Cisco IP Voice Media Streaming App

I just built a new 10.5 cluster. In this cluster I have a publisher and a single subscriber. If I use the CM Group that has only the publisher, or at least the publisher first, the annunciator and software conference bridge register fine. If I change them to the CM Group where the subscriber is listed first then they remain unregistered. I have checked and both servers are running the Cisco IP Voice Media Streaming App service.
This is the same configuration I am using on my 8.6.2 clusters, which are working.

You need to check the trace debug level from Cisco Unified serviceability > Trace configuration and collect it from RTMT after a restart of IPVMSA service
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/10124-cm-trace.html
HTH
Manish

Similar Messages

IP Voice Media Streaming App won't start

I'm using CUCM 7.1.5, but this was occurring in 7.1.3 also. I can start all of my other services except this one. CUCM says it starts the service successfully, but it still says "Not Running." This is the output of the service's log file. At the bottom it says it can't find the device driver. Any thoughts on how to fix this? Thanks.
05/21/2010 07:27:31.552 VoiceStrm|   PITrace: *ERROR* [CRITIAL INFO]: platform install api isDeployedStandalone()= true|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* [CRITIAL INFO]: platform install api prods[0]=CallManager|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* [CRITIAL INFO]: Platform Box Configuration :0 (CCM:0,UNITY:1,CoRes:2,Unknown:3)|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[Cisco Tomcat]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[DBMon]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CallManager]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CmiManager]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CtiManager]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[Cisco TcdSrv]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[Cisco TFTP]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CiscoAXLWebService]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[Cisco Media Streaming App]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CiscoEM]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CiscoWebdialer]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[CiscoIPMA]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.553 VoiceStrm|   PITrace: *ERROR* Important info only, Category[RISSystemAccess]: productid_enable_flag=1, system_enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.554 VoiceStrm|   PITrace: *ERROR* [IMPORTANT INFO] AppCrashCheckBase::AppCrashCheckBase() the RW_Process_Mutex is :/dev/shm/Category_CiscoMediaStreamingApp_Mutex_crash_checker
|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.554 VoiceStrm|   PITrace: *ERROR* [IMPORTANT INFO] AppCrashCheckBase::AppCrashCheckBase() acquire lock successed. : errno = 0; strerr = Success|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.554 VoiceStrm|   PITrace: *ERROR* [IMPORTANT INFO]WriterSideCheck() open existing flag_check_mmf left from last run. : errno = 17; strerr = File exists|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.554 VoiceStrm|   PITrace: *ERROR* [IMPORTANT INFO]ExAppCrashCheck::WriterSideCheck() create/open flag_check_mmf successed!!! : errno = 17; strerr = File exists|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.555 VoiceStrm|   PITrace: *ERROR* {Debug infomation}: look from writer side, writer crashed last time : true|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.555 VoiceStrm|   PITrace: *ERROR* CPICategory::loadXmlFile() Category: [Cisco Media Streaming App] xmlfile: [/usr/local/cm/conf/MediaAppPerfmon.xml] |<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.558 VoiceStrm|   PITrace: *ERROR* CPIClass::createMMF() MMF file : [/Category_CiscoMediaStreamingApp_PIClass_MEDIAAPP_GLOBAL_OBJECT_MDB] is started.(no error)|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.562 VoiceStrm|   PITrace: *ERROR* ExObjectUpdateTracker::ExObjectUpdateTracker::_synchronizeMAP() m_nIndex :0 m_nRoundCount : 0 m_nTick: 0|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.562 VoiceStrm|   PITrace: *ERROR* [SIGNIFICANT INFO]Category Name: Cisco Media Streaming App file: /usr/local/cm/conf/MediaAppPerfmon.xml enbled_flag: 1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.562 VoiceStrm|   PITrace: *ERROR* [SIGNIFICANT INFO]Add ITraceObj for Category Name: Cisco Media Streaming App, tid:8971l|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.562 VoiceStrm|   PITrace: *ERROR* [SIGNIFICANT]::statsInit() enable_flag=1|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.563 VoiceStrm|   CIpVMSMgr::Init *ERROR* Failed Open device driver.|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>
05/21/2010 07:27:31.564 VoiceStrm|kIPVMSDeviceDriverNotFound - Cisco IP voice media streaming driver not found. App ID:Cisco IP Voice Media Streaming App Cluster ID:StandAloneCluster Node ID:cucmpub|<CLID::StandAloneCluster><NID::10.0.1.13><CT::Alarm><LVL::All><MASK::ffff>
05/21/2010 07:27:31.564 VoiceStrm|   CIpVMSMgr::MgrThreadProc *ERROR* Failure in Init, tell ipvmsd to quit|<CLID::StandAloneCluster><NID::10.0.1.13><LVL::Error><MASK::0002>

Rob,
I used those directions and found some other ones that worked better for me and it solved my issue! Here are the steps for the archives:
Download the CentOS 5.x CD
Restart the VM with that CD
At the "boot:" prompt type "linux rescue"
Then after it finishes booting up type "chroot /mnt/sysimage"
That will try to mount the CUCM file system to the root directory.
Then navigate to /lib/modules/cm_ipvms/ and identify directory xyz.i686
Create a new directory called xyz.athlon and copy all contents from xyz.i686 to the new directory xyz.athlon
Do the same for all directories ending .i686, so there is a .athlon equivalent for each i686 folder.
Restart CUCM vmware and boot normally
Then try to start the service.

IP Voice Media Streaming App

Hello everybody,
If you have a hardware conference bridge, does it make any problem to turn off the "IP Voice Media Streaming App" service? If I am not wrong, if you turn this service off, software conference bridge stops working.
Thanks,
Fatih.

Yes, because also MTP and ANN depend on it. Just assign the SW CFB to a MRG and do not use it anywhere.
Sent from Cisco Technical Support iPad App

Cisco 2911 Voice Gateway SIP PSTN Calls Fail

Hello All,
    I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smith

i have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=90555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x30CF41D4, Call Info(
   Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Aysar Mohamed
   Account Number=2217156, Final Destination Flag=TRUE,
   Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
   Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
   Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
   Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
   Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                    ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                    tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                    tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
   Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP

Media Streaming Problem

My z10 is not working well. The media streaming between my Sante Fe and z10 doesn't appear to be working. I downloaded the latest OS and I believe that it's since then that i've had the problem. The phone is paired with the car fine. The phone works through the car speaker system..... but my map app and music won't play (the car says "no media"). I've been to the dealership because the car is under warranty and the service manager was able to connect his android fine (played music). I then went to my mobile service provider and the service rep came to my car and he couldn't get it working either. He suggested this site. Anyone know how to correct this? Thanks

Hi Matthew
Welcome to Apple Discussions
Those sites are driven by Adobe Flash.
Make sure you are using the most recent plug-in, which is version 10.0.12.036. You can determine your machine's version here. If you have an earlier version update the plug-in. You can do so here.
Whether or not you have the most recent version, go to your Applications>Utilities folder and open Disk Utility. Select "repair permissions". You'll probably see Flash items being repaired.
When complete, close DU, then restart Safari and try some Flash videos.
Post back
Message was edited by: Hawaiian_Starman

RTP Media Stream with "Third Party SIP Device" always through CUCM

Hello,
i have i quite strage problem on one of my customers locations:
we have a cucm 7.1.5(SU4) with the cucm in the datacenter. And we have a small location(branch office) which has a small wan connection to the datacenter (1MBit/s).
In this location we have several Kirk (Polycom) Dect phones which register as "Third Party SIP Device - Basic" on the CUCM.
(The problem is the same if i use the X-Lite SIP Client instead)
When this SIP Phones or the X-Lite Client dials a internal Number of the same location the RTP Media Streams goes directly from the SIP Client to the phone. But if they dial an external number the RTP Stream goes from the SIP Client via the wan connection to the CUCM and back via the wan connection to the 2901 H.323 Gateway (on the same location).
and of course if i now start a big download or upload i'm no longe able to complete the phone call because we have no QOS on the wan connection, because we don't want to make calls over this connection.
When i look at the Sniffer files with Wireshark i see that the CUCM sends his own ip adresse in the SDP Header for the RTP Stream to the SIP Client. And this of course is wrong because the RTP Stream should always reside on the branch office.
i tested this in my lab (CUCM 8.5) and it is the same. i used the "Standard SIP Profile" and a Basic Third Party SIP Device"
The SCCP Phones on the same location which are configured with the same Region, Location, Device Pools, Media Resources and which use the same Gateway for external calls do not have this problem.
In the Gateway configuration "MTP Required" is not activated and i tested it in my lab with some Cisco SIP Phones (9971) and they are also not affacted with this problem.
any ideas?

do you have a SIP trunk to the external devices with MTP required checked ?

Injecting DTMF event in the midst of RTP media streaming?

Hi all, I am attempting to implement a mechanism that allows me to inject DTMF RTP events in an RTP media stream. This is useful in telephony applications where the users are prompted to enter digits while being served by automated voice services such as answerinng services or tele-bankings etc. So basically my approach is to extend Sun's RTP packetizer and intercept outgoing packets when necessary and replace them with the appropriate DTMF RTP (RFC 2833) packets. Sounds simple enough. So I derived my custom packetizer from com.sun.media.codec.audio.ulaw.Packetizer, and over-ridden method process(). Normally my packetizer's method process() simply delegates the functionality to its parent class's process() method. When a DTMF digit is required I'd take the outputbuffer generated by the parent's process() and modify its header and payload to turn it into a DTMF RTP packet. The problem is that the class RTPHeader is so limited, there is no way to set the payload type, sequence number, ... And the documentation is precious few and far in between. If someone has solved this issue, or if you have a reference to some documentation that describes the inner working of JMF's codec chaining, I would appreciate some pointers. What I need to know is:
- What JMF does with the Buffer objects between stages (from one codec to the next)?
- The data portion of Buffer is an Object of arbitray class, what on earth does JMF do with that?
- How does JMF take a Buffer object and turn it into a UDP packet?
- How do I go about creating an RTP header and fill it with the information (payload type, timestamp, sequence number, ...) that I want?
Thanks in advance,

I don't know what the heck RTPHeader represents but it sure doesn't seem to conform to RFC1889. And also, it seems the UDP RTP packets are formed somewhere after the packetizer and before the RTP connector, someone oughta jot all this down in a book. So anyway, using my own RTPConnector implementation I have some control over the outgoing/incoming RTP packets, to access and manipulate the real RTP header (not RTPHeader), I devised a new class that takes the RTP packet buffer and provides an API to examine and manipulate some RTP info directly on the buffer without doing any unecessary data copy (you guys can extend it to do more as per your requirement):
package com.mycompany.media;
import javax.media.rtp.RTPConnector;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.io.IOException;
import javax.media.protocol.ContentDescriptor;
import javax.media.protocol.SourceTransferHandler;
import java.net.DatagramPacket;
import javax.media.protocol.PushSourceStream;
import javax.media.rtp.OutputDataStream;
import java.net.SocketException;
import mitel.utilities.MiQueue;
import java.nio.ByteBuffer;
import java.util.LinkedList;
class MiRtpHeader
     byte[] data;
     int myoffset;
     public MiRtpHeader(byte[] buf, int offset, int len) throws ArrayIndexOutOfBoundsException
          if(len < 12)
               throw new ArrayIndexOutOfBoundsException("Buffer not large enough to contain a basic RTP header");
          data = buf;
          myoffset = offset;
     public boolean getExtension()
          return (0 != (data[myoffset] & 0x10));
     public void setExtension(boolean state)
          if(state)
               data[myoffset] = (byte)(data[myoffset] | 0x10);
          else
               data[myoffset] = (byte)(data[myoffset] & 0xef);
     public boolean getMarker()
          return (0 != (data[myoffset + 1] & 0x80));
     public void setMarker(boolean state)
          if(state)
               data[myoffset + 1] = (byte)(data[myoffset + 1] | 0x80);
          else
               data[myoffset + 1] = (byte)(data[myoffset + 1] & 0x7f);
     public int getTs()
          ByteBuffer tsBuf = ByteBuffer.wrap(data, myoffset + 4, 4);
          return tsBuf.getInt();
     public void setTs(int ts)
          ByteBuffer tsBuf = ByteBuffer.wrap(data, myoffset + 4, 4);
          tsBuf.putInt(ts);
     public int getPayloadType()
          ByteBuffer ptBuf = ByteBuffer.wrap(data, myoffset + 1, 1);
          return ptBuf.get();
}

Looking For Field Trial Customers to Test Audio & Video Using Cisco's Intercompany Media Engine (IME)

IntelePeer, Inc. (www.intelepeer.com) is a leading provider of on-demand cloud-based IP communications services, and is currently looking for a limited number of existing users of Cisco's CUBE, CUCM, or CME call processing systems to work with IntelePeer in testing our new hosted audio and video federation applications, including interworking with CUCM systems enabled with Cisco's Intercompany Media Engine (IME) platform. The specific supported Cisco call processing and audio/video specs mentioned above are as follows:
Call Processing:
Cisco Unified Border Element (CUBE) - preferred
Cisco Unified Communications Manager (CUCM) 6.X or higher
Cisco Unified CallManager Express (CUCME) 7.1 or higher
Audio Codecs Supported:
G.722
G.711
Video Codecs Supported:
H.264
H.263
IntelePeer will provide those willing to participate with a no cost SIP trunk, as well as free IME based voice and video usage during the trial. The specific details of the requested test case scenarios, as well as timeframes for testing, will be outlined in detail upon being approved for participation in the field trial program.
Those willing to participate are urged to notify us immediately, via response to this form.
Thank you in advance for your time and consideration. We very much look forward to working with you during this field trial.
Sincerely,
IntelePeer, Inc.

Hello there and yes I would be interested in learning more about your testing program. How can I share a little bit about my environment and we can see I'd we would be a good candidate. Please feel free to reach me at [email protected] or [email protected]
Sent from Cisco Technical Support iPhone App

Looking For Field Trial Customers to Test Audio & Video Using Cisco's Intercompany Media Engine

IntelePeer, Inc. (www.intelepeer.com) is a leading provider of on-demand cloud-based IP communications services, and is currently looking for a limited number of existing users of Cisco's CUBE, CUCM, or CME call processing systems to work with IntelePeer in testing our new hosted audio and video federation applications, including interworking with CUCM systems enabled with Cisco's Intercompany Media Engine (IME) platform. The specific supported Cisco call processing and audio/video specs mentioned above are as follows:
Call Processing:
Cisco Unified Border Element (CUBE) - preferred
Cisco Unified Communications Manager (CUCM) 6.X or higher
Cisco Unified CallManager Express (CUCME) 7.1 or higher
Audio Codecs Supported:
G.722
G.711
Video Codecs Supported:
H.264
H.263
IntelePeer will provide those willing to participate with a no cost SIP trunk, as well as free IME based voice and video usage during the trial. The specific details of the requested test case scenarios, as well as timeframes for testing, will be outlined in detail upon being approved for participation in the field trial program.
Those willing to participate are urged to notify us immediately, via response to this form.
Thank you in advance for your time and consideration. We very much look forward to working with you during this field trial.
Sincerely,
IntelePeer, Inc.

Hi, Stephen.
Yes, we would be more than happy to speak with you about becoming a test participant for IME. If you could please email me with your contact information, I will get in touch to set up a meeting to discuss.
Please advise.
Thanks much.
Regards,
Scott
Scott Davidson
Product Manager
http://www.voex.com/index.php
2300 15th Street
Suite 100
Denver, CO 80202
303.222-0090 Office
303-818-9183 Mobile
[email protected]

Airport Extreme, 5Ghz connectivity problems and poor media streaming

Hi,
I purchased an extreme from the apple store a little over a week ago, and before I send the device back I would like to solicit recommendations from this forum.
Some information about the setup:
- Single airport extreme (supplying wifi services for my non-wifi DSL router and a connected USB disk)
- Airport Extreme Firmware: 7.4.2
- iTunes: 9.0
- OS: Snow Leopard (10.6.1), Leopard (10.5.8), Window Vista.
- Harddrive: 2TB lacie, USB2.0 connection only
- 2.4 and 5 Ghz broadcasting mode
- The device is broadcasting on all supported protocols.
- Multiple clients (various: apple, microsoft, fujitsu)
- Small space, no major concrete/brick/stone walls, signal strength is 5 bars of goodness
- Country: Spain (I know there were 5Ghz restrictions here a while ago, but the device broadcasts on 5Ghz and is therefore not crippled).
If anything else could be of use, let me know and I'll update this list.
And now the problems:
1) 5Ghz connections fail
When connecting over the 5Ghz freq my connectivity to the extreme and the services it is offering (internet, disk sharing) is intermittent, at best. Basically, it has not worked reliably since I took it out of the box. This is equal for both of my macbooks and a fujitsu laptop (all with n support).
I've tried switching channels to avoid congestion - I don't live in a wifi-heavy area, and there are no other likely sources of interference (e.g. microwaves, cordless phones).
My connections to mounted disks come and go, I have to re-authenticate, and so on.
A general scan of the environment with KisMac does not show any major interference from other wifi networks (only 2 others in range, one is wimax).
I've tried to do some wifi analysis to see if I can spot any frame/packet related issues. Some rudimentary analysis indicates that when the connection stops working on the 5Ghz channel my laptop continually sends request-to-send packets to the extreme, but does not receive a clear-to-send. This occurs after a-few-seconds to a-couple-of-minutes of connectivity - the connection initially 'works' (i.e. wifi packets are sent/received correctly and I can access the internet).
2) Data access and media streaming from the connected USB drive is very poor
I've mostly had problems with itunes. I thought this might be related to the size of my library, but this shouldn't effect single track playing. Even enabling large buffering in itunes results in intermittent pauses and long track loading.
Note: this occurs over b/g connections; as stated already, n connections don't work. iTunes simply goes into a coma for while and then eventually fails. See above. Also, if I just stream from my macbook (i.e. HD connected via USB), it works flawlessly.
The extreme was running 7.4.2 - updated on first connection. It's now running 7.4.1, as I read on the forums that there were some perceived problems with 7.4.2 (although these perceptions may not be accurate.
Concl(f)usion:
I believe I've ruled out the following:
- Distance
- Interference
- Laptop related issues (the same problem occurs with 2 macbooks and a fujitsu laptop).
- Latest firmware
- Software (apps, OS, etc)
Which leaves me to ponder two possibilities:
a) The extreme is damaged.
b) The extreme is not a good product (surely this can't be so?!)
There is, of course, a c-possibility ... that the collective experience of the online community might be able to offer me an alternative path?
I would be very grateful if anyone could offer any suggestions, tips, tricks, spells or simply a hug before I contact apple to begin the process of explaining the issues and seeking some care.
Many thanks in advance,
Dave.

I have had the same problem. Really frusrtating, expecially since the reason I bought the airport was because of all the wireless congestion in my neigborhood with only 11 channels on the 2.4 GHz spectrum.
Bought my airport dual band about a week ago. It had the firware 7.4.2. After extensive searching, I found other with similar issues. Apparently the last stable firmware may be 7.3.8. I am on 7.4.0 and not having any issues any longer (only testing one day now).
This may solve your issue. Login to your airport and choose manual setup. on the summary page, find 'Version' 7.x" and click on the word 'version'. This will allow you to rollback your firmware. I went with 7.4. Depending on what options you need with airdisk, Time Capsule, rolling back may remove those options, so experiment. You can always go back to 7.4.1 or 7.4.2 from within the interface.
Hope this helps.

EA-4500 Media Streaming Issues?!

Hi everybody,
I bought EA-4500 3 weeks ago, apart from its fantastic performance in all fields ,the media streaming server is glitchy.
I google about it and found that the built-in media streaming server is twonky with the limitation of 8000 files only.
when I had more than 8000 media file on my hard drive some of them( Randomly!!) were not shown on none of my DLNA certified devices in my network.
I contacted the Support group and they told me that "it is true!!!"
Has anybody any experience in the field ?
It is good to notice that 1- My Firmware is uptodate .
2- My ext-HDD attached to ea-4500 is compatible with it, as I have no problem with its FTP server.
Twonky Config page of your router is: http :// your router ip:9999/config
I hope it would be solved soon by some firmware tweak or sth.
Thanks in Advance.

Your firmware is uptodate. Are you using the Cisco Connect Cloud software? or the classic user interface?

Sony Media Remote App

Hi
I have a Sony BDP s380 blu-ray player, wired (via devolo dLAN 200 AVmini powerline adaptors) to a v2 hub.
I downloaded the Sony media remote app to my iphone 4. This should allow me to control the blu-ray player from the app, but the app/iphone will not "see" the blu-ray player, so it does not work. From checking the home homepage, I can see that the iphone and blu-ray player are both connected to the network.
From some googling, there is a suggestion that it is a UPnP issue, so I was wondering whether setting up port forwarding would help (although I'm not sure this is right, given that it is not about access from outside the network).
Can someone tell me what I can do to get the devices talking to each other?
I have checked that the blu-ray player is compatible with the app - and am driving myself mad trying to get this thing to work, so and suggestions gratefully received.
Thanks

Thanks Viv
No joy (yet). Sony told me I needed to download "third party software" without being any more specific of helpful, then said I needed to speak to Apple. Apple (of course) said it was to do with Sony, but that I might want to check with BT. BT were helpful but couldn't ultimately do anything more than send me back to Sony. A much more helpful Sony person at least appeared to understand the issue and said they'd escalate it and get back in a couple of days, but I've heard nothing back from them.
I did find this site:
http://www.roomieremote.com/ip-compatibility/
which at least seems suggest a port that is used but I don't know if that is the only one, and I haven't been able to make it work by forwarding that port. If you have any more luck, I'd be keen to know.
It's hardly the biggest deal that it doesn't work (the player itself, and streaming of iplayer etc is great) but it is slightly driving me mad that there must surely be an answer out there...
Bob

Does media streaming server work with IPAD client

What if I access video delivered through media streaming server on IPAD client. Does it supports?

Hey hi Petro, first of all sorry for delayed response.
Now coming to your query - Mac OS per se allows flash , i mean if you Mac machine flash would still work as its officially supported there as standalone, in authoring and as Safari Plug-in. Its only iPhone and iPad which don't support flash.
Now how they block it - in order to play any flash content - you need to have FP Plug-in installed - and that is what iPad/iPhone won't allow you to install.
And if I am not mistaken they do block Flash Apps on thier store too ( though i am not completely sure about this as I am not that expert in this area but high chances are there that i might be true , atleast till someone wiser corrects me )
I hope my answer is helpful and gives little insight to what you asked

I tried using the Voice Memos iPhone app and ended up with a red banner on top of the screen that flashes "recording 00.00" but I can't locate what I thought I was recording. can you help me?

I tried using the Voice Memos iPhone app and ended up with a red banner on top of the screen that flashes "recording 00.00" but I can't locate what I thought I was recording. can you help me?
jacknil

I found an answer to this on another page (I was having the same problem) This worked for me:
Double-click the Home button.
Swipe left or right until you have located the app you wish to close.
Swipe the app up to close it.
more here: https://discussions.apple.com/thread/5596831

I tried to download a Cisco webex application. The app did not download and I also can't open the app. The problem is that I can't get the webex screen to close, it is stuck open and I can't get back to the app home page!

I tried to download a Cisco website application. The app did not download and it is stuck on the window where it has the option of open.....it will not open. The problem is that I cannot get the window off the screen and I can't get back to the App Store home page! Help! Thanks

Try this.
Reset the iPad by holding down on the sleep and home buttons at the same time for about 10-15 seconds until the Apple Logo appears - ignore the red slider if it appears on the screen - let go of the buttons. Let the iPad start up.

Cisco IP Voice Media Streaming App

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