Can you please figure out the configuration issue in my freepbx ?
On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind <[email protected]> wrote: > 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 >
-- _____________________________________________________________________ -- 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
