Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Steve Edwards wrote:


On Wed, 13 Apr 2016, Jeremy Kister wrote:

is there a way i can use the asterisk cli (or some other asterisky method) 
to recreate that extensions.conf ?


Will 'dialplan save' help?


I just tried this one. It writes the dialplan, but without the application 
arguements. Worthless.


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Re: [asterisk-users] recreating extensions.conf from live dialplan ?

[Steve Edwards]

On Wed, 13 Apr 2016, Jeremy Kister wrote:

is there a way i can use the asterisk cli (or some other asterisky 
method) to recreate that extensions.conf ?


Will 'dialplan save' help?

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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

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Re: [asterisk-users] Best timing source?

[Steve Edwards]

On Tue, 5 Apr 2016, Mamadou NGOM wrote:

I am doing a configuration for connecting my server asterisk to a SIP 
provider. I ask if somebody can give me a basic code or a link to begin 
well;


0) Don't top-post.

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Re: [asterisk-users] Best timing source?

[Steve Edwards]

On Tue, 5 Apr 2016, Mamadou NGOM wrote:

I am doing a configuration for connecting my server asterisk to a SIP 
provider. I ask if somebody can give me a basic code or a link to begin 
well;


1) You should start a fresh thread rather than hijacking an existing 
unrelated thread.


2) You should show us what you have done so far.

Most SIP providers have sample snippets for your sip.conf and 
extensions.conf files.


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Re: [asterisk-users] Lost outgoing SIP packets

[Steve Edwards]

On Thu, 31 Mar 2016, Dovid Bender wrote:

Just guessing I would verify that the out of : iptables -L -nv Shows no 
dropped packets...


Doesn't tcpdump 'see' packets before iptables?

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Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

[Steve Underwood]

On 03/30/2016 08:23 PM, Vitor Mazuco wrote:

Hi!

Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or
any others digium card FXO for use Fax modem?

Thanks.

Asterisk + iaxmodem gives you a bunch of soft FAX modems. Add one of the 
analogue PSTN interface cards you listed and you have a multi-channel 
PSTN connected FAX modem. This arrangement is widely used with HylaFAX, 
although people do use it with other FAX software, such as the stuff 
built into Windows (using ethernet virtual terminals to connect the 
windows box to the linux box).


Regards,
Steve

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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
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Sun, 20 Mar 2016, Trey Hilyard wrote:


On Mar 18, 2016 8:27 PM, "Steve Edwards"  wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-lrn]
>         exten = _x.,1,                  verbose(1,[${EXTEN}@${CONTEXT}])
>         same = n,                       set(DID=${CUT(EXTEN,\;,1)})
>         same = n,                       set(LRN=${CUT(EXTEN,\;,2):3:12})
>         same = n,                       execif($["${LRN:0:1}" = 
"+"]?set(LRN=${LRN:1}))
>         same = n,                       execif($["${LRN:0:1}" = 
"1"]?set(LRN=${LRN:1}))
>         same = n,                       goto(${LRN},${DID},1)
>         same = n,                       hangup()

That's a good one. One thing it doesn't do is actually validate that the 
LRN is mine, but that shouldn't be tough to add now the the LRN is in 
its own variable. Thanks for the help!


If the LRN is not yours, you will not have a matching context so the 
goto() will run the invalid handler (the 'i' extension). You could play an 
appropriate message there.


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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Trey Hilyard wrote:


I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})


Have you tried the '_!.' pattern?

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Re: [asterisk-users] Using Asterisk to play Icecast streams

[Steve Edwards]

On Wed, 16 Mar 2016, Dovid Bender wrote:

1) I want to be able to add a few hundred streams per box. Not all 
streams are being listed to at once. Once you add a MOH class to 
musiconhold.conf it stays up forever (which I can understand why). When 
trying realtime madplay wont be loaded until it's called in the dial 
plan.  After that it is in Asterisk until I restart asterisk. If we have 
20-30 streams it's OK but once that grows it can bog down the machine.


2) I found that it for some reason the stream returns a 404 (it goes off 
line etc.) then 10-15% of one core gets locked up until the stream comes 
back online. The issue is that if a few  streams have an issue then I am 
locking up one care.


For these and other reasons, I think streaming is dumb (in most cases).

I have a client that wanted to allow his customers to enter URLs for their 
moh. Bad idea. They kept entering invalid URLs which would cause bits to 
crash. Further, we started getting complaints from URL providers as to why 
we were streaming 24x7 and from our colo about the bandwidth.


My solution* (which the customers either haven't figured out yet or don't 
mind) was:


1) Customers submit URLs to my client.

2) Once entered into the system, the customer can select the 'station' 
from a web page.


3) I record 24 hours of the stream and then use sox to break into separate 
files on the silence between songs.


4) I create a moh class that random plays the files.

*) My client approved the solution which may be of questionable legality 
depending on the stream.


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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Steve Edwards wrote:


Have you tried the '_!.' pattern?


The '_x.' pattern works fine.

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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Trey Hilyard wrote:


I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})


How about something like:

[parse-lrn]
exten = _x.,1,  verbose(1,[${EXTEN}@${CONTEXT}])
same = n,   set(DID=${CUT(EXTEN,\;,1)})
same = n,   set(LRN=${CUT(EXTEN,\;,2):3:12})
same = n,   execif($["${LRN:0:1}" = 
"+"]?set(LRN=${LRN:1}))
same = n,   execif($["${LRN:0:1}" = 
"1"]?set(LRN=${LRN:1}))
same = n,   goto(${LRN},${DID},1)
same = n,   hangup()

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-----
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Re: [asterisk-users] Need help with my dial plan - general logic flaws

[Steve Edwards]

On Sat, 12 Mar 2016, Ivan Demkovitch wrote:


I created very simple automated attendand (with a help of book), below is code.
But logic is simple:

Depending on time - I want:
If during business hours - give them menu and handle extensions
If after hours - give them message and take to voicemail.

This dial plan accomplishes just that.

What I don’t like is side behavior of this plan
When after hours - I can still press 1/2/3 and go into Sales/Support 
queue or to operator, etc.
It’s probably OK to allow dialing of extensions of actual users (101, 102, etc)
But I don’t want them to dial operator (0) or 1,2,3 and such.


A new context is a great way to limit the scope of trouble after hours 
callers can get into.


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Re: [asterisk-users] Dialplan question: Variables in GoTo() ?

[Steve Edwards]

On Thu, 10 Mar 2016, A J Stiles wrote:


Can you use variables in the target of a GoTo() statement?


Yes. Here are a few examples from one of my dialplans:

; invalid template
[i](!)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)

; look up (set) DNIS (DID) channel variables
exten = _x.,n,  goto(lookup-dnis,${EXTEN},1)

; dispatch to selected application
exten = _[123456],n,
goto(${PRODUCT-${EXTEN}-APPLICATION},s,1)

This particular dialplan uses the invalid template in around 30 contexts 
and 'goto(${CONTEXT},s,1)' about 15 times. Note that the last example 
'nests' the variable expansion -- a variable within a variable.


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Re: [asterisk-users] Asterisk Call Forwarding

[Steve Edwards]

On Fri, 4 Mar 2016, Madushan Geethanga wrote:


What is redacted means?

same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)


Censored. Ususally for political reasons. In this case, the OP didn't want 
to put a real phone number in a public list.


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Re: [asterisk-users] load test docker images?

[Steve Edwards]

On Fri, 19 Feb 2016, Jeff LaCoursiere wrote:

Has anyone created any docker images I might be able to use on EC2 for load 
testing an asterisk platform?  I started an instance this morning and was 
about to load sipp and other tools, and then thought surely someone must have 
done this already.  I'd like to hammer a platform we have created with 
multiple EC2 images until it breaks, to test capacity.


I'm surprised no one has this available.

If it's in your skill set, I could make use of it.

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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

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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

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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 <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
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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

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Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

[Steve Edwards]

On Mon, 25 Jan 2016, waqas.mehmood90 wrote:

I am working on asterisk ivr .i am facing problrm in crontab.when i run 
example it give bash 5:command not found then i check and found that no 
crontab for root user kindly guide me please


If you start your thread with a relevant subject you may get better 
responses -- better bait, better fish :)


Your question is somewhat cryptic.

Is this an Asterisk issue or a system administration (bash/crontab) issue?

If you don't have a crontab for root, how are you executing your script?

The usual issues in running scripts are:

1) file and/or directory permissions and/or ownership.

2) PATH.

3) 'she-bang' errors.

4) different environment between shell and cron.

5) different environment between your user ID and the executing user ID.

6) Incorrect line endings -- editing the script/crontab on Windows (cr/lf) 
and copying it without conversion to Unix (lf).


None of which can be identified with the information provided.

The 'bash 5:command not found' snippet implies that something is wrong on 
the 5th line of your script.


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Re: [asterisk-users] how to flush user input before READ()

[Steve Edwards]

On Tue, 19 Jan 2016 09:02:30 -0800 (PST) Steve Edwards 
 wrote:


How about a read() to a dummy variable with a 1 second timeout to 
consume the octothorpe and password?


If 1 second is too long, you could write an AGI to use the 'wait for 
digit' AGI command which allows the timeout to be specified in 
milliseconds.


On Tue, 19 Jan 2016, D'Arcy J.M. Cain wrote:

If I understand it the OP has un-consumed input and is just looking for 
the shortest possible time to read it.  Would a read with a timeout of 
zero do the job or would Asterisk optimize away the call?


Good idea. I was looking at the 'read()' dox for 1.2 which doesn't say 
much about the timeout values -- only that if it is 0, the default is 
used. The default may be changed using the TIMEOUT() function. Maybe 
setting the default to 0 and specifying 0 in read() may be fruitful.


More modern Asterisks (11.17) say the timeout can be a floating point 
number. Maybe setting it to 0.1 (or 0.01) may do the trick.


Unfortunately, I don't have the time to play right now.

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Re: [asterisk-users] how to flush user input before READ()

[Steve Edwards]

On Mon, 18 Jan 2016, Ethy H. Brito wrote:


how to flush user input before READ()?



On Mon, 18 Jan 2016 09:38:52 -0800 (PST)
Steve Edwards  wrote:


How about a read() to a dummy variable with a 1 second timeout to 
consume the octothorpe and password?


On Tue, 19 Jan 2016, Ethy H. Brito wrote:

To close the thread, the solution did "flush" the input as a side effect 
of the short timeout.


If 1 second is too long, you could write an AGI to use the 'wait for 
digit' AGI command which allows the timeout to be specified in 
milliseconds.


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Re: [asterisk-users] how to flush user input before READ()

[Steve Edwards]

On Mon, 18 Jan 2016, Ethy H. Brito wrote:


how to flush user input before READ()?


How about a read() to a dummy variable with a 1 second timeout to consume
the octothorpe and password?

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Re: [asterisk-users] Help me please i am facing much trouble

[Steve Edwards]

On Sat, 16 Jan 2016, waqas.mehmood90 wrote:

How to get user extension number in agi php scrip from which he's 
calling on ivr i am using cid and able to get his name but not his 
extension no please help me thanx in advance


You can use the 'agi set debug on' CLI command to enable AGI debugging. 
This will show you the AGI variables passed to your script.


You can use the 'dumpchan()' application to display the available channel 
variables.


There are also CLI commands to interrogate the Asterisk database.

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Re: [asterisk-users] Call Recording

[Steve Edwards]

On Sun, 10 Jan 2016, Ian Harding wrote:


Inbound route: Don't Care
Queue: Yes
Extension: Don't Care


What front end are you using?

What version of Asterisk, OS, etc?

You may get more interest on a mailing list specific to that front end.

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Re: [asterisk-users] No joy with my first AGI Python script

[Steve Edwards]

On Wed, 6 Jan 2016, D'Arcy J.M. Cain wrote:


It's very simple but it doesn't work.  Here's the entire script.

#! /usr/bin/python

import sys

env = {}

def comm(cmd):
   sys.stdout.write(cmd.strip() + '\n')
   sys.stdout.flush()
   return sys.stdin.readline().strip()

while 1:
  line = sys.stdin.readline().strip()

  if line == '': break

  key,data = line.split(':')
  if key[:4] == 'agi_':
  key = key.strip()[4:]
  data = data.strip()
  if key: env[key] = data

#comm("Verbose(0,pyast: %s)" % sys.argv)
comm('SAY NUMBER 123 ""')

sys.stderr.write("AGI Environment Dump:\n");
for key in env.keys():
  sys.stderr.write(" -- %s = %s\n" % (key, env[key]))

sys.stderr.flush()

The extension is;

exten => *22,1,Verbose(0,${CHANNEL(peername)} calling 22 TEST)
 same => n,AGI(/home/darcy/pyast,Hello world)
 same => n,Hangup

What happens when I dial it is that the dialplan Verbose statement runs
but nothing else is logged and the number is not said.  When I turn on
AGI debugging I get this:

... (a bunch of agi_ variables)
AGI Tx >> agi_arg_1: Hello world
AGI Tx >>
AGI Rx << SAY NUMBER 123 ""
AGI Tx >> 200 result=0

There is a delay between the last Rx and Tx suggesting that it thinks
that the numbers are being played but I don't hear them.  Also, the
output to stderr does not appear in the logs.

Here is my environment:
- Asterisk 11.20.0
- NetBSD 7.0
- Python 3.4


In no particular order (except #0):

0) Use an existing Python library. Nobody gets it right the first time. I 
wrote my C library 100 years ago, so I don't remember all the specifics of 
the AGI protocol.


1) Is the space after the 'she-bang' significant?

2) Your 'sys.stderr.write' may be violating the AGI protocol. Writing to 
stdout definitely does. I don't remember what writing to stderr does.


3) If you dump your AGI environment before 'say number' do you get a 
different outcome?


4) Any chance the 'digits' directory is missing or that your channel 
language is set weird? Does the saynumber() dialplan application work?


5) Can you play any audio to the channel? Does playback(demo-congrats) 
work?


(That was my last straw to grasp -- need another cup of tea.)

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Re: [asterisk-users] Manipulating of a dialed sequence

[Steve Edwards]

On Sat, 5 Dec 2015, Frank wrote:


Given, as an example, the following sequence

012345*543210

I would like to store into a variable all digits before "*" (012345) and
in a different variable all digits after the "*" (543210) for further
processing in the dial plan.


Have you tried the 'cut' function?

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Re: [asterisk-users] SIP calls dropping at 15 minutes

[Steve Edwards]

On 11/20/15 11:13 AM, Steve Edwards wrote:


I have a problem where SIP calls from some providers are dropping at 15 
minutes.


The topology is: Client sends calls to a host running OpenSIPS, 
OpenSIPS sends calls to an Asterisk server.


1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to 
route calls in OpenSIPS? It works most of the time.


2) Can (or should) I configure Asterisk to not send the INVITE at 15 
minutes?


On Sat, 21 Nov 2015, Andres wrote:

Looks like session timers are kicking in and a Re-Invite is being sent. 
I would disable them in sip.conf and try again:


session-timers=refuse

http://doxygen.asterisk.org/trunk/sip_session_timers.html


3) Should OpenSIPS be responding differently to the INVITE at 15 
minutes?


This appears to work, but it feels wrong. Shouldn't I be configuring 
Asterisk or OpenSIPS  to respond or receive the re-invite correctly?


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[asterisk-users] SIP calls dropping at 15 minutes

[Steve Edwards]

)   |
|907.868231|   | BYE   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|907.869337|   | 481 Call leg/transac  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|907.869412|   | 481 Call leg/transac  |SIP 
Status
| |   |(5060)   -->  (5061)   |
|1140.290782| INVITE SDP (g711U te  |   |SIP From: 
"760xxx"   (5060)   |   |
|1140.291032|   | INVITE SDP (g711U te  |SIP 
Request
| |   |(5060)   -->  (5061)   |
|1140.292338|   | 481 Call/Transaction  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|1140.292445| 481 Call/Transaction  |   |SIP 
Status
| |(5060)   <--  (5060)   |   |
|1140.339890| ACK   |   |   |SIP 
Request
| |(5060)   -->  (5060)   |   |
|1140.340011|   | ACK   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|1140.452758| BYE   |   |   |SIP 
Request
| |(5060)   -->  (5060)   |   |
|1140.452893|   | BYE   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|1140.453470|   | 481 Call leg/transac  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|1140.453541| 481 Call leg/transac  |   |SIP 
Status
| |(5060)   <--  (5060)   |   |

My knowledge of SIP is limited, but it appears that Asterisk is sending an 
INVITE at 907.588019, OpenSIPS responds with an INVITE at 907.590261, but 
Asterisk thinks the call doesn't exist and sends a BYE.


1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to route 
calls in OpenSIPS? It works most of the time.


2) Can (or should) I configure Asterisk to not send the INVITE at 15 
minutes?


3) Should OpenSIPS be responding differently to the INVITE at 15 minutes?

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Re: [asterisk-users] Asterisk Mobile Dialer

[Steve Edwards]

On Mon, 2 Nov 2015, Helvio Junior wrote:

We are launching a new product to help-us to reduce mobile call costs 
using Asterisk.


Commercial products belong on asterisk-biz.

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Re: [asterisk-users] Is there any API on Free PBX to connect to the extensions?

[Steve Edwards]

On Sat, 24 Oct 2015, Thyda ENG wrote:

I wonder about free pbx does it has any api to get the register 
extensions or the api to create the extensions or not ?


You may have better luck asking on a FreePBX mailing list.

I don't understand the question you keep asking.

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Re: [asterisk-users] Remote UNIX connection / disconnected.

[Steve Edwards]

On Sun, 25 Oct 2015, Bryant Zimmerman wrote:


Anyone know how to suppress the -- Remote UNIX connection / disconnected 
messages.


I have not tried it, but according to:

http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/additional_configuration_tasks-asterisk-conf-file.html

You can set 'hideconnect = yes'

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Re: [asterisk-users] Remote UNIX connection / disconnected.

