Re: [asterisk-users] bash: asterisk: command not found

[Steve Howes]

On 07/12/16 04:56, christopher kamutumwa wrote:
Ive installed asterisk 14.2 on centos 6.8 but i am not able to start 
it below is what am executing and those are the errors anything am 
doing wrong?


It doesn't look like it is installed to me... Check the install actually 
worked etc. I've never had to do any path changes or anything for 
asterisk on centos so I suspect it just isn't there...


Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)

[Steve Howes]

On 28/10/16 16:38, Markus wrote:
I'm using Asterisk2Billing (v2.0.16) and it appears to have an 
annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 
(UK premium) and a fraudster manages to somehow dial 44-870 instead of 
44870 the rate for 44 will match, not the one for 44870.


44 is *not* UK landline, you shouldn't even have a rate for it. 44 is 
the country code. A (very) brief summary is:


441 Landline
442 Landline
443 Landline (at least for billing)
447 Mobile
4470 Personal rate (rarely used, expensive)
44800/8 Freephone
4484/5 Special Services Lower Rate
4487 Special Services Higher Rate
449 Premium rate

Having a correct rates table / normalising and validating your inputs 
(as in FILTER) would both have potentially stopped the attack.


Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Steve Howes]

On 01/08/16 09:08, Nabeel wrote:

I am yet to test this behaviour in Asterisk during the 
Unavailable/Busy message. However, if this is the case, then this 
seems to be an illogical security hole in Asterisk's design. Why does 
Asterisk allow accessing another person's mailbox by pressing the '*' 
key, while listening to /the other person's/ unavailable message?




So you can access your own voicemail remotely.

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] implementing asterisk call center.

[Steve Howes]

On 06/04/16 20:58, Goke Aruna wrote:
Can someone help me with a kind of howto build call center around 
asterisk with all the necessary features like CTI, call recordings, 
call spying, real time monitoring etc?

What is your budget? I'm sure there are many contractors who can help.

Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP trunk with whatsapp

[Steve Howes]

On 28/03/16 12:46, bilal ghayyad wrote:

Does anyone has information if possible to setup SIP trunk with whatsapp?
How can we let asterisk send and receive calls from whatsapp?


I don't think you can. Whatsapp is a closed system.

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice recognition IVR Is it possible?

[Steve Howes]

On 22/02/16 23:58, Frank wrote:

On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote:
...
Speech API key from Google
Yes... OK... but... where and how can I obtain this API Key?

Google?...

Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Authenticate() 11.21.0

[Steve Howes]

On 10/02/16 14:20, Jerry Geis wrote:

I am trying to use Authenticate() in the dialplan
for something other than "my password".
The message says "Please enter YOUR password followed by the pound key".

I'm not using this for my password.
Is there any way to change the message to "please enter the password 
followed by the pound key"?


or is there another version of Authenticate() that I'm not aware of or 
another way to prompt for a password?


READ() ?

Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to deal with error messages passed as Early Media

[Steve Howes]

 On 03/02/16 15:29, Olivier wrote:
2016-02-03 15:59 GMT+01:00 Steve Howes <steve-li...@geekinter.net 
<mailto:steve-li...@geekinter.net>>:




On 03/02/16 14:41, Olivier wrote:

How can I best deal with error messages passed as Early Media.

Tell the ITSP to give you proper signaling, if they wont then get
a new ITSP. I suspect if they can't handle this correctly, there
will be a lot more they're doing wrong as well. Long term you'll
save yourself a whole lot of bother.


Yes but I'm afraid that, in this industry, the rule is to pass 
anything received to the other party.
In that case I wish you the best of luck. You can't process audio and 
turn it into a proper signal. If they don't send a SIP/ISDN signal then 
you're stuffed.


I still maintain the best way is to get the right thing sent to you in 
the first place - it's a basic interop requirement that data is 
consistent (even if it's not exactly the format what you want)


Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to deal with error messages passed as Early Media

[Steve Howes]

On 03/02/16 14:41, Olivier wrote:

How can I best deal with error messages passed as Early Media.
Tell the ITSP to give you proper signaling, if they wont then get a new 
ITSP. I suspect if they can't handle this correctly, there will be a lot 
more they're doing wrong as well. Long term you'll save yourself a whole 
lot of bother.


Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fwd: Sublime Text License Key

[Steve Howes]

Wonder what happens when an entire mailing list tries to use that key?...

On 05/10/15 15:28, Optical Phoenix wrote:

-- Forwarded message --
From: *Sublime HQ Pty Ltd* >

Date: Wednesday, July 25, 2012
Subject: Sublime Text License Key
To: "opticalphoe...@gmail.com " 
>



Hello,

Thanks for purchasing a copy of Sublime Text! Your license key is:

- BEGIN LICENSE -
Dennis Wright Jr
Single User License
EA7E-819939
356F68A3 BDE42447 A0B7E2C4 9429E490
1760A71B C59AF641 94066F0A 04146120
6F5FC041 A95B5175 139BB680 4EB40EFD
C50C4829 806BCC12 E2C80B94 77474B29
D1224F42 F916634C 68CE1BBB 96F1D6D0
EA547ED4 2E695093 CC474A9B 755D3E9E
00CAF5FB 77AA4C22 12FC089C 17A0B891
61DDD391 808E58EE 2F9AA80E B04E344A
-- END LICENSE --

Entering the license details:

1. Open Sublime Text, and select Help/Enter License from the menu.
2. Copy the license above (including the BEGIN LICENSE and END LICENSE 
lines) and paste them into the license box.
3. Press the Use License button and Sublime Text will enter into 
licensed mode.


Regards,

Jon
SUBLIME HQ PTY LTD






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fwd: Sublime Text License Key

[Steve Howes]

On 05/10/15 16:18, Mitul Limbani wrote:


The company making sublime text gets few thousands of dollars of 
notional loss :)


I was thinking more about if they'd built in software activation type 
stuff. But yea, stealing bad etc too.


Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] uptime

[Steve Howes]

On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
 minipbx*CLI show uptime
 System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
 Last reload: 8 hours, 3 minutes, 51 seconds

What's the highest current 'genuine' one on-list?..

klein*CLI core show uptime
System uptime: 2 years, 1 week, 4 days, 21 hours, 52 minutes 
Last reload: 41 weeks, 6 days, 16 hours, 6 minutes, 39 seconds 


That's the best I can come up with..

S

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hide the plain text password

[Steve Howes]

On 15 Feb 2011, at 13:17, Richard Kenner wrote:
 Of course not!  It would be useless if that were the case: the whole
 point here would be that you need the master encryption key.
 
 Here's a possible design:
 
 - There's optionally a file in the config
  directory called master_key.  It contains just a string.
 
 - A CLI command core encrypt string is added to Asterisk.  It takes the
  provided string, encrypts it using the string in master_key, and outputs
  a string of the form {enc:encrypted_version_of_string}.
 
 - The config file reader looks for strings of the form {enc:string}:
  and replaces them, before otherwise parsing the line, with the decrypted
  version of the string using the key in the master_key file.

Let us know when you've made the patch..

S

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] On-Hold Music

[Steve Howes]

On 11 Feb 2011, at 22:37, Danny Nicholas wrote:
   In 500 words or less (if possible), please explain what is a legal 
 music-on-hold file?

Depends on the country, and what licence you posses. Googling 'countryname 
hold music regulations' may help.

S--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ${HANGUPCAUSE} in CDR

[Steve Howes]

On 8 Feb 2011, at 13:30, Shariq Khan wrote:
 Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I 
 want to add the Hangup reason of call in userfield of CDR.

http://www.google.com/search?q=asterisk+hangupcause+cdr

Top result... Should do it

Steve


Steve Howes
SMTP to Google proxy Inc
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] forward calls by the ports

[Steve Howes]

On 8 Feb 2011, at 14:52, mehran khajavi wrote:
 i searched a lot but i couldn't find the answer

.

 i have two openvox(fxo/fxs) card so I have 24 ports!

Ok!

 on first card i have 12 fxs and on the second i have 12 fxo
 i want to then one person calling from  dahdi/13 forward it to dahdi/1
 when a person calling from  dahdi/14 forward it to dahdi/2
 when a person calling from dahdi/15 forward it to dahdi/3
 
 how can i do this?

You dont need a PBX for that... Just plug the phones into the line?..

 i should make an AGI? or can i make it with extentions.conf? how can i get 
 the caller's port number?

You could do either. extensions.conf is more sensible. Put ports in different 
contexts / use channel variables. How to do this is probably in the extensive 
documentation you've been studying.

S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] faxter

[Steve Howes]

On 30 Jan 2011, at 09:21, Pezhman Lali wrote:
  Faxter is an opensource email to fax gateway, 
 please check it, let me know if any bug.

Only bug i can see is the attitude of the developer... 

As for the bugs, having the config variables liberally scattered throughout the 
script makes it's use (and then subsequent update) near impossible. There are 
even context names towards the end of the file. Ideally you'd want a separate 
config.php which you then include from your main script. A readme would then 
document what you'd put in here (and their default values if you dont). The 
tabbing is pretty random, and the commented out test data is pretty decorative.

chmod($save_dir.$filename,0777);

Is a slightly interesting idea.

Not actually run it to see if it works, wouldn't know how..

S


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Unable to insert cdr-data into mysql-DB

[Steve Howes]

On 25 Jan 2011, at 09:36, Andrew Thomas wrote:
 Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the
 cdr_mysql.conf.  I seem to remember a problem I had when '127.0.0.1' and
 'localhost' didn't marry up never did find out why.

I believe localhost means it can use a socket, where as 127.0.0.1 forces IP.

S

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk stops responding

[Steve Howes]

On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
 Cannot allocate memory

Have you tried looking at memory?

S

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Mailing list question 2

[Steve Howes]

On 20 Jan 2011, at 17:13, Andrew Thomas wrote:
 Sorry about this - testing this disclaimer problem :)

I can give you a POP3 account on my server if it stops you spamming the list?..

S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Continuously core dumping of 1.8 on SLES

[Steve Howes]

On 17 Jan 2011, at 11:29, Hans Witvliet wrote:
 Missing something obviously,

core dump / backtrace? ;)

Might be worth knocking a few of the modules out that were listing errors to 
see if any of them are causing it. It's possible something not loading isn't 
being handled gracefully.

S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to check a number online or offline

[Steve Howes]

On 10 Jan 2011, at 10:17, Phuong Hoang wrote:

 Thanks enkillar, but this is`nt thing that i need. I want to check number 
 online, offline or unreachable on asterisk using AMI(Asterisk Manager 
 Interface) by java but i have`nt found a solution yet. I hope you can help me 
 do this.
 Thanks in advance !

http://www.voip-info.org/wiki/view/Asterisk+manager+API

There are a number of commands there that would help if you'd bothered to 
look.. Can retrieve sip peers with one, or a generic 'command' command would do 
it too in most cases..


S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to check a number online or offline

[Steve Howes]

On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to resolve this. 
Can you say clearlier for me?

Not really. It's a list of manager commands. There is 'SIPshowpeer' which will 
work for sip stuff. Try the command 'Command' action and you can send any CLI 
command, like sip/iax2 show peers etc. 'ExtensionState' might work in some 
cases..

S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call sip:u...@domain.com?

[Steve Howes]

On 21 Dec 2010, at 14:20, A J Stiles wrote:
  Well, every Free and Open Source telephony system is using Asterisk  (and 
 Linux)  under the bonnet.  The differences are in the user configuration 
 tools.

Uh, no?

S

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Mail Integration

[Steve Howes]

On 13 Dec 2010, at 14:25, Danny Nicholas wrote:
 (god forbid) postal mail

Haha, I'm kind of tempted to write an app_cups module to print envelopes ;)

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Push central phone book to phones

[Steve Howes]

On 7 Dec 2010, at 11:35, Jonas Kellens wrote:
 When on a public server, I find this insecure.

Then secure it? Tie down by IP address, or some phones support the 
username:password@ in a URL.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Abandon events in cdr

[Steve Howes]

On 3 Dec 2010, at 13:47, Rodrigo Lang wrote:
 unansweredy = yes

Remove the extra y.

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)

[Steve Howes]

On 1 Dec 2010, at 10:18, Michael Nausch wrote:
 If I start asterisk 1.8 with service asterisk start or 
 /etc/init.d/asterisk start, I can't load chan_misdn.so
 
 If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card 
 and I be able to dial out to my PSTN provider! ;)

File permissions? If you run with init.d script it may be running under 
'asterisk' user not 'root'. If files are not readable by asterisk it wont load 
it.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem

[Steve Howes]

On 30 Nov 2010, at 09:28, bilal ghayyad wrote:
 If I ran IAX in TCP port, and in case my network was having a lot of users 
 doing browse on the internet and downloading, so in that case and if the IAX 
 used TCP port, so the voice will be better than using UDP (because in TCP the 
 lost packets will be resend while in TCP it will not which will cause the 
 voice to be cutting)?

The re-sending would introduce massive latency and jitter. That's why UDP is 
used. In real-time voice, by the time the packet is 'missed' it's too late to 
retransmit it.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)

[Steve Howes]

On 30 Nov 2010, at 09:47, Michael Nausch wrote:
 I tried to configure Asterisk 1.8 on one of my test-hosts.
 
 I've installed from centos-asterisk.repo 
 (http://packages.asterisk.org/centos/$releasever/tested/$basearch/):
 snip
 [Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No channel type 
 registered for 'mISDN'
 [Nov 30 10:35:53] WARNING[7281]: app_dial.c:2030 dial_exec_full: Unable to 
 create channel of type 'mISDN' (cause 66 - Channel not implemented)
 
 Is there no misdn-support activeted in the latest version, cause if I use the 
 help-command on Asterisk's command-line-interface, I can't see misdn?!

You installed the module, but did you load it in modules.conf? If you have, 
unload from CLI and re-load it and see if you get any errors.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stability..

[Steve Howes]

On 28 Nov 2010, at 22:26, dotnetdub wrote:
 
 It could be an extension name Where is the error trapping if this is the 
 case.. Who writes this shit?
  

A dedicated bunch of volunteers who don't appreciate you being a dick about 
bugs, which you report without so much as a log entry or a core dump.

At least update to the latest version.

HTH.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit

[Steve Howes]

On 18 Nov 2010, at 10:36, Phuong Hoang wrote:
 I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but 
 not successful, Can anyone help me to do it?

How is this different to the other two posts? Please stop repeatedly sending 
messages! If nobody replies you're probably not giving enough information!

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit

[Steve Howes]

On 18 Nov 2010, at 10:33, Phuong Hoang wrote:
 I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but 
 not successful, Can anyone help me to do it?

Given that you haven't given any error messages, any logs, or your sip.conf, or 
the manner in which it is not working No?

Going to assume by the fact you said 'registered' rather than 'trying to 
register' - it's registered and you can't make a call? if so it could be 
something codec related I suppose. Please post your asterisk log, and  SIP 
traces of when the problem occurs.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] install

[Steve Howes]

On 7 Nov 2010, at 20:59, Thomas Perron wrote:
 I have installed Asterisk before w/ no issues but while trying today
 (1.6.2.13 and centors 5.4) I receive the following at the CLI:
 
 The configure script must be executed before running 'make'.
    Please run ./configure.
 
 Any tricks on getting through this?

Type ./configure

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MixMonitor

[Steve Howes]

On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote:
 Have you noticed a marked increase in CPU load when using MixMonitor?

Since when? 1.6.2.9-1? 1.6.2.8? 1.0?

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] What is digium doing on port 113?

[Steve Howes]

On 31 Oct 2010, at 01:29, Joel Maslak wrote:

 Probably doing an ident lookup when you send mail to the list.  Standard 
 sendmail behavior. 

Agreed. Nothing to worry about.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Auto provisioning from public server

[Steve Howes]

On 26 Oct 2010, at 16:31, Jonas Kellens wrote:
 has anyone experience with auto provisioning IP-phones on different locations 
 through a central public provisioning server ? You use http or https ?

What handset? That's rather what controls your options. Some support HTTPS with 
client certificate authentication. Some support passwords. Some don't.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dial Plan Conf

[Steve Howes]

On 21 Oct 2010, at 10:16, Jigar Joshi wrote:
 I have attached the dial plan file.

In what format?

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician

[Steve Howes]

On 21 Oct 2010, at 15:56, JR Richardson wrote:
 These are full time positions in Dallas, no telecommuters please.

A very vast majority of people on here are not in Dallas (and indeed probably a 
majority in the US). So stop filling their mailboxes with this crap.

Incase you hadn't noticed Asterisk Users Mailing List - Non-Commercial 
Discussion

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP Blacklisting

[Steve Howes]

Hi,

Given the recent increase in SIP brute force attacks, I've had a little idea.

The standard scripts that block after X attempts work well to prevent you 
actually being compromised, but once you've been 'found' then the attempts seem 
to keep coming for quite some time. Older versions of sipvicious don't appear 
to stop once you start sending un-reachables (or straight drops). Now this 
isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth 
costs - and for people running on lower bandwidth connections. The tool to 
crash sipvicious can help this, but very few attackers seem to obey it..

The only way I can see to alleviate this, is to blacklist hows *before* they 
attack. This means you wont ever be targeted past an initial scan.

Is there any interest in a 'shared' blacklist (similar to spam blacklists, but 
obviously implemented in a way that is more usable with Asterisk/iptables)?. 
Clearly it raises issues about false positives etc, but requiring reports from 
more than X hosts should alleviate this. There's all the usual de-listing / 
false-listing worries as with any blacklist, but the SMTP world has solutions 
we could learn from.

Leaving a 'honeypot' running on a single IP address has revealed a few hundred 
addresses in less than a month. I am fairly certain these are all 'bad' as this 
host isn't used for anything else. There is obviously a wealth of data (and 
attacks) out there that would be good to share.

Anyone have any thoughts?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP Blacklisting

[Steve Howes]

On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
 I'll subscribe, that is for sure.  What is the best way to dist the 
 blacklist?  iptables include file?  Or something more integrated to 
 asterisk... just thinking off the top of my head that a module that vetted 
 inbound connections against an external list would be a very cool thing.

I was thinking some sort of script to pull via HTTP to update whatever you 
wanted (output as iptables etc). I know its not an instant 'lookup', but an 
hour delay between updates is nothing. Also means whoever is running the server 
isn't getting hammered by everyone ;) Realtime lookups from Asterisk would be 
quite a load (and would introduce latency).

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP Blacklisting

[Steve Howes]

On 21 Oct 2010, at 17:03, Zeeshan Zakaria wrote:
 But the problem is how to make sure that only legitimate users are 
 contributing to this list. Contributors to this list somehow need to verify 
 to an admin that they are not hackers, and this the hard part.

I was thinking of having a threshold of number of people reporting an address 
before it's approved (perhaps from X countries to stop someone with their own 
subnet abusing it). Clearly it's not an easy thing to guarantee, but a 'report 
false positive' with human intervention at this point might be useful.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP Blacklisting

[Steve Howes]

On 21 Oct 2010, at 17:32, Jeff LaCoursiere wrote:
 I agree in principle - some cron job pulling the list by http would 
 certainly be simple.  But just to continue my thoughts to the brick wall, 
 I don't see a lookup adding latency to the call other than what should 
 be a very brief addition to the time taken for a call to be accepted. 

Yea that's what I was referring to. Say some evil people attacked the server, 
you could add a few second delay to someone's call setup. I know it's not a 
major problem but it might just be opening another attack vector.

 Once accepted you would just continue to accept the packets.  How about 
 something DNS based?  Load could potentially be distributed that way if a 
 number of people agreed to participate.  I'll mull this over a bit more.

DNS is a possibility. It would require an Asterisk module I guess. There's 
nothing saying we could publish the same data in multiple ways (store it in SQL 
somewhere and output files to HTTP and generated zone files for bind to pick 
up).

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk router

[Steve Howes]

On 7 Oct 2010, at 23:57, steve casto wrote:
 A Crisco RVS4000 installed now has real problems with Sip, one-way audio and 
 throughput not up to the WAN speed.

ALG? (Assuming you mean Cisco..)
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

[Steve Howes]

On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:

 Hi All,
 Please refresh my memory. I am trying to install asterisk after 2 years. I 
 hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 
 1.8.0-rc2 on centos 5.5 but getting the following errors.
 snip
 Plz help.

You need mysql-devel. You might also find that most things are case sensitive, 
maybe your malfunctioning caps-lock is causing problems? ;)

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?

[Steve Howes]

On 5 Oct 2010, at 15:13, Gordon Henderson wrote:
 $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk
 $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk
 
 I did that before I moved to LXC, but you can't use the standard port 5060 
 for all instances, only one - might be OK in testing, but you can't 
 realistically expect punters to change the port their equipment uses...

More than one IP on the box. Change the bind address..

Easy, no?

Steve

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk/Realtime and MySQL

[Steve Howes]

On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
 That's work and in my extension.conf, i have:
 [as5300-incoming]
 switch = Realtime
 
 and in extconfig.conf
 extensions = mysql,general,VOIP_Extensions
 A lot of Extension are into the table VOIP_Extensions.
 
 I am search to know if i can add a :
 [beta-incoming]
 switch = Realtime
 
 but not use the table VOIP_Extensions but VOIP_Extensions_Beta
 
 
 Anyone know if it's possible ? (use two table for extension)

Dont think you can use two tables.. But you're using two contexts there right? 
Just have your 'beta' stuff in the same table, but different context.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] should trixbox system hang when ISP dropsconnection?

[Steve Howes]

On 24 Sep 2010, at 16:09, Danny Nicholas wrote:
 The BOBW solution I would suggest is that you run your
 Trixbox/Asterisk using a local DCHP provider/server so you aren't as
 vulnerable to how efficient your ISP is at staying up.

DNS. Not DHCP.

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digest Username/auth name mismatch ‏

[Steve Howes]

On 23 Sep 2010, at 17:29, t. k wrote:
 Isn't there any way to configure the username in the hardphone to be 
 just ?
 Yes.there is no way to cofigure as  in the hardphone.It will cost and 
 spend time a lot to implement

Then I think the short answer is that it's not compatible.

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] http://www.asterisk.org/downloads naming schema

[Steve Howes]

On 22 Sep 2010, at 16:45, Klaus Darilion wrote:
 Since some time the download of the newest Asterisk does not contains 
 the version number anymore, but is just called asterisk-1.4-current.tar.gz
 
 This gives me a tarball where I do not know the version without looking 
 into the tarball.
 
 Thus, IMO it would be very useful to switch back to old schema war the 
 download contained the version number.

http://downloads.asterisk.org/pub/telephony/asterisk/
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Realtime semi-colon

[Steve Howes]

On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
 Does anyone know how to send * a semi-colon from a realtime database.  I
 know that * uses the semi-colon as a 'seperator' - but I need to be able
 to use one in a command.  I know I can use \; in the non-realtime
 configs, but this doesn't work in realtime.

in /etc/asterisk/extensions.conf

[globals]
SEMICOLON=\;

Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at the 
moment..

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Steve Howes]

On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
 I have indeed found the core file in /tmp (that is where 'locate' does 
 not look huh...)

'updatedb'?

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip show channels

[Steve Howes]

On 14 Sep 2010, at 17:32, Dan Journo wrote:
 I'm trying to view a list of the active calls to see if I can restart 
 Asterisk.

Don't?. 'core restart when convenient' will wait until there are no calls.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

[Steve Howes]

On 14 Sep 2010, at 19:27, Jonas Kellens wrote:
 And again !! Without me doing anything !!

Yea, you didn't even enable any kind of debugging or anything. Amazing..

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)

[Steve Howes]

On 3 Sep 2010, at 10:07, Roger Burton West wrote:
 Also: I've heard good things about the PAP2T for getting analogue
 handsets to talk to a VoIP server. But all the ones I can see on eBay
 are PAP2T-NA models. Will these work with British handsets? (Obviously
 with a plug adaptor to put the BT jack into an RJ11 socket, but that's
 relatively easy to arrange.)

PAP2 was discontinued a long time ago. Use a 2102. There are fake PAP2's out 
there so avoid. 2102 works fine with a UK handset for me.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk routing to SoftSwitch

[Steve Howes]

On 1 Sep 2010, at 10:30, Pratik Shrestha wrote:

 Any Idea??

Read 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf

But I'm guessing you knew that and are just after getting someone else to do 
the work...

Just create a catch-all pattern to match anything your specific dialplan 
doesn't (what it does match is completely unknown as you have yet to 
satisfactorily describe any of your system to us). I'd guess as _X. but i have 
no idea of your setup...

Then to add the prefix just make it dial SIP/softswitchpeer/4327${EXTEN}

EXTEN will be filled with the existing number.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

[Steve Howes]

On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote:
 I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) 
 to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 
 simultaneous calls), g729 all the way through

Sounds fine to me. Reckon you could do that on a toaster ;)

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

[Steve Howes]

On 31 Aug 2010, at 17:58, jmilli...@sentinelcommunications.com wrote:
 On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote:
 I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) 
 to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 
 simultaneous calls), g729 all the way through
 Sounds fine to me. Reckon you could do that on a toaster ;)
 That is what I was thinking.  I have an eeebox at home that does fine with a 
 single core Atom(1 or 2 simultaneous calls) but I do not have any real world 
 testing/experiance for these so I thought I would get some expert opionions.

Please don't reply to me directly, reply to the list address (contained within 
the Reply-To field).

Thanks.

S


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

[Steve Howes]

On 31 Aug 2010, at 18:10, Andrew Latham wrote:
 Sounds fine to me. Reckon you could do that on a toaster ;)
 Thanks, I needed to clean this keyboard anyway

Hehe. It's true though. I was amazed what our atom boards would do. We even 
chucked transcoding/conferencing at them and they worked amazingly. I almost 
didn't believe it when I measured their power usage too...

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec choice

[Steve Howes]

On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote:
 Does anyone has an idea how to tell asterisk to use codec A for first 50 
 calls and then codec B for rest of the calls.

You could create two separate trunks, one for each codec?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] install asterisk

[Steve Howes]

On 13 Aug 2010, at 14:08, Albert Bonomo wrote:
 How come nowhere in the internet, nor in Digium.com docs, blogs, or whatever, 
 anybody mention that yum install is available ? Why nobody ever make a small
 note telling that asterisk is available from repositories to install, and 
 that is so easy ?

Fedora made the packages for their yum I'd guess. Their job to document.

For CentOS there is documentation:

http://www.asterisk.org/downloads/yum

Which is top result for 'asterisk yum install'. I figured asterisk.org would be 
a good place to look for Asterisk stuff..

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.8.0-beta1 Connectedline

[Steve Howes]

On 24 Jul 2010, at 13:21, unsero...@aol.com wrote:

 Hi,
 
 i just tried to use the CONNECTEDLINE() feature but it does not work, at 
 least with my softphones (zoiper, 3CX, Xlite)
 
 in sip.conf under general I have:
 trustrpid = yes
 sendrpid = rpid,pai
 rpid_update = yes
 
 in extensions.conf I have:
 exten = 2000,1,Set(CONNECTEDLINE(number,i)=98)
 exten = 2000,n,Set(CONNECTEDLINE(name,i)=test)
 exten = 2000,n,Set(CONNECTEDLINE(pres)=allowed)
 exten = 2000,n,Dial(SIP/2000,20)
 
 It seems to be executed correctly
 
 -- Executing [2...@default:1] Set(SIP/1000-002e, 
 CONNECTEDLINE(number,i)=98) in new stack
 -- Executing [2...@default:2] Set(SIP/1000-002e, 
 CONNECTEDLINE(name,i)=test) in new stack
 -- Executing [2...@default:3] Set(SIP/1000-002e, 
 CONNECTEDLINE(pres)=allowed) in new stack
 -- Executing [2...@default:4] Dial(SIP/1000-002e, SIP/2000,20) in 
 new stack
 -- Called 2000
 -- SIP/2000-002f is ringing
 -- SIP/2000-002f answered SIP/1000-002e
 -- Remotely bridging SIP/1000-002e and SIP/2000-002f
   == Spawn extension (default, 2000, 4) exited non-zero on 'SIP/1000-002e'
 
 
 but neither the number is changed on the calling softphone nor the name is 
 displayed.

Do a sip debug?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BLF - Realtime Asterisk

[Steve Howes]

On 16 Jul 2010, at 09:17, Danny Dias wrote:
 [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766
 handle_request_subscribe: SUBSCRIBE failure: unrecognized format:
 'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0,
 dialogver: 0, subscribecont: 'pbx9', subscribeuri: ''

Looks like the SUBSCRIBE from your phone is not a type Asterisk supports. What 
phone is it, and what type of subscribe is it set to send (and can it be set to 
send)?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MyFuel Express FO - Shortcomings

[Steve Howes]

Did you mean to send this to a mailing list?..

S

On 13 Jul 2010, at 13:33, Alphonse Ogulla wrote:

 Re-sent copying UNON and Expand Technologies. Apologies for the omission.
 
 Rgds,
 Alphonse
 
 On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote:
 Dear Esther,
 
 The foregoing mail notes sent to Expand Technology refer, in view that you 
 were not copied in the initial correspondence.
 
 We were hoping that Expand Technology would make a comeback today with a 
 course of action to resolve certain shortcomings flagged in MyFuel Express 
 but unfortunately this has not been the case. 
 
 Kindly contact Expand Technology and make it clear that we need the five 
 critical elements resolved in the next 7 days to enable us progress with the 
 system upgrade as planned.
 
 An improved version of MyFuel Express should be released speedily without 
 unnecessary mention to the ToR, more so because we are requesting minute 
 revision to code in beta stage and not significant modification or new 
 functionality in a final product.
 
 Please follow up on our behalf and revert with firmed up dates when we can 
 get a new version of MyFuel Express without the listed drawbacks.
 
