similar to: Endless loop with asterisk directory

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello everyone. I am wondering whether there is a certain technique I should use to identify all log lines in the asterisk/full logfile that are related to a single call. If a user reports that something strange happened with a certain call, I'd like to be able to easily go back and look at the asterisk/full logfile, and look at only the lines that are relevant. I am having some difficulty

1. Does "Wrap-Up-Time" apply to all queue agents/extensions that just rang,= or only the one who actually answered the call (I assume the latter)? 2. Does the "Member Delay" delay the ringing of new calls to agents, or onl= y come into play AFTER the agent answers the ringing call? Any other suggestions for how I can resolve this issue? I am wondering whet= her "Agent

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

What are the pros and cons of both? Especially in regards to performance, stability, management & maintenance? I really appreciate feedback. We're on a time-crunch to migrate from a debian 5 box w/ dovecot 1.1 to a debian 6 box w/ dovecot 2.0.9 (built from source). Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ .

Today I found our dovecot 2.x gracefully logged: dovecot: master: Warning: service(imap-login): process_limit reached, client connections are being dropped I am confident that we had the very same problem on our previous dovecot 1.x box. Of course with dovecot 1.x, the same relative setting is login_max_processes_count. I believe that I turned up all dovecot logging & debugging to the max

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Today I found out we are having users w/ problems because: Mar 3 09:57:33 jlgray dovecot: master: Warning: service(imap-login): process_limit reached, client connections are being dropped Mar 3 09:58:42 jlgray dovecot: master: Warning: service(imap-login): process_limit reached, client connections are being dropped Mar 3 10:02:51 jlgray dovecot: master: Warning: service(imap-login):

Here's part of the output of running an AGI file: -- Playing 'degrees' (escape_digits=) (sample_offset 0) -- Playing 'fahrenheit' (escape_digits=) (sample_offset 0) -- Playing 'wx/humidity' (escape_digits=) (sample_offset 0) -- <DAHDI/1-1> Playing 'digits/40.ulaw' (language 'en') -- <DAHDI/1-1> Playing

This morning on our newly built server, the following was logged twice: auth: Error: pam(username,127.0.0.1): pam_authenticate() failed: Authentication failure (/etc/pam.d/dovecot missing?) This also happened to be during a time of 100+ imap-login processes, where we were seeing: master: Warning: service(imap-login): process_limit reached, client connections are being dropped The initial error

Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said that he has "no real data" on why I shouldn't use 1.8. They just follow a practice of not jumping on the newest version. But I

I am wondering if anyone knows of a way to do this, as it would be much more meaningful for our CDR reports. We use FreePBX under the Elastix distro. We are able to set the CALLER's CID on inbound calls by using the "Asterisk Phonebook" module in FreePBX, then configure the Inbound Route settings to use it for CID. I haven't seen anything like this to apply those same rules to

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi, We have clients with various security & compliance requirements. Although not required, it would be ideal to have messages encrypted at rest. We already use SSL/TLS to secure the transmission of most email. However, it would be nice to have them encrypted sitting on our server. Is anyone doing this? I think that ideally, rather than full-disk encryption, we should use an encryption that

I am interested in IMAP Voicemail storage for some of my customers. Does anyone know of any vendors of asterisk appliances (physical PBXs) that provide this as a "standard feature" (or an optional standard feature)? Ultimately, I'd like to be able to have a single point of accountability for the system as a whole. I would like an intuitive & powerful configuration GUI (such as

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its