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| 01:21.35 | spengler1 | is it necessary to recompile and reinstall asterisk after updating the kernel? |
| 01:22.32 | WIMPy | No. |
| 01:22.42 | WIMPy | But You have to reinstall DAHDI if you use that. |
| 01:22.58 | spengler1 | i do and i did ; thanks ; just forgot if i should |
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| 01:43.46 | rue_more | http://paste.debian.net/143665/ |
| 01:43.53 | rue_more | I need help with a basic logistical error |
| 01:45.25 | WIMPy | For logistics, try UPS, FedEx or the like. |
| 01:45.33 | WIMPy | But what's your question to that PB? |
| 01:45.44 | rue_more | not working, |
| 01:45.55 | rue_more | it said it couldn't find the timeout |
| 01:46.13 | rue_more | but it should go back to part 1 of that section |
| 01:46.16 | rue_more | ? |
| 01:46.28 | WIMPy | 1. The filename looks funny. |
| 01:46.36 | rue_more | its just a temp file |
| 01:46.47 | WIMPy | 2. There is no Application named ResoonseTimeout. |
| 01:46.50 | rue_more | that bit works when I stick it in a continious loop |
| 01:47.01 | WIMPy | Take a look at function TIMEOUT. |
| 01:47.18 | WIMPy | 3. there is nothing happening after the playback. |
| 01:47.33 | rue_more | http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 01:47.37 | rue_more | example 2 is wrong? |
| 01:47.56 | rue_more | its uspposed to keep recording and playing back toyou till you hit a key |
| 01:48.08 | rue_more | so I'm using the invalid as the catch-all for a key |
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| 01:49.37 | WIMPy | I see nothing that waits for input. |
| 01:49.53 | rue_more | sounds like a logistical errror |
| 01:50.07 | rue_more | can you suggest a fix before I can figure it out? |
| 01:50.39 | rue_more | WaitExten()? |
| 01:50.47 | WIMPy | there are many ways. |
| 01:51.11 | rue_more | this sounds like the voice of experience |
| 01:51.33 | WIMPy | You can use WaitExten, Read, or the right option to Record. |
| 01:52.55 | rue_more | k, workign with that... |
| 01:53.33 | WIMPy | And if you want to be able to exit by pressing a key, you may want to use Background or Read instead of Playback. Otherwise you have to wait for Playback to finish before being able to enter anything |
| 01:53.40 | rue_more | perfect |
| 01:53.56 | rue_more | ah |
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| 01:54.28 | rue_more | how can.. I make... will background ... how do I make it loop right after the recording is finished? |
| 01:56.08 | WIMPy | Recording? Playback? |
| 01:56.41 | rue_more | http://paste.debian.net/143666/ |
| 01:56.44 | rue_more | this is where I'm at |
| 01:56.47 | WIMPy | With Background/WaitExten you do it with extensions t and i like you already have there. |
| 01:56.59 | WIMPy | With Read, you'd use a GotoIf. |
| 01:57.28 | rue_more | I want it to loop as soon as the playback is finished if the user hasn't hit an escape key |
| 01:57.28 | WIMPy | That should work. |
| 01:57.44 | rue_more | this works, if it can be improved, then all the better |
| 01:58.07 | WIMPy | As I said, that doesn't work with Playback. That will not accept input while playing. |
| 01:58.15 | rue_more | yup, I get that |
| 01:59.26 | rue_more | in one hand I'v got background that will play audio while letting other things happen, on the other I'v got a timer that will repeat the loop, but I'd like to wait for the audio to stop before timing out :) |
| 02:00.16 | WIMPy | The timeout starts after playing. |
| 02:00.26 | rue_more | ah! |
| 02:00.38 | rue_more | so I can use background and make the timeout 0 if I want |
| 02:01.05 | WIMPy | Err, wait. No. If you give the timeout to WaitExten, it starts immediately. |
| 02:01.24 | rue_more | ah |
| 02:01.28 | WIMPy | With Read it starts after playing. |
| 02:01.41 | rue_more | so if I use read, .. ok |
| 02:02.14 | WIMPy | That is you use Read to play the file. |
| 02:02.52 | rue_more | ah |
| 02:03.42 | rue_more | exten => s,n,Read(junk,cracker,1,1,0) |
| 02:04.14 | rue_more | put digits in a variable I dont care about to dispose of, play the file cracker, 1 digit, 1 attempt, and 0 timeout |
| 02:04.22 | rue_more | or does 0 disbale and I need to use 1 |
| 02:04.35 | rue_more | 1 |
| 02:04.41 | WIMPy | Ther's a comma missing. |
| 02:05.03 | rue_more | oh, option... |
| 02:05.15 | rue_more | exten => s,n,Read(junk,cracker,,1,1,1) |
| 02:05.26 | rue_more | exten => s,n,Read(junk,cracker,1,,1,1) |
| 02:05.36 | WIMPy | Yes |
| 02:05.48 | rue_more | ;) its shotgun coding |
| 02:06.09 | rue_more | I wish I worked with * every day |
| 02:06.17 | rue_more | wait, maybe |
| 02:06.37 | rue_more | I only work on this poor things every 2 or 3 years |
| 02:06.38 | WIMPy | Think what you ask for. You might get it. |
| 02:07.26 | WIMPy | I hope there are some upgrades in between. |
| 02:07.37 | rue_more | ... no |
| 02:08.02 | rue_more | Connected to Asterisk 1.6.0.22 |
| 02:08.06 | rue_more | whats current? |
| 02:08.21 | rue_more | 1.8? |
| 02:08.31 | WIMPy | 13 |
| 02:08.37 | MaliutaLap | rue_more: current as in current long term support? or current current |
| 02:08.37 | rue_more | hah! |
| 02:08.50 | WIMPy | You missed 1.6.1, 1.6.2, 1.8, 10, 11 and 12. |
| 02:08.53 | rue_more | bleeding edge current |
| 02:09.21 | rue_more | 2 thru 9 appear to be missing? |
| 02:09.26 | WIMPy | I hope it's not reachable from the internet. |
| 02:09.36 | WIMPy | yes |
| 02:09.41 | rue_more | its not |
| 02:09.43 | rue_more | its all analog |
| 02:09.57 | rue_more | pots->t1 channelbank |
| 02:10.06 | rue_more | t1 channelbank -> phones |
| 02:10.13 | rue_more | *->t1 channelbank |
| 02:10.38 | WIMPy | I still wouldn;t want to use anyhting <1.8 and there's not really any excuse for using anythign <11. |
| 02:11.08 | rue_more | all the config formats changed, I think, I dont want to rewrite them and everything works |
| 02:11.33 | WIMPy | There hasn't been much change since 1.4. |
| 02:11.38 | MaliutaLap | rue_more: didn't take too much effort to get things up to scratch |
| 02:11.56 | rue_more | I realize that I dont have the latest apps like FlushTiolet() and MakeTea(), but thats ok, I'm not a person who wants my pbx to do those things |
| 02:12.07 | WIMPy | Not to the old stuff that is. Lots of new stuff, however. |
| 02:12.49 | MaliutaLap | I still have to migrate to pjsip |
| 02:13.06 | MaliutaLap | waiting until I get some hardware I'd lent out back |
| 02:13.49 | rue_more | hmm that didn't work |
| 02:14.19 | rue_more | http://paste.debian.net/143667/ |
| 02:14.20 | rue_more | ? |
| 02:15.44 | WIMPy | You want EITHER Background or Read. |
| 02:16.03 | rue_more | its commented |
| 02:16.07 | WIMPy | And with Read you need a GotoIf instead of the t and i extensions. |
| 02:16.17 | WIMPy | Ooops, right. |
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| 02:16.37 | rue_more | ah, gotoif to check if a key was pressed? |
| 02:16.53 | WIMPy | Either you check READSTATUS or junk for being empty. |
| 02:17.02 | WIMPy | Exactely. |
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| 02:20.12 | ldc | hi! anyone here with a spa122? |
| 02:20.38 | WIMPy | ~polls |
| 02:20.39 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 02:20.54 | ldc | ah, I was trying to figure out whether it could be used as a ATA only |
| 02:21.00 | ldc | instead of ata+router, before buying |
| 02:21.39 | WIMPy | Now, that's a question. But none I can answer. |
| 02:22.13 | MaliutaLap | ldc: what does <search_engine> say? |
| 02:22.38 | ldc | MaliutaLap: cisco doc appears, and it's unclear |
| 02:23.00 | ldc | Bridge: Bridged mode is used if the ATA is acting as a bridge device to another router. |
| 02:23.16 | ldc | I guess, by doing this the "WAN" port becomes a standard switchport? |
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| 02:24.07 | WIMPy | Uncertain, but it surely looks like you can use it as an ATA only. |
| 02:25.28 | MaliutaLap | Looking at the docs I'd say it's similar to some of the WAP's I use - if you don't configure the internet ports, and only configure the the LAN it should be fine |
| 02:25.57 | ldc | okay thanks |
| 02:26.06 | ldc | for 27 eur it's worth trying out then |
| 02:26.22 | WIMPy | I have seen routers that wouldn't even try to register or call if they didn't think they were "online". |
| 02:26.48 | WIMPy | But on most of them it's no problem. |
| 02:28.10 | rue_more | exten => s,n,GotoIf($[$["$READSTATUS" = "TIMEOUT"]]?1:${Origin},s,1) |
