similar to: Problem with Queue periodic announcemnets

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

Grretings to all, I am having a problem with a customer's queue setup that I don't really understand. Background: Customer has 5+ queues and is using dynamic login to the queues based on SIP/XXX for example. There is a litle script that runs that allows agents to log into particular queues via the keypad. The user can log in to any queue that he wants, including multiple queues. The

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

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 =>

When I boot a machine from disc 1 of 2, Centos 6.5 install dvd, I get to a grub prompt. I have no idea what to do from there, but clearly something isn't right. Shoudl I try to download centos 6 again and burn new discs? thanks, -chuck --

Suppose i want to compute a 95% highest density for a beta distribution beta(a,b) the two end points x1 and x2 shoudl satisfy the following two equations: pbeta(x1,a,b)-pbeta(x2,a,b)=95% dbeta(x1,a,b)=dbeta(x2,a,b) Is there any fast way to compute x1 and x2 in R? [[alternative HTML version deleted]]

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

(Posted few days ago on qemu group but no reactions) Do I understand correctly that ssl shoudl be configured independently for libvirt and each hypervisor? I asked because I configured libvirt connection as qemu+tls://bambus.kjonca/system?pkipath=... (and on bambus in /etc/libvirt/libvirtd.conf) I set key_file = ... cert_file = ... ca_file = ... But after connect and lauching (on bambus) vm I

hi, I am just about to try (next week) some streaming to Icecast2 over a satellite uplink in a remote location. I was wondering if anyone has tried this and if there are any issues that I shoudl be prepared for. One issue I am wary of is the possible role latency will play in this system. Any pointers on streaming over high latency or satellite internet connections most appreciated! adam

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! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP

Hi All, I recently just took the time to play with the OGG format and compare it to mp3..and my conclusion is: SWEEEEET Where can I learn about writing a source for Icecast2? I written sources for Shoutcast..that was pretty straight forward..rip through an mp3 file and send the bytes to shoutcast at specific intervals. OGG is VBR...so I'm having trouble wrappign my head around how I shoudl

Most documents around there talk abour "dsync", but the modern way is "doveadm backup". This command is not documented in the wiki and there are a few details missing, like how to use it thru SSH. I am currently doing some tests about how to backup my mdbox. I can do tests in local using: $ doveadm backup -u jcea -m proveedores/dovecot mdbox:/tmp/aa/ This will