similar to: dovecot Digest, Vol 116, Issue 38

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

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:

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

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

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

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

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):

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

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

Every so often, I have an asterisk 1.4.22-4 system that goes into an endless loop with the following: [Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch' (escape_digits=) (sample_offset 0) [Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame [Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame [Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing

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+ .

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

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

hi folks, in my (unlucky) first test of ogg vs other encoders, i found a case where wma and mp3pro sound much better than ogg at 64k. can anyone suggest a setting that i haven't tried yet that can rival the wma and mp3pro samples at 64k? it's the "gravel effect" that is troublesome. the part in question is the first 15 seconds of this wave file:

I have an asterisk system where the asterisk daemon dies typically at least once per day. It is running in the wrapper safe_asterisk, which automatically starts the daemon back up. But we find this unacceptable because when the daemon dies, we usually have active calls drop, and sometimes we have to run asterisk -r -x "module reload" after the daemon starts back up before everything is

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that