[asterisk-users] cross ivr is comming in my ivr system

2012-04-04 Thread Jagadish Thoutam 
hi all,


   i have gradwell DID i am using it for inbound dialing with IVR when ever
customer call my DID some times other IVR is cumming on my IVR that IVR is
not even related with my server .can u please help me on this
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Re: [asterisk-users] cross ivr is comming in my ivr system

2012-04-04 Thread A J Stiles 
On Wednesday 04 April 2012, Jagadish Thoutam wrote:
> hi all,
> 
> 
>i have gradwell DID i am using it for inbound dialing with IVR when ever
> customer call my DID some times other IVR is cumming on my IVR that IVR is
> not even related with my server .can u please help me on this

Sorry.  This list is only for questions that make sense.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] cross ivr is comming in my ivr system

2012-04-04 Thread Arthur Stanfield 
- Original Message -
From: "A J Stiles" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, 4 April, 2012 10:48:02 AM
Subject: Re: [asterisk-users] cross ivr is comming in my ivr system

>On Wednesday 04 April 2012, Jagadish Thoutam wrote:
>> hi all,
>> 
>> 
>>i have gradwell DID i am using it for inbound dialing with IVR when ever
>> customer call my DID some times other IVR is cumming on my IVR that IVR is
>> not even related with my server .can u please help me on this

>Sorry.  This list is only for questions that make sense.

>-- 
>AJS

>Answers come *after* questions.

Not everyone who comes here is going to speak English perfectly as their first 
language. Taking snide little digs at someone because of their English skills 
is not what this userlist is about and doesn't benefit the community in the 
slightest (If anything, It damages it.)

For what its worth, I understood his question entirely. He has a DDI provided 
by Gradwell, Which when dialed leads into an IVR (I assume running on an 
Asterisk server, Hence why he's posted the question here.) Occasionally this 
number is hitting an IVR system that is not his own (Upstream Call routing 
issue?).

Jagadish, When the unknown IVR is being played are you seeing any traffic at 
all through the Asterisk CLI (Or in the logs?) - Does the call hit the server 
at all?.

Which Gradwell service are you using, And how are you connecting it to your 
server? 

Which version of Asterisk are you using?.

Cheers,
AJ.



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[asterisk-users] Asterisk 1.8 and DeadAGI

2012-04-04 Thread bilal ghayyad 
Dears;

In asterisk 1.8, it is not more possible to use DeadAGI?

Also, I found the below commands in the a2billing and I would to ask why it set 
the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How?

[a2billing-callingcard]
exten => _X.,1,NoOp(A2Billing Start)
exten => _X.,n,Answer()
exten => _X.,n,Wait(2)
exten => _X.,n,DeadAgi(a2billing.php,1)
exten => _X.,1,Hangup()

>From the other hand, what is the benifit of using NoOp here? Because I tried 
>it without NoOp and it was working? 

Regards
Bilal

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Re: [asterisk-users] Asterisk 1.8 and DeadAGI

2012-04-04 Thread SamyGo 
>
> exten => _X.,1,Hangup()

its wrong causing priority conflict.

NOOP is just for fun no benefit aside from printing something for you info
on CLI.


On Wed, Apr 4, 2012 at 6:35 PM, bilal ghayyad  wrote:

> Dears;
>
> In asterisk 1.8, it is not more possible to use DeadAGI?
>
> Also, I found the below commands in the a2billing and I would to ask why
> it set the sequence 1 for the Hangup()? Maybe because it is related to the
> NoOp? How?
>
> [a2billing-callingcard]
> exten => _X.,1,NoOp(A2Billing Start)
> exten => _X.,n,Answer()
> exten => _X.,n,Wait(2)
> exten => _X.,n,DeadAgi(a2billing.php,1)
> exten => _X.,1,Hangup()
>
> From the other hand, what is the benifit of using NoOp here? Because I
> tried it without NoOp and it was working?
>
> Regards
> Bilal
>
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Re: [asterisk-users] Asterisk 1.8 and DeadAGI

2012-04-04 Thread Alex Balashov 
Look up the definition of NoOp.  A moral and practical ambivalence inheres in 
that definition.  It is neither more nor less beneficial to use or not use it, 
for it is a NoOp.

--
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Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Decatur, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

bilal ghayyad  wrote:

>Dears;
>
>In asterisk 1.8, it is not more possible to use DeadAGI?
>
>Also, I found the below commands in the a2billing and I would to ask why it 
>set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? 
>How?
>
>[a2billing-callingcard]
>exten => _X.,1,NoOp(A2Billing Start)
>exten => _X.,n,Answer()
>exten => _X.,n,Wait(2)
>exten => _X.,n,DeadAgi(a2billing.php,1)
>exten => _X.,1,Hangup()
>
>From the other hand, what is the benifit of using NoOp here? Because I tried 
>it without NoOp and it was working? 
>
>Regards
>Bilal
>
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Re: [asterisk-users] Asterisk 1.8 and DeadAGI

2012-04-04 Thread Steve Edwards 

On Wed, 4 Apr 2012, SamyGo wrote:

NOOP is just for fun no benefit aside from printing something for you 
info on CLI.


It's not fun, it's misleading. Especially if English is not your primary 
language.


The name is not obvious. The Wikipedia entry for 'noop' says a 'no 
operation' instruction should be 'nop,' not 'noop.'


Even the 'help' description is contradictory. It says 'this application 
does nothing' and then it describes what it does.


There's a perfectly obvious application specifically designed for console 
output. It's named 'verbose' and it has more features.


A 'best practice' would be to use the obvious application with the 
intended action instead of the obtuse application with the convenient 
side-effect.


A 'noop' does have it's place as a 'place holder' for another application 
or to make a syntactically valid place for a label. If that's your intent, 
use it.


Otherwise, 'raise the bar' in your dialplan coding skills and do the right 
thing. Type the extra 3 characters and make your intent clear. Isn't that 
one of the 'hallmarks' of a skilled programmer?


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Voicemail crashs asterisk

2012-04-04 Thread Vik Killa 
http://asterisk-sucks.blogspot.com/

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Re: [asterisk-users] Voicemail crashs asterisk

2012-04-04 Thread Steve Edwards 

On Wed, 4 Apr 2012, Vik Killa wrote:

Please don't feed the trolls.

--
Thanks in advance,
-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] issue with Digium TDM410P

2012-04-04 Thread Alec Davis 
The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that
option was for the old phones that have a neon light (or equivalent
LED+ZENER ciruit).

Are other phones off the TDM410P (other than the VTECH) working, or is the
Vtech the only model with VMWI available to you.

I'm not able to check at the moment, I have copied the asterisk-users list,
someone else may have had the same problem.

Alec


> -Original Message-
> From: ve2...@gmail.com [mailto:ve2...@gmail.com] On Behalf Of 
> Mathieu Therrien
> Sent: Thursday, 5 April 2012 7:21 a.m.
> To: siva...@paradise.net.nz
> Subject: Fwd: issue with Digium TDM410P
> 
> Hi I have issue with my Digium TDM400 card, all working good 
> except VMWI on phones connected on FXS ports.
> I user with regular analog phone VTECH 5.6 GHz. I am unable 
> to activate VMWI. It's a red flashing LED on phone with 
> message display "MESSAGE WAITING".
> 
> With PAP2 it works good, but now I tried lot config, but none 
> works good.
> 
> I try lot configs, and search on forum and wiki of Asterisk 
> and Digium.
> 
> Could you help my with it, please?
> 
> Best regard.
> 
> 
> 
> 
> 
> folloing my config.
> 
> chan_dahdi.conf
> [...]
> ; The following keyword 'mwisendtype' enables various VMWI 
> methods on FXS lines (if supported). The default is to send 
> FSK only. The following options are ; available; ; 'fsk' or 
> undefined for exisiting FSK only support.
> ; 'none' no FSK spill heard every second time you go off hook.
> ; 'rpas' for existing support ring pulse before FSK ; 'lrev' 
> line reversed while messages exist ; 'hvdc' high voltage, 
> 90Vdc idle voltage ; 'hvac' high voltage neon generation, 
> Following Silicb AN33 ; 'nofsk' Disables FSK MWI spills from 
> being sent out.
> ; It is feasible that multiple options can be enabled.
> ;
> mwisendtype=fsk,lrev,hvac
> [...]
> 
> root@io:~# cat /sys/module/xpd_fxs/parameters/vmwi_ioctl
> Y
> root@io:~# cat /sys/module/xpd_fxs/parameters/reversepolarity
> N
> root@io:~# cat /sys/module/xpd_fxs/parameters/poll_digital_inputs
> 1000
> root@io:~# cat /sys/module/xpd_fxs/parameters/ring_trapez
> N
> root@io:~# cat /sys/module/xpd_fxs/parameters/dtmf_detection
> Y
> root@io:~# cat /sys/module/xpd_fxs/parameters/debug
> 0
> 
> root@io:~# dahdi_scan
> [1]
> active=yes
> alarms=OK
> description=Wildcard TDM410P
> name=WCTDM/0
> manufacturer=Digium
> devicetype=Wildcard TDM410P
> location=PCI Bus 01 Slot 08
> basechan=1
> totchans=4
> irq=0
> type=analog
> port=1,FXS
> port=2,FXS
> port=3,FXS
> port=4,FXS
> 
> this is my software info :
> io*CLI> core show version
> Asterisk 1.8.10.1 built by root @ io on a i686 running Linux on
> 2012-03-27 00:53:01 UTC
> io*CLI> dahdi show version
> DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2
> 
> and this is my kernel info :
> root@io:~# lsmod
> Module Size Used by
> dahdi_echocan_mg2 4691 4
> xpd_fxs 29178 0
> xpp 138396 1 xpd_fxs
> wctdm24xxp 97993 2
> dahdi_voicebus 44395 1 wctdm24xxp
> dahdi 192410 9 dahdi_echocan_mg2,xpd_fxs,xpp,wctdm24xxp,dahdi_voicebus
> [...]
> 
> root@io:~# uname -a
> Linux io 2.6.33.4-smp #2 SMP Wed May 12 22:47:36 CDT 2010 i686
> Intel(R) Pentium(R) 4 CPU 2.60GHz GenuineIntel GNU/Linux
> 
> 
> 
> 
> 
> --
> Mathieu Therrien, VE2TMQ / VA2IO
> B. Sc. A. Genie Logiciel
> www.ve2tmq.ca
> VoIP: sip:*99...@sipbroker.com iNUM: +883-5100-099-01841
> PSTN: +1-514-316-9498
> 
> PGP id : 0xA2882612
> PGP FingerPrint : FC7C 9D8A 11BE 7C0E C95B 1D2A E372 338C A288 2612


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[asterisk-users] Change extension for international ?

2012-04-04 Thread Olivier CALVANO 
Hi

i am search a solution for "change" the number called.

Sample:

I have a Linksys SPA942 connected in SIP with my server.

When this phone call a number: 043112
automatiquely change in 3343112

because my carrier want a number in international format.

It's possible ?

thanks
Olivier

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Re: [asterisk-users] Change extension for international ?

2012-04-04 Thread Danny Nicholas 
Simple
Exten => _043112.,1,Dial(SIP/blah,3343112${EXTEN:6})


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Wednesday, April 04, 2012 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Change extension for international ?

Hi

i am search a solution for "change" the number called.

Sample:

I have a Linksys SPA942 connected in SIP with my server.

When this phone call a number: 043112 automatiquely change in
3343112

because my carrier want a number in international format.

It's possible ?

thanks
Olivier

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Re: [asterisk-users] Change extension for international ?

2012-04-04 Thread Phil Frost 

On 04/04/2012 04:50 PM, Olivier CALVANO wrote:

Hi

i am search a solution for "change" the number called.

Sample:

I have a Linksys SPA942 connected in SIP with my server.

When this phone call a number: 043112
automatiquely change in 3343112

because my carrier want a number in international format.

It's possible ?

Yes. You can use ${EXTEN:N} to strip the leading N digits off the dial 
extension. Probably something like:


exten => 043122,1,Goto(33${EXTEN:1})

or

exten => 043122,1,Dial(SIP/33${EXTEN:1}@your-provider)


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[asterisk-users] FollowMe and billsec field of the CDRs

2012-04-04 Thread Sunny 
Hi list,

The billsec field of the CDRs not updated on followme app.
Lets suppose following configuration:

extensions.conf
[phones]
exten => _123,1,Answer()
exten => _123,2,Dial(SIP/123,12)
exten => _123,3,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?:4:5)
exten => _123,4,Followme(123)
exten => _123,5,Hangup()

exten => _101,1,Answer()
exten => _101,2,Dial(SIP/101)
exten => _101,3,Hangup()

exten => _102,1,Answer()
exten => _102,2,Dial(SIP/102)
exten => _102,3,Hangup()

followme.conf
[123]
context => phones
number => 101,30
number => 102,2


User A dial 123, after 12 seconds if the call is not answered then the
followme app is triggered.
According to the configuration on followme.conf 1st the call goes to 101
and after to 102.

The call is correctly connected on both cases (101 or 102).
On CDRs I have entries for each call, ie
User A  -> 123
User A  -> 101
User A  -> 102

The call is connected in User A  -> 102, but the field billsec of
the CDRs is always 0.
Same thing happens if the call is connected in User A  -> 101
Only the billsec of User A  -> 123 is correct.

I have tested on following versions and got same result on both.
asterisk-1.8.7.1
asterisk-1.8.11.0

Is this a expected behaviour or a bug?


Thanks,
Sunny
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Re: [asterisk-users] Asterisk 1.8 and DeadAGI

2012-04-04 Thread SamyGo 
Thanks for the lesson, I appreciate that.

On Wed, Apr 4, 2012 at 8:02 PM, Steve Edwards wrote:

> On Wed, 4 Apr 2012, SamyGo wrote:
>
>  NOOP is just for fun no benefit aside from printing something for you
>> info on CLI.
>>
>
> It's not fun, it's misleading. Especially if English is not your primary
> language.
>
> The name is not obvious. The Wikipedia entry for 'noop' says a 'no
> operation' instruction should be 'nop,' not 'noop.'
>
> Even the 'help' description is contradictory. It says 'this application
> does nothing' and then it describes what it does.
>
> There's a perfectly obvious application specifically designed for console
> output. It's named 'verbose' and it has more features.
>
> A 'best practice' would be to use the obvious application with the
> intended action instead of the obtuse application with the convenient
> side-effect.
>
> A 'noop' does have it's place as a 'place holder' for another application
> or to make a syntactically valid place for a label. If that's your intent,
> use it.
>
> Otherwise, 'raise the bar' in your dialplan coding skills and do the right
> thing. Type the extra 3 characters and make your intent clear. Isn't that
> one of the 'hallmarks' of a skilled programmer?
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
> --
> __**__**_
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>  
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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