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Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Oop's. You were talking about the Fax build. Disregard my previous mail. -- I want a WESSON OIL lease!! -------------- next part -------------- An embedded message was scrubbed... From: marrandy <marrandy@chaossolutions.org> Subject: Re: [Asterisk-Users] A software FAX modem Date: Fri, 24 Oct 2003 13:30:26 -0400 Size: 1511 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20031024/8ccd5d00/attachment.eml

...64") in new stack NOTICE[426004]: File app_dial.c, Line 516 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -------------------------------------------------- show version gives :- show version Asterisk CVS-10/24/03-09:55:23 built by root@carol.chaossolutions.org on a i686 running Linux ----------- After the restart ----------- .....rest of startup [cdr_csv.so] => (Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> -- Starting simple switch on 'Zap/2-1' -- Executing D...

That'd be the 3CNJP24SE, we have one that powers 3COM NJ-200's. Works well. -----Original Message----- From: Martin [mailto:marrandy@chaossolutions.org] Sent: Tuesday, January 20, 2004 9:23 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea On Tuesday 20 January 2004 10:30 am, Kevin Ragsdale wrote: > > 3Com makes a 24-port midspan box that sells fo...

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

...ating asterisk/sounds/digits cvs server: Updating asterisk/stdtime [root@carol src]# -- "...Unix, MS-DOS, and Windows NT (also known as the Good, the Bad, and the Ugly)." (By Matt Welsh) -------------- next part -------------- An embedded message was scrubbed... From: Martin <marrandy@chaossolutions.org> Subject: new cvs build failure Date: Fri, 9 Jan 2004 18:51:44 -0500 Size: 750 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/3d4d23ff/attachment.eml

Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just

...n analog with a CID built in, you could have that, too? (Haven't tried that yet). What is the justification for VOIP? just total cost reductions (if the phone is cheap enough?)? Or are there some nicenesses that only VOIPs can supply? murf --__--__-- Message: 9 From: marrandy <marrandy@chaossolutions.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] NuFone International Calls Date: Mon, 17 Nov 2003 09:43:46 -0500 Reply-To: asterisk-users@lists.digium.com On Sunday 16 November 2003 12:52 pm, wasim@convergence.com.pk wrote: Hello Jasim. > lag just took on a whole ne...

Hello. Second question. Should I be asking this on the dev list (that's not the question by the way). Q. - there are several mentions on the list that asterisk :- "can interoperate with almost all standards-based telephony equipment" "interconnection with digital and analog telephony equipment" "visual message waiting indicator" etc. etc, That seems

Hello. Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx Because I'm experimenting, I would like to do things in a certain order :- 1) VOIP inside the private LAN from one computer to another. e.g. 192.168.1.1 to 192.168.1.2 etc. 2) VOIP

Just wondering. http://www.vorbis.com/ Regards...martin -- Maslow's Maxim: If the only tool you have is a hammer, you treat everything like a nail.

Hello. Does asterisk detect fax tones so it can switch to an extension where a regular fax machine is ? I've looked at the archives, and there seems to be a lot of debate about the fax issue in general. The last one, in January, I think, talked about a user that had fax tones in his voicemail, but asterisk never switched to the ext. There was no published solution to enlighten us to the

Hello. I suspect the answer is no, but I'll ask anyway. Commercial phone systems have phones with multiple extension buttons e.g. 20, that can be programmed so that when you press one, it will call the extension. Is there any 'open' phone that can do this with asterisk, does an 'open' phone even exist. Further to that, commercial phone systems have a operators

Hello. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Hello Barry. I was searching the archives and google for information on connecting to a proprietry PBX and came across an response from you dated Oct 31st, 2002. Quite good information. Since then, has more information become available ? As in, what PBX's have been successfully connected, and more importantly, how ? Regards...Martin -- Where will it all end? Probably somewhere near

*CLI> NOTICE[81926]: File chan_sip.c, Line 5144 (handle_request): Unknown SIP command 'NOTIFY' from '192.168.1.1' I think my settings on the budgetone (192.168.1.70) might be wrong as I'm getting this message in the console. Any hints/tips ? Regards...martin

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.

Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there.

Hello. Lost my test hardware, being used for other purposes. Any idea what the mimimum hardware for a 1-FXO (X100P), 4 FXS (TDM400, 4 analog phones) setup ? Regards...martin -- One nice thing about egotists: they don't talk about other people.