Hi, Please see the sample.
A ) Analog HardwareType Ports Action FXO Ports 1 Edit<http://192.168.1.134/admin/config.php?type=setup&display=dahdi&dahdi_form=analog_signalling&ports=fxo> FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog* * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID: CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXXXXXX Dial Rules Wizards: Outbound Dial Prefix: Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXXXXXX 8|NXXXXXX Dial patterns wizards*: * Trunk Sequence ZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration ====================== Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI> dahdi show status* Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro("SIP/199-0000003a", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-0000003a", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-0000003a", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-0000003a", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-0000003a", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-0000003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-0000003a' -- Executing [h@from-internal:1] Macro("SIP/199-0000003a", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-0000003a", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-0000003a", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-0000003a", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-0000003a", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-0000003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-0000003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind <[email protected]> wrote: > Some CLI logs will get you better help on the issue ! also paste the FXO > configurations and how you configured it ! > > On Wed, Sep 28, 2011 at 2:11 PM, michael k <[email protected]> wrote: > >> Hi All, >> >> I am trying to connect my asterisk box with freepbx to PSTN. I >> have purchased x100p FXO card and installed in my asterisk server. My >> freepbx detected the x100p FXO card and i can see the card specific details >> in command line. I have configured the following things. >> >> 1. OUTBOUND caller id and Dialing rules in Freepbx. >> >> 2. INBOUND route >> >> When i call to the PSTN number before connecting to the FXO card, i am >> getting a ringing. But i get a message like the "number is out of order" >> when i just connect the line to FXO card. >> >> Please some one help me to resolve his issue >> >> -- >> _____________________________________________________________________ >> -- 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
