IMHO the Festival application is slightly broken since it doesn't interface
to the asterisk playback routines in a standard way. I've never had much
luck with caching but have experienced the problem you outline on direct
text conversions. This issue has been discussed on the bug tracker and
--On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink
[EMAIL PROTECTED] wrote:
http://ipphones.utelisys.net/
http://ipphones.utelisys.net/includes/cisco.inc.phps
There are some perl classes on this topic too (even for image
generation!). I didn't had the time to made a GD patch to use it
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson
Sent: Friday, 11 June 2004 19:53
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] XML How To for Cisco 7960
--On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink
[EMAIL PROTECTED] wrote:
http
sip cnf files?
-Original Message-
From: Iain Stevenson [mailto:[EMAIL PROTECTED]
Sent: 11 June 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] XML How To for Cisco 7960
Ah, how? Which SIP version do you have - 'cos I've made innumerable
tests of my own (and using Cisco code
to indicate that you don't know what HylaFAX and
spandsp actually do :-)
Regards,
Steve
Iain Stevenson wrote:
... might as well use hylafax.
Iain
--On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote:
Hi all.
I'm looking to set up a fax via email service so that users can email a
specific
transformed into an email.
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson
Sent: 08 June 2004 09:10
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fax via email
Maybe not. However, if the user is primarily interested in fax to email
... might as well use hylafax.
Iain
--On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote:
Hi all.
I'm looking to set up a fax via email service so that users can email a
specific mailbox and receive fax's to a specific mailbox. Can this be
done? I've had a look an SpanDSP and I
--On Friday, May 28, 2004 2:57 pm -0400 Timothy R. McKee
[EMAIL PROTECTED] wrote:
My SIP users need to transmit the # key as part of data entry. Asterisk
intercepts and initates a transfer function. I'm almost positive I've
seen this discussed somewhere, but none of my searches are finding
Yes, I've read and implemented all the stuff on IAX. It's the local SIP
connection and its RTP streams that's the problem. For instance I noted
the strange timestamp behaviour from * on local traffic earlier.
Iain
--On Tuesday, May 18, 2004 1:56 pm -0600 Rich Adamson
[EMAIL PROTECTED]
--On Tuesday, May 18, 2004 1:42 pm -0500 Nik Martin
[EMAIL PROTECTED] wrote:
Out of context, this isn't much information. Is your network connection
OK?
Yes, AFAIK - I'm running all the traffic shaping / prioritisation stuff
mentioned on the list
Is your broadband provider having troubles?
--On Tuesday, May 18, 2004 1:43 pm -0500 brian [EMAIL PROTECTED] wrote:
Strange I do 7960 = * = IAX all day long without one jitter or any bad
audio. Now if both ends are NOT running the very latest(within the last
month or so) CVS-head for example if you have say a 2 month old
chan_iax2.c on
:
Out of context, this isn't much information. Is your network
connection
OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber
followed by a valid string of arguments. Do a show application saynumber
in *.
Iain
--On Thursday, May 20, 2004 7:14 am -0400 Jer [EMAIL PROTECTED] wrote:
Dear all
I am just getting started with AGI
so I wrote the following
anything relevant here.
Enable the maximum debugging support in *.
Iain
--On Thursday, May 20, 2004 2:44 pm +0300 Apollon Koutlides
[EMAIL PROTECTED] wrote:
Iain Stevenson wrote:
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber
followed by a valid string of arguments. Do
--On Tuesday, May 18, 2004 12:30 pm -0400 Stephen R. Besch
[EMAIL PROTECTED] wrote:
P.S. Grandstream, if you are listening, then Early Dial is still broken!
It's been many months now to fix what apparently is just a counter bug.
Come on, let's get this fixed.
Here, here!
Iain
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless. I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it
This isn't really the issue. Up until a week ago or so everything worked
fine with a hallf duplex hub. Now it doesn't - so I suspect some code
change made in * is responsible. I think * must maintain backwards
compatibility with existing hardware or many people will get fed up with
I've had this too, reported it as a bug last week and got my butt kicked
for not being responsive enough in providing support to sort it out. You
could file another bug report but be sure to have a thick book ready to
stuff down your trousers.
Iain
--On Friday, May 7, 2004 10:43 am -0400
You've probably got callerID enabled in zapata.conf. That will cause a
wait of several rings whilst * looks for the caller ID info. Since this
only works in the US (or pkaces with similar phone systems), disabling it
in other territories saves the ring delay.
Make sure you have this in
Listening to the options on the voicemail system it seems to be missing a
feature for users to turn voicemail off completely. This seems a rather
glaring omission. Does the feature of turning off message recording via
the phone exist - or does it need a patch?
Iain
--On Saturday, April 10, 2004 10:42:26 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
Thanks for all the replies.
Can someone tell me if it is possible to put two of these X100P cards
into the same machine to order to gain access to two BT landlines ?
I believe so although problems have been
--On Saturday, April 10, 2004 11:55:26 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
What I want to do is have the asterisk server sat in my house and used by
my family to access the BT landline and to recieve calls made to that
landline. If it is not possible to do the auto attendant thing
--On Saturday, April 10, 2004 17:47:24 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
Sorry to sound stupid, but where can I get copied of the Asterisk manual
?
http://www.asterisk.org/index.php?menu=support#handbook_project
What is the VoIP wiki and where can I get that too ?
The wiki is a
--On Wednesday, March 31, 2004 2:00 pm -0500 Hall, Eric M.
[EMAIL PROTECTED] wrote:
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking at just
using IP Phones or IP Softphones just
--On Monday, March 29, 2004 8:24 am -0500 Kevin [EMAIL PROTECTED] wrote:
Hi All-
As I'm doing this, I'm considering installing an asterisk box at my
office (about 6-10 different phone stations) and would like to get
opinions on the best quality and/or most well-supported SIP hard phones
and
--On Monday, March 29, 2004 2:09 pm + Hermann Wecke [EMAIL PROTECTED]
wrote:
Which one? I'm running one the latest image available at
http://www.grandstream.com/BETATEST/ (b14p4.54.zip) and my * and my GS are
working OK.
The 4.53 was buggy, but I can't find a problem (so far) with 4.54
Welcome to the very much less than wonderful world of Cisco software
support. When will those guys simply make the software downloadable
straight away from their website for a modest fee?
Iain
--On Saturday, March 27, 2004 1:43 am -0600 Mitchell S. Sharp
[EMAIL PROTECTED] wrote:
I just
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: Saturday, March 27, 2004 4:06 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
Welcome to the very much less than wonderful world of Cisco software
support. When will those guys simply make
--On Saturday, March 27, 2004 4:52 pm -0500 Ray Burkholder
[EMAIL PROTECTED] wrote:
Iain Stevenson wrote:
.. not sure this applies outside the US - or I'd reach for
the credit card.
Iain
--On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA
[EMAIL PROTECTED] wrote:
If you pay 8 USD
--On Wednesday, March 24, 2004 11:13 am -0600 Steven Sokol
[EMAIL PROTECTED] wrote:
I have seen a number of postings cross this list that mention the
possibility of standards-tracking IAX2 with the IETF (generating an RFC,
etc.). Has that gone anywhere? What would it take to make it happen?
I think this has been discussed a lot in the last 3 days - do some legwork
before posting!
Iain
--On Wednesday, March 24, 2004 3:53 pm -0800 Ron McMillin
[EMAIL PROTECTED] wrote:
I am using the wildcard X100P with *. PSTN line comes in to the FXO port
of this card. Everything works fine most
--On Sunday, March 21, 2004 8:11 pm + Dee Lowndes [EMAIL PROTECTED]
wrote:
If I find the voltage drop out can I configure the x100p to do it based on
the new voltage drop. If so where and how?
To a certain extent yes. Im fact, in the absence of measurements you could
just try a couple of
--On Monday, March 22, 2004 12:51 pm +0100 Matteo Rancilio
[EMAIL PROTECTED] wrote:
You're right :)
I'm using Asterisk 7.2 on a SuSE 8.2 installation.
Hardware:
Dual Intel PIII
1Gb ram
AVM Fritz! ISDN card
SIP
CISCO Phones
Codec g711 (switching today to g729)
... and what applications? AGI,
I assume you're using Dial with the Tt options to enable transfer? If
you need to keep the transfer you may need something like the double hash
patch I posted last week.
Iain
--On Friday, March 19, 2004 1:39 am -0600 Mitchell S. Sharp
[EMAIL PROTECTED] wrote:
Haven't been able to find
cant find
it, can you give me the ID please.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Friday, March 19, 2004 4:00 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Using the pound (#) key while in a call
I assume you're using
into the transfer.
A little odd I'd say.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Friday, March 19, 2004 7:19 AM
To: Asterisk Users
Subject: RE: [Asterisk-Users] Using the pound (#) key while in a call
It went to the list 'cos Mark's
I guess that means every * server needs to play old Victorian Music Hall
favourites:
Bicycle Built For Two
Daddy Wouldnt Buy Me A Bow Wow
Hello, Hello, Whos Your Lady Friend?
The Man on the Flying Trapeze
... and many more
Iain
--On Friday, March 19, 2004 12:59 pm -0800 George Pajari
Look here ...
http://www.cstr.ed.ac.uk/projects/festival/
Iain
--On Friday, March 19, 2004 4:10 pm -0600 Justin Carlson [EMAIL PROTECTED]
wrote:
I am sorry if this is a silly question but I can not seem to locate the
festival binaries. does this come with asterisk or is it another
... just installed this. The database updates OK but status.php shows no
active channels (either SIP to SIP or SIP to voicemail).
Iain
--On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED]
wrote:
I just pushed out a snapshot of the -devel version of monastery.
I'll answer my own question ...
If you don't call the database asterisl you need to edit in the name you
do use to status.php otherwise monastery behaves as though nothing is
happening rather than flagging an error ;-)
Iain
--On Thursday, March 18, 2004 5:51 pm + Iain Stevenson
[EMAIL
What sort of phone line are you using? Connecting an X100P to a PBX line
or ISDN TA can cause the problems you mention.
Iain
--On Wednesday, March 17, 2004 7:37 am -0600 [EMAIL PROTECTED] wrote:
Hullo!
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
Sample
It's quite easy to write an LDAP interface. There are code snippets on the
web and I can send you my very quick hack, if you like.
Iain
--On Thursday, March 11, 2004 4:06 pm -0600 Brian R. Swan
[EMAIL PROTECTED] wrote:
Hi gang,
I'm looking into writing a some phone book XML/PHP software
I hacked the Wait command to wait in increments of 100ms. The 7960 needs
about 300ms delay after answer to play the sound properly. ATA186's work
fine without any delay for me.
A finer grained 'Wait' would be helpful in developing workarounds for this
sort of problem.
Iain
--On Wednesday,
--On Thursday, March 11, 2004 3:17 am -0500 James Golovich
[EMAIL PROTECTED] wrote:
As of 3/4/2004 in cvs head and stable the Wait application has accepted
time with fractions of a second. So 0.1 would be 100ms, 0.3 would be
300ms, etc.
James
Thanks, that makes a workaround for the 7960
Try the attached patch. Go to your asterisk root directory and type:
patch -p0 path_to_patch/Parking.patch
.. then rebuild asterisk.
Iain
--On Wednesday, March 10, 2004 7:43 am -0500 John Congdon [EMAIL PROTECTED]
wrote:
I have applied the patch and restarted Asterisk.
But it still
,
MATT---
-Original Message-
From: Iain Stevenson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outbound Transfer and the # key
Try the attached patch. Go to your asterisk root directory and type:
patch -p0
The root cause of the problem is the 3 wire phone wiring in the UK compared
to the 2 wire wiring in the US. I've had the problem you mention just
using ordinary phones! I suspect that a socket somewhere has been wired up
with wires crossed. Your X100P probably needs to go straight across the
Anyone else seeing SIP registration requests rejected by FWD? I don't seem
to be able to register any longer - even though my SIP config remains the
same.
Iain
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Edit the top level * Makefile to enable this:
OLD_DSP_ROUTINES
then rebuild and reinstall *
Iain
--On Monday, March 1, 2004 7:09 pm -0300 listas iPfone
[EMAIL PROTECTED] wrote:
Hi!
Every time i make or receive a call with my x100p i receive that notice:
NOTICE[1234379840]:
Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream
uses the latest firmware and SIP INFO.
Iain
--On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED]
wrote:
I cannot get the Message Waiting Light (MWL) on my Grandstream phone to
turn on when I leave
I'd reach for the Oxometer on that one - 36k shouldn't make any difference.
However, the X100P may be introducing some capacitance on the line that
would affect the ADSL signals - but the purpose of filters is to stop this
problem. Maybe it's worth trying another filter between the X100P and
Looking at the reference design for the chipset used in an X100P a fair
chunk of capacitance is slapped straight across the line which would
present a significant load to DMT signals. I guess the fax machine
introduces some inductance in series with the phone to compensate.
