similar to: SIP registration issues - Ugly workaround

Just out of curiosity, I''m running shorewall on a machine that has 4 nic''s and 4 different VPN tunneled subnets. When I want to define a service that is available from any source to a certain destination, instead of making a matrix of all the different combinations possible, is there an easier way? Something like,: ACCEPT any loc tcp ssh Which

Hello, I have a server to which is defined with static nat in Shorewall, and on that server, I''m running a http on a non-standard port (lets say, port 1234). I would like to use on of my free IP addresses, and map port 80 on the public side to port 1234 on the private side (forget about binding my services on a separate IP on the server, if it was feasible, I would have done that).

Hi, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/ae5eea65/attachment.html>

For those of you that have had/are having problems getting Win 98 clients to correctly authenticate, I have identified a stumbling block which I had to deal with. Specifically, EnablePlainTextPassword *must* be a DWORD value. In following the instructions found here, http://us1.samba.org/listproc/samba/October1998/0321.html and on the Microtheft web site, I installed the ptxt_on.inf file from the

Hello, I need some help in understanding few things related to llvm/clang. What I want to do is to take a C source code and then get some information from source code like the number of branches, load/store count etc. and then instrument the source code with this obtained information. At the moment, I can think of 2 approaches: (1) Clang: Generate AST from the source code, obtain these counts

> Is your Asterisk server listening on port 5061? If not, just change > the > entry to 5060. Also, I'm not sure how your sip.conf is set up for asterisk, but if you've set it up like: [203] type=friend username=blah secret=blah etc... Your Polycom config file will generally look like this. <PHONE_CONFIG> <OVERRIDES

Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they "think" packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

Dear Wine Team, I believe that I got the correct e-mail adress, however, if I don't, please redirect me. Anyway, I have a problem with wine that I'm wondering if there is a fix to. everytime I try to run a program under it, it comes up with a funny language, kind of like a Arabic font. I'm not sure how to correct this. This happens even when I try to open the control panel for

I use the following code to do the depth first search on the DSGraph. And I always got this error message. Couldn't figure out what's wrong. Could you please let me know? Besides, I really have no idea about what those functions do. Is there any documentation about DSGraph functions besides the pool allocation paper? Thanks, xiaodong Code: for( df_iterator<DSNode*> I =

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

Hi Farhan, What is the purpose of the protocol? Just to be a binary form a SIP? I mean conferencing, PTT, chat we have in h.323, sip, jingle, iax? What is the reason for a new one? The NAT traversal has been fixed for a while for all those protocols also. Diana farhan@phonestack.com wrote: > Dear All, > > I have finished documenting my Lightweight Telephony Protocol. It is > now

Hi all, I have a data frame with column over which I would like to run repeated functions for data analysis. Currently I am only running recursively over two columns where I column 1 has two states over which I split and column two has 3 states. The function therefore runs 2 x 3 = 6 times as shown when running the following code: mydata <- data.frame(userid = c(5, 6, 5, 6, 5, 6), taskid =

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

Dear Samba developers, We have an unexpected problem with Samba in Combination with M$ Terminal Server. Setup: ------ - Terminal Server with Office Applications (MS-Word, ...) used by multiple concurrent users (30) - Linux box with Samba 2.0.6 as print / fax server Goal: ----- - To send FAXes from Office applications How we did it: -------------- in smb.conf: [fax] comment = FAX Service

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

FYI Kris ----- Forwarded message from Kris Kennaway <kris@FreeBSD.org> ----- X-Original-To: kkenn@localhost Delivered-To: kkenn@localhost.obsecurity.org Delivered-To: kris@freebsd.org Delivered-To: ports-committers@freebsd.org From: Kris Kennaway <kris@FreeBSD.org> Date: Thu, 17 Apr 2003 14:45:03 -0700 (PDT) To: ports-committers@FreeBSD.org, cvs-ports@FreeBSD.org,

FYI ----- Forwarded message from Kris Kennaway <kris@FreeBSD.org> ----- X-Original-To: kkenn@localhost Delivered-To: kkenn@localhost.obsecurity.org Delivered-To: kris@freebsd.org Delivered-To: ports-committers@freebsd.org From: Kris Kennaway <kris@FreeBSD.org> Date: Thu, 4 Dec 2003 06:45:06 -0800 (PST) To: ports-committers@FreeBSD.org, cvs-ports@FreeBSD.org, cvs-all@FreeBSD.org

Hi, ThoughtWorks just released the auto test toolset Twist, the "tagging" feature seems pretty useful, http://studios.thoughtworks.com/twist-agile-test-automation/twist-features-and-benefits "Tags allow you to organize test suites in multiple dimensions. With tags, you can include a test in a smoke test suite by just tagging it smoke, or exclude unfinished tests from running by

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip: