Hi, what do you now get in the way of error messages? Robert Jenkins.
_____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: 19 January 2007 23:03 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79 Hi, I checked by changing to from-zaptel, but no luck yet. Pls guide me on this. Regards, vudura senadeera ------------------------------ Message: 9 Date: Fri, 19 Jan 2007 16:47:18 -0000 From: "Robert Jenkins" < [EMAIL PROTECTED]> Subject: RE: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380 To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED] > Content-Type: text/plain; charset="us-ascii" Hi, your zapata.con has 'context=from-pstn' Try changing this to 'context=from-zaptel' Robert Jenkins. _____ From: [EMAIL PROTECTED] [mailto: <mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: 19 January 2007 15:19 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380 Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity is OK. The digium TE110p LED state green. zttool also OK. Toshiba stata configured to make outbound call via E1 link with pressing 9 and then the out side number. I was able to make call from soft phone to analog extension at toshiba pbx. A==> B way as shown above. But when trying to dial from Toshiba PBX analog extension to asterisk extension, by pressing 9 the call rejected. In the asterisk command prompt I'm having following error message. -- Extension '' in context 'from-pstn' from '' does not exist. Rejecting call on channel 0/31, span 1 Is there any wrong in my setup. dial plan??, additional configuration if i required to put please guide me. I will be greately appreciated your feedback on this regard. configuration details /etc/zaptel.conf # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 /etc/asterisk/zapata.conf signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=euroisdn ;switchtype=national echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived overlapdial=no pridialplan=unknown immediate=no ;rxwink=300 callprogress=no loadzone=au context=from-pstn ; Points to the default context of your extensions.conf group=2 channel=>1-15 channel=>17-31 ;PRI/E1 link [trunkgroups] trunkgroup=>2,16 spanmap=1,2,1 /etc/asterisk/extension.conf [from-zaptel] exten => _X.,1,Set(DID=${EXTEN}) exten => _X.,n,Goto(s,1) exten => s,1,NoOp(Entering from-zaptel with DID == ${DID}) ; If ($did == "") { $did = "s"; } exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})}) exten => s,n,NoOp(DID is now ${DID}) exten => s,n,GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:notzap) exten => s,n(notzap),Goto(ext-did,${DID},1) ; If there's no ext-did,s,1, that means there's not a no did/no cid route. Hangup. exten => s,n,Macro(hangup) exten => s,n(zapok),NoOp(Is a Zaptel Channel) exten => s,n,Set(CHAN=${CHANNEL:4}) exten => s,n,Set(CHAN=${CUT(CHAN,-,1)}) exten => s,n,Macro(from-zaptel-${CHAN},${DID},1) ; If nothing there, then treat it as a DID exten => s,n,NoOp(Returned from Macro from-zaptel-${CHAN}) exten => s,n,Goto(ext-did,${DID},1) -- Thanks & Regards, Vidura B. Senadeera. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070119/0dd5e0 be/attachment-0001.htm ------------------------------ Message: 10 Date: Fri, 19 Jan 2007 11:46:57 -0500 From: "Chris Earle \(CBL\)" < [EMAIL PROTECTED]> Subject: [asterisk-users] Disconnect Supervision UK / BT solution? To: < <mailto:asterisk-users@lists.digium.com> asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P & Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear Time" setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. "Disconnect Clear Time" is BT's name for CPC. " Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist
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