[asterisk-users] PSTN connectivity

michael k michael at inapp.com
Thu Sep 29 01:57:09 CDT 2011 

Can you please figure out the configuration issue in my freepbx ?




On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind <govoiper at gmail.com> 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 <michael at inapp.com> 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 at 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 at 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 at from-internal:1] Macro("SIP/199-0000003a",
>> "hangupcall") in new stack
>>     -- Executing [s at macro-hangupcall:1] GotoIf("SIP/199-0000003a",
>> "1?skiprg") in new stack
>>     -- Goto (macro-hangupcall,s,4)
>>     -- Executing [s at macro-hangupcall:4] GotoIf("SIP/199-0000003a",
>> "1?skipblkvm") in new stack
>>     -- Goto (macro-hangupcall,s,7)
>>     -- Executing [s at macro-hangupcall:7] GotoIf("SIP/199-0000003a",
>> "1?theend") in new stack
>>     -- Goto (macro-hangupcall,s,9)
>>     -- Executing [s at 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 at from-internal:1] Macro("SIP/199-0000003a",
>> "hangupcall") in new stack
>>     -- Executing [s at macro-hangupcall:1] GotoIf("SIP/199-0000003a",
>> "1?skiprg") in new stack
>>     -- Goto (macro-hangupcall,s,4)
>>     -- Executing [s at macro-hangupcall:4] GotoIf("SIP/199-0000003a",
>> "1?skipblkvm") in new stack
>>     -- Goto (macro-hangupcall,s,7)
>>     -- Executing [s at macro-hangupcall:7] GotoIf("SIP/199-0000003a",
>> "1?theend") in new stack
>>     -- Goto (macro-hangupcall,s,9)
>>    -- Executing [s at 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'
>>
>>
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>> On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind <govoiper at gmail.com> 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 <michael at inapp.com> 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
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>>
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>>> 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
>
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