Skype for SIP Configuration

Answered Question
Dec 10th, 2009
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Hello,


I've been trying to get this Skype for SIP trunk configured but have had no success. Can't make or recieve calls although the SIP trunk is showing as registered.


Debug ccsip messages shows the following output. I highlighted some errors messages that are obvious. Can anyone help me?



===================================================================================================

978481: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
From: <sip:[email protected]>;tag=BA003490-1F25
To: <sip:[email protected]>;tag=05aed4eb43523e287156e2da6464d890.fb34
Call-ID: D10F7D71-E2D411DE-AD10C05A-8E0992B3
CSeq: 2366 REGISTER
Via: SIP/2.0/UDP 173.161.141.9:5060;branch=z9hG4bK639525C5
WWW-Authenticate: Digest realm="sip.skype.com", nonce="4b20f6f1000107cefee269e09ad178fa87362e016ba26eba", algorithm=MD5
Server: OpenSIPS
Content-Length: 0



978482: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.skype.com:5060 SIP/2.0
Date: Thu, 10 Dec 2009 13:25:39 GMT
Authorization: Digest username="99051000000916",realm="sip.skype.com",uri="sip:sip.skype.com:5060",response="7ea1fe67ac36d4149fb89b02f5bb189c",nonce="4b20f
6f1000107cefee269e09ad178fa87362e016ba26eba",algorithm=MD5
From: <sip:[email protected]>;tag=BA003490-1F25
Timestamp: 1260451539
Content-Length: 0
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Expires: 3600
Call-ID: D10F7D71-E2D411DE-AD10C05A-8E0992B3
Via: SIP/2.0/UDP 173.161.141.9:5060;branch=z9hG4bK639667F
CSeq: 2367 REGISTER
Max-Forwards: 70



UC500-CUE#
978483: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
From: <sip:[email protected]>;tag=BA003490-1F25
To: <sip:[email protected]>;tag=05aed4eb43523e287156e2da6464d890.0f10
Call-ID: D10F7D71-E2D411DE-AD10C05A-8E0992B3
CSeq: 2367 REGISTER
Via: SIP/2.0/UDP 173.161.141.9:5060;branch=z9hG4bK639667F
Server: OpenSIPS
Content-Length: 0

Correct Answer by brian.russell31... about 7 years 5 months ago

Hi Guys,

I have the Skype for Sip working well receiving and making calls...

However when the voicemail is supposed to kick in I receive a message saying "There is no mailbox associated with this extension" which is not the case.

When I call the extension from another one it drops into vmail as it should...

Can someone help?

Thanks

Brian

Correct Answer by Maulik Shah about 7 years 5 months ago

The translation for an internal MWI SIP message - do not bother trying to decipher it. The root of the issue is a known issue with CCA when the leading access code digit (9 in your case for FXO) matches the leading digit for DID (9905xxxxxxxx).


http://www.cisco.com/cgi-bin/Support/Bugtool/launch_bugtool.pl - ID is CSCtc98096.


Workaround via CLI for now that you can try:


no access-list 1

access-list 1 permit 64.34.175.158
access-list 1 permit 192.168.10.1
access-list 1 remark CCA_SIP_SOURCE_GROUP_ACL
access-list 1 remark SDM_ACL Category=1
access-list 1 permit 63.209.144.201

!

access-list 2 permit 10.1.10.0 0.0.0.3
access-list 2 permit 192.168.10.0 0.0.0.255
access-list 2 permit 10.1.1.0 0.0.0.255
access-list 2 deny   any

!

voice source-group CCA_SIP_SOURCE_GROUP
access-list 1

!

voice source-group CCA_SIP_CUE_WORKAROUND

access-list 2

translation-profile incoming SIP_Incoming


See if that fixes the issue and also ensure MWI works.

Correct Answer by Maulik Shah about 7 years 5 months ago

I think this may have to do with the access-code 9 being in use - as a test can you do the below CLI and re test inbound calls:


config term

voice source-group CCA_SIP_SOURCE_GROUP
no translation-profile incoming SIP_Incoming


Note this may impact MWI for now but its only to test a theory. If this works then will take it from there.

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Maulik Shah Thu, 12/10/2009 - 09:19
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The below log seems to show registration is failing but not sure if that is relevant as you are trying to register an internal extension (1105) which will fail. What would help is to provide logs for a failed call inbound or outbound based on section 5.3 on guide below:


https://www.myciscocommunity.com/servlet/JiveServlet/previewBody/10787-102-2-18503/Skype4sip-uc500.pdf

jaydien1358 Thu, 12/10/2009 - 12:25
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Ok i see.


In my CUE GUI I can see where the E.164 is set to register for most of my extensions. I guess that would explain those messages now. I went ahead and turned that off for those extensions. Not quite sure what that does anyways?


I've attached the logs as per the troubleshooting section.


Thanks for your help.


Brian

www.jaydien.com

Maulik Shah Fri, 12/11/2009 - 11:56
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The log shows nothing in there for SIP calls in or out. Not sure if you ran the log when you made a failed call. Also please gather logs when you make one failed inbound call and separate log for one failed outbound call and make a note of these.

jaydien1358 Fri, 12/11/2009 - 13:42
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Not sure what happened with the logs?


After some many hours of tinkering, I have got outbound SIP calls to work. But inbound sip calls are not processed as they should. My inbound dial plan is set to dump SIP trunk calls to the AA, but the UC doesn't accesp the calls.


Here is the output from from my Debug ccsip message. I will resubmit my logs also.


Do you see anything here that would indicate why the UC would not handoff the call to the AA?

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++


Received:
INVITE sip:[email protected]:5060 SIP/2.0
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 INVITE
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af02-4b22bbc4-7934f6c9-6368e698
Max-Forwards: 12
User-Agent: sipgw-1.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
Contact:
Content-Type: application/sdp
Content-Length: 268


v=0
o=+17326745057 1260567491 1260567491 IN IP4 63.209.144.201
s=Skype call
c=IN IP4 63.209.144.201
t=0 0
m=audio 28844 RTP/AVP 18 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 pcma/8000
a=fmtp:18 annexb=no


008391: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
008392: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x87DE8F80) with key=[406] to table
008393: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 63.209.144.201,Port 5060, Transport 1, SentBy Port 5060
008394: //-1/50C8BB4E8160/SIP/State/sipSPIChangeState: 0x87DE8F80 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
008395: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 63.209.144.201,Port 5060, Transport 1, SentBy Port 5060
008396: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone EST to SIP default timezone = GMT
008397: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 63.209.144.201,Port 5060, Transport 1, SentBy Port 5060
008398: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
008399: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCSTA: No CSTA found in inbound container
008400: //-1/50C8BB4E8160/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
008401: //-1/50C8BB4E8160/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x87DE8F80 key=CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075990510
00000916
008402: //-1/50C8BB4E8160/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
008403: //-1/50C8BB4E8160/SIP/Info/sipSPIMatchSrcIpGroup: Match found on access list
008404: //-1/50C8BB4E8160/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name sip_profile:99051000000916, number +17326745057, Calling oct3 0x
00, oct_3a 0x80, Called number ABCD99051000000916
008405: //-1/50C8BB4E8160/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : +17326745057
008406: //-1/50C8BB4E8160/SIP/Info/sipSPIGetFromCalledPartyId: P-Called-Party-ID header not found
008407: //-1/50C8BB4E8160/SIP/Info/sipSPIGetPeerByCalledPartyId: P-Called-Party-ID not found or parse error
008408: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
008409: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Peer tag 1003 matched for incoming call
008410: //-1/50C8BB4E8160/SIP/Error/sipSPICheckReliableProvStringtag: Unable to access supported header values
008411: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
008412: //-1/50C8BB4E8160/SIP/Error/sipSPICheckReliableProvStringtag: Unable to access supported header values
008413: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
008414: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec
008415: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
008416: //-1/50C8BB4E8160/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
008417: //-1/50C8BB4E8160/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
008418: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled
008419: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH
008420: //-1/50C8BB4E8160/SIP/Info/sipSPIGetCallConfig: Media forking disabled
008421: //-1/50C8BB4E8160/SIP/Info/sipSPIContinueNewMsgInvite: Calling name sip_profile:99051000000916, number +17326745057, Calling oct3 0x00, oct_3a 0x80
, ext_priv 0x00, Called number 99051000000916, oct3 0x00
008422: //-1/50C8BB4E8160/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
008423: //-1/50C8BB4E8160/SIP/Error/sipSPICheckReliableProvStringtag: Unable to access supported header values
008424: //-1/50C8BB4E8160/SIP/Info/sipSPIValidateRequestUri: Not Enabled
008425: //-1/50C8BB4E8160/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
008426: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_IsSDPPassthruEnabled:  - 0
008427: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
008428: //462/50C8BB4E8160/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
008429: //462/50C8BB4E8160/SIP/Info/sipSPIProcessDiversionHeader: No diversion headers recvd from app container
008430: //462/50C8BB4E8160/SIP/Info/sipSPIProcessReplacesHeader: No replaces hdr found
008431: //462/50C8BB4E8160/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
SIP: (462) Attribute mid, level 1 instance 1 not found.
008432: //462/50C8BB4E8160/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.161.141.9
008433: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec No Codec
008434: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 96
008435: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved
                          as its in use by other codec No Codec
008436: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (101) is  reserved by another application
008437: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 98
008438: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 101
008439: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 98 for Codec:
008440: //462/50C8BB4E8160/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
008441: //462/50C8BB4E8160/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or multiple ptime attributes that can't be handled
008442: //462/50C8BB4E8160/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
008443: //462/50C8BB4E8160/SIP/Info/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!
008444: //462/50C8BB4E8160/SIP/Info/sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode: Inband Voice
008445: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: V150 NSE payload = 0, SSE payload = 0, SPRT payload=0
008446: //462/50C8BB4E8160/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
008447: //462/50C8BB4E8160/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled f
or m-line:1 and num-a-lines:0
008448: //462/50C8BB4E8160/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
        payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=inband-voice
        stream_type=voice-only (0), dest_ip_address=63.209.144.201, dest_port=28844
008449: //462/50C8BB4E8160/SIP/State/sipSPIChangeStreamState: Stream (callid =  -1)  State changed from (STREAM_DEAD) to (STREAM_ADDING)
008450: //462/50C8BB4E8160/SIP/Media/sipSPIUpdCallWithSdpInfo:
        Preferred Codec        : g711ulaw, bytes :160
        Preferred  DTMF relay  : sip-notify
        Preferred NTE payload  : 98
        Early Media            : No
        Delayed Media          : No
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No


