similar to: Announce option for meetme - is it used?

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.

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 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

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

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

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

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

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

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

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

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

HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call?" Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

I have an issue with call parking and hope there is some undocumented feature for this. ;-) We are replacing our legacy PBX with asterisk, but to save money over time (handsets and network), I am trying to maintain the use of our legacy PBX. Asterisk extensions can not use the call parking features (not usable over trunk cards) of the old PBX, so I have to get the old PBX to use asterisk's.

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I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

My Dell 2800 Dual 64bit Proc. machine came in with hyperthreading enabled. (they call it virtual processor??) I have been intending for a month to disable it. Tonight, I rebooted, turned it off, and let the system come up. zaptel loaded and asterisk loaded, but both of my t1s were red. (it is a TE411P) /proc/interrupts looked OK, zttest gave OK numbers. I doublechecked all of the files in case

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

Hello frieds, do I need to do 'net rpc join -U Administrator' (to selfjoin the localhost to the domain) if I have samba3.0.4 running in ROLE_DOMAIN_PDC and with ldapsam-Backend and want to use winbindd for ntlm_auth squid authentification? -- Best regards, Malte mailto:malte.woelky@gmx.de _________________ Malte Woelky -=[SkyNet]=-