Unified Mobility, no audio on Send call to Mobile Phone
I'm using UCM 9.1 with Unified Mobility (xfer to alternate number) with good success if I follow a typical call flow:
Inbound call -> Ext -> Rings deskphone x seconds -> Rings mobile phone -> Answer mobile phone -> hangup -> Resume call on desk phone.
But if I pick up a call on my desk phone, and use the Mobility button to xfer a call to my mobile phone I get no audio:
Inbound call -> Ext -> Pickup desk phone -> Mobility soft button, 'Send call to Mobile Phone' -> Answer mobile phone, no audio -> Hang up mobile phone -> Resume call on desk phone (two-way audio).
Device wise the call flow is:
ITSP SIP trunk -> CUBE -> CUCM -> 7965 IP Phone.
Recently I reconfigured CUCM to use the CUBE for any MTP resources instead of the software option and I think I may have missed something.
CUBE config:
voice-card 0 dspfarm dsp services dspfarm!!!voice service voip ip address trusted list ipv4 173.46.30.218 ipv4 173.46.30.202 ipv4 10.0.6.30 ipv4 10.0.6.31 ipv4 10.0.6.33 ipv4 10.0.6.32 ipv4 10.1.1.4 ipv4 10.0.250.0 255.255.255.0 mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip midcall-signaling passthru media-change early-offer forced no call service stop registration passthrough!voice class codec 1 codec preference 1 g711ulawsccp local GigabitEthernet0/0.42sccp ccm 10.0.6.30 identifier 1 version 7.0 sccp!sccp ccm group 1 bind interface GigabitEthernet0/0.42 associate ccm 1 priority 1 associate profile 1 register MTP_2951-01!dspfarm profile 2 transcode universal codec pass-through codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 11 associate application SCCP shutdown!dspfarm profile 3 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP!dspfarm profile 1 mtp codec pass-through codec g711ulaw maximum sessions hardware 15 associate application SCCP!
In CUCM I removed the software MTP from the MRG:
Not sure where to start troubleshooting this problem, any help is appreciated.
Steve
Thanks for the help gents. I couldn't get to this until we're out of office hours on the weekend.
Interestingly, I have no mid-call option in my dial-peers. This is a 2951 running 15.2(4)M2.
I double checked the MRGL, my phone is associated with it.
Codec is G711 on ITSP side, and on phones - I'm not sure I fully understand the use cases for MTP, this is something I need to research more.
I've included two ccsip message debugs, the first one is the existing issue of no audio (in either direction).
The second I've changed the midcall-signaling passthru option, dropping the media-change bit and we get audio in both directions for mobility except we use Unity call handlers for IVR functionality, and now when an inbound caller is forwarded to an extension we get no audio - obviously this is a game stopper.
In Unity I have the port group configured as SCCP - Maybe I should be using SIP instead?
No Audio:
voice service voip
sip
midcall-signaling passthru media-change
early-offer forced
no call service stop
registration passthrough
Dec 8 21:07:35.842: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 3244526336-0000065536-0000003600-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:07:35.850: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 3244526336-0000065536-0000003600-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536855Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9952 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:35.850: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:35.858: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Dec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120698 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:43.938: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120699 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:43.938: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:43.942: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274C4F2From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:07:43.958: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1d288c5e53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40265 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:44.158: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.158: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:07:44.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationSession-Expires: 1800;refresher=uacP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:44.214: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2151dd5c40From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40265 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 25834 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:52.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120700 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9953 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:52.610: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK8m4hbg10c8ag4kg723g0.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:07:52.630: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:07:52.634: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386536872CSeq: 101 BYEReason: Q.850;cause=16P-RTP-Stat: PS=0,OS=0,PR=420,OR=67200,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.634: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=2,OS=320,PR=100,OR=16000,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.646: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 101 BYEContent-Length: 0
bi-direcitonal audio:
voice service voip
sip
early-offer forced
midcall-signaling passthru
no call service stop
registration passthrough
Dec 8 21:09:44.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 0239559040-0000065536-0000003601-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:09:44.339: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536984Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesSession-Expires: 1800Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2507 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:44.339: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:44.347: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536984Dec 8 21:09:52.535: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674502 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:52.539: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:53.007: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:09:53.007: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674503 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:54.967: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274F1060From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:09:54.979: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2876c28ab9From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40276 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.007: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:09:55.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 102 INVITEMax-Forwards: 70Timestamp: 1386536995Contact: Diversion: ;privacy=off;reason=follow-me;screen=yesExpires: 180Allow-Events: telephone-eventContent-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:55.019: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Dec 8 21:09:55.031: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 279v=0o=root 1961674502 1961674504 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:55.035: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:55.175: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2c8e71c96From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.179: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK2751A88From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 102 ACKAllow-Events: telephone-eventContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674505 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:10:05.271: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITEMax-Forwards: 70Timestamp: 1386537005Contact: Expires: 180Allow-Events: telephone-eventContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.271: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceSupported: replacesSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=called;screen=yes;privacy=offContact: Content-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:10:05.279: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.279: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27531E8DFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4d5qq910785h6ks9f3c0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:10:05.295: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386537005CSeq: 102 BYEReason: Q.850;cause=16P-RTP-Stat: PS=511,OS=81760,PR=505,OR=80800,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=505,OS=80800,PR=634,OR=101440,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.303: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 BYEContent-Length: 0
Similar Messages
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Sending text message/SMS from mobile phone to Skyp...
Hi,
This is going to sound rather complicated but I'll try to explain my question as clearly as I can. I'm currently in Singapore and using Skype to keep in touch with my boyfriend in the U.S. I set up a Skype caller ID number several days ago so that when I call my boyfriend through Skype, he can see that caller ID number and know it's me. My question is, is it possible for my boyfriend to text me on that caller ID number using his mobile phone? And if so, will I have to be signed into Skype to receive his message? How will I be charged?
Thanks.hi,
SMS API(WMA) is an optional package. It is not a MIDP1.0 or MIDP2.0 api's.
There are phones which has WMA api with MIDP1.0 support .... Nokia 3650
Seimens has some phone with their own api's to send sms.Check out seimens site for more info
BTW, What do you mean buy sending SMS to Server????
If you want to send message to server you can do it with Http.
HTH
phani -
Can't we get calls to normal mobile phones from apple iPad 3 wifi+4g
I want to know weather we can get calls to normal mobile phones from apple I pad3 wifi+4g
The iPad is not a mobile phone and therefore cannot make or receive calls. The iPad can send a message to another iOS device using the Messages app but the iPad cannot send or receive cellular text messages using the SMS system that all cell phones use. There may be an app that gets around this limitation though so check the App Store.
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IPhone - Cisco Unified Mobile Communicator
I used to have Cisco Mobile 8.0 with ( Cisco Dual mode for iphone devices ) that start with TCT<username>
Within this App, I had to configure this in the settings of the app on the Iphone
Device ID : TCT....
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The Mobile 8.0 was working fine
I cant find Mobile 8.0 on the apps store anymore, I can find Cisco Unified Mobile Communicator, it seems to be the new version
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password
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Cisco 3905 failed but outgoing call still ringing on mobile phone
Dear all,
We setup Cisco Call Manager 10 with Cisco Voice gateway 2911 ( E1 port) and 3905. All outgoing call & incoming call are ok. But we meet a small issue as below:
-Using 3905 do outgoing call to mobile number. Mobile number ringing.
-3905 failed without power but the call still ringing on mobile phone.
Anyway to fix that ?
Thank you & regards,
ThànhNTHello,
So, your config looks good.
Pelase, can you explain what you mean when you say:
"-3905 failed without power but the call still ringing on mobile phone."
You can try a "debug isdn q931" on your gateway and see if the disconnect is ok. Use the documentation bellow to understand the debug:
http://www.cisco.com/c/en/us/td/docs/ios/12_2/debug/command/reference/122debug/dbfipx.html#wp1018126
If the disconnection is good, you will need to understand your call flow.
Phone -> CUCM -> Gateway -> PSTN -> Called Party
The final disconnection point is provided by your telco, and it can send the disconnect late.
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best regards. -
Lumia 925, earpiece audio trouble (voice calls)
Background: I am a T-Mobile USA subscriber. Previously owned a Lumia 710, then switched to 810 when it became available. Great products; I've been very happy with both. Two weeks ago gave the 810 to another family member and got the Lumia 925.
I love the 925. The screen is great, the phone is light, and the camera amazing. Except for one thing. The first day I had the 925, I noticed that the audio from the earpiece did not no sound right. While on calls, the earpiece had a nasty buzz to it, sounding like a damaged speaker. The audio quality while on speakerphone was perfect, so I could not blame T-Mobile's network. I took the phone back to T-Mobile and they replaced it, no questions asked.
The second 925 sounded great. The earpiece audio was clear, without any buzzing or audio glitches. I was a happy man and everything was great, that is, until yesterday. Now the second 925 developed a similar bad audio quality during calls, but only from the earpiece. The speakerphone sounds fine, and using a bluetooth headset sounds fine also. It is not easy to describe the "badness" of the audio but there is a significant decrease in quality. The audio level seems to be down; with the volume all the way up, you can barely hear the other caller. The audio also has a buzzy, muffled quality to it, almost as if it is coming out of a tin can, and it is not easy to understand the caller. I tried calling my old Lumia 810 and the sound on the 810 is just fine. The trouble seems to be only the earpiece on the 925.
This morning I went back to T-Mobile and they replaced the SIM card. I'm not sure how that would help, but they said it should help if this is a network issue. That was 5 hours ago and the audio quality has not improved at all. I have also performed a full phone reset, then reloaded my previous settings, but the problem persists. At this point it appears to be a hardware issue, but it is very odd that it happened on the first 925 right away, and on the replacement one as well, but after two weeks. Beside the bad earpiece audio the phone works great. The apps work as they should, and the camera is lovely, but if I first need the device to function as a phone.
I am debating if I should try a third Lumia 925. I am a bit reluctant, as I'm afraid that it might break again. At least now it is within the 30 day return period, but what if this issue happens later? I would love to stick with Lumia but these issues with the 925 are worrisome.
Any help or suggestions would be much appreciated.
LumiaJohn
Solved!
Go to Solution.supersape wrote:
I'm having the same problem with my Nokia Lumia 925. Voice in the earpiece audio is otherwise ok but when the other person is saying the S-letter in a word the earpiece audio makes it sound like other person is having a lisp or something. It's really annoying. Same effect comes when other person on the line is inhaling or exhaling. There is this whoosh sound the earpiece is making on S-letters, inhale - exhale sounds and on other low frequency background noices. Phone was like this from my first calls after unboxing. ...
I was thinking how to describe my audio problems with 925 but you did it perfectly! I have exactly the same problems with the S-letters, inhaling, exhaling etc. which makes every voice call more or less annoying experience.
I was thinking to replace the unit but the longer I read this thread, the more I start to think this is "a feature" i.e. bad design and/or poor HW choices. That said, ff this is so common issue, I wonder howcome there wasn't anything about this in those several reviews L925 was subjected to when it was released. Or are the reviewers so busy in testing the 'smart phone' feature that they do not spend time to mundane stuff like voice call quality.
I love the phone otherwise; looks great, has fantastic camera, crisp and cleat display - anyhow, had I known about this, I would have left this to the store.
EDIT:
supersape wrote:
I was making a Skype call from my Lumia 925 yesterday and with Skype the earpiece audio was clear and there was no whooshing disturbing the call. So i guess the speaker is working properly in Lumia 925 but the problem might be in 3G/4G reception (or somewhere else) that disturbs the call quality creating this whooshing in a call.
Havent' tried myself but if this is true, this starts to sound like some voice codec issue. ...which would be strange since one would think that the same speech encoding/decoding implementations are used in all Lumias so why would there be problems specifically in 925.
The good with this option would be that it's probably possible to fix with SW update. Provided that Nokia is aware of the problem or better yet, interested to anything about it. -
My iphone will not receive/send calls or texts when my computer is on? Help?
Recently I've noticed my iphone is having reception problems while I'm in my apartment. I've noticed that when my computer is on, my iphone will not receive or send any texts or calls and has problems connecting to the internet. This is a pretty new development, I've had my phone for years with the same computer and no problems before. Any ideas?
Your computer and iPhone 4 are two seperate elements really using different connections.
i.e your computer will probably use your router to connect to the internet where as your iphone uses your mobile phone carrier to send and receive texts and calls.
So its pretty difficult to judge how they can effect each other. You try resetting your router, but ive never heard of anything like this happening before... -
Timing of NetStream.send() calls
Hello,
we use NetStream.send() calls to remote control clients
subscribed to a live stream. We observed that the live video &
audio stream can get out of sync with the remote calls sent via
NetStream.send(). In bad cases the remote calls where executed 4
seconds before the audio/video signal was shown.
My idea is that NetStream.send() events are created when the
stream "arrives" at the Flash Player, while the Audio/Video Data
might be buffered. Can anybody confirm this implementation detail
of FMS 3.0?
Any help on this is greatly appreaciated,
JuergenThanks for reporting. I can reproduce the bug in house. We will investigate.
Calise -
4FXS-DID configuration problem using BRI to receive/send calls from/to PSTN
Hi to all.
Follow is the partial configuration of my CME.
It has a 4FXS card to use analog phones and faxes, and 4 ISDN Basic Rate interfaces.
On each port of the 4FXS DID card happens the following:
If I enable direct-inward-dial
I can receive calls from PSTN.
Off hooking the analog phone I cannot hear any free line tone. I cannot make any call from that phone.
If I disable direct-inward-dial
I cannot receive directly calls form PSTN.
Callers on PSTN after typing my number, they hear a free tone. Then just typing the extension desired the phone ring.
Off hooking the analog phone I hear a free line tone. I can make normal calls to everyone.
I have half the configuration working.
Where am I wrong?
I really appreciate your help.
Thanks to all.
Giorgio.
CME#sh ver
Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2)
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)
CME uptime is 2 days, 15 hours, 49 minutes
System returned to ROM by power-on
System restarted at 18:54:45 CDT Fri Aug 2 2013
System image file is "flash:/c2800nm-ipvoicek9-mz.124-24.T3.bin"
Cisco 2811 (revision 53.50) with 249856K/12288K bytes of memory.
Processor board ID FTX1120A08L
2 FastEthernet interfaces
4 ISDN Basic Rate interfaces
16 terminal lines
4 Voice FXS interfaces
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
1948656K bytes of USB Flash usbflash0 (Read/Write)
497448K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
CME#sh telephony-service
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
isdn switch-type basic-net3
voice translation-rule 1
rule 1 /^2929091\(..\)/ /\1/
rule 2 /^2929091\(.\)/ /\1/
rule 3 /^02929091\(..\)/ /\1/
rule 4 /^02929091\(.\)/ /\1/
voice translation-rule 2
rule 2 /\(.*\)/ /02929091\1/
voice translation-rule 10
rule 1 /\(^......$\)/ /0\1/ type national national plan isdn isdn
rule 8 /\(^......$\)/ /00\1/ type international international plan isdn isdn
voice translation-profile PSTN-IN
translate calling 10
translate called 1
voice translation-profile PSTN-OUT
translate calling 2
interface BRI0/0/0
description Isdn channels 1 & 2
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/0/1
description Isdn channels 3 & 4
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/1/0
description Isdn channels 5 & 6
