[Asterisk-Users] Transfer call error

asterisk183 asterisk183 at yahoo.it
Wed Nov 30 07:57:57 MST 2005


 When I call a internal Sip telephone, the calling transfer to Teleohone external, but Asterisk show this error:
 
 Executing Dial("SIP/201-1e2a", "ZAP/g1/3472543320|60") in new stack
     -- Requested transfer capability: 0x00 - SPEECH
     -- Called g1/3472543320
 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 2: Red Alarm
 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 2
 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1
 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8458 pri_dchannel: pri_shutdown
 Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 2
 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1
 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 1: No Alarm
 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 1
 Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 1
     -- Hungup 'Zap/1-1'
   == No one is available to answer at this time (1:0/0/0)
     -- Executing Hangup("SIP/201-1e2a", "") in new stack
   == Spawn extension (local, 203, 2) exited non-zero on 'SIP/201-1e2a'
 
 
 My extension.conf:
 
 exten => 203,1,Dial(ZAP/g1/3472543320,60)
 exten => 203,2,Hangup
 
 Why?
 
 Thanks
 
 

		
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