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...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] >...

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

...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&amp;data=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259&amp;sdata=JdT9Yvi7ml%2FqzIYMO39ks68rdMKY2P2DFIAGKCCh6a8%3D&amp;reserved=0> ] - dial (API): Storage of dialed target uses AST_MAX...

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?

...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

...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&amp;data=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424140187&amp;sdata=ypez4T%2BQyKSK3MC7RYvPuPHlPw5Xjur8cE4tEh85BA0%3D&amp;reserved=0 -- Joshua C. Colp Digium - A Sangoma Co...

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

...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

...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...

...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...

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,

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

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 --

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

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.

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

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

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