| 02:28.24 | rue_more | that took a lot of educated guesswork, is it right? |
| 02:28.34 | WIMPy | But does it really make sense to give 27 EUR to be able to connect an ancient phone? |
| 02:29.15 | WIMPy | No. Why do you have two []? and where are the {} around the variable? |
| 02:30.22 | rue_more | ah |
| 02:31.01 | rue_more | that expalins why it wasn't working right |
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| 02:34.47 | ldc | WIMPy: it's an italian designed ancient phone. :p |
| 02:35.04 | [TK]D-Fender | <ldc> ah, I was trying to figure out whether it could be used as a ATA only <- yes |
| 02:35.12 | rue_more | ok! I got it working good 'nuff! |
| 02:35.20 | WIMPy | ldc: Take it apart and put something else inside :-) |
| 02:35.26 | rue_more | thanks for your help WIMPy |
| 02:35.32 | ldc | http://www.museoscienza.org/dipartimenti/catalogo_collezioni/images/12397_01.jpg |
| 02:36.14 | WIMPy | He. It already says SIP. Make that real. |
| 02:36.32 | WIMPy | What's RP? |
| 02:37.46 | ldc | WIMPy: redial last |
| 02:37.58 | ldc | R is hold |
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| 04:16.57 | hebber | In cdr_adaptive_odbc.conf there is a filter function to discard CDR's: filter src != 456. This doesn't seems to work. In manager.conf it has this format: eventfilter=!Channel: DAHDI* |
| 04:17.12 | hebber | Is the conf files up to date? |
| 04:18.56 | [TK]D-Fender | != is not =! either |
| 04:19.28 | hebber | aha |
| 04:22.03 | hebber | actually, could be correct, but in the context of manager.conf I don't think its _either_ |
| 04:25.23 | hebber | <PROTECTED> |
| 04:25.43 | [TK]D-Fender | odbc requires a 100% match |
| 04:25.59 | [TK]D-Fender | wheras manager is partial |
| 04:26.30 | [TK]D-Fender | so " filter src != 456" should get all except 456 |
| 04:27.03 | hebber | ok, thats what I need - will try and test some more - thanks Fender |
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| 05:07.34 | hebber | Using asterisk 11.15.0: on an originate call using AMI I'm not able to set user defined parameters in ODBC adaptive CDR. The Set(CDR(userfields)=something) get discarded. |
| 05:08.09 | hebber | On a normal call I can set any value with no problem |
| 05:09.16 | hebber | the problem is limited to CONGESTED originated calls and the cdr.conf unanswered=yes |
| 05:12.56 | hebber | Basically want to discard ODBC adaptive CDR records which are not ANSWERED. The setting cdr.conf unaswered=yes must be included. |
| 05:14.07 | [TK]D-Fender | Show us exactly how you're placing the call and the odbc config and table to back it |
| 05:14.42 | [TK]D-Fender | That filter sounds like it might work... |
| 05:16.13 | hebber | agree, but the value is not set, so the filter is rendered useless. It works on a normal call as I just tested it. |
| 05:17.15 | hebber | You want to take a look at the whole dialplan? It got very complex to support what's needed. |
| 05:21.35 | [TK]D-Fender | no, the Originate, configs, and CLI for now |
| 05:23.49 | hebber | The originate commands in dialplan: http://pastebin.com/teuFSpQf |
| 05:25.14 | hebber | cdr_adaptive_odbc.conf: http://pastebin.com/Pava2YcQ |
| 05:28.38 | hebber | sending command to AMI to originate: http://pastebin.com/hVXTMGbg |
| 05:29.58 | hebber | Most of the dialplan's complexity was to be able to send variables back and forward between legs, to be able to capture the parameters at the end. |
| 05:32.05 | [TK]D-Fender | I think I see why it isn't getting set.... |
| 05:32.41 | [TK]D-Fender | You're using a Local channel to place the call-out, but when it gets answered the local channel (which is what had the vars set against it) gets optimised away |
| 05:32.59 | [TK]D-Fender | You need to call it with /n so it stays in the chain. |
| 05:33.14 | [TK]D-Fender | IINM |
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| 05:36.09 | hebber | could be, but I'm sending a LOCAL...@autodial_call_out/n. So the LOCAL channel should stay in the chain? |
| 05:36.24 | hebber | Code here: http://pastebin.com/hVXTMGbg |
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| 05:38.18 | hebber | and in this case the problem is when the call is not answered and unable to use Set(CDR(userfield)=blabla) or NOCDR() |
