Hi Noc,
I had similar problem. Check If you have netjetpci module and try to
delete it...this solved my problem.
Giorgio Incantalupo
Noc Phibee wrote:
Hi
i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect
my asterisk to a french analog line.
In my zaptel.conf, i have:
l
Thanks Giogio,
but no i don't have this module
bye
Giorgio Incantalupo a écrit :
Hi Noc,
I had similar problem. Check If you have netjetpci module and try to
delete it...this solved my problem.
Giorgio Incantalupo
Noc Phibee wrote:
Hi
i have buy a Digium TDM400P with 1 fxo modules T
Hi,
I'm installing Asterisk on Ubuntu 6.10
When I first compiled the zaptel package I used:
make clean
make
make install
So far so good, but the following command failed:
make b410p
I did some digging on google and found a guide on how
to install it manually, but the result was the same.
I g
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use the
Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls
can I handle simultaneously.
I want to use the cards with the following Configurations:
Intel® Xeon 3.00GHz/800MHz, 2M Processor
1GB
Noc,
it seems your zap*.conf are okhave you tried to do all the modules
loading procedure step by step?
ztcfg -s
modprobe -r wctdm
modprobe wctdm
ztcfg -vv
Giorgio Incantalupo
Noc Phibee wrote:
Thanks Giogio,
but no i don't have this module
bye
Giorgio Incantalupo a écrit :
Hi
Noc Phibee wrote:
Thanks Giogio,
but no i don't have this module
bye
Check your zapata.conf. Your signalling and channel settings are wrong
for FXO module.
signalling=fxs_ls
channel=> 4
FXO module use fxs signalling, FXS module use fxo signalling.
Leo.
___
On 23 Nov 2006, at 11:18, Zeeshan Zakaria wrote:
Hi all,
All of a sudden all my IAX DIDs have gone down. I couldn't find any
reason other than that the ISP is blocking port 4569. DIDs register
fine from my home server, but not from office server, which is not
behind any NAT. SIP register
Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ?
Those 2 seem to get average reviews from users (short range, autonomy, ...).
On paper, it seems to me a decent WiFi phone is still lacking today.
Maybe this Gigaset SL75 could fill the void.
_
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4:
No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp->channel = 4, channel = 4
Nov 24 10:32:42 ERROR
Noc Phibee wrote:
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4:
No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp->channel = 4, channel = 4
N
Noc Phibee wrote:
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4:
No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp->channel = 4, channel = 4
No
On 23 Nov 2006, at 11:36, Zeeshan Zakaria wrote:
iax2 debug is giving following messages repeatedly.
Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: REGREQ
Timestamp: 1ms SCall: 00010 DCall: 0 [xxx.xxx.
157.230:4569]
USERNAME: XXX9072835
RE
On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension
Hello;
Maybe this is a little off-topic, but I need help. I need to repair a
cisco 7970, but in my country(spain) cisco is only selling, they don't
repair if you're not client. Because I bought on ebay, I'm not client,
so I have no chance.
I tried to repair by myself, the problem is on the LCD s
Leo Ann Boon a écrit :
Noc Phibee wrote:
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel
4: No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp->
Hi Ricardo,
Could you post a specific example where your problem happens.
That way would be easier for me to try to help you on this.
Does asterisk is registred into SER , or you have trust based relationship
between them?
On 11/23/06, Ricardo Carvalho <[EMAIL PROTECTED]> wrote:
Hi,
I'm d
Pranav Peshwe a écrit :
Hi,
Check your /etc/zaptel.conf and ensure that it has the right kind of
signalling set for the same channel number as that in you zapata.conf.
do : cat /proc/zaptel/1
and it should show channels and the effective signalling settings for
them.
If signalling does not app
Now i'm receiving this and my phone no longer can dial out...
ERROR[4391]: chan_sip.c:11169 handle_request: Missing Cseq. Dropping this
SIP message, it's incomplete.
I'm having to use the configs that came with the zip because apparently my
previous configs no longer are valid and lock the phon
Hi,
Thank you for response. I configured DISA and its working sometimes and not
working sometimes. Here I am sending the configuration and output on Asterisk
server console:
Extensions.conf file content:
[custom-CLID]
exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeou
Hi all,
anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45
plug?
My telco left active the db9 port, but on my te407p card i have rj45 connection.
Anyone can help me pls ?
Thanks in advance
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Versi
Hi Noc,
why channel 5 in zapata? It must be channel 1 the same as in zaptel.conf.
