search for: leif

...that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the...

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Leif, Did You faced that with the exact same Chrome version? Since this version is the only one having this issue. I have had this kind of arrangement (intro + live stream) for decades. Technology changes but the idea is the same. I have tested a lot of hardware and combinations. I do have a fail over...

...ing to implement a show/hide details function. I have the show part done, but I''m not sure how to get it to hide. My app looks like this so far: View: <%= link_to_remote "Show/Hide", :update => "details", :url => { :action => "show", :id => leif } %> <p id="details"></p> Action: def show @leif = Leif.find(params[:id]) render_partial ''show'' end _show.rhtml: <% for column in Leif.content_columns %> <p> <b><%= column.human_name %>:</b> <%=h @leif.send(column...

...s 256 kbits/s 48 kHz ISO-MPEG2 L3 - live is 256 kbits/s 48 kHz ISO-MPEG2 L3 http://185.139.168.34:8000/vara - live is 128 kBits/s 48 kHz, possible ISO-MPEG2 L3 - there is no intro Of cause there is a certain difference between a file and a stream. Yours, Oskar > On 18 Apr 2018, at 17.38, Leif Scriba <leif at scriba-bb.de> wrote: > > > Hello Oskar, > > No i didn’t face it with this exact version of Chrome but with different versions and other players/ Webbrowsers. > IMHO you should first try to set encoding of intro and stream identical. And try different pla...

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media...

...nnels/ patch -p0 < /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen <leif@hacklocalhost.com> http://www.hacklocalhost.com

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

...xTS[i]<-max(DF$TS[DF$abc==abclist[i]],na.rm=TRUE) } maxTSdf<-data.frame(device=I(abc),maxTS=maxTS ) DF<-merge(DF,maxTSdf,by="abc",all.x=TRUE) DF<-Df[DF$TS==DF$maxTS,,drop=TRUE] DF$maxTS<-NULL This seems a bit lengthy for such a simple task. Any simpler suggestions? -Leif K. Leif Kirschenbaum Senior Yield Engineer Reflectivity, Inc. (408) 737-8100 x307 leif at reflectivity.com

...n + txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +----------------------------------...

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

...documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Thanks, -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus ICQ: 3445119 FWD: 18924

Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? Thanks, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi

Hi: Which was released for free download under a Creative Commons license for "The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen". Regards. --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pi...

I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif

...risk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -----END PGP SIGNATURE-----

...s 48 kHz ISO-MPEG2 L3 > > http://185.139.168.34:8000/vara > - live is 128 kBits/s 48 kHz, possible ISO-MPEG2 L3 > - there is no intro > > Of cause there is a certain difference between a file and a stream. > > Yours, > > Oskar > > > On 18 Apr 2018, at 17.38, Leif Scriba <leif at scriba-bb.de> wrote: > > > Hello Oskar, > > No i didn’t face it with this exact version of Chrome but with different > versions and other players/ Webbrowsers. > IMHO you should first try to set encoding of intro and stream identical. > And try differen...

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one