search for: jtodd

...register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re: Message from iptel.org SIP admin > >At 11:42 PM 2/23/2003, John Todd wrote: >>While I understand SIP to a "reasonable" degree, I don't see what >>the problem is below with the >ACK...

...could be written: http://www.redorbit.com/news/technology/1699391/rtx_releases_dectcatiq_20_usb_dongle/index.html?source=r_technology DECT really is a nice protocol if properly done - wireless phones should really just work like this everywhere. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...hipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all possible symbols on a North American keyboard, as it was getting exhausting and possibly silly. calls abandons staffing average Speed of Answer Sorry, but the user's mailbox can't accept more messages. Please enter the conference call number for t...

...ited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial "purchase" page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...eo material. In the interests of keeping this thread from getting out of control, please limit yourself to factual, content-rich posts. "I hate YouTube" or "Why didn't you film blah" is something we can discuss off- list. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...st can't get without meeting face-to-face. Washington DC is convenient from Europe, with direct flights to IAD (Dulles), DCA (Reagan International), and BWI (Baltimore Washington) airports from most major European and South American cities. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...e a good match, we'll pass them to you! If you're an integrator, please don't fill out the form - have your customers fill out the form - press folks are always looking for end users, not integrators. http://bit.ly/asterisk-press JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan

[sorry for cross-post here from -biz, but there are significant non- cross-subscribed audiences] I'm looking for some case studies for people who are implementing Open Source Asterisk in Federal, State, and Local governments in the United States. Please reply PRIVATELY to jtodd at digium.com with contact data, and I'll follow up. I have some requests from various directions for governmental uses of Asterisk, for people who are willing to talk to a magazine/journal about their use of Asterisk and how it works for them. This kind of press contacts are vital to k...

...personally if we run out of space before we run out of project proposals! There are so many great OSS things out there, but so few square feet. :-) Please email me directly with your project details to be included on the list of applicants. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

...all for this, and I've been swamped with other things to do and haven't gotten around to it yet. Would you be interested in such a list that is specifically for discussing implementation issues for such higher educational institutions? JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

This email is off topic. If it offends you then suck it :-) But seriously, I know this will interest USA readers (unfortunately for now primarily on the west coast only). How cool is this concept http://techshop.ws/index.html Cheers, Dean ________________________________ From: Dean Collins Sent: Sunday, 3 August 2008 7:53 PM To: 'newtech-1 at meetup.com' Subject: TechShop

[moving to asterisk-users by request] On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote: > > On Jan 26, 2009, at 7:38 PM, James Lamanna wrote: > >>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote: >>> >>>> Hi, >>>> Is it just me, or does DTMF queuing not work properly? >>>> I'm consistently fa...

..."0", so this is not an issue of the digits being dropped due to ending with "0" or "00". Other notes: if I swap out priority 5 with other non-agi applications, there is no difference in results. [foo] exten => _X.,1,AbsoluteTimeout(4) exten => _X.,2,AGI(/home/jtodd/agi/timer1) exten => _X.,3,SetVar(STARTTIME=${HIRESTIMESTAMP}) exten => _X.,4,NoOp(${HIRESTIMESTAMP}) exten => _X.,5,AGI(some-harmless-other-agi-here) exten => _X.,6,AGI(/home/jtodd/agi/timer1) exten => _X.,7,NoOp(${HIRESTIMESTAMP}) exten => _X.,8,SetVar(ENDTIME=${HIRESTIMESTAMP})...

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY > requests) > > At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: > >>[snip] > >Sending NOTIFY to Aste...

Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master -------------- next part -------------- An HTML attachment was scrubbed... URL:

...rse, the talk/slides will be available with Creative Commons Attribution-Noncommercial-Share Alike 3.0 United States License, and I expect that I'll probably use all or parts of this talk a few times this year, given the focus on the economy. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have ind...