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/1/1
description not used
no ip address
shutdown
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
voice-port 0/0/0
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 1 & 2
voice-port 0/0/1
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 3 & 4
voice-port 0/1/0
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 5 & 6
voice-port 0/1/1
description Disponibile
voice-port 0/2/0
input gain 14
connection plar 83
description Interphone 40
station-id name Citofono
station-id number 40
caller-id enable
voice-port 0/2/1
cptone IT
description Ced 35
station-id name CED
station-id number 35
caller-id enable
voice-port 0/2/2
cptone IT
description FAX_1 50
station-id name FAX_1
station-id number 50
caller-id enable
voice-port 0/2/3
cptone IT
description FAX_2 60
station-id name FAX_1
station-id number 60
caller-id enable
dial-peer voice 1001 pots
description **Sends call to PSTN line 1-2**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0
dial-peer voice 2001 pots
description **Receives calls coming from PSTN line 1-2**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/0
dial-peer voice 1003 pots
description **Sends call to PSTN line 3-4**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/1
dial-peer voice 2003 pots
description **Receives calls coming from PSTN line 3-4**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/1
dial-peer voice 1005 pots
description **Sends call to PSTN line 5-6**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/1/0
dial-peer voice 2005 pots
description **Receives calls coming from PSTN line 5-6**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/1/0
dial-peer voice 2035 pots
description **Receives calls coming from PSTN to Extension 35**
answer-address 35
destination-pattern 35$
incoming called-number 35
port 0/2/1
dial-peer voice 2040 pots
description **Receives calls coming from PSTN to Extension 40**
destination-pattern 40
incoming called-number 40
no digit-strip
direct-inward-dial
port 0/2/0
dial-peer voice 2050 pots
description **Receives calls coming from PSTN to Fax 50**
destination-pattern 50
incoming called-number 50
no digit-strip
direct-inward-dial
port 0/2/2
dial-peer voice 2060 pots
description **Receives calls coming from PSTN to Fax 60**
destination-pattern 60
incoming called-number 60
no digit-strip
direct-inward-dial
port 0/2/3
CME#Hi Paolo.
Sorry for the delay. I was on holiday.
I would like to keep separate incoming calls from outgoing calls.
So I decided to keep two dial peers for every BRI interface.
I followed your suggestion on eliminate commands:
"incoming called-number" on FXSs,
various "progress_ind" on BRIs
Also I eliminated "direct-inward-dial" on FXSs,
Today I reconfigured and right tested the dial peers as following:
! Bri Interfaces
dial-peer voice 1001 pots
description **Sends call to PSTN line 1-2**
destination-pattern 0T
port 0/0/0
dial-peer voice 2001 pots
description **Receives calls coming from PSTN line 1-2**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/0
dial-peer voice 1003 pots
description **Sends call to PSTN line 3-4**
destination-pattern 0T
port 0/0/1
dial-peer voice 2003 pots
description **Receives calls coming from PSTN line 3-4**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/1
dial-peer voice 1005 pots
description **Sends call to PSTN line 5-6**
destination-pattern 0T
port 0/1/0
dial-peer voice 2005 pots
description **Receives calls coming from PSTN line 5-6**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/1/0
! FXS interfaces
dial-peer voice 2035 pots
description **Receives calls coming from PSTN to Extension 35**
destination-pattern 35
no digit-strip
port 0/2/1
dial-peer voice 2040 pots
description **Receives calls coming from PSTN to Extension 40**
destination-pattern 40
no digit-strip
port 0/2/0
dial-peer voice 2050 pots
description **Receives calls coming from PSTN to Fax 50**
destination-pattern 50
no digit-strip
port 0/2/2
dial-peer voice 2060 pots
description **Receives calls coming from PSTN to Fax 60**
destination-pattern 60
no digit-strip
port 0/2/3
Thanks a lot Paolo. :-)
Giorgio. -
How Can I send SMS to a Mobile Phone within Oracle
How Can I send SMS to a Mobile Phone within Oracle
It depends on how you are planning to use this feature.
You can set up a mobile using special software in order to interface to it programatically - you can then, via this software interface, use this mobile to send SMS's.
Some service providers support an e-mail interface. Where you send the SMS as a the e-mail content and the mobile number as the recipient's address, or as the subject of the e-mail. However, many service providers have long since stopped providing this type of mail interface due to abuse. Remember that service providers are there to make money - not offer one a service via bulk SMS's can be send.
Some service providers have dial-up servers that works like the old style bulletin board systems (BBS's) of the 90's. You use a telnet like interface, dial into this BBS, and is prompted for mobile number, message to deliver, and even delivery confirmation. You can expect the cost of this phone call to this system, to also cover the cost of sending the SMS.
Some companies provide a software server to interface with one or more service providers - this allows you to fairly easily use this server gateway to send SMS's for you. Such a gateway server will typically support a telnet interface - allowing you to send SMS's via it from inside PL/SQL.
I've used all these options, and I've found the best one to be the last option. It is the easiest to use from a programming perspective. It comes with various management and reporting tools. So you need to do the very minimal in managing this whole SMS interface. -
Sending command to a mobile phone blugged into a pc to send an sms
hi.
i am new to this field, but i heard of "at command" that are helpful to control the mobile phone from a pc. And Java supports this...
this is the case:
1- a mobile with a valid sim card is connected via USB with the pc.
2- an application is running at the pc (java application)
on this application we need to add a module/class that sends command "at command" to our connected mobie with 2 parameters:
1- the phone number
2- the message
MyClass.sendViaMobile("+962795940824", "we want to inform you that u got the help :-)");what i need is how to start this, what do i need to know,and what api's i need to use
regards,
bilal
Message was edited by:
bilal_RDhi mlk
i found the rxtx api, but i got now a question,
how to connect to a usb port?
it seems that it is easy to deal with com port or serial port, but until this moment usb didnt work,
in my case, the mobile is connected via usb port.
this a code that uses the RXTX API, whaere do i need to modify to connect using usb?
*note, the following code is written by someone else, it works via com port...
// SendMessage.java - Sample application.
// This application shows you the basic procedure needed for sending
// an SMS message from your GSM device.
// Include the necessary package.;c:\classpath\smslib.jar;c:\classpath\comm.jar
package examples;
import org.smslib.*;
class SendMessage
public static void main(String[] args)
// Define the CService object. The parameters show the Comm Port used, the Baudrate,
// the Manufacturer and Model strings. Manufacturer and Model strings define which of
// the available AT Handlers will be used.
CService srv = new CService("COM1", 19200, "Nokia", "6630");
System.out.println();
System.out.println("SendMessage(): Send a message.");
System.out.println(" Using " + srv._name + " " + srv._version);
System.out.println();
try
// If the GSM device is PIN protected, enter the PIN here.
// PIN information will be used only when the GSM device reports that it needs
// a PIN in order to continue.
srv.setSimPin("0000");
// Normally, you would want to set the SMSC number to blank. GSM devices
// get the SMSC number information from their SIM card.
srv.setSmscNumber("");
// OK, let connect and see what happens... Exceptions may be thrown here!
srv.connect();
// Lets get info about the GSM device...
System.out.println("Mobile Device Information: ");
System.out.println(" Manufacturer : " + srv.getDeviceInfo().getManufacturer());
System.out.println(" Model : " + srv.getDeviceInfo().getModel());
System.out.println(" Serial No : " + srv.getDeviceInfo().getSerialNo());
System.out.println(" IMSI : " + srv.getDeviceInfo().getImsi());
System.out.println(" S/W Version : " + srv.getDeviceInfo().getSwVersion());
System.out.println(" Battery Level : " + srv.getDeviceInfo().getBatteryLevel() + "%");
System.out.println(" Signal Level : " + srv.getDeviceInfo().getSignalLevel() + "%");
// Lets create a message for dispatch.
// A message needs the recipient's number and the text. Recipient's number should always
// be defined in international format.
