Cisco 2811 SIP-to-SIP GW T.38 does not work!
Hello!
Diagram is something like this: Softswitch(MERA) -->>-- Cisco2811 -->>-- Softswitch(MERA) It's needed to limit traffic if one of SSWs is hacked. But it is not a subject. We just need such "construction".
Previously there was Cisco 1760 instead of 2811, result was the same. So I exclude platform and IOS.
1. Voice calls are sent and received fine in diagram above.
2. Fax are passed good between two Softswitches if I exclude Cisco2811.
3. Faxes are stopped immediately when I re-route voice traffic through Cisco2811 (in the same conditions on both Softswitches as in above paragraph 2. That is, Faxes are passed between Softswitches directly, I don't change anything on Softswitches, and I just re-route Voice from both Softswitches on C2811 - Faxes stop immediately).
4. Relevant configuraion:
voice service voip allow-connections sip to sip!!voice class uri Centrex sip host ^10\.0\.99\.111$!voice class uri RTU1 sip host ^10\.0\.99\.121$!voice class uri RTU2 sip host ^10\.0\.99\.221$!!voice class codec 1 codec preference 1 g711alaw bytes 80 codec preference 2 clear-channel!!voice translation-rule 112 rule 1 /^000112\(.*\)$/ /\1/!voice translation-rule 999 rule 1 /^999\(.*\)$/ /000\1/!voice translation-rule 999112 rule 1 /^\(.*\)$/ /999112\1/!voice translation-profile 112 translate called 112!voice translation-profile 999 translate called 999!voice translation-profile 999112 translate called 999112!!interface FastEthernet0/0.18 encapsulation dot1Q 18 ip address 10.0.99.29 255.255.255.0 no snmp trap link-status!!dial-peer voice 999112 voip translation-profile incoming 999112 voice-class codec 1 session protocol sipv2 incoming uri from Centrex dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711alaw no vad!dial-peer voice 999 voip translation-profile outgoing 999 destination-pattern 999.+ voice-class codec 1 session protocol sipv2 session target ipv4:10.0.99.99 session transport udp dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711alaw no vad!dial-peer voice 112 voip translation-profile outgoing 112 destination-pattern 000112.+ voice-class codec 1 session protocol sipv2 session target ipv4:10.0.99.100 session transport udp dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711alaw no vad!dial-peer voice 901 voip voice-class codec 1 session protocol sipv2 incoming uri from RTU1 dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711alaw no vad!dial-peer voice 902 voip voice-class codec 1 session protocol sipv2 incoming uri from RTU2 dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711alaw no vad!
5. TSHARK from left-side Softswitch:
16:10:51.680764 10.0.99.221 -> 10.0.99.29 SIP/SDP Request: INVITE sip:[email protected];user=phone, with session description16:10:51.721616 10.0.99.29 -> 10.0.99.221 SIP Status: 100 Trying16:10:55.413288 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 183 Session Progress, with session description16:10:55.418718 10.0.99.29 -> 10.0.99.221 SIP Status: 180 Ringing16:10:59.090481 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:10:59.091451 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:04.296532 10.0.99.29 -> 10.0.99.221 SIP Status: 488 Not Acceptable Media16:11:04.296708 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:04.793058 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:04.793262 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:05.793043 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:05.793261 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:07.793042 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:07.793300 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:11.793077 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:11.793264 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:15.793316 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:15.793541 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:19.793289 10.0.99.29 -> 10.0.99.221 SIP/SDP Status: 200 OK, with session description16:11:19.793538 10.0.99.221 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:11:23.794963 10.0.99.29 -> 10.0.99.221 SIP Request: BYE sip:[email protected]:5061;user=phone16:11:23.795650 10.0.99.221 -> 10.0.99.29 SIP Status: 200 OK
6. TSHARK from right-side Softswitch:
16:10:12.071247 10.0.99.111 -> 10.0.99.29 SIP/SDP Request: INVITE sip:[email protected];user=phone, with session description16:10:12.113708 10.0.99.29 -> 10.0.99.111 SIP Status: 100 Trying16:10:12.843352 10.0.99.29 -> 10.0.99.111 SIP/SDP Status: 183 Session Progress, with session description16:10:16.328955 10.0.99.29 -> 10.0.99.111 SIP/SDP Status: 200 OK, with session description16:10:16.329808 10.0.99.111 -> 10.0.99.29 SIP Request: ACK sip:[email protected]:506016:10:51.721600 10.0.99.29 -> 10.0.99.100 SIP/SDP Request: INVITE sip:[email protected]:5060, with session description16:10:51.723145 10.0.99.100 -> 10.0.99.29 SIP Status: 100 Trying16:10:55.384493 10.0.99.100 -> 10.0.99.29 SIP/SDP Status: 183 Progress, with session description16:10:55.392178 10.0.99.100 -> 10.0.99.29 SIP Status: 180 Ringing16:10:59.069771 10.0.99.100 -> 10.0.99.29 SIP/SDP Status: 200 OK, with session description16:10:59.088587 10.0.99.29 -> 10.0.99.100 SIP Request: ACK sip:[email protected]:5060
7. Debug output for "debug ccsip all" and "debug voice dialpeer all"
Router#*Sep 19 12:27:55.107: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetAgeingTimer: Aging timer initiated for holder=0x4654DA30,addr=10.0.99.111*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.0.99.221:5061*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.0.99.221,Port 5061, Transport 1, SentBy Port 5061*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:INVITE sip:[email protected];user=phone SIP/2.0Via: SIP/2.0/UDP 10.0.99.221:5061;rport;branch=z9hG4bK-3628481038-3792786178-436258467-408012644From: <sip:[email protected]:5061;user=phone>;tag=4095425038-3792786178-436258467-408012644To: <sip:[email protected];user=phone>Call-ID: [email protected]: 1 INVITEContact: <sip:[email protected]:5061;user=phone>Content-Type: application/sdpAllow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATEMax-Forwards: 70User-Agent: MERA MVTS3G v.4.4.0-15Cisco-Guid: 237931618-38998498-2747662362-1690784024Category: 10Content-Length: 313v=0o=- 1348056651 1348056651 IN IP4 10.0.99.221s=-c=IN IP4 10.0.99.221t=0 0m=audio 17294 RTP/AVP 8 0 18 4 96a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:4 G723/8000a=fmtp:4 annexa=yesa=rtpmap:96 telephone-event/8000a=fmtp:96 0-15a=sendrecv*Sep 19 12:27:55.267: //-1/0E2E8C62A3C6/SIP/State/sipSPIChangeState: 0x4627A3B8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.0.99.221,Port 5061, Transport 1, SentBy Port 5060*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT*Sep 19 12:27:55.267: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.0.99.221,Port 5061, Transport 1, SentBy Port 5061*Sep 19 12:27:55.271: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4627A3B8 [email protected]*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 0001124957887603*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 4991589848*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name , number 4991589848, Calling oct3 0x00, oct_3a 0x80, Called number 0001124957887603*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerCore: Calling Number=4991589848, Called Number=0001124957887603, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected];user=phone*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Dial String=, Expanded String=, Calling Number= Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Result=-1*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype:exit@5392*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected];user=phone*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Dial String=, Expanded String=, Calling Number= Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Result=-1*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype:exit@5392*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected]:5061;user=phone*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE*Sep 19 12:27:55.271: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Dial String=, Expanded String=, Calling Number= Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/DPM/MatchNextPeer: Result=Success(0); Incoming Dial-peer=902 Is Matched*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/DPM/dpMatchPeertype:exit@5392*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_FROM_URI; Incoming Dial-peer=902*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/DPM/dpAssociateIncomingPeerSPI:exit@5926*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/SIP/Info/sipSPIGetCallConfig: Peer tag 902 matched for incoming call*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/SIP/Info/sipSPIGetCallConfig: Using Voice Class Codec, tag = 1*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/SIP/Info/sipSPICopyPeerDataToCCB:From CLI: Modem NSE payload = 100, Passthrough = 0, Modem relay = 0, Gw-Xid = 1SPRT latency 200, SPRT Retries = 12, Dict Size = 1024 String Len = 32, Compress dir = 3*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number 4991589848, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number 0001124957887603, oct3 0x00*Sep 19 12:27:55.275: //-1/0E2E8C62A3C6/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or multiple ptime attributes that can't be handled*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(96) could not be reserved.*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY payload (96) is reserved by another application.