Hi Michel,
Michel Brabants wrote:
> I didn't see a bri-adapter on the digium-site, only pri it seems. Any
> recommendation on that or is there a bri-adapter from digium. I'm also
> open to other vendors. I saw that there are others which ar ecompatible
> with asterisk, but I don't have a lot of t
Hi,
I've tried your suggestion but the result is still the same...
Have another suggestion?
Best regards,
Stevanus
Wilson Pickett wrote:
Is it possible to bypass incoming ring on asterisk so that incoming
calls come to asterisk box will be directed straight into did?
Try
Hi List,
I have a test asterisk box with a TDM400P with 4 FXO modules plugged in.
Yesterday I could use the box without any issues - no problems.
This morning, the sound on the box was absolutely horrible. After some
fiddling about, I have rebooted the box, and now asterisk refuses to start!
Bugger, thanks for replying and telling me, might send a request through
to Grandstream and see when they intend on releasing it.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Thursday, 9 June 2005 3:54 PM
To: Asterisk Users
Hi,
We’re looking for an experienced Asterisk engineer/programmer to
configure and install Asterisk systems.
This is a full time position and the person will be based in Asia. Share options are available and we are open to
negotiate.
Minimum of 1 year experience installing and
Hi,
try xlite if you have enough bandwitdh for G711 codec requirement..
try firefly if you want to use G729 codec freely (linked via dll)..
both of them are the best freeware softphone for windows.
Best regards,
Stevanus
infra struct wrote:
I have been searching for the necessary compon
> Is it possible to bypass incoming ring on asterisk so that incoming
> calls come to asterisk box will be directed straight into did?
Try setting callerid=no on the FXO channel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists
> so i was looking at the internet and i read a lot, the cheapest are the
> Grandstream BudgetTone
> but some reviews of this list says they are not so good ... so i found
Many people hate these phones, yet I've found my 3 BT100 to be
excellent for a network of friends and associates (not everyday
I have been searching for the necessary components for my setup from sometime back;
yet to install Asterisk and will be installing softphones on Linux Server and on all windows PCs(most of them are Windows Xp,others are Windows 2000 professional,Windows 98); but could not decide which softphone
> what we needed and I want to ask if I understand the naming correctly:
> FXS = pstn-signals for calling someone (towards central pbx/server) and
> knowing that someone is calling you
> FXO = ...?
FXS has a phone plugged in it
FXO hgas a phone line plugged in it
http://www.onlamp.com/pub/a/onlamp/
On Thu, 9 Jun 2005, James Bean wrote:
> Has anyone got the hint function working, and maybe with the GXP2000.
I don't think the current firmware release for the GXP-2000 supports
SUBSCRIBE/NOTIFY. That functionality is to be released at a later date.
Peter
Hi all,
Does Asterisk support multi thread? I mean:
Is it possible to do one of the 2 following scenarios:
1. Play a low background music when the user record his/her voice
2. If the first scenario is not possible, can we play two music stream at
the same time? i.e: using MP3Player to play a musi
I had that with a TDM400P if I had it set for incoming caller id in
zapata.conf
It was allowing time to detect CID.
We weren't interested in CID, so I just turned it off. Now incoming
calls ring the phones in about 1 second rather than a few seconds.
Might be what you are getting?
On Thu, 2005
Asterisk 1.0.7
Has anyone got the hint function working, and maybe with the GXP2000.
I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment
trying to get the LED's to light up.
On ext 690, button 1 is setup for ext 691, I did this using both methods
691, and
On ext 691, button 1 is
Hi,
After days tinkering with this digium card (TDM04B), I notice that this
card has a slow response in detecting ring signal from pstn and hanging
up when the call is over.
The delay can consume up to several seconds...
Is this normal?
