similar to: log messages...

Hello If I try to open COM in a app. I got: BaudRate 9600 fixme:comm:SetupComm insize 1024 outsize 1024 unimplemented stub BaudRate 5760 fixme:comm:SetupComm insize 131072 outsize 8192 unimplemented stub err:comm:SetCommState baudrate 5760 BaudRate 5726 fixme:comm:SetupComm insize 1024 outsize 1024 unimplemented stub err:comm:SetCommState baudrate 5726 Use newest Wine, Kernel 2.4.18-7, SuSE

Alex: Any idea why the list is still bouncing these? "noreply at rubyforge.org" is in the list of addresses that are supposed to be allowed to post even though not subscribed. What is that about "Message has implicit destination"??? Kevin -------- Original Message -------- Subject: Wxruby-development post from noreply at rubyforge.org requires approval Date: Sat, 21 Oct

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

Bugs item #6262, was opened at 2006-10-21 18:13 You can respond by visiting: http://rubyforge.org/tracker/?func=detail&atid=218&aid=6262&group_id=35 Category: Incorrect behavior Group: None Status: Open Resolution: None Priority: 3 Submitted By: Alex Fenton (brokentoy) Assigned to: Kevin Smith (qualitycode) Summary: MDIParentFrame#add_child not working on OS X Initial Comment:

https://bugzilla.samba.org/show_bug.cgi?id=6262 Summary: single-file rsync fails without further options set Product: rsync Version: 3.0.3 Platform: x86 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: anders.henke@1und1.de

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0

Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: > As a

Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"

Hi, I'm trying to compile dovecot under Solaris 10 with native Mysql support. Couple changes are needed before configure can detect and compile mysql support. Native Solaris Mysql is installed under /usr/sfw directory. Configure can't find libraries and when these are found -R runtime path is needed for linking. Also if library name is given before library path it can't find it.

Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

Hi there! I'm researching a project for my company that would involve a proprietary SSH tunnel client running on Win32. I was wondering if OpenSSH had a portable library component that I could leverage to manage the connection and tunneling functionality without any interface components...? Thanks in advance for any help or direction you can provide! ========== Robert Bates Cobra

Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten => s,1,Background(my/age) ;;Play recorded message to enter age exten => s,n,WaitExten(10) exten => _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten => s,n,NoOp(${AGE}) exten => s,n,GotoIf($[${LEN(${AGE})} >

I have a Centos 5.4 machine that has, for the past two weeks, apparently been shut off over the weekend. It's just sitting there turned off on Monday morning and when someone hits the power switch it comes right back on and everything works again. This happened last weekend, and again over this past weekend. Here is /var/log/messages from shortly before it apparently shut down this weekend.

Any ideas? Thanks. [root@localhost root]# asterisk -r Unable to connect to remote asterisk [root@localhost root]# asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Asterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Parsing

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in).

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Please forgive me, if I misunderstand the problem completely. Following instructions in several german blogs, I want to configure Asterisk with a hfc-pci card, an old NTBA and an ISDN phone as a SIP device. It seems that I have to set signalling in zapata.conf to bri_net_ptmp. When I do this, Asterisk will hang if started with -vvvvc, the last lines of output being: [res_features.so] =>