Hi,
did the digium registration server for g.729 change
or is it currently just down?
Anyone else has currently problems registering
G.729 channels?
Roger.
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Howard Lowndes schrieb:
...
Just was wondering if anyone else was able to reach them today.
No, their G.729 registration server was also down today,
and even the email I sent today return with routing problems.
Roger.
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.
You can expand this example with further fixed ip address
and further users, which authenticate via your gatekeeper.
Roger.
--
Dr. Roger Schreiter
PLANinterNET GmbH
Geschäftsführer
Hauptstrasse 6
D-74391 Erligheim
Tel.: +49 7143 408091-5
oder: +49 7143 87205-6 (Vermittlung)
Fax: +49 7143 87205-7
Andreas Czerniak schrieb:
...
Has anyone an usable (capi.conf?) configration ?
Hi,
I didn't yes investigate your logs, but I'm sending you
my capi.conf for the C4-card, which works fine. Maybe it
helps.
Roger.
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
Roy Sigurd Karlsbakk schrieb:
...
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a good
choice, but this seems sold out. Does anyone know about another book
Hi,
since it's sold out for a longer time, I sold mine used at
amazon.
Roger.
Hi,
whatever dialplan I'm using for outgoing calls via
PRI (Digium card, chan_zap), the callerid when receiving
calls has no leading zeros, which are normally used to distinguish
local, national and international calls in Europe.
The number has always the area code in front, but the
country code
Hi,
I'm in the need to use a line extender. I found a device, which
should best be placed in the middle of the line, which would make
power supplying difficult, unless I'd use its telco power supply
features:
The device is capable of taking current from one signal wire pair
or voltage from the
Hi,
I put a
pickupgroup=0
line for each user in sip.conf.
After restarting asterisk I called my collegues phone
with my cell phone, I heard it ringing and saw ringing
in the asterisk console.
Then I dialed *8 with my phone and got on the console:
Jan 24 20:41:45 NOTICE[13747]: chan_sip.c:7321
Carlos Chavez schrieb:
I want to buy a new server to run Asterisk and after looking at prices
for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed
rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64?
...
Yes, it will compile and work.
Roger.
Hi,
I'm using mysql to define my sipfriends.
When authenticating, the calleridname from the calling
SIP user (phone) seems getting lost.
With sip debug I can see in the SIP messages:
From: myName sip:[EMAIL PROTECTED];tag=22668125
To: sip:[EMAIL PROTECTED]
but I can't find myName in any channel
Hi,
I have some mySQL-sipfriends and connectivity to PSTN.
When a call from PSTN comes, it shows a callerid,
and that callerid is displayed at the called sip phone.
When the call comes from another sip user (defined as
mySQL-sipfriend), no callerid is displayed at the called
sip phone.
I turned on
Hi,
this seems to be a bug in chan_sip.c (asterisk-1.0.3).
Whenever a column restrictcid is defined, the
variable restrictcid was set to 1 in chan_sip.
Now I changed line 1067 to
u-restrictcid = *(rowval[x])-'0';
Now restrictcid really reads the value from the database.
Thus setting this value in
Ginel Tudorache schrieb:
Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to
find a working combination between asterisk,asterisk-oh323,pwlib and
openh323.
Hi,
I'm using SuSE, not Redhat, but imho you'll succeed, if
you strictly follow the version hints mentioned in
Calin Serbanescu schrieb:
...
e164 numbers in my h.323 network and my ISDN provider doesn't accept
those identities (CIDs). So, i have to spoof the outgoing CID depending
on incoming CID.
Is there any possible way of doing this by AGI? How? any examples are
welcome.
Hi,
I'm not sure, whether
Calin Serbanescu schrieb:
...
the net about this... do you have any link to such script ?
No, I don't have such a link.
But on the voip-wiki pages there are some examples
for agi-scipting.
There are APIs for some common languages, e.g. Perl,
which is maybe one of the fastet ways to code simple
Calin Serbanescu schrieb:
unfortunatelly i have to accept their terms and rewrite caller id... but
again, i am newbie in scripting with agi and i can't find any example on
the net about this... do you have any link to such script ?
... what I should maybe also mention:
My script in the recent
Robert Rozman schrieb:
...
centrala:~/Asterisk/h323/asterisk-oh323-0.6.5 # make
for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory
Hi,
I have installed chan_sip on asterisk-1.0.3 / 5 (tried
both, same result).
My sip phone registers fine.
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
...
Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686
Hi,
I'm still trying to understand that Unable to create/find channel
problem on chan_sip, I've never seen on my other gateways.
Please can anyone tell me possible reasons for too old packets,
and maybe how to avoid them!
I'm using the same SIP with other asterisk gateways without problems.
Hi,
I'm installing asterisk 1.0.5 for a partner in China.
Since the ping round trip takes typically 600 msec, I doubt,
whether voice quality will we satisfying, but that is currently
not my concern.
The problem is, that most SIP phones or software (e.g. SJPhone)
do resend the invite request, after
Michael Manousos schrieb:
asterisk-oh323-0.7.0 is for Asterisk CVS.
How did you manage to compile it with Asterisk-1.0.3?
Hi,
sorry, I just checked again: I'm using asterisk-oh323-0.6.4
on that machine, where asterisk-stable runs.
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Ok, I will try it
Michael Manousos schrieb:
...
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Hi,
may I ask, whether that combination runs really stable
at your machine?
I have now those versions installed.
I have asterisk crashes at least once every hour, when
several simultanious calls take place.
Roger.
Hi,
I've installed some asterisk based SIP gateways for some
VoIP providers during the last weeks. Since those providers
want to allow their customers free calls to subscribers of
other VoIP providers, I had to put the provider codes into
the dialplan, like +0700 for IAXtel.
I found it helpful
Dragos Ungureanu schrieb:
...
The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
accountCode=AN_APPROPRIATE_ACCOUNTNAME
into your oh323.conf!
Roger.
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Hi,
it is the same hardware, but with a firmware by Brian F. G. Bidulock.
It has nothing to do with the libisup project, Steve Underwood wrote
several times within this mailing list and soon will be made public
as SS7 support for asterisk with that Digium card.
Roger.
Martijn van Oosterhout schrieb:
...
There was also an SS7 status report[2] last June but it's doesn't seem to
have lead anywhere either. There was post saying an SS7 release was
immenent last September[3], but then silence.
Hi,
yes, in the beginning, when we looked for a SS7 solution
for
Hi,
I also had some problems using chan_oh323 together
with gnugk.
* - gnugk - h323-phone
When I called the phone and hang up, befor the phone
was picked up, the h323-phone continued ringing.
The same, when the h323- and some sip-phones were
called, and the sip-phone picked up the call first.
(It
Hi,
does anyone know hardware, which supports an S2M
(E1/PRI) via C7 (CCIT7) (instead of DSS1) and is
supported by asterisk?
Roger.
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Hi,
one month ago, I announced, that I will look at the openss7
project in order to use it together with asterisk.
It took a while for me to check the capabilities
of and around the project. Since the openss7 project consists
of only one person, and when there was just silence for a long
time in
Senad Jordanovic schrieb:
...
Are you aware of bounties posted to get SS7 working with *?
If not look at http://bugs.diuim.com .
...
No, I'm not.
Unfortunately, the link you mention, does not work.
I tried http://bugs.digium.com and found a login form.
I logged in with anonymous and looked
Michael M. Saunders schrieb:
I am willing to donate some programming time
Fine!
I will tell you, when one can start coding.
Roger.
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Hi,
I've read your question also in the OpenSS7 mailinglist,
and I hoped, Brian F.G. Bidulock would answer. Brian is the
expert concerning SS7 support of those cards.
If I have understood right, Brian developed for his company
(OpenSS7.com) a special firmware for the digium T400P and
sells it now
...
I've replaced extension name s by _. and now
everything is fine.
Nevertheless I'm confused, since I think I have
already used s as a synonym for _. (asterisk 0.5 or 0.7)
and it also worked fine.
Maybe I did not remember right. It works now fine anyway.
Roger.
Keith Waters schrieb:
...
I configured:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
I consider a short key for [1-9] at least
as useful as N for [2-9], maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short
for [1-9]?
Hi,
I have configured chan_oss and chan_capi (asterisk from CSV on Monday),
and I can dial from the console:
*CLI dial 830774
-- Executing Dial(OSS/dsp, CAPI/872058:b830774|60) in new stack
-- Called 872058:b830774
-- CAPI[contr1/872058]/3 is making progress passing it to OSS/dsp
...
...
Certainly not! :-) Create a [request] entry at bugs.digium.com
Done.
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Aaron J. Angel schrieb:
...
Does Z not work?
...
Yes, it does. I had to look in the most recent docs first.
Sorry!
Roger.
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Action Command, which I just discovered, is now my friend:
Action: Command
Command: Dial 123456
seems to fit well for a softphone application
with the local soundcard.
At least it has solved my problem and works well for me.
Roger.
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Joseph schrieb:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
...
Hi,
your question is not very asterisk related. So you can
use any signal converter on the telephone device market.
I you are looking for something below 10 kEUR, there is not
much choice, e.g.
-
Hi,
there are still some questions to be answered by OpenSS7.com
in order to decide, whether E400P-SS7 is a good choice for
the asterisk SS7 support.
In the meanwhile I'm also in negotiations with another
manufacturer (whose name I currently may not tell due to
a NDA) of SS7 hardware, who gets
Hi,
I'm using asterisk with chan_h323 together with gnugk.
chan_h323 and gnugk were recently compiled with pwlib-1.5.2
and openh323-1.12.2 as advised.
When connecting asterisk directly by ohphone
(without gatekeeper), everthing is fine.
When using gnugk for usage control in routed mode, I find
a
Kai Militzer schrieb:
...
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile
Roger Schreiter schrieb:
I have currently the same problem with my E1 card and I wonder,
...
SetCallerID(0${CALLERIDNUM})
O.k. this works fine for me too.
I hope, I won't have to take special care, when
calls came from local or from international.
Roger
Dennis Cartier schrieb:
Does anyone know if the 'externip=' in sip.conf is resolved just once
at startup or on an on going basis? I would like to use a DNS name
Hi,
at least at chan_h323 it does resolve only once
at startup.
Any method to get asterisk looking up certain
H323 hosts would be
Hi,
I have to develop a phone application using asterisk's
chan_oss.
When the phone is idle, i.e. the last command was a hangup,
one hears a toot, toot, toot, ...
But unforuntaly its use is in Germany, where one expects
a continous too ...
before dialing.
Is there
.
Roger Schreiter schrieb:
...
I'm using asterisk with chan_h323 together with gnugk.
chan_h323 and gnugk were recently compiled with pwlib-1.5.2
and openh323-1.12.2 as advised.
When connecting asterisk directly by ohphone
(without gatekeeper), everthing is fine.
When using gnugk for usage control
Hi,
I'm using asterisk as softphone for a certain application.
It uses chan_capi for PSDN connection and chan_oss and
the manager as user interface.
When calling someone, who is busy, I can hear at the
speaker the busy indication, but the manager command
Status still tells Ringing (chan_oss) or
Hi,
is there any mean to get ringing tones for the
E100P, according the signals from the telco?
When I use the r-option in extensions.conf, I have
too early ringing tones, i.e. when busy, I have
ringing tones for a while and afterwords the busy
tone.
If I don't use the r-option I have no tone at
Hi,
I have asterisk configured for german ISDN
and SIP. SIP only for intranet connections.
In our office there is a snom 100 and a snom 200
phone.
When I'm calling a (public) telephone number
which is busy, asterisk chan_modem hangs.
Busy is never indicated to the calling SIP phone.
And
Hi,
I have a problem with asterisk-0.5.0 which I don't
understand. The monitor says when making a call:
*CLI -- Executing Dial(SIP/roger-c456,
Modem/ttyI0:BYEXTENSION|60|tTm) in new stack
-- Called ttyI0:1234567890
WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
Tais M. Hansen schrieb:
...
Now that I've been unable to register 2 hardware SIP phones and one software
(Kphone), I'm beginning to doubt that chan_sip works at all.
...
Hi,
we have 2 snom phones running with sip. (Asterisk-0.5.0).
The sip part seems to be very stable.
We have currently some
WipeOut . schrieb:
I have the same but everything still seems to be working so I haven't worried about
it.. maybe there has been an extention to the SIP protocol??
...
me too.
Roger.
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Tilghman Lesher schrieb:
I'm trying to get a Snom100 configured with H.323. Right now, the
...
Hi,
I had Snoms (100 and 200) configured with H.323 working with
asterisk-0.4.0.
Since I upgraded to asterisk-0.5.0 and I had problems with H.323
I switch to sip.
(The problem was: when I
Hi,
my asterisk experiences with isdn cards supported by i4l
are not very good, but with avm a1 and capi everything
works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20,
german ISDN).
Now I want to connect a T1. Should I use an AVM T1-B
for approx 6000 EUR or is it ok to use one of Eicon's
Anthony Wood schrieb:
...
If you are in Germany, I think you want E1 = ISDN PRI outside US/Japan (30 lines)
and for that you want a E100P for US$595 from digium.
Hi,
that would be very interesting, of course. I assume, there are
experiences about stability and reliability under high load
and
Hi,
it is now 3 months ago, that I told here, I were beta testing
SS7 for asterisk.
I promised to give a result afterwords - here it is:
There are still some minor problems (maybe more a zaptel
or hardware problem that a SS7 one), but in general it is
running very stable.
I assume the author will
Hi,
I have one asterisk box with chan_oh323 to an external
carrier.
In order to have G.729 with this carrier, I use:
setGlobalVar(OH323_OUTCODEC=g729)
I have that asterisk box connected to another asterisk
box via IAX, with
disallow=all
allow=alaw
in the respective peer/friend chapters in
Jay Brussels schrieb:
...
When trying to upgrade to 1.0.2, I get several compile warnings such as:
chan_zap.c:3515: warning: implicit declaration of function
`__use_ast_pthread_create_instead__'
Hi,
I assume you mixed some parts of an older asterisk version with
the new asterisk kernel. Did you
Hi,
I've got an IAXy, and provided it with connection
data - unfortunately didn't provide a fixed ip address,
but left with dhcp.
Everthing went fine, and I could work a while with it.
After a restart of IAXy, it didn't negotiate a new
IP address, nor used the old one. I even restarted
the DHCP
Hi,
after loading the zaptel driver wct4xxp I have strange
log lines in the syslog:
Out of storage space.
free tells, that more than 900 MB are still free.
Disk space is also available.
I'm using a dual opteron in 64 bit mode.
Any ideas?
Roger.
Syslog:
Dec 1 02:18:37 ipphone4 kernel: TE410P:
jim wrote:
In this particular case what you'll probably need to do is run a sniffer
program such as Ethereal to see what the IAXy is up to. Whatever state
Yes,
worked perfectly. It had an ip address in
the 192.168.0. network - I assume its default.
Roger.
Whisker, Peter schrieb:
Partly is is down to the fact that G.711u (mu-law) is primarily used in the
USA and G.711a (a-law) is used in Europe.
Like you, I am not sure if the exact differences - they have the same
bitrate and audio, although there are minor differences in the format.
Hi,
it is just
Eng. Vanzetti Walter schrieb:
...
The cdr table is the same reported in Wiki's link.
where is the problem?
.
...
Hi,
when everthing is according to the Wiki's mysql_cdr pages,
it should work.
Please check permissions!
(Insert permission granted to the user as defined in
cdr_mysql.conf? Even if
Hi,
I intend to change the cdr_mysql-field uniqueid,
which seems not to be used so far, to an (not unique)
indexed field and use it later for my own hints and infos.
I don't have very much traffic so far, and I wonder,
if there will appear problems when asterisk is under high
load (100
Jay Milk schrieb:
...
note, you probably wouldn't re-use existing fields (whether they're used
or not) but rather add new fields if needed. In order to keep
...
I assume, I may not add nor change fields to the cdr table,
which cdr_mysql uses. Maybe I should consider patching
the source code in
Hi,
has anybody out there experience with those
server grade connections?
What hardware do you use to connect your asterisk
box to a PSTN carrier via C7/SS7 (instead of ISDN PRI)?
Thanks for any hints!
Roger Schreiter.
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Hi,
I would like to define a h323 user with serveral
ip address, like:
[roger1]
type=user
host=217.94.99.216,217.94.99.217,217.94.99.218
context=default
accountcode=schreiter
Ok, the above sample does not work.
Is it possible in any way to define a h323 user with
seveval (but not undefined) ip
W. Kevin Hunt schrieb:
I am looking to integrate * w/ ss7 also.
I've looked at the openss7.org site, but there are not integration
...
Hi,
the project asterisk-ss7 seems to be stalled.
I'm currently verifying, whether there is nevertheless
already the stuff inside openss7 to build a preliminary
Hi,
maybe, now there are really enough mirrors.
If not, or another european mirror is appreciated:
(including zaptel)
http://voip.planinternet.net/asterisk
or
ftp://voip.planinternet.net/asterisk
Roger.
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Benjamin on Asterisk Mailing Lists schrieb:
...
Is there a release of the zaptel drivers too for 1.0 release?
...
People always seem to forget the Zaptel drivers when they put up mirrors :-(
...
I asked John Bigelow from Digium about that in relation to RC1 and RC2
and he said we should only use
Klaus-Peter Junghanns schrieb:
Hi,
if i understand german telco regulations right (even for a german that's
not an easy task...) then a provider may not assign a DID to a non-local
Hi,
it's right, that german RegTP, the authority, who assigns number
ranges to telcos, now explicitely forbids to do
Alfred Nurnberger schrieb:
Try sipgate.de.
They have free DIDs in many german citys and their rate into Germany is
very affordable (aprx. $0.02 / min.)
...
Hi,
I'm sure, they won't anymore. For it were Sipgate and Nikotel,
who got those letters from RegTP, forbidding to give DIDs to
non-local
Hi,
when I do ps aux | grep asterisk I can find very different
results on my various machines I'm running asterisk on.
One of my machines just shows one asterisk job running,
others are shown approx 12.
Is it defined by a parameter?
Does it affect quality or reliablity?
Roger.
Hi,
the structs ast_channel and / or ast_callerid seems to have recently
changed, and now chan_capi does no more compile.
I commented lines 1073, 1072 and 1724 in chap_capi.c and compilation
succeeded, and - good luck - I didn't comment essential things,
we just don't have any caller id infos
Peer Oliver Schmidt schrieb:
...
as I am living in Germany, let me advise you against using VoIP
providers in Germany. Most of the time they do work, but they are not
as reliable as a regular phone company.
Hi,
on the one hand, I should ignore this thread, because it is not
asterisk
Hi,
incoming SIP calls have a channel name in the form of:
SIP/ip-adresss-of-peer-handle
This is a way to get fetch the IP address of the remote side
of a SIP call - in most cases.
However, sometimes, instead of the IP address, there is a host
name in the channel name. I assume, this value in
Steve Sobol schrieb:
Low, Adam wrote:
We can offer SIP based VoIP call termination in The Netherlands,
Austria and Norway. If you'd like to speak to an account representative
please contact me personally by email.
Hmmm, this information should be on a website somewhere...
Hi,
I have a new phone in our IP phone network: Planet VIP-101T.
When calling from that Planet phone to anybody, everthing is
fine.
But when calling from anybody to that Planet phone, I
get a mashine gun noise and the following msg in asterisk log:
NOTICE[262161]: File channel.c, Line 1091
Hi,
I just bought the E100P from digium. It has both
keys: 3.3V and 5V, so it would fit both, in a 5V-PCI
slot and in a 3.3V PCI slot.
Is it true, that I can plug it without destroying it in an
ordenary 5V PCI slot?
Roger.
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Hi,
I have asterisk running with gnugk.
When a call comes via gnugk and is
cancelled befor the called party picks up,
the called phone continues ringing, asterisk seems to
ingore that the call was cancelled.
In the gnugk-log I find:
DCF|192.168.1.1|1237_endp|29659|normalDrop; (immedeately after
Hi,
I currently have a yellow/red alarm on one span of a Digium card.
It is not the first time, this already happened some
months ago, and I expect to clear the alarm when
rmmoding and insmoding the zaptel and wct4xxp modules.
Unfortunately I can't rmmod while asterisk is running
and I can't
Hi,
app_milliwatt is a nice tool for a quick check of the
line quality.
Anyway, hearing to that tone for more than a minute is
painful.
Does anyone know the opposite application, i.e. an
application, that hears and listens for a 1000 Hz
tone and displays the quality in any unit?
If not, I'll
Matt Roth wrote:
...
What is being discussed here is basically what I was planning on
...
This sounds like a programming project. Something like a stripped
down softphone (or possibly a plugin to an existing phone) with
Hi,
since I need rather a tool not that versatile but within some
Douglas Garstang schrieb:
I'm trying to find a way in extensions.conf to match ANYTHING dialled,
Hi,
your subject is probably not correct. You want to catch
anything except h, t, ...?
Maybe you want to get matched the digits and *.
Thus try:
_[*0-9].
This will match any dialed string,
Douglas Garstang schrieb:
If dial() doesn't return until after the call completes,
it means the channel status AGI command is a waste of time.
Hi,
you are right, dial will block, so you won't get the channel
status by that method when having an outbound call.
You can use the manager. But
Douglas Garstang schrieb:
...
HOWEVER, if the CALLER hangs up the call, it seems
Hi,
did you try the dial command option g?
I did not neither, but when I understand the voip-wiki right,
it might help you.
Roger.
Voip-wiki page about dial:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Hi,
some days ago we discused here the need for an analyzer
for the 1000 Hz tone, as opposite application to Milliwatt.
Here it is: Mwanalyze
http://planinternet.net/download/voip/asterisk/app_mwanalyze.c
It performs a Fourier analysis for a fixed frequency
and tells the amplitude.
The
Hi,
when I get a SIP call from an unknown user, I can
see the IP address in the channel name.
When the call comes from a known user (sip friend),
I can see only the username in the channel name.
Ok, most users will use the IP address, which they also
register, thus can be lookup up in the
amaury BOSSE schrieb:
Is there a free linux tool which can test voip call quality between two
Asterisk PBX.
It will help me to test the WAN network between them.
I have only found commercials ones, so if you know a free one, let me know.
Hi,
just some hours ago I published in this list:
Juan Carlos Castro y Castro schrieb:
Could I use this to distinguish human voices from machine beeps and/or
ambient noise etc, by (after a few adaptations) calling it a number of
times on the same set of samples with some representative set of
frequencies? Or is there a better, less
Ed Nuñez schrieb:
Is anyone else having trouble going into voip-info.org today?
Yes. Me.
Roger.
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Hi,
I'm looking for a mean to send digital data over
an E1 line, just like isdn4linux or Capi via AVM's FritzCard
is able to do it with BRI lines (e.g. for PPP or ISDN raw
connections).
I'm not looking for modulated audio data representing
digital data, like fax or the analogue modems of former
Shane Spencer schrieb:
point to point E1 lines? Or are you interfacing to a PSTN network for
local calling/receiving?
Hi,
yes, PSTN. Normal operation is ordinary voice.
Hm, the hybrid configuration mentioned in your link
may serve as a workaround anyway. I should read this further.
Hi,
I'm using app_pppd with a Digium-PRI-card for PPP connections.
I had some strange problems with some IP packets passing
and some not, e.g. ftp login went well, but as soon as
I tried to up- or download a file, noting was transferred.
I finally guessed, it must have to do something with the
Eric \ManxPower\ Wieling schrieb:
ICMP is used to determine maximim packet size. If you or the other end
are blocking all ICMP then MTU Path Discovery will not work. It's a
Hi,
the problem is, the other side (ISDN-router) does not negotiate
the MTU while setting up PPP. I can see this in
Hi,
I figured out, that app_pppd suffered from
overruns under high out traffic.
(ping -s 600 destip was already high in this context.)
I've just made a quick and dirty hack to fix it.
If interested, just download the original package
by Sirrix (as mentioned on VoIP-Wiki) and the replace
their
Jeff LaCoursiere schrieb:
Is it ready for prime time?
He Jeff,
at least version 1.6.0-beta9 was not yet very stable.
We are also used to handle serveral Mmin/month with
asterisk 1.4, but in our test environment, our asterisk
1.6.0-beta9 consumed file handles without releasing,
and even a
Ricardo Carvalho schrieb:
...
tries with the following codec preferences like G.711. On the other side
there is PSTN, as I deliver my traffic in IP to a Telco that uses also
Hi,
that is not passthrough! You will need something to translate T.38 to
one of the ordinary fax/modem-modulations,
Matthias Fechner schrieb:
...
I use here mgetty+sendfax with a modem to receive and send fax
messages. Is it possible to receive and send a fax with asterisk
directly?
Hi,
did google for asterisk and fax show no results?
Strange!
Ok, what you need is Steve Underwood's package
spandsp and
Matthias Fechner schrieb:
...
yes I found spandsp but it will do everything in software.
Is it not a good idea to use my modem for the fax stuff?
Hi,
ok, you want to use an external faxmodem?
Something like that:
outside (PSTN or anythin else)
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