search for: davidduffett

Hey all, I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. I've traced this same box on two different analog providers so it probably isn't a problem with them. I DO have callerid enabled and not sure I can turn it off (if they will let me). Any other ideas of making it ring through faster? By the

...sterisk-users > -- [image: Digium logo] *David Duffett* Digium, Inc. ? Director, Worldwide Asterisk Community 6 Landscape Close ? Weston on the Green ? Bicester ? Oxfordshire OX25 3SX ? UK direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com ? http://asterisk.org <http://www.asterisk.org/> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150630/47adbf3c/attachment.html>

...sterisk-users > -- [image: Digium logo] *David Duffett* Digium, Inc. ? Director, Worldwide Asterisk Community 6 Landscape Close ? Weston on the Green ? Bicester ? Oxfordshire OX25 3SX ? UK direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com ? http://asterisk.org <http://www.asterisk.org/> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160823/96008e48/attachment.html>

...e: Digium logo] > *David Duffett* > Digium, Inc. ? Director, Worldwide Asterisk Community > 6 Landscape Close ? Weston on the Green ? Bicester ? Oxfordshire OX25 3SX > ? UK > direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 > twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett > Check us out at: http://digium.com ? http://asterisk.org > <http://www.asterisk.org/> > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the As...

...hank you, David Digium logo David Duffett Digium, Inc.??? Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green ??? Bicester, Oxfordshire OX25 3SX ??? UK direct/fax: ? +1 256 428 6119 ?? mobile: ? +44 7722 442236 twitter:? dduffett ? ? linkedin:? www.linkedin.com/in/davidduffett ? Check us out at:? http://digium.com ??? http://asterisk.org ? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120912/2decac77/attachment.htm>

...ister All the best, David Digium logo David Duffett Digium, Inc. ? Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green ? Bicester, Oxfordshire OX25 3SX ? UK direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130516/4f805390/attachment.htm>

On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider. I haven't been able to reproduce the problem, and the log doesn't contain anything useful. How can I increase the log verbosity for SIP registration-related events? I've looked through logger.conf and tried searching with Google, but I haven't been able to find a clear answer. This

...ards. To the future... David Digium logo David Duffett Digium, Inc. ? Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green ? Bicester, Oxfordshire OX25 3SX ? UK direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120731/06bb6c2f/attachment-0001.htm>

I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and the caller's number was Asterisk at 10 or or Astrk at 10 similar. (I don't recall the exact

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so I'm now working with distro packages (Ubuntu) and hand edited configurations files. I have three SIP "trunks", each providing me with a UK telephone number. They are for my home (1), my wife's home-based

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >