After reading these links:
Using Objective C to read log messages posted to the device console
https://developer.apple.com/library/ios/#documentation/System/Conceptual/ManPages_iPhoneOS/man3/asl.3.html
I've successfully posted messages to the ASlog using
aslmsg m = asl_new(ASL_TYPE_MSG);
asl_log(NULL, m, ASL_LEVEL_INFO, result);
The problem is that when I go to query the log there is extreme lag in getting the results. It seems to be searching everything since I started printing with NSLog earlier today.
My current code to get the information is:
q = asl_new(ASL_TYPE_QUERY);
asl_set_query(q, ASL_KEY_SENDER, "db_poc", ASL_QUERY_OP_EQUAL);
asl_set_query(q, ASL_KEY_TIME, "1306768140", ASL_QUERY_OP_GREATER);
I'm trying to get my app to send messages to the console (from javascript/UIWebview). I want to then watch the console for these messages so I can send data back to the UIWebviews javascript code..
I wonder are there any extra flags I can set on either send or receive side to speed up things? Also, is there a way to clear this ASlog?
Any ideas..?
Thanks.
Have you tried creating your own aslclient using ASL_OPT_NO_DELAY?
Related
I implemented two different solution to discover service on my BLE device. One use a handler then return what .discoverService have found, the other one is really similar but give the size of the service discovered list that is always 0. I tried it with my realme buds 2 as test and some other device publically visible. The result is always 0. What can the problem be?
Handler(Looper.getMainLooper()).post {
var temp = bluetoothGatt?.discoverServices()
addGlog("discordservice() returned ${temp.toString()}")
}
addGlog("handler discover service reached an end")
val gattServices: List<BluetoothGattService> = gatt.getServices()
addGlog("Services count: " + gattServices.size)
for (gattService in gattServices) {
val serviceUUID = gattService.uuid.toString()
addGlog("Service uuid $serviceUUID")
}
edit: AddGlog is a simple log function to print results
answer: The code is not wrong but it take some time to discover those services so i put this code in a button. In this way there is 3-4 second of time between connecting with the device and make a discoveryservice operation. So a button make the conneting operations and another one the service discovery operations. I am sorry if my answer is pretty lame but I am still a noob on this topic
I'm trying to get a BG95 to activate on hologram.
Here are my commands:
AT+QCFG="band",F,180A,180A OK
AT+QCFG="iotopmode",2 OK
AT+QCFG="nwscanseq",020301 OK
AT+QCFG="nwscanmode",0 OK
AT+QCFG="snrscan",0 OK
AT+QICSGP=1,1,"hologram","","",1 OK
AT+QIACT=1 ERROR
At first I thought it was antenna/signal related so I ran AT+CSQ and got this:
+csq: 11,99
This tells me I have a good signal I believe.
Next I tried AT+QNWINFO and get this:
+QNWINFO: "eMTC","311480","LTE BAND 13",5230
In my mind this is saying it's connected to a network.
After trying that I tried to activate again and got this:
AT+QIACT=1
ERROR
The weird thing is it activated just fine about a week ago with pure AT commands. I did try and use an Arduino library with it (WisLTEBG96TCPIP) which may have changed a setting in it. I've done a factory reset but the it still woln't activate.
Another strange thing is the hologram dashboard. Every once and a while it will show the SIM as connected, even though I can't activate.
I have tried with 2 different SIM cards any get the same activation error.
Any help would be greatly appreciated!
Verizon has cut off all non ODI products. If your hardware has not been Verizon ODI 'certified' it will no longer be allow to be connected to their network, I have 5 new pet rocks thanks to them. The solution is to purchase new modems from vendors that have been through the Verizon ODI program or switch carriers.
I had the same problem before, after a lot of maling with network operator I find out that there isn't a LTE-CAT-M1 (eMTC) network in my area, I tested in another area successfully
Also before setting AT+QCFG commands try AT+CFUN = 0
and after setting AT+QCFG commands try AT+CFUN = 1 .
before sending AT+QIACT, try 'AT+CEREG?' command several times and tell me what is the return of it
i've been trying to develop a project taht uses gps and cell location with SIM5320A, i have successfully implement the GPS location, but when i try to use the AT commands related to the Cell location the SIM responses with an error message, for that reason i bougth a FONA 3g to make some tests with the cell location, but im algo getting error messages.
Hope you can help me with that
thank you
`
at
OK
at+cpin=0000
OK
+STIN: 25
+STIN: 25
+CPIN: READY
a
SMS DONE
+VOICEMAIL: INIT_STATE, 0, 0
a
PB DONE
START
at+cassistloc?
ERROR
AT+ CASSISTLOCTRYTIMES=3,2
ERROR
`
UPDATE
I've seen in some forums taht some AT doesnt work depending on the firmware version, mine is 1575B12SIM5320A , should i update to the last one?
when taking a shot using a long shutter speed >15 seg an error message is returned instead of the result containing the pocture address.The error message is "error": [40403, "Long Shooting"] .
the camera is a nex-6 the api realease v = 1.6
Please use getAvailableShutterSpeed method to get current possible values.
Best Regards,
Prem, Developer World team
See the special note in the API documentation in the "actTakePicture" section. For very long exposure times it will return error code 40403 in which case you can poll the camera using the "awaitTakePicture" method. The camera will continue to return 40403 when queried with "awaitTakePicture" until it's finished capturing, and then it will return the address of the postview image.
I'm pulling my hair out with this one. A month or so ago, I was able to put together a proof-of-concept WebRTC demo, using some sample code from the good folks at SignalR. The demo is located here, the source for it is here, and it does what it's supposed to do.
But when I took that code and moved it into our actual application, I haven't been able to get it to work. Of course the code had to be changed significantly - different backends, different set of frameworks and supporting code, supporting multiple simultaneous connections, that sort of thing - but the core logic is very similar. But I can't get it to work.
I've put together a sample app here that demonstrates the problem:
https://bitbucket.org/smithkl42/signalr.webrtc
The core WebRTC logic is all in this TypeScript file:
https://bitbucket.org/smithkl42/signalr.webrtc/src/tip/SignalR.WebRTC/Scripts/Media/WebRTC.ts?at=default
It's several hundred lines long, so I won't bother posting it here, but you can see it by clicking on the link above.
When it runs, it produces output like this:
12:17:58.531 WebRTCController.call(): Calling 7d9e0d39-5047-4afe-86e5-e6e01b9f5955 when preparations have finished
12:17:58.533 WebRTCController.prepareForCall(): Preparing for call: localSessionId='39d2df53-6854-415a-8748-b5230eda2eb1'; remoteSessionId='7d9e0d39-5047-4afe-86e5-e6e01b9f5955'
12:18:0.139 Object.(): The user has granted media device access, so proceeding to prepare for call
12:18:0.141 Connection.createPeerConnection(): Creating peer connection; using stunServer stun:stun1.l.google.com:19302
12:18:0.144 (): Preparations finished. Creating and sending JSEP offer. Util.js:21
12:18:0.272 Connection.handleIceCandidate(): STUN server has found an ICE candidate (event.type='icecandidate').
12:18:0.282 Connection.handleIceCandidate(): STUN server has found an ICE candidate (event.type='icecandidate').
(More like that)
12:18:0.655 WebRTCController.handleJsepAnswer(): Handling JsepAnswer from 7d9e0d39-5047-4afe-86e5-e6e01b9f5955
12:18:0.694 Object.(): Sending ICE candidate to the remote machine: {"sdpMLineIndex":0,"sdpMid":"audio","candidate":"a=candidate:2999745851 1 udp 2113937151 192.168.56.1 62978 typ host generation 0\r\n"}
12:18:0.706 Object.(): Sending ICE candidate to the remote machine: {"sdpMLineIndex":0,"sdpMid":"audio","candidate":"a=candidate:2999745851 2 udp 2113937151 192.168.56.1 62978 typ host generation 0\r\n"}
(More like that)
But then it never connects, i.e., the video from the other side never starts playing. At the signaling layer, I can tell by the logs and by stepping through the code that the first browser is sending a JSEP offer; the second browser is receiving it, storing it and sending back an appropriate JSEP answer; and the first machine is storing that answer. Each peerConnection is then finding the ICE candidates and sending them to the remote machine; and each peerConnection is receiving and apparently trying those ICE candidates; and the peerConnections are even raising the onaddstream event. But the video never starts playing.
The state of the peerConnection object all the way through looks like this:
(iceGatheringState=new; iceState=starting; readyState=active)
The frustrating bit is that every so often, maybe one time out of 20, it does work, i.e., both videos show up. So I'm not doing everything wrong. It sounds like a timing issue of some sort - but I can't figure out what it is. And so far as I can tell, there's not much in the WebRTC objects (specifically RTCPeerConnection) to tell you what's going wrong.
