| 00:00.15 | phix | The reason I asked is I see a few s,1,NoOp(log something here)'s in my asterisk console |
| 00:01.34 | Penguin | Hangup(), which is at priority 2, will be executed after priority 1. |
| 00:02.35 | Penguin | I will also say that I have never used the old-style caller ID matching at priority 1. |
| 00:03.00 | phix | that is old style caller ID matching? what is new style? |
| 00:03.13 | phix | CALLERID(number) ? |
| 00:04.31 | phix | s,1,gotoif($[CALLERID(number) == 100]?Dial(SIP/trunk1/${ARG1})\n s,n,gotoif($[CALLERID(number) == 101]?Dial(SIP/trunk2/${ARG1})\n |
| 00:04.34 | phix | ? |
| 00:04.59 | [TK]D-Fender | phix: Forgetting to reference your functions.... |
| 00:05.18 | phix | yeah I was short cutting it :) |
| 00:05.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.87) |
| 00:05.25 | phix | exten => blah blah |
| 00:05.34 | [TK]D-Fender | phix: And you don't shove a command like Dial() into a GotoIf.... |
| 00:05.54 | [TK]D-Fender | phix: GotoIf JUMPS. that is all, it does not execute apps conditionally |
| 00:06.00 | Penguin | You could use ExecIf(). |
| 00:06.19 | phix | [TK]D-Fender: yes it does, but you get what I meant |
| 00:06.27 | [TK]D-Fender | phix: No... it doesn't |
| 00:06.39 | phix | just the callerid matching is what I was trying to get accross |
| 00:06.41 | [TK]D-Fender | phix: You cannot pass an application to GotoIf and get it to execute |
| 00:07.17 | phix | [TK]D-Fender: yes I know, but that wasn't the example I was trying to get across, it was the callerid matching in a macro-dialout |
| 00:07.36 | Penguin | GotoIf() goes to labels, not execute other applications. |
| 00:07.42 | phix | Correct |
| 00:07.51 | phix | It does do that, but will that of matched? |
| 00:07.59 | [TK]D-Fender | phix: Don't make another mistake when trying to find your first.. it make it look like you have NO clue :) |
| 00:08.19 | phix | it may of errored :) but would it of matched? |
| 00:08.25 | Penguin | may have |
| 00:08.27 | phix | :) |
| 00:08.37 | Penguin | have matched |
| 00:09.05 | phix | (assuming the callerid had been set for the sip phone, which it has been) |
| 00:09.24 | phix | basically I want to dial a different trunk depending on which phone is used to make the call |
| 00:09.34 | *** join/#asterisk unstable (unstable@tor/regular/sid) |
| 00:09.38 | Penguin | If you're having a problem with the old callerID matching, try moving it to a priority after 1. I've never had a problem with it, and I've also never used it at priority 1. |
| 00:09.54 | [TK]D-Fender | phix: You didn't reference your functions properly... |
| 00:10.13 | [TK]D-Fender | phix: So no, forget about "before the ?" working |
| 00:10.13 | phix | I currently use the old style callerid matching but it looks like that also matches s,1,... as well as s/100,1,... as I get both entries in my asterisk console (asterisk -rvvvvdddd) |
| 00:10.19 | unstable | I have general Internet phone questions, thougH I don't know if asterisk is what I need. I just want to have a phone number where people can call me over the Internet, and not need cellular service. I want a dedicated number, what is a convenient and cheap way to get his on a linux operating system? |
| 00:10.33 | phix | [TK]D-Fender: <3 |
| 00:10.41 | Penguin | Move it to 2. See if you get a different result. |
| 00:11.21 | Penguin | s,1,NoOp(); s/100,2,Whatever(); s,2,SomethingElse(); |
| 00:11.56 | phix | will s/100,2 ever be reached? |
| 00:12.07 | Penguin | Only if 100 is the callerid number. |
| 00:12.08 | phix | hmmm I would rather not try that out atm, as the system is currently in use |
| 00:12.42 | phix | I mean it is working but I also get s,1,NoOp's in my log too, so just trying to track down what is causing that |
| 00:13.08 | phix | if phone100 calls out it uses trunk1 and phone101 uses trunk2, that bit works |
| 00:13.11 | Penguin | I've never ever put the callerID match at priority 1, and I've never ever had a problem with it executing properly. |
| 00:13.35 | phix | hmmm, what you reckon [TK]D-Fender ? should I have callerid match at priority 1?> |
| 00:14.23 | [TK]D-Fender | Show how you'd like to do it and we'll critique from there |
| 00:14.33 | WIMPy | The callerID match failed for me some time in the 1.6 area. I never tried since. |
| 00:14.40 | phix | when does callerid match execute? or does it matter in what order I have it in? |
| 00:14.43 | Penguin | I used it back in 1.4. |
| 00:15.00 | Penguin | I can only assume it still works. |
| 00:15.16 | WIMPy | Ok, I was unclear. It failed after some upgrade. |
| 00:15.35 | Penguin | Actually, I still have some in my dialplan, but I think it's on a DID I no longer have. |
| 00:16.12 | WIMPy | That should make sure it doesn't fail :-) |
| 00:16.14 | Penguin | I'd bet it still works if I try it. I'm using 1.8. |
| 00:16.17 | phix | [TK]D-Fender: I want to use it as a fallback, I have s/100, s/101, s/etc... then at the last line I have exten => s,1,NoOp(Use default trunk) exten => s,2,Dial(SIP/defaultSipTrunk/${ARG1},,Tr) |
| 00:17.01 | phix | basically for all of the callerids I care about, use specific sip trunks, but if none match then use a default / fallback sip trunk |
| 00:17.07 | [TK]D-Fender | phix: it's an override at that priority level |
| 00:19.10 | phix | I guess I could rewrite it and use gotoIf($[CALLERID(number) == 100]?dial-100-out) and specify dial-100-out in the macro, rinse and repeat for other matches? |
| 00:19.14 | Max_E | unstable, do you want a pstn number or only a way to people call you using internet? |
| 00:21.06 | phix | Penguin, [TK]D-Fender: or is there a better way to do this? |
| 00:21.22 | Max_E | because you can be reached by a sipuri, by the PHONOSDK, by webrtc, etc etc |
| 00:21.48 | Penguin | phix: http://pastebin.com/rRETZfhH |
| 00:22.04 | Penguin | It works. Don't break it. |
| 00:23.18 | Penguin | I can't say it works in a macro, but it works from Gosub() on every single inbound call I receive. |
| 00:23.27 | [TK]D-Fender | phix: If you're sure to use CID for this, multiple CID matches looks cleaner than app-based conditional |
| 00:23.46 | Penguin | I've provided a nice, clean way. |
| 00:23.52 | [TK]D-Fender | phix: if you can afford to use a match for it.. in a macro, etc I'd use straight dialplan |
| 00:23.57 | phix | [TK]D-Fender: yes I prefer the look of that too but Penguin said that was old match style |
| 00:24.32 | [TK]D-Fender | Not a question of "old". Both ways have always existed. It's a question of which is cleaner, and which works best with where you're going to use it. |
| 00:24.38 | Penguin | My dad's old, but he still works. |
| 00:24.44 | phix | [TK]D-Fender: I want to use a macro as I have my dialplan filtering allowed number patterns |
| 00:25.26 | [TK]D-Fender | [19:21]PenguinIt works. Don't break it. <--- ummm.. it doesn't work.. you already broke it ;) |
| 00:25.28 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 00:25.46 | Max_E | i dont have a dad :'( :'( |
| 00:26.01 | phix | so I want to block out 1900 numbers, and use a different trunk for 1800 and 13 numbers, and for numbers that are greater than 6 digits then use the dialout macro for the trunk filtering |
| 00:26.06 | Penguin | [tk]d-fender: What are you talking about? What I pasted works 100%. |
| 00:26.21 | [TK]D-Fender | Penguin: not quite... |
| 00:26.27 | phix | Max_E: Are you the second coming of Jesus/ |
| 00:26.30 | [TK]D-Fender | Penguin: You "oopsed" |
| 00:26.39 | Penguin | Tell me why you think so. |
| 00:26.57 | [TK]D-Fender | Penguin: Penguin In what world is "s" a priority on lines 2 & 3? |
| 00:27.01 | Penguin | And I'll tell you why you're incorrect. |
| 00:27.16 | Penguin | Priority s means same. Similar to how n means next. |
| 00:27.26 | phix | lol s priority |
| 00:27.33 | [TK]D-Fender | They made a synonym for it? |
| 00:27.39 | [TK]D-Fender | makes no sense |
| 00:27.45 | phix | never used that before |
| 00:27.47 | Max_E | phix, alive or is missing dont know |
| 00:27.55 | phix | Max_E: oh, ;( |
| 00:28.03 | [TK]D-Fender | Penguin: I've got to go see some doc for this... |
| 00:28.39 | Penguin | If you've ever needed to use numbered priorities because you wanted to have several priority 3 instances, you could have used n and s rather than numbers. |
| 00:29.07 | phix | A psychiatrist? |
| 00:29.27 | WIMPy | Wow. Someone told the fender something new about dialplan syntax. |
| 00:29.37 | phix | hai WIMPy! |
| 00:29.58 | WIMPy | o/~ |
| 00:30.10 | Penguin | s,1,NoOp(prio 1); s,n,NoOp(prio 2); s/100,s,(Still prio 2); s/101,s,NoOp(Still prio 2); s,n,NoOp(prio 3); |
| 00:30.19 | Max_E | where is the popup about the unlock achievement |
| 00:31.08 | Penguin | I can't remember if I used the s priority back in 1.4 or not, but it works well in 1.8. |
| 00:31.10 | [TK]D-Fender | Penguin: Yup, just looked it up... that is an odd thing to see... |
| 00:31.18 | [TK]D-Fender | Penguin: You get a pass :) |
| 00:31.23 | Penguin | Yay me! |
| 00:31.49 | Max_E | is taking a screenshot |
| 00:31.53 | Penguin | hahahaha |
| 00:31.57 | [TK]D-Fender | Penguin: But this is something that twits have huge odds of screwing up worse than screwing up hard numbered priorities, etc |
| 00:32.07 | Penguin | I won't argue that. |
| 00:32.30 | [TK]D-Fender | Penguin: And in you same (functional as it is), it's still a lazy shit substitute for just putting "1" there instead :) |
| 00:33.10 | [TK]D-Fender | Penguin: You didn't even save a byte and you can't do multiple steps for the same pattern without scrapping your use of "s" on the subsequent ones... |
| 00:33.22 | Max_E | dont apply he just won |
| 00:33.26 | Max_E | let it go |
| 00:33.48 | [TK]D-Fender | Penguin: But I'll concede that your suggestion appears to work in as (equally) small a space as anything else :) |
| 00:33.49 | Penguin | It allows for the flexibility of non-numbered priorities just like n does. Let's imagine you wanted to add somewhere between 6 and 53 more lines between 1 and those lines with s... |
| 00:34.09 | [TK]D-Fender | Penguin: mixing s & n become problematic... |
| 00:34.17 | Penguin | Only if you can't read dial plan! |
| 00:34.48 | Penguin | It may not be good for everyone, but it's good for asterisk. |
| 00:34.48 | Max_E | can we go back to the phix solution, i need to add it to my book |
| 00:34.58 | Max_E | of possible future slutions |
| 00:38.56 | [TK]D-Fender | Max_E: Remember, copying from 1 person is plagiarism. Copying from a hundred is RESEARCH. |
| 00:38.58 | Penguin | You can follow my example, or you can do it the hard way. I don't care either way. |
| 00:38.58 | Max_E | sometimes you are so mean :( |
| 00:38.58 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:38.58 | Penguin | Do as I say, not as I do. |
| 00:38.59 | Max_E | now i need icecream |
| 00:38.59 | Penguin | I could use some ice cream. |
| 00:38.59 | Penguin | After I got back from the store the other night, I regretfully made the realization that I didn't get any ice cream. |
| 00:38.59 | *** join/#asterisk jetlag (~jetlag@pool-71-168-196-225.cmdnnj.east.verizon.net) |
| 00:50.13 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:50.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 01:48.01 | margeas | hello ppl, it seems that my h extension in dialplan does not trigger if I hang-up the phone while ringing....what could be the problem ? |
| 01:48.45 | margeas | I'm trying something very simple as "same => h,1,NoOp('h' extension executed)" |
| 01:49.40 | margeas | before that I have a simple same => n,Dial() line in the context and nothing else... |
| 01:50.09 | margeas | I don't see the message in console though... |
| 01:50.21 | margeas | any hints? |
| 01:56.02 | *** join/#asterisk D30 (~deo@58.71.19.178) |
| 01:56.11 | LiuYan | margeas: "same => h,1,NoOp" <-- that can't be right, you can try "exten => h,1,NoOp" |
| 01:58.02 | pabelanger | exten => h,1,NoOp() |
| 01:58.06 | pabelanger | same => n,Blah() |
| 01:58.13 | pabelanger | that is the syntax |
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| 01:58.17 | margeas | LiuYan: ups(!!)...OK it's triggering now... |
| 01:58.20 | margeas | the fact is |
| 01:59.31 | margeas | that I am calling a macro from that context...and in the macro the last line is: exten => h,1,NoOp('h' extension executed) but in that case the h does not trigger.... |
| 02:00.09 | margeas | I'm assuming something wrong? i.e. that the macro has "its" h extension |
| 02:00.36 | pabelanger | step 1, stop using macro |
| 02:00.37 | pabelanger | use gosub |
| 02:01.13 | margeas | I see a lot of confusion about this around and dozens of references contradict themselves |
| 02:02.31 | margeas | pabelanger: OK, to be clear: i'm following straight here http://zwizwa.be/-/asterisk/20120418-152928 look at the last macro exten... |
| 02:03.25 | margeas | I'm quite novice, any best practices in handling macros currently is appreciated |
| 02:04.10 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 02:04.23 | pabelanger | margeas, macros have been deprecated in Asterisk, so save yourself some pain and use GoSub. |
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| 02:04.30 | pabelanger | There is also a limit of like 10 levels deep |
| 02:04.35 | pabelanger | then asterisk crashes |
| 02:04.47 | margeas | when using macros? |
| 02:04.51 | pabelanger | yes |
| 02:05.13 | margeas | OK, i will try to adapt it...does it exist a command reference online for asterisk? |
| 02:06.16 | WIMPy | ~wiki |
| 02:06.52 | WIMPy | Hmm. No. http://wiki.asterisk.org/ |
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| 02:07.06 | margeas | https://wiki.asterisk.org/wiki/display/AST/Application_Gosub ? |
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| 02:12.48 | margeas | thank you all, goodbye! |
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| 03:01.37 | Greek-Boy | I have a problem with SIP signalling over a openvpn tunnel. RTP audio seems to be just fine but when I hangup the call it doesn't send the hangup to the other side. Any ideas? |
| 03:03.38 | Penguin | Can no one correctly spell signaling? |
| 03:03.46 | WIMPy | 1. your sip does not go over that tunnel - 2. you have a firewall on the tunnel - 3. some component is broken. |
| 03:04.51 | WIMPy | Penguin: You use the wrong english. |
| 03:04.59 | Penguin | Oh really? |
| 03:05.16 | Penguin | Which one do you prefer? |
| 03:05.51 | WIMPy | The one where signalling is spellt signalling. |
| 03:06.42 | WIMPy | But you can correct me on the spelling of spell. |
| 03:06.48 | Penguin | I'm not familiar with that one. Although it must exist -- so many other people seem to use it. |
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| 03:08.04 | WIMPy | en_GB vs en_US |
| 03:09.54 | Greek-Boy | lol |
| 03:10.43 | Greek-Boy | WIMPy: Thanks for those pointers |
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| 06:14.43 | phix | hey gang |
| 06:15.38 | ChannelZ | yo yo bust a cap in yo ass |
| 06:15.55 | phix | werd up |
| 06:29.04 | *** part/#asterisk unstable (unstable@tor/regular/sid) |
| 06:32.24 | phix | http://pastebin.com/P9Lfsfk6 |
| 06:32.44 | phix | That is what I got so far, Penguin mentioned it was old school matching, does that mean it is deprecated? |
| 06:34.07 | *** join/#asterisk trox_t (~trox@cm147.kappa24.maxonline.com.sg) |
| 06:35.02 | trox_t | hello guys, i need some help on setting up my asterisk and digium card... can anyone help me? |
| 06:35.11 | trox_t | thank you in advance |
| 06:35.58 | phix | I will be in and out but I can give you some pointers if you like |
| 06:36.11 | phix | I have setup a digium card a few time (TDM400p) |
| 06:36.19 | trox_t | @phix, thanks that would be great |
| 06:36.25 | trox_t | i have 410p |
| 06:36.26 | phix | What type of card? |
| 06:36.41 | phix | so one FXO port? |
| 06:36.44 | phix | or one FXS port? |
| 06:36.53 | trox_t | 2/2 ports |
| 06:37.00 | trox_t | 2 FXO and 2 FXS |
| 06:37.00 | phix | so you have a 422p :P |
| 06:37.17 | trox_t | i think i have TDM 410P let me double check |
| 06:37.21 | phix | ok and what OS you got? |
| 06:37.28 | phix | doesn't matter :) |
| 06:37.32 | trox_t | ok :) |
| 06:37.36 | trox_t | ubuntu server |
| 06:37.51 | trox_t | 12.04.3 LtS |
| 06:38.13 | trox_t | basically i setup the basic configuration, i can call softphones |
| 06:38.18 | phix | ok, well you can use asterisk and dahdi from the ubuntu repo or from digium, personally it doesnt matter, it works on both |
| 06:38.37 | phix | so sudo apt-get install asterisk dahdi-source |
| 06:39.01 | trox_t | that one i alredy took care, basic configuration is done already.. |
| 06:39.02 | phix | it will install build-essential as well and a few other stuff |
| 06:39.21 | phix | you installed dahdi-source too and compiled the kernel modules? |
| 06:39.27 | trox_t | yes |
| 06:39.39 | phix | and they load at boot time? |
| 06:39.40 | trox_t | i can see modules from asterisk already |
| 06:39.44 | phix | ok great |
| 06:40.04 | phix | you have genarated the dahdi-channel files for use in asterisk? |
| 06:40.17 | trox_t | hold on let me chck |
| 06:40.19 | trox_t | check |
| 06:41.00 | trox_t | dahdi-channels.conf right? |
| 06:41.04 | phix | ok and /etc/dahdi/system.conf has the correct country code on the network you will ne connecting it to? |
| 06:41.08 | phix | s/ne/be/ |
| 06:41.18 | phix | haha stupid regex |
| 06:41.36 | trox_t | :) my system.conf i haven't touchd yet |
| 06:42.18 | trox_t | dahdi-channels.conf is generated already |
| 06:44.11 | trox_t | i did some sample configuration and im getting this error when i try to call the dahdi channel |
| 06:44.12 | trox_t | Executing [9001@default:1] Dial("IAX2/8001-9938", "DAHDI/1") in new stack |
| 06:44.13 | trox_t | [Dec 10 13:50:27] WARNING[9737][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 06:44.13 | trox_t | <PROTECTED> |
| 06:46.11 | trox_t | system.conf shows loadzone=us, defaultzone=us -- |
| 06:47.59 | trox_t | let me know if you need more information guys, thanks |
| 06:50.54 | trox_t | basically for now id like to configure my phone to call any extension, like i can call any sip to iax extensions that i created |
| 06:51.30 | trox_t | then later on setup to call outside (phone line) |
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| 07:06.22 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:21.28 | DataWraith | how can i tell asterisk to pickup an external call from a sip-trunk on a specific extension? |
| 07:23.42 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:24.58 | kaldemar | DataWraith: you don't. instead, you tell the other end what extension they should dial in your register statement in sip.conf. |
| 07:25.27 | kaldemar | DataWraith: unless you really mean a device when you say extension. |
| 07:26.04 | kaldemar | DataWraith: then you dial it just like any other in the extension that the call lands in. |
| 07:29.55 | kaldemar | trox_t: why are you trying to dcc me? |
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| 07:35.46 | trox_t | kaldermar, thought i could ask something in private |
| 07:35.56 | trox_t | its ok, i can ask here |
| 07:37.59 | trox_t | same question ive posted earlier |
| 07:38.09 | trox_t | im a little lost :( |
| 07:38.37 | trox_t | just need some directions on how to configure my dahdi channel to call sip trunk or vice versa |
| 07:38.41 | trox_t | sip extension i mean |
| 07:42.10 | trox_t | if anyone could point me in the right direction that would be appreciated :) |
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| 07:45.31 | DataWraith | i get the error: Call from '0621231001' (193.84.65.66:5060) to extension '43621231001' rejected because extension not found in context 'default' |
| 07:46.18 | MaliutaLap | DataWraith: that is a self explanatory one |
| 07:46.47 | MaliutaLap | look at what context the call is coming in on, and where the extension you want to match is |
| 07:46.53 | DataWraith | and my extensions.conf looks like this: http://pastebin.com/a1qvEJDx |
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| 07:52.55 | DataWraith | MaliutaLap: but i have that extenstions 'default' and '43621231001' |
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| 08:02.03 | trox_t | for anyone who could help, here are some conf files i have and the error output when i try to call the dahdi channel http://pastebin.com/S36YCK1L |
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| 08:16.28 | DataWraith | now it works. |
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| 08:16.48 | DataWraith | finally i have to figure out, how to manage that incoming calls ring on every phone |
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| 08:21.43 | trox_t | hi DataWraith, do you have digium card installed? |
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| 08:31.52 | kaldemar | DataWraith: do you now know what an extension is and what a context is? |
| 08:32.44 | kaldemar | also, stop using the default context and configure your peer in sip.conf to use something else. |
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| 08:33.47 | kaldemar | trox_t: dahdi show channels |
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| 08:55.04 | trox_t | kaldemar: here it is thanks |
| 08:55.06 | trox_t | http://pastebin.com/2ezYwRNi |
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| 08:59.26 | kleszcz | trox_t: DAHDI/r0/${EXTEN} |
| 09:00.43 | trox_t | klesszcz: sorry where do i enter that? |
| 09:00.50 | kleszcz | extensionsc.onf |
| 09:01.00 | kleszcz | exten => 9001,1,Dial(DAHDI/1) |
| 09:01.10 | kleszcz | exten => 9001,1,Dial(DAHDI/r0/${EXTEN} |
| 09:01.24 | kleszcz | or 9001,1,Dial(DAHDI/1/${EXTEN} |
| 09:02.34 | trox_t | so it needs to be sequence 1 |
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| 09:02.42 | kleszcz | 9001,1,Dial(DAHDI/1/${EXTEN}) |
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| 09:07.09 | kaldemar | trox_t: one of your outputs show you trying to dial channel 1. "dahdi show channels" tells you don't have such a channel configured in chan_dahdi.conf or another file included from it. |
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| 09:15.27 | escube | HI guys, I need some help with asterisk |
| 09:15.39 | escube | I'm trying to connect avaya sip phone to asterisk, so far i made two phones talk to each other |
| 09:15.49 | escube | now I'm trying to implement the function of redirect a call, and put the phone in redirection mode |
| 09:16.53 | trox_t | kaldermar: i have channel 1 and channel 2 on chan_dahdi.conf |
| 09:17.29 | escube | I don't know how to save a phone state in a permanent way in my dialplan |
| 09:17.43 | trox_t | if im doing it wrong do you have a sample config for me? thanks |
| 09:17.50 | escube | and how to visualize this information on the phone when it's in idle mode |
| 09:18.15 | trox_t | kleszcz: i already modifid my dial plan still no go when i try to dial my dahdi channel |
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| 09:53.47 | skrusty | morning |
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| 10:04.29 | escube | Hi guys, anyone can tell me how i can display something on the phone display ? |
| 10:04.41 | escube | I need to set a D when forward is active |
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| 10:11.06 | escube | no one? |
| 10:11.34 | kaldemar | trox_t: does not look like you do. pastebin your system.conf and chan_dahdi.conf. |
| 10:11.40 | bulkorok | escube: set callerid(name) |
| 10:11.47 | WIMPy | escube: Take a look at the phones manual. |
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| 10:22.41 | escube | WIMPy: sorry just lost my connection |
| 10:23.28 | escube | bulkorok: i tried that, doesn't work |
| 10:24.12 | escube | I need a permanent D, and my avaya phone manual doens't say much |
| 10:24.43 | bulkorok | it's a phone thing... |
| 10:24.45 | WIMPy | If the manual doesn't tell you, connect it to an Avaya and do some reverse engeneering. |
| 10:24.45 | escube | do you know where I can find information on this topic, just need to knwo where to look |
| 10:25.50 | escube | ok if you say it's a phone function and is not settable from dialplan I'll look in that direction |
| 10:26.23 | WIMPy | What protocoll are you using anyway? |
| 10:26.38 | WIMPy | Are they SIP phones? |
| 10:27.26 | escube | SIP phones yes |
| 10:28.30 | WIMPy | If you're lucky it's using SIP-I. |
| 10:28.52 | WIMPy | But Asterisk doesn't. |
| 10:34.09 | escube | WIMPy : I need to implement a forward function, in dial plan i did that, I wish only that the phone in witch forward is active, I can visualize something about the state |
| 10:34.35 | escube | WIMPy : something like "Forward Active" |
| 10:34.40 | WIMPy | As we all do. |
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| 10:35.33 | escube | WIMPy: I'm very new to Asterisk, so not sure yet what can be done and what can't |
| 10:35.33 | WIMPy | Maybe you have keys with LEDs (BLF style) that you can use for that purpose? |
| 10:36.33 | WIMPy | As a general rule: The more sophisticated, the easier. That also means that some standard stuff is (next to) impossible. |
| 10:37.38 | escube | WIMPy: I can visualize callerid name for example when call is active |
| 10:37.59 | escube | WIMPy: but how to visualize something when call is not active? |
| 10:38.19 | WIMPy | As you said yourself, only a call has a caller ID. |
| 10:38.31 | WIMPy | >>Maybe you have keys with LEDs (BLF style) that you can use for that purpose? |
| 10:38.51 | escube | WIMPy: isn't there another function similar to that for other purposes' |
| 10:38.53 | escube | ? |
| 10:39.11 | WIMPy | No |
| 10:39.43 | WIMPy | Many phones have proprietary options to change the idle screen. |
| 10:39.59 | WIMPy | Possibly using graphics. |
| 10:40.33 | WIMPy | So that's more like changing your desktop background image to a version that includes some text. |
| 10:41.09 | escube | WIMPy: thank you very much |
| 10:41.17 | escube | WIMPy: I'll go look for that |
| 10:41.40 | WIMPy | Your best bet is to use a direct dialling key with status LED for such things. |
| 10:41.59 | WIMPy | That's also done in a standard way then. |
| 10:42.47 | WIMPy | And you can also use that key to (de)activate the forwarding then. |
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| 11:52.26 | jkroon | hi guys, with asterisk 11.6.0, after a reload it seems that not all members of a queue is receiving calls. |
| 11:52.46 | jkroon | this is a general asterisk -rx reload style reload, after queue reload all everything seems to be working correctly again |
| 11:56.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 12:02.13 | jkroon | hmm, it would seem that app_queue starts to think that the one (of two) endpoints in the queue has gone into Ringing, but there aren't even channels pointing to that SIP endpoint, so that doesn't seem sensible. |
| 12:03.19 | jkroon | it would seem that whatever the state is at time of reload sticks... |
| 12:09.02 | *** join/#asterisk oquidave (~oquidave@41.77.79.182) |
| 12:09.33 | oquidave | hello has anyone ever configured a huawei 3G modem to work with asterisk using chan_dongle module? |
| 12:09.55 | kleszcz | yes |
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| 12:14.42 | oquidave | kleszcz: cool, am having trouble making it work. let me send you my configs |
| 12:19.17 | oquidave | kleszcz: here http://pastebin.com/cApCfffF when i call the dongle, it calls continously and the server doesn't pick the call at all |
| 12:22.20 | oquidave | am i missing some configs in the dongle.conf file or exxtensions.conf file? |
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| 12:40.45 | oquidave | any help |
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| 12:55.15 | file | Arret demande! |
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| 13:16.12 | WIMPy | Show us what happens. |
| 13:21.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:25.10 | jkroon | hi, ok, tracked a definite app_queue reload issue |
| 13:25.34 | jkroon | when reloading the queue member's status goes into some state as per time of reload |
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| 13:26.39 | Greenlight | hmm that explains some stuff |
| 13:32.30 | jkroon | haha, ok, so I'm not the only one. |
| 13:32.35 | jkroon | <PROTECTED> |
| 13:32.42 | jkroon | extension is DEFINITELY not in use. |
| 13:33.16 | jkroon | some kind of race condition between handle_statechange and the reload mechanism. |
| 13:33.57 | Greenlight | I'm not the biggest fan of app_queue |
| 13:34.15 | Greenlight | it was my colleague who mentioned some odd stuff happening if they reloaded sometimes |
| 13:34.32 | jkroon | lol, i wonder why ... :p |
| 13:35.06 | jkroon | what alternatives are there to get the same functionality in a sane way though? |
| 13:35.29 | Greenlight | FOr our application, I now handle all queueing via our AMI connection |
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| 13:35.36 | jkroon | ouch |
| 13:35.41 | jkroon | i think i'll rather fix this bug thanks. |
| 13:35.54 | Greenlight | It was actually a breath of fresh air to have it work how users expect |
| 13:36.13 | Greenlight | But for our "just pbx" installs, we still need app_queue |
| 13:36.13 | jkroon | and how's that |
| 13:36.44 | Greenlight | Like just little things, if a user finishes a call, and there's a call in a queue, their phone starts to ring, instead of waiting till the next "retry" |
| 13:37.25 | jkroon | i set wrapup time anyway ... |
| 13:37.55 | Greenlight | Yea, but imagine a back office setup, where a user is hurridly ending there call because they hear a phone ringing and want to answer it |
| 13:38.13 | Greenlight | *their |
| 13:38.33 | jkroon | *8 |
| 13:38.37 | jkroon | oh, you have ringall :p |
| 13:38.41 | Greenlight | Yea |
| 13:38.55 | jkroon | leastrecent mostly hear, so we just *8 them |
| 13:39.42 | Greenlight | It's also cool to have control over the queuing priorities inside my application though - opens up a load of doors |
| 13:40.42 | Greenlight | Attempting to connect callers to the agent they've been dealing with before, if we can. Giving certain callers priority. That kind of thing |
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| 13:41.58 | Greenlight | I'm also rather convinced app_queue has some sort of resource type leak that was the cause of all those issues we used to get |
| 13:43.21 | jkroon | i wouldn't be surprised either which way, but for my requirements it's adequate |
| 13:43.30 | jkroon | very seldomly have >3 callers in a queue. |
| 13:43.44 | jkroon | this member status == in_use though is a serious issue. |
