Hallo,
ich versuche mit folgendem dialplan alle Telefone im Haus klingeln zu lassen wenn ein call ueber die 10 hereinkommt.
Das funktioniert auch.
Allerdings bekomme ich einen Fehler wenn einer der angerufenen Clients den call annimmt
ich versuche mit folgendem dialplan alle Telefone im Haus klingeln zu lassen wenn ein call ueber die 10 hereinkommt.
Code:
[ Context 'internal' created by 'pbx_config' ]
'10' (CID match '10') => 1. Dial(${AllRing},${RINGTIME_DOOR}) [pbx_config]
2. Hungup() [pbx_config]
'10' => 1. Dial(SIP/${EXTEN}) [pbx_config]
2. Hangup() [pbx_config]
'800' => 1. Answer() [pbx_config]
2. Wait(1) [pbx_config]
3. VoiceMailMain(100) [pbx_config]
4. Hangup() [pbx_config]
'_2XX' => 1. Dial(SIP/${EXTEN},${RINGTIME}) [pbx_config]
2. VoiceMail(100,u) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
Das funktioniert auch.
Allerdings bekomme ich einen Fehler wenn einer der angerufenen Clients den call annimmt
Code:
-- Executing [10@phones:1] Dial("SIP/10-00000006", "SIP/200&SIP/201&SIP/202&SIP/203&SIP/204,5") in new stack
[Mar 19 08:52:32] WARNING[100459][C-00000003]: app_dial.c:2520 int dial_exec_full(struct ast_channel *, const char *, struct ast_flags64 *, int *): Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Mar 19 08:52:32] WARNING[100459][C-00000003]: app_dial.c:2520 int dial_exec_full(struct ast_channel *, const char *, struct ast_flags64 *, int *): Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Mar 19 08:52:32] WARNING[100459][C-00000003]: app_dial.c:2520 int dial_exec_full(struct ast_channel *, const char *, struct ast_flags64 *, int *): Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
-- Called SIP/200
-- Called SIP/204
-- SIP/204-00000008 is ringing
-- SIP/200-00000007 is ringing
> 0x8becf09000 -- Strict RTP learning after remote address set to: 192.168.20.205:41402
-- SIP/204-00000008 answered SIP/10-00000006
-- Channel SIP/204-00000008 joined 'simple_bridge' basic-bridge <1c2037a8-9b2d-42c5-b272-45d3e3f85637>
-- Channel SIP/10-00000006 joined 'simple_bridge' basic-bridge <1c2037a8-9b2d-42c5-b272-45d3e3f85637>
> Bridge 1c2037a8-9b2d-42c5-b272-45d3e3f85637: switching from simple_bridge technology to native_rtp
> Remotely bridged 'SIP/10-00000006' and 'SIP/204-00000008' - media will flow directly between them
-- Got SIP response 486 "Busy here" back from 192.168.20.205:45356
> 0x8becec0000 -- Strict RTP learning after remote address set to: 192.168.20.218:7078