The Panasonic will not allow you to dial an extension directly from the “outside”.  You need to have Asterisk setup as a Panasonic extension and use an extension to extension format in your dial plan.

 

Try making connections like this.

 

 

  ----------     --------------

  |        |     |            |

  |  SIP   |-----|  Asterisk  |

  |        |     |            |

  ----------     --------------

                    |        |

                   FXS      FXO

                    |        |

                 Pana Ext   Pana CO

  ----------    ------------------------

  |        |    |                      |

  | PSTN   |----|      Panasonic       |

  |        |    |                      |

  ----------    ------------------------

 

 

Now for outbound calls via PSTN, you would use a ZAP FXS, dial “9w” (w=wait a second) then out the PSTN.  To dial a Panasonic ext, you would pick up the same ZAP FXS channel and simply dial the number of the Panasonic extension you want.  When a SIP call comes in, your dial plan would ring the ZAP FXO port so as far as the Pana KSU is concerned, its just another inbound ring.

 

I wrote a pretty detailed paper about this integration and put it on the WIKI along with the full EXTENSIONS.CONF I used to do a pretty tight integration the Panasonic KSU.  Take a look and see if maybe this might solve your problems.

 

 http://www.voip-info.org/wiki/view/Panasonic+KSU

 

G.Henderhsot

 


From: Pablo L. Arturi [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 29, 2006 10:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232

 

Hello Pablo, I think you should decribe with details how are you routing the call between the SIP device and the extensions.

 

Pablo

----- Original Message -----

From: Pablo Mora

Sent: Thursday, July 27, 2006 10:18 PM

Subject: [asterisk-users] Strange behaviour Panasonic KX-TD1232

 

Hello,

 

I’ve got asterisk running and almost working with Panasonic KX-TD1232

I said almost, because there’s a strange behaviour when I make calls.

 

 -------              -----------------              ---------------------               -----------

| SIP | ---------- | ASTERISK | ---------- | PANASONIC | ------------ | PSTN |

 -------              -----------------              ---------------------                ----------

                                                           |         |

                                                       -------     -------

                                                      | Ext1|  | Ext2|

                                                       -------     -------

 

When I make a call from PSTN to SIP, the call goes on successfully.

When I make a call from SIP to PSTN, the call goes on successfully.

When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully.

When I make a calla from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn’t hear anything. 

 

It seams appear like Asterisk doesn’t detect the answer on Ext1

 

Is there any way to figure it out??

 

Thanks


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