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Hi all! I am from the Windows world and am trying to migrate to Linux and have done a fairly good job so far. My recent challenge is that I have built a Samba file/print server that works very well on my internal network but when I VPN into the network remotely I am unable to access the server via it's server name. What is driving me crazy is the fact that the last of my Windows

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

To ANYONE that can help :), we where using samba version 2.0.7 and everything worked great. we just upgraded to version 2.2.7a. and now we get this weird/wicked problem. sometimes when a user logs out and then trys to log back in it says "//servername/.ntprofile/share/file" cannot be copied over. i can delete the file and then another file cant be copied over. i at times cant delete

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, Overall, everything works fine, and all the emails are delivered without anything messing up. But then, from time to time, the transaction logs for some users become corrupted, meaning emails cannot be read at all from dovecot. With a bit of a search on google, it seems to be a problem with locking on the transaction log. The individual files are fine - it's maildir, so the chances

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

I try to setup SJphone on my PDA, but it does not register with Asterisk. I have setup a sip account on asterisk, ... Can anybody give me a hint? bye Ronald

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

I need to install Puppy linux as a guest. Usually I boot from the Puppy CD and have so a life system which I can than install to hard disk. How do I do that as a Xen guest? bye Ronald _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

http://bugs.freedesktop.org/show_bug.cgi?id=23223 Summary: Regression: Nvidia Geforce 7300 (NV4B) fails to suspend Product: xorg Version: 7.2 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

https://bugs.freedesktop.org/show_bug.cgi?id=56757 Priority: medium Bug ID: 56757 Assignee: nouveau at lists.freedesktop.org Summary: [regression][bisected][screenshots] nvfx -> nv30 driver rework causes regression in UrbanTerror4.2 (beta5) Severity: normal Classification: Unclassified

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea