It was a pending draft I forgot to send.. sorry. On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins <mcoll...@ccdservice.net>wrote:
> 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 >
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