Hello,
we are running an Eicon Diva Server card with chan_capi
and Asterisk-1.4.8.
When we put in capi.conf softdtmf=off, the local command
read() is recognizing dtmf digits from cell phone and from
ISDN phones and from VoIP phones (via PSTN) very well,
and asterisk is forwarding those digits
...@192.168.1.150;party=called */somewhere ??
Jonas.
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Tel.: +49 7143 36476
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Hello,
the SIP header now should be sent. What the remote device
is doing with this header, or whether the syntax of the
header is as the remote device expects it, is another
question.
You can check with
sip set debug on
whether the header is now sent as you expect!
If it does, I cannot tell
Hello,
if the remote side (the public IP side) is capable to do
something like asterisk's nat=yes (in sip.conf), than
a mascerading router (like every cheap DSL router) would
do enough NAT do let SIP work.
If the remote side does not support that nat-hack (which
is not SIP standard), than you
Nivin Kumar schrieb:
Is there a tool that will allow me to automatically change sip headers
in realtime?
Hi,
imho changing the SIP headers will not be sufficient, since
the old IP addresses are now private IP addresses (only in
your network, outside, there are still public, but pointing
not to
...
Hi,
I've found the solution.
I remembered, that with IAX2 - DAHDI everything is fine.
Only SIP - DAHDI showed the problem.
It seems, that chan_sip does not open ealry audio,
if progressinband=yes in sip.conf.
progressinband=no is needed for early audio.
Strange!
Anyway, that's ok for me
Hello,
if have a problem since I switched to asterisk-1.6:
When making an outgoing call through chan_dahdi, I
cannot hear anymore early audio, the asterisk generated
sound (as defined in indications.conf) is played.
Thus, I cannot hear announcements by the operator,
and when the line is busy,
Ken D'Ambrosio schrieb:
...
pretty pricey. Is there any reason that a BRI can't do exactly the same
stuff, but on 2B+D instead of 23B+D?
Hello,
this depends on your operator and the telcom regulation in
your country.
In Germany, the main difference (besides the number of channels)
is the
Hello,
I do remember having read some weeks ago something about
a virtual device provided by asterisk, behaving like
an ISDN device, i.e. like /dev/isdn0.
I know iaxmodem, but iaxmodem imho unfortunately does not transport
raw ISDN data (HDLC frames), but only voice.
Do I remember right, and
. Is
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Tel.: +49 7143 36476
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Hello,
I've recently bought a new Sangoma A104d PCI-card.
WANPIPE release is 3.4.7.
Machine is a dual Xeon with Debian 5.0.3.
Asterisk is 1.6.11 with recent libpri and dahdi.
When I boot the machine (including hardware and wanpipe
and dahdi drivers) and start asterisk, everthing runs
fine for
Hi,
mea maxima culpa.
I've found the cause, however Sangoma card and driver
are working excellent.
Just for the archive, to help someone else, maybe having the
same problem:
In order to test my system before bringing it to the data
center, I plugged in a loop cable. Therefore I had to
switch
/dahdi/4 ==
Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4
If there are no alarms there, the wanpipe driver probably did not report
them to DAHDI.
Probably. But why? (When I turn of wanpipes, I can
see them disapear with dahdi_tool.)
How can I investigate the reason?
Roger.
--
Roger Schreiter
Hi,
I'm using Sangoma's wanpipe together with dahdi, all
software downloaded today at most recent version.
Hardware is Sangoma A104, a 4xE1 card.
Installation went well.
Anyway, wanrouter status shows a different result than
dahdi_tool or dahdi_scan.
I've just put a hardware loop on port 1. All
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
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
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 just installed Antonio Gallo's agx-ast-addons package
in order to use app_txfax with asterisk-1.4.
Compiling according to docs went well.
However, I'm getting an error after the first page
of fax:
/usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec:
Transmission loop error
The (very
Jaswinder Singh schrieb:
Can you post the part of your dialplan which causes this behaviour
Hi,
I've found, what's causing the problem:
My dialcommands are always of the type:
Dial(IAX2/user:[EMAIL
PROTECTED]/12345678,120,gS(${maxduration})M(connect^${some_params}))
or
Dial(SIP/[EMAIL
Hi,
some months ago, I had the problem with an asterisk-1.4.x-
Version, that some calls (but not all) were interrupted
64 seconds after connect (a call limit of 86400 seconds
was installed using the S()-parameter).
It was just a test machine, and later, I switched to callweaver,
and the problem
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
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
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.
Virmones Pereira Tavares de Miranda schrieb:
How to configure asterisk and zaptel for ISDN ? EWSD?
Hi,
below is the ISDN part of my zaptel.conf.
Imho crc4 is software selectable in EWSD, thus
ask your provider! The D-channel could be found
at another location, thus ask your provider!
For
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)
|
V
asterisk box
|
| (via
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,
Ronald Wiplinger schrieb:
Is there a table available, which tells me if a zip code, city and area
code matches?
I doubt, that such a table does exist.
Imho you will have to look for individual tables for each country.
For Germany, look at:
http://w3logistics.com
Roger.
Thameem Ansari schrieb:
...
understand why do they give this 01801 if it is similar to local number?
I am also exploring the options to get some landlines for flat rate.
Hi,
local numbers make the other party think to call someone
in the respective area.
Thus reaching someone at a local
Hi,
I have a sipphone behind a router doing NAT, an asterisk
box in the middle and another asterisk box, which works
as gateway to further destinations.
The asterisk box in the middle should do all call setup
and tear down, but no RTP. RTP should flow directly between
the sipphone via the
Hi,
I have following setup:
++ ++
| asterisk A |-| asterisk B |-- PSTN-gateways ...
++ ++
| .|
| .
=Router (NAT)=.==
|.
Hi,
is several 1000s of extensions in a context a problem?
Roger.
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Hi,
I want to implement a lookup for valid extensions
using agi.
Thus I want chan_zap to accept some digits,
then check via agi if the number is complete,
run waitexten if necessary and check again ...
Unfortunately waitexten only accepts one digit, regardless
how may key strokes I did on
Dovid Bender schrieb:
several thousand extensions or several extensions called 1000 ?
Several thousend extensions.
exten = 497111234,1,goto(...)
exten = 497111235X,1,goto(...)
exten = 497111236XX,1,goto(...)
exten = 497111237,1,goto(...)
Several thousend extensions of maybe different
Roger Schreiter schrieb:
...
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Hi,
I've found the reason. Imho it has not yet been discussed
here, and imho it is a bug in chan_sip.
Thus, I will start a new thread, please
Hi,
I did not yet study the newest chan_sip.c versions, but
it seems, that chan_sip treats mysql-peers different from
other peers, concerning the variable canreinvite.
If this variable is not explicitely set for a peer or user in
sip.conf, the global value for canreinvite in sip.conf is
taken
Hoa Thai Duy schrieb:
Roger
If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
issue no re-INVITE, for sure.
Pls. change
Disallow=all
Allow=gsm (only one codec)
Hi,
yes, to avoid transcoding problems I only have one
codec, just alaw. Anything else is disallowed.
Hi,
I have in my sip.conf
disallow=all
allow=alaw
in order to avoid any codec problems disturbing reinvite.
And of course I have:
canreinvite=yes
In extensions.conf there is only one Dial command. It
has no qualifiers like t or T.
Just Dial(SIP/[EMAIL PROTECTED])
Anyway, asterisk does not
BJ Weschke schrieb:
...
I have no modifiers in my dial command.
...
One reason might be is if you are passing parameters in app_dial (eg.
Hi,
sorry, I did use the wrong expression. No, there
is no parameter like tT in the Dial command.
I think, I've made everything according to the docs.
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately,
Hi,
is it possible to let asterisk issue a SIP redirect?
A SIP invite command by a SIP client should be answered
by 30X Temporarly moved to SIP/
Is this possible with asterisk, maybe from within the dialplan?
(reinvite is not what I'm looking for, because it does not
completely release
Alex Mosburger schrieb:
...
SIPGate works also fine (better than SIPCall) but the ECHO is terrible.
My side (* server) connected with X-Lite Softphone has a great quality,
but the PSTN caller hears his voice with an echo.
Did anybody already had such a problem? Do you think that my * server
...
I forgot in my previous email one further issue:
Maybe the ping round trip to your SIP provider and thus
to the PSTN gateway is too long. The echo cancellation
is typically limited to a reasonable echo run time.
If the time is too long, echo cancellation will fail,
because it would be too
Alex Mosburger schrieb:
...
It is not my end hearing or producing echo. My voice is heard correctly
without any echo, but the other side hears his OWN voice several msec
...
Yes, this is, what I meant.
The other's voice is fed back by your device and running
back to the other side. That's why
Oliver Vermeulen schrieb:
...
We have ...
Hi,
I'm sure, there are a lot of providers of very interesting
and useful and helpful products and offers reading and writing
to this group - including our company.
Nevertheless, noone is offering his products here, because it is not
fair, if
James W. Brinkerhoff schrieb:
...
My question for you is: When testing a tone as given in the example,
Mwanalyze(8|8000|60|328), assuming there were no issues with the
connection and the tone came through perfectly, what would you expect
the 3 variables to hold?
Hi,
assuming slinear
Douglas Garstang schrieb:
Wow. If Asterisk could return SIP response codes that would be AWESOME.
... and the remote IP address (which may differ from the
address who registered).
Btw: Isn't the SIP response translated into a Q.931 code,
which can be read by ${HANGUPCAUSE}?
Roger.
Hi,
how can I disable event_log in order to reduce
hard disk activity?
I can't find any hints in conf files.
Must I hack the source code or even use brutal
methods like creating a dir called event_log in
the log dir, in order to prevent asterisk from
creating an event_log file? (Just chmod a-w
Hi,
I have unpleasent short audio gaps when a
perl based agi scripts starts.
Thus, I now started to put all those things in C programmed
daemons for fast-agi.
Anyway I'm looking for another mean, which would help me
more quickly.
I noticed, that all agi scripts are running with system
Michael Gaudette schrieb:
I've been warned many times that Fax over VoIP is unreliable.
How should I consider Fax over PRI channels with Asterisk? Is the quality
and reliability good, or should I be prepared for alot of grief?
Hi,
we tested faxing a lot. With spandsp (app_rxfax/txfax) we
Anton Krall schrieb:
...
Might be good for faxing though
Hi,
faxing suffers mainly from jitter, not from logarithmizing
of the audio data.
In PSTN G.711 is the standard and does not seem to impose
problems on faxes.
G.711 is ISDN quality!
I assume, for fax quality and reliability the
Anton Krall schrieb:
Guys, how much bandwidth does slinear comsume and what quality can it be
compared with? g711, gsm, g729?
Hi,
the bandwith is approx double compared to G.711, since it uses 16bit
(signed) integers, whereas G.711 uses 8bit integers.
The human ear has approximately a
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
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
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
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
Hi,
how can I know in the dialplan, whether and when I received
the ringing event?
Imho, the only way is to parse all events using the manager
and to forward this information by an application to the
dialplan. The application would have to be called on
connect or on hangup.
Hints for a more
Hi,
the uniqueid obviously consists of a timestamp part
and an continously incremented part, separated by a dot.
The two channels of a call in most cases have the
same number before the dot (timestamp) and consecutive
numbers after the dot.
Now I wonder, whether I can rely on that scheme.
I
Stefan Reuter schrieb:
To propose the best solution we must know more about your actual use
case.
Thanks for your answer!
I want to track the ringing event of the outgoing channel.
Unfortunatelly the link event is fired not before connect.
Thus, I see first the incoming channel (SIP or IAX)
Stefan Reuter schrieb:
...
So having a look at Asterisk 1.2-beta2 is probably the way to go.
Great! Yes, this will solve my problem.
Let's upgrade ...
Thanks!
Roger.
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Usman schrieb:
anyone running SS7 with Asterisk ? Please help me out.
I need to know the hardware used for SS7 with Digium E1 cards...
google shows for the words asterisk and ss7 amongst the
top five:
http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7
This will give you the desired
Hi,
please don't bother me continuing trying to fax, even if
I've got convinced, that it generally won't work!
I've found a strange behaviour, when sending
from IAXCLIENT1 - asterisk - IAXCLIENT2
or
from IAXCLIENT1 - asterisk - SIPCLIENT2
When IAXCLIENT1 is sending an absolutely constant
Kresimir Petrovic schrieb:
...
What do you mean more supporters. t.38 is only *reliable* way for transporting
fax over ip. Fax over g711 is pure luck...
Hi,
it is rather a question of IP quality than good luck.
I think, 99.9% of all faxes are transported via G.711.
Is there any telecom
Roy Sigurd Karlsbakk schrieb:
...
see http://soft-switch.org/foip.html for a brief explaination of why
this generally doesn't work...
Hi,
maybe one should update this link.
I think, you agree, that VoIP is somewhat similar to ISDN, as it
transports analog audio data in a digitally coded
Carlos Alperin schrieb:
...
Bank, and send all the faxes through that system, and forget to try anything
through Asterisk.
We're already tired of the complainings from customers that never can send
faxes, or sometimes some pages.
Wow, strong words!
Since our experiences with asterisk are
Andy Kuo schrieb:
...
It sounds like Asterisk should handle T.38 fax (at least at very light
...
Can you please tell me what ATA you are using? Could it be that the
Hi,
are you sure, your ATA does support T.38?
If so, you'll need the most recent CVS versions of asterisk,
and rather
Lee Howard schrieb:
...
See: https://sourceforge.net/projects/iaxmodem/
Hi,
great! I googled a lot, before we started our project.
A iaxclient which works as modem (aka tty device) would
be great.
Do you think, one can connect a hylafax server to
iaxmodem?
Yes, I know, that jittering will
Lee Howard schrieb:
...
to go through all of the effort of doing it yourself. But you actually
seem to be looking for something beyond just knowing if it's possible or
Hi,
yes, we consider offering our PSTN-gateways to fax providers
during off peak times, in order to get a higher average
Hi,
I've had a strange problem several times during the last days:
A call is established, both parties have audio in both directions,
but asterisk is still waiting for connect.
Thus after timeout (120secs) the call is terminated with either
busy or no answer.
This is annoying for the both
Hi,
I have some experience in sending SMSs using smsclient.
I call the german Vodafone SMSC (01722278020),
and smsclient takes approx 20 secs to send a SMS.
The hardware is an Sedlbauer ISDN card.
Now, I want to do the same using asterisk and a digium PRI card.
I dialed using the manager with:
Chris Shipman schrieb:
What build of SpanDSP did you use?
spandsp-0.0.2pre18
I'm working on a windows program
so users can print to a local printer which will be forwarded to the
asterisk server to be faxed.
So far the program FTPs a Tiff to the Asterisk server to be faxed
Hi,
I found some contradicting infos about pass through of
T.38 data.
Are there any experiences of just passing T.38 via SIP from one T.38
application or gateway trough asterisk to another T.38 application
or gateway?
Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip
(without
Il Neofita schrieb:
Is there some way to know if the fax was received correctly or not?
Hi,
after also having asked here some days ago without an answer,
I assume, there is currently no way.
Thus, I started to study a little bit the source code.
Maybe I found a solution, which will work.
I
Hi,
I tried to use txfax to send several faxes at the
same time.
It seams, that one can't send more than 3 faxes at once,
or one risks to get 50% and more aborted faxes due to
errors.
The CPU usage is below 97%.
I tried with Opteron and IntelP4: same result.
Ok, I know, that faxing via a
Hi,
using Zap, I have several messages to pass when
terminating a successful or unsuccessfull call,
indicating the reason e.g., why a call failed.
Using SIP or IAX2, I know only
Hangup
Busy
Congestion
without passing any more detailed information.
Am I right, that I can't tell the caller in
Hi,
I read here in this mailing list about the debug info
from txfax.
I plaid a lot, but didn't get debug infos.
I added the debug argument to txfax, I enabled
debug in logger.conf, what else should I do?
In which file or medium can I then expect the debug
infos from txfax?
