Solved: Hi all,

devraj123 · ‎10-19-2016

Hi All,

Calling Number : 27214921100
Called Number : 0827718822

CUCM---------------------CUBE (2900 |15.4 M3)------------------ITSP

182.17.3.5 182.30.50.13

when making oubound calls, cube is sending 503 service unavailable. we have working site as well with same model cube and configuration, its working there.

Cucm send invite and it gets 100 trying after that 503 service available is coming from CUbe imeediately. even not reaching to external leg.

Oct 17 12:14:33.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0827718822@182.17.3.5:5060 SIP/2.0
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bK51212b3efc1
From: <sip:8834@182.30.50.13>;tag=572717~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30598611
To: <sip:0827718822@182.17.3.5>
Date: Mon, 17 Oct 2016 10:15:14 GMT
Call-ID: 96cbea80-8041a4b2-28fb-d321eac@182.30.50.13
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:182.30.50.13:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 2529946240-0000065536-0000000952-0221388460
Session-Expires: 1800
P-Asserted-Identity: <sip:8834@182.30.50.13>
Remote-Party-ID: <sip:8834@182.30.50.13>;party=calling;screen=yes;privacy=off
Contact: <sip:8834@182.30.50.13:5060;transport=tcp>
Max-Forwards: 69
Content-Length: 0

Oct 17 12:14:33.836: //-1/96CBEA800000/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
Oct 17 12:14:33.840: //16773/96CBEA800000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 182.17.3.5
Oct 17 12:14:33.840: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16820 for stream 1
Oct 17 12:14:33.840: //16773/96CBEA800000/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
Oct 17 12:14:33.840: //16773/96CBEA800000/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_IDLE (1)
          Stream address type    : 1
          Callid                 : -1
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 0
          Negotiated Codec       : No Codec   , bytes :0
          Nego. Codec payload    : 255 (tx), 255 (rx)
          Negotiated DTMF relay : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload : 0
          Media Srce Addr/Port   : [172.17.3.5]:16820
          Media Dest Addr/Port   : [ - ]:0

Oct 17 12:14:33.840: //16773/96CBEA800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bK51212b3efc1
From: <sip:8834@182.30.50.13>;tag=572717~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30598611
To: <sip:0827718822@182.17.3.5>
Date: Mon, 17 Oct 2016 12:14:33 GMT
Call-ID: 96cbea80-8041a4b2-28fb-d321eac@182.30.50.13
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0

Oct 17 12:14:33.844: //16774/96CBEA800000/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
Oct 17 12:14:33.844: //16774/96CBEA800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x229FC460
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 27214921100
Called Number            : 0827718822
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : :0
Destn SIP Resp Addr:Port : :0
Destination Name         :

Oct 17 12:14:33.844: //16774/96CBEA800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 188
Disconnect Cause (SIP)   : 200

Oct 17 12:14:33.844: //16773/96CBEA800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bK51212b3efc1
From: <sip:8834@182.30.50.13>;tag=572717~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30598611
To: <sip:0827718822@182.17.3.5>;tag=A0DFD58-784
Date: Mon, 17 Oct 2016 12:14:33 GMT
Call-ID: 96cbea80-8041a4b2-28fb-d321eac@182.30.50.13
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=0
Content-Length: 0

Oct 17 12:14:33.868: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0827718822@182.17.3.5:5060 SIP/2.0
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bK51212b3efc1
From: <sip:8834@182.30.50.13>;tag=572717~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30598611
To: <sip:0827718822@182.17.3.5>;tag=A0DFD58-784
Date: Mon, 17 Oct 2016 10:15:14 GMT
Call-ID: 96cbea80-8041a4b2-28fb-d321eac@182.30.50.13
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

Oct 17 12:14:33.868: //16773/96CBEA800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x229F5DB0
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 8834
Called Number            : 0827718822
Source IP Address (Sig ): 182.17.3.5
Destn SIP Req Addr:Port : 182.30.50.13:5060
Destn SIP Resp Addr:Port : 182.30.50.13:33439
Destination Name         : 182.30.50.13

Oct 17 12:14:33.868: //16773/96CBEA800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 182.17.3.5
Source IP Port    (Media): 16820
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Oct 17 12:14:33.868: //16773/96CBEA800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503

Thanks in advance

Manish Gogna · ‎10-20-2016

Hi Devraj,

In the INVITE message there is no message body, hence we have a Content-Lenght of "0".