IOutgoingMessage msg = new COutgoingMessage("+5550000", "Message from SMSLib for Java.");
// Set the message encoding.
// We can use 7bit, 8bit and Unicode. 7bit should be enough for most cases. Unicode
// is necessary for Far-East countries.
msg.setMessageEncoding(IMessage.MESSAGE_ENCODING_7BIT);
// Do we require a Delivery Status Report?
msg.setStatusReport(true);
// We can also define the validity period.
// Validity period is always defined in hours.
// The following statement sets the validity period to 8 hours.
msg.setValidityPeriod(8);
// Do we require a flash SMS? A flash SMS appears immediately on recipient's phone.
// Sometimes its called a forced SMS. Its kind of rude, so be careful!
// Keep in mind that flash messages are not supported by all handsets.
// msg.setFlashSms(true);
// Some special applications are "listening" for messages on specific ports.
// The following statements set the Source and Destination port.
// They should always be used in pairs!!!
// Source and Destination ports are defined as 16bit ints in the message
// header.
msg.setSourcePort(10000);
msg.setDestinationPort(50000);
// Ok, finished with the message parameters, now send it!
// If we have many messages to send, we could also construct a LinkedList with
// many COutgoingMessage objects and pass it to srv.sendMessage().
srv.sendMessage(msg);
// Disconnect - Don't forget to disconnect!
srv.disconnect();
catch (Exception e)
e.printStackTrace();
System.exit(0);
} -
UCCX Agent with Unified Mobility
Hi all,
We have UCCX 8.5 and CUCM 8.6
Maybe somebody knows.
Can we make next:
Incomnig call to UCCX ---> UCCX Script--->CAD and Agent IP Phone
But we woud like configure Unified Mobility feature and use this feature with UCCX agents phone.
There is moment when agent leavs his workplace, and we want forward incoming calls UCCX to agents mobile phone.
I would be very grateful for the helpful advice.
Perhaps there are other solutionsI have just configured Mobile Connect on a ACD line and tested it and it seems to work fine The Agent phone rings then if not answered the mobile rings, the call is answered. Call is seen on the Agent phone aswell and in CAD and the call gets included in the Agent call stats.
I know it's not supported, but what issues should I expect to see?
Thanks,
Stuart -
CWMS Call to IP Phone via ICT trunk, after answered no audio
CWMS ->CUCM 9.1 -> CUCM 8.6-> CUCM 8 IP Phone.
We have installed the CWMS which has SIP trunk to the CUCM 9.1 for audio to phone. The CUCM 9.1 has an Intercluster Trunk to CUCM 8.6. These two are separate CUCM clusters. The customer bought these systems from different vendors.
When the CWMS do a call back to the CUCM 8 IP Phone, the phone is ringing and can answer. However, on the CWMS meeting page, it shows call back failed and no answer. This is strange as when the CUCM 8 IP phone call to WebEx number the CWMS able to answer and user can join the meeting and works normally.
CWMS ->CUCM 9.1 -> CUCM 8.6-> Voice Gateway -> PSTN
In this scenario, when CWMS call to PSTN, e.g. mobile number, the mobile phone will ring and can answer. But the call drop after that without any audio heard. Same thing happens when the mobile phone call to CWMS DID number. It answered bu no sound, then the call drop.
The reason is connected above to the CUCM 8.6 as the PRI is connected to the voice gateway which is control by CUCM 8.6 via MGCP.
Any idea what's wrong in both scenario?? The weird issue is in both scenarios the call is ringing and after answered, there is no audio. IT seems the CWMS no getting any signal the call has been answered.
The CUCM 9.1 has IP phone registered to it as well and no issues for those IP phone to call WebEX CWMS and have conference.Hi Yong,
What is the version of your CWMS system?
What is the size of the solution? (50, 250, 800, 2000 users)?
Do you have High Availability (HA)?
Keep in mind that for the appropriate setup of CWMS and CUCM integration you need to have at least 2 SIP trunks between CUCM and CWMS. If you have 2000 users system or smaller systems with HA, you may need to create more SIP trunks to fulfill the integration. Please, review the following documentation for configuring CUCM for CWMS deployment: http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_5/Planning_Guide/Planning_Guide/Planning_Guide_chapter_0111.html
If that is verified, and still doesn't work properly, without looking at the SIP negotiation traces for the affected calls, I can't tell you what the issue might be. Hence, if the configuration is verified, you may need to open a call with CUCM TAC to take a look at the SIP exchange between CWMS and IP phone and see if there is an issue with the call setup.
I hope this will help.
-Dejan -
8961 phone having issue receiving and calling mobile phone
Ext XXX is having issue receiving and calling mobile phone. It stated that 'Download Status -Failed' attached is the screenshot of the error in Cisco Unified CM Administration. Any idea where to look into?
If i want to grab the log, should be from Trace & Log Central under CUCM?Phone goes "offline" for about 7-10 minutes at a time (every couple of hours) usually in the afternoon (near ~60% charge).
In an office most of the day with coverage going from 4 bars to 0 bars every few minutes or so. I never miss or drop a call though given the appearance of erattic signal. RSSI usually -83. EvDO and 1x will flicker on and off througout the day. WiFi is on all the time with WPA2. Bluetooth is on without anything connected. I will sometimes get the issue if I wander out of WiFi range (out for lunch)
Reseting the phone will clear the problem. Toggling the Airplane, WiFi, and Bluetooth on/off dont clear the problem.
I have:
- Exchange calendar setup with owa.mailseat.com (I believe they have Exchange 2003)
- Google standard calendar (personal)
- Google standard shared calendar (wifes)
My wife has a Pre as well and to my knowledge has never had the phone offline problem. (I would certainly hear about it if she did )
She has:
- Google standard calendar (personal)
- Google standard shared calendar (mine)
* No exchage setup.
Message Edited by zonyl on 08-29-2009 07:04 AM
Message Edited by zonyl on 08-29-2009 07:06 AM -
How to config unified mobile agent
hello everyone
i lookup some doc. to config the Unified MA
i use IPCC 8.0 (CUCM ICM CVP is 8.0) i config MA but when i call in the ICM return no device targets...but i really config it ....
next is my config:
in CUCM 8.0
i creat a local CTI port :LCP5000F0030 and give it a DN 60030 (my CUCM pim is 5000)
i creat a network CTI port : RCP5000F0030 and give ti a dn 913311173269 (9 is the number for target the number is goto PSTN... and 13311173269 is a mobile phone number)
in ICM
i open the MA in agent desk setting and CTI OS
i creat a device target
name:60030
global address:60030
configuration parameters:/devtype ciscophone /dn 60300
and i creat a lable : 60030.CM_pg (CM_pg is my CUCM routing client)
now i login the agent use agent ID 50000
agent ID :50000
instrument:60030
phone number 913311173269
if i use call by call login ,when i ready and call in the CC the ICM return no device target
if i use nailed connection the agent desktop return possible causes are invalid instrument;media termination problem or other CM issueEnable mobile agent in CTI OS installation setup yes i enable it
Create two CTI port as below,
LCP5000F0030 DN 60030
RCP5000F0030 DN 60031
Enable mobile agent check on ICM agent desk setting.
Add device target or agent targetting rule for the CTI port DN i look at some doc. say only need creat LCP DN
the TEST.nic is a routing client connect to another instance ,and use that instance goto CVP ,so this device target i also creat to that instance
Add static route in CVP ops console 600>cucm ip address
Use the below login method
agent ID :50000
instrument(LCP Port DN):60030
phone number 913311173269
nailed connection
when i login the error
if i use call by call when the call goto agent , the router log viewer show no device target
and i find another problem the RCP ip address is wrong the 10.10.10.33 is right and 10.10.10.37 is another instance icm...i dont know how the CTI port regester to CUCM...
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