similar to: Asterisk and Client NAT

Hello, I''m stuck IPSECing my wireless network at home and would appreciate any comments. I appologize in advance if I''m wasting your time with trivia - I''m not a professional and staring at the problem for days from various angles hasn''t done me any good ... My home server/firewall (morannon) is hooked up through an USB to ethernet adapter (eth1) to my DSL

Dear Centos-Virt: Help! Guest=XP Pro, SP2 Host=CentOS5; # uname -r; 2.6.18-8.1.8.el5 VirtualBox-1.4.0_21864_rhel5-1.i586 http://www.virtualbox.org/download/1.5.0/VirtualBox-1.5.0_24069_rhel5-1.i586.rpm.run I have two nic: eth0 is the internal network; eth1 connects to a DSL modem. I also have a fully functioning DHCP server on eth0 (works perfectly with Parallels for Linux). The directions

on sip show channels I do get a lot of entrys like 192.168.1.47 11 07ba5a490b3 00102/00000 unkn No Init: INVITE 192.168.1.47 11 19090f115b8 00102/00000 unkn No Init: INVITE 192.168.1.47 11 7d8b8fde46f 00102/00000 unkn No Init: INVITE How do they appear? How can they be removed? "core show channels" does not list them. Elmar

I want to DENY access to port 5190 for all but two computers on my network DROP lan wan tcp 5190 Accept lan:192.168.1.48 wan tcp 5190 Accept lan:192.168.1.47 wan tcp 5190 Will this work? If not, what is the correct way to do it? Thanks

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

On 2007-08-19, at 15:41, Duncan Sands wrote: > can you please explain more about what restrict means: it may help > in improving code quality for Ada. In Ada you have runtime > constants that are really constant, for example array bounds. The > bounds are passed around by pointer, which causes LLVM to think > they may be aliased and changed by function calls (which is

I am trying to install Football Manager 2009 Demo. The file can be downloaded here: http://community.sigames.com/showthread.php?t=54626 When I run the setup.exe, I can see the first "loading %" screen. However, after that I get a blank window (probably the "next, next, next, install" set of screens). The shell output is as follows: > wine: configuration in

Hello all, i recently finished setting up my asterisk with sipura 3102 using PSTN. this is my dial plan relevant to wht i want: exten =>_01,1,Dial(SIP/$(EXTEN)@200) right now as u see i made my dial plan on a 2 stage dialing mode. tht means i dial 01, i get the pstn dial tone, and then i call whichever number i want through it. i want to have the option for my call to directly go through

I want to take the exams of redhat.. I?m starting now.. What advices I can get from you list? _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Timo, I've noticed that since the Courier to Dovecot migration have had few problems with the %usage of email accounts as they do not reflect the real use, to fix this I have to manually delete the file maildirsize and run the command vuserinfo user at domain.com, depending on your experience may be the cause of this problem? thanks Jose Luis

I already understood that a packet enters chains in the ''nat'' table only if it is the _first_ packet of a connection. In that case, we may do SNAT in the POSTROUTING chain of the ''nat'' table. So, the packets arrive to the POSTROUTING chain of the ''mangle'' table with the source IP address changed (if it''s said by the rules of the

Hi All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP

Hi, I wonder what does this warning 399 mean and how to workaround? "sip show peers" says that sip client is unreachable althought it works with some eexceptions ... I saw posts in this list about setting codec to ilbc, is this right action ? Also, I'm very interested if anyone succeded on Asterisk behind one firewall and Grandstream behing another ? I' almost got it working

Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext

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So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 10000 - 20000 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph

Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the

Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER

Could someone form the list please let me know an equalent pop connector to fetchmail?. I have problem with fetchmail that it downloads only 99 mails and in the event of mails exceeding this limit will result a socket error. Fetchmail: Query status=2 (SOCKET) Thanks Rajeev R. Veedu -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 2007-08-19, at 14:53, Torvald Riegel wrote: > does anyone keep track of performance results? Benchmark info is submitted by the nightly testers. http://llvm.org/nightlytest/ — Gordon -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20070819/636214ba/attachment.html>