search for: andersonfam

...at. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.org

...uot;call-by-call" trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org

...l only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage scenario in mind, is it worth the extra $800 for the echo cancellation? As a sidenote, I assume these cards are PRI-compatible, correct? Thanks! -erik -- Erik Anderson http://andersonfam.org

...nd file completes, I get this: -- Playing 'logic-main' (language 'en') == Auto fallthrough, channel 'SIP/445-0815e1d0' status is 'UNKNOWN' Any ideas? This seemed like it should be simple, but it's getting the best of me. Thanks- Erik -- Erik Anderson http://andersonfam.org

...higher load than this system would ever see, but I just want to be safe. Is there currently any script out there that would facilitate this sort of testing? Here's my current config: linux-2.6.21 asterisk-1.4.10 zaptel-1.4.4 wanpipe-3.1.3 libpri-1.4.1 Thanks! -Erik -- Erik Anderson http://andersonfam.org

...saged into working just fine. Thoughts? I'm currently negotiating with our TW Sales Rep. to see if they could provision a few test DIDs on a SIP trunk so I can verify compatibility, so I *should* hopefully have answers soon for many of these questions. Thanks! -erik -- Erik Anderson http://andersonfam.org

...lementation would involve a few different contexts for orginization and future flexibilty, as well as perhaps needing some code to set CID information correctly (I'm not sure if that's passed through automatically). Am I heading down the right path? Thanks! -erik -- Erik Anderson http://andersonfam.org

...nk interface [logic] exten => 270,1,Dial(IAX2/erika_idefisk) exten => 271,1,Dial(IAX2/erika_idefisk) exten => _ZZZ,1,Dial(${TRUNK}/${EXTEN}}) iax.conf: [erika_idefisk] type=friend username=erika secret=foobar host=dynamic context=logic Any ideas? Thanks! -Erik -- Erik Anderson http://andersonfam.org

...to do a completely raw dump of the d-channel? Here are my specs: linux-2.6.16 libpri-1.3.5 zaptel-1.2.19 asterisk-1.2.21.1 The PRI interface is a Sangoma A102...it's running the latest firmware and I'm running wanpipe-2.3.4-12 for the sangoma drivers. Any ideas? -- Erik Anderson http://andersonfam.org

Hi asterisk-users at lists.digium.com, Add me as a friend on Last.fm so we can share our music taste :)? Check out what I'm listening to: http://www.last.fm/user/shina01/?invitedby=shina01&tp=ff_tp_b -------- I also sent you a personal note: "boo!" Signing up is free and takes less than a minute. Just click the link to automatically become my friend.

Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? -------------- next part

...ick out a bluetooth conference phone that I could use with a softphone along with my asterisk server. I've been looking at the TrendNet TVP-SP4BK. Have any of you used this device or any other bluetooth conference phone? How have your experiences been? Thanks! -Erik -- Erik Anderson http://andersonfam.org

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

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

...at device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org

Andrew D Kirch Indianapolis, United States Good day, I am Mr. Andrew D Kirch, a native of Indianpolis, United States and I am an Asterisk Hacker with the Summit Open Source Development Group. First and foremost,I apologized using this medium to reach you for a transaction/business of this magnitude, but this is due to Confidentiality and prompt access reposed on this medium. Be informed

...59-1 On 10/22/07, Vincent <vincent.delporte at bigfoot.com> wrote: > > 2008 might be a good year to update "* - The future of telephony" :-) Version 2 of TFOT was just released a few weeks ago... http://downloads.oreilly.com/books/9780596510480.pdf -- Erik Anderson http://andersonfam.org ------------------------------ Message: 18 Date: Mon, 22 Oct 2007 17:57:44 -0400 From: Jared Smith <jsmith at digium.com> Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists....