similar to: Confused about Asterisk server with regards to Linux NAT Firewall

Dear All, I think adding metadata and expecting transforms to repect it is a bad idea. It is just too easy for someone who does not know about the metadata to add a transform that ignores it. As for SAFECode, I think we have one of several options for handling load-widening. The most obvious one is to have a pass that just boosts the allocation size of any alloca with an align 16 attribute;

On Wed, Dec 28, 2011 at 12:40 PM, Criswell, John T <criswell at illinois.edu>wrote: > Dear All, > > I think adding metadata and expecting transforms to repect it is a bad > idea. It is just too easy for someone who does not know about the metadata > to add a transform that ignores it. > > As for SAFECode, I think we have one of several options for handling >

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

________________________________ From: Kostya Serebryany [kcc at google.com] Sent: Wednesday, December 28, 2011 2:46 PM To: Criswell, John T Cc: llvmdev at cs.uiuc.edu Subject: Re: [LLVMdev] load widening conflicts with AddressSanitizer On Wed, Dec 28, 2011 at 12:40 PM, Criswell, John T <criswell at illinois.edu<mailto:criswell at illinois.edu>> wrote: Dear All, I think adding

I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

I inadvertently unplugged my Budgetone phone tonight and I think it went out and upgraded its firmware. It is now at 1.0.3.81. Does anyone know how new this might be? Suddenly things that have been working wonderfully before don't work--basically it seems to do the SIP stuff just fine but then all the RTP stuff breaks, even something as simple as retrieving my voicemail from a server on

I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from

Hi, [resurrecting an old mail thread about AddressSanitizer false positive caused by load widening] Once the Attribute::AddressSafety is set by clang (a separate patch), fixing this bug may look as simple as this: --- lib/Analysis/MemoryDependenceAnalysis.cpp (revision 148708) +++ lib/Analysis/MemoryDependenceAnalysis.cpp (working copy) @@ -323,6 +323,14 @@

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

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I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

We're using CentOS samba for about 20 workstations, mostly XP, some 2000 and two win98. Samba shares a data directory with Word, Excel and Access documents/db's. Also Exact for Dos is running from this share (executable and data). The users experience a long delay every morning opening the first Word document (Word 97, Word 2000). This could take 10 minutes or so. If they close

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

I have a question with regards to using Unix groups with samba. This might be easier to explain with an example. Lets say I have two users UserID: GID UserA:Group1 UserB:Group2 etc..... I now have an entry in /etc/group that looks like groupdev::UserA, UserB If I have a directory with group ownership set to "groupdev" and group mode "rws" UserA & UserB cannot create a

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

Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo.

Something tells me this was covered recently, but i didn't find it again in gmail. The nodes see each other but if i have something misconfigured that is decreasing my speed. All my nodes are behind NATs. i have a remote node that connects to a central node via port forwarding from public port 7777 to hp821 running on port 655. remoteNode ----> public ip 1.2.3.4:7777 ---->

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