test message
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Thanks a lot, I'll look into it from here.
AJ
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Thanks a lot I might give it a try. Any specific instructions for running
it with asterisk?
AJ
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Hello all,
In browsing through my latest version of Linux Journal I noticed the LipZ4
Sip Soft Phone advertised. I was wondering if anyone had yet used it with
asterisk? And if so, what were the results? Thanks.
AJ
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Hi all,
I'm having a bit of a problem using the # sign to transfer when using a
soft IAX2 client. Has anyone else experienced this problem or know of a
possible work around / fix of this problem. The following is a snippet
from my extensions.conf file. This is how the file is setup for inboun
Many thanks to Mike, Steve and all the IaxComm and Diax users who
contributed to tracking down and solving this problem.
AJ
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When I get a chance I will zip over to their website and give you the
absolute url. I was looking at the hard copy.
AJ
On Wed, 14 Jan 2004, Franz Edler wrote:
> > From: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004
>
> > For anybody who didn't know there is an article on asterisk in Fe
For anybody who didn't know there is an article on asterisk in February's
Linux Journal.
AJ
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Hello all, Can someone send me a current working example of what the
entries need to look like for iaxtel and the iax.conf file inclusive of
the register line? It's been quite awhile since I have done anything with
iaxtel and I need to do a little bit of testing. Currently it appears
that I can m
Since the list community has done so much for me in my humble asterisk
beginnings I have put together a simple little script written in php that
serves as a paging reminder script. If anyone is interested in a copy of
it contact me off list and I'll forward you a copy.
The basics of the script
Also to add a bit to what I said earlier, the reason I thought RedHat /
Fedora was a good howto was the mere fact that RedHat / Fedora has been
known to present it's own distinct installation problems because of
packages / pachage dependencies.
AJ
___
First of all regarding my social ettiquette, it has nothing to do with my
lack of it but more to do with my philosophy of chew them up, spit them
out regardless of the venue they choose. I'll play on field, "You lay it,
I'll play it." As far as your thought of a generalized howto, it happens
Quite frankly, I don't need a howto, I have it running on my Fedora core 1
system as well as my RH 9 system. I just thought it might be a good idea
for newbies or other people not very familiar with Linux or asterisk,
needed packages, dependencies, etc. Personally I think thorough
documentati
Happy New Year Wipeout and all the other Asterisk Users around the globe
from the Shore Linux Solutions Team
Special kudos and a Happy New Year to the Digium Team
P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9
Howto of 2003?
AJ
__
Brian
I have a 24 channel fxs channel bank. The asterisk server has 2 digium
T100 cards. One connects to the channel bank and one connects to the
Dmark (PRI). As I understand it the line signalling=pri_cpe is for the
card that talks to the Dmark (PRI), channel => 1-23 then signalling=fxo_ks
In my zapata.conf file I have:
channel => 1-23
signalling=pri_cpe
signalling=fxo_ks
I have a 24 channel channelbank on the inside and a pri coming from
Verizon on the outside. All previously worked fine with my zaptel.conf
file and zapata.conf file.
On Mon, 29 Dec 2003, Tim Thompson wrote:
The only things uncommented in my /etc/zaptel.conf file are as follows:
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
loadzone=us
defaultzone=us
On Mon, 29 Dec 2003, zoa wrote:
> try running ztcfg first
>
>
>
>
> At 09:26 29/12/2003 -0500, you wrote:
> >Hello all
> >I just checked out the lates
Ok what do I do with ztcfg?
On Mon, 29 Dec 2003, zoa wrote:
> try running ztcfg first
>
>
>
>
> At 09:26 29/12/2003 -0500, you wrote:
> >Hello all
> >I just checked out the latest
> >zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through
> >the entire make procedures. E
Hello all
I just checked out the latest
zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through
the entire make procedures. Everything seemed to go fine however now when
I attempt to start asterisk, it says ok but it seems to be immediately
crashing. The following messages a
Merry Christmas Ollie from all of us Asterisk people in the US/East Coast
region.
AJ
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Quite a few of us have noted this problem with not only DIAX but also
iaxcomm. It just seems that no one has figured out what the problem is
yet. Though I believe that there are no shortage of people trying.
AJ
On Sun, 21 Dec 2003 [EMAIL PROTECTED] wrote:
> Yes,I often get the same result, b
Speaking for me personally, I have experienced the problem when not using
NAT. The particular incidents I referenced dealt with an asterisk server
and client, both being on the 192.168.22 network.
AJ
On Tue, 16 Dec 2003, Patrick Cantwell wrote:
> Just a thought: Out of everyone having proble
Hello All
When I open up iaxcomm, it registers fine with the asterisk server. If I
call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
for awhile (I haven't figured out exactly how long) it seems to miss
calls. I can see the calls coming in on the asterisk server but they
nev
I haven't personally switched to Fedora but I did decide to upgrade a lot
of the packages on my * box from RH9 to Fedora. I have not spent a lot of
time monitoring how it has handled the load but it does seem to run quite
smoothely. After having installed many of the packages to satisfy library
The new versions of iaxcomm and DIAX are both now using the iax2 protocol.
So in order to receive incoming calls on either of them in your
extensions.conf file change IAX/clientname to IAX2clientname. Then you
should be able to receive incoming calls on either iaxcomm or DIAX. Also
there is
Steve Kahn and I were having this very discussion the other day on the
iaxclient-devel list. I know that Steve is now aware of it and I believe
he's going to pass the the same suggestion to Mike VanDoselaar. I can't
speak for either Steve or Mike but I think you will probably be seeing it
in
>From the console, I see where the call comes in and I can see where the
party from the outside hangs up. The next thing that is said is as
follows:
"libgcc_s.so.1 must be installed for pthread_cancel to work".
Now I've taken a look on my system and I do in fact have the libgcc_s.so.1
on my s
I deleted all the asterisk related directories and their subdirectories
from /usr/src/ and did a brand new check out of zaptel, zapata, libpri,
asterisk-addons and asterisk.
AJ
On Sat, 29 Nov 2003, Tilghman Lesher wrote:
> On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote:
> > Quotin
I just deleted all the old directories, did a fresh checkout and make
install and the problem still persists.
AJ
On Sat, 29 Nov 2003 [EMAIL PROTECTED] wrote:
> Quoting [EMAIL PROTECTED]:
> > In the zaptel zapata and libpri directories I executed a make clean and
> > did a cvs update and then r
In the zaptel zapata and libpri directories I executed a make clean and
did a cvs update and then ran make install. In the asterisk directory I
did a make clean, a cvs update and a make upgrade. So I guess the answer
to your question is yes I did take care of the other things as well. At
lea
I've been running an asterisk server for several months on my RedHat 9
box. Today I decided to update some of the glibc packages on my box and
upgrade it to the latest asterisk cvs as well. I can start the asterisk
server without incident. I can place a call in from the outside and it
takes
Take a look at John Todd's configuration files:
http://www.loligo.com/asterisk
There should be some iconnect stuff there as well. One of the things that
I noticed from your sip.conf file is that you appear to be missing a
register line in the [general] section. If you are behind Nat it should
In my extensions.conf file I'm attempting to distinctively ring one of my
zap channels with a different ring depending upon whether the call is
received from the DID or inside extension. The DID extension looks like
so:
exten => 5551236543,1,Dial,Zap/28r1|20
However, when I dial in on
Dan
I seem to be having the same problem as some of the other guys. With all
the previous versions I could make outgoing and receive incoming calls;
however with this latest version even if I have Diax open the call drops
through to the busy priority in my extensions.conf file. It's like it's
Actually I'm only using it for incoming calls; however I believe John Todd
has sample configs posted on his site and these should include some
examples for iconnecthere. His site is http://www.loligo.com/asterisk.
AJ
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I'm receiving calls on my asterisk server from iconnecthere. My asterisk
server is behind nat but it still seems to be working fine.
AJ
On Fri, 21 Nov 2003, Chris HARIGA wrote:
> Hi,
>
> Is anyone using the iconnect on Asterisk to receive and to place calls?
