search for: transnexus

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

...g calls. You cannot self-sign, you cannot get > around it, the calls will either go to straight to voicemail or fail. Even > worse, the carries wil play a fake voicemail and charge you a fee, > something that some already a are doing when they detect robocallig. Don't even think about Transnexus, because they use 302 Redirect with a header, and no version of Asterisk supports it. I am the only game in the world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is literally true. If you need to sign your calls to get through, with Asterisk, you need to connect to my service. I...

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

...lls. You cannot self-sign, you cannot get around it, the calls will either go to straight to voicemail or fail. Even worse, the carries wil play a fake voicemail and charge you a fee, something that some already a are doing when they detect robocallig. > > > > Don't even think about Transnexus, because they use 302 Redirect with a header, and no version of Asterisk supports it. I am the only game in the world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is literally true. If you need to sign your calls to get through, with Asterisk, you need to connect to my service....

A few weeks... like in a year and a few weeks: https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/ Some interesting bits in there as well, like: "These rules do not apply to providers that lack control of the network infrastructure necessary to implement STIR/SHAKEN." See also: https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN *J...

...inating calls. You cannot self-sign, you cannot get around it, the calls will either go to straight to voicemail or fail. Even worse, the carries wil play a fake voicemail and charge you a fee, something that some already a are doing when they detect robocallig. > > Don't even think about Transnexus, because they use 302 Redirect with a header, and no version of Asterisk supports it. I am the only game in the world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is literally true. If you need to sign your calls to get through, with Asterisk, you need to connect to my service....

Mr. Lindstrom, I am trying to upgrade from ssh1 to ssh2. I am getting the following error when I start sshd: Disabling protocol version 2. Could not load host key SSHD does start, however, ssh1 is functional, but ssh2 is not. Please advise, William 770-671-1888, ext-240 -------------- next part -------------- An HTML attachment was scrubbed... URL:

...plan functions replacing old applications * Significant deadlock and performance upgrades for the Manager interface * An upgrade to the 'new' dialplan expression parser for all users * New Zaptel echo cancellers with improved performance * Support for the latest OSP toolkit from TransNexus * Support user-controlled volume adjustment in MeetMe application * More dialplan applications now return status variables instead of priority jumping * Much more powerful ENUM support in the dialplan * SIP domain support for authentication and virtual hosting * Many PRI protocol up...

...OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet. If you haven't registered yet please do so ASAP so we can make sure to reserve you a room! Thanks, Brian West http://www.cluecon.com _______...

...OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet. If you haven't registered yet please do so ASAP so we can make sure to reserve you a room! Thanks, Brian West http://www.cluecon.com _______...

...and then, but don't expect any major progress. If you have ideas on how to get the community to help fund a major overhaul like this, please send me e-mail off list. To find out more about Codename Pineapple, please visit http://www.codename-pineapple.org A big Thank You to Voop, Nuvio, TransNexus and Peter Gradwell for your support! Best regards, /Olle

Can someone guide me on what package I can use to do least cost routing in asterisk-1.4 without going through the prepaid calling card platforms. I have tried Asterisk::LCR and LCDial without success, if more help on either too. I will be glad. I will be glad for good pointers. Thanks. Goksie

hello everyone. i am concerned about security to the PBX and i would like to discuss different fraud detection methods. Apart from making everything to secure the PBX (latest patches, iptables, firewalls, no outside users, strongs passwds,...) i would like to find out if there are any fraud detection techniques. As for my setup i do have a PBX running asterisk 11.4 and it has 3 sip trunks (over