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,
Juergen
Thanks for reporting. I can reproduce the bug in house. We will investigate.
Calise
Similar Messages
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NetStream.send not working in Flash Player 11.2 Beta with Cirrus, Please confirm if it is a bug
Title
NetStream.send not working in Flash Player 11.2 Beta with Cirrus, Please confirm if it is a bug or feature
Description
Problem Description:
NetStream.send can not send data to peerstreams when using with cirrus. Conflict with documents.
Sorry for tag the build as 11.0.1.3 while the bug is actually on 11.2 beta since the bug report system didn't have 11.2 beta yet.
If you are not responsible for 11.2 beta bug fix, please help a hand to handle this bug to 11.2 team.
This bug is "killing" to your application, so we really appreciate your help. Thanks.
==Publisher==
nc.connect("rtmfp://");
var ns:NetStream = new NetStream(nc, NetStream.DIRECT_CONNECTIONS);
ns.publish("sendtest");
...//after connection success.
ns.send("clientfunction", "ok"); // this line cannot reach subscribers. even if subscribers have client object correctly.
==Subscriber==
nc.connect("rtmfp://");
var ns:NetStream = new NetStream(nc, cirrusid);
var client:Object = new Object();
client.clientfunction = clientfunction; // target function
ns.client = client;
ns.play("sendtest");
Steps to Reproduce:
1. compile the code in the attachment to SendTestExample.swf (not be able to paste it here)
2. run it under flash player 11.2.202.19 beta
3. run it under flash player 11
Actual Result:
HeartBeat is:
Start HeartBeat:
send hello
send hello
send hello
which means NetStream.send was not able to call "clientfunction" as expected.
Expected Result:
Start HeartBeat:
send hello
in client function: hello
send hello
in client function: hello
send hello
in client function: hello
which can call into the clientfunction as flash player 11 did.
Any Workarounds:
I can not find it out since it's an api level bug. But this can be very important for lots of applications which rely on send to do rpc.
Test Configuration
IE8, Firefox under Windows 7
Also have problem under Windows XP (but not well tested on this platform)
App Language(s)
ALL
OS Language(s)
ALL
Platform(s)
Windows 7
Browser(s)
Internet Explorer 8.0
==Attachment==
package {
import flash.display.Sprite;
import flash.events.Event;
import flash.events.NetStatusEvent;
import flash.events.TimerEvent;
import flash.media.Video;
import flash.net.NetConnection;
import flash.net.NetStream;
import flash.text.TextField;
import flash.utils.Timer;
import flash.utils.setTimeout;
public class SendTestExample extends Sprite
public static var statusArea:TextField;
var ncServer:NetConnection = new NetConnection();
var nsServer:NetStream;
var ncClient:NetConnection = new NetConnection();
var nsClient:NetStream;
var timer:Timer = new Timer(1000);
public function SendTestExample() {
ncServer.addEventListener("netStatus", onNCStatusServer);
ncServer.connect("rtmfp://p2p.rtmfp.net","99f72ccbed0948d7589dc38a-3ce1b2616680");
statusArea = new TextField();
status("status");
statusArea.x = 0;
statusArea.y = 0;
statusArea.border = true;
statusArea.width = 200;
statusArea.height = 350;
addChild(statusArea);
function onNCStatusServer(event:NetStatusEvent):void {
status("Step 1:");
status("server: " + event.info.code);
status("id: " + ncServer.nearID);
switch (event.info.code) {
case "NetConnection.Connect.Success":
nsServer = new NetStream(ncServer, NetStream.DIRECT_CONNECTIONS);
nsServer.addEventListener(NetStatusEvent.NET_STATUS, onNSStatusServer);
nsServer.publish("sendtest");
ncServer.removeEventListener(NetStatusEvent.NET_STATUS, onNCStatusServer);
ncClient.connect("rtmfp://p2p.rtmfp.net","99f72ccbed0948d7589dc38a-3ce1b2616680");
ncClient.addEventListener("netStatus", onNCStatusClient);
case "NetStream.Publish.BadName":
//status("Please check the name of the publishing stream" );
break;
function onNCStatusClient(event:NetStatusEvent):void {
status("Step 2:");
status("client: " + event.info.code);
status("id: " + ncClient.nearID);
switch (event.info.code) {
case "NetConnection.Connect.Success":
nsClient = new NetStream(ncClient, ncServer.nearID);
var c:Object = new Object();
c["clientfunction"] = clientfunction;
nsClient.client = c;
nsClient.play("sendtest");
ncClient.removeEventListener(NetStatusEvent.NET_STATUS, onNCStatusClient);
//setTimeout(sendHello, 5000);
case "NetStream.Publish.BadName":
//status("Please check the name of the publishing stream" );
break;
protected function onNSStatusServer(event:NetStatusEvent):void {
status("nsserver: " + event.info.code);
if (event.info.code == "NetStream.Play.Start") {
status("Start HeartBeat:");
this.timer.addEventListener(TimerEvent.TIMER, function (e:Event):void {
sendHello();
this.timer.start();
protected function sendHello():void {
status("send hello");
nsServer.send("clientfunction", "hello");
protected function status(msg:String):void
statusArea.appendText(msg + "\n");
trace("ScriptDebug: " + msg);
protected function clientfunction(event:Object):void {
status("in client function: " + event);Thanks for reporting. I can reproduce the bug in house. We will investigate.
Calise -
Netstream.send only text messages or any type?
In Netstream.send() documentation it says: " ... arguments
— Optional arguments that can be of any type."
In the documentation to the Netstream class it says "You can
also use NetStream objects to send text messages to all subscribed
clients (see the NetStream.send() method)."
I tried to send a Rectangle with Netstream.send:
Code on Sender:
var rect = new Rectangle(1,2,5,5);
nc.call("serverShowRect",null, rect);
Code on Flash Media Server: (FMS 2.0.4)
STREAM_NAME = "myStream";
Client.prototype.serverShowRect= function(msg)
myStream = Stream.get(STREAM_NAME);
myStream.send("@setDataFrame","onRectangle",msg);
Code on Client:
function onRectangle(infoObject:Object):void {
trace ("Rectangle received.");
trace ("tracing properties:");
var key:String;
for (key in infoObject) {
trace(key + ": " + infoObject[key]);
try {
var rect:Rectangle = infoObject["rectangle"] as Rectangle;
trace("Rect as rect:" +rect);
catch (error:Error){
trace(error.message);
The function onRectangle() ist called ok. The infoObject's
properties can be shown by trace. But I cannot convert infoObject
into an Rectangle. var rect will be either null or I get an
type-conversion error.
Does anyone know how to send other data than Strings using
Netstream.send()? Or can one actually only send Strings using FMS2You should receive an email receipt with every purchase that you make in iTunes on the computer or in iTunes, the App Store or the iBookstore on an iOS device. I always get a receipt via email. You are not getting them?
If you are asking if Apple will send you a detailed purchase history, the answer to that is no since you can access that information in your account in iTunes on your computer. Click on Account under the Quick links on the right side of iTunes, enter your password at the prompt, and then click on "See all" to the right of Purchase History in the next window. -
How can I restrict my iPhone 4S to only receive or send calls or send receive texts from specific contacts and not all contacts in the phone.
Contact the carrier and see if they offer such a service.
This is not a feature of the iPhone and no App could restrict calls/sms. -
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.
SteveThanks 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 -
Incoming Phone Call DND (Do Not Disturb) or ability to send call to voicema
Does anyone know a good way to put a DND (do not disturb) on incoming phone calls? The idea is to send calls directly to voice mail while still leaving the phone on to receive email & text messages. While at work or in meetings, it would be nice to not be bothered by calls especially from specific people. Ideally it would be really nice to one or all of the following:
-Function to send all calls to voice mail without ringing
-An option when an incoming call is coming in to send the person to voicemail with the choice of a prerecorded message such as "In a meeting" or "At work" etc.
-Send specific people to voice mail without ringing, possibly with a prechosen message.The only option for the iPhone not ringing when a call is received and the same when a text message is received while in a meeting is by turning the silent switch on located above the volume buttons.
-
Username and password for SOAP sender call
Hello,
does anyone know how to provide the username and password to a SOAP sender call, e.g. XI receives the ws via soap and needs to know which userid and password to check. When you use a SOAP client they use basic authentication which sends the request first to XI and XI send a request back for password. This would work for an online app but not for ws from machine to machine. I read some docu about query strings but no where it has an example what to put either on the request URL, the adapter or the SOAP envelope. SOAP 1.1 seems to have left that open and IBM has an example using SOAP Header which did not work with XI.
Thanks
StefanHi,
do you use the javax.xml.rpc.Call class? Because then
you can supply username and password to the call via
the addParameter method. I think I did that with XI 3.0
and it worked. If you need more information please consult
the javax.xml.rpc.Call javadoc.
Best regards,
Hermann -
Netstream.send() privacy settings
I'm using Netstream.send() to transfer a file between two users.
A previous discussion can be found here:
http://forums.adobe.com/thread/890122?tstart=0
It works well except of flash settings issue.
When trying to access a camera or mic the Flash ask the user for permissions automatically.
When trying to send a file, the Flash player doesn't show the privacy popup.
If the user doesn't set his privacy to Allow, the file transfer fails.
There are two issues:
1. The privacy settings popup isn't shown automatically.
2. The privacy popup ask permissions to access the camera and mic and doesn't say anything about data transfer.
Am I missing something or are the above real issues?
Thanksthe P2P access dialog will only show when using RTMFP Groups (NetGroup or a group NetStream). if you're just doing NetStream.send() on a 1-to-1 (NetStream.DIRECT_CONNECTIONS and new NetStream(netConnection, peerID)) P2P connection, then there is no P2P access dialog.
if you're doing groups and the P2P access dialog isn't showing up, there are a few possible causes:
1) for some reason you at one point selected "Deny" *and* checked "Remember"; in that case, Flash Player won't present the dialog and will behave as though you clicked "Deny"
2) Flash Player's window or region in the browser page is too small for the P2P access dialog. if the P2P access dialog can't be presented because the window is too small, it will behave as though you clicked "Deny"
3) there is a known bug in Flash Player when using wmode=direct (which you would use for Stage Video) where, when you set wmode=direct, the P2P access dialog won't be displayed, and the behavior will be as though you clicked "Deny". this issue will be fixed in a future release of Flash Player. to work around this, set wmode to a different value. -
Hi
I used NetStream.send("
Sender ", " parm " ) function to send text .How to define
the function
Sender in client file using AS3.Sends a message from the server to all subscribing clients is
the short answer.
The long, and awesome, answer can be found here:
http://livedocs.adobe.com/flashmediaserver/3.0/hpdocs/help.html?content=Book_Part_33_cs_as d_1.html -
NetStream.send() with datareliable=false
Does NetStream.send() and setting NetStream.datareliable = false make the flash player use UDP? Or does it just use TCP for 1 second and then stop?
Thanksthe dataReliable flag doesn't control whether you're using TCP or UDP. if you're using RTMP, you're using TCP because that's what RTMP goes over. if you're using RTMFP, you're using UDP because that's what RTMFP goes over.
if you're using RTMP, the videoReliable, audioReliable, and dataReliable flags have no effect because TCP is fully reliable all the time.
those flags only have meaning for RTMFP connections. when video/audio/dataReliable is set to true, RTMFP sends the corresponding message with full reliability, retransmitting it if necessary. if a reliable flag is false, RTMFP will not retransmit the corresponding message, and the message will only remain in the transmission queue for 1 second (so depending on network congestion/throughput, the message might not be transmitted at all).
messages of different types are sent with different priority in RTMFP. audio is highest priority, then NetStream.send() (data), then video. all RTMFP Groups modes (multicast, posting, directed routing, object replication) are lower priority than video. priority matters when bandwidth is tight. -
CUC 8.0.2 AA sends calls to operator straight to VM
Auto-attendant sends calls to the designated operator's number (hitting 0 or dialing the individual's extension) straight to VM. I temporarily changed the operator to my extension (which works dialing through AA as did every other extension I checked) and it worked. The user was originally a local account on CUC. I renamed this account and changed the extension to a nonexistent number, and then imported the operator's account from CUCM (like the majority of the accounts are setup). The symptoms remained the same as if the problem is tied to that extension. Any thoughts/suggestions/next steps?
Hi William,
Can you have a look at this page under User>Edit User>and check the Transfer Calls to setting
Unity Directory > Edit Transfer Rule
Edit Transfer Rule (Standard)
Search Users Edit User Basics (buncle) Transfer Rules Edit Transfer Rule (Standard)
User
Search Users
New User
Save Transfer Rule
Transfer Rules
Edit
User Basics
Password Settings
Change Password
Roles
Message Waiting Indicators
Transfer Rules
Message Settings
Caller Input
Mailbox
Phone Menu
Playback Message Settings
Send Message Settings
Message Actions
Greetings
Notification Devices
Alternate Extensions
Alternate Names
Private Distribution Lists Launches CPCA
External Service Accounts
SMTP Proxy Addresses
Refresh
Help
Contents
This Page
Save
Rule Name
Status
Disabled
Enabled With No End Date and Time
Enabled Until:
Date January February March April May June July August September October November December 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 2010 2011 2012 2013 2014
Time 01 02 03 04 05 06 07 08 09 10 11 12 00 05 10 15 20 25 30 35 40 45 50 55 AM PM
Transfer Action
Transfer Calls To:
Greeting
Extension
Transfer Type
Release to Switch Supervise Transfer
Rings to Wait For
Play the "Wait While I Transfer Your Call" Prompt
