similar to: Using Asterisk as a softphone

This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka

I wrote this script to have an easier time looking at the doc paths referenced from the default configuration files (currently using dovecot in Debian oldstable). Feel free to include in the distribution or wherever. Probably a better solution would be to publish the config files with the paths already expanded to the online URL, or a local filesystem path that works (in the latter case, I'd

I'm cvsupping 4.8-STABLE to 4.8-STABLE. The cvsup run finished about 23:32 on 7/13. After make buildworld, make buildkernel, mergemaster -p, make installworld, and mergemaster -v, I do (per mostgraveconcern, RIP): cd /usr/src/release/sysinstall make clean make all install The 'make all install' gives errors: In file included from keymap.c:40: keymap.h:4436: `keymap_be_iso'

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't

On 5/24/23 08:03, Steve Matzura wrote: > > ***  extensions.conf  *** > > > [general] > > [globals] > > ; Make sure to include inbound prior to outbound because the > _NXXNXXXXXX handler will match the incoming call and create a loop > include => voipms-inbound > include => voipms-outbound > > [voipms-outbound] > exten =>

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

A problem here getting winbind traffic to be encrypted using Kerberos. I have set up a test environment with a pair of servers (actually lxc containers): - samba server (ubuntu 16.04, stock samba 4.3.11) - client machine (ubuntu 16.04) joined with "net ads join" and winbind The client machine has the following in /etc/samba/smb.conf: ------- [global] #netbios name = client-ad

I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing.... I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right

I'm not really looking for working configurations as much as I am looking for people who can say "This is a solid product and I trust my business to a solution running Asterisk." As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time

Hi. I'm having problems with chan_mISDN configuration. Line "echotraining=yes" causes warning, when Asterisk is parsing misdn.conf and I'm confused why the PBX doesn't accept the setting. No matter which section I try to offer it, it is always invalid or out of range. The setting itself is supposed to be valid, it is in the sample configuration file of chan_mISDN 0.3.1.

I?m trying to follow the instructions on the wiki to move from antispam to IMAPSieve, but it?s not working at all. A bit of a background: dovecot-core 2:2.3.8-4~buster dovecot-imapd 2:2.3.8-4~buster dovecot-lmtpd 2:2.3.8-4~buster dovecot-managesieved 2:2.3.8-4~buster dovecot-pgsql 2:2.3.8-4~buster dovecot-pop3d 2:2.3.8-4~buster

I have Centos 6.1 and i'm configuring cluster with conga, lucci and ricci these services are installed and running without problem. I need this type configuration for a high availability of virtual machines which is run on kvm. My problem is at the time of configuring "service group", when i click to add appears the new windows wich say "add service to cluster" and i fill

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb

>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its not doing the hangup or the system. jerry --------------

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same

I've tried this issue and could reproduce this issue: 1.Convert a guest with 2 disks to rhev. # virt-v2v -ic xen+ssh://10.66.106.64 -o rhev -os 10.66.90.115:/vol/v2v_auto/auto_export rhel6.6-i386-hvm -on 2disk-test -of qcow2 [ 0.0] Opening the source -i libvirt -ic xen+ssh://10.66.106.64 rhel6.6-i386-hvm [ 16.0] Creating an overlay to protect the source from being modified [ 47.0]

On Tue, 22 Nov 2016 15:19:34 +0000 Brian Candler via samba <samba at lists.samba.org> wrote: > On 21/11/2016 17:21, Brian Candler wrote: > > I'd quite like to be able to fetch a ticket using the keytab > > I found a solution to that part by using a different form of > principal name with "hostname$" > > root at client-ad:~# kinit -k -t