similar to: Colorado Asterisk Users

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

There was some talk a couple of weeks ago about voip devices w/ ethernet and fxo/fxs configurations. I found this while out and about and thought it might be interesting. Configurable fxo/fxs with dual ethernet. http://www.tekdigitel.com/website/htmlPages/content/products/product_introductions/introduction_to_V-SERVER_iGATE_Dual_Ethernet.htm -- Karl Putland <karl at

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

For those of you that have dlink VoIP hardware, I thought I'd post this link. It's a whole lot easier that navigating their support site trying to find anything. There is updated firmware for most of their products. ftp://ftp.dlink.com/VoIP/ A word of warning about the DG_104S firmware. It's better than the firmware that came with my box, but it still has quirks. Some settings

Is this a known problem? Niels. ----- Forwarded message from Judah Levine <jlevine at utcnist.colorado.edu> ----- Date: Fri, 5 Jul 2002 08:58:46 -0600 (MDT) From: Judah Levine <jlevine at utcnist.colorado.edu> To: provos at citi.umich.edu Subject: Privilege separation Hello, I have just installed openssh-3.4p1 on a COMPAQ/DEC/HP Alpha running True64 UNIX v4.0F. The privilege

David Geary will be speaking at the Pikes Peak Java Developers group meeting, Tuesday, November 8. The meeting starts at 6:00pm. So if you live in the Colorado Springs area, come out and help support rails! For the detailed agenda, visit, http://www.ppjdg.org Also, if you are interested in starting a Ruby Group in Colorado Springs, please send me an email and let me know. -- Joe Clarke

ok all. I have sent the PHP web gui code to Mark at Digium but have not heard back yet. I dont know that status if he wants to CVS it or not. maybe if he does not answer in a this next week ill just upload it to a website for all to DL. we could add to this as a base and get it to work better for all again the current URL for it is http://rads.netcom.utah.edu/openconf/openconf.php Dave P

Details below. Please contact me for more info. Thanks --Duncan Job Details: Release Automation Engineer CyberSource provides businesses with services and software designed to automate the processing of web, phone and other customer-not-present credit card and check transactions. Our complete suite of hosted transaction services and software products includes real time fraud screening,

Hello everyone, I have contacted providers listed on the Wiki and other lists of providers offering SIP or IAX service... So far I'm having trouble finding an IAX provider offering the 303 or 720 area codes. Can anyone suggest a IAX provider that offers these areas codes? I would prefer to use IAX due to NAT but I am willing to work through a solution for SIP if it's available to me.

Do you want the best performance from your Lustre implementation? Lustre Administration & Support training will be offered June 5-7 in Boulder, Colorado. Attendance is limited! Click here to register Today! http://www.acteva.com/booking.cfm?bevaid=132736 Cluster File Systems offers Lustre? Training programs that provide the all the knowledge required to start building high performance

Hi, I havn't actually tried this yet, but would it be possible to use asterisk as a mgcp <-> h.323 translator? For example, I have mgcp service from Next Gen telephone company. But i only have a h.323 phone. Would there be a way to the mgcp signalling to hit asterisk, and then have it fire the call out h.323? And vice versa? Just brainstorming. Sean Watkins

Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? Ta SJ

Hi, I'm a new asterisk user developing an AGI application. As part of my application I'd like to play sounds on the server's speakers, but it seems that I can't do this while asterisk is running. When I try to play sounds using the play or aplay command, it blocks until I stop asterisk. My guess is that asterisk is using the sound device and this means that other programs

Bill I have adtran750 with Vista/Aastra 350, 390, & 480's never seen that one tim -----Original Message----- From: asterisk at billheckel.com <asterisk at billheckel.com> To: asterisk-users at lists.digium.com <asterisk-users at lists.digium.com> Date: February 21, 2003 12:52 PM Subject: [Asterisk-Users] Fun new problem with Aastra 390 ADSI phones >I seem to have

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do?

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

Hi there.. Compiling Asterisk (CVS from 10.05.04) with gcc 3.3.2 works fine. But I have to compile the Kernel with GCC 3.4. Trying to compile Asterisk with gcc 3.4 will result in a compiler-crash. Any idea? Yours, Karl

Hi Karl Im running for years with samba now with bind_dlz and on stock debian samba and bind runs fine. I went through your mails on the list and i noticed the following. > client006\$\@MY.DOMAIN.DE: updating zone 'MY.DOMAIN.DE/NONE': update > failed: rejected by secure update (REFUSED) > samba_dlz: disallowing update of signer=client006\$\@MY.DOMAIN.DE >