similar to: Re: 911 and lawsuits and redundancy

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

I have updated the voice prompts (mostly small fixes, but still work left to be done - however, it's usable now, I'd say) and the patch file (to current CVS) for Asterisk-in-Dutch. As far as I can tell, all of the grammar work is now in - if anyone has feedback to share, please do so before I spend lots of time cleaning up the patches so they can be integrated back.

Hi, Most companies we work with, have 'designated' crisis management teams. These vary from the insignificant crisis', through to life-threatening crisis'. There is always an assigned emergency services contact, whose job it is in an emergency, to maintain communication with the emergency services. One of our corporate functions is crisis management - so we have to consider

I have just completed a set of voice files in Dutch, plus a patch that forces Asterisk to sane (i.e. Dutch ;-)) behaviour when composing dates, times, numbers, etcetera. The current release, 0.0.1, is a sort of pre-release - some known issues have been identified, but I nevertheless would like to have some feedback so we can do a minimum number of polishing rounds. The patch is a bit

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone else has seen them: - Now and then, * just exits. Until now I had lowish-level verbosity on, so all I saw was 'Executing last minute cleanups'. What can trigger * exits? (in other words, what should I pay attention to when attempting to

Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of

Hardly finished building our phone system for our school district and I have an opportunity to sell and install a system for a local small business. We are competing against a bid for an integrated voicemail/switch that runs about $1300 (without phones and cabling) and will work with analog phones. Is there hardware configuration (either using analog or IP phones) that would meet these needs and

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

I saw on the wiki that asterisk supports a *0 dial code to flash the external trunk. When I try to use this on my system using a t100p card connected to a channel bank that is agregating 6 POTS lines the code doesn't seem to do anything. Do I need to set a config value somewhere to enable this code? Is anyone using this feature successfully? -- Jonathan Moore Director of Technology Winfield

I just purchased 30 of these after testing one for a few months and would like to quickly purchase another 40. We really like these phones: good sound quality, good echo control (no echo in speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys. Unfortunately we missed on big problem with Call waiting in our testing. When using asterisk 0.9.1 or rc2 the phone will reboot

I am wondering what is the best way to send the smallest files with the vm to emai l integration? I am not sure what order the three lines of the format command take, so I have just tried trial and error swapping. I think when set to "gsm" I get the smallest sizes. I can get my Windows Media player to play at least part of the file (get missing codec message from Realplayer), but get a

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

I have experienced lots of apparently dropped packets (in other words, lots of short interruptions of what the other party tries to tell me) with a GS102 and chan_capi. The GS102 is connected through a lightly-loaded switch directly connected to the * server, so bandwidth/latency shouldn't pose a problem. Funny thing is that the switch indicates 10mbit on the GS102 port - is that correct?

When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try

Hi! I'm new here, and I was wondering if there is any newsgroup gateway to the Asterisk lists? Thanks! Testa

Hi, I've been asked to see whether it is possible to do call logging for call center environments at a lower budget than the usual $1000 per channel. Afaik, with PRI this is possible through a high-impendance Y connection, but I wonder whether this would work with the Zapata cards. Anyone ever tried this? Regards, Cees -- XP SP2 can cause cancer in rats