similar to: asterisk and cisco 7970 - multiple lines

Hi I have a problem that I can't pass. I have asterisk and cisco 7970 phones with 8.0.3 sip firmware. I registered two extensions: Line1: 260 Line2: 160 Regardless of which extension I call, always Line 1 on cisco is blinking. This makes impossible to recognize which extension is calling. Also, I've set Line 2 to be automatically answered with speaker phone (intercom). Even though I

Hi all, I have had (what I consider) an odd request. The installation I'm working on now 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

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

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 /

Hi list, I'm having a problem with CallerID names not showing up when calls come in. I have dialplan code to store the callerid(name) away and it is blank (null). However, the voicemail variable ${VM_CALLERID} has the name field populated. For example, here is some of the dialplan code: 2. Set(CALLER_ID_INFO_ALL=${CALLERID(all)}) 3. Set(CALLER_ID_INFO_NAME=${CALLERID(name)}) 4.

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root at Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or

Hello. When trying to sort a folder by thread (with pine4.61), I get a sig11. The mailstore is Maildir, indexes stored in /var/indexes/%u. The syslog on the server shows this: Nov 29 11:37:55 olan dovecot: IMAP(kowalski): Corrupted index cache file /var/indexes/kowalski/.INBOX/dovecot.index.cache: record points outside file Nov 29 11:37:56 olan dovecot: child 14344 (imap) killed with signal 11

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

Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: ? checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking 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

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Dear Members, I am working with 6-dimensional Student-t distribution with 4 degrees of freedom truncated to [20; 60]. I have generated 100 000 samples from truncated multivariate Student-t distribution using rtmvt function from package ?tmvtnorm?. I have also calculated mean vector using equation (3) from attached pdf. The problem is, that after summing all elements in one column of rtmvt result

Dear OpenSSH developers, I was looking at the fail2ban project and had an idea that instead of parsing log files it could be possible to notify interested parties (like fail2ban) via (for instance) D-bus about a failed login attempt. Other application could also use this protocol to notify about suspect behaviors. A central functionality will allow for other (new) projects to integrate

Hello. I am currently testing dovecot (0.99.10.8, Debian woody from backports), with the maildir storage. I have noticed that when I use the mailutil tool from uw-imap distribution to transfer my IMAP mailboxes from our current server (uw-imap, mbx format) to the dovecot one, the mails arrival time are apparently lost. I can see this in Pine, using the Arrival sort: the mailboxes on the

This novice coder roughly understands the difference between instance and local variables thanks to David Black''s excellent book, but I''m still unclear as to when and why a either is more desirable to use. In general, I use instance variables in my controller and local in my views, but I''m not sure as to why or if this is correct. Thanks Joe Kowalski

Hello, I have a win32 DLL library, which I can't decompile, and can't get it's source code, but which I'd like to use in my Linux program. I wondered about making a win32<->Linux interface to use this library, using ready-to-use wine libraries. Is it possible, using wine libraries, to have access to functions from this DLL? If yes, can you tell me how? Thank you in

Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA