similar to: Maybe a NAT problem

Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the

Hi list: First post, hope find a solution.... This is my problem, i've got some chm (Microsoft html help files) and i can't see them from my host clients (windows) i can open them but i cant see them... this is the error log from samba: facu (192.168.0.48) couldn't find service e-books - oreilly & cisco smbd/service.c:make_connection(798) Any ideas? Many thanks.

Hi: Is there an easy way to add a HP 5si printer using a Jetdirect interface, to share with my lan?? Maybe some config help?? Thanks a lot -- Facundo Agustin Barrera IT Management. Buenos Aires - Argentina.

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me

Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works

Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: > Channel Zap/1-1 was answered. -- Executing

Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal.

Hi all there. I am trying to set up an icecast2 server (wich is already running) to send some stream mount point to another server. I mean, my server is streaming at /radio1.mp3 mount point. We can listen it at http://myserver:8000/radio1.mp3; so, i want to know if is there any way to "clone" that strem and send it to another server, such as http://giss.tv:8000/radio1.mp3 Thanks. --

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

Hi all there. I am trying to set up an icecast2 server (wich is already running) to send some stream mount point to another server. I mean, my server is streaming at /radio1.mp3 mount point. We can listen it at http://myserver:8000/radio1.mp3; so, i want to know if is there any way to "clone" that strem and send it to another server, such as http://giss.tv:8000/radio1.mp3 I have been

Hello everone, I am a computer science researcher in the UK - and have a quick question regarding importing data into R and more specifically latentnet. I am trying to import data such as a text file containing: (This is based on the Sampson data set) ----------- Network attributes: vertices = 3 directed = TRUE hyper = FALSE loops = FALSE multiple = FALSE total edges=

Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of

Hello everone, In my Company we are going through a network redesign and Planning to retire our Novel edirectory, and Novel Servers and replace them with Samba4 (Over 150 Servers). We have setup a Samba4 test environment which seems to be working well so far. We are an organization with multiple locations and over 1200 users, we are also very heavy users of google apps. I have couple of questions

On 5 August 2010 02:14, Tobias Grosser <grosser at fim.uni-passau.de> wrote: > On 08/05/2010 01:45 AM, David Greene wrote: > > Do pushes to the LLVM git mirror get forwarded to the svn trunk > > automatically? For various reasons it may be more convenient > > for me to work on LLVM through git but I don't want to go through > > a lot of pain to send patches

Hello everone,            Anyone who knows how to force a cubic smoothing spline to pass through a particular point?            I found on website  someone said that we can use "cobs package" to force the spline pass through certain points or impose shape           constraints (increasing, decreasing). However,  this package is using  B-spline and can only do linear and quadratic

Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I would help) I

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

I ordered 10 7960's with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, that's not really applicable in a SIP setup? I'm guessing it's going to be a more XML

Hi everone, I currently try to test the Opus codec. It seems, however, that it is more easy to do that in Linux. Nevertheless, I have compiled the sources in Visual Studio 2010. Here are some hints: 1) "static inline int function(...)" (combination of static and inline) does not work with Visual Studio 2010. I defined a macro which translates all "inline" to nothing. 2) I