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_____ From: Cullin J. Wible [mailto:cullin@emailanalyst.com] Sent: Monday, July 25, 2005 3:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'asterisk-dev@lists.digium.com' Cc: 'David Rule' Subject: Voicemail Send Message (Options 3, 5) Patch We run Asterisk 1.0.9 with mu...

I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD

...wered" is probably being transmitted before the "ringing" signal and is therefore being ignored by asterisk and iaxComm. If anyone could give this a shot and see what works or doesn't work that would be great. We have reproduced this with Asterisk 1.0.7 and 1.0.9. Thanks, Cullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050712/cab0408f/attachment.htm

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

...that for the purposes of the configuration tests, --includedir and --includelib were not being passed to gcc. The easiest fix is simply to alias gcc and put in the correct options - but I thought you might want to know anyway. This was using the latest source tarball as of a few days ago. Thanks, Cullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.mindrot.org/pipermail/openssh-unix-dev/attachments/20001023/4c5863dc/attachment.html

We're passing PSTN traffic on to a SIP proxy. The SIP phone customers have voicemail that will answer if their phone isn't picked up in a certain amount of time. However, if their phone is not on the network, a caller will get nothing but dead air as Asterisk keeps attempting to initiate the SIP connection. Is there a way to generate a ringtone for the caller while Asterisk is trying to

I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.

Is there a way to edit the options available in the voicemail menu trees? My users are complaining that it's too complicated (I know, it's not really complicated), and I wanted to remove some of the options if this is possible. So far I havent' found any info on the wiki or searches, not that it isn't out there.. I just cant' seem to find it.. Any pointers? Thanks Dan

>We did it by comment out a number of lines in the code and then re-compiled >just that module. Thx Cullin for the reply, has anyone made a flow chart or end user instructions for comedian mail? Jus trying not to reinvent the wheel if it's already been done. Thanks! Dan

I intend to add to my asterisk system a feature similar to cisco call manager's extension mobility so that agents can log in to any phone in the office and keep their profile (ex. the agent's specific directory number). But before doing that, I need to confirm that asterisk doesn't have a native solution for that (ex. application/addon), and that nobody has come up with their own

I have a ring group set up with 3 extensions. we'll use 14, 15 and 16. When a call comes in, it rings all three extensions. If one particular extension already is on the phone, it completely skips that phone and only rings the other 2. Example to explanation sake is: Call comes in, ext. 14 is already in the middle of a call, 15 and 16 will ring normally, but 14 does not have any

Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not