I hate to ask anybody else to do my troubleshooting for me, but... well, I'm running out of options. Does anybody else see anything I'm doing obviously wrong?
Update 2012-12-19: I'm making some progress. I realized I was calling peerConnection.setLocalDescription() synchronously, i.e., without specifying callbacks. So now I've got some lines of code that look like this:
// Answer the call by sending a JsepAnswer message.
connection.peerConnection.createAnswer(
answer => {
connection.peerConnection.setLocalDescription(answer, () => {
var signalState: mData.SignalState = {
FromSessionId: connection.localSessionId,
ToSessionId: connection.remoteSessionId,
Message: JSON.stringify(answer)
};
me.roomHub.server.jsepAnswer(signalState);
mUtil.log("Sent JSEP answer: " + signalState.Message);
connection.readyForIceCandidates.resolve();
},
error => {
mUtil.error("Error setting local description from created answer: " + error + "; answer=" + JSON.stringify(answer));
});
},
error => {
mUtil.error("Error creating answer: " + error);
}, me.mediaConstraints);
And the setLocalDescription() error callback is showing this error:
16:14:42.439 WebRTCController.handleJsepOffer(): Error setting local description from created answer: SetLocalDescription failed.; answer={"sdp":"v=0\r\no=- 439659381 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjf\r\nm=audio 1 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:vOKflTJ56gV0R9i0\r\na=ice-pwd:9nuXPMDvQ2mZATFCQyEzPRQz\r\na=sendrecv\r\na=mid:audio\r\na=rtcp-mux\r\na=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:m9q9pmLgLuFnfFC09KXKW5p8TjsKk+VdqX0OWv77\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:111 opus/48000/2\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:107 CN/48000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:548068416 cname:IXg8QRisWrd7+7f8\r\na=ssrc:548068416 msid:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjf a0\r\na=ssrc:548068416 mslabel:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjf\r\na=ssrc:548068416 label:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjfa0\r\nm=video 1 RTP/SAVPF 100 116 117\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:vOKflTJ56gV0R9i0\r\na=ice-pwd:9nuXPMDvQ2mZATFCQyEzPRQz\r\na=sendrecv\r\na=mid:video\r\na=rtcp-mux\r\na=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:m9q9pmLgLuFnfFC09KXKW5p8TjsKk+VdqX0OWv77\r\na=rtpmap:100 VP8/90000\r\na=rtpmap:116 red/90000\r\na=rtpmap:117 ulpfec/90000\r\na=ssrc:1460425980 cname:IXg8QRisWrd7+7f8\r\na=ssrc:1460425980 msid:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjf v0\r\na=ssrc:1460425980 mslabel:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjf\r\na=ssrc:1460425980 label:u9fhVrWeLLweqb5ubLkw61Ijsh6BM6vZLhjfv0\r\n","type":"answer"}
Now I just need to figure out why that particular SDP - which comes straight from the createAnswer() method - is failing.
Update 2012-12-20: I've created an online demonstration of the problem here: http://srdemo.alanta.com/. I've also turned on Chrome debug logging, with the result that I see a bunch of errors that look like this:
[6584:7308:1220/091356:ERROR:rtc_peer_connection_handler.cc(84)] Native session description is null.
[6584:7308:1220/091356:ERROR:rtc_peer_connection_handler.cc(84)] Native session description is null.
[6584:7308:1220/091356:ERROR:rtc_peer_connection_handler.cc(84)] Native session description is null.
[6584:7308:1220/091356:ERROR:rtc_peer_connection_handler.cc(84)] Native session description is null.
[6584:7308:1220/091356:ERROR:rtc_peer_connection_handler.cc(84)] Native session description is null.
Not sure what relationship they have to my problem, but I'm continuing to look into it.
*Edit 2012-12-20: I've managed (I think) to narrow the problem down. See this question for more precise details.
Figured it out. Turns out that SignalR 1.0 RC1 has a bug in it that changes any "+" in a string into a space. So lines in the SDP that looked like this:
a=ice-pwd:qZFVvgfnSso1b8UV1SUDd2+z
Were getting changed into this:
a=ice-pwd:qZFVvgfnSso1b8UV1SUDd2 z
But because not every SDP had a "+" in it on a critical line, sometimes it would work. Everything explained.
The bug has been reported to the good folks working on SignalR (see https://github.com/SignalR/SignalR/issues/1194), and in the meantime, a simple encodeURIComponent() and decodeURIComponent() around the strings in question fixed it.