Damn, where were you 6 months ago? ;) Daniel - Asterisk wrote: > Just if it is helps someone, based on information at the blog: > http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html > > I've summarized the following steps: > > *Step 1:* > Configure audiocodes to have registration account with asterisk, this > can be done easily with "Protocol Management -> Protocol Definition -> > Proxy&Registration", fill on "Proxy IP Address", "Enable Registration > : Yes", "Username", "Password", and "Authentication Mode : Per Endpoint". > > *Step 2:* > Configuring "Protocol Management -> Endpoint Phone Number", this is > important part for make each FXO port on audiocodes registered with > asterisk, in here, under "Channel", you can fill with either 1, 1-2, > 1-8, 3-4, or whatever you want to have, this means that port 1, or > port 1-2, etc will registered on astersik with userid/username filled > on "Phone Number", yes, that is correct, "Phone Number" on this > configuration page is AlphaNumeric, the password is using global > "Password" on First step. > > next, on same page configure "Hunt Group ID", this is another > important configuration which make audiocodes forward incoming call > from asterisk to any available FXO. Hunt Group ID is number from 0 to > any, I put 1. > > *Step 3:* > to make audiocodes forward call from FXO to asterisk, configure > "Endpoint Settings -> Automatic Dialing", I have 777 number on > asterisk to handle all incoming call, so I put "Destination Phone > Number" as 777 so every incoming call on FXO will be forwarded to 777 > on my Astersik. > > *Step 4:* > this is the last configuration that everyone need, forward call from > asterisk to any available FXO. in "Routing Tables -> IP to Hunt Group > Routing Table" configure under "Dest. Phone Prefix" with "*" (or any > prefix that you might have), "Source Phone Prefix" with "*", "Source > IP Address" with "*", "Hunt Group ID" with any number you configure on > Step 2, in my case, 1. > > /I add here addiiotnal steps needed for me to get ready/*: > Step 5:* > Add port by port authentication at Protocol Management -> Endoint > Settings -> Authentication > > *Step 6:* > Choosing Channel Selection Mode: Protocol Management -> Hunt Group > Settings, choose the hunt group number and the way you prefer. > > *Step 7:* > Choosing Dialing Mode: Protocol Management -> FXO Settings, I select > One Stage. > > Hope it helps. > > Elder Daniel > > > > On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk > <earohua...@gmail.com <mailto:earohua...@gmail.com>> wrote: > > 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 <j...@psu.edu > <mailto:j...@psu.edu>> 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:* asterisk-users-boun...@lists.digium.com > <mailto:asterisk-users-boun...@lists.digium.com> > [mailto:asterisk-users-boun...@lists.digium.com > <mailto:asterisk-users-boun...@lists.digium.com>] *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