[Asterisk-Users] Asterisk & Alcatel PBX

[Trasschaert Karl]

hi,,

Thanks for your help

Concernig the Analog port i've to ask our operator to check that, we have no 
acces to the pbx :(

asterisk1*CLI> zap show channels
   Chan Extension  Context         Language   MusicOnHold
pseudo            from-pstn       en
      1            from-pstn       en
I only have 1 card in the server. I see 2 Chan, could you explain me?

you said "with only one analogue line into the * server who are you
expecting the Alcatel users to address the SIP phones?"

I don't understand what you mean, could you developpe more?

Actually, i've configured the incoming call to ring a group wich include all 
sip extension.

>From: Mark Phillips <g7ltt at g7ltt.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] Asterisk & Alcatel PBX
>Date: Tue, 23 Aug 2005 10:19:35 -0400
>
>Make sure you have your Alcatel's analogue port setup for fxo so that you 
>can plug the X100P's fxs port (line) into it.
>
>As for the ringing, that's artificially gernerated by the Alcatel.
>
>You do have the Zaptel drivers compiled and loaded right? What do you get 
>when you do zap show channels?
>
>Also, with only one analogue line into the * server who are you expecting 
>the Alcatel users to address the SIP phones?
>
>Mark
>
>Trasschaert Karl wrote:
>>Hello everybody,
>>
>>I just buy a X101p clone and i'm new in asterisk.
>>
>>Here is my configuration :
>>
>>ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone 
>>------sip phone ext 200 - 203
>>                      |||
>>                  ISDN phones ext 60-67
>>
>>>From sip phone to ext 60-67 it works. 9+extnumber
>>>From sip phone to Land lines it works. 9+0+phone number
>>
>>
>>>From ext 60-67 to sip doesnt works. simply dial 68. I only ear the 
>>>ringing
>>
>>tone in the fone but no SIP phone ring.
>>Of course from landline to sip it doesn't works.
>>I've dial 7777 fro a sip phone to simulate incoming call and the sip phone 
>>ring.
>>If i replace the asterisk server by a simple analog phone, it ring.
>>Apparently, the asterisk server doesn't unhook.
>>
>>I also activate the debug
>>notting happen when i try to make a call.
>>
>>I already exchange the card, same problem.
>>
>>I think it's a configuration problem but i don't know where.
>>
>>Did someone can help me?
>>
>>Thx
>>
>>jme
>>
>>
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>
>--
>
>Mark, G7LTT/KC2ENI
>Randolph, NJ
>http://www.g7ltt.com
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