On 15 Apr 2006, at 06:53, George Pajari wrote:
Kevin:
You wrote:
FAX transmission is massively more complex than modem
transmission. At
higher speeds, it involves 3 or 4 different 'carrier' frequencies and
signaling rate shifts, and these are done with very critical timing
requirements.
I'm having a problem with my Cisco 7960 phones with the SIP image. When i
try to dial a international number i keep getting a busy signal but i dont
see anything on the asterisk console (-vc) like i do when i dial
local or long distance numbers. It makes me think the phone is doing
rpms for Fedora Core 1-5, RHEL 3-4 and RHL 7.3-9 have been updated:
http://atrpms.net/name/asterisk/
http://atrpms.net/name/zaptel/
http://atrpms.net/name/libpri/
http://atrpms.net/name/asterisk-addons/
http://atrpms.net/name/asterisk-sounds/
I've recently swapped my router out for a slightly different router, and
now everything works fine. I guess my other router must not be very good
or have some issue with this protocol/setup/something.
Thanks for the help.
Regards,
Carey O'Shea.
On Wed, 2006-04-12 at 20:16 +1000, Carey O'Shea
Rudolf Ladyzhenskii wrote:
I think you are right. 1.0.7
I connect via VoIP providers -- via Internet only. No direct PSTN
connection. (Well I do have TDM400, but did not have time to set ot up
yet).
I use Polycom SP300 phones
I even have problems when talking to people with softphones
Hi,
I have try install the TE110P or TDM400P on HP Proliant DL380 Server. And
use the AAH 2.7 distribution. When I reboot the server. The CentOS always
display some hardware removed - Tiger pci card. Found new
hardware - Tiger pci card. Is it the digium cards have some
compatiable
Hi,
On Sat, 2006-04-15 at 20:07 +0800, kevin ling wrote:
Hi,
I have try install the TE110P or TDM400P on HP Proliant DL380 Server. And
use the AAH 2.7 distribution. When I reboot the server. The CentOS always
display some hardware removed - Tiger pci card. Found new
hardware -
George Pajari wrote:
I'm sure you didn't quite mean to write what you have said above. Fax
transmission builds upon exactly the same ITU-T standards as data
transmission. For example, 33.6 kbps fax transmission (so called Super
G3) uses the same V.34 standard as 33.6 data modems. At slower
Kevin P. Fleming wrote:
Rusty Dekema wrote:
If this works, I don't see why a fax transmission wouldn't work. Is it
because the fax protocol doesn't have error correction? Is that even
true?
FAX transmission is massively more complex than modem transmission. At
higher speeds, it
Kevin P. Fleming wrote:
Jeff Gustafson wrote:
Is there any reason an easier implementation of the same, basic, idea
could be created for the Asterisk generation? According to a quick
search of H.100 it's just a TDM bus. It handles 2,048 full duplex
calls. Would a lightweight
On Fri, 14 Apr 2006, Kevin P. Fleming wrote:
What about a new line of Digium cards that have bridge cables that run
between the various cards and bypass the PCI bus? Since one of the best
aspects of using Asterisk is standards. This bridge cable should be
standardized and published so
So, to document this, the likelihood of a fax working goes in this
order best to worse:
1. POTS - fax
2. POTS - FXO-TDM400P-FXS - fax
3. T1 - TE410P - channel bank - fax
4. T1 - TE110P - PCI - TE110P - channel bank - fax
5. T1 - TE110P - PCI -
Hi Steve,
Thank you for your very enlightening message!
On Sat, 15 Apr 2006, Steve Underwood wrote:
[...]
modem it must be applied end to end by the modems themselves. The
real killer, though, is imperfect timing.
[...]
and its not always always available within
On Sat, 15 Apr 2006, Kevin P. Fleming wrote:
Actually, I did. During a FAX transmission, there are many shifts to
different carriers and signaling rates as pages are transmitted and
acknowledged. It is _not_ as simple as a single carrier, like a normal
data modem connection. In addition to
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.
Is there something that would speak against it?
cheers,
tom
Thomas Artner wrote:
Hi!
A few months ago
Begumisa Gerald M wrote:
Hi Steve,
Thank you for your very enlightening message!
On Sat, 15 Apr 2006, Steve Underwood wrote:
[...]
modem it must be applied end to end by the modems themselves. The
real killer, though, is imperfect timing.
[...]
and its not always
Remco Barende wrote:
On Sat, 15 Apr 2006, Kevin P. Fleming wrote:
Actually, I did. During a FAX transmission, there are many shifts to
different carriers and signaling rates as pages are transmitted and
acknowledged. It is _not_ as simple as a single carrier, like a normal
data modem
record the sound fax machines make when negotiating (specifically the part
where they try to negotiate anything above
9600 baud) and make a provision in asterisk (an extra letter added to the
Dial command?) that will make Asterisk monitor
the channel and listen for the fax nego sounds and have
Remco Barende wrote:
record the sound fax machines make when negotiating (specifically
the part where they try to negotiate anything above
9600 baud) and make a provision in asterisk (an extra letter added
to the Dial command?) that will make Asterisk monitor
the channel and listen for the fax
Hello list,
When new voicemail comes and i pick up the phone i hear special tones
indicating that the new voicemail arrived.I've never had any problems
with this feature, but several days ago it begin to behave strangely:
1. new voimcemail arrives, but i dont hear the special indicating tones
Check your voicemail.conf and sip.conf - I suspect that you have multiple
mailboxes and they are not associated with the right SIP device.
MS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tofik
Suleymanov
Sent: Saturday, April 15, 2006 2:48 PM
To:
Does asterisk look for all known incoming callerID name messages, or do I have
to tell it how the Telco is sending it?
--
--
Steven
http://www.glimasoutheast.org
Matthew Fredrickson [EMAIL PROTECTED] wrote in message news:[EMAIL
PROTECTED]
It is in fact required for some implementations
Thank you for doing that.
With a little luck someone will be able to fix it, assuming they
understand it NEEDS to be fixed.
Damon Estep stated proper transfer operation well yesterday
Sustitution of another key for pound in features.conf might also be
desirable
To be typical it would act
That fix would be great!!!
To press # and be able to get the call back and terminate the transfer...
I had to implement an horrible workaround to emulate this functionality
Dov
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de John Novack
Enviada em:
What phone are you using?
On 4/15/06, Technical Support [EMAIL PROTECTED] wrote:
Check your voicemail.conf and sip.conf - I suspect that you have multiple
mailboxes and they are not associated with the right SIP device.
MS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hmm not so sure of that. I have an HP all-in-one thingy. It is not possible
to set the TX/RX speed hard in the config at a certain speed. Through the
developers menu in the beast it is possible to do this temporary.
Faxing at max 9600 bps works, anything higher fails miserably after the
hi,
SER is less about the number of callers than it is about the number of registered sip clients. Without NAT issues a pizza box server with SER can essentially register an unlimited number of SIP clients.
With larger numbers of SIP clients i find SER handles them much better than asterisk.
Now,
here's the reported issue: http://bugs.digium.com/view.php?id=6973
cheers,
tom
Thomas Artner wrote:
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.
Thanks Matt...
D.K.
On 4/13/06, Don Pobanz [EMAIL PROTECTED] wrote:
Thank you Matt!!!
Matt Roth wrote:
Try switching to native MOH. You'll eliminate the decoding of the MP3s
and the host of problems that come along with using mpg123. The MOH is
handled by the same thread that's
Matt:
Some great scripts on your site! I read the one about blacklisting a
callerID. As an alternative (albeit more complex), we have a posted a
utility which looks up caller phone numbers in a mySQL database and takes
action based on one field of the database (Drop it, say busy, or let caller
I have complied the latest releases, patched with spandsp and iaxmodem
support. If you upgrade to the provided kernel you will have support for
the zaptel modules and sangoma drivers without the need to recomplie
anything.
ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS4/
Have fun...
Andrew
I'm using a macro such that if a call is forwarded to my mobile, I can
choose to accept it or not. This is done with the following Dial command:
Dial(IAX2/voiptalk/44${ARG1:1},15,trM(screen)ow)
and the following macro:
[macro-screen] ; Allow called user to accept/reject the call
exten =
Matt Roth wrote:
Don Pobanz wrote:
I believe I am using MP3. My musiconhold.conf file looks like this
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
Daniel and Don,
Try switching to native MOH. You'll eliminate the decoding of the MP3s
and the host of problems that come along
Is anyone using FreePBX in production level systems because I'm just wondering if its stable enough to use. Currently I'm editing my own *.conf scripts but it sure would be nice if there were some sort of web interface for other people to use. The only thing holding me back is the stability of the
I can see in some cases where you might want to start all callers from the beginning, but I can also see in some cases where you would not. This sounds to me like we need an option.
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A client wants to record all calls to a specific extension. MixMonitor
seems to do the job, but is there a way to get it to append something to the
filename for each call? Right now it overwrites the file every time a call
comes in.
I realize there is an append option, but I'd prefer a separate
Hello,
On 4/11/06, Shad Mortazavi [EMAIL PROTECTED] wrote:
I'm working on a call recording solution and would like to have the ability
to initiate a 3 way call based on an incoming call.
One party will be an AGI that I have other will be an outbound call via a
second T1 interface.
Does
Tom Vile wrote:
Here is one that was made for [EMAIL PROTECTED] but you should be able to use
it.
http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html
Hiya,
I've given this a go - and got it working - in a fashion. My web server
is on a different box to my asterisk
Seems like mixmonitor app uses the extension to determine what format to
save the file as. (ie raw, gsm, etc.)So I think you want to leave
the extension alone. But you have lot's of control over the filename
itself using variables functions. Here's what I'm doing which seems
marginally
I'm currently using it at 2 offices- each one is about 40 phonesOn 4/15/06, Min Hwan Chang [EMAIL PROTECTED]
wrote:Is anyone using FreePBX in production level systems because I'm just wondering if its stable enough to use. Currently I'm editing my own *.conf scripts but it sure would be nice if
Remco Barende asterisk at barendse.to writes:
I tried lots of different settings but none really seemed to help.
The line is ISDN BRI with an HFC-S card. Software is bristuff with florz
patch. Echo can, silence suppr. etc all disabled.
The HP is connected to a Sipura SPA 2000 with the
Shaun wrote:
I'm having a problem with my Cisco 7960 phones with the SIP image. When i
try to dial a international number i keep getting a busy signal but i dont
see anything on the asterisk console (-vc) like i do when i dial
local or long distance numbers.
sip debug peer
We had an issue at an install of [EMAIL PROTECTED] - where if you use the
external extensions the machine is unable to start Asterisk after a reboot.
Which in the end begged a question - it was nice have customers who could edit
their box, but was it worth it for the angry calls when their
Hi, everyone,
We've been reasonably happy with Polycom SoundPoint phones, but we only
have them installed on the LAN. I've read that they have problems
working across NAT. So ... I guess I have a few questions. First, is
there a way to get Polycoms to work well over NAT? If not, then are
Technical Support wrote:
Matt:
Some great scripts on your site! I read the one about blacklisting a
callerID. As an alternative (albeit more complex), we have a posted a
utility which looks up caller phone numbers in a mySQL database and takes
action based on one field of the database (Drop
On Saturday 15 April 2006 21:12, jennyw wrote:
We've been reasonably happy with Polycom SoundPoint phones, but we only
have them installed on the LAN. I've read that they have problems
working across NAT. So ... I guess I have a few questions. First, is
there a way to get Polycoms to work well
Remco Barende wrote:
Hmm not so sure of that. I have an HP all-in-one thingy. It is not
possible to set the TX/RX speed hard in the config at a certain
speed. Through the developers menu in the beast it is possible to do
this temporary.
Faxing at max 9600 bps works, anything higher fails
I have so far found 2 ATA's that seem to be able to handle FAX reasonably
well. The first one is the Grandstream ATA-286 (firmware up to 1.0.6.7,
have not tried any firmware later than this), I have used these at multiple
customer sites and no one has ever reported problems. They handle G3
On 4/15/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Saturday 15 April 2006 21:12, jennyw wrote:
We've been reasonably happy with Polycom SoundPoint phones, but we only
have them installed on the LAN. I've read that they have problems
working across NAT. So ... I guess I have a few
Polycoms (the IP501s at any rate) work EXCEPTIONALLY well through NAT. It's
as literally dead-simple as plug-and-go. No configuration on the phone, and
all you want is a nat=yes in their sip.conf entry. That's it. Seriously.
In addition to nat=yes, I recommend adding qualify=yes for all
use feauters.conf and the application map section.
On 4/14/06, nzrh [EMAIL PROTECTED] wrote:
Hi all,
I urgently need a solution in a part of a project.
I appreciate all types of help.
The thing I absolutely need is. To play a background
music in call.
If I have the opportunity to stop it
It's all in the buttons.cfg BTW flashpanel has it's own list.
On 4/14/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
is it possible to remove the no timeout combo box in flash operator panel?
How can I reduce the flash area? I set small buttons and half of the
area is white and I want
One way would be to use a queue to initiate the call and use the
context defined in the queue as where to drop the call. I'm however
not sure if the context setting in queues work when ring is used as an
option in the app_queue. Also you would have to make sure that the
member in the queue is
Yes, just configure your sendmail to do it.
On 4/13/06, nik600 [EMAIL PROTECTED] wrote:
is it possibile to set up an external smtp server for the relay to the
users of the mails?
thanks
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Disregard my previous email about this problem, I guess this is what I
get for not RingTFM
On 4/12/06, Obelix [EMAIL PROTECTED] wrote:
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because
Hello i am a asterik user from Indiai can't receive the caller id information in my WildCard X100P.in India we are using CLIP for Caller ID Can any body help me on this matter..Thanks in advance
Raju
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How can I change this:
Asterisk PBX [EMAIL PROTECTED]
to:
London PBX [EMAIL PROTECTED] ??
I tried several settings in voicemail.conf, without success!
bye
Ronald Wiplinger
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