. He will not
s
> I think no one should abuse Asterisk and make it into a telemarketer
> tool. In fact, it is designed to supposedly drive telemarketers away!
There's telemarketing and then there's telemarketing. Everyone's opinion
is different but I think the type you are referring to are probably the
ones that
Anybody know if there are any issues with these phones and * ? Does the MWI
work and such?
Henry Devito
Telephone Connection, Inc
Network Design / Implementation
Phone: 402.330.7510
Fax: 402.330.8586
Toshiba CTX/DK/Stratagy Certified
Cisco Certified Internetwork Expert (CCIE) Voice ( VoIP)
C
Thanks for the input.
Downloaded kernel sources, recompiled, re installed Asterisk from scratch.
Now I have no sound at all.
Must be the box.
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Monday, January 10, 2005 12:41 PM
To: Asterisk Users Mailing List -
> does anybody knows from where I can get an list of
> international area codes incl. the mobile numbers?
The way I did it is to get the rate tables of one of the IAX providers
(LiveVOIP, VoipJet, NuFone come to mind).
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york)
Hi Sam,
Did you build libunicall with
./configure
make
make install
If so, the library will be in /ustr/local/lib. Is this in your search
path? Wither add this directory to /etc/ld.so.conf, or build with:
./configure --prefix=/usr
make
make install
This is an issue common to most packages which u
Chris Miller wrote:
Dave Green wrote:
I've downloaded the latest CVS as of yesterday. Zaptel and libpri
compile and link OK but after issuing the "make asterisk" command I
get the following:
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
mak
Hi!
> - No internal Nikotel call (phone number beginning with 99) reaches my
> friends (which have similar sip.conf and extensions.conf). Somewhere I
> read that the section must be named like the host "calamar0.nikotel.com"
> so that asterisk finds it. It didn't help. Did someone manage to get
>
> But this also means that after 20 seconds of ringing it goes
> on the next dialpeer. I would like to be able to set the
> timeout Asterisk wait to establish a connection, any
> connection, with the gateway to something much shorter than it is now.
I have a similar problem. I asked the same quest
Barry Flanagan wrote:
> > - Although the cards' credit seems to be maintained correctly, I
> > cannot see the call details in astcc-admin. When I try to view
> > information on the card, it's just blank. Any ideas?
> There is a bug in the CREATE statement for the cdrs table.
> you need to create a
> > I can do the dial command like this to give me a 20 second timeout
> >
> > exten => _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20)
> >
> > But this also means that after 20 seconds of ringing it goes on the next
> > dialpeer. I would like to be able to set the timeout Asterisk wait to
All,
Capi config ok, but I cant get a incoming number displayed on the screen
of the snoms; only asterisk appears. What do I need in my extension.conf
to make it display the number received??
Mike
___
Asterisk-Users mailing list
Asterisk-Users@lists.d
Adi Linden wrote:
I can do the dial command like this to give me a 20 second timeout
exten => _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20)
But this also means that after 20 seconds of ringing it goes on the next
dialpeer. I would like to be able to set the timeout Asterisk wait to
establi
I can do the dial command like this to give me a 20 second timeout
exten => _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20)
But this also means that after 20 seconds of ringing it goes on the next
dialpeer. I would like to be able to set the timeout Asterisk wait to
establish a connection,
> >I noticed the following came into cvs head yesterday:
> >
> >>Modified Files:
> >>fxotune.c wctdm.c wctdm.h
> >>Log Message:
> >>More TDM card echo API modifications. Making the fxotune program
> >>automatically
> >>find the correct coefficients for the module. Lots of neat stuff.
> >>
Having read the majority of the doc for the chipset used on the TDM
card, I'm fairly certain (about 80% sure) the reference to echo can
is for near-end cancellation. The far-end is handled by * code.
Still a 20% chance of me being wrong though.
>
> wctdm.c
>
> LogMessage
I'm using a Grandstream IP phone to call someone through our asterisk
pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses
2x ZAP-HFC cards.
When I call someone, if the call isn't answered and then I hang up, I
get "487" coming up on the grandstream phone. If I pick up the receiv
Christopher L. Wade wrote:
Andrei (MPI) wrote:
Brian West wrote:
Just an FYI to all out there that are upgrading after this weekend's
run of
CVS updates that are in now... MAKE SURE YOU DO "make clean". If
you don't
and asterisk acts funny this is why. Anytime any struct like
ast_channel
(whi
Dave Green wrote:
I've downloaded the latest CVS as of yesterday. Zaptel and libpri
compile and link OK but after issuing the "make asterisk" command I get
the following:
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: leaving direct
> >Just an FYI to all out there that are upgrading after this weekend's run of
> >CVS updates that are in now... MAKE SURE YOU DO "make clean". If you don't
> >and asterisk acts funny this is why. Anytime any struct like ast_channel
> >(which was changed over the weekend) and you don't make clea
>From the Asterisk Daily News:
wctdm.c
LogMessage: Adding FXO module support for onboard echo cancellation
Comment: /* Set the digital echo canceller registers */
Details: It would appear that the chip on board the TDM400P cards is
capable of hardware echo cancellation. This has just been impl
OK, I've done all of the suggestions, and had already read and met the
requirements. I'm running Asterisk Stable 1.01 on Red Hat Enterprise 3.0
Server. We're using AgentCallBackLogin to log the agents in, and then
pointing them to a SIP channel (Agent/4102 logs in, and enters 4102 (SIP/4102)
as h
The SPA-3000 is your best bet then. It works great. Both lines will register
with Asterisk, and then you can have Asterisk push calls to the 2nd line
(the FXO port) and it will be placed as normal. Works great. The phone
connected to line1 will act perfectly normal, just like the SPA-2000, or the
C
Rich Adamson wrote:
I noticed the following came into cvs head yesterday:
Modified Files:
fxotune.c wctdm.c wctdm.h
Log Message:
More TDM card echo API modifications. Making the fxotune program automatically
find the correct coefficients for the module. Lots of neat stuff.
If I modify the
Doesn't that depend on the converter? With my Cisco ATA186, when I call a
party, then flash, call another party, flash back so we're in a 3-way call,
when I hangup, everybody gets dropped.
With my Sipura SPA-2000, when I do the same trick, the other 2 stay
connected. I found the configuration opti
Nabeel Jafferali wrote:
Often, when someone tries to dial any internal extension or
external number, they get the "Reorder" message. If they try
again, they get another "Reorder" message. If they try a
third time, the call gets through.
I have one Cisco 7960 inside a NAT with an * box on a public
I have no experience with the Sipura product. Does it work well with
Asterisk? It looks like it would definately fit my needs. I need one FXO
and one FXS interface and it looks to fit that requirement. This
particular 4801 (I have a few) is currently running debian off of a 512
MB CF card (with
Miguel,
you can try using # as a way of transfering the call, but that's a blind
transfer meaning that you will be prompted an extension number and the call
will be transfered and that's it, on the other hand pressing flash put's the
call on hold and then let's you dial another call and if you pres
Andrei (MPI) wrote:
Brian West wrote:
Just an FYI to all out there that are upgrading after this weekend's
run of
CVS updates that are in now... MAKE SURE YOU DO "make clean". If you
don't
and asterisk acts funny this is why. Anytime any struct like ast_channel
(which was changed over the week
I suspect if someone recompiled the module that handles the transcoding
it could take advantage of say a AMD64 bit cpu. Although I bet cache
has a big impact on performance with this.
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
Jason,
This problem may be happening because asterisk server and PC or hardware
phone clock are out of sync .
You need to find a way (e.g. ntp with atomic clock etc) to sync time up
to a second on all the devices involved in the network communication.
Andrei
Jason Goecke wrote:
Hello,
I was mon
I wouldn't mind seeing the g729 codec written to take advantage of 64bit
processing. Might make a machine capable of handling more PRI -> g729 calls.
Right now the limit is about 80 on a dual 3.2Ghz Xeon machine.
-Matthew
- Original Message -
From: "Chris Miller" <[EMAIL PROTECTED]>
To:
I've downloaded the latest CVS as of yesterday. Zaptel and libpri
compile and link OK but after issuing the "make asterisk" command I get
the following:
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: leaving directory '/usr/src/aste
I have a problem:
When I'm in a call and a second call arrive (call waiting) I can't transfer
the first call. If I press flash the line change to the second call, if I
press flash again the line change to the first call.
How I can transfer a call in this kind of situation ?
Kind regards,
Migue
Brian West wrote:
Just an FYI to all out there that are upgrading after this weekend's run of
CVS updates that are in now... MAKE SURE YOU DO "make clean". If you don't
and asterisk acts funny this is why. Anytime any struct like ast_channel
(which was changed over the weekend) and you don't mak
She already has the if all else fails button. It is like a life alert and
seizes the line in such a manor that blocks any internal device from
"hanging the line".
In the circumstance you describe, Asterisk would be back up within a few
minutes so would not be a real problem. Also the box would h
> ignorepat => ${DIAL_OUT}
> exten => _011.,1,SetGroup(${CALLERIDNUM})
> exten => _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com
> WORLD exten => _011.,3,Congestion
> exten => _011.,103,Macro(outisbusy)
> exten => _${DIAL_OUT}011.,1,SetGroup(${CALLERIDNUM})
> exten => _${DIAL_OUT}011.,2
Be very careful with your 4801 - Soekris boards are designed to only
support 3.3V PCI at very low power levels - putting a TDM card in there
would very much exceed the allowed power use on the PCI connector.
My setup will be using my Soekris 4801, a 40G 2.5 IDE drive for
voicemail storage/boot
> Often, when someone tries to dial any internal extension or
> external number, they get the "Reorder" message. If they try
> again, they get another "Reorder" message. If they try a
> third time, the call gets through.
I have one Cisco 7960 inside a NAT with an * box on a public IP (running
Stab
no *XX in my extensions.conf.
In features.conf:
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 1355 ;Number of sec
I'm running * on an AMD 64 system with FC3 x86_64, everything works fine
so far. Programs can be rewritten to take advantage of the the 64 bit
architecture and the extra computing power. Having seen that many high
end systems are using 32 bit Xeon based systems for call capacity, I'm
wondering
I did try and another PCI card, an old 3com NIC, it works fine. I'm
starting to think that the X100P is 5 volt only (can't find the specs
anywhere). 5 volt cards do not work in the Soekris :( I think a TDM will
but it's too big to fit in the soekris case. I'm starting to think that
I'm going to hav
Lessons sometimes show us how silly we are to post to a list of 8000
users before exhausting our own endeavors.
I never tried to test this by creating the context and pressing a number
not defined.
I just setup the "Context" to jump to, and Wolla
If I press 0 the operator is dialed.
If I press 1
What do you mean "it also applies to busy signal"? Can you elaborate?
My dial-plan is something similar, I have like
BroadVoice/VoipJet/NuFone/LookieLoo and if I set them in order of my
preference, I've never had the primary fail so I've never witnessed this 60
second delay. But am interested in w
Hi list,
I have some nontrivial questions. I am no telecommunication guru and I
will explain it with my simple words. I hope someone can help me with
these issues (with Asterisk 1.0.3):
- If I call outside (with Nikotel to German Telekom) there is a remote
hangup after 2 minutes. I've seen other
Just an FYI to all out there that are upgrading after this weekend's run of
CVS updates that are in now... MAKE SURE YOU DO "make clean". If you don't
and asterisk acts funny this is why. Anytime any struct like ast_channel
(which was changed over the weekend) and you don't make clean you'll end
Is there a *88 in your extensions.conf? Or any * codes for that matter? If
it's not listed in the Sipura, it should send it to Asterisk, and if it
works, then it has to be in there somewhere. Otherwise it's an "unlisted" or
unconfigurable Sipura feature.
-Original Message-
From: [EMAIL PRO
Joe Dennick wrote:
The hype and documentation for the last couple of releases of the Flash
Operator Panel claim that the Panel can be configured to either change the LED
for a phone, or the name of a phone to indicate when that phone is logged into
a queue. I've tried on two different versions (0.
>-Original Message-
>From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
>Sent: Monday, January 10, 2005 8:46 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] ACD Queue question.
>
>
>Queues.conf
>
>Is it possible to have asterisk only drop out of the queu
Hi,
This is the motherboard:
SM X6DAE-XG2 Dual XEON 800FSB EMT64 w/2-Ch SATA-R 0&1,SVGA,2xGb LAN
Dual Intel® Xeon EM64T Support up to 3.60 GHz
Intel® E7520 (Lindenhurst) Chipset
1(x8) PCI-Express on (x16) Slot, 3 x 64-bit 133MHz PCI-X, 2 x 64-bit 100MHz
PCI-X Slots
ATI RageXL 8MB Graphics
Actual
I moved my TDM400B cards (first two cards are 40's, third is a 31, last
is an 04) from one computer to another, copied all the config files, and
now the LED on the line 11 - third line of the third card doesn't go on
(it used to on the previous computer). I can get by telling * not to
use this
Got it now, was a stupid error, I suppose make install would copy the
modules but this wasn't the case.
Thank you very much,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: lundi 10 janvier 2005 16:26
To: Asterisk Users Mailing List
Michael Lyszczek wrote:
Anyone have any issues like thisI am fwding broadvoice to zaptel,1
with my t100p and the t1 goes to a zhone zplex10b.. I can ring
extension 1, which is pair 1 of the channel bank, but it doesnt
recognize offhook and it keeps ringing the phone after I pick up.
Also, its
We have a strange issue here - we have the following setup:
Asterisk CVS-HEAD-12/15/04-07:42:16
40 SIP Cisco 7940 phones, linking to PSTN via EuroiSDN 30 channels.
Often, when someone tries to dial any internal extension or external
number, they get the "Reorder" message. If they try again, they g
On Mon, 2005-01-10 at 14:48 -0600, Shawn L. Djernes wrote:
> Hello List,
>
> Does anyone know of a device that works with the TDM400P FXO/FXS Modules to
> provide line backup for power failure?
>
> I have an idea for such a device but do not have enough vision to do the
> soldering of the parts.
Thanks Paul.
I understand and can agree with that.
Does that also hold true for a remapped (*8 to *88), PickupGroup?
I ask, because this isn't working either from firefly, which is what
started the head scratching.
*88 is not listed in the regional settings on my sipura 2000's, which
is why I cho
Hello List,
Does anyone know of a device that works with the TDM400P FXO/FXS Modules to
provide line backup for power failure?
I have an idea for such a device but do not have enough vision to do the
soldering of the parts.
My ideal device would be able to sense when asterisk has brought the FX
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