You have to Answer the exten first.
bkw
- Original Message -
From: Freddy Setiawan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 11:23 PM
Subject: RE: [Asterisk-Users] please mail me wave.cc and tts.scm
i'm using RH9. Yes, after you send the file, i'll recompile
Hi Hekuran,
You can post an example account how did you populate the database on it?
thanks
regards
- Original Message -
From: Hekuran Doli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 9:23 AM
Subject: [Asterisk-Users] Fixed! Midifyed-Prepaid-Application
Hi
Jay Milk wrote:
This question isn't entirely Asterisk related, but I'm hoping that
someone here may have the knowledge to respond to me anyway. I'm
using Asterisk with several Sipura SPA-2000 SIP devices as FXS
adapters. I would like to have my SPA's automatically provisioned
through http
On Jun 20, 2004, at 9:16 PM, [EMAIL PROTECTED]
wrote:
I am new to Asterisk, but though that I would need calls being
answered in
different contexts.
How can I direct one line to a given context and the other one to
another,
or is there a better way???
I use this at the end of my zapata.conf
Hi all
I simply added a resller account then cardtype cid and card I also filled
the order and sale table. I didnt populate the providers* tables because I
didnt need it. Populated the country, countryprefix to. keep checking ID`s
couse if you give wrong ID to tables you cant link them!
Try this
Yes, the telewest side (span1) is cpe and the GDK side (span2) is net.
I didn't bounce it, but I did stop asterisk, stop zaptel (i.e. unloaded the
modules) then reloaded and restarted.
I left it over the weekend, and am about to check the logs now, I get
evenings and weekends to do the
On Mon, 2004-06-21 at 11:40, Nik Martin wrote:
Adam Goryachev wrote:
On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote:
So, if someone could brief me on the GPL issue, and (perhaps someone
else) offer a distribution point, it's free for the asking, VB sources
and all.
Stephen R.
On Sat, 2004-06-19 at 19:27, Storer, Darren wrote:
Hi Kevin,
KW By the way, it's useful to map 911 and 112 onto your 999
KW route for the benefit of foreigners who don't know any better.
Well, while you are at it, you might as well add-in 000, because that is
what we use.
(BTW, why is it
Hi Adam,
AG (BTW, why is it that people used 000, 999, 911, etc for EMERGENCY
AG calls (every second counts) when we used to dial from rotary dial
AG phones, where dialling a 0 or 9 takes a long time compared to
AG dialling a 1Why didn't we all use 111, or something similar?)
In the UK
Here's a diff that will patch it properly.
-Original Message-
From: Damian Minkov [mailto:[EMAIL PROTECTED]
Sent: 21 June 2004 10:14
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem compiling fax applications
I'm tring to compile fax applications on Debian system.
the spandsp
At 18:58 21/06/2004 +1000, you wrote:
On Sat, 2004-06-19 at 19:27, Storer, Darren wrote:
Hi Kevin,
KW By the way, it's useful to map 911 and 112 onto your 999
KW route for the benefit of foreigners who don't know any better.
Well, while you are at it, you might as well add-in 000, because
Hi all,
Is there a way of disabling ALERTING message on a PRI
channel? I have a problem .* is sending ALERTING message to the Nortel telco switch
of my local provider BEFORE it dial the number it has to .if the number is busy
or invalid there is no way we can tell this to the switch
Hi all,
I've been through the wiki and the archives and I've been unable to find what I'm
looking for. Basically I have a phone that I don't want to be able to dial out. I've
only got one context at the moment; but from my investigations I think I might need to
create another for this
That sound's like the right thing to do, you'd probably have contexts
related to what phones could do
unrestricted,localdialling,extensionsonly and then within those
contexts include the relevant contexts.
If you look at the sample configs you can see how this is done for the
international and
Hi,
Has anyone written a management application
(WinXP)which can display simple QueueStats on a
monitor.