 Best Regards,
 Alphonse Ogulla
 
 
 -- Forwarded message --
 From: Jacques de Gersigny j.degersi...@expand-technology.com
 Date: Mon, Jul 12, 2010 at 8:34 PM
 Subject: Re: MyFuel Express FO - Shortcomings
 To: Alphonse Ogulla aogu...@gmail.com
 Cc: Lovena Modelly l.mode...@expand-technology.com, James Gathoga 
 j.gath...@expand-technology.com, Simon Beamish simon.beam...@unon.org, 
 Sanjita Sehmi sanjita.se...@unon.org, Sheila Cardovillis 
 sheila.cardovil...@unon.org
 
 
 Hi Alphonse, Simon,
 I'm in a Business trip and I will get back on monday next.
 Rgds,
 JDG
 
 
 On 8 July 2010 16:05, Alphonse Ogulla aogu...@gmail.com wrote:
 Dear Jacques,
 We tried getting you on phone in the office at noon (Kenya time) but 
 unfortunately you had stepped out for lunch. We however managed to get hold 
 of Lovena and briefly deliberated the critical items in the ensuing email. In 
 principle, we agreed to address these drawbacks in the following manner:
 
 1) Expand Technologies to resolve items 1a (card printing) and 1c (bank card 
  cheque payment currency) without further deliberations..
 2) Refer to the final signed TOR for items 1b (card transfer) and 1d (FO 
 direct topup). I shall get the final TOR from Easther Wanjoga of Kenya Shell 
 Ltd.
 3) Lastly, Expand Technologies to check if the chip card has sufficient 
 space to store the expiry date in order to implement item 1e (card validity).
 
 I'm also made to understand that you called Simon Beamish and discussed 
 further the items listed above. Kindly look into these issues keenly and 
 revert with a proposal on headway latest by Monday 12th July AM. Please 
 remember to copy Shell in your rejoinder.
 
 Looking forward to hearing from you soon.
 
 Best Regards,
 Alphonse Ogulla
 Tel: +254 20 7621510
 Mobile: +254 723 465172
 
 
 On Mon, Jul 5, 2010 at 12:15 PM, Alphonse Ogulla aogu...@gmail.com wrote:
 Dear Jacques et alia,
 
 We have identified certain shortcomings in MyFuel Express Front Office (FO) 
 software that we need rectified as soon as possible and in-time for the go 
 live scheduled for next month. The critical elements should be given 
 uttermost priority as it is impossible to commence printing and issuing of 
 cards with the listed drawbacks still in place.
 
 1) CRITICAL ELEMENTS
 
 a) Card Printing and Personalisation
 Increase padding on the left margin so that the card holder name, description 
 and vehicle registration do not print on the UN logo.
 
 b) Card Transfer
 The required functionality should be transfer of card value on the e-purse 
 and not transfer of the card-holder particulars to another card as is 
 currently the case.
 
 c) Local Epurse Remote Top-up (Bank card  cheque payment)
 Currently only the cash top-up function has the option of selecting the 
 paying currency. A similar option is required for the bank card and cheque 
 top-up since many clients run US$ transactions on their credit/debit bank 
 cards. Similarly, US$ account holders have cheque books for US$ transactions 
 only.
 
 d) Front Office Direct Top-up
 The FO lacks direct top-up capability whereas there is a card reader/writer 
 directly connected to the FO. A work-around has been implemented by 
 connecting a hand held POS to the Ethernet network to download and effect the 
 actual top-up on the card. This two step procedure is time consuming and 
 shall drastically slow down the top-up process at the station.
 
 e) Client Management - Card Validity
 Provide entry for expiry date i.e. dd/mm/ instead of number of years 
 since client contracts expire on specific dates and not at the end of the 
 year.
 
 2) IMPROVEMENTS
 
 a) User Management - User Rights
 Group the list of functions into various roles to ease creation of a new 
 user. We acknowledge 

Re: [asterisk-users] Remote-Party-ID party=called

[Steve Howes]

On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
 I want to set the SIP-header Remote-Party-ID to display the name of the 
 calling party on my phone in stead of the number.

Am I missing something or is this waht CALLERID(name) and sendrpid is for?..

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote-Party-ID party=called

[Steve Howes]

On 12 Jul 2010, at 16:35, Jonas Kellens wrote:
 On 07/12/2010 05:01 PM, Steve Howes wrote:
 On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
 
 I want to set the SIP-header Remote-Party-ID to display the name of the 
 calling party on my phone in stead of the number.
 
 Am I missing something or is this waht CALLERID(name) and sendrpid is for?..
 
 If I'm not mistaken, sendrpid is an option in the sip.conf file that 
 only can be set on/off. Can it be dynamically set in the dialplan ?

No. It merely makes it generate it from CALLERID

 Does this 'sendrpid' display the name of the person I am calling on my 
 phone (in stead of the extension I am calling) ?

No, that wasn't what you asked for.. display the name of the calling party on 
my phone

 To be clear : when I dial 20, I want to see 'eric' on my display and not 
 '20'. The dialed number needs to be transformed to a name without me 
 having to use the phonebook-option of the IP-phone.

So you walled CALLED party displayed on your phone. There are a number of 
threads about this recently. You need asterisk trunk (soon to be 1.8). Doesn't 
exist in 1.6

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ARA : Realtime or not ?

[Steve Howes]

On 6 Jul 2010, at 10:34, Jonas Kellens wrote:
 what is the use of realtime SIP peers when you always need to reload the sip 
 configuration as if you were just putting your SIP peers in sip.conf ??

Did you enable caching by any chance?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Party ID issue

[Steve Howes]

On 2 Jul 2010, at 12:29, John Novack wrote:
 regardless, people will post either way, and wasting archive space 
 complaining about either one is pointless.

I was mainly pissed off about him directly replying to people (i.e. me) rather 
than the list. troll It was you lot that started the religious war ./troll

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Party ID issue

[Steve Howes]

On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:

 [Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function 
 CONNECTEDLINE not registered
 Same happens trying function CALLEDID.
  
 I am using Asterisk 1.6.1.20.
  
