similar to: Asterisk Tech Job Posting Dallas Texas

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

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

Ntegrated Solutions in Dallas, TX is still looking for voice guy. This position is for US hire only, will not sponsor H1B work visa. http://www.ntegrated.net/careers/ Thanks. JR -- JR Richardson Engineering for the Masses -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am wondering if any of the Texas user groups have members in the North West part of the state. I am in the Amarillo area and would like to find some othere in this area, maybe even start a user group in this area. -- Bruce Reeves Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Must have event-driven processing expertise. Linux environment. Mobile apps dvlpmt. NW suburban location. Recruiter: Liz Klein 469-223-4900 adfwrecruiter-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org or lizk-IElRh2z3wdXDP7Uc+MJFAgC/G2K4zDHf@public.gmane.org www.talentlegends.com LinkedIn profile at *www.linkedin.com/in/elizabethmklein/*<http://www.linkedin.com/in/elizabethmklein/> --

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

All, Just to get the word out: We are looking for a new Statistical Consultant at the Division of Statistics and Scientific Computation here at the University of Texas at Austin. Please pass along to any colleagues who might be interested... Statistical Consultant Purpose of Position: Provide professional statistical consulting and instruction to UT faculty, staff and students. Apply knowledge

All, Just to get the word out: We are looking for a new Statistical Consultant at the Division of Statistics and Scientific Computation here at the University of Texas at Austin. Please pass along to any colleagues who might be interested... http://ssc.utexas.edu/people/employment Thanks, Michael

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, 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.

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

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

FYI Kristian. http://www.astlinux.org/ Unable to connect to database server This either means that the username and password information in your settings.php file is incorrect or we can't contact the MySQL database server. This could mean your hosting provider's database server is down. The MySQL error was: Can't connect to local MySQL server through socket