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...le 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 side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org

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

...a "magic" spot in Free PBX's configs to add the wait for all calls on that PRI, or do I need to alter the FreePBX code to add it when creating the conf. Files? Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org "Steven" <asterisk@tescogroup.com> wrote in message news:e1e54h$2f5$1@sea.gmane.org... > Thanks for the info. > > I went to add the Wait(2), but am unsure where to do it. > My context is "from-pstn". > > M...

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.

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

I will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

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

I have seen conflicting references in regards to the Digium Wildcard TE411P echo settings in zapata.conf. Does anyone have the official word on this? Should echo cancel be enabled in zapata.conf if the card has built in EC? If so, should a particular EC method be compiled into the zaptel build? My reference, which has echo: My zaptel is 1.2.5 context=from-pstn switchtype=national

>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

>From discussions with the receptionist staff, this is what we need: X number of buttons for parking slots. These buttons should be lit when a call is parked there. When on a call, just pushing an unlit button will park the call. (The do not want to hit hold, transfer, etc.) Hitting Transfer and the button would most likely be OK. To pick up a call, they can push the parking slot button. The

hot to asterisk multiport...??? example 5060, 5061, 5080 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070815/8a515309/attachment.htm

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris & Linux explode when loading login_page.php.

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

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.

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as

Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0,

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