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00:42.24 | pclerie | Hello all! I am new to Asterisk and inevitably I need a little bit of help. |
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00:44.08 | kunwon1 | pclerie: ask your questions and those who see them will likely do their best to answer |
00:44.23 | pclerie | I've set up a test system and I'm using a GrandStream 1625 IP phone to test. The phone appears to work only some of the time. Just after a restart of Asterisk I can receive and place call. After a while, I can receive but every outgoing call fails. |
00:44.49 | kunwon1 | do you have sip qualify set? |
00:45.03 | pclerie | As a matter of fact I do. |
00:45.13 | kunwon1 | i think it's a different setting i'm thinking of, actually |
00:45.15 | kunwon1 | hang on |
00:45.40 | pclerie | I replaced the GS with a Polycom 331 and that appears to work correctly. |
00:46.09 | pclerie | The system on Ubuntu 16.04 and it's version 13.1. |
00:46.57 | Samot | Sounds like an issue with the phone not handling the auth properly for the call |
00:47.51 | pclerie | samot: why would it work some of the time then? |
00:48.02 | Samot | I don't know. |
00:48.21 | Samot | There could be various reasons. Can't really tell without seeing it fail. |
00:50.14 | kunwon1 | pclerie: some tips- run asterisk -r -vvvvv and watch the console to see if there is any evidence - and after the problem manifests and you're unable to dial, see if you see anything on the console |
00:50.24 | kunwon1 | and is this pjsip or chan_sip? |
00:51.52 | pclerie | kunwon1: chan_sip. Asterisk won't start with pjsip. But that's another story for now. |
00:52.20 | kunwon1 | well, chan_sip is much easier to troubleshoot |
00:52.29 | kunwon1 | sip show peer XXX where XXX is your peer |
00:52.36 | kunwon1 | when the problem manifests |
00:52.43 | kunwon1 | that could give hints as well |
00:53.06 | kunwon1 | and if you want to use pjsip, use the version bundled with asterisk |
00:53.11 | kunwon1 | using separate pjsip is deep magic |
00:53.36 | pclerie | kunwon1: I've used the console with up to 5 v's and sip debug (or something). I've seen the registration packets come and go but I'm not too sure yet about the details of that. |
00:53.59 | kunwon1 | sip debug is less helpful to people who aren't intimately familiar with sip |
00:54.20 | kunwon1 | the console output should only show you things that are meaningful, sip debug will show you a crapload of stuff that is not |
00:54.33 | [TK]D-Fender | Actually the opposite |
00:54.44 | [TK]D-Fender | generic verbose is just that .. generic |
00:54.51 | kunwon1 | at the moment you attempt a call, the output from the console is important |
00:55.01 | [TK]D-Fender | if doesn't show you where the screwup is and often barely alludes to the cause |
00:55.53 | pclerie | I think I would agree with that. |
00:56.51 | [TK]D-Fender | <pclerie> I've set up a test system and I'm using a GrandStream 1625 IP phone to test. The phone appears to work only some of the time. Just after a restart of Asterisk I can receive and place call. After a while, I can receive but every outgoing call fails. |
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00:57.05 | [TK]D-Fender | WHY does it fail? You'd have to look at the packets for the attempt and response |
00:59.23 | pclerie | fender: I've tried that. I've tried matching packets to what is happening. But as near as I can tell, there are simply missing packets when it fails that are present when it works. Which is kind of expected... :-( |
00:59.48 | kunwon1 | do you have the latest firmware for your phone? |
01:00.07 | pclerie | OK! I was sort of hoping it was a known issue... :-) |
01:00.37 | kunwon1 | it probably is, but there aren't enough details to determine |
01:01.17 | pclerie | Then I should go back and gather more info... |
01:01.58 | kunwon1 | console output, sip debug, and sip show peer are useful for others to help you to determine the issue |
01:02.04 | kunwon1 | phone config too |
01:02.22 | pclerie | OK! Will do and will revert ASAP. |
01:02.24 | pclerie | Thanks all! |
01:02.25 | [TK]D-Fender | Forget configs right now |
01:02.44 | [TK]D-Fender | That's like seeing a picture of a car before a crash. You don't see the fact that it gets driven into a wall |
01:02.58 | pclerie | :-D |
01:02.59 | [TK]D-Fender | Get proper debug for a filure |
01:03.07 | [TK]D-Fender | You want an autopsy? Give us a body |
01:03.11 | pclerie | Gotcha! |
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08:12.46 | sotoz | Hi, recently I sent an email at the asterisk-dev list. http://lists.digium.com/pipermail/asterisk-app-dev/2017-October/000892.html |
08:13.00 | sotoz | If anyone has an answer on that I would be grateful. Thanks for checking. |
08:18.10 | drmessano | Well, there is a -dev channel |
08:38.59 | sotoz | sorry my mistake. The message was in the asterisk-app-dev list :) |
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13:43.19 | _8bits | Hi, how to set System() result to variable in dialplan? |
13:43.24 | _8bits | I try same => n,Set(test=${System(/var/www/rankinis/test.php)}) |
13:43.36 | _8bits | but it doesnt work :< |
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13:49.54 | sibiria | is that php-file executable? |
13:50.01 | sibiria | is it +x, does it have a proper shebang? |
13:52.00 | sibiria | use SHELL function instead i think |
13:52.04 | [TK]D-Fender | System isn't a function |
13:52.09 | sibiria | if you need the result and not just the exit status |
13:52.14 | [TK]D-Fender | and functions are all uppercase |
13:52.16 | sibiria | the output* |
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13:52.35 | [TK]D-Fender | 2 mistakes from the start |
13:52.50 | [TK]D-Fender | System() is an app and retuns nothing. |
13:52.55 | [TK]D-Fender | SHELL() is a function and can |
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13:54.21 | _8bits | sibiria, i get pbx.c:4358 ast_func_read: Function System not registered |
13:54.31 | _8bits | but without Set - System is working fine |
13:55.16 | _8bits | ERROR[5736][C-000a1dd2]: pbx.c:4358 ast_func_read: Function Shell not registered |
13:55.27 | sibiria | SHELL |
13:55.30 | sibiria | not Shell |
13:56.01 | _8bits | **** me.... |
13:56.09 | _8bits | I was trying solve this like for 2 hours |
13:56.18 | _8bits | and always tried Shell not SHELL |
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14:25.40 | [TK]D-Fender | <[TK]D-Fender> System() is an app and retuns nothing. <---------------------- |
14:25.49 | [TK]D-Fender | APPLICATION |
14:26.07 | [TK]D-Fender | <[TK]D-Fender> and functions are all uppercase <-------------------------- |
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14:48.38 | domlaz11 | hi guys, trying to understand this part of log |
14:48.49 | domlaz11 | [2017-10-04 11:46:06] VERBOSE[12928][C-00005ab4] pbx.c: Executing [**301@from-internal:4] Hangup("SIP/303-00005caf", "") in new stack |
14:48.49 | domlaz11 | 303 tried to pickup the call from 301 |
14:48.56 | domlaz11 | and why its saying hangup? |
14:49.39 | Samot | One line of log doesn't tell anything 98% of the time. |
14:50.03 | Samot | That just says the Hangup happened. |
14:50.12 | Samot | Everything BEFORE IT tells you why it would be hungup. |
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14:50.54 | Samot | You asked this yesterday |
14:51.01 | Samot | You were given multiple answers yesterday |
14:51.55 | Samot | Including how to determine which side initiated the hangup. |
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15:06.38 | domlaz11 | sorry it wasnt the same call |
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15:19.07 | Samot | It doesn't mean the method for checking for the information is different. |
15:19.16 | domlaz11 | i have one question |
15:19.31 | domlaz11 | when i see the call coming in, is there a reference number i can look at when a trunk has left the bridge |
15:20.06 | domlaz11 | "SIP/GoldLineOut-00005ca9 Is the 0005ca9 a unique ID of the call? |
15:21.45 | Samot | No. |
15:21.53 | Samot | It's the trunk name |
15:22.02 | Samot | The unique ID is different |
15:22.58 | domlaz11 | what it look like? |
15:23.10 | domlaz11 | [2017-10-04 11:45:05] VERBOSE[12842][C-00005ab1] pbx.c: Executing [in@sub-record-check:3] ExecIf("SIP/GoldLineOut-00005ca9", "10?Set(FROMEXTEN=5143276372)") in new stack |
15:23.16 | domlaz11 | this is the first line i can see the phone number |
15:23.26 | Samot | What? |
15:23.37 | Samot | First you have to call on it in the dialplan to "see it" |
15:23.47 | Samot | It's a channel variable. |
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15:24.52 | domlaz11 | in the full log i cannot have a way to figure out when trunk left the bridge and associate it with the call i am lookin at? |
15:25.21 | Samot | It's in there. |
15:25.37 | Samot | The problem is, you don't understand how to interrupt/read the log. |
15:25.52 | Samot | Because you have no idea what any of it really means. |
15:25.52 | domlaz11 | thats why i am asking |
15:26.01 | Samot | You need to go and learn Asterisk's dialplan. |
15:26.06 | domlaz11 | ok |
15:26.18 | Samot | You need to understand how dialplan works in order to understand what the log output means. |
15:27.23 | Samot | Even more, you're using FreePBX.. |
15:27.40 | domlaz11 | yes i am |
15:27.46 | Samot | so not only do you have to understand dialplan, you have to understand the dialplan *it* created |
15:28.32 | domlaz11 | i started to read the book asterisk the feature of telephony but i am not done yet |
15:28.45 | Samot | Reading it is a good start |
15:28.55 | domlaz11 | i usally understand what i read |
15:29.11 | Samot | But *just* reading it doesn't give you real-time experience. |
15:29.27 | Samot | Because the book doesn't teach troubleshooting |
15:29.45 | domlaz11 | i have to start somewhere |
15:29.50 | Samot | Yes |
15:29.52 | domlaz11 | the way you talk i should quit |
15:30.01 | Samot | But supporting customer systems is not the place. |
15:30.18 | Samot | You need to have a system you can *literally* break. |
15:30.20 | domlaz11 | its small systems |
15:30.25 | Samot | Doesn't matter. |
15:30.33 | Samot | People are paying you for an expectation of service. |
15:30.41 | Samot | And knowledge, experience, skillsets. |
15:30.56 | Samot | Not to have you come into IRC and ask random people to fix their issues. |
15:31.05 | domlaz11 | you are very special my friend |
15:31.23 | Samot | You want to learn Asterisk, I 100% encourage that. |
15:31.30 | Samot | On your own time and dime. |
15:31.47 | Samot | Not at the expense of unknowing end users. |
15:32.34 | domlaz11 | thats not the point |
15:32.43 | Samot | Oh? |
15:32.47 | Samot | What is the point? |
15:32.48 | domlaz11 | i am not asking you to fix my issue |
15:32.55 | domlaz11 | just asking question |
15:32.59 | Samot | No, you're asking me to explain it. |
15:33.06 | Samot | You can't figure out how something happened. |
15:33.12 | Samot | You need to go and learn. |
15:33.15 | Samot | We had this talk. |
15:33.29 | domlaz11 | what is the purpose of this channel |
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15:33.35 | Samot | I said don't do calculus before you learn algebra.. |
15:33.43 | Samot | To help. |
15:33.49 | domlaz11 | many people use freepbx without knowing asterisk |
15:34.02 | domlaz11 | i am learning both |
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15:34.15 | Samot | While having people paying you services. |
15:34.22 | Samot | That you have to ask questions about. |
15:34.34 | Samot | Why do you need to know why the hangup or when the hangup happened? |
15:34.38 | domlaz11 | no |
15:34.57 | Samot | Why did you need to ask the question you asked? |
15:35.17 | domlaz11 | because apparently when she tried to pickup the call no one was there |
15:35.28 | domlaz11 | i was wondering when the call ended from the sip trunk side |
15:35.29 | Samot | And the issue yesterday? |
15:35.39 | Samot | Because it's the same trunk. |
15:35.40 | domlaz11 | ok let it go |
15:35.42 | Samot | Same customer. |
15:35.51 | Samot | So you're asking a question because your customer has an issue |
15:35.52 | domlaz11 | im ok thanks |
15:35.57 | Samot | And you don't know how to answer it. |
15:36.12 | Samot | So you're asking us. |
15:36.17 | Samot | So you can tell your customer. |
15:36.39 | domlaz11 | nope |
15:36.44 | domlaz11 | i can ask the senior |
15:37.02 | domlaz11 | i just wanted to have some tips |
15:37.06 | domlaz11 | not you to tell me what to say\ |
15:37.21 | domlaz11 | im not using you to get money |
15:37.24 | domlaz11 | thi sis ridiculous |
15:37.26 | [TK]D-Fender | So far what you've shown doesn't tell us anything about that call |
15:37.43 | domlaz11 | it's ok i will stop |
15:38.28 | [TK]D-Fender | <domlaz11> in the full log i cannot have a way to figure out when trunk left the bridge and associate it with the call i am lookin at? <- Channel name |
15:39.23 | domlaz11 | thanks fender |
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20:37.15 | SoBlindWolf | Do you guys know if there is a free automated voice assistant for FreePBX 13 and Asterisk 13? Such as Speech Recognition for voice prompt? |
20:38.52 | SoBlindWolf | Sorry version 14 for both |
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22:12.27 | eharris | is there a way to make Asterisk cdr log inbound calls that ring for only a few seconds, not long enough to be answered by voicemail? There don't seem to be cdr records generated for those. |
22:12.56 | [TK]D-Fender | You can log whatever you want in your dialplan |
22:13.28 | [TK]D-Fender | And if you wnt to log unanswered calls there is a setting in CDR for that too |
22:13.59 | eharris | have a keyword for me to lookup? |
22:14.15 | [TK]D-Fender | those are the keywords |
22:14.20 | eharris | ah, I see it. Thanks. |
22:14.33 | [TK]D-Fender | That's what the sample configs are for |
22:14.35 | [TK]D-Fender | read them |
22:15.13 | [TK]D-Fender | They are all commented in plain writing |
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23:56.54 | hdon | hi all :) when my asterisk sends SIP INVITE i'm not getting the Contact header that I want sent. what's the dialplan interface to the Contact header? is that Set(CALLERID(...)=...) ? |