Thanks for
Hi,
I have asterisk 1.0.7 and spandsp-0.0.2_pre18.
txfax return a non-zero return code only if the
fax file is not found.
Unfortunately I can't get any information, whether
the fax was transmitted completely or not.
Will an update to a newer version change this?
Thanks for telling me your
Asterisk schrieb:
I'm looking to develop some custom AGI that will be MySQL intensive. It
appears Asterisk supports many different development environments. Which
would be best suited for Asterisk and MySQL?
Hi,
this is the same question as for other applications:
Use C, if you want to
Ma Zhiyong schrieb:
...
Trace shows that the fax is received successfully.
Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX(Zap/94-1,
Hi,
sorry, I don't know the solution to your problem, but I would like
to know, how did you get that trace?
I'm looking for a reliable way to
Hi,
I noticed a strange behaviour:
Faxing using spandsp (TxFax) from my asterisk box to my
old, common fax machine at home works fine.
Faxing from the same box to my office pc-fax (Hylafax)
also worke fine.
Receveiving faxes on my asterisk box using spandsp (RxFax)
also works fine. It is a PSTN
Hi,
there are some messages indicating, that TxFax is able to return
-1 on failure.
Well, I tried a lot but didn't succeed.
I even sent a fax to a phone set, picked up the hand set and
waited until timeout of TxFax.
There is no difference to success.
The only thing I could determine, is,
Steve Underwood schrieb:
...
If the call really dialed out through a PSTN port and back in it should
work. It is was a pure internal connection between 2 processes it will
Hi,
the setup is:
TxFax (Box A)
Dial(Zap...) (Box A, Digium Card)
v
PSTN
v
Box B, Digium Card
Dial(IAX2...) (Box
Hi,
I searched a while about T.38 decoding, and learned about the
bounty for T.38 support for asterisk and some softdecoders and
some hardware de- and encoding T.38.
Now I wonder, if there is already any (almost) ready to use solution
for decoding of T.38 faxes?
My szenario would be:
-
Hi,
asterisk compiled fine and is running very stable on
our dual opteron in 64 bit mode.
When loading G.729 library we have to peload libz manually for any
reason, but besides that minor issue, everthing is fine.
We didn't yet test the limits of that machine.
Roger.
Allan Kamau schrieb:
...
I am looking for a GSM VoIP gateway for use with
Hi,
do you think of something to interconnect
to GSM carriers via cable (GSM-A) or do you
think about using a GSM-modem with all its
limitations?
For the first option I could forward your email address
to someone
Hi,
I'm trying to use the manager cmd Getvar.
Unfortunately I always get (null) as variable
content.
I'm using asterisk 1.0.7
When calling a non existant channel, I get an
appropriate result.
This is what I tried and got:
Action: Getvar
Channel: SIP/01234567-5242
Variable: CALLERID
Hi,
if I understand right, the best way to indicate a PSTN
line busy, is something like that in extension.conf:
... background(busy-tone)
... busy
So the caller will first hear my busy-tone, and
after some seconds, when PSTN honours the busy
indication (cmd busy), he hears the busy sound by
Marco Parmeggiani wrote:
...
i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only
Hi,
where did you get that version?
On libtiff.org, 3.6.1 is the most recent one.
Roger.
They keep breaking the FAX support in libtiff. 3.6.1 is broken, a
...
Hi,
thanks for the information about libtiff 3.6.1.
I had to search a while in order to find the old libtiff 3.5.7,
which now works fine in my asterisk installation.
For those being in the need of libtiff 3.5.7: Pay
Hi,
package tiff-v3.5.7 contains the currently recommended version
of libtiff in order to run spandsp (fax support for asterisk).
Imho tiff-v3.5.7 is not very easy to find in the internet, and
maybe will almost disappear, because it is an old version,
I put it on our little asterisk download
Hi,
I've bought a Siemens GSM-modem based on the Siemens TC35-module.
I studied the operation manual of the modem and found, that for
transferring voice via the RS232 wire, the module supports
RS232-mulitplexing and wires the voice data on a separate
channel (whatever this means on RS232?).
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