Why there is a SIP settings if VoIP is not present...
Hello,
Kindly ask you to explain subject of this post. There is SIP settings on my C5-03 but I can find how to make an internet call ! Moreover nothing about SIP in official attached phone manual.
There is an advise to install "SIP_VoIP_3_1_Settings_S60_5_x_v1_0_en.sis" which will create additional menu like "Advanced SIP seetings" but it doesn't help in my case. Probably that old plugin just not capable with new C5-03.
I can see that this is not only my problem. Please try to resolve this issue, it's very stupid to have SIP settings but not able to make internet VoIP call.
Best regards,
another department.
Hello,
Thank you for quick reply. But seems that you didn't read my post carefully.
I have already perform SIP settings and made my phone registered with configured SIP profile.
Problem is that I can't find how to make VoIP call after that.
In Nokia E63 it was very simple After dialing the number you need just to choose voice, video or Internet call (using created SIP profile) option.
But seems R&D just forgot to put it in call options.
Also given link are leading to already installed "SIP VoIP Settings (SIP_VoIP_3_1_Settings_S60_5_x_v1_0_en.sis)".
However mentioned "Menu > Settings > Connectivity > Admin.settings > Net settings > Advanced VoIP settings > VoIP services"
still not present.
Same was indicated by other users, shown in my previous post.
I'm sure that somebody from Nokia is present here, guys, just tell how to make VoIP call using already registered and configured SIP profile, so simple.
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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 : SHi 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. -
N900 - "public user name" in sip settings!
on symbian n95/n82 phones: sip settings: there is a field "public user name".
the info there is what the caller id sends to the receiver of the call.
so i had "sip:[email protected]" (nonoh is my voip service and my mobile is
registered with them) and the receiver of my calls sees my mobile # calling ...
not private #, or a strange # that he/she would not care to call back ...
how can i add this to my settings
i tried to play with accounts.cfg
in
/home/user/.rtcom-accounts/
but with no effect
please help!!
thank u!anyone can help?
thank you!! -
HI All
Has any one managed to set X3-02 with sip settings from third part provider not supported in Internet Phone settings?
Tried to follow this
Nokia X3-02 with VOIP other than Gizmo5
and I have set it up without any problems and got logged in.
The problem starts when calling as the other person cannot hear me. I can here with no problem but I also hear my own voice (just like echo).sudude wrote:
Has worked fine for me and on my friends phones as well.
Take a look at this, /t5/Pool-of-Knowledge/Series40-phones-and-VoIP/td-p/1146267
Tried this already and got the it logged in for 12voip as well as justvoip. I can call and I get connected but the communication works only one way which means that the person who I call can not hear a thing. On my site I can hear with no problem and quality is fine.
I presume that there is something wrong with my settings, please see xml uploaded in my phone attached.
sudude Did you create you own xml or used n0kVoIP? I Google it but I cannot get it anywhere.
Attachments:
example.zip 2 KB -
Why there is not an option to unable the 3G and use 2G on iPhone 4S ? It's awful! When will you fix that ? What are we gonna do now? I cant believe Apple can do it, like the battery of iPhone 4S is amazing. Everyday I should charge my phone! It doesnt make sense...
YOu can't disable the camera - this feature came out with IOS 5.
You will have to set your passcode lock to immediately. settings - general - passcode lock. -
How to Delete Locked SIP Settings.
Hello,
I tried adding one profile to my SIP Settings. It got created successfully but a LOCK SIGN is appearing along with it. and I am not able to DELETE the profile or edit it.
Can anyone kindly guide me how to DELETE/EDIT it and send the requested information to my mail id [email protected]
Thanks,
Pawan.try resetting device with the following codes
With all these operations I recommend you TAKE OUT MEMORY CARD and DO A BACK UP OF YOUR STUFF!!!
NOTE: In front of the hash key ( #) there is also a STAR key that people often miss and state that the code does not work.
1.. *#7780# - Restore factory settings - resets all the settings to the default ( you will not lose any data) Make sure you back up your data as you will lose all of it when you perform option 2 or 3 on this list. The default code for ALL operations listed here is 12345.
2.*#7370# - Reformat your phone (out of the box, tho keep in mind that most newer nokia phones at least n series, e series and s60 based phones have udp - user data preservation so not ALL data may be lost. still it is a good idea to always do a back up of your stuff.)3. This you perform as a last resort. Nothing else is working.If the phone is not showing any activity, proceed with hard formatting , turn off your phone, hold the following buttons while pressing the power button. (the default code is 12345)
3. hard reset - hold the following buttons * (star key), 3 (number button) and talk/green key. turn on the phone and do not release those buttons until you see the Nokia boot up screen. once you feel the phone power up you can let go off the power button while still holding all three buttons ( for Nokia 5800XM - use the following buttons to do a hard reset, /GREEN/RED/CAMERA keys pressed all at once on power up. this will only work on 5800's that have firmware version 20... and up. ) For Nokia N97 (unofficial version but works) is SHIFT /SPACE BAR / BACKSPACE while holding the power button. Again do not let go until you see the handshake screen.
If these codes are not working the only thing for you to do is contact your Nokia Care Center/Service for assistance.
If i have helped at all a click on the white star below would be nice thanks.
Now using the Lumia 1520 -
Why the dependency of "Site settings" for the "About me"-button in the top menu?
Hello, I recently had a customer case where the customer asked why certain users couldn't see the "About me"-button in the top personal actions menu, after they clicked at the user name/avatar. Only user details, "Sign out" and "My
settings" was visible for the user, who had an user license and a my site. There was also no customizations whatsoever in either javascript or css that could cause this, and the masterpage or the personal actions control doesn't seem to actually have
anything to do with the "About me"-button.
After some tracing and testing of Microsoft's code so we now know why the "About me" is not visible to all users. "About me" button depends on whether the "Site
Settings" is available in the menu or not. For the "Visitors" then there is no "Site Settings" because of their "Read"-permission level and thus does the "About me" button not appear in the menu.
In sposuitenav.js there is a loop for all ie:menuitem and inside of that there is these lines:
if(menuItem.id.indexOf("MenuItem_Settings") > 0) {
hasSiteSettingsMenuItem = true;
And a bit further down the code there is this:
if(!hasSiteSettingsMenuItem) {
a.AboutMeLink = null;
Which will cause so that the "About me"-link is null and is not rendered in the personal actions menu. Otherwise the AboutMeLink would have had the default value that is set by the code earlier.
So my question really, is why have Microsoft made the "About me"-button depending on whether or not the "Site settings" is visible for the user?
Because neither I nor my client can understand why there should be any relationship between Site Settings and the About Me link. Shouldn't anyone be able to go to their "my site"? As long as they have one, I can't see any reason to prohibit
anyone else than site administrators to see and use the "About me"-link.It's a recent change it seems, people on the Office 365 Yammer network have noticed this as well. Microsoft has confirmed this as an issue and will work to fix it.
Join there if you haven't already :)
https://www.yammer.com/itpronetwork/#/Threads/show?threadId=466244664 -
How to configure SIP settings on Nokia E51
Hello to all.
I need help. I have
Username
Authentication Username
SIP address
SIP Password
Domain
Outbound Proxy
in one of big SIPs provider but i cant configure my phone work with it. Please help me. I allready installed SIP Voip settings from nokia store.Have you checked out this post upon forum http://discussions.nokia.com/t5/Connectivity/Nokia-E51-SIP-Settings/m-p/481535/highlight/true#M24073 or this resource?http://www.voip-connections.com/wp-content/uploads/2011/06/How-to-configure-Nokia-E51-and-E71-to-con...
Happy to have helped forum in a small way with a Support Ratio = 37.0 -
Hello, I wonder why there isn't Finnish language in the adds-on list. Could you please make it available to Firefox users. Thank youin advance. Rappe
(1) Firefox's built-in post-crash page has not been a real HTML page for a long time (for example, from the time of Firefox 22, see: [https://support.mozilla.org/en-US/questions/968212 Want to save LOTS of versions of "Restore Session.xht" from the "oops ..." page for later use]). If you had this working differently with Firefox 25, that might have been created by an extension.
You can check to see whether extensions are disabled or need an update on the Add-ons page. Either:
* Ctrl+Shift+a
* orange Firefox button (or Tools menu) > Add-ons
In the left column, click Extensions. The disabled extensions cluster toward the bottom of the list. To poll for updates, use the "gear" button above the list and choose Check for Updates.
If you used the Reset feature (or Firefox automatically did a reset due to some problem during upgrading), you will need to reinstall missing extensions. The reset feature creates a folder on the desktop named Old Firefox Data. Do you have that folder? There may be data you can recover from it.
(2) There are many ways for history to get cleared, both internal to Firefox and external. Could you double-check your Privacy settings?
orange Firefox button (or Tools menu) > Options > Privacy
* The "Firefox will" drop-down says Remember History: Firefox shouldn't be clearing history, but an add-on or external software could do it
* The "Firefox will" drop-down says Use custom settings for history: inspect the "Clear history when Firefox closes" setting to make Firefox isn't set to clear history. Also check your add-ons and consider external software.
Firefox normally accumulates months of history. However, some of Firefox's database sizes are based on disk space available. If your hard drive is very full, Firefox might reduce the amount of history stored. -
Why there are two IMG steps provided for Home Page configuration?
Dear experts,
I wonder why there are two IMG steps provided for configuring ESS home page.
1. Cross Application Components --> Employee Self Service --> General Settings
2. Personnel Management --> Employee Self-Service --> General Settings
Both tables contents are also similar.
Why SAP has provided same configuration steps in two areas?Your first one is wrong. It is under Cross App. Components but not in an ESS node....I am sure it is a typo on your part. Anyways......
It's NOT two steps. It's just the same configuration in two places. It makes it easier to locate for instance if you are in ESS to just have it there. BUT the Homepage Framework is used elsewhere (MSS for example) as well as you can use it for whatever else you like (building an easier interface for other things) therefore, it has it's "true" place up as a "cross application component".
This is very similar to say Benefits. If you look under the Benefits config area in the IMG, you will actually see a lot of ESS settings there that are the exact same ones you see under the ESS node too.
Don't let it confuse you. -
There are no export settings for DVCPRO HD
Hello,
I've tried to export video from Final Cut Pro and After Effects in DVCPro HD 720. I've noticed that for any DVCPROHD format there are no quality settings in the "Compressor" box of the export window. What's strange is that on my old computer I did have those options.
Because I cannot specify 16X9 or 4X3, I have to render anything exported from After Effects into FCP.
I am trying to figure out if I am missing something on the install?
Any help would be greatly appreciated.I just read this week...and DVCPROHD may be in this category... When a quality side shows up in some codec...the slider has no effect on the quality of the video. It just shows up... I've seen a quality slider show up when using the AIC codec. The slider has no effect on the video
DVCPro HD 720 doesn't have a quality slider. And why would it...HD right. And if in some program it does show up, I bet it does nothing.
What problems are you having with aspect ratio 4x3, 16x9. If you trying to set aspect ratios in Compressor and not getting what you want, it can be done. If your changing from one to another there is some math to do. And sometimes you have to crop off some of the video. -
why don't my apps automaticaly update? i keep trying to turn it on under settings but it will not stay on
Sounds like we have a similar problem. Four days ago I was playing music through a wireless Bose speaker via bluetooth and everything was fine. Then I was texting while listening and got 2 texts about the same time as I was typing one and my phone went black. I pushed the home button and it worked, went into my texts and all were there. However ever since then the time isn't right, so it can't keep time no matter what I do. The bluetooth works only at times, the wifi keeps jumping on and off, sometimes my data won't work and the other day I couldn't even turn it off, so had to wait until it died. The battery life display at that moment wasn't correct either. Took to Verizon, they couldn't help. Suggested a factory reset, which I did last night and still the same issues! One day I couldn't even text my husband on his iPhone. Just wouldn't work at all. Did everything suggest and still not good. Seems like after stalking these pages today a lot of people are having this problem in Dec/Jan and of course also I'm 3 weeks out of my 1 year warranty and so were many others, by days. Very strange, and all on the 5C. What did you do? Is it working now?
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I cannot get an e65 to successfully register to a 3com NBX SIP server. My settings are
service profile: IETF
default access point: AP Profile (this WLAN works and I can connect to the server from internet browsing)
Public User name: sip:[email protected]
Compression: no
Registration: Always on
use security: no
Proxy Server
Address: sip:10.10.10.1
Realm: None (do not add this to other SIP phones)
username: 422
password: xxxx
Loose Routing: No
Transport Type: UDP (Have tried all three)
Port: 5060 (This is as set on Server)
Registrar Server
address: sip:10.10.10.1
Realm: None
Username: 422
Password: xxxx
Loose Routing: No
Transport: UDP
Port: 5060
Both Proxy and Registration Servers are set the same as they are the same Box. I'm not sure what to do with "Realm", my other SIP phones do not require this setting. Memory cards are no in, Is this a problem?
Regards,
AdamI dont think I can remove the password, the user profile created on the 3Com NBX must have a password set. I tried the following setting but still get a failed to connect message. Is there documentation on what all the SIP settings are? eg. use security? use compression?
service profile: IETF
default access point: AP Profile (this WLAN works and I can connect to the server from internet browsing)
Public User name: sip:[email protected]
Compression: no
Registration: Always on
use security: no
Proxy Server
Address: sip:10.10.10.1
Realm: None 10.10.10.1
username: 422
password: xxxx
Loose Routing: Yes
Transport Type: UDP (Have tried all three)
Port: 5060 (This is as set on Server)
Registrar Server
address: sip:10.10.10.1
Realm: 10.10.10.1
Username: 422
Password: xxxx
Loose Routing: Yes
Transport: UDP
Port: 5060
Regards,
Adam -
Hi All,
Can anybody provide detailed guide on how to make SIP settings for N95 so that I can use internet telephony thru my router?
Thanks in advance.It'll involve using a proxy put at your disposal by your VoIP provider. Contact them for more details. People here can't help you because they have no reason to know your provider's settings.
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