*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode: Inband Voice*Sep 19 12:27:55.275: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1 payload_type=8, codec_bytes=80, codec=g711alaw, dtmf_relay=inband-voice stream_type=voice-only (0), dest_ip_address=10.0.99.221, dest_port=17294*Sep 19 12:27:55.275: //19/0E2E8C62A3C6/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :80 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*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_report_media_to_peer: callId 19 peer 0 flags 0x201*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:CallID 19, sdp 0x45A61FCC channels 0x4627BC80SIP: (19) Attribute ptime, level 1 instance 1 not found.*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 8 mline 1*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw*Sep 19 12:27:55.279: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=0,stream->negotiated_codec_bytes=80, coverted ptime=10 stream->mline_index=1, media_ndx=1*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 6 ptype 8 time 10, bytes 80 as channel 0 mline 1 ss 0 10.0.99.221:17294*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 0 mline 1*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulawSIP: (19) Attribute ptime, level 1 instance 1 not found.*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=0, media_ndx=1*Sep 19 12:27:55.279: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :0, codecbytes: 0*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Codec bytes 0, use default packet rate 160*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 5 ptype 0 time 0, bytes 160 as channel 1 mline 1 ss 0 10.0.99.221:17294*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 18 mline 1*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPISelectCodecVersion: Codec (g729r8) is not in preferred list*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIRouter#P/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8 pre-ietf*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8 pre-ietfSIP: (19) Attribute ptime, level 1 instance 1 not found.*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=0, media_ndx=1*Sep 19 12:27:55.279: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g729r8 pre-ietf ptime :0, codecbytes: 0*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Codec bytes 0, use default packet rate 20*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 0 ptype 18 time 0, bytes 20 as channel 2 mline 1 ss 0 10.0.99.221:17294*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 4 mline 1*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPISelectCodecVersion: Codec (g723ar63) is not in preferred list*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8 pre-ietf*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8 pre-ietfSIP: (19) Attribute ptime, level 1 instance 1 not found.*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=0, media_ndx=1*Sep 19 12:27:55.279: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g729r8 pre-ietf ptime :0, codecbytes: 0*Sep 19 12:27:55.279: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Codec bytes 0, use default packet rate 20*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 0 ptype 4 time 0, bytes 20 as channel 3 mline 1 ss 0 10.0.99.221:17294*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 96 mline 1*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_report_media_to_peer:Report initial call media*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/copy_channels: callId 19 size 296 ptr 0x46646D94)*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_report_media_to_peer:CCSIP: Unable to report channel ind*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Callid : -1 Negotiated Codec : g711alaw, bytes :80 Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 Negotiated CN payload : 0 Media Srce Addr/Port : 10.0.99.29:0 Media Dest Addr/Port : 10.0.99.221:17294*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIHandleInviteMedia:Negotiated Codec : g711alaw, bytes :80Preferred Codec : g711alaw, bytes :80Preferred DTMF relay 1 : 6Preferred DTMF relay 2 : 0Negotiated DTMF relay : 0Preferred and Negotiated NTE payloads: 101 0Preferred and Negotiated NSE payloads: 100 0Preferred and Negotiated Modem Relay: 0 0Preferred and Negotiated Modem Relay GwXid: 1 0*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active*Sep 19 12:27:55.283: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19570 for stream 1*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=19570*Sep 19 12:27:55.283: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 19570*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = [email protected]*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo*Sep 19 12:27:55.283: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS*Sep 19 12:27:55.283: //19/0E2E8C62A3C6/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4627A3B8 [email protected]*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 13 to table*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: msg=0x4654E450, addr=10.0.99.221, port=5061, sentBy_port=5061, is_req=0, transport=1, switch=0, callBack=0x00000000*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0*Sep 19 12:27:55.287: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4654E450, addr=10.0.99.221, port=5061, connId=0 for UDP*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/State/sipSPIChangeState: 0x4627A3B8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE)*Sep 19 12:27:55.287: //19/0E2E8C62A3C6/SIP/Info/sipSPIProcessContactInfo: Previous Hop 10.0.99.221:5061*Sep 19 12:27:55.287: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/dpMatchPeersCore: Calling Number=, Called Number=0001124957887603, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=0001124957887603*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/dpMatchCore: Dial String=0001124957887603, Expanded String=0001124957887603, Calling Number= Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/MatchNextPeer: Result=Success(0); Outgoing Dial-peer=112 Is Matched*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST*Sep 19 12:27:55.291: //-1/0E2E8C62A3C6/DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=112*Sep 19 12:27:55.291: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)*Sep 19 12:27:55.291: //20/000000000000/SIP/Info/ccsip_call_setup_request: This a IPIP call: Chan 0, codec 6 channel 17294, ip A0063DD:17294 params 0x465F9EF4 caps 0x44ED30C8*Sep 19 12:27:55.291: //20/000000000000/SIP/Info/ccsip_call_setup_request: This a IPIP call: Chan 1, codec 5 channel 17294, ip A0063DD:17294 params 0x465F9EF4 caps 0x44ED30C8*Sep 19 12:27:55.291: //20/000000000000/SIP/Info/ccsip_call_setup_request: This a IPIP call: Chan 2, codec 0 channel 17294, ip A0063DD:17294 params 0x465F9EF4 caps 0x44ED30C8*Sep 19 12:27:55.291: //20/000000000000/SIP/Info/ccsip_call_setup_request: This a IPIP call: Chan 3, codec 0 channel 17294, ip A0063DD:17294 params 0x465F9EF4 caps 0x44ED30C8*Sep 19 12:27:55.291: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP*Sep 19 12:27:55.291: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:*Sep 19 12:27:55.291: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 137)*Sep 19 12:27:55.291: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: peer ID 20 chans 0x44EE3BB0 event 137 flags 0x10020038 0x601 data 0x44EE3BB0*Sep 19 12:27:55.291: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 20 chans 0x44EE3BB0 event 137 flags 0x10020038 0x601 data 0x44EE3BB0*Sep 19 12:27:55.295: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 20 chans 0x44EE3BB0 event 137 flags 0x10020038 0x601 data 0x44EE3BB0, type = 3*Sep 19 12:27:55.295: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 14 to table*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/act_idle_continue_call_setup:*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec bytes: 0*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIGetCallConfig: Using Voice Class Codec, tag = 1*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPICopyPeerDataToCCB:From CLI: Modem NSE payload = 100, Passthrough = 0, Modem relay = 0, Gw-Xid = 1SPRT latency 200, SPRT Retries = 12, Dict Size = 1024 String Len = 32, Compress dir = 3*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:callid 20, channels 0x44E96BE0 caps 0x44ED30C8*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:pref dtmf 96*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4627C64C [email protected]*Sep 19 12:27:55.295: //20/000000000000/SIP/Info/sipSPIUsetBillingProfile: sipCallId for billing records = [email protected]*Sep 19 12:27:55.295: //20/000000000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:55.295: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16926 for stream 1*Sep 19 12:27:55.299: //20/000000000000/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101*Sep 19 12:27:55.299: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:27:55.299: //20/000000000000/SIP/Info/sip_generate_sdp_xcapsRouter#_list: Modem Relay and T38 disabled. X-cap not needed*Sep 19 12:27:55.299: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.0.99.100,Port 5060, Transport 1, SentBy Port 5060*Sep 19 12:27:55.299: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT*Sep 19 12:27:55.299: //20/000000000000/SIP/Event/sipSPICreateRpid: Received Octet3A=0x80 -> Setting ;screen=no ;privacy=off*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x4654D520, addr=10.0.99.100, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x41086470*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0*Sep 19 12:27:55.299: //20/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4654D520*Sep 19 12:27:55.299: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4654D520, addr=10.0.99.100, port=5060, connId=3 for UDP*Sep 19 12:27:55.299: //20/000000000000/SIP/Info/sentInviteRequest: Sent Invite in state STATE_IDLE*Sep 19 12:27:55.303: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteRequest: Transaction active. Facilities will be queued.*Sep 19 12:27:55.303: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_SENT_INVITE, SUBSTATE_NONE)*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 20) to the VOIP RTP library*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.0.99.29, lport = 16926, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 20, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY media_ip_addr = 0.0.0.0*Sep 19 12:27:55.303: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one*Sep 19 12:27:55.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent:SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.99.221:5061;rport;branch=z9hG4bK-3628481038-3792786178-436258467-408012644From: <sip:[email protected]:5061;user=phone>;tag=4095425038-3792786178-436258467-408012644To: <sip:[email protected];user=phone>;tag=114FC0-1F24Date: Wed, 19 Sep 2012 12:27:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-12.xCSeq: 1 INVITEAllow-Events: telephone-eventContent-Length: 0*Sep 19 12:27:55.307: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent:INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.99.29:5060;branch=z9hG4bK1C14C0From: <sip:[email protected]>;tag=114FE0-26C0To: <sip:[email protected]>Date: Wed, 19 Sep 2012 12:27:55 GMTCall-ID: [email protected]: 100rel,timer,replacesMin-SE: 1800Cisco-Guid: 237931618-38998498-2747662362-1690784024User-Agent: Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTERCSeq: 101 INVITEMax-Forwards: 70Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=offTimestamp: 1348057675Contact: <sip:[email protected]:5060>Expires: 180Allow-Events: telephone-eventContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3284 8564 IN IP4 10.0.99.29s=SIP Callc=IN IP4 10.0.99.29t=0 0m=audio 16926 RTP/AVP 8 101c=IN IP4 10.0.99.29a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:10*Sep 19 12:27:55.307: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.0.99.100:5060*Sep 19 12:27:55.307: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000*Sep 19 12:27:55.311: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.99.29:5060;branch=z9hG4bK1C14C0From: <sip:[email protected]>;tag=114FE0-26C0To: <sip:[email protected]>Call-ID: [email protected]: 101 INVITEContact: <sip:[email protected]:5060>Server: MERA MVTS3G v.4.4.0-15Timestamp: 1348057675Content-Length: 0*Sep 19 12:27:55.311: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)Router#*Sep 19 12:27:58.971: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.0.99.100:5060*Sep 19 12:27:58.971: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000*Sep 19 12:27:58.971: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 183 ProgressVia: SIP/2.0/UDP 10.0.99.29:5060;branch=z9hG4bK1C14C0From: <sip:[email protected]>;tag=114FE0-26C0To: <sip:[email protected]>;tag=2318849048-3792786178-436251047-2287060836Call-ID: [email protected]: 101 INVITEContact: <sip:[email protected]:5060>Content-Type: application/sdpServer: MERA MVTS3G v.4.4.0-15Content-Length: 239v=0o=- 1348056655 1348056655 IN IP4 10.0.99.111s=-c=IN IP4 10.0.99.111t=0 0m=audio 21550 RTP/AVP 8 101a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:10a=sendrecva=silenceSupp:off - - - -*Sep 19 12:27:58.971: //20/000000000000/SIP/Info/HandleSIP1xxSessionProgress: Content-Disposition NOT received in 18x response - using default Content-Disposition values*Sep 19 12:27:58.971: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.*Sep 19 12:27:58.971: //20/000000000000/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1*Sep 19 12:27:58.971: //20/000000000000/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1*Sep 19 12:27:58.971: //20/000000000000/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:10*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :10, codecbytes: 80*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPIDoDTMFRelayNegotiation: Payload type (101) is reserved for requested dtmf relay mode.*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1 payload_type=8, codec_bytes=80, codec=g711alaw, dtmf_relay=rtp-nte stream_type=voice+dtmf (1), dest_ip_address=10.0.99.111, dest_port=21550*Sep 19 12:27:58.975: //20/000000000000/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :80 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*Sep 19 12:27:58.975: //20/000000000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_report_media_to_peer: callId 20 peer 19 flags 0x7*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:CallID 20, sdp 0x45C92F44 channels 0x4627DF14*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 8 mline 1*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=10,stream->negotiated_codec_bytes=80, coverted ptime=10 stream->mline_index=1, media_ndx=1*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 6 ptype 8 time 10, bytes 80 as channel 0 mline 1 ss 1 10.0.99.111:21550*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 101 mline 1*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/sipSPI_ipip_report_media_to_peer:Report initial call media*Sep 19 12:27:58.975: //20/000000000000/SIP/Info/copy_channels: callId 20 size 80 ptr 0x46655B7C)*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:*Sep 19 12:27:58.975: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 131)*Sep 19 12:27:58.975: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: peer ID 20 chans 0x46655B7C event 131 flags 0x10020038 0x403 data 0x46655B7C*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: CC_EV_H245_OPEN_CHANNEL_IND: peer ID 20 chans 0x46655B7C event 131 flags 0x10020038 0x403 data 0x46655B7C*Sep 19 12:27:58.979: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_NEW_MEDIA*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_event_handler: ccsip_event_handler: set event->type = SIPSPI_EV_CC_NEW_MEDIA!: peer ID 20 chans 0x46655B7C event 131 flags 0x10020038 0x403 data 0x46655B7C*Sep 19 12:27:58.979: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=-1, current_seq_num=0x1A8*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=-1, current_seq_num=0x0*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711alaw, Bytes=80*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=1161273728, from CLI config=0*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/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*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Media/sipSPISetStreamInfo: 0 Active Streams*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Error/sipSPISetStreamInfo: Number of active streams is zero (0)!*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Media/sipSPISetStreamInfo:caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711alaw, Bytes=80*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->flags_ipip = 0x403*Sep 19 12:27:58.979: //20/000000000000/SIP/Info/ccsip_caps_ack: Set forking flag to 0x0*Sep 19 12:27:58.979: //20/000000000000/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Callid : 20 Negotiated Codec : g711alaw, bytes :80 Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 Negotiated CN payload : 0 Media Srce Addr/Port : 10.0.99.29:16926 Media Dest Addr/Port : 10.0.99.111:21550*Sep 19 12:27:58.979: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.*Sep 19 12:27:58.979: //20/000000000000/SIP/Info/HandleSIP1xxSessionProgress: ccsip_api_call_cut_progress returned: SIP_SUCCESS*Sep 19 12:27:58.979: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)*Sep 19 12:27:58.979: //20/000000000000/SIP/Info/HandleSIP1xxSessionProgress: Transaction Complete. Lock on Facilities released.*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_handle_channel_info:CCSIP:callID 19 ft: 1, inc 8, 10.0.99.111:21550, codec 6*Sep 19 12:27:58.979: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:callid 19, channels 0x46655B7C caps 0x44E8F284*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:pref dtmf 101*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: nego mline 1 dtmf 101 ss 1 ret 0*Sep 19 12:27:58.983: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: retreive codec 6 ptype 8 time 10 bytes 80*Sep 19 12:27:58.983: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 19570*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPIProcessMediaChanges: sipSPIProcessMediaChanges*Sep 19 12:27:58.983: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROGRESS*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/ccsip_bridge: confID = 10, srcCallID = 19, dstCallID = 20*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/InfRouter#o/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 19/20*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=-1, new streamcallid=19*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 19) to the VOIP RTP library*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1*Sep 19 12:27:58.983: //19/0E2E8C62A3C6/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.0.99.29, lport = 19570, raddr = 10.0.99.221, rport=17294, do_rtcp=TRUE src_callid = 19, dest_callid = 20, stream type = voice+dtmf, stream direction = SENDRECV media_ip_addr = 10.0.99.221*Sep 19 12:27:58.987: //19/0E2E8C62A3C6/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one*Sep 19 12:27:58.987: //19/0E2E8C62A3C6/SIP/Info/sipSPIUpdateRtcpSession: Process Media changes is still pending.*Sep 19 12:27:58.987: //19/0E2E8C62A3C6/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0*Sep 19 12:27:58.987: //20/000000000000/SIP/Info/ccsip_bridge: confID = 10, srcCallID = 20, dstCallID = 19*Sep 19 12:27:58.987: //20/000000000000/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 20/19*Sep 19 12:27:58.987: //20/000000000000/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=20, new streamcallid=20*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 20) to the VOIP RTP library*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.0.99.29, lport = 16926, raddr = 10.0.99.111, rport=21550, do_rtcp=TRUE src_callid = 20, dest_callid = 19, stream type = voice+dtmf, stream direction = SENDRECV media_ip_addr = 10.0.99.111*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update*Sep 19 12:27:58.987: //20/000000000000/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0*Sep 19 12:27:58.991: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.0.99.100:5060*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Info/sipSPISendInviteResponse183: Session Type is Media/Qos/Security/RTR SDP body is attached*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Transport/sipSPISendInviteResponse: Sending 183 Response to the Transport Layer*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: msg=0x4654DFD0, addr=10.0.99.221, port=5061, sentBy_port=5061, is_req=0, transport=1, switch=0, callBack=0x41086D90*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0*Sep 19 12:27:58.991: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4654DFD0, addr=10.0.99.221, port=5061, connId=0 for UDP*Sep 19 12:27:58.991: //19/0E2E8C62A3C6/SIP/Info/sentInviteResponse18x: Sent a 18x Response*Sep 19 12:27:58.991: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000*Sep 19 12:27:58.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.99.29:5060;branch=z9hG4bK1C14C0From: <sip:[email protected]>;tag=114FE0-26C0To: <sip:[email protected]>;tag=2318849048-3792786178-436251047-2287060836Call-ID: [email protected]: 101 INVITEContact: <sip:[email protected]:5060>Server: MERA MVTS3G v.4.4.0-15Content-Length: 0*Sep 19 12:27:58.995: //20/000000000000/SIP/Info/ccsip_api_call_alert: SDP Body either absent or ignored in 180 RINGING:- will wait for 200 OK to do negotiation.*Sep 19 12:27:58.995: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.*Sep 19 12:27:58.995: //20/000000000000/SIP/Info/HandleSIP1xxRinging: ccsip_api_call_alert returned: SIP_SUCCESS*Sep 19 12:27:58.995: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)*Sep 19 12:27:58.995: //20/000000000000/SIP/Info/HandleSIP1xxRinging: Transaction Complete. Lock on Facilities released.*Sep 19 12:27:58.995: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent:SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.99.221:5061;rport;branch=z9hG4bK-3628481038-3792786178-436258467-408012644From: <sip:[email protected]:5061;user=phone>;tag=4095425038-3792786178-436258467-408012644To: <sip:[email protected];user=phone>;tag=114FC0-1F24Date: Wed, 19 Sep 2012 12:27:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-12.xCSeq: 1 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTERAllow-Events: telephone-eventContact: <sip:[email protected]:5060>Content-Disposition: session;handling=requiredContent-Type: application/sdpContent-Length: 268v=0o=CiscoSystemsSIP-GW-UserAgent 4191 6681 IN IP4 10.0.99.29s=SIP Callc=IN IP4 10.0.99.29t=0 0m=audio 19570 RTP/AVP 8 101c=IN IP4 10.0.99.29a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:10a=silenceSupp:off - - - -*Sep 19 12:27:58.999: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_ALERTING*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Transport/sipSPISendInviteResponse: Sending 180 Response to the Transport Layer*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: msg=0x4654DFD0, addr=10.0.99.221, port=5061, sentBy_port=5061, is_req=0, transport=1, switch=0, callBack=0x41086D90*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0*Sep 19 12:27:58.999: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4654DFD0, addr=10.0.99.221, port=5061, connId=0 for UDP*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/Info/sentInviteResponse18x: Sent a 18x Response*Sep 19 12:27:58.999: //19/0E2E8C62A3C6/SIP/State/sipSPIChangeState: 0x4627A3B8 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_SENT_ALERTING, SUBSTATE_NONE)*Sep 19 12:27:59.003: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent:SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.99.221:5061;rport;branch=z9hG4bK-3628481038-3792786178-436258467-408012644From: <sip:[email protected]:5061;user=phone>;tag=4095425038-3792786178-436258467-408012644To: <sip:[email protected];user=phone>;tag=114FC0-1F24Date: Wed, 19 Sep 2012 12:27:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-12.xCSeq: 1 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTERAllow-Events: telephone-eventContact: <sip:[email protected]:5060>Content-Length: 0Router#*Sep 19 12:28:02.655: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.0.99.100:5060*Sep 19 12:28:02.655: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000*Sep 19 12:28:02.655: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.99.29:5060;branch=z9hG4bK1C14C0From: <sip:[email protected]>;tag=114FE0-26C0To: <sip:[email protected]>;tag=2318849048-3792786178-436251047-2287060836Call-ID: [email protected]: 101 INVITEContact: <sip:[email protected]:5060>Content-Type: application/sdpAllow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATEServer: MERA MVTS3G v.4.4.0-15X-mera-expires: 86460Content-Length: 239v=0o=- 1348056655 1348056655 IN IP4 10.0.99.111s=-c=IN IP4 10.0.99.111t=0 0m=audio 21550 RTP/AVP 8 101a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:10a=sendrecva=silenceSupp:off - - - -*Sep 19 12:28:02.659: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIhandle200OKInvite: Transaction active. Facilities will be queued.*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIhandle200OKInvite: *** This ccb is the parent*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:10*Sep 19 12:28:02.659: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :10, codecbytes: 80*Sep 19 12:28:02.659: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.*Sep 19 12:28:02.659: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sip_do_nse_negotiation: Remote NSE payload = local one = 0, Use it*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1 payload_type=8, codec_bytes=80, codec=g711alaw, dtmf_relay=rtp-nte stream_type=voice+dtmf (1), dest_ip_address=10.0.99.111, dest_port=21550*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPICompareStreams: stream 1 dest_port: old=21550 new=21550*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPICompareStreams: Flags set for stream 1: RTP_CHANGE=No CAPS_CHANGE=No*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPICompareSDP: Flags set for call: NEW_MEDIA=No DSPDNLD_REQD=No IPIP_MEDIA=No*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :80 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*Sep 19 12:28:02.659: //20/000000000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.0.99.29*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPI_ipip_report_media_to_peer: callId 20 peer 19 flags 0x407*Sep 19 12:28:02.659: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:CallID 20, sdp 0x45CB1F40 channels 0x4627DF14*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 8 mline 1*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw*Sep 19 12:28:02.663: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :80, ptime: 10*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=10,stream->negotiated_codec_bytes=80, coverted ptime=10 stream->mline_index=1, media_ndx=1*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Adding codec 6 ptype 8 time 10, bytes 80 as channel 0 mline 1 ss 1 10.0.99.111:21550*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:Hndl ptype 101 mline 1*Sep 19 12:28:02.663: //20/000000000000/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ACTIVE (5) Callid : 20 Negotiated Codec : g711alaw, bytes :80 Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 Negotiated CN payload : 0 Media Srce Addr/Port : 10.0.99.29:16926 Media Dest Addr/Port : 10.0.99.111:21550*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPIProcessMediaChanges: sipSPIProcessMediaChanges*Sep 19 12:28:02.663: //20/000000000000/SIP/Info/sipSPIhandle200OKInvite: ccsip_api_call_connect_media returned: SIP_SUCCESS*Sep 19 12:28:02.663: //20/000000000000/SIP/State/sipSPIChangeState: 0x4627C64C : S
Hi Ellad.
Why don't try to use the 2811 as a SIP signalling proxy only?
In this way the media (RTP or T.38) will be handled only from the two MERA SoftSwitch.
To do this you must enable CUBE on your 2811 and use these special commands:
voice service voip
media flow-around
allow-connections sip to sip
signaling forward unconditional
sip
rel1xx disable
header-passing
midcall-signaling passthru
pass-thru headers unsupp
pass-thru content unsupp
pass-thru content sdp
I don't remember if we have already try this solution.
Regards.
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Dear All,
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My Current Config is as below
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2013.06.09 17:55:23 =~=~=~=~=~=~=~=~=~=~=~=
exit
Gateway#show run |
Building configuration...
Current configuration : 25109 bytes
! Last configuration change at 13:33:57 UTC Sun Jun 9 2013 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname Gateway
boot-start-marker
boot system flash c1841-advsecurityk9-mz.151-2.T1.bin
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crypto pki trustpoint TP-self-signed-1981248591
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1981248591
revocation-check none
rsakeypair TP-self-signed-1981248591
crypto pki trustpoint VMWare
enrollment terminal
revocation-check crl
crypto pki trustpoint OWA
enrollment terminal pem
revocation-check crl
crypto pki certificate chain TP.StartSSL.CA
certificate ca 01
(removed the certificate info for clarity)
quit
crypto pki certificate chain TP.StartSSL-vpn
certificate 0936E1
(removed the certificate info for clarity)9
quit
certificate ca 18
(removed the certificate info for clarity)
quit
crypto pki certificate chain TP-self-signed-1981248591
certificate self-signed 01
(removed the certificate info for clarity)
quit
crypto pki certificate chain VMWare
certificate ca 008EDCE6DBCE6B
(removed the certificate info for clarity)
quit
crypto pki certificate chain OWA
(removed the certificate info for clarity)
license udi pid CISCO1841 sn FCZ122191TW
archive
log config
hidekeys
username admin privilege 15 password 7 1304131F02023B7B7977
username ali password 7 06070328
redundancy
crypto isakmp policy 10
encr 3des
authentication pre-share
group 2
lifetime 84000
crypto isakmp key admin_123 address 0.0.0.0 0.0.0.0
crypto isakmp keepalive 10
crypto ipsec security-association lifetime seconds 28800
crypto ipsec transform-set vpnset esp-3des esp-sha-hmac
crypto ipsec transform-set strongsha esp-3des esp-sha-hmac
crypto dynamic-map mydyn 10
set transform-set strongsha
crypto map Dxb-Auh 1000 ipsec-isakmp dynamic XXXXXXXXXX
interface FastEthernet0/0
description Internal Network (Protected Interface)
ip address 10.236.5.254 255.255.255.0
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
pppoe enable group global
pppoe-client dial-pool-number 1
interface ATM0/0/0
no ip address
shutdown
no atm ilmi-keepalive
interface BRI0/1/0
no ip address
encapsulation hdlc
shutdown
interface Virtual-Template1
ip unnumbered Dialer1
peer default ip address dhcp-pool ContosoPool
ppp encrypt mppe auto required
ppp authentication ms-chap ms-chap-v2 eap
interface Dialer1
ip ddns update hostname XXXXXXX.dyndns.org
ip ddns update dyndns
ip address negotiated
ip nat outside
ip virtual-reassembly in
encapsulation ppp
ip tcp adjust-mss 1450
dialer pool 1
ppp pap sent-username vermam password 7 13044E155E0913323B
crypto map Dxb-Auh
interface Dialer2
mtu 1460
ip address negotiated
ip nat outside
ip virtual-reassembly in
encapsulation ppp
dialer in-band
dialer idle-timeout 0
dialer string 123
dialer vpdn
dialer-group 2
ppp pfc local request
ppp pfc remote apply
ppp encrypt mppe auto
ppp authentication ms-chap ms-chap-v2 callin
ppp eap refuse
ppp chap hostname hasanreza
ppp chap password 7 070E2541470726544541
interface Dialer995
no ip address
ip local pool webssl 10.236.6.10 10.236.6.30
ip forward-protocol nd
ip http server
ip http secure-server
ip nat inside source list nat interface Dialer1 overload
ip nat inside source static tcp 10.236.5.12 25 interface Dialer1 25
ip route 0.0.0.0 0.0.0.0 Dialer1
ip route 172.21.51.0 255.255.255.0 10.236.5.253
ip access-list extended internal
permit ip any 10.236.5.0 0.0.0.255
ip access-list extended nat
deny ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
deny ip 10.236.5.0 0.0.0.255 172.19.19.0 0.0.0.255
permit ip 10.236.5.0 0.0.0.255 any
ip access-list extended nonat
permit ip 10.236.5.0 0.0.0.255 172.19.19.0 0.0.0.255
permit ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
ip access-list extended sslacl
ip access-list extended webvpn
permit tcp any any eq 443
logging esm config
access-list 101 permit ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
control-plane
line con 0
line aux 0
line vty 0 4
exec-timeout 0 0
login local
transport preferred ssh
transport input telnet ssh
line vty 5 15
exec-timeout 0 0
login local
transport preferred ssh
transport input telnet ssh
scheduler allocate 20000 1000
webvpn gateway gateway1
ip interface Dialer1 port 443
ssl encryption rc4-md5
ssl trustpoint TP.StartSSL-vpn
inservice
webvpn install svc flash:/webvpn/anyconnect-win-3.1.00495-k9.pkg sequence 1
webvpn install csd flash:/webvpn/sdesktop.pkg
webvpn context webvpn
ssl authenticate verify all
url-list "Webservers"
heading "SimpleIT Technologies NBNS Servers"
url-text "Google" url-value "www.google.com"
url-text "Mainframe" url-value "10.236.5.2"
url-text "Mainframe2" url-value "https://10.236.5.2"
nbns-list "ContosoServer"
nbns-server 10.236.5.10
nbns-server 10.236.5.11
nbns-server 10.236.5.12
port-forward "PortForwarding"
local-port 3389 remote-server "10.236.5.10" remote-port 3389 description "Server-DC01"
policy group policy1
url-list "Webservers"
port-forward "PortForwarding"
nbns-list "ContosoServer"
functions file-access
functions file-browse
functions file-entry
functions svc-enabled
svc address-pool "webssl"
svc default-domain "Contoso.Local"
svc keep-client-installed
svc split include 10.236.5.0 255.255.255.0
svc split include 10.236.6.0 255.255.255.0
svc split include 172.31.1.0 255.255.255.0
svc split include 172.21.51.0 255.255.255.0
svc dns-server primary 172.21.51.10
default-group-policy policy1
gateway gateway1
inservice
end
Gateway#Dear All,
I have Cisco 1841 router running the below roles
1) SSL VPN Server
2) PPTP Server
3) Site to Site Connection with Sonicwall router
I want the router to be configured a pptp client to internet vpn server (so that i will get a fixed public ip )
Once i get this ip address i want to use this connection to accept in coming connection and forward ports to internal host,
I went through below
http://www.mreji.eu/content/cisco-router-pptp-client
https://supportforums.cisco.com/thread/2167562
But it does not work as i do not have the option for the below 2 commands in vpdn-group 2 section.(Please see section in blue)
protocol pptp
rotary-group 4
Please Advise and Help
Regards
Hasan Reza
My Current Config is as below
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2013.06.09 17:55:23 =~=~=~=~=~=~=~=~=~=~=~=
exit
Gateway#show run |
Building configuration...
Current configuration : 25109 bytes
! Last configuration change at 13:33:57 UTC Sun Jun 9 2013 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname Gateway
boot-start-marker
boot system flash c1841-advsecurityk9-mz.151-2.T1.bin
boot-end-marker
logging buffered 4096
no logging console
enable secret 5 $1$SciF$TlX1tR5qaG9ZE7pdZHcRJ/
no aaa new-model
dot11 syslog
ip source-route
no ip dhcp use vrf connected
ip dhcp excluded-address 10.236.5.1 10.236.5.20
ip dhcp excluded-address 10.236.5.21 10.236.5.50
ip dhcp excluded-address 172.21.51.2 172.21.51.50
ip dhcp pool ContosoPool
network 10.236.5.0 255.255.255.0
default-router 10.236.5.254
dns-server 213.42.20.20 195.229.241.222
ip dhcp pool DMZ
network 172.21.51.0 255.255.255.0
dns-server 172.21.51.10
default-router 172.21.51.1
domain-name contoso.local
ip cef
ip domain name contoso.local
ip name-server 213.42.20.20
ip name-server 195.229.241.22
ip name-server 195.229.241.222
ip ddns update method dyndns
HTTP
add http://xxxxxx:[email protected]/nic/update?system=dyndns&hostname=<h>&myip=<a>
remove http://xxxxxx:yyyyy@@members.dyndns.org/nic/update?system=dyndns&hostname=<h>&myip=<a>
interval maximum 0 1 0 0
multilink bundle-name authenticated
vpdn enable
vpdn-group 2
request-dialin
protocol l2tp
initiate-to ip 173.195.0.42
vpdn-group RAS-VPN
! Default PPTP VPDN group
accept-dialin
protocol pptp
virtual-template 1
l2tp tunnel timeout no-session 15
crypto pki token default removal timeout 0
crypto pki trustpoint TP.StartSSL.CA
enrollment terminal pem
revocation-check none
crypto pki trustpoint TP.StartSSL-vpn
enrollment terminal pem
usage ssl-server
serial-number none
fqdn ssl.spktelecom.com
ip-address none
revocation-check crl
rsakeypair RSA.StartSSL-vpn
crypto pki trustpoint TP-self-signed-1981248591
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1981248591
revocation-check none
rsakeypair TP-self-signed-1981248591
crypto pki trustpoint VMWare
enrollment terminal
revocation-check crl
crypto pki trustpoint OWA
enrollment terminal pem
revocation-check crl
crypto pki certificate chain TP.StartSSL.CA
certificate ca 01
(removed the certificate info for clarity)
quit
crypto pki certificate chain TP.StartSSL-vpn
certificate 0936E1
(removed the certificate info for clarity)9
quit
certificate ca 18
(removed the certificate info for clarity)
quit
crypto pki certificate chain TP-self-signed-1981248591
certificate self-signed 01
(removed the certificate info for clarity)
quit
crypto pki certificate chain VMWare
certificate ca 008EDCE6DBCE6B
(removed the certificate info for clarity)
quit
crypto pki certificate chain OWA
(removed the certificate info for clarity)
license udi pid CISCO1841 sn FCZ122191TW
archive
log config
hidekeys
username admin privilege 15 password 7 1304131F02023B7B7977
username ali password 7 06070328
redundancy
crypto isakmp policy 10
encr 3des
authentication pre-share
group 2
lifetime 84000
crypto isakmp key admin_123 address 0.0.0.0 0.0.0.0
crypto isakmp keepalive 10
crypto ipsec security-association lifetime seconds 28800
crypto ipsec transform-set vpnset esp-3des esp-sha-hmac
crypto ipsec transform-set strongsha esp-3des esp-sha-hmac
crypto dynamic-map mydyn 10
set transform-set strongsha
crypto map Dxb-Auh 1000 ipsec-isakmp dynamic XXXXXXXXXX
interface FastEthernet0/0
description Internal Network (Protected Interface)
ip address 10.236.5.254 255.255.255.0
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
pppoe enable group global
pppoe-client dial-pool-number 1
interface ATM0/0/0
no ip address
shutdown
no atm ilmi-keepalive
interface BRI0/1/0
no ip address
encapsulation hdlc
shutdown
interface Virtual-Template1
ip unnumbered Dialer1
peer default ip address dhcp-pool ContosoPool
ppp encrypt mppe auto required
ppp authentication ms-chap ms-chap-v2 eap
interface Dialer1
ip ddns update hostname XXXXXXX.dyndns.org
ip ddns update dyndns
ip address negotiated
ip nat outside
ip virtual-reassembly in
encapsulation ppp
ip tcp adjust-mss 1450
dialer pool 1
ppp pap sent-username vermam password 7 13044E155E0913323B
crypto map Dxb-Auh
interface Dialer2
mtu 1460
ip address negotiated
ip nat outside
ip virtual-reassembly in
encapsulation ppp
dialer in-band
dialer idle-timeout 0
dialer string 123
dialer vpdn
dialer-group 2
ppp pfc local request
ppp pfc remote apply
ppp encrypt mppe auto
ppp authentication ms-chap ms-chap-v2 callin
ppp eap refuse
ppp chap hostname hasanreza
ppp chap password 7 070E2541470726544541
interface Dialer995
no ip address
ip local pool webssl 10.236.6.10 10.236.6.30
ip forward-protocol nd
ip http server
ip http secure-server
ip nat inside source list nat interface Dialer1 overload
ip nat inside source static tcp 10.236.5.12 25 interface Dialer1 25
ip route 0.0.0.0 0.0.0.0 Dialer1
ip route 172.21.51.0 255.255.255.0 10.236.5.253
ip access-list extended internal
permit ip any 10.236.5.0 0.0.0.255
ip access-list extended nat
deny ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
deny ip 10.236.5.0 0.0.0.255 172.19.19.0 0.0.0.255
permit ip 10.236.5.0 0.0.0.255 any
ip access-list extended nonat
permit ip 10.236.5.0 0.0.0.255 172.19.19.0 0.0.0.255
permit ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
ip access-list extended sslacl
ip access-list extended webvpn
permit tcp any any eq 443
logging esm config
access-list 101 permit ip 10.236.5.0 0.0.0.255 172.31.1.0 0.0.0.255
control-plane
line con 0
line aux 0
line vty 0 4
exec-timeout 0 0
login local
transport preferred ssh
transport input telnet ssh
line vty 5 15
exec-timeout 0 0
login local
transport preferred ssh
transport input telnet ssh
scheduler allocate 20000 1000
webvpn gateway gateway1
ip interface Dialer1 port 443
ssl encryption rc4-md5
ssl trustpoint TP.StartSSL-vpn
inservice
webvpn install svc flash:/webvpn/anyconnect-win-3.1.00495-k9.pkg sequence 1
webvpn install csd flash:/webvpn/sdesktop.pkg
webvpn context webvpn
ssl authenticate verify all
url-list "Webservers"
heading "SimpleIT Technologies NBNS Servers"
url-text "Google" url-value "www.google.com"
url-text "Mainframe" url-value "10.236.5.2"
url-text "Mainframe2" url-value "https://10.236.5.2"
nbns-list "ContosoServer"
nbns-server 10.236.5.10
nbns-server 10.236.5.11
nbns-server 10.236.5.12
port-forward "PortForwarding"
local-port 3389 remote-server "10.236.5.10" remote-port 3389 description "Server-DC01"
policy group policy1
url-list "Webservers"
port-forward "PortForwarding"
nbns-list "ContosoServer"
functions file-access
functions file-browse
functions file-entry
functions svc-enabled
svc address-pool "webssl"
svc default-domain "Contoso.Local"
svc keep-client-installed
svc split include 10.236.5.0 255.255.255.0
svc split include 10.236.6.0 255.255.255.0
svc split include 172.31.1.0 255.255.255.0
svc split include 172.21.51.0 255.255.255.0
svc dns-server primary 172.21.51.10
default-group-policy policy1
gateway gateway1
inservice
end
Gateway# -
Cisco Aironet FW 15.2 Does not work with Non-Cisco Media Bridges
I have a Cisco Aironet 1142i that was just updated from 12.4(23c)JY to 15.2(4)JA1 (don’t think model matters as the issue seem to be the firmware) and now I cannot get my media bridges (3 different ones) to either connect to the 1142 AP or obtain and pass the DHCP addresses to other device connected to the built in switch. If I reload the 1142 AP firmware to 12.4, than this works fine. I have not seen anything in the release notes that changed how this works or if there is I could not find it.
Does anyone know why this changed and if there is any settings that I need to enabled / disable?
Any help on this would be greatly appreciatedMore info to add to this.
AIR-AP1142N-A-K9 Hardware Version of v06 works with firmware 15.2.
AIR-AP1142N-A-K9 Hardware Version v05 does not work with firmware 15.2, but will when downgraded to firmware 12.4.
I'm also having this issue with Cisco Aironet 3602 Fw 15.2(2)JB and 3502 Fw 15.2(2)JB$ that's on a Cisco 2500 WLAN Controller Sw Ver. 7.4.100.0.
Any help on this would be greatly appreciated -
Cisco Catalyst 4507R+E with WS-X4748 blade, ports not working
I have a Cisco Catalyst 4507R+E (with Sup7-E) and two blades; one WS-X4748-RJ45V+E and one WS-X4648-RJ45-E.
When I connect a device to a port on the WS-X4748-RJ45V+E blade the port will not come up, show interface shows the status as "notconnect". When I connect the same device to the WS-X4648-RJ45-E blade the interface comes up.
The WS-X4748-RJ45V+E blade seems to have initialised okay, it appears in the output of "show module" as OK.
I get exactly the same effect on a second, identically configured Catalyst 4507R+E.
The software version is IOS XE 3.1.0SG, which according to the release notes supports the WS-X4748-RJ45V+E blade. Has anybody else seen anything like this?I have 2 Core switches, single SUP on each.
Line cards #1 and #2 randomly stop forwarding packets, only solution is to reload the switch ( hw-module reset does not work ). I have cat4500e-universalk9.SPA.03.05.00.E.152-1.E.bin and the sympton looks the same as the described one,
although the code is newer than 3.2.2SG.
Now we are downgrading to cat4500e-universalk9.SPA.03.04.02.SG.151-2.SG2.bin to check if that solves the issue,
otherwise we´ll try removing/re-inserting the modules, and if issue persists, most probably RMA the Core1.
We´ve sent a show tech to Cisco support while the issue was happening.
Current modules on the Core switches.
Mod Ports Card Type Model Serial No.
---+-----+--------------------------------------+------------------+-----------
1 48 10/100/1000BaseT EEE (RJ45) WS-X4748-RJ45-E CAT1627L48B
2 48 10/100/1000BaseT EEE (RJ45) WS-X4748-RJ45-E CAT1629L0ZY
3 4 Sup 7-E 10GE (SFP+), 1000BaseX (SFP) WS-X45-SUP7-E CAT1629L1PD
5 12 1000BaseX (SFP) WS-X4612-SFP-E JAE163007EO
M MAC addresses Hw Fw Sw Status
--+--------------------------------+---+------------+----------------+---------
1 30f7.0d57.80c0 to 30f7.0d57.80ef 1.1 Ok
2 30f7.0dac.fd40 to 30f7.0dac.fd6f 1.1 Ok
3 30f7.0dca.6c00 to 30f7.0dca.6c03 2.1 15.0(1r)SG10 03.05.00.E Ok
5 a493.4c44.13e8 to a493.4c44.13f3 1.1 Ok
Mod Redundancy role Operating mode Redundancy status
----+-------------------+-------------------+----------------------------------
3 Active Supervisor SSO Active -
Cisco/IPsec VPN built-in service of 10.6.1 does not work!
Hello,
I have been using for a while in Leopard (10.5) the Cisco VPN client delivered by Cisco company until I upgrade to Snow Leopard (10.6.1) which comes with a native built-in Cisco VPN client and I gave it a try in order to replace my dedicated Cisco app.
I set up the Cisco VPN service in System Preferences > Network with the same settings than those used in the Cisco client but the connection fails when it is launched from the 10.6 network VPN service... while it works perfect when launched from the Cisco app itself.
I need to activate a VPN connection in order to connect from home to my enterprise server and I have to respect the VPN settings the network administrator of my enterprise put in place.
Those are very common:
1. Host name
2. Group name + Group password
3. Domain name\Userid + User password
4. RSA pass code (random code provided by a specific RSA keyfob)
5. Transport is IPsec over UDP
According to my testings, I would say that:
1. The connection to the host is OK.
2. The validation of the group name + group psw is OK.
3. The validation of the userid + user psw is OK.
4. The RSA pass code is rejected.
According to my enterprise network engineer's investigations, the possible reasons of the connection failure could be:
1. the UDP protocol is not (well) supported by the client service.
2. and/or the extended authentification phase (aka "Xauth") is not working as it should.
As far as I can see in other VPN clients, there is usually an option to select whether the transport is run over TCP or over UDP. Unfortunately, I have not been able to find such option in 10.6.
In my opinion, it could be either a bug or an (undesired) limitation of the Apple VPN service. In both cases, it requires a quick fix from Apple as, for time being, this issue prevents me and many of us connecting to our enterprise servers when we are far from its local network.Thank you for your answers which confirm the limitation of the Apple VPN solution to the TCP transport only.
I have to say that I do not understand such a decision from Apple since the UDP protocol is very common in the enterprise world.
I will thus have to rely on the Cisco app itself. Is the version 4.9.01 (0080) the correct one for SL as well?
Thank you! -
USB Connect Cisco for Mac does not work
I created a USB for Cisco Connect on my desktop with Windows XP but it will not work to connect a Mac to the Internet.
When I insert the USB, the Connect Icon will not execute. Is there something special that has to be done?Hey! Are you referring to the Cisco Connect software installed in the router? What's the model number of your router? Or if you're referring to the Smart Wifi software's USB feature, you may check this link:
http://homekb.cisco.com/Cisco2/ukp.aspx?vw=1&docid=f2fd60559043499c8643c3deea7c8ede_Overview_of_the_... -
using the lync client connectivity tester on a pc on the same lan as my mobile client everything is green and it says its ready for use.
using my android galaxy s5 client on wifi on the same lan i get a screen with waiting to sign in spinning and an error at the top "we cant connect to the server check your network connection and server address, and try again."
i have uploaded the full client log files
here: client log file
some errors that stand out from this log file are:
1. ERROR HttpEngine: Certificate check fails: java.security.cert.CertPathValidatorException: Trust anchor for certification path not found.
2. <h2>401 - Unauthorized: Access is denied due to invalid credentials.</h2>
<h3>You do not have permission to view this directory or page using the credentials that you supplied.</h3>
i am using the correct creds, same creds i used on the analyzer tool.
in the analyzer tool i did have to fill in the username field because my sip domain is different then my users UPN. which from what ive read its required to use the username field.
i also filled in the username field in the mobile app with domain\username
3. ERROR LYNC: ERROR TRANSPORT /Volumes/ServerHD2/buildagent/workspace/200604/tps/ucmp/platform/networkapis/privateandroid/CHttpConnection.cpp/295:CHttpConnection exception: java.lang.NullPointerException
Jan 14, 2015 8:40:49 AM INFO LYNC: INFO TRANSPORT /Volumes/ServerHD2/buildagent/workspace/200604/tps/ucmp/ucmp/transport/requestprocessor/private/CHttpRequestProcessor.cpp/173:Received response of request(UcwaAutoDiscoveryRequest) with status = 0x22020001
Jan 14, 2015 8:40:49 AM INFO LYNC: INFO TRANSPORT /Volumes/ServerHD2/buildagent/workspace/200604/tps/ucmp/ucmp/transport/requestprocessor/private/CHttpRequestProcessor.cpp/201:Request UcwaAutoDiscoveryRequest resulted in E_ConnectionError (E2-2-1). The retry
counter is: 0
4. Jan 14, 2015 8:40:50 AM ERROR LYNC: ERROR TRANSPORT /Volumes/ServerHD2/buildagent/workspace/200604/tps/ucmp/ucmp/transport/authenticationresolver/private/CAuthenticationResolver.cpp/431:Failing the original request as we weren't able to get the token
this is the same type of error i was getting in the lync connectivity analyzer until i filled in the username field. but its filled in, in my client.
again you can see the full log file is `HERE
thank you in advance for any help. im trying to get internal working before i try external.Eric,
I am trying to configure a reverseproxy on my netscaler which is in a 2 arm mode(dmz/internal) but I keep getting an error when configuring the monitor.
i used this guide to configure it
http://www.lynced.com.au/2014/04/configure-citrix-netscaler-vpx-as.html
but continue to get this error in the netscaler monitor "Failure - TCP connection successful, but application timed out"
so the virtual server is never up, thinking about just changing it to tcp as a monitor so it stays up and i can at lesat get the vip up.
Also your link to the diagram shows it going to the reverse proxy but the one im using has it going directly to the front end servers.
http://www.lync-solutions.com/Documents/Lync_2013_protocol_poster_v6_7.pdf
I'm guessing Microsoft's is the correct one but wonder why the config differential?
I see that your diagram says "mobility url", what is the mobility url? i though that was the lyncdiscoverinternal.internal.com
current setup is
2 fe servers on internal
1 edge server on dmz
1 almost done reverse proxy netscaler load balancer.
also this ms link i used to configure dns entries, along with the pdf linked above.
http://technet.microsoft.com/en-us/library/jj945644.aspx
i currently have these external dns entries and they all point to the edge server on the dmz.
dialin .external.com
lync .external.com
lyncweb .external.com
lyncdiscover.external.com
meet .external.com
sip .external.com
webconf .external.com
av .external.com
_autodiscover._tcp.external.com.
the internal dns links point to 1 of the front end servers
1. lyncdiscoverinternal.internal.com
2. lyncdiscover.internal.com
3. _sipinternaltls._tcp.internal.com
4. _sipinternal._tcp.internal.com
5. sipinternal.internal.com
6. sip.internal.com
thanks again for your help. -
Cisco WS-2960S-48TS-L does not work after firmware update
Hello.
After purchasing the Cisco WS-2960S-48TS-L decided to do a firmware upgrade using the CISCO NETWORK ASSISTANT. Unfortunately update failed. Currently the switch when you run can not load the system image.
Since up to now I had to deal with the other switches do not really know how to upload the correct image now. The more that you connect the switch through the console acts as to a limited extent.
Please helpThank you Leo Laohoo.
Your suggestion to usbflash0 helpful.
It is true that I was struggling because of usbflash latest iso system also did not want to go, but I recorded version of one lower and went.
By the way, I learned that the USB must be formatted to FAT file system.
Again, thank you for your help.
Darek Rynkowski -
Cisco 871 to Cisco ASA 5545 Site-to-Site VPN Split Tunnel not working.
Tunnel comes up and can see and access protected traffic but cannot access web (Split Tunnel). Don't know if access problem or route issue.
Listed below is configuration for Cisco 871, any help very much appreciated.
crypto isakmp policy 1
encr 3des
authentication pre-share
group 2
crypto isakmp key test address x.x.x.x
crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha-hmac
crypto map SDM_CMAP_1 1 ipsec-isakmp
description Tunnel to x.x.x.x
set peer x.x.x.x
set transform-set ESP-3DES-SHA
match address 100
interface FastEthernet0
interface FastEthernet1
interface FastEthernet2
interface FastEthernet3
interface FastEthernet4
ip address 4.34.195.193 255.255.255.192
no ip redirects
no ip unreachables
no ip proxy-arp
ip route-cache flow
duplex auto
speed auto
crypto map SDM_CMAP_1
interface Vlan1
description $ETH-SW-LAUNCH$$INTF-INFO-HWIC 4ESW$$ES_LAN$$FW_INSIDE$
ip address 172.200.1.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip route-cache flow
ip tcp adjust-mss 1452
ip route 0.0.0.0 0.0.0.0 4.34.195.193 permanent
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
logging trap debugging
access-list 100 remark SDM_ACL Category=4
access-list 100 remark IPSec Rule
access-list 100 permit ip 172.200.1.0 0.0.0.255 172.16.2.0 0.0.0.255I don't see any NAT configuration above. Check you can PING out to the internet (8.8.8.8 for example) from the router itself as it won't need NAT to PING from the outside interface.
Have a look at this document on setting up NAT for your inside devices:
http://www.cisco.com/c/en/us/support/docs/ip/network-address-translation-nat/13772-12.html -
Cisco Prime Infrastructure 2.2 Rest API XML does not add LF
I am using several scripts that dig the data from XML files created through Cisco Prime Infrastructure Rest API. It worked fine until 2.1. Now, with 2.2, seems Cisco stopped adding LF to each line and everything is 'one' line.
Here is an example with 2.1, there is a LF to the end of each line:
?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<queryResponse type="AccessPointDetails" rootUrl="https://server/webacs/api/v1/data" requestUrl="https://server/webacs/api/v1/data/AccessPointDetails?type=contains(UnifiedAp)&.full=true&.maxResults=1000&.firstResult=0" responseType="listEntityInstances" count="715" first="0" last="714">
<entity url="https://server/webacs/api/v1/data/AccessPointDetails/1505569" type="AccessPointDetails" dtoType="accessPointDetailsDTO">
<accessPointDetailsDTO id="1505569" displayName="1505569">
<adminStatus>ENABLE</adminStatus>
<apType>AP1240</apType>
<cdpNeighbors>
<cdpNeighbor>
<capabilities>Switch IGMP </capabilities>
<duplex>Half Duplex</duplex>
<interfaceSpeed>100Mbps</interfaceSpeed>
<localPort>2</localPort>
<neighborIpAddress>10.1.1.1</neighborIpAddress>
<neighborName>switch</neighborName>
<neighborPort>FastEthernet0/1</neighborPort>
<platform>cisco WS-C3560-8PC</platform>
</cdpNeighbor>
Now, with 2.2, there is not LF anywhere:
?xml version="1.0" encoding="UTF-8" standalone="yes"?><queryResponse type="AccessPointDetails" rootUrl="https://server/webacs/api/v1/data" requestUrl="https://server/webacs/api/v1/data/AccessPointDetails?type=contains(UnifiedAp&.full=true&.maxResults=1000&.firstResult=0" responseType="listEntityInstances" count="715" first="0" last="714">
<entity url="https://server/webacs/api/v1/data/AccessPointDetails/1505569" type="AccessPointDetails" dtoType="accessPointDetailsDTO">
<accessPointDetailsDTO id="1505569" displayName="1505569"><adminStatus>ENABLE</adminStatus><apType>AP1240</apType><cdpNeighbors><cdpNeighbor>
Does anyone know, is this intentional or a mistake? Or is there a way how I control this?
