Solved: incoming sip call with no audio drop after 10 seconds - Page 5

mohammed asif suleman · ‎03-31-2013

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

mohammed asif suleman · ‎04-02-2013

just i signed up for web ex, its asking for mail id for meeting

my mail id is

md.asifsuleman@gmail.com

Ayodeji Okanlawon · ‎04-02-2013

You need to send me a webex invite...

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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mohammed asif suleman · ‎04-02-2013

i am extermly sorry,i am not able send invitation , its asking for your mail id to invite, which i don't have.

can you join me on

https://meetings.webex.com/collabs/meetings/join?uuid=M59BDC9JKI8CV4781VT9WKT01W-1IIH

Ayodeji Okanlawon · ‎04-02-2013

my mail is deji_ok@hotmail.co.uk

Please rate all useful posts

mohammed asif suleman · ‎04-02-2013

you got the invitation.

Ayodeji Okanlawon · ‎04-02-2013

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"

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mohammed asif suleman · ‎04-03-2013

Dear aokanwon,

i am extermly sorry that i am bringin this issue to you,

there is no audio on the call from sip line for outgoing and incoming

the greeting were playing and the calls are connecting to reception but there is no audio.

for internal calls also there was no audio from the new device pool to other device pools

please can you spare some time on this issue, it will be very helpful for me

Ayodeji Okanlawon · ‎04-03-2013

send me a debug ccsip messages.

can your ip phones ping the cube ip address used for the sip trunk.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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mohammed asif suleman · ‎04-03-2013

hi aokanlawon,

i don't think so it can ping the ip address.

but i am using cipc, is there any option to check that on cipc.

debug attached

Ayodeji Okanlawon · ‎04-03-2013

Ping from the PC. is the IP communicator using IP in the subnet of the PC? You need to configure a route for the network your phones are to that CUBE IP

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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mohammed asif suleman · ‎04-03-2013

i cannot ping from my pc to the ip 172.29.25.246

shall i add a route towards the IP

sorry the MPLS IP add is 172.18.16.182 what we given for sip trunk. for me not able to ping this ip also.

mohammed asif suleman · ‎04-03-2013

it a mpls connection so i think the local network data will not be able to travel on that.

i have one doubt can we able to use the ip 192.168.33.5 instead of this ip 172.18.16.182

Ayodeji Okanlawon · ‎04-03-2013

Mohammed,

Ok..send me a webex invite again..

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

mohammed asif suleman · ‎04-03-2013

Hi Mr.Deji , i sent the webex invite

duclan_hp · ‎04-03-2013

Hi mohammed,

I assume the audio data will go through voice gatway be fore reach your CICP, if it is true, then:

Let use CICP to make test call again, before the call drops, let issue below command:

sh call active voice compact

sh control-plane host open-port

sh udp

and post the log along with you CICP Ip address!