Hello everyone!
So now I'm testing with chan_sip and I discovered, that I can make calls,
even if they're only listed as active channels.
But JACK just emmits white noise, with a highger frequency than 8kHz, in my
believe.
A call with app_record shows, that the signal is clear and very
Hello Jared!
OK, now calls go in and out. Even with the syntax:
channel originate sip/mu...@iptel.org application ...
it works. I've tested that with application record.
But, the channel only displays ACK and core show channels doesn't list it as
a call or a processed call afterwards.
Hello everyone!
So now I found someone to forward the ports 5060 and 16000-16100 on my
router and made sure to enter these ports 16000-16100 in rtp.conf. Still I get
no calls going.
The call is initiated. sip show channels shows the call with status ACK
and then the dialog with method
Hi Ira!
Sorry, can't use any softphone, to my knowledge. They all come with GUIs or
don't support JACK or have so limited ALSA support, that they don't fit my
card (which has a lot of channels and some other HD-recording stuff).
Still I did try the sip call with app playback as well.
Hello all!
Hm, I just examined the output of chan_sip's debug again and found this,
might that be the problem:
Warning: 392 213.192.59.75:5060 Noisy feedback tells: pid=3955
req_src_ip=91.58.9.172 req_src_port=24002 in_uri=sip:sip.iptel.org
out_uri=sip:sip.iptel.org via_cnt==1
I don't
Thanks anyway, Ira. It was very kind of you to help me along as far as you
could. I appreciate it.
anyone else here, who might be able to help me along with my problem?
Warmly yours
Julien
Music was my first love and it will be my last (John Miles)
FIND MY
Hello again!
So I tried again, experimented a bit more and got this:
channel originate sip/e...@iptel.org
[Jun 5 12:14:29] WARNING[8537]: chan_sip.c:17882 handle_response_invite:
Re-invite to non-existing call leg on other UA. SIP dialog
'3b39b40240b6126a61c7ad16108be...@91.58.24.59'. Giving
Hello everyone!
I still am not much further along with my sip calling. I changed my sip.conf
taking suggestions from the net (voip-info.org in particular). I changed
iptel's position from friend to peer. I turned on and off nat, I chose
different codecs in first place, entered my outward IP
Hello Lyle!
Thanks for your answer!
I don't know, if the server sees me at the local-ip or not. I only know,
that I'm able to register at iptel.org successfully. So asterisk tells me.
I believe my router is a Samsung router 3010 phone SL. Samsung it tells me,
the rest I had to search on
Hello Ira!
I will have a look at my rtp.conf and change the rtp-port range there. As to
forwarding: Well it remains to be seen - pardon the pun - if I can find
someone willing and patient enough to be my pair of eyes. :-)
Kindly yours
Julien
Music was my first love
Hello Motiejus!
Thank you very much. I'll try to setup an account and see what these numbers
give me. then I'll know more about the quality of the JACK module, since I
suppose, sip is very well tested, actively maintained and widely used.
Knd regards
Julien
Music was
Hello again!
I just got a SIP account and it seems - from a config on the net -, that
I've configured it correctly. But I get no call to the outside. Registration
was OK.
I tried:
channel originate sip/1/e...@iptel.org Application ...
I see the channel active for a while, but no call
Hello Kyle!
Earlier I was listening to my voicemail using my ISDN-card and a simple
telephone. But this card is no longer supported. So I just go to:
/var/spool/asterisk/voicemail/default/1234/INBOX
Then of course I use JACK. I seems some doesn't work as expected. I have the
feeling, the
Hello Russell!
That's a pitty.
But even so, this time the problem is a different one. Before we had the
problem, that I couldn't really make a call at all, or that the call crashed
quickly. This time, the call connects and seems to be stable enough, but I
have no sound from the other
Hello everyone!
So I hacked app_jack.c today, as best I could. Whic came mostly down to
inserting ast_log() messages.
I discovered the following with JACK:
When it starts, it tries to read 512 bytes and only gets 0. That clears up
after a while.
Sometimes a good time later than the
Oh P.S.:
I changed my jackd startup options as well from:
jackd --tmeout 4500 -R -d alsa -d hw:1 -r 48000 -z shaped
and then in case a:
-p 64 -n 2
Case b:
-p 1024 -n 3
case c;
-p 128 -n 2
Case c is my default setting.
The rountrip time to talk.google.com is 60.440ms average. My upstream
Hello Leif!
The issue about gtalk and jack was originated by me. Yet in those days, the
problem was of a different nature.
Still, why do I use these two and no other channel or software? Well I've
mentioned, that Asterisk is the only commandline phone on Linux I know, that
supports JACK.
Hello everyone!
So I've just scanned through the debug log, defined like this in
logger.conf:
full = notice,warning,error,debug,verbose
I couldn't see any reason for the connection not working. I called my
friend, he heard ringing, accepted the call and then it got hungup. I didn't
see
Hello everyone!
OK I'm a step farther now. Don't ask me how and why, but it seems, I can
call someone and he gets audio from me. I tried the echo-test on:
echo.test.collabora.co.uk
yet I got only noise in response. Might this have something to do with my
adress? You know, I had to specify
Hello everyone!
I have a problem with my voicemail. When someone leaves a message - using
googletalk at least - the message file starts silnet, stays that way for a few
seconds and then is cut short at the end.
The last test we did ended up more than 10 seconds missing in the left
Greetings!
I now found someone to test gtalk with and found out, that app_jack has a
problem here. My voice gets transmitted fine, but I only get white noise from
the other party. I tried to set my JACK samplerate to 8000 to make sure it's
no libresample problem, the results were the same.
Hi!
GT conversations don't go well. I can only use JACK as output - for
realtime- and that doesn't work. I'm invesitagting and have asked for help.
I'm currecntly only operating from the CLI. I could try record. I assume it's
an application as well?
Kindly yours
JUlien
Hello Steve!
I can do that. But I've seen, that the voicemail is first saved in a folder
tmp in the directory of my inbox. It's all on one local filesystem. Yet I'll
still try.
Thanks for this hint!
Warm regards
Julien
Music was my first love and it will be my
Hello Motiejus!
My jack setup is fine. I'm a musician, so I use it regularly for much more
demanding tasks than simple one-channel I/O.
I just installed asterisk 1.6.2.9-rc1, it looked newer and fix-richer than
1.6.2.8.
So any other ideas? Do you remember, when you had these problems?
Hello everyone!
I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be
very grateful, if someone could help me here. I'd be very glad, if one of you
could test googletalk with me. Last time I tried (in 1.6.0.x times) it
wouldn't work in the end.
But here are my gtalk and
Hello everyone!
So I tried to test gtalk with a friend. We could both see each other. He
uses the gtalk application for Windows.
So I tried to call him and he got a ringtone. But when he picked up, he got
a missed.
When he called me, he got a dial tone and then after one ring he got a
Hi again!
Just a short addition: I've looked a bit closer at the buddies list as well
and I see, that I ahve no resources. Though my friend could see me and I tried
sending a text message from freetalk (jabber-client) and it worked as well. I
just thought, that might be of use.
Can
Hello Motiejus!
Sorry I'm a bit late with this, but I was quite busy. But here's the
procedure with the latest JACK svn:
svn co http://subversion.jackaudio.org/jack/trunk/jack
cd jack
./autogen.sh
[edit ./configure]
Search for;
not_overwriting=$(expr $not_overwriting +
[exclude the in
Hi Motiejus!
I'll look for JACK's configure script and send it off-list, unless someone
else here wants it?
Now about my programs. Scenario: Start Asterisk. Then directly use CLI to
dial. And if possible use asterisk only to pick up calls.
problem: I didn't find an easy way to let the
Hi Motiejus!
If all else fails for the moment, it should be quite simple to move JACK.
Move all jack applications from /usr/local/bin to /usr/bin.
In /usr/local/lib move the dir jack and libjack* to /usr/lib.
That should be it for the moment. another thing is to hack the JACK
confiugre
Hi Motiejus!
I of course menat prehacking the installation script. I can unpack my
jack-tarball or get a new svn and see, what I did. It was some time ago. And I
usually don't have to hack the installation prefix.
Information: I'm running my asterisk and JACK on a simple desktop system (no
Hi!
