similar to: asterisk linkedin group

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.

Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. ? Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any

Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. ? Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA

Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode, Connect to IVR, List all options and then if they dial 0 or timeout, ring every phone in the

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion dollar company in just 2 years. This company's compensation plan is the best and quickest that I have

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can?t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

Hi, Already posted once but I need more feedback. What kind of servers is everyone using for asterisk and what problems have you ran in to ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050630/dd52bf35/attachment.htm

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

Hello group, I'm running asterisk @ home 1.5 - I would like to change these messages(call attend) to Spanish, how I can do that. Thanks, Nelson

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: