On Sat, 2003-11-22 at 17:01, God Knows Well wrote:
Hi
Have any body experienced Asterisk with Speex??May i know the result i.e
Voice quality or echo problems and wats frame size and other settings are
compataible with asterisk .
Before you rub someone the wrong way, please understand
Hello list!
I'm using Asterisk CVS-11/22/03-04:28:51 and try to route my normal
(classic) phone calls via nikotel (www.nikotel.com). I can talk about 1
minute and get then disconnected. Here my current configuration parts
which affect nikotel:
register = chabrol:[EMAIL PROTECTED]/500
I seriously doubt these things are possible.. not without recoding some
of the Asterisk components..
The whisper thing might take some work, but wouldn't it be possible to
forcibly park a call and have the manager pick it up to achieve #2?
Regards,
Andrew
It seems that there's a non-printable character at the beginning of the
DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off,
everything works right.
Would you mind sharing with the rest of the list your patch to drop off the
character? Or was it simply $EXTEN:1? How did
For anyone trying to make an E1/T1 crossover, here's a nice diagram from
NMS that may help. The only pins that are needed in a short cable for
testing are 1,2,4,5.
Aren't those the only wires that need to be connected for _any length_ of
cable since the others are non-connects on the jacks?
i am having callerid problems with *. i have the
callerid from my telco and it shows up in my normal
phone when i connect it directly to the line but if i
connect the same phone thru * server the callerid is
not shown. i am using X101p and tdm400p. i have
everything defined in my zapata.conf
Do you really want all those spans going down cause someone tripped over
a power cable or your hard drive nukes itself?
You usually don't worry about either of those problems when you've got
redundant power supplies and drives in the rackmount system in a locked
room.
We only use 2
How's this worse than an as5300? I could install ata-flash and get
high-ish end pc hardware (rcc serverworks boards, etc). Heck, if it
was such an evil thing, I'm sure Voicepulse (along with every other
clec) and friends wouldn't be doing it!
The dialup ISP I worked at used MaxTNTs with
On Sunday 23 November 2003 09:38, Harry McGregor wrote:
On Sun, 2003-11-23 at 07:57, Andrew Kohlsmith wrote:
Do you really want all those spans going down cause someone
tripped over a power cable or your hard drive nukes itself?
You usually don't worry about either of those problems
Third and last question for now: The phonebook used to be in the
ini-file, but
it seems to be somewhere else now ??? I'd like to
preprogram other entries in there :-)
It's a separate file now. I have put in a feature request to have TWO
phonebooks paths in the ini file, a global and
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
i am having callerid problems with *. i have the
callerid from my telco and it shows up in my
normal
phone when i connect it directly to the line but
if i
connect the same phone thru * server the callerid
is
not shown. i am using X101p
Hi,
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 23, 2003 6:22 PM
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
Hi Dan,
Some preliminary testing with this version
Hi,
- the phonebook is now in a separate file;
Duh, I can't read :-)
:-))
Another issue I've just seen, however :-)
When I'm passing a call from a Zap channel (PRI) I get an error: STATUS:
Bad
or incomplete voice
This is strange someone else with this issue?
I've also noticed
Hi,
I have put in a feature request to have TWO
phonebooks paths in the ini file, a global and personal (with
personal
overriding global for duplicate entries) so that you could handle a global
directory and still be able to have personal entries without a LOT of
duplication work.
I intend
Citeren Dan [EMAIL PROTECTED]:
However, is
there an option so that the sound-dialog can be forced to just use the
defaults it preselected for me ?
At first run, you are asked to select the audio devices... one time only..
Then it will be used at each run
Yes, this was clear to me.
On Sun, 2003-11-23 at 08:52, Andrew Kohlsmith wrote:
i am having callerid problems with *. i have the
callerid from my telco and it shows up in my normal
phone when i connect it directly to the line but if i
connect the same phone thru * server the callerid is
not shown. i am using X101p
i can't get callerid in anyway thru *.
If * can't get it how do you think you're phone's gonna get it if it's
conneected via *?
whats demarc? can u explain more?
point of demarcation -- where the telco says anything up to this exact
spot, we'll look after and take responsibility for.
Hi,
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 23, 2003 7:39 PM
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
Citeren Dan [EMAIL PROTECTED]:
However, is
there
i believe the more accepted term is 'basics' as in asterisk-basics
Grzegorz Nosek wrote:
On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
Maybe asterisk-install ?
i think that is a bad idea, atm i have the option to use my screen speakers
for ringing and my headset for the actual audio. My pc speaker sux bigtime
(too quiet) but i agree that putting an option for the pc speaker is a good
idea.
Zoa.
Ok... you're right. I'll make it to take the default
I never used Bayonne as a PBX of any kind, only for IVR, that's what it was
designed for. You can put other packages with Bayonne to get it to work with
some VOIP protocols supposedly , but it would be much more work to do that
than to just set up Asterisk.
MATT---
-Original Message-
On Sat, 22 Nov 2003, marrandy wrote:
Now, I am getting the stutter/dialtone, that tells me I have a message, But
when I check both the new and old messages on each of the 3 mailbox accounts,
there are no messages.
I had this problem the other day. I think it was caused by me trying to
Hello:
I have installed *. I configured my SIP account and
my X100P. But whenI call from SIP or from PSTN. The SIP extension hear an
echovoice of its conversation. Anyone can help me???
Thanks,
voipfan
Ya learn to search the archives. This has been covered MANY MANY times.
bkw
On Sun, 23 Nov 2003, VoIP Fan wrote:
Hello:
I have installed *. I configured my SIP account and my X100P. But when I call from
SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone
System Diagram:
+- PRI -- modem bank
|
LEC -- PRI -- * with --+
T400P --+
| (pri)
+- TDMoE -- *
with -- SIP --
Well, SIP to SIP with no intervening analogue should produce no echo at
all. Echo on SIP to analogue calls has been covered extensively on this
list. Do a search on echo.
Iain
Hello:
I have installed *. I configured my SIP account and my X100P. But when I
call from SIP or from PSTN.
Hey Brian... when resetting transmission levels in zapata.conf, is restarting *
enough?
Ya learn to search the archives. This has been covered MANY MANY times.
bkw
On Sun, 23 Nov 2003, VoIP Fan wrote:
Hello:
I have installed *. I configured my SIP account
I was wondering if you can pick up a ringing channel by dialing *8# when
you and the other phones are in pickupgroup. Could you do something to
the effect of If the caller was put on a certain extension and just
sitting there... Could you grab the caller by doing something like
*8exten where the
On Sat, 2003-11-22 at 21:25, [EMAIL PROTECTED] wrote:
This book will be available in electronic form under some sort
of open publishing license, in addition to being sold in bookstores,
right?
Yes, that's the plan. I personally am a lot more interested in this
from the standpoint of having
On Sat, 2003-11-22 at 21:22, [EMAIL PROTECTED] wrote:
please please please if you are going to write something like that,
write it using something like texinfo or groff or docbook or whatever
so that you can make it available in a wide variety of formats.
you should not have to run non-free
Hi Dan,
From the log, it looks like something wrong is in the modem software. I
have been adding a V.27ter modem to the FAX software, to do the slow
2400, and 4800 bps FAX modes. I've been busy with other things the last
couple of weeks, but I am finishing this off now. I have also made the
existing V.29 modem more robust on bad lines. When I get this released
(should be this week) I will look into the issues that you and some
others are having. It seems the current software works OK with some FAX
machines and not others. Perhaps I need to make it more tolerant. :-)
I have a
Hi Mark
Did you or anyone else ever find a satisfactory solution to this? Are there any
phones which provide voice through the serial connection?
What about the nokia card phone - does it have open source drivers?
Cheers
Rob
On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote:
Greetings...
We've been having some interoperability issues between Asterisk and an
AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000
somewhere. So, I've been pondering using iptel.org's SIP server (SIP
Express Router) as a front end for PSTN calls going out to the Mediant,
hi folks.
(apologies in advance if this is a particularly stupid question)
just getting my feet wet with asterisk / agi, and am a little stuck using
EXEC. it works fine for applicaitons that take simple arguments, but
chokes on applications that require multiple words as arguments.
for example,
Robert Murray wrote:
Hi Mark
Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection?
What about the nokia card phone - does it have open source drivers?
Cheers
Rob
On Sat, Jul 13, 2002 at 10:31:53AM -0500,
Hello!
Is there a phone compatibility list anywhere? I know Asterisk is
supposed to be compatible with IP phones that support SIP, h.232, and
IAX, but a list of phone known to be supported would be a nice addition
to the documentation. I need a buisiness phone, and the Cisco is
_EXPENSIVE_
actually, i do have a workaround which bypasses the exec command entirely:
system(asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local');
but it's ugly. seems like it should be possible to do this with exec.
.t
-- Forwarded message --
Date: Sun, 23 Nov 2003
On Sunday 23 November 2003 20:17, tad wrote:
hi folks.
(apologies in advance if this is a particularly stupid question)
just getting my feet wet with asterisk / agi, and am a little stuck
using EXEC. it works fine for applicaitons that take simple
arguments, but chokes on applications that
asterisk*CLI show agi
answer Asserts answer
wait for digit Waits for a digit to be pressed
send text Sends text to channels supporting it
receive char Receives text from channels supporting it
tdd mode Sends text to channels supporting
What is the goal of this? It doesn't make much sense to me. Care to
share some insite into what your goal is?
bkw
On Sun, 23 Nov 2003, tad wrote:
actually, i do have a workaround which bypasses the exec command entirely:
system(asterisk -r -x 'add extension s,3,Playback(demo-congrats) into
Late Sunday night, getting
cvs update asterisk
? asterisk/doc/api
cvs server: Updating asterisk
M asterisk/app.c
cvs [server aborted]: missing expected branches in
/usr/cvsroot/asterisk/asterisk-ng-doxygen,v
[EMAIL PROTECTED] src]#
Checkout does same thing
What did I mess up?
Works fine from here... blow your src tree away and start fresh.
bkw
On Sun, 23 Nov 2003, Jonathan Biggs wrote:
Late Sunday night, getting
cvs update asterisk
? asterisk/doc/api
cvs server: Updating asterisk
M asterisk/app.c
cvs [server aborted]: missing expected branches in
On Sun, 2003-11-23 at 21:30, Brian West wrote:
asterisk*CLI show agi
answer Asserts answer
wait for digit Waits for a digit to be pressed
send text Sends text to channels supporting it
receive char Receives text from channels supporting it
hi,
my * box is behind a NAT. i am using netgear router.
it seems that my toll free number is unable to
register with nufone because my * box is behind NAT
firewall. the outgoing calls are working fine and
nice. what can be done?
cm
configurations are fine.
=
Designs
This is not a private issue as far as i know. u
misundersood me. the nufone account is fine. the real
problem is with the asterisk NAT issue. i was asking
for help if any one had similar problem with nufone
account. i am using IAX. is there anything like
nat=yes as in sip.conf?? i read iax should
Hi,
- Original Message -
From: zoa [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 23, 2003 8:32 PM
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
i think that is a bad idea, atm i have the option to use my screen
47 matches
Mail list logo