[Steve Edwards]

On Sun, 25 Oct 2015, Bryant Zimmerman wrote:

Anyone know how to suppress the -- Remote UNIX connection / disconnected 
messages. I have a monitoring application that calls asterisk from the 
command line to verify some uptime stats. I would like to not have the 
console log the connections.. Any ideas are appreciated.


Use AMI instead?

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Re: [asterisk-users] asterisk core dumped after UBUNTU 14.04 dist-upgrade

[Steve Edwards]

On Tue, 20 Oct 2015, Ethy H. Brito wrote:

I had a bad experience upgrading Ubuntu a few months ago. Today I made a 
"dd" copy to another harddisk and tried to dist-upgrade.


I get "Illegal instruction (core dumped)" running "service asterisk 
debug" at random places.


Different CPU?

Different kernel version?

Was the install from packages or did you compile it yourself?

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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

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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 " 
mailto: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






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Re: [asterisk-users] pedantic=yes in sip.conf

[Steve Davies]

Alan,

A little more context would be useful. Where are you putting the '#' and
why? ( If all else fails, print it out and mail it to them ;-) )

%23 is the correct encoding for a hash '#' symbol in many SIP contexts, and
should be decoded by a properly functioning far-end.

Regards,
Steve


On Wed, 30 Sep 2015 at 19:20 sysad...@reed-media.com <
sysad...@reed-media.com> wrote:

> Hi guys
> i'm using asterisk 11.18.0.
> I need to send the pound # sign to my SIP provider, but each time it's
> reencoded in %23.
> I try to put pedantic=yes in the sip.conf as advised here:
> http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic
>
> but nothing's changed.
>
> Have someone already met this issue please ?
>
> thanks a lot,
>
>
> regards,
>
> Alan
>
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Re: [asterisk-users] Real-time payments platform - looking to invite a few users

[Steve Edwards]

On Tue, 29 Sep 2015, Shamir Allibhai wrote:


Our platform blah, blah, blah.


You already pitched on -biz.

Asterisk-users is the wrong place to pitch your business.

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Re: [asterisk-users] sox convert gsm file

[Steve Edwards]

On Wed, 23 Sep 2015, Jerry Geis wrote:

If I use the rasterisk convert file method - that works and has PCM not 
GSM but I need to use sox.


Never been a fan of using a sledge hammer when a tack hammer will do.


How does one correctly convert gsm to wav for play in the browser?


I use:

sox\
input.xxx\
--bits=16\
--channels=1\
--encoding=signed-integer\
--rate=8000\
output.wav

Followed by a trip through:

normalize\
--amplitude=-22dB\
output.wav

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Re: [asterisk-users] Fail2ban

[Steve Edwards]

On Mon, 14 Sep 2015, Gokan Atmaca wrote:

Another problem is too late to do the ban. The reason for this yetmemse 
of CPU power. I'm simulating an attack. Of course, eating CPU. One 
reason, now forbids. Abstracts must be strong if we are eating our 
resources is a serious attack.


The problem with fail2ban is it is an 'after the fact' approach. It 
depends on packets already going where they don't belong and put the 
responsibility on the application (Asterisk) to log the offending packets 
so fail2ban can scan the logs and create rules.


Years ago (2010?) Gordon Henderson published an iptables script that 
handled things like invite and registration flooding.


If you take care of these things before they eat resources and before they 
get to the logging that fail2ban depends on you will save a lot of cycles.


If Gordon is still on list, maybe he can re-publish. I'd be interested to 
see if he has any new tricks included.


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Re: [asterisk-users] phones behind nat

[Steve Edwards]

On Fri, 11 Sep 2015, Jerry Geis wrote:

My issue is if I call phone to phone in the office the phone doesnt even 
ring. The CLI shows I'm calling the correct extension like SIP/524.


The lack of sufficient connectivity to signal ringing suggests taking a 
peek with wireshark may be fruitful as well as reviewing the configuration 
of the endpoints.


Can you check the web page on the phones to confirm the IP addresses and 
netmasks are as expected? (What you think your configuration does is less 
important than what the phone thinks it does.)


I recently broke a lot of things between my office and my home when I 
decided to split 192.168.0.0 with a 255.255.255.128 net mask.


I also recently broke a working configuration by running Asterisk and 
OpenSIPS on 5060. The phones would ring but could not answer. Lost a lot 
of time until I started confirming the really basic stuff and entered 
'sudo netstat -a -n -p | grep 5060'


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Re: [asterisk-users] Looking for Asterisk Consultants & Experts

[Steve Edwards]

On Wed, 2 Sep 2015, Shahid H wrote:

Can someone recommend me where is best place to find Asterisk 
Expert/Consultant for freelance work?


Please repost to the asterisk-biz list.

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[asterisk-users] AMI 'meetme list concise' hanging

[Steve Edwards]

I have a problem with AMI 'meetme list concise' hanging. I'm running 
Asterisk 11.15.1, and PHPAGI 2.20.


The AMI stuff is in the file phpagi-asmanager.php, which says it is v 1.10 
2005/05/25.


Here's the relevant snippet of my PHP code:

// get list of conferences
if  ($debug)
{
echo 'getting list of conferences' . PHP_EOL;
}
$response = $ami->command("meetme list concise");
if  ($debug)
{
echo 'got list of conferences' . PHP_EOL;
}

I never get to 'got.'

1) Is this the current / best version of the PHP AMI interface?

2) Is this a 'known issue' with either Asterisk 11.15.1 or PHPAGI 2.20?

3) Can I set some sort of timer to abort the AMI command if it takes 
longer that a second or two?


4) Is there a better 'work-around?'

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Re: [asterisk-users] Anyone doing speech to text?

[Steve Edwards]

On Fri, 28 Aug 2015, Philippe Sultan wrote:


Lefteris,


Thanks a lot for your detailed answer and for the valuable work you've 

been doing on this topic for quite some time now.

+1

googletts-cli.pl is now my 'go to' for creating prompts for prototyping 
systems.


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Re: [asterisk-users] Anyone doing speech to text?

[Steve Edwards]

On Thu, 27 Aug 2015, Tiago Geada wrote:

I had been using google tts, but it started requiring a captcha for my 
browser, and via linux I can't 
access http://translate.google.com/translate_tts?q=test (redirects to 
captcha)


I'm confused. Your subject says 'speech to text' but the URL you reference 
does 'text to speech.'


'speech to text' is where your caller speaks and Asterisk gets the text.

'text to speech' is where Asterisk (your dialplan) has a string of text 
and your caller hears the spoken words.


What are you trying to do?

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Re: [asterisk-users] asterisk server stress test

[Steve Edwards]

Please don't top post.

On Wed, 19 Aug 2015, James Cass wrote:


Steve, would you be willing to share that "quick bash script"?


There's no magic in the script, but here it is, embarrassing myself:

cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10

Here's what's wrong with this snippet:

1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol 
may have been involved.


2) I hate single character variable names. I love alcohol.

3) cp is ill advised. For a testing script, it was easy. For a production 
application, use mv.


In use, I would execute it specifying how many call files to create, like 
50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to 
get to your goal.


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Re: [asterisk-users] asterisk server stress test

[Steve Edwards]

On Wed, 19 Aug 2015, Dominique Haeber wrote:


Hi Barry Flanagan,

Barry Flanagan  schrieb am Mit, 19. Aug 11:06:

SIPP is probably what you seek. http://sipp.sourceforge.net/

Hope this helps.


That looks pretty like what I'm looking for! Many thanks!


Another approach is to use another Asterisk system.

Recently, a customer wanted to confirm his platform would support 500 
simultaneous calls.


I wrote a quick bash script to dump 500 call files (at a leisurely pace) 
into another host that originated calls to the target host.


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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

[Steve Edwards]

On Tue, 11 Aug 2015, Stefan Viljoen wrote:


I suspected something like that, though repeatedly running

lsof | wc -l

Always stays quite low - 100 000 open files, which is still 8 times less
than the system maximum as confirmed by running ulimit -n


What the 'h' are you doing that takes x00,000 open files?

I'm running Asterisk 11.17.0 on CentOS 6.7 and my 'numbers' seem 
insignificant by comparison.


sudo /usr/sbin/asterisk -r -x 'core show channels' | grep active
347 active channels
344 active calls

sudo lsof | wc -l
3945

sudo lsof | grep asterisk | wc -l
2161

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Re: [asterisk-users] AgentRequest() and which agent id?

[Steve Edwards]

Please don't top post.

On Fri, 7 Aug 2015, Shahid H wrote:

Is it possible to build a list of agent-id in MySQL Database rather than 
agent.conf?


[snip]


I have two options.

- Build hundreds static agent-id in agents.conf 
- Dynamic agent-id in mysql table (Not associated with agent.conf).


Or, how about (periodically or on demand) creating agents.conf from the 
MySQL table?


Sometimes it makes more sense (from a resource perspective) to use a 
static file even if you can read directly from the database.


I have a system that uses OpenSIPS to route incoming calls between 
Asterisk servers based on the DNIS. The routing changes less that once a 
month.


It makes more sense to create OpenSIPS routing configuration as a static 
file than to hit the database 10,000 times a day for the same information.


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Re: [asterisk-users] Asterisk uses "Anonymous", but why?

[Steve Edwards]

On Thu, 6 Aug 2015, Steve Edwards wrote:


On Thu, 6 Aug 2015, Steve Edwards wrote:


Would comparing an INVITE from X-Lite or X-Pro with the INVITE from
Asterisk yield any clues?


On Thu, 6 Aug 2015, Murthy Gandikota wrote:


For Asterisk INVITE please view

http://pastebin.com/v15vMax4

For X-Lite INVITE please view

http://pastebin.com/rmHZKu3N


Just a quick glance (because I'm not a SIP expert)...

(Asterisk)

Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0

(X-Lite)

Request-Line: INVITE sip:732xxx@69.59.234.67 SIP/2.0

This seems relevant to me.

Some ISPs want the country code and some don't. Also, the port difference 
(5060 vs 5061) strikes me as curious. Not a smoking gun, just curious.


On the production systems I run, I run OpenSIPS on 5060 and Asterisk on 5061 
so you may be talking to 2 different endpoints.


This also seems relevant:

(Asterisk)

Proxy-Authorization:...
Authentication URI: "sip:69.59.234.67"

(X-Lite)

Proxy-Authorization:...
Authentication URI: "sip:732xxx@69.59.234.67"

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Re: [asterisk-users] Asterisk uses "Anonymous", but why?

[Steve Edwards]

On Thu, 6 Aug 2015, Steve Edwards wrote:


Would comparing an INVITE from X-Lite or X-Pro with the INVITE from
Asterisk yield any clues?


On Thu, 6 Aug 2015, Murthy Gandikota wrote:


For Asterisk INVITE please view

http://pastebin.com/v15vMax4

For X-Lite INVITE please view

http://pastebin.com/rmHZKu3N


Just a quick glance (because I'm not a SIP expert)...

(Asterisk)

Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0

(X-Lite)

Request-Line: INVITE sip:732xxx@69.59.234.67 SIP/2.0

This seems relevant to me.

Some ISPs want the country code and some don't. Also, the port difference 
(5060 vs 5061) strikes me as curious. Not a smoking gun, just curious.


On the production systems I run, I run OpenSIPS on 5060 and Asterisk on 
5061 so you may be talking to 2 different endpoints.


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Re: [asterisk-users] Asterisk uses "Anonymous", but why?

[Steve Edwards]

On Thu, 6 Aug 2015, Murthy Gandikota wrote:

[trimming cruft nobody cares about anymore]

I use the same password for INBOUND and it works fine! Something amiss 
with Asterisk OUTBOUND  because I used the same password with X-Lite and 
X-Pro Vonage soft phones with successful calls.


Would comparing an INVITE from X-Lite or X-Pro with the INVITE from 
Asterisk yield any clues?


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Re: [asterisk-users] Asterisk uses "Anonymous", but why?

[Steve Edwards]

On Thu, 6 Aug 2015, Murthy Gandikota wrote:

[trimming cruft irrelvant to the current issue]

[Aug  6 11:20:28] NOTICE[25977][C-001a]: chan_sip.c:23147 
handle_response_invite: Failed to authenticate on INVITE to '"Vonage 
User" ;tag=as0bf485e8'        > Channel 
SIP/vonage202-0019 was never answered.    I don't understand the 
"Channel SIP/vonage202-0019 was never answered" your kind 
clarification is sought.


"Failed to authenticate on INVITE"

Sounds like something you could work out with wireshark and Vonage 
support.


My SIP needs are small, but I've always been happy with vitelity.com.

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Re: [asterisk-users] Call Center

[Steve Edwards]

On Sat, 1 Aug 2015, Murthy Gandikota wrote:

Has anyone used Asterisk for a Call Center operation? What I mean is: 
given a list of phone numbers, can Asterisk dial each number, play a 
message and accept some DTMF? I ask because I am an employee of a 
non-profit company based in San Diego, CA. I already evaluated Voicent 
and Voxeo. The former has expensive licensing terms and the latter is 
not best suited for a call center. I would appreciate your kind 
comments.


On Mon, 3 Aug 2015, Murthy Gandikota wrote:

We make only solicited calls. That means, the people we are going to 
call have signed up with our service. As for technical details, I can 
think of a while loop in .ael to dial out. If you can please point me to 
a URL, I'd be grateful.


When I hear 'call center' I think of agents and queues.

What you describe sounds more 'automated' -- no humans involved.

I don't think looping in a dialplan is the right approach, since this 
process (originating calls) is not executing in the context of a channel.


I think an external 'scheduler' either creating call files or issuing 
originate requests via AMI is the way to go.


Something like:

// read the list of numbers from a text file or database

// for each number...

// write a 'call file' in /tmp/

// move the call file to the outgoing spool directory

// sleep a bit so you don't overwhelm Asterisk or your SIP provider

// lather, rinse, repeat

The call file asks Asterisk to dial the number. Once the call is answered, 
the call continues at the context, extension, and priority specified in 
the call file.


The dialplan plays the file, asks the questions, and writes the responses 
to the database. You can pass variables (donor name, last year's donation) 
in the call file that you can access as channel variables.


Based on the 'project' description, that's how I would approach it. For 
specifics, feel free to break out your check book and contact me off-list 
:)


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Re: [asterisk-users] Why no CentOS 7 repos?

[Steve Edwards]

On 27/07/2015 1:51 PM, Steve Edwards wrote:



Any particular reason CentOS 7 repos aren't available?

I'm finding integration issues with CentOS 6's ancient versions of 
MySQL and PHP with third party applications.


On Mon, 27 Jul 2015, Ron Wheeler wrote:

You might have o upgrade MySQL and PHP outside of the Centos 
distribution. I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with 
FreePBX 2.11.


I really prefer to keep to the repos. It's a numbers thing:

) I don't want to spend the time (aka $$$) to track patches to packages.

) I don't want to be 'different.' I want to run the same versions as 
others so I don't get to discover and resolve incompatibilities all by 
myself.


CentOS 7 was released over a year ago. Seems overdue to me.

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[asterisk-users] Why no CentOS 7 repos?

[Steve Edwards]

Any particular reason CentOS 7 repos aren't available?

I'm finding integration issues with CentOS 6's ancient versions of MySQL 
and PHP with third party applications.


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Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

[Steve Edwards]

On Fri, 3 Jul 2015, Jerry Geis wrote:

Ok digging deaper... I was always trying to run the session as su myuser 
-c "asterisk -fn" 


This does not seem to work.

If I login as myuser and run "asterisk fn" it worked... I got a lot of 
crackly noise that I normally dont have but it worked.


Any thoughts on why I cannot run the command as 'su myuser -c "asterisk 
-fn"' ?


Maybe there is a difference in the environment -- like a search [XXX]PATH 
or resource limit?


Does comparing the output from:

su myuser -c 'set | sort --unique >set-su'
sudo --user=myuser sh -c 'set | sort --unique >set-sudo'
set | sort --unique >set-shell # logged in as myuser

su myuser -c 'ulimit -a | sort >ulimit-su'
sudo --user=myuser sh -c 'ulimit -a | sort >ulimit-sudo'
ulimit -a | sort >ulimit-shell # logged in as myuser

yield any clues?

Maybe the 'crackly noise' is because myuser cannot access resources the 
same way root can -- like an elevated priority or 'real time' or ???


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Re: [asterisk-users] Asterisk dialplan best practices syntax

[Steve Edwards]

On Sun, 28 Jun 2015, Ludovic Gasc wrote:

It's interesting because from my point of view, I prefer to use '=>', 
because, to me, '=' is for config files. The dialplan is a programming 
language, not a config file.


The only language I'm familiar with (I'm primarily a 'c' guy) that uses 
'=>' is PHP. In PHP, '=>' is used with associative arrays.


I tend to think of a dialplan as a group of one dimensional linear arrays 
with the name of the array being the context concatenated with the exten 
so for me, a simple assignment makes sense.


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Re: [asterisk-users] Asterisk 13 logging to two places

[Steve Edwards]

Please don't top-post.

On Fri, 26 Jun 2015, Dale Noll wrote:


I turned on the messages that he had in the file again, all the logs were in 
/var/log/asterisk and it does not show anything for syslog.
asterisk -rx 'logger show channels'
Channel                             Type     Status    Configuration
---                                  --    -
/var/log/asterisk/full              File     Enabled    - DEBUG NOTICE WARNING 
ERROR VERBOSE DTMF FAX 
/var/log/asterisk/messages          File     Enabled    - NOTICE WARNING ERROR 

I wonder if there is a weird parsing error in the logger.conf file that is 
causing it to verbose log, but I will need to do more testing.
It may also be related to the verbose=3 in the asterisk.conf file.
I want to try to replicate it on a test system.


Weird. I was expecting something like:

Channel Type StatusConfiguration
---  ---
syslog.local0   Syslog   Enabled- WARNING ERROR VERBOSE

This may be a bit off the wall, but any chance the script that starts 
Asterisk is piping to logger?


Does 'pidof logger' show anything?

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Re: [asterisk-users] Asterisk 13 logging to two places

[Steve Edwards]

On Fri, 26 Jun 2015, Dale Noll wrote:


I added a filter to the /etc/rsyslog.conf file

:syslogtag, contains, "asterisk" stop

Syslog is still receiving the messages, but is discarding them.


Nice to learn a new (to me) feature of rsyslog.

What does 'logger show channels' show?

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Re: [asterisk-users] Asterisk dialplan best practices syntax

[Steve Edwards]

On Fri, 26 Jun 2015, Ludovic Gasc wrote:

1. What's the "official" notation of each line: "=>" or "=" ? In the 
wiki of Asterisk, I see very often "=>", however, what's the reason for 
both syntaxes authorized ? Historical ?


I'm not 'official,' but I have a strong preference for just '=.' Using 
'=>' seems clunky, ugly, and unnecessary.


2. To write info in logs/console, you have two commands: NoOp and 
Verbose. Verbose seems to be better, because you can define a log level. 
Are you agree or another command fits better for logs ?


This is kind of a pet peeve of mine. Why use the misguided 'side effect' 
of an application when there is a specific, 'more featured,' application 
for that purpose? A 'noop' is a contraction of 'no operation' meaning it 
should do nothing but act as a placeholder.


Two other areas of 'best practices' I'm a strong believer in are: 
alphabetize wherever possible, and use white space to improve readability.


For example, here's a 'sanitized' sip.conf snippet from a popular 
provider's web site:


[xxx-inbound]
type=friend
dtmfmode=auto
host=xxx.yyy.zzz
context=inbound

username=xxx
secret=yyy

allow=all
insecure=port,invite
canreinvite=no

[xxx-outbound]
type=friend
dtmfmode=auto
host=xxx.yyy.zzz
username=xxx
fromuser=xxx
secret=xxx
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no

Pretty ugly and difficult to read. With a little whitespace and 
alphabetizing we get:


[xxx-inbound]
allow   = all
canreinvite = no
context = inbound
dtmfmode= auto
host= xxx.yyy.zzz
insecure= port,invite
secret  = yyy
type= friend
username= xxx

[xxx-outbound]
allow   = all
canreinvite = no
dtmfmode= auto
fromuser= xxx
host= xxx.yyy.zzz
secret  = xxx
sendrpid= yes
trustrpid   = yes
type= friend
username= xxx

Now, the major sections are easy to 'visually delineated.' Finding the 
'secret' is much easier now. Comparing a 'working' extension with a 
'broken' extension will be much easier as well.


I use the same formatting in the dialplan. This snippet is from 
extensions.conf.sample:


[outbound-freenum2]
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

With a little whitespace, this becomes much more readable:

[outbound-freenum2]
exten = _X!,1,  Verbose(2,Performing ISN lookup for 
${EXTEN})
same = n,   Set(SUFFIX=${CUT(EXTEN,*,2-)})
same = n,   GotoIf($["${FILTER(0-9,${SUFFIX})}" != 
"${SUFFIX}"]?fn-CONGESTION,1)
same = n,   Set(TIMEOUT(absolute)=10800)
same = n,   
Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,   GotoIf($["${isnresult}" != ""]?from)
same = n,   Set(DIALSTATUS=CONGESTION)
same = n,   Goto(fn-CONGESTION,1)
same = n(from), Set(__SIPFROMUSER=${CALLERID(num)})
same = n,   GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = 
""]?dial)
same = n,   
Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same = n(dial), Dial(SIP/${isnresult},40)
same = n,   Goto(fn-${DIALSTATUS},1)

exten = fn-BUSY,1,  Busy()

exten = _f[n]-.,1,  NoOp(ISN: ${DIA

Re: [asterisk-users] setting outbound caller ID

[Steve Edwards]

On Thu, 18 Jun 2015, Matt Riddell wrote:


Did you buy the number from your carrier?


I prefer using 'rent' instead of 'buy' :)

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Re: [asterisk-users] howto copy a voicemail message to another machine ?

[Steve Edwards]

On Tue, 16 Jun 2015, sean darcy wrote:

There's no problem setting up vm on *. I can't use email off the instance, 
since the assigned ip address doesn't have a PTR.


It looks too much like spam. The mail relays drop it.


Would configuring your own (or your ISP's) smarthost help?

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Re: [asterisk-users] Variable variables

[Steve Edwards]

On Tue, 16 Jun 2015, Ishfaq Malik wrote:


Can asterisk handle asterisk variable variables?

For example:

If I were to set

FOO300=BAR111

and I had something in a dialplan like:

_3XX,1,NoOp(${FOO${EXTEN}})

And the user had entered 300, it would output BAR111


Yes.

1) Rather than relying on the 'side effect' of the noop() application, you 
should use the application specifically designed for the purpose of 
outputting to the console, verbose().


2) You could have had your answer in 2 minutes instead of 2 hours if you 
had just tried it yourself.


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Re: [asterisk-users] small homebrew pbx

[Steve Edwards]

On Mon, 15 Jun 2015, Steve Edwards wrote:

Although, if you lose power, you've probably lost your Internet 
connection as well so you could only make calls between extensions.


And you would lose the Italian equivalent of 911. In the US, everybody 
over the age of 6 has a cell phone stapled to the side of their head, so 
it is kind of a 'non-issue' :)


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Re: [asterisk-users] small homebrew pbx

[Steve Edwards]

On Mon, 15 Jun 2015, lu...@sulweb.org wrote:

I'm new here and I'm interested in building a small PBX with asterisk at 
home. I have one single PSTN line and ethernet cabling in place. I 
already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM 
and RAID 10 SATA disks). I make and receive 10 calls a day on average. I 
want 4 IP phones connected to the ethernet network. When there is a 
incoming call, all phones must ring and the first that takes the call 
makes the others stop ringing, but lets them available for internal 
calls.


Given the requirements above, what's a cheap but working PCIe card / USB 
adapter I could buy for this kind of PBX? Do I need things like echo 
cancellation? Do I need FXS ports?


I don't know this 'translates' to Italy, but this is what I would advise 
somebody in the US to consider, assuming you have a reliable Internet 
connection.


0) I hope you mean you want to run Asterisk at home instead of 'Asterisk 
at Home.' A@H was an ancient distribution from around 2005.


1) Rent a DID (a 'PSTN number') from a reputable SIP provider. This 
eliminates the need for a PCI/USB interface and you won't disrupt your 
'business' while you figure out how to configure and test your Asterisk 
server.


In the US, you can rent a DID for about $1.50 per month and about a $0.01 
per minute of 'talk time.' For 10 calls per day, this should beat the hell 
out of a 'landline' monthly standing fee.


In the US, it costs less than $20.00 to 'port' your existing number if you 
are really in love with it.


2) Ditch the 'room warmer' and find something really small and cheap to 
run. I live in San Diego and we pay $0.32 per kWh. I'd guess running your 
rig would cost me $50.00 to $100.00 per month just in electricity -- and 
probably that much again in the summer for additional Air Conditioning.


Take a look at Soekris net4801. It's pretty old (but very reliable) and 
it's CPU will limit you on what OS you can run, but it will give you an 
idea of how small (and cheap to power) an 'Asterisk server' capable of 
handling a couple of simultaneous calls can be.


For a more modern server, look for something small and cheap based on 
something like an Atom processor. Maybe a used laptop. If the battery is 
still good, you've solved your UPS problem as well. Although, if you lose 
power, you've probably lost your Internet connection as well so you could 
only make calls between extensions.


3) For the IP phones, check out ebay.com. Last year, I picked up 3 Polycom 
SP 501's for $20.00 each. A little dated, but a great phone.


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Re: [asterisk-users] Calling multiple phones at ones

[Steve Edwards]

On Sun, 14 Jun 2015, Ivan Demkovitch wrote:

I want all those 3 phones to be “one”. So, if someone calls our company 
number and dials my extension - I’d like 3 phones to ring at the same 
time.


Check out the 'followme()' application.

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Re: [asterisk-users] Am I cracked?

[Steve Edwards]

On Mon, 8 Jun 2015, Michelle Dupuis wrote:

You're definitely under attack (based on the 0123456 ID) so be sure to 
take preventative steps to avoid a $50k phone bill..


Don't enable 'auto-replenish' in your provider account and don't keep a 
balance you can't afford to lose.


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Re: [asterisk-users] Am I cracked?

[Steve Edwards]

On Mon, 8 Jun 2015, Kevin Larsen wrote:


Better to fail and fix than to permit and pay for it later.


That would make a great T-shirt:

Deny and Fix
 vs
   Permit and Pay

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Re: [asterisk-users] Am I cracked?

[Steve Edwards]

On Mon, 8 Jun 2015, Luca Bertoncello wrote:

This is not really possible, since I'll login on my Asterisk from many 
Providers...


many < all

So make a list of the 100 or so providers you have active accounts with. 
It's still way less than 'all.'


Also, I'm willing to bet you won't be using providers from China, North 
Korea, Russia, Iraq, etc, etc, etc. (Sorry if that steps on anybody's 
toes.)


Look for address blocks (class A, B, C) that are allocated to geographic 
regions you do not have any providers. If you limit your 'attack surface' 
you make your security problem manageable.


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-----
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Re: [asterisk-users] Connecting two Asterisk

[Steve Edwards]

On Sun, 7 Jun 2015, Luca Bertoncello wrote:


Now the problem: on my phones at "wrt" I can hear what the mobile phone at
"lucabert" sends (with a very good audio-quality), but on this mobile phone
I cannot hear a single word spoken with the phone at "wrt", not even the music
on hold I configured...



   -- Call accepted by X.Y.Z.K (format gsm)
   -- Format for call is gsm


I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw?

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Re: [asterisk-users] Connecting two Asterisk

[Steve Edwards]

On Sun, 7 Jun 2015, Luca Bertoncello wrote:


I think, my UMTS-Provider doesn't want to connect to dynamic IP or my 
DSL-Provider
does not want it, too, since I have no problem to connect and get a very good
audio quality if I connect to other SIP-Provider or to an Asterisk (SAME
configuration!!) installed on my Server...


Maybe fiddling with the SIP and RTP ports would help.

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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello 
wrote:

> Zitat von Steve Totaro :
>
>  Are you using the wifi on on the cellphone?  The peer IP is showing as
>> 192.168.200.3 which is not a routable address.  Unless things have
>> changed,
>> double NAT configurations do not work.
>>
>
> Hi Steve,
>
> My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct
> in Internet.
> But maybe my Provider does a NAT, too...
>
> Very strange is, that I have a very poorly audio-quality, if I use my
> cellphone in my WLAN and connect to my Asterisk.
> With THE SAME USER, but from a PC in the same Network, the audio quality
> is perfect.
>
> Any idea?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
>
>
>
Not without seeing some SIP debug output.

Thanks,
Steve T
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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro  wrote:

>
>
> On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello 
> wrote:
>
>> Ashwin Surendran  schrieb:
>>
>> > What settings have you got for directmedia?
>> >
>> > Could you try
>> >
>> > nat=force_rport,comedia
>> > directmedia=no
>>
>> Tried. Peer always unreachable, call not possible... :(
>>
>> Other idea?
>>
>> Thanks
>> Luca Bertoncello
>> (lucab...@lucabert.de)
>>
>>
>
> Are you using the wifi on on the cellphone?  The peer IP is showing as
> 192.168.200.3 which is not a routable address.  Unless things have changed,
> double NAT configurations do not work.
>
> Thanks,
> Steve T
>

You could try using your carrier's internet access instead of wifi.

OpenVPN for Android looks like it could work to eliminate your NAT issues
as well.

Thanks,
Steve T
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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello 
wrote:

> Ashwin Surendran  schrieb:
>
> > What settings have you got for directmedia?
> >
> > Could you try
> >
> > nat=force_rport,comedia
> > directmedia=no
>
> Tried. Peer always unreachable, call not possible... :(
>
> Other idea?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
>

Are you using the wifi on on the cellphone?  The peer IP is showing as
192.168.200.3 which is not a routable address.  Unless things have changed,
double NAT configurations do not work.

Thanks,
Steve T
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Re: [asterisk-users] sedwa...@sedwards.com causes me to be knocked off the list

[Steve Edwards]

"sedwa...@sedwards.com causes me to be knocked off the list"

I feel so powerful :)

On Wed, 3 Jun 2015, Giles Coochey wrote:


Someone on this list uses the address @sedwards.com


That would be me. Although, I prefer to use asterisk@sedwards.com for 
'list related' emails.


I doubt this is their actual email address as there is no MX record for 
sedwards.com and I can't find registration for their domain either.


It is, there is, and please look harder.

Part of my mail servers reject these emails because they cannot be replied 
to, or are likely to be spam.


Every so often I get a mail from the list management to say that I've been 
unsubscribed because of excessive bounces and it takes a single click to 
re-register.


It's a bit of a niggle for me. What do you think I should do? Change my 
servers so that I don't check sender domains?


DNS and email servers are not my area of expertise, so if you think I've 
misconfigured something on my side, please let me know.


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-----
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Re: [asterisk-users] Signaling incoming call

[Steve Edwards]

On Sun, 31 May 2015, Luca Bertoncello wrote:

Now, it would be nice, if I can signaling on the phone which number will 
be called, so that, for example, if I receive a call for +4935 I 
get a message on the display or the phone ring with a particular tone, 
and if I receive a call for +49351222 the phone write something 
other on the display or ring with another tone.


Is it possible? Maybe it depends from phone... I use a Thomson ST2022.


You can fiddle with the caller ID to change what is displayed on the 
phone.


You can fiddle with the ring tone by phone specific configuration and 
phone specific SIP headers (sipaddheader(Alert-Info: ...)).


These seem relevant:

http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the 
discussion looks relevant as well).


http://www.asteriskguru.com/tutorials/thomson_st2030.html

http://www.freepbx.org/support/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular-

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Re: [asterisk-users] Signaling incoming call

[Steve Edwards]

On Sun, 31 May 2015, Luca Bertoncello wrote:


register => 004935:MYSECRET@pbxluca/004935
register => 0049351222:MYSECRET@pbxfax/0049351222
register => 0049351333:MYSECRET@pbxanika/0049351333
register => 44:MYVERYSECRET@messagenet/44

[pbxluca]
type=peer
defaultuser=004935
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=004935
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxfax]
type=peer
defaultuser=0049351222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600


[snip]

Just a few 'stylistic' and 'ease of maintenance' suggestions:

1) Group the 'register' with the stanza.

2) Add a bit of whitespace to increase 'readability.'

3) Sort the parameters to make it easier to maintain.

4) Move 'common' settings to 'general.'

Thus, I'd write your sip.conf as:

[general]
canreinvite = no
context = luca_incoming
dtmfmode= rfc2833
insecure= invite
outboundproxy   = 172.16.34.132
port= 5060
qualify = yes
qualifyfreq = 600
usereqphone = yes

[general](+)
register= 
004935:MYSECRET@pbxluca/004935
[pbxluca]
defaultuser = 004935
fromdomain  = 172.16.34.132
fromuser= 004935
host= 172.16.34.132
secret  = MYSECRET
type= peer

[general](+)
register= 
0049351222:MYSECRET@pbxfax/0049351222
[pbxfax]
defaultuser = 0049351222
fromdomain  = 172.16.34.132
fromuser= 0049351222
host= 172.16.34.132
secret  = MYSECRET
type= peer

5) I'd try and move more of the common settings to general, but these were the 
ones listed on voip-info.org.

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Re: [asterisk-users] chanspy and mixmonitor

[Steve Edwards]

On Fri, 29 May 2015, sysad...@reed-media.com wrote:

You may get better replies if you start a new thread rather than replying 
to an unrelated thread.


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Re: [asterisk-users] Debugging dialplan

[Steve Edwards]

On Fri, 29 May 2015, Steve Edwards wrote:


; admin functions
   exten = _[456],1,   verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
   exten = _[456],n,   gotoif($["TRUE" = 
"${ADMIN}"]?meetme-star-admin-menu,${EXTEN},1)
   exten = _[456],n,   goto(enter-room,s,1)


This is an old dialplan. Now I would use 'same = n.'

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Re: [asterisk-users] Debugging dialplan

[Steve Edwards]

Please don't top post.

Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello 
:



Zitat von jg :


Yes, it is called "core set verbose 42", the other options is "core 
set debug 42".  Enjoy the show!


I know you can specify a level to the verbose application, but is anything 
in Asterisk 'hard-coded' for debug or verbose above 6? (And yes, I know 
the significance of '42' in pop culture.)


OK, thanks, but with this option I can just debug what happens if I 
call an extension right now... I'd like to have a command to ask 
Asterisk how it will handle a call...


You can use the 'dialplan' command to get a clue. For example, I have this 
context in a dialplan:


; meetme-star-menu
; 1 say private meeting number
; 3 enter private room
; 456 go to the admin menu
[meetme-star-menu](h,s)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = i,n,goto(enter-room,s,1)
exten = t,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = t,n,goto(enter-room,s,1)
; say private meeting number
exten = 1,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = 1,n,saydigits(${PRIVATE-CODE})
exten = 1,n,goto(enter-room,s,1)
; enter private room
exten = 3,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = 3,n,goto(private-lounge,s,1)
; admin functions
exten = _[456],1,   verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = _[456],n,   gotoif($["TRUE" = "${ADMIN}"] 
?meetme-star-admin-menu,${EXTEN},1)
exten = _[456],n,   goto(enter-room,s,1)

I can ask Asterisk what happens if the caller enters '5' like:

joy10:joy:08:50:18> dialplan show 5@meetme-star-menu
[ Context 'meetme-star-menu' created by 'pbx_config' ]
  '_[456]' =>   1. verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])[pbx_config]
2. gotoif($["TRUE" = "${ADMIN}"] 
?meetme-star-admin-menu,${EXTEN},1) [pbx_config]
3. goto(enter-room,s,1)   [pbx_config]

If I ask what happens if a caller enters 7, I get:

joy10:joy:08:51:42> dialplan show 7@meetme-star-menu
There is no existence of 7@meetme-star-menu extension

In which case, I could ask what Asterisk will do with an invalid 
extension:


joy10:joy:08:52:19> dialplan show i@meetme-star-menu
[ Context 'meetme-star-menu' created by 'pbx_config' ]
  'i' =>1. verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])[pbx_config]
2. goto(enter-room,s,1)   [pbx_config]

Note the format of my verbose() arguments. It makes it easy to 
'cut-n-paste' in a 'dialplan show' command.


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Re: [asterisk-users] Asterisk "virtual hosting"

[Steve Edwards]

On Sun, 17 May 2015, martin f krafft wrote:


You know the Henry Ford quote about faster horses, right? ;)


I do now :)

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Re: [asterisk-users] Asterisk "virtual hosting"

[Steve Edwards]

On Sun, 17 May 2015, martin f krafft wrote:

also sprach Steve Edwards  [2015-05-16 23:22 
+0200]:


I use a preprocessor 
(http://software.hixie.ch/utilities/unix/preprocessor/) to tailor 
dialplans and configuration files to each host based on the client (or 
project) and the hostname.


On Sun, 17 May 2015, martin f krafft wrote:

Yeah sure, templating works, but it introduces a layer of complexity 
that can make debugging hard(er).


While preprocessing could be called 'templating,' this may be confusing 
because Asterisk already as a configuration file feature called 
'templates.'



I just had the following alternative ideas.

 - when #include parses a file, prefix all stanzas found therein
   with text derived from the path, e.g.

 * #include foo/extensions.conf  →  "foo-"
 * #include bar.conf →  "bar-"
 * #include foo/bar/moo.conf →  "foo-bar-moo-"

 - if e.g. a context includes another context using a path
   separator, then the [common] context is looked up in a different
   location:

 * include foo/common→ "foo/extensions.conf"
 * include foo/bar/common→ "foo/bar/extensions.conf:foo/bar.conf"

   The same logic could be applied e.g. in the arguments of the
   Dial() application or local channels or registry instructions in
   sip.conf.

The first is probably easier to implement, while the second is clearer 
to the user.


And you find preprocessing/templating complex?


Is this something to consider?


I don't think so, primarily because it is specific to your problem. The 
audience is too small.


Let's take a closer look at preprocessing using the preprocessor I 
referenced above to make sure I understand your needs.


If we had extensions.conf.pre containing:

#filter substitution

#define PREFIX  a
#includegeneric.conf.pre

#define PREFIX  b
#includegeneric.conf.pre

# (end of extensions.conf.pre)

(Note that '#include' is seen by the preprocessor, not by Asterisk's 
configuration file parsing code.)


and generic.conf.pre containing:

[@PREFIX@-long-distance]
exten = s,1,
verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
#if PREFIX==a
same = n,   verbose(a specific code)
#endif
same = n,   hangup()

# (end of common.conf.pre)

we could process these files with a command like:

./preprocessor.pl extensions.conf.pre >/etc/asterisk/extensions.conf

which would create extensions.conf containing:

[a-long-distance]
exten = s,1,
verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
same = n,   verbose(a specific code)
same = n,   hangup()

[b-long-distance]
exten = s,1,
verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
same = n,   hangup()

This lets you write generic contexts that will be prefixed as well as 
'tailor' code specific to the value of the prefix. Isn't this what you're 
looking to accomplish?


Also note, the 'real' extensions.conf contains no additional complexity so 
it is as easy to understand and maintain as you want it to be.


--
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-
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Re: [asterisk-users] Asterisk "virtual hosting"

[Steve Edwards]

On Sat, 16 May 2015, martin f krafft wrote:


I am in the peculiar situation to have to set up a PBX for two
independent sites, but operated by the same entity. Yes, I could set
up two VPSs and install Asterisk to each, put common stuff (e.g.
conferencing setup) into Git and share between both using includes,
but for various reasons (among them simplicity and cost), I'd prefer
a single Asterisk instance.

I know I can #include files from sip.conf and extensions.conf, so
making a extensions.conf that consists of

 #include ext-common.conf
 #include foo/extensions.conf
 #include bar/extensions.conf

is trivial. Unfortunately, the contexts in each of these files must
not clash, and so I will be forced to use e.g. [bar-incoming] in
bar/extensions.conf.

That's a bit of redundancy here (which I am always trying to avoid
like the plague) and I am wondering if there are better ways. Do you
know of any, short of writing a script to "compile" the files and
change the contexts based on path (which will be dirty and hard to
get right)?


I use a preprocessor 
(http://software.hixie.ch/utilities/unix/preprocessor/) to tailor 
dialplans and configuration files to each host based on the client 
(or project) and the hostname.


This lets me do stuff like:

[globals]
DATABASE-DATABASE   = @DATABASE_DATABASE@
DATABASE-PASSWORD   = @DATABASE_PASSWORD@
DATABASE-PRODUCTION-SERVER  = @DATABASE_PRODUCTION_SERVER@
DATABASE-REPORTING-SERVER   = @DATABASE_REPORTING_SERVER@
DATABASE-ROOT-PASSWORD  = @DATABASE_ROOT_PASSWORD@
DATABASE-USERNAME   = @DATABASE_USERNAME@

and

#ifdef  A11
exten = s,n,execif($[${ACCEPT-COUNTER} < 0]?hangup)
#else
exten = s,n,execif($[${ACCEPT-COUNTER} < 0],hangup)
#endif

--
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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds

[Steve Davies]

Hi,

In my experience, all Yealink phones work just fine with Asterisk, we have
hundreds (perhaps even low-thousands) out there with customers on Asterisk
1.2, 1.6.2, 1.8 and 11.

If you are accurately representing the SIP trace on the phone and the SIP
trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
network between the two devices and that SIP ALG does not understand SIP
properly. The two halves simply do not match, so something must surely be
interfering.

In my experience it is often an innocent looking Cisco router. Cisco's SIP
implementation is "SIP By Cisco" rather than "RFC compliant SIP". If that
is the case Cisco call it a "SIP fixup" and you just need to disable it.

Hope that helps,
Steve


On Wed, 13 May 2015 at 16:59 Andrew Martin  wrote:

>
>
> - Original Message -
> > From: "Joshua Colp" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Wednesday, May 13, 2015 10:50:02 AM
> > Subject: Re: [asterisk-users] "Retransmission Timeout" results in
> dropped calls   after 32 seconds
> >
> > Andrew Martin wrote:
> > > Since some packet loss is a possibility, I assume the protocol has
> > > mechanisms
> > > for dealing with it. What should be happening differently in the
> > > communication
> > > when packet loss occurs? Should the phone just be re-sending the OK,
> > > instead of
> > > printing "<0>  | ERROR | receive a request with same cseq??" to its
> log? Or
> > > should
> > > Asterisk be starting with a new cseq on each INVITE retry?
> >
> > The 200 OK should be retransmitted until an ACK is received. It honestly
> > looks like the phone can't talk to Asterisk and it's just generally
> > screwing up signaling.
> >
>
> Thanks for the clarification and help debugging this problem. I will work
> with the phone vendor to see if they can resolve this from their end. If
> you
> have any other ideas about how to disable re-INVITEs on the asterisk side,
> beyond what I have done already, please let me know.
>
> Thanks,
>
> Andrew
>
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Re: [asterisk-users] adding area code

[Steve Edwards]

On Mon, 27 Apr 2015, Chad Wallace wrote:


On Mon, 27 Apr 2015 14:30:07 -0700 (PDT)
Steve Edwards  wrote:


On Mon, 27 Apr 2015, Bryant Zimmerman wrote:


exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})


Missing a colon?

${EXTEN:-1}



Does that work?


No. I was so focused on the tree, I missed the forest :)

${EXTEN:-x} means 'return the last x digits'

${EXTEN:x} means 'return all but the first x digits'

So, ${EXTEN:1} is correct for this use.

Something like (tested!):

exten = _9nxx,1,verbose(The 'raw' exten is ${EXTEN})
same = n,   set(MY-AREA-CODE=760)
same = n,   set(DNIS=${MY-AREA-CODE}${EXTEN:1})
same = n,   verbose(The full DNIS is ${DNIS})
same = n,   dial(sip/1${DNIS}@vitel-outbound,60,r)
same = n,   hangup()

is closer to what the OP needs. Note the 'n' in the pattern.

--
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-----
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] adding area code

[Steve Edwards]

On Mon, 27 Apr 2015, Bryant Zimmerman wrote:


exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})


Missing a colon?

${EXTEN:-1}

--
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-----
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] MixMonitor Files Always Empty

[Steve Edwards]

Please don't top post.

On Wed, 22 Apr 2015, Mark Farmer wrote:


The file is always created but is always zero size. This is the dial plan that 
records the call:

exten = 
_0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID})
exten = _0[1-8]X.,2,MixMonitor(${CALLFILENAME}.gsm,b)

The dial plan then calls a macro that makes the call.
I’ve tried adding a StopMixMonitor after calling the macro but that did not 
help and I have tried putting an Answer() at the start of the dial plan.

I am currently using the Monitor application instead which is working OK but 
I’d much rather stick with MixMonitor.


Change something and see if it yields clues.

1) Change the file location to /tmp/, without the STRFTIME subdirectory.

2) Change the codec by changing the file type.

3) Don't use a macro.

What's different between your monitor() call path and your mixmonitor() 
call path?


--
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-----
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Re: [asterisk-users] Multicast to polycom from asterisk

[Steve Edwards]

On Mon, 13 Apr 2015, Kevin Larsen wrote:

I hesitate to promote the name here since this is non-commercial 
discussion...



but Polycom...



Polycom phones...


If mentioning Polycom is OK, I think mentioning a possible commercial 
solution is OK.


--
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-
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Newline  Fax: +1-760-731-3000

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[asterisk-users] meetme vs confbridge max user comparison wanted

[Steve Edwards]

I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme 
and I'd like to switch to confbridge to service more callers.


Can anyone reply with their experience along the lines of 'using meetme I 
was only getting x callers per server but with confbridge I now get y 
callers per server?'


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-
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[asterisk-users] error retrieving a video voicemail in asterisk 11

[Steve Dolloff]

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video 
attachment while using any video phone.  This does work in my 1.8.23.1 
installation.  The file is skipped with the ast_streamfile error (and moved to 
OLD), and the prompts following that message display the ast_best_codec error.

[Apr  7 16:05:50] WARNING[17497][C-6fdd]: file.c:1017 ast_streamfile: 
Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg (format 
(ulaw|h264)): No such file or directory [Apr  7 16:05:50] 
WARNING[17497][C-6fdd]: app_voicemail.c:8609 play_message: Playback of 
message /var/spool/asterisk/voicemail/default/2036/INBOX/msg failed
[Apr  7 16:05:50] --  Playing 'vm-advopts.gsm' (language 
'en')
[Apr  7 16:05:50] WARNING[17497][C-6fdd]: channel.c:940 ast_best_codec: 
Don't know any of (h264) formats

The file does exist in h264 format

-rw-r--r-- 1 root root 298102 Apr  7 16:05 msg.h264
-rw-r--r-- 1 root root301 Apr  7 16:05 msg.txt
-rw-r--r-- 1 root root 124524 Apr  7 16:05 msg.wav

Passthrough h264 video does work.  I do have h264 and ulaw codecs on the peer 
and videosupport=yes in sip.conf.  I also tried enabling h264 in the general 
section of sip.conf and gsm in voicemail.conf with the same results.

If I disable the h264 codec for the peer, I can listen to the audio portion of 
the message:

[Apr  7 16:41:05] --  Playing 
'/var/spool/asterisk/voicemail/default/2036/Old/msg.slin' (language 'en')

Any guesses what I might be doing wrong?  Did something related change in 
asterisk 11?

-- Stephen


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Re: [asterisk-users] exten versus EXTEN

[Steve Edwards]

On Mon, 6 Apr 2015, thufir wrote:


p 176 has exten => 1NXXNXXX,1,Dial(SIP/${EXTEN}@myprovider)

how is "exten" distinct from "EXTEN"? What is this line of code doing?


This is a line of 'classic' dialplan (as opposed to AEL, Lua, or ...).

It defines a single step in the dialplan.

It 'says' if the extension matches the pattern 1NXXNXXX, dial 
myprovider using the SIP protocol and pass the value of the current 
extension.


exten is not case sensitive. exten is the same as EXTEN. If you write 
EXTEN, Asterisk will clobber it ('dialplan save') and rewrite it as exten 
so you might as well just use exten.



https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

says that EXTEN is the current extension.


You can reference the current value of EXTEN like:

same = n,verbose(The current value of EXTEN is ${EXTEN})

Another good reason to use exten so you won't confuse 'the next guy' who 
has to maintain your code.



In ruby, you this:

H = Hash["a" => 100, "b" => 200]

The => is a mapping, or at least that's my understanding.  What does it mean 
in Asterisk?  I didn't

fully appreciate that Asterisk is, apparently, its own language.


'=>' is the same as '=' when used in the dialplan. Personally, I always 
use '=' as the dialplan doesn't seem to be the place for some object 
oriented mumbo-jumbo -- at least to my 'C programmer till I die' eyes.


--
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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] [OT] switches

[Steve Edwards]

On Mon, 23 Mar 2015, thufir wrote:

so how does a client pc find the server if there's no NAT?  by IP 
address?? That makes no sense, to me, if the switch isn't assigning 
addresses.


The 'endpoint' (pc, softphone, mobile, desk set, etc.) 'finds' the 
server's IP address when:


) You configure the endpoint with the IP address or host name of the 
server. This happens either by a web page you fill out on the endpoint or 
a configuration file that is downloaded by TFTP, FTP, HTTP, etc.


) You configure SRV records in your DNS.

I think the old IAXy did some sort of discovery on port , but I don't 
remember if it was device or server discovery.


--
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-----
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Re: [asterisk-users] Asterisk 13 : SILK codec ?

[Steve Murphy]

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy  wrote:

> On 10/29/2014 08:06 PM, Matthew Jordan wrote:
>
>> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy  wrote:
>>
>>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
>>>
>>>
>> codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A
>> number of performance improvements in the media handling in Asterisk
>> required some codec compatibility changes.
>>
>> I would expect said modules to be available in the next few weeks.
>>
>>  Great. Thanks.
> sean
>
>
​I just checked at http://digium.com/en/products/asterisk/downloads,
and of the "Add-On Voice Codecs", g.729 is the only one available
for Asterisk 13.

The Siren 7/14 and SILK codecs offer only 12.X selections, and I proved
today that the 12.x codecs will not load on Asterisk 13.

a "few weeks" has turned into almost half a year now. Are these
codecs no longer going to be available for 13 and up? Or, were they
just overlooked in the day-to-day rush called life?

murf​

-- 

Steve Murphy
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Re: [asterisk-users] Regarding Text To Speech conversion

[Steve Edwards]

On Monday 09 Mar 2015, janani m wrote:



The Error Which I face I have attached.


Please do not attach pictures. Please cut and past text.

On Mon, 9 Mar 2015, A J Stiles wrote:

That's a very common error and what it means is, the AGI script 
"/var/lib/asterisk/agi-bin/googletts.agi"  either has an incorrect #! 
line, or needs chmod +x run on it.


What do you get if you run
# ls -l /var/lib/asterisk/agi-bin/googletts.agi
and
# head -n1 /var/lib/asterisk/agi-bin/googletts.agi
respectively?


Another possibility if Windows boxes have been involved in 
downloading/extracting/copying the script -- incorrect line endings.


What does:

od -c /var/lib/asterisk/agi-bin/googletts.agi | head --lines=2

look like. Any '\r \n' (Windows line ending) where '\n' (Unix line ending) 
should be?


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-----
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Re: [asterisk-users] hangup call gw FXO

[Steve Davies]

Looking at the pastebin, the Vega device sends a CANCEL with reason:

Reason: Q.850 ;cause=16.

Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs

Regards,
Steve


On Thu, 5 Mar 2015 at 11:41 ricky gutierrez  wrote:

>
>
> On Wednesday, March 4, 2015, ricky gutierrez 
> wrote:
>
>> I'm having some problems with a vega sangoma, if a call comes into my
>> ivr and hangs up, the call continues to ring and leaves hanging the
>> channel, I have to restart Asterisk and everything works Ok
>>
>> my sangoma is a vega 50 , 4 FXO .
>>
>> I tried different tone of countries and does not work,
>>
>> this is the trace of which is for hanging up the channel:
>>
>> http://pastebin.com/y410Rhzt
>>
>> I was thinking that might help rpt timeout , I have put in 30s, but
>> does not work
>>
>> any advice?
>>
>> regardss
>>
>>
>>
>>  something strange, I have some extensions not connected to Asterisk and
> if I call, I get the message busy, the version I'm using is asterisk 11.15
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
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Re: [asterisk-users] System() command refuses to execute bash script

[Steve Edwards]

On Mon, 2 Mar 2015, Stefan Viljoen wrote:

So the problem was not Asterisk or BASH or permissions, but rather that 
it appears that all paths in any System() script must be absolutely, not 
relatively, specified.


Not quite.

The 'base' for relative paths would be the 'cwd' (current working 
directory) of the Asterisk process.


You can show the cwd for your running Asterisk by:

sudo ls -l /proc/$(pidof asterisk)/cwd

which is a link to the process's cwd.

I suspect if you search your file system ('sudo find / -name 
wireless-executed'), you will find 'wireless-executed' -- probably in the 
directory shown by the above command.


You can set this in the script that starts Asterisk. I set mine to /tmp/ 
('cd /tmp/') so I know where any random file access will occur, relatively 
speaking.


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-----
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Re: [asterisk-users] System() command refuses to execute bash script

[Steve Edwards]

Please don' top post.

On Mon, 2 Mar 2015, Tech Support wrote:


I'm surprised that you didn't have to specify the full path to the 'touch'
command. When writing AGI scripts, I always do something like
$touch = which( 'touch' ). I guess it's over kill.


The AGI process inherits the environment of the parent, Asterisk.

You can set the Asterisk environment in the script that starts Asterisk.

For example:

cd /tmp/
ulimit -n 8192
nice --adjustment=-20\
env --ignore-environment\
HOSTNAME=${HOSTNAME}\
PATH=${PATH}\
$ASTERISK $START_OPTIONS

--
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-----
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