| 13:47.18 | Greenlight | Only started happening in 11.6.0 ? |
| 13:47.50 | jkroon | well, no, client originally reported in 11.3.0 |
| 13:48.07 | jkroon | but could very well have affected earlier versions too. |
| 13:48.54 | Greenlight | I'm going to be very cautious about in hours reloads now |
| 13:49.17 | jkroon | i've reported and fixed probably around 10 reload issues this year to date. |
| 13:51.00 | *** join/#asterisk dxd828 (~dxd828@217.39.7.254) |
| 13:56.20 | jkroon | ok, problem is worse than that |
| 13:56.32 | jkroon | once i've reloaded state updates simply doesn't happen at all ... |
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| 14:27.48 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:27.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 14:30.57 | *** join/#asterisk Draecos (~Draecos@124-169-108-149.dyn.iinet.net.au) |
| 14:31.03 | mmyers | Hello. Asterisk server is behind NAT, and a road warrior also behind NAT. Both of them have the right ports forwarding. The sip info for the RW is canreinvite=no, nat=yes, qualify=yes. Yet, calls drop in 20-30 seconds. In the SIP debug I'm seeing the RW client is registering fine but I see the client address shows up at 6010@192.168.0.15 instead of RW's public IP. So I'm guessing because the communication is trying to go to an int |
| 14:31.06 | mmyers | How can I fix it? |
| 14:31.15 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 14:31.50 | mmyers | From: Unknown@asterisk server IP To: 6010@192.168.0.15 lol |
| 14:32.40 | mmyers | The forwarding port at the RW side is fine - but just the IP. So it shows up like 6010@192.168.0.15:9876 <- correct port, wrong IP. |
| 14:36.40 | [TK]D-Fender | You should not be forewarding remote clients |
| 14:37.08 | [TK]D-Fender | Undo that, confirm precisely what you have forwarded on our server side, and show us a call with SIP DEBUG enabled at CLI |
| 14:37.09 | [TK]D-Fender | ~pb |
| 14:37.10 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:37.12 | [TK]D-Fender | ^^^^^ |
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| 14:40.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:45.44 | mmyers | thank you [TK]D-Fender |
| 14:46.05 | mmyers | Let me remove the port forwarding on the RW's router |
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| 14:55.59 | mmyers | [TK]D-Fender: http://pastebin.ca/2495434 (Forwarded the SIP port and UDP ports for RTP on the Asterisk server) |
| 14:56.15 | mmyers | [TK]D-Fender: Also, this call was made after disabling forwarding on the RW's side |
| 15:01.55 | [TK]D-Fender | mmYou aren't debuggin your call OUT. |
| 15:02.05 | [TK]D-Fender | And I don't trust that is done properly in this picture |
| 15:03.23 | *** join/#asterisk insha (~insha@mobile-166-147-096-108.mycingular.net) |
| 15:05.57 | mmyers | What do you mean [TK]D-Fender? The call was made through the RW's client |
| 15:06.17 | mmyers | Only inbound calls drop |
| 15:06.19 | mmyers | Not outbound |
| 15:06.25 | mmyers | (sorry I forgot to mention that part earlier) |
| 15:06.52 | kaldemar | what does [6010] in sip.conf look like? |
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| 15:09.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:12.29 | mmyers | kaldemar: One sec |
| 15:12.43 | *** join/#asterisk jetlag (~jetlag@pool-71-168-196-225.cmdnnj.east.verizon.net) |
| 15:14.04 | mmyers | kaldemar: http://pastebin.ca/2495438 |
| 15:18.11 | Penguin | That could use some work. |
| 15:18.21 | mmyers | What work |
| 15:18.21 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:19.06 | Penguin | Firsr thing I noticed was both a deny and permit of 0.0.0.0 |
| 15:19.18 | kaldemar | Penguin: freepbx... |
| 15:19.27 | mmyers | kaldemar: yeah lol |
| 15:19.36 | Penguin | Why are we talking about it here, then? |
| 15:20.23 | mmyers | Because regardless of the FreePBX used to setup the initial system, the underlying nat issues that are part of Asterisk has been misconfigured :( |
| 15:20.49 | Penguin | I guess I'll buy that. |
| 15:22.06 | mmyers | Thank you Penguin |
| 15:24.13 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ryyiprwzmdftifly) |
| 15:24.14 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:26.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.87) |
| 15:27.02 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:29.36 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:30.12 | mmyers | So, my googling is also failing vehemently and I'm left with no solutions |
| 15:30.13 | mmyers | :( |
| 15:32.11 | navaismo | try ?natsettings in #freepbx |
| 15:32.42 | mmyers | navaismo: Thanks, I just did. Nada. |
| 15:33.09 | navaismo | I hate that stupid frog |
| 15:33.31 | mmyers | lol |
| 15:33.56 | navaismo | well basically you need to go to the asterisk sip settings |
| 15:34.09 | [TK]D-Fender | Found peer '6010' for '6010' from 99.XX.XX.XX:62712 Peer audio RTP is at port 192.168.0.15:40002 <--- Transmitting (NAT) to 99.XX.XX.XX:62712 ---> |
| 15:34.17 | [TK]D-Fender | this looks like an Asterisk RTP issue |
| 15:34.41 | navaismo | add a externip or externhost set your localnet etc then set your peer with nat or not depends on where is it |
| 15:34.47 | [TK]D-Fender | Peer is confirmed as NAT=YES, but still accepts the SDP IP for audio. |
| 15:35.18 | [TK]D-Fender | And chan_sip devs awake? |
| 15:35.21 | [TK]D-Fender | Any* |
| 15:35.37 | mmyers | [TK]D-Fender: 62712 is a SIP port set on RW's SIP client... |
| 15:35.49 | [TK]D-Fender | mmnot the port... the ***IP*** |
| 15:35.56 | [TK]D-Fender | mmyers: not the port... the ***IP*** |
| 15:36.09 | mmyers | Ah ok, yeah that's what I was wondering why it's showing the private IP |
| 15:36.17 | mmyers | navaismo: It's all set and ready to rumble |
| 15:36.57 | mmyers | [TK]D-Fender: That's what I see in my sip debugs as well, the 6010 peer that is the RW is connecting as a NAT IP and not showing the public IP, that's why I did some port forwarding on the router of the RW |
| 15:37.16 | Penguin | externaddr/externhost and localnet are both set? |
| 15:37.35 | Penguin | correctly? |
| 15:38.08 | mmyers | Penguin: Yup, lemme paste bin it forya |
| 15:38.32 | Penguin | And you're sure any SIP ALG crap on all involved routers has been disabled? |
| 15:39.27 | mmyers | Penguin: It's not on the RW side. And the Asterisk's edge router side I'm positive but lemme check |
| 15:39.58 | mmyers | Penguin: externip: http://pastebin.ca/2495446 |
| 15:40.47 | [TK]D-Fender | mmyers: Won't matter... * will try senting to a private IP that'll never route |
| 15:41.45 | [TK]D-Fender | [10:37]Penguinexternaddr/externhost and localnet are both set? <- * is advertising a QAN IP, the problem is it isn't ignoring the SDP IP as it should for the peer's audio |
| 15:41.50 | [TK]D-Fender | WAN* |
| 15:43.02 | Penguin | What is the asterisk version? |
| 15:44.44 | *** join/#asterisk jwww (~jmm@LPuteaux-156-14-28-187.w82-127.abo.wanadoo.fr) |
| 15:44.48 | jwww | Hello. |
| 15:44.58 | mmyers | Penguin: 1.8.20.0 |
| 15:45.15 | mmyers | Penguin: Also checked SIP stuff, it's disabled. But the edge router hasn't been rebooted and I just read that it should be rebooted. |
| 15:45.28 | Penguin | Can you do that now? |
| 15:46.30 | mmyers | Penguin: Amen, going for it |
| 15:46.42 | Penguin | The only time I had a problem with NAT, the RTP packets all had the private IP address of the remote device instead of the public address. I got rid of the router and the problem went away. |
| 15:47.09 | mmyers | This is an "Untangle" box as edge. |
| 15:47.09 | Penguin | It was a Cisco router. |
| 15:48.17 | [TK]D-Fender | * is choosing to use the SDP for this though, and basedon NAT=YES it shouldn't... still looks like it's a core issue |
| 15:49.57 | jwww | I'm very new to asterisk, I followed some basic tutorial, and got 2 softphones calling each others.but If I try to change the usernames in sip.conf, then calls become rejected.here is what I added to sip.conf : http://dpaste.com/1499096/ , can somebody help me please ? |
| 15:50.57 | mmyers | You think this SIP thing may fix it, [TK]D-Fender ? I sure hope so lol |
| 15:51.34 | navaismo | jwww when you change the usernames didi youalso change from the softphones? |
| 15:51.42 | [TK]D-Fender | mmyers: What "SIP thing"? To my eyes this looks like an Asterisk bug.... |
| 15:51.45 | Penguin | jwww: Get rid of the username lines in your peers. |
| 15:51.52 | Penguin | jwww: Those don't belong there. |
| 15:51.55 | navaismo | jwww, and make a sip reload? |
| 15:52.09 | jwww | Penguin && navaismo I try this. |
| 15:52.16 | [TK]D-Fender | jwww: PASTEBIN <- |
| 15:52.18 | [TK]D-Fender | ~pb |
| 15:52.18 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:52.20 | [TK]D-Fender | ^^^^ |
| 15:52.33 | [TK]D-Fender | jwww: that is only part of the picture BTW... |
| 15:52.39 | Penguin | jwww: The peer username is found between the square brackets. |
| 15:52.43 | elguero | [TK]D-Fender, mmyers: There have been some NAT cleanups and fixes after 1.8.20, you may want to check the ChangeLog for 1.8 |
| 15:53.12 | [TK]D-Fender | mmyers: You are a few releases behind... update to the latest 1.8 and attempt to replicate the error |
| 15:53.32 | mmyers | [TK]D-Fender: Alright, let's do that. Be back soon guys. |
| 15:54.43 | Penguin | elguero: Unless they broke it right before 1.8.20 so that it was good before that and bad in 1.8.20, I don't think that's the problem. I used asterisk <1.8.20 for a long time without that SIP problem. |
| 15:55.07 | Penguin | And I still do. |
| 15:55.43 | elguero | Penguin: It doesn't affect everyone, if I recall... but we did do some fixes for nat settings not being handled properly |
| 15:56.16 | Penguin | jwww: You changed the name in asterisk, and then you changed the name in the phone as well? |
| 15:56.18 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 15:56.28 | jwww | Penguin: umm still I can't make call . I update the config there : http://dpaste.com/1499109/ , I also located an error in asterisk's console I added it. |
| 15:56.46 | jwww | Penguin: I did. linphone say I registred successfully. |
| 15:56.47 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 15:56.50 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 15:57.06 | Penguin | jwww: You're calling extension "chris" but it does not exist. |
| 15:57.07 | [TK]D-Fender | jwww: you are not showing us your DIALPLAN |
| 15:57.17 | Chainsaw | hands Fender the megaphone |
| 15:57.20 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 15:57.21 | jwww | [TK]D-Fender: the one in extension.conf ? |
| 15:57.30 | Penguin | jwww: That's the only dialplan. |
| 15:57.34 | [TK]D-Fender | jwww: the call is accepted for the source.. and then rejected because your dialplan has nothing to process what they were requesting |
| 15:57.45 | Penguin | (unless you use some other one, of course) |
| 15:57.50 | [TK]D-Fender | jwww: extensions.conf |
| 15:58.06 | jwww | sorry for my noobness, but I just started this morning. |
| 15:58.16 | [TK]D-Fender | jwww: Doing OK so far... |
| 15:58.29 | Penguin | jwww: It's plain and simple. You've made a call to extension 'chris' which does not exist. Fix that or dial a different extension. |
| 15:58.30 | Chainsaw | jwww: Fender is a like a drill instructor. There will be a lot of yelling, but the end result is worth it. |
| 15:59.00 | [TK]D-Fender | jwww: [Dec 10 16:54:29] NOTICE[2221]: chan_sip.c:20325 handle_request_invite: Call from 'jmm' to extension 'chris' rejected because extension not found in context 'internal'. <- means exactly what it says. The caller is accepted, and they want to match "chris" in [internal] in your extensions.conf. There is no match for it there |
| 15:59.22 | Penguin | I don't even need to see the dialplan. |
| 15:59.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.87) |
| 15:59.39 | [TK]D-Fender | Chainsaw: No, there's pretty much no yelling left on my side anymore |
| 16:00.02 | [TK]D-Fender | Penguin: I'd like to see it just to see whathe did have at one point in time at least |
| 16:00.07 | jwww | I updated the pastebin : http://dpaste.com/1499118/ |
| 16:00.25 | [TK]D-Fender | jwww: exten=> _XXXX,1,Dial(SIP/${EXTEN}) <- that is a NUMERIC PATTERN |
| 16:00.38 | [TK]D-Fender | jwww: that is expecting to match a 4-digit number... not the word "chris" |
| 16:00.54 | Penguin | You have only one single extension in the internal context. And it requires a 4-digit extension. |
| 16:01.05 | Penguin | 4-digit cannot be chris. |
| 16:01.09 | jwww | I think I understand. |
| 16:01.30 | Penguin | jwww: I'd like to make a suggestion to you... |
| 16:01.34 | [TK]D-Fender | jwww: Generally you make numbered extension that specific dial your SIP peers by name |
| 16:02.00 | [TK]D-Fender | jwww: exten => 1000,1,Dial(SIP/chris,20) |
| 16:02.03 | Penguin | jwww: Abstract a numbered extension to reach your people. Dial their phone's name from the numbered extension. |
| 16:02.03 | jwww | Penguin: go on. |
| 16:02.15 | Penguin | jwww: Exactly what [tk]d-fender just said. |
| 16:02.19 | [TK]D-Fender | jwww: You';d then dial "1000" and it would call the SIP device named "chris" |
| 16:02.28 | Penguin | ~devices |
| 16:02.29 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 16:02.34 | [TK]D-Fender | jwww: for 20 seconds in this case |
| 16:03.42 | [TK]D-Fender | jwww: You don't dial what you think their name is in sip.conf. You dial what you dial and extensions.conf processes that and chooses what to do. Dial() is just one thing you can do in the process. |
| 16:03.45 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:03.52 | Penguin | jwww: For example, my name is Rob, so my extension is 762 (ROB on the keypad). Extension 762 dials my phone: exten => 762,1,Dial(SCCP/0000111222,30) |
| 16:04.54 | Penguin | Maybe Chris's extension could be 2474. |
| 16:05.13 | Penguin | exten => 2474,1,Dial(SIP/chris,26) |
| 16:05.38 | Penguin | 2474 is CHRI on the keypad, in case you didn't catch that. |
| 16:05.40 | jwww | I get it ! |
| 16:06.34 | *** join/#asterisk dash__ (~d45h@unaffiliated/dash-/x-7576607) |
| 16:08.11 | jwww | thanks for the explanations guys. |
| 16:08.57 | mmyers | Penguin: [TK]D-Fender elguero : Running into dependency issues, it might take a tad bit longer sorry. |
| 16:09.33 | Penguin | jwww: Now that you understand that part of it, you can used named extensions, but it makes it harder to dial it from a phone that doesn't use a keyboard. :/ |
| 16:10.55 | [TK]D-Fender | mmyers: NP, it'll take the time it takes and this is the process to do... |
| 16:11.13 | jwww | I just noticed that linphone client doesnt have the letters on the dialing numpad. |
| 16:11.17 | jwww | doh |
| 16:11.40 | Penguin | jwww: Now you see why numbered extensions make more sense. |
| 16:12.02 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 16:12.03 | Penguin | Most phones have 0-9. |
| 16:12.20 | Penguin | Some have 0-9 and A-D. |
| 16:12.23 | WIMPy | Most phones have 0-9 and * and #. |
| 16:12.24 | [TK]D-Fender | jwww: You can use a number like the one Penguin is suggestion which you can "think out" to get the number", or you can use a simple numbering scheme if you don't have too many people to track. If you have 3 people perhaps remembering that Chris is "10" isn't so hard. |
| 16:13.07 | Penguin | Abstract your extensions any way you want, but be sure it makes sense in your situation. |
| 16:13.19 | jwww | [TK]D-Fender: that's right, it's just for 6 peoples. |
| 16:13.34 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:14.20 | WIMPy | 6 peoples sounds like millions of people. |
| 16:14.34 | [TK]D-Fender | jwww: You can probably make speed-dial entries in your phone anyway for this... |
| 16:15.18 | [TK]D-Fender | jwww: And good old paper lists work very well too |
| 16:21.21 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:32.21 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:35.38 | *** join/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 16:37.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:43.27 | *** join/#asterisk hardwire (~hardwire@cl-101.anc-01.us.sixxs.net) |
| 16:46.11 | mmyers | Wow these updates are hardcore. About a gigs worth. |
| 16:46.14 | mmyers | Been too long I guess |
| 16:49.06 | [TK]D-Fender | that's clearly far more than just * |
| 16:50.54 | *** join/#asterisk outtolunc (~me@c-67-170-214-55.hsd1.ca.comcast.net) |
| 17:00.10 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 17:04.35 | mmyers | lol yup |
| 17:04.59 | mmyers | Mirror speed sucks too |
| 17:14.08 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:25.59 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 17:26.07 | polysics | hello |
| 17:26.34 | polysics | is there still no way aside form ChanSpy to play a sound to a bridged channel? |
| 17:27.09 | [TK]D-Fender | Bridge() <- |
| 17:27.43 | navaismo | I saw a failed demo of asterisk 12 trying to do that in the Asterisk youtube channel |
| 17:27.54 | polysics | we are having horrible performance issues with ChanSpy |
| 17:28.10 | polysics | [TK]D-Fender: won't that boot the current "other party"? |
| 17:29.22 | file | polysics, what is showing the indications that there are performance issues? |
| 17:29.38 | file | that was a poorly written sentence. |
| 17:31.02 | polysics | we have been timing the app with and without the chanspy. The app is fully instrumented with timing and chanspy setup takes up 90% of the non-talk time in an app. |
| 17:31.44 | Greenlight | We use ChanSpy extensively to whisper sounds to briged channels, and havn't noticed any issues... |
| 17:32.37 | file | Greenlight, I've been meaning to tell you... my approach for doing it in the new REST interface got in for 12 :) |
| 17:32.51 | Greenlight | Nice! |
| 17:32.57 | polysics | that's good news |
| 17:33.03 | polysics | but we are stuck with 11 here :D |
| 17:33.14 | Greenlight | The ChanSpy does feel a little hacky, but hey it does work :) |
| 17:33.21 | polysics | it does work |
| 17:33.37 | Greenlight | polysics: So, what that's actual issue you're getting ? |
| 17:34.03 | polysics | the ChanSpy originate + invocation seem to be very CPU intensive, and the whole system slows dow |
| 17:34.04 | mjordan | navaismo: which one? |
| 17:34.05 | polysics | *n |
| 17:34.46 | navaismo | mjordan, let me check my history |
| 17:35.50 | mjordan | if you're referring to david's demo at AstriCon, that was (a) playing a sound to a channel outside of a bridge; and (b) was a problem with the laptop setup, not Asterisk 12 |
| 17:36.06 | polysics | so we are exploring alternatives |
| 17:36.17 | polysics | I could make do with playing a sound to the conference bridge before joining the customer |
| 17:36.46 | polysics | and we might try that to see if the issue is really with ChanSpy |
| 17:36.46 | navaismo | mjordan, yep that one-->http://www.youtube.com/watch?v=D1yagQys0_0 |
| 17:37.18 | file | yeah that was his Mac being a Mac |
| 17:37.20 | mjordan | yeah, that isn't really what demo'ing what polysics is referrring to. And that functionality worked just fine - it was just a bad setup |
| 17:37.30 | navaismo | we all hate murphy's law |
| 17:38.33 | navaismo | "Mac being a MAc" <--- <3 |
| 17:39.10 | navaismo | so, asterisk 12 can playsounds to a bridgeid like in the demo? |
| 17:39.34 | navaismo | s/can/can't/ |
| 17:39.58 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:40.07 | polysics | asterisk 11 can't |
| 17:40.31 | file | Asterisk 12 can play sounds via the REST interface to a bridge you've created via the interface fine |
| 17:40.33 | Greenlight | polysics: But what is the issue you're seeing, and how have you verfied it? |
| 17:43.39 | polysics | Greenlight: the issue is "ChanSpy generates high CPU load", we verified it by removing the ChanSpy and running the same scenarios again, noting that performance improved |
| 17:44.05 | navaismo | virtual machine? |
| 17:44.35 | Greenlight | How did you originate the ChanSpy ? |
| 17:45.00 | Greenlight | And, for what duration, and define "high load" |
| 17:46.44 | Greenlight | I've a box I'm looking at now, which at some points of the day is originating more than 300 chanspys a minute, and I'm not seeing a performance difference that I could atribute to them |
| 17:48.06 | polysics | navaismo: iron box, 16 cores, 16Gb RAM |
| 17:48.11 | polysics | sorry |
| 17:48.15 | polysics | it's 48 Gb :D |
| 17:48.42 | polysics | originate is done with this: https://gist.github.com/JustinAiken/c63f41be77addb5dc45a |
| 17:48.43 | mjordan | what formats are on the channels in question? |
| 17:48.47 | *** join/#asterisk imox (~imox@91-65-182-111-dynip.superkabel.de) |
| 17:48.52 | polysics | and this is the extension: https://gist.github.com/JustinAiken/bf8cae569cb0a737bf2f |
| 17:48.58 | *** join/#asterisk imox (~imox@91-65-182-111-dynip.superkabel.de) |
| 17:49.01 | polysics | the target is a channel that is bridged on a confbridge |
| 17:49.14 | polysics | goal is for only the agent to hear a "beep" when a user is put into the room |
| 17:49.35 | polysics | we use this to fast-join dialed users instead of dialing agents |
| 17:49.42 | Penguin | 48 gigabits of RAM isn't all that great. |
| 17:50.11 | Greenlight | Penguin never misses a thing |
| 17:50.16 | Penguin | You know it! |
| 17:50.39 | Penguin | And now to check the pastes. |
| 17:50.41 | Greenlight | polysics: Let me check our whisper code... two secs |
| 17:50.45 | polysics | well, it should be sort of enough |
| 17:51.23 | Greenlight | Hmm... we actually orignate to an extension that does Playback, rather than the playback application. |
| 17:51.34 | Greenlight | I wrote this a few years back now; perhaps there was a reason |
| 17:51.45 | Greenlight | Ahh I see |
| 17:51.54 | Greenlight | We originate to the ChanSpy application |
| 17:52.01 | Greenlight | SO, we do it the other way around to you |
| 17:52.01 | [TK]D-Fender | Penguin: Always got your .02 cents worth to throw in ;) |
| 17:52.07 | [TK]D-Fender | </verizon> |
| 17:52.11 | Penguin | snickers |
| 17:52.17 | Greenlight | Although not sure why that should make a difference |
| 17:52.36 | polysics | Greenlight: maybe it does, always happy to try. What would happen in this case? |
| 17:53.02 | polysics | the extension has Playback() with the specified file |
| 17:53.11 | Penguin | [tk]d-fender: I see that written on various forums all the time. |
| 17:53.16 | polysics | and the originate has Chanspy() with which params? |
| 17:53.17 | Greenlight | Also, is your application running on the same box ? |
| 17:53.21 | polysics | yeah |
| 17:53.25 | polysics | the Ruby app |
| 17:53.34 | Greenlight | Are you *sure* it's not a problem there ? |
| 17:53.58 | file | I will say that for a short sound file you'll have a channel creation/destruction happen close to eachother |
| 17:53.59 | Greenlight | http://pastebin.com/RSfZuy8k <-- A snipped of what we submit |
| 17:53.59 | polysics | can't be *sure* but I am trying solutions to differentially diagnose it |
| 17:54.17 | Greenlight | We also play a beep |
| 17:54.46 | Greenlight | polysics: What sort of scale are we talking here... does your box have >500 channels? Are you doing this ChanSpy >100 times a minute> |
| 17:54.52 | mmyers | [TK]D-Fender: You won't believe this, there's a segfault now lol |
| 17:55.13 | [TK]D-Fender | mmyers: I have no issue believing that.... |
| 17:55.21 | [TK]D-Fender | mmyers: Now what the fix is... is another matter |
| 17:55.35 | [TK]D-Fender | mmyers: I'd make sure your upgrade job was clean... |
| 17:55.42 | mmyers | [TK]D-Fender: Yeah, I'm revisiting the upgrades.. |
| 17:57.03 | polysics | Greenlight: 100ish channels up |
| 17:57.38 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 17:58.14 | polysics | and 40 Chanspy a minute |
| 17:58.17 | Greenlight | polysics: I've a box I'm looking at here, about same spec as yours; I was pushing over 1000 channels earlier today. That was doing 200+ ChanSpy whispers a minute at peak. It didn't miss a beat; Asterisk 11. |
| 17:58.52 | Greenlight | So my gut feeling is that it's not a ChanSpy issue |
| 17:58.53 | polysics | how do you run logic on those? Dialplan only? |
| 17:59.03 | file | ^^^ hearing stuff like that makes me happy! |
| 17:59.14 | Greenlight | A little dialplan, mostly AMI |
| 17:59.21 | Greenlight | file: Me too! :) |
| 17:59.35 | file | Greenlight, did you get those issues long ago sorted? |
| 17:59.56 | Greenlight | file: Ditched app_queue as a mechnism of routing calls to agents, and all was good :) |
| 18:00.06 | file | Greenlight, ah! cool |
| 18:00.12 | Greenlight | We not originate calls to a Wait(), and then use the AMI to fire a Bridge |
| 18:00.16 | Greenlight | *now |
| 18:00.30 | Greenlight | Seriously impressive how well it's scaling |
| 18:01.22 | file | that's actually not a bad approach |
| 18:01.40 | file | it's sorta what ARI does |
| 18:01.44 | Greenlight | Yea, lets me do all the "logic" off on another box, and let Asterisk handle the basics |
| 18:02.33 | Greenlight | My only concern was any delay between outbound calls hitting that Wait() and it bridging, but all the measurements I can get are showing it mostly <15ms |
| 18:03.15 | polysics | so your agents are sitting in a Wait()? |
| 18:03.33 | Greenlight | Well, they are as well, yes. |
| 18:04.13 | polysics | and when they finish a call they go back to the Wait()? |
| 18:04.21 | polysics | I might even want to look at changing the whole architecture |
| 18:04.37 | Greenlight | Indeed |
| 18:04.43 | Greenlight | We used to use ConfBridge, and app_queue |
| 18:04.46 | polysics | but first I think I should just try swapping Chanspy and Playback |
| 18:07.18 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:08.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:10.37 | polysics | what about playing sound into a Confbridge using an Originate? |
| 18:11.25 | Penguin | I've never had any problem doing that. |
| 18:15.49 | polysics | Penguin: got an example, please? I think i am a little lost in teh order of events |
| 18:17.11 | Penguin | channel originate Local/random-conf-extension@conferences-context application Playback my-sound |
| 18:18.11 | Penguin | You could use Originate() in a similar way. |
| 18:22.23 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:24.00 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:24.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:31.34 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 18:38.24 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 18:43.58 | mmyers | All done! |
| 18:44.10 | mmyers | [TK]D-Fender: Penguin here's the new paste bin with the call: http://pastebin.ca/2495491 |
| 18:44.16 | mmyers | Same issue, 21 second call drop. |
| 18:44.30 | mmyers | Upgraded, removed SIP ALG and everything else we spoke of |
| 18:45.22 | [TK]D-Fender | Peer audio RTP is at port 192.168.0.15:40004 |
| 18:45.37 | [TK]D-Fender | Still see * IDing the wrong address. This looks like incorrect processing... |
| 18:45.43 | [TK]D-Fender | I'd file a bug for it. |
| 18:45.56 | mmyers | What shall I include? |
| 18:46.18 | file | that's what is in the SDP |
| 18:46.19 | [TK]D-Fender | call with SIP debug, sip.conf entries, etc |
| 18:46.40 | mmyers | My configs are all correct? Jeez. Unbelievable that it's a bug .. FML |
| 18:46.42 | [TK]D-Fender | file: SDP ofers the private address we see but the peer is clearly nat=yes so it should not be taking that address |
| 18:46.43 | file | if the NAT settings are enabled it'll change when audio actually flows to the source IP address/port |
| 18:47.00 | file | [TK]D-Fender, it takes the address for RTP until it gets RTP, cause there is nothing else to do |
| 18:47.15 | [TK]D-Fender | file: Peer audio RTP is at port 192.168.0.15:40004 <--- Transmitting (NAT) to 99.xx.xx.xx:62712 ---> |
| 18:47.21 | mmyers | file: There was two way audio |
| 18:47.34 | [TK]D-Fender | file: Well it has the source... it should never say that it's private just to change it's mind later |
| 18:47.39 | file | yes, it's sending the signaling to the source IP address and port for signaling |
| 18:47.41 | [TK]D-Fender | its* |
| 18:47.47 | file | the source of signaling is NOT the source of RTP |
| 18:48.00 | file | you could best effort use the same IP address, but if behind NAT that is still useless until they send you media |
| 18:48.01 | Penguin | If there is two-way audio, what was the problem? |
| 18:48.11 | mmyers | Penguin: Call drops in 20 or so seconds |
| 18:48.13 | [TK]D-Fender | file: Peer audio RTP is at port 192.168.0.15:40004 <- saying this was never really right though... it's not supposed to think about it... |
| 18:48.13 | Penguin | I thought this was an audio problem. |
| 18:48.13 | file | the problem is that the client never sent an ACK for the 200 OK |
| 18:48.18 | mmyers | Only outbound, all inbound calls work fine. |
| 18:48.49 | [TK]D-Fender | mmyers: Have you checked the router at the remote side? In case it's interfering in some way... |
| 18:49.13 | mmyers | [TK]D-Fender: Yup. The SIP ALG that Penguin mentioned got disabled. All ports sip and ftp are being forwarded.. |
| 18:49.23 | Penguin | That's wrong. |
| 18:49.29 | Penguin | You were told to stop doing forwarding. |
| 18:49.38 | mmyers | Stop forwarding on the RW's side |
| 18:49.40 | mmyers | And that's done |
| 18:49.44 | [TK]D-Fender | mmyers: Server: FPBX-2.8.1(1.8.20.0) |
| 18:49.45 | Penguin | "all ports" |
| 18:49.53 | [TK]D-Fender | mmyers: That is not looking like you upgraded as requested |
| 18:50.10 | [TK]D-Fender | mmyers: and FTP has nothing to do with this |
| 18:50.13 | mmyers | mmyers: Where did you see that? Wait up... |
| 18:50.18 | Penguin | Never forward to the phone. If you are not forwarding to the phone, good. |
| 18:50.35 | [TK]D-Fender | [13:44]mmyers[TK]D-Fender: Penguin here's the new paste bin with the call: http://pastebin.ca/2495491 |
| 18:50.40 | [TK]D-Fender | line 249 |
| 18:50.58 | Penguin | or 48 |
| 18:51.06 | mmyers | Very weird! |
| 18:51.10 | mmyers | I'm on the latest Asterisk |
| 18:51.43 | mmyers | buendia*CLI> core show version Asterisk 11.6.0 |
| 18:52.12 | [TK]D-Fender | mmyers: I didn't suggest you jump branches there |
| 18:52.15 | Penguin | Oh boy. |
| 18:52.23 | [TK]D-Fender | mmyers: and FreePBX 2.8 doesn't support * 11 |
| 18:52.29 | Penguin | shakes his head |
| 18:52.38 | mmyers | Wait wait, I did the yum update |
| 18:52.45 | mmyers | Elastix does all the update in the backend apparently |
| 18:53.00 | mmyers | FreePBX isn't the distro, it's running within elastix |
| 18:53.46 | mmyers | [TK]D-Fender: FreePBX shows 2.8.1.4 |
| 18:54.05 | mmyers | Here I thought I was winning doing everything right! |
| 18:54.06 | mmyers | lol |
| 18:57.30 | mmyers | Elasix 2.4.0-11, FreePBX 2.8.1-17, Asterisk 11.6.0-1 |
| 18:58.21 | *** join/#asterisk serafie1 (~erin@nat/digium/x-gqclobwcgdtnmpao) |
| 18:59.04 | Penguin | Was the problem the same before you upgraded? |
| 18:59.23 | mmyers | yup |
| 18:59.27 | mmyers | Yup Penguin |
| 19:01.57 | *** join/#asterisk vlad_sta_ (~vlad_star@77.50.56.177) |
| 19:06.00 | mmyers | In a long time, my call just dropped at 19 seconds.. |
| 19:06.22 | mmyers | Normally its at 20+ |
| 19:07.10 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 19:12.02 | *** join/#asterisk dxd828 (~dxd828@94.116.77.190) |
| 19:12.26 | mmyers | [TK]D-Fender: Oh, new thing. Inbound calls that were working still continue to work but now there's a new thing. If the caller hangs up, or the RW hangs up. The call still continues. |
| 19:15.46 | [TK]D-Fender | mmyers: 2.8 is still not made for * 11... |
| 19:16.04 | [TK]D-Fender | mmin sip_general_custom.conf make sure you have "directmedia=no" |
| 19:16.11 | [TK]D-Fender | mmyers: in sip_general_custom.conf make sure you have "directmedia=no" |
| 19:16.14 | mmyers | one sec [TK]D-Fender |
| 19:17.32 | mmyers | [TK]D-Fender: done |
| 19:17.41 | [TK]D-Fender | mmgo retest |
| 19:17.48 | mmyers | ok |
| 19:19.33 | mmyers | [TK]D-Fender: no dice |
| 19:19.35 | mmyers | :( |
| 19:21.33 | mmyers | X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 |
| 19:22.03 | mmyers | set_destination: Parsing <sip:6010@192.168.0.15:62712;ob> for address/port to send to set_destination: set destination to 192.168.0.15:62712 |
| 19:22.08 | mmyers | grr it's still showing internal IP |
| 19:22.43 | [TK]D-Fender | mmyers: Provide configs & full SIP DEBUG and file a bug report... |
| 19:23.08 | mmyers | Ok, thanks [TK]D-Fender for sticking it out with me |
| 19:28.21 | file | from the pastebins I saw previously it does not appear to be Asterisk |
| 19:29.35 | [TK]D-Fender | file: * keeps setting destinations for private there.... am I missing something? |
| 19:30.04 | file | it may set the destinations, but that doesn't mean it sends there |
| 19:30.18 | file | the sip debug clearly shows it sending to the source IP address and port for signaling |
| 19:33.03 | *** join/#asterisk Dumby (~dumby@204.246.140.162) |
| 19:36.37 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 19:36.59 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:37.54 | mmyers | Just spent over 3 mins looking for the bug tracker (no offence digium brass) |
| 19:38.26 | file | https://issues.asterisk.org/ |
| 19:41.05 | *** join/#asterisk bchamberlain (~brian98@unaffiliated/brian98) |
| 19:43.14 | mmyers | Yeah file, it doesn't work via safari for some reason. Used Chrome |
| 19:44.03 | mmyers | And now my username from eons ago doesn't work, email isn't with me anymore. |
| 19:44.05 | mmyers | Just great. |
| 19:47.48 | mmyers | What if my domain that I registered the email with is no more.. I kiss goodbye to my account? :( |
| 19:48.46 | navaismo | i guess its easier to create new one |
| 19:49.58 | mmyers | Yeah, I'm a packrat of a special kind. Hate throwing away old accounts. Like my ICQ. |
| 20:02.16 | *** join/#asterisk dxd828 (~dxd828@212.183.128.250) |
| 20:08.19 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:09.42 | Qwell | mmyers: why would you throw that out? |
| 20:09.52 | Qwell | 18082910, sup? |
| 20:10.11 | mmyers | Qwell: I wouldn't. I just emailed the JIRA admins to help me restore my account so I can file this bug. |
| 20:10.27 | Qwell | mmyers: I meant the ICQ account. :p |
| 20:10.57 | mmyers | Qwell: It will go down in the books of history one day for something or the other, and I want to be part of it. Weirdly enough. :P |
| 20:11.18 | Qwell | oh, so you didn't throw it away. got it |
| 20:12.12 | mmyers | Wouldn't want to! |
| 20:15.44 | *** join/#asterisk serafie (~erin@nat/digium/x-wsgemcrnrfdygdvo) |
| 20:32.55 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 20:36.34 | *** join/#asterisk serafie (~erin@nat/digium/x-jkjjebiuczlalrvm) |
| 20:41.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.75) |
| 20:52.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.100) |
| 20:54.12 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:08.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.100) |
| 21:13.17 | Kobaz | System uptime: 1 year, 36 weeks, 1 day, 22 hours, 39 minutes, 16 seconds |
| 21:13.18 | Kobaz | wow |
| 21:13.27 | Kobaz | i think that's my longest running system |
| 21:25.10 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 21:26.56 | [TK]D-Fender | checkout time, BBIAB |
| 21:28.56 | polysics | anyone here uses SIPp, please? |
| 21:29.15 | Max_E | ?ask |
| 21:29.19 | Max_E | ~ask |
| 21:29.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:29.25 | polysics | I know it is OT, but I am trying to understand if it is feasible to have a call that runs indefinitely |
| 21:29.37 | polysics | ie. SIPp pretending it is an agent in a confbridge |
| 21:36.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.100) |
| 21:40.16 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 21:43.13 | Max_E | polysics, amm not sure about the media/rtp |
| 21:43.51 | Max_E | pabelanger, the other day recommend somthing IIRC puppy/zippy or something |
| 21:44.26 | pabelanger | pabelanger, for what? |
| 21:45.14 | Max_E | for use with sipp |
| 21:45.18 | Max_E | or im crazy |
| 21:45.28 | Max_E | not sure if that was 2 months ago |
| 21:46.23 | polysics | that sounds interesting |
| 21:47.37 | Max_E | digging in logs |
| 21:51.01 | pabelanger | Max_E, what do you want to do |
| 21:51.52 | leifmadsen | points at sippy_cup @ https://github.com/mojolingo/sippy_cup |
| 21:52.02 | leifmadsen | makes SIPp easier to work with at least |
| 21:52.17 | polysics | I built part of that :P |
| 21:52.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:52.28 | leifmadsen | but you can just reflect RTP or whatever with it; just configure SIPp to be on a super long call and setup 1 call |
| 21:52.56 | leifmadsen | I admit to not having actually done this, but it seems reasonable :) |
| 21:53.43 | polysics | I think RTP echo plus simply setting up the first half of an UAS scenario should work |
| 21:53.47 | polysics | ie. never actually hanging up |
| 21:55.19 | Max_E | cant find anything only a reference to https://github.com/mojolingo/sippy_cup |
| 21:55.36 | polysics | then it's us : |
| 21:55.39 | polysics | :D |
| 21:55.52 | *** join/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 21:58.02 | Max_E | hmm nope cant find it, and sippy_cup only send dtmf |
| 21:58.25 | Max_E | argh i need to trust less in my memory |
| 21:59.42 | polysics | no problem, I think RTP reflect + waiting will do it |
| 22:00.55 | Max_E | brb going to cook |
| 22:02.29 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 22:11.52 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 22:12.57 | *** part/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 22:15.24 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 22:22.54 | *** join/#asterisk Scorp1us (~jason@pool-98-117-60-146.bltmmd.fios.verizon.net) |
| 22:23.20 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 22:23.37 | Scorp1us | hi all my asterisk provider moved me to another server and now my AGI can't dial out (via SIP) |
| 22:23.57 | Scorp1us | I am wondering where I should start looking? The scripts are there |
| 22:24.13 | [TK]D-Fender | that tells us nothing really |
| 22:24.19 | Scorp1us | My sip extension is registered |
| 22:24.21 | Kobaz | nope, not a whole lot |
| 22:24.25 | [TK]D-Fender | "Asterisk provider" isn't really a "thing". |
| 22:24.30 | Kobaz | ~details |
| 22:24.31 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:24.51 | [TK]D-Fender | we'd need to see you failed calls with AGI debug for starters |
| 22:24.58 | [TK]D-Fender | then at some point possibly the code itself. |
| 22:25.05 | Scorp1us | well, here's the problem, it used to work, now it doesn't. |
| 22:25.13 | [TK]D-Fender | Assuming that "AGI" is even part of the reason that you aren't calling out. |
| 22:25.22 | Scorp1us | When I put a call file, I see asterisk try to use it |
| 22:25.34 | Kobaz | Scorp1us: generalities aren't going to help at all |
| 22:25.34 | Scorp1us | but nothing shows up in SIP as a dialed call. |
| 22:25.35 | [TK]D-Fender | that is another completely separate matter |
| 22:25.38 | [TK]D-Fender | we need debug |
| 22:25.47 | [TK]D-Fender | And code to match |
| 22:26.04 | [TK]D-Fender | ~pb |
| 22:26.04 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:26.06 | [TK]D-Fender | ^^^^ |
| 22:26.11 | Kobaz | Scorp1us: it's like going to a doctor and saying "i feel sick", and not telling the doctor you ate a box of nails yesterday |
| 22:26.27 | Kobaz | Scorp1us: be *specific* |
| 22:26.37 | Scorp1us | ok i just didn't know where to begin |
| 22:26.43 | Kobaz | begin with what tk asked for |
| 22:26.47 | Scorp1us | now that you said that Ill get that collected |
| 22:26.48 | Kobaz | console logs of your failed call |
| 22:29.01 | Scorp1us | http://pastebin.com/z3GTWNfF |
| 22:29.25 | Scorp1us | that's call file and the console |
| 22:30.13 | Scorp1us | actually, those might not even match. arg. so lost |
| 22:30.28 | [TK]D-Fender | they don't |
| 22:30.30 | Kobaz | you're going to have to start showing your agi code, this startservice.py |
| 22:30.34 | [TK]D-Fender | not yet |
| 22:30.55 | [TK]D-Fender | lets just see something that matches at all with teh level of debug requested |
| 22:30.57 | Kobaz | and yeah, show the actual log |
| 22:31.22 | newtonr | someone could tell him how to get those logs :D https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 22:31.37 | [TK]D-Fender | newtonr: he's clearly grabbing output already |
| 22:31.49 | [TK]D-Fender | newtonr: Just not paying attention that it was the RELEVANT call |
| 22:32.34 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 22:33.40 | Kobaz | so sad |
| 22:33.42 | Kobaz | root 19610 0.0 1.4 68060 59328 ? Ss 2012 180:55 AgiDaemon |
| 22:33.48 | Kobaz | i have to restart it for an upgrade |
| 22:33.52 | Kobaz | it's been running since 2012 :( |
| 22:37.04 | newtonr | Scorp1us, you don't have the "DEBUG" message type enabled to your console (logger.conf), and may not have the debug level turned up (asterisk.conf). You probably want debug and verbose both turned up to 5, and either present on your console or the log file that you grab from. |
| 22:37.26 | Kobaz | you dont need DEBUG |
| 22:37.29 | Kobaz | just verbose |
| 22:37.57 | newtonr | For what? |
| 22:38.12 | Kobaz | for Scorp1us dialplan/agi problem |
| 22:38.20 | Kobaz | debug is going to spew lots of useless info related to what his issue is |
| 22:38.38 | Kobaz | unless he's running into an asterisk bug (unlikely), all he needs is core set verbose 3 |
| 22:41.29 | Scorp1us | ok. let me start here, if I have a call file in outgoing, how do I get asterisk to process it? |
| 22:41.37 | newtonr | Eh, I find it really helpful to see what is really going on and I always rather have more info than not enough. |
| 22:41.45 | Scorp1us | because right now its just sitting there |
| 22:42.19 | [TK]D-Fender | Scorp1us: it will process it immediately unless it is POST-DATED |
| 22:43.01 | Kobaz | newtonr: in this case, the debug output is just going to cause extra work in ignoring it when reading the log file |
| 22:43.11 | Kobaz | netmax: debug output is useful in many cases, this isn't one of them |
| 22:43.18 | Max_E | or maybe someone need to tell him that is not the normal dialplan behavior dialing using AGIS |
| 22:43.29 | Scorp1us | well there is a call file and its not processing it. all i see are register messages |
| 22:43.33 | Kobaz | Max_E: sure it is |
| 22:43.43 | Max_E | O_o |
| 22:43.43 | Kobaz | people use AGI all the time for call control |
| 22:43.47 | *** join/#asterisk [sr] (~kvirc@213.228.163.73) |
| 22:43.50 | [sr] | hi |
| 22:43.53 | Kobaz | myself included |
| 22:43.54 | [sr] | bug: https://issues.asterisk.org/jira/browse/PRI-152 |
| 22:44.11 | Max_E | people use it* |
| 22:44.19 | [sr] | says its fixed in 1.4.13, fact is that i have latest 1.4.14 and it still happens, any idea? |
| 22:44.21 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:44.21 | Kobaz | newtonr: whoops prefixed the wrong person... debug output is useful in many cases, this isn't one of them |
| 22:44.25 | Max_E | LOL @kobaz |
| 22:44.54 | Kobaz | [sr]: you should probably think about upgrading 1.4 is quite EOL'd |
| 22:45.14 | [sr] | Kobaz: pri version not asterisk |
| 22:45.20 | Kobaz | oh |
| 22:45.27 | Max_E | based on that we can stop blamig people asking on freepbx here because a lot of people use it to dial out and in the end it use asterisk dialplan logic |
| 22:45.31 | Kobaz | right |
| 22:45.46 | Max_E | ¬¬ |
| 22:45.58 | Kobaz | [sr]: i would say post back on the bug and say it's still happening. contact the person who wrote the fix |
| 22:46.20 | [TK]D-Fender | Scorp1us: how are they being placed there? |
| 22:46.31 | Kobaz | Max_E: depends what you're doing |
| 22:46.54 | Max_E | nein |
| 22:47.01 | Kobaz | Max_E: writing anything more than Dial()... Voicemail()... in dialplan logic is bloody nightmare... so might as well use a general purpose scripting language |
| 22:47.40 | Max_E | im not telling about that im just talking about what "many people use" and that convert in normal stuff |
| 22:47.57 | Max_E | so freepbx apply perfeclty like the agi dialing |
| 22:48.03 | Kobaz | many people use all kinds of things |
| 22:48.32 | Max_E | dont you say, i guess you are missing my point... |
| 22:48.50 | Kobaz | probably |
| 22:49.03 | Max_E | lets back to the issue |
| 22:49.04 | tm1000 | little bit of a language barrier isn't helping either Max_E :-) |
| 22:49.04 | Kobaz | my point is... best tool for the job |
| 22:50.35 | Scorp1us | my script moves them there |
| 22:50.54 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 22:51.04 | Scorp1us | I'm prepareing anoother pastegin |
| 22:52.02 | Max_E | ~freepbx |
| 22:52.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:53.27 | Scorp1us | ok so here is. it's right this time. |
| 22:53.29 | Scorp1us | http://pastebin.com/dZQBPw2d |
| 22:53.42 | Scorp1us | sorry about before |
| 22:54.56 | [TK]D-Fender | Scorp1us: I see the confirmation of it attempting the call... and you not sticking around long enough to show it ANSWER |
| 22:55.19 | Scorp1us | hrm ok |
| 22:55.34 | [TK]D-Fender | Scorp1us: so it IS taking your call files from the look of it |
| 22:56.22 | [TK]D-Fender | Scorp1us: It would probably be a very good this to have full SIP DEBUG enabled.... |
| 22:56.27 | [TK]D-Fender | "sip set debug on" <- |
| 22:59.26 | Kobaz | yeah make sure you have ample wait time |
| 22:59.38 | Kobaz | if the other side doesnt answer then your own side will terminate the call |
| 23:00.09 | Scorp1us | i did type that |
| 23:00.33 | Scorp1us | SIP Debugging re-enabled |
| 23:00.46 | outtolunc | probably missing timeout |
| 23:04.05 | Kobaz | Scorp1us: if your own asterisk is sending a BYE, then your timeout/wait is not long enough |
| 23:05.26 | Scorp1us | hmm. |
| 23:08.23 | [TK]D-Fender | We don't see the call out yet... |
| 23:08.27 | [TK]D-Fender | or the end |
| 23:13.53 | outtolunc | sorry didn't look at pb (i have too much AMI on the brain).. does he even pass register.. (didn't look like it) if he is now on a new asteris box, did the account get updated with the new ip? |
| 23:16.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 23:17.07 | Scorp1us | what do you mean? |
| 23:17.31 | Scorp1us | because the IP did change, that's the only thing that should have changed |
| 23:18.58 | outtolunc | well if you term provider (callcentric) was setup with IP based auth.. then they need to be informed of the new asterisk box ip |
| 23:19.42 | Scorp1us | well the device is registered |
| 23:21.46 | Scorp1us | my collectdigits.py script (dial, play message, collect digit) isn't dialing out either |
| 23:21.53 | Kobaz | registered doesn't always mean you can make calls |
| 23:21.56 | outtolunc | i've never used callcentric.. i do not know if they allow open reg .. if your sip entry for them has a user/pass, then you are not using ip auth. |
| 23:21.57 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 23:22.24 | *** join/#asterisk nickfennell_ (~nickfenne@unaffiliated/nickfennell) |
| 23:24.03 | Scorp1us | yes, they use a secret |
| 23:24.11 | Scorp1us | my new ip is registered |
| 23:24.56 | outtolunc | then it is something else.. have you done an agi debug (and retest) yet? (sorry, wasn't following the pb's) |
| 23:41.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 23:48.57 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 23:49.55 | phix | Hey, is there a way to programmatically add in a sip account and local extension? |
| 23:50.29 | phix | Basically I want to modify an Add User script so when I add a user onto my server it also creates an extension for them and assigns them to a phone |
| 23:50.42 | WIMPy | Do it |
| 23:51.04 | phix | so I need to script it myself? or is there an API I can use to interact with asterisk and tell it to do that? |
| 23:51.19 | phix | without having to hack around configuration files? |
| 23:51.28 | WIMPy | You have to do it yourself. |
| 23:51.37 | Chainsaw | phix: Well, either you hack around configuration files, or you serve a configuration file equivalent from a database. |
| 23:51.42 | WIMPy | You could use dynamic realtime. |
| 23:51.45 | Chainsaw | phix: (And that is known by the odd name of "realtime") |
| 23:52.13 | phix | ok so there is a way I can use a database as a backend for configuration files? |
| 23:52.30 | WIMPy | Yes. |
| 23:52.36 | phix | Does asterisk cache it incase the database isn't available? |
| 23:52.47 | phix | or can i use sqlite or something? |
| 23:52.47 | WIMPy | For risks and side effects ask your doctor. |
| 23:52.49 | Chainsaw | phix: Yes, "realtime". But I would start with flat config files and convert that setup, to ensure any problems you experience are not simply due to the database setup. |
| 23:53.06 | Chainsaw | phix: I'd stick the MySQL server on the same box and not tempt fate. |
| 23:53.10 | phix | ok so I can use multiple flatfiles and import them? |
| 23:53.26 | Chainsaw | Or get over your irrational fear of configuration files, of course. |
| 23:53.28 | phix | s/MySQL/PostgreSQL/ |
| 23:53.54 | Max_E | or use FreePBX |
| 23:54.10 | phix | Chainsaw: no the issue I had is I don;t want to edit the main extensions.conf or sip.conf incase it puts an entry into the wrong spot, but you are saying I can have multiple conf files and import them? |
| 23:54.11 | Max_E | it use MySQL |
| 23:54.11 | WIMPy | You can generate files or include other files. You can even include configuration generated by a script being called when reading the config. |
| 23:54.30 | Chainsaw | phix: You can #include, yes. |
| 23:54.57 | phix | WIMPy: ok I like that better, I just didn't want to hack a single conf file, but importing multiple ones sounds a bit safer, unless there are multiple sip / extension entries |
| 23:55.03 | Chainsaw | phix: This is why I have per-queue config files. Make the problem smaller and keep it contained. |
| 23:55.05 | phix | that are the same that is |
| 23:55.28 | phix | Chainsaw: ok, do you have any examples of this? |
| 23:56.01 | WIMPy | You can even #include myconfig/*.conf or the like. |
| 23:56.04 | phix | also, do I need to reload asterisk everytime I change a conf file that is imported? |
| 23:56.16 | WIMPy | yes |
| 23:56.21 | WIMPy | Well, not the whole Asterisk. |
| 23:56.22 | Chainsaw | phix: You can reload the specific components that you updated. The SIP stack, the queue rules, etc. |
| 23:56.30 | WIMPy | You can reload the part that was changed. |
| 23:56.31 | phix | and is there a config file checker that I can run on the new conf first before getting asterisk to use it? |
| 23:56.44 | phix | WIMPy: ok cool |
| 23:56.56 | Chainsaw | You have no sense of adventure. I can tell. |
| 23:57.02 | Chainsaw | No, I'm not aware of such functionality. |
| 23:57.13 | phix | Not when this is a clients machine :) If I break it they break me :) |
| 23:57.25 | Scorp1us | crazy question. how do you do agi debug? |
| 23:57.30 | WIMPy | No risk, no fun. |
| 23:58.41 | newtonr | Scorp1us, "agi set debug on" |
| 23:59.27 | Scorp1us | ok i did that. is till don't see agi output |