Linksys SPA3102, busy line on incoming calls

Hi there,
I'm trying to set this device up in this specific scenario:
- PSTN: 1st incoming call, the user gets it.
- PSTN: 2nd incoming call, but the 1st call is still ongoing.
Actual result: 2nd incoming call obtains ring tones, and he/she thinks there is nobody home
Desired result: 2nd incoming call obtains busy line tone, and he/she thinks there is somebody home and is busy
Is this device able to do this?, any tip about which configuration parameter should I change?
Many thanks, Ibon.

Sorry, wrong forum
Posting in the right one

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Analog line (FXO) Incoming calls getting connected after 3 rings

     HI,
we are having 4 Analog line (FXO)...Every time when callers call the number they hear 3 rings & after that call frwds to AA or any extension.
In show voice port summary, we can see that voice port is getting connect at the first ring but after 3 rings only phone rings.
here is the o/p of voice port.
Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar
Connection Number is 250
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for AE
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming): INCOMING_CallerID_PROFILE
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabled

hostname VGUAE001
no aaa new-model
clock timezone UAE 4 0
ip cef
ip domain name yourdomain.com
no ipv6 cef
multilink bundle-name authenticated
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
voice-card 0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class cause-code 1
no-circuit
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 3265
rule 1 // /9\1/
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 50
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
license udi pid CISCO2901/K9 sn FCZ173992Z8
hw-module pvdm 0/0
hw-module pvdm 0/1
username cisco privilege 15 secret 4 opjnnkXqCr4kCOa9DuALcNpBOMetBAc/usnpSWADsCI
username godiva privilege 15 secret 4 cH8b8z.ioYu/pMv/AKuEcBd/f6g9v/vm/s3aXeqUAd6
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.31.2 255.255.255.0
ip helper-address 192.168.31.11
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.31.2
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 192.168.31.1
control-plane
voice-port 0/0/0
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
groundstart auto-tip
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/1
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/2
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/3
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 250
caller-id enable
mgcp profile default
dial-peer voice 2000 voip
destination-pattern 2..
session target ipv4:192.168.31.11
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 10 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fire**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 997
forward-digits all
no sip-register
dial-peer voice 11 pots
trunkgroup ALL_FXO
description **CCA*UAE*International Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 900T
forward-digits all
no sip-register
dial-peer voice 12 pots
trunkgroup ALL_FXO
description **CCA*UAE*Eitisalat**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9101
forward-digits all
no sip-register
dial-peer voice 13 pots
trunkgroup ALL_FXO
description **CCA*UAE*Water or electrical emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 971
forward-digits all
no sip-register
dial-peer voice 14 pots
trunkgroup ALL_FXO
description **CCA*UAE*Police and emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 999
forward-digits all
no sip-register
dial-peer voice 15 pots
trunkgroup ALL_FXO
description **CCA*UAE*National area codes**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-579].......
forward-digits all
no sip-register
dial-peer voice 16 pots
trunkgroup ALL_FXO
description **CCA*UAE*Mobile Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 90[5-6][0-7].......
forward-digits all
no sip-register
dial-peer voice 17 pots
trunkgroup ALL_FXO
description **CCA*UAE*toll-free**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]00T
forward-digits all
no sip-register
dial-peer voice 18 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fixed Line Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-8]T
forward-digits all
no sip-register
dial-peer voice 19 pots
trunkgroup ALL_FXO
description **CCA*UAE*808**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9808T
forward-digits all
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/0/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/0/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/0/2
no sip-register
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/0/3
no sip-register
Debug vpm signal:
Nov 23 19:31:31.556: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Nov 23 19:31:31.556: htsp_timer - 125 msec
Nov 23 19:31:31.684: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Nov 23 19:31:31.684: htsp_timer - 10000 msec
Nov 23 19:31:31.684: htsp_timer3 - 5600 msec
Nov 23 19:31:31.684: [0/0/0] htsp_start_caller_id_rx:Mode BELLCORE. Alerting 0x1
Nov 23 19:31:31.684: htsp_start_caller_id_rx create dsp_stream_manager
Nov 23 19:31:31.684: [0/0/0] htsp_dsm_create_success returns 1
Nov 23 19:31:33.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:33.604: fxols_ringing_not
Nov 23 19:31:33.604: htsp_timer_stop
Nov 23 19:31:33.604: htsp_timer - 10000 msec
Nov 23 19:31:37.284: htsp_process_event: [0/0/0, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Nov 23 19:31:37.284: htsp_timer_stop3
Nov 23 19:31:37.516: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
Nov 23 19:31:39.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:39.604: fxols_ringing_not
Nov 23 19:31:39.604: htsp_timer_stop
Nov 23 19:31:39.604: htsp_timer_stop3
Nov 23 19:31:39.604: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Nov 23 19:31:39.604: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Nov 23 19:31:39.604: [0/0/0] get_local_station_id calling num= calling name= calling time=11/23 23:31 orig called=
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3CE27724, Call Info(
   Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=50, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: cc_get_feature_vsa count is 1
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218944,feature_id:83
Nov 23 19:31:39.604: //83/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown))
Nov 23 19:31:39.608: [0/0/0] htsp_dsm_close_done
Nov 23 19:31:39.608: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Nov 23 19:31:39.608: fxols_wait_setup_ack:
Nov 23 19:31:39.608: [0/0/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   Event=0x22ACD828
Nov 23 19:31:39.608: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 250
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230F9C10
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 83 with tag 50 to app "_ManagedAppProcess_Default"
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=2000, Params=0x230FB0D0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination Pattern=2.., Called Number=250, Digit Strip=FALSE
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=B583C95F-53AC-11E3-8093-C8EEBDE4256A, Outgoing Dial-peer=2000
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x22847B14, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2000, Call Count On=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, tg_label_flag=1, Application Call Id=)
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: cc_get_feature_vsa count is 2
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218720,feature_id:84
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230FB080
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2000
Nov 23 19:31:39.612: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Nov 23 19:31:39.612: htsp_timer - 120000 msec
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGetMediaClassTag:
   media class tag 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_call_proceeding:
   Interface=0x22847B14, Progress Indication=NULL(0)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_delay_xport:
   CallInfo(delay xport=TRUE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Interface=0x22847B14, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)htsp_alert_notify
Nov 23 19:31:39.628: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x5, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x5, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.672: //84/B583C95F8093/CCAPI/ccIsInfoRingback:
   Returning dpRingBack=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Interface=0x22847B14, Data Bitmask=0x1, Progress Indication=NULL(0),
   Connection Handle=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x7, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Network, Params=0x0, Call Id=83
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID1=0x53, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID2=0x54, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Conference Id=0xFFFFFFFF, Call Id1=83, Call Id2=84, Tag=0x0
Nov 23 19:31:39.700: htsp_call_bridged invoked
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0xFFFFFFFF
Nov 23 19:31:39.700: //84/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Nov 23 19:31:39.700: cc_api_get_xcode_stream : 4819
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=84)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=83)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Nov 23 19:31:39.700: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=0x1, Fax Rate=0x1, Fax Version:=0, Vad=0x1,
   Modem=0x2, Codec Bytes=20, Signal Type=3)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x21, Call Id1=83, Call Id2=84
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=0x1, Fax Rate=0x2, Fax Version:=0, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x1
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Nov 23 19:31:39.704: htsp_timer_stop
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Id=83
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Entry(Context=0x230F9C10)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Nov 23 19:31:39.932: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x22847B14, Call Id=84
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
   Conference Id=0x21, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Nov 23 19:31:48.860: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: htsp_timer_stop3
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x22847B14, Tag=0x0, Call Id=84,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: :cc_free_feature_vsa freeing 3D1B9898
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: vsacount in free is 1
Nov 23 19:31:48.884: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Nov 23 19:31:48.884: htsp_timer_stop
Nov 23 19:31:48.884: htsp_timer_stop2
Nov 23 19:31:48.884: htsp_timer_stop3
Nov 23 19:31:48.884: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 23 19:31:48.884: htsp_timer - 2000 msec
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3CE27724, Tag=0x0, Call Id=83,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: :cc_free_feature_vsa freeing 3D1B9978
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: vsacount in free is 0
Nov 23 19:31:49.156: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

Biz transaction type Incoming Call blocked for further biz transactions

Gurus:
When I go txn cic0 at CRM6.0, I get the message :
Business transaction type Incoming Call is blocked for further business transactions
Message no. CRM_ORDERADM_H501
I searched SDN and SSO but found no fix.
Could you tell me how to fix this?
Thanks!

Hi,
What must be hapening here is that you are accesing a IC profile, which has a default sales transaction assigned which is not active in customizing.
What is the source of this error, or better, is this happening with all users, is this result of any upgrade?
With these answers it will be easier for us to help you find the error. At first moment, it seems that the user you are logged in is not assigned to any position in PPOMA_CRM, or the position to which he is assigned doesn't have any IC profile, so the system logs in standard IC profile from SAP, and this IC Profile calls transaction type 0001 (Incoming Call) which probably you deactivated because you must have being using your own Z document types.
Regards,
Susana Messias

Linksys SPA3102 "flashing" and incoming call problem.

I tried to do as much research as I could before posting, however i'm stuck =/.
My Setup:
Phone Port on wall Split into three phone ports, one with a fax in it, one with a splitter, one empty.
From the splitter I have a line into a D-link modem and a line into the back of the SPA3102.
From the D-Link modem i have an ethernet cable plugged into the SPA3102 [im not using the inbuilt modem, should i be ?]
Also from the D-link modem i have a D-link wireless router.
Then I have a line into a set of Telstra V900 cordless phones from the SPA3102.
I also have another phone connected directly to the PSTN line in another room.
The problem is when a call is coming in on the PSTN line the phone connected directly to the PSTN line rings however the phones connected to through the SPA3102 don't register the call until much after the PSTN phone does. I'm not sure how to go about fixing that.
Also should I pick up on the wired PSTN handset, how do I go about switching to the wireless handset. "flashing" i think it's called.
NOTE: I'm in the market for new wireless phones anyway as the telstra V900 ones are crap, any recmomendations ?
Any help would be appreciated.

First I didn’t understand what you meant for “I’m not using the inbuilt modem” for the SPA-3102. Did you mean you set the SPA-3102 as a bridged?
If you are using a phone splitter where one of the 3 phone ports is connected to the PSTN portion of the SPA-3102 and the other phone splitter port is into another phone and someone calls the PSTN line, the phone connected to the splitter will really ring for it gets a share of the signal coming from the splitter. The phone connected to the phone port (FXS) port for the SPA-3102 will ring only if the PSTN signal has successfully transferred the call to the from the PSTN line registration to the actual Line 1 registration. If that is not happening, there are some registration settings you need to check in the SPA-3102. One important parameter is the registration status of the Line 1 and PSTN lines. You can check that under the Info section just make sure you are login as admin advanced login with the SPA-3102. You should be getting registration state “registered” confirming that the VoIP information you set in the SPA-3102 is valid. If registrations are correct, maybe you need to know is the PSTN Line is transferring the call to the VoIP line section. Setting up this portion is discussed in this link:
http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159&p_created=11686...
Hope this helps.

ISDN E1 PRI incoming calls busy out after 2nd b channel picks up

I have an ISDN E1 PRI line coming in from the Telco. D Channel comes up nicely and B channels accept calls, but only the first two calls. Once two callers are online, any further incoming calls get a busy tone. ISDN Debug data reports:
**ERROR**: call_incoming: Remote Protocol Error: No b channel assigned by switch, call id 0xCE
Hardware I'm using? Cisco 2650, 12.3(7T) and NM-HDV, VWIC-2MFT-E1-DI
Anybody please help?

Problem sorted - believe it or not, it was the telco. They changed their switch setting for the line to preferrable B channel - and voila!
Thanks,
Brian

SPA3102 can you process incoming PSTN calls without translation into VOIP?

On the SPA3102 is it possible to avoid translation of incoming calls into VOIP and out again. If so does anyone know the SPA3102 setting?
I realise that this would probably lose me quite a lot on incoming call processing options, but the echo on the PSTN line - revealed by the delay caused by the translation - is terrible, and I have tried most things to resolve it!
Many thanks in anticipation for any help you can give.
Mouse

try setting PSTN-To-VoIP Gateway Enable parameter under the PSTN tab to NO -- this should disable the gateway functionality for PSTN to VoIP
btw: try playing with Network Jitter Level under the PSTN tab to resolve the the echo
| isolate! isolate! isolate! |

An incoming call from VoIP. The SPA122 generate Dial Tone after the far end hung up rather busy tone.

Hi All!
I have a problem with the SPA122 telephony adapter, uncorrectly process the subscriber signaling at the end of the call.
1) Outbound call from FXS port SPA122 . When a remote caller hangs up first , the subscriber SPA122 Reorder Tone played with a delay specified in the Reorder Delay. This circuit is working properly.
2 ) Incoming call from VoIP to the SPA122. When a remote caller hangs up first , subscriber on the FXS port of the SPA122 hears silence ~ 3-4 seconds , then SPA122 plays Dial Tone, as if he had just picked up the phone and he 's going to call . No signal lights out (Busy Tone or Reorder Tone) will not play .
Config is attached.
Model: SPA122, LAN, 2 FXS
Hardware Version: 1.0.0 Boot Version: 1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version: 1.3.2-XU (014) Jul 2 2013
Recovery Firmware: 1.0.2 (001)
WAN MAC Address: 6C:20:56:55:3A:B6
Host Name: SPA122
Domain Name: (none)
Serial Number: CCQ16450LG3
However, other VoIP terminals registered to Huawei, including older versions of the Linksys SPA2102 work in these scenarios correctly.
Where to kick it?

[2] is misconfiguration on your's side. You have CPC turned on, but no CPC capable device. Set CPC Duration to zero to turn off CPC.
By the way, wrong forum for your question. You should consider to move it to space.

Multi incoming calls (spa 525g - 5 lines)

Hello,
I got a SPA 525G2 color with 5 lines.
I don't know how correctly configurate into (web browser IP panel of my phone) to use incoming calls with it's rule :
When line 1 is busy, i would like that next incoming calls goes to the 4 other lines empty.
Cause actually, all incoming calls goes to line 1. When i'm ever on phone into line 1 you imagine well that's it's not easy to manage.
Thanks for your help.
Regards,
Denis

In the Phone tab, under Miscellaneous Line Key Settings, you can
configure line mapping. Each LED (line/extension) can hold two calls. You can
assign an extension to two LEDs. The first call always causes the assigned LED to
flash. Choose one of the following:
• Vertical first—The next LED on the phone flashes with the second incoming
call.
• Horizontal first—The same LED to flashes with the second incoming call.
Vertical First

PAP-2T, make incoming calls ring in both phones/Lines?

I have a PAP-2T. On this device I have connected two different phones, one connected to Line 1 and the other to Line 2. I have one SIP account with my provider. My provider has configured my account so that I can make two concurrent outgoing calls. My Lines are configured as the following;
Line 1:
Proxy: xxxx.sipserver.xxxx.xx
Outbound Proxy: <blank>
Register: Yes
Register Expires: 3600
Use Outbound Proxy: No
Use OB Proxy In Dialog: No
Make Call Without Reg: No
Ans Call Without Reg: No
Line 2:
Proxy: <blank>
Outbound Proxy: xxxx.sipserver.xxxx.xx
Register: No
Register Expires: 3600
Use Outbound Proxy: Yes
Use OB Proxy In Dialog: Yes
Make Call Without Reg: Yes
Ans Call Without Reg: No
Now, when I receive an incoming call, the phone connected to Line 1 rings. If I configure Line 2 like this,, Ans Call Without Reg: Yes,, the phone on Line 2 rings but not the one on Line 1.
My question is,, can I make both phones ring when I receive an incoming call?
If so, how do I configure the ATA to make that happen?

Cat2k1 what kind of probs do you hve with your phones ?
- assume it's a DECT phone, try to change the ring voltage to 75V and ring waveform to trapezoid - this works better with problematic DECT phones
- assume you're having an old analog phones, make sure the PAP2T's Line1 socket is only connected via 2wire interconnect cable (4wire cable on Line1 plus certain analog phones is a trouble-maker)
- assume you use multiple analog phones on one line, you may consider increase the FXS power from 3 to let's say 5
- assume you're having an old analog phone, it's ability to ring (if all the SIP comm works OK) is also affected by the FXS port impedance which must (shall) fit the phone's one.
====
On the other hand, shall you have problems with SIP communication between PAP2T an SIP operator, you shall enable NAT traversal and NAT keeplalive for the first on PAP2T settings
Message Edited by Scorpio-cz on 12-24-2008 10:50 PM

How can I be notified of incoming calls while on-line?

I am currently using dial-up and I'm wondering if there is already an application in my hard drive which would allow me to be notified when I was receiving an incoming call while I'm online? (I'm looking to get rid of my 2nd phone line which I only use for internet connection.)
Any suggestions? I also know there are services out there: CallWave, Pagoo, BuzMe but I hear they will bombard your waiting callers with advertisements. Isn't there also a device which you can plug in which lets you know if the incoming call is fax, phone or whatever? Any suggestions, advice, recommendations would be very helpful.
Thank you!
lablady

tinpan,
Thanks. I haven't heard of any of these programs. I'll also check out those websites. But, I'm not concerned about caller ID, per se. I don't care if I know who is calling me before I pick up the phone, I just want to be alerted that a phone call is coming in while I'm online. I thought I saw some sort of "box" that could let you know this but it was years ago and I don't know if it's available anymore nor can I remember the name. If I recall correctly, it was about $40. I was hoping it was cheaper by now.
And, I wasn't sure if Apple actually had this capability on any of their Macs but wanted to check. Sure would be an added plus, don't you think?
Thanks again!
lablady

N71 freezes when calling busy line

Hi,
I am witnissing since several months the fact, that when I call a busy line, I hear the occupied or line busy signal, then my N71 by itself closes the connection but the busy tone keeps on beeping for up to 3 minutes and all other phone functions freeze. no matter what I do, the phone does not react at all. Very recently I updated the OS, but no change so far.
Does anybody have an idea, what this could be?
thanks for your help
rolfester

I had this problem for years with my N95, N96, E71. To fix this simply turn off automatic redialling. I appreciate this info is probably no more use to you now as you posted the problem in 2007, but it is an ongoing problem so I thought I'd put up the solution for others to see!

Switched over to MGCP from H323, no incoming calls fast busy

Hello, I'm on the network side crossing over to the Voice side. We replaced a 3825 Voice Router at a branch office with a 2921. The 3825 was setup with a T1 and had a PRI connected to the FXO ports. The 2921 is now connected via fiber and TAC helped get the router registered to the Call Manager. I'm trying to match up the old dial peers on the new router. I can't make out going or receive incoming calls, I get a This Call Can't be completed at this time.
When it was on the T1, the branch office was using H323. Now that's connects to the same CUCM, it's on MGCP. Shouldn't the old Dial Peers work on the new router?

I had to configure the FXO ports for the DN to route the incoming calls. I learned just because the Call Manager configured the Voice Gateway as a client, you still must configure the FXO ports to route the main DID to a DN on the LAN.
Great advice, learning alot about telephony and VoIP.
Wish I had more experience troubleshooting the CUCM and DID portion, also learned the phone compnay doesn't turn up their ISDN switch until you configure your PBX. And a PRI testing isn't the same as the data T1, it's about having the Signaling channel turned up and or configured.
Lessons learned, wished I would have crossed over to VoIP earlier.

Unity Express - Incoming calls wont get voice mail

CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
I searched for another post which suggested the following commands:
telephony-service
call-forward pattern .T
voice service voip
allow connections from h323 to sip
I've double checked them and there's still something wrong.
Here's my current configuration:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
max-ephones 24
max-dn 24
ip source-address 192.168.20.1 port 2000
auto assign 1 to 24
system message Comtek
voicemail 3000
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
time-webedit
transfer-system full-consult
transfer-pattern 2...
transfer-pattern 3...
directory last-name-first
directory entry 2 2001 name Phone Two 7912
directory entry 3 2000 name Phone One 7970
ephone-dn 1 dual-line
number 2000 secondary 441833000000
call-forward busy 3000
call-forward noan 3000 timeout 10
no huntstop
ephone 1
no multicast-moh
device-security-mode none
mac-address 0017.0EF0.3642
type 7970
button 1:1
So pros, any suggestions?
Thanks

I made a new dial-peer to handle incoming calls as follows.
dial-peer voice 1000 voip
description Incoming SIP
translation-profile incoming SIPin
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
no vad
The translation-profile puts the call through to my 2000 extension.
This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
dur 00:00:00 tx:0/0 rx:0/0
Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
This is the "show call active voice brief" for an external incoming call when the call is established.
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
dur 00:00:02 tx:105/16800 rx:104/16640
IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
dur 00:00:02 tx:0/0 rx:105/16800
Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Not too sure where to go from here.

Display both company AND name in contacts, incoming calls, messages, etc

Is there no way to have both a person's full name, AND company appear on the display of contacts and incoming calls??
I have over 2,000 contacts in my address book, many from the same company, and many with the same (or similar) names.
When someone with a very common name calls or texts, it's impossible for me to remember who they are without listing their company as well. If I have both Sam Michaels and Samantha Mitchells in my address book, sometimes the only way to distinguish between the two is knowing that Sam works for Viacom and Samantha works for Universal.
I've heard that the only way to do this is to include the company name in the person's main "name" field. This is problematic because A) it would take me a million years to go through and personally edit all of my 2k+ contacts, and B) this is completely counter-intuitive of having the company/office/job title lines available as fields in the address book in the first place.
Does anyone know a way around this at all?? Apple really needs to find a way to fix this, as it's super inconvenient for business professionals. This is just another one of my many frustrations with Apple products being not intuitive for workers in the corporate world.

I'm not aware of a way to do this, however I'm certain that when my phone rings, it shows the person's name and company information that I have written in contacts. I've not taken time to check lately, but that seems to be the case for contacts that have a company name in them. Ones that do not have a company name show the person's name and then the label for the phone number they are using, such as home, work, mobile, etc.
Otherwise, feedback to Apple goes HERE, and then just click on the appropriate subject.

Linksys SPA3102, busy line on incoming calls

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