[Asterisk-Users] Dialing pstn-asterisk

Matthias Leeb
2004-09-09T01:09:44Z

Hello list

When i'm trying to dial into our pstn the following errors occure:

 -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack
Sep  9 10:02:22
WARNING[59409]: channel.c:1901 ast_request: No channel type registered for
''
Sep  9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create
channel of type ''
  == Everyone is busy/congested at this time
    -- Executing Congestion("SIP/snomsip-dbd0", "") in new
stack
  == Spawn extension (default, 02100, 2) exited non-zero on
'SIP/snomsip-dbd0'

When i type

metis:~# lsmod
Module                  Size  Used by    Not tainted
apm                     8892   0  (unused)
wct1xxp                12384   1
zaptel                177408   6  [wct1xxp]
soundcore               3556   0  (autoclean)
eepro100               17104   1
af_packet              11432   1
rtc                     5368   0  (autoclean)
ext2                   30400   1  (autoclean)
ide-disk                6592   2  (autoclean)
ide-probe-mod           7968   0  (autoclean)
ide-mod               129420   2  (autoclean) [ide-disk ide-probe-mod]
ext3                   56544   0  (autoclean)
jbd                    34968   0  (autoclean) [ext3]
unix                   13316  12  (autoclean)

Everything seems to be allright. Here is a part of my extensions.conf:

CONSOLE=Console/dsp
TRUNK=Zap/g1

ignorepat => 0
exten => _0.,1, Dial(${TRUNK}/${EXTEN:1}) exten => _0.,2,Congestion

Has anybody got some hints for me?

Beste regards

matthias