[asterisk-users] PSTN connectivity
michael k
michael at inapp.com
Thu Sep 29 07:51:16 CDT 2011
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 <govoiper at gmail.com> 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 <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'
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> 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
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110929/bc9ad563/attachment-0001.htm>
More information about the asterisk-users
mailing list