Giorgio Incantalupo
Noc Phibee wrote:
Pranav Peshwe a écrit :
Hi,
Check your /etc/zaptel.conf and ensure that it has the right kind of
signalling set for the same channel number as that in you zapata.conf.
do :
Noc Phibee wrote:
Thanks for your help,
a cat:
[EMAIL PROTECTED] zap]# cat /proc/zaptel/1
Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
1 WCTDM/0/0 FXSKS
2 WCTDM/0/1
3 WCTDM/0/2
4 WCTDM/0/3
in my zaptel.conf, i have only:
loadzone=fr
defaultzone=fr
fx
Dear all,
a newbie question...
I have two external lines (PSTN & SIP) and two internal lines.
When receiving an incoming call, I correctly get the CID, but it's not
propagated to the internal lines. My analog phones shows "External call"
instead of the CID.
My analog device is a TDM400P (2 FXO +
On Thu, Nov 23, 2006 at 05:31:03PM -0300, Ard wrote:
> This is the output.
>
> [EMAIL PROTECTED] ~]# ps auxw | grep asterisk
> root 4392 0.0 0.6 50604 13968 ? Ssl 11:02 0:00 asterisk
> root 5050 0.0 0.4 38416 9268 ?S11:07 0:00 asterisk
> root 5242 0.0 0.
A summary of some of the mistakes in this thread, as it was full of
them.
On Fri, Nov 24, 2006 at 06:32:14AM +0100, Noc Phibee wrote:
> Hi
>
> i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect
> my asterisk to a french analog line.
>
> In my zaptel.conf, i have:
>loadzone=fr
Slightly off-topic:
On Fri, Nov 24, 2006 at 09:30:57AM +0100, Timothy Parez wrote:
> Hi,
>
> I'm installing Asterisk on Ubuntu 6.10
> When I first compiled the zaptel package I used:
Zaptel has nothing to do with misdn.
>
> make clean
> make
> make install
Sadly. This was unnecessary for you
Tzafrir Cohen wrote:
>
> And probably fxsks=3: I'm not sure whether or not KS is used in the
> French telco .
signalling=fxs_ks
works Ok in France with both: France Telecom and Freebox.
AF.
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On Thu, Nov 16, 2006 at 07:57:12AM +0300, Rogers Ochieng wrote:
> Am trying to make zaptel with ztdummy uncommented in FC5 but am
> getting make error. Has anyone gotten this to work?
You should post the errors :)
But there are ready-to-use packages at http://ATrpms.net/name/zaptel/,
so you don't
Tzafrir Cohen a écrit :
* Use genzaptelconf from xpp/utils/genzaptelconf to save you from this
guesswork.
Hi,
thanks ;=) with genzaptelconf, now that's works ...
correct channel are put into zaptel.conf and zapata.conf
small question if you know the TDM400P: if the fxo module are
at the
Hello all,
I was wondering whether the only way to store the
passwords for the voicemail is in extensions.conf. Is
it perhaps possible to store it (encrypted) in a DB or
store the hash of the password (as is standard in
unix) in the extensions.conf file?
Finally, it does not seem to me that the p
> >>
> >> In my zaptel.conf, i have:
> >>loadzone=fr
> >>defaultzone=fr
> >>fxols=3
loadzone=fr
defaultzone=fr
fxsls=4
> >>
> >> my zapata.conf:
> >> [trunkgroups]
> >>
> >> [channels]
> >> context=default
> >> signalling=fxo_ls
signalling=fxs_ls
> >> usecallerid=yes
> >> hidecall
Hi,
we just upgraded from 1.2.10 to 1.2.13 and now encounter strange problems with
our snom phones (FW 6.2.3 to 6.5.1). Upon phone boot everything works fine.
Phone registers and asterisk is happy. Soon afterwards the registration is lost
however. Sometimes after a few minutes the phone reregis
Hello;
Maybe this is a little off-topic, but I need help. I need to repair a
cisco 7970, but in my country(spain) cisco is only selling, they don't
repair if you're not client. Because I bought on ebay, I'm not client,
so I have no chance.
I tried to repair by myself, the problem is on the LCD s
Its a hosted server with public IP. I also noted that it doesn't get replies
from the service provide. (But my home computer does get replies with no
problem from the same service provide).
SIP registers fine. I can login using SSH. Apache server on port 80 works
ok.
Only IAX is giving trouble.
On 24 Nov 2006, at 11:34, Giordano Grandis wrote:
Hi all,
anyone known how can i create an adapter from a DB9 port of a E1 to
an RJ45 plug?
My telco left active the db9 port, but on my te407p card i have
rj45 connection.
Probably not.
