similar to: Small site survivability

> ata1: port is slow to respond, this delay is known to occur on vacant SATA ports > ata1: port failed to respond (30 secs) > ata1: SRST failed (status 0xFF) > ata1: SRST failed (err_mask=0x100) > ata1: softreset failed, retrying in 5 secs > ata1: SRST failed (status 0xFF) > ata1: SRST failed (err_mask=0x100) > ata1: softreset failed, retrying in 5 secs > ata1: SRST

Hi all, I installed CentOS 5 on DELL machine. Then, When the system comes up, I always get below erros. Could you pls help me to solve this out before I setup this box for productiuon use ? these are errors. usb 1-2: device not accepting address 2, error -71 ata2: port failed to respond (30 secs) ata2: SRST failed (status 0xFF) ata2: SRST failed (err_mask=0x100) ata2: SRST failed (status

As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn

Hello, If anyone is using a TE121B card and it works reliably (i.e. no "HDLC Bad FCS" or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a

I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call IN/OUT through the gateway (without asterisk in the middle), but it is not working. I'm working with tech. support from the source I purchase the unit from they we are just emailing back and forth and the unit is still not working. Can

Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix

How do you require a password for a voicemail box? I have been searching all day, and can't find any type of "security" setting for voicemail. I am looking for some what to have some minimum security like "no blanks, can't be the same as the extension, can't be sequential numbers or repeated numbers". I know that not all of these options may exist, but there has

We have 3 sites located across the US. Each has its own Asterisk PBX with a stand-alone installation. The sites are connected via VPN, fully messed, with fractional DS3s to the same service provider. We'd like to set it up so that if the PBX at site A fails, it fails over to B or C, if the PBX at B fails it fails over to A or C, and so on. I know Cisco CallManager supports this using

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten => 558,1,Answer exten => 558,2,Playback(message.wav) exten => 558,3,Dial(SIP/439@CallManager) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error :

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

Hello Folks, The official email address for this list is `freebsd-security@freebsd.org'. Due to convention, there is an email alias for this list: security@freebsd.org, just as there is for hackers@ & freebsd-hackers@, arch@ & freebsd-arch@, and so on. The security@freebsd.org alias has been the source of occassional problems. Several times in the past, postings have been made to