<p>Hi!</p>
<p>Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone directories, and if so, how?</p>
<p>Thanks in advance!</p>
<div class="gmail_quote">On Feb 17, 2013 6:40 PM, <<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<br>Today's Topics:<br>
<br>
1. Re: ODBC and SQLIte3 (Yves A.)<br>
2. Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing<br>
on incoming calls (Administrator TOOTAI)<br>
3. Re: Asterisk 1.8, Siemens C610IP with 3 handsets: all are<br>
ringing on incoming calls (Chris Bagnall)<br>
<br><br>---------- Forwarded message ----------<br>From: "Yves A." <<a href="mailto:yves030@gmx.de">yves030@gmx.de</a>><br>To: Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Cc: <br>Date: Sun, 17 Feb 2013 15:07:43 +0100<br>Subject: Re: [asterisk-users] ODBC and SQLIte3<br>
<div text="#000000" bgcolor="#FFFFFF">
<div>looks like a mistake in your
extconfig.conf...<br>
do you want to use realtime extensions too? <br>
<br>
for further instructions show us your extensions.conf and the
verbose output of the cli showing the dialattempt...<br>
<br>
regards,<br>
yves<br>
<br>
Am 17.02.2013 14:31, schrieb termo termosel:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi,<br>
<br>
I have add this options into Sip.conf but the CLI continues
telling the same message:<br>
<br>
ubuntu*CLI> sip show peers<br>
Name/username Host Dyn Forcerport ACL Port Status Description
Realtime<br>
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0
online, 0 offline]<br>
<br>
I have two users in my slite3.db. but Asterisk doesn't show
me. It is how asterisk can't access into this database.<br>
<br>
When I go to call, Asterisk tells me that extension xxx is not
found in phones context.<br>
<br>
Thanks,<br>
Jordi <br>
<div>
<hr>Date: Sun, 17 Feb 2013 13:00:44 +0100<br>
From: <a href="mailto:yves030@gmx.de" target="_blank">yves030@gmx.de</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] ODBC and SQLIte3<br>
<br>
<div>hi,<br>
<br>
if you use realtime peers, and you want to see their states,
you have to look in the database...<br>
if you want to see their states via cli, you have to set
rtcachefriends=yes in your sip.conf...<br>
there are other settings that you might be interested in...
:<br>
<br>
<small><small><font face="Courier New, Courier, monospace">rtcachefriends=yes
; Cache realtime friends by adding them to the
internal list<br>
; just like friends
added from the config file only on a<br>
; as-needed basis?
(yes|no)<br>
<br>
rtsavesysname=yes ; Save systemname in
realtime database at registration<br>
; Default= no<br>
<br>
rtupdate=yes ; Send registry updates
to database using realtime? (yes|no)<br>
; If set to yes, when
a SIP UA registers successfully, the ip address,<br>
; the origination
port, the registration period, and the username of<br>
; the UA will be set
to database via realtime.<br>
; If not present,
defaults to 'yes'. Note: realtime peers will<br>
; probably not
function across reloads in the way that you expect, if<br>
; you turn this option
off.<br>
rtautoclear=yes ; Auto-Expire friends
created on the fly on the same schedule<br>
; as if it had just
registered? (yes|no|<seconds>)<br>
; If set to yes, when
the registration expires, the friend will<br>
; vanish from the
configuration until requested again. If set<br>
; to an integer,
friends expire within this number of seconds<br>
; instead of the
registration interval.<br>
<br>
ignoreregexpire=yes ; Enabling this setting
has two functions:<br>
;<br>
; For non-realtime
peers, when their registration expires, the<br>
; information will
_not_ be removed from memory or the Asterisk database<br>
; if you attempt to
place a call to the peer, the existing information<br>
; will be used in
spite of it having expired<br>
;<br>
; For realtime peers,
when the peer is retrieved from realtime storage,<br>
; the registration
information will be used regardless of whether<br>
; it has expired or
not; if it expires while the realtime peer<br>
; is still in memory
(due to caching or other reasons), the<br>
; information will not
be removed from realtime storage<br>
</font></small></small><br>
regards,<br>
yves<br>
<br>
<br>
Am 17.02.2013 12:51, schrieb termo termosel:<br>
</div>
<blockquote>
<div dir="ltr"> Hi,<br>
<br>
I had configured Asterisk to use default database located
in /var/lib/asterisk/sqlite3dir/sqlite3.db. When I put
odbc show in Asterisk's cli, It returns me that I have
conected but when I put "sip show peers",Asterisk doesn't
found any peer or user.<br>
<br>
ubuntu*CLI> odbc show<br>
<br>
ODBC DSN Settings<br>
-----------------<br>
<br>
Name: asterisk<br>
DSN: asterisk-connector<br>
Last connection attempt: 1970-01-01 01:00:00<br>
Pooled: No<br>
Connected: Yes<br>
<br>
ubuntu*CLI> sip show peers<br>
Name/username
Host Dyn Forcerport ACL
Port Status Description
Realtime<br>
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0
online, 0 offline]<br>
<br>
<br>
This mi configuration,<br>
<br>
/etc/odbci.ini<br>
<br>
[asterisk-connector]<br>
Description = SQLite3 database <br>
Driver = SQLite3<br>
Database =
/var/lib/asterisk/sqlite3dir/sqlite3.db<br>
<br>
/etc/odbcinst.ini<br>
<br>
[SQLite3]<br>
Description= SQLite3 ODBC Driver<br>
Driver=/usr/local/lib/libsqlite3odbc.so<br>
Setup=/usr/local/lib/libsqlite3odbc.so<br>
Threading=2<br>
<br>
/etc/asterisk/extconfig.conf<br>
<br>
[settings]<br>
<br>
sipusers => odbc,asterisk,sip_buddies<br>
sippeers => odbc,asterisk,sip_buddies<br>
sipregs => odbc,asterisk,sip_buddies<br>
<br>
/etc/asterisk/func_odbc.conf<br>
<br>
[SQL]<br>
dsn=asterisk<br>
readsql=${ARG1}<br>
<br>
/etc/asterisk/modules.conf<br>
<br>
autoload=yes<br>
;preload => res_odbc.so<br>
;preload => res_config_odbc.so<br>
noload => pbx_gtkconsole.so<br>
;load => pbx_gtkconsole.so<br>
noload => pbx_kdeconsole.so<br>
noload => app_intercom.so<br>
noload => chan_modem.so<br>
noload => chan_modem_aopen.so<br>
noload => chan_modem_bestdata.so<br>
noload => chan_modem_i4l.so<br>
noload => chan_capi.so<br>
load => res_musiconhold.so<br>
noload => chan_alsa.so<br>
;noload => chan_oss.so<br>
noload => cdr_sqlite.so<br>
noload => app_directory_odbc.so<br>
;noload => res_config_odbc.so<br>
;noload => res_config_pgsql.so<br>
<br>
/etc/asterisk/res_odbc.conf<br>
<br>
[asterisk]<br>
enabled => yes<br>
dsn => asterisk-connector<br>
pre-connect => yes<br>
<br>
<br>
Can someone help me?<br>
<br>
Thanks,<br>
Jordi<br>
</div>
<br>
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<br><br>---------- Forwarded message ----------<br>From: Administrator TOOTAI <<a href="mailto:admin@tootai.net">admin@tootai.net</a>><br>To: Asterisk-Users <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Cc: <br>Date: Sun, 17 Feb 2013 18:02:24 +0100<br>Subject: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls<br>Hi everybody,<br>
<br>
We installed the Gigaset C610IP to one of our customer, those phone are natted and connects to our Asterisk 1.8.19. Each handset has is own account on our Asterisk, lets say Handset_102, Handset_103 and Handset_104.<br>
<br>
Problem is this one, taken from a sip show peers:<br>
<br>
customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms)<br>
customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms)<br>
customer104/Handset_104 xxx.yyy.zzz.153 D N 5062 OK (66 ms)<br>
<br>
As you see, all handsets are identified with the same port, which means that on incoming call to one handset or when transfering a call with the asterisk transfer feature, all 3 handsets are ringing :-(<br>
<br>
We tried using fixed port (sample above with port 5062) as well as random, no changes.<br>
<br>
We know that few of you are using those phones, how did you manage to solve this problem? Would be great if you could share.<br>
<br>
Regards<br>
<br>
-- <br>
Daniel<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Chris Bagnall <<a href="mailto:asterisk@lists.minotaur.cc">asterisk@lists.minotaur.cc</a>><br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Cc: <br>Date: Sun, 17 Feb 2013 17:27:38 +0000<br>Subject: Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls<br>On 17/2/13 5:02 pm, Administrator TOOTAI wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
customer102/Handset_102 xxx.yyy.zzz.153 D<br>
N 5062 OK (80 ms)<br>
customer103/Handset_103 xxx.yyy.zzz.153 D<br>
N 5062 OK (70 ms)<br>
customer104/Handset_104 xxx.yyy.zzz.153 D N 5062<br>
OK (66 ms)<br>
</blockquote>
<br>
That's perfectly normal with these phones, and shouldn't pose a problem.<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
As you see, all handsets are identified with the same port, which means<br>
that on incoming call to one handset or when transfering a call with the<br>
asterisk transfer feature, all 3 handsets are ringing :-(<br>
</blockquote>
<br>
You can specify which SIP account correlates to each handset in the Gigaset web interface.<br>
<br>
Go to Settings -> Telephony -> Number Assignment<br>
You want Handset 1 to use Connection 'Handset_102' for outgoing calls and for incoming calls (untick everything else except this for incoming calls).<br>
Likewise Handset 2 should use Connection 'Handset_103' for outgoing and incoming (again, untick everything but this option).<br>
<br>
Rinse and repeat for other handsets.<br>
<br>
I can confirm it does work properly - we have dozens of clients with Gigaset phones and separate SIP registrations per handset.<br>
<br>
Kind regards,<br>
<br>
Chris<br>
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