Thanks for the update. but how do i resolve this issue ? can you help me please ?
On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind <[email protected]> wrote: > 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 >
-- _____________________________________________________________________ -- 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
