Displaying 20 results from an estimated 26 matches for "oosterhout".
2005 Mar 21
2
Permission issue with outgoing calling
I have created a call file which has been moved into the outgoing directory.
However the log file displays the following message: Unable to open
/var/spool/asterisk/outgoing/1.call: Permission denied, deleting
I have executed chmod 777 1.call on the file prior to moving it to the
outgoing directory but is there something else I need to do before the file
can be used by Asterisk?
Any help
2005 Feb 25
3
How does the g.729 registration program work?
...options I
can think of are:
- There's a config file, though I've seen no mention of it
- The actual binary shared library is modified
- The system contacts Digium every time you start asterisk
In the last case nothing is changed at all and I'm fine.
Thanks in advance,
--
Martijn van Oosterhout
Ecomtel Pty Ltd
2005 Feb 24
4
What is an E400P-SS7??
...nyone would like to share?
Thanks in advance,
[1] http://lists.digium.com/pipermail/asterisk-users/2004-September/062882.html
[2] http://lists.digium.com/pipermail/asterisk-users/2004-June/052198.html
[3] http://lists.digium.com/pipermail/asterisk-users/2004-September/062872.html
--
Martijn van Oosterhout
Ecomtel Pty Ltd
2004 Jun 17
1
Calling the firefly network?
Is there a way to register with or call the firefly network from an Asterisk
server. It would be pretty cool if you could gateway calls onto it.
Have a nice day,
--
Martijn van Oosterhout
2005 Jan 27
1
Digium and Intel Chipset compatability
...#39;t find any confirmed details on the mailing list
about it. Also, the email seems to imply that the TE405P will be fine,
though it doesn't say that explicitly.
Basically, is anyone using a 4 port E1 card successfully on an IntelĀ®
E7221 Chipset or similar?
Thanks in advance,
--
Martijn van Oosterhout
Ecomtel Pty Ltd
2005 Mar 09
1
Providing a dialtone
...ones when you have a physical line,
like the alsa channel, or a zap channel. But I'm just thinking of if
they've selected an option that allows them to dial a normal number, to
also provide a normal dialtone. Should I just record one and use
Background()?
Thanks in advance,
--
Martijn van Oosterhout
Ecomtel Pty Ltd
2004 Jun 16
1
asterisk/netmeeting works, asterisk/ohphone doesn't?
...somewhere.
So my question, has anyone had ohphone work for them and if so, what
versions did they use at each end. Are any known compiler issues (g++ 3.0.4,
gcc 2.95.4)?
Are there any clients (other than ohphone) which one can use with asterisk
to test it out?
Thanks in advance,
--
Martijn van Oosterhout
IT Manager
Ecomtel
2005 Mar 02
4
timing/clock problem
Hi all,
We have been fighting with telco for a entire week.
Today they came here with a LITE3000 to analyze what is going on.
When I configure zaptel with no external clock, E1 gets aligned/synchronized
with bit rate in 2048000 bps, both me and telco.
span=4,0,0,ccs,hdb3,crc4
But when I configure span4 to get clock source from telco they become
unsynchronized. TElco bit rate stays in
2005 Mar 13
5
possible bug in chan_capi concerning context handling
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section "[interfaces]" the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in
2005 Feb 01
1
choppy sound after 15 minutes in a call
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 Transmission error to
85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN
2005 Feb 17
1
Problems compiling pridump utility
I do 'make pridump' from the libpri source directory and receive the
following:
# make pridump
cc -o pridump pridump.o -L. -lpri -lzap -Wall -Werror
-Wstrict-prototypes -Wmissing-prototypes -g
/usr/bin/ld: cannot find -lzap
collect2: ld returned 1 exit status
make: *** [pridump] Error 1
I am new to all of this, so I am sure I am missing something obvious,
any help will be appreciated.
2005 Feb 26
2
ERROR: compile asterisk(from CVS HEAD) and got an error
Dear ALL:
I got an error message lists below.
Does anyone have the same problem? How to solve it?
Best Regard
Charles
In file included from config.c:34:
include/asterisk/app.h:62: array size missing in `options'
make: *** [config.o] Error 1
2005 Mar 16
1
live monitoring of SIP calls chan_spy
hello there,
I have searched lists about an application chan_spy, people talked about
it on lists that we can use it to monitor sip to sip calls. but I am
unable to find any clue of it.
can some one please tell me from where I can get this chan-spy application
thank you
regards,
--
Atif
2005 Mar 24
1
Error cannot record voicemail
I tried to share my spool directory so I could get monitored calls, and now
this error comes up when I try to leave a message in any of my voicemail
boxes.
Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error
opening text file for o
utput
-- Recording the message
Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open
file /var/spool/asterisk/v
2005 Mar 09
1
Should ICMP port unreachable generate a BYE request?
Hi all,
I'm researching random call drops on our Asterisk and would like to
make sure whether it's something wrong with our VoIP provider or with
the Asterisk. I sniffed traffic between Asterisk and our VoIP
provider's SIP gateway, and observed that in the middle of the
conversation an RTP stream originating from Asterisk gets an ICMP port
unreachable from provider's SIP gateway
2005 Feb 16
2
Sip Notify PAP2-NA?
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
I was thinking of just setting a cron job or something to check every minute
for voicemail and set our sip NOTIFY messages as needed.
Also, the PAP2-NA has the ability to reboot via a sip notify and I would
like to be
2005 Mar 02
3
[OT] stupid firmware question...
I know this is a really stupid question, but I just have to ask...
Where would I start if I wanted to try and develop my own firmware for a
particular phone. Namely, I want to try and 're-write' the SIP firmware
for Cisco 7940's. Any ideas?
-Chris
PS: [* put on flame suit *] why won't any of the phone manufacturer's
just open-source the firmware for their phones? [*
2005 Mar 03
5
country/city codes
Some country codes are three digits long. Some are two.
e.g. UK 44 , Bermuda 441
Does anyone know a formula for determining which part of a dialled number is the country code and city code ?
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2005 Mar 12
2
Unable to create channel of type 'IAX2'
Hi all,
I'm a newbie and I have a configuration problem with Asterisk.
Seems that I'm not able to call an outbound number. I'm quite sure that it
is a configuration problem, but I'm not able to find out where is the
mistake, even reading several docs to www.voip-info.org.
I do not have a good knowledge of Asterisk, I'm not very familiar with its
configuration and I've a
2005 Mar 05
2
Getting asterisk-addons installed on Debian?
Hi,
I am having some trouble installing asterisk addons on Debian. I wish to do
this to use mysql billing.
I have mysql and mysql-devel packages installed I think!?
pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev
Desired=Unknown/Install/Remove/Purge/Hold
| Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed
|/