-- Added extension '2' priority 1 to demo -- Added extension '2' priority 2 to demo -- Added extension '3' priority 1 to demo -- Added extension '3' priority 2 to demo -- Added extension '1000' priority 1 to demo -- Added extension '1001' priority 1 to demo -- Added extension '1234' priority 1 to demo -- Added extension '1234' priority 2 to demo -- Added extension '1235' priority 1 to demo -- Added extension '1236' priority 1 to demo -- Added extension '1236' priority 2 to demo -- Added extension '#' priority 1 to demo -- Added extension '#' priority 2 to demo -- Added extension 't' priority 1 to demo -- Added extension 'i' priority 1 to demo -- Added extension '500' priority 1 to demo -- Added extension '500' priority 2 to demo -- Added extension '500' priority 3 to demo -- Added extension '500' priority 4 to demo -- Added extension '600' priority 1 to demo -- Added extension '600' priority 2 to demo -- Added extension '600' priority 3 to demo -- Added extension '600' priority 4 to demo -- Added extension '8500' priority 1 to demo -- Added extension '8500' priority 2 to demo -- Registered extension context 'default' -- Including context 'demo' in context 'default' -- Added extension '8500' priority 1 to default -- Added extension '8500' priority 2 to default -- Added extension '8600' priority 1 to default -- Registered extension context 'sip' -- Added extension '101' priority 1 to sip -- Added extension '102' priority 1 to sip -- Added extension '103' priority 1 to sip -- Added extension '104' priority 1 to sip -- Added extension '105' priority 1 to sip -- Added extension '106' priority 1 to sip -- Added extension '_X.' priority 1 to sip -- Added extension '8500' priority 1 to sip -- Added extension '8500' priority 2 to sip -- Added extension '8600' priority 1 to sip -- Added extension '_NXXXXXX' priority 1 to sip -- Added extension '_1NXXNXXXXXX' priority 1 to sip -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found -- Reloading module 'app_queue.so' (True Call Queueing) == Parsing '/etc/asterisk/queues.conf': Found -- Reloading module 'app_enumlookup.so' (ENUM Lookup) == Parsing '/etc/asterisk/enum.conf': Found -- Reloading module 'app_txtcidname.so' (TXTCIDName) == Parsing '/etc/asterisk/enum.conf': Found -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend) -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend) == Unregistered 'ODBC' CDR backend == Parsing '/etc/asterisk/cdr_odbc.conf': Found -- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend) == Unregistered 'pgsql' CDR backend == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Oct 22 16:04:20 ERROR[1076298368]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Un able to connect to database server localhost. Calls will not be logged! Oct 22 16:04:20 ERROR[1076298368]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Re ason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 3306? Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found -- Executing Dial("SIP/Phone2-a272", "Sip/g1/8178290827") in new stack Oct 22 16:05:06 WARNING[1118864576]: chan_sip.c:1375 create_addr: No such host: g1 Oct 22 16:05:06 NOTICE[1118864576]: app_dial.c:742 dial_exec: Unable to create c hannel of type 'Sip' == Everyone is busy/congested at this time Oct 22 16:05:16 WARNING[1118864576]: pbx.c:1933 ast_pbx_run: Timeout, but no rul e 't' in context 'sip' Oct 22 16:05:16 ERROR[1118864576]: cdr_pgsql.c:66 pgsql_log: cdr_pgsql: Unable t o connect to database server localhost. Calls will not be logged! Oct 22 16:05:16 ERROR[1118864576]: cdr_pgsql.c:67 pgsql_log: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 3306? reload == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/manager.conf': Found Oct 22 16:11:21 NOTICE[1076298368]: manager.c:1437 init_manager: Use of portno i n manager.conf deprecated. Please use 'port=5038' instead. == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 20000 -- Reloading module 'cdr_addon_mysql.so' (MySQL CDR Backend) == Unregistered 'mysql' CDR backend == Parsing '/etc/asterisk/cdr_mysql.conf': Not found (No such file or director y) Oct 22 16:11:21 WARNING[1076298368]: cdr_addon_mysql.c:264 my_load_module: Unabl e to load config for mysql CDR's: cdr_mysql.conf -- Reloading module 'res_adsi.so' (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) -- Reloading module 'res_indications.so' (Indications Configuration) -- Unregistered indication country 'cl' -- Unregistered indication country 'tw' Oct 22 16:11:21 NOTICE[1076298368]: indications.c:397 ast_unregister_indication_ country: Removed default indication country 'us' -- Unregistered indication country 'us' -- Unregistered indication country 'au' -- Unregistered indication country 'fr' -- Unregistered indication country 'de' -- Unregistered indication country 'nl' -- Unregistered indication country 'uk' -- Unregistered indication country 'fi' -- Unregistered indication country 'no' -- Unregistered indication country 'br' -- Unregistered indication country 'za' -- Unregistered indication country 'it' -- Unregistered indication country 'us-o' -- Unregistered indication country 'gr' -- Unregistered indication country 'ru' -- Unregistered indication country 'nz' == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Registered indication country 'us-o' -- Registered indication country 'gr' -- Registered indication country 'ru' -- Registered indication country 'nz' -- Setting default indication country to 'us' -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) -- Reloading module 'chan_agent.so' (Agent Proxy Channel) == Parsing '/etc/asterisk/agents.conf': Found -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP)) -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) == Parsing '/etc/asterisk/iax.conf': Found == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' -- Reloading module 'chan_local.so' (Local Proxy Channel) -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny )) -- Reloading module 'pbx_config.so' (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Setting global variable 'CONSOLE' to 'Console/dsp' -- Setting global variable 'IAXINFO' to 'guest' -- Setting global variable 'TRUNK' to 'Zap/g1' -- Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'iaxtel700' -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700 -- Registered extension context 'iaxprovider' -- Registered extension context 'trunkint' -- Added extension '_9011.' priority 1 to trunkint -- Added extension '_9011.' priority 2 to trunkint -- Registered extension context 'trunkld' -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld -- Registered extension context 'trunklocal' -- Added extension '_9NXXXXXX' priority 1 to trunklocal -- Added extension '_9NXXXXXX' priority 2 to trunklocal -- Registered extension context 'trunktollfree' -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree -- Added extension '_91800NXXXXXX' priority 2 to trunktollfree -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree -- Added extension '_91888NXXXXXX' priority 2 to trunktollfree -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree -- Added extension '_91877NXXXXXX' priority 2 to trunktollfree -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree -- Added extension '_91866NXXXXXX' priority 2 to trunktollfree -- Registered extension context 'international' -- Including context 'longdistance' in context 'international' -- Including context 'trunkint' in context 'international' -- Registered extension context 'longdistance' -- Including context 'local' in context 'longdistance' -- Including context 'trunkld' in context 'longdistance' -- Registered extension context 'local' -- Including context 'default' in context 'local' -- Including context 'parkedcalls' in context 'local' -- Including context 'trunklocal' in context 'local' -- Including context 'iaxtel700' in context 'local' -- Including context 'trunktollfree' in context 'local' -- Including context 'iaxprovider' in context 'local' -- Registered extension context 'macro-stdexten' -- Added extension 's' priority 1 to macro-stdexten -- Added extension 's' priority 2 to macro-stdexten -- Added extension 's-NOANSWER' priority 1 to macro-stdexten -- Added extension 's-NOANSWER' priority 2 to macro-stdexten -- Added extension 's-BUSY' priority 1 to macro-stdexten -- Added extension 's-BUSY' priority 2 to macro-stdexten -- Added extension '_s-.' priority 1 to macro-stdexten -- Added extension 'a' priority 1 to macro-stdexten -- Registered extension context 'demo' -- Including context 'sip' in context 'demo' -- Added extension 's' priority 1 to demo -- Added extension 's' priority 2 to demo -- Added extension 's' priority 3 to demo -- Added extension 's' priority 4 to demo -- Added extension 's' priority 5 to demo -- Added extension 's' priority 6 to demo -- Added extension '2' priority 1 to demo -- Added extension '2' priority 2 to demo -- Added extension '3' priority 1 to demo -- Added extension '3' priority 2 to demo -- Added extension '1000' priority 1 to demo -- Added extension '1001' priority 1 to demo -- Added extension '1234' priority 1 to demo -- Added extension '1234' priority 2 to demo -- Added extension '1235' priority 1 to demo -- Added extension '1236' priority 1 to demo -- Added extension '1236' priority 2 to demo -- Added extension '#' priority 1 to demo -- Added extension '#' priority 2 to demo -- Added extension 't' priority 1 to demo -- Added extension 'i' priority 1 to demo -- Added extension '500' priority 1 to demo -- Added extension '500' priority 2 to demo -- Added extension '500' priority 3 to demo -- Added extension '500' priority 4 to demo -- Added extension '600' priority 1 to demo -- Added extension '600' priority 2 to demo -- Added extension '600' priority 3 to demo -- Added extension '600' priority 4 to demo -- Added extension '8500' priority 1 to demo -- Added extension '8500' priority 2 to demo -- Registered extension context 'default' -- Including context 'demo' in context 'default' -- Added extension '8500' priority 1 to default -- Added extension '8500' priority 2 to default -- Added extension '8600' priority 1 to default -- Registered extension context 'sip' -- Added extension '101' priority 1 to sip -- Added extension '102' priority 1 to sip -- Added extension '103' priority 1 to sip -- Added extension '104' priority 1 to sip -- Added extension '105' priority 1 to sip -- Added extension '106' priority 1 to sip -- Added extension '_X.' priority 1 to sip -- Added extension '8500' priority 1 to sip -- Added extension '8500' priority 2 to sip -- Added extension '8600' priority 1 to sip -- Added extension '_NXXXXXX' priority 1 to sip -- Added extension '_1NXXNXXXXXX' priority 1 to sip -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found -- Reloading module 'app_queue.so' (True Call Queueing) == Parsing '/etc/asterisk/queues.conf': Found -- Reloading module 'app_enumlookup.so' (ENUM Lookup) == Parsing '/etc/asterisk/enum.conf': Found -- Reloading module 'app_txtcidname.so' (TXTCIDName) == Parsing '/etc/asterisk/enum.conf': Found -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend) -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend) == Unregistered 'ODBC' CDR backend == Parsing '/etc/asterisk/cdr_odbc.conf': Found -- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend) == Unregistered 'pgsql' CDR backend == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Oct 22 16:11:21 ERROR[1076298368]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Un able to connect to database server localhost. Calls will not be logged! Oct 22 16:11:21 ERROR[1076298368]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Re ason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 3306? Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found -- Executing Dial("SIP/Phone2-032e", "Zap/g1/8178290827") in new stack Oct 22 16:11:40 NOTICE[1118864576]: app_dial.c:742 dial_exec: Unable to create c hannel of type 'Zap' == Everyone is busy/congested at this time Oct 22 16:11:50 WARNING[1118864576]: pbx.c:1933 ast_pbx_run: Timeout, but no rul e 't' in context 'sip' Oct 22 16:11:50 ERROR[1118864576]: cdr_pgsql.c:66 pgsql_log: cdr_pgsql: Unable t o connect to database server localhost. Calls will not be logged! Oct 22 16:11:50 ERROR[1118864576]: cdr_pgsql.c:67 pgsql_log: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 3306? Chan Extension Context Language MusicOnHol pseudo sip 1 sip 2 sip 3 sip 4 sip 5 sip 6 sip 7 sip 8 sip 9 sip 10 sip 11 sip 12 sip 13 sip 14 sip 15 sip 16 sip 17 sip 18 sip 19 sip 20 sip 21 sip 22 sip 23 sip ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group => ,[,...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will have the ; d-channel for the trunk. ; backup1 is an optional list of backup d-channels. ; ;trunkgroup => 1,24,48 ; ; Spanmap: Associates a span with a trunk group ; spanmap => ,[,] ; ; zapspan is the zap span number to associate ; trunkgroup is the trunkgroup (specified above) for the mapping ; logicalspan is the logical span number within the trunk group to use. ; if unspecified, no logical span number is used. ; ;spanmap => 1,1,1 ;spanmap => 2,1,2 ;spanmap => 3,1,3 ;spanmap => 4,1,4 [channels] ; ; Default language ; ;language=en ; ; Default context ; switchtype=national context=sip signalling=pri_cpe group=1 channel => 1-23 ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: AT&T 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1: Old National ISDN 1 ; switchtype=national context=sip signalling=pri_cpe group=1 channel => 1-23 ; ; Some switches (AT&T especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;pridialplan=national ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;prilocaldialplan=national ; ; Overlap dialing mode (sending overlap digits) ; ;overlapdial=yes ; ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones ; ; priindication = outofband ; ; ; Signalling method (default is fxs). Valid values: ; em: E & M ; em_w: E & M Wink ; featd: Feature Group D (The fake, Adtran style, DTMF) ; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; featb: Feature Group B (MF (domestic, US)) ; fxs_ls: FXS (Loop Start) ; fxs_gs: FXS (Ground Start) ; fxs_ks: FXS (Kewl Start) ; fxo_ls: FXO (Loop Start) ; fxo_gs: FXO (Ground Start) ; fxo_ks: FXO (Kewl Start) ; pri_cpe: PRI signalling, CPE side ; pri_net: PRI signalling, Network side ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side ; sf: SF (Inband Tone) Signalling ; sf_w: SF Wink ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US)) ; The following are used for Radio interfaces: ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank) ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank) ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank) ; em_rx: Receive audio/COR on an E&M interface (1-way) ; em_tx: Transmit audio/PTT on an E&M interface (1-way) ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way) ; em_rxtx: same as em_txrx (for our dyslexic friends) ; sf_rx: Receive audio/COR on an SF interface (1-way) ; sf_tx: Transmit audio/PTT on an SF interface (1-way) ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way) ; sf_rxtx: same as sf_txrx (for our dyslexic friends) ; ;signalling=fxo_ls ; ; A variety of timing parameters can be specified as well ; Including: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes ; ; Whether or not to use caller ID ; usecallerid=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands ; ;cidsignalling=bell ; ; What signals the start of caller ID ; ring = a ring signals the start, polarity = polarity reversal signals the start ; ;cidstart=ring ; ; Whether or not to hide outgoing caller ID (Override with *67 or *82) ; hidecallerid=no ; ; Whether or not to enable call waiting on FXO lines ; callwaiting=yes ; ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user) ; Mostly use with FXS ports ; ;restrictcid=no ; ; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending ; usecallingpres=yes ; ; Support Caller*ID on Call Waiting ; callwaitingcallerid=yes ; ; Support three-way calling ; threewaycalling=yes ; ; Support flash-hook call transfer (requires three way calling) ; transfer=yes ; ; Support call forward variable ; cancallforward=yes ; ; Whether or not to support Call Return (*69) ; callreturn=yes ; ; Stutter dialtone support: If a mailbox is specified without a voicemail ; context, then when voicemail is received in a mailbox in the default ; voicemail context in voicemail.conf, taking the phone off hook will ; cause a stutter dialtone instead of a normal one. ; ; If a mailbox is specified *with* a voicemail context, the same will ; result if voicemail recieved in mailbox in the specified voicemail ; context ; ; for default voicemail context, the example below is fine: ; ;mailbox=1234 ; ; for any other voicemail context, the following will produce the ; stutter tone: ; ;mailbox=1234@context ; ; Enable echo cancellation ; Use either "yes", "no", or a power of two from 32 to 256 if you wish ; to actually set the number of taps of cancellation. ; echocancel=yes ; ; Generally, it is not necessary (and in fact undesirable) to echo cancel ; when the circuit path is entirely TDM. You may, however, reverse this ; behavior by enabling the echo cancel during pure TDM bridging below. ; echocancelwhenbridged=yes ; ; In some cases, the echo canceller doesn't train quickly enough and there ; is echo at the beginning of the call. Enabling echo training will cause ; asterisk to briefly mute the channel, send an impulse, and use the impulse ; response to pre-train the echo canceller so it can start out with a much ; closer idea of the actual echo. Value may be "yes", "no", or a number of ; milliseconds to delay before training (default = 400) ; ;echotraining=yes ;echotraining=800 ; ; If you are having trouble with DTMF detection, you can relax the ; DTMF detection parameters. Relaxing them may make the DTMF detector ; more likely to have "talkoff" where DTMF is detected when it ; shouldn't be. ; ;relaxdtmf=yes ; ; You may also set the default receive and transmit gains (in dB) ; rxgain=0.0 txgain=0.0 ; ; Logical groups can be assigned to allow outgoing rollover. Groups ; range from 0 to 31, and multiple groups can be specified. ; group=1 ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 ; ; Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no ; ; CallerID can be set to "asreceived" or a specific number ; if you want to override it. Note that "asreceived" only ; applies to trunk interfaces. ; ;callerid=2564286000 ; ; AMA flags affects the recording of Call Detail Records. If specified ; it may be 'default', 'omit', 'billing', or 'documentation'. ; ;amaflags=default ; ; Channels may be associated with an account code to ease ; billing ; ;accountcode=lss0101 ; ; ADSI (Analog Display Services Interface) can be enabled on a per-channel ; basis if you have (or may have) ADSI compatible CPE equipment ; ;adsi=yes ; ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D ; etc, it can be useful to perform busy detection either in an effort to ; detect hangup or for detecting busies ; ;busydetect=yes ; ; If busydetect is enabled, is also possible to specify how many ; busy tones to wait before hanging up. The default is 4, but ; better results can be achieved if set to 6 or even 8. Mind that ; higher the number, more time is needed to hangup a channel, but ; lower is probability to get random hangups ; ;busycount=4 ; ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. ; ; Few zones are supported at the time of this writing, but may ; be selected with "progzone" ; ; This feature can also easily detect false hangups. The symptoms of this ; is being disconnected in the middle of a call for no reason. ; ;callprogress=yes ;progzone=us ; ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF ; ;pulsedial=yes ; ; For fax detection, uncomment one of the following lines. The default is *OFF* ; ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ; ; Select which class of music to use for music on hold. If not specified ; then the default will be used. ; ;musiconhold=default ; ; PRI channels can have an idle extension and a minunused number. So long ; as at least "minunused" channels are idle, chan_zap will try to call ; "idledial" on them, and then dump them into the PBX in the "idleext" ; extension (which is of the form exten@context). When channels are needed ; the "idle" calls are disconnected (so long as there are at least "minidle" ; calls still running, of course) to make more channels available. The ; primary use of this is to create a dynamic service, where idle channels ; are bundled through multilink PPP, thus more efficiently utilizing ; combined voice/data services than conventional fixed mappings/muxings. ; ;idledial=6999 ;idleext=6999@dialout ;minunused=2 ;minidle=1 ; ; Configure jitter buffers in zapata (each one is 20ms, default is 4) ; ;jitterbuffers=4 ; ; You can define your own custom ring cadences here. You can define up to ; 8 pairs. If the silence is negative, it indicates where the callerid ; spill is to be placed. Also, if you define any custom cadences, the ; default cadences will be turned off. ; ; Syntax is: cadence=ring,silence[,ring,silence[...]] ; ; These are the default cadences: ; ;cadence=125,125,2000,-4000 ;cadence=250,250,500,1000,250,250,500,-4000 ;cadence=125,125,125,125,125,-4000 ;cadence=1000,500,2500,-5000 ; ; Each channel consists of the channel number or range. It ; inherits the parameters that were specified above its declaration ; ; For GR-303, CRV's are created like channels except they must start ; with the trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 ; ; ;callerid="Green Phone"<(256) 428-6121> ;channel => 1 ;callerid="Black Phone"<(256) 428-6122> ;channel => 2 ;callerid="CallerID Phone" <(256) 428-6123> ;callerid="CallerID Phone" <(630) 372-1564> ;callerid="CallerID Phone" <(256) 704-4666> ;channel => 3 ;callerid="Pac Tel Phone" <(256) 428-6124> ;channel => 4 ;callerid="Uniden Dead" <(256) 428-6125> ;channel => 5 ;callerid="Cortelco 2500" <(256) 428-6126> ;channel => 6 ;callerid="Main TA 750" <(256) 428-6127> ;channel => 44 ; ; For example, maybe we have some other channels ; which start out in a different context and use ; E & M signalling instead. ; ;context=remote ;sigalling=em ;channel => 15 ;channel => 16 ;signalling=em_w ; ; All those in group 0 I'll use for outgoing calls ; ; Strip most significant digit (9) before sending ; ;stripmsd=1 ;callerid=asreceived ;group=0 ;signalling=fxs_ls ;channel => 45 ;signalling=fxo_ls ;group=1 ;callerid="Joe Schmoe" <(256) 428-6131> ;channel => 25 ;callerid="Megan May" <(256) 428-6132> ;channel => 26 ;callerid="Suzy Queue" <(256) 428-6233> ;channel => 27 ;callerid="Larry Moe" <(256) 428-6234> ;channel => 28 ; ; Sample PRI (CPE) config: Specify the switchtype, the signalling as ; either pri_cpe or pri_net for CPE or Network termination, and generally ; you will want to create a single "group" for all channels of the PRI. ; ; switchtype = national ; signalling = pri_cpe ; group = 2 ; channel => 1-23 ; ; Used for distintive ring support for x100p. ; You can see the dringX patterns is to set any one of the dringXcontext fields ; and they will be printed on the console when an inbound call comes in. ; ;dring1=95,0,0 ;dring1context=internal1 ;dring2=325,95,0 ;dring2context=internal2 ; If no pattern is matched here is where we go. ;context=default ;channel => 1 Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=,,,,[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1 # # Note: "d4" could be referred to as "sf" or "superframe" # # The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1 # # E1's may have the additional keyword "crc4" to enable CRC4 checking # # If the keyword "yellow" follows, yellow alarm is transmitted when no # channels are open. # span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=,
,, # # Where is the name of the driver (e.g. eth),
is the # driver specific address (like a MAC for eth), is the number # of channels, and is a timing priority, like for a normal span. # use "0" to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # = # # Valid devices are: # # "e&m" : Channel(s) are signalled using E&M signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # "fxsls" : Channel(s) are signalled using FXS Loopstart protocol. # "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol. # "fxsks" : Channel(s) are signalled using FXS Koolstart protocol. # "fxols" : Channel(s) are signalled using FXO Loopstart protocol. # "fxogs" : Channel(s) are signalled using FXO Groundstart protocol. # "fxoks" : Channel(s) are signalled using FXO Koolstart protocol. # "sf" : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: # channel# => sf:,,,,, # rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # "unused" : No signalling is performed, each channel in the list remains idle # "clear" : Channel(s) are bundled into a single span. No conversion or # signalling is performed, and raw data is available on the master. # "indclear": Like "clear" except all channels are treated individually and # are not bundled. "bchan" is an alias for this. # "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the # bundle, and the resulting data is communicated via the master # device. # "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the # bundle and also performs incoming and outgoing FCS insertion # and verification. "dchan" is an alias for this. # "nethdlc" : The zaptel driver bundles the channels together into an # hdlc network device, which in turn can be configured with # sethdlc (available separately). # "dacs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon # "dacsrbs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon and # also performs the DACSing of RBS bits # # The channel list is a comma-separated list of channels or ranges, for # example: # # 1,3,5 (channels one, three, and five) # 16-23, 29 (channels 16 through 23, as well as channel 29 # # So, some complete examples are: # e&m=1-12 # nethdlc=13-24 # fxsls=25,26,27,28 # fxols=29-32 # #fxoks=1-24 #bchan=25-47 #dchan=48 #fxols=1-12 #fxols=13-24 #e&m=25-29 #nethdlc=30-33 #clear=44 #clear=45 #clear=46 #clear=47 #fcshdlc=48 #dacs=1-24:48 #dacsrbs=1-24:48 # # Finally, you can preload some tone zones, to prevent them from getting # overwritten by other users (if you allow non-root users to open /dev/zap/* # interfaces anyway. Also this means they won't have to be loaded at runtime. # The format is "loadzone=" where the zone is a two letter country code. # # You may also specify a default zone with "defaultzone=" where zone # is a two letter country code. # loadzone = us #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=us