Call the telco and get them to fit the correct termin
Hi Marco,
Ser has IP of Asterisk server in his "trusted" table, Asterisk isn't
registered in Ser. When Ser needs to use Asterisk, it simply rewrites
the IP destination with Asterisk's IP, and routes them to him.
For example, here's one failed attempt in transferring a call PSTN ->
VoIP -> Vo
Noc Phibee wrote:
>
> small question if you know the TDM400P: if the fxo module are
> at the slot 4, the RJ11 connector are the number 4 ?
accordingly to the documentation I have, yes, it's the connector #4.
btw, why don't you put your module to the first slot? In this case you
would avoid th
do you have created Asterisk views to SER database? Are you using sip
realtime on asterisk?
please post your extensions.conf.
By the way, I'm Portuguese:)
Qualquer coisa manda mail pode ser q consiga ajudar.
On 11/24/06, Ricardo Carvalho <[EMAIL PROTECTED]> wrote:
Hi Marco,
Ser has IP of Ast
from my aging memory.
Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim
DB9 and make standard RJ45 jack.
Also,
http://www.pccables.com/01910.htm
http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1
On 11/24/06, Giordano Grandis <[EMAIL PROTECTED]> wrote:
Hi all,
I forgot to mention that in the db9 part I guess pins 2-3(tx) and
6-8(rx) are used. I'm sorry I dont recall ground.
On 11/24/06, Erick Perez <[EMAIL PROTECTED]> wrote:
from my aging memory.
Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim
DB9 and make standard RJ45 jack.
Hey everybody,
I wanted to know what other may be doing to stem the flood of inbound
junk faxes?
We currently using Asterisk/iaxmodem/Hylafax for fax services and get a
number of junk faxes daily. Most (If not all) of them have caller-id
blocked and have a TSI of "". I was hoping that, sin
Unfortunately a lot of people don't bother to set TSI and have blocked
Caller ID on their fax line so you would get false positives if you filtered
out those faxes. I just did a HylaFAX install last week where the enduser
was extremely pleased about the fax-to-email - when a junk fax came in
(about
Hi :) We currently have a single * box with 4-port E1 card terminating
60 channels:
[PSTN]
| | 2 x E1
[Asterisk]
| | 2 x E1
[Legacy PBX]
What I'd like to have is this:
[PSTN]
| \__
||
[*1]- - - -[*2] - DUNDi peering between 2 * boxes
||
[Legacy PBX]
Whereby a
> Hey everybody,
>
> I wanted to know what other may be doing to stem the flood of inbound
> junk faxes?
>
> We currently using Asterisk/iaxmodem/Hylafax for fax services and get a
> number of junk faxes daily. Most (If not all) of them have caller-id
> blocked and have a TSI of "". I was ho
Joe Greco wrote:
You can also do a much better job of getting "caller-id" by subscribing
to an 800# service that puts ANI information in the caller-id field
We have our own 1800 lines, hm I'll have to look into this.
Doug
-- Ben Franklin quote: "Those who would give up Essential Li
for see if the card answer, what is the process ?
since your port is configured to be in the "interne" context, just add
this to this context
exten => s,1,Answer
exten => s,2,Playback(tt-monkeys)
exten => s,3,Hangup
watch the console and dial-in. if you get monkeys screaming at you, it worked !
Hi,
i receive a call on my analog line but my asterisk don't answer ;=)
do you know if they hae a solution for know if the card see the call ?
for see if it's not my cable don't work ..
thanks bye
___
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I had some troubles once with db9 and RJ45... It was due to an impedance pb.
As I remember, you have to change the impedance in the adaptor (with adding
some R-C)... I haven't done it, and my line was half-working (sometimes yes,
sometimes not...).
So, I suggest you to check if the impedance are
On Fri, Nov 24, 2006 at 02:25:38PM +0100, Anton Frolov wrote:
>
>
> Tzafrir Cohen wrote:
> >
> > And probably fxsks=3: I'm not sure whether or not KS is used in the
> > French telco .
>
> signalling=fxs_ks
Nitpicking:
'signalling=fxs_ks' is the line to add to /etc/asterisk/zapata.conf .
'fxsk
Hello,
I'm discovering asterisk, it seem to be a great soft.
I have seen a fonction to record calls that's a great fontion but there is
something disturbing me.
When the record start, except if the recorder prevent the other part, he is not
aware of the recording...
I dont find a way from the
On Fri, Nov 24, 2006 at 06:15:12AM -0800, je . wrote:
> Hello all,
>
> I was wondering whether the only way to store the
> passwords for the voicemail is in extensions.conf.
voicemail.conf ?
