Keine CallerID auf ISDN-Tel über SIP

michaelxy

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Hallo,

über ISDN bekomme ich Nummern auf ein Siemens ISDN-Telefon angezeigt. Sendtext geht auch auf das Telefon.
Ruft jemand über SIP an, kommt immer "unbekannt". Die CallerIDs sind aber in den Logs vorhanden.

Warum zeigt mir das Telefon die CallerID nicht an ?


Gruß
Michael
 
Zuletzt bearbeitet:
Hallo michaelxy,

kannst du dzu mal die CLI-Ausgabe anhängen?
 
Hi chaos2000,

falls man da was erkennen kann ?:

Connected to Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j currently running on esel (pid = 22456)
Verbosity was 0 and is now 3
Jan 4 17:38:57 DEBUG[22457]: acl.c:176 ast_apply_ha: ##### Testing 217.10.79.9 with 192.168.0.0
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:636 ast_sip_ouraddrfor: Target address 217.10.79.9 is not local, substituting externip
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:4187 transmit_register: Scheduled a registration timeout # 2
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:842 __sip_ack: Stopping retransmission on '[email protected]' of Request 102: Found
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:842 __sip_ack: Stopping retransmission on '[email protected]' of Request 103: Found
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:6812 handle_response: Registration successful
Jan 4 17:38:57 DEBUG[22457]: chan_sip.c:6814 handle_response: Cancelling timeout 2
Jan 4 17:39:07 DEBUG[22457]: acl.c:176 ast_apply_ha: ##### Testing 217.10.79.9 with 192.168.0.0
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:636 ast_sip_ouraddrfor: Target address 217.10.79.9 is not local, substituting externip
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:5559 check_user_full: Setting NAT on RTP to 4
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:7382 handle_request: Check for res for
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:1597 update_user_counter: is not a local user
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:[email protected];ftag=c3709051a39f4a89aeb982e90ac3966b;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:[email protected];ftag=c3709051a39f4a89aeb982e90ac3966b;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:194.97.54.97;ftag=c3709051a39f4a89aeb982e90ac3966b;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:194.97.59.146;ftag=c3709051a39f4a89aeb982e90ac3966b;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:194.97.54.97;ftag=c3709051a39f4a89aeb982e90ac3966b;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4646 build_route: build_route: Record-Route hop: <sip:84.163.31.82:20166;lr=on>
Jan 4 17:39:07 DEBUG[22457]: chan_sip.c:4671 build_route: build_route: Contact hop: <sip:[email protected]:20166>
-- Executing System("SIP/freenet.de-0816e000", "echo "04012006-17:39:07: "michaelxyxy" <michaelxyxy> rief an." | mail -s VOIP-Anrufbenachrichtigung mail@... ") in new stack
-- Executing GotoIfTime("SIP/freenet.de-0816e000", "23:50-7:00|mon-thu|*?7") in new stack
-- Executing GotoIfTime("SIP/freenet.de-0816e000", "00:30-7:00|fri-sat|*?7") in new stack
-- Executing GotoIfTime("SIP/freenet.de-0816e000", "23:30-7:00|sun|*?7") in new stack
-- Executing Dial("SIP/freenet.de-0816e000", "Zap/g1/3527737|60|tTr") in new stack
Jan 4 17:39:07 DEBUG[22457]: app_dial.c:519 dial_exec: SIMPLE DIAL (NO URL)
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/3527737
Jan 4 17:39:08 DEBUG[22457]: chan_zap.c:8382 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1
-- Zap/1-1 is making progress passing it to SIP/freenet.de-0816e000
Jan 4 17:39:08 DEBUG[22457]: app_dial.c:367 wait_for_answer: Dunno what to do with control type 15
-- Zap/1-1 is ringing
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:2680 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:2178 zt_hangup: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:2315 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:1421 zt_disable_ec: disabled echo cancellation on channel 1
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:2592 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:1360 update_conf: Updated conferencing on 1, with 0 conference users
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:2674 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Jan 4 17:39:12 DEBUG[22457]: chan_zap.c:1421 zt_disable_ec: disabled echo cancellation on channel 1
-- Hungup 'Zap/1-1'
Jan 4 17:39:12 DEBUG[22457]: app_dial.c:1072 dial_exec: Exiting with DIALSTATUS=CANCEL.
== Spawn extension (pbx-trunk, 3527737, 5) exited non-zero on 'SIP/freenet.de-0816e000'
Jan 4 17:39:12 DEBUG[22457]: chan_sip.c:7851 sipsock_read: Failed to grab lock, trying again...
Jan 4 17:39:12 DEBUG[22457]: cdr_addon_mysql.c:178 mysql_log: cdr_mysql: inserting a CDR record.
Jan 4 17:39:12 DEBUG[22457]: cdr_addon_mysql.c:197 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2006-01-04 17:39:07','\"michaelxyxy\" <michaelxyxy>','michaelxyxy','3527737','pbx-trunk', 'SIP/freenet.de-0816e000','Zap/1-1','Dial','Zap/g1/3527737|60|tTr',5,0,'NO ANSWER',3,'')
Jan 4 17:39:12 DEBUG[22457]: chan_sip.c:7851 sipsock_read: Failed to grab lock, trying again...
Jan 4 17:39:12 DEBUG[22457]: chan_sip.c:1728 sip_hangup: update_user_counter() - decrement inUse counter
Jan 4 17:39:12 DEBUG[22457]: chan_sip.c:1597 update_user_counter: is not a local user
Jan 4 17:39:12 DEBUG[22457]: chan_sip.c:842 __sip_ack: Stopping retransmission on '[email protected]' of Response 2: Found
 
scheinen ja wirklich infos übertragen zu werden.
Ich kann mir nur noch vorstellen, dass es ein paar einstellungen in der zapata.conf dies unterdrücken.
 
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