similar to: NAT solutions

Displaying 20 results from an estimated 300 matches similar to: "NAT solutions"

2012 Feb 16
2
help with e+01 number abbreviations
Dear List, I will appreciate any advice regarding how to convert the following numbers [I got in return by taxondive()] in numeric integers without the e.g. 6.4836e+01 abbreviations. Thank you very much in advance, Gian > taxa_dive Species Delta Delta* Lambda+ Delta+ S Delta+ Nat1 5.0000e+00 6.4836e+01 9.5412e+01 6.7753e+02 8.7398e+01 436.99 Nat2
2008 Dec 17
1
Asterisk and NAT one way audio
Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone
2010 May 11
3
Improving loop performance
R-users, I have the following piece of code which I am trying to run on a dataframe (aga2) with about a half million records.  While the code works, it is extremely slow.  I've read some of the help archives indicating that I should allocate space to the p1 and ags1 vectors, which I have done, but this doesn't seem to improve speed much.  Would anyone be able to provide me with advice on
2003 Sep 08
5
Help needed with IAX behind NAT
Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has
2003 May 07
2
SIPPROXD for SIP thru NAT
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2003 Apr 06
5
SIP Testing
We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation
2003 Oct 15
1
SER vs STUND with Asterisk..
One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to
2005 Oct 16
2
Looking for advanced consultant services
Hi, I have a meeting with an important customer in a couple of days and I am aware that most of their questions are going to be related about scability of Asterisk. We want to propose this customer to integrate Asterisk with SER, but I have a loot of complex doubts that I would like to known before this meeting. I would like to contact with a busines that has experience with large
2004 Nov 01
1
SSHD with PAM question
Let me refine my previous question: > Can anyone provide some insight into why the auth-pam module uses a fork in > pthread_create (auth-pam.c)? This completely breaks the ability of one PAM function > to pass data to others via the pam_set_data/pam_get_data functions. Can anyone tell me how to #define USE_POSIX_THREADS when building --with-pam defined? The autoconfig stuff
2005 Aug 11
1
MS Live Communication Server
Hi List! does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heared about that, please let me know. The fact i figured out is that the Border Controler from Jasomi can be used as a gateway from MS-LCS-SIP to regular SIP. But that is not really handy and expensive too. Thank you
2004 Oct 06
10
Asterisk and SIP phones
I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP
2004 Sep 22
2
SSHD with PAM question
Greetings All, I am trying to get sshd to authenticate using PAM in a situation where there is no password entry (as found by getpwent et. al.) for a user. Setting: AllowUsers * UsePAM yes causes the right PAM stuff to be invoked, but as soon as the PAM module tries to have a conversation with the (illegal) user (in order to get the password) sshd throws out the authentication context.
2004 Dec 14
1
Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?
Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a
2003 Sep 25
1
Help me to understand BDC
hi short question -> if i set up samba 3.0 as BDC with an NT pdc and a client wants to login to the domain, but his account is only stored (for tests) on the BDC: should he be able to authenticate successfully? or can he just login, if the *PDC* responds positive? thx in advance greez
2003 Sep 28
3
FYI-New ATA clone out
was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286
2005 Jun 28
4
How do you handle NAT?
We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something
2003 Oct 29
3
Am I missing somthing?
Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??
2005 Feb 06
0
re: difference between STUN servers and far-end solutions
Hi asterisk list, this is a bit off topic, but can anyone explain the point of the commercial far-end solutions floating around (jasomi, for example)? or are the far-end things just hyped up media proxies? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short,
2009 Nov 06
1
Function Value Not Being Overwritten?
Part of my problem is that I am in the middle of starting to use functions, so this was unexpected behavior for me.? Maybe there is a work-around because I like declaring some of the variables prior to using them.?? I have two separate files: # File: Test.R dog<-function(x) { ??? templeton<-NULL ??? source("Cat.R") ??? print(templeton) ??? print(x) ??? ??? print(bobby) } #
2008 Jun 23
2
Correlation Help
Hi, I have recently been using the R program and encountered a recurring problem. I have been trying calculate the correlation of a 16 column table. Everytime I type in cor(test), where test is data that I uploaded into R using the read.table function, I get an error: Error in cor(test) : missing observations in cov/cor In addition: Warning message: In cor(test) : NAs introduced by coercion