search for: bdarcy

...into our office, so having a seperate extension for each user who travels is not really the preferred solution. It's much easier to just hand out one phone number on your business cards... My setup in extensions.conf looks like the following (simplified) for one example user: [globals] BDARCY => SIP/bdarcy 3209 => ${BDARCY} [macro-stdexten] exten => s,1,Dial(${ARG2},20,tTr) exten => s,2,Voicemail(u${ARG1}) exten => s,3,Wait(4) exten => s,4,Hangup exten => s,102,Voicemail(b${ARG1}) exten => s,103,Wait(4) exten => s,104,Hangup [inbound] include => defau...

...sm would be most welcome as to the layout and configuration of the soon to be app_findme. Thanks! Spec for app_findme Have a .conf file (findme.conf?) which contains multiple contexts, each context's name should match the naming convention used with sip, or iax.conf. For example, if I have [bdarcy] as one of my sip peer entries, in findme.conf I would have, [bdarcy] also listed as an entry. Values within each entry would be labeled something like, [bdarcy] ExternalNum1: 91235551212 ExternalNum2: 91235551213 etc... app_findme would be used as the unavailable behaviour within the dialplan (o...

...uld be more than happy to implement odbc connectivity. I need to become more familiar with post and mysql first however. Up to this point, I've been strictly a MSSQL DBA due to my job functions. Thanks again for your feedback. Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: bdarcy@akiva.com Web: http://www.akiva.com Phone: 760-710-3209 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Goryachev Sent: Tuesday, June 01, 2004 5:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-...

...: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to ;externip = 216.9.32.42 ;localmask=255.255.254.0 ;localnet=192.168.0.0 context = default ; Default context for incoming calls ;srvlookup = yes [bdarcy] type=friend username=bdarcy secret=blah host=dynamic qualify=400 mailbox=3209 callerid="Brian D'Arcy" <3209> nat=1 disallow=all allow=ulaw If anyone can provide any feedback on what works for you, or what's recommended, it would be highly appreciated. Than...

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

I ordered 10 7960's with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, that's not really applicable in a SIP setup? I'm guessing it's going to be a more XML

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

The new voicemail features are great, cheers to those who worked on it! Un-related to this though (maybe?) When using the voicemail features (not happening in meetme or other apps that I can tell), I'm getting a lot of error messages in the CLI regarding IAX: Apr 28 14:05:45 WARNING[9226]: chan_iax2.c:5487 socket_read: Received mini frame before first full voice frame Sometimes this causes

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

I've got a strange issue where I get no ringing on outbound PRI calls using the TE410P span 1. The call actually goes out and works, you just hear no ringing. The quality and features on the call are pristine, no cracks pops or any weirdness like that. If I specify the 'r' option on the Dial() string, I get a half of a ring, sometimes a full ring as soon as the dial completes.

Hello all, One of my remote employees is using a 7960 we sent him, on a public IP address at his home office. I've run pings and traceroutes both from the server to his phone, and from the cable modem to our server, there's never a high ping time, or a dropped packet, however about every 30 minutes to an hour into his calls (not all of them, it's random) he can no longer hear the

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Shawn, That's a complete load of manure. I have an office full of 7960's, they work great with asterisk with the SIP images loaded. I'm about to pick up a lot of 7912's (simple one line phones, same as the 7905 but it has a built in switch). These phones have also been confirmed to work with Asterisk. I would recommend not going directly to cisco, and just find a reseller who