Re: [Asterisk-Users] asterisk to asterisk config
Well do you yahoo? har har har j/k bkw - Original Message - From: Gonzalo Servat [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 12:13 AM Subject: RE: [Asterisk-Users] asterisk to asterisk config On 9/07/2004 6:06 AM +0100, Kevin Walsh wrote: Eugen Cristea [EMAIL PROTECTED] wrote: Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com What does Yahoo have to do with it? Have you considered trimming your quotes? Clearly not. Have you considered maybe his webmail provider (Yahoo) is automatically inserting the advertisement footer? Clearly not ;) Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
It is a good resource for neck tie non-geeks in small offices and will hopefully evangelize many of the uhh, it's open source and it is for free = so this could not be good heathens. Michael On Jul 8, 2004, at 11:19 PM, usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
On Thu, 2004-07-08 at 22:02, Kevin Walsh wrote: In my experience, the royalty is 10 to 15% of the *wholesale* price, which means it's more like two to three bucks a book. Trust me, unless you're Stephen King, John Grisham, or someone like that, you don't make all that much writing books (at least directly). All the more reason for the author to consider OpenDoc publishing. At the time I started the first book (which technically, someone else started, I ended up taking over), I don't think that was an option. By the second book (which was just an update of the first), I couldn't do OpenDoc because I was basically under contract with the publisher. Authors of technical books tend to make more money out of consulting anyway, as they're seen as an expert in their field. That assumes, of course, the author still wants to work in that field by the time the book is published. :) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem
Glen Hinkle wrote: I assume the pstn is your * system. Can you get audio both ways if you send the traffic back to *? pstn - as5350 - pstn ? -g Iuse the as5350 for termination at my telco, so it's physicly located there. When I call pstn - as5350 - (sip) asterisk, I can hear the audio from the asterisk, but audio from pstn will not get through. I tried: psth -- as5350 -- sipphone. and the same result. I can hear the sipphone but the sipphone cannot hear me. the as5350 is connected to my telco with dual trunked E1's /Micke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated Grandstream configurator
Neil Cherry wrote: Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Thanks, I've been having real trouble with those stupid DLLs. I can't upgrade some of them no matter what I do (WIN2K)! hmm , even with this version on two pc's (win xp) i can't get it working. you must restart after restart run setup again and same message ;) Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff - hfc card + x100p
hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bristuff - hfc card + x100p
Yes, me. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomaz Sent: 09 July 2004 09:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bristuff - hfc card + x100p hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sound quality IAX client GSM to ALAW with oh323
Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 - asterisk - DIAX :: sound is ok. When I perform an outbound call: DIAX - Asterisk - h323 :: sound is terrible and CPU load is 80% When I perform an asteisk internal call with DIAX: DIAX - asterisk IVR :: sound is good and cpu OK. Does anyone else have this problem ? Know how to solve it ? regards, Arne. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW ASTERISK WORKS
http://www.voip-info.org/ see you in two months ;-) Giscard Fernandes Faria wrote: Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with comments) of a simple example working with sip and h323. Thanks. Giscard ___ Yahoo! Mail agora com 100MB, anti-spam e antivírus grátis! http://br.info.mail.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wake-up call script in wiki
when i try and run this i just get a 403 error and i have set the chmod to
0777
- Original Message -
From: Gonzalo Servat [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 09, 2004 4:58 AM
Subject: Re: [Asterisk-Users] wake-up call script in wiki
On 9/07/2004 10:21 AM +0700, Isianto Istiadi wrote:
Dear guys,
I'm searching the wake-up call script in wiki, found one, but I have no
idea how to use it. Can you give some direction how to install it?
Thanks
I presume you're talking about this wake up call script:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20wake-up
Stick the following in cron:
* * * * * root /path/to/run_wakeups.sh
/path/to/run_wakeups.sh contains:
= cut ==
#!/bin/bash
PENDING=/tmp/wakeups
OUTGOING=/var/spool/asterisk/outgoing
SLEEP=5
TIME=$(/bin/date +%H%M)
for fn in $PENDING/$TIME.*.call
do
if test -r $fn
then
/bin/mv -f $fn $OUTGOING/
sleep $SLEEP
fi
done
= cut ==
The following is my wakeup.agi. Changes to the original version are: some
debugging functionality (as I was troubleshooting an issue where it would
read out the wrong time when the script tells you what time the wake up
call was set to), and it also creates the /tmp/wakeups directory if it
doesn't already exist. I suggest using the one on the voip-info.org page
first, and if you decide to use my version then use at your own risk :)
= cut ==
#!/usr/bin/perl
use Asterisk::AGI;
use Date::Manip;
use strict;
#
# Settings:
my $pending_dir = '/tmp/wakeups';
unless (-d '/tmp/wakeups') {
mkdir('/tmp/wakeups');
}
my $local_context = 'default';
# values for the call file:
my $maxretries = 60;
my $retrytime = 30;
my $waittime = 35;
my $debug = 1;
#my $application = 'MusicOnHold';
my $application = 'Playback';
my $data = 'wake-up';
my $callerid = 'Wakeup Call Service 297';
#
my ($sec,$min,$hour,$mday,$mon,$year,$wday,$yday,$isdst) =
localtime(time);
if ($debug) {
my $log = '/tmp/wakeup.log';
unlink($log);
open (DBG,$log) or die Cannot open debug file: $!;
print DBG \n . - x 50 . \n;
print DBG Logging started: . join('/', $mday, $mon, $year) .
. join(':', $hour, $min, $sec) . \n;
print DBG - x 50 . \n;
}
my $agi = new Asterisk::AGI;
my %stuff = $agi-ReadParse;# MUST DO THIS! -- (add this to
constructor!)
# this says 1 to create, 2 to confirm, 3 to cancel
my $func = $agi-get_data('wakeup-menu', 2, 1);
exit if $func == -1;
my ($caller) = $stuff{callerid} =~ /(\d+)/;
if ($func == 1)
{
my $time = $agi-get_data('time', 15000, 4);
exit if $func == -1;
if ($time =~ /^(\d{2})(\d{2})$/)
{
my $hour = $1 * 1;
my $min = $2;
print DBG 'HOUR entered: ' . $hour . \n if $debug;
print DBG 'MINUTE entered: ' . $min . \n if $debug;
if ($hour 0 $hour = 12 $min 60)
{
my $time;
# $agi-stream_file('pls-enter');
# $agi-stream_file('digits/1');
# $agi-stream_file('for');
# $agi-stream_file('digits/a-m');
# $agi-stream_file('or');
# $agi-stream_file('digits/2');
# $agi-stream_file('for');
# my $ampm = $agi-get_data('digits/p-m', 15000, 1);
my $ampm = $agi-get_data('am-or-pm', 15000, 1);
exit if $ampm == -1;
if ($ampm == 1)
{
$time = ParseDate(sprintf(%s:%02s AM, $hour, $min));
print DBG 'TYPE entered: AM' . \n if $debug;
print DBG '$time is set to: ' . $time . \n if $debug;
}
elsif ($ampm == 2)
{
$time = ParseDate(sprintf(%s:%02s PM, $hour, $min));
print DBG 'TYPE entered: PM' . \n if $debug;
print DBG '$time is set to: ' . $time . \n if $debug;
}
else
{
$agi-stream_file('vm-sorry');
}
if ($time)
{
my $h = UnixDate($time, %I) * 1;
my $m = UnixDate($time, %M);
my $a = UnixDate($time, %p);
foreach my $fn ($pending_dir/*.$caller.call)
{
unlink $fn;
}
my $filename = sprintf(%s/%04s.%s.call, $pending_dir,
UnixDate($time,
%H%M), $caller);
open(FILE, $filename);
printf FILE q{#
Channel: Local/[EMAIL PROTECTED]
MaxRetries: %s
RetryTime: %s
WaitTime: %s
Application: %s
Data: %s
Callerid: %s
},
$caller, $local_context,
$maxretries,
$retrytime,
$waittime,
$application,
$data,
$callerid,
;
close(FILE);
# say Your wakeup call
$agi-stream_file('has-been-set-to');
print DBG 'UnixDate $time translates to ' . UnixDate($time, %o) .
\n if $debug;
print DBG 'localtime (UnixDate $time) translates to ' .
localtime(UnixDate($time, %o)) . \n if $debug;
$agi-exec('SayUnixTime', sprintf(%s||IMp, UnixDate($time, %o)));
$agi-stream_file('for');
$agi-stream_file('extension');
$agi-say_digits($caller);
$agi-stream_file('auth-thankyou');
}
}
else
{
$agi-stream_file('vm-sorry');
}
}
else
{
Re: [Asterisk-Users] sample config file for GS BT101?
