similar to: User authentication and privileges

If I want to send outgoing DTMF over a BRI interface, can I do it with 'isdn4linux' or must I use the 'capi' library? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part -------------- A non-text attachment was

I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys.... Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has

What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can

I'm using a SNOM 360 with Ver 4.3 software. Asterisk is.... Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

I have a Grandstream GSX-2000 with .. Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.0.3 I tried to do an HTTP update from the Grand Stream web site... After half an hour, I recycled power and now its dead... LED's come on and stay on, screen and buttons are dead. Connectivity to Grandstream.com was always good - whenever I checked (I downloaded the "User Manual" in a

I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part

I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the

I've got asterisk able to make and receive calls via the Internet via E164 lookups. If I get such a call - I'd like to display the original phone number on my phone. In the log is the following - which displayed '601' on my phone. The caller was +886288097680 - am I getting the wrong ClID because of my end or the caller end?

Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI> -- Executing VoiceMailMain("SIP/phone1-e0dd", "") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027

I have just installed one of the new TDM400 cards with an FXS and an FXO module into my * server. I also checked out the latest cvs head. I am using 7940 phones. Now I have some strange problems: 1. When in the VM menus, key presses do not register. 2. When I press "hold" on the 7940, it hangs up. Has anyone got any ideas? TIA Simon Brown

Hi there, could anyone drop a short line on what "cidinternalcontexts" exactly does in voicemail.conf? The Wiki explanation isn't sufficient - at least not for me... :-> Also: How/where do I define an Operator extension? Cheers, Philipp

I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? Have anyone tried chan_mISDN on a 2.6 box ? How does it run ???

We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear,

I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not "server" divas) with asterisk. First thing to do is to load the module. If you have two of these cards, you should do it like that: modprobe -v hisax protocol=2,2 type=11,11 And now you can have up to 4 incoming calls with two cards (try calling yourself and see if anything gets into

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten

Greetings all, I'm trying to build a justification case to get the firm I work for to start working with Asterisk more. How could I build this case? The argument I'm raising is that people need phones. PBX systems are too expensive for fewer options and less expansion capabilities. Leveraging Asterisk in the business plan would allow for more consulting revenue and resale

I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? Thanks John Fullington

Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of "restricted cone". Immediately reboot GS, get results "full cone". I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on

Playing with time ranges - using the examples found in one of the asterisk cook books... (pdf - page 17) ; After Hours include => night_menu|00:00-08:00|Tue-Fri|*|* include => night_menu|17:00-24:00|Mon-Thu|*|* this gives... ... pbx.c:2962 get_timerange: 24:00 isn't a valid end time.... -- Including context 'night_menu|17:00-24:00|Mon-Thu|*|*' in context 'default'