similar to: SER and NAT

Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support) Both signalling and media channels are kept in the path of SER+rtpproxy and ASTERISK . I can

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Using SUSE Linux, Windows 32 bit and Windows 64 bit R 3.0.2 , I am unable to use R CMD check successfully. Here is the Windows 64 bit report: Z:\R\source\effects>R CMD check pkg * using log directory 'Z:/R/source/effects/pkg.Rcheck' * using R version 3.0.2 (2013-09-25) * using platform: x86_64-w64-mingw32 (64-bit) * using session charset: ISO8859-1 * checking for file

Hi folks. I am new to the forums and have only minimal experience with wine. I am trying to convert completely to Ubuntu but have been unable to get a few programs to work. One is a program for downloading audio books from my local library. It is an .msi file. I tried installing it using both the "wine msiexec /i program.msi" command and the "wine start program.msi"

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

https://bugzilla.samba.org/show_bug.cgi?id=10668 Summary: Remote rsync daemon still showing deleted file Product: rsync Version: 3.1.1 Platform: x86 OS/Version: Mac OS X Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: ebsanford at

Hi guys, I am new to QEMU-KVM, libvmi and libvirt stuff. Libvmi uses libvirt. I am trying to to run process-list example of libvmi and getting error as below. It seems that this error may be due to libvirt as it is not able to find domain. I seek your kind help on below error: spanhal1@seclab2:~/KVMModule/libvmi-0.10.1$ sudo ./examples/process-list ubuntu libvir: QEMU error : Domain not found:

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?

Hello, I am new to R, and I am trying to figure out how to use it for a random effects model. I am using version 2.4.0, and I also have the book Applied Linear Regression by Sanford Weisberg. I have four variables: Swimmer, Sex, Swim, and Difference. Swimmer identifies the number assigned to a particular person. Sex is male/female. Swim identifies the number swim from 1 to 6. Difference

I would like to write a function that computes Tukey's 1 df for nonadditivity. Here is a simplified version of the function I'd like to write: (m is an object created by lm): tukey.test <- function(m) { m1 <- update(m, ~.+I(predict(m)^2)) summary(m1)$coef } The t-test for the added variable is Tukey's test. This won't work: data(BOD) m1 <- lm(demand~Time,BOD)

I found that the documentation for -L and -R was hard to understand. So I made some changes to try to make it clearer. I started with Revision 1.328 from http://www.openbsd.org/cgi-bin/cvsweb/src/usr.bin/ssh/ssh.1 Comments welcome. ================ ssh.1.patch ================ --- ssh.1 2012/09/15 16:08:48 1.1 +++ ssh.1 2012/09/15 20:23:35 @@ -51,13 +51,13 @@ .Op Fl F Ar configfile .Op Fl I

Hi Bjoern, CCing cfg-dev (since that's where the conversation started) and llvm-dev (since it's relevant there). Do you know if there is a way to obtain the fully resolved obj-code? I > wanted to load the functions into a shared memory, but how? The only thing > I receive is a function pointer, but I don't know how large the function > 'behind' is. Even a call to

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

Full_Name: Sanford Weisberg Version: 1.9.1 OS: Win XP Submission from: (NULL) (160.94.148.2) > wm <- read.table(url("http://www.stat.umn.edu/~sandy/wmdata0.txt"), header=TRUE) > mean(wm$Spd1) [1] 7.7773 > summary(wm$Spd1) Min. 1st Qu. Median Mean 3rd Qu. Max. 0.222 4.780 7.550 7.780 10.200 21.600 The mean of this variable DOES NOT ROUND to the value

Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -------------- next

Can anyone explain what I'm doing wrong here: > fred <- data.frame() > class(fred) [1] "data.frame" > test.fn <- function(x,class=class(x)) {class} > test.fn(fred) Error in test.fn(fred) : promise already under evaluation: recursive default argument reference or earlier problems? R 2.5.1 on both Windows and SUSE Linux. -- Sanford Weisberg, sandy at

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven