similar to: openssh vulnerability WITH TCP DUMP!

Hello, my apologies for not being sure whom to tell this problem, but it is very strange. Let me tell the story: I''m using XEN (3.0.2) with SLES10 (x86_64, SunFire X4100). On one machine I have three virtual machines ("DomU") that are very identically configured (SLES10 x86_64 also). There is also a SLES9 (i386) acting as a multi-homed NFS server. I can mount and access a

Hi, Jeremy: > That isn't any use. We don't need to know what is going on at the client, we need to know what is going on at the server. I pasted the samba log here: https://raw.githubusercontent.com/CaoShuFeng/paste/master/samba_log The file I fsync()ing is called `aaa`. I can see some log about this file. Thanks Cao 在 2018年02月02日 21:07, Jeremy Allison 写道: > On Fri, Feb 02,

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Using Wireshark I can see that ZoIPer always send a 160 byte payload. First few payloads contain 20 bytes of data (what I believe to be a mode 3 frame, ie first byte in 0x18 - 0x1F range), followed by 0x7B (21st byte), ie. 5 bit 0x0F terminator padded with 011. ... and then zeroes all the way up to 160 bytes. ... but then after a few payloads there are more following these 21 bytes ...

Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register => username:password@192.168.11.6 But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The

Greetings! Being a Perl hacker for some time, and wanting to leverage what R provides, I've been trying to work with Statistics::R and RSPerl. The former has a race condition that breeds some unreliability and the latter seems to have issues all around, and neither has been updated in some time. Are these projects are abandoned, or is there some effort currently being undertaken to

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

Hi all, I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)? Greetings, Evert This message posted from opensolaris.org

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert

Sure, and thanks for trying to help! These are the two correct answers when SNI is included. The certificates are fully chained. Both certificates carry the same subject mail.cs.sbg.ac.at but differ in Subject Alternative Name (SAN). X509v3 Subject Alternative Name:? ? DNS:mail.cs.sbg.ac.at, DNS:smtp.cs.sbg.ac.at, DNS:imap.cs.sbg.ac.at, DNS:pop.cs.sbg.ac.at X509v3 Subject Alternative Name:? ?

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled

Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.

Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this

Hi all! Am I the only one with this problem? I doubt it... The problem is that I have a couple of shares of a W2K server mounted with Samba on my (Gentoo) Linux. This works fine, until the W2K server gets rebooted. After that the shares are just timing out, and they are impossible to unmount/remount... :-/ How do I prevent/fix this problem? Regards, Evert

What rigs do you folks use to listen to your music? I have a P-III 500 with Altec Lansing speakers in the dining room and a P-II 350 with Labtec speakers in the Guestroom/office. Sorry, I can't remember what model the Lansings are off the top of my head. The Labtec speakers are fairly cheap. I have a PCI ensonique sound card in the P-III system. I not sure what kind of sound card is in the