I found this link
... looks like a case for the System application or AGI. Check out AGI
on the VoIP wiki.
Iain
--On Wednesday, February 18, 2004 12:41 pm +0100 Alessio Focardi
[EMAIL PROTECTED] wrote:
Hello asterisk-users,
just a simple question: I'm looking for a way to execute an external
script (php) on
[EMAIL PROTECTED] wrote:
Iain Stevenson wrote:
The problem with the Ofcom consultation as I see it is that it seems
to be regressive wrt to the position now being taken by the FCC.
There are probably not many more than 250,000 VoB users worldwide so
now is not the time to impose significant market
The problem with the Ofcom consultation as I see it is that it seems to be
regressive wrt to the position now being taken by the FCC. There are
probably not many more than 250,000 VoB users worldwide so now is not the
time to impose significant market constraints.
The new EU regulatory
Well, since they restricted attendance to service providers and
representatives of consumer organisations I wouldn't be too optimistic for
a balanced outcome ;-)
Iain
--On Monday, February 16, 2004 4:51 pm + WipeOut
[EMAIL PROTECTED] wrote:
Steve Kennedy wrote:
On Sat, Feb 14, 2004 at
Yes - not much seems to be creeping out of the list servers.
Iain
--On Friday, February 13, 2004 07:54:50 -0600 Rich Adamson
[EMAIL PROTECTED] wrote:
Are others seeing hugh delays and/or lack of connectivity to Digium?
Rich
___
Asterisk-Users
You can probably use the festival text2wave utility in a cron job to create
a speech file from your source text and then use asterisk's Playback
function to play it as required.
Iain
--On Saturday, February 14, 2004 9:41 pm +0100 Lars Fredriksson
[EMAIL PROTECTED] wrote:
Hello!
I wan't to
Search the list - there's a detailed answer on it.
I have two of the I1 version (at least that's what they say they are -
ProductId: ATA186I1) and they work with UK spec phones. All you need to
watch for is that UK phones are three wire and US phones are 2 wire.
Maplin sells an adapter to
--On Monday, February 9, 2004 8:35 am -0700 Erick Schmidt
[EMAIL PROTECTED] wrote:
When I try to make Asterisk I get the following error:
In file included from aescrypt.c:39:
aesopt.h:156:22: endian.h: No such file or directory
aesopt.h:157:24: byteswap.h: No such file or directory
make: ***
file if and only if you ref an extension
and not an application.
bkw
On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:
There is no CDR for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED
for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:27 PM
Subject: [Asterisk-Users] Back to front logging for calls placed
through /var
This is similar to the last version and applies against the current cvs.
cd asterisk
patch -p0 Parking.patch
Then the double has transfer should be back.
Iain
--On Friday, January 16, 2004 6:10 pm -0500 mattf [EMAIL PROTECTED]
wrote:
Hello,
I was using the doublehash.patch that Iain
I've just noticed that if you start a call by writing a file to
/var/spool/asterisk/outgoing the cdr created on termination logs the call
placed to the local extension - not to the destination in the PSTN. Hence
there is no record of the PSTN number dialled. I guess most people want to
log
check to make sure your
accually connetcing to the database.
bkw
On Sat, 17 Jan 2004, Iain Stevenson wrote:
I've just noticed that since swapping from the direct mysql cdr driver to
cdr_odbc, the call duration (and anything else that's an integer) isn't
logged - anyone else seen this and know
I've just noticed that since swapping from the direct mysql cdr driver to
cdr_odbc, the call duration (and anything else that's an integer) isn't
logged - anyone else seen this and know the reason. The cdr_odbc driver
gives no error messages and records any string based fields correctly.
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax
detected, but no fax extension
... and then redirected to voicemail. An extract from extensions.conf is
attached below. Is there any way to stop * even
app_festival currently seems to chop the start of sound it plays back -
probably something to do with rtp and maybe the same problem that was
present in voicemail prompt plauback.
Iain
--On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu,
Will the driver support big endian systems (PPC) - most ISDN cards seem to
ship only with Wintel drivers. I have ISDN but at the moment have to use an
analogue interface through a TA.
Iain
--On Wednesday, January 14, 2004 3:11 pm +0100 Klaus-Peter Junghanns
[EMAIL PROTECTED] wrote:
Hi,
It may not be you, I think the Festival driver is buggy. Specifically,
I've found that the the way in which you pass the text to Festival matters.
If I use the Festival () suntax then it won't work. If I use the wrong
sort of quotation mark instead of ' there are problems. Asterisk will
Looks familiar to me - check this:
http://bugs.digium.com/bug_view_page.php?bug_id=695
Iain
--On Tuesday, January 13, 2004 4:55 pm + Tristan 'Minty' Colgate
[EMAIL PROTECTED] wrote:
Hi,
I'm trying to use the say-ani agi asterisk-perl script and am
experiencing crashes, I am also
Prices? Are we talking a 7960 for the price of a SNOM?
Iain
--On Friday, January 9, 2004 6:00 pm + Adthrawn
[EMAIL PROTECTED] wrote:
Hi,
I know it's not really the place, but if anybody in the UK (or US) is
interested, I'm clearing out lots of new Cisco stock...
7970G's (colour LCD),
--On Thursday, January 8, 2004 11:25 am +0100 Olle E. Johansson
[EMAIL PROTECTED] wrote:
Well, mailing list growth is not only a good thing. It's getting almost
impossible to handle. As I've stated before, we need to change
Asterisk.org so we can help people in a better way and avoid a lot of
--On Wednesday, January 7, 2004 5:24 pm + WipeOut
[EMAIL PROTECTED] wrote:
The GS phones have a setting for Voice Frames per TX with a default
value of 10.. This causes the phone to use a 100ms packet size and
Asterisk is set to use a 20ms pachet size.. The result is a choppy sound
when
You can use the asterisk management interface to query for extension status
etc - see http://www.voip-info.org/wiki-Asterisk+manager+API. You may
need to supply a channel number for the device you want to monitor. This
is usually derived from the name you supplied for the extension in the
It is a problem - but the call recording is saved by * when you hang up.
So you need to look for new files in whichever directory the call
recordings are saved and pick them up eg with a script.
Iain
--On Sunday, January 04, 2004 12:07:35 -0500 Kevin [EMAIL PROTECTED]
wrote:
There was a
--On Monday, December 29, 2003 11:28 am +0100 Cees de Groot [EMAIL PROTECTED]
wrote:
Lubomir Christov [EMAIL PROTECTED] said:
Yes, I know that the Grandstream firmware have problems (I have here 15
phones with some beta version already installed :( and waiting for bug
fixing in the new beta)
--On Monday, December 29, 2003 11:58 am -0700 [EMAIL PROTECTED] wrote:
Lubomir Christov [EMAIL PROTECTED] said:
Yes, I know that the Grandstream firmware have problems (I have here
15
phones with some beta version already installed :( and waiting for
bug
fixing in the new beta) but the
Maybe you just need to dump a file to the spool directory that has your
phone number and an asterisk extension that goes to a voicemail check.
You'd still need to patch app_voicemail to create the call file.
Iain
--On Sunday, December 28, 2003 4:07 pm -0500 Kevin [EMAIL PROTECTED]
wrote:
--On Thursday, December 25, 2003 9:13 pm -1000 Ron Fox [EMAIL PROTECTED]
wrote:
Also, is there a script or makefile target that will fully un-install
asterisk, zaptel, zapata and libpri so that I can try again?
You could install the utility checkinstall. It creates a RPM for
software that
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED]
wrote:
I've got it running through Asterisk - all working fine from a SIP
standpoint. I can dial FWD numbers like 612/613/etc and everything works.
However, if I dial *18005551212 or *408xxx (say, a USA number), I
--On Wednesday, December 24, 2003 6:35 pm +0100 Arnold Ligtvoet
[EMAIL PROTECTED] wrote:
Read the fwd announcement. Jeff Pulver mentioned the fact that * users
cannot use the free holiday calls, since FWD cannot be sure that * is not
being used by more than 1 user at the same time.
Where in
--On Wednesday, December 24, 2003 10:06 pm +0100 rnc Info Lists
[EMAIL PROTECTED] wrote:
Still, there seems to be a you get what you pay for theme to many of
today's posts and this clearly applies to support on FWD. Naybe we
should remove the signature from * that enables FWD to identify *
I tried running the festival app today with little success. I have a
working festival installation that does TTS to the linux sound output
perfectly.
With * it just produces a sort of hissing sound. The length of hissing is
proportional to the length of text string that it is given to speak.
VoIP watchers may like to take a look at this:
http://www.btbroadbandvoice.com/broadband_voice/bb_voice_home.html
BT has launched a consumer VoIP service in the UK using ATA 186s (judging
by the picture). Now if only I could connect the service to my * server
without the ATA
Iain
Not quite - I want to use SIP directly from * - I don't need a locked
ATA186 as a paperweight ;-) That is, assuming BT locks the config as
Vonage does.
Iain
--On Tuesday, December 9, 2003 3:59 pm + Senad Jordanovic
[EMAIL PROTECTED] wrote:
You can!!! :)
Use one of those FXS to FXO
--On Sunday, December 07, 2003 09:36:14 -0500 TeleSIP [EMAIL PROTECTED]
wrote:
Its the VT1000
http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212
We have looked everywhere for it but looks like no distributor sells it
right now.
Maybe because it's a new variant of the
I've been tinkering with ENUM and found that the lack of a debug message in
enum.c that says it has actually succeeded in resolving an address is a bit
of a nuisance. It makes it difficult to see if failures with ENUM are due
to problems with parsing NAPTR records (in enum.c) or mistakes in
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.
AFAIK the 7920 needs CallManager to work - if you haven't got that you'll
have to wait for Cisco to make a general purpose version - or maybe buy a
Pulver phone http://www.pulverinnovations.com/ - assuming that works with
*
Iain
--On Monday, November 17, 2003 6:31 am -0500 Jeremy McNamara
Yes - the aggressive suppressor does tend to clip speech although I don't
think it is half duplex.
The MEC3 echo suppressor seemed to be heading in the right direction but
last time I tried it it went funny after a while causing speech
interruption.
Iain
--On Saturday, November 15, 2003
You'll probably need clean builds of zaptel and asterisk - I tried with
updates earlier today and the echotraining option wasn't recognised until I
did a complete clean install.
Iain
--On Saturday, November 15, 2003 13:59:13 -0800 Andrew Gillham
[EMAIL PROTECTED] wrote:
Andrew Joakimsen
I have been running asterisk on an old PowerMac 9600 and YellowDog Linux
for about a year now. Asterisk software builds fine most of the time -
there seem to be some trivial issues with the Makefiles for codecs at the
moment.
I have an X100P card as the PSTN interface. I suspect that the
I'll own up to a patch - bug report 110. However, Mark peremptorily
dismissed my suggestion putting forward a solution I find illogical. I
guess more people need to ask for this feature!
I think my original patch was a bit over-engineered. The one below is
simpler.
Iain
---
--On Monday, August 18, 2003 10:31 pm +1200 Roger De Salis
[EMAIL PROTECTED] wrote:
Interesting menu options implying mechanisms to take the 11
channels of WiFI, and dedicate 1-3 for voice, and turn the
rest over to data. There were some rumours that they only
work on Cisco Aironet base
It should work with the standard PSTN but you can get problems if you
connect a X100P to a PBX or ISDN TA. Try editing the wcfxo.c file and
enabling support for ZERO_BATT_RING (uncomment the #define) then rebuild
and reinstall the zaptel modules - you will need to unload and reload the
wcfxo
Assuming this is on incoming calls, the most usual source of the problem is
that the telco exchange either doesn't send a disconnect pulse or the wcfxo
driver doesn't recognise the format used. I've unfortunately forgotten the
exact situation but, when a call finishes, a telco exchange in the
I was in a call through an ATA 186, * and the PSTN today when someone
dialled me over FWD. I got a tone in the earpiece more than once which was
jolly annoying. Is this the problem you're getting? I think an option to
turn this tone off is needed.
Iain
--On Thursday, August 7, 2003 7:51
--On Thursday, August 14, 2003 12:58 pm +0200 Dave Cotton
[EMAIL PROTECTED] wrote:
Last night I posted showing that the problem is repeatable and only
occurs in one certain circumstance. I think it is within voicemail.c. If
the caller exits voicemail by pressing # the line is dropped
The chipset used in the X100P - at least the one I have - is designed for
the US/Japan market only. The reference design in the datasheet for the
chipset does not include facilities for the detection of line voltage
reversal. Hence the only way to detect caller ID sent before ringing would
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