similar to: newbie needs advice

When I try to copy a directory of files from Windows XP to my samba server, I get additional files on the samba server that look like Windows registry class IDs. Here is a sample: -rw-r--r-- 1 jfayre jfayre 0 Mar 17 06:31 IMG_0470.JPG:{4c8cc155-6c1e-11d1-8e41-00c04fb9386d}:$DATA -rw-r--r-- 1 jfayre jfayre 461574 Mar 24 07:15 IMG_0478.JPG -rw-r--r-- 1 jfayre jfayre

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

Miguel, How many simultaneous calls (incoming and outgoing) you did ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

I've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Several things there are very concerning to me, and I'm interested in what other people here think of them. * The ToS specifically forbids use for any call relating to medical, financial, or government matters -- as well as any

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Somebody know if they are down? Let me know, Julius C. Barber ventas@gringotel.com www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/924605b6/attachment.htm

Steve, Do you know if there are many differences between the Mexican and Brazilian variant ? Kind regards, Miguel Antonio Date: Tue, 05 Oct 2004 22:51:53 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] RES: Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

No Dice so far, anyone now how to add anIAX trunk? What are the settings exactly? I have added everything but I do not know what are the registration strings? Jonathan

Hi all, I'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

Hi, Does anyone know what is going on with voipjet? This morning/afternoon they just seem to have gone down.... no word on their website.....

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is

Hi, Has anyone experienced intermittent echo issues with voipjet lately?