Re: [asterisk-users] DIDs required of Paris and Gottenburg Sweden
On Fri, Jul 4, 2008 at 7:25 AM, Kashif Naeem [EMAIL PROTECTED] wrote: We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ? IdeaSIP.com can provide this. I don't know their rates, see the site for that. Their call quality is excellent. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
Aside from the 5g phone that will come out as soon as you plunk down
$300 for the 3g ($800 if you calculate your 2 year contract
obligation), don't forget to join us today for The S Word: Security
Most of you on this list will be more qualified than I am to discuss
or even list the issues involved, but I would start with these:
* What are the principal risks?
DoS
Fraudulent usage of your minutes
Compromising your user accounts (example, getting all the emails, CID, etc)
Making your life miserable in various ways through resource abuse
* What's wrong with running as root?
* How to lock down your server
Denying access using standard *nix tools
Authentication
Checking against known attackers
Those are just a few ideas. Please join us for the call this Friday 4th of July.
See http://VoipUsersConference.org
IRC.Freenode.net #voip-users-conference
PSTN;: Call (724) 444-7444 and enter 22622# 1#
Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; by default
your PIN is 1#
ts.x2z.eu resolves to the above IP
http://food4wine.ning.com has news, forums, blogs, etc
http://voipuserstv.com has videos of Asterisk Tag and other asterisk
and voip stuff
RSS http://feeds.feedburner.com/AstUser
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] removing == Parsing '/etc/asterisk/manager.conf': Found from CLI!
hi all, is there any way of removing this line from showing on the console? my verbosity level is 3. and this is the following output on cli 24/7 unless its interrupted by the msgs tht really counts like connected sip and so on.. [Jul 4 10:32:38] NOTICE[18542]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:48] NOTICE[18543]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:58] NOTICE[18544]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:08] NOTICE[18545]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate -- Registered SIP '179' at 192.168.0.2 port 27780 expires 3600 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:18] NOTICE[18547]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] removing == Parsing '/etc/asterisk/manager.conf': Found from CLI!
RoLaNd RoLaNd wrote: hi all, /is there any way of removing this line from showing on the console? my verbosity level is 3. and this is the following output on cli 24/7 unless its interrupted by the msgs tht really counts like connected sip and so on../ [...] [Jul 4 10:32:38] NOTICE[18542]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:48] NOTICE[18543]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' [...] How about you start off by correcting your configuration and configuring your system correctly so that you don't have Flash Operator Panel trying to connect all the time using a non-existent password? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] removing == Parsing '/etc/asterisk/manager.conf': Found from CLI!
2008/7/4 RoLaNd RoLaNd [EMAIL PROTECTED]: hi all, is there any way of removing this line from showing on the console? my verbosity level is 3. and this is the following output on cli 24/7 unless its interrupted by the msgs tht really counts like connected sip and so on.. [snip] Stop reloading the manager every 10 seconds??? That would stop the messages. :) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug tracker having issues
( ! ) Fatal error: Maximum execution time of 30 seconds exceeded in
/var/www/insects.digium.com/core/config_api.php on line 32
Call Stack
# Time MemoryFunctionLocation
1 0.0006 98216{main}( )../bugnote_add.php:0
2 0.242013136160email_bugnote_add( long )
../bugnote_add.php:48
3 0.242013136160email_generic( long, string(7), string(53),
??? )../email_api.php:585
31.666024270536email_build_visible_bug_data( long, long,
string(53) )../email_api.php:449
532.482724621888relationship_get_summary_text( long )
../email_api.php:1134
632.583524630712relationship_get_details( long,
object(stdClass)[293], bool, ???, ??? )../relationship_api.php:668
732.587724635552bug_prepare_display( object(BugData)[296]
)../relationship_api.php:539
832.606724636776string_display_links( string(1354) )
../bug_api.php:1411
932.612124640016string_process_bug_link( string(1552), ???,
???, ??? )../string_api.php:101
1032.622424661480string_get_bug_view_link( string(1), null,
bool, bool )../string_api.php:264
1132.623124662448get_enum_element( string(6), string(2) )
../string_api.php:496
1232.623124662552config_get( string(18), ???, ???, ??? )
../helper_api.php:81
1332.623524665088auth_is_user_authenticated( )
../config_api.php:69
1432.623524665128auth_get_current_user_cookie( )
../authentication_api.php:55
1532.623524665288config_get( string(13), ???, ???, ??? )
../authentication_api.php:378
Dump $_SERVER
Dump $_GET
Dump $_POST
Dump $_COOKIE
Dump $_FILES
Dump $_ENV
Dump $_SESSION
Dump $_REQUEST
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Removing voicemail messages
Hello, I want to create an script which remove all the old voicemail messages. I make a simple Bash script to delete all the new messages for the extension 100. Something like, rm /var/spool/asterisk/voicemail/defaul/100/INBOX Should I update any index file or something after reemove them? Thanks in advance VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing voicemail messages
You don't want to remove the INBOX folder. Also just rm will want you to hit y to confirm. Try doing rm -rf /var/spool/asterisk/voicemail/default/100/INBOX/*.* On Fri, Jul 4, 2008 at 2:57 PM, voip crazy [EMAIL PROTECTED] wrote: Hello, I want to create an script which remove all the old voicemail messages. I make a simple Bash script to delete all the new messages for the extension 100. Something like, rm /var/spool/asterisk/voicemail/defaul/100/INBOX Should I update any index file or something after reemove them? Thanks in advance VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Forwarding Lopp Prevention
i have two extensions which have call forwarding enabled when they are busy to forward the caller to each other. 11 ==on busy== 12 12 ==on busy== 11 when both extensions are Busy a large number of stale calls will be made in the system! how can i prevent this mess in my system? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new install of asterisk appliance.
I have 1 nic card which is linked to the router. Then I use 1 port on the router which is linked to the asterisk appliance. It will work via WAN which ive now got. SO I can access the asterisk appliance via 192.168.1.15 The problem is now…How do I connect the phone. Ive got the phone (Ethernet) connected from the LAN port on the phone to a LAN port in the asterisk appliance. The short answer is that if you're not using the AA50 as a router you cannot use the LAN ports on the AA50. You'll have to connect your phone to a LAN port on your switch. As Rob mentioned, you'll need to configure the Grandstream manually. Also, if your SBS server or your router is providing DHCP be sure to turn off DHCP on the AA50. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding Lopp Prevention
Paradise Dove wrote:
i have two extensions which have call forwarding enabled when they are
busy to forward the caller to each other.
11 ==on busy== 12
12 ==on busy== 11
exten = 11,1,Set(GROUP()=Loop11_Detect)
exten = 11,n,NoOP(Loop Detect for Extension 11:
${GROUP_COUNT(Loop11_Detect)})
exten = 11,n,GotoIf($[ ${GROUP_COUNT(Loop11_Detect)} 2 ]?11,100)
exten = 11,n,Dial(SIP/12)
exten = 11,100,Voicemail([EMAIL PROTECTED]|b)
exten = 11,101,Hangup(17)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spoofing CID
On Jul 3, 2008, at 10:45 PM, Alex Balashov wrote: Alexander Lopez wrote: I may create an IVR Hell for them, so that I can transfer the calls to, Hey, Its their dime. If you do end up going that route, please share the details, and possibly the code. We could all benefit from a good IVR from hell. I certainly could use one. No, I'm not being sarcastic. I tried an IVR once before but they hang up immediately... instead I have a recorded message of me saying Hello then a wait 5 sec... and then an I'm sorry, ... yatta yatta yatta if youuse real number I'll take your call. I use a blacklist script to add numbers to and record the call. During that 5 sec wait they normally talk, so it provides me some sense of amusement (sadly). Fred Posner [EMAIL PROTECTED] www.teamforrest.com FWD#: 902963 smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] background noise
Hi all, I would like to know how can I immunize the background noise in my case. Anyone can help? I have adjusted txgain rxgain in different value but the result is the same. ango Below is my configuration. asterisk1.4.21.1 zaptel1.4.11 addon1.4.7 TDM400 (FXOx4) There is a very large background noise when a call from sip to PSTN. Below is the test case. A - sip phone B - PSTN case 1: result is normal caller A1 callee A2 case 2: A hears B is ok but A hears much background noise from his/her area when A say nothing. The sound volume of A is very low when B hears A. caller A callee B case 3: same as case 2 caller B callee A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing voicemail messages
I want to create an script which remove all the old voicemail messages.
I make a simple Bash script to delete all the new messages for the
extension 100. Something like,
rm /var/spool/asterisk/voicemail/defaul/100/INBOX
Should I update any index file or something after reemove them?
In my solution, I archive messages that haven't been touched in x number
days and send the compilation of messages being archived to the
voicemailbox's owner. You could modify to delete them and leave it at that.
Since I'm leaving newer messages in the directory, and Asterisk doesn't like
nonsequential filenames - I renumber all the remaining files.
I also had an issue with abandoned voicemail - where the caller would hangup
at the exact right moment leaving a zero length message. The .txt file
would be created but none of the audio formats. It was a rare occurrence -
but the user was unable to fix it from the phone interface meaning I had to
manually correct it every time, so I automated it with a script and a
cronjob.
My crack at these is below, these are a work in progress - use at your own
risk, suggested improvements are welcome.
--- age_move_voicemail.sh ---
#!/bin/bash
# Script to automatically archive old voicemail messages
# Based on modified date of files, when messages are listened to the date is
updated.
# Not an issue in this implementation as users mostly listen to e-mail
attachment and don't
# touch the phone.
# Archived messages are also e-mailed to the voicemailbox owner.
# Possible future iteration: Read the date of the messages directly from the
.txt file
# Future Improvement: Read options from config file to allow defaults and
mailbox
# specific modifications.
# Possible future improvement: Use AGI or other interface to allow users to
modify archive
# options via the telephone. If this proves too difficult, perhaps an e-mail
# or web interface would work.
archivefolder=/home/asterisk/oldvoicemail
voicemailroot=/var/spool/asterisk/voicemail/default
process=yes
logging=yes
defaultdaysold=60
sendemail=yes
function logit() {
if [ $logging = yes ]; then
echo $1
fi
}
function processold() {
if ( test -d /tmp/$maindir ); then
logit /tmp/$maindir Already exists. Deleting contents
rm /tmp/$maindir/*
else
mkdir /tmp/$maindir
fi
find . -type f -name 'msg*' -daystart -mtime +$daysold -exec mv {}
/tmp/$maindir \;
# Let's make one .txt file for an e-mail that we will send out.
# first a message explaining what's going on
cd /tmp/$maindir
echo -e Messages older then $daysold days in voicemail box $maindir folder
$subdir are being archived and removed from the phone system.\n\nThese have
been previously sent to your e-mail when they were received, and are
attached here for reference email.txt
echo -e \n\n email.txt
for txtname in msg*.txt; do
echo -n `echo $txtname|awk -F . '{print $1}'` email.txt
echo -n `grep -e callerid= -e origdate= -e duration= $txtname`
email.txt
echo seconds email.txt
done
echo -e \n\n email.txt
# now the fun stuff - uuencode all the .WAV files that were previously
e-mailed
for filename in msg*.WAV; do
uuencode $filename $filename email.txt
done
# who do we e-mail this to?
sendtoemail=`grep $maindir = /etc/asterisk/voicemail.conf| awk -F ,
'{print $3}'`
if [ $sendemail = yes ]; then
logit Sending e-mail to $sendtoemail
todaydate=`date +%Y%m%d`
/usr/bin/mail -s Archived Voicemail Messages $todaydate -a 'From: Asterisk
PBX [EMAIL PROTECTED]' $sendtoemail email.txt
fi
logit Tarballing the old messages.
tarfile=$maindir\_$todaydate\_$subdir.tar.gz
tar -czf $tarfile --no-recursion --remove-files ./msg*
chmod 600 $tarfile
mv ./$tarfile $archivefolder
rm /tmp/$maindir/*
rmdir /tmp/$maindir
}
#for maindir in `find $voicemailroot -type d -maxdepth 1|awk -F / '{print
$7}'`; do
# Currently only processing specific mailboxes:
for maindir in 1234 204 250 205 214 220; do
processme=$process
daysold=$defaultdaysold
case $maindir in
1234) processme=no;;
200) processme=no;;
250) let daysold=7;;
205) let daysold=14;;
204) let daysold=5;;
esac
if [ $processme = no ]; then
echo skipping $maindir
continue
fi
cd $voicemailroot/$maindir
for subdir in `find . -type d|awk -F / '{print $2}'`; do
# to only process specific directories:
#for subdir in INBOX; do
cd $voicemailroot/$maindir/$subdir
if ( test -e msg.txt ); then
messages=`ls *.WAV|grep -c msg`
oldmessages=`find . -type f -name '*.WAV' -daystart -mtime +$daysold|grep -c
msg`
if [ $oldmessages = 0 ]; then logit $maindir $subdir $oldmessages/$messages
No old messages to process;continue;fi
logit $maindir $subdir $oldmessages/$messages Processing old messages
processold
else
logit $maindir $subdir 0/0 No messages to process
fi
done
done
/home/asterisk/renumber.sh
--- renumber.sh ---
#!/bin/sh
# Asterisk doesn't handle out of order messages very well.
# Script written assuming worst case scenario that patches of messages
# Have been removed, instead of a solid block.
# Future improvement: Read in
Re: [asterisk-users] The S word: Asterisk security
Hope you get some beach time in today! Thanks for being a part of what I consider to be a gathering of friends with a common interest. I'm always happy to see you :) Best, Randy On Tue, Jul 1, 2008 at 5:56 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 7/1/08, randulo [EMAIL PROTECTED] wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy Randy, I'd love to participate as long as no one minds me calling in from the beach... :) I'm interested in developing my SIP DoS script (and any similar solutions). While I'm reluctant to claim that it or anything like it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDs required of Paris and Gottenburg Sweden
Kashif Naeem wrote: We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ? Please reply with rates. Do these cities have official, representative telephone numbers? What is the referent of the DIDs as opposed to merely DIDs? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
Randy,
Do you record these sessions?
Thanks,
Steve T
On Fri, Jul 4, 2008 at 2:31 AM, randulo [EMAIL PROTECTED] wrote:
Aside from the 5g phone that will come out as soon as you plunk down
$300 for the 3g ($800 if you calculate your 2 year contract
obligation), don't forget to join us today for The S Word: Security
Most of you on this list will be more qualified than I am to discuss
or even list the issues involved, but I would start with these:
* What are the principal risks?
DoS
Fraudulent usage of your minutes
Compromising your user accounts (example, getting all the emails, CID, etc)
Making your life miserable in various ways through resource abuse
* What's wrong with running as root?
* How to lock down your server
Denying access using standard *nix tools
Authentication
Checking against known attackers
Those are just a few ideas. Please join us for the call this Friday 4th of
July.
See http://VoipUsersConference.org
IRC.Freenode.net #voip-users-conference
PSTN;: Call (724) 444-7444 and enter 22622# 1#
Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; by default
your PIN is 1#
ts.x2z.eu resolves to the above IP
http://food4wine.ning.com has news, forums, blogs, etc
http://voipuserstv.com has videos of Asterisk Tag and other asterisk
and voip stuff
RSS http://feeds.feedburner.com/AstUser
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Fri, 4 Jul 2008 14:21:59 -0400, Steve Totaro wrote: Randy, Do you record these sessions? Thanks, Steve T All the VUC calls are recorded and available via http://voipusersconference.org/ning/. The post-call session is often not recorded, but the main body of the call is available for download. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
On Fri, Jul 4, 2008 at 6:41 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jul 04, 2008 at 04:56:41PM -0400, Steve Totaro wrote: On Fri, Jul 4, 2008 at 4:25 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jul 04, 2008 at 02:25:59PM -0400, Steve Totaro wrote: Call me Mr. Obvious, but why not use locks like Callweaver (the entire reason it was created)? It's not Mr. obvious. It's Captain Obvious. http://uncyclopedia.org/wiki/Captain_Obvious (this one has a mask) OK. And now we can resume the normal discussions/flames/whatever on the list. No flames, no mask, just discussion. I am what I am and that is all that I am, I always use my real name so your mask comment is out the window, no agenda or false pretenses. Really. I was just making a Captain-Obvious-like fact-correction comment. I have not intended to hint anything about you. I did omit a :-) Gotcha, I guess your lack of sense of humor, at l on the lists made me think you were putting me down. A little humor goes a log way. There has to be a reason why FreeSwitch bows away Asterisk as far as switching. Tell me how please? Could it have something to do with locking? Or missing functionality? It is immature and and is only at v1.0. Give it time, and it will blow Asterisk away, even more than it does now, plus there is no reason to use Again, please re-read S. Davies's reply there. If you want to rewrite Asterisk, have fun. But don't complaind about petty little 1.4-1.6 API canges :-) because the breakage here will be much more complete. As I said, I pay people to do the grunt work or even better, I wait for someone to GPL their mature software. BKW is doing work for me righ now.. FreeSwitch will be the clear winner, or at least the heart of large scale systems with a few Asterisk boxen here and there until it becomes more mature. Thanks, Steve T -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
On Friday 04 July 2008 19:59:55 Steve Totaro wrote: FreeSwitch will be the clear winner, or at least the heart of large scale systems with a few Asterisk boxen here and there until it becomes more mature. If you want to be a Freeswitch fanboy, that's fine, but please keep it off this list. This list is for usage questions of Asterisk, not for fanboyism of other software projects. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 AVM ISDN Fritzcards
On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED] wrote: Yes, with Suse 10.2/10.3 and chan_misdn. Just to follow up on this. SLES 10.2 SP2 worked bang on. The two cards are configured and working correctly and recognised by Asterisk. Question: I guess you were meaning openSUSE 10.2/10.3... will openSUSE 11 work here? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue welcome message
Hello! just fyi. If anyone wants to use the same function - here is the solution I found: Before calling the queue() do a queue_member_count and if 1 do not call the queue. Works fine with dynamic members. if you use static members - I think there is no solution ;-) best regards Martin - Original Message - From: Martin Schrott - thinking:systems To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, July 01, 2008 6:18 AM Subject: Re: [asterisk-users] queue welcome message Hello Tarek, thank you for your idea. But this only would work for the first caller - when the moh starts. all other callers go directly into moh on the position where the first caller is in moh. So this does not work. :-( Anyone an other idea? thank you Martin - Original Message - From: Tariq .. To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 30, 2008 5:13 PM Subject: Re: [asterisk-users] queue welcome message the only suggestion i would have for you is to use a SINGLE file for your MOH .. and you record the welcoming note in the begining of the file.. so whenever a caller comes in .. they will hear the MOH .. which has the welcoming note before the music starts... i know it's a stupid trick but it does the work for your needs... Salam Tarek Sawah -- From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 30 Jun 2008 09:19:27 +0200 Subject: [asterisk-users] queue welcome message Hi all! I would like to ask, how you realize the following in a call queue: When a caller gets into a queue how can I play a welcome Message to this caller first, before he starts hearing the music? We now use a playback before the caller gets into the queue. But when the queue is closed the caller heared already that somebody will pick up soon, but then gets into voicemail - because the queue has no members. So it would be great to let the queue welcome the caller at first and then start music. Is this possible? specifiing a queue-youarenext does only work when the caller is not the first in line. specifiing a periodic announce does play the message after the periodic-announce-frequency has been over. Is there also something else we can use? Thank you Martin Earn cashback on your purchases with Live Search - the search that pays you back! Learn More ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition
Hi On Fri, Jul 04, 2008 at 08:59:55PM -0400, Steve Totaro wrote: On Fri, Jul 4, 2008 at 6:41 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: (in an off-list message) On Fri, Jul 04, 2008 at 04:56:41PM -0400, Steve Totaro wrote: On Fri, Jul 4, 2008 at 4:25 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jul 04, 2008 at 02:25:59PM -0400, Steve Totaro wrote: Call me Mr. Obvious, but why not use locks like Callweaver (the entire reason it was created)? It's not Mr. obvious. It's Captain Obvious. http://uncyclopedia.org/wiki/Captain_Obvious Just to state the obvious: I don't normally use Uncyclopedia as a reliable reference. Anyway, have some http://uncyclopedia.org/wiki/Captain_Obvious_Cereal and have a good morning! [snip] As I said, I pay people to do the grunt work or even better, I wait for someone to GPL their mature software. BKW is doing work for me righ now.. It's not GPL. It's MPL. Somewhat comparable to LGPL. Though has a few nasties clauses. For instance, it can be subverted by patents. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
