similar to: permission problem

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

http://yro.slashdot.org/article.pl?sid=04/08/04/2212251&tid=158&tid=95&tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

I was hoping someone can help with a problem with my GrandStream Budgetone "hanging" after a while. Problem seems related to the SIP registration - I am using dynamic registration (host=dynamic in sip.conf). Static IP is not an option in my case. I start Asterisk and all is groovy, phone works fine and can dial around. Then, at the time specified on the phone in the "Register

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

---------- In?cio da mensagem original ----------- De: asterisk-users-admin@lists.digium.com Para: asterisk-users@lists.digium.com Cc: Data: Mon, 12 Jul 2004 11:48:05 -0500 Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide

>>> TL;DR: please try the patch out and report if it causes "Did not receive >>> identification string" log messages. I believe it does not. Aw crap. My homegrown anti-dos tool for ssh looks for either DNRIS or if logging is verbose enough a connection that didn't result in a login. I give the attacker a few tries and whitelist any successful candidate so I

Damien Miller <djm at mindrot.org> writes: > We don't plan to remove this check. Accidental key exposure is still an > unfortunately common problem and, while this check isn't perfect, I'm > pretty sure that it avoids enough real-world misconfiguration to > justify it's continued existence. Maybe the check could have a configuration option to disable it? That

In a previous mail, Jeremy Fitzhardinge wrote: > The softlockup watchdog is currently a nuisance in a virtual machine, > since the whole system could have the CPU stolen from it for a long > period of time. While it would be unlikely for a guest domain to be > denied timer interrupts for over 10s, it could happen and any > softlockup message would be completely spurious. I wonder

In a previous mail, Jeremy Fitzhardinge wrote: > The softlockup watchdog is currently a nuisance in a virtual machine, > since the whole system could have the CPU stolen from it for a long > period of time. While it would be unlikely for a guest domain to be > denied timer interrupts for over 10s, it could happen and any > softlockup message would be completely spurious. I wonder

This patch against OpenBSD -current adds a simple form of PKI to OpenSSH. We'll be using it at work. See README.certkey (the first chunk of the patch) for details. Everything below is BSD licensed, sponsored by Allamanda Networks AG. Daniel --- /dev/null Wed Nov 15 15:14:20 2006 +++ README.certkey Wed Nov 15 15:13:45 2006 @@ -0,0 +1,176 @@ +OpenSSH Certkey + +INTRODUCTION + +Certkey allows

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC

Hm, look what I got hold of today.. Works if sendmail is mode 4111 or similar: #! /bin/sh # # # Hi ! # This is exploit for sendmail smtpd bug # (ver. 8.7-8.8.2 for FreeBSD, Linux and may be other platforms). # This shell script does a root shell in /tmp directory. # If you have any problems with it, drop me a letter. #

Ok, I've been away from the list for sometime now and feel as though I'm going to ask something that's been asked many times before, yet I find nothing in the wiki, so here goes: I notice that allison has done recordings for weather forecasts, yet I find no agi's that parse forecasts and use this recordings to piece together the forecast. I find plenty of weather agi's, but

I believe a good amount of them (if not most of them) have already been ported! Off the top of my head, I remember that asan, tsan, msan, hwasan, the kernel santizers, and sancov have been ported. I don't think ubsan has been ported yet though. You can also check if other passes you need run under the new PM by checking PassRegistry.def. On Fri, Aug 2, 2019, 14:20 Jordan Rupprecht