Hello I am using a Junghans quadBRI ISDN card and it is loaded and working. In Asterisk if I connect to ISDN line it is detected and tells me so. In my zapata.conf I have (abbreviated): [channels] switchtype=euroisdn signalling = bri_cpe context=default group=1 channel => 1-2 ;plus group 2 - 4 zaptel.conf: loadzone=uk defaultzone=uk # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Then in extensions.conf I have: [default] ; this below for internal extensions - works OK exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm) ; for dialing outbound - over ISDN line - this bit does not work exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) exten => _9XX.,2,Hangup Error I get is: -- Executing Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stack Jul 25 11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time -- Executing Hangup("SIP/200-e433", "") in new stack == Spawn extension (default, 902088787367, 2) exited non-zero on 'SIP/200-e433' I am dialing with sip phones. They work if dialing extensions internally but not if try to dial outside - eg dial 9 followed by number. What have I not done right? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050725/f2e9357a/attachment.htm
Angus Comber wrote:> ; for dialing outbound - over ISDN line - this bit does not work > exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) > exten => _9XX.,2,Hangup > Error I get is: > -- Executing Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stack > Jul 25 11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create > channel of type 'ZAP' > == Everyone is busy/congested at this time > -- Executing Hangup("SIP/200-e433", "") in new stack > == Spawn extension (default, 902088787367, 2) exited non-zero on > 'SIP/200-e433'You should rewrite the calling rule so that the 'outgoing' extension is not "ZAP/g1/902088787367|60" but "ZAP/g1/02088787367|60". The 9 should only be used to 'indicate an outgoing line'. Samples can be found on the wiki. I'm not 100% sure how to rewrite it so best check voip-info.org. Cheers, Kristf.
Have you defined the context "default" in the extensions.conf for outbound dialing in the globals section? For example, I have my ZAP channels identified as OUTBND1 not ZAP in the global section. This new global identifier is pointed to ZAP/g1 [globals] OUTBND1=Zap/g1 Instead of ZAP in my dial plan to call out, I use ${OUTBND1}. Yours: ; for dialing outbound - over ISDN line - this bit does not work exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) exten => _9XX.,2,Hangup Mine would look like this exten => _9XX.,1,Dial(${OUTBND1}/${EXTEN},##) exten => _9XX.,2,Hangup This helps me to keep track of inbound T1s and outbound T1s. Also, you have 2 (2) priorities listed in your example. You can't really do this. JASON WALKER ----- Original Message ----- From: Angus Comber To: asterisk-users@lists.digium.com Sent: Monday, July 25, 2005 8:11 AM Subject: [Asterisk-Users] Should this work? Hello I am using a Junghans quadBRI ISDN card and it is loaded and working. In Asterisk if I connect to ISDN line it is detected and tells me so. In my zapata.conf I have (abbreviated): [channels] switchtype=euroisdn signalling = bri_cpe context=default group=1 channel => 1-2 ;plus group 2 - 4 zaptel.conf: loadzone=uk defaultzone=uk # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Then in extensions.conf I have: [default] ; this below for internal extensions - works OK exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm) ; for dialing outbound - over ISDN line - this bit does not work exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) exten => _9XX.,2,Hangup Error I get is: -- Executing Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stack Jul 25 11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time -- Executing Hangup("SIP/200-e433", "") in new stack == Spawn extension (default, 902088787367, 2) exited non-zero on 'SIP/200-e433' I am dialing with sip phones. They work if dialing extensions internally but not if try to dial outside - eg dial 9 followed by number. What have I not done right? Angus ------------------------------------------------------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050725/d3db489d/attachment.htm
On Tue, 2005-07-26 at 12:03 +0900, Vic wrote:> Hi, everybody, > > can someone please clarify Digium cards for me? > > I was looking through the cards, and could not tell the difference > between TE410P and TDM400P. >The TDM400P is for FXS and FXO ports, up to 4 in any combination. The TE410P card is a 4 E1 Card. This means that you can connect up to 120 digital trunks to a single card. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050726/5d075e72/attachment.pgp