Hi again,
I've been struggling a little with the ISDN card and drivers and found
out that CAPI doesn't work fine with it, so I switched to ISDN4Linux and
it works like a charm: both dial-in and dial-out is possible, which is
what I was looking for.
Thanks again and sorry for the bandwidth
Hi all,
this is a rather newbie-oriented question, so please bear with me...
The system running Asterisk has been provided with an AVM FRITZ!Card
PnP. SuSE Linux 9.0 recognizes it right after booting the system and it
seems to be configured (MSN) correctly...
The hwinfo looks like this:
---
Thomas Niesel wrote:
[ snip ]
Does the phone had the same MSN?
I think so. It could dial outside without a problem...
Is there maybe a PBX needs a leading Digit to get outside line?
No, those are direct lines to the PSTN, so no leading 0 (or whatever)
is needed ...
Try your settings by
: *** [chan_capi.o] Error 1
What's going wrong? (well, it doesn't compile...)
Is there any chance to find the RPM for chan_capi 0.3.5?? :)
TIA,
Martin
--
Martin Mielke
Senior UNIX SysAdmin[EMAIL PROTECTED]
THALES Information Systems http://www.thales-is.com/
Tel
Hi all,
due to the rather big email traffic regarding this issue, I decided to
publish the script so people can download it at their own risk... :-)
Please, visit:
http://www.leals.com/~mm/asterisk
for further information.
Regards,
Martin
___
Hi all,
I just modified one of the startup scripts provided on the tarball to
fit on my SuSE 9.x system to start/stop Asterisk when the system boots
or goes down.
Maybe I'm overseeing the answer but could't find where to
post/(cvs)upload the changes I made...
TIA,
Martin
Tony Nichols wrote:
I would be interested in the script.
OK. I'll send it off the list...
Did you do zaptel drivers too?
Nope ;)
Martin
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Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Huddleston, Robert wrote:
I'd like a startup script for redhat... should be just some small changes..
do you have one?
It's already there... :-)
Take a look at .../asterisk_v1_0_stable/contrib/init.d to find a file
called rc.redhat.asterisk. This one should do the trick... ;)
HTH,
Martin
Hi there,
this is just a me too... well, not exactly. I get jitter when trying
to make SIP calls through Asterisk using a GPRS connection... can this
be done actually?
TIA,
Martin
Storm D. J. Petersen wrote:
Hello,
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
Hi all,
I noticed that all incoming calls come from the user [EMAIL PROTECTED], so
I just can't hit the Call button on my SJphone for Linux to return the
call...
Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ?
Thanks and regards,
Martin
Hello again,
sorry for the delay in replying; I've been off for some weeks at a
customer's offices and couldn't read my email at work...
ePyron Felix Deierlein wrote:
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with VoIP from PC to
ePyron Felix Deierlein wrote:
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with VoIP from PC to PC, we'd like that our
people abroad could dial company internal extensions through Asterisk
using a SIP client. On a second approach, the same
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with old PBX...
http://www.voip-info.org/wiki-Asterisk+legacy+integration
...but I couldn't find anything on integration with a Siemens HiCom
150E. Later on we'll migrate to a HiPath 3750 so information
Hi all,
can Asterisk be used as a videoconference server or the like when
enabling 'videosupport=yes' ? if so, how do I use it? is there any
recommended SIP/Video-client for both Windows and Linux?
Thanks,
Martin
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Asterisk-Users mailing list
[EMAIL
Hi all,
for some strange reason, our still-under-test Asterisk deployment wants
to contact the outside world and that raised some eyebrows here...
Just a sample of our firewall log:
--
...a=DROPIN=eth0 OUT=eth2 SRC=192.168.36.199 DST=195.77.113.194 LEN=476
TOS=0x10 PREC=0x00 TTL=62 ID=39572 DF
Hi there,
because yesterday I had a problem with my email, I wanted to check the
replies (if any) to my question Needed Open ports on the archives
but... where are the ones from may??
http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html
I only see 3 posts.. is this the
Hi list,
surely this has been posted before but the archives don't offer a
'search' functionality and I need an answer really soon on this
subject... so, my apologies.
Which ports (range) must be open on a firewall, either TCP and/or UDP,
for Asterisk to work correctly?
TIA,
Martin
Dear all,
Mayybe this is a little off-topic but I don't know of any other place to
ask for it... my apologies in advance!
I'm looking for a softphone (SIP) with multiple profiles support.
Right now I use SJPhone on SuSE 9.0 Pro, which allows to create several
profiles but, AFAIK, it's not
Hi Markus,
Markus Miertschink wrote:
The one I know of is X-Pro/X-Lite from http://www.xten.com/
I doubt that there is a Linux version available...
Markus
I contacted X-Ten and they told me they are working on a Linux version
of X-Lite... let's see...
Martin
[ snip ]
William Suffill wrote:
Would it be possible to use an IAX softphone in your situation?
I know iaxcomm is available for both Windows and Linux and can handle
multiple accounts.
yes, iaxComm works for both Linux and Windows, but the sound quality is
poor compared to SIP softphones such as
Fran Boon wrote:
Gavin Hamill wrote:
I'm using Mozilla 1.7a installed from a tarball. The needed libraries
are just there:
You've answered your own question. You installed Mozilla from a
tarball. RPM therefore doesn't know about it. You need to install a
recent Mozilla RPM :)
Why do I need to
Hi Troy,
Troy Settle wrote:
Can anyone point me to some sample Cisco QoS configurations suitable for
IAX2? I've looked through Cisco's site, and get overwhelmed with the level
of documentation (too much of a good thing).
Take a look at this and see if you can use it for IAX2 as well:
Hi Jeremy,
Jeremy Hall wrote:
Actually, the short answer any more is yes, you can use a modem.
Cool! that could make my life easier when setting up a demo system to
sell Asterisk to my bosses... :-)
I know it is better for several reasons to use an actual Digium X100P.
The main reason being
Hi all,
I installed all needed RPMs by GnoPhone to be installed without problems
but when attempting to install GnoPhone itself I get this message:
# rpm -Uvh gnophone-0.2.4-1.i386.rpm
error: Failed dependencies:
mozilla = 0.9.2 is needed by gnophone-0.2.4-1
libgtkembedmoz.so is
Hallo Sacha, :-P
Sascha Knific wrote:
Hi
capiinfo gives:
---
capi not installed - No such device or address (6)
---
It´s not just about installing the apropriate package but you have to load
the capi kernel module for your isdn card.
The module to load on boot time is set in
James Moran wrote:
Do normal modems work with asterisk?
Taken from the FAQ:
Can I use my modem to connect to the PSTN?
The answer is short: No you cannot. You'll need special telephony
hardware.
Further info under: http://www.voip-info.org/wiki-Asterisk+FAQ
HTH,
Martin
Hi all,
I compiled/installed chan_capi.so without problems. When I launch
Asterisk, I get the following error:
---
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
Hi there,
Martin List-Petersen wrote:
Hi Martin,
Have you checked the rights of your /dev/capi20* interfaces ?
pbx:~ # ls -l /dev/capi*
crw-rw1 root dialout 68, 0 Sep 23 2003 /dev/capi20
crw-rw1 root dialout 68, 1 Sep 23 2003 /dev/capi20.00
crw-rw1
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva Server BRI
2M ISDN card attached and it seems to be recognized by the system.
I added the following lines to:
* modem.conf
driver=i4l
...
-(hard)phone...
...rch!... I need more tea! ;)
As always, ideas/suggestions/hints are much appreciated.
Regards,
Martin
Martin Mielke wrote:
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva
Steven Critchfield wrote:
[ snip ]
You should have a / instead of a : in the dial.
It doesn't help...
See error message:
---
Mar 29 20:34:06 WARNING[393232]: chan_modem.c:181 modem_call:
Destination g1/y requres a real destination (device:destination)
---
btw, __TRIM__ the
Hi all,
I have installed Asterisk and SIP calls are successfull inside our office.
Then I created some extensions for my colleagues in other city. As our
offices are connected trough a dedicated point-to-point line, by now
I'll just create the extensions for the remote people in the Asterisk
Hello,
I'm still doing some tests with Asterisk before reaching a production state.
To do some VoIP-PSTN tests I'd like to know how to configure Asterisk
to use an ISDN card such as Eicon Diva Server BRI 2M.
Any hints are much appreciated.
Martin
Hi Mieria,
Mireia Munoz de jesus wrote:
Hi!
When I try to call from a SIP phone to a PBX phone I get this error:
chan_oh323.c [1004] Couldn`t call 483377839
and if I get the messages from SIP debug, I have a 403 message. The
configuration of my system is:
SIP Phone ASterisk
Hi all,
I just installed Asterisk and access the preconfigured demos using
Kphone on Linux. It works but the recorded speech sounds choppy sometimes...
The Asterisk box has a 100 Mbps NIC...
Any clues?
TIA,
Martin
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Asterisk-Users mailing list
Hi all,
is there any reasonably good management tool for Asterisk out there? all
I've found under
http://www.voip-info.org/tiki-index.php?page=Asterisk+GUI are not so
complete utils, as some have the same functionality others do...
Does such ideal tool exist or do I have to type ahead all
website, has anyone any experience with NMS boards
(http://www.nmscommunications.com/)?
Thanks in advance!
Best regards,
Martin
--
Martin Mielke [EMAIL PROTECTED]
THALES Information Systems http://www.thales-is.com/
UNIX is user-friendly... It´s just selective about who its
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