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fxsks=1
loadzone=es
defaultzone=es
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[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[miprimerejemplo]
exten => 20000,1,Dial(SIP/20000,30,Ttm)
exten => 20000,2,Hangup
exten => 20000,102,Voicemail(20000)
exten => 20000,103,Hangup
exten => 20100,1,Dial(SIP/20100,30,Ttm)
exten => 20100,2,Hangup
exten => 20100,102,Voicemail(20100)
exten => 20100,103,Hangup
exten => 20200,1,Dial(SIP/20200,30,Ttm)
exten => 20200,2,Hangup
exten => 202000,102,Voicemail(20200)
exten => 20200,103,Hangup
exten => 20300,1,Dial(SIP/20300,30,Ttm)
exten => 20300,2,Hangup
exten => 203000,102,Voicemail(20300)
exten => 20300,103,Hangup
exten => 20400,1,Dial(SIP/20400,30,Ttm)
exten => 20400,2,Hangup
exten => 204000,102,Voicemail(20400)
exten => 20400,103,Hangup
exten => 30000,1,VoicemailMain
exten => _9XXXXXXXX,1,Dial(SIP/0034${EXTEN}@VoipBuster)
exten => _9XXXXXXXX,2,Hangup
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[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[20000]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
mailbox=20000@miprimerbuzon
[20100]
type=friend
secret=some
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=miprimerejemplo
mailbox=20100@miprimerbuzon
[20200]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
mailbox=20200@miprimerbuzon
[20300]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
mailbox=20300@miprimerbuzon
[20400]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
mailbox=20400@miprimerbuzon
[VoipBuster]
type=peer
host=sip.voipbuster.com
username=somesi3
fromuser=somesi3
secret=some
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[channels]
language=es
context=incoming
switchtype=euroisdn
usercallid=yes
hidecallerid=no
musiconhold=default
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
inmediate=no
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbriged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxs_ks
context=incoming
channel=4
Hi all, I have installed Asterisk in my PC. I am running one LDAP server. I could not get enough documents which would help me to intergrate the existing user Database. Say I have a LDAP directory which has all the numbers and user details I should not edit the sip.conf again. Asterisk should be made aware to contact the LDAP directory for user info or Voicemail passwords etc. Help on this would be highly appreciated. Thanks and Regards, Sravana
On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote:> hello, I have a A400P01 PCI from OpenVox. > > I have installed some extension and a VoipBuste account to callo out of my LAN. > > How can I receive and send calls from a nd to outside by my analog line??? > > I want to receive dthe calls from 20100 extension. > > Here you have my config files, thanks for all.A few things unrelated to your issue that may help you to get more effetive answers from this list: 1. Please give more descriptive subject lines. The subject of your first message ("asterisk on Debian") was good. The subject of your more recent messages are rather poor: "please help me" gives no hint as to what the problem is. 2. You have already started a thread, and another list member has asked you for some details. The files attached to this message appear to be replies to that message. If they are, please follow-up the same thread. 3. You did not write what is actually wrong: "I do XYZ. I expect it to cause ABC but instead I get DEF" See also the document on how to ask questions effectively: http://www.catb.org/~esr/faqs/smart-questions.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir@jabber.org +972-50-7952406 mailto:tzafrir.cohen@xorcom.com http://www.xorcom.com iax:guest@local.xorcom.com/tzafrir
[ Subject manually fixed. Maybe my threading manipulation even worked...] On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote:> hello, I have a A400P01 PCI from OpenVox. > > I have installed some extension and a VoipBuste account to callo out of my LAN. > > How can I receive and send calls from a nd to outside by my analog line??? > > I want to receive dthe calls from 20100 extension. > > Here you have my config files, thanks for all.Two problems are obvious: 1. /etc/zaptel.conf defines channls no. 1, whereas /etc/asterisk/zapata.conf defines channel no. 4 . This should cause chan_zap to fail loading on whatever configuration you have. Did I mention genzaptelconf before? 2. Your sip.conf sets the mailboxes in a non-default context, but the VoicemailMain call in extensions.conf checks in the default context. Get rid of the useless context unless you really have a multi-domain setup. -- Tzafrir Cohen icq#16849755 jabber:tzafrir@jabber.org +972-50-7952406 mailto:tzafrir.cohen@xorcom.com http://www.xorcom.com iax:guest@local.xorcom.com/tzafrir
On 24/04/07, sravana <sravana@it.iitb.ac.in> wrote:> Hi all, > I have installed Asterisk in my PC. I am running one LDAP server. I > could not get enough documents which would help me to intergrate the > existing user Database. Say I have a LDAP directory which has all the > numbers and user details I should not edit the sip.conf again. Asterisk > should be made aware to contact the LDAP directory for user info or > Voicemail passwords etc. > > Help on this would be highly appreciated.http://bugs.digium.com/view.php?id=5768> > Thanks and Regards, > Sravana > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >