I've set at Protocol Management >> FXO Settings >> Dialing Mode ==> One Stage and everything is fine now
Thank you very much John, EDA On Wed, Dec 2, 2009 at 1:43 PM, John Balogh <[email protected]> wrote: > > I want to do single-stage dialing. I've just realized I > > > have the two-stage running now (I get dial tone and then, > > > when i introduce the number, the call get through). > > > > Right. > > > > According to the SIP User's Manual > > LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf > > page 67/294 > > > > " > > Enable Digit Delivery to Tel [EnableDigitDelivery] > > Disable [0] = Disabled (default). > > Enable [1] = Enable Digit Delivery feature for MediaPack/FXO & FXS. > > The digit delivery feature enables sending of DTMF digits to the gateway’s > port after the line is offhooked (FXS) or seized (FXO). For IP->Tel calls, > after the line is offhooked / seized, the MediaPack plays the DTMF digits > (of the called number) towards the phone line. > > [...] > > To use this feature with FXO gateways, configure the gateway to work in one > > stage dialing mode. > > " > > > > You probably need to set the above. > > > > The FXO parameter (from page 107/294): > > > > " > > Dialing Mode [IsTwoStageDial] > > One Stage [0] = One-stage dialing. > > Two Stage [1] = Two-stage dialing (default). > > Used for IP->FXO gateways calls. > > > > If two-stage dialing is enabled, then the FXO gateway seizes one of the > PSTN/PBX lines without performing any dial, the remote user is connected > over IP to PSTN/PBX, and all further signaling (dialing and Call Progress > Tones) is performed directly with the PBX without the gateway’s > intervention. > > > > If one-stage dialing is enabled, then the FXO gateway seizes one of the > available lines (according to Channel Select Mode parameter), and dials the > destination phone number received in INVITE message. Use the ‘Waiting For > Dial Tone’ parameter to specify whether the dialing should come after > detection of dial tone, or immediately after seizing of the line. > > " > > > > So you probably need to clear that parameter (it is not configured in your > .INI file now, so you need to add it, or change the web interface drop-down > control). > > > > Let us know if this helps. > > > > JDB > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Daniel - Asterisk > > *Sent:* Wednesday, December 02, 2009 12:33 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Help configuring Audiocodes MP-104 FXO > > > > Hi list, > > > I'm trying to get ready the MP-104 FXO to use qith my box, but when I send > calls I hear only dial tone and after a few seconds I get busy signal. > > I very appreciate your advices. > > Command line results and SIPconfigurations follows: > > *CLI>* > -- Executing [7991696...@total:1] Playback("SIP/101-09dd8918", "beep") > in new stack > -- <SIP/101-09dd8918> Playing 'beep' (language 'es') > -- Executing [7991696...@total:4] Dial("SIP/101-09dd8918", > "SIP/201/991696900") in new stack > -- Called 201/991696900 > -- SIP/201-09ddc890 answered SIP/101-09dd8918 > > > *sip.conf* > [201] > secret = **** > callerid = Mobile_01 <201> > type = friend > host = dynamic > context = total > dtmfmode=rfc2833 > qualify = yes > call-limit=5 > disallow = all > allow = gsm > allow = ulaw > allow = alaw > allow = g729 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
