similar to: Monitor and * 1.2.3: Sync issues?

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 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

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

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'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

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:

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:

we apologize if you receive multiple copies of this CfP ================================================================= CALL FOR PAPERS 8th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '13) as part of SC 13, Denver, Colorado | sponsored by ACM sighpc ================================================================= Date: November 22, 2013 Workshop URL:

we apologize if you receive multiple copies of this CfP ================================================================= CALL FOR PAPERS 8th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '13) as part of SC 13, Denver, Colorado | sponsored by ACM sighpc ================================================================= Date: November 22, 2013 Workshop URL:

we apologize if you receive multiple copies of this CfP ================================================================= CALL FOR PAPERS 8th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '13) as part of SC 13, Denver, Colorado | sponsored by ACM sighpc ================================================================= Date: November 22, 2013 Workshop URL:

we apologize if you receive multiple copies of this CfP ================================================================= CALL FOR PAPERS 8th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '13) held in conjunction with SC 13, Denver, Colorado | in cooperation with ACM SIGHPC ================================================================= Date: November 22, 2013

we apologize if you receive multiple copies of this CfP ================================================================= CALL FOR PAPERS 8th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '13) held in conjunction with SC 13, Denver, Colorado | in cooperation with ACM SIGHPC ================================================================= Date: November 22, 2013

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Has anyone implemented HOBIC SMDR output from *? Can someone point me to the Bell HOBIC specification? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

I finally got my ASUSCOM (Cologne chip) ISDNLink card working here in the US. When a call arrives, I get "Unknown IE 42 (Unknown Information Element)" and "Unknown IE 21 (Unknown Information Element)". IE 21 (0x15) is defined as Q931_CALL_STATE_SUSPEND_REQUEST. IE 42 (0x2a) is not defined in the code and I couldn't google it. Is this something perculiar with ISDN in

I'm having an intermittent problem with fax transmissions, something I call the streaking mascara syndrome. TxFax will be part way through a transmission when something fails--on the fax printer, part of the page will have correctly printed, but the rest of the page is "streaked" to the bottom. I have several successful multi-page transmissions interspersed with streaked

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Dear Samba Team member, I?m afraid I am not a fluent Samba user and but desperately need to upgrade the version we have. Do you know where I can find Upgrading instructions for Samba or do I need to reinstall the whole program with the latest one? I haven?t been able to find upgrading instructions anywhere, which is why I'm bothering you for assistance. If you have any recommendations or know

I need to test QoS on an IAX link between a server in Colorado and a server in Europe. I know I could install a Milliwatt extension on the European server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of "Page Not Found" on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters