Re: [asterisk-users] Question on AEL2 string comparisons

2013-07-03 Thread Satish Barot 
On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne wrote:

> I have this code in a dial plan:
>
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
> "SIP/41799"]?notfromlocal)
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
> "SIP/41700"]?notfromlocal)
>
> The value of "${CALLERID(num)}" appears to be "SIP/41712-0181"
>
> -- Executing [41720@from-internal:5] GotoIf("SIP/41712-0181",
> "0?notfromlocal") in new stack
> -- Executing [41720@from-internal:6] GotoIf("SIP/41712-0181",
> "1?notfromlocal") in new stack
> -- Goto (from-internal,41720,8
>
> This value is evidently comparing to be less than "SIP/41799" as
> expected but also is considered less than "SIP/41700" as well, which
> is not expected (by me).  What am I doing wrong here?
>
> What I am attempting to accomplish is to detect calls originally made
> from internal extension numbers in the range 41700..41799 inclusive.
> What is the correct method to accomplish this?
>
> James B. Byrne
>

${CALLERID(num)} should give you only number and not technology i.e. 41712.

Give this a shot,

exten => _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten => _417XX,n,GotoIf($[$["${CALLERID(num)}" > "41799"] |
$["${CALLERID(num)}" < "41700"]]?notfromlocal:)

--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Executing Script after MixMonitor is called

2013-07-03 Thread Satish Barot 
On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> I tried with hangup cause but my script is not executed... also I tried
> the same script with mix monitor itself no sucess.
>
> The script what I have is, am converting wav file to flac format..
> On 11 Jun 2013 11:17, "Satish Barot"  wrote:
>
>> And yes if you want to use System application in your dialplan then have
>> System in your h extension
>>
>> System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
>> /PathToMp3FileToBE Stored/filename.mp3)
>>
>> On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot > > wrote:
>>
>>> Hi Gopamkrishnan,
>>>
>>> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
>>> facility to execute a command when recording is over.
>>>
>>> *In my case, 'wav2mp3' is a script which gets executed and converts 
>>> recorded wav audio file to mp3. I pass ${FILENAME} as an argument to my 
>>> script.*
>>>
>>> *You should have something like 
>>> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) 
>>> in your dialplan.
>>>
>>> Hope this helps.
>>>
>>> --Satish Barot
>>>
>>> Ahmedabad, India
>>>
>>>
>>> On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 Hi Satish,

 I tried with sox, without any parameter, just sox filename.wav to
 filename.mp3, in linux shell prompt... the file is been converted...

 Now If i want to run that command using dialplan,

 MixMonitor(filename.wav,m)
 Monitor_Exec(sox filename.wav filename.mp3)

 Or to use System command?

 Regards..


 On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot <
 satish4aster...@gmail.com> wrote:

> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
> (Asterisk 1.8.7.0).
> ...
> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
> ^{FILENAME})
> ...
> and my script is...
>
> #!/bin/bash
>
> WAV="/var/spool/asterisk/monitor/$1"
> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
> MP3DEST="/var/spool/asterisk/mp3/$MP3"
> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m
> --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>
> --SATISH BAROT
> Ahmedabad,India.
>
>
> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib 
> wrote:
>
>> Hello Guys,
>> I am trying to convert files that are .wac to mp3 after mixmonitor
>> command is called but it doesnt execute the command, I tried the command 
>> in
>> terminal it worked, any help please ... below is my dial plan
>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b
>> 8 -t -F -m m --bitwidth 8 --quiet
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>
>>
>
Show your latest dialplan and script.

--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot 
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI wrote:

> Le 03/07/2013 15:07, Satish Barot a écrit :
>
>> [...]
>>
>> Then you should add Local channel as a queue member and dial your SIP
>> member from Local channel context.  A little hint here. Suppose you have a
>> support queue configured in queues.conf
>>
>> ;queues.conf
>> [support]
>> ... ...
>> member => Local/1000@memberconnector,0,**John Smith,SIP/1000
>> ... ...
>> Now In your dialplan add a context for local channel,
>> [memberconnector]
>> exten => _X.,1,Noop(Connecting to Member at ${EXTEN})
>> same => n,Dial(SIP/${EXTEN})
>> ;Check the Dialstatus for Member
>> same => n,Noop(DIALSTATUS=${**DIALSTATUS})
>>
>
> Already tested, I tried again as the option passed to queue was changed (n
> option)
>
> Logs:
>
> -- Started music on hold, class 'default', on SIP/gw-005e
> -- Executing [909@memberconnector:1] 
> Dial("Local/909@**memberconnector-0002;2",
> "SIP/s-ntfe_909,60,") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/s-ntfe_909
> -- SIP/s-ntfe_909-0060 is ringing
> -- Local/909@memberconnector-**0002;1 is ringing
> -- SIP/s-ntfe_909-0060 is ringing
> -- Stopped music on hold on SIP/gw-005e
>   == Spawn extension (macro-toQueue, s, 11) exited non-zero on
> 'SIP/gw-005e' in macro 'toQueue'
>   == Spawn extension (incoming-Swiss-itech, 1, 204) exited non-zero on
> 'SIP/gw-005e'
> -- Executing [h@incoming-Swiss-itech:1] NoOp("SIP/gw-005e", "Call
> ended with QUEUESTATUS= and DIALSTATUS= and HANGUPCAUSE=0") in new stack
>
> From extension:
>
> [memberconnector]
> ;
> exten => _XXX,1,Dial(SIP/${peerPrefix}$**{EXTEN},${TIMERINGQUEUE},)
>   same => n,NoOp(DIALSTATUS=${**DIALSTATUS})
>
> As you can see, all status are empty,
>
>
> --
> Daniel
>
>

QUEUESTATUS will contain different values in different scenarios. i.e. If a
call gets answered then the value is CONTINUE, If a call doesn't get
answered and Queue timeout happens then TIMEOUT. If a caller hangs up when
call is in Queue then QUEUESTATUS will be blank.

Have something like this,
... ...
same => n,queue(support,c,,,20)
same => n,Noop(QSTATUS=${QUEUESTATUS})
... ...
exten => h,1,Noop(QSTATUS=${QUEUESTATUS})

[memberconnector]
exten => _X.,1,Noop(Connecting to Member at ${EXTEN})
same => n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same => n,Noop(DIALSTATUS=${DIALSTATUS})

exten => h,1,Noop(DIALSTATUS=${DIALSTATUS})

--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread Eric Wieling 
See cli.conf.sample

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ian Pilcher
Sent: Wednesday, July 03, 2013 7:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk stops registering

On 07/03/2013 05:51 PM, David Duffett wrote:
> I would enable SIP debugging, but only for that provider.

Will that increase the logging verbosity?

> This can be done on the Asterisk command line by using either of the
> following:
> 
> *> sip set debug peer 
> 
> or
> 
> *> sip set debug ip 

Can this be set in a config file?

This is a sporadic problem, so I need to set the server up to log the 
appropriate information when it occurs.

Thanks!

--

Ian Pilcher arequip...@gmail.com
Sometimes there's nothing left to do but crash and burn...or die trying.



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Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread Ian Pilcher 
On 07/03/2013 05:51 PM, David Duffett wrote:
> I would enable SIP debugging, but only for that provider.

Will that increase the logging verbosity?

> This can be done on the Asterisk command line by using either of the
> following:
> 
> *> sip set debug peer 
> 
> or
> 
> *> sip set debug ip 

Can this be set in a config file?

This is a sporadic problem, so I need to set the server up to log the
appropriate information when it occurs.

Thanks!

-- 

Ian Pilcher arequip...@gmail.com
Sometimes there's nothing left to do but crash and burn...or die trying.



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Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread David Duffett 
I would enable SIP debugging, but only for that provider.

This can be done on the Asterisk command line by using either of the
following:

*> sip set debug peer 

or

*> sip set debug ip 


On 3 July 2013 23:25, Ian Pilcher  wrote:

> On several occassions lately, my home Asterisk box has stopped
> registering with my VoIP provider.  I haven't been able to reproduce the
> problem, and the log doesn't contain anything useful.
>
> How can I increase the log verbosity for SIP registration-related
> events?  I've looked through logger.conf and tried searching with
> Google, but I haven't been able to find a clear answer.
>
> This is Asterisk 1.8.20.0, BTW.
>
> Thanks!
>
> --
> 
> Ian Pilcher arequip...@gmail.com
> Sometimes there's nothing left to do but crash and burn...or die trying.
> 
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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>



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[asterisk-users] Asterisk stops registering

2013-07-03 Thread Ian Pilcher 
On several occassions lately, my home Asterisk box has stopped
registering with my VoIP provider.  I haven't been able to reproduce the
problem, and the log doesn't contain anything useful.

How can I increase the log verbosity for SIP registration-related
events?  I've looked through logger.conf and tried searching with
Google, but I haven't been able to find a clear answer.

This is Asterisk 1.8.20.0, BTW.

Thanks!

-- 

Ian Pilcher arequip...@gmail.com
Sometimes there's nothing left to do but crash and burn...or die trying.



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Re: [asterisk-users] Executing Script after MixMonitor is called

2013-07-03 Thread Gopalakrishnan N 
I tried with hangup cause but my script is not executed... also I tried the
same script with mix monitor itself no sucess.

The script what I have is, am converting wav file to flac format..
On 11 Jun 2013 11:17, "Satish Barot"  wrote:

> And yes if you want to use System application in your dialplan then have
> System in your h extension
>
> System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
> /PathToMp3FileToBE Stored/filename.mp3)
>
>
>
>
>
>
> On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot 
> wrote:
>
>> Hi Gopamkrishnan,
>>
>> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
>> facility to execute a command when recording is over.
>>
>> *In my case, 'wav2mp3' is a script which gets executed and converts recorded 
>> wav audio file to mp3. I pass ${FILENAME} as an argument to my script.
>> *
>>
>> *You should have something like 
>> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in 
>> your dialplan.
>>
>> Hope this helps.
>>
>> --Satish Barot
>>
>>
>> Ahmedabad, India
>>
>>
>>
>>
>>
>> On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Hi Satish,
>>>
>>> I tried with sox, without any parameter, just sox filename.wav to
>>> filename.mp3, in linux shell prompt... the file is been converted...
>>>
>>> Now If i want to run that command using dialplan,
>>>
>>> MixMonitor(filename.wav,m)
>>> Monitor_Exec(sox filename.wav filename.mp3)
>>>
>>> Or to use System command?
>>>
>>> Regards..
>>>
>>>
>>> On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot <
>>> satish4aster...@gmail.com> wrote:
>>>
 This is how I use a wav to mp3 script on Mixmonitor in my dialplan
 (Asterisk 1.8.7.0).
 ...
 same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
 ^{FILENAME})
 ...
 and my script is...

 #!/bin/bash

 WAV="/var/spool/asterisk/monitor/$1"
 MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
 MP3DEST="/var/spool/asterisk/mp3/$MP3"
 /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m
 --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1

 --SATISH BAROT
 Ahmedabad,India.


 On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib 
 wrote:

> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor
> command is called but it doesnt execute the command, I tried the command 
> in
> terminal it worked, any help please ... below is my dial plan
> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b
> 8 -t -F -m m --bitwidth 8 --quiet
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


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>>>
>>>
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>>
>>
>
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[asterisk-users] Question on AEL2 string comparisons

2013-07-03 Thread James B. Byrne 
I have this code in a dial plan:

exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
"SIP/41799"]?notfromlocal)
exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
"SIP/41700"]?notfromlocal)

The value of "${CALLERID(num)}" appears to be "SIP/41712-0181"

-- Executing [41720@from-internal:5] GotoIf("SIP/41712-0181",
"0?notfromlocal") in new stack
-- Executing [41720@from-internal:6] GotoIf("SIP/41712-0181",
"1?notfromlocal") in new stack
-- Goto (from-internal,41720,8

This value is evidently comparing to be less than "SIP/41799" as
expected but also is considered less than "SIP/41700" as well, which
is not expected (by me).  What am I doing wrong here?

What I am attempting to accomplish is to detect calls originally made
from internal extension numbers in the range 41700..41799 inclusive. 
What is the correct method to accomplish this?

-- 
***  E-Mail is NOT a SECURE channel  ***
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[asterisk-users] Calls drop after transfer

2013-07-03 Thread Carlos Chavez 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I have an Asterisk 11.4 SIP only system.  We are using a SIP trunk
for outside calls.  We are having a problem with calls dropping after
a transfer.

Outside call awswered by phone 101
101 transfers to 100 (attended transfer)
call is dropped after a few seconds

I cannot really think of anything else to check in sip.conf.
Incoming calls never drop if they are not transferred.

- -- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-BEGIN PGP SIGNATURE-
Version: GnuPG/MacGPG2 v2.0.18 (Darwin)
Comment: GPGTools - http://gpgtools.org
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/

iEYEARECAAYFAlHUbhQACgkQqmNh+MyHzx7h4QCeNTjexC8a0RPQHeSR2lzc3Q+c
BWMAoMnpDtLJ4TQK+ZZTY7/XoPZyWYKV
=0sYb
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[asterisk-users] Custom dial plan for internal transfers of external calls

2013-07-03 Thread James B. Byrne 
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2

We use Snom870 handsets with firmware v.8.7.3.19.

I am trying to develop a custom dial plan to invoke a distinctive
ring-tone when an external call is transferred internally.  Based on
an earlier solution I discovered I am attempting this:

[from-internal]
include => set-alert-if-local

[from-internal-original]
include => from-internal-xfer
include => bad-number

[set-alert-if-local]
.  . .
exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
"SIP/41799"]?notfromlocal)
exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
"SIP/41710"]?notfromlocal)
;If we reach here then the caller is within the upper and lower bounds
exten => _417XX,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten => _417XX,n(notfromlocal),Goto(from-internal-original,${EXTEN},1)
;The following three lines must not be changed!
exten => _.,1,Goto(from-internal-original,${EXTEN},1)
exten => s,1,Goto(from-internal-original,s,1)
exten => h,1,Macro(hangupcall)


This context appears to be entered only when the call originates from
another extension.  When a transfer of an external call is attempted
theis context does not seem to be entered.

The following abstracted asterisk trace log shows this for an incoming
call answered on 41712 and then transferred to 41720.

-- SIP/41711-0165 is ringing
-- SIP/41713-0167 is ringing
-- SIP/41712-0166 is ringing
-- SIP/41720-0169 is ringing
-- SIP/41718-0168 is ringing

  == Extension Changed 41712[ext-local] new state InUse for Notify
User 41710
  == Extension Changed 41712[ext-local] new state InUse for Notify
User 41711
-- SIP/41712-0166 connected line has changed. Saving it until
answer for DAHDI/1-1
-- SIP/41712-0166 answered DAHDI/1-1
  == Extension Changed 41712[ext-local] new state InUse for Notify
User 41715
  == Extension Changed 41712[ext-local] new state InUse for Notify
User 41717
  == Extension Changed 41712[ext-local] new state InUse for Notify
User 41718
-- Executing [s@macro-auto-blkvm:1] Set("SIP/41712-0166",
"__MACRO_RESULT=") in new stack
-- Executing [s@macro-auto-blkvm:2] Macro("SIP/41712-0166",
"blkvm-clr,") in new stack

At this point the original incoming call is answered.

.  .  .
-- Executing [s@macro-auto-blkvm:3] ExecIf("SIP/41712-0166",
"0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=41712)") in new stack

And then transferred to 41720

[2013-07-03 13:43:03] WARNING[7954][C-4685]: res_srtp.c:406
ast_srtp_unprotect: SRTP unprotect failed with: authentication failure
110
-- Stopped music on hold on DAHDI/1-1
  == Extension Changed 41712[ext-local] new state Idle for Notify User
41710
  == Extension Changed 41712[ext-local] new state Idle for Notify User
41711
  == Extension Changed 41712[ext-local] new state Idle for Notify User
41715
  == Spawn extension (from-internal-xfer, 41720, 1) exited non-zero on
'DAHDI/1-1' in macro 'dial'
  == Spawn extension (from-internal-xfer, 41720, 1) exited non-zero on
'DAHDI/1-1'
-- Executing [41720@from-internal-xfer:1] Set("DAHDI/1-1",
"__RINGTIMER=20") in new stack

And finally answered on 41720

[2013-07-03 13:43:16] DEBUG[29747][C-4685]: sip/sdp_crypto.c:310
sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:5D038u88tI6PLyruDovyQIku9PH7exEAL3Qolc9m
  == Extension Changed 41720[ext-local] new state InUse for Notify
User 41711
  == Extension Changed 41720[ext-local] new state InUse for Notify
User 41715
-- SIP/41720-016a answered DAHDI/1-1

It is evident from the trace that the context [set-alert-if-local] is
not entered on internal transfers and I lack the experience to
understand why.  Can someone here enlighten me as to what is going on
in this instance and how I should change my contexts in order to check
for internal transfers of external calls?

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte & Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

2013-07-03 Thread Amit Patkar | ATPL 
Faced some issues with size. So deleted some content. But this was 
full file and we are using default file.

Hi Matt,

I have pasted entire say.conf here. It has datetime extension.
;
; language configuration
;

[general]
mode=new; method for playing numbers and dates
; old - using asterisk core function
; new - using this configuration file


[digit-base](!); base rule for digit strings
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}

[date-base](!); base rules for dates and times
; the 'SAY' variable contains MMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4}; year, 19xx
_date:[Bbh]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1}; day of week
_date:[de]:. => num:${SAY:6:2}; day of month
_date:[H]:. => digits/oh, num:${SAY:8:2} ; hour (oh one, oh two, 
..., oh nine, ten, eleven, ..., twenty-three)

_date:[I]:. => num:$[${SAY:8:2} % 12]; hour 0-12
_date:[M]:. => num:${SAY:10:2}; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: 
"a-m"]; am pm

_date:[pP]:. => digits/p-m; am pm
_date:[S]:. => num:${SAY:13:2}; seconds
_date:[Ii]:. => num:$[${SAY:8:2} % 12]; hour 0-12
_date:[k]:. => num:${SAY:8:2}; hour (one, two. 
three, ...,twenty three

_date:[m]:. => num:${SAY:4:2}; month number
_date:[Q]:. => date:dby ;incompleat ; "today", "yesterday" or ABdY
_date:[q]:. => date:dby ;incompleat ; "" (for today), 
"yesterday", weekday, or ABdY
_date:[R]:. => date:HM${SAY}; 24 hour time, 
including minute
_date:[T]:. => date:HMS${SAY}; 24 hour, minure, 
seconds


[en-base](!)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 =>  digits/${SAY}
_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:X00 => num:${SAY:0:1}, digits/hundred
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}

_[n]um:X000 => num:${SAY:0:1}, digits/thousand
_[n]um: => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XX000 => num:${SAY:0:2}, digits/thousand
_[n]um:X => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
_[n]um:XX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}

_[n]um:X00 => num:${SAY:0:1}, digits/million
_[n]um:XXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XX00 => num:${SAY:0:2}, digits/million
_[n]um: => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXX00 => num:${SAY:0:3}, digits/million
_[n]um:X => num:${SAY:0:3}, digits/million, num:${SAY:3}

_[n]um:X0 => num:${SAY:0:1}, digits/billion
_[n]um:XX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XX0 => num:${SAY:0:2}, digits/billion
_[n]um:XXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXX0 => num:${SAY:0:3}, digits/billion
_[n]um: => num:${SAY:0:3}, digits/billion, num:${SAY:3}

; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]00 => num:${SAY:0:1}, digits/h-hundred
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/h-hundred, enum:${SAY:1}

[en_GB](date-base,digit-base,en-base)
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, vm-and, num:${SAY:1}

[it](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 =>  digits/${SAY}
_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:100 => digits/hundred
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}

_[n]um:1000 => digits/thousand
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]000 => num:${SAY:0:1}, digits/thousands
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XX000 => num:${SAY:0:2}, digits/thousands
_[n]um:X => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXX000 => num:${SAY:0:3}, digits/thousands
_[n]um:XX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}

_[n]um:100 => num:${SAY:0:1}, digits/million
_[n]um:1XX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]00 => num:${SAY:0:1}, digits/millions
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Administrator TOOTAI 

Le 03/07/2013 15:07, Satish Barot a écrit :

[...]
Then you should add Local channel as a queue member and dial your SIP 
member from Local channel context.  A little hint here. Suppose you 
have a support queue configured in queues.conf


;queues.conf
[support]
... ...
member => Local/1000@memberconnector,0,John Smith,SIP/1000
... ...
Now In your dialplan add a context for local channel,
[memberconnector]
exten => _X.,1,Noop(Connecting to Member at ${EXTEN})
same => n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same => n,Noop(DIALSTATUS=${DIALSTATUS})


Already tested, I tried again as the option passed to queue was changed 
(n option)


Logs:

-- Started music on hold, class 'default', on SIP/gw-005e
-- Executing [909@memberconnector:1] 
Dial("Local/909@memberconnector-0002;2", "SIP/s-ntfe_909,60,") in 
new stack

  == Using SIP RTP CoS mark 5
-- Called SIP/s-ntfe_909
-- SIP/s-ntfe_909-0060 is ringing
-- Local/909@memberconnector-0002;1 is ringing
-- SIP/s-ntfe_909-0060 is ringing
-- Stopped music on hold on SIP/gw-005e
  == Spawn extension (macro-toQueue, s, 11) exited non-zero on 
'SIP/gw-005e' in macro 'toQueue'
  == Spawn extension (incoming-Swiss-itech, 1, 204) exited non-zero 
on 'SIP/gw-005e'
-- Executing [h@incoming-Swiss-itech:1] NoOp("SIP/gw-005e", 
"Call ended with QUEUESTATUS= and DIALSTATUS= and HANGUPCAUSE=0") in new 
stack


From extension:

[memberconnector]
;
exten => _XXX,1,Dial(SIP/${peerPrefix}${EXTEN},${TIMERINGQUEUE},)
  same => n,NoOp(DIALSTATUS=${DIALSTATUS})

As you can see, all status are empty,

--
Daniel

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Re: [asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread I.Pavlov 
Thanks for answer. For correct dialstatus I use now:

 

Set(DIALSTATUS=${IF($[ "${SIPPEER(${EXTEN},curcalls)}" >=
"${SIPPEER(${EXTEN},limit)}" ]?BUSY:${DIALSTATUS})});

 

I tried to use Busy app and got CDR(disposition)=BUSY, but in this way I
can't redirect *calling* channel to voicemail, because it hang up both
channels.  

Finally, in voicemail I cause ResetCDR(wav) to get two records in cdr (with
BUSY (in ideal, for called party) and ANSWER (for voicemail) dispositions).

 

My asterisk 1.6.2.20.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Wednesday, July 03, 2013 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP. Call-limit dialstatus

 

On 3 Jul 2013, at 12:28, I.Pavlov wrote:

[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1

-- Couldn't call 0014

  == Everyone is busy/congested at this time (0:0/0/0)

-- Executing [0014@sub_pbxdialco:50] NoOp("SIP/1295-01f8",
"CHANUNAVAIL") in new stack

 

I think that isn't correct. Is it possible to change dialstatus and
CDR(disposition) to BUSY-value when call-limit reached?

 

You could look for CHANUNAVAIL in dialplan and run Busy(), bit of a
workaround but may work I've not actually used the Busy() app recently,
but I assume the CDRs would work with that?

 

S

 

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Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot 
On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI wrote:

> Hi Satish
>
> Le 03/07/2013 09:15, Satish Barot a écrit :
>
>
>> On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI 
>> > ad...@tootai.net>> wrote:
>>
>> Hi all,
>>
>> I have to questions about queues. Member is a phone like
>> SIP/myphone and only one member in the queue.
>>
>> At first, DIALSTATUS doesn't return any status. How to now if a
>> call in queue has been answered or if caller just hangup?
>>
>> Queue application uses QUEUESTATUS and not DIALSTATUS
>>
>
> QUEUESTATUS returns the status of the queue not the one from the dial
> command used in the queue. And there is no such information I'm looking for
> in QUEUESTATUS.
>
> Other clue?
>
Then you should add Local channel as a queue member and dial your SIP
member from Local channel context.  A little hint here. Suppose you have a
support queue configured in queues.conf

;queues.conf
[support]
... ...
member => Local/1000@memberconnector,0,John Smith,SIP/1000
... ...
Now In your dialplan add a context for local channel,
[memberconnector]
exten => _X.,1,Noop(Connecting to Member at ${EXTEN})
same => n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same => n,Noop(DIALSTATUS=${DIALSTATUS})


>
>
>> Second, how to deal with timeout, I have strange behaviors. If I
>> put timeout=60 in queue.conf and I call the queue passing also 60
>> as timeout value, asterisk is returning after 5000ms the 4000 then
>> 2000 then 2000 aso. I can replace the 60sec value on both place,
>> or 60 in queue conf and 10 when calling queue, I never have a
>> stable behavior and more, not what I want.
>>
>> Exemple: let say asterisk should try all 20 seconds to call the
>> member for 8 seconds: how to configure this? What I found, is to
>> put timeout=0 in queue conf and passing 20 to queue, so caller
>> stays in queue 20 seconds before timeout. But asterisk rings 20
>> seconds :-(
>>
>> A snippet from queues.conf...
>>  "A Queue has two different "timeout" values associated with it. One is
>> the  timeout parameter configured in queues.conf. This timeout specifies
>> the  amount of time to try ringing a member's phone before considering the
>> member to be unavailable. The other timeout value is the timeout argument
>> to the Queue() application. This timeout represents the absolute amount of
>> time to allow a caller to stay in the queue before the caller is removed
>> from the queue. "
>>
>> Correct me if I am wrong but if you want your caller to be in Queue for
>> 20 seconds and try calling member for 8 seconds you should have 20 as a
>> timeout argument in Queue application in your dialplan and in timeout in
>> queues.conf should be 8. Also check the 'retry' and 'timeoutpriority'
>> parameters for queues.conf
>>
>
> OK, I got it: parameter n was send to queue. Now everything is working
> like it should.
>
> Thanks for your help.
>
>
> --
> Daniel
>
>
--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

2013-07-03 Thread Matthew Jordan 
On Tue, Jul 2, 2013 at 11:03 AM, Amit Patkar | ATPL  wrote:

>  Hi Matt,
>
> As required, please find DEBUG trace for datetime function. I have used
> this function in Dialplan to capture DEBUG trace. I hope, this can help us
> in resolving the issue.
>
> [Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
> 1001
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 -
> state 2 (In use)
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
> [Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'Answer'
> [Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:1]
> Answer("SIP/1001-", "") in new stack
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking
> channel drivers for SIP - 1001
> [Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
> 1001
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 -
> state 2 (In use)
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: SIP answering channel:
> SIP/1001-
> [Jul  2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting the marker bit
> due to a source update
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Setting framing from config on
> incoming call
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: ** Our capability: 0x4 (ulaw)
> Video flag: True Text flag: True
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: -- Done with adding codecs to SDP
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Done building SDP. Settling with
> this capability: 0x4 (ulaw)
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'SIP/2.0 200' onto
> UDP socket destined for 192.168.2.18:7490
> [Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> [Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> [Jul  2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> [Jul  2 15:54:44] DEBUG[2722] chan_sip.c: = Looking for  Call ID:
> YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY. (Checking From) --From tag
> 226b515a --To-tag as6e727cd7
> [Jul  2 15:54:44] DEBUG[2722] chan_sip.c:  Received ACK (6) - Command
> in SIP ACK
> [Jul  2 15:54:44] DEBUG[2722] chan_sip.c: Stopping retransmission on
> 'YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.' of Response 2: Match Found
> [Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'DateTime'
> [Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:2]
> DateTime("SIP/1001-", "136512,,YBd") in new stack
> [Jul  2 15:54:44] DEBUG[2737] app_playback.c: string
>  depth <0>
> [Jul  2 15:54:44] DEBUG[2737] app_playback.c: try
>  in 
> [Jul  2 15:54:44] DEBUG[2737] pbx.c: Launching 'Hangup'
> [Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:3]
> Hangup("SIP/1001-", "") in new stack
> [Jul  2 15:54:44] DEBUG[2737] pbx.c: Spawn extension (avhan,,3) exited
> non-zero on 'SIP/1001-'
> [Jul  2 15:54:44] VERBOSE[2737] pbx.c:   == Spawn extension (avhan, ,
> 3) exited non-zero on 'SIP/1001-'
> [Jul  2 15:54:44] DEBUG[2737] channel.c: Soft-Hanging up channel
> 'SIP/1001-'
> [Jul  2 15:54:44] DEBUG[2737] channel.c: Hanging up channel
> 'SIP/1001-'
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Hangup call SIP/1001-,
> SIP callid YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Updating call counter for
> incoming call
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking
> channel drivers for SIP - 1001
> [Jul  2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
> 1001
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001 -
> state 1 (Not in use)
> [Jul  2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '1'
> [Jul  2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting RTCP address on
> RTP instance '0x98ac7f0'
> [Jul  2 15:54:44] DEBUG[2737] netsock2.c: Splitting '192.168.2.18:7490'
> into...
> [Jul  2 15:54:44] DEBUG[2737] netsock2.c: ...host '192.168.2.18' and port
> '7490'.
> [Jul  2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'BYE sip:100'
> onto UDP socket destined for 192.168.2.18:7490
>
>
>
So, these statements show that it is actually using the config file to try
to say the datetime:

[Jul  2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:2]
DateTime("SIP/1001-", "136512,,YBd") in new stack
[Jul  2 15:54:44] DEBUG[2737] app_playback.c: string
 depth <0>
[Jul  2 15:54:44] DEBUG[2737] app_playback.c: try
 in 

The DEBUG statements in app_playback indicate the following:

[asterisk-users] CEL events

2013-07-03 Thread Jairo 
Dear list.

This is probably a complex subject but is that right to consider:

a) each distinct linkedid field value in a mysql CEL table as a unique call?

b) the duration of a call as the period (eventtime fields) between
BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not
considering transfers)

Just a start ...

Tks.
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Re: [asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread Steven Howes 
On 3 Jul 2013, at 12:28, I.Pavlov wrote:
> [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: 
> Call to peer '0014' rejected due to usage limit of 1
> -- Couldn't call 0014
>   == Everyone is busy/congested at this time (0:0/0/0)
> -- Executing [0014@sub_pbxdialco:50] NoOp("SIP/1295-01f8", 
> "CHANUNAVAIL") in new stack
>  
> I think that isn’t correct. Is it possible to change dialstatus and 
> CDR(disposition) to BUSY-value when call-limit reached?

You could look for CHANUNAVAIL in dialplan and run Busy(), bit of a workaround 
but may work I've not actually used the Busy() app recently, but I assume 
the CDRs would work with that?

S

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[asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread I.Pavlov 
Hi all. We have a problem with correct dialstatus and cdr(disposition) when
using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and
CDR(disposition)='NO ANSWER'

 

-- Executing [0014@sub_pbxdialco:49] Dial("SIP/1295-01f8",
"SIP/0014,12,tTkK") in new stack

  == Using SIP RTP CoS mark 5

[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1

-- Couldn't call 0014

  == Everyone is busy/congested at this time (0:0/0/0)

-- Executing [0014@sub_pbxdialco:50] NoOp("SIP/1295-01f8",
"CHANUNAVAIL") in new stack

 

I think that isn't correct. Is it possible to change dialstatus and
CDR(disposition) to BUSY-value when call-limit reached?

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Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Administrator TOOTAI 

Hi Satish

Le 03/07/2013 09:15, Satish Barot a écrit :


On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI 
mailto:ad...@tootai.net>> wrote:


Hi all,

I have to questions about queues. Member is a phone like
SIP/myphone and only one member in the queue.

At first, DIALSTATUS doesn't return any status. How to now if a
call in queue has been answered or if caller just hangup?

Queue application uses QUEUESTATUS and not DIALSTATUS


QUEUESTATUS returns the status of the queue not the one from the dial 
command used in the queue. And there is no such information I'm looking 
for in QUEUESTATUS.


Other clue?



Second, how to deal with timeout, I have strange behaviors. If I
put timeout=60 in queue.conf and I call the queue passing also 60
as timeout value, asterisk is returning after 5000ms the 4000 then
2000 then 2000 aso. I can replace the 60sec value on both place,
or 60 in queue conf and 10 when calling queue, I never have a
stable behavior and more, not what I want.

Exemple: let say asterisk should try all 20 seconds to call the
member for 8 seconds: how to configure this? What I found, is to
put timeout=0 in queue conf and passing 20 to queue, so caller
stays in queue 20 seconds before timeout. But asterisk rings 20
seconds :-(

A snippet from queues.conf...
 "A Queue has two different "timeout" values associated with it. One 
is the  timeout parameter configured in queues.conf. This timeout 
specifies the  amount of time to try ringing a member's phone before 
considering the member to be unavailable. The other timeout value is 
the timeout argument to the Queue() application. This timeout 
represents the absolute amount of time to allow a caller to stay in 
the queue before the caller is removed from the queue. "


Correct me if I am wrong but if you want your caller to be in Queue 
for 20 seconds and try calling member for 8 seconds you should have 20 
as a timeout argument in Queue application in your dialplan and in 
timeout in queues.conf should be 8. Also check the 'retry' and 
'timeoutpriority' parameters for queues.conf


OK, I got it: parameter n was send to queue. Now everything is working 
like it should.


Thanks for your help.

--
Daniel

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Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot 
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI wrote:

> Hi all,
>
> I have to questions about queues. Member is a phone like SIP/myphone and
> only one member in the queue.
>
> At first, DIALSTATUS doesn't return any status. How to now if a call in
> queue has been answered or if caller just hangup?
>
Queue application uses QUEUESTATUS and not DIALSTATUS

>
> Second, how to deal with timeout, I have strange behaviors. If I put
> timeout=60 in queue.conf and I call the queue passing also 60 as timeout
> value, asterisk is returning after 5000ms the 4000 then 2000 then 2000 aso.
> I can replace the 60sec value on both place, or 60 in queue conf and 10
> when calling queue, I never have a stable behavior and more, not what I
> want.
>
> Exemple: let say asterisk should try all 20 seconds to call the member for
> 8 seconds: how to configure this? What I found, is to put timeout=0 in
> queue conf and passing 20 to queue, so caller stays in queue 20 seconds
> before timeout. But asterisk rings 20 seconds :-(
>
A snippet from queues.conf...
 "A Queue has two different "timeout" values associated with it. One is
the  timeout parameter configured in queues.conf. This timeout specifies
the  amount of time to try ringing a member's phone before considering the
member to be unavailable. The other timeout value is the timeout argument
to the Queue() application. This timeout represents the absolute amount of
time to allow a caller to stay in the queue before the caller is removed
from the queue. "

Correct me if I am wrong but if you want your caller to be in Queue for 20
seconds and try calling member for 8 seconds you should have 20 as a
timeout argument in Queue application in your dialplan and in timeout in
queues.conf should be 8. Also check the 'retry' and 'timeoutpriority'
parameters for queues.conf


> I read documentation on voip-info.org, not very clear. timeout in
> queue.conf is only for calling agent, not members? If I put a timeout=8 in
> queue conf, how to tell asterisk to retry each 20 seconds playing MOH to
> the caller?
>

> Thanks for any hint
>
> --
> Daniel
>
>
--Satish Barot
Ahmedabad, India
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