Can you see if MTP is allocated on SIP trunk and early offer is being used , reset the trunk and try again.

We should get an INVITE with SDP on the CUBE from cucm for outbound call.

Manish

View solution in original post

Aseem Anand · ‎10-20-2016

Hi,

Are the calls from the working CUBE also using delayed offer? If the calls in working site are using early offer can you enable early offer under the sip trunk and then try a call?

Also send the running configuration from CUBE.

Aseem

View solution in original post

Manish Gogna · ‎10-20-2016

Hi Devraj,

In the INVITE message there is no message body, hence we have a Content-Lenght of "0".

Can you see if MTP is allocated on SIP trunk and early offer is being used , reset the trunk and try again.

We should get an INVITE with SDP on the CUBE from cucm for outbound call.

Manish

devraj123 · ‎10-20-2016

Hi Manual

I tried with setting mtp checked and also unchecked on cucm. Same problem.

I also checked with DTMF on sip trunk no preference and RFC 2833. But not.

Any clue

Mohammed al Baqari · ‎10-20-2016

Can you make sure that the CUBE has the command allow-connection sip-to-sip? If so, please post CUBE config.

devraj123 · ‎10-20-2016

Hi All,

the one difference which is highlighted i am getting in non-working

Nonworking-

*Oct 20 09:50:50.519: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0834699693@182.30.50.5:5060 SIP/2.0
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bKe4957d45d2df
From: "Stefan Viljoen" <sip:1175@182.30.50.13>;tag=1256106~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30612681
To: <sip:0834699693@182.30.50.5>
Date: Thu, 20 Oct 2016 09:47:25 GMT
Call-ID: 333bfa00-808192ad-7caf-d321eac@182.30.50.13
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:182.30.50.13:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 0859568640-0000065536-0000002958-0221388460
Session-Expires: 1800
P-Asserted-Identity: "Stefan Viljoen" <sip:1175@182.30.50.13>
Remote-Party-ID: "Stefan Viljoen" <sip:1175@182.30.50.13>;party=calling;screen=yes;privacy=off
Contact: <sip:1175@182.30.50.13:5060;transport=tcp>
Max-Forwards: 69
Content-Length: 0

working--
*Oct 20 09:35:56.713: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0720433475@182.17.3.5:5060 SIP/2.0
Via: SIP/2.0/TCP 182.30.50.13:5060;branch=z9hG4bKe3df6dd5fac7
From: <sip:8834@182.30.50.13>;tag=1254468~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-30612623
To: <sip:0720433475@182.17.3.5>
Date: Thu, 20 Oct 2016 09:38:33 GMT
Call-ID: f6233800-80819099-7c54-d321eac@182.30.50.13
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 4129503232-0000065536-0000002945-0221388460
Session-Expires: 1800
P-Asserted-Identity: <sip:8834@182.30.50.13>
Remote-Party-ID: <sip:8834@182.30.50.13>;party=calling;screen=yes;privacy=off
Contact: <sip:8834@182.30.50.13:5060;transport=tcp>
Max-Forwards: 69
Content-Length: 0

Mohammed al Baqari · ‎10-20-2016

This is fine. it means that your cucm is offering NOTIFY dtmf relay. A mismatch in dtmf relay wont cause 503 error.

The problem lies from cube side.

Did you verify the command allow-connection sip-to-sip configured in you cube.

voice service voip

sip

allow-connection sip-to-sip

Also post your cube config

devraj123 · ‎10-20-2016

Hi,

Dial peer hitted incoming 3000 and outgoing 901 for this outbound call.

voice-card 0
!
!
voice rtp send-recv
!
voice service voip
ip address trusted list
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
bearer-capability clear-channel udi bidirectional
early-offer forced
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g722-64
codec preference 4 g729r8

interface GigabitEthernet0/0
description LINK to CORE
ip address 182.17.3.5 255.255.255.0
ip mtu 1428
ip tcp adjust-mss 1370
duplex auto
speed auto
!

dial-peer voice 3000 voip
description Outbound to CUCM JNB
preference 1
destination-pattern 88..
session protocol sipv2
session target ipv4:182.30.50.13
voice-class codec 1
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
fax-relay sg3-to-g3
no vad

dial-peer voice 901 voip
description OUTBOUND TO Vodacom
translation-profile outgoing outbound-CARRIER
destination-pattern .T
session protocol sipv2
session target ipv4:91.91.137.11
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
fax protocol none
ip qos dscp cs3 signaling
no vad
!

Dev

Mohammed al Baqari · ‎10-20-2016

Hi Dev,

1. Do u need bearer capability command under sip. If not remove it

2. I see options enabled can you share sh dial-pee vo summary

3. You need to have both rtp nte and sip notify under your dialpeers

devraj123 · ‎10-20-2016

Hi Mohammed,

I just checked sh voice dial-peers, it is showing that outgoing dial-peer 901 was busy out. i then disabled the option from the dial-peer.

need to do test. Then will confirm

Thanks

Mohammed al Baqari · ‎10-20-2016

Exactly. That's the reason for cube to send 503 cuz the peer is dead so no need for cube to initiate INVITE

devraj123 · ‎10-21-2016

Hi all,

The call is reaching now to service provider but the call still failing.Now ITSP is sending 503 service unavailable

*Oct 21 06:59:07.692: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0216724006@182.17.3.5:5060 SIP/2.0

Via: SIP/2.0/TCP 182.17.2.39:5060;branch=z9hG4bK58a450dc1fb5

From: <sip:8809@182.17.2.39>;tag=869131~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-33744165

To: <sip:0216724006@182.17.3.5>

Date: Fri, 21 Oct 2016 07:01:44 GMT

Call-ID: 38598180-8091bd58-2a0c-270211ac@182.17.2.39

Supported: timer,resource-priority,replaces

Min-SE: 1800

User-Agent: Cisco-CUCM10.5

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback,X-cisco-original-called

Call-Info: <sip:182.17.2.39:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP

Cisco-Guid: 0945389952-0000065536-0000001591-0654447020

Session-Expires: 1800

P-Asserted-Identity: <sip:8809@182.17.2.39>

Remote-Party-ID: <sip:8809@182.17.2.39>;party=calling;screen=yes;privacy=off

Contact: <sip:8809@182.17.2.39:5060;transport=tcp>

Max-Forwards: 69

Content-Length: 0

*Oct 21 06:59:07.696: //-1/385981800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=8809
   ----- ccCallInfo IE subfields -----
   cisco-ani=8809
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0216724006
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

*Oct 21 06:59:07.696: //-1/385981800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3E6B7A5C, Call Info(
   Calling Number=8809,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0216724006(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=3000, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=12710
*Oct 21 06:59:07.696: //-1/385981800000/CCAPI/ccCheckClipClir:
   In: Calling Number=8809(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Oct 21 06:59:07.696: //-1/385981800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=8809(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Oct 21 06:59:07.696: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.696: :cc_get_feature_vsa malloc success
*Oct 21 06:59:07.696: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.696: cc_get_feature_vsa count is 1
*Oct 21 06:59:07.696: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.696: :FEATURE_VSA attributes are: feature_name:0,feature_time:1052249600,feature_id:919
*Oct 21 06:59:07.696: //12710/385981800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=8809(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0216724006(TON=Unknown, NPI=Unknown))
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/cc_process_call_setup_ind:
   Event=0x3EBA3820
*Oct 21 06:59:07.700: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 0216724006
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCallSetContext:
   Context=0x21A4E7E8
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 12710 with tag 3000 to app "_ManagedAppProcess_Default"
*Oct 21 06:59:07.700: //12710/385981800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/TCP 182.17.2.39:5060;branch=z9hG4bK58a450dc1fb5

From: <sip:8809@182.17.2.39>;tag=869131~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-33744165

To: <sip:0216724006@182.17.3.5>

Date: Fri, 21 Oct 2016 06:59:07 GMT

Call-ID: 38598180-8091bd58-2a0c-270211ac@182.17.2.39

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.4.3.M3

Content-Length: 0

*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=901, Params=0x21A4FDE0, Progress Indication=NULL(0)
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCheckClipClir:
   In: Calling Number=27214921100(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=27214921100(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=.T, Called Number=0216724006, Digit Strip=FALSE
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccCallSetupRequest:
   Calling Number=27214921100(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0216724006(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=8809, Final Destination Flag=TRUE,
   Guid=38598180-0001-0000-0000-0637270211AC, Outgoing Dial-peer=901
*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=8809
   ----- ccCallInfo IE subfields -----
   cisco-ani=27214921100
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0216724006
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

*Oct 21 06:59:07.700: //12710/385981800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x3E6B7A5C, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=27214921100,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0216724006(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=901, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Oct 21 06:59:07.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.700: :cc_get_feature_vsa malloc success
*Oct 21 06:59:07.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.700: cc_get_feature_vsa count is 2
*Oct 21 06:59:07.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Oct 21 06:59:07.700: :FEATURE_VSA attributes are: feature_name:0,feature_time:1052249376,feature_id:920
*Oct 21 06:59:07.704: //12711/385981800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Oct 21 06:59:07.704: //12711/385981800000/CCAPI/ccCallSetContext:
   Context=0x21A4FD90
*Oct 21 06:59:07.704: //12710/385981800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=901
*Oct 21 06:59:07.704: //12711/385981800000/CCAPI/cc_api_call_proceeding:
   Interface=0x3E6B7A5C, Progress Indication=NULL(0)
*Oct 21 06:59:07.708: //12711/385981800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0216724006@91.21.137.11:5060 SIP/2.0

Via: SIP/2.0/UDP 182.17.3.5:5060;branch=z9hG4bK22E6464

Remote-Party-ID: <sip:27214921100@182.17.3.5>;party=calling;screen=yes;privacy=off

From: <sip:27214921100@182.17.3.5>;tag=A165D0C-9B2

To: <sip:0216724006@91.21.137.11>

Date: Fri, 21 Oct 2016 06:59:07 GMT

Call-ID: B1FEE6FC-969211E6-BF85BBE3-B72D0EBC@182.17.3.5

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0945389952-0000065536-0000001591-0654447020

User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1477033147

Contact: <sip:27214921100@182.17.3.5:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

Session-Expires: 1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 357

v=0

o=CiscoSystemsSIP-GW-UserAgent 7297 4173 IN IP4 182.17.3.5

s=SIP Call

c=IN IP4 182.17.3.5

t=0 0

m=audio 17460 RTP/AVP 8 0 9 18 101

c=IN IP4 182.17.3.5

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=fmtp:9 bitrate=64

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

*Oct 21 06:59:07.736: //12711/385981800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 182.17.3.5:5060;branch=z9hG4bK22E6464

From: <sip:27214921100@182.17.3.5>;tag=A165D0C-9B2

To: <sip:0216724006@91.21.137.11>

Call-ID: B1FEE6FC-969211E6-BF85BBE3-B72D0EBC@182.17.3.5

CSeq: 101 INVITE

Timestamp: 1477033147

*Oct 21 06:59:07.736: //12711/385981800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 182.17.3.5:5060;branch=z9hG4bK22E6464

From: <sip:27214921100@182.17.3.5>;tag=A165D0C-9B2

To: <sip:0216724006@91.21.137.11>;tag=aprqngfrt-5q8hd630000a6

Call-ID: B1FEE6FC-969211E6-BF85BBE3-B72D0EBC@182.17.3.5

CSeq: 101 INVITE

Timestamp: 1477033147

*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/cc_api_call_disconnected:
   Cause Value=63, Interface=0x3E6B7A5C, Call Id=12711
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=63, Retry Count=0)
*Oct 21 06:59:07.736: //12710/385981800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=12711
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/ccCallDisconnect:
   Cause Value=63, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=63)
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/ccCallDisconnect:
   Cause Value=63, Call Entry(Responsed=TRUE, Cause Value=63)
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3E6B7A5C, Tag=0x0, Call Id=12711,
   Call Entry(Disconnect Cause=63, Voice Class Cause Code=0, Retry Count=0)
*Oct 21 06:59:07.736: //12711/385981800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Oct 21 06:59:07.736: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Oct 21 06:59:07.736: :cc_free_feature_vsa freeing 3EB80D18
*Oct 21 06:59:07.736: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Oct 21 06:59:07.736: vsacount in free is 1
*Oct 21 06:59:07.736: //12711/385981800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3F380388
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 27214921100
Called Number            : 0216724006
Source IP Address (Sig ): 182.17.3.5
Destn SIP Req Addr:Port : 91.21.137.11:5060
Destn SIP Resp Addr:Port : 91.21.137.11:5060
Destination Name         : 91.21.137.11

*Oct 21 06:59:07.736: //12711/385981800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 182.17.3.5
Source IP Port    (Media): 17460
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

*Oct 21 06:59:07.736: //12711/385981800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503

*Oct 21 06:59:07.736: //12710/385981800000/CCAPI/ccCallDisconnect:
   Cause Value=63, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Oct 21 06:59:07.736: //12710/385981800000/CCAPI/ccCallDisconnect:
   Cause Value=63, Call Entry(Responsed=TRUE, Cause Value=63)
*Oct 21 06:59:07.740: //12710/385981800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error

Via: SIP/2.0/TCP 182.17.2.39:5060;branch=z9hG4bK58a450dc1fb5

From: <sip:8809@182.17.2.39>;tag=869131~9ef3738a-e9a5-419e-997d-cad9d7f84eb1-33744165

To: <sip:0216724006@182.17.3.5>;tag=A165D30-6EE

Date: Fri, 21 Oct 2016 06:59:07 GMT

Call-ID: 38598180-8091bd58-2a0c-270211ac@182.17.2.39

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.4.3.M3

Reason: Q.850;cause=63

Content-Length: 0

*Oct 21 06:59:07.740: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:0216724006@91.21.137.11:5060 SIP/2.0

Via: SIP/2.0/UDP 182.17.3.5:5060;branch=z9hG4bK22E6464

From: <sip:27214921100@182.17.3.5>;tag=A165D0C-9B2

To: <sip:0216724006@91.21.137.11>;tag=aprqngfrt-5q8hd630000a6

Date: Fri, 21 Oct 2016 06:59:07 GMT

Call-ID: B1FEE6FC-969211E6-BF85BBE3-B72D0EBC@182.17.3.5

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Mohammed al Baqari · ‎10-21-2016

This is a problem at itsp. You need to contact them. Nothing wrong at your end

Graham Old · ‎10-23-2016

I have seen a similar problem when you offer codecs the ISP does not support.

Look at an incoming INVITE from the service provider and see what codecs they are offering, make sure you only offer the same ones.

Graham

devraj123 · ‎10-23-2016

Hi All,

The issue is resolved. we applied early offer over sip trunk using sip profile. and its working

Thanks to all of you for help.

Aseem Anand · ‎10-20-2016

Hi,

Are the calls from the working CUBE also using delayed offer? If the calls in working site are using early offer can you enable early offer under the sip trunk and then try a call?

Also send the running configuration from CUBE.

Aseem

Cube sending 503 service not available to cucm