similar to: bugetone 200's

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports working fine. I am unable to get the other to work. Does anybody have an example configuration to make both work. Both are registering fine but there's just no dialtone on the non working port. TIA

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

I know I heard people looking for inexpensive ADSI phones a while back. Can't vouch for these guys, but this looks like a reasonable deal for what perhaps are new phones. No experience with ADSI, myself, but thought I'd pass it along. These folks also carry the Nortel analog units that do intercom and 2-line, as has been discussed in prior threads. JT ---------------------

what is a typical server / processor / memory configuration for approx 50 user install? phone -> asterisk -> pri thanks Phil.

Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and

Still finding my feet here. I need a server which can take approx 100 sip users accessing 24 channels through pri. Approx 20 concurrent calls. 6000 calls a month. Are there any "rules of thumb" when it comes to sizing hardware. I've checked the wiki but nothing close to what I need plus some of the information is really old now and may not be relevant. Also, what server

"People are always blaming circumstances for what they are. I don't believe in circumstances. The people who get on in this world are the people who get up and look for the circumstances they want and if they can't find them, make them." George Bernard Shaw What can I say "Document or Die". Without the broader base Asterisk will just be a L33T toy. Cheers, Dean

Hi, Just got a couple of Bugetone's yesterday.. I updated the firmware to 1.0.3.72 but now it seems to have problems getting the time from the default NTP server.. I also tried time-b.nist.gov but still the time does not seem to work.. I am able to contact the time server from my PC.. Anyone got any ideas? -- ______________________________________________ http://www.linuxmail.org/ Now

asterisk-users-request@lists.digium.com is believed to have said: >> Haha....well the MWI is the blinking blue LCD. The message button >> is "reserved for future use" Hang in there. There will soon to be some >No the message button call the number you configure in the web >interface. Presumably voicemail, but could be your mistress. Then it's easy to get

Guys. How and which tools to use to load test an asterisk install? Say for example, you need to see how many calls can be routed thru before losing quality and making the cpu jump to the roof?

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

Hello list members! I joined this list on May 1st and I can see I have a lot to learn. Thanks to everyone - especially those *@samba.org folks. You are so appreciated. Background in my plea for help: I'm the IT guy for a family owned community bank in Southern Colorado. I am very tired of the limitations and security problems of Micro$oft and have committed to learning and using Linux

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>