[asterisk-users] Calls from talkonaut to pstn Phone
Darin Iv
adariniv at gmail.com
Sat Sep 8 13:12:11 CDT 2012
Hi I need to call from talkonaut to PSTN number using our asterisk server.
VoIP service which allows utilizing GPRS, 3G or WIFI mobile data
connections to make free of cheap VoIP calls to phones through my SIP
network, to Google Talk, MSN, AIM or Yahoo through GTalk2VoIP soft-switch
in which I can set up on my asterisk server in INDIA servers to deliver
VoIP calls from branded-Talkonaut users to your SIP network, to VoIM
clients and vice versa.
I have configured like this
Dial plan (extension.conf) is
[gtalk_incoming]
exten => s,1,Verbose(2,Incoming Gtalk call from ${CALLERID(all)})
same => n,Answer()
same => n,Dial(SIP/1000,30)
same => n,Hangup()
[google_out]
exten => 1010,1,Verbose(2,Extension 1010 calling darinfaststream at gmail.com)
same => n,Dial(Gtalk/asterisk/adariniv at gmail.com,30)
same => n,Hangup()
[LocalSets]
exten => XXXXXXXXXX,1,Verbose(2,Placing call to ${EXTEN} via Google Voice)
same => n,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same => n,Hangup()
Jabber.conf
[general]
debug=no
autoprune=no
autoregister=yes
[username]
type=client
serverhost=talk.google.com
username=darinfaststream at gmail.com/Talk
secret=xxxxx
priority=1
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage="I am an Asterisk Server"
timeout=100
keepalive=yes
Gtalk.conf
[guest]
disallow=all
allow=ulaw
context=google-in
[buddy]
username=my at gmail.com
disallow=all
allow=ulaw
context=google-in
connection=gtalk_account
But when I tried to make call from talkonaut after configure the own dial
plan in talkonaut voip settings I cant make calls through asterisk or vice
versa.
On Sat, Sep 8, 2012 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote:
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> Today's Topics:
>
> 1. Automatic reconnect to ODBC sources? (Stefan at WPF)
> 2. Rolm T1 not passing caller ID to asterisk (Kohler, Ed)
> 3. how to load our own .wav sound files in the dial plans for
> playback (upendra)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 7 Sep 2012 19:17:58 +0200
> From: Stefan at WPF <stefan.at.wpf at googlemail.com>
> Subject: [asterisk-users] Automatic reconnect to ODBC sources?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <CADBM-whttJf1uMs1xj8V=
> PwAALHGKWnqJj_qkSmi+Z4qDgxjog at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I am using Asterisk 1.8.10.1 on Ubuntu Server 12.04. I use MySQL to store
> CDR records using cdr_adaptive_odbc.
> The problem: When the MySQL Server fails for whatever reason, Asterisk
> never reconnects automatically! So I loose all CDR informations even after
> the MySQL server works again.
> isql (used as ODBC testtool) tells me, that ODBC reconnects or does a new
> connection on request, so ODBC seems not to be the problem, but Asterisk.
> Is there any automatic reconnect option on Asterisk concercing ODBC /
> cdr_adaptive_odbc? My complete configuration including ODBC is as follows:
>
> *ODBC:
>
> /etc/odbcinst.ini
> *
> >
> > [MySQL]
> > Description = ODBC for MySQL
> > Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
> > Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
> > FileUsage = 1
> > OPTION = 4194304
> >
> >
> */etc/odbc.ini *
>
> > [asterisk-connector]
> > Description = MySQL connection to 'asterisk' database
> > Driver = MySQL
> > Database = asterisk
> > Server = localhost
> > UserName = root
> > Password = password
> > Port = 3306
>
> *
>
> ASTERISK CDR*:
>
> */etc/asterisk/res_odbc.conf*
>
> > [asterisk]
> > enabled => yes
> > dsn => asterisk-connector
> > username => root
> > password => password
> > pre-connect => yes
> > pooling => no
> > limit => 1
> > connect_timeout => 1
> > idlecheck => 1
> >
>
> */etc/asterisk/cdr_adaptive_odbc.conf *
>
> > [mytable]
> > connection = asterisk
> > table = asterisk_cdr
> >
>
> */etc/asterisk/cdr.conf*
> >
> > [general]
> > enable = yes
> > unanswered = yes
> >
>
>
> Thanks for any hint!
>
> Best regards
> Stefan
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> ------------------------------
>
> Message: 2
> Date: Fri, 7 Sep 2012 14:49:21 -0400
> From: "Kohler, Ed" <ekohler at siena.edu>
> Subject: [asterisk-users] Rolm T1 not passing caller ID to asterisk
> To: "'asterisk-users at lists.digium.com'"
> <asterisk-users at lists.digium.com>
> Cc: "Berman, Mark" <mberman at siena.edu>, "McHugh, Kim"
> <kmchugh at siena.edu>
> Message-ID:
> <DE01B7D81028C145884D624C2D361F710137DDEE3B20 at mb-2.siena.edu>
> Content-Type: text/plain; charset="us-ascii"
>
> We have a Rolm 9751 connecter to our asterisk box via a straight T1. The
> Rolm cannot do PRI. Has anyone figured out how to configure this link
> (probably on the Rolm side) to pass caller ID? Any Help or suggestions,
> aside from forklifting the Rolm, would be appreciated.
>
> Thanks,
>
> Edward Kohler
> Network Technician
> 101Hines Hall
> Siena College
> 515 Loudon Rd.
> Loudonville, NY 12211
> 518-783-2391
> Fax 518-783-2590
> ekohler at siena.edu<mailto:ekohler at siena.edu>
>
> Siena College is a learning community advancing the ideals of a liberal
> arts education, rooted in its identity as a Franciscan and Catholic
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> ------------------------------
>
> Message: 3
> Date: Sat, 8 Sep 2012 14:19:51 +0530
> From: upendra <uppi.me at gmail.com>
> Subject: [asterisk-users] how to load our own .wav sound files in the
> dial plans for playback
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <CAFogui8j2Ny3wChAu-t29PyPT=CnZqpedtAJzDn3R=
> Lrc+AKgw at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> i am trying to add my own sound file in the asterisk dial plan extension
> for playback option , i dont no where to put the file and how to give the
> path in extension file and all so is need that the sound file should be
> convert in asterisk as .wav file???
>
>
> regards
> Upendra
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