02-24-2010 03:00 AM - edited 03-15-2019 09:31 PM
When i have the hunt pilot configured the calls are not completed.
if i change the hunt pilot number to not match the DNIS then i am able to complete the call.
i don´t understand why the router is matching the DNIS.
in attach i send a debug i did. (ccapi inout; isdn q931 and voice translation)
voice translation-rule 1
rule 1 /0/ /236200230/
rule 2 /1/ /4419/
rule 3 /2/ /4420/
rule 4 /3/ /4420/
rule 5 /4/ /4410/
rule 6 /5/ /4420/
rule 7 /6/ /4420/
rule 8 /7/ /4420/
rule 9 /8/ /4420/
rule 10 /9/ /4420/
!
!
voice translation-profile DDI
translate called 1
!
!
voice-port 0/2/0:15
translation-profile incoming DDI
!
dial-peer voice 2200 voip
destination-pattern 236200230
voice-class codec 2
session protocol sipv2
session target ipv4:10.162.100.70
session transport udp
dtmf-relay rtp-nte
!
!
!
ephone-hunt 1 sequential
pilot 2
list 4420, 4421, 4422
timeout 45
statistics collect
!
Solved! Go to Solution.
02-24-2010 07:13 AM
One more think, when i call i get a dialtone one the caller side.
You should have said that before, and I should have inferred it from the ISDN trace. Configure
dial-peer voice pots 1
incoming called-number .
direct-inward-dial
And everything will work then.
02-24-2010 03:41 AM
If calls comes in via PRI, why are you mentioned voip DP?
In your country, do you have overlap receiving ?
02-24-2010 03:52 AM
i have a voip DP because that is where i want the call to go.
it has to leave the router to a SIP server.
interface Serial0/2/0:15
description *** Acesso Primario 236200230 ***
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
02-24-2010 04:59 AM
Then prepare a tranaltion-rule that makes sense, as the current one does not.
If still trouble, include only debug isdn 931 and ccsip message.
There is no need to use ZIP or attachments.
02-24-2010 05:33 AM
the translation doesn´t make sense???
Could you please tell me why? Because if there is no pilot number it works fine....
has you can see in the debug, the call never gets to the voip dialpeer so ccsip debug does not help.
001463: Feb 24 13:30:04.274: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0044
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8083 - Origination address is non-ISDN
Calling Party Number i = 0x0081, '213848218'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x81, '0'
Plan:ISDN, Type:Unknown
001464: Feb 24 13:30:04.290: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8044
Channel ID i = 0xA98381
Exclusive, Channel 1
001465: Feb 24 13:30:04.290: ISDN Se0/2/0:15 Q931: TX -> CONNECT pd = 8 callref = 0x8044
001466: Feb 24 13:30:04.418: ISDN Se0/2/0:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0044
GPS_Pombal_RT#
001467: Feb 24 13:30:04.422: %ISDN-6-CONNECT: Interface Serial0/2/0:0 is now connected to 213848218 N/A
GPS_Pombal_RT#
001468: Feb 24 13:30:13.122: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0044
Cause i = 0x8090 - Normal call clearing
Progress Ind i = 0x8288 - In-band info or appropriate now available
Display i = 'Chamada desligada'
001469: Feb 24 13:30:13.126: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0 disconnected from 213848218 , call lasted 8 seconds
GPS_Pombal_RT#
001470: Feb 24 13:30:13.126: ISDN Se0/2/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0xCE
001471: Feb 24 13:30:13.134: ISDN Se0/2/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x8044
001472: Feb 24 13:30:13.206: ISDN Se0/2/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0044
GPS_Pombal_RT#
GPS_Pombal_RT#
GPS_Pombal_RT#sh deb
GPS_Pombal_RT#sh debugging
The following ISDN debugs are enabled on all DSLs:
debug isdn error is ON.
debug isdn q931 is ON. (filter is OFF)
CCSIP SPI: SIP Call Statistics tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call State Machine tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call Events tracing is enabled (filter is OFF)
CCSIP SPI: SIP error debug tracing is enabled (filter is OFF)
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: SIP media debug tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call preauth tracing is enabled (filter is OFF)
CCSIP SPI: SIP Call transport tracing is enabled (filter is OFF)
GPS_Pombal_RT#u all
All possible debugging has been turned off
02-24-2010 06:50 AM
Something is wrong with you voip DP. Try for example "show dialplan number xxxx" where xxx is the number to which 0 is translated.
The ephone pilot has nothing to do with voip DP.
02-24-2010 06:55 AM
The dial-peer 198 is the backup of the VOIP dial-peer.
GPS_Pombal_RT#sh dialplan number 236200230
Macro Exp.: 236200230
VoiceOverIpPeer2200
peer type = voice, system default peer = FALSE, information type = voice,
description = `Ligacao VContact',
tag = 2200, destination-pattern = `236200230',
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 2200, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
URI classes:
Incoming (Request) =
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
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
type = voip, session-target = `ipv4:10.162.100.70',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,
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
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19
fax rate = fax, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
Fax Relay SG3-to-G3 Enabled (by system configuration)
fax NSF = 0xAD0051 (default)
voice-class codec = 2
codec = g729r8, payload size = 20 bytes,
text relay = disabled
Media Setting = flow-through (global)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 250 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
redirect ip2ip = disabled
local peer = false
probe disabled,
Secure RTP: system (use the global setting)
voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 1925269, Charged Units = 0,
Successful Calls = 129, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "10 ",
Last Disconnect Text is "normal call clearing (16)",
Last Setup Time = 8415975.
