[asterisk-users] Can Cisco 5XX phones share asterisk phone directory?
Sina Owolabi
shinacalypse at gmail.com
Sun Feb 17 13:09:43 CST 2013
Hi!
Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone
directories, and if so, how?
Thanks in advance!
On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: ODBC and SQLIte3 (Yves A.)
> 2. Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing
> on incoming calls (Administrator TOOTAI)
> 3. Re: Asterisk 1.8, Siemens C610IP with 3 handsets: all are
> ringing on incoming calls (Chris Bagnall)
>
>
> ---------- Forwarded message ----------
> From: "Yves A." <yves030 at gmx.de>
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> Cc:
> Date: Sun, 17 Feb 2013 15:07:43 +0100
> Subject: Re: [asterisk-users] ODBC and SQLIte3
> looks like a mistake in your extconfig.conf...
> do you want to use realtime extensions too?
>
> for further instructions show us your extensions.conf and the verbose
> output of the cli showing the dialattempt...
>
> regards,
> yves
>
> Am 17.02.2013 14:31, schrieb termo termosel:
>
> Hi,
>
> I have add this options into Sip.conf but the CLI continues telling the
> same message:
>
> ubuntu*CLI> sip show peers
> Name/username Host Dyn Forcerport ACL Port Status Description Realtime
> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
> offline]
>
> I have two users in my slite3.db. but Asterisk doesn't show me. It is how
> asterisk can't access into this database.
>
> When I go to call, Asterisk tells me that extension xxx is not found in
> phones context.
>
> Thanks,
> Jordi
> ------------------------------
> Date: Sun, 17 Feb 2013 13:00:44 +0100
> From: yves030 at gmx.de
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] ODBC and SQLIte3
>
> hi,
>
> if you use realtime peers, and you want to see their states, you have to
> look in the database...
> if you want to see their states via cli, you have to set
> rtcachefriends=yes in your sip.conf...
> there are other settings that you might be interested in... :
>
> rtcachefriends=yes ; Cache realtime friends by adding them to
> the internal list
> ; just like friends added from the config
> file only on a
> ; as-needed basis? (yes|no)
>
> rtsavesysname=yes ; Save systemname in realtime database at
> registration
> ; Default= no
>
> rtupdate=yes ; Send registry updates to database using
> realtime? (yes|no)
> ; If set to yes, when a SIP UA registers
> successfully, the ip address,
> ; the origination port, the registration
> period, and the username of
> ; the UA will be set to database via
> realtime.
> ; If not present, defaults to 'yes'. Note:
> realtime peers will
> ; probably not function across reloads in
> the way that you expect, if
> ; you turn this option off.
> rtautoclear=yes ; Auto-Expire friends created on the fly on
> the same schedule
> ; as if it had just registered?
> (yes|no|<seconds>)
> ; If set to yes, when the registration
> expires, the friend will
> ; vanish from the configuration until
> requested again. If set
> ; to an integer, friends expire within
> this number of seconds
> ; instead of the registration interval.
>
> ignoreregexpire=yes ; Enabling this setting has two functions:
> ;
> ; For non-realtime peers, when their
> registration expires, the
> ; information will _not_ be removed from
> memory or the Asterisk database
> ; if you attempt to place a call to the
> peer, the existing information
> ; will be used in spite of it having
> expired
> ;
> ; For realtime peers, when the peer is
> retrieved from realtime storage,
> ; the registration information will be
> used regardless of whether
> ; it has expired or not; if it expires
> while the realtime peer
> ; is still in memory (due to caching or
> other reasons), the
> ; information will not be removed from
> realtime storage
>
> regards,
> yves
>
>
> Am 17.02.2013 12:51, schrieb termo termosel:
>
> Hi,
>
> 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.
>
> ubuntu*CLI> odbc show
>
> ODBC DSN Settings
> -----------------
>
> Name: asterisk
> DSN: asterisk-connector
> Last connection attempt: 1970-01-01 01:00:00
> Pooled: No
> Connected: Yes
>
> ubuntu*CLI> sip show peers
> Name/username Host Dyn
> Forcerport ACL Port Status Description
> Realtime
> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
> offline]
>
>
> This mi configuration,
>
> /etc/odbci.ini
>
> [asterisk-connector]
> Description = SQLite3 database
> Driver = SQLite3
> Database = /var/lib/asterisk/sqlite3dir/sqlite3.db
>
> /etc/odbcinst.ini
>
> [SQLite3]
> Description= SQLite3 ODBC Driver
> Driver=/usr/local/lib/libsqlite3odbc.so
> Setup=/usr/local/lib/libsqlite3odbc.so
> Threading=2
>
> /etc/asterisk/extconfig.conf
>
> [settings]
>
> sipusers => odbc,asterisk,sip_buddies
> sippeers => odbc,asterisk,sip_buddies
> sipregs => odbc,asterisk,sip_buddies
>
> /etc/asterisk/func_odbc.conf
>
> [SQL]
> dsn=asterisk
> readsql=${ARG1}
>
> /etc/asterisk/modules.conf
>
> autoload=yes
> ;preload => res_odbc.so
> ;preload => res_config_odbc.so
> noload => pbx_gtkconsole.so
> ;load => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> noload => app_intercom.so
> noload => chan_modem.so
> noload => chan_modem_aopen.so
> noload => chan_modem_bestdata.so
> noload => chan_modem_i4l.so
> noload => chan_capi.so
> load => res_musiconhold.so
> noload => chan_alsa.so
> ;noload => chan_oss.so
> noload => cdr_sqlite.so
> noload => app_directory_odbc.so
> ;noload => res_config_odbc.so
> ;noload => res_config_pgsql.so
>
> /etc/asterisk/res_odbc.conf
>
> [asterisk]
> enabled => yes
> dsn => asterisk-connector
> pre-connect => yes
>
>
> Can someone help me?
>
> Thanks,
> Jordi
>
>
> --
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>
>
> ---------- Forwarded message ----------
> From: Administrator TOOTAI <admin at tootai.net>
> To: Asterisk-Users <asterisk-users at lists.digium.com>
> Cc:
> Date: Sun, 17 Feb 2013 18:02:24 +0100
> Subject: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets:
> all are ringing on incoming calls
> Hi everybody,
>
> 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.
>
> Problem is this one, taken from a sip show peers:
>
> customer102/Handset_102 xxx.yyy.zzz.153 D N
> 5062 OK (80 ms)
> customer103/Handset_103 xxx.yyy.zzz.153 D N
> 5062 OK (70 ms)
> customer104/Handset_104 xxx.yyy.zzz.153 D
> N 5062 OK (66 ms)
>
> 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 :-(
>
> We tried using fixed port (sample above with port 5062) as well as random,
> no changes.
>
> 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.
>
> Regards
>
> --
> Daniel
>
>
>
>
> ---------- Forwarded message ----------
> From: Chris Bagnall <asterisk at lists.minotaur.cc>
> To: asterisk-users at lists.digium.com
> Cc:
> Date: Sun, 17 Feb 2013 17:27:38 +0000
> Subject: Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3
> handsets: all are ringing on incoming calls
> On 17/2/13 5:02 pm, Administrator TOOTAI wrote:
>
>> customer102/Handset_102 xxx.yyy.zzz.153 D
>> N 5062 OK (80 ms)
>> customer103/Handset_103 xxx.yyy.zzz.153 D
>> N 5062 OK (70 ms)
>> customer104/Handset_104 xxx.yyy.zzz.153 D N 5062
>> OK (66 ms)
>>
>
> That's perfectly normal with these phones, and shouldn't pose a problem.
>
> 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 :-(
>>
>
> You can specify which SIP account correlates to each handset in the
> Gigaset web interface.
>
> Go to Settings -> Telephony -> Number Assignment
> You want Handset 1 to use Connection 'Handset_102' for outgoing calls and
> for incoming calls (untick everything else except this for incoming calls).
> Likewise Handset 2 should use Connection 'Handset_103' for outgoing and
> incoming (again, untick everything but this option).
>
> Rinse and repeat for other handsets.
>
> I can confirm it does work properly - we have dozens of clients with
> Gigaset phones and separate SIP registrations per handset.
>
> Kind regards,
>
> Chris
> --
> This email is made from 100% recycled electrons
>
>
>
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