Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important.
But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI. On Thu, Sep 29, 2011 at 11:57 AM, michael k <[email protected]> wrote: > 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 >
-- _____________________________________________________________________ -- 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
