search for: afficionado

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

My apologies if this post offends anyone but I wasn''t sure where else this might be appropriate to reach the largest group of Ruby on Rails afficionados. I have 2 tickets to the Canada on Rails Conference (in Vancouver, April 13&14) for sale. Any interested, please email me directly at: doris_on_rails@mac.com Thanks to everyone for their patience.

It would be great if the man page had examples for each of the commands. I've got R installed, but info R brings up the page for the GNU regular expression library. Any chance of having a complete set of man pages? Plain text documentation is good. I don't use Latex, and no one I know uses it. Thanks for a great package! Tom -- Tom Laudeman mailto:twl8n@virginia.edu (434)

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

...ure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak michiel@vanbaak.info http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

Hello! Those of you, who use Sweave a lot, will probably find my shell script usable. You can get it at: http://www.bfro.uni-lj.si/MR/ggorjan/programs/shell/Sweave.sh No warranty, however don't hesitate to contact me if you find an error or have a patch! -- Lep pozdrav / With regards, Gregor GORJANC --------------------------------------------------------------- University of

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,

Hi, I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI blinks and I do press retrieve button I get "Unknown" on display and busy tone. From the sip debug it looks like that Snom does not send credentials to Asterisk which responds with 407 Proxy Auth required. I have loaded Snom with latest 5 firmware. No change. I'm

...I set up an icecast server back in early 1999, and it seemed to work fine for months. We had a server crash here that meant I had to go and reget the sources to rebuild, and ever since then it has been nothing but grief. I've reached the end of my very, very long tether, and not being an afficionado of the online mp3 streaming scene, I'd just like to know:- - icecast is the open source equivalent of something. - can I pay some money and get the non-open source version of that something that will run on my Linux box and work ?? I just want to stream some mp3 files. It can't surely b...

On Tuesday, 24 July 2001 at 23:51, Mike Prudence wrote: > I've reached the end of my very, very long tether, and not being an > afficionado of the online mp3 streaming scene, I'd just like to know:- > > - icecast is the open source equivalent of something. > - can I pay some money and get the non-open source version of that > something that will run on my Linux box and work ?? you can get the not-open-source version...

Is it possible to start asterisk in the foreground ("asterisk -fc") and later detach from the terminal but leave asterisk running? thanks, James

Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian

Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phones....on lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail trough internet the quality goes down...it hears rally bad... i am on a