[asterisk-users] PSTN connectivity
Sam Govind
govoiper at gmail.com
Thu Sep 29 02:30:00 CDT 2011
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 <michael at inapp.com> wrote:
> 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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>>>>>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>>
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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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>>> 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
>>
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>> 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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