Best regards,
Stevanus
__
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
G
Sent: Thursday, 9 June 2005 1:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI
Thanks for your answer. Googling in the lists I found what you
Robert Goodyear wrote:
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote:
So, anyone else have any ideas? I tried the below suggestion and
it's still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make install
On 5/2/05, Chris Stins
On Thu, 2005-06-09 at 00:51 -0300, Alejandro G wrote:
> Thanks for your answer. Googling in the lists I found what you are telling
> that maybe there is a synchro problem with the E1, but I'm not so sure that
> this could be. I am configuring zaptel.conf like this:
>
> span=1,0,0,ccs,hdb3
> bchan=
Thanks for your answer. Googling in the lists I found what you are telling
that maybe there is a synchro problem with the E1, but I'm not so sure that
this could be. I am configuring zaptel.conf like this:
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
But I also changed to test to:
span=1,1,0,c
Tried that. Didn't work.
Robert Goodyear wrote:
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote:
So, anyone else have any ideas? I tried the below suggestion and it's
still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make ins
Would an admin please contact me off list? I tried to subscribe from
another address and it failed--I got no email to confirm the
subscription. I would rather use the other address and need to know if
there is a problem with my mail server.
Thank you,
David Koski
[EMAIL PROTECTED]
[EMAIL PROTE
On Jun 8, 2005, at 7:19 PM, Shidan wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shidan
Sent: Thursday, June 09, 2005 11:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Ringing a few phones
I have a client requirement th
On Mon, 6 Jun 2005 11:17:20 -0600, Colin Anderson wrote:
>
>http://www.pcmag.com/article2/0,1759,1812887,00.asp
>
>Specifically, his assertion that ISP's would sniff traffic and block, say,
>the SIP port. You could play wack-a-mole with port numbers, no?
>
>Also a community based, Freenet style o
Finally I've got the ivr to work..
The workaround I've found so far is record and edit the voice file
through Adobe Audition or cool edit or sound recorder, etc and then
convert it to gsm using sox..
Hope that might help someone ;)
Best regards,
Stevanus
stevanus wrote:
Hi,
Recently, I'
On Jun 8, 2005, at 6:15 PM, Chris Mason (Lists) wrote:
I have had good success with my efforts to send faxes over voip using
ulaw,
surprisingly, and I want to move it from testing to reality. I have an
account with Teliax, who have been very good. For voice I use g729 and
ulaw,
but for faxing
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote:
So, anyone else have any ideas? I tried the below suggestion and it's
still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make install
On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> w
Hi Jen thanks for the info but I already knew that, what I want is for
it to not get picked up by voicemail on one of the channels. dialing
them in sequence is not an option either, and as I mentioned changing
the settings on the actual phones isn't an option either. I remember
there was an option
Hi,
I've tried your suggestion but it still yield no result :(
Personally, I think the problem lies in zaptel driver, because when I
see the interactive asterisk log (as I set asterisk to run in verbose
mode) , any call to the asterisk box will result in one ring then the
digium TDM04B start
I have had good success with my efforts to send faxes over voip using ulaw,
surprisingly, and I want to move it from testing to reality. I have an
account with Teliax, who have been very good. For voice I use g729 and ulaw,
but for faxing I can only allow ulaw. However, Teliax only sets the codec
p
If you want to dial a number of phones at the same time do "exten =>
5000,1,Dial(SIP/5000&SIP/5001&SIP?5002). The & value is what does the job.
Kind regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shidan
Sent: Thursday, June 09, 2005 11:01 A
So, anyone else have any ideas? I tried the below suggestion and it's
still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make install
On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> wrote:
Still only doing 20 voicemails. Thanks for
I haven't gone all the way back to the original poster, but I noticed
mention of a TDM400 in a couple of places.
If you are not in North America, you need to pass an option to the wctdm
driver when it loads to set it in the right mode. Default is FCC mode.
This leaves the card with an impedance m
I have a client requirement that multiple phones can be dialed,
however they don't want the pstn phone to pick up automatically
because of voicemail etc, nothing can be changed on the phones, how
can I handle this requirement, by the way no zap channels are
involved, all the pstn phones are behing
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, June 08, 2005 11:27 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Echo problem
>
>
> On Wednesday 08 June 2005 13:37, Martin Roy wrote:
> > rxgain= I tried from -8.0 to 10
On Tue, 2005-06-07 at 23:28 -0300, Joshua Colp wrote:
> A network booboo occurred and and just like it warns (note the word
> WARNING), it received a mini frame before the first full voice frame...
