Brian,
Try these:
...
-A INPUT -s x.x.x.x -p udp -m udp --dport 5060 -j
ACCEPT
-A INPUT -s x.x.x.x -p udp -m udp --dport 1:2
-j ACCEPT
-A INPUT -s x.x.x.x -p tcp -m tcp --dport 22 --syn -j
ACCEPT
...
Sunny
--- Brian West [EMAIL PROTECTED] wrote:
I'm trying to get some iptables rules
Ok it its working but what extra steps besides port 1720 need to be open
for chan_h323 to play nice with this setup? Because right now its missing
something... :/
bkw
On Sat, 20 Sep 2003, Sunny Woo wrote:
Brian,
Try these:
...
-A INPUT -s x.x.x.x -p udp -m udp --dport 5060 -j
ACCEPT
-A
I don't know whether this ought to go to the bugtracker.
I downloaded the current CVS last night and then again just a few
minutes ago.
In both cases I can crash asterisk very easily by the following method:
1. Call up and leave a voicemail.
2. Log in and listen that I have a new message.
3.
Hi,Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in
2 offices.I am able to make call between two offices. But the problem is
that calldosen't hangup.Office A [Asterisk+2FXO+SamsungPBX]
- I A X Office B
[Asterisk+2FXO+SamsungPBX]Configuration files are
On Sun, 2003-09-21 at 08:50, Brian Capouch wrote:
I don't know whether this ought to go to the bugtracker.
I downloaded the current CVS last night and then again just a few
minutes ago.
In both cases I can crash asterisk very easily by the following method:
1. Call up and leave a
I'm trying to get some iptables rules that work with asterisk but for some
reason I keep blocking everything and or locking myself out of the box..
mybad does anyone have any configs they would like to share that allow
asterisk and ssh from x ip?
This is what I use for Asterisk form my
googled: yes, asked #asterisk: yes..
I need to extract headers of the SIP call signalling protocol on
outbound calls (especially call setup response message/ 200 OK).
Example:
phone with extension 33 place a call to the number 12345 (or
sip:[EMAIL PROTECTED]:5060) via SIP. The SIP proxy then
Last week I did aCVS update and since then I
haven´t been able to run asterisk normally.The strange thing is that I
have even go back to previous versions (0.5.0) andI am seening the same
problems.
Basically, when I try to load the zap module I get
the following error:
WARNING[16384]:
Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql
/Olle
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This is already fixed in CVS.
Mark
On Sun, 21 Sep 2003, Brian Capouch wrote:
I don't know whether this ought to go to the bugtracker.
I downloaded the current CVS last night and then again just a few
minutes ago.
In both cases I can crash asterisk very easily by the following method:
1.
On Saturday 20 September 2003 14:29, Steve Totaro wrote:
...i think the nec runs dos...
NEC PBX' run a derivative of BSD.
-d
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The implementation of *72 is done for FXS port (the one that gives the
dialtone). However you could implement that with some extensions.conf
logic.
regards
Martin
On Sat, 20 Sep 2003, Rich Adamson wrote:
Martin,
That makes sense... but how would one actually use *72#, as an example,
when *
Does this thread help?
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
Thanks, this is exactly what I was looking for. I tried experimenting with
different codecs myself, and GSM seems to be the only one that works...
neither iLBC or Speex went thru. I'm using XLite
Hi,
Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm
thinking of getting a BRI in my house to deliver more advanced signaling to
my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux.
Is there any particular BRI card that works better with Asterisk than any
We sell:
AVM B1 for development environment
Eicon Diva Server BRI card for live system (on-board echo canceller)
Tan
www.telappliant.com
- Original Message -
From: Mark Hagler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, September 21, 2003 10:43 PM
Subject: [Asterisk-Users]
The echo canceller algorithms aren't doing anything. We get extreme
echo during the conversation, it appears even before the call connects,
the echo is there...
This only happens with SIP to/from WCFXO (analog POTS). Looking at the
Zaptel configuration:
/etc/asterisk/zapata.conf:
I'm having trouble with a WX100USB adapter and a Siemens Gigaset
cordless phone.
If I select fxols as a signalling method, calls are being
disconnected. Usually after about 4 minutes, and asterisk just says that
the phone has hung up.
If I choose fxogs, I immediately get a LINE IN USE message on
Hi,
Uncomment the following line in the makefile in /usr/src/zaptel
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
Do a make clean install in this directory and reload wcfxo driver (rmmod and
then modprobe). Try again and see if there is any improvement.
Tan
www.telappliant.com
- Original Message
I bet your jack is wired backwards.. :) Try checking that out.
bkw
On Sun, 21 Sep 2003, Asterisk PBX wrote:
The echo canceller algorithms aren't doing anything. We get extreme
echo during the conversation, it appears even before the call connects,
the echo is there...
This only happens
I have a question regarding MySQL CDR's:
For a given extension I need to limit the number of minutes it can use in a
given week.
I was thinking about using the CDR information in the MySQL table to see the
usage for the week and then if exceeded, STOP the call and play a message.
does anybody have
Oh, I forgot to say, zaptel/wcfxo is compiled with:
KFLAGS+=-DECHO_CAN_MARK2
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
(and, Brian, my jack is wired correct..)
-Original Message-
From: Lenny Tropiano [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
PBX
Sent: Sunday, September 21, 2003 5:02 PM
To:
Hi,
I just tried to test this feature with fwd. I defined an incoming and
outgoing limit of 1. The following comand verifies it:
*CLI sip show inuse
UsernameincomingLimit outgoingLimit
fwd 0 1 0 1
1010
I'm trying to get some iptables rules that work with asterisk but for
some
reason I keep blocking everything and or locking myself out of the box..
mybad does anyone have any configs they would like to share that
allow
asterisk and ssh from x ip?
This is what I use for Asterisk
Hi,
I am having the same issue with the echo wit that configuration. Were
you able to resolve it?
Thanks,
Kevin
-Original Message-
From: Asterisk PBX [mailto:[EMAIL PROTECTED]
Sent: Sunday, September 21, 2003 6:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Very bad echo
So 'zap show chanenl channel-no' shows that the echocan is turned on ?
Martin
On Sun, 21 Sep 2003, Asterisk PBX wrote:
Oh, I forgot to say, zaptel/wcfxo is compiled with:
KFLAGS+=-DECHO_CAN_MARK2
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
(and, Brian, my jack is wired correct..)
-Original
You could have an AGI script that runs after an outbound call to update a
running-total figure with the amount of either the last call or all calls to
date in the current period?
That way you're just checking a stored value before allowing/denying an
outbound call?
My partner found it!!
Problem solved...
The error was a syntax error in the zapata.conf
channel=1
Should have been written as:
channel=1
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Agree, I can run an AGI script after the outbound call.
But where do I invoke the AGI script?
it can't be in extensions.conf since, I believe, when either party hang-up,
the next priority is not invoked, or am I mistaken?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
You are kidding,I hope.
This typo would manifest itself as an echo problem?
May be the parser needs to put out a warning of some kind.
That is my 2cents.
URiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk PBX
Sent: Sunday, September 21, 2003 8:39
Just on a side note can you please put a realname in your name field on
your email client. Everytime I see Asterisk PBX I think gee more
voicemail.
bwk
On Sun, 21 Sep 2003, Asterisk PBX wrote:
My partner found it!!
Problem solved...
The error was a syntax error in the zapata.conf
I wouldn't mess with the gains if I were you.
Mark
On Sun, 21 Sep 2003 [EMAIL PROTECTED] wrote:
Hi,
I am having the same issue with the echo wit that configuration. Were
you able to resolve it?
Thanks,
Kevin
-Original Message-
From: Asterisk PBX [mailto:[EMAIL PROTECTED]
TC wrote:
I have seen nufone die, if the callerid is not
a cid from us 48 try setting your sic to
This seems to not be an issue any more as we have many Canadian
customers sending Canadian caller*id's.
Jeremy McNamara
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Asterisk-Users mailing
First off the Asterisk mailing list is not a proper place to be
discussing these details. Secondly, I have sent you an example config to
the email address that you sent the payment from, when you signed up
with us.
The solution to your problem: You need to register to our system. As
of
You have to enable ring indications
exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr
Jeremy McNamara
Roy Sigurd Karlsbakk wrote:
hi
seems like things are closing in to something that might look like
success. I have one problem left: I don't get ring indicator when I dial
out from the
On Sunday 21 September 2003 18:16, Uriel Carrasquilla wrote:
I have a question regarding MySQL CDR's:
For a given extension I need to limit the number of minutes it can
use in a given week.
I was thinking about using the CDR information in the MySQL table to
see the usage for the week and
Hi,
Is there anyway to use xlite though a nat
I have a xlite - nat- asterisk.
* is on a public IP.
When I do this, I get an error on the asterisk server because it is trying
to use the dirty ip of the computer running xlite.
All of the settings in xlite seem to have no effect!
Michael
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this something that Asterisk or channel bank is capable of? I just need
someone to give me a brief
Hi Leif,
Hunting, or roll-over has to be done at the CO. The
CO is the only place that knows if a line is busy or not
**AND** have the ability to redirect the call setup request
to a different line
On Sun, Sep 21, 2003 at 10:46:20PM -0400, Leif Madsen wrote:
-BEGIN PGP SIGNED
Check the zapata.conf.sample for the keyword 'group'
Jeremy McNamara
Leif Madsen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this
how does a PBX control the call setup of inbound calls from
the PSTN??
unless you are doing something like ATM an your switch is going to
handle processing a call setup request, I don't see how * can
deal with hunting from a PSTN side.
Certainly from the station or SIP or IAX or H323 side it
I like it.
I am thinking of putting this query in a C++ but I am a bit concern on
1) scalability
2) delays in setting up the calls
shoud I be concerned?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: Sunday, September 21, 2003
I am planning on getting 4 analog trunk lines from my carrier (SBC).
~US$14/month/each
And a block of 20 DID numbers for these trunk lines.
(~US$15/month/block of 20) (a block of 20 is the smallest)
Inbound calls come in, and the lines (on the * side) are set to the
same context. (which contain
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