similar to: Problem with CLI output on Asterisk@Home

I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received

Hello everyone. Is it possible to do some very basic voice recognition from within Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I want to dial from my mobile phone. Dialing digits on my mobile phone while driving is not all that safe... Thanks for any input, Cosmin Prund

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT

Hello everyone. I'm working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? I already tried using the local channel for dialing (so I can put in

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

hi, i was able to get this working with google talk. i entered myusername@gmail.com using the gtalk2voip.com website's "invite" box, and as a result, saw a request from service@gtalk2voip.com to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten => 3501, 1,

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hello: I'm using: - samba-common-3.0.9-1 - kernel 2.6.5-1.358 - FC 2 - openldap-servers-2.1.29-1 We're running an NT4 domain using an LDAP backend, and everything was running fine until recently. The first thing that I noticed that new users were suddenly being assigned SambaSID's that were previously being assigned to machines. Previously: Typical User Entry: uid: john

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

Hello. I've got a network of FreeBSD servers which traditionally hosted a classic domain. I upgraded some months ago, removing the old PDC and BDC and migrating to an AD DC controller in a jail. This is working fine with Samba 4.4.13. Now I'm trying to add a second DC, so I created a new jail on another physical server and went on with the setup, following: >

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

mbox messages gets header corruption caused by an extra linefeed after Content-Length Users sees their mails in Sent mbox folder without the from and to fields, without attachments and with the date of 1/1/1970 Diego. --- Here is an anonymized header: >From xxxxxxxx at xxxxxx.xxxxxx.xxxxx.xx.xx Tue Jun 03 09:14:33 2008 Message-ID: <xxxxxxxx.xxxxxxx at xxxxxx.xxxxx.xx.xx> X-UID: 3913

On 26/06/2019 15:28, Andrea Venturoli via samba wrote: > Hello. > > I've still got a couple of NT domains and I'd like to upgrade them to > AD. In these days I had a chance to try to migrate one of them, but I > ran into troubles and had to go back. I don't know when I'll have the > chance to try again (probably not before some months), but I need to >

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

I'm trying to set up our test box here. Identical versions and setup to our devel box. It is part of the domain (has already been joined). And there was a problem with the secrets.tdb file (corrupted or whatever). winbindd.log: --- [2004/10/20 08:33:46, 1] nsswitch/winbindd_util.c:add_trusted_domain(166) Added domain XXXXXXXX XXXXXXXXX S-1-5-21-1645522239-1202660629-725345543

Hello. I've still got a couple of NT domains and I'd like to upgrade them to AD. In these days I had a chance to try to migrate one of them, but I ran into troubles and had to go back. I don't know when I'll have the chance to try again (probably not before some months), but I need to understand what went wrong before that. The starting situation: Samba 4.8 running as AD

As the subject line says: Is PING a good indicator of network latency? If not, how can I measure latency? Thanks, Cosmin Prund

What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07]

Hi, I've setup samba 3.4.7 to use idmap_rid as per the documentation: idmap backend = rid:DOMAIN=500-100000000 idmap gid = 500-100000000 imap uid = 500-100000000 It seems to work for groups: wbinfo --group-info="domain admins" domain admins:x:100512 PsGetSid v1.43 - Translates SIDs to names and vice versa Copyright (C) 1999-2006 Mark Russinovich Sysinternals -