>
> Best regards,
>
> Chris HARIG
Hi guys
I just registered an incoming number with iconnecthere and I'm trying to
set up incoming calls from icconnecthere on my asterisk server. I took a
look at john todds sample sip.conf and extensions.conf file but for some
reason my incoming is still not working. At this point I wish to us
HI guys
I do have usb-uhci. How do I build ztdummy? I think once its built I
just have to do a modprobe to load it, I just don't know how to load it.
AJ
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Hi guys,
Having a bit of a problem trying to get conference bridges working. In my
meetme.conf file I have the following line
[rooms]
conf => 6000
In my extensions.conf file I have:
exten => 1000,1,MeetMe,6000
My problem is that when I dial into extension 1000 it is telling me "this
is not a
Can someone please aid me in getting the transfer feature to work on my
IAX clients. I tried putting the ,t options on my dial line of
extensions.conf, but this does not work. Below is what my extensions.conf
dial line looks like:
exten => _91NXXNXX,1,Dial($IAX2/[EMAIL PROTECTED]/${EXTEN:1})
Good Job buddy, I'll be checking it out myself. Also, can you help me
figure out how to transfer calls from an IAX client?
A.J.
On Fri, 14 Nov 2003,
Michael Van Donselaar wrote:
> On Fri, 14 Nov 2003 05:36:25 + (GMT), mukta vasudeva
> <[EMAIL PROTECTED]> wrote:
>
> >Hi,
> >I have conf
Well in pine I'm sending it out as [EMAIL PROTECTED] In asterisk in the
actual voicemail.conf file I set the From field to a valid user name like
[EMAIL PROTECTED] However for the loopback I have several names like local
host.localdomain and myhost.mydomain.com which actually is probably
unres
So what do you use instead of s,1? My s extensions set things like
response timeout, digit timeout, etc. Thanks again.
AJ
On Thu, 13 Nov 2003, Master Abi wrote:
> I had experienced this problem before. I found this to be related to 2
> items. Firstly, try not to use the s,1 starting each subm
Ok thanks. I'll try to shorten the digit timeout.
On Wed, 12 Nov 2003, David Carr wrote:
> Without looking at your extensions.conf I can only guess that maybe the
> first digit(s) of your exten aren't unique and asterisk is waiting for a
> digit timeout. You can shorten your timeout or make your
Hi again,
I'm attempting to figure out how to transfer calls from an IAX client. I
have read and seen on the list where if you put a ,t at the end of the
dial portion in the extensions.conf file that you should be able to use
the # to park and transfer calls. I have not found this to be the cas
Hi guys
I've set up a layered menu system on one of my asterisk servers where
there is a main menu and several submenus; one for each department. Each
menu plays a background intro message giving its various options. My
problem is when I'm in the main menu and press the option to go to one of
On my asterisk server I have placed valid email addresses in the
voicemail.conf file as to allow mailbox users to receive message
notification. My problem is it appears that the messages are attempting to
be sent but instead they are bouncing with a fatal error message like the
one below:
(re
Ok thanks I'll try that.
AJ
On Tue, 11 Nov 2003, Dan wrote:
> Hi,
>
> - Original Message -
> From: <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, November 11, 2003 9:09 PM
> Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
>
>
> > I unders
I understand that it can only be registered with one asterisk server. I
also understand it can only take 2 incoming calls. My question is,
within the dial plan (extensions.conf) when you put the t at the end of
the exten line, does this have to be in each context that the user has
in exten li
Dan,
My question here is, what about when you have an IAX client that
actually answers to multiple extensions to multiple contacts? Does the
t argument need to follow the exten line that the IAX client answers to
in each context? Or is just one of the context ok?
Say for instance, I have S
Ok for those of you all up in a tizzy over my subject line, please don't
take it literally because I'm certainly not saying that asterisk is the
problem here. I just got a little nightmare problem that I need a bit of
help figuring out. I installed an asterisk system a few months ago for a
client
When is the expected rollout of this?
AJ
On Wed, 5 Nov 2003, Dan wrote:
> Hi Steven,
>
> - Original Message -
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, November 05, 2003 5:36 PM
> Subject: Re: [Asterisk-Users] IAX clients and the flas
Hi guys
As usual I am playing around with IAX soft clients. I was wondering with
the various IAX clients, IAX client, DIAX, etc how's one park calls,
transfer calls, etc since there is no flash key? Is there something I
must do in the iax.conf or is it something I must do with the individual
Any thoughts or plans on making it available on the asterisk key *NIX?
AJ
On Sun, 2 Nov 2003, Senad Jordanovic wrote:
> Finaly, someone has started the IAX soft phone ball :)
>
> Thanks, Dan...
>
>
>
> ___
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> [EMAIL PRO
>From them I figured it would be proprietary but I was just thinking since
now their offering a softphone it may lead to some other interesting
possibilities.
On Mon, 27 Oct 2003, Phillip Jackson wrote:
> Yes, your correct in the fact that they offer a softphone now.
> However, with it, the
They have something on their website advertising that you can purchase a
softphone now to use with their service.
AJ
On Mon, 27 Oct 2003, Phillip Jackson wrote:
> Where'd you see that?
>
> Regards,
> Phil
>
> Quoting [EMAIL PROTECTED]:
>
> > In taking a cursory browse at Vonage's site today
In taking a cursory browse at Vonage's site today, I noticed they are now
offering a soft phone. Has anyone had any experience using this? And does
this possibly open new opportunities for using Vonage with Asterisk? Just
thinking outloud on the list, soliciting thoughts and experiences from
Thanks, got it working, Thanks to everyone!
AJ
On Sat, 25 Oct 2003, CW_ASN wrote:
> Modify /etc/asterisk/modules.conf
>
> load => cdr_mysql.so
>
> or
>
> load => cdr_addon_mysql.so
>
> Then, modify your cdr_mysql.conf, like this:
>
> [global]
> hostname=localhost
> dbname=astcdr
> password=1
Thanks a lot your earlier suggestion worked. The system was lacking
zlib-devel. Now where do I insert the lines for it to load the
cdr_mysql.so since I have it built? Can you give me an exact example of
what to put here?
AJ
On Sat, 25 Oct 2003, Eric Wieling wrote:
> gzip is not zlib. On my
Thanks a bunch
On Sat, 25 Oct 2003, Eric Wieling wrote:
> Install the zlib and zlib devel packages
>
> On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote:
> > Yes I do have the mysql and mysql-devel packages installed. With my very
> > limited knowledge of C/C++ here's what seems to be the
Yes I do have gzip installed on that box. Any other ideas?
On Sat, 25 Oct 2003, WipeOut wrote:
> [EMAIL PROTECTED] wrote:
>
> >Yes I do have the mysql and mysql-devel packages installed. With my very
> >limited knowledge of C/C++ here's what seems to be the culpret line right
> >before the e
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the culpret line right
before the error:
"/usr/bin/ld: cannot find -lz"
Any suggestions here?
AJ
On Sat, 25 Oct 2003, WipeOut wrote:
> [EMAIL PROTECTED] wrote
I just went over and grabed the asterisk-addons directory from the CVS,
changed into the directory and executed a make install and got the
following error:
"make: ***[cdr_addon_mysql.o] Error 1"
You have any suggestions about this?
AJ
___
Aster
Can anyone give me presise instructions on how to compile cdr_mysql.so?
When I initially installed asterisk on the system, I didn't have mysql
installed. Since then I have installed mysql, created the database and
table structure for cdr_mysql and placed the appropriate settings in the
cdr_my
Okay, at the CLi i did a show version and it's still showing the old
version. What I'm attempting to prevent the overwriting of my already
established config files and sound files. Any further suggestions?
On Fri,
24 Oct 2003, Leif Madsen wrote:
> [EMAIL PROTECTED] wrote:
>
> > In attemptin
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the files
completed ran make upgrade. My problem is that when I pull up the CLI the
cvs version
Mike
If you see this mail, please drop me an email off list. I've been trying
to shoot you mail regarding iaxcomm but it keeps bouncing so I must have
the wrong addy.
To the rest of the list, I apologize for taking up list space just no
other option available right now.
AJ
_
I've gotten a lot of unwanted, unsolicited mail today as well. Most
probably with the subject line "wicked screensaver". I guess the bad guys
are mining the asterisk list. Guess I'll have to play with iptables and
the mirror arguement.
AJ
On Tue, 19 Aug 2003, Steven Critchfield wrote:
> S
Hi guys,
About a month ago, I went from asterisk infancy to diving in to my first
production level install. I have a Verizon ISDN/PRI coming in to my
asterisk box, pretty standard configuration with AT&T analog phones. In
the almost 3-4 weeks of evaluating, asterisk seems to perform superbly
While you gugs are on the subject of MOH and SIP, exactly where in my
configs do I turn on MOH for my SIP clients?
Also while we're on the XLite would anyone like to help in getting my
XLite client to work. I worked with several people on the irc channel the
other night and still can't seem to
I use asterisk behind my Linux firewall with no problem. For IAX I have
the firewall forward udp port 5036 to the asterisk box. Really simple nat
setup. My asterisk box also connects to other asterisk/IAX servers
through nat from behind the firewall. I'm not sure about sip at all but I
know
I'm sorry I'm coming in on the rear end of this but is there a difference
between Voicemail and Voicemail2 in asterisk? If so what is it?
AJ
On Sun, 17 Aug 2003, Paul Cheng wrote:
> I haven't tried the patches, but they sounds very useful! My 2 cents...
>
> BTW, there have been some recent bug
Does anyone know what the Grandstream Budgetone is going for $$$ in the
US? I didn't immediately see pricing on the phones page.
AJ
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Can anyone provide me with a step by step on how to set up Windows
Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm
interested in doing various tests of my asterisk server from the Windows
perspective of the world. In the alternative if someone could provide
information on
In my recent new Asterisk installation I'm having users complain that if
they answer a call on call waiting while talking on an existing line they
are then unable to park a call without one of the two parties hanging up.
Is there anyway whatsoever to be on a call, answer a call on call
waiting
I'm a bit interested in an intercom system as well. I'm using asterisk
with analog phones. Is there any way I can do this?
AJ
On Fri, 8 Aug 2003, cwitte wrote:
> There was a thread a few months ago that tossed around some ideas for
> using a cisco phone for intercom or paging. I don't have any
Thanks anyway
On Fri, 8 Aug 2003, WipeOut . wrote:
> Hmm.. then i don't know.. If you were using the GS IP phones I may have had an
> idea.. sorry.
>
> > AT&T 957 analog sets
> > AJ
> >
> > On Fri, 8 Aug 2003, WipeOut . wrote:
> >
> > > What phones are you using?
> > >
> > > > In my recent
Try kill -9. Also make sure there's not more incidence of it running. I
don't know if this is the case with asterisk but some programs spawn
multiple processes. If your using RedHat and you've installed the startup
scripts you can execute /etc/rc.d/init.d/asterisk stop. You can replace
stop
AT&T 957 analog sets
AJ
On Fri, 8 Aug 2003, WipeOut . wrote:
> What phones are you using?
>
> > In my recent new Asterisk installation I'm having users complain that if
> > they answer a call on call waiting while talking on an existing line they
> > are then unable to park a call without one
Jeremy MacNamara an active list contributor is involved in the nufone
project. In dealing with him personally via email, I have found him very
quick and a pleasure to deal with. He seems to give people help on the
list all the time reguarding various asterisk issues. Being the owner of
a sma
So you mean a just simple blank line at the end of the musiconhold.conf
file or the extensions.conf file?
Second question, though it might seem a bit stupid, do I perhaps need a
sound card on the box that asterisk is running on? I don't think this
should be the case but I'm just wondering.
Is
Is there something I need to do to make asterisk start?
On Sat, 26 Jul 2003, WipeOut . wrote:
> Only things I can suggest is..
>
> 1. Execute it from a command line and make sure it runs.. If not you may hevr to
> compile it from source..
>
> 2. Make sure you have a new line at the end of your
No instances of it running when I look at processes.
AJ
On Sat, 26 Jul 2003, WipeOut . wrote:
> Sorry I though you had compiled from source...
>
> When * is running do "ps-aux | grep mpg123" and make sure it is actually running..
>
> Later..
>
> > Wipeout
> > I'm using the exact mpg123 bina
Wipeout
I'm using the exact mpg123 binary that you sent me. When I execute a
"whereis mpg123" it returns /usr/bin. To take it a step further I've done
"whereis mpg321" and "rpm -q mpg321" just to make sure mpg321 is not on
the system. The one thing that's confusing the heck out of me is the f
I can't seem to get musiconhold to work. I'm running asterisk on a RH9
box, I have the mpg123 package installed. In my zapata.conf file I have
the line MusicOnHold=default . In my musiconhold.conf file, in the
classes section I uncommented default and loud. In my extensions.conf
file I hav
The main number extension (_X) is in a different context than the other 4
extensions plus what do I do with the timeout arguement and other things
such as that? Just point them to _X.?
AJ
On Wed, 23 Jul 2003, Todd Lieberman wrote:
> Put the _X below the first 4 extensions.
>
>
> -Ori
I would like to know how to define the s extension when I have an incoming
PRI line? Currently I have 5 incoming DID numbers. Four of these DID
numbers I have going to specific extensions, the fifth number which is the
main number I wish to go to a background sound where callers can hear
messa
I just purchased 24 of the AT&T's that Steven Chritchfield recommended.
Got them for just over $26.00 a pop but I will keep this site handy for
future reference.
AJ
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Okay do you have any suggestions as to a reasonably priced / inexpensive
analog phone set that can be used in a business setting with asterisk?
AJ
On 18 Jul 2003, Steven Critchfield wrote:
> On Fri, 2003-07-18 at 08:36, [EMAIL PROTECTED] wrote:
> > Has anyone on the list actually purchased any
Has anyone on the list actually purchased any H3500CW from
http://lktelecom.zoovy.com? I need to purchase quite a few phones and I
was looking to get the H3500CW but it appears they are in China so I was
wondering 1.) if they are a reputable company (before I put the $ out)
2.) about how long i
Is there a way for me to set my outgoing callerid string so that all
callers outside of my pbx see our callerid string as company name main
company number but callers inside our telephone network see extension
holders name extension number? In looking at the references it looks like
I can do o
I'm interested in offering fee based support services by telephone. Does
anyone have any suggestions on how I could obtain a 900 number to do this?
My initial thought is to have the 900 number terminate to my asterisk
server which will connect callers to cus service reps / techs in various
lo
Ok, my channel bank is configured. The signaling is configured for
fxsloopstart. The framing is esf. The line is b8zs.
When I plug in a crossover cable from the T100P to the channel bank my T1
line status indicator shows a red light. However if I plug a crossover
cable straight from the Dmar
I have a premisys slimline 24 channel fxs channel bank. I'm attempting to
get it configured to work with my asterisk server. I have 2 T100P cards
in the asterisk box. One is connected to an incoming pri, the second is
connected to the channel bank.
In my /etc/zaptel.conf file I have the li
Hello all,
I have premisys 24 port fxs channel bank that is connected to a Wildcard
T100 card. I also have a pri coming in from verizon that is going to
another T100 card in my box. I've downloaded the latest CVS, I went into
the file /etc/zaptel.conf and added the following lines:
span=1
Thanks for the explanation.
AJ
On Sat, 21 Jun 2003, Steven Critchfield wrote:
> On Sat, 2003-06-21 at 07:39, [EMAIL PROTECTED] wrote:
> > Greetings all,
> > As most of you probably know from my previous questions on the list, I'm
> > still in the newbie category. My question today is pretty br
Someone recommended that I purchase the H3500CW analog handsets from
lktelcom.zoovy.com; however it appears the page or product has moved. Can
anyone tell me where I might find these?
AJ
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Greetings all,
As most of you probably know from my previous questions on the list, I'm
still in the newbie category. My question today is pretty brief, as I
told you all a few weeks ago I ordered a PRI from Verizon. I understand
that there is a "B" channel that comes with this. The question
Hello guys
Is there anyway for me to change the sounds that are presented in
VoicemailMain? For instance, instead of it saying "mailbox", I would like
it to say something like "please enter your mailbox number now". Is there
a way for me to do this?
I also noticed that when in some of the menu
Are there any self described or otherwise gnophone experts out there?
Maybe I'm a stickler for pain but I'm having a lot of gnophone problems
and conventional wisdom tells me that they should be able to be figured
out. However, after reading all the documentation, reviewing the files, I am left
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