I do not have the possibility to check does TDM800 works with asterisk 1.6.2. I checked i have this code in chan_dahdi file. But when I try to call, I get only
chan_dahdi.c: Using channel 11 devicestate.c: device 'DAHDI/11-1' state '2' rtp.c: Channel 'DAHDI/11-1' has no RTP, not doing anything channel.c: Not copying variable DIALEDTIME. channel.c: Not copying variable ANSWEREDTIME. channel.c: Not copying variable DIALEDPEERNAME. channel.c: Not copying variable DIALEDPEERNUMBER. DEBUG[9730] channel.c: Not copying variable DIALSTATUS. DEBUG[9730] channel.c: Not copying variable SIPCALLID. DEBUG[9730] channel.c: Not copying variable SIPDOMAIN. DEBUG[9730] channel.c: Not copying variable SIPURI. DEBUG[9548] app_queue.c: Device 'DAHDI/11-1' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Ignore possible polarity reversal on line seizure DEBUG[9730] chan_dahdi.c: Dialing '8685XXXXX' DEBUG[9730] chan_dahdi.c: Deferring dialing... VERBOSE[9730] app_dial.c: -- Called 11/8685XXXXX DEBUG[9544] devicestate.c: No provider found, checking channel drivers for DAHDI - 11 DEBUG[9544] channel.c: Avoiding initial deadlock for channel '0x1d81910' DEBUG[9730] channel.c: Set channel DAHDI/11-1 to read format slin DEBUG[9544] devicestate.c: Changing state for DAHDI/11 - state 2 (In use) DEBUG[9544] devicestate.c: device 'DAHDI/11' state '2' DEBUG[9730] channel.c: Set channel SIP/XXX-0000000b to write format slin DEBUG[9730] channel.c: Set channel SIP/XXX-0000000b to read format slin DEBUG[9730] channel.c: Set channel DAHDI/11-1 to write format slin DEBUG[9548] app_queue.c: Device 'DAHDI/11' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: Sent deferred digit string: T8685XXXXXw DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Dial Complete(9) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: No echo cancellation requested DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event On hook(1) on channel 11 (index 0) DEBUG[9730] channel.c: Hanging up channel 'DAHDI/11-1' DEBUG[9730] chan_dahdi.c: dahdi_hangup(DAHDI/11-1) DEBUG[9730] chan_dahdi.c: Hangup: channel: 11 index = 0, normal = 15, callwait = -1, thirdcall = -1 DEBUG[9730] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/11-1 DEBUG[9730] chan_dahdi.c: Updated conferencing on 11, with 0 conference users VERBOSE[9730] chan_dahdi.c: -- Hungup 'DAHDI/11-1' On Wednesday, April 07, 2010, at 12:17AM, "Alec Davis" <siva...@paradise.net.nz> wrote: >Does TDM800 with FXO ports work with 1.6.2? > >You should have also got other 'polarity related messages' during the call >setup. >One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get >fired. > >Code below. > >ast_debug(1, "Polarity Reversal event occured - DEBUG 2: channel %d, state >%d, pol= %d, aonp= %d, honp= %d, pdelay= %d, tv= %d\n", p->channel, >ast->_state, p->polarity, p->answeronpolarityswitch, >p->hanguponpolarityswitch, p->polarityonanswerdelay, >ast_tvdiff_ms(ast_tvnow(), p->polaritydelaytv) ); > >If it doesn't work with the TDM800, file a bug report. > >Alec > > >-----Original Message----- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas >Gulbinskas >Sent: Wednesday, 7 April 2010 8:00 a.m. >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] polarity reverse > >call not succsessful. > >I use nokia gsm gw witch have polarity reverse i try on my old asterisk >1.4.17 with digium tdm800 with fxo ports card polarity reverse works fine. >But then i connect to asterisk 1.6.2 with sangoma a400 with fxo ports card >polarity don't work. >polarity reverse is 600 milliseconds set on nokia gsm gw > >On Apr 6, 2010, at 10:08 PM, Alec Davis wrote: > >> Is the call successfull? >> The 'Ignore polarity reversal on line seizure' may just be a warning. >> >> What equipment, which Telco is the FXO card connected to? >> >> Alec Davis >> >> >> -----Original Message----- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas >> Gulbinskas >> Sent: Wednesday, 7 April 2010 12:03 a.m. >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] polarity reverse >> >> Hi, >> >> I have a problem with polarity reverse >> >> this my dahdi config >> >> [channels] >> context=default >> usecallerid=yes >> hidecallerid=no >> callwaiting=yes >> usecallingpres=yes >> callwaitingcallerid=yes >> threewaycalling=yes >> transfer=yes >> canpark=yes >> cancallforward=yes >> callreturn=yes >> echocancel=yes >> echocancelwhenbridged=yes >> relaxdtmf=yes >> rxgain=0.0 >> txgain=0.0 >> group=1 >> callgroup=1 >> pickupgroup=1 >> immediate=no >> answeronpolarityswitch=yes >> >> I use asterisk 1.6.2 and sangoma a400 fxo ports. >> Then i try call i get chan_dahdi.c: Ignore possible polarity reversal >> on line seizure >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- New to >Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >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 -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users