> Nothing too serious, audio might sound odd for less then a second but it
> should recover.
Actuall
Hi,
I'm having a problem with one of our 7960. They all run latest 7.4
SIP firmware.
The problem appears on the other end. The other end constantly hears
a 'crackling' noise. I have tested using phone set, headset and
speaker and the noise appears on all cases. I have other 7960 setup
exactly
I was having problems and your tip helped, my handset showed a polarity
reversal... Now we'll see how well it works...
Thanks,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, June 08, 2005 2:27 PM
To: asterisk-user
Title: Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the incoming call to work all the time, the outgoing does not seem to have a problem.
I have worked with their support but since they say that we are getting the initial call to
On Wed, 2005-06-08 at 14:58 -0600, Rich Adamson wrote:
> > I've just had polarity reversal provisioned by our telco to test hangup
> > detect with a TDM400P
> >
> > I've set hanguponpolarityswitch=yes in zapata.conf
> >
> > When I start Asterisk I get "ignoring hanguponpolarityswitch"
> > in /var
> 2. I have started to use realtime and have the hard disk with the mysql
> data.
> How can I use them now on the new machine.
>
> In fact, I would like to use it on my database server directly, that
> would help me to add two Asterisk boxes to one database.
You can just copy the MySQL files to th
On Wed, 2005-06-08 at 12:49 -0400, Andrew Kohlsmith wrote:
> On Wednesday 08 June 2005 12:00, Neil and Fiona wrote:
> > /var/log/messages seems to be indicating that the wctdm driver thinks
> > that the polarity of the line is reversed on start. (ie incorrect
> > polarity)
> >
> > Polarity reversed
Hi,
I have just signed up with Voxee.com and have attached my Asterisk
server to dial them via IAX2.
Below is the start of the log which dials the number and promply
hangs up when the call is answered, with the logs saying that the
channel is not compatiable.
I have traced this down to the g
Angus
a "BT" socket with a capacitor in is commonly refered to as a "Master
socket", and are very cheap even without wholesale. It gets its name
from being the socket that BT installed into the house for the line,
all other sockets in the house will be slave or secondary (ie no
capacitor) (and it
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jorge Alayon
> Sent: Wednesday, June 08, 2005 2:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Multiple E1s on one box
>
>
> Answering both ques
Anyone know if
- it is possible to limit 1 agent per extension where
the last agent to log in overrides any previous agents
or
- a Command/application to clear all agents logged in
on extension
Does this look like it would require a custom mod to
do it?
J
[EMAIL PROTECTED] wrote:
I would recommend rebuilding the zaptel modules
rebuild_zaptel
then run the genzaptelconf -s
Have you used the yum update command to update your installation ?
If not do that first then run the rebuild_zaptel command.
this is my output
STOPPING ASTERISK
STOPPING FOP S
Your ringtone seem to have gone bad. You have to upload a new ringtone
file to correct your phone problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Wednesday, June 08, 2005 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial
This Phone is same as Netweb-X100 made by Yuxin. These phones are
reliable. It has PA168 Chipset.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Brown
Sent: Wednesday, June 08, 2005 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
Hi Angus,
If you connect the phone directly to the outside line will it ring ?
The ring from the C.O. provides a >90volt AC (30cps) and is capable of
ringing a standard phone ( a real two tone gong bell) My guess is that
the TDM400 card does not supply enough current to actually do this. Most
Answering both questions:
1) I am connecting to a Meridian usin a SIP E1 Gateway in R2. I just bought
the card and one of my test will be direct R2 connection. Have not tried
yet.
2) I was told you can do 12 E1 as long as it is G.711, but nobody is telling
me how many E1s per box doing G.729. I ha
On 09:05, Tue 07 Jun 05, Florian Overkamp wrote:
> Hi Michiel,
>
> > -Original Message-
> > I been searching on the wiki and google for ENUM in NL.
> > All I could find were some docs from the Dutch Financial
> > Department about taskforces and plans. But it all links to
> > dead pages an
I'm trying to gauge the amount of overhead for idle users (NOT in the middle of a phone call) per user, per server. These are a combination of SIP and IAX2 clients, with "qualify=yes".
On, for example, a dual 2.4 Ghz Pentium server (with plenty of RAM), how many hundreds, or thousands (rough
I would recommend rebuilding the zaptel modules
rebuild_zaptel
then run the genzaptelconf -s
Have you used the yum update command to update your installation ?
If not do that first then run the rebuild_zaptel command.
this is my output
STOPPING ASTERISK
STOPPING FOP SERVER
Unloading zaptel hardw
Hi,
I have used the Budgetone 102's extensively on Asterisk and found then
quite reliable as long as you update the firmware.
The GXP2000 is quite a mess at the moment as the current firmware does
not support 3 quarters of the advertised functions and codec support is
extremely limited. I have test
On 6/8/05, Jorge Ortega Perez <[EMAIL PROTECTED]> wrote:
> Good day, im gonna buy 2 IP phones to test Asterisk, i don't wanna spend too
> much $$$ on then,
> so i was looking at the internet and i read a lot, the cheapest are the
> Grandstream BudgetTone
> but some reviews of this list says they ar
Roman Zhovtulya wrote:
Dear all,
I've noticed some significant voice quality deterioration when calling US
landline via VoIPjet.com in the last week or so.
Before that the quality was pretty good.
Has anyone else experienced any voice quality problems with voipjet
recently?
I've been using VOIP
I had problems with their West coast server so I switched to their East
coast server and better success.
-Scott
- Original Message -
From: "Roman Zhovtulya" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, June 08, 2005 2:08 PM
Subject: [Asterisk-Users] Anyone noticed Voipjet voice quality probl
Dear all,
I've noticed some significant voice quality deterioration when calling US
landline via VoIPjet.com in the last week or so.
Before that the quality was pretty good.
Has anyone else experienced any voice quality problems with voipjet
recently?
Thanks,
Roman
Franco: Why is not possible to handle 8 E1?? then is not possible to
use 3 PCI cards with 4XE1 ports, hence having 12 E1? i have never
installed an E1, but i tought it was possible when i saw this:
http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE405P.
thanks
On 6/8
Seems I figured this out. My rxgain and txgain settings were at 0.
I adjusted them properly using ztmonitor and now it seems to be working.
On Jun 8, 2005, at 1:25 PM, Jay Austad wrote:
I have asterisk 1.0.7 and I made the required patch and got
everything installed. I have libtiff 3.7.0
> I use Digium TDM400 cards as well. Asterisk's software echo cancellation
> sucks. From what I've heard on the IRC channel, you'll never completely
> eliminate echo with it. And unfortunately, hardware echo cancellation starts
> out at a full T1. They don't seem to have any solution for someone wi
Good day, im gonna buy 2 IP phones to test Asterisk, i don't wanna spend too
much $$$ on then,
so i was looking at the internet and i read a lot, the cheapest are the
Grandstream BudgetTone
but some reviews of this list says they are not so good ... so i found
iareaphones but i can't find review
We use it like this if that is what you are looking for:
exten => s,4,GotoIfTime(8:30-17:00|mon-fri|*|*?open,s,1)
-Greg
On Wed, 2005-06-08 at 11:24 -0400, Henry Coleman wrote:
> This feature is called "attendant - night answer position". Is it not
> possible to switch the incoming call to an a
[EMAIL PROTECTED] wrote:
Dean,
Here are the results of the genzaptelconf -s -d. As you can see, it is
throwing some errors, but I am a bit of a newbie so any help you could provide
would be greatly appreciated!
[EMAIL PROTECTED] /]# genzaptelconf -s -d
STOPPING ASTERISK
Asterisk ended wi
I'm sorry all, lines means config lines of code.
