smime.p7s
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New to Asterisk? Join us for a live introductory webinar every Thurs:
its kinda tangential to the original post,
but check this link
http://www.voip-info.org/wiki/index.php?page=Asterisk%20Native%20Sounds
since the link for download doesn't seem to work I can post or send
the files in many codecs for you
On Jan 24, 2008, at 9:56 AM, Erik Anderson wrote:
On Jan
On Jan 22, 2008, at 10:23 AM, Philipp Kempgen wrote:
Andre Herrlich wrote:
any one advise a good, strong and free softphone that can work
with SIP
or/and IAX lines and supports attended transfer ?
IMHO there are no good softphones - at least not for
Mac OS X and I think that is true
On Jan 7, 2008, at 8:53 AM, daniele visaggio wrote:
2008/1/7, map [EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put
a number greater than 200 :-). I suggest 10.
Sorry, i'm using a Linux
On Jan 2, 2008, at 12:33 AM, bilal ghayyad wrote:
Hi List;
I heared that IAX is good for NATing issues, but I do
not know if it can help me in that senario:
I have two Asterisks machines in different sites and
both are behind NAT (both have private IP address), I
need to link these two
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Andres Paglayan
--Harmony is more important than
make sure your telco agrees with your channels, like d being in the 16th
try using sangoma tools (wanpipemon) and even calling sangoma,
Rory
On 16/08/07, Andres Paglayan ([EMAIL PROTECTED]) wrote:
On Aug 16, 2007, at 8:58 AM, Rory Campbell-Lange wrote:
However both incoming and outgoing calls
/listinfo/asterisk-users
Andres Paglayan
--Harmony is more important than being right
Bapak
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Andres Paglayan
--Harmony is more important than being right
Bapak
,
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Andres Paglayan
--Harmony is more important than being right
Bapak
or update options visit:
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Andres Paglayan
--Harmony is more important than being right
Bapak
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Hi,
is it possible to fork from a dial plan?
meaning, is there a way to redirect to two different
context,extension,priority
without waiting for the first to finish?
Andres Paglayan
--Harmony is more important than being right
Bapak
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Andres Paglayan
--Harmony is more important than being right
Bapak
(or if there's an specific
instruction for)
forking out of the dial plan,
On 8/9/07, Andres Paglayan [EMAIL PROTECTED] wrote:
Hi,
is it possible to fork from a dial plan?
meaning, is there a way to redirect to two different
context,extension,priority
without waiting for the first to finish
://lists.digium.com/mailman/listinfo/asterisk-users
Andres Paglayan
--Harmony is more important than being right
Bapak
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-users mailing list
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Andres Paglayan
--Harmony is more important than being right
Bapak
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Andres Paglayan
--Harmony is more important than being right
Bapak
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andres Paglayan
Sent: Friday, June 29, 2007 12:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX over T1
On Jun 22, 2007, at 3:43
options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Andres Paglayan
--Harmony is more important than being right
Bapak
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to
(or at least one of the how tos)
Andres Paglayan
--Harmony is more important than being right
Bapak
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vant output of show dialplan.
Note that the sip calls come in on extension 666.
it's cursed,
Thanks much in advance.
Ryan
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wireshark can further filter out what you don't want,
you can also pipe the dump to grep and match only what you want
On May 1, 2007, at 11:32 AM, CSB wrote:
I want to capture all my Asterisk traffic (including RTP) and then
analyse it.
My plan was to use tcpdump and then analyse with
yep,
a nice setup though is run * as the man in the middle,
if you have a telco T or E, use a dual TE so you can inbound in one end,
do your * stuff, and forward to the legacy box in the other,
On Feb 5, 2007, at 7:29 AM, Noc Phibee wrote:
Hi
it's possible to use a Digium TE110P Single T1 /
you gotta check festival site,
it works for Spanish too through external language modules,
On Feb 5, 2007, at 7:57 AM, voip crazy wrote:
Hello all,
I am looking for software for text to speech in spanish witch works
with asterisk (1.2.13).
