sorry.. but you got a heads up
On 6/27/06,
Josué Conti <[EMAIL PROTECTED]> wrote:
Hi Mike, all good? I thank its attention. Where I modify these parameters that you said? Best RegardsJosué
2006/6/27, Mike Lynchfield <[EMAIL PROTECTED]>:HERE IS answer [EMAIL PROTECTED]
had same problem..
make the settings for 90 volt.. not 70 volt ringer..
make it trapezoidal not sinusoisal
make it 900 ohm not 600 impedence..
that worked for pap2's
seem siemens are made for europe style ring voltage not north american.
On 6/27/06, Herchi Silviu < [EMAIL PROTECTED] > wrote:Hello,The main differences I can see:- in zaptel.confyou have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a timer for the PBX - on my setup the PBX is the master clock and Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4)- in zapata.confI have switchtype=EuroISDN. Generally speaking, try using other switchtypes.Regards,Silviu
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Josué Conti
Sent: 27 June 2006 14:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000
Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf#zapte.confspan=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us
#zapata.conf[trunkgroups]
[channels]
language=pt_BR
context=default
switchtype=qsig
pridialplan=private
prilocaldialplan=private
facilityenable = yes
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
immediate=no
callerid=asreceived
musiconhold=default
group=1
channel=>1-15
channel=>17-31
Best RegardsJosué
2006/6/27, Herchi Silviu <[EMAIL PROTECTED]>:Hi,
Could you post your /etc/zaptel.conf and zapata.conf?
Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)?
Silviu
----
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me?Best Regards
Josué
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Making it happen
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Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.888.470.7253
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