search for: marrandi

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.

[Posted here becasue your mail server is rejecting my direct reply to you.] Hi Martin, AFAIK SIP can run on both UDP and TCP but I have only seen it used over UDP.. :) To setup the GS phones you need to open up the following ports (If its still set at the defaults)... UDP/5060 UDP/5004 UDP/5005 UDP/5006 UDP/5007 I have not tested the GS phone through a firewall yet but this config should

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).

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.

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:

Hello. I've seen some mentions of asterisk possibly being used as an inexpensive voicemail attachment to a commercial PBX etc. Does anyone here, have experience of using it in this fashion ? What commercial systems have been successfully attached too ? How is the attachment made ? Analog, digital ? If anyone has successfully accomplished this, I would like to hear the make and model of

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

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. 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. 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.

Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the

Hello. I have been searching the archives for a simple, clear listing of the available codecs with total size, plus the data and overhead sizes. Does anyone have this handy, and can it be added somewhere, even the wiki. Regards...Martin -- The system will be down for 10 days for preventive maintenance.

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:

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

I just rebuilt it and watched this time. What are the ? about ? [root@carol src]# cvs checkout zaptel libpri asterisk ? libpri/libpri.so.1.0 ? libpri/pri.lo ? libpri/prisched.lo ? libpri/q921.lo ? libpri/q931.lo ? asterisk/doc/api cvs server: Updating zaptel cvs server: Updating libpri cvs server: Updating asterisk cvs server: Updating asterisk/agi cvs server: Updating asterisk/apps cvs server: