similar to: 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping

hello, I have a asterisk box (Slackware 9.1.0, Linux 2.4.31) connected to a Ericsson Businessphone PBX on the internal S0 bus with HFC-S card and zaphfc driver - point2point mode -. --------- | TELCO | | BRI | --------- | | PBX external S0 -------- | PBX | -------- | PBX internal P2P S0 NT Mode | | HFC-S Card P2P TE Mode

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

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'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

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?

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 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 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

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'

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

-- Executing DateTime("SIP/phone1-07ff", "") in new stack -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en') This works - the pathname is complete - Joy.

I have a AEX800 PCI Express card - sort of a TDM800 with PCI-Express. (or AEX844 - 4FXS & 4FXO) I downloaded Asterisk Now - and have got this loaded on a new motherboard (Intel with 3 PCI, 3 PCI Express - etc). (Downside on PCI-Express is the physical support the express slot gives (very little) compared with an 'old' standard PCI slot!!!) With only this card in the box....

I have * with i4l installed and working - on a dumb eicon card. It seems in order to get DTMF out of the BRI (for business banking - etc) - I should change from i4l drivers to capi drivers. wiki help seems to be for the Fritz card only...??? I have ticked the suggested boxes in 'menuconfig', explored the capi4linux websites - etc... but am missing some magic. I've "modprobe

Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the

I think what is missing with asterisk is what I'd call a 'working' demo. Its real cool getting the 'Welcome to asterisk' demo running... I think that there should be a 'make basic-plan' that would generate some well commented '.conf' files that set up a basic working systems with.. Two phones for Support Two phones for Sales Two phones for Accounting Voice

I have some Grandstream BT101 SIP phones. Work great (so far). I have some "Planet VIP-101T" H323 phones... how do I make them look/feel/act like a SIP phone ???? I can dial to them from both Trunk + SIP's (ie - I've added 'oh323' libraries) What config do I add so that if I dial the * IP - they then at least act as an extension? Ideally I'd like to just pick up

After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP -

Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register.... Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '****' in the the two password fields... This is what caught me - I

Hi, does the hint extension work together with the Snom phones in stable? I don't get an error in the dialplan, but it does not work either. On SIP/26 I want to monitor SIP/22. This is what I do right now: extension.conf [incoming] exten => 955,hint,SIP/22 exten => 955,1,Dial(SIP/22) sip.conf [26] ... subscribecontext=incoming ... Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b TIA