Intermittent diconnect CC cause 47

[Rob Dubois]

Hi,

I got a weird behavior on my UC560. 

We have some extension forwarded to external number we are using as quick dial. Ex 751 goes to 95145555555

About half of the time, when we dial one of those "speed dial", the call disconnects as soon as it's answered by the other party, with disconnect cause CC 47. 

Here's the logs of a non-working call : 

002913: //881/07584A1F816F/SIP/Xlate/ccsip_call_setup_request: Apply outgoing redirect (diversion) Tag(1048) [751] to [751]
002914: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
002915: //881/07584A1F816F/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
002916: //881/07584A1F816F/SIP/Error/sipSPI_ipip_set_history_info_header: ccb->src_addr_str is NULL
002917: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE
002918: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002919: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19296 for stream 1
002920: //881/07584A1F816F/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
002921: //881/07584A1F816F/SIP/Media/sipSPIProcessRtpSessions: No active streams.
002922: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 198.38.7.34,Port 5065, Transport 1, SentBy Port 5065
002923: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
002924: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:8D11577C
SIP: (881) Group (a= group line) attribute, level 65535 instance 1 not found.
002925: //881/07584A1F816F/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
002926: //881/07584A1F816F/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
002927: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: msg=0x8D51D970, addr=198.38.7.34, port=5065, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x814AA734
002928: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
002929: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
002930: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:198.38.7.34, rport:5065 with laddr:

002931: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x8D51D970
002932: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8D51D970, addr=198.38.7.34, port=5065, local_addr=, connId=3 for UDP
002933: //881/07584A1F816F/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 881) to the VOIP RTP library
002934: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002935: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002936: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 173.231.97.100, lport = 19296, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 881, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1
002937: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
002938: //881/07584A1F816F/SIP/Media/sipSPICreateRtpSession: stun is disabled
002939: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:5145555555@sip.babytel.ca:5060 SIP/2.0
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK3941D71
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
To: <sip:5145555555@sip.babytel.ca>
Date: Thu, 11 Feb 2016 17:38:00 GMT
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0123226655-3491566053-2171578581-1041417351
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: 1455212280
Contact: <sip:5146666666@173.231.97.100:5060>
Diversion: <sip:751@sip.babytel.ca>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287

v=0
o=CiscoSystemsSIP-GW-UserAgent 2644 346 IN IP4 173.231.97.100
s=SIP Call
c=IN IP4 173.231.97.100
t=0 0
m=audio 19296 RTP/AVP 0 18 101
c=IN IP4 173.231.97.100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

002940: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x8746D12C
002941: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
002942: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
To: <sip:5145555555@sip.babytel.ca>
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK3941D71
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
CSeq: 101 INVITE
Content-Length: 0

002943: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x8746D12C
002944: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
002945: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
To: <sip:5145555555@sip.babytel.ca>;tag=dlzv2xamcP
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK3941D71
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.babytel.ca", nonce="7rstMl8Xa5yjmlbDvLAGGKLNaTS2Dno0jv32uWh9"
Content-Length: 0

002946: //881/07584A1F816F/SIP/Transport/sipSPISendAck: Sending ACK to the transport layer
002947: //881/07584A1F816F/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
002948: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: msg=0x8D53D9C0, addr=198.38.7.34, port=5065, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x0
002949: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
002950: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
002951: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:198.38.7.34, rport:5065 with laddr:

002952: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x8D53D9C0
002953: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8D53D9C0, addr=198.38.7.34, port=5065, local_addr=, connId=3 for UDP
002954: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
002955: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:8D11577C
SIP: (881) Group (a= group line) attribute, level 65535 instance 1 not found.
002956: //881/07584A1F816F/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
002957: //881/07584A1F816F/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
002958: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: msg=0x8CD80080, addr=198.38.7.34, port=5065, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x814AA734
002959: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
002960: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
002961: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:198.38.7.34, rport:5065 with laddr:

002962: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x8CD80080
002963: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8CD80080, addr=198.38.7.34, port=5065, local_addr=, connId=3 for UDP
002964: //881/07584A1F816F/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 881) to the VOIP RTP library
002965: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002966: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002967: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 173.231.97.100, lport = 19296, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 881, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1
002968: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
002969: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D4C2F40
002970: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:5145555555@sip.babytel.ca:5060 SIP/2.0
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK3941D71
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
To: <sip:5145555555@sip.babytel.ca>;tag=dlzv2xamcP
Date: Thu, 11 Feb 2016 17:38:00 GMT
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

