similar to: Alex Tew interview made possible because of Simon @ Simwood eSMS

I just upgraded a box to 1.2.3 this morning after encountering the issues noted earlier on the list. Everything is great. In fact, a LOT better. In the past few weeks, I've been battling with BV to address dropped outgoing voice packets (the flipside is that I haven't experienced this with other providers during tests), and an annoying mechnical 'chirp' at the start of a call.

I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood, and I was hoping someone on this list might have managed to do this. I have configured some numbers to route to a SIP endpoint %e164 at customer's server and convinced the customer to open up UDP ports 5060 and 10000 - 20000. Calling the number gets a SIP request from Simwood. The customer's machine

Dear Asterisk Community: With more than 10,000 readers worldwide, I've refreshed my free Asterisk PBX on Amazon EC2 ebook for 2011. It has been used by Avaya, Polycom, universities, and consultants everywhere. Did I mention it's free? If you have suggestions for its improvement or things you'd like to see, please let me know! It's online here:

I upgraded a box to 1.2.3 today after the bridging issues. I also had a big interview planned that I was recording. Well, I had to redo the interview, because the in/out channels (when combined) were out of sync. I didn't experience this until this update -- I am going to revert to 1.2 stable, and see if there's a difference. I am curious to know if anyone's experiencing the same.

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download

I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

So says "The Voice of Asterisk," Allison Smith in this new and informative interview: http://www.ronaldlewis.com/interviews/2007/01/interview-with-allison-smith-north.html (I know this isn't the most appropriate place, but Allison is about as relevant as Mark Spencer and the community) -------------- next part -------------- An HTML attachment was scrubbed... URL:

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would

Simon Matter has graciously volunteered to provide RPMs taylored for Redhat and Fedora. You can download Simon''s RPMs from http://www.invoca.ch/pub/packages/shorewall/ Thanks, Simon! -Tom -- Tom Eastep \ Nothing is foolproof to a sufficiently talented fool Shoreline, \ http://shorewall.net Washington USA \ teastep@shorewall.net PGP Public Key \

Dear all I am a PhD student of Computer Scince at Simon Fraser University ( http://www.cs.sfu.ca) interested in applying to GSoC. My PhD is focused on theoretical computer science, but since Sep. 2008 I have started working on Software projects again. Currently I am working in COSTAR lab on a high performance regular expression engine based on Parallel bit streams technology. A considerable part

Hi, I understand that Simon may be discontinuing his OpenSSH work. Does anyone know if someone plans to maintain the patch? Thank you, -- ******************************************************* Quellyn L. Snead UNIX Effort Team ( unixeffort at lanl.gov ) CCN-2 Enterprise Software Management Team Los Alamos National Laboratory (505) 667-4185 Schedule B

Hey, RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail beolow will not get a Security Advisory. This only affects applications using DTLS, and I doubt there are many of those, but users should still upgrade to get this fix, just in case. See the OpenSSL advisory for some more details: http://www.openssl.org/news/secadv_20071012.txt If anybody were wondering, and

Hey, RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail beolow will not get a Security Advisory. This only affects applications using DTLS, and I doubt there are many of those, but users should still upgrade to get this fix, just in case. See the OpenSSL advisory for some more details: http://www.openssl.org/news/secadv_20071012.txt If anybody were wondering, and

Dear Simon, Thank you for your quick reply! I used to perform the GAMsetup in the following manner: GAMsetup sintax: x.summer: vector used for construct the spline knots<-14 N<-length(x.summer) x<-array(x.summer,dim=c(1,N)) G<-list(m=1,n=N,nsdf=0,df=knots+1,dim=1,s.type=0,by=0,by.exists=FALSE,p.order=0,x=x,n.knots=knots,fit.method="mgcv") H<-GAMsetup(G) with the

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing

Felix du Plessis Tel:               +27 (0) 12 640 0135  Fax:              +27 (0) 12 640 0151 Mobile:       +27 (0) 83 457 8718 -----Original Message----- From: xen-users-bounces@lists.xen.org [mailto:xen-users-bounces@lists.xen.org] On Behalf Of xen-users-request@lists.xen.org Sent: Friday, May 18, 2012 2:00 PM To: xen-users@lists.xen.org Subject: Xen-users Digest, Vol 87, Issue 32 Send

I've tried getting this running but mpg123 won't spawn. It spawns fine for the files but if I try streaming she doesn't work. I've tried with just about every stream at somafm.com w/o success. I can play them locally though. When I try to play them from the server from the command line I get: # mpg123 -s --mono -r 8000 -b 2048 http://160.79.128.40:8052 High Performance MPEG

I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register.

Yes, and they are aware of the problem. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan Tew Sent: Monday, February 02, 2004 1:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoicePulse IAX2 lag Is anyone else noticing high lag on their voicepulse IAX2 connections? We're seeing