[asterisk-users] Asterisk and Siemens Legacy PBX

[James Arscott]

Hi, thanks for this, something I had totally looked over because I saw the
span 2 had gone from RED to OK.

Zapata.conf
-------------

[channels]
language=en

; Default context
context=inbound-from-pstn
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
usecallerid=yes
idecallerid=no
callwaiting=no
;restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=no
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=9
pickupgroup=9
immediate=no
musiconhold=default
busydetect=no
callprogress=no

channel=>1-15,17-31

context=inbound-from-siemens
signalling=pri_net
switchtype=euroisdn
priindication=outofband
group=2
channel=>32-46,48-62


Zaptel.conf
-------------

loadzone=uk
defaultzone=uk

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

I now assume framing and timing are not right.....

Any help would be appreciated ! :)

James


On 7/8/06 03:48, "(AstATN)" <chan at isysnetsolution.com> wrote:

> Hi James,
>> >James wrote;
>> >When I hit 9 on the siemens it does not get a dial tone from asterisk, I
>> assume this is
>> >because I have not told asterisk to give it one.
> I might be wrong;
> My question is, are you sure your ISDN ( Asterisk span to Siemens ) is up
> logically?
> ISDN is no tone given, dial tone is actually produced by Legacy Side, when L1,
> L2 and L3 signals is up (eg coding, framing and timing), Legacy PBX will
> automatically make it self ready, and simulating the dial tone when user hit
> ³9² to call out. 
> I did try with Alcatel and Ericsson MD machine; both are simulating dial tone
> once L2 and L3 are working properly, so I assume that this is the Europe PBX
> standard.
> As fall as ISDN Legacy PBX is concern, it will throw out the digits if nothing
> wrong with the link.
> If possible, share with your Zapata.conf setting, may be group of us can help.
>  
> Tq
>  
>  
> James wrote;
> Hi, I just realised I think I have missed a step....
>  
> Asterisk is not matching the extension from the siemens because the siemens
> has not even sent one yet, it is still waiting for a dial tone. When I hit 9
> on the siemens it does not get a dial tone from asterisk, I assume this is
> because I have not told asterisk to give it one....(dur!) How should I tell
> asterisk how to handle this, I have defined it a context and I know its
> making it this far, but I don9t know how to get the next bit coded. Any help
> appreciated !
>  
> Thanks
>  
> James
> 
> 
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