[Asterisk-Users] Problems with app_transfer
Alexandru Coseru
alex_spam at distinctgroup.net
Fri Mar 19 03:50:06 MST 2004
Hello... I can't make the app_transfer application to work:
I have 2 extensions in the same context : 9612 , which plays a prompt , and 9613 which tranfers a call to 9612... The transfer fails with timeout...
*CLI>
-- Accepting call from '890003' to '9612' on channel 17, span 1
Mar 19 12:45:29 DEBUG[131081]: chan_zap.c:1099 zt_enable_ec: Enabled echo cancellation on channel 17
-- Executing Answer("Zap/17-1", "") in new stack
-- Executing Playback("Zap/17-1", "demo") in new stack
*CLI>
-- Accepting call from '890003' to '9613' on channel 18, span 1
Mar 19 12:44:14 DEBUG[131081]: chan_zap.c:1099 zt_enable_ec: Enabled echo cancellation on channel 18
-- Executing Answer("Zap/18-1", "") in new stack
-- Executing Transfer("Zap/18-1", "9612") in new stack
Mar 19 12:44:24 WARNING[589842]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default'
[CDR] Zap/18-1 ('890003' -> '9613') Dur: 10s Bill: 10s Disp: ANSWERED Flags: DOCUMENTATION Account: []
/etc/asterisk/extension.conf:
............
exten => 9612,1,Answer
exten => 9612,2,Playback(demo)
exten => 9613,1,Answer
exten => 9613,2,Transfer(9612)
.............
show dialplan gives me that:
[ Context 'alex' created by 'pbx_config' ]
'9612' => 1. Answer() [pbx_config]
2. Playback(demo) [pbx_config]
'9613' => 1. Answer() [pbx_config]
2. Transfer(9612) [pbx_config]
Can somebody tell me what I'm doing wrong ?
Regards
Alex
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040319/d3a5611a/attachment.htm
More information about the asterisk-users
mailing list