I tried with 0.109.* up to 0.112 or so. I once managed an ISDN call. but
with all newer versions googletalk somehow failed. There is a bugreport in the
system, but unfortunitely it still seems unresolved:
http://bugs.digium.com/view.php?id=13812
Now I'm running Jack 0.116.2, which
Hello everyone!
I haven't seen anything new about the JACK support in Asterisk and I was
wondering, if anyone has experience with a current release of Asterisk, JACK
and mISDN/googletalk etc. I'm thinking of installing a new version
(havingcurrently 1.60-beta9. But the excercise would be
Hello Wilton!
Thanks for your looking after my problems.
No I meant the usual asterisk call park. Yes, it should be independet of the
trunk. But I wondered how to activate it from the asterisk CLI? Should I send
some special DTMFs (dialing digits) and be done with it? Or should I use some
Thanks Brent! I'll have a look there in features.conf.
Warm regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT:
Hello!
I'm still wondering, how to park a call with an ISDN line. The setup is the
asterisk server only, controlled via the CLI. I can originate a call and I can
tell asterisk to start the JACK application. But I can't then park the call. I
tried it with sending DTMFs with misdn send digit,
Hello everyone!
I'm sorry I can't be more specific. So here's the setup:
a Samsung router with analog and ISDN ports. the phone company says the
outgoing line is analog landline, but I'm sure it's some VOIP.
so connected to the ISDN port of the router is a Fritz AVM card, used with
mISDN.
Hello!
Thanks Wilton! You pointed me to the fact, that my initial post was a bit
unspecific. so here's the setup (hopefully acurately detailed).
The phone company promsed a full DSL+phone package. So there's the phone
jack in the wall, which is connected to a box, which I called router.
Hello!
I've installed asterisk and now it crashes on me, here are three core-dumps
plus a note, saying which command created them and all I could gather in
/var/log/asterisk.
http://juliencoder.de/ast_debug.tar.bz2
the asterisk version running now is:
1.6.2.0.beta3
But it seems, that
Hello Wilton!
OK, now it works. The ISDN port is for voice telephony. the router, fyi, is
a piece of shit. That's a rough translation of what I found about it.
I've regressed to asterisk 1.6.0-beta9 again and I had to reconfigure jackd
a bit to work. Byut - as before when I used it - the
Hello!
I've configured Music on Hold in asterisk, the only, most certainly, stupid
problem I have is, which DTMFs to send to activate and deactivate it.
If I use the cli, I can establish a call with originate. With the misdn
send digit command I can send a number of digits to the other
Hi!
First thought: try to debug it. Use debugging options while building, if
there are any, and I believe there are.
Then run asterisk with gdb:
gdb asterisk # optionally try asterisk with full path
Then in gdb:
gdb set args your_options
e.g.:
gdb set args -c
Good luck!
Hello everyone!
This is my problem: I try to do gtalk, but my asterisk server uses the local
IP 127.0.0.1 or perhaps the 192.168.*.*.
Now I've heard, that a NAT router can help there. I was told it's the way
the windows-world does the trick, when they sit behind a
router/phonebox/modem.
I'm using gtalk.
So I can try to configure my router (it's got a lot of javascript :-) ) to
forward 5222 to my server and the same thing backwards?
Thanks for responding so fast!
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
I'm not completely sure about the things my router can do. It's from the
telephone company and it's supposed to do a lot of stuff. I've just heard,
that windows people could solve such things. After all my setup isn't too
strange or rare? Or is it for running asterisk?
Kndest regards and
Thanks for the answers. I have to read those more carefully, when I'm properly
awake and concentrated, but it sounds as if this might be of help.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
Hello everyone!
I just installed the latest asterisk from svn. Now I'm retrying my luck with
gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not
sure if it helps or hurts.
I tried this:
call myself:
channel originate gtalk/gtalk_account/julienco...@googlemail.com
Hello!
Sorry for not being able to phrase the problem in one line. My phone
situation is this:
The calls go over analog line (or NGN/vip) I don't really get to see it. I
have got a router with a lot of jacks. One or two of them are for ISDN phones
or other ISDN capable devices. Can I use
Hi!
It looks like the correct adress. I'm not sure, why this is. I'm sorry, I'm
out of it.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased
Hi!
I'm part of two mailinglists. I haven't heard much from one, but in the
other there are some mails now and then. isdn4linux and then there's a
specific misdn mailinglist (I think in connection with asterisk). I'll check
my address book if you're interested in more info.
Kindest
Thanks!
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de
___
Hello all!
I just noticed, that since installing the latest SVN branch (152803), I
receive the following error, when loading/reloading the misdn.conf file
misdn reload
[...]
[Nov 3 16:20:37] WARNING[5267]: misdn_config.c:938 _build_general_config:
misdn.conf: misdn_init=/etc/misdn-init.conf
Hi!
I think I saw a command !, which would escape to a shell. But I'm not
sure. Unfortunitely I can't look it up at the moment, because I compiled my
asterisk for full debug. Just enter your CLI and type TABTAB at the
prompt. I think I only saw this in the latest SVN.
But your client
hello everyone!
I just got the newest asterisk SVN:
trunk# svnversion
152803
and compiled it. then I made some test-calls.
1. Calling my mailbox. It worked, but quality was not good, in comparison to
1.6.0-beta9.
I called via mISDn.
2. Just call myself.
Result: Ringing and asterisk
Hello Philippe!
Would you by any chance have asterisk running with gtalk? I saw your mail
there. If so perhaps we could test. Because all others I have found either
don't have gtalk, so we tried jingle, which was still a bit problematic or if
they had pure gtalk, they weren't really upto the
Hello Philippe!
I've set the strictrtp to no. Where should I have scrambeld my address? I'm
not aware of it. The only thing that maybe is that I'm on a small LAN behind
the firewall. May this be relevant?
About the bugtracker. I've had a look there, but couldn't really find the
place
Hello everyone!
Is there a way to send a text-message using the CLI? I didn't find anything,
I could use from within the CLI, just the command of the manager and I don't
know how to use that interactively.
Kindest regards
Julien
Music was my first love and it will be my
Hello everyone!
Philippe, you told me to make a bugreport. Well, here it comes, I'm still
not sure, if tis is a bug or a miss-configuration.
So I've put up a collection of configurations/output/debug files from a
simple asterisk session testing the gtalk call.
You can download it here:
Hi!
I just tried to call a friend using jingle, but I got refused. Errorcode was
502, he tried to call me, heard it ringing once and then it stopped.
I used:
originate jingle/gtalk_account/[EMAIL PROTECTED] [application]
I'm registered to googletalk, but this should mean no harm, or
Hello Philippe!
Do I need a googletalk client? Or can I just use asterisk's originate CLI
command? I was under the illusion I could. Otherwise it's a bit problematic. I
canonly use text-based applications and they better support JACK audio
Connection Kit, for my soundcard is not simple
Hi!
There's something strange. I have entered a couple of buddies. On has Jingle
capability and two have resources (Home and Telepathy), but my own account
does have no resource, I put myself in the buddies list. Is tat supposed to
be?
And again about those ports: Accept the 5222 port, do
Well, so asterisk seems to think, that I'm not connected, for I don't see a
resource Asterisk or Talk with my name.
That shouldn't really be. :-(
Any ideas on fixing this?
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
Evening Philippe!
Here's what jabber show connected says:
Jabber Users and their status:
User: [EMAIL PROTECTED]/Talk - Connected
Number of users: 1
I'll have to ask my friends, what their clients say. Although I suppose as
my friend already send me a text message he
Hello everyone!
Perhaps I missed something: But where can one download the beta-version of
the new asterisk skype channel? Can it work with 1.6.0-beta9?
I tried to browse the digium downloads, but it's dificult, if you're blind
and only have a text-based (almost no javscript) browser.
Hi!
So no way to get in anymore? that's too bad. It would have been the first
real skype accessibility for blind people working like myself on linux.
If anyone does remember or can retrieve the URL for the signup form, please
tell me anyway, perhaps it's still possible.
Kindest regards
Hello everyone!
Would someone be willing to test googletalk with me? Please reply in
private, I'll get you my account and we can exchange anything else, that's
necessary.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND
Hi everyone!
I couldn't find anything expressive about gtalk dialstrings. It doesn't seem
to work. I'm not sure why, so I'll start at the easiest point.
The syntax I found was:
gtalk/my_account_name/[EMAIL PROTECTED]
Is this correct?
And does any of you googletalkers know, if a simple
Hi!
OK, this seems to work. But still I can't find anything to talk to for a
test. I don't know, if the firewall might be in the way. Isn't there some
echo-test service for google?
And about the buddies: I found usernames in two places
1. in jabber.conf
[gtalk_account]
...
[EMAIL
Hello everyone!
I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not
working. Here's what happens, if I try to call the line:
bach P[ 1] -- !! lib: No free channel!
P[ 1] -- we have already send Release_complete
I haven't changed the configuration fles. Should I
Hello Robert and all others of course! :-)
Here are the ringtones (ring files):
http://juliencoder.de/ringtones.tar.bz2
All are in .wav-format, once in CD-quality and once in phone-line-quality.
All means all 6 of them. If the console/dsp only plays gsm or some other
special format, which
Hi!
One further notice about ringtone6: This existed long before today. It's
called schon wie-der drei-zehn To-te, the dashes are there to mark
syllables. The translation is: Again 13 dead people. the the melody of a
German radiostation's traffic news. A German commedian came up with the
Hi!
I have a different approach. I wrote a small application which simply starts
an audio player. You can write a very small script to answer fast or just use
telnet like this:
telnet localhost 8642
At the moment everything is hardcoded, but can be changed in any case. I use
15s
Hi!
I just uploaded a small tarball of my ast_picker application with a few
extras and an example_dialplan. You can find it here;
http://juliencoder.de/ast_picker-0.1.tar.bz2
As I said before: It's still early stage and not too customiseable, but you
can manage. If you need help, just tell
Hello Robert!
I don'texactly know, what you need for a ringing file. but if it is the
matter of just some announcement sound, I could make you one. It's easy.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT
No problem... I'll whomp something up. I'll upload a tarball tomorrow or
thrusday morning at the latest.
Quality: desired samplingrate, bit-depth, channel number? Any particular
needs, or will CD quality just be fine for you?
Kindest regards
Julien
P.S.: Did you get to my
i!
Not about this directly, but an alternative. If you need the length of
finished calls, work with the system. Use a specific call to the date command,
so it's easy to evaluate the time info or some other tool to give you an
absolute of time. Then at the end of the call use another system
Hi!
I think all - at least all PSTN - calls have the same quality in means of
bitrate, number of channels and samplerate.
It's 8kHz, 16bit and mono.
About noise, I didn't have problems with that. Seems it's not really about
quality. Probably it would be helpful, if you tell us, which
Hi!
I'm also still new to this. but perhaps:
Did you do make examples, or did you have an earlier asterisk installation,
so the configuration files were present? If so did you make sure, that you
mISDN card was properly configured, using:
1. misdn-init scan
2. misdn-init config
3.
Hi!
I wouldn't know a proper way to check for off-hook. But, couldn't you change
your dialplan?
1. answer the call
2. check for CID
3. branch with a gotoif
4. Enter CID
5. Look up CID in your DB and whatever
6. Playback the mainmenu welcome
[go on]
Something like this?
Kindest regards
Hi!
for which feature? I'm relatively new, but I guess, if it is dependent on
some dialplan related stuff, you coudl always use a:
System(soxmix Options)
in the appropriate place. From what I've experienced upto now, you can setup
a lot, which doesn't seem obvious.
Kindest regards
Hello!
I'm currently rebuilding my dialplan. Now I have the following
[Official]
;statements
[Extern]
include = Official|8:00-18:00|mon-fri|*|*
;other statements
when I want to reload the dialplan asterisk tells me:
CLI [Sep 16 17:38:10] WARNING[5125]: pbx.c:7981
Thanks!
That did it.
Plea for documentation update...
Kidnest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL
Hello!
Thank you all! The VPN-solution sounds nice. If I can find a provider/friend
offering that, it might be better then IAX, for I know, that a lot of people
have SIP and thus it would be easier to communicate.
In addition, I found free SIP-providers offering SIP conversations to people
This is cool! So that's the way to do things in parallel...
Btw.: saw you on the LAD today, Russell.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased
Thanks! You're the best!
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
Russell!
This time it's really a problem:
when I use application Jack I get input and output. When I use
functionJACK_HOOK with the same options, just copied from the Jack call, I
only get one way. the o-option doesn't work. I connect it to my microphone,
sstem:capture_1. So nothing
Some addition...
Something I find even stranger is that jack_lsp shows, that the asterisk
input AND output ports do exist and ARE CORRECTLY connected. So I should get
audio from my microphone and still I don't.
Hope that helps...
Kindest regards
Julien
Music was my
Hello Russell!
Certainly, here's the shortened dialplan:
exten = NUM,1,System(ast_picker ring.wav)
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${SYSTEMSTATUS} = SUCCESS]?4:7)
exten = \
NUM,4,Set(JACK_HOOK(manipulate,i(sstem:playback_1)o(system:capture_1)=on)
exten = NUM,5,System(ast_connect)
Hello!
I'll classify the subject. :-) I have a nasty firewall, I don't have to much
power over. It's javascript based in configuration and I can't use any
graphical browser. The only other person at my home, doesn't know too much
about computers.
So I know, from experience, that SIP is
Hello!
IAX I can basically understand, although I wasn't aware in the slightest,
that other standard softphones supported it.
But why SIP? Correct me if I'm wrong. there's a standard SIP-port. Then you
send out the request to talk, then server and client negotiate a port for the
audio
Hello Stefan!
Sorry for the miss-understanding. I didn't refer to your mail about IAX, but
about the one sayng SIP. I read your links and it seems I'll delve into it.
I'll try to quote next time. I hate doing this, it always looks a bit
unorganised, while writing... :-(
Kindest regards
Greetings!
I have got a systematic problem with the SHELL function. Consider this
dialplan snippet:
*** CUT ***
exten = NUM,1,Set(myreturn=${SHELL(ast_picker sound_file)})
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${myreturn} = 0]?4:6)
[...]
*** CUT ***
ast_picker does simultaneously
Hi!
Thanks! I changed course and reworte the program code to interoperate in
other ways. Now it works.
Is there a way to do something based on some other phone taking a call? Or
if the caller stops ringing?
It's too bad asterisk can't run applications/functions in parallel...
Kindest
Hi!
AGI... Oh I was never good at perl. I'mhappy, that I finally completed my
personal ringing application as a standalone program.
Though I'd be only too willing to share the code, if someone wants/could
make an asterisk application of it.
I think I may still just bore down on AGI and
Greetings!
Does application Jack run the whole time, the conversation is going?
If so: is there a SIMPLE extensions.conf-only-based way to put it in the
background? I know AGI and other applications... :-(
Kindest regards and thanks
Julien
Music was my first love
Hello!
I just wonder, I've uncommented a few part of my features.conf, to park
calls and the like. But now I wonder, how it can be done? I tried the #72
(park call) from both asterisk with
CLI misdn send digit mISDN/1-102 #72
Never mind if the channel-name looks wrong here, asterisk
Hello!
I wondered could I (mis)use an AGI program to decide if I pickup. At the
moment asterisk has to pck up, when the ring tone has stopped playing.
The dialplan looks like this:
*** CUT ***
exten = NUM,1,System(mplayer file /dev/null)
exten = NUM,n,Answer()
exten =
Hello again!
Thanks Sven, for your private advice. My question now is: If the Fritz is
going out of bussiness: Which card to purchase for the future?
I have a very small budget, a simple european ISDN line (three numbers, two
similar channels?). I want to use it for asterisk.
Features
Hello!
I'm wondering which is the best choice (kernel version and mISDN) to get my
AVM Fritz A1 PCI card to work properly?
Does anyone have an AVM Fritz runningunder Linux? Or has anyone deep
knowledge of mISDN? Please I need some hellp here, for after a whole lot of
testing, reading,
Hello!
Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple
ISDN-card, now finally running. :-)
I'm using application Jack and asterisk (CLI) only to do my bidding. Now I
can make calls. But how ca I setup my extensions.conf to receive a call? I've
had an example like
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