from:"\-\- \[ UxBoD \] \-\-"

Hello all, 

I have recently upgraded to version 1.8.7.2 and have started to see the 
following errors in the logs: 

[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: SELECT failed 
[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: must be in 
SELECTED state 

They are not having a detrimental effect on the storing of VMs in IMAP just 
filling up the logs quickly :) What do they mean please ? 

-- 
Thanks, Phil 

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Re: [asterisk-users] VoiceMail and IMAP

1.8.7.0 ... am using Zimbra as the backend IMAP storage.
-- 
Thanks, Phil

- Original Message -
 On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
  Hello all,
 
  I have recently upgraded to version 1.8.7.2 and have started to see
  the following errors in the logs:
 
  From what version?
 
 
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org
 

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Re: [asterisk-users] VoiceMail and IMAP

Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder 
contains a space in the name and it errors; so that could be the cause of it 
all.  Is is valid to have a space in an IMAP folder name ?
-- 
Thanks, Phil

- Original Message -
 1.8.7.0 ... am using Zimbra as the backend IMAP storage.
 --
 Thanks, Phil
 
 - Original Message -
  On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
   Hello all,
  
   I have recently upgraded to version 1.8.7.2 and have started to
   see
   the following errors in the logs:
  
   From what version?
  
  
  --
  Paul Belanger
  Digium, Inc. | Software Developer
  twitter: pabelanger | IRC: pabelanger (Freenode)
  Check us out at: http://digium.com  http://asterisk.org
  
 
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Re: [asterisk-users] VoiceMail and IMAP

Okay, though removing the space and reloading the module still throws the same 
error messages.
-- 
Thanks, Phil

- Original Message -
 Generally speaking, no.  if you need the space, use quotes.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[
 UxBoD ]--
 Sent: Monday, December 12, 2011 11:12 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] VoiceMail and IMAP
 
 Hmmm, just tried leaving a voicemail on a new mailbox where the
 imapfolder
 contains a space in the name and it errors; so that could be the
 cause of it
 all.  Is is valid to have a space in an IMAP folder name ?
 --
 Thanks, Phil
 
 - Original Message -
  1.8.7.0 ... am using Zimbra as the backend IMAP storage.
  --
  Thanks, Phil
  
  - Original Message -
   On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
   
I have recently upgraded to version 1.8.7.2 and have started to
see the following errors in the logs:
   
From what version?
   
   
   --
   Paul Belanger
   Digium, Inc. | Software Developer
   twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at:
   http://digium.com  http://asterisk.org
   
  
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Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

2011-11-10 Thread --[ UxBoD ]--

Could we have more hours in the day to play with all the goodiness ? cannot 
keep up with everything at the moment :)
-- 
Thanks, Phil

- Original Message -
 I know what you mean - I'd rather have a working x-beta1 that a
 failing x.0
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
 Madsen
 Sent: Thursday, November 10, 2011 11:24 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available
 
 On 11-11-10 12:12 PM, Danny Nicholas wrote:
  Yeah!  My boss will be much happier having a system that doesn't
  have
  the -tail on it.
 
 I hear this kind of statement every once in a while, which makes
 absolutely
 no sense to me. If you're blindly running a version of any software
 in
 production (regardless as to it being tagged a -beta, -rc,
 -magic_candy,
 etc) without prior testing, then you're pretty much at the same risk
 regardless.
 
 I could take a random snapshot from a branch and name it something
 without a
 tailing hyphen+name, and it'd be pretty much the exact same thing
 without
 prior testing in your environment.
 
 Leif.
 
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[asterisk-users] [OT] Yealink T26/28/38 and Open-VPN

2011-08-24 Thread --[ UxBoD ]--

Hi, 

Sorry for an OT post but striking out a bit at the moment trying to get a 
response from Yealink RD. Has anybody successfully managed to get a Yealink 
phone to work across Open-VPN when using tlsauth ? We really do hope that it is 
possible due to the benefits tlsauth offers against DoS. 
-- 
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Re: [asterisk-users] Trouble with *8 Pickup

2011-08-11 Thread --[ UxBoD ]--

Ah, now this is interesting as one of our clients had the same problem the 
other day; in our case when they performed the *8 they got an extension 
unavailable from a completely different dialplan! This was on Asterisk 1.6 
though with Snom phones.
-- 
Thanks, Phil

- Original Message -
 We have a client that has sporadic problems with the *8 pickup
 facility.
 The server they are using is 1.8.5 and they are using Snom phones.
 
 Every now and then when they try to do a pickup from another phone
 they
 get a forbidden message on the phone and I can see the following in
 the
 logs.
 
 [Aug  8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
 [Aug  8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
 [Aug  8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup
 of SIP/-0404 failed.
 [Aug  8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel
 SIP/-0404MASQ, strange things may happen.
 
 Does anyone know what this warning means?
 
 Thanks
 
 Ish
 
 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd
 
 Office:   0161 660 3062
 
 
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Re: [asterisk-users] CEL and MySQL

2011-08-08 Thread --[ UxBoD ]--

cel_odbc.conf and then use adapative odbc I think.
-- 
Thanks, Phil

- Original Message -
 Is anyone using CEL with a MySQL backed at all?
 
 I've found a table schema but I'm guessing I need some sort of
 cel_mysql.conf and don't even have a sample for that.
 
 Can anyone give me any pointers as to what files I need to change to
 get
 this logging to my MySQL table?
 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd
 
 Office:   0161 660 3062
 
 
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[asterisk-users] X86_64 Compilation Issue

2011-07-29 Thread --[ UxBoD ]--

Hi, 

compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am 
seeing the following when running the make: 

/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam 
/usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl 
/usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl 
/usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for 
-lcrypto 
/usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for 
-lcrypto 

How can I get Asterisk to pick up the 64 bit version of the libraries instead 
of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ? 
-- 
Thanks, Phil 

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Re: [asterisk-users] X86_64 Compilation Issue

2011-07-29 Thread --[ UxBoD ]--

Thank you Dave.
-- 
Thanks, Phil

- Original Message -
 On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
  Hi,
 
  compiling up a new installation of Asterisk 1.8.5 on CentOS 6
  X86_64 and
  am seeing the following when running the make:
 
  /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when
  searching for
  -lpam
  /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when
  searching for
  -lssl
  /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching
  for
  -lssl
  /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when
  searching
  for -lcrypto
  /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when
  searching
  for -lcrypto
 
  How can I get Asterisk to pick up the 64 bit version of the
  libraries
  instead of the 32 bit ones ? Is it just a case of updating
  LD_LIBRARY_PATH ?
  --
  Thanks, Phil
 
 
 Did you run configure with --libdir=/usr/lib64 ?
 
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Re: [asterisk-users] Securing Asterisk

2011-07-27 Thread --[ UxBoD ]--

Simple answer to all this is to install http://lync.microsoft.com/ ... good 
luck ;)
-- 
Thanks, Phil

- Original Message -
 Kevin P. Fleming wrote:
  
  'alwaysauthreject' in not imcompliant with any RFCs; the RFCs
  define
  response codes that *can* be used to indicate (for example) that
  the
  Request URI does not represent a target known to the receiver (404
  Not
  Found), but does not mandate that the server respond with that code
  in
  that situation.
 
 
 Kevin,
 
 Thanks for the correction and I apologize if I'm propagating a
 misconception.  Am I misunderstanding this Asterisk Security
 Advisory?
 
 http://lists.digium.com/pipermail/asterisk-announce/2009-April/000177.html
 
In 2006, the Asterisk maintainers made it more difficult
to scan for valid SIP usernames by implementing an
option called alwaysauthreject...
 
...What we have done is to carefully emulate exactly the
same responses throughout possible dialogs, which should
prevent attackers from gleaning this information. All
invalid users, if this option is turned on, will receive
the same response throughout the dialog, as if a
username was valid, but the password was incorrect.
 
It is important to note several things. First, this
vulnerability is derived directly from the SIP
specification, and it is a technical violation of RFC
3261 (and subsequent RFCs, as of this date), for us to
return these responses...
 
 I am asking out of genuine curiosity, because I trust your assessment
 more than my interpretation of the advisory.
 
 Thank you,
 
 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer
 
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Re: [asterisk-users] Securing Asterisk

2011-07-26 Thread --[ UxBoD ]--

That is pretty interesting. I am writing a similar tool but using OSSEC to 
identify the attacks and then share the data between nodes using Memcached and 
AnyEvent. Both Asterisk and Apache, or any other server that can run OSSEC, 
will be able to feed into the shared ban database.
-- 
Thanks, Phil

- Original Message -
 Why not firewall hack attempts after 3 tries?  When we started doing
 that the quantity of hacking attempts dropped right off.  We also
 setup
 our own fail2ban sharing server so that we could share the bans
 across
 multiple servers.  Have a look at
 http://www.f2bshare.org/index.php?title=Main_Page if you want to do
 something similar.  Why try to make Asterisk into something it's not
 intended to be?  Just use your firewall for what it's good at.
 
 --
 Darren Wiebe
 
 
 On 7/23/11 11:38 AM, CDR wrote:
  I beg to differ. Digium is hiding from the real world and somebody
  is
  going take the software and run with it. My customers lost in
  excess
  of $50.000 and cut my pay in half, because of hackers. The hackers
  figured out how to scan every asterisk for weak passwords or open
  ports, and bang them real good. We need two things: a) disable in
  sip.conf the reply for INVITES that have wrong user information,
  and
  also, b) disable any response to any REGISTER packet altogether.
  Can
  somebody please write  patch? Or should we go broke trying to stop
  the
  flood of criminals coming from abroad?
  Federico
 
  On Sat, Jul 23, 2011 at 1:00 PM,
  asterisk-users-requ...@lists.digium.com  wrote:
  Send asterisk-users mailing list submissions to
  asterisk-users@lists.digium.com
 
  To subscribe or unsubscribe via the World Wide Web, visit
  http://lists.digium.com/mailman/listinfo/asterisk-users
  or, via email, send a message with subject or body 'help' to
  asterisk-users-requ...@lists.digium.com
 
  You can reach the person managing the list at
  asterisk-users-ow...@lists.digium.com
 
  When replying, please edit your Subject line so it is more
  specific
  than Re: Contents of asterisk-users digest...
 
 
  Today's Topics:
 
 1. Re: use dahdi for local terminal modem access? (Lyle Giese)
 2. dialplan pattern help (Armand Fumal)
 3. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603
Declined (Patrick Lists)
 4. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603
Declined (Paul Belanger)
 
 
  --
 
  Message: 1
  Date: Sat, 23 Jul 2011 09:29:26 -0500
  From: Lyle Giesel...@lcrcomputer.net
  Subject: Re: [asterisk-users] use dahdi for local terminal modem
  access?
  To: asterisk-users@lists.digium.com
  Message-ID:4e2adac6.4010...@lcrcomputer.net
  Content-Type: text/plain; charset=ISO-8859-1; format=flowed
 
 
  On 07/22/11 22:47, William Stillwell wrote:
  Um, no VOIP involved here.
  Wrong.  What do you think Asterisk is?  Chopped meat?  It's a VoIP
  switch.  All traffic inside Asterisk is VoIP.
 
  I have an asterisk server with 2 23B+D PRI's
 
  I want to telnet/ssh into the asterisk server, and make an
  outbound call
  serial based modem/terminal connection (Like the 80/90's BBS
  Days).
 
  No TCP/IP or PPP or crazyness
 
  (ie, dialing into a Modem set to AA hooked to a Cisco Console
  Port)
 
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Lyle Giese
  Sent: Friday, July 22, 2011 8:07 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] use dahdi for local terminal modem
  access?
 
  On 07/22/11 18:13, William Stillwell wrote:
  I have some terminals that have phone lines.
 
  One of my tech had an idea of using IAXmodem or something
  similar to
  use
  existing PRI/DAHDI Trucks for dial out via the asterisk/Linux
  console.
  Anybody ever heard of doing this?
 
  I would think maybe would use iaxmodem maybe and a shell
  terminal
  app?
  (basically I'm dialing into a remote access device that uses a
  pots
  like
  for remote administration, and don't want to string a channel
  bank
  off
  my asterisk box, and a hook to a modem)
 
 
 
  --
  Depends on your expectation.  Because of compression in the
  codecs, it
  will be hard to get fast dialup.  If you mean ssh or telnet, it
  might
  work.  If you mean vnc or RDP over this, you may not get enough
  usable
  bandwidth to do that.
 
  Given this, I have in an emergency dialed into a RAS server via
  a VoIP
  line. My laptop connected at 14,400bps.  All I needed to do was
  telnet
  into an APC masterswitch to toggle power on one outlet.  It
  worked.
 
  I was surprised at getting a 14,400bps connect.  I was not
  expecting
  that high and really did not need that high.  300 baud probably
  would
  have been fast enough to telnet into an APC masterswitch.
 
  Lyle Giese
  LCR Computer Services, Inc.
 
  --

Re: [asterisk-users] Functions not autoloading

2011-07-26 Thread --[ UxBoD ]--

Have filed https://issues.asterisk.org/jira/browse/ASTERISK-18167 as its always 
repeatable.
-- 
Thanks, Phil

- Original Message -
 Is anybody else seeing this at all ?
 --
 Thanks, Phil
 
 - Original Message -
  Just received a call and on checking messages I now see:
  
  ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered
  
  Grrr, looks like time to go back to 1.8.3 as all the apps and
  functions exist in /usr/lib/asterisk/modules.
  
  How could I help to debug this please ?
  --
  Thanks, Phil
  
  - Original Message -
   On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
Since upgrading to 1.8.5.0 I have had to add into modules.conf:
   
load =  func_callerid.so
load =  func_cdr.so
   
otherwise they do not get loaded even though I have set
autoload=yes.
   
Is this something you would expect as it is different behavior
to
1.8.3.0 and I do not see any issues in
/var/log/asterisk/messages
?
   
   No, this is not expected behavior.
   
 
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Re: [asterisk-users] file2ban

2011-07-26 Thread --[ UxBoD ]--

If you are using OSSEC here are some rules:

rule id=1 level=5
  decoded_aslocal-asterisk-denied/decoded_as
  descriptionAsterisk Potentially Under Attack/description
/rule

rule id=10001 level=8 frequency=5 timeframe=10
  if_matched_sid1/if_matched_sid
  same_source_ip /
  descriptionAsterisk Under Brute Force Attack/description
/rule

and for the local_decoder:

decoder name=local-asterisk-denied
  prematchNOTICE[\d+] \S+: Registration from /prematch
  regex offset=after_prematch^\S+ failed for '(\d+.\d+.\d+.\d+)'/regex
  ordersrcip/order
/decoder

OSSEC can then use Active Response to block the IP using IPtables.
-- 
Thanks, Phil

- Original Message -
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
  Sent: Tuesday, July 26, 2011 3:22 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] file2ban
  
  I want to add an entry to a database every time a brute force
  registration
  attempt is done.
  from this database we are updating cisco routers with our ban list
  so our
  entire network is protected.
  The database side of things is working and has been for some time.
  I really
  would like to add the file2ban side of it to protect our asterisk
  system
  better.
 
 Look at the /etc/fail2ban/action.d/   Actions in the default config
 runs an iptables command to insert the ban into IPTables, but you
 can have it run most any command.
 
 
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Re: [asterisk-users] Functions not autoloading

2011-07-22 Thread --[ UxBoD ]--

Is anybody else seeing this at all ?
-- 
Thanks, Phil

- Original Message -
 Just received a call and on checking messages I now see:
 
 ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered
 
 Grrr, looks like time to go back to 1.8.3 as all the apps and
 functions exist in /usr/lib/asterisk/modules.
 
 How could I help to debug this please ?
 --
 Thanks, Phil
 
 - Original Message -
  On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
   Since upgrading to 1.8.5.0 I have had to add into modules.conf:
  
   load =  func_callerid.so
   load =  func_cdr.so
  
   otherwise they do not get loaded even though I have set
   autoload=yes.
  
   Is this something you would expect as it is different behavior to
   1.8.3.0 and I do not see any issues in /var/log/asterisk/messages
   ?
  
  No, this is not expected behavior.
  

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[asterisk-users] Functions not autoloading

2011-07-21 Thread --[ UxBoD ]--

Since upgrading to 1.8.5.0 I have had to add into modules.conf:

load = func_callerid.so
load = func_cdr.so

otherwise they do not get loaded even though I have set autoload=yes.

Is this something you would expect as it is different behavior to 1.8.3.0 and I 
do not see any issues in /var/log/asterisk/messages ?
-- 
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Re: [asterisk-users] Functions not autoloading

2011-07-21 Thread --[ UxBoD ]--

Just received a call and on checking messages I now see:

ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered

Grrr, looks like time to go back to 1.8.3 as all the apps and functions exist 
in /usr/lib/asterisk/modules.

How could I help to debug this please ?
-- 
Thanks, Phil

- Original Message -
 On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
  Since upgrading to 1.8.5.0 I have had to add into modules.conf:
 
  load =  func_callerid.so
  load =  func_cdr.so
 
  otherwise they do not get loaded even though I have set
  autoload=yes.
 
  Is this something you would expect as it is different behavior to
  1.8.3.0 and I do not see any issues in /var/log/asterisk/messages
  ?
 
 No, this is not expected behavior.
 
 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
 kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org
 
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[asterisk-users] Converged phones; akin to SwitchVox ?

2011-07-19 Thread --[ UxBoD ]--

Been looking at SwitchVox and how it handles mobility using virtual extensions. 
Does somebody have any examples on how this can be achieved with Asterisk ? I 
have Bria on my Android and it would be nice if I could get my office phone 
and/or cell to ring. 
-- 
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Re: [asterisk-users] Macro issue under 1.8.5

- Original Message -
 On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
 --[ UxBoD ]-- ux...@splatnix.net wrote:
 
  - Original Message -
   On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing
as
app_macro has been installed okay?
   
Macro was depreciated in 1.6 and most likely removed in 1.8.5
   
   Removed, no.  However in future version of Asterisk it will not
   be
   enabled in menuselect by default.
   
   @OP: *CLI module load app_macro.so
   
  
  Same problem even after performing the above load. module does
  exist:
 
 Watch the console carefully for errors when you run that command.
  They
 should tell you exactly what's wrong.
 
 Also, it may help to inspect the differences in apps/app_macro.c
 between
 1.8.3 and 1.8.5.
 

Well it seems like its getting worse!

[Jul 18 11:36:00] WARNING[28936]: pbx.c:4071 pbx_extension_helper: No 
application 'Playback' for extension (home, 400, 1)

Looking in pbx.c it would appear it cannot find the application in some sort of 
cache:

if (!e-cached_app)
  e-cached_app = pbx_findapp(e-app);
  app = e-cached_app;
  ast_unlock_contexts();
  if (!app) {
 ast_log(LOG_WARNING, No application '%s' for extension (%s, %s, %d)\n, 
e-app, context, exten, priority);
 return -1;
  }

Any thoughts ?
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Re: [asterisk-users] Macro issue under 1.8.5

- Original Message -
 - Original Message -
  On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
  --[ UxBoD ]-- ux...@splatnix.net wrote:
  
   - Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
 --[ UxBoD ]-- wrote:
 I back leveled to 1.8.3 and that works fine. What am I
 missing
 as
 app_macro has been installed okay?

 Macro was depreciated in 1.6 and most likely removed in 1.8.5

Removed, no.  However in future version of Asterisk it will not
be
enabled in menuselect by default.

@OP: *CLI module load app_macro.so

   
   Same problem even after performing the above load. module does
   exist:
  
  Watch the console carefully for errors when you run that command.
   They
  should tell you exactly what's wrong.
  
  Also, it may help to inspect the differences in apps/app_macro.c
  between
  1.8.3 and 1.8.5.
  
 
 Well it seems like its getting worse!
 
 [Jul 18 11:36:00] WARNING[28936]: pbx.c:4071 pbx_extension_helper: No
 application 'Playback' for extension (home, 400, 1)
 
 Looking in pbx.c it would appear it cannot find the application in
 some sort of cache:
 
 if (!e-cached_app)
   e-cached_app = pbx_findapp(e-app);
   app = e-cached_app;
   ast_unlock_contexts();
   if (!app) {
  ast_log(LOG_WARNING, No application '%s' for extension (%s, %s,
  %d)\n, e-app, context, exten, priority);
  return -1;
   }
 
 Any thoughts ?

Okay, I cleared out /usr/lib/asterisk/modules plus my build directory and 
started with a fresh extract of asterisk tar file. This time all seems a lot 
better apart from:

[Jul 18 12:25:38] ERROR[14082] pbx.c: Function CALLERID not registered

for which I need to add into modules.conf:

load = func_callerid.so

Why is this need now as it was not necessary in 1.8.3 ?
-- 
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[asterisk-users] chan_gtalk load error

Hi, 

When starting Asterisk (1.8.5.0) I see in messages:

[Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 
'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: 
ast_aji_get_client
[Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be 
loaded.

Yet I do have iksemel installed:

ls -l /usr/local/lib/libik*
-rw-r--r-- 1 root root 281994 Jul 18 16:14 /usr/local/lib/libiksemel.a
-rwxr-xr-x 1 root root822 Jul 18 16:14 /usr/local/lib/libiksemel.la
lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so - 
libiksemel.so.3.1.1
lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so.3 - 
libiksemel.so.3.1.1
-rwxr-xr-x 1 root root 165132 Jul 18 16:14 /usr/local/lib/libiksemel.so.3.1.1

and checking whether they have been linked okay:

ldd chan_gtalk.so 
linux-vdso.so.1 =  (0x7fff01523000)
libiksemel.so.3 = /usr/local/lib/libiksemel.so.3 (0x2b6fbeb09000)
libssl.so.6 = /lib64/libssl.so.6 (0x2b6fbed15000)
libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b6fbef62000)
libpthread.so.0 = /lib64/libpthread.so.0 (0x2b6fbf2b3000)
libc.so.6 = /lib64/libc.so.6 (0x2b6fbf4ce000)
libgnutls.so.13 = /usr/lib64/libgnutls.so.13 (0x2b6fbf827000)
libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 
(0x2b6fbfaab000)
libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000)
libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b6fbff6f000)
libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b6fc0171000)
libdl.so.2 = /lib64/libdl.so.2 (0x2b6fc0396000)
libz.so.1 = /usr/lib64/libz.so.1 (0x2b6fc059b000)
/lib64/ld-linux-x86-64.so.2 (0x003ac420)
libgcrypt.so.11 = /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000)
libgpg-error.so.0 = /usr/lib64/libgpg-error.so.0 (0x2b6fc0a21000)
libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 
(0x2b6fc0c25000)
libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b6fc0e2d000)
libresolv.so.2 = /lib64/libresolv.so.2 (0x2b6fc102f000)
libselinux.so.1 = /lib64/libselinux.so.1 (0x2b6fc1245000)
libsepol.so.1 = /lib64/libsepol.so.1 (0x2b6fc145d000)

Any thoughts on why this is happening as I could not find many references to it 
when searching ?
-- 
Thanks, Phil

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Re: [asterisk-users] chan_gtalk load error

- Original Message -
 - Original Message -
  From: --[ UxBoD ]-- ux...@splatnix.net
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Monday, July 18, 2011 11:42:25 AM
  Subject: [asterisk-users] chan_gtalk load error
  Hi,
  
  When starting Asterisk (1.8.5.0) I see in messages:
  
  [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module
  'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined
  symbol: ast_aji_get_client
  [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so'
  could not be loaded.
  
  Yet I do have iksemel installed:
  
  ls -l /usr/local/lib/libik*
  -rw-r--r-- 1 root root 281994 Jul 18 16:14
  /usr/local/lib/libiksemel.a
  -rwxr-xr-x 1 root root 822 Jul 18 16:14
  /usr/local/lib/libiksemel.la
  lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so
  -
  libiksemel.so.3.1.1
  lrwxrwxrwx 1 root root 19 Jul 18 16:14
  /usr/local/lib/libiksemel.so.3
  - libiksemel.so.3.1.1
  -rwxr-xr-x 1 root root 165132 Jul 18 16:14
  /usr/local/lib/libiksemel.so.3.1.1
  
  and checking whether they have been linked okay:
  
  ldd chan_gtalk.so
  linux-vdso.so.1 = (0x7fff01523000)
  libiksemel.so.3 = /usr/local/lib/libiksemel.so.3
  (0x2b6fbeb09000)
  libssl.so.6 = /lib64/libssl.so.6 (0x2b6fbed15000)
  libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b6fbef62000)
  libpthread.so.0 = /lib64/libpthread.so.0 (0x2b6fbf2b3000)
  libc.so.6 = /lib64/libc.so.6 (0x2b6fbf4ce000)
  libgnutls.so.13 = /usr/lib64/libgnutls.so.13 (0x2b6fbf827000)
  libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2
  (0x2b6fbfaab000)
  libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000)
  libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b6fbff6f000)
  libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3
  (0x2b6fc0171000)
  libdl.so.2 = /lib64/libdl.so.2 (0x2b6fc0396000)
  libz.so.1 = /usr/lib64/libz.so.1 (0x2b6fc059b000)
  /lib64/ld-linux-x86-64.so.2 (0x003ac420)
  libgcrypt.so.11 = /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000)
  libgpg-error.so.0 = /usr/lib64/libgpg-error.so.0
  (0x2b6fc0a21000)
  libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0
  (0x2b6fc0c25000)
  libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b6fc0e2d000)
  libresolv.so.2 = /lib64/libresolv.so.2 (0x2b6fc102f000)
  libselinux.so.1 = /lib64/libselinux.so.1 (0x2b6fc1245000)
  libsepol.so.1 = /lib64/libsepol.so.1 (0x2b6fc145d000)
  
  Any thoughts on why this is happening as I could not find many
  references to it when searching ?
  --
  Thanks, Phil
  
 
 Do you have res_jabber installed?
 

That would help :) Thanks David.
-- 
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Re: [asterisk-users] Macro issue under 1.8.5

2011-07-16 Thread --[ UxBoD ]--

- Original Message -
 On 11-07-15 02:18 PM, Doug Lytle wrote:
  --[ UxBoD ]-- wrote:
  I back leveled to 1.8.3 and that works fine. What am I missing as
  app_macro has been installed okay?
 
  Macro was depreciated in 1.6 and most likely removed in 1.8.5
 
 Removed, no.  However in future version of Asterisk it will not be
 enabled in menuselect by default.
 
 @OP: *CLI module load app_macro.so
 

Same problem even after performing the above load. module does exist:

[root@voip asterisk]# ls -l /usr/lib/asterisk/modules/app_macro.so 
-rwxr-xr-x 1 root root 218156 Jul 16 10:56 
/usr/lib/asterisk/modules/app_macro.so

If I back level to 1.8.3 again everything starts working fine.
-- 
Thanks, Phil

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[asterisk-users] Macro issue under 1.8.5

2011-07-15 Thread --[ UxBoD ]--

Have just tried to test an upgrade to 1.8.5 and when making an outbound call I 
get:

[Jul 15 18:48:52] WARNING[21038]: pbx.c:4071 pbx_extension_helper: No 
application 'Macro' for extension (context, XX, 1)

I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has 
been installed okay?
-- 
Thanks, Phil

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[asterisk-users] Next Asterisk 1.8 Release

2011-06-17 Thread --[ UxBoD ]--

Hi, When is the next release planned for as very keen to get it into Production 
but require the call pickup fix. 
-- 
Thanks, Phil 

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[asterisk-users] Unusual message

2011-05-13 Thread --[ UxBoD ]--

Hi, 

Needed to test follow-me this evening on Asterisk 1.6.2.17 and received the 
following message: 

== Spawn extension (international-US, 0114407590XX, 5) exited non-zero on 
'Local /0114407590XX@aXX-a62a;2' 
-- no live channels left. exiting. 

I have not seen that before. What does it mean ? -- 
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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread --[ UxBoD ]--

- Original Message -
 On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
  Are you not seeing issues with *8 call pick up then ?
 
 Nope, I double checked it after seeing someone saying they had issues
 with it and it is fine on the installation I have.
 

Which release are you running as this is still open 
https://issues.asterisk.org/view.php?id=18654
-- 
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Re: [asterisk-users] Configuring Voicemail in Asterisk 1.8

2011-05-07 Thread --[ UxBoD ]--

- Original Message -
 Dear;
 
 Where I can find a new documentation for Asterisk 1.8?
 
 Where is the wrong in that line? I see it is as 1.8 version !
 
 500 = 1234,Operator,opera...@gama.com
 
 Regards
 Bilal
 
 
 ---
 
  You are using an old format for specifying the
  mailbox.  See core show application voicemail for the
  correct usage.   Also read ALL the
  UPGRADE*.txt files.
  
   -Original Message-
   From: asterisk-users-boun...@lists.digium.com
   [mailto:asterisk-users-boun...@lists.digium.com]
  On Behalf Of
   bilal ghayyad
   Sent: Friday, May 06, 2011 12:49 PM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] Configuring Voicemail in
  Asterisk 1.8
  
   Hi All;
  
  
   Already in the voicemail.conf file, I added the
  extension 500
   and kindly find below my voicemail configuration:
  
   [Internal]
  
   0 = 1234,Gama Operator,opera...@gama.com
   500 = 1234,Operator,opera...@gama.com
   501 = 1234,Employer Name,employer_em...@gama.com
   502 = 1234,Employer Name,employer_em...@gama.com
  
  
   Asterisk version is 1.8 and currently I am getting
  this
   warning message:
  
   [May  7 19:32:46] WARNING[4328]:
  app_voicemail.c:5535
   leave_voicemail: No entry
                 
                 
            in voicemail config
   file for 'u500'
  
   So what I might be missing?
  
   Regards
   Bilal
 
 

No error message in /var/log/messages/asterisk ? By the looks of the previous 
message, file for 'u500', you have a typo somewhere in your dialplan. Check all 
the VoiceMail() directives.
-- 
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[asterisk-users] [SOT] Virtualising Asterisk

2011-05-07 Thread --[ UxBoD ]--

I know a lot has changed over the past couple of years, and even monthly, and 
that Asterisk running within a virtualised environment is very happy indeed. If 
one would only be using SIP/IAX would Xen/KVM be the best solution ? / or 
perhaps VServer/LXC maybe advantageous due to binary hashing. Your thoughts 
would be very welcome. 
-- 
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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-06 Thread --[ UxBoD ]--

- Original Message -
 On Thu, 2011-05-05 at 14:13 +, satish patel wrote:
  Hi All,
  
  Just wondering is it safe to use asterisk 1.8 latest branch on
  production ?
  
  http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision
  317100
  
  -S
 We've been running 1.8.3.2 with the patch to fix the local channel
 issue
 (https://issues.asterisk.org/view.php?id=18818) For about a month in
 our
 test environment and it's been pretty stable. I would strongly advise
 that you run and version you wish to migrate to in a test environment
 for a good while as there are differences between 1.4 and 1.8 that
 are
 quite subtle and hard to pick up on (e.g. how to set outbound CLI
 correctly in CDR).
 
