hello yes this is a fresh install [trunkgroups] trunkgroup => 1,16 spanmap => 1,1,1
[channels] #include dahdi-channels.conf context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel => 1-15,17-31 the issue h=just with group 1 can not call via G1 with group 2 theris no problem group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=520xxxxxx immediate=no channel => 32-46,48-52 thanks and regards 2013/10/21 John Novack <jnov...@stromberg-carlson.org> > A VERY OLD and beyond EOF version. > If you MUST, due to some driver issue, use Asterisk 1.4, then please use > 1.4.44 > Otherwise I suggest you move to something more current, either version > 1.8.current or beyond. > Also, CLI says 1.4.43, your message says 1.4.32 ??? > > Some examination of chan_dahdi and your dialplan would help someone give > you some assistance. > Is this a fresh install, or one that has been working for years? > > What Digium card? > > John Novack > > Salaheddine Elharit wrote: > > i need your help regarding some issue related to the outband calls > > i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim > with 2 ports > when i try to call my phone number all time i receive message busy number > > > this error just with g1. > > with g2 there is no problem i can call without issue > > can anyone see the CLI and tell me what is the problem > > thanks and regards > > == Parsing '/etc/asterisk/asterisk.conf': Found > == Parsing '/etc/asterisk/extconfig.conf': Found > Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12 currently running on > SRVRADI > O (pid = 4147) > Verbosity is at least 3 > -- Executing [0661049303@agents:1] Set("SIP/223-00000021", > "CALLERID(number) > =520460587") in new stack > -- Executing [0661049303@agents:2] Dial("SIP/223-00000021", > "DAHDI/g1/066104 > 9303|30") in new stack > -- Requested transfer capability: 0x00 - SPEECH > -- Called g1/0661049303 > -- Moving call (DAHDI/3-1) from channel 3 to 2. > [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438 pri_fixup_principle: > Can't mo > ve call (DAHDI/3-1) from channel 3 to 2. It is > already in use. > [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558 > pri_find_fixup_principle: Spa > n 1: PRI requested channel 1/2 is > not available. > -- Hungup 'DAHDI/3-1' > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [0661049303@agents:3] Hangup("SIP/223-00000021", "") in > new sta > ck > == Spawn extension (agents, 0661049303, 3) exited non-zero on > 'SIP/223-0000002 > 1' > -- Executing [h@agents:1] GotoIf("SIP/223-00000021", "0?3:2") in new > stack > -- Goto (agents,h,2) > -- Executing [h@agents:2] AHEventsProxy("SIP/223-00000021", > "MSG_TYPE_TERMIN > ATE_CALL::::1382377407") in new stack > AHEventsProxy: Channel [SIP/223-00000021]. Data > [MSG_TYPE_TERMINATE_CALL::::138 > 2377407] > -- chan is SIP/223-00000021 > AHEventsProxy: Send To CtiServer: socket:[89]. > message:[41,1382377407^^^^stcrpb > x^~] > -- Executing [h@agents:3] Hangup("SIP/223-00000021", "") in new stack > == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-00000021' > -- SIP/224-00000020 is ringing > SRVRADIO*CLI> > Disconnected from Asterisk server > Executing last minute cleanups > > > > > > > -- > > Dog is my Co-pilot > > > -- > _____________________________________________________________________ > -- 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