uhh, gafachi only does new york DID. I have my oick of IAX outgoing,
im talking incoming DID
On Tue, 10 Aug 2004 01:52:28 -0400, Luke Catranis [EMAIL PROTECTED] wrote:
gafachi
This mailbox protected from junk email by MailFrontier Desktop
Hi,
I'm using TE410P card with four T1 lines. I've configured all the channels in my
/etc/zaptel.conf file. In zttool i'm getting OK for the Span-1 but the other three
spans giving RED alarms. Pls give me your help where is the mistake.
Regards
Monster
Oh, sorry, for not reading that far.
In DeStar at http://www.holgerschurig.de I have a manager.py script where
I digged out all possible manager commands that I found. Here is an
excerpt:
manager.c:
# Action: Ping
# Parameters: none
res = conn.action('Ping')
In article [EMAIL PROTECTED],
Travis Conway [EMAIL PROTECTED] wrote:
I am having problems getting the latest CVS right now. A cvs checkout asterisk -t
gets to
this part and sits forever:
S- server_register(fpm-world-mix.mp3, 1.1, , , , , )
S- Register(fpm-world-mix.mp3, 1.1, , , )
That
odd
my broadvoice has been working fine over here.
- Original Message -
From: lists-jmhunter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 09, 2004 10:37 PM
Subject: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs
Hey there,
I don't know who else has suffered
Hi all,
I have two problems with h.323 in *
The first one is, I can call my voip-phone,
(exten = 59305004,1,Dial(H323/[EMAIL PROTECTED]))
BUT, I hear nothing
in h.323 debug mode:
*CLI Allowed Codecs:
Table:
GSM-06.10{sw} 1
Set:
0:
0:
GSM-06.10{sw} 1
-- Making
On a SuSE 9.1 installation I have severe problems using asterisk
with the active AVM B1. Just making some outgoing call works without
problems, but if both isdn channels are used asterisk or the
complete server may hang or start showing very strange behavior.
It might not even be possible to
Hi there..
Could someone here please explain what these codes mean..
Aug 10 10:22:22 NOTICE[1146227632]: PRI got event: 6 on Primary D-channel of
span 2
Aug 10 10:25:22 NOTICE[1146227632]: PRI got event: 8 on Primary D-channel of
span 2
I have two E100P installed in the machine, but the problem
dear all users
hi !! , I m using Asterisk as a call manager,
i have made two windows clients and a linux server on which asterisk is running ,
calls are succesfully authenticate in asterisk ...
but the problem is with voice jittering when , i record my voice and then play it (by
using Playback
Hi Claus,
CF Could someone here please explain what these codes mean..
CF
CF Aug 10 10:22:22 NOTICE[1146227632]: PRI got event: 6 on Primary
D-channel
CF of span 2
CF Aug 10 10:25:22 NOTICE[1146227632]: PRI got event: 8 on Primary
D-channel
CF of span 2
Event 6: Abort HDLC Frame
Event 8: Bad HCS
Here you go.
loadzone = no
defaultzone = no
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
bchan=48-62
dchan=47
Both E100P are connected to PSTN.
Kind Regards
Claus Futtrup
- Original Message -
From: Storer, Darren [EMAIL PROTECTED]
To:
Hello List!
I just read an article about asterisk, and i would like to ask a few
questions to see if i understood the principle right.
Reciving Calls:
---
- To be able to recive calls, i need to have an VoIP-Provider.
- I need a Static IP, so that the VoIP-Provider can redirect the
Hi,
I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI. Then
I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.
The error message given by asterisk is that it
maybe you can if try this..
span=2,1,0,ccs,hdb3,crc4
Claus Futtrup wrote:
Here you go.
loadzone = no
defaultzone = no
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
bchan=48-62
dchan=47
Both E100P are connected to PSTN.
Kind Regards
Claus Futtrup
Try specifying your number you want to dial with b in front of, e.g.
Dial(CAPI/01824708169:b01824708752,60) in your extensions.conf!
Regards,
roland
Roland Zagler
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent:
[EMAIL PROTECTED] wrote:
Hello List!
I just read an article about asterisk, and i would like to ask a few
questions to see if i understood the principle right.
Reciving Calls:
---
- To be able to recive calls, i need to have an VoIP-Provider.
No, You can use one but you could also
Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.
Regards.
Jonathan
Original Message
== From: Roland Zagler [EMAIL PROTECTED]
== Date: Tue, 10 Aug 2004 11:18:32 0200
Try specifying your number you want to dial with b in front of,
e.g.
- To be able to recive calls, i need to have an VoIP-Provider.
No, you can receive calls by other means, e.g. ISDN line, analog line
etc
In my case, i wouldnt have a telephoneline.
You didn't write this, you asked if you *NEED* an VOIP-Provider.
Yes, you CAN have an VOIP-Provider :-)
On Tue, 10 Aug 2004 20:18:29 +1000, David MacKinnon [EMAIL PROTECTED] wrote:
Okay, I've read as much as I can, and I think i've followed
instructions, but I'm still having problems with * and firefly... I can
get outgoing to other freshtel working, but not incoming (I get the not
available
Hi,
here is something that is bugging me for some time now...any pointers would
be great.
I am running linux on 1 pc 192.168.x.x and my softphone (Sjphone ) can
connect to it from 192.168.x.y without a problem on port 5060.
However when i run a softphone on the same linux box where i run
Robert Barnes wrote:
On Tue, 10 Aug 2004 20:18:29 +1000, David MacKinnon [EMAIL PROTECTED] wrote:
Okay, I've read as much as I can, and I think i've followed
instructions, but I'm still having problems with * and firefly... I can
get outgoing to other freshtel working, but not incoming (I get
On Mon, 9 Aug 2004, hank wrote:
can you use .wav files or does it have to be gsm?
I thought about that possibility too. Unfortunately I was not
successful when putting .wav Files instead of gsm Files in
/var/lib/asterisk/sound
Which Formats will * accept and what extensions may be used ? Is
On Tue, 10 Aug 2004 21:07:53 +1000, David MacKinnon [EMAIL PROTECTED] wrote:
gateway.freshtel.net gives Number is disconnected messages for all the
PSTN numbers I try, cts-au.freshtel.net gives me the no such
context/extension errors. Are you using firefly for PSTN calls?
Yes, I am using
Hi Roland,
Nothing on that message helps me unfortunately. I can make calls
from
SIP to ISDN I just can't get call transfer to work.
Regards, Jonathan
Original Message
== From: Roland Zagler [EMAIL PROTECTED]
== Date: Tue, 10 Aug 2004 12:21:29 0200
Here's the post i
Robert Barnes wrote:
[outgoing-std]
exten = _0[238],1,Macro(outgoingfreshtel,61${EXTEN:1},70) ; Freshtel
Got it, thanks :) Not sure what it was, I'll have to fiddle a bit.
-David
___
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[EMAIL PROTECTED]
Can you post your extensions.conf, maybe i can find something!
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM
To: [EMAIL PROTECTED]
Subject: Re:
My extensions.conf is:
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2
Yang,
(B
(BI am now running rc1 and I am only getting one call record when calling from
(Ba
(Ba sip channel to a zap channel
(B
(BWhat is you configuration?
(B
(B/ed
(B
(B-Original Message-
(BFrom: [EMAIL PROTECTED]
(B[mailto:[EMAIL PROTECTED] Behalf Of VoIP
(BSent: Tuesday,
Have you tried removing ${CALLERIDNUM} from your 1st line in context
[SIP] in extensions.conf?
Is your ISDN Line configured to transfer the Extensions to you
(Provider-dependent)?
And try to put Answer before calling to CAPI!
I do it like this:
[MyContext1]
exten = _.,1,Answer
exten =
Hello!
Is there a asterisk mirror?
ftp://ftp.asterisk.org/pub/telephony/asterisk/ is slwww
;)
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To UNSUBSCRIBE or update options visit:
On Mon, Aug 09, 2004 at 11:52:51AM +0100, Nick Barnes wrote:
The reason I ask is that I installed a BRI system (Single Fritz! AVM card
using chan_CAPI) last week which refused to work - turned out that British
Telecom had provisioned the line as a point-to-point and not
point-to-multipoint as
Hi Roland,
Still no difference. The call works fine but the transfer fails with
the same error message as before:
-- Executing Dial(CAPI[contr1/01824708169]/0,
CAPI/01824708169:b170) in new stack
Aug 10 13:34:34 NOTICE[294930]: chan_capi.c:1172 capi_request: didn't
find capi device with
On Mon, 2004-08-09 at 17:41, Jason Kawakami wrote:
is there something that needs to be set up to make the 'called' and
'callers' buttons work on this phone?
all i get is the backlight to switch on and off.
Jason Kawakami
Lift up the handset and try it. It doesn't work when it is on-hook.
Ty Oliver that seemed to do the trick! :-)
-Original Message-
From: Oliver [mailto:[EMAIL PROTECTED]
Sent: Monday, August 09, 2004 3:52 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)
No, i made a mistake ... the symlink is actually linux-2.6 -
On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
Hello!
Is there a asterisk mirror?
ftp://ftp.asterisk.org/pub/telephony/asterisk/ is slwww
Use CVS, it is mirrored.
--
Steven Critchfield [EMAIL PROTECTED]
___
Asterisk-Users mailing
I have been following this thread with great interest, because I am also
an asterisk newbie and I had many of the same questions.
I did not see one of my questions, though:
Suppose we retain our analog phones and PSTN lines, but we wish to make
VoIP long distance calls. Can Asterisk do this?
On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
Hello!
Is there a asterisk mirror?
ftp://ftp.asterisk.org/pub/telephony/asterisk/ is slwww
Use CVS, it is mirrored.
thanks!
But how stable is the cvs checkout?
___
Asterisk-Users
Another nibbage question:
modprobe tor2 -v
install /sbin/modprobe --ignore-install tor2 /sbin/ztcfg
insmod /lib/modules/2.6.5-1.358smp/misc/tor2.ko
ZT_SPANCONFIG failed on span 1: No such device or address (6)
FATAL: Error running install command for tor2
dmesg states that the module is
Hello,
I cannot find any inforamtion about distinguish rining tone on Channl Banks.
Can somebody point me somewehre?
Thank you in advance.
BArtosz
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[EMAIL PROTECTED]
Has anyone successfully configured an Asterisk box which is
behind a (NAT) router for Sipgate? I have mine behind a Linksys router and can
successfully register and apparently call out but I get no incoming audio and
can't be called from the outside.
Any help would be greatly appreciated.
All,
This is related to the following bug reports:
http://bugs.digium.com/bug_view_page.php?bug_id=0002024
http://bugs.digium.com/bug_view_page.php?bug_id=0002017
This is not an iConnect specific problem, but a chan_sip change. As it
turns out, type=user does not seem to work in the latest CVS
Good day all,
I have started playing around with using the TxtCIDName lookup tool to
replace the CID name field on inbound calls. The DNS side of things
seems to be working perfectly. When I trace the lookup that Asterisk
performs, the name it gets back is perfect. But, when I use
Since I joined the list in 2003, there has been an order of magnitude
increase in functionality in Asterisk. Kudos to Mark and the developers.
I'm finding that when I Google the list, posts that are more than a few
months old are no longer relevant to me. Without using the Google API,
is
All,
I just had the fortune to take one of the new Senao Wifi SIP phones for
a short test drive. First look - it's a nice, compact phone. Weighs
around 87g and roughly the size of a Nokia 6210. More on the those
later. The thing that struck me was the RF power, it's rated at 100mw
(20dBm).
Title: Can Incomming CallerID be fowarded to a SIP phone extension?
Hi All,
I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID to the SIP phone that is setup as an extension? We have a Voice
Suppose we retain our analog phones and PSTN lines, but we wish to make
VoIP long distance calls. Can Asterisk do this? That is, convert the
analog outgoing calls to IP. I assume we would need a VoIP provider to
do that (assuming the destination is a PSTN number) ?
When you want to do
It's difficult to give you help when you provide so little information.
Please post your zaptel.conf and zapata.conf at least, and say what your 4
spans are connected to.
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From:
On Tue, 2004-08-10 at 09:19, [EMAIL PROTECTED] wrote:
On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
Hello!
Is there a asterisk mirror?
ftp://ftp.asterisk.org/pub/telephony/asterisk/ is slwww
Use CVS, it is mirrored.
thanks!
But how stable is the cvs checkout?
On Tue, 2004-08-10 at 10:05, Michael Welter wrote:
Since I joined the list in 2003, there has been an order of magnitude
increase in functionality in Asterisk. Kudos to Mark and the developers.
I'm finding that when I Google the list, posts that are more than a few
months old are no
On Mon, 9 Aug 2004, Kevin Johnson wrote:
When dialing 8437624, I get the following output:
-- Executing NoOp(SIP/office1-b727, call for 843762 43762
6) in new stack
on the following line:
exten = _8.,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
this is really odd. I've got a
I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF
working just fine for internal extensions, voicemail, etc. If making an
outgoing call like this spa -- * -- Cisco AS5350 -- PSTN, I get no dial
tone. I am working unsuccessfully with Cisco right now on this, but they
cant
Hello *,
I try to establish a Asterisk-Server for internal and external usage.
Perfect use case for a DMZ, or not?
My configuration:
I N T E R N E T |
| | E
|
the mirrors of rc1 are also listed in the wiki as well
On Tue, 10 Aug 2004 10:43:36 -0400, Seth Remington
[EMAIL PROTECTED] wrote:
On Tue, 2004-08-10 at 09:19, [EMAIL PROTECTED] wrote:
On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
Hello!
Is there a asterisk mirror?
Hello!
I set up asterisk and its running well.
Now i would like to call asterisk and just mess around with the dial plans
a little, like described
here:http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/c469.html
But how do i connect to asterisk in the first place.
I would like to made a H323 Gateway with asterisk.
-- remote
gateway B --PBX-B
|
PBX---* gateway -remote gateway APBX-A
I have installed all channel
I've had an e-mail into pulver for a few days on getting access to the
firmware w/o a response. Anyone have the most recent wisip firmware they
wouldn't mind posting or e-mailing me?
Thanks,
Bill
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 10 August 2004 08:45 am, Seth Remington wrote:
On Mon, 2004-08-09 at 17:41, Jason Kawakami wrote:
is there something that needs to be set up to make the 'called' and
'callers' buttons work on this phone?
all i get is the backlight
On Tue, 2004-08-10 at 10:55, Bastian Schern wrote:
But now the IP-Phones could not communicate with Asterisk because the
Server (a Linux host) will NAT the internal IP-Addresses.
Is there a good way to solve this Problem?
Not the best solution, but you could tell the server not to
nat when
hi all,
just wondering if once a agent logs into a queue, that instead of them
sitting there and just getting the call without the phone ringing, is there
a way for them to hangup, and then the phone will ring when a call gets put
through from the queue?
thankyou in advance,
Greg Smith
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 09 August 2004 11:06 pm, Matt Fontaine wrote:
Hello-
I am new to the list, so forgive me if this has been answered, but I
haven't seen it on google yet.
Hi Matt,
Welcome to the list!
A note. Most of us here use software which
a) use a transparent bridge firewall
b) Use redirect with multiport of the sip ports to the * box IP.
c) And the most effective for your topology, don't use nat, use only
the routing properties of linux...
can u post ur firewall rules and routing table?
Bastian Schern wrote:
Hello *,
I try to
I have one of those SMC High Power W-LAN cards for special war driving
applications :-) It is rated at 200mW and is sold over the counter. Ok,
I usually don't put the card on my ear...
Leo Ann Boon wrote:
All,
I just had the fortune to take one of the new Senao Wifi SIP phones for
a short test
Hello list
I don't get to compile h323. I have the mistake:
asteriskaudio.cxx: In destructor `virtual
PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function)
asteriskaudio.cxx:167: (Each undeclared identifier is reported only
On Wed, 11 Aug 2004, Greg Smith wrote:
just wondering if once a agent logs into a queue, that instead of them
sitting there and just getting the call without the phone ringing, is there
a way for them to hangup, and then the phone will ring when a call gets put
through from the queue?
Why not use a public address for * ? A firewall, if properly configured
can protect your * server the same way as it would with NAT in a DMZ.
Dominique
Bastian Schern wrote:
Hello *,
I try to establish a Asterisk-Server for internal and external usage.
Perfect use case for a DMZ, or not?
My
-Original Message-
From: Greg Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 11:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] agent login
hi all,
just wondering if once a agent logs into a queue, that
instead of them sitting there and just getting
Anybody have any experience with blocking numbers in the U.S's Do Not
Call list?
We have a customer that will be getting their own Asterisk server from
us, and they want it to be check outbound numbers against the do not
call list; this is for a backup, in case there's a slip up and one of
their
Why use AGI? Why not just use the builtin DBGet() and DBPut() functions in
*?
-Chris
- Original Message -
From: drodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 9:22 AM
Subject: [Asterisk-Users] Blocking the 'Do Not Call List
Anybody have any
AsteriskList wrote:
Hello list
I don't get to compile h323. I have the mistake:
asteriskaudio.cxx: In destructor `virtual
PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function)
asteriskaudio.cxx:167: (Each undeclared
Has anyone seen the following problem?
Until recently, I couldn't understand why some extensions on my * system
would have a congestion tone as soon as I picked up the handset.
A little sleuthing through the logs and the source code led me to understand
that * thought it had seen the extension go
Christopher L. Wade wrote:
Hi all,
Is it just me and not reading the docs right, or has anybody else had
problems with the AbsoluteTimeout application and the 'T' extension when
used inside a macro?
[macro-attended]
; ARG1 is the device to dial out on, SIP or Zap, or whatever
; ARG2 is the
For one thing it's 't' not 'T', just like invalid is 'i' not 'I'
-Chris
- Original Message -
From: Christopher L. Wade [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 10:03 AM
Subject: Re: [Asterisk-Users] AbsoluteTimeout Inside A Macro
Christopher L. Wade
either DBget()[if you have mysql support set up in asterisk] or a simple
perl AGI script could do the DB search and either play the message itself or
change the extension to a default DNC number recording. There are dozens
of Perl AGI examples out there on the web that do MySQL lookups, and I have
I was curious if anyone had any ideas as to what could be causing this
error I did not see any responses and I did not know if it was
because no one knew of if I made someone mad... :)
On Aug 9, 2004, at 5:41 PM, Stephen Malenshek wrote:
I have searched all over the web and have not
Steve Szmidt wrote:
This one is one of those really dumb design errors some make.
The fact that it has survived all this time is the scary part.
The only two buttons that work off-hook are Speakerphone and Messages.
Why they would not have Called and Callers is beyond me.
Actually half of that
Chris, While you are thinking logically, This will just as
un-effective as putting them all in the dialplan, as the DBGet() and
DBPut() functionality deals with the internal astdb (db1 database).
I would reccomend going the AGI route at this time, until we have better
functionality for DB
Absolute timeout is 'T', and your standard timeout is 't'. If he's
looking for absolute timeout, he is, indeed, looking for the T extension.
They are case sensitive, and should work.
Mr. Wade: Have you tried using the T extension outside of the macro?
Although it *SHOULD* work within the
On Tue, 10 Aug 2004, AJ Grinnell wrote:
I am now at a total loss. Using Sipura spa-2000s connected to *, I get
DTMF working just fine for internal extensions, voicemail, etc. If
making an outgoing call like this spa -- * -- Cisco AS5350 -- PSTN, I
get no dial tone. I am working unsuccessfully
Hello-
Sorry if you see this
again, I am new to this list and accidentally put this message in another
thread.
What I want to know is, is there any way to send a
hook-flash signal from a cell phone (and then have Asterisk pass it up to the
PSTN?)
I suspect we need an example.
I have an
Bill,
please email me privately.
The latest 'released' version wf.00.11.
of course, we have some special versions in house
that haven't passed QA yet.
/ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004
On Tue, 2004-08-10 at 13:11, Chris Shaw wrote:
For one thing it's 't' not 'T', just like invalid is 'i' not 'I'
-Chris
Christopher L. Wade wrote:
Hi all,
Is it just me and not reading the docs right, or has anybody else had
problems with the AbsoluteTimeout application
Mind sharing how you got asterisk working with callmanager as an h323 gateway?
I have configured callmanager with the ip address of my asterisk server and
have setup asterisk with h323 and routed a call pattern to asterisk box but
im not getting anything at the asterisk end.
i have loaded
Josh Roberson wrote:
Absolute timeout is 'T', and your standard timeout is 't'. If he's
looking for absolute timeout, he is, indeed, looking for the T extension.
They are case sensitive, and should work.
Mr. Wade: Have you tried using the T extension outside of the macro?
Although it
You could try to specify incomingmsn *NOT* to * and outgoingmsn in
your capi.conf
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:38 PM
To: [EMAIL
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
Do anyone have a clue on how they do this ??
QOVIA FILES PATENTS FOR VOICE SPAM BLOCKING TECHNOLOGY
http://www.qovia.com/company/news/06.28.2004_voip_spam_patent_app_final.htm
Qovia ready to take on VoIP spam
I hadnt heard of that setting until today either, but it still doesnt work.
I am using dtmfmode-rfc2833 in sip.conf, and I have my spa's set to avt or
auto. The internal dtmf used for voicemail or anything within * works just
fine. I hust cant get tones out to the PSTN. The DTMF sounds very
Christoph == Christoph Rothe [EMAIL PROTECTED] writes:
Christoph Which Formats will * accept and what extensions may
Christoph be used? Is there a page in the wiki about that ?
Look in the formats dir in the asterisk src. Each of those formats
can be used.
They are well documented in terms of
We're not doing the entire list, just the key area codes this customer
will be calling. But yes, it's over a million and I definitely did not
intend on putting it in the dialplan.
We can make the specs of the machine nice, and put an optimized MySQL on
it, this should help with the query. Maybe
This appears to have been the magic bullet for me.
Thank you very much.
So, the bottom line is that there is a bug that ends up making inbound
calls use type=peer rather than type=user.
Correct?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 10 August 2004 10:55 am, Bastian Schern wrote:
Hello *,
I try to establish a Asterisk-Server for internal and external usage.
Perfect use case for a DMZ, or not?
Yes, use OpenBSD and make the server firewall a bridge. Then you can
I got to thinking about that too after I posted... nevermind, AGI would be
the way to go here anyway... I was thinking ZapaTeller or personal block
lists, which are WAY easier to use with DBPUT/GET... This is too massive for
that...
Good Luck! :)
-Chris
- Original Message -
Hmm you're right, sorry about that...
Have you seen this comment in the wiki?
if you connect a client to a sip peer with the option canreinvite=yes, then
absolutetimeout command has no effect.
Is it happening in both SIP and ZAP or just SIP?
-Chris
- Original Message -
From: Josh
dump tcpdump and bring out google.
http://www.google.com/search?hl=enie=UTF-8q=asterisk+sip+configuration
Returns three good matches in the first four results.
Start here:
http://www.google.com/search?hl=enie=UTF-8q=asterisk+sip+configuration
-Original Message-
From: [EMAIL
I'm a big fan of PHP for this kind of stuff myself, it's really easy to get
something like this going, probably would take you like a half an hour to
get a working program that simply checks against a MySQL database...
Perl works good too, I just like PHP because of all the built-in MySQL
voip spam?
I have never gotten any yet.
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 11:13 AM
Subject: [Asterisk-Users] Re: VoIP SPAM, what's next ?
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
Do anyone have a
Interesting thing...
If I put my 'T' extension in the context that called the macro, yet put
the AbsoluteTimeout command inside the macro, my 'T' extension from
outside the macro gets called...
Is this a bug? (Or a feature?)
Thanks,
Chris
___
On Tue, 10 Aug 2004, AJ Grinnell wrote:
I hadnt heard of that setting until today either, but it still doesnt work.
I am using dtmfmode-rfc2833 in sip.conf, and I have my spa's set to avt or
auto. The internal dtmf used for voicemail or anything within * works just
fine. I hust cant get tones
An AGI script can exit any way you want it to. You can have it set the
extension and the Priority to whatever you desire, you're not limited to +1
or +101 you can have it be anything.
As for speed, AGI scripts that we use on a daily basis do thousands of
searches a day through a 800,000 record
Chris Shaw wrote:
Hmm you're right, sorry about that...
Have you seen this comment in the wiki?
if you connect a client to a sip peer with the option canreinvite=yes, then
absolutetimeout command has no effect.
Is it happening in both SIP and ZAP or just SIP?
-Chris
Combination of SIP and ZAP,
Hey all,
I've got a PRI used for data calls right now, terminated in a MAX 4000.
We're only using around 12 channels on average and 16 max, so I'd like to
split off the remaining channels to terminate in an Asterisk box.
Does anyone know of a device that'll take a PRI in, and spit out two PRI's
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