On Monday 17 March 2003 05:07, Andres Tello Abrego wrote:
Does anyone over here use spamassasin, I belive the * list messages get
treated as spam...
I've set up spamassassin to put spam in a separate folder, and I don't think
I've ever seen any * mail go there
--
Roy Sigurd Karlsbakk,
1 DS3
How many Dell PowerEdge 2600 PIII 1.13GHz (could add a second
processor), gig RAM
Would I be able to put 3 T400P in each server. Or is that too much? if
so, I better buy a couple more 2600's.
Asterisk will function as an PBX/IVR
I have 2 T1's I can play with for the next couple of weeks
Carlos Crembil wrote:
Hi guys,
I'm new here, so, greatings for all... (i'll give you the candies in a
future meeting :-).
I've installed asterisk and opengk in my server, and I'm in the
experimenting phase. Also I have a Cisco 800 series to play (4 FXS
interfaces), and a netmeeting client.
My
According to James Sizemore:
Problem will all XML configs:
1. They are nearly imposable for a human to read, for any non trivial
config.
It's about like reading HTML source -- it doesn't have prose-like flow, but
it can be read and understood without a complex editor. In my experience,
Hi,
I have some problem with ztdummy in order to use Conference in asterisk.
When I do modprobe ztdummy I have this :
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
proc_mkdir_Rsmp_220b03b4
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
create_proc_entry_Rsmp_3a9bfbd2
On Tue, 2003-03-18 at 00:10, Chris Albertson wrote:
The best thing might be to seporate the two types of data.
The simple thing to do is use a preprocessor like M4.
Defin a macro for a type-D phone and then have lines like
extn-type_D(6578)
extn-type_D(6579)
Then to change the flw
Call it a RDBMS, LDAP server or even an NIS map.
Asterisk is designed to support multiple switch backends, so you can put a
statement like this in your dialplan:
switch = IAX2/host/context
Which pulls in extensions from another remote host. But just as easily
one could create a switch for
Oops, the previous patch had a little
bug... now it's fixed. please use this one.
Matteo
Il lun, 2003-03-17 alle 18:06, Matteo Brancaleoni ha scritto:
I've a sip phone that when called,
if not answered and you hangup
on the caller side, sends out
487 Request Terminated .
But seems that *
In case anybody is interested, here's a patch to store voicemail
passwords as an md5sum in the astdb. Change the first character
of the password field in voicemail.conf to a '!' in order to use.
-Tilghman
Index: apps/app_voicemail.c
Matteo, your fix looks like a good temporary solution, but I don't want to
merge it with CVS because the *right* thing that I *need* to do is to
implement retransmissions on SIP. Once retransmissions are in place, then
we can keep the channel around until we receive the 487 back that we
expect.
We have need of a ringdown circuit in an elevator. If someone picks up
the phone, it should dial another extension without any keys being
pressed. (There are no keys on the phone)
If it was an incoming call to asterisk, the following lines in
extensions.conf would do the trick.
exten =
What I would like to see would be a configurable option the admin could
enable a SOAP client in Asterisk that would make calls to a SOAP server
requesting the various pieces of information he needs. The SOAP server
would have the DB connection(s) (or other back ends) to store
persistent data.
Many SIP phones will not pass audio until the call is answered. This
means that you may need to explicitly Answer() before doing Dial, etc.
Further, MusicOnHold is based on sending audio at the rate it is
*received*.
Mark
On Tue, 18 Mar 2003, Masakazu Nakano wrote:
I was tried following.
If XML is important to your needs, why not write a translation script to
parse XML and write the asterisk configs? Scripting languages abound and are
appropriate to the task. Obviously, the transaltion script could grab your
XML and write fresh asterisk configs every time you started asterisk.
use immediate=yes in zapata.conf
matteo
Il mar, 2003-03-18 alle 16:04, Don Pobanz ha scritto:
We have need of a ringdown circuit in an elevator. If someone picks up
the phone, it should dial another extension without any keys being
pressed. (There are no keys on the phone)
If it was an
set it to immediate mode in zapata.conf
I have a door phone (real phone but keypad ignored) that does exactly the
same thing.
At 09:04 AM 3/18/2003 -0600, you wrote:
We have need of a ringdown circuit in an elevator. If someone picks up
the phone, it should dial another extension without any
On Tue, 2003-03-18 at 08:21, Rattana BIV wrote:
Hi,
I have some problem with ztdummy in order to use Conference in asterisk.
When I do modprobe ztdummy I have this :
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
proc_mkdir_Rsmp_220b03b4
/lib/modules/2.4.7-10/misc/zaptel.o:
You have to add
immediate=yes
to zapata.conf to the declaration of this channel.
Then right after someone picks up the phone asterisk will just
right to 's' extension of the specified context.
regards
Martin
On Tue, 18 Mar 2003, Don Pobanz wrote:
We have need of a ringdown circuit in an
That worked.
I had tried setting immediate with some other settings but I guess
never to 'yes'.
Thanks Tilghman.
Don Pobanz
On Tuesday, March 18, 2003 9:55 AM, Tilghman Lesher
[SMTP:[EMAIL PROTECTED] wrote:
On Tuesday 18 March 2003 09:04, Don Pobanz wrote:
We have need of a ringdown
On Tue, 2003-03-18 at 09:04, Don Pobanz wrote:
We have need of a ringdown circuit in an elevator. If someone picks up
the phone, it should dial another extension without any keys being
pressed. (There are no keys on the phone)
If it was an incoming call to asterisk, the following lines in
I know this isn't really related to *, but I thought some of you might
be able to shed a little light.
I have a 7960 running SIP. No matter what I send the phone, I get
CMXML parse errors. I have the services_url set to a perl app that I
keep changing with no luck. I have set the correct
Something bad seems to happen after speaking for ~30 minutes. After that, I
receive really bad sound, with glitches and 'brzts'. I just got it twice
today (yeah - I was doing support. sheit). It works fine if I hang up and
dial again.
I'm using MGCP phones and AVM C2 with CAPI with
hi. read my mail 'bout 487 response.
I wrote a patch to fix that in chan_sip .
It's good for a occasional fix, until
mark updates chan_sip to handle retransmissions.
matteo.
Il mar, 2003-03-18 alle 16:58, Christoph Frei ha scritto:
Hello Guys,
i'm using a Cisco 7960 and a Cisco 2610 with a
Hi,
Asterisk is not sending the CANCEL when you hang up back to the gateway
that Matches the ORIGNAL INVITE Request.
It looks like it was fixed with the last cvs release.
Dave
On Tue, 2003-03-18 at 10:58, Christoph Frei wrote:
Hello Guys,
i'm using a Cisco 7960 and a Cisco 2610 with a
The problem is here:
if ((first=strchr(cp3,':'))) {
*first='\0';
offset=atoi(first+1);
if ((second=strchr(first+1,':'))) {
*second='\0';
offset2=atoi(second+1);
} else {
It's fixed now.
what's the intended behaviour of ${variable:a:b}?
it's the same as substring application
given that ${exten} = 501234
until yesterday ${exten:2} would give '1234'
and it does now
with current CVS ${exten:2} is '50' while ${exten:2:4} is '1234'
how do I just strip
His 481 issue is not the same as your 487 issue.
Mark
On 18 Mar 2003, Brancaleoni Matteo wrote:
hi. read my mail 'bout 487 response.
I wrote a patch to fix that in chan_sip .
It's good for a occasional fix, until
mark updates chan_sip to handle retransmissions.
matteo.
Il mar,
There appears to be an Asterisk WebMin module on the Digium FTP site, in
some state of development. Is this being officially developed?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stefano Finetti
Sent: Monday, March 17, 2003
On Tue, 2003-03-18 at 11:00, Stefano Finetti wrote:
Wow...