I would like to see;
No of Calls currently in the queue
Hold time
Completed
Abandoned
I know the info is available via API,'Action:
QueueStatus', I just don't want to re-invent the
On Mon, 21 Jun 2004, Matt wrote:
I've been through the wiki and the archives and I've been unable to find
what I'm looking for. Basically I have a phone that I don't want to be
able to dial out. I've only got one context at the moment; but from my
investigations I think I might need to
content extensions.conf of mine :
[local]
exten = 4001,1,Dial(SIP/4000)
exten = 4002,1,Dial(SIP/4001)
exten = 555,1,Answer
exten = 555,2,Festival(good morning)
exten = 555,3,Wait(2)
exten = 555,4,Hangup
should be alright.
Regards,
Freddy Setiawan
::Simple is Everything,Nothing is Complex::
I've got a Digium TE405P feeding three Carrier Access AB-II units located
in three separate suites of the complex my company resides in. I've had
one AB-II connected over about 15' of cable and running for several
weeks now, and it seems to be perfectly happy.
Over the weekend, I added two more
On Mon, 21 Jun 2004 10:53:42 +0100, Matt [EMAIL PROTECTED] wrote:
Hi all,
I've been through the wiki and the archives and I've been unable to find what I'm
looking for. Basically I have a phone that I don't want to be able to dial out.
I've only got one context at the moment; but from
Title: Message
This
is a known issue that needs fixing URGENTLY...!
Markster, please can you have a look at this asap? I can call you
if you need further info on this aspect of operation - it is a major issue for
us and if we had more time we would fix it...but I think you are probably
I'm trying to compile cdr_addon_mysql but keep getting this error.
Again, searching the Wiki and the mailing list archive didn't bring up
anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to
switch back to 3.23?
# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
Hi,
Another helpless person like me, I had the same problem a few days ago and a very
helpful person suggested putting CFLAGS+=-I../asterisk/include in the Makefile,
which worked fine for me.
Maybe this is a problem with asterisk CVS ??? Don't really know but it might be
worth looking at for
Looks like half a patch has been applied to app_dial in cvs head could
someone with commit rights fix it.
Thanks
Chris
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Yeah its broken... i just re-checkout the * then try to compile and it show
error
regards,
Freddy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton
Sent: Monday, June 21, 2004 8:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] app_dial
Hello,
Have you checked the CPU load on your asterisk server?
If you have no change of codec between the two sides, asterisk just
pasts the audio straight through. If there is a change, conversion has
to take place and conversion from g729 to anything else does require cpu
power.
Another
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 21 June 2004 14:22, Chris Stenton wrote:
Looks like half a patch has been applied to app_dial in cvs head could
someone with commit rights fix it.
Here's a patch from Michael Manousos on the Asterisk developer list, which
fixes the
Senad Jordanovic wrote:
Jay Milk wrote:
This question isn't entirely Asterisk related, but I'm hoping that
someone here may have the knowledge to respond to me anyway. I'm
using Asterisk with several Sipura SPA-2000 SIP devices as FXS
adapters. I would like to have my SPA's automatically
I cant use this t410 card to send calls to my telcos
switch bse the network manager is complaining about this message being
sent to their switch saying that the call is going on even before the number is
dialed
Thanks
Title: Message
Please
can you send debug traces with comments so we (and other wiser members of the
group) can see what is going on?
Thanks
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AimableSent: 21 June 2004 14:18To:
[EMAIL
Thanks Andrew, I will give it a try.
I assume I can put channel = x for each context.
Have a nice day,
Francois
In zapata.conf include the line
context=contextname
for each port on the card.
Andrew
_
Andrew Yager
Real World Technology Solutions
Real People,
On Mon, 21 Jun 2004 14:24:06 +0530, Navnit Chachan [EMAIL PROTECTED]
wrote:
Hi,
I am not sure whether this is the right forum but anyway am posting my woes.
I am trying to compile iaxclient on win2k.
The iaxclient-devel list is three doors down on the left. You can subscribe
here:
Hi all,
Odd... I did a make update and how the MailboxExists works fine.
However, it works just as the docs say: add 101 to priority if the box
*does* exist, add 1 if not. I have tested it and this seems to be how
it works. You may wish to test your flow and make sure it works as
you
| Since caller ID does not work with my FXO card, I am wondering if Asterisk
| supports the following extensions functionality.
|
| When a call comes in, I'd like to give the caller an opportunity to enter
an
| extension if he/she knows it; if not, Asterisk will dial one or more
default
|
Not sure I understood the comments... the existing situtation (as shown in
the original email) is a remote 7960 uses g729, and an internal 7960 that
uses g711 (both forced via sip.conf entries). The audio is oftentimes
choppy, but changing the g729 side to g711 (still passing through * with
Stephen R. Besch wrote:
I've just finished a general purpose configuration utility for the GS
phones:
1) Generates files from scratch (using MAC), from HTML config
listing, or by directly downloading from the phone.
2) Does multiple simulteneous edits.
3) Can reboot as many or as
hi;
I compiled asterisk including the zaptel folder without any problem on a
redhat 9.0 machine but no i am trying to compile it on a mandrake 8.3
machine but the zaptel make fails with error regarding the veriosn.h file.
I think am missing some linking option or possibly some libraries. Any
Because whenever I place call from the phone without it going through the
prepaid application, I don't have any audio issues and the route between the
gateway and the phone is built correctly with NAT taken into account. When
the call goes through app_prepaid, a new dial command is issued via
Within the last week or so there were several changes in CVS related to
alerting
On Mon, 2004-06-21 at 05:52, Robinson Tim-W10277 wrote:
This is a known issue that needs fixing URGENTLY...!
Markster, please can you have a look at this asap? I can call you if
you need further info on
Any suggestions on how to resolve this problem? :)
Thanks,
Wojtek
- Original Message -
From: Aaron J. Angel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 12:01 AM
Subject: RE: [Asterisk-Users] enum problem with latest cvs
Try RFC3761. It specifies E2U+spec
I'm configuring call forwarding and I want to allow the user to enter a
call-forward number of any length, followed by the # key.
I have tried _X.#, but that will only end when I get a digit timeout.
Presumably, this is because the . will match the pound and it is
waiting fro more input.
Thanks for the tip, but adding the CFLAGS directive doesn't work either, same error
message. I'll try to have a look in -dev, but if anyone comes up with a solution, a
reply would be appreciated.
-Manuel
-Messaggio originale-
Da: Luckcuck Nick-LCKN001 [mailto:[EMAIL PROTECTED]
Hi folks,
I use safe asterisk to startup and run asterisk in the background. In
Safe_asterisk script, there is a parameter (right at the top ), CONSOLE
which I can set to no or something. If it is no asterisk startup as
asterisk -vvvg , if it is set to something the asterisk startup as
asterisk
Hi there,
I just give you the script (in Python) I have just written in case of
someone would like to implemant this. I think it is more simple than the
one we can see over the net... It uses DISA (security issues == limit
access with contexts and the password !!) and CAPI but it should work
Holger Schurig wrote:
Unless someone does something serious about the flakiness of libtiff, I
don't think either spandsp or Hylafax will ever be very stable. :-(
Delete the word unless.
And then create a subdirectory spandsp/tiff where you put a libtiff into
it that actually works. Create
Senad,
I would appreciate your help -- I'm running 2.0.6(c) for the most part,
but I have one box which is still on 1.0.30 -- but that I can upgrade
easily. I'm certain I can work it out from the admin guide (for 2.0.x),
which doesn't seem available to the end-user.
Regards,
-- Jay
On Monday 21 June 2004 11:02, Steve Underwood wrote:
That isn't what I meant. libtiff is really flaky when you give it a bad
tiff file. It used to core dump on almost any fault. I fed updates back
to the libtiff guys a few years ago, which I think went in. Others must
have contributed most
Hi!
I'm configuring call forwarding and I want to allow the user to enter a
call-forward number of any length, followed by the # key.
...
Is there a straightforward way to do this?
show application read
http://www.voip-info.org/wiki-Asterisk+cmd+Read
Cheers, Philipp
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Holger Schurig wrote:
|He says he will release the sourcecode when he gets to a stable
|working release.
|
|
| He said the same to me, almost a month ago before his holiday ...
|
| However, I confess that QtIAX didn't come forward during this month
Lee Howard wrote:
I've never seen this kind of flakiness of libtiff cause any problems
for HylaFAX. As far as I'm aware, there has only been two instances
when libtiff caused HylaFAX any grief. The 3.6.1 release problem with
G3/G4 is a given. And then there was the 16-to-32 bit type change
Philipp von Klitzing wrote:
Hi!
