On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote:
On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote:
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote:
Interesting is, that I receive an INFO_IND *before* the CONNECT_IND.
This looks like an interesting
Hi, all,
we are building a forwarding station in Japan where we
would be receiving and forwarding over 3000 SIP calls at
the same time.
The calls will be offered to us via a carrier as SIP and
we will forward the call via the same carrier as SIP.
The callflow would look like this:
1. SIP call
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote:
On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote:
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote:
Interesting is, that I receive an INFO_IND *before* the
Ok 1 for Debian, any Fedoras Core 3 out there?
fc3, and it doesn't work.
If you check the archives, this has all been discussed before. The
issue seems to be more oriented to the specific pci bus implementation
on the motherboard. You might also want to run /usr/src/zaptel/zttest
and read the
On Tue, 28 Feb 2006, Paolo Prandini wrote:
I am trying to use chan_capi with an Eicon Diva Server BRI.
I installed the Eicon drivers from source including CAPU and
I can use the board correcly using tty_test and minicom over /dev/ttyds01
or /dev/ttyds01.
I need to insmod capi ( why? it is not
A thread on running 5000 simultaneous cllas ran on this list recently and it
did generate a lot of heat. You might want to look it up the archives - but
make sure you read as many posts on it as possible because lots of different
opinions formulated over time.
-Original Message-
From:
Hello Paolo,
I put together this page which has instructions on getting Asterisk
working with a Diva Server card. Follow the steps for Option 0...
http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On Tue, Feb 28, 2006 at 09:20:05AM +0100, Armin Schindler wrote:
OK, I've reported a bug to mISDN. With the patch from the Karsten Keil
in the mantis tracker:
issue: https://www.isdn4linux.de/mantis/view.php?id=40
I'm now gettig connect_ind and info_ind in the correct order (asterisk
Calling any unreachable mobile from Fastweb or Telecom PSTN I don't get
any courtesy message. The same happens when calling an inexistent
number.
I'm configuring two PBX's, connected to two different phone lines, both
behave this way.
Perhaps there's some missing zapata parameter?
Regards,
_fangi_
I am having problems with a Zoom 5801 and *.
It does not appear possible to route voip calls out the FXO, all voip
calls get routed to the FXS no matter what.
There are tons of menus in the webconfig but about 1/3 of them have no
help page, and there is no documentation from Zoom on this
I was wrong.
The problem was with chan_sccp library and was solved downgrading from version
20060207 to 20060204.
_fangi_
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi
Inviato: venerdì 24 febbraio 2006 10.00
A: Asterisk Users Mailing
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
To pick-up with SIP phone, it has to be defined in sip.conf. Same goes for zap
and iax2.
--
Tomislav
Hi,
I have installed an TE110P but forgot to change the jumper
settings to E1. I don't have easy physical access to ther server
at the moment so I wonder if it will be possible to run it without changing
the jumper settings with a configuration like below or will it be
impossible
to use the card
Hi All,
I installed Asterisk recently and it was working from 2 weeks without a problem
until today. Today it started showing strange error
Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received
from 'sip:[EMAIL PROTECTED]'
Whatever number I call it displays this,
Situation. I call out from SIP phone over h323 trunk and called person decides
not to pick up (on mobile phone they press red button - NO - hang-up). Until
the called person press the NO button, I can hear ringing. When called person
press the button, I don't hear anything. Asterisk waits until
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 09:20:05AM +0100, Armin Schindler wrote:
OK, I've reported a bug to mISDN. With the patch from the Karsten Keil
in the mantis tracker:
issue: https://www.isdn4linux.de/mantis/view.php?id=40
I'm now gettig
a better way is to to load the driver with all spans set to E1 by running
modprobe wcte11xp t1e1override=15
or edit wcte11xp.c and change 'static int t1e1override = -1;' to 'static int
t1e1override = 15;'
Diyanat
From: Robert Andersson [EMAIL PROTECTED]
Reply-To: Asterisk Users
Hi,
I'm having problems getting our server
to work with our BT ISDN30 box. We are using a Digium TE110P card
to connect to the ISDN box on the wall. The card is configured as
an E1 (strap on). I've made the T1 crossover cable ( well, made two
variations ) and neither work. The light on the
Thanks. Might have saved me a lot of trouble...
best regards
Robert
Diyanat Ali wrote:
a better way is to to load the driver with all spans set to E1 by
running
modprobe wcte11xp t1e1override=15
or edit wcte11xp.c and change 'static int t1e1override = -1;' to
'static int t1e1override =
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
And how do you turn on Asterisk's debug facilities?
Edit logger.conf and uncomment full.
Start Asterisk with the the -d option.
View debugging information in the /var/log/asterisk/full
Doug
--
Ben
Hello,
AFAIK the feature CD (call deflection) is only possible on
point-to-multipoint links, is this correct?
I've heard about the feature partial rerouting which should do the
same on point-to-point-links. Is this implemented in either bristuff or
chan-capi(-cm)?
Thanks in advance,
Karsten
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
To pick-up with SIP phone, it has to be defined in sip.conf. Same goes
for zap and iax2.
callgroup and pickupgoup is configured in the config-files
Hi,
I have a few questions that I have been researching for a while. Sorry
if it is a bit long winded?
I have a huge need for VAD and CNG support in Asterisk, as my bandwith
is *very * limited and expensive, and VAD, CNG and DTX will save me
alot, at least 30 - 50%. I have installed and am
Hi Phil, we have a very similar setup... ISDN30 plus TE110P... I used a
standard cat5 patch lead... Worked a treat...
[EMAIL PROTECTED] wrote:
Hi,
I'm having problems getting our server to work with our BT ISDN30 box.
We are using a Digium TE110P card to connect to the ISDN box on the
Have you crc check enabled in
zaptel.conf?
Mimmus
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
What is the outcome of this finding on f3000.
goksie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews
Sent: Thursday, November 24, 2005 10:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Hi,
I have just joined this mail list yesterday and have
been searching the Asterisk wiki prior to posting this question.
Unfortunately I am not sure if I am searching at the
correct places, so I do apologise if this has been posted before.
I have currently been tasked to roll out
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Edit logger.conf and uncomment full.
Start Asterisk with the the -d option.
View debugging information in the /var/log/asterisk/full
Is -d option necessary?
Anyway, done that. Just thought that you think about something else.
Thank you!
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
callgroup and pickupgoup is configured in the config-files (zap/sip/sccp)
- is anything else needed ?
Sorry, I'm not up to this.
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
On Tue, Feb 28, 2006 at 11:43:55AM +0100, Karsten Wemheuer wrote:
Hello,
AFAIK the feature CD (call deflection) is only possible on
point-to-multipoint links, is this correct?
At least on my ptp link capiinfo reports:
[...]
Supplementary services support: 0x0033
Hold / Retrieve
hi
i'm migrating a callcenter to asterisk, inbound calls, queue monitorig
is ok, but how can i monitot outgoing calls?
for example my agent can be associated with more than one campaigns,
so if i monitor his calls in a day, how can i learn about how many
calls has he made for campaings A or
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Ah, I see. Not very nice to send such a confusing log ;-)
I'm sorry.
Anyway, your config is set to DID mode. So chan_capi will wait for more
digits (an INFO_IND with called-party-number) and if the already given
destination
Hi Anton (et al) -
Well.. I already sent my email to them :)
Kind of OT here, but just out of curiosity, how do you email them? Do
you have an actual address, or do you just use the form on their web
site? I've sent a bunch of requests via that form, and even though it
says I should receive
The only time I see recorded in your log is that of the recording check
-- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060227-131600|1141046151.2: Inbound recording not
Phil
To connect a BT ISDN30e to the TE110P card you do NOT require a T1
crossover. You need a straight through cable, and any cat5 cable will
be just fine.
Rgds
Tim
[EMAIL PROTECTED] wrote:
Hi,
I'm having problems getting our server to work with our BT ISDN30 box.
We are using a
Hi Mimmus,
I have just ordered a Sangoma A101, that I should receive pretty soon.
As you may remember, I will try to connect it to an Alcatel 4400, using
either EuroISDN or Q.Sig.
In order to save me some time and effort, would you mind sending me some
sample configuration, like the wanpipe and
Goksie - I have found the F3000 works fine with Asterisk, however, the
general release of this phone has been pushed back several times by
UTStarcom. At present, we have none of these available. I might suggest
the Linksys WIP300 as an alternative.
Cory J Andrews
VOIPSupply.com
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
When I try to call from asttapi one number, I get message No one is
available to answer at this time (1:0/0/0). Immediately after that I try to
call the same number from SIP phone (the same one that is used with asttapi)
and call goes
Hi guys,
I am trying to step our asterisk server. All the internal phones /
extensions work and I had the outgoing / incoming calls working before.
But for some reason, unknown to me, it has stopped working.
I have switched on sip debug and the main thing I notice is the
recurring appearance of
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Ah, I see. Not very nice to send such a confusing log ;-)
I'm sorry.
Anyway, your config is set to DID mode. So chan_capi will wait for more
digits (an INFO_IND with
Hi All,
I was able to install Asterisk and make outgoing calls. Recently I purchased two
DID's and I am facing a problem configuring them to my Asterisk, I hope with
the help I get from this list I will be able to configure successfully. Mu
errors are
Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331
On 28/02/06, Alexander Lopez [EMAIL PROTECTED] wrote:
read STDIN
while [ x$STDIN != x ]
do
export VARNAME=`echo $STDIN | cut -f1 -d :`
export VARVALUE=`echo $STDIN |cut -f2 -d : | cut -c2-255`
case $VARNAME in
(agi_request) export
As the thread from the other mailing list he sent this to states, it is
illegal to share the file(s) he is asking for. Below is the thread from the
sccp users mailing list that he sent this to.
sccp mailing list
2006/2/28, picciuX
In fact: the one you mention is not a config file; it is
I've seen in the asterisk configuration the way to call some internal
variables like caller-id-number, caller-id-name, language, etc. but..
What is the variable for changing the DID?
Is there a manual with this details?
--
Alejandro Vargas
___
Hi,
I have a PRI line with DID (from 100 to 499) in Italy.
Now I'm seeing all calls from same DID 'main' number. Can I set outgoing
CallerIdNum to the right extension?
Thanks
Mimmus
___
--Bandwidth and Colocation provided by Easynews.com --
In the 1.0.x branch asterisk does not always send SIGHUP to agi scripts on
call hangup. In 1.2.x a SIGHUP is always sent, even using DEADAGI - From
the UPGRADE.txt in the source:
AGI:
* AGI scripts did not always get SIGHUP at the end, previously. That
behavior has been fixed. If you do
Greetings fellow list members,
I have what I think is a
relatively simple question, but it did not appear to be addressed on the
wiki. I am trying to setup a queue so that it plays an estimated holdtime
announcement, but not a queue position announcement. Currently my dialplan
does both,
Hi,
Mimmus wrote:
I have a PRI line with DID (from 100 to 499) in Italy.
Now I'm seeing all calls from same DID 'main' number. Can I set outgoing
CallerIdNum to the right extension?
Yes, assuming your telco allows you to. Be sure to figure out what
number format is required in your case.
On Tue, Feb 28, 2006 at 02:32:27PM +0100, Armin Schindler wrote:
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Anyway, your config is set to DID mode. So chan_capi will wait for more
digits (an INFO_IND with
This is third time today that my Asterisk hangs up. It seams that I have
problems with h323. I'm using ooh323 from Asterisk add-ons. I have the
following configuration
Asterisk 1.2.1
Asterisk-addons 1.2.1
Fedora Core 4
I'm using SIP phones and
h323 trunk to my VoIP provider
Like I said this is
Using Asterisk 1.2.1, why not 1.2.4?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Ioan Indreias
|Sent: Tuesday, February 28, 2006 1:29 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] fax receive
That what worries me, the 2 systems Im testing are completely different. One
has x100p cards (2) and the other has 2 TDM400P with 4 FXO and 1 TE110P..
All same results... No go.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent:
Hi there!
I've set up an [EMAIL PROTECTED] with AVM C2 P2P ...
Everything works fine ...
BUT ... If someone is dialing the PBX head number without any
extension, asterisk can't handle this ... the DID in this case is
empty
Any ideas how to handle this?
Regards,
Marcus Hofbauer
--
|**
Yep, been there, done that.
How about this results:
[EMAIL PROTECTED] asterisk]# /usr/src/zaptel-1.2.4/zttest -v
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8192 sample intervals 100.00%
8192 samples in 8192 sample
No, same as you, thru the form on their website... :(
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Noah I. Miller
|Sent: Tuesday, February 28, 2006 6:56 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users]
Hi all,
hard for me to explain this, but it keeps happening on a number of machines
I attempt to upgrade zaptel, or do something to zaptel modules. and then
I reboot the machine, and for whatever reason, it hangs on loading the
modules
Either the install wasn't complete, the zaptel modules
Hi Mark,
Thanks for your reply.
For the phase you have indicated the time it took was immediate, no delays
there.
Regards
Ash
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hulber
Sent: 28 February 2006 13:00
To: Asterisk Users Mailing List -
We are using an Asterisk box to do conferencing right now. I
have had about sixteen active lines in conference and the quality was
acceptable. We now have a need for 50 people to conference at one time. Does
anyone have enough experience doing this to give me some pointers. Will it even
be
FC2 SpanDSP -pre25, Te110P. Works perfect.
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 9:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] fax receive using TDM400P
Ok 1 for Debian, any
Marcus Hofbauer [EMAIL PROTECTED] writes:
BUT ... If someone is dialing the PBX head number without any
extension, asterisk can't handle this ... the DID in this case is
empty
Any ideas how to handle this?
Try the WaitExten application.
cu,
Wolfgang
Anyone have a clue how to get the voicemail pager
notification (actually, text message) source email address to change?
We use both the email and pager feature, so just using the
email feature to send test messages is not an option.
We also do not manage the users email, so creating
What happens if you take out the Zaptel I/F's? If it boots, you can correct
whatever you did then replace them.
hth
-Original Message-
From: Chris Earle (CBL) [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 28, 2006 7:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
On Tue, Feb 28, 2006 at 03:40:28PM +0100, Marcus Hofbauer wrote:
I've set up an [EMAIL PROTECTED] with AVM C2 P2P ...
Everything works fine ...
BUT ... If someone is dialing the PBX head number without any
extension, asterisk can't handle this ... the DID in this case is
empty
Do you
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 02:32:27PM +0100, Armin Schindler wrote:
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Anyway, your config is set to DID mode. So chan_capi will wait
Thanks
Yeah, you would think so wouldn't you.
Tried that , and still wouldn't boot
Really annoying. beacuse I've been doing work with the zaptel drivers
and such and this happened once already...
Thanks for the suggestion,
Chris
- Original Message -
From: Colin Anderson
On Tue, 28 Feb 2006, Marcus Hofbauer wrote:
Hi there!
I've set up an [EMAIL PROTECTED] with AVM C2 P2P ...
Everything works fine ...
BUT ... If someone is dialing the PBX head number without any
extension, asterisk can't handle this ... the DID in this case is
empty
Any ideas how
Robert Andersson wrote:
Hi,
I have installed an TE110P but forgot to change the jumper
settings to E1. I don't have easy physical access to ther server
at the moment so I wonder if it will be possible to run it without changing
the jumper settings with a configuration like below or will it be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 28.02.2006, at 15:44, Chris Earle ((CBL)) wrote:
Hi all,
hard for me to explain this, but it keeps happening on a number of
machines
I attempt to upgrade zaptel, or do something to zaptel modules.
and then
I reboot the machine, and
I have also issues with jitter over wan (cdma),
I'm trying to debug how dejitter buffer is working (using iax2 jb
debug), but nothing happens/no debug output on asterisk console :-(
is any way how to monitor iax jitter buffer? thx
PJ
I'm really hoping to see some working settings from some
I had a similar issue here in Aus where I
was chasing crossover cables around. Eventually the cows actually did come home
and I called up the telco. They rebuilt (or reinitialized) the
ISDN service and everything worked a treat from there on in. Took a couple of
days to get to this point.
Paul,
Ah, I see. Our echo is largly under control now. It took me a while to
figure out the gains and get them tuned, and now the echo only leaves very
small artifacts. Nonetheless, this still provokes the odd complaint here and
there. We use VOIP for outgoing calls when our POTS lines are
Hi,
I am trying to determine the actual call duration (billsec) when is
used Attended Transfer but this is very dificult because there is no
relation between channels. Are there any suggestions how can be solved
this?
I have an idea where in the CDR must be added new column where to
be
Ah! A spandsp pre25... Ok.. The plot thickens :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Colin Anderson
|Sent: Tuesday, February 28, 2006 8:55 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users]
When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise
that this is DATA call, but answering anyway (playing IVR messages,
etc...)
How to stop that? I want that only VOICE calls are answered, and
DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f,
How about this:
--- Results after 33 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988163
Faxing is working just fine. Mabe it's mother board related?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent:
Brent-There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be
One board is intel and the other is also intel (supermicro). :(
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Cosmin Prund
|Sent: Tuesday, February 28, 2006 10:28 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE:
Hi Pasqualotto,
Actually, I've seen your post on Asterisk-Users list yesterday, but I could
not understand back then. Now, I've checked your sip configuration again, I
think you make a mistake in type of sip account. I use friend not
peer. I am not sure though.
Following is what I had in my
Boot up with this:
http://www.sysresccd.org/Main_Page
Mount the partition in question and remove the Zaptel module. Reboot, and
you should be good (except for Zaptel of course)
hth
-Original Message-
From: Christoph Eicke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 28, 2006 8:32
Do anyone know who can
provide some cheap PH routes/.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Ron, keep in mind, that yoy mix parameters for new and old iax
jitterbuffer implementation, these:
dropcount=2
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
are ae valid only for _old_ implementation, and I thing, that asterisk
1.2 use new iax buffer by
On Feb 28, 2006, at 1:22 AM, [EMAIL PROTECTED] wrote:
I am having problems with a Zoom 5801 and *.
It does not appear possible to route voip calls out the FXO, all voip
calls get routed to the FXS no matter what.snip
If there is a routing function of some kind on the modem setup,
perhaps
is it possible
resetcdr and/or start newcdrAFTER pickup of
dialout?
[dialthru]exten
= s,1,Answer()exten = s,2,DigitTimeout,4exten =
s,3,ResponseTimeout,10 exten = s,4,Playtones(dial);exten =
i,1,Playback(invalid)exten = i,2,Goto(dialthru,s,2);exten =
t,1,Playback(timeout)exten =
Sam Tam wrote:
Do anyone know who can provide some cheap PH routes/.’
I've been looking myself. Cheapest DIDs in Metro Manila I've seen are
$27.50/month; cheapest termination to same (non-mobile) from US I've
seen is $0.23/minute.
Expensive chismis :-)
-Johnathan
On 08:05, Tue 28 Feb 06, Colin Anderson wrote:
I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD,
but no go
Do this, pick the kernel you want to load, and add: single
So in my laptops case it sais: Linux single
That will boot your pc into singleuser mode and it won't
i think we can help. we do have there some contacts and if the total volume would be significant, we can give a nice quotation.
let me know off list what you exactly need.
BTW, $0.23/minute is much much high compared to our solution.
On 2/28/06, Johnathan Corgan [EMAIL PROTECTED] wrote:
Sam
Can be as low as 15€cents from us on fix and 20€cents for mobiles
We don't have dids yet for Philipine
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Johnathan
Corgan
Envoyé : mardi 28 février 2006 18:07
À : Asterisk Users Mailing List -
Dear all,
Did anyone successfully test T38 fax pass thru to Cisco
as53xx? Weve tried 1.2.4 with latest patch and latest svn trunk
and T38 patch but still not work. Reinvites from Cisco are correctly passed
back to the originating gateway, but fax never able to connect.
Cisco IOS
I'm chasing down a pop/click type of disturbance on a PBX system.
Strangely, the disturbance is only heard by the outside caller, the
internal recipient hears the caller crystal clear. This seems to have
crept up when upgrading the zaptel driver to the 1.2 series while
running 1.0.10. I went
We have just installed Asterisk in our new office and we have some
teething problems, but so far nothing we did not
expect/could not handle. However, our CEO was very attached to a
function in our old Nortel PBX that I am not sure
how to approach. If someone could point me in the right
I think I have seen a
post about that before. But cant find it again
Can some people light me
up with the detail
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
On Feb 28, 2006, at 6:50 AM, Ash Thakrar wrote:
Hi Mark,
Thanks for your reply.
For the phase you have indicated the time it took was immediate, no
delays
there.
I have seen on the list several discussions of how additional delay on
ringing can be due to Asterisk trying to get caller ID
Hi Jordan,We are planning on building the same thing. We are still waiting for the hardware. We are using a Dell PE 2850 3GHz with 2G of RAM and a TE210P.I asked Digium support if this can suupport 50 users in one conference and the tech support guy said yes. Here's also a response
Paul,
Just curious - what kind of stuff are you reading from the file?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi guys,I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk.Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller.
Immediately, I though into
On Wed, Mar 01, 2006 at 12:45:45AM +0800, Sam Tam wrote:
I think I have seen a post about that before. But can't find it
again
Can some people light me up with the detail
GSM extenders I don't think are legal in the UK, except if
installed/operated by a GSM network operator (as
Hi,
Im
trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with
other PBX. The port use to connect is TCP 1720 but I cant configure this
port on my * box. Im using a H.323.conf file sample to activate the port
but the * isnt listening there. Somebody have any idea
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
...
- But I guess the workaround would yield to my current situation (I'm
running a patched version of 0.35 currently as mentioned at the start
of this tread): When a caller uses overlap sending (e.g from a POTS
line) instead of block
)
-- Executing AGI(SIP/300-3bb9,
recordingcheck|20060228-133504|1141151704.8) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060228-133504|1141151704.8: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
1 - 100 of 182 matches
Mail list logo
