Hello all,
My problem is that when I call from an extension to another, I ear the
dial tones, but when I make a call using the Zap or Capi channels I do
not ear the dial tones.
Why this could be happen?
Any clue will be appreciated.
Thanks.
Ismael gil.
Hello all,
I am having a lot of problems with zaptel channels,
I have got an TDM02B, and I don't know how setup /etc/zaptel.con and
/etc/asterisk/zapata.conf for use it on asterisk.
Some one could help me with this configuración?
My problem is about the type of signalling
Thanks,
Regards.
Ismael
Hello all,
Recently I bought a TDM02B digium card to conect to the PSTN.
We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try
to start asterisk, the box hangs.
Someone have the same card running with asterisk in a similar machine?
Could you tell me your box hardware details?
Hello all,
I recently install a TDM04B with only 2 FXS modules.
I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of
these stable packages.
I configured /etc/zaptel.conf like the following
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
.
But when I try to start asterisk, I get a box crash just after parsing
musiconhold.
Any clue?
Thanks
Ismael Gil.
ismaelg wrote:
Hello all,
I recently install a TDM04B with only 2 FXS modules.
I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of
these stable packages.
I
It crashes, here the last messages asterisk wrote in stdout before hangs.
[app_hasnewvoicemail.so] = (Indicator for whether a voice mailbox has
messages in a given folder.)
== Registered application 'HasVoicemail'
== Registered application 'HasNewVoicemail'
[format_wav_gsm.so] = (Microsoft
Hello,
I am trying to setup an asterisk to store users datails in a mysql
database. Explained here
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
All works well, I create a user in the database, but asterisk seems that
can't read the data, I create a mysql user with enough
Hello,
I trying to configure asterisk to store sip and iax2 user in a mysql
database.
All goes well, but my problem is when i try to add a new user (sip or iax).
I have look for an aplication with a web interface that lets us manage
the user account in asterisk without success .
How could I
Hello,
I am getting this kind of Warning in the Asterisk console, but i don't
know why.
WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 102
(Non-critical Request)
Could you give some clue to solve this problem?
Thanks in advice.
Ismael.
Hello all,
I want to setup Asterisk to forward a call if the dialed extension is
busy. I do not want to wait on the line until the extension timeout
expired. What I want is when I dial am extension currently Busy (Talking
with someone), asterisk inmediately forwards my call to an extension I
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i want.
Anybody could helpme?
ani clue will be appreciated.
Regards.
Ismael.
Hello all,
I just configure Bind 9 in our LAN to resolve the Asterisk name
sip.bussines.com for our phones.
I want that when a local extensión calls to another local extension, the
phone shows Extension@DNS name instead of Extension@ip address
like now happens.
In all my phones I configure
Hello all,
I'm still having problems with the IVR options.
When I press on my mobile phone one of the digits related in the IVR
options, press 1 for .,press 2 for.., press 3 for..
After I press the one, the second or the tirth key on my mobile phone, I
can't hear nothing more, I can't
Hello all,
I'm trying setting up an IVR on a Asterisk Soho PBX.
My problem is when I dial the IVR extensin from an Asterisk internal extension
all goes well, but when I dial the external number of the IVR, e.g. 119235656,
the PSTN number of my asterisk, I get the same IVR menu but when I
Hello,
I'm having the following problem in my asterisk config.
I have a little voice menu, with two options,
The welcome message looks like that,
1- press 1, to dial an extension
2- press 2, to speak with an operator.
If I press 1, I get the following message
Dial the extensión number
Hello all,
I am tring to change the default language in Asterisk, exactly for the
Voicemail messages.
I trying with the option Language=fr in the voicemail.conf global
section, without success.
I trying with the Setlanguage(fr) in the extensions.conf global section,
but without success too.
Hello all,
I having a lot of troubles to configure a simple voice menu.
In extensions.conf I have the following.
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,10
exten = s,4,ResponseTimeout,20
exten = s,5,Background(itranser/msg_bienvenida)
exten =
Thank you Christopher,
Imade the changes you told me, but, when I try to make an incoming call,
in the Asterisk console, I get
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9'
-- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r")
in new stack
-- Called
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