 Most of the big issues we found were due to our use of RealTime
 architecture. I get the impression that RealTime is not that widely
 used
 and therefore not that widely tested.
 
 To any development people out there, one we get these 1.8 servers
 into
 production I may well offer my services for testing with an emphasis
 on
 RealTime...

Are you not seeing issues with *8 call pick up then ?
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Re: [asterisk-users] [OT] Yealink Phones

2011-04-13 Thread --[ UxBoD ]--

- Original Message -
 
 I've just started deploying these (well the T28P model) after years
 of
 Snom issues and they look pretty good (although the documentation is
 execrable; if you thought the Snom stuff was obtuse Yealink have got
 them knocked into a cocked hat!).
 
 Anyway, for provisioning I use HTTP with a DHCP entry like:-
 
   #
 #   Yealink Phones
 #
 group {
 #
 # The phone should pickup the
 # model config file (y0.cfg for the
 # T28P) first and then the MAC.cfg file
 #
   # Yes tftp-server-name to set the DHCP option 
 but
   # the http:// tells the phone to get it's files 
 via
   # http.
   option tftp-server-name 
 http://192.168.1.13/yealink;;
 #
 host yealinkT28P {
 hardware ethernet 00:15:65:1b:d9:12;
 fixed-address 192.168.1.33;
 option host-name yealinkT28P;
 }
 }
 
 As the comments say, the phone's first pick up the model dependant
 config file (y0.cfg for the T28P model) and then the
 MAC.cfg file.
 
 This is nice as you have one model.cfg file for the site-wide config
 and
 then fine tune specific phones (setup different BLF keys and,
 obviously,
 SIP logins for each device) in the MAC.cfg files.
 
 In the y0.cfg file I have:-
 
   #
   #   Auto Provision
   [ autoprovision ]
   path = /config/Setting/autop.cfg
   server_address = http://192.168.1.13/yealink
   [ autop_mode ]
   path = /config/Setting/autop.cfg
   # Mode 7 = at Power On and Weekly
   mode = 7
   #   Sunday between 0100 and 0500
   schedule_dayofweek=0
   schedule_time = 01:00
   schedule_time_end = 05:00
   #
 
 
 Re non-web based access.  Obviously the config files are on your
 DHCP/Apache/Asterisk server so you can edit them however you like.
 
 You can also enable telnet access to the phones with a 'hidden'
 config option of:-
 
   #
   [ telnet ]
   path=/config/Network/Network.cfg
   telnet_enable=1
   #
 
 but the login/password are the admin defaults so a bit of a security
 hole there.  Not really found much useful telnetting into the phone
 but
 I've not played around with it much.
 
 One other useful tip:  If you play around in the web interface, set
 the
 phone up and then export the config, you end up with a config.bin
 file
 which is just tar of the config files.  A quick diff and you can
 easily
 find out what you need to tweak in your Autoprovision config files.
 
 Hope that helps.
 
 PS - anyone else with useful Yealink tips?
 

We are looking to switch to Yealink from SNOM and that last tip for saving the 
configuration is one I have recently asked them about. All sounds very 
promising and we hope to get some eval units soon :)
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Re: [asterisk-users] Asterisk 1.8.3

2011-04-07 Thread --[ UxBoD ]--

- Original Message -
 On 11-04-07 08:20 AM, Satish Patel wrote:
  Is it ture 1.8.3 is unstable? We are planning to put this in
  production.
  Please suggest me what should I do?
 
 This is a loaded question, since it really depends on what you plan
 to
 do.  What does your migration plan look like?  What sort of testing
 have
 you done with Asterisk?  Blindly moving into production with
 _anything_
 is a recipe for trouble.
 

And don't forget that call pickup crashes Asterisk from what would appear 
release 1.8.1 upwards! We have had to back level to that latest 1.6 branch.

https://issues.asterisk.org/view.php?id=18654
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Re: [asterisk-users] problem with crashing Asterisk 1.8

2011-03-10 Thread --[ UxBoD ]--

- Original Message -

 My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2.
 It just keeps restarting.
 Any pointers on log files to watch? I tried to debug it but i
 couldn't find a good reason for the crashes.
 Maby the box is just overloaded or something like that but there
 should be a log file telling me that, right?

 Thanks,
 Peter
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Which version of 1.8 are you using ? If you are using call pickup that can 
generate a segfault and crash Asterisk in version 1.8.3. Am hoping 1.8.4 will 
be out soon. 
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[asterisk-users] Multiple SIP endpoint registrations

2011-03-09 Thread --[ UxBoD ]--

Hi, 


With Asterisk 1.8 is it now possible to register the same SIP account at 
multiple endpoints and for both to ring when the associated extension is dialed 
? 
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Re: [asterisk-users] doorphone?

2011-03-09 Thread --[ UxBoD ]--

- Original Message -
 Hi,
 
 could anybody suggest a usable doorphone and magnetic door opener
 hardphone system for me, please? Of course should be connectable to
 asterisk. I am in the EU, should be available here.
 
 thank you,
 Csaba
 
 --

http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html

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[asterisk-users] CEL and PGSQL

2011-02-28 Thread --[ UxBoD ]--

Hi, 


Would someone know where I can download the CEL schema for (create commands) 
for PostgreSQL please ? 
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Re: [asterisk-users] CEL and PGSQL

2011-02-28 Thread --[ UxBoD ]--

- Original Message - 

 Pretty sure I saw those on wiki.asterisk.org .

 Thanks,
 --Warren Selby, dCAP

 On Feb 28, 2011, at 10:39 AM, --[ UxBoD ]--  ux...@splatnix.net 
 wrote:

  Hi,
 

  Would someone know where I can download the CEL schema for (create
  commands) for PostgreSQL please ?
 

Doh! Thank you Warren, appreciated.

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[asterisk-users] SRTP Error Message

2011-02-26 Thread --[ UxBoD ]--

Apologies in advance if this has come up a thousand times before but is there 
any way to stop this error in 1.8 ? 


[ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup 
SRTP session. 

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[asterisk-users] [OT] Yealink IP Phones

2011-02-25 Thread --[ UxBoD ]--

Hello all, 

After numerous issues with Snom phones (360/370/870) potentially looking to 
migrate too Yealink as their product range looks very promising indeed. 

Are any of you using them with Asterisk ? How do they perform ? Do you use mass 
deployment at all ? 

Would be very interested to hear from you. 
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Re: [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number

2011-02-15 Thread --[ UxBoD ]--

- Original Message -

 I found some great pieces of script on the internet that I've
 combined to allow Asterisk to send voicemails as an MP3 file, and
 encode the sender name and number as well as message number as tags
 into the MP3 file. I even include a cover art image which has our
 company logo and PBX symbol in it.

 Mobile phone users love it, and Android phones can now play the
 attachments (without having to move to the larger WAV format).

 If anyone wants to try it out let me know!

 Michelle
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[asterisk-users] app_calendar and SSL

2011-01-17 Thread --[ UxBoD ]--

Hi,

Over the weekend tried to setup a test using the new app_calendar code but 
receiving the following error:

[Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar: 
Unable to retrieve iCalendar 'testcal' from 
'https://office.test.net/home/teamsh...@test.net/Calendar/': Server certificate 
verification failed: issuer is not trusted

The target server is using a self signed cert so where would one store the PEM 
on the Asterisk server for the calendar app to find it ?
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Re: [asterisk-users] app_calendar and SSL

2011-01-17 Thread --[ UxBoD ]--

- Original Message -
 Try to disable certificate verification on the app. I had never tried
 it personally but check for that option.
 
 Sent from my iPhone
 
 On Jan 17, 2011, at 5:51 PM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
 
  Hi,
 
  Over the weekend tried to setup a test using the new app_calendar
  code but receiving the following error:
 
  [Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146
  fetch_icalendar: Unable to retrieve iCalendar 'testcal' from
  'https://office.test.net/home/teamsh...@test.net/Calendar/': Server
  certificate verification failed: issuer is not trusted
 
  The target server is using a self signed cert so where would one
  store the PEM on the Asterisk server for the calendar app to find it
  ?

I could not see anything in the code so have filed 
https://issues.asterisk.org/view.php?id=18630 as it should be possible using 
ne_ssl_trust_cert to specify a not default CA file to trust.
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Re: [asterisk-users] Voicemail Forwarding

2011-01-04 Thread --[ UxBoD ]--

- Original Message -
 --[ UxBoD ]-- wrote:
  - Original Message -
 
 
  Yes exactly that indeed. Though Asterisk appears to ignore which
  context the user is in and selects default instead. Beginning to
  think that it is a bug.
 
 
 I got it figured out.
 
 In your voicemail.conf, search for the option
 
 searchcontexts=yes
 
 And enable it.
 
 Doug
 

Sorry for the late reply! While that does allow it to work it is not 
appropriate in a multi-tenant environment where the same extension could exist 
in different contexts. Will file a bug for this and the configuration we are 
using looks correct.
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Re: [asterisk-users] Voicemail Forwarding

2010-12-18 Thread --[ UxBoD ]--

- Original Message -
 Is that user trying to forward to xxx in the same context?
 
 On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
  Experiencing a problem when users attempt to forward a voicemail
  from within VoiceMailMain(Option 8) I see the console message:
 
  Couldn't not find mailbox XXX in context default
 
  As why are running in a multi-tenant environment voicemail.conf has
  been separated into individual contexts. The users retrieve their
  email by dialing an extension which calls
  VoiceMailMail(x...@vmcontext) so how do I instruct Asterisk to use
  that context when forwarding voicemails ?

Yes exactly that indeed. Though Asterisk appears to ignore which context the 
user is in and selects default instead. Beginning to think that it is a bug.
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[asterisk-users] Voicemail Forwarding

2010-12-17 Thread --[ UxBoD ]--

Experiencing a problem when users attempt to forward a voicemail from within 
VoiceMailMain(Option 8) I see the console message:

Couldn't not find mailbox XXX in context default

As why are running in a multi-tenant environment voicemail.conf has been 
separated into individual contexts.  The users retrieve their email by dialing 
an extension which calls VoiceMailMail(x...@vmcontext) so how do I instruct 
Asterisk to use that context when forwarding voicemails ?
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Re: [asterisk-users] How to find , internal, external inbound or outbound

2010-12-17 Thread --[ UxBoD ]--

- Original Message -
 reply please
 
 On 12/17/2010 10:03 AM, Nikhil wrote:
  Hi
  Does anyone knows how to find out a call in a asterisk is
  external incoming ,external out going or internal
 
  Thanks
  Nikhil
 

Perhaps if you were clearer in the question you are asking ?
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Re: [asterisk-users] Attack problem

2010-12-17 Thread --[ UxBoD ]--


- Original Message -





HI, 



My system been attacked from someone I guess, kindly check the link below 

How can I stop the ircd attack 

http://pastebin.com/tbjh5qzP 



regards 






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[asterisk-users] Asterisk 1.8 crashing

2010-11-26 Thread --[ UxBoD ]--

Hi,

Has anybody had 1.8 crashing for no reason at all ? It has happened a couple of 
times so far and when I check /var/log/asterisk/messages nothing is in there at 
all :(
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Re: [asterisk-users] Asterisk 1.8 crashing

2010-11-26 Thread --[ UxBoD ]--

- Original Message -
 On Fri, Nov 26, 2010 at 8:00 AM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
  Hi,
 
  Has anybody had 1.8 crashing for no reason at all ? It has happened
  a couple of times so far and when I check /var/log/asterisk/messages
  nothing is in there at all :(
  --
  Thanks, Phil
 
  --
 
 1.8 is pretty stable. There are some issues with postgresql / openssl
 that have gotten a lot of work lately. Are you using any external
 programs?
Ah okay, yes am using Postgres for the CDR. Have upgraded to RC1 and see how 
that fairs.
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Re: [asterisk-users] astcanary ?

2010-11-24 Thread --[ UxBoD ]--


- Original Message -


Hello, 

I notice that the following proces is running : 

astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 


What is this ?? 


Kind regards, 
Jonas. 

You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf 
for the line highpriority = yes ; Run realtime priority (same as -p at startup) 
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[asterisk-users] Asterisk 1.8 Release Schedule

2010-11-23 Thread --[ UxBoD ]--

I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta 
or release candidate ?
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Re: [asterisk-users] Asterisk 1.8 Release Schedule

2010-11-23 Thread --[ UxBoD ]--

- Original Message -
 On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote:
  I have read the wiki entry but unsure when we would likely see a
  1.8.0.1 beta or release candidate ?
 
 It will be Asterisk 1.8.1-rc1 and that is now available (as of a few
 minutes ago)
 
 http://www.asterisk.org/node/51466
 
 Leif.
 
Talk about timing :)
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Re: [asterisk-users] URGENT Help needed

2010-11-22 Thread --[ UxBoD ]--


- Original Message -
 Hello,
 
 
 We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after
 trying
 to install iksemel (jabber support) and spandsp, but now Asterisk
 doesn't work anymore and we can't get it to run, althorugh we tried to
 remove it completely and reinstall 1.6.2.13.
 
 
 when trying to start it via /etc/init.d/asterisk start we get the
 following error:
 
 Asterisk died with code 1.
 Automatically restarting Asterisk.
 Asterisk ended with exit status 1
 
 When just trying to run it as asterisk from the command line, we don't
 see the process being active and we get this message when running
 asterisk -r, although the file is present:
 Unable to connect to remote asterisk (does
 /var/run/asterisk/asterisk.ctl exist?)
 
 Any help would be highly appreciated.
 
 Thank you in advance,
 
 Michael
 

What is being reported in /var/log/asterisk/messages ? Do you see any errors 
when you run asterisk from the command line in foreground ?
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Re: [asterisk-users] SIP Extensions and loss of Internet connection

2010-11-22 Thread --[ UxBoD ]--

- Original Message -
 Hi all!
 
 A few days I have problems connecting to the Internet on my house and
 since then my local SIP extensions are no longer registered against
 the
 local Asterisk server.
 
 I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
 found that it can be related to a bug of chan_sip, can it be? In this
 case, is there a possible workaround?
 
 Thanks in advance for your reply.
 
 Regards,
 Daniel
 

Does you Asterisk server point to an internal DNS or to your router ?
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Re: [asterisk-users] SIP Extensions and loss of Internet connection

2010-11-22 Thread --[ UxBoD ]--

- Original Message -
 Hi, Phil.
 
   A few days I have problems connecting to the Internet on my house
   and since then my local SIP extensions are no longer registered
   against the local Asterisk server.
  
   I'm using Asterisk 1.4.24.1. I was researching on the Internet and
   I
   found that it can be related to a bug of chan_sip, can it be? In
   this case, is there a possible workaround?
 
  Does you Asterisk server point to an internal DNS or to your router
  ?
 
 The /etc/resolv.conf of the host on which I installed Asterisk points
 to
 an internal DNS. Is there a parameter in the Asterisk configuration
 where also I have to force the use of an internal DNS server?
 
 Thanks for your reply.
 
 Regards,
 Daniel
 
Do your SIP extensions use your internal DNS server ? are they able to resolve 
the IP of your Asterisk server ? If you enable SIP debugging do you see them 
even try and connect ?
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[asterisk-users] Asterisk 1.8 and Zimbra

2010-11-09 Thread --[ UxBoD ]--

Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber 
service ?  I have opened http://issues.asterisk.org/view.php?id=18198 as it 
keeps failing for me. Am wondering whether it is due to using a self signed 
cert.
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Re: [asterisk-users] SIP Load Balancing

2010-10-28 Thread --[ UxBoD ]--


- Original Message -


I have a very simple setup with two SIP routes to my carrier. I need to have 
every other phone call placed to that carrier go to a different address. 

This is what I need the call flow to look like. I have spent many hours 
searching and have not found a working example. 
Call1 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 ) 
Call2 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 ) 
Call3 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 ) 
Call4 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 ) 
Call5 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 ) 
Call6 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 ) 
Call7 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 ) 
Call8 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 ) 
.. 


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and give those IPs a single name ? 
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Re: [asterisk-users] fraud advice (Also advice on using ipbanning)

2010-10-17 Thread --[ UxBoD ]--


- Original Message -


When we designed our systems on asterisk we designed it to me multi-tenant. Se 
we use customer prefixes on all extensions. This allows us to have multiple 
customers using the same extension pools. It also reduces the hack foot print 
as hackers must know the prefix for a customer to try and brute force things. 
All passwords use 8+ characters with alfa/numeric and special characters. 

As I see it Asterisk does very good keeping out the hackers if you use a solid 
design in your peer and dialplans. At the least put an alpha character post or 
pre other wise you are just asking for it. Use your head you can be smarter 
then they are. 

We are looking into ipban as well. If any one has an example of ipban I would 
love to see how best to implement it. In a 4 year period we have not had a 
breach but we do get about 10 to 15 hack attempts a week. We have blocking 
scripts that block ip's at the primary firewall but I would like to trigger the 
ipban at each switch level. Could I also use the ipban method to trigger the 
audo updates to our primary firewalls? Any advice is appreciated. 


Bryant 



You could also use OSSEC http://www.ossec.net and a custom decoder and rule: 

decoder name=local-asterisk-denied 
prematchNOTICE[\d+] \S+: Registration from /prematch 
regex offset=after_prematch^\S+ failed for '(\d+.\d+.\d+.\d+)'/regex 
ordersrcip/order 
/decoder 

rule id=110005 level=5 
decoded_aslocal-asterisk-denied/decoded_as 
descriptionAsterisk Potentially Under Attack/description 
/rule 

rule id=110006 level=10 frequency=5 timeframe=10 
if_matched_sid110005/if_matched_sid 
same_source_ip / 
descriptionAsterisk Under Brute Force Attack/description 
/rule 
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[asterisk-users] Phantom phone ringing

2010-10-04 Thread --[ UxBoD ]--

Hi,

Running 1.6.2.11 and getting the odd occation that all phones will start 
ringing with nobody on the other end.  From the information we have received 
from the client we can see that a call comes in, it is either answered or not 
answered, but at the same time a second call comes in and it handled by the 
Background auto-attendant.  At that point all their office phones ring.

Any thoughts on go about diagnosing this issue as unless we can catch it when 
it exactly happens we miss the window of opportunity.

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Re: [asterisk-users] Help me Out!!!!

2010-09-15 Thread --[ UxBoD ]--

- Original Message -
 Rough area. Consider yourself lucky you haven't been ripped apart :P
 
 Pete wrote:
  I hope someone has helped poor Rob, I would as I am just over the
  bridge
  in Bristol, UK but some evil internet scammer has stolen all my
  money! ;)
 
  Cheers!
 
 
  On 15/09/10 12:14, Rob Fugina wrote:
  It is with deep sorrow and broken heart that am sending you this
  mail.
  Am in deep need and my situation is lamentable. my family and I
  decide to come visit Wales,United Kingdom for a short vacation. To
  our greatest dismay we were attacked and ripped apart at the park
  of
  the hotel where we were lodging,all cash,credit cards and cell
  phone
  were forcefully robbed off us at gun point but we still have our
  passports with us.
 
  We've seek help at embassy and high commission,the Police too,
  unfortunately they have been unable to help or offer any reasonable
  support whatsoever. Our flight leaves in couple of hour from now
  but
  we are being held to ransom by the hotel management because we
  cannot
  settle the hotel bills. It is clear we would not be allowed to
  leave
  until pay the bill. Word cannot explain the anguish in my heart
  now. I
  am in need of immediate assistance.
 
  Rob
 
 
 
Makes me want to jump on a train and head down to London and help ... 
Unfortunately some unscrupulous person has ran off with my wallet!
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Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)

2010-09-04 Thread --[ UxBoD ]--

- Original Message -
 Roger Burton West wrote:
 
  I want to hook one of them to the PSTN. Given that I am in
  the UK, what is a reasonably easily-available device to
  provide an FXO interface from a Linux box, with a minimum of
  faffing around with drivers? Just one line is needed, though
  in theory two might eventually be useful. My usual white-box
  hardware suppliers don't seem to play in this field.
 
 I've had good experiences with an OpenVox A400P, once you've done the
 Dahdi
 dance, it settles down to be very reliable. Reasonable price, too. I
 bought
 mine from Voipon, although I'm sure a bit of shopping around will find
 other
 vendors.
 
 Cheers,
 Ade.
 

Snap. Same card and supplier. Have had no issues at all.
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Re: [asterisk-users] DAHDI not detecting caller hangup

2010-08-24 Thread --[ UxBoD ]--

- Original Message -
 On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
  Odd problem have just noticed in that when I call into the PBX DAHDI
  detects the call and hands it off to the extension, if I then hang
  up it still continues to process through the dialplan.
 
 It is common for telco not to provide a disconnect tone for Analog.
 You'll need to confirm one is there, either ask your telco or have
 Asterisk record the line. Then update your indications.conf with your
 disconnect tone.
 
 Also be sure you set a TIMEOUT in your dialplans.
 

I am pretty sure that BT (British Telecom) do provide a disconnect tone. 
Hopefully somebody from the UK, Gordon, will be able to confirm this and 
whether they have experienced this issue ?
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Re: [asterisk-users] DAHDI not detecting caller hangup

2010-08-24 Thread --[ UxBoD ]--

- Original Message -
 On Tue, 24 Aug 2010, --[ UxBoD ]-- wrote:
 
  - Original Message -
  On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]--
  ux...@splatnix.net
  wrote:
  Odd problem have just noticed in that when I call into the PBX
  DAHDI
  detects the call and hands it off to the extension, if I then hang
  up it still continues to process through the dialplan.
 
  It is common for telco not to provide a disconnect tone for Analog.
  You'll need to confirm one is there, either ask your telco or have
  Asterisk record the line. Then update your indications.conf with
  your
  disconnect tone.
 
  Also be sure you set a TIMEOUT in your dialplans.
 
 
  I am pretty sure that BT (British Telecom) do provide a disconnect
  tone.
  Hopefully somebody from the UK, Gordon, will be able to confirm this
  and
  whether they have experienced this issue ?
 
 I've not heard a hangup tone myself and I've never enabled disconnect
 tone
 support... Just checked on my home line (bog-standard BT line) -
 calling
 from a mobile and from another VoIP account - Asterisk detected the
 line
 hangup almost immediately... However my systems are tried, tested and
 trusted asterisk 1.2 and Zaptel.
 
 If I don't answer it, but hangup, it does seem to sometimes take a
 second
 or 2 to detect the hangup, but I'm not sure if that's just BT.
 
 Analogue lines. Bah. Hate them. Use VoIP.
 
 Gordon
 

Hi Gordon,

Thanks for that .. Yes I do have VoIP lines as-well ... This is our old number 
on BT which is tied to DSL. Some people still use the number, even though we 
have given them the VoIP number, so had to put in a TDM to handle it.  I know 
what you mean though :)
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Re: [asterisk-users] All phones ringing when temporary loss of Internet

2010-08-24 Thread --[ UxBoD ]--

- Original Message -
 On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
  This is a real strange one and trying to phantom it out. One of our
  clients is connected to our Asterisk installation, from two sites,
  via VPN which works great. Every so often one of the sites VPN
  tunnel goes does for say a couple of seconds. When that happens all
  the extensions, including both sites, ring which is bizarre. Has
  anybody seen this before ? I only see two places in the dial plan
  where all phones are called; and neither should be tripped :(
 
 Enable some SIP debugs and reproduce the problem. Should be simple
 enough to resolve.
 
Thanks Paul, will see if I can get it to happen again.
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[asterisk-users] EMail on Missed Call

2010-08-23 Thread --[ UxBoD ]--

Hi,

Running Asterisk 1.6.2.11 and wondering what would be the best way to send an 
email when a missed call has occurred ? I believe you can modify [stdexten] is 
this still the case in V1.6 ?
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[asterisk-users] DAHDI not detecting caller hangup

2010-08-23 Thread --[ UxBoD ]--

Hi,

Odd problem have just noticed in that when I call into the PBX DAHDI detects 
the call and hands it off to the extension, if I then hang up it still 
continues to process through the dialplan.

This is what I have in chan_dahdi.conf:

[channels]
language=en
echocancel=yes
usecallerid=yes
cidsignalling=v23
sendcalleridafter = 2
hanguponpolarityswitch=yes
rxgain=2.0
txgain=3.0
progzone=uk
signalling=fxs_ks
callerid=asreceived
group=0
context=inbound-dahdi
channel = 1
callerid=
group=
context=inbound-dahdi

and using Asterisk 1.6.2.11 and DAHDI 2.3.0.1.  Dahdi_scan looks okay as-well:

[1]
active=yes
alarms=OK
description=Wildcard TDM400P REV E/F Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV E/F
location=PCI Bus 02 Slot 02
basechan=1
totchans=4
irq=169
type=analog
port=1,FXO
port=2,none
port=3,none
port=4,none
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[asterisk-users] All phones ringing when temporary loss of Internet

2010-08-23 Thread --[ UxBoD ]--

Hi,

This is a real strange one and trying to phantom it out.  One of our clients is 
connected to our Asterisk installation, from two sites, via VPN which works 
great. Every so often one of the sites VPN tunnel goes does for say a couple of 
seconds. When that happens all the extensions, including both sites, ring which 
is bizarre. Has anybody seen this before ? I only see two places in the dial 
plan where all phones are called; and neither should be tripped :(
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[asterisk-users] OT: Grandstream GXV3140

2010-08-06 Thread --[ UxBoD ]--

Hi,

Do any of you have these phones ? How have you found it ? Are you using them 
over WiFi or hard wired ? Does it play nicely with Asterisk ?

Need to replace my Snom M3s and this phone maybe a contender.
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Re: [asterisk-users] OT: Grandstream GXV3140

2010-08-06 Thread --[ UxBoD ]--

- Original Message -
 On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote:
 
  Hi,
 
  Do any of you have these phones ? How have you found it ? Are you
  using them over WiFi or hard wired ? Does it play nicely with
  Asterisk ?
 
  Need to replace my Snom M3s and this phone maybe a contender.
 
 Full of bugs. Stay away. Vendors have started dropping this phone. I
 use them personally when traveling (to talk to my baby girl), but as a
 commercial product this phone is AWFUL.
 
 j

Oh, that is very disappointing indeed; especially some of the others bits I 
read on the net :( Desperately trying to find a new phone that will support:

* Centralised management and deployment
* OpenVPN
* Asterisk support

The company I work for have got very despondent with Snom and their lack of 
support.
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Re: [asterisk-users] SIP response 482 Loop Detected

2010-07-06 Thread --[ UxBoD ]--

- Original Message -
 - Original Message -
  On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
  wrote:
  
   - Original Message -
   Hi,
  
   We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
   that
   we are unable to URI dial our clients. We run a multi-tenant
   server
   and have set sip.conf to forward calls to a public context based
   on
   incoming domain name. This was all working before but not it is
   complaining of a loop back as the source and target server are
   the
   same.
  
   Any ideas on how to overcome this problem as we dial our clients
   based
   on their email address.
  
   Grabbing a SIP debug I see:
  
   --- Transmitting (no NAT) to 10.172.120.5:5060 ---
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
   10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
   From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
   To: sip:us...@seconddomain.com
   Call-ID: 66b3314cc6d1-jxu0nhluv4zt
   CSeq: 2 INVITE
   Server: secret
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
   NOTIFY,
   INFO
   Supported: replaces, timer
   Require: timer
   Session-Expires: 1800;refresher=uas
   Contact: sip:us...@172.30.14.8
   Content-Length: 0
  
   And am guessing that as the source from IP matches the Contact:
   address Asterisk sees that as a loop ?
 
  I don't know these things, but you should probably post more of a
  SIP
  trace. Maybe turn on full sip debug to a file for long enough to see
  what the SIP conversation looks like that asterisk 1.6.2.9 is having
  with itself.
 
 
 From what I have read hairpin calls are not supported by asterisk;
 so am guessing something has been fixed in the 1.6.2.X branch that
 should have not worked in 1.6.1.X anyway :) While I continue the
 research have implemented using a workaround via the AstDB and the
 following changes to the uri-dial plan:
 
 exten =
 _[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${ext...@${sipdomain})}?inturi:exturi)
 exten =
 _[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${ext...@${sipdomain})})
 exten = _[a-zA-Z0-9].,n(exturi),Macro(uridial,${ext...@${sipdomain})
 
 This is a bit of pain as we have to make sure we update the DB when a
 new inbound URI is added; though it works and means we can stick with
 the 1.6.2.X branch.
 
 Would be interested to hear from a dev though as to whether they think
 it should work as we originally had it configured ?

Do you think this should be raised as a issue in bugtraq or at least brought up 
on the asterisk-dev mailing list ?
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[asterisk-users] SIP response 482 Loop Detected

2010-07-05 Thread --[ UxBoD ]--

Hi,

We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are 
unable to URI dial our clients. We run a multi-tenant server and have set 
sip.conf to forward calls to a public context based on incoming domain name. 
This was all working before but not it is complaining of a loop back as the 
source and target server are the same.

Any ideas on how to overcome this problem as we dial our clients based on their 
email address.
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Re: [asterisk-users] SIP response 482 Loop Detected

2010-07-05 Thread --[ UxBoD ]--


- Original Message -
 Hi,
 
 We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
 we are unable to URI dial our clients. We run a multi-tenant server
 and have set sip.conf to forward calls to a public context based on
 incoming domain name. This was all working before but not it is
 complaining of a loop back as the source and target server are the
 same.
 
 Any ideas on how to overcome this problem as we dial our clients based
 on their email address.

Grabbing a SIP debug I see:

--- Transmitting (no NAT) to 10.172.120.5:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
To: sip:us...@seconddomain.com
Call-ID: 66b3314cc6d1-jxu0nhluv4zt
CSeq: 2 INVITE
Server: secret
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:us...@172.30.14.8
Content-Length: 0

And am guessing that as the source from IP matches the Contact: address 
Asterisk sees that as a loop ?
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Re: [asterisk-users] SIP response 482 Loop Detected

2010-07-05 Thread --[ UxBoD ]--

- Original Message -
 On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
 wrote:
 
  - Original Message -
  Hi,
 
  We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
  that
  we are unable to URI dial our clients. We run a multi-tenant server
  and have set sip.conf to forward calls to a public context based on
  incoming domain name. This was all working before but not it is
  complaining of a loop back as the source and target server are the
  same.
 
  Any ideas on how to overcome this problem as we dial our clients
  based
  on their email address.
 
  Grabbing a SIP debug I see:
 
  --- Transmitting (no NAT) to 10.172.120.5:5060 ---
  SIP/2.0 100 Trying
  Via: SIP/2.0/UDP
  10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
  From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
  To: sip:us...@seconddomain.com
  Call-ID: 66b3314cc6d1-jxu0nhluv4zt
  CSeq: 2 INVITE
  Server: secret
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
  INFO
  Supported: replaces, timer
  Require: timer
  Session-Expires: 1800;refresher=uas
  Contact: sip:us...@172.30.14.8
  Content-Length: 0
 
  And am guessing that as the source from IP matches the Contact:
  address Asterisk sees that as a loop ?
 
 I don't know these things, but you should probably post more of a SIP
 trace. Maybe turn on full sip debug to a file for long enough to see
 what the SIP conversation looks like that asterisk 1.6.2.9 is having
 with itself.
 

From what I have read hairpin calls are not supported by asterisk; so am 
guessing something has been fixed in the 1.6.2.X branch that should have not 
worked in 1.6.1.X anyway :) While I continue the research have implemented 
using a workaround via the AstDB and the following changes to the uri-dial 
plan:

exten = 
_[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${ext...@${sipdomain})}?inturi:exturi)
exten = _[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${ext...@${sipdomain})})
exten = _[a-zA-Z0-9].,n(exturi),Macro(uridial,${ext...@${sipdomain})

This is a bit of pain as we have to make sure we update the DB when a new 
inbound URI is added; though it works and means we can stick with the 1.6.2.X 
branch.

Would be interested to hear from a dev though as to whether they think it 
should work as we originally had it configured ?
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[asterisk-users] OT: Physical SIP phone with inbuilt VPN support

2010-06-18 Thread --[ UxBoD ]--

Hi, all

Would any of you be able to suggest physical SIP phones that support inbuilt 
VPN capabilities; akin to the Snom 370/870 ?
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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread --[ UxBoD ]--

 * Skype for Asterisk needs to run on this - so this means x86, right?

or x86_64 is fine

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Re: [asterisk-users] Dual Atom mobo - call capacity

2010-06-11 Thread --[ UxBoD ]--


- Original Message -
 On Thu, 10 Jun 2010, Michelle Dupuis wrote:
 
  I'm looking for a small formfactor mobo for an install that needs to
  handle 25 phone sets (no transcoding). I found a new dual atom
  1.66GHz
  mobo - anyone know what kinds of call volume that will handle?
 
 On Thu, 10 Jun 2010, mgra...@mstvp.com wrote:
 
  Based on comments from Ward Mundy during a recent VUC call I'd
  expect
  even a single CPU Atom system to handle that many phones in an
  office
  application. Perhaps there may be merit in dual CPU in more of a
  call
  center application.
 
 Assuming you're talking about something like the Atom 330...
 
 My guess is you will have plenty of horsepower for 25 phone sets --
 probably even 25 simultaneous calls.
 
 The 330 is dual-core and hyper-threaded so it shows up as 4 CPUs in
 top.
 
 Asterisk is multi-threaded and should distribute the workload. Another
 advantage is that if you have something CPU heavy like bzip2'ing your
 database dump or compiling Asterisk from source, there are still
 several
 CPUs available for Asterisk.
 

I have a single rack server with a Atom 330 and 2GB RAM, six phones connected 
and probably a couple of simultaneous calls at one time.  This is how it looks 
at the moment:

 total   used   free sharedbuffers cached
Mem:   20498561346480 703376  0 181920 990376
-/+ buffers/cache: 1741841875672
Swap:  4095992  04095992

top - 10:41:59 up 12 days, 16:03,  1 user,  load average: 0.01, 0.00, 0.00
Tasks: 122 total,   1 running, 121 sleeping,   0 stopped,   0 zombie
Cpu0  :  0.0%us,  0.0%sy,  0.0%ni, 99.9%id,  0.0%wa,  0.0%hi,  0.0%si,  0.0%st
Cpu1  :  0.0%us,  0.0%sy,  0.0%ni, 98.4%id,  0.0%wa,  1.5%hi,  0.0%si,  0.0%st
Cpu2  :  0.1%us,  0.0%sy,  0.1%ni, 99.8%id,  0.0%wa,  0.0%hi,  0.0%si,  0.0%st
Cpu3  :  0.1%us,  0.0%sy,  0.0%ni, 99.8%id,  0.0%wa,  0.0%hi,  0.0%si,  0.0%st
Mem:   2049856k total,  1346232k used,   703624k free,   181920k buffers
Swap:  4095992k total,0k used,  4095992k free,   990376k cached

Have a TDM card in the server and also use G729 codec and Skype for Asterisk.
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Re: [asterisk-users] Asterisk 1.6.2.8 Now Available

2010-06-02 Thread --[ UxBoD ]--

- Original Message -
 The Asterisk Development Team has announced the release of Asterisk
 1.6.2.8. This release is available for immediate download at
 http://downloads.asterisk.org/pub/telephony/asterisk/

Which release will http://issues.asterisk.org/view.php?id=17135 make it into; 
was it to late for this one ?
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Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

2010-05-29 Thread --[ UxBoD ]--



- Original Message -
 On Friday 28 May 2010 12:50:19 --[ UxBoD ]-- wrote:
  - Original Message -
 
   You're missing this in your chan_dahdi.conf:
  
   #include dahdi-channels.conf
 
  Hmm, I changed the signalling as per a previous post and now it is
  okay. Why is there chan_dahdi.conf and dahdi-channels.conf ?
 
 Clearly, he's using something like FreePBX and not pure Asterisk. It's
 not needed on most systems.
 
 -- Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org

I am using Asterisk 1.6.2.7 and that file was generated by dahdi_genconf. Am I 
safe to remove the file then ?
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[asterisk-users] DAHDI Help (made a cardinal sin :()

Looking for some help from the UK please.  I backed up all my Asterisk 
configuration before re-installing the server from 32 - 64 bit.  Unfortunately 
I did not transfer the backup to another machine!

I now have a TDM400P that is not picking up the line.  Can you see what I have 
done wrong when I have rebuilt the config please:

dahdi_scan
--
[1]
active=yes
alarms=OK
description=Wildcard TDM400P REV E/F Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV E/F
location=PCI Bus 02 Slot 02
basechan=1
totchans=4
irq=169
type=analog
port=1,FXO
port=2,none
port=3,none
port=4,none

/etc/dahdi/system.conf
--
fxsks=1
echocanceller=mg2,1
loadzone= uk
defaultzone = uk

dmesg when loaded dadhi
---
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.3.0.1
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXO (UK mode)
Module 1: Not installed
Module 2: Not installed
Module 3: Not installed
Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules)
dahdi: Registered tone zone 0 (United States / North America)
dahdi_echocan_mg2: Registered echo canceler 'MG2'
dahdi: Registered tone zone 4 (United Kingdom)

/etc/asterisk/chan_dahdi.conf
-
[channels]
language=en
usecallerid=yes
cidsignalling=v23
sendcalleridafter = 2
rxgain=2.0
txgain=3.0
progzone=uk
signalling=fxo_ks
callerid=asreceived
group=0
context=inbound-dahdi
channel = 1
callerid=
group=
context=inbound-dahdi

/etc/asterisk/dahdi-channels.conf
-
; Autogenerated by /usr/sbin/dahdi_genconf on Thu May 27 13:25:14 2010
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
;;; line=1 WCTDM/4/0 FXSKS
signalling=fxs_ks
callerid=asreceived
group=0
context=inbound-dahdi
channel = 1
callerid=
group=
context=default

Really need some help please, Gordon ;)

Thank you all.
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Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message -
 --[ UxBoD ]-- wrote:
  Looking for some help from the UK please. I backed up all my
  Asterisk configuration before re-installing the server from 32 - 64
  bit. Unfortunately I did not transfer the backup to another
  machine!
 
  I now have a TDM400P that is not picking up the line. Can you see
  what I have done wrong when I have rebuilt the config please:
 
  dahdi_scan
  -- [1]
  active=yes alarms=OK
  description=Wildcard TDM400P REV E/F Board 5
  name=WCTDM/4 manufacturer=Digium
  devicetype=Wildcard TDM400P REV E/F
  location=PCI Bus 02 Slot 02
  basechan=1 totchans=4
  irq=169 type=analog
  port=1,FXO port=2,none
  port=3,none port=4,none
 
  /etc/dahdi/system.conf
  -- fxsks=1
  echocanceller=mg2,1 loadzone = uk
  defaultzone = uk
 
  dmesg when loaded dadhi
  ---
  dahdi: Telephony Interface Registered on major 196
  dahdi: Version: 2.3.0.1
  Freshmaker version: 71
  Freshmaker passed register test
  Module 0: Installed -- AUTO FXO (UK mode)
  Module 1: Not installed
  Module 2: Not installed
  Module 3: Not installed
  Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules)
  dahdi: Registered tone zone 0 (United States / North America)
  dahdi_echocan_mg2: Registered echo canceler 'MG2'
  dahdi: Registered tone zone 4 (United Kingdom)
 
  /etc/asterisk/chan_dahdi.conf
  - [channels]
  language=en usecallerid=yes
  cidsignalling=v23 sendcalleridafter = 2
  rxgain=2.0 txgain=3.0
  progzone=uk signalling=fxo_ks
  callerid=asreceived group=0
  context=inbound-dahdi channel = 1
  callerid= group=
  context=inbound-dahdi
 
  /etc/asterisk/dahdi-channels.conf
  - ; Autogenerated by
  /usr/sbin/dahdi_genconf on Thu May 27 13:25:14 2010
  ; If you edit this file and execute /usr/sbin/dahdi_genconf again,
  ; your manual changes will be LOST.
  ; Dahdi Channels Configurations (chan_dahdi.conf)
  ; ; This is not intended to be a complete chan_dahdi.conf. Rather,
  it is intended
  ; to be #include-d by /etc/chan_dahdi.conf that will include the
  global settings
  ;
 
  ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
  ;;; line=1 WCTDM/4/0 FXSKS
  signalling=fxs_ks callerid=asreceived
  group=0 context=inbound-dahdi
  channel = 1
  callerid= group=
  context=default
 
  Really need some help please, Gordon ;)
 
  Thank you all.
 
 in chan_dahdi.conf are you sure you have the correct signalling
 defined? Shouldnt it be fxs_ks in both places?

Yup, that was spot on.  Thank you so so much  :)
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Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message -
 On Fri, 28 May 2010, --[ UxBoD ]-- wrote:
 
 
 [NON-Text Body part not included]
 
 
 Er, my mailer's obviously struggled to interpret this, however did you
 do
 the include and I also have this in /etc/modprobe.d in a file:
 
 options wctdm opermode=UK
 
 
 Gordon
 

Yep; sorry missed that off the post but that is in.
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Re: [asterisk-users] DAHDI Help (made a cardinal sin :()

- Original Message -
 - --[ UxBoD ]-- ux...@splatnix.net wrote:
  /etc/asterisk/chan_dahdi.conf
  - [channels]
  language=en usecallerid=yes
  cidsignalling=v23 sendcalleridafter = 2
  rxgain=2.0 txgain=3.0
  progzone=uk signalling=fxo_ks
  callerid=asreceived group=0
  context=inbound-dahdi channel = 1
  callerid= group=
  context=inbound-dahdi
 
  /etc/asterisk/dahdi-channels.conf
  - ; Autogenerated by
  /usr/sbin/dahdi_genconf on Thu May 27 13:25:14
  2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf
  again, ; your manual changes will be LOST.
  ; Dahdi Channels Configurations (chan_dahdi.conf)
  ; ; This is not intended to be a complete chan_dahdi.conf. Rather,
  it is
  intended ; to be #include-d by /etc/chan_dahdi.conf that will
  include the
  global settings
  ;
 
  ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
  ;;; line=1 WCTDM/4/0 FXSKS
  signalling=fxs_ks callerid=asreceived
  group=0 context=inbound-dahdi
  channel = 1
  callerid= group=
  context=default
 
 
 You're missing this in your chan_dahdi.conf:
 
 #include dahdi-channels.conf
 

Hmm, I changed the signalling as per a previous post and now it is okay.  Why 
is there chan_dahdi.conf and dahdi-channels.conf ?
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Re: [asterisk-users] is my PHPAGI Soap code right?

2010-05-17 Thread --[ UxBoD ]--

- Original Message -
2010/5/14 --[ UxBoD ]-- ux...@splatnix.net:

- Original Message -
Hello,

i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.

this is my phpagi script, Could you show me what's wrong ? becasue
i can't excute it successfully.

please open the following url to see my code:

http://pastebin.com/uzvWSxPy

Thanks!

Perhaps if you explained what errors you were seeing would help ?
Have you tried running it from the CLI to see if the syntax is
correct ?

Thanks! the systax is right in my php code. but when excute the php
script. there is errer happend in the server side.

as follows, i don't know what's wrong with it, please help me. thank
you:

2010-05-17 14:08:19,359 INFO
[org.codehaus.xfire.handler.DefaultFaultHandler] - Fault occurred!
org.codehaus.xfire.fault.XFireFault: Not enough message parts were
received for the operation.
at
org.codehaus.xfire.service.binding.ServiceInvocationHandler.fillInHolders(ServiceInvocationHandler.java:238)
at
SNIP
Well that to me looks like the remote Servlet is expecting additional
parameters which your SOAP call has not supplied; it does not appear to me to
be a Asterisk AGI issue.
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Re: [asterisk-users] is my PHPAGI Soap code right?

2010-05-14 Thread --[ UxBoD ]--


- Original Message -
 Hello,
 
 i try to use soap in the phpagi.
 i want to call a function from a web service
 when a user call a telephne failed.
 
 this is my phpagi script, Could you show me what's wrong ? becasue i
 can't excute it successfully.
 
 please open the following url to see my code:
 
 http://pastebin.com/uzvWSxPy
 
 Thanks!
 

Perhaps if you explained what errors you were seeing would help ? Have you 
tried running it from the CLI to see if the syntax is correct ?

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Re: [asterisk-users] Interesting email project.

2010-05-04 Thread --[ UxBoD ]--

- Original Message -
 mike mosier wrote:
 
  Hey all.
 
  My boss asked me to implement the following
 
  When DID 713xxx is dialed send an email to mmos...@xxx.com
  mailto:mmos...@xxx.com. with the time date and CID included in the
  email. I know how to code some but am looking for the best way to do
  this.
 
  Sorry I might have asked this a couple months back. I forgot.
 
  Mmosier
  Houston
 
  Respectfully
  Michael D Mosier
  Ftoc Certified
 
 
 Here is the script I am using for email alert. Form Asterisk dialplan:
 exten = h,1,System(/path/to/the/script/emailnotice.sh
 some...@gmail.com ${CALLERID(num)} ${CALLERID(name)}
 ${DIALSTATUS} ${VMSTATUS} ${MYEXTEN}
 ${STRFTIME(${EPOCH},,%Y/%m/%d %H:%M)})
 

you could always use the PHP AGI interface to send the email and log 
information to a database ?

eg.

exten = h,1,AGI(sendemailandlog.php)
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

2010-05-01 Thread --[ UxBoD ]--

- Original Message -
 Randy-
 
  On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy m...@parsetree.com
  wrote:
  Assuming that every such spamming/hacking/attack site is funded on
  a stolen identity/CC number, it will soon sink into Amazon that
  they are
  getting a bad rep, and losing money on such problems, as all such
  charges are reversed when the identity theft is discovered... How
  they overcome
  the problem, should be a tribute to the marvelous power of human
  ingenuity.
 
  Interesting point about the stolen CC numbers. If that is true, then
  they will be forced to investigate for their own internal damage
  control.
 
 You are nothing if not persistent, an excellent quality in a case like
 this. By now I'm sure Amazon execs are
 wondering who is this Randulo guy, hehe.
 

Slammed again last night by a A-WS server; see if anything comes back from 
their abuse department!

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message -
 Think we need some solution WITHIN the Asterisk core. Roderick A.
 suggested something that looks nice using iptables, some others have
 pointed out using RBL or fail2ban, but the best would be to have some
 generic solution not dependant on third party programs.
 
 I'm not aware of the asterisk.dev list but maybe someone can tell if
 they can help us here?
 
 Alyed
 
 
 
 2010/4/13 Randy R  randulo2...@gmail.com 
 
 
 
 On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman
  dhart...@djhsolutions.com  wrote:
  That only addresses EC2 (and assumes that Amazon has any interest in
  protecting their reputation). What about attacks that come from
  other locations? Granted it's pretty easy to buy time on an EC2
  server so
  this may be the primary source for a period of time.
 
 With the growth of the cloud offerings, this problem will likely grow,
 so yes, a generic solution is needed. What I want to see though, and
 no provder has done much if anything about it, is REPORTING and
 INVESTIGATION. It is easy to use a script to report and submit, we can
 all do that, even I could (if I had a box running and needed to). The
 hard part is them having their tech/sys people actually look at the
 network and see, Oh, ya, there's some shit happening that on that
 instance...
 
 If Amazon's form submit didn't even work, that's a really bad
 reflection on their brand in a lot of ways, including tech competence.
 If that is know to geeks like us, it won't hurt them which is why,
 like a broken record, I keep saying: put your Amazon experience out to
 the public. When it starts being mentioned in Wired, Storm Cloud or
 something, THEN Amazon will have to do something.
 
 I do not believe Amazon is taking reasonable measures now in doing
 their job, and that they should be working towards that goal,
 reasonable measures as opposed to NO measures.
 
 /r
 
 
 
 

DNS lookup capability appears to be required on a Asterisk installation and 
hence a DNSRBL would appear to be a good solution. A alternative, similar to 
the SaneSecurity AV sigs, would be to have a pool of rsync servers for 
downloading a list of known IPs.  Again this would require community 
contribution in both time and resources.  I would be happy to allocate some 
spare memory and CPU cycles and hopefully my employer would as-well.
-- 
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message -
 On Tue, 13 Apr 2010, Alyed wrote:
 
  Think we need some solution WITHIN the Asterisk core. Roderick A.
  suggested something that looks nice using iptables, some others have
  pointed out using
  RBL or fail2ban, but the best would be to have some generic solution
  not dependant on third party programs.
 
 I'd strongly disagree with this. (And I was the OP of this thread and
 had my home/office network connection taken down due to it)
 
 But then, I'm an old worldy Unix sysadmin and the philosophy of having
 a program do one thing well is still etched into my core...
 
 http://en.wikipedia.org/wiki/Unix_philosophy
 
 So get asterisk to do what it does well, then get something else that
 does what you need to do just as well - built-in to Linux are the
 iptables firewall rules. Use them! They are very effective and do
 work. (And you
 have a choice!)
 
 The biggest issue I see is that people are installing Asterisk and
 other high-level applications on top of Linux (and other *nix'es)
 without the
 experience of sysadmin - then when something goes wrong they want
 the application to fix it rather than apply some basic and pretty
 fundamental sysadmin techniques to solve the issue.
 
 And that means that even having permit= and deny= in sip.conf and
 iax.conf, etc. is too much. With proper OS level firewalling they're
 simply not needed and do nothing more than add another potential point
 of failure and add yet more code to maintain.
 
 Gordon
 

Gordon,

Completely agree with what you are saying though I believe the proposal of some 
sort of shared IP list is a valid one.  If you had not brought this to the 
attention of the list then this discussion would have not taken place.  I am 
guilty in that when a EC2 server attempted to break into my PBX I did not share 
it with the list.  We, large assumption, are all at some point subjected to 
probing attacks against our Asterisk deployments and I feel it would be great 
if there was some mechanism where we were able to share those hackers IPs for 
blocking by one means or another.
-- 
Thanks, Phil

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

- Original Message -
 Speaking of all these attacks, are there any good web managed security
 monitor tools for CentOS out there that can be installed on the system
 so that it can give us a visual of let's multiple failed attempts
 against SSH or HTTPd?
 
 
 Something nice that is simple and doesn't eat a lot resources and
 spits out everything on the screen?
 
 
 Thanks,
 Bruce

How about http://www.ossec.net which you could later integrate with 
http://www.splunk.com/.

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Thanks - Phil

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...