similar to: Odd MeetMe bahaviour with MoH ...

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

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

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

Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do

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,

Hi. I have a problem with my audio in meetme conference under asterisk 13 using Debian buster compiled from source. The problem is that there is some sort of distortion in the audio -- a workaround is always to lower the listen volume (*4). I see nothing in the log and so I wonder what is happening. I have dahdi loaded so I can record the conferences. Thanks in advance for any suggestions and

On Wed, 25 Mar 2020 12:42:00 -0400, Doug Lytle wrote: > > > >>> he problem is that there is some sort of distortion in the audio > > Has been been going on for a while or is this a new setup? Do you have a timing source? > > I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as horrible as I thought it would be to setup. Well, this has

On Thu, 26 Mar 2020 06:54:37 -0400, Doug Lytle wrote: > > >>> I never moved to confbridge because they don't have an option for controlling the volume of other > >>> participants audio > > I have menu options in my confbridge configs that has increase and decrease conference volume. > > I'd still configure a small confbridge and test if you still

Hi list, I was going through my dialplan today and found these 2 oddities with meetme using DAHDI to join the conference. 1. Although music on hold is indicated, I don't get any sound until I press *. Then the conference menu plays and all is well except - 2. According to the instructions, pressing 8 should leave the conference. No dice. Problem persists with 1.4.30 Any

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

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

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten => 300,1,MeetMe(300|M(class)) But that did not work. Thx, -Rob.

I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference "room", ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to

So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up the link, but it was from the voip-info wiki). My musiconhold.conf looks like this: ; ; Music on