search for: viles

http://grandstream.com/BETATEST/HT488_496_386/

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

On 2019-12-16 05:34, Roger Price wrote: > On Sun, 15 Dec 2019, Phil Stracchino wrote: > >> I switched a while back to a Cyberpower PR3000LCDRTXL2U UPS with an >> external battery chassis. We've lost power twice since I installed it, >> and both times, the UPS has killed power to everything while still >> indicating 80% capacity before NUT has even initiated

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

> From: Akim Demaille <akim at epita.fr> > Date: 17 Sep 2001 11:22:19 +0200 > > | 2001-09-16 Tim Van Holder <tim.van.holder at pandora.be> > | * lib/autoconf/functions.m4: (AC_FUNC_GETPGRP) Don't rely on > | setpgrp() being present. I looked into that problem a bit more. Several programs use AC_FUNC_GETPGRP and GETPGRP_VOID, including Bash, Gawk, and

Hi all, I was initially going to send this email to a thread on llvm-commits where David explained the issue with poison and select to me, but then some quick googling led me to this thread. I've been thinking about a possible semantics for poison values in a certain way: The key semantic difference between undef and poison, as I understand it, is that when justifying an execution trace the

Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

On Mon, 16 Dec 2019, Phil Stracchino wrote: > One observation: > > "WARNING: I am unable to call script nut-journal to > read the NUT activity records in the systemd journal. > > .... OR, you're not running systemd. (Out, out, vile daemon. You shall not > pass.) I definitely need to check that the vile daemon is present and running. In upsmon.conf there are no

Hello. I would like ask about install last firefox version 1.5.0.3 under centos, what is thee better option, as rpm pachage I didn?t found nothing. Thanks in advance JM --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20060531/965ae660/attachment-0002.html>

...| | | \----------/ \----------/ \----------/ domain1.com domain2.com domain3.com 10.0.0.2 10.0.0.3 10.0.0.4 --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <htt...

Hello. I?m newbie in centos, I would like know more repositories as DAG repository and if there is conflicts using several repositories. Thanks Josep --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.centos.org/pipermail/centos/attachments/20060329/b40c07ab/attachment.htm

The option se.fit in predict.nls is currently ignored. Is there any other function available to calculate the error in the predictions? Thanks, Manuel ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m??viles desde 1 c??ntimo por minuto.

...exten => _0.,2,Set(CALLERID(Name)='Gil Estarellas') exten => _0.,3,Dial(Zap/g1/${EXTEN:1}||rf) exten => _0.,4,NoOp(${HANGUPCAUSE}) exten => _0.,5,NoOp(${PRI_CAUSE}) ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y m?viles desde 1 c?ntimo por minuto. http://es.voice.yahoo.com

I've had the same excellent responsiveness from telasip, on the rare occasion that I've had issues. YMMV -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Vile Sent: Thursday, April 06, 2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telasip

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the