 What do i have to do to use this function or alternatively the function 
 CALLEDID() described in bug 8824?
 

Isn't CONNECTEDLINE only in trunk?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Party ID issue

[Steve Howes]

On 1 Jul 2010, at 16:25, unsero...@aol.com wrote:

 Sorry, what does this mean? Only in trunk?

If you look in the post you quoted

This feature is in Asterisk trunk and will be present in the upcoming 1.8 
release.

First sentence.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote Party ID issue

[Steve Howes]

On 1 Jul 2010, at 16:56, unsero...@aol.com wrote:
 
 Sorry, i wanted to know what is in trunk means.
 So it seems to mean is in the pipeline for the next version.

DON'T reply to people off list. And stop bloody top posting.

Steve

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Echo problem in VoIP-calls

[Steve Howes]

On 30 Jun 2010, at 13:48, Gareth Blades wrote:
 By ITSP do you mean a SIP provider?

ITSP: Internet Telephony Service Provider

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip add header

[Steve Howes]

On 28 Jun 2010, at 13:08, Jerry Geis wrote:
 It works fine with I call the SIP phone directly - however -
 when I first call the Local channel - then Dial the SIP phone
 the SIPADDHEADER doesnt seem to do anything.

Are you adding the header before or after you dial the local channel?

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Pickup a ringing Queue member

[Steve Howes]

On 28 Jun 2010, at 15:36, Jonas Kellens wrote:

 Does this mean I have a patched asterisk ? (I ask this because some
 applications require a non-patched asterisk version)
 Yes.
 What is then the unpatched version of Asterisk 1.4.30 ??

The one you have before you apply the patch?..

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Very strange registration problem

[Steve Howes]

On 24 Jun 2010, at 12:49, Jonas Kellens wrote:
 It seems as if some SIPaccounts could register and others could not. I don't 
 think a firewall distinguishes between phone brands or SIP accounts.

Alas 'stabbing in the dark' is all we can do until you actually provide some 
information for us. SIP traces, sip.conf, log extracts etc.

Your theory about the MySQL stuff is probably wrong. You'd still see the SIP 
packets coming to you would you not?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one for your filters

[Steve Howes]

On 23 Jun 2010, at 18:39, Steve Edwards wrote:

 Ouch. 82.0.0.0/8 is on my block list, available at:
 
   http://www.sedwards.com/class-a-block-list

Would advise people in the UK do not use that list... 82.0.0.0/8 would block a 
reasonable chunk of my users for starters..

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one for your filters

[Steve Howes]

On 23 Jun 2010, at 19:26, Steve Howes wrote:

 
 On 23 Jun 2010, at 18:39, Steve Edwards wrote:
 
 Ouch. 82.0.0.0/8 is on my block list, available at:
 
  http://www.sedwards.com/class-a-block-list
 
 Would advise people in the UK do not use that list... 82.0.0.0/8 would block 
 a reasonable chunk of my users for starters..

Infact, your list includes 88 subnets that are /8's. I can't find an IP address 
on any server I manage in the UK that isn't covered by it. Thats just over a 
third of the internet.. Perhaps this list is only advisable for those in the 
US/wherever you are?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Slightly OT: Cisco SPA525G and network errors

[Steve Howes]

On 17 Jun 2010, at 15:58, Mike wrote:

 I have a Cisco SPA525G  latest firmware, and very often when I attempt a 
 transfer I get a network error message when I press Dial on the transfer. I 
 never get that erroron a simple call out   Asterisk is configured for that 
 phone exactly the same as my other phones (Polycom) are (same sip.conf 
 settings, except for user and pw of course), so it likely isnt an Asterisk 
 problem.

Do you see anything in a SIP trace?

https://supportforums.cisco.com/community/netpro/small-business/voiceandconferencing/ipphones?view=discussions

Those guys are also good.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Out of Office

[Steve Howes]

Hi,

It isn't a problem with the list. And it is not 'mine'. It is a problem with 
your software. I am just one of the thousands of people it is annoying! Perhaps 
your IT staff could help fix it?

CCing the list so everyone is aware of your wonderful customer service. ;)

S

On 10 Jun 2010, at 11:27, Mary wrote:

 He is away with no cell phone or e-mail so either be helpful and tell me how 
 to change (step by step) this to take him off your list or write a progrm for 
 your list to fix this so it doesnt happen!
 
 Mary Shubert
 Accessgate.net, Inc.
 Suite 106 
 8600 Commodity Circle
 Orlando, FL 32819
 
 m...@accessgate.net
 Office Toll Free: (888) 227-9337
 Fax: (407) 352-2717 
 
 
 
 From: Steve Howes steve-li...@geekinter.net
 Sent: Thursday, June 10, 2010 4:26 AM
 To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne
 Subject: Re: [asterisk-users] Out of Office
 
 
 On 10 Jun 2010, at 06:20, d...@accessgate.net wrote:
 
  I will be out of the office starting
  Wed June 9th and returning Wed June 16th.
  Please contact Mary at m...@accessgate.net cell 407-267-1463
  or Jonathan at jsny...@accessgate.net cell 407-267-0056
  or call our main number 888-227-9337.
 
 Several thousand people DO NOT need spamming with this daily because you 
 can't configure your mail client/server to reply to a mailing list. Please 
 FIX THIS.
 
 S


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] reloading realtime sip peers

[Steve Howes]

On 8 Jun 2010, at 16:40, Jonas Kellens wrote:
 I noticed that changes to realtime sip peers are not applied until a 
 'reload'. A 'sip reload' does not make any changes to realtime sip peers.

sip prune ?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to get call duration

[Steve Howes]

On 3 Jun 2010, at 14:24, Necati Demir wrote:
 I want to ask how to get call duration.

Go on then

When you do ask the question you might want to include a few details. Are you 
trying to get call duration during a call? If so then the cli will help 'core 
show channels'. If it's after the call has happened then billsec in the CDR 
will help.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voicemail : mail attachment to multiple mail-addresses

[Steve Howes]

On 1 Jun 2010, at 16:53, Jonas Kellens wrote:
 Sounds... p

Perhaps you could contribute a patch? ;)

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Softphones on thin clients...

[Steve Howes]

On 20 May 2010, at 18:35, Carlos Chavez wrote:
   I am worried about conflicts when running 10 softphones on the same
 server since they will all try to use por 5060.

And the fact most terminal services servers/clients still don't support audio 
input.. only output..

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: NAT in SPA922

[Steve Howes]

On 6 May 2010, at 14:16, Sebastian Milioto wrote:
 Ok..So what ip phone model do NAT?

I think you'd struggle to find one. If it's a requirement you're probably doing 
something wrong...

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: NAT in SPA922

[Steve Howes]

On 5 May 2010, at 14:39, Sebastian Milioto wrote:
 However, when I connect a PC to that port, SPA922 works as bridge.
 
 Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist 
 such LAN tab for setting up parameters as port forwarding?  
 (by the way, version is 5.1.15(a). I'll appreciate links for downloading new 
 firmware)

It's a phone not a router. It doesn't do nat. You can get new firmware from 
www.cisco.com (believe free CCO login will get you the SMB stuff). The 'My 
Cisco Community' forums are also good. Has real Cisco people who appear to know 
their stuff.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Channel failover

[Steve Howes]

On 4 May 2010, at 03:44, Jack Bates wrote:
 We recently got VoIP, so when we make a call, Asterisk should first try
 to make the call with VoIP, but in case either our VoIP or our internet
 service are down, Asterisk should then try to make the call with our old
 school analog phone line

Well, first you try to dial it with the VoIP line.. Then the analogue one... So 
you just put the two dial commands on separate lines..

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

[Steve Howes]

On 30 Apr 2010, at 09:41, Vieri wrote:
 As far as having an internal fan for cooling, I don't know if that's actually 
 better... In general, these devices shouldn't need to rely on mechanical 
 cooling which tends to fail in time (sure, you can open the case and replace 
 it but that's extra maintenance).

The fan in the 8000 and the 8800 is horribly loud. Taking the screws out and 
mounting it on sticky pads helps..

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Issue with (pattern) matching extension

[Steve Howes]

On 29 Apr 2010, at 22:56, Leif Madsen wrote:
 Danny Nicholas wrote:
 Good snippet, Leif.  It's easier to read 100 threads on this forum than the
 100 pages of the infamous Asterisk Book PDF.
 Infamous? Ouch :)

He's insulting our holy book! Stone him!

;)

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dial plan question.

[Steve Howes]

On 28 Apr 2010, at 06:53, Aditya Kumar wrote:

 exten = bob,1,Dial(SIP/${exte...@ext-sip,20)


Where did you define EXTERN?

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk choking on voice messages announcements

[Steve Howes]

On 22 Apr 2010, at 00:36, bruce bruce wrote:
 Opened pseudo dahdi interface, measuring accuracy...
 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
 -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%
 
 What can one tell from these?

Thats.. Interesting...

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help with FastAGI server in Windows

[Steve Howes]

On 19 Apr 2010, at 17:00, Edwin Quijada wrote:
 Hi! I am trying to do a FastAGI server in windows. I am using the example 
 from their page but I dont get anything. Anybody here has experienced with 
 Fastagi in windows and perl that give a rigth direction to do this. I have 
 experience with AGI but fastagi dont

This is the third thread you have created for this. You're boring me now.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Realtime changes not reflected realtime

[Steve Howes]

On 17 Apr 2010, at 10:25, Jonas Kellens wrote:
 When changing the secret, the old secret is still the one to use until a sip 
 reload.
 When changing the name, the old name is still the one to use for 
 registrations until a sip reload.

So it's being cached? Does 'sip prune realtime all' clear it too?

 rtcachefriends=yes

By that?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Delay the HungUp

[Steve Howes]

On 16 Apr 2010, at 14:39, cbulist wrote:
 We need to delay the HungUp because some calls that we dial are so short 
 (3 or 4 seconds) and our provider requests 8 seconds. That is the reason

Sounds like you need a decent provider ;)

S

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is svn.asterisk.org down ?

[Steve Howes]

On 13 Apr 2010, at 15:22, Olivier wrote:
 Is it me or is svn.asterisk.org down ?

issues. too

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Steve Howes]

On 12 Apr 2010, at 17:30, Tom Stordy-Allison wrote:

 Good article - might solve our problems for now: 
 http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood
 
 He got the bots to stop by writing a ruby script that responds back to them 
 with a SIP 200 OK. 
 
 I'm going give it a go when I'm back home...

Send a 'moved temporarily' SIP message and redirect it back to them? ;)

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] cause 66 - Channel not implemented

[Steve Howes]

On 12 Apr 2010, at 20:00, David Backeberg wrote:
 chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade
 and check it out.

Terrible latency, and seems susceptible to packet loss where shotguns are 
involved.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk start with php

[Steve Howes]

On 2 Apr 2010, at 18:49, salaheddine elharit wrote:
 thank so much again for your response ,
  
 i don't understand what shoud i do if you can please give me more information 
 how to do in oreder to excute this script

He's damned near written it for you. Try researching the terms he used, and try 
doing a few of the bits. This is a community of people providing support to 
each-other, not a bunch of slaves to do your work for free.

Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

[Steve Howes]

On 25 Mar 2010, at 10:18, Asterisk wrote:
 How is it possible that the peer becames UNREACHABLE eventhough Wireshark 
 logged its proper response? 

Wireshark received it, doesn't mean Asterisk did. what does a sip debug in 
Asterisk show?

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] new server install errors starting asterisk

[Steve Howes]

On 25 Mar 2010, at 13:08, Ott Rose wrote:
 Can't find indications config file indications.conf.

Thats the last line. Probably the problem... Amazing what reading instructions 
does...

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] new server install errors starting asterisk

[Steve Howes]

On 25 Mar 2010, at 14:02, Ott Rose wrote:

 well i followed the same directions i used like 3 weeks ago with 1.6.0 and 
 didn't have any issue.  Not sure what went wrong. That why i posted it. 
 
 how can it work one time and not the next. 

Does the file exist? If not, then something is different. You probably missed 
the step to make the sample config files.

S
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users