and cisco sell ciscoworks but they still show me how the basic interface works in case i want to do something crazy like write a custom app.. Steve On Tue, 8 Jun 2004, Steve Totaro wrote: I am not really sure what you are trying to accomplish. If its the GAPS alternative, the reason why its not on there is they sell GAPS so yeah, its reverse engineering unless you care to pay for their system. - Original Message - From: Stephen J. Wilcox [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 4:01 PM Subject: Re: [Asterisk-Users] sample config file for GS BT101? I was wondering about that too.. Following the instructions on that page for config did not work for me. Setting up a config file like the sample one made no difference to the phone (I can confirm it did tftp it okay). Also the method references md5 checks and I dont see that at all. I tried the downloads, we wouldnt do this from windows so need to know how to do this to write for *nix but I couldnt get the windows app to run on XP/2000 machines altho apparently it will run on 98 but I wasnt able to test that with a phone. So - is it literally just supposed to be a case of creating a blah=blah style config file at mac.txt ?? I note not all the options are listed in the sample, what about the others? And finally.. why doesnt this info appear to be available from the manufacturer, surely we shouldnt be reverse engineering? Steve On Thu, 8 Jul 2004, Steve Totaro wrote: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone the wiki seems to be VERY complete when it comes to GS - Original Message - From: Bruce Komito [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 9:31 AM Subject: [Asterisk-Users] sample config file for GS BT101? If you have an example of a config file for a Grandstream BT101/102, I would appreciate if you would share it with me. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Audiocodes MP124
I have problem in configuring MP124 FXS Gateway to work with *. Can anaybody help me in this way? - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with cdr_csv
Hello All, It seems that this question is very stupid, but anyway. Do I need any additional configuration for cdr_csv.so? This module is loaded by default at Asterisk's startup (asterisk -fvvv): [cdr_csv.so] = (Comma Separated Values CDR Backend) But when I place call I didn't see anything in /var/log/asterisk/cdr-csv. There is also no errors or warnings regarding this module on console. P.S.: Asterisk CVS-HEAD-07/07/04-11:48:42 P.P.S.: I'm using chan_oh323.so channel driver (by InAccess Networks). -- Best regards, Oleg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with cdr_csv
In oh323.conf set: amaFlags=billing Michael. Oleg A. Arkhangelsky wrote: Hello All, It seems that this question is very stupid, but anyway. Do I need any additional configuration for cdr_csv.so? This module is loaded by default at Asterisk's startup (asterisk -fvvv): [cdr_csv.so] = (Comma Separated Values CDR Backend) But when I place call I didn't see anything in /var/log/asterisk/cdr-csv. There is also no errors or warnings regarding this module on console. P.S.: Asterisk CVS-HEAD-07/07/04-11:48:42 P.P.S.: I'm using chan_oh323.so channel driver (by InAccess Networks). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
On 08/07/2004 at 22:19 usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. Late last year I was approached by a publisher asking if I would be interested in writing an asterisk book. I said a polite no (after some discussion) for a number of reasons: 1. Precisely what the author of this book is experiencing. Being bitchslapped by the asterisk community, for no apparent good reason. Since very few of the people on this list have actually read the book this early critism and mud slinging appears unfounded. Let's face it - the biggest failing of asterisk is it's lack of documentation. Sure there are guides, documentation projects.. but all of these rely on people giving up their free time... and since we don't have much of that, progress is slow. Anything that helps document asterisk and how to get it set up can't be all that bad. 2. I hand't heard of the publisher before, and a google search didn't turn up the most favourable links. 3. Asterisk changes day by day.. If I'd gone with it the book would have been out by now and (aside from being bitchslapped) I'd probably immediately have had to start a 2nd edition.. I'm not a writer... I can't even spell properly. I don't know what the author was offered, but if it was just 15% then perhaps the deal I was offered wasn't as bad as I thought... At $49 it is quite expensive, however, when funds allow I'll more than likely buy a copy out of interest - I consider myself fairly a well seasoned asterisk person, but hey it might teach me something too... I'm prepared to give it a chance. All IMHO of course... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Thursday 08 July 2004 22:49, Ian D. Wlloughby wrote: I am guessing the problem is that your internal clients can see the external SIP clients but not the other way round. The clients have to be able to make a physical connection to each other. You are not using any NAT capabilities I guess as your internal clients have their own network to access the server on. If you set nat on in sip.conf for one of your internal clients and get it to register on the public network, does this work? Yes, the internal clients can see the external but not the other way round. I thought that canreinvite=no meant that the clients didn't need to be able to talk directly - just be registered on the same * box. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP Monitoring
GIBERT Frdric wrote: Hello, Does someone know how to setup snmp monitoring on asterisk. Ive plan to deploy 50 asterisk, so I need some monitoring tools. I try with nagios as I read in the wiki, there is some project on it, but I cant reach the end. Can someone help me? Thanks. ***GIBERT Frdric* *Ste VigiNetworks* *Mobile: +33 6 72 08 35 16*** You could try ast_snmpd at http://faino.it/ast-ax-snmpd.html I hope to get some substantatial progress in it during the august holiday. Best regards, Andrea Fino -- Andrea Fino 8-) - Sistemi su misura di qualita' industriale Handcrafted systems with industrial quality [Phone: +39 071 2070104]+[Fax: +39 071 2077919]+[iaxtel:17002876594] [Web: http://www.faino.org]+[Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rollover oddity
Hello Jay,
I had something similar happen -- or so I thought. Turns out my *
wasn't configured right, and the call-waiting blip was generated by
Asterisk as it was detecting ring on the second line.
In this case, to try to see where the problem lay, I connected
just the first line to Asterisk. The second line went directly to
a handset, and Asterisk wasn't involved with it at all - so I
don't think it's the same issue you experienced.
I made a call using the first line through Asterisk, to busy the
line.
Then I called in from outside onto the first line. The telco
should have detected the line was busy, and rung on the second
line instead. They did not. The caller just hears the line
ringing.
If I take Asterisk out completely, and busy the first line with a
handset directly connected to the line, and repeat the
experiment, the rollover happens correctly.
Without your
extensions.conf and as much info as you can provide (hardware, extension
phones, etc) nobody's going to be able to tell you more about your
problem, though.
That's fair enough. I'm fairly new to Asterisk, so wasn't sure
what sort of debug info is of most use in this type of
scenario...
extensions.conf starts like this, and goes off into other
contexts. I can post the whole file if you feel it's relevant.
|
| [inbound]
| exten = s,1,Wait,1
| exten = s,2,Answer
| exten = s,3,NoOp,${CALLERID}
| exten = s,4,ResponseTimeout,10
| exten = s,5,AbsoluteTimeout(60)
| exten = s,6,BackGround(g-us-f-enstarwelcome)
| exten = s,7,BackGround(g-us-f-dialextension)
| exten = s,8,BackGround(g-us-f-choose)
| exten = s,9,BackGround(g-us-f-mailtraq1)
| exten = s,10,BackGround(g-us-f-nc2)
| exten = s,11,BackGround(g-us-f-other3)
| exten = s,12,BackGround(g-us-f-again)
|
| exten = 1,1,Goto(mailtraq,s,1)
| exten = 2,1,Goto(neatcomponents,s,1)
| exten = 3,1,Goto(other,s,1)
| exten = 8,1,Goto(inbound,s,1)
| exten = t,1,Goto(inbound,s,6)
| include = localextensions
Hardware is perhaps a bit on the low side - a 'recycled' Windows
box, PIII, 350Mhz. The extension phones are standard POTS phones,
nothing fancy at all.
I am wondering whether it may be somehow related to the line
length: we are a long way from the exchange (beyond DSL reach,
for example), so I wonder if there are some settings I can use to
compensate for that?
Bob
-Original Message-
From: Bob Bailey [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 4:55 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Rollover oddity
Hello,
I've got 2 analogue lines (from SBC) coming into a TDM22B.
SBC have put rollover from the first to the second line. The
rollover works fine when handsets are connected directly to
the lines (ie when Asterisk is not involved), but when the
lines are connected to Asterisk, the rollover fails: the
caller just hears the line ringing, and the person on the
first (busy) line hears call waiting interrupts.
I have proved that it's the rollover that's not taking place
(ie it is not that the rollover happens, but the second line isn't
answered)
So how (and why) does Asterisk affect the rollover in this
way, and how can a busy line going through Asterisk 'look'
different to the telco from a normal handset?
And, of course... how do I solve this?
