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Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very unfortunate) :). I was there demoing AstLinux on the Soekris Net4801, and Brian Capouch was showing off the WRT54GS run...

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

Mark and Ed received word today that the long-awaited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http...

For the past several days I can no longer use iconnecthere with asterisk. It is broken in BOTH directions; I can neither make nor receive calls. On outbound calls I get an immediate error: -- Got SIP response 400 "Bad or Missing To" back from 213.137.73.140 On incoming calls, the call switches through OK, and for a few seconds I get audio in both directions, although much

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

I just got a TDM30B, and the documentation doesn't indicate what driver I should be loading for it. Does anyone know? Any tricks I should know about before rolling this thing out? Thanks. B.

It seems like I've seen how to do this, but so far my archives search has come up dry. I have two X100Ps working with a single instance of asterisk, and I want to do different things with incoming calls depending on which line has rung in. Could someone provide me with a quick pointer? Thx. B.

I inadvertently unplugged my Budgetone phone tonight and I think it went out and upgraded its firmware. It is now at 1.0.3.81. Does anyone know how new this might be? Suddenly things that have been working wonderfully before don't work--basically it seems to do the SIP stuff just fine but then all the RTP stuff breaks, even something as simple as retrieving my voicemail from a server on

I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a "parking site." My accounts still seem to work, but I wonder WTH is going on? Thx. B.

Digium is sending a couple of Asterisk representatives to ISPcon next week in Baltimore: Greg Boehnlein (author of AstWind) and Brian Capouch (professor at St. Joseph's College and author of an upcoming Asterisk book by Prentice-Hall). We will also have a 10 x 10' booth there dedicated to Asterisk. *Greg and Brian need some help manning the booth.* If you have a business centered around Asterisk and would like to help them o...

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me,

In my opinion there ought to be a configuration option that allows for those who have their whole OpenSSL distro located in /usr/local/ssl to configure without what now must be done: moving libraries and include files around, etc. I am reasonably certain that I built my version of OpenSSL (0.9.5a) with the default install paths and names, and it breaks horribly when I try to ./configure in your

FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I

I hope this isn't something newbiesque and Steve will denigrate me. . . I just built gnophone, and I'm having trouble figuring out just how to sync the "Telephony Preferences" in gnophone and iax.conf on my asterisk server. I am heading on a trip and I'm pretty sure I'm going to be behind a NAT gateway, and my asterisk server is *separately* behind a NAT gateway. I

FYI to all you Grandstream users out there. I just fetched and installed the 1.0.5.0 firmware, and it appears they have removed the option to either do or not do SIP registration. Now it appears that one is going to register with the server specified in the "SIP Server" field, without any ability to disable it. Bug or feature? Just thought I'd put this out there and see if

I don't know whether this ought to go to the bugtracker. I downloaded the current CVS last night and then again just a few minutes ago. In both cases I can crash asterisk very easily by the following method: 1. Call up and leave a voicemail. 2. Log in and listen that I have a new message. 3. Hit "1" to listen, and 'Hasta la vista' asterisk. I also noticed that the normal

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is