The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide.
On Thu, Sep 29, 2011 at 11:01 AM, michael k <[email protected]> wrote: > 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 >
-- _____________________________________________________________________ -- 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
