03-12-2012 07:08 AM - edited 03-16-2019 10:04 AM
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
Solved! Go to Solution.
03-14-2012 08:22 AM
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
03-12-2012 07:29 AM
Make sure the codec is what is needed from remote site to UM server or use transcoders at the main location.
HTH,
Chris
03-12-2012 08:36 AM
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
So I think that we are using a valid codec.
03-12-2012 08:53 AM
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
03-13-2012 04:38 AM
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:6227@10.37.50.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95
Remote-Party-ID: "Stephan Bergmann" <3236484207>;party=calling;screen=no;privacy=off3236484207>
From: "Stephan Bergmann" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>6227>
Date: Tue, 13 Mar 2012 09:45:26 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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: <323648>323648>
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" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>;tag=C86C19A8-1AF16227>
Date: Tue, 13 Mar 2012 10:55:44 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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:6227@10.37.50.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95
From: "Stephan Bergmann" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>;tag=C86C19A8-1AF16227>
Date: Tue, 13 Mar 2012 09:45:26 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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:
TO: <10.37.50.20:5060>10.37.50.20:5060>
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:
To: <10.37.50.20:5060>;tag=C86C2614-72010.37.50.20:5060>
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:6227@10.37.50.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95
Remote-Party-ID: "Stephan Bergmann" <3236484207>;party=calling;screen=no;privacy=off3236484207>
From: "Stephan Bergmann" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>6227>
Date: Tue, 13 Mar 2012 09:45:26 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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: <3236484207>3236484207>
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" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>;tag=C86C19A8-1AF16227>
Date: Tue, 13 Mar 2012 10:55:44 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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:6227@10.37.50.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.37.0.25:5060;branch=z9hG4bK9CAE95
From: "Stephan Bergmann" <3236484207>;tag=C8456D40-24653236484207>
To: <6227>;tag=C86C19A8-1AF16227>
Date: Tue, 13 Mar 2012 09:45:26 GMT
Call-ID: 18BE4C78-6C2811E1-BB7BFCE0-7412EC76@10.37.0.25
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:
TO: <10.37.0.25:5060>10.37.0.25:5060>
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:
To: <10.37.0.25:5060>;tag=C84579C4-215C10.37.0.25:5060>
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
03-13-2012 06:28 AM
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
03-13-2012 07:28 AM
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
03-13-2012 07:32 AM
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
03-14-2012 08:18 AM
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.
03-14-2012 08:22 AM
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
03-15-2012 02:29 AM
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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