similar to: Asterisk 1.0 released

Hi all, Anyone got a link or the file to Mr. Guy's notes from his speech at Astricon.. I looked for them or relevant info he showed ad FWD's webpage but have not been able to find them. He had a working FWD registered thru Asterisk behing a NAT and a couple of samples of their new scripts, talking weblog (from Mr. Pulver) and other cool new features. If anyone got these please send em

Hi, Just finished watching the season finale of '24' the TV series. Throughout the series they have been showcasing Cisco hardware especially Cisco IP phones (7970's). On the last episode or two they showed what seemed to me a new cisco IP video phone. It stands just as a 12" lcd screen with the cisco branding/logo and letters just as the 79xx series. I wonder if this is a

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Has anybody been able to get in touch with anybody at digium today? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just

Hi, I'm having a problem with one of our 7960. They all run latest 7.4 SIP firmware. The problem appears on the other end. The other end constantly hears a 'crackling' noise. I have tested using phone set, headset and speaker and the noise appears on all cases. I have other 7960 setup exactly same way (using same asterisk, firmware, etc) so it looks like a hardware issue.

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Hi, I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the

Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once

Apparently asterisk cannot reboot gracefully (unattended) when using realtime.... MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug. Since asterisk starts before mysql it will never start any workaround for this? g Regards Greg Cirino

I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to

It's been hours since I've seen a post from this list Must be broken again. Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You

does anybody have an idea what the difference and significance of sip seeding and registration is. g Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Hi, Has anybody had any problems getting digium hardware lately? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a

it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When

Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to transfer the call to this extension Currently they must pus "flash" for a new line speak to the

Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

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