Hi,
I want to
automatically send the sound files generated by asterisks monitor functions to a
certain email address. My knowledge of shell scripting leaves a lot to desire,
so I was hoping maybe on of you guys already did this and might provide me with
an example of what to do :)
Best
That issue is fixed in the CVS HEAD version of asterisk.
There are a couple of workarounds possible with 1.0.6. Check
the bugtracker for the bug where it was implemented for more
information. (sorry, don't remember the bug-number and don't
have time to look it up right now).
You might
That issue is fixed in the CVS HEAD version of asterisk.
There are a couple of workarounds possible with 1.0.6. Check the
bugtracker for the bug where it was implemented for more
information. (sorry, don't remember the bug-number and don't have
time to look it up right now).
The problem is the Got event Ring/Answered(2) line.
Normally, a ring should not be detected and the DTMF-cid
times out and no incoming call is registered.
Make sure you load wctdm with the parameter
'opermode=SWEDEN', it might help.
You might also try to increase the 'RING_DEBOUNCE' define
The problem is the Got event Ring/Answered(2) line.
Normally, a ring should not be detected and the DTMF-cid
times out and no incoming call is registered.
Make sure you load wctdm with the parameter
'opermode=SWEDEN', it might help.
You might also try to increase the 'RING_DEBOUNCE' define
That is the expected behaviour. =)
It is so because when YOU hang up, asterisk hangs up the
channel and destroys it. A moment later (actually up to 90sec
in sweden) the pstn disconnects the call and signals a
polarity reversal. That reversal causes the above.
I have a solution but I don't
Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine.
I bought the card mainly to get caller ID to work properly in Sweden, and
that works just fine.
However, if the called or calling party hangs up after I hangup my SIP
channel, polarity CID detection kicks in and dials a
The only issue I have had with it so far, and it may be a
misconfiguration problem as I am certainly a newb, is that
when I dial a number that sends it over to my POTS line, I
get ringing from the softphone and the POTS line. When to
POTS line answers, the softphone continues to ring.
With an X100P card I get the PSTN line ringtone and/or busy
tone in DIAX when an outgoing call is in progress.
No need to have an Answer before...
I forgot to mention that I'm connecting to the PSTN with a SIP connection to
my provider.
/Anders
I'm using X-Pro
connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15
minutes in a call I get a lot of noise in my end. I don't think the other part
of the call hears it. After some 10 seconds or so everything is fine
again.
In my CLI I get
NOTICE[32322]: RTP
Check out http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM.
/Anders
A while ago, I saw some threads on booting linux w/ asterisk
from a CF card.
I have also seen CD installs of Asterisk, which require a hdd.
Has anyone come up with a bootable cd (like a Live CD), that
creates
I think you might have to add the line below to [sip.broadvoice.com]:
insecure=very
I know that it's required for other services, and probably with broadvoice
as well.
/Anders
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip =
If you use RH or Fedora the included mpg123 doesn't work. Check
http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat for
info on how to fix it.
/Anders
Having played with the music on hold settings for 2 days now
trying to get it to work I am stumped and need help from the
Are there any good
SIP softphones with support for SUBSCRIBE/NOTIFY, like the SNOM phones have? I
tried one from Estara, but it costs $100 and it crashed when I tried to run it
(maybe I could have make it run, but any software that crashes after running
install/config wizard is not
Hi,
I think you should remove the [iaxtel_out] from iax.conf
This is a snip from mine iax.conf:
[general]
register = user:[EMAIL PROTECTED]
[iaxtel]
type=user
context=incoming
auth=rsa
inkeys=iaxtel
You then can modify extensions.conf to handle outgoing calls. See
Hmm... For outgoing connections (from the softphone) that's true, but if you
want asterisk to send a call to the softphone the default would be to send
it to port 5060 (which is already taken by asterisk). If the softphone is
setup to register to asterisk on another port, there should be no
I've never tried softphones on Linux, but my guess is that since you run
kphone and asterisk on the same server you get a port conflict. If the
client uses port 5060 (default sip port) it would defenitely have problem
connecting to an asterisk on the same port.
Maybe you can change the kphone
There is no way for asterisk to know which extension you want to call when
you have a single analog line. You can either send the call to an extension
or group, or you could create a menu system that allow the caller to select
which extension to call (press 1 for Steve, 2 for Dave etc.).
/Anders
Hi,
I don't think any SIP server would allow you to register more than once with
the same login information. What you can do in asterisk is setup two
different entries in sip.conf and then use extensions.conf to dial both.
Example from extensions.conf
[default]
exten =
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