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<TITLE>Re: [asterisk-users] Asterisk and Siemens Legacy PBX</TITLE>
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<FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>Hi, thanks for this, something I had totally looked over because I saw the span 2 had gone from RED to OK.<BR>
<BR>
Zapata.conf<BR>
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<BR>
[channels]<BR>
language=en<BR>
<BR>
; Default context<BR>
context=inbound-from-pstn<BR>
switchtype=euroisdn<BR>
signalling=pri_cpe<BR>
rxwink=300<BR>
usecallerid=yes<BR>
idecallerid=no<BR>
callwaiting=no<BR>
;restrictcid=no<BR>
usecallingpres=yes<BR>
callwaitingcallerid=yes<BR>
threewaycalling=no<BR>
transfer=no<BR>
cancallforward=no<BR>
callreturn=yes<BR>
echocancel=yes<BR>
echocancelwhenbridged=no<BR>
echotraining=yes<BR>
rxgain=0.0<BR>
txgain=0.0<BR>
group=1<BR>
callgroup=9<BR>
pickupgroup=9<BR>
immediate=no<BR>
musiconhold=default<BR>
busydetect=no<BR>
callprogress=no<BR>
<BR>
channel=>1-15,17-31<BR>
<BR>
context=inbound-from-siemens<BR>
signalling=pri_net<BR>
switchtype=euroisdn<BR>
priindication=outofband<BR>
group=2<BR>
channel=>32-46,48-62<BR>
<BR>
<BR>
Zaptel.conf<BR>
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loadzone=uk<BR>
defaultzone=uk<BR>
<BR>
span=1,1,0,ccs,hdb3,crc4<BR>
bchan=1-15,17-31<BR>
dchan=16<BR>
<BR>
span=2,0,0,ccs,hdb3,crc4<BR>
bchan=32-46,48-62<BR>
dchan=47<BR>
<BR>
I now assume framing and timing are not right.....<BR>
<BR>
Any help would be appreciated ! :)<BR>
<BR>
James<BR>
<BR>
<BR>
On 7/8/06 03:48, "(AstATN)" <chan@isysnetsolution.com> wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE="4"><FONT FACE="Courier New"><SPAN STYLE='font-size:13.0px'>Hi James,<BR>
>James wrote;<BR>
>When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is<BR>
>because I have not told asterisk to give it one. <BR>
I might be wrong;<BR>
My question is, are you sure your ISDN ( Asterisk span to Siemens ) is up logically?<BR>
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 “9” to call out. <BR>
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.<BR>
As fall as ISDN Legacy PBX is concern, it will throw out the digits if nothing wrong with the link. <BR>
If possible, share with your Zapata.conf setting, may be group of us can help.<BR>
<BR>
Tq<BR>
<BR>
<BR>
James wrote;<BR>
Hi, I just realised I think I have missed a step....<BR>
<BR>
Asterisk is not matching the extension from the siemens because the siemens<BR>
has not even sent one yet, it is still waiting for a dial tone. When I hit 9<BR>
on the siemens it does not get a dial tone from asterisk, I assume this is<BR>
because I have not told asterisk to give it one....(dur!) How should I tell<BR>
asterisk how to handle this, I have defined it a context and I know its<BR>
making it this far, but I don9t know how to get the next bit coded. Any help<BR>
appreciated !<BR>
<BR>
Thanks<BR>
<BR>
James<BR>
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