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CME - Windows Unifed Messaging Server

Answered Question
Mar 12th, 2012
User Badges:

Hello,


I have a big problem and I hope that someone can help me.

We have a CME 2921 which has a Sip-Trunk to a Windows Unified Messaging Server (Exchange 2010).

Calls from outside can can be redierected to the Voicemailbox on the Exchange Server and calls from the internal users on the CME can call the Voicemailbox.


!!But calls over a IP from another CME (VPN connected) get a disconnect when you redirected it to the voicemail!!

And I do not known why?



Config for the Sip-Trunk


dial-peer voice 6669 voip

description ** Exchange Unified Messaging **

destination-pattern 6669

session protocol sipv2

session target ipv4:10.37.0.63

session transport tcp

voice-class codec 1

dtmf-relay rtp-nte sip-notify h245-alphanumeric

no vad



voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g711alaw

codec preference 4 g726r16

codec preference 5 g726r24

codec preference 6 g726r32

codec preference 7 g729br8



Here, you have a Error from the CCSIP all debug modus


*Mar  8 15:44:58.036: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceHandleConnAgeing: Holder=0x2A7EA544 Connection=0x296A33B0, addr=10.37.0.63, port=17703, connid=2 has been REFRESHED

*Mar  8 15:44:58.036: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetAgeingTimer: Aging timer initiated for holder=0x2A7EA544,addr=10.37.0.63

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Info/sip_tcp_sockerror_to_spi: Sending TCP Remote Closure to SPI, connid: 3

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 55

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x30B127C4, addr=10.37.48.24, port=38226, connid=3, transport=TCP

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipCreateConnInstance: Created new accptd conn=0x296A347C, connid=3, addr=10.37.48.24, port=38226, transport=TCP

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 56

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWSocketException: context=0x0

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessSocketExceptions: gConnTab=0x30B127C4, addr=10.37.48.24, port=38226, connid=3, transport=TCP

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostCloseConnection: Posting TCP conn close for addr=10.37.48.24, port=38226, connid=3

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Transport/sipDeleteConnInstance: Deleted conn=0x296A347C, connid=3, addr=10.37.48.24, port=38226, transport=TCP

*Mar  8 15:45:13.824: //-1/xxxxxxxxxxxx/SIP/Info/sip_tcp_purge_entry: Socket fd: 2 closed for connid 3 with address: 10.37.48.24, remote port: 38226

*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x3147C3B0) with key=[32032] to table

*Mar  8 15:45:32.076: //34271/000000000000/SIP/State/sipSPIChangeState: 0x3147C3B0 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)

*Mar  8 15:45:32.076: //34271/000000000000/SIP/Info/ccsip_call_setup_request: Before processing SETUP REQccb->pld.flags_ipip = 200

*Mar  8 15:45:32.076: //34271/000000000000/SIP/Info/ccsip_call_setup_request:

This a IPIP call: Chan 0, codec 16 channel 19430, ip 10.37.0.25:19430  params 0x2A721C30 caps 0x2A9CC9D4

*Mar  8 15:45:32.076: //34271/000000000000/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-H323

*Mar  8 15:45:32.076: //34271/000000000000/SIP/Info/ccsip_call_setup_request: After processing SETUP REQccb->pld.flags_ipip = 200000

*Mar  8 15:45:32.076: //34271/000000000000/SIP/Info/ccsip_call_setup_request: Copy over rediectNumber from call info

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/ccsip_call_setup_request: Copy over rediectNumber from ssInfo

*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : 10.37.0.63 target_port : 5065


*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/ccsip_call_setup_request: Incrementing call counter in dial-peer [6669]

*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 2

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 85DF to table

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec    bytes: 0

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: Media forking disabled

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIGetCallConfig: Media forking disabled

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/preprocessSetup:

This is a not a SIGO Call -, could be DM call

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_call_setup: No video caps posted by peer

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_call_setup: xcoder high-density disabled

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_call_setup: Flow Mode set to FLOW_THROUGH

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:

callid 34271, channels 0x31DC3C20 caps 0x2A9CC9D4

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: Peer cap provided: callid = 34271, peer dtmf = 0

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: callid = 34271, peer not doing RFC2833, peer dtmf = 0, enable NTE_ASSUMED

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_codec_byte_transrating: NOT SIP-SIP CALL. Will be addressed in future.

