similar to: Avoiding deadlock

Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this?

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hi all If I have the following in my dialplan: exten=>25001,hint,SIP/25001 Doing a core show hint 25001 results in 25001 at local : SIP/25001 State:Idle Watchers 0 1 hint matching extension 25001 in the Asterisk CLI. What does the Watchers 0 mean? I use the hints table output via core show hints for logic in my dialler application - but

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

Hi, Any of you have any experience with SIP softphone dialler that capable of local recording? (recording to files in harddrive) So far I only know eyeBeam and Express talk. eyebeam fine but there are known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *. Cheers Benni-

This is probably the most eternal question concerning rsync, but I'm really in doubt since I can't find the appropriate answer or solution anywhere for many months. I'm using rsync for simple daily back up data from one HD to another. It takes about 10 minutes daily under "nice -n 19". The problem is well-known as I see - several other services such as

We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing "hello ? Hello ?" and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

https://bugzilla.samba.org/show_bug.cgi?id=3491 Summary: throttle disk IO during filelist/directory parsing Product: rsync Version: 2.6.4 Platform: All URL: http://vilius.multiply.com/video/item/10 OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core

I have a samba server acting as a domain controller. Is there a way that I can Have a script that delete the shortcuts on the desktop,quicklaunch and startmenu for Internet Exploder. At the same time installing Mozilla Fire Fox. Maybe like a little vbscript or something that gets ran from the server when they login. Thanks

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

On Sat, Jan 7, 2017 at 2:30 PM, Peter Moody <mindrot at hda3.com> wrote: > so I spent a bit of time looking at this and it seems like the only > way to go, at least if I want to keep it in ssh_connect_direct(), is > to use pthreads. further, it seems like getting that accepted is > something of a long shot: Sorry, pthreads is a non-starter. I would have thought that using

The Asterisk Development Team has announced the release of Asterisk 10.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello, I know that in Windows Server there is functionality to specify Full domain name (for ex. "company.com") and NetBIOS domain name (for ex. "company"). Does samba provide such functionality and if so how? I haven't found anything on manual. -- Best regards, Vilius

The Asterisk Development Team has announced the release of Asterisk 11.0.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.0.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

Hello, We are considering to migrate from Novell eDirectory to Apple Open Directory + Windows Services from Apple as Domain Controller. I've read a lot of books and manuals and forums about Apple Directory Services, Domain Controllers and Samba, but there are lots of unanswered questions still. Mainly about naming and planning of different things on the network. I also tried to post to Apple

The Asterisk Development Team has announced the release of Asterisk 10.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello Sir, I have few clarifications as we are planning to work with asterisk. If you don't mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and it's usage). Q2. From the list of features listed in www.asterisk.org , I see "Predictive dialler" is listed

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira