Re: [asterisk-users] Beggining asterisk

[Esteban Cacavelos]

2011/9/6 Esteban Cacavelos 

>
>
> 2011/9/6 Esteban Cacavelos 
>
>>
>>
>> 2011/9/6 Leif Madsen 
>>
>>> On 04/09/11 02:51 PM, Tamer Higazi wrote:
>>>
 the 3rd edition is available, but that book covers every thing to run
 the asterisk PBX.

>>>
>>> You can read the 3rd edition online at http://ofps.oreilly.com/**
>>> titles/9780596517342/ 
>>>
>>> HTH!
>>> Leif.
>>>
>>> --
>>> Leif Madsen
>>> http://www.oreilly.com/**catalog/asterisk
>>>
>>>
>>> --
>>> __**__**
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>>>  
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>>>
>>
>>
>> Thanks for all the responses !. I will try with ubuntu bundleded packages
>> first.
>>
>> I will post my results.
>>
>>
>>
>> Esteban
>> --
>> Esteban L. Cacavelos de Amoriza
>> Cel: 0981 220 429
>>
>
>
> finally i decided to install from source because the documentation suggest
> that.
>
> I've installed successfully asterisk+dahdi+libpri. I tested a basic SIP
> configuration and there were no problems.
>
> Now i have problems with pstn termination and origination. I have one fxo
> module from witch i want to make and receive calls. Can I do that ?. I'll
> post my configuration files.
>
> I want to make calls from my android phone (where i have a SIP client) and
> recieve calls from my analog line through my androi.
>
> My country code is 595, city code 21, number , xxx xxx
>
>
> chan_dahdi.conf
>
> [channels]
>
> ;
> ; To apply other options to these channels, put them before "channel".
> ;
> signalling=fxs_ks  ; in Asterisk, FXO channels use FXS signaling
>  ; (and yes, FXS channels use FXO signaling)
> context=from-pstn
> channel => 1   ; apply all the previously defined settings to this
> channel
>
>
> extensions.conf
> [LocalSets]
>
> exten => 100,1,Dial(SIP/android-esteban) ; Replace 0001 with your
> device name
>
> exten => 101,1,Dial(SIP/recepcion) ; Replace 0002 with your device
> name
>
>
> exten => 200,1,Answer()
> same => n,Playback(hello-world)
> same => n,Hangup()
>
> ; TERMINATION
> [from-voip-network]
> exten => _X.,1,Verbose(2, Call from VoIP network to ${EXTEN})
>same => n,Dial(DAHDI/g0/${EXTEN})
>
> ORIGINATION
> [from-pstn]
> ; This is the context that would be listed in the config file
> ; for the circuit (i.e. chan_dahdi.conf)
>
> exten => _X.,1,Dial(SIP/android-esteban)
>
> [number-mapping]
> ; This context is not strictly required, but will make it easier
> ; to keep track of your DIDs in a single location in your dialplan.
> ; From here you can pass the call to another part of the dialplan
> ; where the actual dialplan work will take place.
>
> exten => 59521xx,1,Dial(SIP/android-esteban)
>
> exten => i,1,Verbose(2,Incoming call to invalid number)
>
>
>
>
> Dahdi system.conf
>
> # Autogenerated by /usr/sbin/dahdi_genconf on Tue Sep  6 14:40:03 2011
> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
> # your manual changes will be LOST.
> # Dahdi Configuration File
> #
> # This file is parsed by the Dahdi Configurator, dahdi_cfg
> #
> # Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
> fxsks=1
> echocanceller=mg2,1
> # channel 2, WCTDM/4/1, no module.
> # channel 3, WCTDM/4/2, no module.
> # channel 4, WCTDM/4/3, no module.
>
> # Global data
>
> loadzone= us
> defaultzone = us
>
>
>
> Thanks in advance !
>
> --
> Esteban L. Cacavelos de Amoriza
> Cel: 0981 220 429
>

dahdi_genconf should generate chan_dahdi.conf file ?. It generates these two
files

dahdi_genconf -v
Default parameters from /etc/dahdi/genconf_parameters
Generating /etc/dahdi/system.conf
Generating /etc/asterisk/dahdi-channels.conf


or i must create manually chan_dahdi.conf and include dahdi-channels.conf ?




BR

-- 
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
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Re: [asterisk-users] Reporting for Asterisk Call Center

[Gerardo Barajas]

There are a lot of reporting tools.
I have used:

Asternic: http://www.asternic.biz/
QueueMetrics: http://queuemetrics.com/index.jsp


On Fri, Sep 9, 2011 at 1:13 PM, bilal ghayyad  wrote:

> Hi All;
>
> Anyone advise for a free (open source) reporting to be used for asterisk
> call center?
>
> Regards
> Bilal
>
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Re: [asterisk-users] Reporting for Asterisk Call Center

[Robert Huddleston]

www.buildityourself.org

:)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Friday, September 09, 2011 2:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Reporting for Asterisk Call Center

Hi All;

Anyone advise for a free (open source) reporting to be used for asterisk
call center?

Regards
Bilal

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[asterisk-users] Reporting for Asterisk Call Center

[bilal ghayyad]

Hi All;

Anyone advise for a free (open source) reporting to be used for asterisk call 
center?

Regards
Bilal

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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

/etc/dahdi/system.conf:

# Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 31 18:10:31 2011
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24
echocanceller=oslec,1-23

# Span 2: WCTDM/0 "Wildcard AEX410 Board 1"
fxsks=25
echocanceller=oslec,25
fxsks=26
echocanceller=oslec,26
# channel 27, WCTDM/0/2, no module.
# channel 28, WCTDM/0/3, no module.

# Global data

loadzone = us
defaultzone = us

~

/etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]

usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=256
echocancelwhenbridged=no

callerid = asreceived
faxdetect=both

signalling=fxs_ks
context=incoming-local
group=1,2
channel => 25-26

context=incoming-ld
switchtype=dms100
signalling=pri_cpe
group=1,3
channel => 1-23

~

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com

On Sep 9, 2011, at 1:05 PM, Doug Lytle wrote:


Stephen H. Gerstacker wrote:
so any help would be appreciated

You could start by supplying snippets of your config for the PRI.  Telling us 
what PRI card you're using.

For example, my dahdi config for my PRI:

cd /etc/dahdi

cat system.conf

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defaultzone=us
loadzone=us

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us


cd /etc/asterisk


Show snippets of your chan_dahdi.conf


switchtype=national
context=definity
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
group=1
rxgain=0.0
txgain=0.0
usecallerid=yes
callerid=asreceived
channel=1-23

switchtype=national
context=cts
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
rxgain=0.0
txgain=0.0
group=2
usecallerid=yes
callerid=asreceived


Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] PRI Issues After Upgrade

[Doug Lytle]

Stephen H. Gerstacker wrote:

so any help would be appreciated


You could start by supplying snippets of your config for the PRI.  
Telling us what PRI card you're using.


For example, my dahdi config for my PRI:

cd /etc/dahdi

cat system.conf

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defaultzone=us
loadzone=us

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us


cd /etc/asterisk


Show snippets of your chan_dahdi.conf


switchtype=national
context=definity
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
group=1
rxgain=0.0
txgain=0.0
usecallerid=yes
callerid=asreceived
channel=1-23

switchtype=national
context=cts
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
rxgain=0.0
txgain=0.0
group=2
usecallerid=yes
callerid=asreceived


Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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[asterisk-users] Console Stereo - One call per ear

[fhirschberg]

Hi list!

I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board
and it works really good.
But I need a feature and don't know how to do this. 
What I need is the ability to have 2 separate calls on each ear on the
console channel. 
Is there a way to get this working? It should be possible to have one call
on both ears or, if another call is made, to hear this on one (selectable
L/R) ear, while the other call stays on the other ear.
Do I need a new console driver? I'm currently using chan_alsa and I already
have Alsa devices for left, right and left + right output. 
It would be great if anybody can help with informations or tips where to
start with my problem.

Greetings
Florian




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[asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I've finally moved from my 5 year old 1.2 installation to a 1.8 installation.

I'm using the packages supplied by asterisk.org for Ubuntu 
10.04, so I am at Asterisk 1.8.6.

On the console, I am seeing:

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/22 got hangup, cause 15

And then the call hangs up.  The cause code changes all the time.  It never 
seems to be consistent.  31, 34, 54, 102, 122, etc.  Before it was a different 
type of error.  I've tried multiple versions of libpri and am now on 1.4.12.  
I'm also on DAHDI 2.4.1.  The card in question is a Digium Wildcard TE110P 
T1/E1 Card.

I've tried IRC and the forum and gotten no help, so any help would be 
appreciated.

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com

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Re: [asterisk-users] Asterisk on Android?

[amit anand]

Hey can you share something on this

On Thu, Sep 8, 2011 at 23:49, Cobra 2  wrote:

> I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've
> gotten asterisk to run on that just fine.
>
>
> On Sat, Sep 3, 2011 at 9:45 AM, Daniel Tryba  wrote:
>
>> On Sat, Sep 03, 2011 at 01:53:54PM +0200, Gilles wrote:
>> > >Do you want to run the entire PBX on the Android client or are you just
>> > >looking for a IAX programm to be installed for receiving calls?!
>> >
>> > The entire PBX so I can have an IVR in the phone.
>>
>> I don't think you can access the radio of the phone (RIL) at this
>> moment. So if you want to use the GSM itself you are out of luck.
>>
>> --
>>
>>   Daniel Tryba
>>
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>>
>
>
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-- 

Amit Anand


+91 9818559898
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Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

[Sam Govind]

Thats goood ! :) thanks for updating.

On Fri, Sep 9, 2011 at 2:16 PM, Ishfaq Malik  wrote:

> Hi
>
> Can you please provide an excerpt of your logs when this happens?
>
> Regards
>
> Ish
>
> On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote:
> > Hi sam,
> >
> > Have solved the problem with your advice. Call drop in 10 seconds without
> disconnecting a-party call. Thank you very much.
> >
> > [TB]
> >
> > exten =>_X.,1,Wait(${INCOMING_WAIT})
> >
> > exten =>_X.,2,Verbose(TB)
> >
> > exten =>_X.,3,Answer()
> >
> > exten =>_X.,4,Set(mainLoop=0)
> >
> > ;exten =>_X.,5,Set(TIMEOUT(absolute)=5)
> >
> > exten =>_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
> >
> > exten => _X.,6,Dial(DAHDI/7/
> >
> > 09501032209,100,L(3[:1][:3000])g)
> >
> > exten =>_X.,7,Noop(${DIALEDTIME})
> >
> > exten =>_X.,8,Goto(TB,_X.,1)
> >
> > exten =>_X.,n,Hangup()
> >
> > Cheers
> > Vinod Dharashive
> > Sent from BlackBerry® on Airtel
> >
> > -Original Message-
> > From: Sam Govind 
> > Sender: asterisk-users-boun...@lists.digium.com
> > Date: Wed, 7 Sep 2011 11:53:33
> > To: Asterisk Users Mailing List - Non-Commercial Discussion<
> asterisk-users@lists.digium.com>
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> >   
> > Subject: Re: [asterisk-users] (no subject)
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> >http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > --
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> >http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
>
> --
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Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

[Ishfaq Malik]

Hi

Can you please provide an excerpt of your logs when this happens?

Regards

Ish

On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote:
> Hi sam,
> 
> Have solved the problem with your advice. Call drop in 10 seconds without 
> disconnecting a-party call. Thank you very much.
> 
> [TB]
> 
> exten =>_X.,1,Wait(${INCOMING_WAIT})
> 
> exten =>_X.,2,Verbose(TB)
> 
> exten =>_X.,3,Answer()
> 
> exten =>_X.,4,Set(mainLoop=0)
> 
> ;exten =>_X.,5,Set(TIMEOUT(absolute)=5)
> 
> exten =>_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
> 
> exten => _X.,6,Dial(DAHDI/7/
> 
> 09501032209,100,L(3[:1][:3000])g)
> 
> exten =>_X.,7,Noop(${DIALEDTIME})
> 
> exten =>_X.,8,Goto(TB,_X.,1)
> 
> exten =>_X.,n,Hangup()
> 
> Cheers
> Vinod Dharashive
> Sent from BlackBerry® on Airtel
> 
> -Original Message-
> From: Sam Govind 
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Wed, 7 Sep 2011 11:53:33 
> To: Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Subject: Re: [asterisk-users] (no subject)
> 
> --
> _
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] (no subject)

[Vinod Dharashive]

Hi sam,

Have solved the problem with your advice. Call drop in 10 seconds without 
disconnecting a-party call. Thank you very much.

[TB]

exten =>_X.,1,Wait(${INCOMING_WAIT})

exten =>_X.,2,Verbose(TB)

exten =>_X.,3,Answer()

exten =>_X.,4,Set(mainLoop=0)

;exten =>_X.,5,Set(TIMEOUT(absolute)=5)

exten =>_X.,5,Playback(/var/callagent/prompts/monitor/thanks)

exten => _X.,6,Dial(DAHDI/7/

09501032209,100,L(3[:1][:3000])g)

exten =>_X.,7,Noop(${DIALEDTIME})

exten =>_X.,8,Goto(TB,_X.,1)

exten =>_X.,n,Hangup()

Cheers
Vinod Dharashive
Sent from BlackBerry® on Airtel

-Original Message-
From: Sam Govind 
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 7 Sep 2011 11:53:33 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] (no subject)

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Re: [asterisk-users] Queue agent login notification

[Ishfaq Malik]

Hi

Is the MySQL database running?
Can you access the database from command line on that server?
Does the res_mysql.conf file have the same details to access the
database as you used from command line?

On Thu, 2011-09-08 at 22:25 +0300, Michael wrote:
> I changed to extconfig.conf to:
> [settings]
> queue_log => mysql,general
> 
> and I get this error:
>  MySQL RealTime: Failed to connect database server asterisk on
> localhost (err 2002). Check debug for more info.
> 
> Seems there's some progress...
> 
> On Thu, Sep 8, 2011 at 9:21 PM, Michael 
> wrote:
> I made test calls and go this:
>  res_config_mysql.c:789 store_mysql: MySQL RealTime: Invalid
> database specified: 'asterisk' (check res_mysql.conf)
> 
> In extconfig.conf I started with:
> [settings]
> queue_log => mysql,asterisk
> 
> 
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Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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