Michael D Schelin wrote:
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is one
> I've just had polarity reversal provisioned by our telco to test hangup
> detect with a TDM400P
>
> I've set hanguponpolarityswitch=yes in zapata.conf
>
> When I start Asterisk I get "ignoring hanguponpolarityswitch"
> in /var/log/asterisk/messages
>
> I assume that the option is either not va
Heya,
I'm going to work on a voip project and we're going to setup an
asterisk-server. We want to test a little bit with a connection to a
pstn-line and a connection to a bri-line. I looked around a bit to see
what we needed and I want to ask if I understand the naming correctly:
FXS = pstn-signa
I get the identical message when I run genzaptelconf -s -d.
I'm running AAH 1.1 w/ X100P clone card.
At 02:14 PM 6/8/2005, you wrote:
Dean,
Here are the results of the genzaptelconf -s -d. As you can see, it is
throwing some errors, but I am a bit of a newbie so any help you could
provide
wo
On Wed, Jun 08, 2005 at 02:15:06PM +0100, Paul Redstone wrote:
> Hi
>
> In the end we found it easy to record our own using this section in
> extensions.conf. This also meant that we could add our own company specific
> ones in the same voice (not shown here). Basically you get someone to dial
Hello
I have played about with a TDM400 card and plugged
in some standard analog phones. I am using the card in FXS mode - for
analog extensions. I did notice that one of my phones did not ring and I
wondered why. I later read in Paul Mahler's book VoIP Telephony with
Asterisk that in h
Dean,
Here are the results of the genzaptelconf -s -d. As you can see, it is
throwing some errors, but I am a bit of a newbie so any help you could provide
would be greatly appreciated!
[EMAIL PROTECTED] /]# genzaptelconf -s -d
STOPPING ASTERISK
Asterisk ended with exit status 0
Asterisk shu
Jorge,
As far as I've read, you won't be able to handle 8 E1 in one box.
By the way, have you had success with interconnecting E1 R2 argentina? I´m
having trouble with a Meridian... I can only make calls from asterisk, but
the other way arround...
Tks
Franco
- Original Message -
From: "
I have seen the same problem. The zaptel hardware looks fine in zttool and
appears to be ok when genzaptel -s -d is run, but when you look at the zap
channels in CLI, you only see the pseudo channel.
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Wednesday, June 08, 2005 11
I use Digium TDM400 cards as well. Asterisk's software echo cancellation
sucks. From what I've heard on the IRC channel, you'll never completely
eliminate echo with it. And unfortunately, hardware echo cancellation starts
out at a full T1. They don't seem to have any solution for someone with 4
pot
did genzaptelconf -s -d say it found any cards?
--- [EMAIL PROTECTED] wrote:
> Dean,
>
> Actually, I have run genzaptelconf -s -d but it
> still didnt seem to modify any
> of the config files that I look at in the AMP
> console. Should I try modifying
> the config files manually?
> Thanks,
On 6/7/05, Johann <[EMAIL PROTECTED]> wrote:
> Hugo,
>
> > 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25"
> > <(716)250-3405>
> 2nd column is not really sure...maybe the duration?
Asterisk UniqueID of the call.
-Brian
___
Asterisk-U
I have asterisk 1.0.7 and I made the required patch and got everything
installed. I have libtiff 3.7.0, and I'm using the zaptel stuff.
When I send a fax to it, it autodetects the fax and starts rxfax, however, the
fax machine just sits at 1% and then disconnects. I don't have any error
messa
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
> rxgain= I tried from -8.0 to 10.0
> txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the results you're
only stabbing in the dark playing with these things.
> by the way I live in Canada and the prov
Hello list.
I'm going te explain my trouble.
I have my asterisk with a TDM400P with 4 FXS channels. Two ports are
connected to a Panasonic PBX (it's working fine), and others two ports
are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected
to a CO port (where i had a pstn line).
W
Is there any metric on the number of AGI's that can run
at the
same time. Shouldnt be a limit in my mind but I am thinking in
terms of system performance.
My AGI is a C program with 3 meg executable size.
