[asterisk-users] PSTN connectivity

[michael k]

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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110929/2925793e/attachment.htm>