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| 00:02.50 | fprior | Hi again, I'm testing Hangup Handler as showed in http://pastebin.com/rQwSwVe8 ; my question is how to pass MYVAR from outbound_handler back to extension 1978. |
| 00:04.12 | talnti | make MYVAR a global |
| 00:04.51 | talnti | SetGlobalVar() |
| 00:09.53 | fprior | talnti if a second call enter does not overwrite MYVAR ? |
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| 00:29.15 | WIMPy | fprior: Variables are local to a channel, unless you use something like GLOBAL(). |
| 00:31.45 | *** join/#asterisk dijib (~dijib@d157-77-106.home4.cgocable.net) |
| 00:31.57 | dijib | GUYS I HAVE AN ASTERISK LEGAL QUESTION |
| 00:32.19 | dijib | HOW do i prove that my mixmonitor recording has not been tampered with? |
| 00:32.25 | ChannelZ-Wk | Well I want to sue you for typing in all caps.. |
| 00:32.32 | WIMPy | Ask the lawyer you nistrust the least. |
| 00:32.38 | MaliutaLap | ChannelZ-Wk: +1 |
| 00:32.58 | MaliutaLap | dijib: mtime? atime? ctime? |
| 00:33.15 | dijib | I have moved the files to a datastore so ext4 file dates are altered |
| 00:33.23 | dijib | mtime atime ctime what? |
| 00:33.23 | ChannelZ-Wk | md5sum it or something at the end and record it somewhere? |
| 00:33.34 | MaliutaLap | dijib: you probably need to talk to someone who does forensics |
| 00:33.55 | MaliutaLap | well mtime would tell you when the file was last modified |
| 00:33.57 | ChannelZ-Wk | The tamper-proofing is only as secure as your system actually is regardless |
| 00:34.02 | dijib | is there any metadata i can pull out of the wav49? |
| 00:34.05 | MaliutaLap | yeah, IDS |
| 00:34.11 | fprior | talnti , WIMPy I solved setting MASTER_CHANNEL(SIPcode) on the handler and retrieving it on master extension |
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| 00:34.35 | dijib | Penguin: in the house? |
| 00:34.47 | dijib | WIMPy: whats nistrust? |
| 00:35.03 | WIMPy | mistrust |
| 00:35.24 | dijib | well my lawyer i would trust the most but he is older and doesnt understand computers |
| 00:35.56 | WIMPy | It's something you obviouselt cannot prove. If there ist something that a court would consider proof? I have no idea. |
| 00:36.11 | dijib | so i have to convince a room full of lawyers my recording has not been tampered with non of which are computer savvy |
| 00:36.26 | dijib | crd + recording length? |
| 00:36.28 | ChannelZ-Wk | Is this something that happened already? |
| 00:36.56 | dijib | no im doing a voir dire tomorrow to submit evidence into a trial im in, |
| 00:37.06 | dijib | same legal situation ive been in for years now |
| 00:37.13 | dijib | the reason i built the * box |
| 00:37.28 | ChannelZ-Wk | I don't know what a voir dire means |
| 00:37.44 | WIMPy | Wikipedia tells you |
| 00:37.49 | WIMPy | (and me) |
| 00:38.05 | dijib | its like a mini trial in a trial |
| 00:38.06 | dijib | http://en.wikipedia.org/wiki/Voir_dire |
| 00:38.08 | ChannelZ-Wk | a deposition |
| 00:38.42 | dijib | look at above link, first sentence in the "use in commonwealtgh countries" |
| 00:38.45 | dijib | section |
| 00:38.52 | ChannelZ-Wk | but I guess what I'm asking is are you trying to prove a piece of audio you already have hasn't been messed with, or you're asking "how would I do this in the future with files I record" |
| 00:39.05 | dijib | already have ChannelZ-Wk |
| 00:39.39 | dijib | is the application's use from my dialplan: same => n,MixMonitor(incoming/IN-${EXTEN}-${CALLERID(all)}-${STRFTIME(${EPOCH},$ |
| 00:40.04 | ChannelZ-Wk | Not very possible then. Besides actual forensic analysis of some sort that would analyize noise patterns or something to prove something new hadn't been inserted, or something cut out, etc. |
| 00:40.27 | dijib | hmmm |
| 00:40.34 | ChannelZ-Wk | Given the motivation I could patch parts of the audio and set the time on the filesystem any way I liked so saying "look, the date matches" doesn't say much |
| 00:41.22 | dijib | well im going to use the call length, datestamp, and my cdr records then |
| 00:41.25 | dijib | all i can do |
| 00:41.33 | ChannelZ-Wk | pretty much |
| 00:42.03 | dijib | k thanks guys, you are indespensible |
| 00:42.06 | ChannelZ-Wk | (any/all of which could be forged, but.. you've got what you've got) |
| 00:42.17 | dijib | exactly |
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| 00:42.50 | dijib | ok question: how many business and organizations are using asterisk currently |
| 00:42.58 | dijib | is there any number available of the userbase |
| 00:43.23 | pabelanger | well, there is like 60k users just on the mailing lists |
| 00:44.17 | MaliutaLap | dijib: the only way to "prove" it to a court is to have a computer forensics expert examine the system |
| 00:44.42 | MaliutaLap | dijib: someone whom the court would consider an "expert witness" in such matters |
| 00:44.57 | dijib | well im going to be that expert |
| 00:45.12 | MaliutaLap | You have credentials in forensics? |
| 00:45.20 | dijib | nope.jpg |
| 00:46.48 | MaliutaLap | you see the problem? you can be considered an expert on the * side of things, and how the files were created. But without something like an IDS and/or a report from a forensics expert you're shit out of luck proving that any electronic file hasn't been altered |
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| 00:52.11 | dijib | ty MaliutaLap |
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| 00:56.41 | MaliutaLap | The benefits of having lawyers as friends :) |
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| 05:53.41 | snmp | eggbot: Сэр, что с вашей шпагой? -это арматура, сэр |
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| 07:05.59 | Zogot | morning! |
| 07:08.24 | fahmad | Morning |
| 07:09.55 | fahmad | same question which was not answered yesterday, i have channels stucked in asterisk when i type `core show channels verbose` it shows me that calls are on phone from past 11 hours or so http://pastie.org/private/afxs2wf9pzyskb6lhrabfq |
| 07:10.06 | fahmad | can someone give me solution to get rid from this issue ? |
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| 07:30.01 | Gugge | fahmad: channel request hangup |
| 07:34.03 | fahmad | Gugge: i know that but is there any automated way :) |
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| 09:07.20 | jakey2 | Hi, what database does astriks use to store call information? |
| 09:21.00 | skrusty | whatever database you configure cdr to use |
| 09:21.23 | skrusty | you can use cdr_odbc to select any db supported by unix odbc, or mysql and a few others are baked right in |
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| 09:23.32 | yottanami | Goto(ext-queues,501,1) will send call to a queue how can I send a call to an extention ? |
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| 09:27.23 | jakey2 | skrusty: thanks |
| 09:29.23 | skrusty | yottanami: https://wiki.asterisk.org/wiki/display/AST/Creating+Dialplan+Extensions |
| 09:34.42 | yottanami | skrusty: tnx, I have a dialplan I just want to do somthing like this http://dpaste.com/2ZHPDN2/ after all calls ( not passing to a queue, I want sent it to an extention) |
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| 09:35.04 | yottanami | skrusty: can I write something like "exten => *,1,Answer" |
| 09:35.12 | andi | Hi, how can I configure a sipconnect 1.1 sip-trunk in asterisk? |
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| 10:52.33 | alami | is that very dangerouse to change asterisk user from root to another user? |
| 11:02.08 | r00f | i've heard people saying about dangers of running asterisk as root, but it's the first time i hear that running asterisk as non-root can be dangerous |
| 11:05.00 | alami | r00f: yes that's what i read also, but i have never do it :-)http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html |
| 11:06.19 | r00f | but it says that it is dangerous to run asterisk as root, not what you said |
| 11:06.30 | r00f | or i did not understand your question |
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| 11:08.19 | alami | r00f: no everythink is nice, i wanted to change the asterisk user from root to www-data so that i can change asterisk configuration from a php script, do you thing that#s a good practice? |
| 11:10.46 | r00f | running everything as non-root is a good practice. i was hacked once, some guy received root access to my box because i was running asterisk as root. since then i never do that |
| 11:11.11 | alami | r00f: okay thanks a lot for the info |
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| 11:20.54 | alami | r00f: do you think this simple methode will be enough? http://forum.voxilla.com/threads/running-asterisk-as-different-user-not-root.31241/ |
| 11:23.26 | r00f | yep. the only thing, if you have selinux enabled, it will not let non-root asterisk start. so you will need to create selinux policy, or disable it (which is again not recommended) |
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| 11:29.00 | alami | r00f: i don't have /etc/init.d/asterisk on my debian wheezy, whenre have you change the AST_USER and AST_GROUP |
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| 11:30.24 | bhavikpatel6842 | Hi Guys |
| 11:30.31 | r00f | asterisk.conf |
| 11:31.20 | bhavikpatel6842 | I want to use ChanIsAvail application and get its output in php file using php asterisk manager. |
| 11:31.40 | bhavikpatel6842 | Can any one idea how to run this application using AMI php |
| 11:31.41 | bhavikpatel6842 | ? |
| 11:32.18 | WIMPy | You can execute functions via GetVar. |
| 11:32.47 | *** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad) |
| 11:36.27 | bhavikpatel6842 | Action: GetVar Channel: SIP/6060,ast Variable: AVAILCHAN |
| 11:36.36 | bhavikpatel6842 | is this right thing ? |
| 11:37.32 | WIMPy | The function is the variable. |
| 11:37.49 | WIMPy | And if it's not channel specific, you don't need a channel. |
| 11:38.06 | bhavikpatel6842 | yes I need to know channel is available or not. |
| 11:38.12 | alami | r00f:i have uncommet this line on asterisk.conf and restart but i still see that the user is root |
| 11:38.15 | alami | runuser = asterisk ; The user to run as. |
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| 11:39.37 | bhavikpatel6842 | @WIMPy $data=$ASManager->GetVar($channel2, 'AVAILCHAN'); print_r($data); |
| 11:39.42 | bhavikpatel6842 | This is right one which I use ? |
| 11:40.05 | alami | http://pastie.org/9195699 |
| 11:40.52 | WIMPy | What do you think you get if you ask for "AVAILCHAN"? |
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| 11:44.13 | bhavikpatel6842 | I have one one channel and I want to verify it status |
| 11:44.43 | bhavikpatel6842 | how can I do it. |
| 11:44.44 | bhavikpatel6842 | for that I just create object for AMi in php and call Getvar function like above |
| 11:46.16 | bhavikpatel6842 | I am passing the channel like sip/6060 and checking this channel status |
| 11:46.36 | WIMPy | Do you see the name of the function ChanIsAvail somewhere in your line of code? |
| 11:47.00 | bhavikpatel6842 | Oh no where. |
| 11:47.23 | bhavikpatel6842 | so please suggest me for this application ? |
| 11:47.45 | bhavikpatel6842 | what should I use for working ChanIsAvail app ? |
| 11:47.52 | WIMPy | ChanIsAvail(NameOfChannel) |
| 11:48.33 | bhavikpatel6842 | should I use something like this ? |
| 11:48.35 | bhavikpatel6842 | $data=$ASManager->GetVar(ChanIsAvail(sip/6060), 'AVAILCHAN'); |
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| 11:49.18 | WIMPy | Almost |
| 11:49.30 | WIMPy | What ist this "AVAILCHAN"? |
| 11:49.49 | bhavikpatel6842 | is this channel is available for use or not ? |
| 11:50.06 | WIMPy | The function is the ``variable'' you want to get, which I guess is the 2nd paramaeter. |
| 11:50.10 | bhavikpatel6842 | it return channel name |
| 11:50.39 | bhavikpatel6842 | above one is not working in my case. |
| 11:51.26 | bhavikpatel6842 | $data=$ASManager->GetVar("ChanIsAvail('sip/6060')", 'AVAILCHAN'); |
| 11:51.42 | bhavikpatel6842 | Not sure what happening but this is not working. |
| 11:52.12 | WIMPy | What ist this "AVAILCHAN"? |
| 11:52.41 | bhavikpatel6842 | ${AVAILCHAN}: The name of the available channel, if one exists |
| 11:52.57 | WIMPy | And as I said in the beginning, the channel name to run that action on can (or in this case needs to) be left blank. |
| 11:53.00 | bhavikpatel6842 | this is channel variable if we run successfully ChanIsAvail application using AMI |
| 11:54.03 | WIMPy | Ah, that's the idea. You can't read variable that yould be set by the function. You can only read the return value of the function. |
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| 11:55.03 | bhavikpatel6842 | Yeah I can able to do using phpagi using like this : |
| 11:55.04 | bhavikpatel6842 | $result=$agi->exec("ChanIsAvail",$channel2); |
| 11:56.06 | bhavikpatel6842 | and new 3 variable are set in channel and then after we can get using get_variable function use for this AVAILCHAN. |
| 11:56.07 | bhavikpatel6842 | but as we know this is php file not AGI one so we have to use ASterisk manager for getting variable. |
| 11:57.06 | WIMPy | I should work if you run the function on an existing channel and then get the variable from the same channel afterwards. |
| 11:57.16 | WIMPy | But you need an active channel for that. |
| 11:58.00 | bhavikpatel6842 | nope I have one simple php script and I just pass my channel and get its status using AMI. |
| 11:58.17 | bhavikpatel6842 | is it possible ?if yes then how ? |
| 12:04.16 | WIMPy | Thinking again, maybe you don;t need a channel. It *might* work in the global space. |
| 12:04.54 | WIMPy | But you do need to operations. One to execute the function and another one to read the contents of that variable that (hopefully) has been set by it. |
| 12:05.16 | bhavikpatel6842 | But I originate call using originate command of AMI but originate call first I have to check this channel status |
| 12:06.07 | bhavikpatel6842 | is the channel status is busy then I have to use another channel |
| 12:06.13 | WIMPy | Well, you could just originate the call and see what happens. |
| 12:06.34 | snmp | he need listen to the events? |
| 12:06.37 | bhavikpatel6842 | Yes its working but I use dahdi channel using span |
| 12:08.28 | WIMPy | Well, the situation can change between your call to ChanIsAvail to your originate action, so you probably want to check what actually happens anyway. |
| 12:09.52 | bhavikpatel6842 | http://pastebin.com/u7urum0S |
| 12:09.57 | bhavikpatel6842 | please check this code. |
| 12:10.14 | bhavikpatel6842 | let me know if I am doing wrong. |
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| 12:12.35 | WIMPy | Well, so far you're not using the result anyway. |
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| 12:14.00 | bhavikpatel6842 | so please suggest me what I have to use : |
| 12:14.00 | bhavikpatel6842 | ) |
| 12:14.04 | WIMPy | And I have no idea what getChannelStatus does, but that somehow looks like a duplicate action. |
| 12:14.25 | WIMPy | Just start the call and see what happens. |
| 12:14.36 | bhavikpatel6842 | Oh this one is my test function for just testing but it is commented.so no worry. |
| 12:15.35 | bhavikpatel6842 | yes call is working but I need status of the channel |
| 12:15.56 | WIMPy | I don't see the point. |
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| 12:16.16 | WIMPy | I'm sure you want the status of the call, not of some channel at some time before the call. |
| 12:16.54 | bhavikpatel6842 | Please check this one : core show application ChanIsAvail |
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| 12:18.05 | bmurt | morning folks |
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| 12:39.35 | Ibrahim22 | Hi, everyone! |
| 12:41.28 | Ibrahim22 | I am trying to develop something with WebRTC, but I am having difficulty setting it up. I have done everything on the Asterisk Page (https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) but I still get this error: "Received SAVP profile in audio offer but AVPF is enabled: audio 33820 RTP/SAVP 8 101" in CLI |
| 12:42.06 | Ibrahim22 | I found something on the tracker, a patch: https://issues.asterisk.org/jira/browse/ASTERISK-22005?jql=text%20~%20%22avpf%22 |
| 12:42.30 | Ibrahim22 | But I'm unsure that it will work, and would like to know if it is possible to reverse a patch ? |
| 12:45.38 | bhavikpatel6842 | which phone your using for create demo of webrtc |
| 12:45.44 | bhavikpatel6842 | ? |
| 12:45.51 | Ibrahim22 | polycom |
| 12:46.04 | bhavikpatel6842 | oh I see I used sipml5 |
| 12:46.55 | Ibrahim22 | Yeah, I want to be able to use either the desk phone and a phone i'm developing myself (via WebRTC) |
| 12:47.13 | Ibrahim22 | so i need to enable avpf, but not via polycom |
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| 12:56.55 | bhavikpatel6842 | Not sure how to |
| 13:05.44 | alami | can any one help with this issue? x-lite and blink tell me that the conference pin is invalid but, with Zoiper everything work very good? |
| 13:08.12 | malcolmd | perhaps the pin isn't being read by asterisk because the dtmfmode presented by the client isn't configured as acceptable in asterisk. |
| 13:08.54 | *** join/#asterisk vNistelroot (~secure@63.Red-88-10-102.dynamicIP.rima-tde.net) |
| 13:08.57 | vNistelroot | hi people |
| 13:10.24 | alami | malcolmd: i have checked the dtmfmode and it's the same on all softphone and i have also put dtmfmode=rfc2833 on sip.conf |
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| 13:12.51 | [TK]D-Fender | alami: Where you put things count, and days ago when you came with this I asked for the proper debug of a working, and non-working call and never got it. |
| 13:12.55 | [TK]D-Fender | alami: Show us the calls |
