03-31-2013 05:43 AM - edited 03-16-2019 04:32 PM
Hi all,
i am facing a problem in sip line configuration.
i am configuring sip line on branch router 2921. Incoming sip calls are disconnecting after 10 sec and there is no audio for either side.
the other end is hearing only call progress tone even after my side answers the call.
and out going call choosing a different dial-peer than what it should use.
the call flow is like below
ITSP----------->Branch Router---------->head office Router------------->call manager.
i am attaching the router config and duebugs here....
any help will be highly appriciated...
thanks in advance.
Out going vall debug voice ccapi inout
what does this means Disconnect Cause=57 Bearer capability not authorized
This cause indicates that the user has requested a bearer capability which is implemented by the equipment which generated this cause but the user is not authorized to use.
*Mar 31 13:23:56.243: //-1/00087A1D0200/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=CIPC
----- ccCallInfo IE subfields -----
cisco-ani=2300
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=90542014824
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=-1
cisco-rdnplan=-1
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Mar 31 13:23:56.243: //-1/00087A1D0200/CCAPI/cc_api_call_setup_ind_common:
Interface=0x31C5F568, Call Info(
Calling Number=2300,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90542014824(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T RUE,
Incoming Dial-peer=304, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS E), Call Id=1026
*Mar 31 13:23:56.243: //-1/00087A1D0200/CCAPI/ccCheckClipClir:
In: Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
*Mar 31 13:23:56.243: //-1/00087A1D0200/CCAPI/ccCheckClipClir:
Out: Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
*Mar 31 13:23:56.243: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.243: :cc_get_feature_vsa malloc success
*Mar 31 13:23:56.243: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.243: cc_get_feature_vsa count is 1
*Mar 31 13:23:56.243: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.243: :FEATURE_VSA attributes are: feature_name:0,feature_time:8 05326136,feature_id:109
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passe d, Presentation=Allowed),
Called Number=90542014824(TON=Unknown, NPI=Unknown))
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/cc_process_call_setup_ind:
Event=0x32503920
*Mar 31 13:23:56.243: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 90542014824
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/ccCallSetContext:
Context=0x2CD7DC18
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 1026 with tag 304 to app "_ManagedAppProcess_Default"
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Mar 31 13:23:56.243: //1026/00087A1D0200/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=304, Params=0x2CD7C7F8, Progress Indication=NULL(0)
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccCheckClipClir:
In: Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccCheckClipClir:
Out: Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=90542014824, Digit Strip=FALSE
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccCallSetupRequest:
Calling Number=2300(TON=Unknown, NPI=Unknown, Screening=User, Passed, Present ation=Allowed),
Called Number=90542014824(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=
Account Number=CIPC, Final Destination Flag=TRUE,
Guid=00087A1D-B938-8115-0200-7401C0A82078, Outgoing Dial-peer=304
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=CIPC
----- ccCallInfo IE subfields -----
cisco-ani=2300
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=90542014824
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=-1
cisco-rdnplan=-1
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x31CB243C, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=2300,(Calling Name=)(TON=Unknown, NPI=Unknown, Scr eening=User, Passed, Presentation=Allowed),
Called Number=90542014824(TON=Unknown, NPI=Unknown), Calling Translated=FALSE ,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=30 4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
*Mar 31 13:23:56.247: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.247: :cc_get_feature_vsa malloc success
*Mar 31 13:23:56.247: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.247: cc_get_feature_vsa count is 2
*Mar 31 13:23:56.247: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 31 13:23:56.247: :FEATURE_VSA attributes are: feature_name:0,feature_time:8 05325912,feature_id:110
*Mar 31 13:23:56.247: //1027/00087A1D0200/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Mar 31 13:23:56.247: //1027/00087A1D0200/CCAPI/ccCallSetContext:
Context=0x2CD7C7A8
*Mar 31 13:23:56.247: //1026/00087A1D0200/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=304
*Mar 31 13:23:56.247: //1027/00087A1D0200/CCAPI/cc_api_call_proceeding:
Interface=0x31CB243C, Progress Indication=NULL(0)
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/cc_api_call_disconnected:
Cause Value=57, Interface=0x31CB243C, Call Id=1027
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
*Mar 31 13:23:56.319: //1026/00087A1D0200/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=1027
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=57)
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x31CB243C, Tag=0x0, Call Id=1027,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Mar 31 13:23:56.319: //1027/00087A1D0200/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 31 13:23:56.319: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 31 13:23:56.319: :cc_free_feature_vsa freeing 30004C50
*Mar 31 13:23:56.319: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 31 13:23:56.319: vsacount in free is 1
*Mar 31 13:23:56.323: //1026/00087A1D0200/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=0)
*Mar 31 13:23:56.323: //1026/00087A1D0200/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Mar 31 13:23:56.323: //1026/00087A1D0200/CCAPI/cc_api_get_transfer_info:
Transfer Number=NULL
*Mar 31 13:23:56.323: //1026/00087A1D0200/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x31C5F568, Tag=0x0, Call Id=1026,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Mar 31 13:23:56.323: //1026/00087A1D0200/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 31 13:23:56.323: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 31 13:23:56.323: :cc_free_feature_vsa freeing 30004D30
*Mar 31 13:23:56.323: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 31 13:23:56.323: vsacount in free is 0
ASIC-KHBR-RTR#
ASIC-KHBR-RTR#u all
All possible debugging has been turned off
Solved! Go to Solution.
03-31-2013 06:34 PM
Its on the h323 gateway page...You need to check enable inbound fast start and also enable outbound fast start
You also need to check "MTP required"
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 06:58 PM
Ok..Then you need to go to your provider,.They have not configured their end correctly. They are rejecting calls from the customer IP address.
Hey, dont forget to rate useful posts!
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-02-2013 04:14 PM
Mohammed,
To resolve this issue we did a few things over the webex session
1. We changed the integration from cucm--h323--cube--sip--ITSP TO cucm--sip--cube--sip--ITSP
2. We removed the bind commands on the gateway because you are using different interfaces to connect to the LAN and your ITSP
3. We configured the SIP trunk IP address to the IP address that the CUBE chose as the closest destination amongst all the ips on it
4. We configured the following on the inbound and outbound dial-peer to resolve the DTMF issues..
dtmf-relay rtp-nte digit-drop sip-kpml
This is because SCCP endpoints/sccp integration with Unity connection does not support inband DTMF..so we used sip-kpml to pass DTMF
dtmf-relay rtp-nte digit-drop sip-kpml
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-03-2013 08:10 AM
Mohammed,
So the problem we discovered was that sip signalling and media were using an interface that is not reachable by the IP subnet fo your phones. To resolve this we configured sip signalling on the dial-peers. All calls to and From CUCM uses the local interface for media and signalling and all calls to ITSP uses the WAN interface for signalling and Media
dial-peer voice 15 voip
description outbound to CUCM
voice-class sip bind control source-interface GigabitEthernet0/1.31
voice-class sip bind media source-interface GigabitEthernet0/1.31
dial-peer voice 1 voip
description inbound from CUCM
incoming called number .T
voice-class sip bind control source-interface GigabitEthernet0/1.31
voice-class sip bind media source-interface GigabitEthernet0/1.31
dial-peer voice 2 voip
description inbound from ITSP
incoming called number 0380623..
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interfacen GigabitEthernet0/2
dial-peer voice 20 voip
description outbound to ITSP
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interfacen GigabitEthernet0/2
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 07:28 AM
Your gateway is not properly configured..
Please configure the following:
dial-peer voice 305 voip
translation-profile incoming INCO_SIP
incoming called number 8062300
session protocol sipv2
rtp payload-type nte 97
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
no vad
voice translation-profile INCO_SIP
translate calling 1
translate called 2
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 12:59 PM
Thankyou very much Mr.aokanlawon,
i am very sorry for the delay in reply , i was heading home after workhours
you are correct i was missing the dial-peer, but the problem is incoming call landing on the extension 2300 but when i answer the call there is no audio and the call getting droped after 10 seconds.
and the caller is hearing the call progress tone only,actually the call not getting established after pressing answer key on the ip phone 2300.
and the out going is also getting droped after maching the dial-peer .
all i am getting in the debug is cause code 57
which i am unable to undestand what the reason is...
please help me to get rid of the issue.
03-31-2013 02:07 PM
The debug you have sent shows that 2300 is the calling number and the called number is 90542014824. Its either you have sent the wrong debug or something...
Please do another test call and send
debug voip ccapi inout
debug ccsip messages
Even this call on this debug matches dial-peer 304 for both inbound and outbound leg...
Can you describe your call flow..like this (put the correct protocols)
ITSP---SIP---branch router---SIP/H323---HQ router---SIP/h323--CUCM
You need to send the config on both routers
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 02:50 PM
Hi Mr.aokanlawon
thankyou very much for your support...
the configs for both routers and debugs for incoming and outgoing calls are attached please check
the callflow is like
ITSP---->SIP----->branch router-------->h323---------->HQ router---------->h323-------->CUCM
03-31-2013 02:55 PM
There is no attachment
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 03:06 PM
03-31-2013 06:19 PM
You are doing EO to h323 leg, hence you are going to need to enable fast start. The logs show that CUBE is doing fast start to CUCM, but I dont think your CUCM is enabled for fast start on the h323 gateway, so make sure that the you select g711 as the codec for fast start. You can enable and make your test call again.. You need to ensure that the region between the MTP and h323 gateway is set to G711
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 06:26 PM
Hi aokanlawon,
can you please guide me from where can i enable fast start option for the gateway in cucm
ok i will prepare this for the next test call
Thankyou so much for helping me, if possible can you please spare some time for this issue tommorrow
ok i enabled the Inbound fast start
03-31-2013 06:33 PM
HI Mr. aokanlawon,
hurray...
thankyou very much .....incoming working now ...after enabling fast start but outgoing still not working.
03-31-2013 06:36 PM
Ok, thas good to hear. Did you enable outbound fast start also? You need both inbound and outbound...
Please enable it and send me
debug h225 asn1
debug h245 asn1
debug ccsip messages
debug voip ccapi inout
03-31-2013 06:34 PM
Its on the h323 gateway page...You need to check enable inbound fast start and also enable outbound fast start
You also need to check "MTP required"
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 06:40 PM
03-31-2013 06:49 PM
I have looked at it and your provider seem to be rejecting the call
Is this call sent directly to your provider on 10.205.20.50?
What ip address do you use to connect to your provider? Which ip address is thsi 172.29.25.246
*Apr 1 01:36:19.416: //3331/001FF36C0200/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK825AA
Record-Route: <10.205.20.50:5060>10.205.20.50:5060>
Call-ID:
62F46BA7-99A311E2-8E6197E8-26CACF4@172.29.25.246
From: <2300>;tag=1824053C-49B2300>
To: <0542014824>;tag=sbc08046515s73k0542014824>
CSeq: 101 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
*Apr 1 01:36:19.416: //3331/001FF36C0200/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK825AA
Record-Route: <10.205.20.50:5060>10.205.20.50:5060>
Call-ID: 62F46BA7-99A311E2-8E6197E8-26CACF4@172.29.25.246
From: <2300>;tag=1824053C-49B2300>
To: <0542014824>;tag=sbc08046515s73k0542014824>
CSeq: 101 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-31-2013 06:55 PM
the details sent by the provider are
Customer IP address= 172.29.25.246/30
STC IP Address = 172.29.25.245/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF with RFC2833
SIP Server IP address = 10.205.20.50
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