Hi there,
on the System is following installed:
SuSE 2.6.27.39-0.2-default #1 SMP 2009-11-23 12:57:38 +0100 x86_64 x86_64 x86_64 GNU/Linux
asterisk 1.6.1.13
dahdi-linux = 2.2.1
dahdi-tools = 2.2.1
following hardware:
Intel Atom 330 Board
A400P with one FXS Green card.
if i am start the dahdi channel driver, the connected analog telefon are under power. so far so good.
Than i start asterisk and in the asterisk-dahdi module says following:
== Registered application 'DAHDISendKeypadFacility'
== Parsing '/etc/asterisk/chan_dahdi.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
-- Registered channel 1, FXO Kewlstart signalling
-- Registered channel 1, FXO Kewlstart signalling
-- Automatically generated pseudo channel
[Jan 30 23:51:04] WARNING[5103]: chan_dahdi.c:15084 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23.
[Jan 30 23:51:04] WARNING[5103]: chan_dahdi.c:15084 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Jan 30 23:51:04] WARNING[5103]: chan_dahdi.c:15084 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35.
[Jan 30 23:51:04] WARNING[5103]: chan_dahdi.c:15084 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Jan 30 23:51:04] WARNING[5103]: chan_dahdi.c:15084 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47.
== Registered channel type 'DAHDI' (DAHDI Telephony Driver w/PRI w/SS7)
== Manager registered action DAHDITransfer
== Manager registered action DAHDIHangup
== Manager registered action DAHDIDialOffhook
* == Manager registered action DAHDIDNDon
== Manager registered action DAHDIDNDoff
== Manager registered action DAHDIShowChannels
== Manager registered action DAHDIRestart
Loaded chan_dahdi.so => (DAHDI Telephony w/PRI & SS7)
asterisk command: "dahdi show status":
DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
asterisk comamnd:
pseudo default default In Service
1 analog de default In Service
so far, so good.
But if i am abolish from the analog telefon the hearer, i hear nothing.
If i am try to dial out or something else on the analog telefone, nothing happend.
If i am try to dial the analog telefone with a SIP Client following happend in asterisk:
*CLI> == Using SIP RTP CoS mark 5
-- Executing [123@analog:1] Dial("SIP/telefon-00000001", "dahdi/g1/123456") in new stack
[Jan 30 23:59:19] WARNING[8025]: chan_dahdi.c:2501 dahdi_call: cidspill already exists??
-- Called g1/123456
The line is ringing, but you can not hear anything!
The line respand about the get-off the line.
Nothing happening in asterisk at this status if i am hangup the line with the analog telephone.
If i am hangup the line with the SIP client following happening in asterisk:
*CLI> -- Hungup 'DAHDI/1-1'
== Spawn extension (analog, 123, 1) exited non-zero on 'SIP/telefon-00000001'
Now i have the situation that the card didn't respond anymore in asterisk and the analog telephone is dead, without power?
If i take now the same call from the sip client to the analog telephone follwing happening in asterisk:
*CLI> == Using SIP RTP CoS mark 5
-- Executing [123@analog:1] Dial("SIP/telefon-00000002", "dahdi/g1/123456") in new stack
[Jan 31 00:10:34] WARNING[8061]: chan_dahdi.c:2539 dahdi_call: Unable to ring phone: Device or resource busy
-- Couldn't call g1/123456
-- Hungup 'DAHDI/1-1'
== Everyone is busy/congested at this time (0:0/0/0)
-- Auto fallthrough, channel 'SIP/telefon-00000002' status is 'CHANUNAVAIL'
I think this is a signalling problem, but i tried any kind of signalling out. With no result.
Did you know this problem ? and can you help me?
Work this card with dahdi 2.2.1 properly ?
with this ducumentaion had i am configured the system:
http://www.henkelm.de/computer/asterisk/tdm400p.html
thx in advance for a answear.
best regards
Patrick
i] Last edited by pemer at 2010-2-8 17:09 [/i