search for: antek

Hi everybody, I found in the company where I work an Antek EGW-804. I googled to see if it can be configured to work with * and I understood that it is possible, but I don't know how. Can someone help me? Thanks Stefano

...h and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code: line 3982 if (p->owner) { if (strlen(buf)) { if (sipdebug) ast_verbose("DTMF received: '%c'\n", buf[0]); event = atoi(b...

...to other office. I don't want to completely upgrade the old equipments ( phones, PBX, etc..), part because the cost, other reason is we have tied too much to POTS in most of our life. What I am confusing about is which hardwares I can choose ? There are lot of things: Gateways : Antek VSP-5004 ( www.antek.com.tw) , D-LINK 1120, Multitech,... Telephony card: Quicknet LineJack, Digium, Dialogic, etc... Those gateways seem to have loaded with lot of features : FXO , Ethernet, NAT, QoS, various codec 723, 729 AB, SIP, H.323, ... and the price (for those made in Taiwan ) is low...

i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten => _.,1,SetVar(areacode=666) exten => _.,2,Background(zz-in-who) ; give them list of extns exten => _.,3,WaitExten(10) ; let them enter extn to call include => extensions include => applications include => speeddials

>From samles config. When user change uavarible message from firefly login by IAX got 3 file .WAV .gsm . wav But no sound record. When login by antek 804 GW (SIP mode) record success. but message can't play when someone need to leave voice mail to this box. But when delete .WAV . gsm and leave only .wav its' work. Is problem in asterisk Or client Dome C. ---------------------------------------------------------------------- This m...

...losing of stdin; ok deraadt - markus at cvs.openbsd.org 2000/07/19 18:33:12 [dsa.c] missing free, reorder - markus at cvs.openbsd.org 2000/07/20 16:23:14 [ssh-keygen.1] document input and output files 20000720 - (djm) Spec file fix from Petr Novotny <Petr.Novotny at antek.cz> -- | "Bombay is 250ms from New York in the new world order" - Alan Cox | Damien Miller - http://www.mindrot.org/ | Email: djm at mindrot.org (home) -or- djm at ibs.com.au (work)

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")