Displaying 20 results from an estimated 28 matches for "neofita".
2006 Jun 28
12
Ajax.Updater
Hi,
someone can help me, I am ot able to find the way how to user
Ajax.updaterto test if the request give some positive or negative
result.
I am able only to return the result inside a div.
An example is appreciated.
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2007 Nov 13
1
[Fwd: Re: VoiceMail hangup]
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug suggested below.
I try to explain my behavior:
- I dial the voicemail extension.
- I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the co...
2006 Apr 01
4
H323 on way voice
Hi,
I installed H323, however when I make a call from SIP Phone -> Asterisk H323
-> Provider H323 the provider can hear me, but I cannot hear nothing.
The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct
to internet with a public IP.
Any thoughts?
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2006 Jun 17
6
Canreinvite
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if
I call the traffic still go throw the asterisk. How come?
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2006 Jun 27
5
WebPhone
Hi,
someone know a good webphone, possibily a free one
Thx
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2006 Jun 16
3
Aiuto
Salve a tutti;
Sono un neofita di Ruby, sinceramente Rails ancora non lo ho utilizzato.
Sono uno studente universitario, ora sto preparando una tesina su un
framework per sviluppare applicazioni web non so se lo conoscete e lo
Spring webflow.
nella tesina ho cercato di convincere il prof sul fatto che Rails ?
migliore.
Solo c...
2005 Aug 16
3
TAFM
Hi,
I installed this program but I am not able to configure, it does not
want to work.
Someone can help me?
2007 Feb 09
2
Chan_Cellphone
Hi,
I download the last svn and I also look around but I cannot find the source,
I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919
any one can help me out.
thx
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2007 Oct 23
2
Force codec order
There is a way to force the order of the codecs in the sip.conf since the
allow seams to let know only the accepted codec.
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2007 Feb 09
4
asterisk 1.4 FC5 and Gtalk
JABBER: gtalk_account OUTGOING: <?xml version='1.0'?><stream:stream
xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='
gmail.com' version='1.0'>
localhost*CLI> jabber show tes
JABBER: gtalk_account INCOMING: <?xml version="1.0"
encoding="UTF-8"?><stream:stream from="gmail.com"
2015 Sep 08
1
Problem with SHA2/Geotrust and dovecot 2.0.9
Hi
I have renew my geotrust certificate using sha2, and I have problem with
Dovecot 2.0.9 and redhat 6.7.
The same certificate is working in Apache.
The error is
dovecot: imap-login: Fatal: Can't load ssl_cert: There is no valid PEM
certificate.
and the configuration file is
ssl_cert=</etc/pki/dovecot/certs/2015/new.2015.pem
ssl_key =</etc/pki/dovecot/private/new.2015.key
ssl_ca =
2015 Sep 17
1
Sieve and forward
Hi
I have already posted to the postfix group, however, I believe that sieve
and dovecot should be able to fix this problem.
When I receive a message from yahoo and the user forward it to a gmail or
yahoo acount this email is considered as spam or rejected.
>From yahoo is rejected since it seems that I am try to send spam since the
email should be signed with dkim.
Is there a way to encpuslated
2009 Feb 19
1
deliver
I am new and for sure I did something stupid
I trying to use dovecot with sendmail
but I am receiving this error
teststat deliver(usertest): net_connect(/var/run/dovecot/auth-master)
failed: Permission denied
stat=Deferred: local mailer (/usr/libexec/dovecot/deliver) exited with
EX_TEMPFAIL
on send mail I added
FEATURE(`local_procmail', `/usr/libexec/dovecot/deliver',`deliver -d
2005 Aug 29
1
TXFAX() status
Hi,
I'm using a script in order to send out my faxes with the application
txfax, therefore, I do not know how to see if the faxes are sent.
Any idea?
2005 Sep 01
1
Problem with include
Hi,
I put on sip.conf the following line
#include "sip.d/*.conf"
inside I have files like that
provider1.conf
provider2.conf
But asterisk does not want to load it
This is the error
Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1
13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found
(No such file or directory)
this
2006 Mar 29
2
H323 behind a Firewall
There is a proble to put an H323 Asterisk server behind an iptables
firewall?
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2006 Apr 28
1
Remote UNIX connection disconnected over and over
Hi,
I am pretty sure that you already answer to this question, but I was not
able to find the solution
on the console I have over and over the following msgs
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection
2006 Apr 29
1
NOTIFY Problem
Hi,
one of my WiFI phone has problem with the notify asterisk signal to me the
following
Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not
implement 'NOTIFY'
In theory the phone support this function.
Any idea?
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2007 Feb 14
1
Strange behaviour with Dial cmd
I have this simple context
I am register to an external provider and when I am not home I would like to
transfer the phone outside
The problem that the call goes in loop
I cannot understand why.
Can you figure out my error?
Thank you
sip.conf
register => user:pass@provider/400
[inside]
exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT)
exten => _4X.,2,hangup
-- Executing
2006 Jun 27
1
Capture click
Hi,
I saw one site (bubbleshare) that it is able to caputer the click on the log
in link, however, I cannot understand how they can do that
Someone can explaint it to me?
Thank you
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