RE: [AsteriskBrasil] digium card não funciona

Fabio Vasco fabiohvgomes em hotmail.com
Domingo Agosto 13 23:59:32 BRT 2006


Ralph,
/etc/zaptel.conf e /etc/asterisk/zapata.conf
zttool
lspci -v
Deixa a gente ver isso...
Que Linux vc. está usando?
Abraços,
FHV

>From: "Ralph Liebessohn" <ralphliebessohn em gmail.com>
>Reply-To: asteriskbrasil em listas.asteriskbrasil.org
>To: asteriskbrasil em listas.asteriskbrasil.org
>Subject: [AsteriskBrasil] digium card não funciona
>Date: Fri, 11 Aug 2006 16:50:28 +0000
>
>boa tarde,
>
>
>pessoal, estou com um probleminha aqui com uma placa da digium.
>Inicialmente comecei com o asterisk em um computador comum sempron com 
>placa
>mãe pcchips e coloquei pra funcionar uma TE406P. Até ai tudo lindo.
>Agora colocando o asterisk pra rodar num P4 Dual com placa mãe intel
>(d101ggc) tudo aparentemente funciona, o asterisk compila, roda, liga de 
>sip
>pra sip, os modulos zaptel compilam, carregam, configuram, mas na hora de
>ligar mesmo PAM !
>Recebo isso.
>
>Connected to Asterisk 1.2.7.1 currently running on astk (pid = 4174)
>Verbosity is at least 99
>Core debug is at least 99
>    -- Executing Dial("SIP/12347-1574", "ZAP/g1/21226551") in new stack
>    -- Requested transfer capability: 0x00 - SPEECH
>    -- Called g1/21226551
>  == Primary D-Channel on span 1 down
>Aug 11 12:05:11 WARNING[4219]: chan_zap.c:2298 pri_find_dchan: No 
>D-channels
>available!  Using Primary channel 16 as D-channel anyway!
>    -- Hungup 'Zap/1-1'
>  == Everyone is busy/congested at this time (1:0/0/1)
>    -- Executing Hangup("SIP/12347-1574", "") in new stack
>  == Primary D-Channel on span 1 down
>Aug 11 12:05:14 WARNING[4219]: chan_zap.c:2298 pri_find_dchan: No 
>D-channels
>available!  Using Primary channel 16 as D-channel anyway!
>
>Saída /var/log/asterisk/full:
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: Allocating new SIP dialog for
>ff3e5bab-e0089797 em 192.168.11.192 - INVITE (With RTP)
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: **** Received INVITE (5) - Command
>in SIP INVITE
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: * SIP extension value: 0 for call
>ff3e5bab-e0089797 em 192.168.11.192
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: Setting NAT on RTP to 524288
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: = Found Their Call ID:
>ff3e5bab-e0089797 em 192.168.11.192 Their Tag 53fbac9bdac9fb67o1
>Our tag: as0681a10d
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: **** Received ACK (6) - Command in
>SIP ACK
>Aug 11 11:53:30 DEBUG[4323] chan_sip.c: Stopping retransmission on '
>ff3e5bab-e0089797 em 192.168.11.192' of Response 101: Match
>Found
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: = Found Their Call ID:
>ff3e5bab-e0089797 em 192.168.11.192 Their Tag 53fbac9bdac9fb67o1
>Our tag: as0681a10d
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: **** Received INVITE (5) - Command
>in SIP INVITE
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: * SIP extension value: 0 for call
>ff3e5bab-e0089797 em 192.168.11.192
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: Setting NAT on RTP to 524288
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: Checking SIP call limits for device
>12347
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: Updating call counter for incoming
>call
>Aug 11 11:53:31 DEBUG[4323] chan_sip.c: build_route: Contact hop: Linksys 2
><sip:12347 em 192.168.11.192:5061>
>Aug 11 11:53:31 DEBUG[4315] chan_sip.c: Checking device state for peer 
>12347
>Aug 11 11:53:31 DEBUG[4315] devicestate.c: Changing state for SIP/12347 -
>state 2 (In use)
>Aug 11 11:53:31 DEBUG[4610] pbx.c: Launching 'Dial'
>Aug 11 11:53:31 VERBOSE[4610] logger.c:     -- Executing
>Dial("SIP/12347-d99a", "ZAP/g1/21226551") in new stack
>Aug 11 11:53:31 DEBUG[4610] chan_zap.c: Using channel 1
>Aug 11 11:53:31 DEBUG[4610] channel.c: Not copying variable
>STACK-default-21226551-1.
>Aug 11 11:53:31 DEBUG[4610] channel.c: Not copying variable SIPCALLID.
>Aug 11 11:53:31 DEBUG[4610] channel.c: Not copying variable SIPUSERAGENT.
>Aug 11 11:53:31 DEBUG[4610] channel.c: Not copying variable SIPDOMAIN.
>Aug 11 11:53:31 DEBUG[4610] channel.c: Not copying variable SIPURI.
>Aug 11 11:53:31 VERBOSE[4610] logger.c:     -- Requested transfer
>capability: 0x00 - SPEECH
>Aug 11 11:53:31 VERBOSE[4610] logger.c:     -- Called g1/21226551
>Aug 11 11:53:31 DEBUG[4610] channel.c: Set channel Zap/1-1 to read format
>slin
>Aug 11 11:53:31 DEBUG[4610] channel.c: Set channel SIP/12347-d99a to write
>format slin
>Aug 11 11:53:31 DEBUG[4610] channel.c: Set channel SIP/12347-d99a to read
>format slin
>Aug 11 11:53:31 DEBUG[4610] channel.c: Set channel Zap/1-1 to write format
>slin
>Aug 11 11:53:31 DEBUG[4611] app_queue.c: Device 'SIP/12347' changed to 
>state
>'2' (In use)
>Aug 11 11:53:31 DEBUG[4315] devicestate.c: Changing state for Zap/1 - state
>2 (In use)
>Aug 11 11:53:31 DEBUG[4315] devicestate.c: Changing state for Zap/1 - state
>2 (In use)
>Aug 11 11:53:31 DEBUG[4612] app_queue.c: Device 'Zap/1' changed to state 
>'2'
>(In use)
>Aug 11 11:53:31 DEBUG[4613] app_queue.c: Device 'Zap/1' changed to state 
>'2'
>(In use)
>Aug 11 11:53:31 DEBUG[4610] rtp.c: Ooh, format changed from unknown to ulaw
>Aug 11 11:53:34 VERBOSE[4364] logger.c:   == Primary D-Channel on span 1
>down
>Aug 11 11:53:34 WARNING[4364] chan_zap.c: No D-channels available!  Using
>Primary channel 16 as D-channel anyway!
>Aug 11 11:53:34 DEBUG[4610] channel.c: Hanging up channel 'Zap/1-1'
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: zt_hangup(Zap/1-1)
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: Set option AUDIO MODE, value: ON(1)
>on Zap/1-1
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: Hangup: channel: 1 index = 0, 
>normal
>= 22, callwait = -1, thirdcall = -1
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: disabled echo cancellation on
>channel 1
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: Set option TDD MODE, value: OFF(0)
>on Zap/1-1
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: Updated conferencing on 1, with 0
>conference users
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: Set option AUDIO MODE, value: 
>OFF(0)
>on Zap/1-1
>Aug 11 11:53:34 DEBUG[4610] chan_zap.c: disabled echo cancellation on
>channel 1
>Aug 11 11:53:34 VERBOSE[4610] logger.c:     -- Hungup 'Zap/1-1'
>Aug 11 11:53:34 VERBOSE[4610] logger.c:   == Everyone is busy/congested at
>this time (1:0/0/1)
>Aug 11 11:53:34 DEBUG[4610] app_dial.c: Exiting with 
>DIALSTATUS=CHANUNAVAIL.
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Launching 'Hangup'
>Aug 11 11:53:34 VERBOSE[4610] logger.c:     -- Executing
>Hangup("SIP/12347-d99a", "") in new stack
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Spawn extension (default,21226551,2)
>exited non-zero on 'SIP/12347-d99a'
>Aug 11 11:53:34 DEBUG[4315] devicestate.c: Changing state for Zap/1 - state
>0 (Unknown)
>Aug 11 11:53:34 DEBUG[4614] app_queue.c: Device 'Zap/1' changed to state 
>'0'
>(Unknown)
>Aug 11 11:53:34 DEBUG[4610] cdr_pgsql.c: cdr_pgsql: inserting a CDR record.
>Aug 11 11:53:34 DEBUG[4610] cdr_pgsql.c: cdr_pgsql: SQL command executed:
>INSERT INTO cdr (calldate,clid,src,dst,dcontext,ch
>annel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield)
>VALUES ('2006-08-11 1
>1:53:31','"Linksys 2" <12347>','12347','21226551','default',
>'SIP/12347-d99a','Zap/1-1','Hangup','',3,0,'NO ANSWER',2,'12347'
>,'1155308011.4','')
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '"Linksys 2" <12347>'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '12347'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '21226551'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'default'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'SIP/12347-d99a'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'Zap/1-1'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'Hangup'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '(null)'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '2006-08-11 11:53:31'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '(null)'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '2006-08-11 11:53:34'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '3'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '0'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'NO ANSWER'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is 'BILLING'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '12347'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '1155308011.4'
>Aug 11 11:53:34 DEBUG[4610] pbx.c: Function result is '(null)'
>Aug 11 11:53:34 DEBUG[4610] channel.c: Hanging up channel 'SIP/12347-d99a'
>Aug 11 11:53:34 DEBUG[4610] chan_sip.c: Hangup call SIP/12347-d99a, SIP
>callid ff3e5bab-e0089797 em 192.168.11.192)
>Aug 11 11:53:34 DEBUG[4610] chan_sip.c: update_call_counter(12347) -
>decrement call limit counter
>Aug 11 11:53:34 DEBUG[4610] chan_sip.c: Updating call counter for incoming
>call
>Aug 11 11:53:34 DEBUG[4610] chan_sip.c: AST hangup cause 16 (no match found
>in SIP)
>Aug 11 11:53:34 DEBUG[4315] chan_sip.c: Checking device state for peer 
>12347
>Aug 11 11:53:34 DEBUG[4315] devicestate.c: Changing state for SIP/12347 -
>state 1 (Not in use)
>Aug 11 11:53:34 DEBUG[4615] app_queue.c: Device 'SIP/12347' changed to 
>state
>'1' (Not in use)
>
>Ele simplesmente liga, chia e corta a ligação imediatamente, a partir daí 
>as
>ligações pelo E1 não se realizam mais.
>
>NOTICE[4981]: app_dial.c:1029 dial_exec_full: Unable to create channel of
>type 'ZAP' (cause 34 - Circuit/channel congestion)
>  == Everyone is busy/congested at this time (1:0/1/0)
>
>Receber ligação de forma nenhuma.
>As mesmas configurações (trocando o hd de máquina) funciona perfeitamente.
>Já tentei usar também com outra placa intel e um p4 HT sem dual core, mas a
>falha permaneceu.
>Já vi algumas coisas parecidas na lista mas nada que chegue ao amago dessa
>questao.
>Alguem já viu algo parecido? Tem alguma idéia do que possa ser feito para o
>sistema completo funcionar?
>
>
>--
>Ralph Liebessohn
>ICQ: 74835911
>Skype: liebessohn


>_______________________________________________
>LIsta de discussões AsteriskBrasil.org
>AsteriskBrasil em listas.asteriskbrasil.org
>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>
>_______________________________________________
>Acesse o  wiki AsteriskBrasil.org:
>http://www.asteriskbrasil.org

_________________________________________________________________
MSN Messenger: converse com os seus amigos online. 
http://messenger.msn.com.br



Mais detalhes sobre a lista de discussão AsteriskBrasil