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Hi All, Ntegrated Solutions is looking for a full-time Asterisk/Telecom Tech and a .net/php developer. http://www.ntegratedsolutions.com/careers/ Forward resume' to jobs at ntegrated.com Thanks. JR -- JR Richardson Engineering for the Masses -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-technician http://www.ntegrated.net/resources/job-opportunities/network-engineering-voice These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses

Greetings, Please review the job descriptions below and if you or any of your associates are interested please submit your resume. This is a full-time position in San Jose, CA or Portland, OR. Senior LLVM Compiler Engineer Responsibilities: - Develop specific code generation and optimizations in the LLVM-based compiler back-end - Implement new programming models for new cores and

Greetings, Please review the job descriptions below and if you or any of your associates are interested please submit your resume. This is a full-time position in San Jose, CA or Portland, OR or Eindhoven, Netherlands. LLVM Compiler Engineer Responsibilities: - Develop specific code generation and optimizations in the LLVM-based compiler back-end - Implement new programming models

Hi, Please forward your resumes to careers-/d9Kxk878xVBDgjK7y7TUQ@public.gmane.org to work on a India-United States (onsite + offsite) RoR development project on Motorola Rhomobile. Preference to H1B holders. If you have experience on Ruby on Rails, but not on RhoStudio, feel free to send your CV along as it won''t take much time to adjust to APIs. Regards, Shantanu Pandey CEO

We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

*Do FiveThirtyEight, priors and random forests mean more to you than a time of day, entries in a rap sheet, and pristine flora? Read on if that was a yes.* CARD.COM is a well-funded financial startup in Santa Monica, CA. We issue Visa debit cards that connect brands with their fans and enable our customers to make a social statement with their payment. Data and scientific insights are at the core

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

Is this with Asterisk in the RTP stream? Is it doing any transcoding? > -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Tuesday, April 18, 2006 9:34 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent > ChannelsWorking Nicely > > > Hi All, > > This is a performance

OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki: ATTENTION: Make sure you take a look at bug report 7144 Just do what Kevin said, include the regcontext in whatever static context you have the priority 2 extension and don't make a static regcontext in extension.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson

On 1/28/08, JR Richardson <jmr.richardson at gmail.com> wrote: > > You need to take a step back and first test the script without using > > MRTG. Execute it like this: > > # /opt/bin/asterisk-mrtg -h localhost -u XXX -p XXXX -1 SIP -2 Zap > > 10 > > 10 > > 10 > > 10 > > > > You should get 4 lines of numbers. That respresents your SIP

Hello, BarZ Adventures is an Austin-based start-up company that creates location-aware tour technology. We are looking for a software developer to work on our Ruby-on-Rails web application. The application also serves mobile clients such as iPhone, Android through web services. We have built a platform for creating location-based applications. It will be a part-time contract position with

Hi, I have a function in R that compares two very large strings for about 1 million records. The strings are very large URLs like:- http://query.nytimes.com/gst/sitesearch_selector.html?query=US+Visa+Laws&type=nyt&x=25&y=8. .. or of larger lengths. The data-frame looks like:- id url 1

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hi All, Strange issue, recently I started getting a lot of .lock files in the voicemail /INBOX folder preventing proper access to voicemail. I can delete the .lock files and everything is normal. After searching around, I found some SIP lock file stuff but nothing specific to voicemail. Can someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge.