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...g List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anyways, I wonder how would it benefit any hacker in any way. -- Zeeshan Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak <jmaslak at antelope.net<mailto:jmaslak at antelope.net>> wrote: No. It seems that opening up some sort of automatic blocking could cause an attacker forging packets to block legitimate endpoints. It also seems like they won't get in with good passwords, so it isn't actually accomplishing...

I'm trying to install both a Sangnoma A102 (with echo cancellation) card and a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the same server. I know I probably shouldn't have mixed vendors - lesson learned for next time. That said, I have everything working fine...except Native Bridging between the Sangnoma and Digium cards. When I do native bridging, I get a very

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, ------------------------------------------------------- Dario Quiroz (71) 9275-9080 gtalk: darioquiroz at gmail.com ------------------------------------------------------- -------------- next part -------------- An HTML attachment was

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit. Asterisk 1.6 makes things a bit easier than what is on that page. I'm integrating an Asterisk server with a Toshiba Strata system at my office. Right now, it is driving some VoIP phones (Cisco ATAs with analog phones plugged into

I didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM & Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any

I did get it to work, and can place and receive calls through the Firefly network via *. Compared to iaxtel or FWD, there is a significantly higher amount of latency, but it is workable. For some reason, this needed to be the last entry in my iax.conf or it would try to authenticate with a different user ID when receiving calls (and obviously would fail. Relevant section from my iax.conf:

I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P.