I have tested festival and the cepstral software,
Any mid-level server (kinda 3ghz 2GB ram) you have been wonderfully
happy with?
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Hi All,
I have a (legacy) Praxton PBX, it has a PRI T1 input card and 64
analog extensions through 4 amphenol connectors.
We receive 12 voice channels (other 12 are idle) and have 100 DIDs.
No caller ID thru PRI though.
The Praxton box is amazing in terms of configuration and flexibility
Hi,
My users are currently using an operator console interface like this:
see it at: http://www.whssf.org/interface.jpg
which came with a Praxon PDX we got about 5 years ago, which is now
unsupported,
it works very good and converts any analog phone plugged into the
system into a powerful
Hi,
My users are currently using a console interface like this:
see it at: http://www.whssf.org/interface.jpg
which came with a Praxon PDX we got about 5 years ago, which is now
unsupported,
it works very good and converts any analog phone plugged into the
system into a powerful console,
they had a short outage today,
it was fixed already,
dunno if related to your issue,
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Hi,
I am looking for, but not able to find:
a wifi voip pager,
meaning a wifi text receiver device which can receive text messages.
I know a wifi voip phone will do, but they cost +$160
it even really doesn't need to be voip,
but if so, it can be integrated within the * PBX and get msgs sent
I am happy teliax user,
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Hi there,
I am writing some Ruby (Rails) app which needs to comunicate some
events straight to phones.
The application runs in one box.
Asterisk runs in another. (I am already able to config a basic *)
My questions are:
Where do I start learning how to send SIP messages to a phone?
any
I have the first edition,
does anyone know if it's worth getting the second too?
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zdummy NEEDS the usb drivers to load the clock,
if you disabled usb from bios, then the kernel is not loading usbci
On 10/13/05, Bruce Ferrell [EMAIL PROTECTED] wrote:
Hi all,
Trying to build ztdummy on an old redhat 7.3 box running kernel
2.4.20-43.7.legacysmp. Yes, I have the kernel
After dealing with a Poly 301 I rather use the FTP server and config
files, even for a single phone,
download the manual and stuff from freedomphones.net/polycom
Tom Vile wrote:
but you do not get all of the features via a web browser to customize.
On 9/22/05, *Tom Hayden* [EMAIL PROTECTED]
This is an excerpt from the log file,
My problem is that randomly, 1out of 3 or 1 out of 2, some calls are
not going out and this is the message in the log file,
The device that should provide the frame is a Sipura 3000 which has its
FXO providing outside connectivity,
24185 Sep 16 10:35:40
Raise both gains from -3 to 5 that solves volume problem,
log in, click admin, advanced, I guess is on the sip tab,
Matthew Harrell wrote:
When I have voip conversations over asterisk through my computer the voice
quality is nice and loud and quite clear. When I go through my Sipura 2K
then
Noah Miller wrote:
Hi Andres -
The two that we have are just used as lobby phones. They're good
little phones, but if you have the money, I'd definitely recommend
the IP501 instead. The screen is MUCH better, and having full
speakerphone is great! Plus the 500/501 just feels a little
Hi All,
I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301
The Polycom misses 1 out of 2 dialout calls, this is the full log from a
call which didn't go through.