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| 13:15.09 | bhavikpatel6842 | is there any other way in phpagi through which we do not need to press enter on script execution? |
| 13:15.39 | alami | [TK]D-Fender: i havn't see that yesterday, i tought you was out :-) |
| 13:15.48 | alami | [TK]D-Fender: http://pastie.org/9195982 |
| 13:18.59 | [TK]D-Fender | alami: you have no basic verbose in there and I asked for 2 calls... |
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| 13:20.26 | alami | [TK]D-Fender: can you tell me wich verbose i should use? |
| 13:20.40 | [TK]D-Fender | "core set verbose 10" |
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| 13:24.33 | alami | [TK]D-Fender: http://pastie.org/9195999 |
| 13:24.44 | alami | i have make now 2 can one with x-lite and one with Zoiper |
| 13:28.49 | [TK]D-Fender | still no verbose in there.... |
| 13:29.57 | alami | [TK]D-Fender: but i have type core set verbose 10 |
| 13:30.41 | Gugge | Dont exit the asterisk console after you set verbose |
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| 13:34.40 | alami | ok thanks Gugge |
| 13:34.44 | alami | so now http://pastie.org/9196021 |
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| 13:41.54 | [TK]D-Fender | alami: Are those multiple machines behind the same IP, or are your switching clients? |
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| 13:47.02 | alami | [TK]D-Fender: i thing that's the Problem, because i have both softphone on my machine, but asterisk in other machine |
| 13:48.12 | [TK]D-Fender | Are you switching, or running both at the same time? |
| 13:50.01 | alami | both softphone at the same time |
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| 13:50.18 | alami | befor a time ago it was working |
| 13:51.00 | anonymouz666 | danjenkins: npm install asterisk-ami |
| 13:51.03 | [TK]D-Fender | never do that |
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| 13:51.16 | danjenkins | hi anonymouz666 |
| 13:51.26 | [TK]D-Fender | Test themn separately |
| 13:51.36 | anonymouz666 | danjenkins: how does NodeJS compares to Python+Twisted regarding performance? |
| 13:51.38 | *** part/#asterisk bhavikpatel6842 (~root@122.169.33.37) |
| 13:51.53 | danjenkins | ive never compared the two directly to be honest |
| 13:51.55 | alami | [TK]D-Fender: do mean run one softphone per every machine? |
| 13:52.23 | anonymouz666 | danjenkins: I ask that because I am planning to use under heavy load |
| 13:52.27 | danjenkins | but i would bet my bottom dollar that node would perform better as long as you wrote it right |
| 13:53.02 | danjenkins | put it this way, my old contact centre was very busy, over 100 seats, active from 8 am til 11 at night |
| 13:53.06 | alami | [TK]D-Fender: ohh right, if i exist Zoiper i can join a conference also with x-lite |
| 13:53.29 | danjenkins | and node dealt with it fine, it was actually the AMI itself that was a limiting factor |
| 13:53.55 | danjenkins | the process ran without needing to be restarted for over 9 months, and then the restart was just bewcause of OS updates etc |
| 13:55.14 | anonymouz666 | danjenkins: I am watching again your presentation, the final part, because I want to setup from scratch exactly the example you did |
| 13:55.31 | danjenkins | ah ok, it should be up on github somewhere |
| 13:55.32 | danjenkins | the example |
| 13:55.42 | danjenkins | maybe on mine |
| 13:56.13 | danjenkins | https://github.com/danjenkins/astricon-2012 |
| 13:56.21 | danjenkins | different branches for the different steps |
| 13:57.32 | Ibrahim22 | Is it possible to have one peer/friend definition in sip.conf for SIP and WebRTC? |
| 13:58.17 | Qwell | Ibrahim22: as opposed to zero or two? |
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| 13:59.27 | anonymouz666 | danjenkins: I have a meeting today with dev guys, my job will be to convince the team to the NodeJS side, after some research I think it's the right way to go. |
| 13:59.53 | anonymouz666 | danjenkins: thanks for the github link |
| 13:59.53 | alami | so it's not possible to run two softphone from the same machine? |
| 14:00.07 | danjenkins | no problem anonymouz666 |
| 14:01.34 | Katty | i'm going to be indianapolis in two weeks |
| 14:01.41 | Katty | if any of you crazies are out that way. |
| 14:02.10 | anonymouz666 | danjenkins: if you don't mind, i will bother you about telling our results with NodeJS |
| 14:02.56 | danjenkins | sure anonymouz666 , alwyas interested! |
| 14:04.24 | alami | [TK]D-Fender: for time a go, that was just working, do have any idea how can i test conference now? |
| 14:05.03 | [TK]D-Fender | alami: Did you go test them individually ? |
| 14:05.37 | alami | [TK]D-Fender: yes, and the conference work now with both softphone |
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| 14:06.28 | alami | but for a time ago, i was able to run the conference from two sofphone at the same time |
| 14:06.41 | alami | for the same machine |
| 14:08.58 | fahmad | same question which was not answered yesterday, i have channels stucked in asterisk when i type `core show channels verbose` it shows me that calls are on phone from past 11 hours or so http://pastie.org/private/afxs2wf9pzyskb6lhrabfq |
| 14:09.21 | fahmad | can someone give me solution to get rid from this issue ? |
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| 14:10.58 | [TK]D-Fender | alami: that is a dumb test and a waste of time. The conference works. The softphones each work on their own. Use 1 per machine. |
| 14:11.49 | alami | [TK]D-Fender: ok thanks you're right i have to go forward :-) thank you so much |
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| 14:11.59 | alami | and sorry for yesterday |
| 14:12.24 | [TK]D-Fender | fahmad: "core show help channel request hangup" |
| 14:12.39 | [TK]D-Fender | alami: You're welcome |
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| 14:13.00 | fahmad | [TK]D-Fender: i know this but i want something automated :$ |
