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SIP-to-SIP transfer to CUE Fails

jcioara
Level 1
Level 1

Hello,

We are converting our environment from a traditional T1 (POTS) incoming phone line to a SIP trunk on our Cisco UC520. DID numbers are configured on each circuit (DID on T1 provisioned and works fine, DID on SIP Trunk provisioned).

A quick diagram

PSTN----\         |----------|

         \--------|          |

         /--------| C-UC520  |-------> IP Phones / CUE

IP SIP--/         |----------|

(10.10.80.252)                 (172.29.0.20)

Inbound calls ring a receptionist extension, then transfer to extension 4700 (auto attendant) if there is no answer. When calls come in on the PSTN line, the transfer to the AA works perfectly when the receptionist doesn't answer. Calls coming in on the SIP line ring the receptionist, but fail when attempting to transfer to the AA.

Your assistance with this would be HUGELY appreciated!

INBOUND SIP CALL: debug ccsip messages (just when the transfer occurs - phone numbers hidden to protect the innocent)

166714: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:6********4@172.29.0.20:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK76661994

Remote-Party-ID: "6********1" <sip:6********1@10.10.80.252>;party=calling;screen=no;privacy=off

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>

Date: Fri, 12 Oct 2012 18:08:13 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

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

Min-SE:  1800

Cisco-Guid: 2673273716-0332272098-2992623594-1219141512

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1350065293

Contact: <sip:6********1@10.10.80.252:5060>

Call-Info: <sip:10.10.80.252:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 284

v=0

o=root 1142925334 1142925334 IN IP4 10.10.80.244

s=Asterisk PBX 1.8.14.0

c=IN IP4 10.10.80.252

t=0 0

m=audio 19234 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

166715: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK76661994

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>

Date: Fri, 12 Oct 2012 19:46:49 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

Timestamp: 1350065293

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Content-Length: 0

166716: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK76661994

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>;tag=32ACC1C8-1BAB

Date: Fri, 12 Oct 2012 19:46:49 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

Timestamp: 1350065293

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

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

Require: 100rel

RSeq: 9534

Allow-Events: telephone-event

Remote-Party-ID: "Main Line" <sip:5455@172.29.0.20>;party=called;screen=no;privacy=off

Contact: <sip:6********4@172.29.0.20:5060>

Content-Length: 0

166717: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

PRACK sip:6********4@172.29.0.20:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK7667827

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>;tag=32ACC1C8-1BAB

Date: Fri, 12 Oct 2012 18:08:13 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

CSeq: 102 PRACK

RAck: 9534 101 INVITE

Allow-Events: telephone-event

Max-Forwards: 70

Content-Length: 0

166718: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK7667827

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>;tag=32ACC1C8-1BAB

Date: Fri, 12 Oct 2012 19:46:49 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 PRACK

Content-Length: 0

166719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK76661994

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>;tag=32ACC1C8-1BAB

Date: Fri, 12 Oct 2012 19:46:49 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

Timestamp: 1350065293

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Diversion: <sip:6********4@172.29.0.20>;reason=no-answer;counter=1

Contact: <sip:4700@172.29.0.20>

Content-Length: 0

166720: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:6********4@172.29.0.20:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK76661994

From: "6********1" <sip:6********1@10.10.80.252>;tag=DC31124-48B

To: <sip:6********4@172.29.0.20>;tag=32ACC1C8-1BAB

Date: Fri, 12 Oct 2012 18:08:13 GMT

Call-ID: 9F5C6A0D-13CE11E2-B265CFEA-48AA9F88@10.10.80.252

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

INBOUND SIP CALL: debug ccsip all (just when the transfer occurs - phone numbers hidden to protect the innocent; I realize this is a little redundant)

166468: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceHandleConnAgeing: Connection=0x855F9E60, addr=10.1.10.1, port=5060, connid=1 has been aged out

166469: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostCloseConnection: Posting UDP conn close for addr=10.1.10.1, port=5060, connid=1

166470: //-1/xxxxxxxxxxxx/SIP/Transport/sipDeleteConnInstance: Deleted conn=0x855F9E60, connid=1, addr=10.1.10.1, port=5060, transport=UDP

166471: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetAgeingTimer: Aging timer initiated for holder=0x84729F98,addr=10.1.10.1

166472: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportStopAgeingTimer: Aging timer stopped for holder=0x84729F98,addr=10.1.10.1

166473: //-1/xxxxxxxxxxxx/SIP/Transport/sipDeleteConnHolder: Deleted holder=0x84729F98, addr=10.1.10.1, count=0

166474: //-1/xxxxxxxxxxxx/SIP/Info/udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060

166475: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_call_forward:

ccsip_call_forward

166476: //2005/FDF54C6DB235/SIP/Info/ccsip_call_forward: Call forward target num 4700

166477: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_FORWARD

166478: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 28

166479: //2005/FDF54C6DB235/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.

166480: //2005/FDF54C6DB235/SIP/Info/ccsip_spi_create_call_forward_contact_list: Calling peeridb not found

166481: //2005/FDF54C6DB235/SIP/Info/ccsip_spi_create_call_forward_contact_list: Call forward dpeer tag: 2001

166482: //2005/FDF54C6DB235/SIP/Info/ccsip_spi_add_redirect_contact: found voip peer, using session target as contact

166483: //2005/FDF54C6DB235/SIP/Info/ccsip_spi_add_redirect_contact: Calling peeridb not found

166484: //2005/FDF54C6DB235/SIP/Info/ccsip_spi_create_call_forward_contact_list: Number of redirect contacts added: 1

166485: //2005/FDF54C6DB235/SIP/Info/act_spi_call_forward:

Performing call forward

166486: //2005/FDF54C6DB235/SIP/Info/act_spi_call_forward:

Sending 3xx response

166487: //2005/FDF54C6DB235/SIP/Info/sipSPISendInviteResponse: Associated container=0x8793841C to Invite Response 302

166488: //2005/FDF54C6DB235/SIP/Transport/sipSPITransportSendMessage: msg=0x87F13D84, addr=10.10.80.252, port=50240, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x80D14908

166489: //2005/FDF54C6DB235/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately

166490: //2005/FDF54C6DB235/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0

166491: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x87F13D84, addr=10.10.80.252, port=5060, connId=0 for UDP

166492: //2005/FDF54C6DB235/SIP/Info/sentInviteResponseRedMovedTemp: Sent an 3456XX Error Response

166493: //2005/FDF54C6DB235/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for incoming call

166494: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[2005], src[6]

166495: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_set_release_source_for_peer: Failed AV set

166496: //2005/FDF54C6DB235/SIP/State/sipSPIChangeState: 0x85427250 : State change from (STATE_SENT_ALERTING, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)

166497: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK7657A75

From: "6********1" <sip:6********1@10.10.80.252>;tag=DA4B8A4-1AC9

To: <sip:6********4@172.29.0.20>;tag=328E69C0-F67

Date: Fri, 12 Oct 2012 19:13:40 GMT

Call-ID: FDFB6695-13C911E2-B23BCFEA-48AA9F88@10.10.80.252

Timestamp: 1350063304

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Diversion: <sip:6********4@172.29.0.20>;reason=no-answer;counter=1

Contact: <sip:4700@172.29.0.20>

Content-Length: 0

166498: //2005/FDF54C6DB235/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.

166499: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

166500: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7

166501: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.10.80.252:50240

166502: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1

166503: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000

166504: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:6********4@172.29.0.20:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.80.252:5060;branch=z9hG4bK7657A75

From: "6********1" <sip:6********1@10.10.80.252>;tag=DA4B8A4-1AC9

To: <sip:6********4@172.29.0.20>;tag=328E69C0-F67

Date: Fri, 12 Oct 2012 17:35:04 GMT

Call-ID: FDFB6695-13C911E2-B23BCFEA-48AA9F88@10.10.80.252

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

166505: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog

166506: //2005/FDF54C6DB235/SIP/Info/sipSPIFindCcbUASRespTable: *****CCB found in UAS Response table. ccb=0x85427250

166507: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.10.80.252,Port 50240, Transport 1, SentBy Port 5060

166508: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.10.80.252,Port 50240, Transport 1, SentBy Port 5060

166509: //2005/FDF54C6DB235/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:84821802 ConnTime 0

166510: //2005/FDF54C6DB235/SIP/State/sipSPIChangeState: 0x85427250 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DEAD, SUBSTATE_NONE)

166511: //2005/FDF54C6DB235/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x85427250

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 6********1

Called Number            : 6********4

Source IP Address (Sig  ): 172.29.0.20

Destn SIP Req Addr:Port  : 10.10.80.252:5060

Destn SIP Resp Addr:Port : 10.10.80.252:50240

Destination Name         : 10.10.80.252

166512: //2005/FDF54C6DB235/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711ulaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 0 (tx), 0 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 172.29.0.20

Source IP Port    (Media): 16668

Destn  IP Address (Media): 10.10.80.252

Destn  IP Port    (Media): 19514

Orig Destn IP Address:Port (Media): 0.0.0.0:0

166513: //2005/FDF54C6DB235/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 302

166514: //2005/FDF54C6DB235/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 7D5

166515: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[599] removed.

166516: //2005/FDF54C6DB235/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.

166517: //2005/FDF54C6DB235/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x85427250 key=FDFB6695-13C911E2-B23BCFEA-48AA9F88@10.10.80.2526********4

166518: //2005/FDF54C6DB235/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.

166519: //2005/FDF54C6DB235/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x85427250 key=FDFB6695-13C911E2-B23BCFEA-48AA9F88@10.10.80.252328E69C0-F67

166520: //2005/FDF54C6DB235/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd

166521: //2005/FDF54C6DB235/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 85427250

166522: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[599]u all

2 Accepted Solutions

Accepted Solutions

paolo bevilacqua
Hall of Fame
Hall of Fame

voice service voip

  no supplementary-service sip moved-temporarily

View solution in original post

Bruno Rangel
Spotlight
Spotlight

Hi jcioara

I had this issue with Disconnect Cause (SIP): 302, this worked when I enabled the following commands:

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

Not sure if is the same for you, anyway if u can show your relevant config someone else can look into this.

PS: Editing: already solved too late from myself!!!

Good Job

Paolo Bevilacqua ( +5)

Cheers
Bruno Rangel
Please remember to rate helpful responses using the star bellow and identify helpful or correct answers

View solution in original post

6 Replies 6

paolo bevilacqua
Hall of Fame
Hall of Fame

voice service voip

  no supplementary-service sip moved-temporarily

You're a genius. What led you to that? Why does the "moved temporarily" kill the connection?

...and where have you been all my life?

Don't worry about.

Thank you for the nice rating and good luck!

FYI - Cisco.com Document ID: 91535 " Cisco CallManager Express (CME) SIP Trunking Configuration Example"  will tell you why the commands in question fix the Disconnect Cause (SIP): 302 and allow pstn sourced calls to CFN to voicemail. 

http://tools.cisco.com/search/results/display?url=http%3a%2f%2fwww.cisco.com%2fen%2fUS%2fcustomer%2fproducts%2fsw%2fvoicesw%2fps4625%2fproducts_configuration_example09186a00808f9666.shtml&pos=1

Bruno Rangel
Spotlight
Spotlight

Hi jcioara

I had this issue with Disconnect Cause (SIP): 302, this worked when I enabled the following commands:

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

Not sure if is the same for you, anyway if u can show your relevant config someone else can look into this.

PS: Editing: already solved too late from myself!!!

Good Job

Paolo Bevilacqua ( +5)

Cheers
Bruno Rangel
Please remember to rate helpful responses using the star bellow and identify helpful or correct answers

Bruno -

I needed your second command there too (no supplementary-service sip refer)! After Paolo's submission, the AutoAttendant answered, but could not perform a dial-by-name transfer.

Thanks again!

Jeremy