002971: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:5145555555@sip.babytel.ca:5060 SIP/2.0
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK39521C
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
To: <sip:5145555555@sip.babytel.ca>
Date: Thu, 11 Feb 2016 17:38:00 GMT
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0123226655-3491566053-2171578581-1041417351
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1455212280
Contact: <sip:5146666666@173.231.97.100:5060>
Diversion: <sip:751@sip.babytel.ca>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="15147777777",realm="sip.babytel.ca",uri="sip:5145555555@sip.babytel.ca:5060",response="edafbf035fafc273ca3cda76a8b98327",nonce="7rstMl8Xa5yjmlbDvLAGGKLNaTS2Dno0jv32uWh9",algorithm=md5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287

v=0
o=CiscoSystemsSIP-GW-UserAgent 2644 346 IN IP4 173.231.97.100
s=SIP Call
c=IN IP4 173.231.97.100
t=0 0
m=audio 19296 RTP/AVP 0 18 101
c=IN IP4 173.231.97.100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

002972: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x8746D12C
002973: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
002974: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
To: <sip:5145555555@sip.babytel.ca>
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK39521C
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
CSeq: 102 INVITE
Content-Length: 0

002975: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x8746D12C
002976: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
002977: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK39521C
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
To: <sip:5145555555@sip.babytel.ca>;tag=1c2009327053
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
CSeq: 102 INVITE
Contact: <sip:CfN92wASVy@198.38.7.34:5065;transport=UDP>
Supported: em,timer,replaces,path,resource-priority
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.049.003
Content-Type: application/sdp
Content-Length: 254

v=0
o=AudiocodesGW 2009402172 2009401795 IN IP4 198.38.7.34
s=Phone-Call
c=IN IP4 198.38.7.34
t=0 0
m=audio 11686 RTP/AVP 0 101
c=IN IP4 198.38.7.34
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

SIP: Attribute mid, level 1 instance 1 not found.
002978: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002979: //881/07584A1F816F/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw
002980: //881/07584A1F816F/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
002981: //881/07584A1F816F/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No

002982: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002983: //881/07584A1F816F/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 881
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101 (tx), 101 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [173.231.97.100]:19296
Media Dest Addr/Port : [198.38.7.34]:11686

002984: //881/07584A1F816F/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 881) to the VOIP RTP library
002985: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
002986: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002987: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 173.231.97.100, lport = 19296, raddr = 198.38.7.34, rport=11686, do_rtcp=TRUE
src_callid = 881, dest_callid = 880, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 198.38.7.34, vrf tableid = 0 media_addr_type = 1
002988: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
002989: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8CC75578
002990: //881/07584A1F816F/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0

002991: //881/07584A1F816F/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
002992: //881/07584A1F816F/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media
line 1 codec g711ulaw
002993: //881/07584A1F816F/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x3, caps.stream_list.xmitFunc=
002994: //881/07584A1F816F/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
002995: //881/07584A1F816F/SIP/Media/sipSPISetStreamInfo: 0x8CC7CDE0 (gccb)
002996: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x8746D12C
002997: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
002998: //881/07584A1F816F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.231.97.100:5060;branch=z9hG4bK39521C
From: "Stephane Plante" <sip:5146666666@sip.babytel.ca>;tag=2D8948-527
To: <sip:5145555555@sip.babytel.ca>;tag=1c2009327053
Call-ID: 75A1E97-D01D11E5-8173A8D5-3E12C487@sip.babytel.ca
CSeq: 102 INVITE
Contact: <sip:CfN92wASVy@198.38.7.34:5065;transport=UDP>
Supported: em,timer,replaces,path,resource-priority
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.049.003
Content-Type: application/sdp
Content-Length: 254

v=0
o=AudiocodesGW 2009402172 2009401795 IN IP4 198.38.7.34
s=Phone-Call
c=IN IP4 198.38.7.34
t=0 0
m=audio 11686 RTP/AVP 0 101
c=IN IP4 198.38.7.34
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

002999: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
003000: //881/07584A1F816F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:200, container:8D115D54
SIP: Attribute mid, level 1 instance 1 not found.
003001: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
003002: //881/07584A1F816F/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw
003003: //881/07584A1F816F/SIP/Media/sipSPICompareStreams: stream 1 dest_port: old=11686 new=11686
003004: //881/07584A1F816F/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0

003005: //881/07584A1F816F/SIP/Media/sipSPICompareStreams: Flags set for stream 1: RTP_CHANGE=No CAPS_CHANGE=No RSVP_ADDR_CHANGE=No RSVP_MEDIA_CHANGE=No
003006: //881/07584A1F816F/SIP/Media/sipSPICompareSDP: Flags set for call: NEW_MEDIA=No DSPDNLD_REQD=No IPIP_MEDIA=No
003007: //881/07584A1F816F/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
003008: //881/07584A1F816F/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : Yes
New Media : No
DSP DNLD Reqd : No

003009: //881/07584A1F816F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.231.97.100
003010: //881/07584A1F816F/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ACTIVE (5)
Stream address type : 1
Callid : 881
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101 (tx), 101 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [173.231.97.100]:19296
Media Dest Addr/Port : [198.38.7.34]:11686

003011: //881/07584A1F816F/SIP/Transport/sipSPISendAck: Sending ACK to the transport layer
003012: //881/07584A1F816F/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
003013: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: msg=0x8CD7B0B4, addr=198.38.7.34, port=5065, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x814ACA1C
003014: //881/07584A1F816F/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
003015: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
003016: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:198.38.7.34, rport:5065 with laddr:

003017: //881/07584A1F816F/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x8CD7B0B4
003018: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8CD7B0B4, addr=198.38.7.34, port=5065, local_addr=, connId=3 for UDP
003019: //881/07584A1F816F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x889FF5D0
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 5146666666
Called Number : 5145555555
Source IP Address (Sig ): 173.231.97.100
Destn SIP Req Addr:Port : 198.38.7.34:5065
Destn SIP Resp Addr:Port : 198.38.7.34:5065
Destination Name : nat5.babytel.ca

003020: //881/07584A1F816F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 173.231.97.100
Source IP Port (Media): 19296
Destn IP Address (Media): 198.38.7.34
Destn IP Port (Media): 11686
Orig Destn IP Address:Port (Media): [ - ]:0

003021: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpSocketWrites: Send failed errno 261
003022: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWSocketException: context=0x8746D12C
003023: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessSocketExceptions: gConnTab=0x8746D12C, addr=198.38.7.34, port=5065, local_addr=, connid=3, transport=UDP
003024: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostCloseConnection: Posting UDP conn close for addr=198.38.7.34, port=5065, local_addr=, connid=3
003025: //-1/xxxxxxxxxxxx/SIP/Transport/sipDeleteConnInstance: Deleted conn=0x87B81004, connid=3, addr=198.38.7.34, port=5065, local_addr=, transport=UDP
003026: //-1/xxxxxxxxxxxx/SIP/Error/act_active_send_msg_failure: Send Error to 198.38.7.34:5065 for transport UDP
003027: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE
003028: //-1/xxxxxxxxxxxx/SIP/Error/sip_dns_type_a_query: TYPE A query failed for 198.38.7.34
003029: //-1/xxxxxxxxxxxx/SIP/Error/_send_dns_fail: DNS Query for 198.38.7.34 failed
003030: //881/07584A1F816F/SIP/Error/sipSPIReqDNSProcessingRequired: FQDN in Contact/RR for next transaction cannot be resolved
003031: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!!
003032: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!
003033: //881/07584A1F816F/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8CC7E39C
003034: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
003035: //881/07584A1F816F/SIP/Media/sipSPIDestroyRtpSession: stream:8CC7E39C
003036: //881/07584A1F816F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x889FF5D0
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 5146666666
Called Number : 5145555555
Source IP Address (Sig ): 173.231.97.100
Destn SIP Req Addr:Port : 198.38.7.34:5065
Destn SIP Resp Addr:Port : 198.38.7.34:5065
Destination Name : nat5.babytel.ca

003037: //881/07584A1F816F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 173.231.97.100
Source IP Port (Media): 19296
Destn IP Address (Media): 198.38.7.34
Destn IP Port (Media): 11686
Orig Destn IP Address:Port (Media): [ - ]:0

003038: //881/07584A1F816F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 200

and here's my related configs :

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no fax-relay sg3-to-g3
sip
registrar server expires max 3600 min 3600
localhost dns:sip.babytel.ca
outbound-proxy dns:nat5.babytel.ca
no update-callerid
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 5 g729r8
!

sip-ua
keepalive target dns:sip.babytel.ca
authentication username 15147777777 password 7 110548731487451817
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:sip.babytel.ca expires 3600
sip-server dns:sip.babytel.ca
host-registrar
g729-annexb override

Any suggestions ??

[Rob Dubois]

Added note : It never does that if I dial the full number directly.