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CUBE dial-peer, problem selecting outbound dial-peer.

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Flemming Ortvald
Contributor
Contributor

‎07-28-2022 03:48 AM

I am unable to determine why my configuration doesn't work. CUBE configuration attached.

Interesting dial-peers are 134, 13401 and 13402.

And "voice class e164-pattern-map 134"

Outbound call are met with this error:

1148: Jul 28 07:46:52.049: //277647/706703800000/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.64.101.121:5060;branch=z9hG4bK21ac30219d9146
From: <sip:204060@10.64.101.121>;tag=5978248~f0410c5e-29ee-481c-be08-28a5ead52bb5-44474760
To: <sip:+4525446346@10.168.2.20>;tag=B35B2EEA-113C
Date: Thu, 28 Jul 2022 07:46:52 GMT
Call-ID: 70670380-2e213eec-21a509-7965400a@10.64.101.121
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Warning: 399 10.168.2.20 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=1
Session-ID: 244091f000105000a000dccec1f13442;remote=e429979460aa57819547f879979f538d
Content-Length: 0

Show dialplan gives these results:

data4uctcube02#show dialplan dialpeer 134 numb +4525446346
Macro Exp.: +4525446346
No match, result=1(MORE_DIGITS_NEEDED)

data4uctcube02#show dialplan numb +4525446346
Macro Exp.: +4525446346

VoiceOverIpPeer13401
peer type = voice, system default peer = FALSE, information type = voice,
description = `*** SOS Test CUBE to TDC ***',
tag = 13401, destination-pattern = `',
destination e164-pattern-map tag = 134 status = valid,
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 = 13401, 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 = enabled,
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 0 oldAddrFamily 0
mailbox selection policy: none
type = voip, session-target = `',
session server-group = `34' status = valid,
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 = enabled,
IPv6 UDP checksum = disabled
session-protocol = sipv2, session-transport = udp,
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=112, 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 = system
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 = disable
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
voice class sip profiles = 344
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 subscribe-notify-events = system,
voice class sip pass-thru content unsupp = system,
voice class sip pass-thru content sdp = system,
voice class sip pass-thru content custom-sdp = system,
voice class sip copy-list = system,
voice class sip g729 annexb-all = system,
voice class sip early-offer forced = system,
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 call-route dest-route-string = 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 content sdp version increment = system,
voice class sip audio forced = system,
voice class sip bind control = enabled, 93.167.254.4,
voice class sip bind media = enabled, 93.167.254.4,
voice class sip bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip encap clear-channel = system,up-interval 0 down-interval 0 retry 0
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 nat = system,
voice class sip conn reuse = 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 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 = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Bandwidth CAC Accepted Calls = 0, Bandwidth CAC Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Last Disconnect Time = 0.
Matched: +4525446346 Digits: 4 Matched pattern: +45[2-9]....... Preference: 0
Target:

VoiceOverIpPeer13402
peer type = voice, system default peer = FALSE, information type = voice,
description = `*** SOS Test CUBE to TDC ***',
tag = 13402, destination-pattern = `',
destination e164-pattern-map tag = 134 status = valid,
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 = 13402, 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 = enabled,
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 0 oldAddrFamily 0
mailbox selection policy: none
type = voip, session-target = `',
session server-group = `34' status = valid,
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 = enabled,
IPv6 UDP checksum = disabled
session-protocol = sipv2, session-transport = udp,
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=112, 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 = system
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 = disable
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
voice class sip profiles = 3420
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 subscribe-notify-events = system,
voice class sip pass-thru content unsupp = system,
voice class sip pass-thru content sdp = system,
voice class sip pass-thru content custom-sdp = system,
voice class sip copy-list = system,
voice class sip g729 annexb-all = system,
voice class sip early-offer forced = system,
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 call-route dest-route-string = 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 content sdp version increment = system,
voice class sip audio forced = system,
voice class sip bind control = enabled, 93.167.254.20,
voice class sip bind media = enabled, 93.167.254.20,
voice class sip bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip encap clear-channel = system,up-interval 0 down-interval 0 retry 0
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 nat = system,
voice class sip conn reuse = 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 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 = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Bandwidth CAC Accepted Calls = 0, Bandwidth CAC Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Last Disconnect Time = 0.
Matched: +4525446346 Digits: 4 Matched pattern: +45[2-9]....... Preference: 0
Target:

 

fortvald
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1 Accepted Solution

Accepted Solutions

Go to solution
b.winter
VIP Advisor VIP Advisor
VIP Advisor

‎07-28-2022 04:22 AM

If you would match DP 134 as your incoming dial-peer, then it would take voice class dpg 340, to select the outgoing dial-peers.

But voice class dpg 340 is empty.

View solution in original post

3 Replies 3

Go to solution
b.winter
VIP Advisor VIP Advisor
VIP Advisor

‎07-28-2022 04:18 AM - edited ‎07-28-2022 04:19 AM

Could you post the debug outputs of following debugs:

debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

Just looking at the config, doesn't help, because it doesn't include a call obviously. So all the relevant details about calling and called number, ... are missing.

0 Helpful

Go to solution
b.winter
VIP Advisor VIP Advisor
VIP Advisor

‎07-28-2022 04:22 AM

If you would match DP 134 as your incoming dial-peer, then it would take voice class dpg 340, to select the outgoing dial-peers.

But voice class dpg 340 is empty.

Go to solution
Flemming Ortvald
Contributor
Contributor

‎07-28-2022 05:16 AM

@b.winter Thank you, I obviously missed that.

fortvald
0 Helpful
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