> Is
> it perhaps possible to store it (encrypted) in a DB or
> store the hash of the password (as i
On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows Media Player:
http://www.mlkj.net/gsm/
No luck with WMP. Anyone else have any suggestions on
playing .gs
Imran M Yousuf wrote:
Dear Users,
*I am fairly new to Digium and Asterisk. I wanted to know that if I
use the Digium product THREE Digium Wildcard TE412Ps’ (Quad E1 Card)
how many calls can I handle simultaneously.*
I want to use the cards with the following Configurations:
/Intel® Xeo
Crazy Boy wrote:
Hi,
Thank you for response. I configured DISA and its working sometimes
and not working sometimes. Here I am sending the configuration and
output on Asterisk server console:
Extensions.conf file content:
[custom-CLID]
exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten =
Doug Lytle wrote:
Hey everybody,
I wanted to know what other may be doing to stem the flood of inbound
junk faxes?
We currently using Asterisk/iaxmodem/Hylafax for fax services and get
a number of junk faxes daily. Most (If not all) of them have
caller-id blocked and have a TSI of "". I w
[EMAIL PROTECTED] wrote:
Hello,
I'm discovering asterisk, it seem to be a great soft.
I have seen a fonction to record calls that's a great fontion but there is
something disturbing me.
When the record start, except if the recorder prevent the other part, he is not
aware of the recording...
I
Hello!
I'm tryuing to bring up an R2 connection but eventhough I've followed
the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems
to be missing.
When an incomming call is generated I get:
Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
M
Doug Lytle wrote:
Joe Greco wrote:
You can also do a much better job of getting "caller-id" by subscribing
to an 800# service that puts ANI information in the caller-id field
We have our own 1800 lines, hm I'll have to look into this.
Doug
-- Ben Franklin quote: "Those who would g
How can i programming a Cisco 7961 to be used as busy lamp field?
my configs :
sccp.conf :
[devices]
type= 7961
tzoffset= 0
autologin = 601
speeddial = *31, Hanna --> other SIP telefon
extensions.conf :
exten => *31,hint,SIP/hanna
exten => *34,hint,SCCP/601
on SIP Telefon
Hi all,
I have a surplus Sangoma 10 port FXO card for sale. This model could be
upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by
changing the grand-daughter cards (each card supports 2 lines). You
could also "downgrade" the card by removing any or all of the daughter
cards.
I'
Please don't cross post FS items to *-users - that's what *-biz is for.
CP
On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote:
Hi all,
I have a surplus Sangoma 10 port FXO card for sale. This model could be
upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by
changing the grand
Steve Totaro wrote:
Try dialing from a phone that lets you block callerid *67 + number
with pri debug span x turned on. You will see all of the info that is
being passed in the IEs. You may already have ANI enabled and not
know it.
I'll give that a try when I have access to that feature.
A
Steve Totaro wrote:
Is this an 800 number? If so
No, they do have 1 800 number though, but it goes to the main inbound
number of the facility.
BTW OT from another thread, I upgraded IAXmodem and used the example
Hylafax modem config files instead of using Hylafax's addmodem and
things see
Why not. Users by stuff too!
On Fri, 2006-11-24 at 11:40 -0800, Anthony Rodgers wrote:
> Please don't cross post FS items to *-users - that's what *-biz is for.
>
> CP
>
> On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote:
>
> > Hi all,
> >
> > I have a surplus Sangoma 10 port FXO card for sale
At 22:07 22/11/2006 -0700, Marco Mouta wrote:
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so
you can override whatever you wanted:)
Thanks everyone. Indeed, it seems like using the embedded Berke
Am Freitag, den 24.11.2006, 22:22 +0100 schrieb Vincent Delporte:
> At 22:07 22/11/2006 -0700, Marco Mouta wrote:
> >You can do it using AstDB, just load the database with callerid names and
> >numbers and then include a lookup on database in all incoming calls, so
> >you can override whatever yo
At 03:14 PM 11/22/2006, you wrote:
The missing piece of the puzzle: I'm extension 203. I want any call I park
to get parked at extension 2203. I want a call my boss parks to park at
2205, since he's ext. 205. In other words, I want calls parked FROM
extension XYZ to be parked AT extension (XYZ+20
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
type=friend
secret=password
qualify=yes
port=4569
notra
Correction: those 'secret' lines are of course all supposed to say '777'. :)
On 11/24/06, Lachek Butalek <[EMAIL PROTECTED]> wrote:
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
p
Two apologies to be made:
#1: Sorry for all the spamming.
#2: Sorry for all the top-posting. Gmail doesn't lend itself well to
email lists since it neatly tucks all the quoted text away
(out-of-sight, out-of-mind for the poster), leaving a mess for those
whose email clients do not. Again, my apol
res |= ast_register_application(app4, vmauthenticate,
synopsis_vmauthenticate, descrip_vmauthenticate);
You need to look more closely at the code. This snippet registers the
dial plan application VMAuthenticate so vmauthenticate is called
wherever you use that function in your dial plan.
static
Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll
need to connect it?
neil
safeharbour IT Ltd
Your IT Department
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On Fri, Nov 24, 2006 at 11:51:41PM -, Neil Tancock wrote:
> Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll
> need to connect it?
It will work with UK ISDN, but ensure it's EuroISDN and NOT UK ISDN
(it's set in the telco switch and can generally be changed).
UK IS
Hi ,
I have a problem with a X100, i do a external call to the asterisk
server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
I'll apreciate all your suggestion. Greetings, txus.
The asterisk outpu
Can u show SER & asterisk configuration or logs?
--
Regards
--
M Emran
Managing Director
InSpiration Software Ltd.
E-mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Web: www.inspiresoftbd.com
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Jesus Jimenez wrote:
Hi ,
I have a problem with a X100, i do a external call to the
asterisk server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
I'll apreciate all your suggestion. Greetings,
Steven wrote:
There are two I can think of.
Hoodahek and asterdex (or asteridex)
We used hoodahek at first, but now use asterdex(sp?)
It has a web interface to enter the new names into.
We use it to fixup, corp. cell phones and used to use it for our leagcy PBX
extensions.
I use some cust
I would like to access my shell environment variable MYIP from within
sip.conf to put in externip.
I've tried some variations of syntax after reading "The Future of
Telephony" but it's not working yet.
Should it be
externip=${ENV{$MYIP}}
or some other syntax??
Larry
--
Larry Alkoff N2LA - A
You need the valet parking module. It's a third pary add-on app.
This is a direct link to the current code
http://www.pbxfreeware.org/app_valetparking.c
Here is a link to the original README
http://www.loligo.com/asterisk/misc/apps/app_valetparking.README
On 11/22/06, Steve Sobol <[EMAIL PROTECT
HI,
thanks for your reply. Here is my ser.cfg and other config files please
guide me.
ser.cfg
--
debug=5
fork=no
log_stderror=yes
listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:[EMA
Something like this should work in your iptables:
iptables -A PREROUTING -t nat -p tcp --dport 1234 -i eth0 -j DNAT
--to-destination 127.0.0.1:4569
iptables -I FORWARD 1 -d 127.0.0.1 -p tcp --dport 4569 -j ACCEPT
This would forward port 1234 to port 4569.
bp
On 11/23/06, Zeeshan Zakaria <[EMAI
Not quite what I'm looking for. I ant to hang up all channels (zap or sip)
in meetme room 5
On 11/23/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:
On 19:18, Thu 23 Nov 06, Eric Bishop wrote:
> Other than rebooting the server or restarting Asterisk from cron does
anyone
> know how to kill a me
You could write an extension which executes meetme kick, for all the
channels, but I am not sure how to execute such a thing at a given
time.
on Saturday 11/25/2006 Eric Bishop([EMAIL PROTECTED]) wrote
> Not quite what I'm looking for. I ant to hang up all channels (zap or sip)
> in meetme room
Hello all.
I have two identically configured asterisk servers each with a TDM2422P
(with two S400M FXS modules and two X400M FXO modules).
*1 works perfectly and the sound quality is great. However, I am having
audio quality problem with *2 when making or receiving calls over the PSTN -
the soun
Hi,
I must say that i'm not very used with customization of FOP. I've a box
runing Flash Op.Panel, and i notice that the screen is full of buttons from
my sip users, as well as Zapata channels.
The problem is that i have more Zapata channels as well as SIP users, is
there any way to get a scroll
On Wed, Nov 22, 2006 at 04:51:26PM -0800, Ira wrote:
> At 03:14 PM 11/22/2006, you wrote:
> >The missing piece of the puzzle: I'm extension 203. I want any call I park
> >to get parked at extension 2203. I want a call my boss parks to park at
> >2205, since he's ext. 205. In other words, I want cal
Upgraded the client this morning, lets hope this works for good :)
Can you tell us what the best way to suggest bug features or additions is?
For example, Arizona is not a option under time zones, we dont have DST and
never change time, would be nice if that was added! Also, a digital clock,
alo
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