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Replies

CME to CUCM connection Error

ali_ebrahimi
Level 1
Level 1

Hi All

I'm busy connecting PBX, MS Lync and CUCM. There is a Cisco 3845 as my central interconnecting point. Up till now I have done Lync to PBX and PBX to CUCM interconnections.

There is a SIP trunk between Lync and CME. I was going to use CME as an intermediary device to avoid another SIP trunk between Lync and CUCM. My Lync numbers are 4XXX and CUCM ones are 9XXX. The call from Lync enters CME.

The following lines are the output of  "debug voice ccapi all":

Dec 22 08:08:00.238: %IP_VFR-4-FRAG_TABLE_OVERFLOW: GigabitEthernet0/1.261: the fragment table has reached its maximum threshold 16

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

   Call Entry Is Not Found

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

   Call Entry Is Not Found

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_api_event_indication:

   CALL_ERROR_INFORMATIONAL; Call Entry Is Not Found

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

   Call Entry Is Not Found

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:

   Call Entry Is Not Found

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

   Call Entry Is Not Found

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=4444

   ----- ccCallInfo IE subfields -----

   cisco-ani=4444

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=942

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=0

   cisco-rdnsi=0

   cisco-redirectreason=-1

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x65A85DA0, Call Info(

   Calling Number=4444(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=942(TON=Unknown, NPI=Unknown),

Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=100, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=1208

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/cc_api_call_setup_ind_common:

   Interface Type=0, Protocol=3

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/ccCheckClipClir:

   In: Calling Number=4444(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/ccCheckClipClir:

   Calling Party Number Is User Provided

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/ccCheckClipClir:

   Out: Calling Number=4444(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Dec 22 08:08:15.842: //-1/FCE5F8E79258/CCAPI/cc_api_call_setup_ind_common:

   After Number Translation Checking:

   Calling Number=4444(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=942(TON=Unknown, NPI=Unknown)

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

   Total Call Count=0, Call Entry(Call Count On=FALSE, Incoming Call=TRUE)

Dec 22 08:08:15.842: //1208/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

   Total Call Count=1

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_insert_guid_pod_entry:

   Incoming=TRUE, Call Id=1208

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=4444(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=942(TON=Unknown, NPI=Unknown))

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_incr_if_call_volume:

   Remote IP Address=172.25.25.27, Hwidb=GigabitEthernet0/0

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_incr_if_call_volume:

   Total Call Count=1, Voip Call Count=1, MMoip Call Count=0

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_process_call_setup_ind:

   Event=0x65CA14F0

Dec 22 08:08:15.842: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:

   Matching Parameters; Called Number=942, Call Transfer Consult Id=

Dec 22 08:08:15.842: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

   Searching Node;

   Called Number=942, Call Transfer Consult Id=

Dec 22 08:08:15.842: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:   No Matching Node

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/ccCallSetContext:

   Context=0x71DF5268

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 1208 with tag 100 to app "_ManagedAppProcess_Default"

Dec 22 08:08:15.842: //1208/FCE5F8E79258/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Dec 22 08:08:15.846: //1208/FCE5F8E79258/CCAPI/ccCallCanGateway:

   CALL_ERROR_INFORMATIONAL; Gateway Is Not Allowed;

   Incoming InterfaceType=3, Outgoing Interface Type=27, Call Id=0x4B8

Dec 22 08:08:15.846: //1208/FCE5F8E79258/CCAPI/ccCallDisconnect:

   Cause Value=3, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Dec 22 08:08:15.846: //1208/FCE5F8E79258/CCAPI/ccCallDisconnect:

   Start Calling Accounting;

   Call Entry(Incoming=TRUE)

Dec 22 08:08:15.846: //1208/FCE5F8E79258/CCAPI/ccCallDisconnect:

   Cause Value=3, Call Entry(Disconnect Cause=0)

Dec 22 08:08:15.846: //1208/FCE5F8E79258/CCAPI/ccCallDisconnect:

   Cause Value=3, Call Entry(Responsed=TRUE, Cause Value=3)

Dec 22 08:08:15.854: //-1/xxxxxxxxxxxx/CCAPI/cc_api_icpif:

   ExpectFactor=0xA

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/ccCallGetVoipFlag:

   Data Bitmask=0x1, Call Id=1208

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/ccCallGetVoipFlag:

   Flag=FALSE

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_decr_if_call_volume:

   Remote IP Address=172.25.25.27, Hwidb=GigabitEthernet0/0

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_decr_if_call_volume:

   Total Call Count=0, Voip Call Count=0, MMoip Call Count=0

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x65A85DA0, Tag=0x0, Call Id=1208,

   Call Entry(Disconnect Cause=3, Voice Class Cause Code=0, Retry Count=0)

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_delete_guid_pod_entry:

   Incoming=TRUE

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_delete_call_entry:

   Total Call Count=1, Call Entry(Call Count On=FALSE, Incoming Call=TRUE)

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_delete_call_entry:

   Total Call Count=0

Dec 22 08:08:15.854: //1208/FCE5F8E79258/CCAPI/cc_delete_call_entry:

   Deleting profileTable[0x6649A22C]terminal no monitor

The following is the output of sh dialplan number command

 

peer type = voice, information type = voice,
        description = `',
        tag = 3, destination-pattern = `9..',
        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 = 3, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        modem transport = system,
        URI classes:
            Incoming (Called) =
            Incoming (Calling) =
            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 = `ipv4:172.25.1.6',
        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 = cisco, 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,
        RTP dynamic payload type values: NTE = 101
        Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
               CAS=123, 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 NSF = 0xAD0051 (default)
        codec = g729r8,   payload size =  20 bytes,
        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
        probe disabled,
        voice class perm tag = `'
        Time elapsed since last clearing of voice call statistics never
        Connect Time = 387407, Charged Units = 0,
        Successful Calls = 39, Failed Calls = 3, 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 = 19242525.

 

Here is also related dial-peers

dial-peer voice 3 voip
 destination-pattern 9..
 session target ipv4:172.25.1.6

dial-peer voice 100 voip
 huntstop
 rtp payload-type comfort-noise 13
 voice-class codec 1
 session protocol sipv2
 session target dns:irisaco.com
 session transport tcp
 incoming called-number .%
 dtmf-relay rtp-nte
 no vad
dial-peer voice 101 voip
 huntstop
 destination-pattern 4...
 rtp payload-type comfort-noise 13
 voice-class codec 1
 session protocol sipv2
 session target dns:irisaco.com
 session transport tcp
 dtmf-relay rtp-nte
 fax rate disable

Thanks all in advance

Ali Ebrahimi

1 Accepted Solution

Accepted Solutions

You're able to do that but you need to allow:

--

voice service voip
allow-connections h323 to sip
allow-connections sip to h323

--

But appart from that there's no reason to use H323 to one system and SIP to another when SIP is able to handle all connected systems.

Why did you choose H323 from CME to CUCM ?

View solution in original post

7 Replies 7

Chris Deren
Hall of Fame
Hall of Fame

Can you also post "debug ccsip messages"?

Let us know what number is calling what and in which direction, i.e. is this inbound or outbound call on CME.

Thanks for your reply

My called number is 920 and it is a CUCM 7911 phone. It comes into mu CME from dialer-peer 100 (MS Lync) and is destined for dialer-peer 3 (CUCM).

I'll post the debug output a couple of days later because I'm on a mission and can not do it now.

Regards

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Hi looking at your outbound dial peer, we can see that it's setup for h323 and not sip and it looks like you didn't enable sip to h323 on the cme

C ALL_ERROR_INFORMATIONAL; Gateway Is Not Allowed;.

dial-peer voice 3 voip
destination-pattern 9..
session target ipv4:172.25.1.6

To start please add session protocol sip v2 on the dial peer

Please rate all useful posts

Thank you for your Reply

I thought that my CME can convert H323 to SIP on the air when routing a call from CUCM to Lync and vice versa. Is it wron?

Regards

You're able to do that but you need to allow:

--

voice service voip
allow-connections h323 to sip
allow-connections sip to h323

--

But appart from that there's no reason to use H323 to one system and SIP to another when SIP is able to handle all connected systems.

Why did you choose H323 from CME to CUCM ?

Thank you Christian

I will soon test what you sent me and will post the result.

But, why H323? I'm new in administration of this telephony system and I had no time to change it. But I will consider your comment in future changes.

Thanks again

Ali Ebrahimi

Dear Christian

Thank you vey much. This solved my problem.