I really didn't mean to create this kind of discussion ;-)
Anyway, i think that the best way without having Mark change something in *,
to make the php script that can with an appropriate set of classes, able to
read the
There appears to be an Asterisk WebMin module on the Digium FTP site, in
some state of development. Is this being officially developed?
No, it hasn't been developed in at least 12 months. If anyone wants to
play with it they're welcome to.
Mark
So far, the overwhelming majority of comments both on and off the list
seem to indicate that the most requested feature is reliability. No
surprise there. :-)
I understand that many folks are frustrated with the lack of technical
support and help for power users being offered by existing
On Tue, 2003-03-18 at 11:35, WipeOut . wrote:
Hi,
What utilities do you guys use to record your IVR messages??
I know there is the record application in * but I was wondering
if there were any freeware Windows or Gnome utilities that would
make it a lot quicker to record a large number
On Tue, 2003-03-18 at 09:35, WipeOut . wrote:
Hi,
What utilities do you guys use to record your IVR messages??
I know there is the record application in * but I was wondering
if there were any freeware Windows or Gnome utilities that would
make it a lot quicker to record a large number
On Tue, 2003-03-18 at 12:24, [EMAIL PROTECTED] wrote:
¡Hola!
We're trying to build a Zapata ISA card and we're having problems finding
the oscillators (12.352 and 16.384 MHz), do you have part numbers for
these?
With the total lack of new hardware supporting ISA, do you think it wise
to
Anybody know the syntax to retrive voicemail using the pingtel xpressa
phones?
I know the parameter is
PHONESET_VOICEMAIL_RETRIEVE=
What I don't know is - what value to set the parameter.
So, when a user pushes the new voicemail button it calls the asterisk
general voicemail.
some examples
check here
http://www.metadecks.org/software/sweep/
---
Masakazu Nakano as [EMAIL PROTECTED]
On Tue, 18 Mar 2003 17:35:39 +
WipeOut . [EMAIL PROTECTED] wrote:
Hi,
What utilities do you guys use to record your IVR messages??
I know there is the record application in * but I was wondering
hi all
is there anyone out there that knows the fcc cert numbers for the Wildcard
T400P or the original Tormenta 2 PCI Card. if we're going to use these
cards in production enviroment we'll need them.
thanks dwayne
Does anyone know if this can be done by any VoIP Technology (SIP, IAX,
IAX2 or MGCP) I don't know the protocols!
On Tue, 2003-03-18 at 09:56, Steven Critchfield wrote:
On Tue, 2003-03-18 at 09:04, Don Pobanz wrote:
We have need of a ringdown circuit in an elevator. If someone picks up
the
I believe the Cisco ATA-186 supports it, but you'd have to do more
digging on their site.
This is really not a protocol issue, but a vendor programming issue.
It all depends on if you can get the hardware to do a hotline call
when the phone is taken off the hook.
JT
Does anyone know if this
Hi,Steve,
I think VCXO and the crystal are the different things.
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 05, 2003 4:41 AM
Subject: Re: [Asterisk-Users] Re: Tormenta ISA E1 card
Hi John,
That is correct. Note
I don't think to build the PCI version of Tormenta is a easy thing either.
We'd better buy it from Digium .
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, March 18, 2003 5:18 PM
Subject: Re: [Asterisk-Users] part number for OSCs
¡Hola!
We plan to
We could implement a ringdown on MGCP or IAX, but SIP would not support
such a system.
Mark
On 18 Mar 2003, Stephen Webb wrote:
Does anyone know if this can be done by any VoIP Technology (SIP, IAX,
IAX2 or MGCP) I don't know the protocols!
On Tue, 2003-03-18 at 09:56, Steven Critchfield
At 15:45 18-3-2003 -0700, you wrote:
You use the browser-based interface to specity a complete SIP URL for
retrieving messages from the target mailbox. Some examples follow.
sip:[EMAIL PROTECTED]
sip:[EMAIL PROTECTED];function=retrieve
sip:[EMAIL PROTECTED];msgId=4
To determine
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