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Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

...ssume ast-1 asks ast-2 to resolve the extension 12010, and I also assume that ast-2 returns "exten => 12010,1,Dial(Zap/1)" then ast-1 tries to Dial(Zap/1) which is not an interface on ast-1 and the call fails. What information am I missing? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

...S}/enterspeednumber) exten => _X.#,1,SetVar(SPEEDNUMBER=${EXTEN}) exten => _X.#,2,Background(${MYSOUNDS}/speedcode) exten => _X.#,3,SayDigits(${SPEEDCODE}) exten => _X.#,4,Background(${MYSOUNDS}/assignedto) exten => _X.#,5,SayDigits(${SPEEDNUMBER}) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

I'm getting the following message when I start Asterisk: WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 Did I mess up the registration key or is something else wrong? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...o see if my asterisk is registering and what 700 number is associated with my iaxtel account? I registered many months ago but never used it. My asterisk shows registered, but I can't seem to receive any calls (callers get a the user is not registered message) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...ttp://www.packet8.net/about/areacodes.asp I have not gotten Asterisk to work with Packet8, however I have put very little effort into making them work together. It's easier to just plug the DTA310 SIP/POTS device that you get from Packet8 into a X100P. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark

Im having problems setting up a trunk between two locations. Heres the setup I have: Server A is connected to the PSTN at my datacenter Server B is connected to a clients e1 line at his datacenter I only want to route calls from Server B to Server A and out through the PSTN. Server A has a lot of other things connecting to it, so I need a very specific context for all calls to go through.

...EN}) exten => s/_XXXX,103,Hangup ; ; Caller*ID is not 4 digits (external call) ; exten => s,1,Dial(${ARG1},${ARG2}) exten => s,2,Voicemail(u${MACRO_EXTEN}) exten => s,3,Hangup exten => s,102,Voicemail(b${MACRO_EXTEN}) exten => s,103,Hangup --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Please forgive me my ignorance ... I've spent two days trying to find out something about the format of the default configuration file, which CCM produces. The only example I have so far is the one from the chan_sccp source. There were tons of references on entering the callmanager commands on a cisco command line - which I don't have (don't need thanks to chan_skinny + chan_sccp).

...ick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the default screen when I'm done making a call. Does anyone know how to fix this? --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free)

...einit.scm but it didn't seem to make any difference. (set! default_after_synth_hooks (list (lambda (utt) (utt.wave.rescale utt .5 t)))) Does anyone have any suggestions? -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) -------------- next part -------------- A non-text attachment was scrubbed... Name: sanitizer.log Type: text/sanitizer-log Size: 1350 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/at...

...39;m sure it will have some bugs. It requires a few modules from CPAN and the asterisk-perl AGI interface. It's a very small script. Available at http://www.fnords.org/~eric/asterisk/ --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) -------------- next part -------------- A non-text attachment was scrubbed... Name: sanitizer.log Type: text/sanitizer-log Size: 1350 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/at...

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line