[asterisk-users] [OT, Job] Senior Software Engineer for exciting, high-growth startup
So we are hiring a senior-level developer with VoIP and preferably sounds and speech processing experience. Please check http://www.smartaction.com/company/careers and the company web site in general. Out solutions are based on Asterisk and Windows servers hosting customer service AI-based virtual agents. See the job requirements below. -kkm Senior Software Engineer for exciting, high-growth startup – SWE-A21 SmartAction has developed a hosted IVR (Interactive Voice Response) service driven by our Artificial General Intelligence technology. We have spent years of research, coding and experimentation on our core AI technology, and we believe that our Smart Call Agents represent a new generation in self-service applications. We are currently seeking a senior software engineer to add to our team. We are most interested in your strength as a programmer, your ability to learn quickly, and your motivation. Key skills and requirements include: * Expert level in VoIP architecture * Deep understanding of SIP, RTP, and related networking protocols. * Experience configuring and maintaining Asterisk deployments with SIP * 5+ years experience with strong software development skills * Experience with .NET * C++, Java or C# * Good communication skills, and ability to work fast alone or in groups * Eager to learn new technologies * Comfortable working in a Microsoft shop Bonus points for: * C# * SQL * Dealing with customers in integration projects * Nuance recognizer and grammars * Bug tracking software (Trac) * Desire to earn equity in the company We are an extremely tightly-knit group; you must be in or willing to relocate to Los Angeles, CA at your own expense. We are generally unable to assist with visa requirements for non-residents (except Canadians). To apply: please email your resume to j...@smartaction.com. Put the keyword “SWE-A21” in the subject line. -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3: Started but no SIP talking
On Sun, Apr 17, 2011 at 1:46 PM, bilal ghayyad wrote: > Hi All; > > I am missing any thing? Should I do any thing? How can I know if my new > asterisk is running sip well? > > Regards > Bilal > Check if you've got a software firewall running, check if SELinux is running, etc. Try running a packet capture using tcpdump to see if your asterisk box is getting any traffic from the phone, etc. Basic network troubleshooting at this point. Can you ping the box from your network, can you ping the phone from your box, etc? -- Thanks, --Warren Selby, dCAP http://www.selbytech.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 1.8.3: Started but no SIP talking
Hi All; I installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet ports. I gave IP address 192.168.0.3 for one Ethernet port. I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there (in the command line) I can type a commands. I have an Polycom IP Phone that is able to register for other Asterisk boxes (and some of them is 1.8.3) but with this new server, I do not see any messages coming to the consol when I give the IP address of this new asterisk server !! What could be? Actually, in the sip.conf file, it is hearing for all IPs 0.0.0.0 and the IP phone sending on port 5060 UDP (every thing default). What could be I am missing? Even if username and password wrong, I should be able to see traffic but without registration ... By the way: on the same network, there is another Asterisk box running with IP address 192.168.0.2 .. does it effect? It should not. I am missing any thing? Should I do any thing? How can I know if my new asterisk is running sip well? Regards Bilal -- _ -- 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
Re: [asterisk-users] sip error logging
On Sun, Apr 17, 2011 at 2:24 AM, Jeremy Kister wrote: > On 4/17/2011 3:16 AM, Sherwood McGowan wrote: > >> This may sound like a stupid question, but what are your verbosity and >> debug >> levels set at currently? >> > > nope, thats exactly the type of thing i'm wondering if i'm missing :) > > but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried > with verbose 10/debug 10 before posting. no dice. > Ah right on mate! Glad to see that you checked it *and* didn't mind being asked (after all, we're all IT/VOIP professionals, and we all know the first thing to ask is the simplest possible solution ;-] ) Cheers! -- _ -- 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
Re: [asterisk-users] sip error logging
On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
Re: [asterisk-users] sip error logging
On Sat, Apr 16, 2011 at 6:05 PM, Jeremy Kister wrote: > bumping once before sending it to the tracker. > > Original Message > Subject: [asterisk-users] sip error logging > Date: Fri, 15 Apr 2011 03:39:23 -0400 > > > I recently noticed that asterisk is not logging unknown sip connections. > I'm not sure if I've broken something or if asterisk itself has been > broken. > > the last entry I have is: > /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: > Registration from '' failed for '10.0.0.228:5060' - > No matching peer found > > > my logger.conf looks like: > # grep -v '^;' /etc/asterisk/logger.conf > [general] > [logfiles] > console => notice,warning,error,dtmf > messages => notice,warning,error,verbose,dtmf,fax > > if i send 'options' or 'register' from a non-configured sip peer, i dont > see anything in the log. am I missing something ? > > * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 > > -- > > Jeremy Kister > http://jeremy.kister.net./ > > This may sound like a stupid question, but what are your verbosity and debug levels set at currently? Sherwood McGowan -- _ -- 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