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/codec_found:

Codec to be matched: 16

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:

need transcoding for codec mismatch

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIDtmfTranscoder: Return upon SCCP version 0

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPISrtpTranscoder:

Checking if transcoder is needed for SRTP-RTP

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:65, category:278

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Error/sipSPI_ipip_copy_channelInfo_to_sdp:

filter mis-match, failing call

*Mar  8 15:45:32.076: //34271/9275AEF081D3/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(65) for outgoing call

*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[34271], src[6]

*Mar  8 15:45:32.076: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!!

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/State/sipSPIChangeState: 0x3147C3B0 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPISrtpTranscoder:

Checking if transcoder is needed for SRTP-RTP

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_call_setup: Container with extended caps previewinfo for peer callid 34270 removed

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.37.50.20

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 29712 for stream 1

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Error/sipSPIAddSDPMediaPayload: Call Origination Failed: None of the selected codec from CLI is supported by SIP

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Error/sipSPIOutgoingCallSDP: Error with codec types on media line : 1

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Error/sipSPICreateOutboundSDP: Error in creating an SDP for the outbound call - Check for supported codecs

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Error/preprocessSetup: Error during outbound SDP creation

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:47, category:180

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(47) for outgoing call

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!!

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/State/sipSPIChangeState: 0x3147C3B0 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_spi_process_ccapi_event: CCAPI Event Preprocessor Failure

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/State/sipSPIChangeState: 0x3147C3B0 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DEAD, SUBSTATE_NONE)

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x3147C3B0

State of The Call        : STATE_DEAD

TCP Sockets Used         : YES

Calling Number           : XXXXXXXX4207

Called Number            : 6669

Source IP Address (Sig  ): 10.37.50.20

Destn SIP Req Addr:Port  : 0

Destn SIP Resp Addr:Port : 0

Destination Name         :


*Mar  8 15:45:32.080: //34271/9275AEF081D3/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): 10.37.50.20

Source IP Port    (Media): 29712

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0


*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 47

Disconnect Cause (SIP)   : 200


*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 85DF

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[32032] removed.

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/ccsip_qos_cleanup: Entry

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISipSdpFree:

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed

*Mar  8 15:45:32.080: //34271/9275AEF081D3/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 3147C3B0

*Mar  8 15:45:32.080: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[32032]exit

Correct Answer by Chris Deren about 5 years 5 months ago

Instead of hard coding the codec on these dial peers can you use the same codec class listing the same codecs on both CMEs?

When you connect to UM from the main site and check the call details (i.e. double click ? on 79XX phone) what codec is showing?


Chris

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Chris Deren Mon, 03/12/2012 - 07:29
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Make sure the codec is what is needed from remote site to UM server or use transcoders at the main location.


HTH,


Chris

Janos.nagy Mon, 03/12/2012 - 08:36
User Badges:

Hello Chris, thanks for your fast answer.


this is the Dial Peer from the other CME to reach my CME, you can see that we are using the codec 711ulaw

dial-peer voice 6200 voip
description NYC ext  62[0-5].
destination-pattern 62[0-5].
session protocol sipv2
session  target ipv4:10.37.50.20
session transport tcp
codec g711ulaw


So I think that we are using a valid codec.

Chris Deren Mon, 03/12/2012 - 08:53
User Badges:
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  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

Can you apply the same codec class on this dial peer that you have on the main location?


Can you post "debug ccsip messages" from both CMEs?


Can you ping the UM server IP address from the remote site? 

Do you have call-forward pattern define on CME at the main site?

Is there any firewall between the systems and/or betweem CME and MSFT UM?


Chris

Janos.nagy Tue, 03/13/2012 - 04:38
User Badges:

Okay Chris,

we have check same of your issues.


1. We can make a Ping successfully.

2. There is no firewallrules between the CME and the Voicemailserver.


But I do not understand, what you mean with pattern define




Here the Log from the CME which was the error


*Mar 13 10:55:42.976: %SEC-6-IPACCESSLOGNP: list 23 permitted 0 10.37.48.24 -> 0.0.0.0, 5 packets

*Mar 13 10:55:44.128: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95

Remote-Party-ID: "Stephan Bergmann" ;party=calling;screen=no;privacy=off

From: "Stephan Bergmann" ;tag=C8456D40-2465

To:

Date: Tue, 13 Mar 2012 09:45:26 GMT

Call-ID: [email protected]

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

Min-SE:  1800

Cisco-Guid: 0407524378-1814565345-3145137376-1947397238

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

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

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1331631926

Contact:

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 209


v=0

o=CiscoSystemsSIP-GW-UserAgent 9907 4481 IN IP4 10.37.0.25

s=SIP Call

c=IN IP4 10.37.0.25

t=0 0

m=audio 18466 RTP/AVP 0 19

c=IN IP4 10.37.0.25

a=rtpmap:0 PCMU/8000   

a=rtpmap:19 CN/8000

a=ptime:20


*Mar 13 10:55:44.132: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent: 

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95

From: "Stephan Bergmann" ;tag=C8456D40-2465

To: ;tag=C86C19A8-1AF1

Date: Tue, 13 Mar 2012 10:55:44 GMT

Call-ID: [email protected]

Timestamp: 1331631926

CSeq: 101 INVITE

Allow-Events: telephone-event

Warning: 304 10.37.50.20 "Media Type(s) Unavailable"

Reason: Q.850;cause=65

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0



*Mar 13 10:55:44.224: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95

From: "Stephan Bergmann" ;tag=C8456D40-2465

To: ;tag=C86C19A8-1AF1

Date: Tue, 13 Mar 2012 09:45:26 GMT

Call-ID: [email protected]

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0



*Mar 13 10:55:47.312: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:10.37.50.20:5060 SIP/2.0

FROM: ;epid=42F9AAFAA9;tag=62817d244e

TO:

CSEQ: 5772 OPTIONS

CALL-ID: 792aab09f0284da59cfe5d5301c66155

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.37.0.63:20715;branch=z9hG4bKc2493d2

ACCEPT: application/sdp

CONTACT:

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.5.0.0



*Mar 13 10:55:47.324: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.37.0.63:20715;branch=z9hG4bKc2493d2

From: ;epid=42F9AAFAA9;tag=62817d244e

To: ;tag=C86C2614-720

Date: Tue, 13 Mar 2012 10:55:47 GMT

Call-ID: 792aab09f0284da59cfe5d5301c66155

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 5772 OPTIONS

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

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

Allow-Events: telephone-event

Accept: application/sdp

Content-Type: application/sdp

Content-Length: 164


v=0

o=CiscoSystemsSIP-GW-UserAgent 5708 1605 IN IP4 10.37.50.20

s=SIP Call

c=IN IP4 10.37.50.20

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 10.37.50.20




________________________________________________________________________________



Here the Log from the other CME (caller)







Mar 13 09:45:26.840: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0


Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95


Remote-Party-ID: "Stephan Bergmann" ;party=calling;screen=no;privacy=off


From: "Stephan Bergmann" ;tag=C8456D40-2465


To:


Date: Tue, 13 Mar 2012 09:45:26 GMT


Call-ID: [email protected]


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


Min-SE:  1800


Cisco-Guid: 0407524378-1814565345-3145137376-1947397238


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


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


CSeq: 101 INVITE


Max-Forwards: 70


Timestamp: 1331631926


Contact:


Expires: 180


Allow-Events: telephone-event


Content-Type: application/sdp


Content-Disposition: session;handling=required


Content-Length: 209




v=0


o=CiscoSystemsSIP-GW-UserAgent 9907 4481 IN IP4 10.37.0.25


s=SIP Call


c=IN IP4 10.37.0.25


t=0 0


m=audio 18466 RTP/AVP 0 19


c=IN IP4 10.37.0.25


a=rtpmap:0 PCMU/8000


a=rtpmap:19 CN/8000


a=ptime:20



Mar 13 09:45:26.928: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 488 Not Acceptable Media


Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95


From: "Stephan Bergmann" ;tag=C8456D40-2465


To: ;tag=C86C19A8-1AF1


Date: Tue, 13 Mar 2012 10:55:44 GMT


Call-ID: [email protected]


Timestamp: 1331631926


CSeq: 101 INVITE


Allow-Events: telephone-event


Warning: 304 10.37.50.20 "Media Type(s) Unavailable"


Reason: Q.850;cause=65


Server: Cisco-SIPGateway/IOS-12.x


Content-Length: 0





Mar 13 09:45:26.936: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:[email protected]:5060 SIP/2.0


Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95


From: "Stephan Bergmann" ;tag=C8456D40-2465


To: ;tag=C86C19A8-1AF1


Date: Tue, 13 Mar 2012 09:45:26 GMT


Call-ID: [email protected]


Max-Forwards: 70


CSeq: 101 ACK


Allow-Events: telephone-event


Content-Length: 0





Mar 13 09:45:29.948: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:10.37.0.25:5060 SIP/2.0