008451: //462/50C8BB4E8160/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
008452: //462/50C8BB4E8160/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.161.141.9
008453: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 462 peer 0 flags 0x201 state STATE_IDLE
008454: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 462, Peer CallID 0, sdp 0x86A93AEC channels 0x87DEA228
008455: //462/50C8BB4E8160/SIP/Info/copy_channels:
callId 462 size 0 ptr 0x85C10B1C)
008456: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 18 mline 1
008457: //462/50C8BB4E8160/SIP/Media/sipSPISelectCodecVersion: Codec (g729r8) is not in preferred list
008458: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8 pre-ietf
008459: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8 pre-ietf
008460: //462/50C8BB4E8160/SIP/Info/codec_found:
Codec to be matched: 0
008461: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 0 mline 1
008462: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
008463: //462/50C8BB4E8160/SIP/Info/codec_found:
Codec to be matched: 5
008464: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5


008465: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
008466: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=0,stream->negotiated_codec_bytes=
160, coverted ptime=20 stream->mline_index=1, media_ndx=1
008467: //462/50C8BB4E8160/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
008468: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 5 ptype 0 time 20, bytes 160  as channel 0 mline 1 ss 1 63.209.144.201:28844
008469: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
008470: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
008471: //462/50C8BB4E8160/SIP/Info/codec_found:
Codec to be matched: 6
008472: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
008473: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_n
dx].codec = 5


008474: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_n
dx].codec = -1


008475: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 462 flags 0x100 state STATE_IDLE
008476: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
008477: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0xC, ccb->pld.flags_ipip 0x201


008478: //462/50C8BB4E8160/SIP/Info/copy_channels:
callId 462 size 240 ptr 0x8A0DEE04)
008479: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
008480: //462/50C8BB4E8160/SIP/Info/ccsip_update_srtp_caps:  5712: Posting Remote SRTP caps to other callleg.
008481: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()
008482: //462/50C8BB4E8160/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : -1
          Negotiated Codec       : g711ulaw, bytes :160
          Nego. Codec payload    : 0 (tx), 0 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [173.161.141.9]:0
          Media Dest Addr/Port   : [63.209.144.201]:28844


008483: //462/50C8BB4E8160/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec       : g711ulaw, bytes :160
Preferred Codec        : g711ulaw, bytes :160
Preferred  DTMF relay 1 : 8
Preferred  DTMF relay 2 : 0
Negotiated DTMF relay   : 0
Preferred and Negotiated NTE payloads: 98 0
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated V150.1 Modem Passthrough: 0 0
Preferred and Negotiated V150.1 Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0


008484: //462/50C8BB4E8160/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: Entry
008485: //462/50C8BB4E8160/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: QOS negotiation for mline_index 1
008486: //462/50C8BB4E8160/SIP/Info/sipSPIDoStreamQoSNegotiation: Best effort
008487: //462/50C8BB4E8160/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
008488: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17352 for stream 1
008489: //462/50C8BB4E8160/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17352
008490: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
008491: //462/50C8BB4E8160/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17352
008492: //462/50C8BB4E8160/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 5, payload type 0


008493: //462/50C8BB4E8160/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
008494: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCPA: No CPA found in inbound container
008495: //462/50C8BB4E8160/SIP/Info/sipSPIProcessCPA: No x-cisco-cpa content found
008496: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
008497: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_IsContentPassthruEnabled:  - 0
008498: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_ExtractPassthruContentFromSipContainer: Passthru Content Not Enabled
008499: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
008500: //462/50C8BB4E8160/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 0,
008501: //462/50C8BB4E8160/SIP/Info/sipSPIShrlCall: Check peer: 1003 for Shared-Line call, callid: 462
008502: //462/50C8BB4E8160/SIP/Info/ccsip_set_bearer_capability:
   Bearer Capability: Speech (0x00)
008503: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
008504: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
008505: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message


008506: //462/50C8BB4E8160/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
008507: //462/50C8BB4E8160/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 1CE to table
008508: //462/50C8BB4E8160/SIP/Info/sipSPISendInviteResponse: Associated container=0x89F5979C to Invite Response 100
008509: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: msg=0x89C62488, addr=63.209.144.201, port=5060, sentBy_port=5060, is_req=0, transport=
1, switch=0, callBack=0x0
008510: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
008511: //462/50C8BB4E8160/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
008512: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x89C62488, addr=63.209.144.201, port=5060, connId=0 for UDP
008513: //462/50C8BB4E8160/SIP/State/sipSPIChangeState: 0x87DE8F80 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, SUBSTATE_NONE)
008514: //462/50C8BB4E8160/SIP/Info/sipSPIProcessContactInfo: Previous Hop 63.209.144.201:5060
008515: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
008516: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
008517: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 164)
008518: //462/50C8BB4E8160/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 463 chans 0x8A046880 event 164 flags 0x40001C 0x100 0x601 data 0x8A046880
008519: //462/50C8BB4E8160/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 463 chans 0x8A046880 event 164 flags 0x40001C 0x100 0x601 data 0x8A046880, type = 1
008520: //462/50C8BB4E8160/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-TDM
008521: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
008522: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3
008523: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af02-4b22bbc4-7934f6c9-6368e698
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>
Date: Fri, 11 Dec 2009 21:38:11 GMT
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0



UC500-CUE#
008524: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
008525: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 164)
008526: //462/50C8BB4E8160/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 464 chans 0x8A046D10 event 164 flags 0x40001C 0x100 0x603 data 0x8A046D10
008527: //462/50C8BB4E8160/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 464 chans 0x8A046D10 event 164 flags 0x40001C 0x100 0x603 data 0x8A046D10, type = 1
008528: //462/50C8BB4E8160/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-TDM
008529: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
UC500-CUE#
008530: //462/50C8BB4E8160/SIP/Info/sip_gw_video_handle_alert: Video caps are not detected in the caps posted by peer leg
008531: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_ALERTING
008532: //462/50C8BB4E8160/SIP/Info/ccsip_bridge: confID = 35, srcCallID = 462, dstCallID = 464
008533: //462/50C8BB4E8160/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 462/464
008534: //462/50C8BB4E8160/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=-1, new streamcallid=462
008535: //462/50C8BB4E8160/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-TDM
008536: //462/50C8BB4E8160/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = -2125326640, ccb xmitFunc = -2125326640
008537: //462/50C8BB4E8160/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
008538: //462/50C8BB4E8160/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 462) to the VOIP RTP library
008539: //462/50C8BB4E8160/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
008540: //462/50C8BB4E8160/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.161.141.9
008541: //462/50C8BB4E8160/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
008542: //462/50C8BB4E8160/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
        laddr = 173.161.141.9, lport = 17352, raddr = 63.209.144.201, rport=28844, do_rtcp=TRUE
        src_callid = 462, dest_callid = 464, stream type = voice-only, stream direction = SENDRECV
        media_ip_addr = 63.209.144.201, vrf tableid = 0 media_addr_type = 1
008543: //462/50C8BB4E8160/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
008544: //462/50C8BB4E8160/SIP/Info/sipSPICreateRtpSession: sess: 8A03843C do_rtcp:1
008545: //462/50C8BB4E8160/SIP/Media/sipSPICreateRtpSession: stun is disabled
008546: //462/50C8BB4E8160/SIP/Info/sipSPICreateAndStartRtpTimer:
008547: //462/50C8BB4E8160/SIP/Info/sipSPICreateAndStartRtpTimer: Media Inactivity Timer is disabled.
008548: //462/50C8BB4E8160/SIP/Media/sipSPIGetNewLocalMediaDirection:
        New Remote Media Direction = SENDRECV
        Present Local Media Direction = SENDRECV
        New Local Media Direction = SENDRECV
        retVal = 0


008549: //462/50C8BB4E8160/SIP/State/sipSPIChangeStreamState: Stream (callid =  462)  State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
008550: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: Entry
008551: //462/50C8BB4E8160/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=462, current_seq_num=0x1931
008552: //462/50C8BB4E8160/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=462, current_seq_num=0x1C15
008553: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711ulaw, Bytes=160
008554: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
008555: //462/50C8BB4E8160/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB
008556: //462/50C8BB4E8160/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=1, from CLI config=0
008557: //462/50C8BB4E8160/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=464
008558: //462/50C8BB4E8160/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
008559: //462/50C8BB4E8160/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
008560: //462/50C8BB4E8160/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
008561: //462/50C8BB4E8160/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
008562: //462/50C8BB4E8160/SIP/Info/sip_set_modem_caps: nse payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy
=200, compres-dir=3, dict=1024, strnlen=32
008563: //462/50C8BB4E8160/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
008564: //462/50C8BB4E8160/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice-only) from media
line 1 codec g711ulaw
008565: //462/50C8BB4E8160/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x1, caps.stream_list.xmitFunc=
008566: //462/50C8BB4E8160/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
008567: //462/50C8BB4E8160/SIP/Media/sipSPISetStreamInfo: 0x8A1B1420 (gccb)
008568: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711ulaw, Bytes=160, payload = 0
008569: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x603
008570: //462/50C8BB4E8160/SIP/Info/sipSPISrtpTranscoder:
Checking if transcoder is needed for SRTP-RTP
008571: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ind: Calling cc_api_caps_ack()
008572: //462/50C8BB4E8160/SIP/Info/ccsip_caps_ack: Set forking flag to 0x7
008573: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_call_service_msg: Outgoing updates for callerid are globally disabled, ignoring update request for callid 462
008574: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 5
008575: //462/50C8BB4E8160/SIP/Error/sipSPIAddCiscoGcid: Fatal Error in parsing CCB/Msg
008576: //462/50C8BB4E8160/SIP/Info/preprocessAlertOrProgress: changing ALERT event to PROGRESS
008577: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIStoreTunnelData: Container /RawMessage Absent
008578: //462/50C8BB4E8160/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
008579: //462/50C8BB4E8160/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
008580: //462/50C8BB4E8160/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x87DE8F80 key=CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-712780754DDFC
A8-1BF3
SIP: (462) Group (a= group line) attribute, level 65535 instance 1 not found.
008581: //462/50C8BB4E8160/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp and dest_sdp available, should be a midcall request
008582: //462/50C8BB4E8160/SIP/Info/sipSPISendInviteResponse: Associated container=0x89F5A76C to Invite Response 183
008583: //462/50C8BB4E8160/SIP/Transport/sipSPISendInviteResponse: Sending 183 Response to the Transport Layer
008584: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: msg=0x85C263E4, addr=63.209.144.201, port=5060, sentBy_port=5060, is_req=0, transport=
1, switch=0, callBack=0x8121071C
008585: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
008586: //462/50C8BB4E8160/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
008587: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x85C263E4, addr=63.209.144.201, port=5060, connId=0 for UDP
008588: //462/50C8BB4E8160/SIP/Info/sentInviteResponse18x: Sent a 18x Response
008589: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af02-4b22bbc4-7934f6c9-6368e698
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>;tag=4DDFCA8-1BF3
Date: Fri, 11 Dec 2009 21:38:11 GMT
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 182


v=0
o=CiscoSystemsSIP-GW-UserAgent 5026 3238 IN IP4 173.161.141.9
s=SIP Call
c=IN IP4 173.161.141.9
t=0 0
m=audio 17352 RTP/AVP 0
c=IN IP4 173.161.141.9
a=rtpmap:0 PCMU/8000