303091 Sep 14 10:45:15 VERBOSE[15427]: -- SIP/pstn_2-1f35 answered
SIP/200-0db1
303092 Sep 14 10:45:15
More information in this thread,
This Poly 301 sometimes rings out sometimes doesn't,
it calls out through * using spa3000's fxo,
I got this log
1129 Sep 14 15:03:55 DEBUG[15702]: Outgoing Call for ww9821642
1130 Sep 14 15:03:55 DEBUG[15702]: ww9821642 is not a local user
1131 Sep 14
Thanks Noah for your time,
I am using rfc2833 as dtmf mode
I already tweaked the dialplan.digitmap= (to an empty string) so
everything gets out.
my phone's sip.cfg codec setting looks like
preferences voice.codecPref.G711Mu=1 voice.codecPref.G711A=2
voice.codecPref.G729AB=3
Hi,
I am having problems on getting the second line to work on a Polycom 301,
this is the phone.cfg file,
the * box is 192.168.1.8 and the phone is 192.168.1.18
I am not 100% sure about what the reg.x.address should be,
with this setting I only get the line number to work,
the second just gives
Hi,
I have an X100P which receives an analog line from another PBX.
These are the relevant entries in extensions,
PHONE1=Zap/1
[macro-extensions]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Voicemail2(u${ARG2})
exten = s,102,Voicemail2(b${ARG2})
exten = s,103,Hangup
[home]
include=tozap
exten =
Thank you very much for all answers.
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Dear all,
Sorry to ask, but...
Do you know where I can find a full list of configuration parameters and
values for each of the .conf files?
Do default .conf files include all options?
Thanks Again
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Dear List,
Besides price, ~$300 against ~$200,
Is there any pros and/or cons on using one or the other approach to
provide 2 FXSs and 2 FXOs (plus 4 IP phone extensions)?
I am about to start building my first ever * production server and would
be nice to have some input from the list.
Many
I buy stuff from VoipSupply online, and will continue, they all were
smooth transactions.
Not need to flame them here. Canadian setup?
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Dell's entry level line of servers is very Linux friendly.
I use poweredge for some production systems (yes, even with a single drive)
but if is only for a proof of concept, then a $50 Compaq deskpro which
are also Linux friendly might be an option.
Stephen McAllister wrote:
Does any one have
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
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To
follow this link
ignore the German and see the commands
http://www.vonloesch.de/node/17 http://www.vonloesch.de/node/17
for the last part be sure that you modprobe the right driver for your
particular device.
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question about this thread,
would a wi-fi voip phone work for this guy?
meaning, he takes it to wherever he goes and it gets registered wherever
it as wireless access.
is that theoretically correct?
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there are cheapo clones of the X100P for the fxo side (up to two will be
ok),
at $20 each = $40
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61841item=5773237792rd=1
and you can get a refurbished sipura 2000 for the (2) fxs part of it.
($70) voipsupply
with an used compaq PIII at $50
follow this link
ignore the German and see the commands
http://www.vonloesch.de/node/17
for the last part be sure that you modprobe the right driver for your
particular device.
one little thing is that in Debian you shouldn't use /usr/local/bin, but
/usr/bin, if you are using the source from
Why the channel bank if he will be routing extensions to ip phones?
The T-1 card should suffice if he isn't serving analog extensions.
Thats $600 (t1) instead of ~ $1000.(3 x 3?? tdm400p)
line cost wise 12 channels on a t1 should be cheaper than 8 pots.
Walt Reed wrote:
On Fri, May 06, 2005 at
We use a Cybermesa (local Santa Fe company) T1
$80 for the transport,
$240 for 12 lines
$12 for 100 dids
$190 tax
total = 522
8 comercial lines * ~40 = 200
So you are right.
Eric Wieling aka ManxPower wrote:
Andres Paglayan wrote:
line cost wise 12 channels on a t1 should be cheaper than 8 pots
Hi List,
I was looking for, but I couldn't find any product or project like BIND
that works with VoIP in an homologous way.
I mean, is there anybody working in a way to register user-ids or domain
name-like information so VoIP calls can be dialed in a number string
format from any IP phone?
try auto-apt for getting dependencies satisfied on the fly while compiling.
Manuel Casal wrote:
During the zaptel configuration at the end of it there is this error:
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Hello,
I am new to * and before diving in I would like to know which will be
the recommended documentation source to study and understand dial plans.
As an starting point my little project will just place automated
'remainder' calls.
Thank you,
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