similar to: Business telephones

This is a little off topic, but maybe someone can help. I have a Win2K laptop that's part of a corporate domain. When I bring it home and add it to my home LAN, it refuses to access any SMB machine (including a Samba server, Win98 or Win95 box) complaining that "There are currently no logon servers available to service the logon request." Is there any way to fix this easily? I

I am getting the following error when i am trying to generate the code. In the optimization pass I have inserted some calls llvm.annotation.i32 class to provide indentification tags to loops to be used in subsequent passes. Codegen is complaining about these annotations. llc -march=c ham1_seq.bc3 Error: Code generator does not support intrinsic function 'llvm.annotation.i32'! llc

Hi, Is there a source code example for using J-Ogg (with a Vorbis Stream) as an Ogg Decoder. The Java API docs are somewhat limited. I was able to extract Comment Header and Indentification Header information from a Ogg file. I was also able to process packets of a Vorbis stream. Unfortunately, the audio sound is just white noise. I don't know if some synchronization is necessary. I am also

I am preparing to setup a system using Cisco 7940 and 7960's I have the 7.1 SIP firmware on them. One issue I have run into is how to silence the ringer if a call comes in and you don't want to take it. Many phones have a DND button. I know the 79XX has the DND in the menu but it is to cumbersome to go into the settings then phone preferences then the DND and select yes. Is there any other

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.

Ok, let‘s stop the open64 "polution". Whether the design is as you stated doesn‘t simpler, the code before and after the change already tells us. We take detailed investigation on gcc support for hwloop, then we come to the conclusion we are essentially the same. So i think the idea can be shared among different compilers, general abstract tripcount, make pseudo operators for

Hi Could someone explain to me the difference between LOGIN and PLAIN? I've been googling for a while, but haven't found anything. Thanks. Simon

Hi! I have installed everything, asterisk, pwlib,openh323, chan_oh323. And now? I want to install ophone to talk, but I don't see what is the asterisk role. I mean, ophone lets us to talk with another phone,... why do we need asterisk? What does ophone do and what dows asterisk do? Thanks for all your help Mireia

I've just got a Snom 190 phone with which I'm really pleased. I can get the LEDs on the keys to light in response to an extension being in use which is cool, but there's a feature I'd like to implement. I'd like to light one of the LEDs on the keys to light if I have night-service enabled. My Night service is currently controlled by a variable in the database. I could go

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

My company is based in Australia and we have a need to be able to present CLI (ANI) and DNIS to an analog extension. Currently our PABX vendor is saying they can only deliver CLI between the first and second cadence. The system will collect calls via an aggregate of ISDN PRI services from the PSTN and then direct them to the analog extensions in a hunt group configuration. It is important that

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

Here is a quick tally of the various things people asked for.. I'm going to go thru the list and weight the results based on my scale of 1-10. This is just a count of each "item", otherwords how many times that item came up. Some things I considered as bugs and lumped them as bug-fixes For the various requests for codec's I broke out which ones people where asking for.

Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud

can anyone recommend a plain ol fashioned telco in north america where i can get a DID with call hunt and voicemail which can be programmed via web interface. thanks and pls pvt joe

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>