search for: stevenberkholz

...hank 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 contact info with my new email address stevenberkholz@hirotecamerica.com

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

>From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

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

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

...ignature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/73 97e46b/attachment-0001.pgp ------------------------------ Message: 13 Date: Mon, 22 Oct 2007 17:12:54 -0400 From: "BerkHolz, Steven" <StevenBerkHolz at hirotecamerica.com> Subject: [asterisk-users] Split asterisk in two ?? One TDM and One IP only?? To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Message-ID: <5C4D356F944BC042A83B3828DF3E0B4412B01CC6D3 at tg12.tescogroup.com> Content-Type:...

This may be slightly off topic. I am using FreePBX, and using it's backup feature. Here is the question part: I would like to copy my backup off the asterisk server. >From your experiences, which approach seems more resilient to failure: Push the backup from asterisk to another server using STFP or FTP? Pull the backups from asterisk from another server using SFTP or other? The

I feel the need to apologize for my previous email to the list. I was thinking that I was sharing something that I am currently exited with, with my associates. I now realize that I was sooo... off-topic, it's ridiculous. Sorry for the improper post to the list. Feel free to keep any further comments, concerning my improper use of the list, off-list. Thank You, Steven B TeamVie at

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 ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if

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

XO fixed my caller ID name. I am using FreePBX and I can include a "wait" to my custom extensions. Is there a way to add a wait to the whole PRI? I assume that if I set immediate to yes, I can then have a "s" extension do the wait, but how would it get from the "s" to the DID extension? (also, I would rather not answer every call) Is there a "magic" spot

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

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.