similar to: Out of trunk data space on call number 16386, dropping

Hi All. Currently I am trying to change a logical volume from linear to stripped because I would like to have a better write throughput. I would like to perform this change "live" without stopping access to this lv. I have found two interesting examples: http://community.hpe.com/t5/System-Administration/Need-to-move-the-data-from-Linear-LV-to-stripped-LV-on-RHEL-5-7/td-p/6134323

Hello! I'm working on a new back-end and have hit a bit of a snag. I'm working on getting selectcc working and have followed the MSP430 model of emitting a custom CMP and SELECT_CC node and matching that with a pseudo-instruction that has useCustomEmitter=1. However, my output ends up very wrong, despite the Machine code being initially correct: # Machine code for function func: Function

Ok so it seg faults when I try to dial out through IAX(voiptalk.org), ofcourse it doesn't if I remove allow=speex :P ---- (gdb) run -c Starting program: /usr/sbin/asterisk -c [Thread debugging using libthread_db enabled] [New Thread 16384 (LWP 28283)] [New Thread 32769 (LWP 28285)] [New Thread 16386 (LWP 28286)] [Thread 16386 (LWP 28286) exited] [New Thread 32771 (LWP 28287)] Asterisk

Hi, My storage admin just assigned a Lun (fibre) to my server. Then re scanned using echo "1" > /sys/class/fc_host/host5/issue_lip echo "1" > /sys/class/fc_host/host6/issue_lip I can see the scsi device using dmesg But mpath device are not created for this LUN Pleas see below. The last 4 should be active and I think this is the problem Kernel:

Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. Thanks lach Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 09:00:07

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

I have an installation that connects in a [very] good day at 22kbps, but the normal is about 18kbps. I use de ILBC codec, and also change in iax.conf the trunkfreq = 20 to trunkfreq = 30 It works, you can understand well the other person, but don't expect miracles or an outstanding sound quality. > Dear Dan; > > Thanks alot for your kindly reply. > > Well, what u advise us

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hi, I need help finding why my system is dropping calls.. I enabled debugging on my box in the hope it would lead me to the answer as to why my system is dropping calls but unfortunately nothing is jumping out at me.. I have attached the portion of the messages file for two calls that were dropped.. (numbers names and ip's have been changes to protect the innocent) Can someone give me a

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

Hello, I want to reduce the bandwidth taken by an IAX trunk when used with a small number of voice channels. When only one call is passed through an IAX trunk, the IP overhead is indecent. I would like to increase the IAX voice packet emission interval (20ms - i'm using speex) to something much larger, in order for the packets to transport as much data as possible. I've been

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <div class="moz-text-flowed" style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi, <br> <br> I try to find a solution for this problem : <br>

SYNOPSIS Erratic red alarm T1 PRI on asterisk, but zttool running concurrently during alarm shows no errors, irq misses, or alarms, on any span. Using asterisk and quad Digium T405P, configured as follows: Span 1 connects to ISDN PRI (fractional 8 B channels, D channel 24). Span 2 connects to T1 Mux and analog stations. Span 3 connects to ISDN PRI Nortel BCM hybrid key system digital trunk.

[redirected from -hackers to -security] Jakub Lach <jakub_lach@mailplus.pl> writes: > http://marc.info/?l=openbsd-tech&m=129236621626462&w=2 http://maycontaintracesofbolts.blogspot.com/2010/12/openbsd-ipsec-backdoor-allegations.html DES -- Dag-Erling Sm?rgrav - des@des.no

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476