008590: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_call_connect: CCSIP_CALL_CONNECT: ccb ptr 87DE8F80


008591: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_CONNECT
008592: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_call_service_msg: Outgoing updates for callerid are globally disabled, ignoring update request for callid 462
008593: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7
008594: //462/50C8BB4E8160/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
008595: //462/50C8BB4E8160/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
008596: //462/50C8BB4E8160/SIP/Info/sipSPIQoSRevertBW: Entry
008597: //462/50C8BB4E8160/SIP/Info/preprocessConnect: Write sdp_info into msg_body
008598: //462/50C8BB4E8160/SIP/Info/sipSPIAppAddCallInfoUI: callinfo UI update request for callid: 462


008599: //462/50C8BB4E8160/SIP/Info/sipSPIShrlGetInstanceInfo: Obtained the call instance 0 for non-shared-line '' with callid: 462
SIP: (462) Group (a= group line) attribute, level 65535 instance 1 not found.
008600: //462/50C8BB4E8160/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp and dest_sdp available, should be a midcall request
008601: //462/50C8BB4E8160/SIP/Info/sipSPISendInviteResponse: Associated container=0x89F5A5B4 to Invite Response 200
008602: //462/50C8BB4E8160/SIP/Transport/sipSPISendInviteResponse: Sending 200OK Response to the Transport Layer
008603: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: msg=0x85C263E4, addr=63.209.144.201, port=5060, sentBy_port=5060, is_req=0, transport=
1, switch=0, callBack=0x812109F8
008604: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
008605: //462/50C8BB4E8160/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
008606: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x85C263E4, addr=63.209.144.201, port=5060, connId=0 for UDP
008607: //462/50C8BB4E8160/SIP/Info/sentInviteResponse200: Sent 200Ok for Invite in state STATE_RECD_INVITE
008608: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteResponse200: Transaction active. Facilities will be queued.
008609: //462/50C8BB4E8160/SIP/State/sipSPIChangeState: 0x87DE8F80 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_SENT_SUCCESS, SUBSTATE
_NONE)
008610: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af02-4b22bbc4-7934f6c9-6368e698
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>;tag=4DDFCA8-1BF3
Date: Fri, 11 Dec 2009 21:38:11 GMT
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 182


v=0
o=CiscoSystemsSIP-GW-UserAgent 5026 3238 IN IP4 173.161.141.9
s=SIP Call
c=IN IP4 173.161.141.9
t=0 0
m=audio 17352 RTP/AVP 0
c=IN IP4 173.161.141.9
a=rtpmap:0 PCMU/8000


008611: //462/50C8BB4E8160/SIP/Info/sipSPIAppAddCallInfoUI: callinfo UI update request for callid: 462


008612: //462/50C8BB4E8160/SIP/Info/sipSPIShrlGetInstanceInfo: Obtained the call instance 0 for non-shared-line '' with callid: 462
SIP: (462) Group (a= group line) attribute, level 65535 instance 1 not found.
008613: //462/50C8BB4E8160/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp and dest_sdp available, should be a midcall request
008614: //462/50C8BB4E8160/SIP/Info/sipSPISendInviteResponse: Associated container=0x89F5A5B4 to Invite Response 200
008615: //462/50C8BB4E8160/SIP/Transport/sipSPISendInviteResponse: Sending 200OK Response to the Transport Layer
008616: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: msg=0x85C263E4, addr=63.209.144.201, port=5060, sentBy_port=5060, is_req=0, transport=
1, switch=0, callBack=0x0
008617: //462/50C8BB4E8160/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
008618: //462/50C8BB4E8160/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
008619: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x85C263E4, addr=63.209.144.201, port=5060, connId=0 for UDP
008620: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af02-4b22bbc4-7934f6c9-6368e698
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>;tag=4DDFCA8-1BF3
Date: Fri, 11 Dec 2009 21:38:11 GMT
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 182


v=0
o=CiscoSystemsSIP-GW-UserAgent 5026 3238 IN IP4 173.161.141.9
s=SIP Call
c=IN IP4 173.161.141.9
t=0 0
m=audio 17352 RTP/AVP 0
c=IN IP4 173.161.141.9
a=rtpmap:0 PCMU/8000


008621: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [63.209.144.201]:5060
008622: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
008623: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
008624: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
From: "sip_profile:99051000000916" [email protected]>;tag=ca90d13f-13c4-4b22bbc4-7934f6c9-7460287b
To: [email protected]>;tag=4DDFCA8-1BF3
Call-ID: CXC-182-694c0a30-ca90d13f-13c4-4b22bbc4-7934f6c7-71278075
CSeq: 1 ACK
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-4af04-4b22bbc8-793507b4-184fc737
Max-Forwards: 70
Contact:
Content-Length: 0



008625: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
008626: //462/50C8BB4E8160/SIP/Info/sipSPIFindCcbUASRespTable: *****CCB found in UAS Response table. ccb=0x87DE8F80
008627: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 63.209.144.201,Port 5060, Transport 1, SentBy Port 5060
008628: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone EST to SIP default timezone = GMT
008629: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 63.209.144.201,Port 5060, Transport 1, SentBy Port 5060
008630: //462/50C8BB4E8160/SIP/Info/act_sentsucc_new_message_request: Transaction Complete. Lock on Facilities released.
008631: //462/50C8BB4E8160/SIP/State/sipSPIChangeState: 0x87DE8F80 : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE)  to (STATE_ACTIVE, SUBSTATE_NONE
)
008632: //462/50C8BB4E8160/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x87DE8F80
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : +17326745057
Called Number            : 99051000000916
Source IP Address (Sig  ): 173.161.141.9
Destn SIP Req Addr:Port  : 63.209.144.201:5060
Destn SIP Resp Addr:Port : 63.209.144.201:5060
Destination Name         : 63.209.144.201


008633: //462/50C8BB4E8160/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    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 173.161.141.9
Source IP Port    (Media): 17352
Destn  IP Address (Media): 63.209.144.201
Destn  IP Port    (Media): 28844
Orig Destn IP Address:Port (Media): [ - ]:0

Thanks.

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

Brian

www.jaydien.com

Maulik Shah Sun, 12/13/2009 - 22:08
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Are you sure the DID 99051000000916 is mapped to the AA in CCA - the logs below the call probably got routed to one of the FXS ports on the UC500 but I could be wrong. Would help to check the UC500 config and look for the number translation rules and see where the below DID is mapped to. By the way - what happens when you call the below number - you hear ringing and then call is answered but not by the AA?

jaydien1358 Mon, 12/14/2009 - 04:49
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I think you must be right cause it is so strange, when you call in, it rings for a couple times and then the call gets transfered to the Verizon long distance operator. Verizon is the carrier for the FXO trunks.


I will go through the config and post what I see.


Thanks.

-Brian

www.jaydien.com

jaydien1358 Mon, 12/14/2009 - 04:54
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Here are the voice translation rules. The AA extension is 500.



voice translation-rule 4
rule 1 /7325734007/ /500/
!
voice translation-rule 6
rule 1 /99051000000916/ /500/
!
voice translation-rule 410
rule 1 /^9\(.......)\)$/ /990\1/
rule 2 /^8\(.......)\)$/ /990\1/
rule 3 /^9\(.*\)/ /\1/
rule 4 /^8\(.*\)/ /\1/
rule 15 /^...$/ /99051000000916/
!
voice translation-rule 411
rule 1 /^9\(.*\)/ /ABCD9\1/
rule 2 /^8\(.*\)/ /ABCD8\1/
!
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
!
voice translation-rule 1111
rule 15 /.*/ /99051000000916/
!
voice translation-rule 1112
rule 1 /^9/ //
rule 2 /^8/ //
!
voice translation-rule 2001
!
voice translation-rule 2222
rule 1 /^9411/ //
!
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
!
voice translation-profile CallBlocking
translate called 2222
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile SIP_Incoming
translate called 411
!
voice translation-profile SIP_Passthrough
translate called 412
!
voice translation-profile SkypeDefault-AA_Called_6
translate called 6
!
voice translation-profile SkypeIn-AA_Called_4
translate called 4


-Brian

Maulik Shah Mon, 12/14/2009 - 09:24
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Do you mind sending me the entire config (private message it to me if you want) - the logs indicate the call is going out the FXO / FXS port but its not clear exactly why without a full config lookup.

jaydien1358 Mon, 12/14/2009 - 11:29
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One other thing.


When I test the incoming dial-plan in CCA with the DID of the SIP trunk, it shows the matching destination as the AA.


-Brian

jaydien1358 Tue, 12/15/2009 - 06:01
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Any luck finding out what is wrong with my config?


-Brian

Correct Answer
Maulik Shah Tue, 12/15/2009 - 10:50
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I think this may have to do with the access-code 9 being in use - as a test can you do the below CLI and re test inbound calls:


config term

voice source-group CCA_SIP_SOURCE_GROUP
no translation-profile incoming SIP_Incoming


Note this may impact MWI for now but its only to test a theory. If this works then will take it from there.

jaydien1358 Tue, 12/15/2009 - 11:22
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Yes that did the trick!


What exactly does that translation profile do? Am I right in assuming that it related to:


voice translation-rule 411
rule 1 /^9\(.*\)/ /ABCD9\1/
rule 2 /^8\(.*\)/ /ABCD8\1/


but I can't really decifer that code.


-Brian

Correct Answer
Maulik Shah Tue, 12/15/2009 - 22:04
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The translation for an internal MWI SIP message - do not bother trying to decipher it. The root of the issue is a known issue with CCA when the leading access code digit (9 in your case for FXO) matches the leading digit for DID (9905xxxxxxxx).


http://www.cisco.com/cgi-bin/Support/Bugtool/launch_bugtool.pl - ID is CSCtc98096.


Workaround via CLI for now that you can try:


no access-list 1

access-list 1 permit 64.34.175.158
access-list 1 permit 192.168.10.1
access-list 1 remark CCA_SIP_SOURCE_GROUP_ACL
access-list 1 remark SDM_ACL Category=1
access-list 1 permit 63.209.144.201

!

access-list 2 permit 10.1.10.0 0.0.0.3
access-list 2 permit 192.168.10.0 0.0.0.255
access-list 2 permit 10.1.1.0 0.0.0.255
access-list 2 deny   any

!

voice source-group CCA_SIP_SOURCE_GROUP
access-list 1

!

voice source-group CCA_SIP_CUE_WORKAROUND

access-list 2

translation-profile incoming SIP_Incoming


See if that fixes the issue and also ensure MWI works.

jaydien1358 Wed, 12/16/2009 - 04:51
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I applied the change via CLI and inbound calls to the SIP are still working as they should. Also, MWI is still working as well (Although it was working with the quick fix before also).


Thank you for your prompt attention to my issue. Much appreciated.


-Brian

www.jaydien.com

Correct Answer
brian.russell31... Thu, 12/17/2009 - 12:06
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Hi Guys,

I have the Skype for Sip working well receiving and making calls...

However when the voicemail is supposed to kick in I receive a message saying "There is no mailbox associated with this extension" which is not the case.

When I call the extension from another one it drops into vmail as it should...

Can someone help?

Thanks

Brian

jaydien1358 Thu, 12/17/2009 - 12:26
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Does your Voicemail extention start with the same digit as your SIP trunk DID?


-Brian

brian.russell31... Thu, 12/17/2009 - 12:29
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Hi Brian,


No it doesnt, I have left the voicemail number as 399 but I have a skype in number assigned to the Skyp for SIP (+44 131 208 5555) and my skype tunk is 9909905**********


Have I done something wrong?

jaydien1358 Thu, 12/17/2009 - 12:35
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Not quite sure as I am a novice as it pertains to troubleshooting SIP trunks.


I thought your issue might have been related to my orginal issue in this post where my SIP trunk DID started with the same digit as my PSTN access code which caused a conflict.


But I have yet to test inbound calls over my Skype SIP trunk when the call is transfered to VM. Let me see if I can replicate on my system.


-Brian

Steven Smith Thu, 12/17/2009 - 12:37
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What IOS are you running?  Is that call being forwarded from anywhere to get to your UC500?  Can you perform a debug ccsip messages?  If you don't want o post the result here, PM me.

Maulik Shah Sat, 12/19/2009 - 15:48
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Ted Trentler reported something similar that I was not able to reproduce on my Skype for SIP here.


Here is what he stated (all comments are on the Skype for SIP guide at https://www.myciscocommunity.com/docs/DOC-10787


My new issue is this.  When A call is redirected to voicemail from an IPphone Skype DID starting with 9905........... the number translation does not work properly.  Unity Express says that there is no number for that extension.


To get a Skype call to successfully redirect to voicemail I went to the web GUI of Unity Express (by default its http://10.1.10.1 from a PC local to the UC500 LAN) and had to define an E.164 number for the user.  The E.164 number that worked was the 9905.......... number



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