12-22-2015 01:37 AM - edited 03-18-2019 11:44 AM
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
Solved! Go to Solution.
12-23-2015 03:52 AM
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 ?
12-22-2015 06:01 AM
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.
12-22-2015 06:59 PM
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
12-22-2015 07:53 PM
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
12-22-2015 10:17 PM
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
12-23-2015 03:52 AM
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 ?
12-23-2015 09:29 PM
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
01-12-2016 03:18 AM
Dear Christian
Thank you vey much. This solved my problem.
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