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.<br> I have configured like this <br><br>Dial plan (extension.conf) is<br><br>[gtalk_incoming]<br>exten => s,1,Verbose(2,Incoming Gtalk call from ${CALLERID(all)})<br> same => n,Answer()<br> same => n,Dial(SIP/1000,30)<br>
same => n,Hangup()<br><br><br>[google_out]<br>exten => 1010,1,Verbose(2,Extension 1010 calling <a href="mailto:darinfaststream@gmail.com">darinfaststream@gmail.com</a>)<br>same => n,Dial(Gtalk/asterisk/<a href="mailto:adariniv@gmail.com">adariniv@gmail.com</a>,30)<br>
same => n,Hangup()<br><br>[LocalSets]<br>exten => XXXXXXXXXX,1,Verbose(2,Placing call to ${EXTEN} via Google Voice)<br>same => n,Dial(Gtalk/asterisk/+${<a href="mailto:EXTEN%7D@voice.google.com">EXTEN}@voice.google.com</a>)<br>
same => n,Hangup()<br><br><br>Jabber.conf<br><br>[general]<br>debug=no<br>autoprune=no<br>autoregister=yes<br>[username]<br>type=client<br>serverhost=<a href="http://talk.google.com">talk.google.com</a><br>username=<a href="http://darinfaststream@gmail.com/Talk">darinfaststream@gmail.com/Talk</a><br>
secret=xxxxx<br>priority=1<br>port=5222<br>usetls=yes<br>usesasl=yes<br>status=Available<br>statusmessage="I am an Asterisk Server"<br>timeout=100<br>keepalive=yes<br><br>Gtalk.conf<br><br>[guest]<br>disallow=all<br>
allow=ulaw<br>context=google-in<br><br>[buddy]<br>username=<a href="mailto:my@gmail.com">my@gmail.com</a><br>disallow=all<br>allow=ulaw<br>context=google-in<br>connection=gtalk_account<br><br><br>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.<br>
<br><div class="gmail_quote">On Sat, Sep 8, 2012 at 10:30 PM, <span dir="ltr"><<a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a>></span> wrote:<br>
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Today's Topics:<br>
<br>
1. Automatic reconnect to ODBC sources? (Stefan at WPF)<br>
2. Rolm T1 not passing caller ID to asterisk (Kohler, Ed)<br>
3. how to load our own .wav sound files in the dial plans for<br>
playback (upendra)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Fri, 7 Sep 2012 19:17:58 +0200<br>
From: Stefan at WPF <<a href="mailto:stefan.at.wpf@googlemail.com">stefan.at.wpf@googlemail.com</a>><br>
Subject: [asterisk-users] Automatic reconnect to ODBC sources?<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<CADBM-whttJf1uMs1xj8V=<a href="mailto:PwAALHGKWnqJj_qkSmi%2BZ4qDgxjog@mail.gmail.com">PwAALHGKWnqJj_qkSmi+Z4qDgxjog@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
I am using Asterisk 1.8.10.1 on Ubuntu Server 12.04. I use MySQL to store<br>
CDR records using cdr_adaptive_odbc.<br>
The problem: When the MySQL Server fails for whatever reason, Asterisk<br>
never reconnects automatically! So I loose all CDR informations even after<br>
the MySQL server works again.<br>
isql (used as ODBC testtool) tells me, that ODBC reconnects or does a new<br>
connection on request, so ODBC seems not to be the problem, but Asterisk.<br>
Is there any automatic reconnect option on Asterisk concercing ODBC /<br>
cdr_adaptive_odbc? My complete configuration including ODBC is as follows:<br>
<br>
*ODBC:<br>
<br>
/etc/odbcinst.ini<br>
*<br>
><br>
> [MySQL]<br>
> Description = ODBC for MySQL<br>
> Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so<br>
> Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so<br>
> FileUsage = 1<br>
> OPTION = 4194304<br>
><br>
><br>
*/etc/odbc.ini *<br>
<br>
> [asterisk-connector]<br>
> Description = MySQL connection to 'asterisk' database<br>
> Driver = MySQL<br>
> Database = asterisk<br>
> Server = localhost<br>
> UserName = root<br>
> Password = password<br>
> Port = 3306<br>
<br>
*<br>
<br>
ASTERISK CDR*:<br>
<br>
*/etc/asterisk/res_odbc.conf*<br>
<br>
> [asterisk]<br>
> enabled => yes<br>
> dsn => asterisk-connector<br>
> username => root<br>
> password => password<br>
> pre-connect => yes<br>
> pooling => no<br>
> limit => 1<br>
> connect_timeout => 1<br>
> idlecheck => 1<br>
><br>
<br>
*/etc/asterisk/cdr_adaptive_odbc.conf *<br>
<br>
> [mytable]<br>
> connection = asterisk<br>
> table = asterisk_cdr<br>
><br>
<br>
*/etc/asterisk/cdr.conf*<br>
><br>
> [general]<br>
> enable = yes<br>
> unanswered = yes<br>
><br>
<br>
<br>
Thanks for any hint!<br>
<br>
Best regards<br>
Stefan<br>
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Message: 2<br>
Date: Fri, 7 Sep 2012 14:49:21 -0400<br>
From: "Kohler, Ed" <<a href="mailto:ekohler@siena.edu">ekohler@siena.edu</a>><br>
Subject: [asterisk-users] Rolm T1 not passing caller ID to asterisk<br>
To: "'<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>'"<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Cc: "Berman, Mark" <<a href="mailto:mberman@siena.edu">mberman@siena.edu</a>>, "McHugh, Kim"<br>
<<a href="mailto:kmchugh@siena.edu">kmchugh@siena.edu</a>><br>
Message-ID:<br>
<<a href="mailto:DE01B7D81028C145884D624C2D361F710137DDEE3B20@mb-2.siena.edu">DE01B7D81028C145884D624C2D361F710137DDEE3B20@mb-2.siena.edu</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
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.<br>
<br>
Thanks,<br>
<br>
Edward Kohler<br>
Network Technician<br>
101Hines Hall<br>
Siena College<br>
515 Loudon Rd.<br>
Loudonville, NY 12211<br>
518-783-2391<br>
Fax 518-783-2590<br>
<a href="mailto:ekohler@siena.edu">ekohler@siena.edu</a><mailto:<a href="mailto:ekohler@siena.edu">ekohler@siena.edu</a>><br>
<br>
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<br>
Message: 3<br>
Date: Sat, 8 Sep 2012 14:19:51 +0530<br>
From: upendra <<a href="mailto:uppi.me@gmail.com">uppi.me@gmail.com</a>><br>
Subject: [asterisk-users] how to load our own .wav sound files in the<br>
dial plans for playback<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<CAFogui8j2Ny3wChAu-t29PyPT=CnZqpedtAJzDn3R=<a href="mailto:Lrc%2BAKgw@mail.gmail.com">Lrc+AKgw@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hi,<br>
<br>
i am trying to add my own sound file in the asterisk dial plan extension<br>
for playback option , i dont no where to put the file and how to give the<br>
path in extension file and all so is need that the sound file should be<br>
convert in asterisk as .wav file???<br>
<br>
<br>
regards<br>
Upendra<br>
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</blockquote></div><br>