If Extension is Busy:
Send Callers to Voice Mail Put Callers on Hold Without Asking Ask Callers to Hold
Call Screening
Tell Me When the Call Is Connected
Tell Me Who the Call Is For
Ask Me If I Want to Take the Call
Ask for Caller's Name
Cheers!Rob -
FaceTime connects ok and rings out when sending call but no ringing sound when receiving incoming calls
Using FaceTime http://support.apple.com/kb/ht4319
Troubleshooting FaceTime http://support.apple.com/kb/TS3367
The Complete Guide to FaceTime + iMessage: Setup, Use, and Troubleshooting
http://tinyurl.com/a7odey8
Troubleshooting FaceTime and iMessage activation
http://support.apple.com/kb/TS4268
Using FaceTime and iMessage behind a firewall
http://support.apple.com/kb/HT4245
iOS: About Messages
http://support.apple.com/kb/HT3529
Set up iMessage
http://www.apple.com/ca/ios/messages/
Troubleshooting Messages
http://support.apple.com/kb/TS2755
Setting Up Multiple iOS Devices for iMessage and Facetime
http://macmost.com/setting-up-multiple-ios-devices-for-messages-and-facetime.htm l
FaceTime and iMessage not accepting Apple ID password
http://www.ilounge.com/index.php/articles/comments/facetime-and-imessage-not-acc epting-apple-id-password/
Unable to use FaceTime and iMessage with my apple ID
https://discussions.apple.com/thread/4649373?tstart=90
My Facetime Doesn't Ring
https://discussions.apple.com/message/19087457
For non-Apple devices, check out the TextFree app https://itunes.apple.com/us/app/text-free-textfree-sms-real/id399355755?mt=8
How to Send SMS from iPad
http://www.iskysoft.com/apple-ipad/send-sms-from-ipad.html
Cheers, Tom -
When I call someone, people on the other end can't see my number. I checked the settings where you get 3 options. Send caller-ID, don't send caller-ID or let the network decide. Obviously "don't send" won't send it, but the other 2 options don't work either.
If I select "yes" the other end still sees "private number".
Is there any other option I'm overlooking?
Solved!
Go to Solution.yes call your operator and ask them to turn on caller id and always set the settings to yes anyway
If i have helped at all a click on the white star below would be nice thanks.
Now using the Lumia 1520 -
Optimizing chunks size for filesending with NetStream.send() method
Hi,
I would like to implement a p2p filesending application. Unfortunately object replication is not the best solution for this so I need to implement it myself.
The process is the following:
Once the user has selected the file for sending, I load it to the memory. After it is completed I cut the file's data into chunks and send these chunks with the NetStream.send() method - to get the progress of filesending -.
How can I optimize the size of the chunks for the fastest filesending method or what is the best size for a chunk?Hi there
Please submit a Wish Form to ask for some future version of Captivate to offer such a feature! (Link is in my sig)
Cheers... Rick
Helpful and Handy Links
Captivate Wish Form/Bug Reporting Form
Adobe Certified Captivate Training
SorcerStone Blog
Captivate eBooks -
Timed out while sending ... message may be send more than once
Hi,
I regular have complaints from some clients receiving my emails 10 -15 times in their mailbox.
When checking the maillog I always see : timed out while sending ... message may be send more than once. See below
Problem can happen at any time, even when there is just 1 person connected to the server, so I rule out overload.
95% of mails send arrive without problem.
When looking the dns mx info of these clients I am having problem with, all have multiple (round-robin ?) mx-hosts (eg mx.mailprotect.be).
Is someone else having these problems ?
gr
Fonny
Nov 20 14:05:05 osxserver postfix/smtp[33385]: 3846CBE9FFC7: conversation with monitor.estrel.com[62.154.155.69] timed out while sending end of data -- message may be sent more than once
Nov 20 14:15:48 osxserver postfix/smtp[33385]: 3846CBE9FFC7: to=<[email protected]>, relay=mx-30.startplus.de[62.154.155.70]:25, delay=1826, delays=486/0.01/716/623, dsn=4.4.2, status=deferred (conversation with mx-30.startplus.de[62.154.155.70] timed out while sending end of data -- message may be sent more than once)
Nov 20 14:15:48 osxserver postfix/smtp[33385]: 3846CBE9FFC7: to=<[email protected]>, relay=mx-30.startplus.de[62.154.155.70]:25, delay=1826, delays=486/0.01/716/623, dsn=4.4.2, status=deferred (conversation with mx-30.startplus.de[62.154.155.70] timed out while sending end of data -- message may be sent more than once)
Nov 20 14:27:06 osxserver postfix/smtp[34737]: 6F586BEA0F09: lost connection with mx1.hotmail.com[65.55.92.136] while sending end of data -- message may be sent more than onceIt means that Cyrus IMAP is not responding or ready to deliver mail (at least temporarily). Most likely reason is that there are too many connections/users.
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