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, Data
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
Title: Message
Just got the last
CVS (Asterisk CVS-03/17/03-10:01:18) which works fine except that sip show
registry now shows nonsense characters (wrong pointer to the
strings?)
BTW: thanks for
cleaning up the $(PROC) settings in the Makefiles! I can now tell gcc to make
code for my Via
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 actua
I was tried following.
exten => 8819,1,Playback,test ; test is my sample voice.
exten => 8819,2,SetMusicOnHold,loud
exten => 8819,3,WaitMusicOnHold,10
exten => 8819,4,Dial,sip/mack|10|mtT
WaitMusicOnHold is works well.
but
Dial,sip/mack|10|mtT isn't sound.
Is it incorrect usage?
I
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 experien
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
/lib/module
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 chang
Ok guys, that's it a patch for chan_sip.c
The problem is: some devices transmit late packets,
even when asterisk has already destroyed the channels.
For example 487 is sent by various ipphones , like
kphone, snom & some fxs-sip gw) when a request
is cancelled by the caller (for eg. when the
caller
> 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//
Which pulls in extensions from another remote host. But just as easily
one could create a switch for MySQL, LDAP,
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 tha
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. T
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 => s,1,Ans
Mark Spencer wrote:
Asterisk is designed to support multiple switch backends, so you can put a
statement like this in your dialplan:
switch => IAX2//
Which pulls in extensions from another remote host. But just as easily
one could create a switch for MySQL, LDAP, NIS, etc.
I have a mysql_swit
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 asteri
On Tuesday 18 March 2003 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
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
that's what immediate mode does for FXO signalled interfaces in
/etc/asterisk/zapata.conf. In immediate mode, as soon as the phone goes
off hook, it starts at the "s" extension.
mark
On Tue, 18 Mar 2003, Don Pobanz wrote:
> We have need of a ringdown circuit in an elevator. If someone picks up
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 key
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/zapte
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 elevat
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 cir
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
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 extensions.conf and the voicemail.conf and to use temporary files
for
On Tue, 2003-03-18 at 02:47, lizardbox wrote:
> 1 DS3
Do you already have a DS3 to T1 gateway, like a CAC widebank?
> 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 bu
My SIP phones still don't have MOH. Just dead quiet
Mark Spencer wrote:
> 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*
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 con
what's the intended behaviour of ${variable:a:b}?
given that ${exten} = 501234
until yesterday ${exten:2} would give '1234'
with current CVS ${exten:2} is '50' while ${exten:2:4} is '1234'
how do I just strip characters/digits in front of the variable without
specifying a length?
I was do
> 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
Hello Guys,
i'm using a Cisco 7960 and a Cisco 2610 with a VIC2-BRI to connect
to POTS. Phone and Gateway are using SIP. Everything works well
(inbound calls, outbound calls etc), the only thing i can't get
to work is hanging up an outbound call *before* the remote end
pick's up the call. When i h
Is that mysql switch available under gpl or for free use?
matteo
Il mar, 2003-03-18 alle 16:25, Jeremy McNamara ha scritto:
> Mark Spencer wrote:
>
> >Asterisk is designed to support multiple switch backends, so you can put a
> >statement like this in your dialplan:
> >
> >switch => IAX2//
> >
>
Mark Spencer wrote:
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 s
More then likely your web server is sending the phone
some environment variables It does not like. The Cisco
phones XML parser is very very picky. There is a perl
module specifically for Cisco phone Cisco::IPPhone;
You should be able to find it in cpan, if not google will.
Mike Reiling wrote:
the handler for ${exten:2} is commented out from current
cvs (pbx.c line 749).
I think that mark is adding substring support with double :
syntax, but that broke the single : syntax.
You can use ${EXTEN-1} ,that works, although is
deprecated.
matteo.
Il mar, 2003-03-18 alle 18:49, Lele Forzani ha
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
One of the creators of the XML standard, a programmer himself, has said
that he is unsatisfied with xml from a programmers point of view
himself.
http://www.tbray.org/ongoing/When/200x/2003/03/16/XML-Prog
He makes good points, ones that I fully concur with.
Unless there is a real need for aster
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 ch
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 1
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 of files..
Also what parameters do the files need to b
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 ext
Last time I talked to him, the project got a little more complex than he
thought to be feasible. So, no, I don't think it's being actively developed.
At 11:38 AM 3/18/2003 -0500, you wrote:
There appears to be an Asterisk WebMin module on the Digium FTP site, in
some state of development. Is thi
¡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?
Thanks,
HoraPe
---
Horacio J. Peña
[EMAIL PROTECTED]
[EMAIL PROTECTED]
___
> 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 provi
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 nu
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 nu
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
I now have (most) of the bugs worked out of my SIP-only asterisk
installation, but one fairly serious issue remains. I am using the
dtmfmode=inband in both my general area as well as the
extension-specific area (for iconnect) in sip.conf, but only DTMF only
works on calls placed out through iconnec
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 of
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 w
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
I think to build a Zapata ISA card it's not the business of wise or not.
It's very inconvenient for us to by goods from US. Just because we are in
the different country.
I have built two pieces of Tormenta ISA card for study. They all work
fine.
- Original Message -
From: "Steven
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
> >
Thanks very much for everyones help today on getting XML working on the
7960. I have made a buch of apps that get weather, stock quotes. I am
working on controlling lights as well. Directory for * is almost done.
As soon as I am done with that, I will release it to all to play with.
So now
Hi,HoraPe
I have built two pieces of Tormenta ISA card for study. They all work
fine. How many oscillators do you want? I can provide it to you as many as
possible.
Regards.
john
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTEC
¡Hola!
We plan to do use Tormenta ISA as a basis for development, and the
PCI card is probably beyond our abilities.
> - Original Message -
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, March 18, 2003 12:33 PM
> Subject: Re: [Asterisk-Users] p
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
You are right with the ATA. Check this doc out
http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataincfg/186ugcc.htm
Look at the section about Pre-Dial.
I wonder if you can do this with a Cisco 7940 or Snom 100/200?
On Tue, 2003-03-18 at 19:34, John Todd wrote:
> I believe the Cisco AT
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. N
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 pl
For Cisco IOS products you can use "connection plar opx " and then set up a dial-peer for that number to go
wherever you want it to. Plar is of course, Private Line Auto
Ringdown. There are various examples on the Cisco web site.
--Eric
On Tue, Mar 18, 2003 at 08:47:22PM -0600, Stephen Webb wrot
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 Critchfie
We've been getting the following message from channel.c:
Exceptionally long queue length queuing to
Now while we still have to fix the base problem of why we're getting
such long queue lengths, I'm wondering the following:
a) Why do we check queue length after we've Q'd and not before?
b) Why do
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 the
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