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Incoming call problem

Jose Goncalves
Level 1
Level 1

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

!

1 Accepted Solution

Accepted Solutions

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.

View solution in original post

11 Replies 11

paolo bevilacqua
Hall of Fame
Hall of Fame

If calls comes in via PRI, why are you mentioned voip DP?

In your country, do you have overlap receiving ?

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

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.

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

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.

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

!

You may have an ACL or routing problem preventing the SIP call to go out.

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.

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.

That was it!!!!!

Sorry for not saying about the dialtone in the first place.

Thank you so much for your help.

No problem. Thank you for the nice rating and good luck!

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