Last Disconnect Time = 8436256.
Matched: 236200230 Digits: 9
Target: ipv4:10.162.100.70
VoiceEncapPeer198
peer type = voice, system default peer = FALSE, information type = voice,
description = `Geral_CC',
tag = 198, destination-pattern = `236200230',
answer-address = `', preference=2,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 198, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
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
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
type = pots, prefix = `210438123',
forward-digits 0
session-target = `', voice-port = `0/2/0:15',
direct-inward-dial = enabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
Time elapsed since last clearing of voice call statistics never
Connect Time = 6014, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 1, Refused Calls = 0,
Last Disconnect Cause is "10 ",
Last Disconnect Text is "normal call clearing (16)",
Last Setup Time = 1354239.
Last Disconnect Time = 1360284.
Matched: 236200230 Digits: 9
Target:
GPS_Pombal_RT#
GPS_Pombal_RT#
GPS_Pombal_RT#sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
100 pots up up 011[257] 0 up 0/2/0:15
101 pots up up 4419 0 up 0/3/1
198 pots up up 21043812- 236200230 2 up 0/2/0:15
3
2000 voip up up 43.. 0 syst ipv4:10.42.2.254
2200 voip up up 236200230 0 syst ipv4:10.162.100.70
2100 voip up up 3... 0 syst ipv4:10.40.2.254
2500 voip down down 09........ 0 syst ipv4:10.40.2.254
1000 pots up up 1 01T 0 up 0/2/0:15
200 pots up up 02........ 0 up 0/2/0:15
300 pots up up 03........ 0 up 0/2/0:15
600 pots up up 06........ 0 up 0/2/0:15
700 pots up up 070[78]...... 0 up 0/2/0:15
620 pots up up 06[129]... 0 up 0/2/0:15
801 pots up up 0800...... 0 up 0/2/0:15
808 pots up up 080[89]...... 0 up 0/2/0:15
900 pots up up 09........ 0 up 0/2/0:15
10000 pots up up 0 00T 0 up 0/2/0:15
110 pots up up 4418 0 up 0/3/0
20001 pots up up 4420$ 0 50/0/1
20002 pots up up 4410$ 0 50/0/2
20003 pots up up 236200234$ 9 50/0/2
20004 pots up up 4411$ 0 50/0/3
20005 pots up up 236200235$ 9 50/0/3
20006 pots up up 4412$ 0 50/0/4
20007 pots up up 236200236$ 9 50/0/4
20008 pots up up 4413$ 0 50/0/5
20009 pots up up 236200237$ 9 50/0/5
20010 pots up up 4414$ 0 50/0/6
20011 pots up up 4415$ 0 50/0/7
20012 pots up up 4416$ 0 50/0/8
20013 pots up up 4417$ 0 50/0/9
20014 pots up up 4421$ 0 50/0/10
20015 pots up up 236200232$ 9 50/0/10
20016 pots up up 4422$ 0 50/0/11
20017 pots up up 236200233$ 9 50/0/11
20018 pots up up 4423$ 0 50/0/12
20019 pots up up 236200238$ 9 50/0/12
20020 pots up up 4424$ 0 50/0/13
20021 pots up up 4425$ 0 50/0/14
20022 pots up up 7620$ 0 50/0/15
20023 pots up up 7720$ 0 50/0/16
20024 pots up up 7820$ 0 50/0/17
20025 pots up up 7920$ 0 50/0/18
20070 pots up up A2A000 0 50/0/1
20071 pots up up 2 0 50/0/1
20072 pots up up A2A001 0 50/0/10
20073 pots up up A2A002 0 50/0/11
dial-peer voice 198 pots
description Geral_CC
preference 2
destination-pattern 236200230
direct-inward-dial
port 0/2/0:15
forward-digits 0
prefix 210438123
!
02-24-2010 07:03 AM
You may have an ACL or routing problem preventing the SIP call to go out.
02-24-2010 07:10 AM
I really don't think so.
The only thing that i have to do to make this work, is to change the pilot number to a different one of the DNIS. In this case it works fine.
That's why i don't think it's a routing problem.
I don't understand why the router is matching the DNIS... there is in no place in the configuration a match calling, that is why this doesn't make any sense to me.
One more think, when i call i get a dialtone one the caller side.
Hope it helps.
02-24-2010 07:13 AM
One more think, when i call i get a dialtone one the caller side.
You should have said that before, and I should have inferred it from the ISDN trace. Configure
dial-peer voice pots 1
incoming called-number .
direct-inward-dial
And everything will work then.
02-24-2010 07:28 AM
That was it!!!!!
Sorry for not saying about the dialtone in the first place.
Thank you so much for your help.
02-24-2010 07:42 AM
No problem. Thank you for the nice rating and good luck!
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: