similar to: Queues - calls going to agents lised as "In use"

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

I have setup several queues for a customer. Their periodic announcement says please wait for the next available agent, or press * to leave a voicemail. This does not work when the message is playing. The message stops, but the user is left in the queue. Q-exit with * works the rest of the time fine. Has anyone seen this or know if it shoudl actually work differently? Regards to all, Joe

currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): "Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer]

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Haile the list ;-) I am new to this list and this is my first post so grretings to all and nice to meet you. I am setting up rsync here to do a MASSIVE copy from one machine to the other. I am moving mail servers so I must copy /var/spool/mail/ over to the other machine. I understand and have used locally the copy methods(manpage is good) but I need to ask what will be the fstest possible

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>

Hi list, I have a strange permission problem with secrets.tdb on an nfs mounted filesystem. A strace of smbd shows the following lines: ######################################################################## open("/etc/samba/private/secrets.tdb", O_RDWR|O_CREAT|O_LARGEFILE, 0600) = 4 fcntl64(4, F_SETLKW64, {type=F_WRLCK, whence=SEEK_SET, start=0, len=1}, 0xbffff010) = -1 EACCES

Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. -------------- next part -------------- An HTML attachment

On 4/8/06, Joe <jrothstein@comcentrixs.com> wrote: > Remove the SIP /400 entry from the Asterisk DB. > > Database del .... At asterisk prompt. > > Or look at the wiki for info on how to remove it. > > Or make sure the SIP/500 uses a different IP address than the old SIP/400. > > Joe > > > > thanks for your reply i've tried to remove the entry in the

Hi, I am about to spend one week of my holidays with transfering our (about 20 clients) NT4 domain to a AD DC one. We are running a samba NT4 PDC on debian buster which is offering dns (bind9) and dhcp (isc-dhcpd), too. I have an older server where I can play with an AD DC setup. If I see I won't make it in a week, I would like to be able to return as smooth as possible to the present

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

I have my Samba PDC setup to use roaming profiles. If the user's profile directory exists (ex: /var/lib/samba/profiles/someuser) with the correct permissions and ownership then it works fine. However, I expect (incorrectly?) that Samba would auto-create the user's profile directory the first time the user logged in. Am I wrong or have I missed something? Thanks, -Mont

Hi, I have an access problem related to local unix groups on a FC2 box running the samba-3.0.10-1.fc2 rpms. The goal is to have a read-only share with sub-folders accessible only by one user. All users lised in a local unix group "admin" shall have access to all folders. The samba box uses an external active directory machine for authentication. Winbind is running on the machine.

Hello, I have set up a Unix workstation as a Samba server. It seems that when the number of pc guests reaches 20, no other pc users can browse the files on the Samba server. Is this a known problem?? My configuration file is lised below: #======================= Global Settings ===================================== [global] # workgroup = NT-Domain-Name or Workgroup-Name workgroup = ZYGOMIS