| 14:13.30 | [TK]D-Fender | fahmad: No automated system is going to make magically correct decisions about what channels you want to kill |
| 14:18.43 | fahmad | hmm |
| 14:19.07 | fahmad | i want to kill the channels which are taking more then one hours on calls regardless of SIP/DHADI/IAX |
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| 14:21.59 | anonymouz666 | danjenkins: the code that listen to 8080 and the webpage.html isn't available, right? |
| 14:22.45 | danjenkins | anonymouz666: you mean https://github.com/danjenkins/astricon-2012/tree/advanced_webpage ? |
| 14:22.55 | danjenkins | theres 3 branches on that repo :) |
| 14:23.03 | anonymouz666 | huhu awesome |
| 14:23.07 | anonymouz666 | thanks! |
| 14:24.33 | mjordan | fahmad: AMI is your friend for things like that. You can listen for Newchannel events, start a timer for each channel, and if they don't get hung up, hang them up yourself |
| 14:25.11 | WIMPy | fahmad, mjordan: Or just set an absolute timeout? |
| 14:27.25 | fahmad | WIMPy: for outgoing calls i have put absolute timeout but for incoming calls ... |
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| 14:28.48 | mjordan | what you're saying however is that you want an absolute timeout for all your calls. You may as well put that on your inbound channels as well. |
| 14:30.47 | jameswf | 6 months from now... All my calls hangup after 10 minutes... why |
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| 14:37.07 | fahmad | mjordan: Ok |
| 14:41.45 | fahmad | the problem is i am using freepbx and if i put absolute timeout it does not work !! |
| 14:42.05 | [TK]D-Fender | It does. |
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| 14:43.28 | fahmad | [TK]D-Fender: where should i place absolute timeout because freepbx modify extensions_addtional.conf |
| 14:50.57 | litn | how should I go about researching/debugging the reason for dropped calls? We get a few a day... I know the phone numbers and around the time it happens, what logs do I need to look at or where do I start? |
| 14:51.00 | [TK]D-Fender | fahmad: you'd make a context in extensions_custom.conf point your calls there, set your timeout and jump back to from-trunk |
| 14:52.46 | WIMPy | litn: '<channeltype> set debug on' and wait for it to happen (again). |
| 14:53.59 | fahmad | Ok |
| 14:58.02 | fahmad | [TK]D-Fender: like this http://pastie.org/private/mxwina0n6cnekfbbrffejw |
| 15:00.53 | anonymouz666 | danjenkins: express has changed a lot between version 3.x and 4.x. :-( |
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| 15:01.13 | danjenkins | oh yeah, it has anonymouz666 |
| 15:01.28 | anonymouz666 | so the server.js doesn't run clean |
| 15:02.04 | anonymouz666 | I have installed all the npm stuff, including the separated errorhandler |
| 15:02.25 | danjenkins | oh anonymouz666 |
| 15:02.31 | fahmad | [TK]D-Fender: when i use it, included it under from-trunk it disconnecting all calls without completing absolute time |
| 15:02.43 | danjenkins | i cant believe i didnt add express to the dependencies list in package.json |
| 15:03.01 | danjenkins | you can do npm install express@3.whatever |
| 15:03.21 | danjenkins | you probably want to do npm install express@3.x.x |
| 15:03.52 | fahmad | i will be back online after sometime !! |
| 15:03.54 | fahmad | need to leave |
| 15:10.29 | anonymouz666 | danjenkins: up and running, beautiful. |
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| 16:12.09 | boom^time | I'm doing an AMI originate to a local context which executes Dial. The Variables I set in the originate are not visible to that Channel, does anyone know of a way to make them accessible? |
| 16:12.43 | [TK]D-Fender | boom^time: Show us |
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| 16:14.20 | modesto916 | Hello everyone. I would like to know if there is a way to configure asterisk to just set up the sip calls and after that traffic be direct from endpoint to endpoint. Is this possible? I think I've read this somewhere but I can't remember where it is. |
| 16:14.25 | boom^time | [TK]D-Fender, here is the channel dump http://pastebin.com/6aNp2mxE |
| 16:14.36 | Qwell | modesto916: directmedia=yes |
| 16:15.07 | boom^time | with a LOCAL/<number>@originate/n |
| 16:15.23 | [TK]D-Fender | boom^time: and rthe AMI call.... |
| 16:15.24 | vlad_starkov | Question: I'm about to install Asterisk 1.9.0 on Ubuntu Server 14.04 64bit. When I try "sudo ./contrib/scripts/install_prereq install" it throws conflict warnings like "gir1.2-atk-1.0 : Conflicts: gir1.2-atk-1.0:i386 but 2.10.0-2ubuntu2 is to be installed." and so on. Anyone know how to deal with that? |
| 16:16.05 | modesto916 | I'm asking this because we would like to save some bandwidth in clients where the phones are in different location from the server. |
| 16:16.24 | modesto916 | Oh, thank you very much. |
| 16:16.27 | modesto916 | I'll try that |
| 16:16.37 | Qwell | modesto916: That'll be $399.98 |
| 16:17.41 | modesto916 | Haha thank you for the 2 cents discount |
| 16:17.54 | boom^time | [TK]D-Fender, http://pastebin.com/3gYdSaZY |
| 16:18.00 | boom^time | it's because I'm setting CDR variables isn't it. |
| 16:18.43 | [TK]D-Fender | boom^time: CDR is a FUNCTION... and the words READ-ONLY come to mind... |
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| 16:19.26 | boom^time | [TK]D-Fender, it's not read only, you can set them for custom cdr values for odbc |
| 16:19.35 | boom^time | it works perfectly on the other leg |
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| 16:20.14 | boom^time | cdr_adaptive_odbc.conf |
| 16:20.33 | fahmad | back |
| 16:21.48 | boom^time | You can create custom columns for cdr values that you set with Set(CDR(column_name)=value) |
| 16:23.17 | fahmad | [TK]D-Fender: still there |
| 16:26.13 | [TK]D-Fender | yes |
| 16:26.37 | [TK]D-Fender | boom^time: probably doesn't work because the channel isn't executing yet |
| 16:26.42 | fahmad | [TK]D-Fender: when i use it into fromm-trunk it disconnects all incoming calls |
| 16:26.48 | [TK]D-Fender | boom^time: pass them as ACTUAL variables, then do Set() in your dialplan |
| 16:27.40 | [TK]D-Fender | fahmad: that is a poor description and doesn't tell me what you did |
| 16:28.09 | fahmad | [TK]D-Fender: hold |
| 16:28.41 | boom^time | [TK]D-Fender, Right. Do you know if a variable on the other leg of the call can be set? One of the issues I'm having is if the call is a congested failure ;2 creates a CDR with a disposition of Congested while ;1 sets it to NO ANSWER. I would love it and greatly simplify things if ;1 would reflect ;2 |
| 16:29.28 | [TK]D-Fender | boom^time: it is... I belive its just an issue of the channel not being "up" enough to set thouse where you are. |
| 16:29.54 | [TK]D-Fender | boom^time: If you want those CDR things set for the called channel You may have to run a M() against it |
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| 16:33.07 | fahmad | [TK]D-Fender: http://pastie.org/private/wceb7ae8yljc97qonaypsa |
| 16:33.50 | [TK]D-Fender | fahmad: You are never ever supposed to touch extensions.conf directly there |
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| 16:34.05 | [TK]D-Fender | fahmad: Your chages get blown away the moment you apply any change to FreePBX |
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| 16:34.17 | fahmad | [TK]D-Fender: hmm |
| 16:34.46 | fahmad | [TK]D-Fender: then where should i add this from-timelimit |
| 16:34.47 | boom^time | [TK]D-Fender, Okay that makes sense and thank you. A separate issue I'm having is where in the case that the called channel has a failure of Congested the other leg ends with a disposition of NO ANSWER. Instead of congested. |
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| 16:36.47 | boom^time | I'm not really sure if there is anything that can be done about that though as they are two separate legs, yet to be bridged. |
| 16:38.06 | [TK]D-Fender | [10:50][TK]D-Fenderfahmad: you'd make a context in extensions_custom.conf point your calls there, set your timeout and jump back to from-trunk |
| 16:38.46 | fahmad | hmm |
| 16:39.12 | fahmad | like i have added one inbound route which have did 920033314 ... |
| 16:39.38 | fahmad | how can i move it to jump extensions_custom.conf |
| 16:39.39 | [TK]D-Fender | fahmad: change your TRUNK |
| 16:40.16 | fahmad | Ok |
| 16:40.24 | fahmad | i can not access Office VPN :( |
| 16:40.36 | fahmad | feeling bad :( |
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| 16:43.39 | modesto916 | Qwell: Do you know if it's possible to use directmedia while allowing call transfers? |
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| 16:49.46 | [TK]D-Fender | modesto916: Not DTMF based ones |
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| 16:56.19 | modesto916 | [TK]D-Fender: Our phones are configured with rfc2833 signalling |
| 16:58.14 | [TK]D-Fender | modesto916: I did not say anything about the mode you use |
| 16:58.40 | [TK]D-Fender | modesto916: if you use DIAL's DTMF transfer features that will force * to stay in the middle of your calls. |
| 17:01.02 | modesto916 | [TK]D-Fender: What's the other option for transfers? |
| 17:01.15 | [TK]D-Fender | modesto916: doing an actual SIP transfer |
| 17:02.15 | modesto916 | [TK]D-Fender: I don't know what that means but I'll take a look =) |
| 17:02.32 | modesto916 | [TK]D-Fender: I think this require some configuration on the endpoint right? |
| 17:02.33 | ChannelZ-Wk | IE the transfer button (hopefully) on your phone |
| 17:07.26 | [TK]D-Fender | ^^^^ |
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| 17:17.49 | philfry | so i have some conference lines setup but i can't seem to transfer to them. It is only looking for 2 digit extensions. Is it possible to allow it to transfer to a 4 digit extension? |
| 17:18.40 | navaismo | uh? sounds like a phone dialplan issue but show us the call and the dialplan |
| 17:20.02 | ChannelZ-Wk | yeah, _what_ is 'only looking for 2-digit extensions?' |
| 17:21.51 | [TK]D-Fender | What is a "conference line"? |
| 17:22.01 | [TK]D-Fender | As for call flow problems... show us the call... |
| 17:25.10 | philfry | ok yeah thats what i thought. One second. |
| 17:26.25 | philfry | Well it doesn't wait for all 4 digits. After 2 digits it accepts the input and tries to go to that extension. A conference line is what freepbx calls it. |
| 17:27.56 | [TK]D-Fender | you need to be specific about the transfer feature being used, how you use it on the device, etc |
| 17:28.16 | ChannelZ-Wk | FreePBX |
| 17:30.26 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 17:31.48 | philfry | okay let me test it again |
| 17:35.38 | philfry | so this is what i found so far. I have a digium D40 phone and it works fine but the polycom phones don't transfer the same way. |
| 17:35.46 | philfry | Where should i put the log from asterisk? paste bin? |
| 17:36.41 | philfry | basically the extension for that conference line is 1033. It tries to transfer to 10 which doesn't exist. |
| 17:38.39 | navaismo | check you phone's dialplan |
| 17:39.28 | mjordan | It's highly unlikely that Asterisk is going to be the cause of or solution to this problem |
| 17:40.12 | *** join/#asterisk zerick (~eocrospom@190.118.43.113) |
| 17:41.17 | [TK]D-Fender | philfry: Again what BUTTONS are you pushing to do these "transfers"? |
| 17:42.00 | [TK]D-Fender | philfry: and if it cuts off your dialing instantly at the 0, then it is your phone dialplan that needs to be adjusted |
| 17:44.25 | philfry | oh okay i didn't think about the phone's dialplan |
| 17:44.34 | philfry | i'll look at that |
| 17:45.11 | navaismo | somtimes feels like is talking to a wall |
| 17:47.08 | philfry | sorry i didn't think about that lol |
| 17:47.11 | [TK]D-Fender | No, typically walls reflect audio. More like talking into studio sound insulation foam :0 |
| 17:47.21 | navaismo | echo |
| 17:47.40 | navaismo | ~echo |
| 17:47.40 | infobot | from memory, echo is an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 17:48.04 | navaismo | awesome i love this bot not the frog |
| 17:51.36 | jameswf | all these years and all i needed was a link... |
| 17:55.43 | *** join/#asterisk that1dude (~Speedy059@67.40.83.104) |
| 17:55.57 | that1dude | Does anyone know if Elastix supports CnetoS 6? |
| 17:56.07 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 17:56.47 | ipengineer | When using mixmonitor is there a way to avoid having recordings that are <1 sec. For example I set monitor on a channel and ring a device. If that device doesnt answer and the call goes to voicemail I get a really short file.. Here is an example: https://gist.github.com/zconkle/eafaa8ccab7e92c3af8d |
| 17:57.10 | [TK]D-Fender | that1dude: What does their website say about their ISO? |
| 17:58.59 | modesto916 | Do you guys know how can I replace dtmf transfer to SIP transfers? |
| 17:59.07 | navaismo | that1dude, test the beta distro |
| 17:59.14 | *** join/#asterisk LemensTS (~t15@8.33.19.98) |
| 17:59.21 | that1dude | I believe their ISO is 5.9 |
| 17:59.52 | navaismo | the stable use 5.10 |
| 18:00.04 | navaismo | try the 3.0 beta |
| 18:00.11 | LemensTS | Howdy. Is there a way to change the cid on my polycom lcd from saying 700(700) when I park a call? It is transfering the call to 700 of course.....but since parked calls is not a dialplan context I can edit I am unsure of how to do this |
| 18:00.26 | LemensTS | *I can't edit |
| 18:00.39 | LemensTS | Or I would just set callerid to something else... |
| 18:00.47 | [TK]D-Fender | LemensTS: it is a context |
| 18:01.22 | that1dude | Any recommendations for a consultancy company or managed service company that can install and manage Elastix for ~800 users |
| 18:01.25 | LemensTS | include = parkedcalls <- i include it, I thought this was built by features.conf on the fly |
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| 18:02.06 | [TK]D-Fender | LemensTS: "core show function CONNECTEDLINE". |
| 18:02.16 | navaismo | that1dude, 800 normal users or anything fancy? |
| 18:02.34 | [TK]D-Fender | LemensTS: You can also make your own exten for parking instead of using the default. |
| 18:02.48 | navaismo | that1dude, which country? |
| 18:02.57 | that1dude | video conferencing, fax to email - email to fax, etc. |
| 18:03.00 | that1dude | In the USA, Arizona |
| 18:03.17 | that1dude | Elastix seems to support all the features needed for the project |
| 18:03.30 | navaismo | yeah, but better use plain asterisk |
| 18:03.47 | that1dude | Doesn't Elastix use Asterisk? |
| 18:04.13 | navaismo | in the botton layer yes, it use a fork for an ancient freepbx with many many modules/addons in it |
| 18:04.27 | navaismo | but yea you can give it a try |
| 18:04.37 | navaismo | or use a freepbx-distro to |
| 18:04.39 | navaismo | o |
| 18:04.41 | that1dude | Customer needs a lot of features, almost similar to MS Lync |
| 18:04.52 | navaismo | thts why i hate customers |
| 18:04.54 | that1dude | Lync was getting to bloated and cost to much, so they are interested in something else |
| 18:06.17 | navaismo | well in the future you may regret the decision of switch that to a "opensource" solution... |
| 18:06.42 | navaismo | an* |
| 18:06.48 | that1dude | Just need something other than MS Lync that can support (for an example): Fax, Conference bridges, standard handsets, soft phones, presence, chat, vm to email, auto attendants, multiparty video, call center agent setup |
| 18:06.53 | that1dude | Anything is better than Lync |
| 18:06.59 | [TK]D-Fender | [14:04]that1dudeCustomer needs a lot of features, almost similar to MS Lync <- when comparing you should have more details on this. |
| 18:07.46 | navaismo | that1dude, comparing that features and using the words "out of the box" elastix seems your best option, but... |
| 18:07.58 | navaismo | NM walks away |
| 18:11.22 | that1dude | HMM |
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| 18:21.55 | LemensTS | <PROTECTED> |
| 18:25.29 | that1dude | Anyone know of any good manage service providers that can setup a Elastix install? |
| 18:26.13 | navaismo | try google |
| 18:26.35 | navaismo | my boss do that but we are in mexico |
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| 19:05.03 | pabelanger | anybody doing asterisk integrations with sugarCRM or vtiger? |
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| 19:07.42 | Chainsaw | pabelanger: Just very simple right now, SugarCRM database lookups for inbound & outbound calls to fill the caller ID/connected line ID in. |
| 19:07.52 | Chainsaw | pabelanger: Which is a MySQL function that the DBAs wrote for me |
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| 19:15.49 | Chainsaw | pabelanger: Based on the silence, I shall assume that this isn't sufficient for your needs. However, if you should change your mind let me know. |
| 19:17.30 | pabelanger | Chainsaw, sorry, dealing with fires. Why sugerCRM? what that existing or did you decided to use it from scratch? |
| 19:17.47 | Chainsaw | pabelanger: The company was already using SugarCRM. |
| 19:17.56 | pabelanger | Ya |
| 19:17.57 | pabelanger | we |
| 19:17.59 | pabelanger | err |
| 19:18.03 | Chainsaw | pabelanger: So having the phone system check calls against it seemed natural. |
| 19:18.27 | pabelanger | Ya, we're trying to decided what to use to house our data, then add hooks for telephony into it |
| 19:18.38 | Chainsaw | pabelanger: It tries to find out caller name & company affiliation if it can. If it can't, it gets you location information based on area/country code instead. |
| 19:19.39 | pabelanger | Chainsaw, Ya, basically what we want to do. We'll tag the call as needed with more specific data, but basically pop using existing software logic. |
| 19:19.50 | pabelanger | does sugarcrm do the pop right in the web app |
| 19:19.53 | pabelanger | or is it an external app? |
| 19:20.15 | Chainsaw | pabelanger: The way we do it is at the database level. |
| 19:20.36 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:20.38 | Chainsaw | pabelanger: So no real UI integration. Just suck the data out of the SQL and push some into XMPP if the user has it enabled. |
| 19:21.05 | Chainsaw | pabelanger: "Your phone is ringing for a call from Joe Bloggs / Asterisk Inc" |
| 19:21.57 | pabelanger | Chainsaw, got it |
| 19:25.06 | boom^time | Can you store an asterisk dial plan entirely in a database, ie postgresql |
| 19:26.28 | pabelanger | yes |
| 19:26.30 | boom^time | maybe a context table with a relation to an extension table with a context foreign key, ext, priority, and function columns |
| 19:27.01 | navaismo | read about realtime in asterisk at the |
| 19:27.03 | boom^time | pabelanger, neat, would you mind pointing me in the right direction. My google search is resulting in astDB. |
| 19:27.05 | navaismo | ~book |
| 19:27.06 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:28.09 | boom^time | navaismo, thank you. |
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| 20:00.01 | jmmills | does anyone have a vim syntax file? |
| 20:00.31 | jmmills | for dialplan formats that is? |
| 20:05.23 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:09.07 | [TK]D-Fender | https://www.google.ca/#q=asterisk+vim+syntax |
| 20:09.15 | [TK]D-Fender | Google seems to be able to find all sorts.... |
| 20:10.07 | newtonr | it comes with vim now right? |
| 20:10.20 | newtonr | i haven't had to install one in years and my .conf files are all highlighted |
| 20:15.12 | TazzNZ | jmmills: you are running "vim" and not "vi" ? |
| 20:17.02 | leifmadsen | yum install vim-enhanced |
| 20:17.07 | leifmadsen | that's all I ever do on CentOS |
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| 20:21.07 | SuperNull | Hey guys, trying to switch from cdr_mysql to cdr_odbc and im not sure if the tables are compatible ? |
| 20:23.59 | skrusty | are you still using mysql through odbc? |
| 20:27.37 | SuperNull | yes ;) i think i got it tho. forgot that cdr_odbc references dsns from res_odbc.conf not from odbc.ini |
| 20:30.36 | SuperNull | trying to rid my self of mysql specific stuff for future ease and conformity. |
| 20:32.21 | mjordan | SuperNull: cdr_adaptive_odbc is the one you want |
| 20:32.37 | mjordan | SuperNull: cdr_odbc is ... sad. |
| 20:33.14 | mjordan | thinks we really should nuke cdr_odbc from high orbit |
| 20:34.37 | file | mjordan, I am going to create a branch one of these days which nukes old stuff |
| 20:34.44 | file | just because I can |
| 20:34.46 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 20:34.57 | mjordan | file: would that branch be trunk? |
| 20:35.07 | file | pfft |
| 20:35.21 | file | Monitor? Gone. Old channel drivers? Gone. |
| 20:35.32 | file | DESTROY |
| 20:35.41 | mjordan | most stable version ever! |
| 20:43.15 | skrusty | :) |
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| 20:49.42 | jmmills | TazzNZ: vim - on a mac |
| 20:51.35 | SuperNull | mjordan any specific reason other than custom fields? |
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| 21:01.22 | *** join/#asterisk lucido (~chris@f051202025.adsl.alicedsl.de) |
| 21:02.26 | lucido | hi, is it possible to fax and implement an answering machine with asterisk? |
| 21:03.07 | ChannelZ-Wk | Yes. |
| 21:03.38 | *** join/#asterisk PHunter (~PHunter@ip72-208-101-51.ph.ph.cox.net) |
| 21:05.58 | PHunter | Anyone know of any tools to check UDP connectivity to a server? |
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| 21:23.18 | TazzNZ | PHunter: do you want to check UDP or SIP (via UDP) ? |
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| 21:27.43 | WIMPy | pong |
| 21:27.59 | PHunter | SIP via UDP |
| 21:28.15 | PHunter | I has suspicion to believe their config is blocking it outbound. |
| 21:28.19 | PHunter | have* |
| 21:28.36 | WIMPy | sipsak |
| 21:32.36 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 21:33.00 | saint_ | Hey Guys.. A quick share with you.. CSF is now able to block failed login attempt into asterisk. Better than Fail2Ban |
| 21:33.04 | saint_ | See http://forum.configserver.com/viewtopic.php?f=6&t=7351&p=22759#p22759 |
| 21:33.05 | PHunter | SIP SAK is Linux, what about Windows? |
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| 21:37.54 | fprior | newtonr, thanks for support of yesterday; finally I resolved using Set(MASTER_CHANNEL(SIPcode)=_____) in hangup_handler; now I can retrieve var SIPcode from originating extension |
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| 21:50.31 | jameswf | saint_: is that scientific or your personal opinion |
| 21:50.55 | jameswf | saint_: geeks like data |
| 21:52.22 | jameswf | PHunter: asterisk is for linux, safe bet you have linux if you use asterisk |
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| 21:53.17 | *** join/#asterisk smaboshe (~smaboshe@unaffiliated/smaboshe) |
| 21:59.20 | PHunter | jameswf: Linux hosted. End clients do not use Linux, Asterisk admins do. |
| 22:00.02 | PHunter | Server in Phoenix, clients in California. We never receive SIP registration packets. |
| 22:00.31 | PHunter | Trying to find the break in communications whether its on their site, or ISP level. |
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| 22:12.38 | jameswf | Netcat runs on windows |
| 22:16.37 | PHunter | I cant seem to find any Windows releases on their site or sourceforge site. |
| 22:17.07 | PHunter | Oh wait, nmap. |
| 22:17.17 | PHunter | Thats what I was trying to use but had no WPCap |
| 22:17.44 | PHunter | Oh wait no that was all precompiled pieces. |
| 22:18.10 | jameswf | Just netcat on both sides talk over the ports. If it works you dont have a routing issue |
| 22:18.57 | PHunter | Nevermind, Im not sure which way is up atm. Late in the workday. |
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