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No is incorrect response on 6945

chandogetrude
Level 1
Level 1

Cisco deskphone with model no 6945 can’t call mobile no with response “The number is incorrect” Note: If we call 90714588510 the extensions just show no 907 with above response Kindly advise

5 Replies 5

Jaime Valencia
Cisco Employee
Cisco Employee

Are you using CUCM?? CME? Version??

Does the call make it to the GW??

What troubleshooting have you done so far?

HTH

java

if this helps, please rate

We are using CME with IOS c3900-universalk9-mz.SPA.155-3.M1.bin

We have tried to update 6945 firmware

Note: The phones can reach their gateway as they have registered and can
calls normal extensions but just cannot call mobile no

This means that you have a wrong translation configured somewhere which strips the rest of the number and sends 907 to telco. 

 

Starting the tracing your call routing and see where the call is dropped. Are you using cucm or cme? How are you connected to telco?

TIB-SMR-CME-RTR#show dialplan number 90653927884
Macro Exp.: 90653927884

VoiceOverIpPeer900
peer type = voice, system default peer = FALSE, information type = voice,
description = `VOIP.CALLS.TO.TIB-TPA',
tag = 900, destination-pattern = `[9]..',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 900, Admin state is up, Operation state is up,
incoming called-number = `',
connections/maximum = 0/unlimited,
bandwidth/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (Via) =
Incoming (To) =
Incoming (From) =
Destination (Diversion) =
Destination (From) =
Destination (Referred-By) =
Destination (To) =
Destination (Via) =
Destination =
Destination route-string = None
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
outgoing LPCOR:
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = voip, session-target = `ipv4:192.168.4.3',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip media rsvp-pass DSCP = ef
ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,
ip defending Priority = 0, ip preemption priority = 0
ip policy locator voice:
ip policy locator video:
UDP checksum = disabled,
IPv6 UDP checksum = disabled
session-protocol = sipv2, session-transport = system,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video = best-effort,
req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,
dtmf-relay = rtp-nte,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124
MP4A-LATM=111, lmr_tone=0, nte_tone=0
h263+=118, h264=119
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = pass-through
fax-relay ecm enable
Fax Relay ans treatment disabled
Fax Relay ans enabled
Fax Relay SG3-to-G3 Enabled (by system configuration)
fax NSF = 0xAD0051 (default)
voice-class codec = 1
codec = g729r8, payload size = 20 bytes,
video codec = None
voice class codec = 1
voice class sip session refresh system
voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
text relay = disabled
Media Setting = forking (disabled) flow-through (global)stats-disconnect (disabled)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = disabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip tel-config = system,
voice class sip rel1xx = system,
tvoice class sip outbound-proxy = system,
voice class sip asserted-id = system,
voice class sip privacy = system,
voice class sip e911 = system,
voice class sip history-info = system,
voice class sip pass-thru headers = system,
voice class sip pass-thru subscribe-notify-events = system,
voice class sip pass-thru content unsupp = system,
voice class sip pass-thru content sdp = system,
voice class sip copy-list = system,
voice class sip anat = system,
voice class sip g729 annexb-all = system,
voice class sip early-offer forced = system,
voice class sip negotiate cisco = system,
voice class sip reset timer expires 183 = system,
voice class sip block 180 = system,
voice class sip block 181 = system,
voice class sip block 183 = system,
voice class sip preloaded-route = system,
voice class sip random-contact = system,
voice class sip random-request-uri validate = system,
voice class sip call-route p-called-party-id = system,
voice class sip call-route history-info = system,
voice class sip call-route url = system,
voice class sip privacy-policy send-always = system,
voice class sip privacy-policy passthru = system,
voice class sip privacy-policy strip history-info = system,
voice class sip privacy-policy strip diversion = system,
voice class sip bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip error-code-override options-keepalive failure = system,
voice class sip error-code-override call spike failure = system,
voice class sip error-code-override cac-bandwidth failure = system,
voice class sip encap clear-channel = system,
voice class sip send 180 sdp = system,
voice class sip map resp-code 181 = system,
voice class sip bind control = system,
voice class sip bind media = system,
voice class sip registration passthrough = System
voice class sip nat mode = System
voice class sip conn reuse = System
voice class sip authenticate redirecting-number = system,
voice class sip referto-passing = system,
voice class sip extension = system,
voice class sip contact-passing = system,
voice class sip requri-passing = system,
voice class sip content sdp version increment = system,
voice class phone proxy name: None
voice class phone proxy config: N/A
redirect ip2ip = disabled
local peer = false
probe disabled,
Secure RTP: system (use the global setting)
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0 rtcp_keepalive = system

voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 218745, Charged Units = 0,
Successful Calls = 11, Failed Calls = 118, Incomplete Calls = 0
Accepted Calls = 42, Refused Calls = 14,
Bandwidth CAC Accepted Calls = 0, Bandwidth CAC Refused Calls = 0,
Last Disconnect Cause is "1 ",
Last Disconnect Text is "unassigned number (1)",
Last Setup Time = 60978145.
Last Disconnect Time = 59980727.
Matched: 90653927884 Digits: 1 Matched pattern: [9].. Preference: 0
Target: ipv4:192.168.4.3

TIB-SMR-CME-RTR#

Hi,

Can you please post the entire config?

 

Thanks

 

Regards

 

 

Carlo

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