[asterisk-users] PSTN connectivity

Sam Govind govoiper at gmail.com
Thu Sep 29 01:05:31 CDT 2011


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
>>
>> 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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