search for: madpilot

Zitat von Guido Falsi <mad at madpilot.net>: > So, trying to bind authentication to originate calls to registrations is > conceptually wrong in the SIP world. Maybe you can do that but that's > not the way the protocols have been engineered to work. Hi Guido, thanks for your answer. Well, I decided to do that, since...

Hi, I'm maintaining the FreeBSD ports for asterisk(With madpilot at FreeBSD.org as identity). Here's a link to the asterisk13 port for your reference: http://www.freshports.org/net/asterisk13/ I performed some tests with RC1 and am doing some final tests with the final release before committing the update. Up to now the ports forced using gcc, version 4.8...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

...isk port in the ports tree, and I routinely test Release candidates when available to speed up updating the port. So I'm especially interested in this issue being investigated, otherwise I'll have to hold up updating the FreeBSD Asterisk port. Thanks in advance. -- Guido Falsi <mad at madpilot.net>

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

On Tue, Nov 29, 2016, at 07:15 AM, Barry Flanagan wrote: > On 29 November 2016 at 10:56, Jonathan H <lardconcepts at gmail.com> wrote: > > > Any ideas why a VPS, cloned from another instance (DigitalOcean > > "droplets" if it matters), won't run on the new instance? > > > > Everything else is the same; region, memory, disk, hypervisor version etc.

...elopers actively working on Asterisk on a BSD platform, though my knowledge isn't comprehensive. It may be worth talking to the people doing the packaging for various BSD platforms, to see how involved they are, or if they know of people developing it directly. jnemeth at netbsd for pkgsrc, madpilot at freebsd for ports/dports, and sthen at openbsd for OpenBSD ports, for example. I know you're developing on NetBSD, but correcting for "not-Linux" would help everyone. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lis...

...sends an invite. That's why it works without being registered. So, trying to bind authentication to originate calls to registrations is conceptually wrong in the SIP world. Maybe you can do that but that's not the way the protocols have been engineered to work. -- Guido Falsi <mad at madpilot.net>

El 29/03/16 a las 08:29, Steve Howes escribi?: > I don't think you can. Whatsapp is a closed system. > > Steve And they change your code every day and make it always obfuscated. https://github.com/tgalal/yowsup/issues/887 Best regards. Emiliano.

Hi all, I've been making a wrapper script for the backup tool 'duplicity', allowing me to create config files for each resource, wherein I define whether a snapshot should be made prior to backing up the resource or not. Now I find that my snapshots never change .... The script creates a snapshot, creates md device, mounts it, runs backup against the mounted snapshot,

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. I have improved the situation by expanding my intrusion detection but it still stops every few days or so. I have a cron job that tests for it and restarts it

Hello list! When em0 has an inet address while bridge0 doesn't, it seems to be OK: ----- bs1% uname -a FreeBSD bs1.sp34.ru 7.0-STABLE FreeBSD 7.0-STABLE #0: Sun May 25 20:15:26 MSD 2008 root@bs1.sp34.ru:/usr/obj/usr/src/sys/BSM i386 bs1% ifconfig em0; ifconfig tap0; ifconfig bridge0 em0: flags=8943<UP,BROADCAST,RUNNING,PROMISC,SIMPLEX,MULTICAST> metric 0 mtu 1500