FROM: ;epid=42F9AAFAA9;tag=2d8631b3f0


TO:


CSEQ: 5771 OPTIONS


CALL-ID: 90d093df9e7b497dab606e7ec6035fb1


MAX-FORWARDS: 70


VIA: SIP/2.0/TCP 10.37.0.63:20716;branch=z9hG4bKeae8c9fb


ACCEPT: application/sdp


CONTACT:


CONTENT-LENGTH: 0


USER-AGENT: RTCC/3.5.0.0





Mar 13 09:45:29.952: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK


Via: SIP/2.0/TCP 10.37.0.63:20716;branch=z9hG4bKeae8c9fb


From: ;epid=42F9AAFAA9;tag=2d8631b3f0


To: ;tag=C84579C4-215C


Date: Tue, 13 Mar 2012 09:45:29 GMT


Call-ID: 90d093df9e7b497dab606e7ec6035fb1


Server: Cisco-SIPGateway/IOS-12.x


CSeq: 5771 OPTIONS


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


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


Allow-Events: telephone-event


Accept: application/sdp


Content-Type: application/sdp


Content-Length: 163




v=0


o=CiscoSystemsSIP-GW-UserAgent 2895 2316 IN IP4 10.37.0.25


s=SIP Call


c=IN IP4 10.37.0.25


t=0 0


m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3


c=IN IP4 10.37.0.25


no      no          no lax#debug ccsip messages

                      ^

% Invalid input detected at '^' marker.


telco1.corp-bo.lax#no deb

telco1.corp-bo.lax#no debug ccs

telco1.corp-bo.lax#no debug ccs   sip

telco1.corp-bo.lax#no debug ccsip mess

telco1.corp-bo.lax#no debug ccsip messages

Chris Deren Tue, 03/13/2012 - 06:28
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This is definitely a codec issue, see the following messages:


SIP/2.0 488 Not Acceptable Media


Warning: 304 10.37.50.20 "Media Type(s) Unavailable"



Your SDP message contains G711 as seen here:

a=rtpmap:0 PCMU/8000



Did you try applying the same codec class on the remote CME as you are using on the head end CME?


Chris

Janos.nagy Tue, 03/13/2012 - 07:28
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Hey Chris, how can I check the codec, because I have entered on every Dial-paar "codec g711ulaw"

But I have copied the Voice class codec from the other CME to mine, but I do not use the voice class codec


voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8



Here is the config dial-peer to reach my CME

dial-peer voice 6200 voip

description NYC ext 62[0-5].

destination-pattern 62[0-5].

session protocol sipv2

session target ipv4:10.37.50.20

session transport tcp

codec g711ulaw




Here is the config from my CME to reach the mailbox


dial-peer voice 6669 voip

description ** Exchange Unified Messaging **

destination-pattern 6669

session protocol sipv2

session target ipv4:10.37.0.63:5065

session transport tcp

dtmf-relay rtp-nte sip-notify h245-alphanumeric

codec g711ulaw

Chris Deren Tue, 03/13/2012 - 07:32
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Perhaps it's matching a default voip dial peer on the incoming side, add the following:


dial-peer voice 6200 voip

incoming called-number 6...



Chris

Janos.nagy Wed, 03/14/2012 - 08:18
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Hey Chris,


I have added your advice to the dial-peer, but it does not help to solve the porblem.

Did you have another advice for me? Perhaps an other logfile where I can check this isssue?



Our problem is still


Calls from another CME (VPN) gets a disconnect, when the call is forwarded to the voicemail

But extrenal and internal calls can be call-forward to the voicemail, without any problems

And you can forwared calls to external attendants.

Correct Answer
Chris Deren Wed, 03/14/2012 - 08:22
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Instead of hard coding the codec on these dial peers can you use the same codec class listing the same codecs on both CMEs?

When you connect to UM from the main site and check the call details (i.e. double click ? on 79XX phone) what codec is showing?


Chris

Janos.nagy Thu, 03/15/2012 - 02:29
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You are my hero!


I found a message in the logs.


*Mar 15 10:13:18.660: //-1/808C477C84AB/SIP/Info/sipSPIGetCallConfig: Peer tag 1000 matched for incoming call

*Mar 15 10:13:18.660: //-1/808C477C84AB/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header

*Mar 15 10:13:18.660: //-1/808C477C84AB/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header

*Mar 15 10:13:18.660: //-1/808C477C84AB/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec


And after that I entered in every Dial-peer "Voice-class codec 1", and it works.

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