similar to: Filesharing + video + voice supported Soft phone

VMukti.com ----- Original Message ----- From: "Anselm Martin Hoffmeister" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 15:06:03 +0200 Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel: > Dear all > > I have setup of

VMukti.com ----- Original Message ----- From: "Goltsios Theodore" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 16:57:25 +0300 There is for sure X-lite and other similar but you won't get file sharing which is meaningless either way. If

I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Regards --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part

Hello, I'm having an issue with usage of libvirt driven qemu where I'm sharing directory with guest machine. virsh domain filesharing configuration: <filesystem type='mount' accessmode='passthrough'> <driver type='path' wrpolicy='immediate'/> <source dir='/media/share'/> <target dir='data'/>

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

I've been trying to set up a server to fileshare to mac and windows clients using Netatalk and Samba. Basic setup and functionality works, but we have two lingering problems, both on the windows clients: - can't change the user field - can't change the group field The windows client permissions dialog allows you to change it, but once you close and reopen it, the changes are gone.

I am having trouble getting Blubster to go from the connecting stage to the connected stage. I need help please in correcting my problem. Thank you very much.

Can anybody help with this issue: My question regarding SAMBA is: we use the file sharing capability from a Windows workstation to a UNIX machine. When we copy files from the workstation to the UNIX box, UNIX places an end of line character in the file that we don't want there. Is there a way to avoid this (a way to tell SAMBA to do a straight ASCII copy)? We don't have this problem when

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi All, I have the following section in my smb.conf file: [Satish' Hindi Alltime Hits - Gems] path = /mnt/d-drive/Music/Satish' Hindi Alltime Hits Gems browseable = yes public = yes readonly = yes guest ok = yes I am able to browse this share through win2k but not from winNT. But If I change the share

Hi, I am working of Intel 82576 device to test the SRIOV functionality. I want to assign VFs to a VMs(domU) and verify the control and data path. I installed Xen-unstable with 2.6.30-rc3(dom0). But the SRIOV functionality isn''t working, because MSI is not supported. Please let me know, what version of Xen and dom0 supports above functionality? Is there any tool available to manage SRIOV

first sorry my english Is posible when reinstalling a server ( changing distro too) , make account machine whith out rejoined all win2k/xp to de domain ?? i create before to login the account machine , but win say can't login account machine does exist I try changin the sid to old , but no work Por si hay gente que hable espa?ol en la lista!!!! Es posible luego de reinstalar un server

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

Hi: Suppose that I have a data frame as below x1 x2 x3 ... x10 wk1 wk2 ... Wk208 (these are the column names) For each record, x1, x2, x3 ... x10 are attributes. and wk1, wk2, ..., wk208 are the sales recoreded for this attribute combination. Suppose that now, that I want to do the following 1. Merge the data frame so that I have a new data frame grouped by values of x2 and x3 (for example).