01-01-2011 10:06 AM - edited 03-16-2019 02:39 AM
Dear all,
I am facing problem in my cme 2800 router .
When anyone calls on coporate number it is hitting autoattendent number & transfered to reception. if the reception is not picking up the call the ip phone rings continously & if the enduser has disconnected the ip phone still rings, & there is no caller id it gives unknow number.
Solved! Go to Solution.
01-02-2011 02:00 PM
We see BELLCORE here...
Jan 2 08:45:17.611: htsp_start_caller_id_rx:BELLCORE
We should be seeing
htsp_start_caller_id_rx: ETSI_DTMF
in the debugs for ETSI DTMF
Can you please configure either of the following "BR", "SE", "DK", "IS",
"NL", "BE" to make the caller-id method ETSI_DTMF.
ie. add
voice-port x/y
cptone BR
shut
no shut
If that doesn't work
try caller-id alerting dsp-pre-alloc
under voice-port
shut/no shut
Collect
debug hpi all
debug vpm signal
debug voip ccapi inout
After each step.
HTH
PS:Rate Useful posts
01-02-2011 02:14 PM
Could you also connect an analog phone directly to the PSTN line and
check whether Caller-id is being displayed on the Phone.
Could you also find out whether PSTN sends the caller-id after 1 ring or
after 2 rings or is it after the line reversal(I would say it should be
the line reversal for ETSI DTMF, Saudi Arabia)
Get the debugs and info from the provider we should be good to go
regarding the Caller-id.
01-01-2011 07:54 PM
This seems to be a supervisory disconnect issue. Check the following document for the configurations required for supervisory disconnect signalling.
Understanding FXO Disconnect Problem:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Hope this helps. Please rate useful posts!
Thanks,
Jyothi
01-01-2011 08:02 PM
For the caller ID issue, please see this similar thread:
https://supportforums.cisco.com/message/306017#306017
Please remember to rate!
Thanks,
Jyothi
01-02-2011 12:02 PM
thanks for ur vaulable comments . today i captured vpm debug but find
the caller is not recieve failed as mentioned below than i enable voicemail caller id in telephone services but still showing smaw unknow number .
"et_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data."
01-02-2011 01:18 PM
I am not able to open the debugs.zip attached to the previous thread.
What country is does the CME deployment in?. Any idea what caller-id standard are you on the in the country(maybe provider dependant),
http://en.wikipedia.org/wiki/FSK_standards_for_use_in_Caller_ID_and_remote_metering
http://www.ainslie.org.uk/callerid/cli_faq.htm
Can you send across the output of "show voice port x/y"
01-02-2011 01:39 PM
thanks for the links
cme is deployed in saudi Arabia they are using esoteric DTMF system.below is vpm debug.
an 2 08:45:10.555: htsp_digit_ready(0/0/0): digit = A
Jan 2 08:45:17.483: htsp_process_event: [0/1/2, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Jan 2 08:45:17.483: htsp_timer - 125 msec
Jan 2 08:45:17.611: htsp_process_event: [0/1/2, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Jan 2 08:45:17.611: htsp_timer - 10000 msec
Jan 2 08:45:17.611: htsp_timer3 - 5600 msec
Jan 2 08:45:17.611: [0/1/2] htsp_start_caller_id_rx:BELLCORE
Jan 2 08:45:17.611: htsp_start_caller_id_rx create dsp_stream_manager
Jan 2 08:45:17.611: [0/1/2] htsp_dsm_create_success returns 1
Jan 2 08:45:18.507: htsp_process_event: [0/1/2, FXOLS_RINGING, E_DSP_SIG_0100]
Jan 2 08:45:18.507: fxols_ringing_not
Jan 2 08:45:18.507: htsp_timer_stop
Jan 2 08:45:18.507: htsp_timer - 10000 msec
Jan 2 08:45:22.427: htsp_process_event: [0/1/2, FXOLS_RINGING, E_DSP_SIG_0000]
Jan 2 08:45:23.211: htsp_process_event: [0/1/2, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Jan 2 08:45:23.211: htsp_timer_stop3
Jan 2 08:45:23.531: htsp_process_event: [0/1/2, FXOLS_RINGING, E_DSP_SIG_0100]
Jan 2 08:45:23.531: fxols_ringing_not
Jan 2 08:45:23.531: htsp_timer_stop
Jan 2 08:45:23.531: htsp_timer_stop3
Jan 2 08:45:23.531: [0/1/2] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Jan 2 08:45:23.531: [0/1/2] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Jan 2 08:45:23.531: [0/1/2] get_local_station_id calling num= calling name= calling time=01/02 11:45 orig called=
Jan 2 08:45:23.535: [0/1/2] htsp_dsm_close_done
Jan 2 08:45:23.535: htsp_process_event: [0/1/2, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Jan 2 08:45:23.535: fxols_wait_setup_ack:
Jan 2 08:45:23.535: htsp_timer - 6000 msec
Jan 2 08:45:23.535: htsp_process_event: [0/1/2, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_prochtsp_alert_notify
Jan 2 08:45:23.575: htsp_process_event: [0/1/2, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Jan 2 08:45:23.591: htsp_call_bridged invoked
Jan 2 08:45:23.595: htsp_process_event: [0/1/2, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Jan 2 08:45:23.595: [0/1/2] set signal state = 0xC timestamp = 0
Jan 2 08:45:23.595: htsp_timer_stop
Jan 2 08:45:23.595: htsp_process_event: [0/1/2, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Jan 2 08:45:31.407: htsp_digit_ready(0/1/2): digit = 1
Jan 2 08:45:35.647: htsp_digit_ready(0/1/2): digit = 2
Jan 2 08:45:36.127: htsp_digit_ready(0/1/2): digit = 2
Jan 2 08:45:36.727: htsp_digit_ready(0/1/2): digit = 2
Jan 2 08:45:37.387: htsp_digit_ready(0/1/2): digit = #
Jan 2 08:45:42.371: htsp_timer_stop3 htsp_setup_req
Jan 2 08:45:42.371: htsp_process_event: [50/0/36.1, EFXS_ONHOOK, E_HTSP_SETUP_REQ]efxs_onhook_setup
Jan 2 08:45:42.375: htsp_ephone_start_caller_id_tx calling num= calling name = called num=222 orig called num=
Jan 2 08:45:42.375: [50/0/36.1] set signal state = 0x0 timestamp = 0
Jan 2 08:45:42.375: efxs_onhook_setup: local target is available
01-02-2011 02:00 PM
We see BELLCORE here...
Jan 2 08:45:17.611: htsp_start_caller_id_rx:BELLCORE
We should be seeing
htsp_start_caller_id_rx: ETSI_DTMF
in the debugs for ETSI DTMF
Can you please configure either of the following "BR", "SE", "DK", "IS",
"NL", "BE" to make the caller-id method ETSI_DTMF.
ie. add
voice-port x/y
cptone BR
shut
no shut
If that doesn't work
try caller-id alerting dsp-pre-alloc
under voice-port
shut/no shut
Collect
debug hpi all
debug vpm signal
debug voip ccapi inout
After each step.
HTH
PS:Rate Useful posts
01-02-2011 02:14 PM
Could you also connect an analog phone directly to the PSTN line and
check whether Caller-id is being displayed on the Phone.
Could you also find out whether PSTN sends the caller-id after 1 ring or
after 2 rings or is it after the line reversal(I would say it should be
the line reversal for ETSI DTMF, Saudi Arabia)
Get the debugs and info from the provider we should be good to go
regarding the Caller-id.
01-03-2011 10:34 AM
Thanks alot dijohn
caller id is working fine now. after configuring Cptone as BE & line reversal for ETSI DTMF
01-03-2011 07:51 PM
Just want to have a seat & learning.
03-10-2011 12:03 AM
Hi dijohn ,
hope ur fine..
i have tested same configuation on anthor router in same country for caller id issue but no sucess i am attaching the debug please have alook & suggest .
03-10-2011 09:56 AM
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