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...is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101115/33625680/attachment.htm

...phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110203/b57ad6d1/attachment.htm>

..., "CALLER_ID_INFO_ANI=2565551212") in new stack Note the first line should have the name field with the number, but does not. HOWEVER the voicemail notification contains: "Just wanted to let you know you were just left a 0:04 long message (number 1) in mailbox 3703 from "SMITH CASSIUS " <2565551212>" So - I know the NAME field is getting into the system, but it's not showing up on the phones (and with telemarketers, that annoys my users). I'm using Asterisk 1.6.2.9, DAHDI 2.3.0 I have added callerid=asreceived to chan_dahdi.conf for my inbound trunks,...

Hi, ? The current SCCP image on the 7942 phone is :SCCP42.9-0-2SR1S. We are trying to convert/upgrade the phone to SIP version of the firmware i.e : cmterm-7942_7962-sip.9-0-3 (Firmware is downloaded from the cisco support site). We have unzipped and placed all the files in the /tftp (root directory) of tftp server. Following files are also placed in the tftp directory. ? The Upgradation /

...pretty simple for these extensions; here is what I am using now: [extension1234] mailbox=1234 at default type=friend context=users host=dynamic secret=verysecret I have looked at the sample sip.conf and did not get any clues, also the SPA900 Admin Manual doesn't say anything about it. Thanks Cassius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101124/4be1ebc6/attachment.htm

...libasteriskssl.so.1 [root at Asterisk12 ~]# ls -l /usr/lib/libasteriskssl.so* lrwxrwxrwx. 1 root root 19 Oct 21 16:08 /usr/lib/libasteriskssl.so -> libasteriskssl.so.1 -rwxr-xr-x. 1 root root 625890 Oct 21 16:08 /usr/lib/libasteriskssl.so.1 [root at Asterisk12 ~]# Any ideas? Many thanks, Cassius

...ing for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting same error. Anyone else run into this? How did you get around it? cheers, Cassius Smith

...ntercom function to page all extensions, the call in progress gets disconnected. I'm wondering if there is a way to either: 1. dynamically figure out the subset of extensions that are not in a call, or 2. use some other function that will not bump a call? Has anyone else run into this? Thanks Cassius Here is my intercom context: [intercom] exten => s,1,Answer exten => s,n,Playback(beep) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,WaitExten(10) exten => t,1,NoOp(timeout) exten => t,n,Playback(sorry-youre-having-problems&goodbye) exten => t,n,Hangup() exten =>...

...ail ${MACRO_EXTEN} -- unavail) exten => s,n,Voicemail(${MACRO_EXTEN}@default,u) exten => s,n,Hangup() I was expecting the system to indicate that ringing was ? I know I can check channel availability to bypass this behavior; just curious why it's happening or whether it's expected. Cassius --

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi install gone wrong * Date: Mon, 23 Aug 2010 10:26:58 -0400...

...eaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious ? has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110506/ace7d87e/attachment.htm>

...ild asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius

...28] Call from '602' to extension '**600' rejected because extension not found" So - how to resolve this? Do I need dialplan code to handle this? I get the clue from "nothing to pickup for blah blah" that I'm close but may be missing something simple. Thanks all Cassius

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

...#39;t get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says " you have no messages". This is true. The message doesn't get moved into the INBOX directory for the mailbox. I am flummoxed. Any ideas welcome! Cassius Smith

...a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and will) use MeetMe to do large conferences. My question is - am I forced to do so by SIP? Or am I missing something? Thanks! Cassius Smith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100722/6f551aa9/attachment.htm

...t. Still no joy. [SPIDR-3758](caryspider) mailbox=3758 at default The above entry works, but: [SPIDR-3749](caryspider) mailbox=3749 at default This one doesn't. [caryspider] looks like this: [caryspider](!) type=friend context=users host=dynamic secret=xxxxxxxxxx Any ideas? I'm stumped. Cassius Smith