Displaying 18 results from an estimated 18 matches for "2fasterisk".
2007 Dec 29
1
Not Able To tar zxvf zaptel-*.tar.gz
...ownloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz
>
> --10:15:59-- http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz
>
> =>
> `elqRedir.htm?ref=http:%2F%2Fdownloads.digium.com%2Fpub%2Fasterisk%2Freleases%2Fasterisk-
> 1.4.16.2.tar.gz'
> Resolving www.digium.com... 216.207.245.16
> Connecting to www.digium.com|216.207.245.16|:80... connected.
> HTTP request sent, awaiting response... 200 OKk/releases/asterisk-
> 1.4.16.2.tar.g
> Length: 2,403 (2.3K) [text/html]
>...
2020 May 01
4
Length of dial string
Hi all
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see
* [ASTERISK-27946
<BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] -
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't
I have been fighting with this issue for months trying to find a solution I
2020 May 04
0
Length of dial string
...g.
Hi all
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see
* [ASTERISK-27946
<BLOCKED::https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fissues.asterisk.org%2Fjira%2Fbrowse%2FASTERISK-27946&data=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259&sdata=JdT9Yvi7ml%2FqzIYMO39ks68rdMKY2P2DFIAGKCCh6a8%3D&reserved=0> ] -
dial (API): Storage of dialed target uses AST_MAX...
2006 Oct 11
1
sending fax with chan-capi
Hi!
Has someone ever used the sendfax option of new chan-capi to send fax? I
need some help regarding the sff format:
How can I generate sff format? I found sfftobmp, not nothing the other
way round.
Is there a nice way to get the sff out of an Windows application (like
virtual printers for hylafax) or at least some scripts which produce the
sff and the asterisk call file out of an pdf?
2008 May 09
1
Asterisk ZRTP?
...ate 2007.
Does any version (1.4.x, 1.6.x) of Asterisk support ZRTP with clients
(or with other servers)? Any successful testing with specific
clients/peers to report? If not, are there any serious efforts underway?
http://www.google.com/search?q=site%3Ahttp%3A%2F%2Flists.digium.com%
2Fpipermail%2Fasterisk-dev%2F+zrtp
http://www.google.com/search?q=site%3Ahttp%3A%2F%2Flists.digium.com%
2Fpipermail%2Fasterisk-users%2F+zrtp
http://bugs.digium.com/view.php?id=10024
--
(C) Matthew Rubenstein
2019 Aug 22
2
h265 codec pass through on asterisk
...The only real reason is that noone has added support for it. Asterisk has to be made aware of it in a few places in order to allow it. I did it for VP9[1] so that could be used as a base.
[1] https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgerrit.asterisk.org%2Fc%2Fasterisk%2F%2B%2F6079&data=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424140187&sdata=ypez4T%2BQyKSK3MC7RYvPuPHlPw5Xjur8cE4tEh85BA0%3D&reserved=0
--
Joshua C. Colp
Digium - A Sangoma Co...
2019 Aug 22
2
h265 codec pass through on asterisk
All,
I'm using asterisk 16.4.0 with h264 and opus quite well using linphone 4.1
client on android and baresip on linux.
I'm exploring use of h265 for improved video quality/lower network
bandwidth. I do not see pass through support on asterisk for h265/hvec. All
my SIP clients and underlying hardware have hvec/h265 encoding and decoding
available.
I would have liked vp9 however, vp9
2018 Apr 10
2
Asterisk behind NAT Early Media Video
...y forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.digium.com%2fmailman%2flistinfo%2fasterisk-users&c=E,1,6VfJH-ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmol-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc&typo=1
2018 Apr 10
2
Asterisk behind NAT Early Media Video
...terisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.di
>> gium.com%2fmailman%2flistinfo%2fasterisk-users&c=E,1,6VfJH-
>> ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jm
>> ol-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc&typo=1
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and C...
2018 Apr 11
2
Asterisk behind NAT Early Media Video
...y forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.digium.com%2fmailman%2flistinfo%2fasterisk-users&c=E,1,6VfJH-ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmol-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc&typo=1
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com<htt...
2006 Jun 17
4
Which phones are good, or at least acceptable, for home and office
I am looking to replace all of the old "Bell" (POTS) phones in my home
and office with IP phones. As you can imagine I don't have a huge
budget to work with but I want phones that will provide acceptable voice
quality and durability.
There are basically three categories as I see it
1. satellite phones (low cost, low function)
2. primary domestic phone (good quality, POE capable,
2006 Apr 21
1
1.2.7.1 on FC5 won't make install
The make seems to go okay.
[root@somebox asterisk-1.2.7.1]# uname -a
Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux
mkdir -p /var/lib/asterisk/sounds/digits
mkdir -p /var/lib/asterisk/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if grep -q "^%`basename $x`%" sounds.txt; then \
install -m 644 $x
2003 Sep 23
3
New kid on block
Hi,
I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution.
1) How does Asterisk compare to VOCAL in terms of support.
2) Is Asterisk free?
3) Where are the docs? Or even better. Where do I start?
4) Will it run on RH9?
Thanks in advance.
Costas
--
Costas Menico
Meezon Software Corp
201-224-8111
costas@meezon.com
--
2003 Jun 11
4
some sip questions AGAIN
I write the email again, the third time!!, cause the other two ones, I have
had problems while sending them. I hope this time it works. Here is the
email again:
Hi (and sorry) everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly
ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN
above all, but I have
2006 Apr 21
5
Separating Asterisk SIP extensions from dialing each other.
This is coming from an * noob. :)
I've got two customers, they both are replacing their phone systems with
VOIP, and we need to retain both their existing dialplans.
One has 5 extensions starting at 100, and the other has 10 extensions,
starting at 100.
Is there a way to have the same extension number twice in the same
asterisk system ?
They will have different incoming DIDs of course.
2018 Apr 09
3
Asterisk behind NAT Early Media Video
wohoo, so if I unterstand it correctly with that patch early media video
works over the Asterisk server? In other words the Asterisk server get's
able to (process/)forward the early media video stream with that patch?
2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>:
> On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote:
> > My understanding based on Wireshark
2006 Mar 02
5
Milliwatt Analyzer available
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 frequency is not limited to 1000 Hz, but can be passed
as argument. The periode duration must be a mulitple
2003 Jun 16
8
SIP REGISTER
Hi!
I have a new problem with my SIP device.I have done some changes and
now I receive continuosly a SIP message: "501" "Not impelmented" back
from the SIP Gateway. I can see that it doesn't register to Asterisk.
I have in the SIP device:
Registrar 1: UnRegistered to: 2222
registrar: 188.208.12.237 5060 expires: 2000
name: gateway passwd: 123
My