similar to: Zap Flash()

Displaying 20 results from an estimated 900 matches similar to: "Zap Flash()"

2008 Jul 16
2
How to extract component number of RMSEP in RMSEP plot
Hi R-listers, I would like to know how can i extract component no. when the RMSEP is lowest? Currently, I only plot it manually and then only feed the ncomp to the jack knife command. However, I would like to automate this step. Please let me know. Many thanks. Rgrds, [[alternative HTML version deleted]]
2011 Mar 31
3
** to disconnect and make a new call
Hi, Does anyone know how to implement the feature in asterisk calling card when a user has dialed the access number and during the IVR or any time during the call, he can press ## or ** to end the current call and dial a new destination number? Please help and give me a step by step help. Thanks. Rgrds-------------Abid -------------- next part -------------- An HTML attachment was
2004 Aug 31
4
winbind problem (?) on samba 3 ADS
Hi, I have installed samba 3.0.6 based on the "Official HOWTO" to join out Active Directory environment, with winbind and pam support. I have join the samba to the domain using "net ads join -Umyloginame". I can do the "wbinfo -g", "getent passwd" and "getent group" correctly. I also can list shares on other machine, using kerberos: # kinit
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no
2004 Sep 12
1
cannot add domain user to directory security on 3.0.6 from Windows Explorer
Hi, I'm trying to run samba 3.0.6 with --with-acl-support on redhat 9.0 with kernel 2.4.25 patched with acl, joined to a ADS w2k environment. Everything is OK, except that I can't add any user to a directory security on a share, with error "unable to save permission changes on folder EmirF_create_tis_fldr. Access is denied". Even as user "EmirF" from windows explorer,
2014 Apr 06
1
rsync --delete-hide
Hello! I was wondering wether or not this would be a viable option to add to rsync: --delete-hide Instead of deleting the file, rsync would rename the file using leading ?.? and trailing ?.deleted-{timestamp}?. rsync -rv --delete-hide /src/ /dst/ For example if /src/readme.txt had already been rsynced to /dst/readme.txt and then later was deleted from /src, rsync would - instead of deleting
2008 Jan 02
7
Two Asterisks behind NAT and need to link them using IAX trunk
Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like
2008 May 19
1
Need help in matrix multiplication error
Hi, I have a data file which is named test.txt as below. Prior to that, I have converted the last row from nominal to numeric using as.integer. Statement T1001 T1002 T1003 T1004 T1005 T100 T1014 T1021 T1022 T1023 1 0 0 0 0 0 0 0 0 1 0 0 2 0 0 0 0 0 0 0 0 1 0 0 3 0 1 0 0 1 0
2006 Nov 01
2
Two Sipura 3000s
I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?
2008 Mar 12
2
Warning: integrate_views and nested description groups
describe MyController do integrate_views describe "A common base senario" do it "no longer integrates views" do be_careful end end end integrate_views affects an attribute in the class formed by the describe factory method. The second describe generates its own class, so integrate_views is OFF at that level. I''ve already spent far, far too much
2010 Jun 25
2
Call drops on group paging asterisk - 1.4.22.1
Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much
2005 Feb 25
3
main effect & interaction in 2-way ANOVA
Hi, I am just a little confused of mian effect in the analysis of variance (ANOVA) when you include or do not include an interaction term. Let's assume a simple case of 2-way ANOVA with 2 factors A and B, each with 2 levels. If it shows that main effect for A is significant when the interaction between A and B is NOT included, and the main effect for A is NOT significant when the interaction
2006 May 01
2
Rebuilding Raid 1
Trying a different approch. Senario Raid 1 setup Bootable raid drive failed Mirror has been working for almost a month and then rebooted Now can't boot mirror drive grub not mirrored from other drive. I Fixed bootable drive. Question? Can I hook up both drives and boot fixed drive then rebuilt mirror from nonbootable drive to bootable drive? Does the raid automatically rebuilt when I
2016 Nov 16
2
Multiple location DC's with same hostnames
Hi, Not sure exactly how I would word the subject line so appologies in advanced. We are trying to accomplish the following scenario: Location 1: PDC: fs01.loc1.example.com IP: 10.0.0.1 Location 2: SDC: fs01.loc2.example.com IP: 10.0.1.1 Clearly when we join the SDC to the PDC there is a naming conflict. The end result would be to have clients at each site resolve the fs01 name to
2006 Jan 13
2
ILBC to G711 transcoding experince ?
Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already
2005 Feb 25
1
msic while ringing
I want to setup a senario in which the callers hears to some music file while the phone is ringing and as soon as the line is answered the music is stopped palying. i.e. instead of the rings the caller listens to some music. Is is possible with asterisk? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Sep 19
1
"Channels got stuck in asterisk 1.4.18.1"
Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do "soft hangup <channel>", it says "Requested for soft hangup" for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to
2009 Sep 25
1
"multiple contexts for multiple locations"
Hi All, I have a senario where we have multiple locations and all have the ability to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nxxxxxxxxx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority
2011 Apr 13
1
Asterisk thread limit
Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---------------[Asterisk]----------------[sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 100000 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r
2008 May 18
1
Bridging a call on hold with an active call
Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on