Analog line Fax to an another number
HI - I have an analog fax line 2580 now that we want to forward it to 3600 another fax line. I am not sure if I am doing this right or not (8.6.2 version) in the directory number configuration I check the box Forward all and entered the full phone number of the new fax line, in the CSS I entered the CFwdAll search space. When do DNA to see if call goes through I get Match Result Blockthispattern, route block cause unallocated number. Not sure why I get this! On the phone configuration I have the old number as a analog phone in line 1, I added a line two as the new 3600. What am i missing?
thank you Murali!
I unchecked the Forward all - did not relize it would go to voice mail.
What I meant about line 1 and line2 is the 2nd line to the fax would be for the 3600 number. But sitting thinking about it the fax only has one line so that will not work either.
any thoughts on how to forward the 2580 to go to 3600? We would like to retire this block of DID's.
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FAX Analog line for the Router
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example : EVM-HD-8FXS/DID ... etc
Thanks
Regards,
KittenAre you connecting 22 analog faxes (FXS), or are you connected to 22 analog POTS lines (FXO)?
EVM can only do 16 max FXS or 8 max FXS + 6 max FXO.
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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 disabledhostname 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] -
Analog Lines to E1 DID/DOD configuration need for CCME
Dear,
I have one of our remote site have the following configuration:
* 2 X CCME running on C3845.
* 1 X CUE installed on one of the C3845
* Around 70 IPT (7910, 7941 & 7941) registres in the CCME.
* 8 * analog lines connected to FXO ports that provide land lind connectivity for the site.
* 4 * FXS ports to provide FAX.
and becasue of unstability issue with the out/ingoing call we requested 4 X voice E1 DID/DOD. and want to know what is the required configuration to consider in migrating my remote site.
regardsHello, thanks for your responses helped me very much.
I have the commands and errors completes:
When I enter the command:
ds0-group 0 timeslots 1-15,17-25 type r2-digital r2-compelled ani into the controller e1, I receive this error:
% Not enough DSP resources available to configure ds0-group 0 on controller E1 0/0/0
% The remaining dsp resources are enough for 20 time slots.
% For current codec complexity, 1 extra dsp(s) are required to create this voiceport.
This the inventory in the router:
GW-VOSS(config-controller)#do sh inv
NAME: "2821 chassis", DESCR: "2821 chassis" PID: CISCO2821 , VID: V05 , SN: FTX1336AHFS
NAME: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1" PID: VWIC2-1MFT-T1/E1 , VID: V01 , SN: FOC133129NF
NAME: "2nd generation two port FXO voice interface daughtercard on Slot 0 SubSlot 1", DESCR: "2nd generation two port FXO voice interface daughtercard"PID: VIC2-2FXO , VID: V02 , SN: FOC13332NDZ
NAME: "3rd generation two port FXS DID voice interface daughtercard on Slot 0 SubSlot 2", DESCR: "3rd generation two port FXS DID voice interface daughtercard" PID: VIC3-2FXS/DID , VID: V01 , SN: FOC132526S0
NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP S IMM with four DSPs" PID: PVDM2-64 , VID: V01 , SN: FOC13341ZE1.
I have the transcoding:
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 15
associate application SCCP
I have CUCM 7.0, Contac Center 6.0, SIP trunk, 1 E1.
I need another PVDM card? The router can be have two PVDM card? Which PVDM card you recommend ?
Regards -
CCME analog line configuration
Hi
I have problem with analog line which is connected to voice port 0/1/0. Outgoing calls working fine, but when somebody is calling from outside there is normal ringing sound but none of telepfones is ringing, problem is with number 0618181143. Part of router configuration:
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp feature access-code
isdn switch-type basic-net3
password encryption aes
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
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
no update-callerid
sip-profiles 100
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice class sip-profiles 100
request INVITE sip-header From modify "(<.*:)(.*@)" "<sip:xxx@"
request INVITE sip-header Remote-Party-ID remove
request CANCEL sip-header From modify "(<.*:)(.*@)" "<sip:xxx@"
voice register global
max-dn 56
max-pool 14
voice translation-rule 4
rule 1 /5990/ /101/
voice translation-rule 5
rule 1 /5995/ /101/
voice translation-rule 1111
rule 1 /^1../ /0618975990/
rule 2 /^1../ /0618975095/
voice translation-rule 1112
rule 1 /^9\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\2/
voice translation-rule 2001
rule 1 /0618975095/ /101/
voice translation-rule 2010
rule 1 /^100/ /0618181143/
rule 2 /^101/ /0618181143/
rule 3 /^102/ /0618181143/
rule 4 /^103/ /0618181143/
rule 5 /^104/ /0618181143/
rule 6 /^105/ /0618181143/
rule 7 /^106/ /0618181143/
rule 8 /^107/ /0618181143/
voice translation-profile AA_Profile
translate called 2001
voice translation-profile Biuro_Called_5
translate called 5
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile FROMSIP
translate calling 5
translate called 3
voice translation-profile Fax_Called_4
translate called 4
voice translation-profile IN_Profile
translate called 2001
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
voice translation-profile TOSIP
translate calling 1
translate called 2
voice translation-profile to_analog_1143
translate calling 2010
voice-card 0
interface FastEthernet0/1.1
description $FW_INSIDE$
encapsulation dot1Q 1 native
ip address 192.168.0.2 255.255.255.0
no ip unreachables
no ip proxy-arp
ip flow ingress
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.100
description $FW_INSIDE$
encapsulation dot1Q 100
ip address 10.10.0.1 255.255.255.0
no ip unreachables
no ip proxy-arp
ip flow ingress
ip nat inside
ip virtual-reassembly
interface BRI0/1/0
no ip address
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
shutdown
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/1
no ip address
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
shutdown
isdn switch-type basic-net3
isdn point-to-point-setup
ip local pool SDM_POOL_1 192.168.99.1 192.168.99.20
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.0.3
ip http server
ip http authentication local
ip http secure-server
ip http path flash:
ip access-list extended tunnel2
deny ip any any log
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 192.168.0.0 0.0.0.255
access-list 2 remark Auto generated by SDM Management Access feature
access-list 2 remark SDM_ACL Category=1
access-list 2 permit 192.168.20.0 0.0.0.255
access-list 2 permit 192.168.0.0 0.0.0.255
access-list 106 remark Auto generated by SDM Management Access feature
access-list 106 remark SDM_ACL Category=1
access-list 106 permit ip 192.168.20.0 0.0.0.255 any
access-list 106 permit ip 192.168.0.0 0.0.0.255 any
route-map SDM_RMAP_1 permit 1
match ip address 103
tftp-server flash:cnu75.8-3-1-22.sbn
tftp-server flash:cvm42sccp.8-3-1-22.sbn
tftp-server flash:cvm45sccp.8-3-1-22.sbn
tftp-server flash:cvm75sccp.8-3-1-22.sbn
tftp-server flash:dsp42.8-3-1-22.sbn
tftp-server flash:dsp45.8-3-1-22.sbn
tftp-server flash:dsp75.8-3-1-22.sbn
tftp-server flash:jar42sccp.8-3-1-22.sbn
tftp-server flash:jar45sccp.8-3-1-22.sbn
tftp-server flash:jar75sccp.8-3-1-22.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term45.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:term75.default.loads
tftp-server flash:APPS-1.0.4.SBN
tftp-server flash:CP7921G-1.0.4.LOADS
tftp-server flash:GUI-1.0.4.SBN
tftp-server flash:SYS-1.0.4.SBN
tftp-server flash:TNUX-1.0.4.SBN
tftp-server flash:TNUXR-1.0.4.SBN
tftp-server flash:WLAN-1.0.4.SBN
tftp-server DistinctiveRingList.xml
tftp-server RingList.xml
tftp-server flash:AreYouThereF.raw
tftp-server flash:Bass.raw
tftp-server flash:CallBack.raw
tftp-server flash:Chime.raw
tftp-server flash:Classic1.raw
tftp-server flash:Classic2.raw
tftp-server flash:ClockShop.raw
tftp-server flash:Drums2.raw
tftp-server flash:FilmScore.raw
tftp-server flash:HarpSynth.raw
tftp-server flash:Jamaica.raw
tftp-server flash:KotoEffect.raw
tftp-server flash:MusicBox.raw
tftp-server flash:Piano1.raw
tftp-server flash:Piano2.raw
tftp-server flash:Pop.raw
tftp-server S00105000100.sbn
tftp-server flash:cmterm_7920.4.0-02-00.bin
tftp-server flash:apps41.8-4-1-23.sbn
tftp-server flash:cnu41.8-4-1-23.sbn
tftp-server flash:cvm41sccp.8-4-1-23.sbn
tftp-server flash:dsp41.8-4-1-23.sbn
tftp-server flash:jar41sccp.8-4-1-23.sbn
tftp-server flash:SCCP41.8-4-2S.loads
tftp-server flash:apps11.8-4-1-23.sbn
tftp-server flash:cnu11.8-4-1-23.sbn
tftp-server flash:cvm11sccp.8-4-1-23.sbn
tftp-server flash:dsp11.8-4-1-23.sbn
tftp-server flash:jar11sccp.8-4-1-23.sbn
tftp-server flash:SCCP11.8-4-2S.loads
tftp-server flash:term61.default.loads
tftp-server flash:term41.default.loads
radius-server host 192.168.0.11 auth-port 1645 acct-port 1646 key 7 094E411B120A00010005
control-plane
voice-port 0/0/0
compand-type a-law
cptone PL
timeouts ringing infinity
description FAX
station-id name FAX
station-id number 108
caller-id enable
voice-port 0/0/1
compand-type a-law
cptone PL
timeouts ringing infinity
description Magazyn
station-id name Magazyn
station-id number 107
caller-id enable
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/2/0
compand-type a-law
cptone PL
timeouts ringing infinity
connection plar opx 100
description FXOport0
voice-port 0/2/1
compand-type a-law
cptone PL
timeouts ringing infinity
connection plar 200
description FXOport1
caller-id enable
voice-port 0/2/2
compand-type a-law
cptone PL
timeouts ringing infinity
connection plar 200
description FXOport2
caller-id enable
voice-port 0/2/3
compand-type a-law
cptone PL
timeouts ringing infinity
connection plar opx 200
description FXOport3
caller-id enable
mgcp fax t38 ecm
sccp ccm 10.10.0.1 identifier 2 version 4.0
sccp ccm group 2
associate ccm 2 priority 1
dial-peer cor custom
name international
dial-peer cor list call-international
member international
dial-peer voice 3001 voip
destination-pattern 3..
session protocol sipv2
session target ipv4:192.168.30.1
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
destination-pattern 108
port 0/0/0
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number .%
direct-inward-dial
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number .%
direct-inward-dial
port 0/1/1
dial-peer voice 53 pots
description ** BRI pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 910T
port 0/1/0
prefix 10
no sip-register
dial-peer voice 54 pots
corlist outgoing call-international
description ** BRI pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 912T
port 0/1/0
prefix 12
no sip-register
dial-peer voice 55 pots
description ** BRI pots dial-peer **-Emergency dial-peer
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 9112
port 0/1/0
forward-digits 3
no sip-register
dial-peer voice 56 pots
description ** BRI pots dial-peer **-Emergency dial-peer
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 112
port 0/1/0
forward-digits 3
no sip-register
dial-peer voice 11 pots
translation-profile outgoing to_analog_1143
destination-pattern 9T
port 0/2/0
dial-peer voice 10 pots
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
destination-pattern 9T
port 0/1/0
dial-peer voice 1 pots
destination-pattern 107
port 0/0/1
dial-peer voice 12 pots
destination-pattern 8...
port 0/2/1
forward-digits 4
dial-peer voice 13 pots
destination-pattern 9T
port 0/2/2
dial-peer voice 14 pots
destination-pattern 9T
port 0/2/3
telephony-service
video
em logout 0:0 0:0 0:0
max-ephones 36
max-dn 100
ip source-address 10.10.0.1 port 2000
auto assign 10 to 19
auto assign 5 to 8 type anl
calling-number initiator
service phone videoCapability 1
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message Borkowski - Gwiazdzista
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
cnf-file location flash:
load 7914 S00105000300
load 7911 SCCP11.8-4-2S
load 7961 SCCP41.8-4-2S
load 7961GE SCCP41.8-4-2S
time-zone 23
time-format 24
date-format dd-mm-yy
voicemail 222
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh vivaldi.au
multicast moh 239.10.16.16 port 2000
web admin system name admin secret 5 $1$02Lp$hrXGO0/qAD9vTsB5YyNUU0
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp 7960 Dec 10 2009 22:07:58
ephone-dn 35 dual-line
number 101 no-reg primary
label 101 G.Hofman
description G.Hofman
name G.Hofman
call-forward all 102
huntstop channel
no huntstop
hold-alert 30 originator
ephone-dn 36 dual-line
number 102 no-reg primary
label 102 A.Nowacki
description A.Nowacki
name A.Nowacki
huntstop channel
no huntstop
hold-alert 30 originator
ephone-dn 37 dual-line
number 103 no-reg primary
label 103 M.Pospieszna
description M.Pospieszna
name M.Pospieszna
huntstop channel
no huntstop
hold-alert 30 originator
ephone-hunt 1 sequential
pilot 100 secondary 1143
list 102, 103, 101
preference 0 secondary 7
timeout 20, 20, 20
I'm quite newbie with voice staff and I don't know how to analyze this issue. Please helpIf you dial 100 from a registered phone does the hunt group work and ring extensions 101, 102, 103?
Have you tried changing the connection plar for 0/2/0 to a DN on a registered phone to take any hunt group issues out of the equation?
Do the other FX0 ports work when dialing in? If you call into 0/2/1, 0/2/2, 0/2/3 does it ring extension 200?
Verify the call is coming into 0/2/0 using:
show voice port summary
debug voip ccapi inout
Could also plug an analog phone into the cable going to 0/2/0. (Connect phone to the wire from the phone company, not to the port on the router). -
Hello All,
When I call from analog line to PSTN, call is going out normally.
But after IVR telling me to dial extension number I can't dial.
Regards
BahlulAre you sure the phone upgraded the firmware to 9.4(1)?
Here's the 200 OK to the phone when the call is answered by IVR.
Aug 6 06:46:18.263: //1857/2DF031058BB2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Aug 6 06:46:18.687: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
This is when the call is disconnected.
Aug 6 06:46:33.335: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 11.11.11.49:5060;rport;branch=z9hG4bKPjGm5IPiXB-KDALUp8RiHMyGhIys0h1vEy
Max-Forwards: 70
From: "Mammadova Svetlana" <sip:[email protected]>;tag=f6d651d6-848e-4716-953c-c6927083f389
To: sip:[email protected];tag=DE91894-218B
Call-ID: 13684f3e-fa5e-4ab6-8939-880026b52655
CSeq: 12871 BYE
User-Agent: Cisco-CP3905/9.2.1
RTP-RxStat: Dur=15,Pkt=575,Oct=98728,LatePkt=0,LostPkt=0,AvgJit=2,VQMetrics="MLQK=3.1396;MLQKav=2.7148;MLQKmn=2.0000;MLQKmx=3.2783;MLQKvr=0.95;CCR=0.2269;ICR=0.0497;ICRmx=0.7383;CS=7;SCS=6"
RTP-TxStat: Dur=15,Pkt=279,Oct=12413
Content-Length: 0
We can see here that the User-agent is 3905 with 9.2.1. Can you check the firmware on the phone?
Also under voice register global, did you mention "load 3905 <9.4.1 loads file>"? -
Hello,
We have a UC540, we bought 10 analog line and about 30 DID or DDI. What we want to achieve is if for example, one DDI number call out, will it correlate to the CLI and be shown in the receiver's party mobile phone the DID number instead of the analog line number? Is it possible for UC540 to do that?
Regards,
MiguelCheck with your telco because normally analog DID circuits are unidirectional, that is, receive calls only.
If you want everthing to work easy, solid and with full features, use ISDN or SIP circuits, not analog of any kind. -
How do I go about transferring my iPhone 4s to another number on my plan?
I want to transfer my iPhone 4s to another number on my account and that phone to this number. In what order should I perform this task and how do I deal with the transfer of contacts. The other device is a LG ENV and has been backed up using vzbackup. The existing iPhone has been backed up to the cloud but will use a different apple ID when transferred.
Use a different phone and call *611 for them to switch them so you can keep the insurance if you have any.
-
TS4006 how do I add another number to my icloud?
I need to locate another phone that is not listed on my icloud how do I add another number?
hi,you have to sign-in into the icloud setting on your device with your apple id to turn on that "FIND MY IPHONE" function....
-
Question about the sensor... just got my 4s yesterday after screwing up my 3 with the laterd version update. EVery call I have been on has either changed to speaker, called another number or ended the call or activated facetime, which I have turned off. never had this trouble with my 3...I don't even want to talk to anyone on this phone! Is the sensor bad? That is what the AT&t rep suggested.
Restore as new... if the problem still continues then there is a hardware issue.
If it stops after a restore as new, then the issue is with the backup the device is currently setup with. -
How to extend analog line using VoIP with SPA8800
I would like to use a SPA8800 to make an analog line available to a VoIP phone (because at the location of the phone only LAN is available and no phone line). This is, when the line rings, the VoIP phone should ring. It would be nice to have if the VoIP phone could also use the line to dial out.
I already have the SPA8800, furthermore I have a SPA941 phone.
PSTN ----- line ----- SPA8800 ----- Ethernet ----- SPA941
Can you give advice on the necessary configuration for the SPA8800 for this setup? The few guides I could found were all for the other way around (Internet ----- SPA8800 ----- analog phone).Sastra;
If the trigger signal you need is TTL type, you can use one of the general purpose counters available at the board to do that.
You just need to configure the counter to do retriggerable single pulse generation, and have the reference signal that clocks the analog inputs routed to the gate of that counter. That will generate a pulse at the event of an active edge at the counter gate.
Hope this helps.
Filipe A.
Applications Engineer
National Instruments -
Sound from headphones and analog line output?
Is there a way to get sound our of the Mac Pro from the analog line output and the headphones at the same time?
dennisThank you Malcolm it works well. The only thing I found out is when doing it this way Audio Hijack Pro has to be in the record mode and on pause or files of what you are listening to will be created, eating up hard drive space.
But it works!
Dennis -
Best practice about dial-peer creating when using analog lines
Hi,
I am trying to find out what is the best practice when creating dial-peer for analog lines on CME, should I use trunk group or create separate dial-peer for each FXO ports? If I use trunk group, is there any advantage ( lesser dial-peer) or disadvantage?
Thanks!The advantage of trunk groups is that a single dial peer can point to for instance PSTN, rather then multiple dialpeers, with varying preference, each pointing to a separate FXO. Funtionally I can't see much difference. So I guess it also comes down to personal preference.
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Hello everyone;
I have an issue with two ports on my cisco VG 3745, in fact the analog lines connected in these ports are instable, that means that the phone user usualy can't use their analog line ( busy tone when they try to call PSTN ).
i notice that when i call on these analog line, the VG port does'nt light up , but if i unplug the line from the VG and put it back and call, the port gets green and the user can use his line.
Any one hhas an idea, about what it could be caused by?
For more information, don't hesitate to ask.
Thanks for your time.
Regards.
CaméliaHello Carlo;
Yes my VG is integrated in a network containing the CUCM ( both are connected to one core switch ) the signaling protocole is H.225 while my VG is a cisco 3745 H.323.
the config of the port in problem is :
voice-port 1/0/1
supervisory disconnect dualtone mid-call
supervisory custom-cptone
input gain 6
output attenuation -4
cptone FR
timeouts initial 0
timeouts interdigit 4
timeouts call-disconnect 0
timeouts ringing 5
timeouts wait-release 5
connection plar opx 1000
impedance complex2
description Manager
caller-id enable
thanks.
Regards. -
What hardware i need to buy if i want to connect 8 analog lines
Hi, i need to connect in a branch office one router supporting :
8 analog lines
1 ethernet interface
1 serial interface
and support security issues.
what model in the 2600 family router i need to buy?
should i need to buy one network module NM-HDV2?
if i need to buy the NM-HDV2,may i connect two VIC2-4FXO?
ThanksYou would need
2611XM chassis
NM-HD-2V card
VIC2-4FXO
VIC2-4FXO
WIC-1T
Appropriate IOS with voice and security feature sets.
The NM-HDV2 card has digital ports and only one VIC slot, so would not handle 8 analogue lines.
Best option is to contact your local Cisco office and speak to the account team to confirm if this system is best suited to your requirements.
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