similar to: asterisk@home V's Asterisk

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-000000a8] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-000000a9] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure

Hi I have the latest a@h installed and everything is working perfectly. I have been told by a collegue that a@h doesn't use the full potential of the machine it is installed on i.e. the CPU & Memory, unless the kernel has modified. He is unsure where he heard this from and I wonderd if anyone had any other information about this or knew where I could find some? Many Thanks Scott

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only

I was quite surprised to find a message in my in box from SellVOIP a day or two ago. It indicated I was running out of credit which was a surprise as I thought they'd gone under a large number of months back. So I ran upstairs, added their entry back to sip.conf, uncommented a couple of lines in extensions.conf and I'm again using sellvoip to make outgoing calls. The reason I was

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine

Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it all seems to work except I have one way audio. I'm still using SIP, not pjsip. As soon as

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/57d1002d/attachment.htm

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

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

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

Hi; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter

hi all, can anyone emailme the .conf of asterisk at home, i cant download the full size tar or iso because of a network problem that fu*** every big file download.... and i just wanna learn not change my distro bye and thanks! -- Luis Diaz - Un obsesivo con proyectos! :oP

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

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