[asterisk-users] Updated: 10 Minutes: Asterisk PBX on Amazon EC2
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: http://ronaldlewis.com/10-minutes-asterisk-pbx-on-amazon-ec2-quickstart-guide http://www.scribd.com/doc/3905321/10-Minutes-Asterisk-PBX-on-Amazon-EC2-A-Quickstart-Guide Thanks for your support! Best, Ronald Lewis Author, 10 Minutes: Asterisk PBX on Amazon EC2 Denver, Colorado http://ronaldlewis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
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 this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GTalk/Jabber passing audio in 1.4.1!
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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is used in U.S. prisons?
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) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [resolved] asterisk 1,4 and google talk
1.4 has been released, and it's still crashing. I guess it hasn't been resolved yet. On 11/10/06, Mik Cheez [EMAIL PROTECTED] wrote: Mani, I've gotten the same result both dialing from a gtalk client to SIP, as well as an SIP call to gtalk. You can run a jabber debug before the call is placed to see more debug info on what's causing the crash. With the module in Beta, I believe it's just a bug that needs to be worked out. Below you'll see the output of one of my calls. :M sysmast01*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/asterisk4273D1E7 type=set id=35 from=[EMAIL PROTECTED]/Talk.v1001EE54E14session type=initiate id=2077360010 initiator=[EMAIL PROTECTED]/Talk.v1001EE54E14 xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= sysmast01*CLI JABBER: gtalk_account INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq sysmast01*CLI *** glibc detected *** /usr/sbin/asterisk: munmap_chunk(): invalid pointer: 0xb7e47b73 *** === Backtrace: = /lib/libc.so.6(cfree+0x1bb)[0x9b667b] /usr/lib/asterisk/modules/chan_gtalk.so[0x82bde5] /usr/lib/asterisk/modules/chan_gtalk.so[0x82c436] /usr/lib/libiksemel.so.3(iks_filter_packet+0x129)[0x278789] /usr/lib/asterisk/modules/res_jabber.so[0x4000c7] /usr/lib/libiksemel.so.3[0x276b55] /usr/lib/libiksemel.so.3(iks_parse+0x5c1)[0x274ad1] /usr/lib/libiksemel.so.3(iks_recv+0x98)[0x276488] /usr/lib/asterisk/modules/res_jabber.so[0x3fbd70] /usr/sbin/asterisk[0x80eadfb] /lib/libpthread.so.0[0xac03db] /lib/libc.so.6(clone+0x5e)[0xa1a06e] Mani Sridhar wrote: hi, it turns out that the iksemel library (which i installed using an rpm) was returning 0 when the function iks_has_tls() was called. it should return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by running a test program i wrote, that calls iks_has_tls . it returned 0. i downloaded iksemel source, compiled it and now the test program returned 1. now, jabber show connected shows the google talk account as connected, but i don't see this buddy online on my other google talk buddy list. i added an extension in extensions.conf that calls Gtalk/buddy, and as soon as i call this extension, asterisk terminates due to a segmentation fault. it didn't seem like a core was dumped - i'm still looking for it. thanks sridhar _ Live the life in style with MSN Lifestyle. Check out! http://content.msn.co.in/Lifestyle/Default ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Lewis Producer, Interviews Founder and CTA, Riverscape http://www.ronaldlewis.com/interviews http://www.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reg errors? Other anomalies? Check those capacitors!
Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing:* The motherboard's capacitor!Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: (Job) Full-Time Asterisk Opportunity
There is currently a permanent, full-time Asterisk opportunity available for the right candidate. The client is seeking to fulfill this position soon. Here are the particulars:* This position requires that you work from home, and be within a reasonable distance to a major airport * You should be comfortable with a moderate amount of travel* You must have good working knowledge of Asterisk, which includes the ability to install and configure the PBX* a dCap (Digium certification) is a plus, but not required * Experience with Python, C++, and/or other scripting languages are helpful, but not requiredPlease submit your resume to ron (at) ronaldlewis.com -- I will not respond to inquiries on the list. Regards,Ronald LewisFounder and CTA, Riverscapewww.ronaldlewis.comwww.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with Monitor in 1.4?
Has anyone noticed any anamolies with Monitor not recording in 1.4 beta2? I just did a half hour interview this morning, and for the FIRST time ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do my interviews -- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviews http://www.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with Monitor in 1.4?
I've used various versions of Asterisk for many things ... this isn't necessarily a production thing. I'm fully aware of the nature of beta software (I've tested a lot of software in my time), and I'm simply asking for feedback ... right now, this type of feedback doesn't help, but thanks anyway. On 9/30/06, Doug Lytle [EMAIL PROTECTED] wrote: Ronald Lewis wrote: ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do Sorry, but I've gotta say it.Then you shouldn't be using BETA software in production.Doug--Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Ronald LewisProducer, Interviews Founder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] University switches to Asterisk
I stumbled upon this yesterday while reading my usual news sites, and added it to Digg.com -- so be sure to digg it for even more exposure -- http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_AsteriskThis is a great example for Asterisk, since most folks remain quiet on its large-scale deployments.-- Ronald Lewis Producer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com On 9/13/06, Doug Lytle [EMAIL PROTECTED] wrote: Interesting article I found linked from Groklaw:Sam Houston State University replaces Cisco CallManagers, Nortel PBXswith Linux-based VoIP and messaging servers http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1Doug--Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help spread the word about Asterisk!
Recently, Network World published an article about a Texas university migrating their 6,000 students from a Cisco VoIP solution to Asterisk. This is the best example to date of a large-scale Asterisk deployment, considering how secretive the numbers are and where. So, help push this news to the top of digg.com by digging the URL below:http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_Asterisk -- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behavior with SIP registration/connectivity
With Asterisk 1.2.* and TRUNK, I've noticed some odd behavior with SIP registrations and connectivity over the past day. First, I noticed Asterisk REFUSED to register any trunks over SIP, prompting a lot of timeout messages. It also refused to accept registration requests from internal phones, rendering any attempt to place a call pointless. Everything was working flawlessly until yesterday. This morning, I narrowed down the SIP registrations from 5 to only 1-3 active -- they all registered fine. When I included all five, nothing registered, including internal clients. Strange behavior -- I've never witnessed this before. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * 1.2.4 1.2.6: Ringing anamoly
I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects,but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned Broadvoice issue. For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I'm FED UP with BroadVoice
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 porting between VoIP providers, but is there a remote chance I can save my phone number? I'd sure hate to change numbers again -- this has been a NIGHTMARE. Everyday, calls are dropping, and I'm calling people back 2 to 3 times to establish a decent connection. And their response (paraphrasing): We've made the best effort to ensure your service is functional ... but there are some things beyond our control with VoIP. Not good enough! I had great service with Vonage, and the times I use VoipJet, it works perfectly! Thanks in advance for any pointers. Ronald Lewis Denver, Colorado http://www.ronaldlewis.com/interviews ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alex Tew interview made possible because of Simon @ Simwood eSMS
I want to personally recommend Simon @ Simwood eSMS for any DID, SIP or IAX needs in the UK!Simon responded PROMPTLY and PROFESSIONALLY to my request to establish a DID for my interview on Jan. 26 with Alex Tew, creator of phenomenon MillionDollarHomepage.com.The only thing I needed to do was register with eSMS's server, and the interview commenced FLAWLESSLY -- the call was CRYSTAL clear to my location here in Denver, Colorado.I am HIGHLY impressed with these guys, so consider them for any UK VoIP needs. Regards,Ronald LewisFounder CTA, RiverscapeDenver, Colorado303-557-0153[EMAIL PROTECTED][EMAIL PROTECTED] www.riverscapecorp.comwww.ronaldlewis.com- Listen to my interviews with Mark Spencer, Alex Tew, and others at http://www.ronaldlewis.com/coffee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BroadVoice subscribers and Asterisk 1.2.3
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. Since 1.2.3, I haven't (so far) noticed anything unusual.Regards,Ronald LewisFounder CTA, RiverscapeIndependent ConsultantDenver, Colorado303-557-0153 [EMAIL PROTECTED][EMAIL PROTECTED]www.riverscapecorp.comwww.ronaldlewis.com -- Listen to my recent interview with Digium's Mark Spencer at http://www.ronaldlewis.com/coffee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor and * 1.2.3: Sync issues?
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.Best,Ronald LewisDenver, Coloradowww.ronaldlewis.com - An interview with Mark Spencer: http://www.ronaldlewis.com/coffee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Announce] Mark Spencer interview
Greetings! On January 19, 2006, I featured VoIP and open source telephony pioneer, Mark Spencer, on my podcast, Technology Coffee. To listen to this interview, visit http://www.ronaldlewis.com/coffee. Also, Tom Keating, CTO and VP at TMC Labs, has blogged about it as well. http://blog.tmcnet.com/blog/tom-keating/asterisk/mark-spencer-podcast-interview.asp http://blog.tmcnet.com/blog/tom-keating/voip/gabcast-audio-blogger-service.asp Regards, Ronald LewisDenver, Coloradowww.riverscapecorp.com www.ronaldlewis.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Trunk) in production
Just out of curiosity, how many of you are using trunk in a production environment? Are you performing regular compilations of the code as well? Do you explicitly prefer trunk over stable, or vice versa?Ronald Lewis www.ronaldlewis.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH engaged while holding for ANOTHER party (1.2.1)
I was just on the phone with BroadVoice support, when the engineer placed me hold. Low and behold, my own MOH was engaged seconds later. I've never experienced this until 1.2.1. Has anyone else experienced such an oddity?-- Ronald LewisDenver, Coloradoron (at) ronaldlewis.comwww.ronaldlewis.com www.riverscapecorp.comwww.voipcentral247.comwww.myspace.com/wysiwyg79Fwd: 520656Gizmo: 747-630-2217GoogleTalk: ronaldl79 (at) gmail.comMSN: ron (at) ronaldlewis.comYahoo: ronaldl79 (at) yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
