similar to: Softphone lag

Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. 2. As an extension to the above, while on call, can the agent request a conference from another agent and later hang him up. 3. Is there any way for a call to be put in the queue destined for a specific agent

Although I see similar, but more complex, questions addressed in the help archive, I'm having trouble adding superscripted text to the y-axis labels of some figures, and I can't find anything in the R documentation on this. I have: ylab="BA (m2/ha)" but I want the "2" to be superscripted. Thanks in advence for the help, or for pointing out the appropriate help file.

Hi all, After a forced reboot of the host machine... my VM lost the eth0 interface... ifconfig eth0 device not found Please help Thanks in advence Yassine _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

i have the following constructed and running very well,, thanks to Gabor Grothendieck for his help. >data.info <- c("station.id", "year", "date", "max.discharge") > > for(i in 1:num.files) { + station.id <- substring(data[i], 1,8) + DF <- read.table(data[i], sep=",", blank.lines.skip = TRUE) + z <- zoo(DF[,4],

I have a samba 3.0.2 working as a PDC with ldap as a authentication backend, how to setup logon hours for users? thanks in advence luk

Hi, I have a B410P card with bri_cpe signalling and two Openvox analog card (A1200p, A800P) with fxo_ks signalling. From the ISDN we have Point-Point 10 connection with a 10 public phone number range. If I receive a public call, the asterisk recevies the last two digit from this range, so it works, I can receive all the 10 numbers. If I'd like to dial from an exten which I have to

Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication (PBX is obviously Asterisk :-) ) concurrently (alaw codec being used)? What problems can be expected in such scenario? Would a good 1

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

On Tue, September 4, 2012 21:47, Michel Donais wrote: > I'm too settig up Hylafax and found ajhfc to be more user friendly > than all > the other cited fax frontend in the Hylafax repository. > > Hylafax-Server 6.0.5-1rhel5.i386.rpm > Hylafax-Client-6.0.5-1rhel5.i386.rpm > > Centos 5.8 > > I downloaded and tried to install but failed >

right now im writing a weblog as my first project in rails. Most tings seems to be working great :), but there is (a least) one thing left i can''t figure out how to do. As most weblogs i would like to have an archive, whick looks like this: January 2006 (42) December 2005 (60) November 2005 (2) October 2005 (101) September 2005 (4) August 2005 (2) July 2005 (101010) where (x) is the

We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested

hello everybody. I am trying to connect to a windows 2003 server in a domain environment (active directory) from a samba box. The samba server successfully joined the domain, but when I try even to list the shares it gives me a sad tree connect failed: NT_STATUS_ACCESS_DENIED I try the same command from another linux box which isn't part of the domain, and everything works. Password and

After doing the updates, my yum and my up2date both tell me the system is totally updated, and it is ... however, my rhn-applet says there are 781 updates ... and it lists as updates packages with exactly the same version ... and even some packages with older versions. Any Ideas? The rhn-applet is: rhn-applet-2.1.4-3.centos.1 Thanks, Johnny Hughes -------------- next part -------------- An

Hi, to all!!! I can't download asterisk-addons...I try with CVS, but i can't. How can I do??? Thank you Fabio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040520/632bcf37/attachment.htm

Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? I'm thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific experiences. Would anyone care to enlighten me? Off list is fine if this is a repeat, thanks very much. Matt -------------- next part

Using SAmba 3.0.2 and Cups 1.1.17. How does one configure a W98 box to print to a samba shared printer? Per my previous postings I can't figure it out, and the methods I've used on older Sambas don't work. I've been thru the lengthy discussion in the samba with cups documentation, which tells how to do it for NT or later, but not 9x. I've posted the smb.conf before,

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Occasionally we get a user who decided they need to pop their account 3 to 4 times a second... usually I just disable their account and shout "no!", but now I have someone trying to do it repeatedly with an invalid username so I don't know who to spank. =) Before if this happened I would block the IP and wait for someone to complain... dovecot doesn't make that very