Thanks
Bob
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Re: [Asterisk-Users] sound quality IAX client GSM to ALAW with oh323
Hi, Do IAX(GSM) - IAX(ALAW) calls sound ok? What is the configuration of OH323 channel (oh323.conf)? Also, run asterisk with '-vvvcd', make a call and send the output. Don't forget to enable the logging of debug messages (logger.conf). Michael. Arne Scheffer wrote: Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 - asterisk - DIAX :: sound is ok. When I perform an outbound call: DIAX - Asterisk - h323 :: sound is terrible and CPU load is 80% When I perform an asteisk internal call with DIAX: DIAX - asterisk IVR :: sound is good and cpu OK. Does anyone else have this problem ? Know how to solve it ? regards, Arne. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff - hfc card + x100p
hi, i have two hfc isdn cards and one x100p modules are loaded in order: - insmod zaptel insmod zaphfc modes=0 insmod wcfxo ztcfg cat /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 # loadzone=nl defaultzone=nl span=2,1,3,ccs,ami bchan=4-5 dchan=6 # loadzone=nl defaultzone=nl fxsks=7 --- and in zapata.conf i have: [channels] switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 1-2 ; switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 4-5 ; signalling=fxs_ks channel=7 so after that i load asterisk -vc and get no errors at all: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 7, FXS Kewlstart signalling == Starting D-Channel on span 1 == Starting D-Channel on span 2 == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) isdn is working ok but with channel 7 (x100p) is something wrong .. but what? Executing Dial(SIP/1-85f5, Zap/7/BYEXTENSION|130|t|r|f) in new stack Jul 9 14:43:32 NOTICE[311316]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time thankyou! Tomaz Robinson Tim-W10277 wrote: Yes, me. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomaz Sent: 09 July 2004 09:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bristuff - hfc card + x100p hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Book
Hello, If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Do I detect some friendly rivalry? ;-) | VoIP Telephony with Asterisk will be available July 22, directly from | Signate and through selected resellers for $49.95 plus shipping. Call | 415-442-4011 to order the book. Seriously, though, the more documentation the better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Hi all! It's great to start with for dummies question, but hey, we all have been human infants also =) Problem is, that I can not log on to this channel and I haven't found anything helpful during the past few days either. 1. The irc.freenode.net server gives me Couldn't look up your hostname and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) 2. Should I register? Where? That's one of the problems. I use mIRC on WIN2k and I should have the On connect, always get: Localhost and IP address on. Please, help me, so I don't have to flood Your inboxes, but ask about basic * things on the channel! BR, Antti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
--- Andy Powell [EMAIL PROTECTED] wrote: On 08/07/2004 at 22:19 usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. Late last year I was approached by a publisher asking if I would be interested in writing an asterisk book. I said a polite no (after some discussion) for a number of reasons: 1. Precisely what the author of this book is experiencing. Being bitchslapped by the asterisk community, for no apparent good reason. Since very few of the people on this list have actually read the book this early critism and mud slinging appears unfounded. Let's face it - the biggest failing of asterisk is it's lack of documentation. Sure there are guides, documentation projects.. but all of these rely on people giving up their free time... and since we don't have much of that, progress is slow. Anything that helps document asterisk and how to get it set up can't be all that bad. 2. I hand't heard of the publisher before, and a google search didn't turn up the most favourable links. 3. Asterisk changes day by day.. If I'd gone with it the book would have been out by now and (aside from being bitchslapped) I'd probably immediately have had to start a 2nd edition.. I'm not a writer... I can't even spell properly. I don't know what the author was offered, but if it was just 15% then perhaps the deal I was offered wasn't as bad as I thought... At $49 it is quite expensive, however, when funds allow I'll more than likely buy a copy out of interest - I consider myself fairly a well seasoned asterisk person, but hey it might teach me something too... I'm prepared to give it a chance. All IMHO of course... Andy You make some very valid points. And to be honest I applaud the authors effort in documentation. However, the price etc I think is a bit on the high side. For something that with the best effort is not going to be complete, simply due to the fluid nature of asterisk, I think it would have been better to have had a PDF offering with a promise of updates for a certain period. Same as you, all of course IMHO ... On a separate note, I would like to see those criticising (myself included) to make an effort and write a book themselves. Umar. ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff - hfc card + x100p
just one thing I forgot: less /proc/zaptel/3 Span 3: WCFXO/0 Wildcard X101P Board 1 RED 7 WCFXO/0/0 /proc/zaptel/3 (END) if this may helps? thankyou, Tomaz Tomaz wrote: hi, i have two hfc isdn cards and one x100p modules are loaded in order: - insmod zaptel insmod zaphfc modes=0 insmod wcfxo ztcfg cat /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 # loadzone=nl defaultzone=nl span=2,1,3,ccs,ami bchan=4-5 dchan=6 # loadzone=nl defaultzone=nl fxsks=7 --- and in zapata.conf i have: [channels] switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 1-2 ; switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 4-5 ; signalling=fxs_ks channel=7 so after that i load asterisk -vc and get no errors at all: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 7, FXS Kewlstart signalling == Starting D-Channel on span 1 == Starting D-Channel on span 2 == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) isdn is working ok but with channel 7 (x100p) is something wrong .. but what? Executing Dial(SIP/1-85f5, Zap/7/BYEXTENSION|130|t|r|f) in new stack Jul 9 14:43:32 NOTICE[311316]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time thankyou! Tomaz Robinson Tim-W10277 wrote: Yes, me. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomaz Sent: 09 July 2004 09:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bristuff - hfc card + x100p hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overlapdial on PRI
Hi Everybody,
Good day to you :)
I am trying to configure Asterisk for overlapdial=yes in zapata.conf
however in the extensions.conf, when specifying ${CALLINGSUBADDR} it
still isnt able to parse the digits through to asterisk.
What I am trying to do is have Asterisk read the digits which are
being keyed in by a mobile phone when the user dials into the system.
For example the user stores 123456789 in their mobile phone for
dialing whereby 123 is the number of our PRI and 456789 is the number
to be called through H323.
Any idea what I may be doing wrong here? Here are my configuration
file settings specified:
extensions.conf
exten = s,1,Answer
exten = s,2,Dial(H323/[EMAIL PROTECTED]/)
zapata.conf
usecallerid=yes
hidecallerid=no
immediate=yes
overlapdial=yes
I have also tried commenting out immediate=yes but that does not help.
Many thanks
___
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Re: [Asterisk-Users] internal external SIP
On Thursday 08 July 2004 23:04, Soren Rathje wrote: bindaddr = 0.0.0.0 ; Local interface externip = xxx.xxx.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks localnet = 10.0.0.0/255.0.0.0; Also RFC1918 localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really without a trace.. Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. I've included a trace below of an incomming call. I don't know which bits are relevant so I've pasted it all. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 12 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7c6b65eb Content-Length: 0 to 82.145.37.29:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 ACK User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=2003, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=7c6b65eb, response=2d2400a30b257419c48ac5dd6747 Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 13 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 2000 in remote list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as17a6c60a Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,
[Asterisk-Users] Dell 6450 / TE405p
I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP Monitoring
You could try ast_snmpd at http://faino.it/ast-ax-snmpd.html I hope to get some substantatial progress in it during the august holiday. I've added your web page to the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+addons ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Book
If either of these publishers are interested in serious sales then I hope that they will be selling through normal distribution channels like amazon.com and bookpool.com which is where I buy my tech books. So far, neither of these distributors are carrying the Asterisk books. I'll buy them both when bookpool.com has them. Tom At 05:51 AM 7/9/2004, you wrote: Hello, If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Do I detect some friendly rivalry? ;-) | VoIP Telephony with Asterisk will be available July 22, directly from | Signate and through selected resellers for $49.95 plus shipping. Call | 415-442-4011 to order the book. Seriously, though, the more documentation the better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
Update - didn't work in the second 6450 either. Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. Install identd and allow TCP port 113 inbound access and it'll work - if you play about with your username it'll probably work too. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Hi! And thanks for helping me out here. Ok, I have an invalid username - how do I get a valid username? Thx Antti -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- [EMAIL PROTECTED] 07/09/04 3:17 PM On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange echo problem
We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS line, I hear everything I say in my earpiece about 1/4 second after I say it. It's very irritating.We have tried 2 different * boxes, using 2 different T1/PRI cards f/ digium. After calling digium about it, we set echotraining to 800 in zapata.conf. It got better but was still there, if I turn the volume down on the phone, it does almost go away, but it's still detectable. No where near as clear as calling a person that has a PRI or channelized T1 for phone service. The POTS persons we call that we do have the echo issue with all say the call sounds perfecto to them. Am I missing something obvious? W. Kevin Hunt CCIE #11841 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Please see my previous post - if you install identd it will give you a valid name. Identd is quite common service and usually very safe to open remotely. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antti Lohikoski Sent: 09 July 2004 1:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net Hi! And thanks for helping me out here. Ok, I have an invalid username - how do I get a valid username? Thx Antti -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- [EMAIL PROTECTED] 07/09/04 3:17 PM On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Book
And I'm going to stop buying digium hardware until I can get it in Wall Mart! :) Joke! :) Panny - Original Message - From: Tom [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:20 PM Subject: Re: [Asterisk-Users] Asterisk Book If either of these publishers are interested in serious sales then I hope that they will be selling through normal distribution channels like amazon.com and bookpool.com which is where I buy my tech books. So far, neither of these distributors are carrying the Asterisk books. I'll buy them both when bookpool.com has them. Tom At 05:51 AM 7/9/2004, you wrote: Hello, If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Do I detect some friendly rivalry? ;-) | VoIP Telephony with Asterisk will be available July 22, directly from | Signate and through selected resellers for $49.95 plus shipping. Call | 415-442-4011 to order the book. Seriously, though, the more documentation the better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:40ee9c7a248451993774873! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Regiter config question
Folks, I been trying to understand how one would register * to a SIP provider that only gives you an Account number, PIN, and phone number. I reviewed some examples on Wiki but those show examples of SIP provides using username and passwords. I did the following under the [general]: Register = [EMAIL PROTECTED]/17135551212 The above line will send a register message to the SIP proxy but it returns a 403 Bad Account/PIN. I also tried various different register = lines using the accounts and pin as user and secret to no avail. My [contexts], under sip.conf, looks as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = voice-mail; Default context for incoming calls dtmfmode=rfc2833 [EMAIL PROTECTED]/17135551212 [17135551212] type=peer context=voiceline dtmfmode=rfc2833 ;secret=1000 ;username=123456789012 host=dynamic defaultip=192.168.0.1 ;accountcode=123456789012 Any suggestion? Kurt __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Please see my previous post - if you install identd it will give you a valid name. Identd is quite common service and usually very safe to open remotely. I'm using happily irc.freenode.net without any identd daemon ... and my firewall does iptables -A INPUT -p tcp --dport 113 -j REJECT iptables -A FORWARD -p tcp --dport 113 -j REJECT anyway ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On 09/07/2004 at 13:25 Chris Bond wrote: On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. Install identd and allow TCP port 113 inbound access and it'll work - if you play about with your username it'll probably work too. Kind Regards, Chris Bond Identd is NOT required Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
I changed from mIRC options to Enable Identd server. Is that enough or is Identd some little script or what? -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- [EMAIL PROTECTED] 07/09/04 3:37 PM Please see my previous post - if you install identd it will give you a valid name. Identd is quite common service and usually very safe to open remotely. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antti Lohikoski Sent: 09 July 2004 1:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net Hi! And thanks for helping me out here. Ok, I have an invalid username - how do I get a valid username? Thx Antti -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- [EMAIL PROTECTED] 07/09/04 3:17 PM On Fri, 9 Jul 2004, Antti Lohikoski wrote: and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) Maybe your username is invalid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
What have you got in /proc/pci? Do you have to do anything funny on the Dell to tell it that a card is there? Maybe it has some kind of health monitoring that you can switch off? -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at
[Asterisk-Users] Cisco MC3810 - Asterisk
Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - Call from a Grandstream (working)- Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing NoOp(SIP/4000-98ec, ) in new stack -- Executing Goto(SIP/4000-98ec, intern-post|4001|1) in new stack -- Goto (intern-post,4001,1) -- Executing Dial(SIP/4000-98ec, SIP/4001|30|Ttm) in new stack Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO URL) Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on RTP to 0 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for 4001 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a local user -- Called 4001 Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel 'SIP/4000-98ec' Call from the Cisco (not working) Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing NoOp(SIP/192.168.1.9-08134bb8, ) in new stack -- Executing Goto(SIP/192.168.1.9-08134bb8, from-sip-post|4001|1) in new stack -- Goto (from-sip-post,4001,1) Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel 'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in context 'from-sip-post', but no invalid handler BTW- Working with a ripped-off version of John Todd's configs... Anyone get this working? It's kicking my ass. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Friday 09 July 2004 06:54, Antti Lohikoski wrote: 1. The irc.freenode.net server gives me Couldn't look up your hostname and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) This is *specifically* why I wish bkw (Brian West) would turn off that flag on the channel. In order to combat spam bots infiltrating the channel, it is set up to only allow freenode-registered nicknames. In order to register your nickname with freenode, send a /msg nickserv help command once you're on freenode. NickServ is a Nickname Server bot -- it will let you register a nickname and set a password so your nickname can't be stolen. Identd is *not* required. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Monitor cmd and Queues
When was this implimented? I am currently running CVS-HEAD-07/07/04-09:34:29 --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of brian Sent: Thursday, July 08, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Monitor cmd and Queues Get the lastest CVS head it can start monitoring from time the agent picks up the phone. So you get ZERO hold music. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Harold Workman Sent: Thursday, July 08, 2004 5:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Monitor cmd and Queues Hi, Id like to record a queue conversation using the Monitor command but the problem im running into is the way i configure it * records the music on hold along with the conversation. Is there a way to start recording when the call is picked up by an agent? Could someone give me a small example of how they set this up. My current extensions.conf portion of the file looks like: [cytelbilling] ;# Cytel Communications Billing Support ### exten = s,1,Background(/usr/src/asterisk-sounds/sounds/this-call-may-be) exten = s,2,Background(/usr/src/asterisk-sounds/sounds/monitored) exten = s,3,Background(/usr/src/asterisk-sounds/sounds/or) exten = s,4,Background(/usr/src/asterisk-sounds/sounds/recorded) exten = s,5,Background(/usr/src/asterisk-sounds/sounds/for-quality-purposes) exten = s,6,Answer exten = s,7,SetMusicOnHold(default) exten = s,8,DigitTimeout,5 exten = s,9,ResponseTimeout,10 exten = s,10,Background(/usr/src/asterisk-sounds/sounds/pls-stay-on-line) exten = s,11,Monitor(wav,,m) exten = s,12,Queue(cytelcs) --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_mISDN test release....
Hi Asterisk knights, we are proud to announce that our brand new chan_mISDN channel driver for Asterisk is now official released for testing. You can download it under http://www.beronet.com/index.php?PageID=3017. Please test it ample, and post bugs or feature requests to www.beronet.com/bugs. Have a lot of fun! Best Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow (bei Berlin) FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed regarding Grandstream phone
Hi there, This is shan. I need help regarding the grandstream Budgetone - 100 phone which i configured. My problem is: I can able to do call --- Softphone(PC) --- * --- Grandstream Budgetone-100 but i'm not able to do call --- Grandstream Budgetone-100 --- * --- Softphone(PC) If anyone knows it pls help it would be very helpful regarding my project work. Regards Shan
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Andrew Kohlsmith wrote: On Friday 09 July 2004 06:54, Antti Lohikoski wrote: 1. The irc.freenode.net server gives me Couldn't look up your hostname and No identd (auth) response followed with Closing Link: StiX (Invalid username [~antti.loh]) This is *specifically* why I wish bkw (Brian West) would turn off that flag on the channel. In order to combat spam bots infiltrating the channel, it is set up to only allow freenode-registered nicknames. In order to register your nickname with freenode, send a /msg nickserv help command once you're on freenode. NickServ is a Nickname Server bot -- it will let you register a nickname and set a password so your nickname can't be stolen. Identd is *not* required. Ok guys, enough FUD and wrong answers. He cannot get on, because his USERNAME has invalid characters antti.loh is not valid. You cannot have . in your username. This is NOTHING to do with the channel requiring you to register with nickserv, this is NOTHING to do with ident. Antti, in your IRC client, you are given a choice of nickname, and realname/username. Make sure both of these are a-z/0-9, no special characters. You should be fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to differentiate a *busy* call from not available?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm trying to find a way to differentiate wether a SIP extension is currently busy (e.g. on the phone) or not registered. So i do something like: exten = 100,1,Dial(SIP/foo,20,tr) exten = 100,2,VoiceMail,u100 exten = 100,102,VoiceMail,b100 If the phone doesn't answer I get the message: User is not available if the phone is currently in used i get the message: User is on the phone But if the phone is unplugged, I also get the message: User is on the phone! Any ideas? Thank you Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA7p9gXeDVKqIr3GURAocmAJsFpy4rlj/wsgNxoGgE+WmzfisMFgCfc8z/ WX6zBH+pg2M25phavis0CYY= =ZxKl -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Regiter config question
Folks, I been trying to understand how one would register * to a SIP provider that only gives you an Account number, PIN, and phone number. I reviewed some examples on Wiki but those show examples of SIP provides using username and passwords. snip Any suggestion? set up a softphone and trace it with ethereal(or the like) You should be able to get an idea of where to put what from the Register requests. Compare the Register requests generated by asterisk with your current config to the ones generated by the softphone. Andrew Thompson http://www.retirequickly.com/43653 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
William, I guess it was a bad idea to have on the website one of the examples stating Much of the information in the book came from the Asterisk Wiki pages. Which is the sole reason for my bashing on the book. I can see having a book like this will ease others concerns on running a free open source pbx that are not familiar with Asterisk. There can never be too much documentation. What are the signate's thoughts on having a lower cost PDF file for download? --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Boehlke Sent: Thursday, July 08, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. As an interested party to the Mahler book VoIP Telephony with Asterisk, I would like to clarify a point about it. We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki instead of buying the book, and we acknowledge the Wiki (that's where the confusion began) as a source of some material. That acknowledgement is in one of the sample pages on our site, which probably began the confusion. But the book is the 320 page product of nine months of independent work. We built a dozen systems during the course of writing it to test and implement features and options. Mark Spencer and the Digium technical staff contributed information, advice and equipment. Eight competent engineers, some of them active here every day, gave it a technical review. As a former beginner, in my opinion the index alone is worth the price compared to the time it took me to locate information. Thanks for listening. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Thursday, July 08, 2004 2:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP Monitoring
Holger Schurig wrote: You could try ast_snmpd at http://faino.it/ast-ax-snmpd.html I hope to get some substantatial progress in it during the august holiday. I've added your web page to the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+addons ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is an English version at http://faino.it/en/ast-ax-snmpd.html sorry, I forgot. Andrea Fino -- Andrea Fino 8-) - Sistemi su misura di qualita' industriale Handcrafted systems with industrial quality [Phone: +39 071 2070104]+[Fax: +39 071 2077919]+[iaxtel:17002876594] [Web: http://www.faino.org]+[Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
This is all I get. Can I send /msg -messages? My name has no symbols like ¤%¤%¤%... * Connecting to irc.freenode.net (6667) - -irc.freenode.net- *** Looking up your hostname... - -irc.freenode.net- *** Checking ident - -irc.freenode.net- *** Couldn't look up your hostname - -irc.freenode.net- *** No identd (auth) response - Closing Link: StiX (Invalid username [~antti.loh]) - * Disconnected -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Friday 09 July 2004 09:33, [EMAIL PROTECTED] wrote: He cannot get on, because his USERNAME has invalid characters antti.loh is not valid. You cannot have . in your username. This is NOTHING to do with the channel requiring you to register with nickserv, this is NOTHING to do with ident. You are correct; I missed the invalid username part, or rather mistook it for the lack of register. Apologies to all. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco MC3810 - Asterisk
I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - Call from a Grandstream (working)- Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing NoOp(SIP/4000-98ec, ) in new stack -- Executing Goto(SIP/4000-98ec, intern-post|4001|1) in new stack -- Goto (intern-post,4001,1) -- Executing Dial(SIP/4000-98ec, SIP/4001|30|Ttm) in new stack Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO URL) Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on RTP to 0 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for 4001 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a local user -- Called 4001 Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel 'SIP/4000-98ec' Call from the Cisco (not working) Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing NoOp(SIP/192.168.1.9-08134bb8, ) in new stack -- Executing Goto(SIP/192.168.1.9-08134bb8, from-sip-post|4001|1) in new stack -- Goto (from-sip-post,4001,1) Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel 'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in context 'from-sip-post', but no invalid handler BTW- Working with a ripped-off version of John Todd's configs... Anyone get this working? It's kicking my ass. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange echo problem
We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS line, I hear everything I say in my earpiece about 1/4 second after I say it. It's very irritating.We have tried 2 different * boxes, using 2 different T1/PRI cards f/ digium. After calling digium about it, we set echotraining to 800 in zapata.conf. It got better but was still there, if I turn the volume down on the phone, it does almost go away, but it's still detectable. No where near as clear as calling a person that has a PRI or channelized T1 for phone service. The POTS persons we call that we do have the echo issue with all say the call sounds perfecto to them. Am I missing something obvious? No, your not missing anything specifically. There seems to be a fair number of people with those same type of echo problems, which are the most difficult things to diagnose. There's no easy way to determine whether this is an * or pstn problem, so most of the previous efforts have been focused on ruling out what is not the problem. It would help all of us better understand the issues if at least some additional data were provided, however. Such things as cvs version, significant portions of 'zap show channel...' with the echo can data, and anything else that seems to narrow the focus. In most cases, turning volumes/gains down is a method to bypass the problem, not solve the root cause. (Certainly not true in all cases.) Personal opinion (based only on the various words posted to this list) is there still is an echo can issue floating in * that none of us have documented with sufficient data that would guide a developer towards the root cause. If you easily recreate the problem, that would go a long ways towards gathering qualifying data. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help needed regarding Grandstream phone
Shanmuganathan Kumaravel wrote: Hi there, This is shan. I need help regarding the grandstream Budgetone - 100 phone which i configured. My problem is: I can able to do call --- Softphone(PC) --- * --- Grandstream Budgetone-100 but i'm not able to do call --- Grandstream Budgetone-100 --- * --- Softphone(PC) If anyone knows it pls help it would be very helpful regarding my project work. Regards Shan I doubt this is a development issue, so no need to crosspost there... How about showing us the work you did? We need extensions.conf and sip.conf or iax.conf for your clients before we can even guess what's going on. Always provide more information rather than less. We can weed through the parts unnecessary parts. - Andrew Thompson http://www.retirequickly.com/43653 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4460 - 14 msgs
Hi! Skip this part - registration is only for dynamic IPs while you work with host=xxx.xxx.xxx.xxx entries here, and - besides - you can't have two *dynamic* servers register with each other. Cheers, Philipp then you need to register with each other in box1 iax.conf register=box1:[EMAIL PROTECTED]:5036 and in box2 iax.conf register=box2:[EMAIL PROTECTED]:5036 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
NO NO NO.change your EMAIL ADDRESS. You cannot have a . in your email address. Try [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: This is all I get. Can I send /msg -messages? My name has no symbols like ¤%¤%¤%... * Connecting to irc.freenode.net (6667) - -irc.freenode.net- *** Looking up your hostname... - -irc.freenode.net- *** Checking ident - -irc.freenode.net- *** Couldn't look up your hostname - -irc.freenode.net- *** No identd (auth) response - Closing Link: StiX (Invalid username [~antti.loh]) - * Disconnected -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to differentiate a *busy* call from not available?
Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm trying to find a way to differentiate wether a SIP extension is currently busy (e.g. on the phone) or not registered. So i do something like: exten = 100,1,Dial(SIP/foo,20,tr) exten = 100,2,VoiceMail,u100 exten = 100,102,VoiceMail,b100 If the phone doesn't answer I get the message: User is not available if the phone is currently in used i get the message: User is on the phone But if the phone is unplugged, I also get the message: User is on the phone! Any ideas? Thank you Jean-Yves Apparently you just ran to the list to beg rather than even atempt to find answer for yourself. This is covered frequently on the list. I could have sworn I just saw some threads about DIALSTATUS in the last few days, but I can't find them. Anyway, see: http://www.voip-info.org/tiki-print.php?page=Asterisk+cmd+Goto - Andrew Thompson http://aktzero.com http://www.retirequickly.com/43653 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
I've attached the /proc/pci below, but I think it's hardware related, not os - the dell does not seem to recognise that there is a card in the slot. Or any slot I put it in :( Thanks for the help, though. Julian. [EMAIL PROTECTED] root]# cat /proc/pci PCI devices found: Bus 0, device 0, function 0: Host bridge: ServerWorks CNB20HE Host Bridge (rev 33). Master Capable. Latency=32. Bus 0, device 0, function 1: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 1). Master Capable. Latency=32. Bus 0, device 0, function 2: Host bridge: ServerWorks CNB20HE Host Bridge (rev 0). Master Capable. Latency=32. Bus 0, device 0, function 3: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 0). Master Capable. Latency=32. Bus 0, device 4, function 0: VGA compatible controller: ATI Technologies Inc 3D Rage IIC (rev 122). Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xfc00 [0xfcff]. I/O at 0xec00 [0xecff]. Non-prefetchable 32 bit memory at 0xfbeff000 [0xfbef]. Bus 0, device 8, function 0: Ethernet controller: Intel Corp. 82557/8/9 [Ethernet Pro 100] (rev 8). IRQ 26. Master Capable. Latency=32. Min Gnt=8.Max Lat=56. Non-prefetchable 32 bit memory at 0xfbefe000 [0xfbefefff]. I/O at 0xe8c0 [0xe8ff]. Non-prefetchable 32 bit memory at 0xfbd0 [0xfbdf]. Bus 0, device 15, function 0: ISA bridge: ServerWorks OSB4 South Bridge (rev 80). Bus 0, device 15, function 1: IDE interface: ServerWorks OSB4 IDE Controller (rev 0). Master Capable. Latency=64. I/O at 0x8b0 [0x8bf]. Bus 3, device 9, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 0, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (#2) (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 1, function 0: SCSI storage controller: QLogic Corp. ISP12160 Dual Channel Ultra3 SCSI Proc essor (rev 6). IRQ 27. Master Capable. Latency=32. Min Gnt=64. I/O at 0xcc00 [0xccff]. Non-prefetchable 32 bit memory at 0xfaaff000 [0xfaaf]. Bus 5, device 0, function 0: RAID bus controller: American Megatrends Inc. MegaRAID (rev 32). IRQ 23. Master Capable. Latency=32. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 14:14 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p What have you got in /proc/pci? Do you have to do anything funny on the Dell to tell it that a card is there? Maybe it has some kind of health monitoring that you can switch off? -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has
[Asterisk-Users] vonage.ca * integration possible?
I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. No. TimeSourceDestination Protocol Info 222 53.601179 172.21.5.102 216.115.25.187SIP Request: REGISTER sip:bspgroup1.bsp.vonage.net:5061 Frame 222 (622 bytes on wire, 622 bytes captured) Ethernet II, Src: 00:0f:9f:86:42:d4, Dst: 00:06:25:db:aa:25 Internet Protocol, Src Addr: 172.21.5.102 (172.21.5.102), Dst Addr: 216.115.25.187 (216.115.25.187) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061) Session Initiation Protocol Request-Line: REGISTER sip:bspgroup1.bsp.vonage.net:5061 SIP/2.0 Method: REGISTER Resent Packet: False Message Header From: sip:[EMAIL PROTECTED]:5061;tag=ac150566-13c5-40eca012-eaee0a8-76e4;user=phone To: sip:[EMAIL PROTECTED]:5061;user=phone Call-ID: ac150566-13c5-40e8ddde-51e6-2327-0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 172.21.5.102:5061;branch=z9hG4bK-40eca012-eaee0a8-474d User-Agent: Motorola VT1000 mac: 000F9F8X sw:VT20_1.1.16e ln:0 cfg:10886711X/10022X Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5061;user=phone;transport=udp Expires: 900 Content-Length:0 No. TimeSourceDestination Protocol Info 224 53.711988 172.21.5.102 216.115.25.187SIP Request: REGISTER sip:bspgroup1.bsp.vonage.net:5061 Frame 224 (713 bytes on wire, 713 bytes captured) Ethernet II, Src: 00:0f:9f:86:42:d4, Dst: 00:06:25:db:aa:25 Internet Protocol, Src Addr: 172.21.5.102 (172.21.5.102), Dst Addr: 216.115.25.187 (216.115.25.187) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061) Session Initiation Protocol Request-Line: REGISTER sip:bspgroup1.bsp.vonage.net:5061 SIP/2.0 Method: REGISTER Resent Packet: False Message Header From: sip:[EMAIL PROTECTED]:5061;tag=ac150566-13c5-40eca012-eaee0a8-76e4;user=phone To: sip:[EMAIL PROTECTED]:5061;user=phone Call-ID: ac150566-13c5-40e8ddde-51e6-2327-0 CSeq: 2 REGISTER Via: SIP/2.0/UDP 172.21.5.102:5061;branch=z9hG4bK-40eca012-eaee10c-2713 Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5061;user=phone;transport=udp Expires: 900 Authorization: Digest username=1905XXX, realm=216.115.25.187, nonce=720170349, uri=sip:bspgroup1.bsp.vonage.net:5061, response=6a2fe5ec7b98a098aaf82a7dfc1340aa, algorithm=MD5 Content-Length:0 No. TimeSourceDestination Protocol Info 234 67.817617 172.21.5.102 216.115.25.187SIP Request: REGISTER sip:bspgroup1.bsp.vonage.net:5061 Frame 234 (622 bytes on wire, 622 bytes captured) Ethernet II, Src: 00:0f:9f:86:42:d4, Dst: 00:06:25:db:aa:25 Internet Protocol, Src Addr: 172.21.5.102 (172.21.5.102), Dst Addr: 216.115.25.187 (216.115.25.187) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061) Session Initiation Protocol Request-Line: REGISTER sip:bspgroup1.bsp.vonage.net:5061 SIP/2.0 Method: REGISTER Resent Packet: False Message Header From: sip:[EMAIL PROTECTED]:5061;tag=ac150566-13c5-40eca020-eaf1830-3c4e;user=phone To: sip:[EMAIL PROTECTED]:5061;user=phone Call-ID: ac150566-13c5-40e8ddde-51e6-2327-0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 172.21.5.102:5061;branch=z9hG4bK-40eca020-eaf1830-216f User-Agent: Motorola VT1000 mac: 000F9F8X sw:VT20_1.1.16e ln:0 cfg:1088671XX/100225 Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5061;user=phone;transport=udp Expires: 900 Content-Length:0 No. TimeSourceDestination Protocol Info 245 76.007450 172.21.5.102 216.115.25.187SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5061, with session description Frame 245 (972 bytes on wire, 972 bytes captured) Ethernet II, Src: 00:0f:9f:86:42:d4, Dst: 00:06:25:db:aa:25 Internet Protocol, Src Addr: 172.21.5.102 (172.21.5.102), Dst Addr: 216.115.25.187 (216.115.25.187) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Method: INVITE Resent Packet: False Message Header From: 905-XXX-sip:[EMAIL PROTECTED]:5061;tag=ac150566-13c5-40eca028-eaf3828-4f5a;user=phone To: sip:[EMAIL PROTECTED]:5061;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP
Re: [Asterisk-Users] internal external SIP
From: Jon Lawrence Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. .. snip .. to 82.145.37.29:5060 Jul 9 12:41:49 WARNING[5126]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 7712 (Response) set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 81.168.4.69, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 81.168.4.67:5060;branch=z9hG4bK201b0b71 From: 2003 sip:[EMAIL PROTECTED];tag=as3f8ccbff To: sip:[EMAIL PROTECTED];tag=0939785f3bc7641e Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Closing Link: StiX (Invalid username [~antti.loh]) Remove the dot in your username. Maybe you make it antilope :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SNMP Monitoring (Andrea Fino)
Thanks for this informations. Do you know where I can find the icd-snmp package for a redhat 9 distri? I can't find it. Thanks. Message: 6 Date: Fri, 09 Jul 2004 15:45:57 +0200 From: Andrea Fino [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SNMP Monitoring Reply-To: [EMAIL PROTECTED] Holger Schurig wrote: You could try ast_snmpd at http://faino.it/ast-ax-snmpd.html I hope to get some substantatial progress in it during the august holiday. I've added your web page to the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+addons ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is an English version at http://faino.it/en/ast-ax-snmpd.html sorry, I forgot. Andrea Fino -- Andrea Fino 8-) - Sistemi su misura di qualita' industriale Handcrafted systems with industrial quality [Phone: +39 071 2070104]+[Fax: +39 071 2077919]+[iaxtel:17002876594] [Web: http://www.faino.org]+[Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP hackers gut Caller ID
On Wed, 7 Jul 2004, brian wrote: Anyone with a PRI/ISDN line can set callerid to anything... Not just voip, not just asterisk. Come on guys. bkw Yes, but the Telco has the ability to either pass or deny that. In my X/O PRI configuration, I can only set the CallerID to a number within the vliad block of DIDs assigned to that PRI group. This prevents willy nilly abuse. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM to iLBC one way audio :-(
Hi, I'm using IAXphone for remote users which limits me to the GSM codec. Internally Ilimit the SIP phones to iLBC codec (GS 101 1.0.5.0) I also use voiptalk.org for external PSTN access again using the iLBC codec. The problem I have is that when the IAXphone dials an internal phone or PSTN number either the line hangs up immediately or there is only one way audio from IAXphone. It works as soon as I allowGSM codec on the GS phones. Is there any way I can debug this issue? Asterisk CVS-04/10/04-15:32:35 built by [EMAIL PROTECTED] on a i686 running Linux Cheers Giles Scott
Re: [Asterisk-Users] vonage.ca * integration possible?
Your problem with doing this is this line right below...you have no idea what your authentication secret is. This is a closely guarded secret of Vonage. They don't have any interest in letting anyone do this. The closest you could do would be a softphone, unlimited inbound and 500 mins outbound calling. There are sample configs floating around out there to make that work. On Fri, 9 Jul 2004 10:28:06 -0400 (EDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Authorization: Digest username=1905XXX, realm=216.115.25.187, nonce=720170349, uri=sip:bspgroup1.bsp.vonage.net:5061, response=6a2fe5ec7b98a098aaf82a7dfc1340aa, algorithm=MD5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and AudioCodes MP124
I have problem in configuring MP124 FXS Gateway to work with *. Can anaybody help me in this way? - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] strange echo problem
I can consistently recreate the problem and will be happy to give a developer or two a sip or iax account on one of my test asterisk boxes to play with. It would take me a week or so to get a new box up w/ a t1 card that we are constantly messing with. W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com Trimmed for electron conservation -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, July 09, 2004 9:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] strange echo problem We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some Personal opinion (based only on the various words posted to this list) is there still is an echo can issue floating in * that none of us have documented with sufficient data that would guide a developer towards the root cause. If you easily recreate the problem, that would go a long ways towards gathering qualifying data. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco MC3810 - Asterisk
Are you also using it for outbound pstn connections? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Fernandez Sent: Friday, July 09, 2004 9:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco MC3810 - Asterisk I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated Grandstream configurator
On Thu, 8 Jul 2004, Neil Cherry wrote: Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Thanks, I've been having real trouble with those stupid DLLs. I can't upgrade some of them no matter what I do (WIN2K)! I downloaded and installed it on a Windows 2000 server box. The install complained about me having a newer .dll file than what was being installed, so I chose to keep my version. Seems to work fine. Does this support the new 1.0.5.x firmware train? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
To fix that, I did have to enable identd in my mIRC options and forward the TCP port to my machine (as noted previously). Some IRC servers in the freenode network require it, some don't. So, get that working and make sure your nickname includes only alphanumeric characters, and you'll be fine. At 06:46 AM 7/9/2004, you wrote: This is all I get. Can I send /msg -messages? My name has no symbols like ¤%¤%¤%... * Connecting to irc.freenode.net (6667) - -irc.freenode.net- *** Looking up your hostname... - -irc.freenode.net- *** Checking ident - -irc.freenode.net- *** Couldn't look up your hostname - -irc.freenode.net- *** No identd (auth) response - Closing Link: StiX (Invalid username [~antti.loh]) - * Disconnected -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 Hardware Echo Can
Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't have a clue what I am talking about. Does anyone have any good companies with competent sales/engineer people who would help put together a solution. Also, for anyone that has hooked up a echo can before. Do you have to buy such a large shelf? Obviously things things are intended for ILEC installs, however, I can't find anything geared towards the PBX realm. It seems everything on Ebay is a 32 module shelf rack. Thats a bit over kill for us. Further, I would imagine this setup. Please correct me if I am off base. I would have a straight T1 cable from the Channel Bank to the Echo Can, and then a X-over from * to the Echo Can. How are these solutions (e.g. Tellabs) wired up? Are there RJ45 connectors on the back of the shelf, or is it a strip the wire and twist method? Any assistance that anyone can provide to myself (and the list) would be greatly appreciated, as there are many people who would benefit from this... Thanks! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Monitor cmd and Queues
[EMAIL PROTECTED] wrote: When was this implimented? I am currently running CVS-HEAD-07/07/04-09:34:29 --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of brian Sent: Thursday, July 08, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Monitor cmd and Queues Get the lastest CVS head it can start monitoring from time the agent picks up the phone. So you get ZERO hold music. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Harold Workman Sent: Thursday, July 08, 2004 5:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Monitor cmd and Queues Hi, Id like to record a queue conversation using the Monitor command but the problem im running into is the way i configure it * records the music on hold along with the conversation. Is there a way to start recording when the call is picked up by an agent? Could someone give me a small example of how they set this up. My current extensions.conf portion of the file looks like: [cytelbilling] ;# Cytel Communications Billing Support ### exten = s,1,Background(/usr/src/asterisk-sounds/sounds/this-call-may-be) exten = s,2,Background(/usr/src/asterisk-sounds/sounds/monitored) exten = s,3,Background(/usr/src/asterisk-sounds/sounds/or) exten = s,4,Background(/usr/src/asterisk-sounds/sounds/recorded) exten = s,5,Background(/usr/src/asterisk-sounds/sounds/for-quality-purposes) exten = s,6,Answer exten = s,7,SetMusicOnHold(default) exten = s,8,DigitTimeout,5 exten = s,9,ResponseTimeout,10 exten = s,10,Background(/usr/src/asterisk-sounds/sounds/pls-stay-on-line) exten = s,11,Monitor(wav,,m) exten = s,12,Queue(cytelcs) --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I upgraded to CVS-HEAD-07/09/04-09:06:28 and with my current configuration it still records my music on hold. Is there any special configuration I need to add to force it not to record the moh before an agent picks up? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] vonage.ca * integration possible?
Sure it's possible. It's probably not trivial though -- for one, you'd need to get the password somehow so that you can produce the md5 response. It probably wouldn't be very stable either, since Vonage keeps changing rotating the passwords (did so with the ATA186 at least) at least once a week, and since you're violating their Terms of Use, they may shut you down anytime. However, the Vonage Softphone service works JUST FINE with asterisk, no special hacks required. I have it running with a $15/month hardline and $10/month softphone so I get my incoming calls in a rate-center that nobody else offers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] vonage.ca * integration possible? I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Unstable Claims Asterisk 1.0-1
Howdy, I just did an apt-get dist-upgrade on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. -rw-rw-r--1 1176 1176 1189696 Jun 7 04:47 asterisk_1.0-1_alpha.deb -rw-rw-r--1 1176 1176 1023342 Jun 1 13:47 asterisk_1.0-1_arm.deb -rw-rw-r--1 1176 1176 27930 May 31 22:17 asterisk_1.0-1.diff.gz -rw-rw-r--1 1176 1176 824 May 31 22:17 asterisk_1.0-1.dsc -rw-rw-r--1 1176 1176 1168060 Jun 1 02:17 asterisk_1.0-1_hppa.deb -rw-rw-r--1 1176 1176 955152 May 31 22:17 asterisk_1.0-1_i386.deb -rw-rw-r--1 1176 1176 1402768 Jun 1 18:17 asterisk_1.0-1_ia64.deb -rw-rw-r--1 1176 1176 973798 Jun 1 02:32 asterisk_1.0-1_m68k.deb -rw-rw-r--1 1176 1176 1008990 Jun 1 00:02 asterisk_1.0-1_mips.deb -rw-rw-r--1 1176 1176 1013994 Jun 1 02:02 asterisk_1.0-1_mipsel.deb -rw-rw-r--1 1176 1176 1149186 Jun 4 02:17 asterisk_1.0-1_powerpc.deb -rw-rw-r--1 1176 1176 1067622 Jun 1 16:17 asterisk_1.0-1_s390.deb -rw-rw-r--1 1176 1176 1028522 Jun 1 02:32 asterisk_1.0-1_sparc.deb -rw-rw-r--1 1176 1176 2808513 May 31 22:17 asterisk_1.0.orig.tar.gz -rw-rw-r--1 1176 1176 67474 May 31 22:17 asterisk-dev_1.0-1_all.deb -rw-rw-r--1 1176 1176 1219912 May 31 22:17 asterisk-doc_1.0-1_all.deb -rw-rw-r--1 1176 1176 1347486 May 31 22:17 asterisk-sounds_1.0-1_all.deb -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SNMP Monitoring (Andrea Fino)
GIBERT Frédéric wrote: Thanks for this informations. Do you know where I can find the icd-snmp package for a redhat 9 distri? I can't find it. Thanks. I guess yoy have to download the sources from the net-snmp site ucd-snmp is not present in actual distribution, because net-snmp 5.1x is more recent. Hopefully we'll have an ast_snmpd stuff net-snmp 5.1.x based in the next months. Regards, Andrea Fino -- Andrea Fino 8-) - Sistemi su misura di qualita' industriale Handcrafted systems with industrial quality [Phone: +39 071 2070104]+[Fax: +39 071 2077919]+[iaxtel:17002876594] [Web: http://www.faino.org]+[Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two outbound calls at once
On Thu, 8 Jul 2004, David Goldfein waxed: Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI 8's It doesn't look like you have a channel collision problem, other than the same far end number being dialed. Are you able to place at least one call with success ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
I have to agree with andy's comments the lack of documentation is * biggest downfall. Andy gave me a lot of help getting * up and running, without much of the help I probably would have not been able to see the full potention of *. Kind Regards, Chris Bond Late last year I was approached by a publisher asking if I would be interested in writing an asterisk book. I said a polite no (after some discussion) for a number of reasons: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Problem of loading the oh-323 module
Remarque : message transféré en pièce jointe. Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com---BeginMessage--- Hello everybody, I am still working on Asterisk, everything worked fine untill now, but now my problem is in loading oh323 module by Asterisk: The error that I have is : [chan_oh323.so]Jul 9 14:11:11 WARNING[1076298368]: loader.c:242 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 9 14:11:11 WARNING[1076298368]: loader.c:374 load_modules: Loading module chan_oh323.so failed! I am using the following versions: asterisk-oh323-0.6.3a openh323-v1_13_5-src pwlib-v1_6_6-src CVS HEAD sources of Asterisk (29/06/2004) Before, I had older versions of asterisk-oh323, openh323 and pwlib, so I had a lot of problems in compiling the different elements, and now that everything compile, I cannot load the oh323 module...maybe should I upgrade aserisk, but if it is the case should I recompile the others again? I am afraid of having bad surprises... Can someone help me urgently please?? Thank you very much Soumaya Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com ---End Message---
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
You're so right Harold. :-) We have a lot on our plate right now, but plan to place information into pdf form over time. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Harold Workman Sent: Friday, July 09, 2004 6:44 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. William, I guess it was a bad idea to have on the website one of the examples stating Much of the information in the book came from the Asterisk Wiki pages. Which is the sole reason for my bashing on the book. I can see having a book like this will ease others concerns on running a free open source pbx that are not familiar with Asterisk. There can never be too much documentation. What are the signate's thoughts on having a lower cost PDF file for download? --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Boehlke Sent: Thursday, July 08, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. As an interested party to the Mahler book VoIP Telephony with Asterisk, I would like to clarify a point about it. We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki instead of buying the book, and we acknowledge the Wiki (that's where the confusion began) as a source of some material. That acknowledgement is in one of the sample pages on our site, which probably began the confusion. But the book is the 320 page product of nine months of independent work. We built a dozen systems during the course of writing it to test and implement features and options. Mark Spencer and the Digium technical staff contributed information, advice and equipment. Eight competent engineers, some of them active here every day, gave it a technical review. As a former beginner, in my opinion the index alone is worth the price compared to the time it took me to locate information. Thanks for listening. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Thursday, July 08, 2004 2:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
Well, you should see an entry like this: Bus 0, device 11, function 0: Communication controller: PCI device 10ee:0314 (Xilinx Corporation) (rev 1). IRQ 5. Master Capable. Latency=64. Non-prefetchable 32 bit memory at 0xda001000 [0xda00107f]. Any curious messages in dmesg when the machine is booted, any settings in the bios related to PCI? At least from this point you can discount any zaptel issues as this shows regardless of whether zaptel is loaded or not. Steve -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 15:30 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p I've attached the /proc/pci below, but I think it's hardware related, not os - the dell does not seem to recognise that there is a card in the slot. Or any slot I put it in :( Thanks for the help, though. Julian. [EMAIL PROTECTED] root]# cat /proc/pci PCI devices found: Bus 0, device 0, function 0: Host bridge: ServerWorks CNB20HE Host Bridge (rev 33). Master Capable. Latency=32. Bus 0, device 0, function 1: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 1). Master Capable. Latency=32. Bus 0, device 0, function 2: Host bridge: ServerWorks CNB20HE Host Bridge (rev 0). Master Capable. Latency=32. Bus 0, device 0, function 3: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 0). Master Capable. Latency=32. Bus 0, device 4, function 0: VGA compatible controller: ATI Technologies Inc 3D Rage IIC (rev 122). Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xfc00 [0xfcff]. I/O at 0xec00 [0xecff]. Non-prefetchable 32 bit memory at 0xfbeff000 [0xfbef]. Bus 0, device 8, function 0: Ethernet controller: Intel Corp. 82557/8/9 [Ethernet Pro 100] (rev 8). IRQ 26. Master Capable. Latency=32. Min Gnt=8.Max Lat=56. Non-prefetchable 32 bit memory at 0xfbefe000 [0xfbefefff]. I/O at 0xe8c0 [0xe8ff]. Non-prefetchable 32 bit memory at 0xfbd0 [0xfbdf]. Bus 0, device 15, function 0: ISA bridge: ServerWorks OSB4 South Bridge (rev 80). Bus 0, device 15, function 1: IDE interface: ServerWorks OSB4 IDE Controller (rev 0). Master Capable. Latency=64. I/O at 0x8b0 [0x8bf]. Bus 3, device 9, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 0, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (#2) (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 1, function 0: SCSI storage controller: QLogic Corp. ISP12160 Dual Channel Ultra3 SCSI Proc essor (rev 6). IRQ 27. Master Capable. Latency=32. Min Gnt=64. I/O at 0xcc00 [0xccff]. Non-prefetchable 32 bit memory at 0xfaaff000 [0xfaaf]. Bus 5, device 0, function 0: RAID bus controller: American Megatrends Inc. MegaRAID (rev 32). IRQ 23. Master Capable. Latency=32. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 14:14 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p What have you got in /proc/pci? Do you have to do anything funny on the Dell to tell it that a card is there? Maybe it has some kind of health monitoring that you can switch off? -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going
[Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726
I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 - the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1478 ast_set_write_format: Unable to find a path from ILBC to G726 Jul 9 13:15:37 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (40)? Jul 9 13:15:37 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to transmit frame type 1024, while native formats is 16 (read/write = 64/1024) I will appreciate any help. Kind regards, Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Unstable Claims Asterisk 1.0-1
On Fri, 9 Jul 2004, Greg Boehnlein wrote: [...] who is responsible for the Debian packages? I believe the responsible is Mark Purcell = msp at debian dot org I sent an email last week and received no reply so far... asterisk*CLI show version Asterisk 0.7.2 built by msp at dell dot purcell dot homeip dot net on a i686 running Linux -- Lista asterisk em portugues: http://groups.yahoo.com/group/asteriskbr/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Unstable Claims Asterisk 1.0-1
apt-cache show is the command you want. pbx:/etc/asterisk# apt-cache show asterisk Package: asterisk Priority: optional Section: comm Installed-Size: 2772 Maintainer: Mark Purcell [EMAIL PROTECTED] Architecture: i386 Version: 1.0-1 Greg Boehnlein [EMAIL PROTECTED] [2004-07-09 11:36:28 -0400]: Howdy, I just did an apt-get dist-upgrade on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. -rw-rw-r--1 1176 1176 1189696 Jun 7 04:47 asterisk_1.0-1_alpha.deb -rw-rw-r--1 1176 1176 1023342 Jun 1 13:47 asterisk_1.0-1_arm.deb -rw-rw-r--1 1176 1176 27930 May 31 22:17 asterisk_1.0-1.diff.gz -rw-rw-r--1 1176 1176 824 May 31 22:17 asterisk_1.0-1.dsc -rw-rw-r--1 1176 1176 1168060 Jun 1 02:17 asterisk_1.0-1_hppa.deb -rw-rw-r--1 1176 1176 955152 May 31 22:17 asterisk_1.0-1_i386.deb -rw-rw-r--1 1176 1176 1402768 Jun 1 18:17 asterisk_1.0-1_ia64.deb -rw-rw-r--1 1176 1176 973798 Jun 1 02:32 asterisk_1.0-1_m68k.deb -rw-rw-r--1 1176 1176 1008990 Jun 1 00:02 asterisk_1.0-1_mips.deb -rw-rw-r--1 1176 1176 1013994 Jun 1 02:02 asterisk_1.0-1_mipsel.deb -rw-rw-r--1 1176 1176 1149186 Jun 4 02:17 asterisk_1.0-1_powerpc.deb -rw-rw-r--1 1176 1176 1067622 Jun 1 16:17 asterisk_1.0-1_s390.deb -rw-rw-r--1 1176 1176 1028522 Jun 1 02:32 asterisk_1.0-1_sparc.deb -rw-rw-r--1 1176 1176 2808513 May 31 22:17 asterisk_1.0.orig.tar.gz -rw-rw-r--1 1176 1176 67474 May 31 22:17 asterisk-dev_1.0-1_all.deb -rw-rw-r--1 1176 1176 1219912 May 31 22:17 asterisk-doc_1.0-1_all.deb -rw-rw-r--1 1176 1176 1347486 May 31 22:17 asterisk-sounds_1.0-1_all.deb -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc - TE mode - callerid trouble
I've got a bit trouble with callerid and zaphfc cards.
Basically zaphfc doesn't add the 0 in front of national numbers
(haven't tried a international call yet).
With chan_capi that allways worked fine, however i had to define the
national and international prefixes in capi.conf.
Is there something similar in zapata.conf ?
Here is my zapata.conf:
[channels]
musiconhold=default
;
; ISDN
;
switchtype = euroisdn ; HFC-S TE mode
signalling = bri_cpe_ptmp
prilocaldialplan = national
pridialplan = unknown
echocancel = yes
immediate= yes
group= 1
context = inbound-zap
channel = 1-2
switchtype = euroisdn ; HFC-S NT mode
signalling = bri_net_ptmp
prilocaldialplan = local
overlapdial = no
echocancel = yes
setcallerid = ( ${CALLERIDNUM})
group= 2
immediate= no
context = inbound-internal
channel = 4-5
;
; PSTN
;
signalling = fxs_ks ; X100P
group = 1
echocancel = yes
usecallerid = yes
context = inbound-zap
immediate = no
channel = 7
signalling = fxo_ks ; TDM400
group = 3
context = inbound-internal
immediate = no
channel = 8-11
A d-channel analyzer on the ISDN line gives me a correct setup (beyond
some Eircom specialities, like a truncated called party MSN):
SETUP
Sending complete
Bearer capability
Coding CCITT
Info. transfer capability Speech
Transfer mode/rate Circuit mode, 64 kbps
Channel identification
Interface identificationImplicitly
Interface type Basic interface
Allocation priority Exclusive
Channel B2-channel
Calling party number
Type of number National number
Numbering plan Isdn/telephony (E.164)
Presentation indicator Presentation allowed
Screening indicator Network provided
Number 876218425
Called party number
Type of number Unknown
Numbering plan Isdn/telephony (E.164)
Number 3987
Any suggestions on what could be wrong ?
I have tried different values for prilocaldialplan and pridialplan on
the TE mode HFC-S card, but no joy.
Kind regards,
Martin List-Petersen
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Re: [Asterisk-Users] Cisco MC3810 - Asterisk
Hi Alberto, I'm wondering if my image might be the problem - I have 12.3.9 on the device - released at some point in may of this year. I've got everything (including the kitchen sink) in terms of feature set. Can you post some of the relevant snippets of your config? I'd love to see how this is done. Graeme On Fri, 9 Jul 2004, Alberto Fernandez wrote: I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - Call from a Grandstream (working)- Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing NoOp(SIP/4000-98ec, ) in new stack -- Executing Goto(SIP/4000-98ec, intern-post|4001|1) in new stack -- Goto (intern-post,4001,1) -- Executing Dial(SIP/4000-98ec, SIP/4001|30|Ttm) in new stack Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO URL) Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on RTP to 0 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for 4001 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a local user -- Called 4001 Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel 'SIP/4000-98ec' Call from the Cisco (not working) Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing NoOp(SIP/192.168.1.9-08134bb8, ) in new stack -- Executing Goto(SIP/192.168.1.9-08134bb8, from-sip-post|4001|1) in new stack -- Goto (from-sip-post,4001,1) Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel 'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in context 'from-sip-post', but no invalid handler BTW- Working with a ripped-off version of John Todd's configs... Anyone get this working? It's kicking my ass. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco MC3810 - Asterisk
Hi Kevin, Not using this one for outbound connections, I have gotten outbound working through a 3810 with an MFT T1 full of channelized voice though. That worked pretty well. This 3810 will hopefully be in my basement running all the non-ip phones in my house. Graeme On Fri, 9 Jul 2004, W. Kevin Hunt wrote: Are you also using it for outbound pstn connections? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Fernandez Sent: Friday, July 09, 2004 9:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco MC3810 - Asterisk I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Ok, then I suggest you have a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - TE mode - callerid trouble
Hi MLP
nationalprefix=0
internationalprefix=00
Regards,
Martin List-Petersen wrote:
I've got a bit trouble with callerid and zaphfc cards.
Basically zaphfc doesn't add the 0 in front of national numbers
(haven't tried a international call yet).
With chan_capi that allways worked fine, however i had to define the
national and international prefixes in capi.conf.
Is there something similar in zapata.conf ?
Here is my zapata.conf:
[channels]
musiconhold=default
;
; ISDN
;
switchtype = euroisdn ; HFC-S TE mode
signalling = bri_cpe_ptmp
prilocaldialplan = national
pridialplan = unknown
echocancel = yes
immediate= yes
group= 1
context = inbound-zap
channel = 1-2
switchtype = euroisdn ; HFC-S NT mode
signalling = bri_net_ptmp
prilocaldialplan = local
overlapdial = no
echocancel = yes
setcallerid = ( ${CALLERIDNUM})
group= 2
immediate= no
context = inbound-internal
channel = 4-5
;
; PSTN
;
signalling = fxs_ks ; X100P
group = 1
echocancel = yes
usecallerid = yes
context = inbound-zap
immediate = no
channel = 7
signalling = fxo_ks ; TDM400
group = 3
context = inbound-internal
immediate = no
channel = 8-11
A d-channel analyzer on the ISDN line gives me a correct setup (beyond
some Eircom specialities, like a truncated called party MSN):
SETUP
Sending complete
Bearer capability
Coding CCITT
Info. transfer capability Speech
Transfer mode/rate Circuit mode, 64 kbps
Channel identification
Interface identificationImplicitly
Interface type Basic interface
Allocation priority Exclusive
Channel B2-channel
Calling party number
Type of number National number
Numbering plan Isdn/telephony (E.164)
Presentation indicator Presentation allowed
Screening indicator Network provided
Number 876218425
Called party number
Type of number Unknown
Numbering plan Isdn/telephony (E.164)
Number 3987
Any suggestions on what could be wrong ?
I have tried different values for prilocaldialplan and pridialplan on
the TE mode HFC-S card, but no joy.
Kind regards,
Martin List-Petersen
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Re: [Asterisk-Users] internal external SIP
[EMAIL PROTECTED] (Jon Lawrence) writes: codec's are set to allow all. Thats your problem. I tried this too as an experiment and asterisk appears to take all to mean all codecs you can think of, not just the ones you have converters for. Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726
I see no errors.. I see three NOTICES and two WARNINGS. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, July 09, 2004 11:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726 I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 - the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1478 ast_set_write_format: Unable to find a path from ILBC to G726 Jul 9 13:15:37 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (40)? Jul 9 13:15:37 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to transmit frame type 1024, while native formats is 16 (read/write = 64/1024) I will appreciate any help. Kind regards, Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco MC3810 - Asterisk
Yes, i was afraid of the digium hardware. its been working for a while. i have a t1 connected to it. I will replace it with a t1 card im buying from digium. On Fri, 2004-07-09 at 11:12, W. Kevin Hunt wrote: Are you also using it for outbound pstn connections? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Fernandez Sent: Friday, July 09, 2004 9:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco MC3810 - Asterisk I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username pw to asterisk when I try to configure it as a client. Eg - http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intermittent SIP 404 Not Found response?
[EMAIL PROTECTED] (Andrew Yager) writes: I believe I'm experiencing the same problem with Grandstream phones, although I haven't had time to track it down yet. When your GS fails, slap a tcpdump on the line and have a look at what it is sending. When my GS fails it forgets how to route stuff on the internet and attempts to ARP for something that is halfway around the world (eg. sends an arp-request for the sip server even if that machine isn't local). I like GS's sound quality and price, but their firmware clearly has some serious corruption problems. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 config help and guidance
I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a euroISDN bearer. This bearer only has 10 channels activated (out of the 30). Obviously, this works - handsets make external calls. What I wanted to do was to add * to the mix, in the middle so that it can intercept inbound / outbound calls and do what it needs to do, as well as providing all the extra functionality that this wonderful product provides. In order to achieve this, I assumed that I needed to take rj45 from the bearer box and plug that into span 2, and take a cable from span 1 into the bearer box. My problem (and blurry eyes) come from not understanding the various protocols to assign to each span. I want the meridian to think that it's still plugged into the EuroISDN bearer. So span 2 should be set up as a EuroISDN link ? What should span 1 be set up as ? What channels should be configured ? Any guidance (I'm not looking for the solution (would be nice!) but for pointers in the right direction). I have previously been able to set up asterisk using the x100p and graduated to BRI isdn. I just got the 405 today and wanted to play! Thanks in advance. Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