Thanks,
Jerry
___
Asterisk-Users maili
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is one of the things also affected. Now I'm using TE110 card in my
system. I hope t
Yes you can. There are some examples @ cisco look for TDM switching.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcelo
Pacheco
Sent: Wednesday, June 08, 2005 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
The configuration in the blog does not depend on the product, it
depend on the IOS used. Should work for your 5300, the only problem you
could have, AFAIR is with the SIP-ua config. Authentication, starts after
12.2.something.
If you have problem come back and I give u a workaround
Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.
With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the
following :
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
> On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
> > Joseph ha scritto:
> >
> > When sending a call to a line defined on chan_sccp, there is an
> > error on the console that says:
> >
> > Jun 7 08:22:29 WARNING[3924]: sccp
I've used that feature in asterisk HEAD, and it has worked for me (i
needed to apply a little patch for it to work for incoming calls
also), but i also used answeronpolarityswitch=yes. Maybe it's a logic
bug in the code. Try with that option and tell us the results ;)
BTW, it doesn't matter is the
Thanks Johann. - that helps out .
Johann wrote:
Hugo,
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25"
<(716)250-3405>
1st column is unixtime stamp for the current date
2nd column is not really sure...maybe the duration?
3rd column is the queue name
4th column is their age
On Wednesday 08 June 2005 12:00, Neil and Fiona wrote:
> /var/log/messages seems to be indicating that the wctdm driver thinks
> that the polarity of the line is reversed on start. (ie incorrect
> polarity)
>
> Polarity reversed (0 -> 1)
Reverse the tip and ring on the line then. :-)
> I'll chec
Tim wrote:
I'm trying to setup remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
Anyone have example of what I need to change to make an asterisk server
log on a remote mysql server?
If you are going to store CDRs on MySQL, why not skip O
Will do ..Thanks Henry
Andrew Kohlsmith wrote:
On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
This feature is called "attendant - night answer position". Is it not
possible to switch the incoming call to an alternate extension based on
time of day ?
You need to read up. This
I'm trying to setup
remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
Anyone have example
of what I need to change to make an asterisk server log on a remote mysql
server?
___
Asterisk-Use
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have Caller
Everyone using CVS head, and owning flex-2.5.31 (or higher)--
Please note that a new version of the expression ( $[ ] constructs used in
extensions.conf ) parser
is automatically built by the makefile if your flex is at 2.5.31 or higher. You
can see what your
flex version is by saying flex -V
On Wed, 2005-06-08 at 11:34 -0400, Andrew Kohlsmith wrote:
> On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
> > I've set hanguponpolarityswitch=yes in zapata.conf
>
> Do you also have the signaling on the channel set to kewlstart? I don't
> believe polarity detection does anything withou
Yeh, this is called line "hunting" all telco's offer this... you get
one published number but say 12 lines each line actually has a
number but just calling the main number will automatically roll-over to
the first available line in that hunting group. By the way, outgoing
calls that use t
On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
> This feature is called "attendant - night answer position". Is it not
> possible to switch the incoming call to an alternate extension based on
> time of day ?
You need to read up. This exact situation is given in the Asterisk Handbook.
ht
On Wednesday 08 June 2005 11:19, Alejandro G wrote:
> When I call to the TDM400 cards from the PAP2 eveything is OK, sound
> quality is perfect.
> When I call to terminate the call in PSTN through E100P I hear clicks which
> aparently are RTP packet looses. This clicks are only heard in the PSTN
>
Dean,
Actually, I have run genzaptelconf -s -d but it still didnt seem to modify any
of the config files that I look at in the AMP console. Should I try modifying
the config files manually?
Thanks,
Marc
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECT
rxfax doesnt work with voip, you need something like NVFaxDetect from
Newman Telecom to detect the incoming fax.
Essentially you sent him an email and he'll send you the code. Once you
compile them into asterisk you can add it.
http://www.voip-info.org/tiki-index.php?page=NVFaxDetect
JD
Antonio
On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
> I've set hanguponpolarityswitch=yes in zapata.conf
Do you also have the signaling on the channel set to kewlstart? I don't
believe polarity detection does anything without this signaling type.
> When I start Asterisk I get "ignoring hangu
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