Often when troubleshooting an issue, having a good test can take a while to get ready. With CSIM, an undocumented IOS command, it is possible to simulate an outbound call, originating from your voice gateway directly. Use the csim start dialstring hidden command to initiate simulated calls to whichever real-world E.164 number is desired. This allows you to determine whether you can properly go offhook, send digits, and complete a call to the destination phone.
2951-VGW#debug isdn q931
debug isdn q931 is ON.
2951-VGW#csim start 19105557245
csim: called number = 19105557245, loop count = 1 ping count = 0
Apr 13 22:19:42.179: ISDN Se0/1/0:23 Q931: Sending SETUP callref = 0x09CD callID = 0x8955 switch = primary-ni interface = User
Apr 13 22:19:42.179: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8 callref = 0x09CD
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Called Party Number i = 0x80, ‘19105557245’
Plan:Unknown, Type:Unknown
Apr 13 22:19:42.227: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x89CD
Channel ID i = 0xA98397
Exclusive, Channel 23
Apr 13 22:19:44.863: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x89CD
Progress Ind i = 0x8188 – In-band info or appropriate now available
Apr 13 22:19:52.376: ISDN Se0/1/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x89CD
Apr 13 22:19:52.376: ISDN Se0/1/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x09CD
csim err csimDisconnected recvd DISC cid(42503)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
Apr 13 22:19:57.144: ISDN Se0/1/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x89CD
Cause i = 0x8290 – Normal call clearing
Apr 13 22:19:57.144: ISDN Se0/1/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x09CD
Apr 13 22:19:57.156: ISDN Se0/1/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x89CDter no mo
Apr 13 22:19:58.741: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Despite the call working in this case CSIM will always display failed=1. There is no real explanation for this.
If the call does not complete, you could perform a “show dial-peer voice sum” to match dialed digits, or my favorite “show dialplan number”
2911-VGW#show dialplan num 19105557245
Macro Exp.: 19105557245
VoiceOverIpPeer1000
peer type = voice, system default peer = FALSE, information type = voice,
description = `SIP_OUT_11_DIGIT’,
tag = 1000, destination-pattern = `1[2-9]………’,
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 = 1000, Admin state is up, Operation state is up,
incoming called-number = `’, connections/maximum = 0/unlimited,
bandwidth/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = disabled,
URI classes:
Incoming (Request) =
Incoming (Via) =
Incoming (To) =
Incoming (From) =
Destination =
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 = `sip-server’,
technology prefix:
settle-call = disabled
ip media DSCP = cs5, ip media rsvp-pass DSCP = ef
ip media rsvp-fail DSCP = ef, ip signaling DSCP = cs4,
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,
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 = sip-notify,
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
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 = disable, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
Fax Relay ans enabled
Fax Relay SG3-to-G3 Enabled (by system configuration)
fax NSF = 0xAD0051 (default)
codec = g711ulaw, payload size = 160 bytes,
video codec = None
voice class codec = `’
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
voice class sip profiles = 1
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 url = system,
voice class sip rel1xx = system,
voice class sip anat = system,
voice class sip outbound-proxy = “system”,
voice class sip associate registered-number = 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 reset timer expires 183 = system,
voice class sip pass-thru headers = 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 g729 annexb-all = system,
voice class sip early-offer forced = enable,
voice calss sip delay-offer forced = disable,
voice class sip negotiate cisco = system,
voice class sip block 180 = system,
voice class sip block 183 = system,
voice class sip block 181 = 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 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 bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip encap clear-channel = system,
voice class sip error-code-override options-keepalive failure = system,
voice class sip error-code-override cac-bandwidth failure = system,
voice class sip calltype-video = false
voice class sip registration passthrough = System
voice class sip authenticate redirecting-number = system,
voice class sip referto-passing = system,
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
voice class perm tag = `’
Time elapsed since last clearing of voice call statistics never
Connect Time = 303893, Charged Units = 0,
Successful Calls = 37, Failed Calls = 9, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Bandwidth CAC Accepted Calls = 0, Bandwidth CAC Refused Calls = 0,
Last Disconnect Cause is “10 “,
Last Disconnect Text is “normal call clearing (16)”,
Last Setup Time = 510566383.
Last Disconnect Time = 510567472.
Matched: 19105557245 Digits: 2 Matched pattern: 1[2-9]………
Target: sip-server