similar to: DTMF problem during read()

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with

> Hello > I complied openssh like this "./configure --with-pam" and I did configure > /etc/pam.conf as follows > # PAM configuration > # > # Authentication management > # > sshd auth required /lib/security/pam_securid.so reserve > sftp auth required /lib/security/pam_securid.so reserve > # > login auth required

To all Employable Asterisk Professionals, We are very pleased to announce the unveiling of the newest incarnation of the popular, OpenSource VOIP jobs forum at http://www.asterisk-jobs.com. We at Asterisk-Jobs.com appologize for the inactivity for the past while. It had come to our attention that the software running the job board was unsecure and allowed for multiple Vulnerabilities. The site

Is it possible to implement in the Asterisk dialplan some way to authenticate a user with a dialed passcode which opens session that stays active enabling the user to make and receive calls, until the user logs off with another dialed passcode? I am aware of the Asterisk application 'Authenticate', but as far as I know, with this application the user meeds to dial his pin at each call he

Hello I am compiled openssh3.6.1p2 with PAM and using RSA Security (ACE) token. the command I used to compile ssh as follow: 1. ./configure --with-pam 2. make 3. make install After starting the sshd daemon, I authenticate using the command "ssh xxx.yyy.nih.gov" On the SecurID server I was watching the log monitor and I saw the following errors : "ACCESS DENIED, syntax

Hi, I would like to do some development on the securid patch to get it to display the passcode prompt when users log in with securid on ssh. Can anyone tell me which scripts currently handle the logon/password prompt in openssh please? Thanks, JS. _________________________________________________________________ Join the world?s largest e-mail service with MSN Hotmail.

Hi, everyone, A quick reminder: This call will start in approximately four hours. See below for how to join. On our current agenda: Agenda * Full restrict patch * Observation and clarification * Issues encountered * Speed up in intersection code - O(nlog(n)) * Calling Verifier after LoopVectorize pass? -Hal Hal Finkel Lead, Compiler Technology and Programming

How do I make a user dial a passcode if he wants to make an international call? ________________________________ This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient,

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it

I need some help with a problem which I'm facing with Asterisk 1.4 final release. I'm using static agents in a queue. Sometimes when an agent answers a call in queue and then releases it, the status for that agent in the queue remains busy where as there is not channel associated to that SIP client. For furthur calls in that queue that particular agent receives no more calls unless you

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

How do I make a user dial a passcode to make calls through asterisk? We would like to place a phone at a client's location for our employee but are afraid it may get abused by the other workers. ________________________________ This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten =>

Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten => 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH

Hi This is a feature request. 1) Make sshd to ignore garbage that may appear before ssh identification string is received. Such "garbage" may be for example telnet negotiation codes. This should be pretty easy task. 2) Make ssh to work in cleartext mode (and have minimum telnet negotiation handling) before it receives ssh identification string. This requires somewhat

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a bigger box is needed? Any suggestions about where to pick up another box? 2) does the Digium TDM100P (already discontinued) fits fine in a

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the '9' comes in and gets queued, but it doesn't start

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you