Sina Owolabi
2013-Feb-17 19:09 UTC
[asterisk-users] Can Cisco 5XX phones share asterisk phone directory?
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:> Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > 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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New > to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ---------- 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 > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... 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