I am reading on-line CPI Rest API docs, but seems cannot find anything.
Thanks,
VladPI 2.0 is in Beta mode for testing in closed group already. However it may still take some time to get released on CCO for FCS.
I think it is expected to be released between aug-sep. BU has also started working on CPI 2.1, which you can see test uploads on CCO.
For what does it have for customers?
> As Cisco have a vision of one network Management software for both wired and wireless infrastructure, which started getting real partially with Cisco PI.
As WCS got evolved to NCS and eventually to Cisco Prime Infrastructure, but it doesnt yet have full Management capability for wired infra., which is there with LMS.
With CPI 2.x Cisco plans to blend entire LMS and WCS features together.
-Thanks -
MWI does not work on Cisco IP Phones using VG248
Hi,
We are facing issue with MWI on/off using SMDI connectivity to FCS voicemail system with Cisco Callmanager (6.1.x). Below is the connectivity.
CUCM ver 6.1.4---->VG248 ver 1.3.1 (SMDI and Asyc1) ----> FCS
We are able to receive and send voicemails on the IP Phones but the light never lit. I verified the MWI numbers on CUCM/VG248 and FCS voicemail system, rebooted all the appliances with no luck.So if have the port numbers with you for Sonos multiroom-system then you can do the port forwarding and follow the steps:-
First of all get your Device IP address for which you want to do the Port Forwarding...
Then open the router's setup page click on the Applications and Gaming Tab and then click on the sub tab Port Forwarding then open the ports that you want...
Refer the link : http://media.photobucket.com/image/XBOX%20360%20NAT%20issues/EricScpoc/linksys_setup.png
Then logout from your Router's setup page.. Unplug your Router from the power, wait 30 seconds. Then plug in the Router again.
Try out these settings it will help you out... -
Cisco Meeting Place Express 2.1.1.2 and not working anymore
Hello, we currently have cisco meeting place express 2.1.1.2; however are now having issues with it. We are unable to dial to the internal extension that was previously working to login to the meetings and only get a busy signal everytime. Furthermore, whenever someone tries to schedule a new meeting via the web form, it says "request to server timed out. please try again".
I tried reaching out to Cisco TAC support; however it's no longer a supported item and am looking for assistance here. The TAC engineer was able to help assist with updating the system clock; however I believe the problems started happening after we updated the clock and rebooted the server. Now we are unable to schedule and attend any meetings (busy signal everytime) at all. I tried restarting the server itself as well as via the command line. I also now see an error during CLI restart, which I have listed directly below:
Stopping MeetingPlace SNMP/R: [ OK ]
Stopping mpe-tomcat-main: bash: /root/.bashrc: Permission denied
[ OK ]
Stopping mpe-tomcat-securewebservices: bash: /root/.bashrc: Permission denied
[ OK ]
Shutting down breeze: [ OK ]
Shutting down fmsmaster: [ OK ]
Shutting down fmsadmin: [ OK ]
Stopping MeetingPlace application: /etc/init.d/mpx_app: line 264: 4955 Segmentation fault (core dumped) ${shutdown_bin} ${shutdown_arg_1} ${shutdown_arg_2} ${shutdown_arg_3} >>${non_verbose_shutdown_output} 2>&1
[FAILED]
ERROR: returncode = 139
Stopping MeetingPlace License Manager: [ OK ]
Stopping MeetingPlace database: [ OK ]
Stopping MeetingPlace Java RMI: Shutting down RMI Registry:
[ OK ]
Stopping Cisco MeetingPlace Express: [ OK ]
Starting MeetingPlace Java RMI: Starting RMI Registry:
starting daemon
mpxadmin
[ OK ]
Starting MeetingPlace database: [ OK ]
Starting MeetingPlace License Manager: [ OK ]
Starting MeetingPlace application: [ OK ]
already running
Starting breeze: [ OK ]
Starting fmsmaster: [ OK ]
Starting fmsadmin: [ OK ]
Starting mpe-tomcat-main: bash: /root/.bashrc: Permission denied
[ OK ]
Both the admin and user web interfaces work on the home pages, but can't seem to schedule any new meetings or login to any existing ones, with busy signal as well. Anyone have any ideas? We need assistance with this ASAP since it's our voice collaboration solution. Thanks in advance for any help.Amlesh,
Here is the pertinent output:
/common/log/taos-log-a/core/core.1947.11.ConfSchd.1183266069
/common/log/taos-log-a/core/core.1837.6.gyromain.bin.1183266161
/common/log/taos-log-a/core/core.1859.6.cpgs.1183266206
/common/log/taos-log-a/core/core.1951.11.ConfSchd.1184475669
/common/log/taos-log-a/core/core.1946.11.ConfSchd.1185685269
/common/log/taos-log-a/core/core.1836.6.gyromain.bin.1185685332
/common/log/taos-log-a/core/core.1880.6.cca.1185685452
/common/log/taos-log-a/core/core.1858.6.cpgs.1185685452
/common/log/taos-log-a/core/core.1950.11.ConfSchd.1185771671
/common/log/taos-log-a/core/core.1863.6.cpgs.1185771865
/common/log/taos-log-a/core/core.1884.6.cca.1185771865
/common/log/taos-log-a/core/core.1991.6.nmpagent.1185771865
/common/log/taos-log-a/core/core.1950.11.ConfSchd.1185858067
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1188018079
/common/log/taos-log-a/core/core.1969.11.ConfSchd.1188190869
/common/log/taos-log-a/core/core.2008.6.vui.1188190988
/common/log/taos-log-a/core/core.2029.6.nmpagent.1188190988
/common/log/taos-log-a/core/core.1998.6.poclient.1188190988
/common/log/taos-log-a/core/core.1970.11.ConfSchd.1189227671
/common/log/taos-log-a/core/core.1862.6.gyromain.bin.1189227762
/common/log/taos-log-a/core/core.1883.6.cpgs.1189227807
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1189918866
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1190005281
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1190178068
/common/log/taos-log-a/core/core.1886.6.cca.1190178237
/common/log/taos-log-a/core/core.2012.6.nmpagent.1190178237
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1191906070
/common/log/taos-log-a/core/core.1843.6.gyromain.bin.1191906162
/common/log/taos-log-a/core/core.1886.6.cca.1191906248
/common/log/taos-log-a/core/core.1865.6.cpgs.1191906248
/common/log/taos-log-a/core/core.1950.11.ConfSchd.1192424479
/common/log/taos-log-a/core/core.1842.6.gyromain.bin.1192424571
/common/log/taos-log-a/core/core.1863.6.cpgs.1192424613
/common/log/taos-log-a/core/core.1951.11.ConfSchd.1192597270
/common/log/taos-log-a/core/core.1969.11.ConfSchd.1192683669
/common/log/taos-log-a/core/core.1903.6.cca.1192683817
/common/log/taos-log-a/core/core.1951.11.ConfSchd.1193029270
/common/log/taos-log-a/core/core.1842.6.gyromain.bin.1193029360
/common/log/taos-log-a/core/core.1864.6.cpgs.1193029400
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1193720467
/common/log/taos-log-a/core/core.1843.6.gyromain.bin.1193720565
/common/log/taos-log-a/core/core.1864.6.cpgs.1193720610
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1195452071
/common/log/taos-log-a/core/core.1843.6.gyromain.bin.1195452170
/common/log/taos-log-a/core/core.1864.6.cpgs.1195452213
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1196834481
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1197007273
/common/log/taos-log-a/core/core.1842.6.gyromain.bin.1197007371
/common/log/taos-log-a/core/core.1864.6.cpgs.1197007413
/common/log/taos-log-a/core/core.1952.11.ConfSchd.1198044080
/common/log/taos-log-a/core/core.1843.6.gyromain.bin.1198044184
/common/log/taos-log-a/core/core.1842.6.gyromain.bin.1199858530
My understanding is these core files are crash dumps and are safe to delete?
Jared
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