I'm configuring call forwarding and I want to allow the user to
enter a call-forward number of any length, followed by the # key. ...
Is there a straightforward way to do this?
show application read
How long has this application been around?
Hi all,
I have got the following problem (E100P, pri_cpe):
My number range is 6digitsxyz. (e.g. 123456-999)
From ISDN phones, everything's fine, but calling in from analogue phones causes the
following problem: Asterisk only receives the first 6 digits.
In zapata.conf, I have
immediate = no.
Is there a means to globally disable displaying the
double-quotes on the caller id sent to a device, SIP, SCCP and analog?
Thanks
I'm using this scenario have had no problems.
When you say that HT-286-2 begets silence, do you mean it does not ring,
or that when you pick up after the ring there is no audio?
-g
On Fri, 2004-06-18 at 14:31, Nathan Martinez wrote:
I have 2 Grandstream HT-286 devices and an Asterisk
Hi,
I had followed the installation-guide to festival
http://www.voip-info.org/wiki-Asterisk+festival+installation
speech-tools compiles OK, but I got this error when compiling asterisk
if I compile without the patch it compiles, but of cause did'nt work with
asterisk.
any clue ?
/Hans-Henrik
On Jun 21, 2004, at 11:19 AM, Philipp von Klitzing wrote:
I'm configuring call forwarding and I want to allow the user to enter
a
call-forward number of any length, followed by the # key.
...
Is there a straightforward way to do this?
show application read
Hi Thomas,
I have got the following problem (E100P, pri_cpe):
My number range is 6digitsxyz. (e.g. 123456-999)
From ISDN phones, everything's fine, but calling in from analogue
phones causes the
following problem: Asterisk only receives the first 6 digits.
Do you have overlapdial=yes in
It doesn't matter what facilities you use. If you get libtiff to open a
bad TIFF file, it may well dump core. TIFF has been one of the most
abused standards over the years. It used to be that hardly any FAX
software would read the TIFFs produced by another package - both because
they were
I have got the following problem (E100P, pri_cpe):
My number range is 6digitsxyz. (e.g. 123456-999)
From ISDN phones, everything's fine, but calling in from analogue
phones causes the
following problem: Asterisk only receives the first 6 digits.
Do you have overlapdial=yes in your
In article [EMAIL PROTECTED],
Manuel Wenger [EMAIL PROTECTED] wrote:
I'm trying to compile cdr_addon_mysql but keep getting this error.
Again, searching the Wiki and the mailing list archive didn't bring up
anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to
switch back
Just a quick question. I setup Directory dial by name, and I read it looks
at the Voicemail config to determine who you want to connect to. The thing
I dont like is when it finds a match it reads the extension instead of their
name. Is there a way to have it read the name in the voicemail
If the user to whom that vm is assigned goes through the setup process
records their name, that is played.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Harold Workman
Sent: Monday, June 21, 2004 11:04 AM
To: [EMAIL PROTECTED]
Directory only reads the number if the voicemail user has not recorded his
name. If the name has been recorded, it plays that, instead.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 21 Jun 2004, Harold Workman wrote:
Just a quick question. I setup
Do you have access to a T-1 analyzer? You more than likely have a 'dirty'
T-1 line that is out of spec based on the length of the run.
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bonzo Armstrong
Sent: Monday, June 21, 2004 5:43 AM
To: [EMAIL
Bruce Komito wrote:
Directory only reads the number if the voicemail user has not
recorded his name. If the name has been recorded, it plays that,
instead.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 21 Jun 2004, Harold Workman wrote:
Just
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote:
Do you have access to a T-1 analyzer? You more than likely have a 'dirty'
T-1 line that is out of spec based on the length of the run.
Sadly, none that I'm aware of, but I'll ask around. I could probably
find a decent scope to put
[Please accept my apology for posting this to the user list.* Please do not
reply on-list to this message.]
Would anybody on the list be interested in doing Asterisk as a career?
Sokol Associates, LLC is seeking qualified applicants to serve as full
time and contract Asterisk consultants,
J Which voicemail is current and latest?
J Voicemail
J or
J Voicemail2
I think Voicemail ist the latest.
Greetings Gunnar
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Hi there,
I tried to get a few Optipoint 400 SIP working with *, but it refused to work
properly.
In my testing-network i have two Sjphones (they are working really fine) and
three optipoints.
I´m able to dial the number of a Sjphone on all of the optipoints.
The call is signalled at the Sjphone
I have looked on the wiki and in the archives for as much info as I can
find about call forwarding.
I have it working based on the sample from TipsTricks except for
handling voice mail.
When I detect a forwarded extension, I call
Dial(Local/[EMAIL PROTECTED]/n,20) to dial the number as
As Felix said: put overlapdial=yes in zapata.conf. Also, make sure you
don't have a match on 123456 (or anything less specific), if you want to
match _123456XXX.
Thilo
On Mon, 2004-06-21 at 19:41, Thomas Schroeter wrote:
Hi *,
I can specify my question:
When calling from analogue phones,
Anyone seen this one before?
Asterisk Box (A):
ATA186 SIP - Asterisk
Asterisk Box (B)
Asterisk - ZAP E1
Boxes linked by IAX2.
ATA user places call to PSTN via box a b. Call fails and user hangsup.
Box A CDRs reports call duration = 2 minutes
Box B CDRs reports call duration = 3 hours
How
I'm looking for a way to give VoIP capabilities to an existing PBX: it's
made by Mitel and it's used in a small/medium environment (a few dozen
phones, but the PBX has capabilities for up to 200, if i remember
correctly).
Any high-level guidelines on how to integrate Asterisk with a PBX that's
Hi!
I have updated the optipoint to the last software version
I can Call the optipoint from other phones and talk.
The optipoint register with asterisk but in the phone display i have
only no server. and no dial tone.
The only way to register was with no password to the optipoint
On Monday 21 June 2004 15:19, Florin Andrei wrote:
Any high-level guidelines on how to integrate Asterisk with a PBX that's
already in use? Probably that particular PBX is not supported directly,
but are there ways to somehow hook-up to it and route calls from IP to
PBX or back?
Depending on
Using a real world example, lets assume you want to add 12 extensions and 12
phone lines that process through the asterisk box. I use those numbers
because they add up to 24 which is the number of channels a single T1 is
capable of signaling. Your hardline PBX would then wire into a channel bank
Here is a good place to look. You either need analog FXO or FXS ports.
http://www.voip-info.org/wiki-Asterisk+legacy+integration
I also have a little info on my site.
Dear Steve
after a lot of compile from spanDSP first version to K and after i have
compile * HEAD and Stable i cant understand where im wrong...
- Mandrake 9.2
- libtiff3-3.5.7-11mdk
I have add Header from your FTP but the result is this error..
Can you give me your opinion?
Thanks in
Any easy way to make festival read the name?
- Original Message -
From: Harold Workman [EMAIL PROTECTED]
To: 'Bruce Komito' [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 1:24 PM
Subject: RE: [Asterisk-Users] Directory dial by name
Bruce Komito wrote:
Directory only
Hi all, I'm looking the way to connect 16 E1/T1 to my * server. How it would be possible ? . Does anyone have already this experience ? I'm thinking to do using two * machine servers. One card E400XX or T400XX from digium on each machine and interconnect one each other using IAX. One * as a
On Monday 21 June 2004 15:57, Angel Diaz wrote:
I'm thinking to do using two * machine servers. One card E400XX or T400XX
from digium on each machine and interconnect one each other using IAX. One
* as a master, and the other which will connect to this (master) to use its
numbering plan... I
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote:
Do you have access to a T-1 analyzer? You more than likely have a
'dirty' T-1 line that is out of spec based on the length of the run.
Sadly, none that I'm aware of, but I'll ask around. I could probably find a
decent scope to
Hi Brian, I have been using a X100P to Packet8 ATA connection for about
3 months, it works fine apart from needing the occasional reset.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Weaver
Sent: Tuesday, 22 June 2004 6:16 AM
To:
On Jun 21, 2004, at 1:16 PM, Brian Weaver wrote:
Question is this, I know the ATA with packet8 is locked down, but is
there any reason I can't use it just like a regular POTS line with
Asterisk if I buy a X100P card? That way I could pick up a SIP device
to talk to Asterisk, and configure the PBX
On Mon, 21 Jun 2004, Angel Diaz wrote:
Hi all,
I'm looking the way to connect 16 E1/T1 to my * server. How it
would be possible ? . Does anyone have already this experience ?
I'm thinking to do using two * machine servers. One card E400XX or
T400XX from digium on each machine and
Hi Stewart, I've tryed all the options for connectionmode, also the
connection progress in cisco, but it didn't work. The way that I solved it
was changing the asterisk source to instead of sending Session Progress
when the call is in progress to send Ringing. Because for the users In
progress
On Tue, 22 Jun 2004, Dean Collins wrote:
Hi Brian, I have been using a X100P to Packet8 ATA connection for about
3 months, it works fine apart from needing the occasional reset.
Working well here also, 2 Packet8 ATAs and no reset necessary so far.
___
I have just get an account on Iaxtel.com, and i woud like to know what can i do to
receive my Iaxtel calls in my asterisk server?
Actually i just can make IAX calls.
Thanks
--
___
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Powered by Outblaze
Hi Charles,
Perhaps I misunderstood your situation.
I thought that your original problem was: Calls from a SIP phone
through Asterisk to an ATA or a 5300 did not provide audible
ringing to the caller when the called party was being alerted.
I also assumed that some calls made through the 5300
I signed up with fonality and so far they've been awesome compared to vonage and
voiceglo (the quality is SWEET, maybe I'm just used to vonage)
I called their 877 number too and actually got to talk to a knowledgeable person...I
asked them for a feature and they added it the same day. BTW,
I've searched WIKI and Archives but nothing.
Im getting:
-- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call
rejected by 69.73.19.178: Unable to negotiate codec
-- Hungup 'IAX2[Iaxtel]/8'
== No one is available to
Has anyone figured out how to get the T100P to handle a channel bank that
does the SLC-96/TR-08 variant of T1 framing?
g.
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Kyle Hagan wrote:
I've searched WIKI and Archives but nothing.
Im getting:
-- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read:
Call rejected by 69.73.19.178: Unable to negotiate codec
-- Hungup 'IAX2[Iaxtel]/8'
== No one is
Hi All,
Do any of you know what the status is for VoiceXML support in * ? Is it
already existing, or is it planned for the future? If it's not in now,
do you know on what type of scale the work would be to integrate VXML
into * ?
Thanks in advance
Gunnar Schaller wrote:
J Which voicemail is current and latest?
J Voicemail
J or
J Voicemail2
I didn't want to reply to the original post with the answer, because:
* This question has been answered numerous times already.
* The poster MESSED UP THREADING by replying, erasing the body, and
Hi all:
I have problems to setup my zaptel E100P hardware.
When I start * after receive the "Asterisk Ready" I see this:
*CLI Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 1
Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event:
This could do with someone from the * community
posting some info on what asterisk does and how good it is!
:-)
http://ask.slashdot.org/askslashdot/04/06/22/0022230.shtml?tid=126tid=137tid=185tid=215tid=95
Matt
Hi,
I've posted this question on OpenSS7 mailing list but I didn't get a
clear answer. I have a system running asterisk using Digium T400P card. Can
this same card be used with OpenSS7 project to carry SS7 links? Is there any
difference between these two cards?
--
Matteo D'Amato
Hi,
VoiceXML support would be great, but I know of any active work on it.
openVXI seems to have spri=ung to life again recently, after years of
languishing. Perhaps it would form a sound base to get VoiceXML up and
running in a reasonable time.
Regards,
Steve
Asterisk User wrote:
Hi All,
Do
I'm thinking to buy an Adtran TSU 600 with 24 FXS ports and I'd like to know
if is it a good business ?
Kind regards,
Miguel
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All,
I was a bit too focused on where I thought the problem was - turns out
I wasn't crazy and the dialplan does work as expected. The problem was
with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for
the premature post for help.
Begin forwarded message:
From: Ben Witso [EMAIL
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