| 05:40.30 | [TK]D-Fender | I think I missed scrolling a bit there.. |
| 05:40.50 | [TK]D-Fender | And it's stil hitting CDR? |
| 05:41.54 | hebber | yes, as with a disposition value other than "ANSWERED" it creates a CDR record using adaptive CDR |
| 05:42.58 | hebber | started with NOCDR(), commented it out then tried a filter using userfield!=nocdr and set the userfield=nocdr in the call. Non works |
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| 05:43.45 | hebber | to be sure I set the CDR values everywhere in the dialplan, still not working. |
| 05:44.20 | hebber | apart from the overuse of CDR records the system works like a charm :) |
| 05:47.20 | [TK]D-Fender | hrm |
| 05:47.36 | [TK]D-Fender | 1 solution... |
| 05:48.03 | [TK]D-Fender | Since the problem is with it not answering.... ASNWER IT. |
| 05:48.19 | [TK]D-Fender | But set variable before you do and check for it on the other side's dialplan |
| 05:48.23 | [TK]D-Fender | And you can call it there. |
| 05:48.36 | [TK]D-Fender | Leaves all the work on the "answer" side |
| 05:51.10 | hebber | hmm, AMI ORIGINATES a call to a context, the context answers it instead of make a DIAL? |
| 05:51.46 | hebber | ANSWER before or after the dial to callee? |
| 05:53.24 | hebber | Or continue with the DIAL, if the disposition is CONGESTED, then set the variables and finally answer the call? |
| 05:54.51 | [TK]D-Fender | No, do your dial.... but on failure to answer, set a variable the ANSwser it hard and wait. |
| 05:55.14 | [TK]D-Fender | On the other side see if that variable got set. If so, set the CDR to nocdr |
| 05:55.45 | [TK]D-Fender | basically it's going to answer no matter what... just if it's "dead" it'll let you know |
| 05:56.36 | hebber | okey, will try that - but that I can't change with 30 concurrent channels, need to test it out tonight. Will let you know |
| 05:56.44 | hebber | I think that may work :) |
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| 07:42.11 | woopstar | Any reason why the scripts shipped for alembic with Asterisk 13 is not containing queue_log , when you want to use realtime? |
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| 09:06.52 | ipalmer | Hi all something really simple for someone, I'm trying to replace a string one,\,two,\three to one&two&three I'm using the replace function to do this, I can replace the comma but not the backslash, can anyone help, the pastebin of my code is here http://pastebin.com/YWTj9X80 |
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| 09:11.47 | eirirs | escape backslash with another backslash |
| 09:12.37 | eirirs | like ,\\, |
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| 09:15.51 | MaliutaLap | leaning trees |
| 09:16.36 | ipalmer | eiris: thanks, trying that now but it doesn't seem to work |
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| 09:18.49 | eirirs | ipalmer: sorry, I meant ,/\/\, |
| 09:18.59 | eirirs | are escape character, not \ |
| 09:19.17 | eirirs | always is confused between / and |
| 09:19.46 | eirirs | no wait |
| 09:19.55 | eirirs | picks another cup of coffee |
| 09:20.31 | ipalmer | eirirs: lol wakey wakey |
| 09:21.11 | MaliutaLap | slash or diarrhea - I mean backslash |
| 09:21.51 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:23.32 | ipalmer | eirirs: so what was the outcome? |
| 09:34.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:36.46 | eirirs | ipalmer: it's \\ , and if it aint working, what exactly are you coding? dialplan? |
| 09:40.36 | MaliutaLap | would depend - scp/ssh needs a double escape foo\\\ bar |
| 09:42.17 | ipalmer | eirirs: I've tried 2 different things 1 and 2 in my pastebin http://pastebin.com/SfgJv25M |
| 09:44.32 | eirirs | ipalmer: use double quotes, this way \ won't be parsed, but read as string |
| 09:44.46 | eirirs | in asterisk dialplan if I ain't wrong |
| 09:47.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:50.21 | ipalmer | eirirs: nope, I'm going to extract the function away from the rest of the dialplan in case something else I'm doing is causing it, I'll be back |
| 09:51.54 | *** join/#asterisk raspberrypifan (~raspberry@186.46.190.243) |
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| 09:52.51 | tparcina | I have been asked to achieve this functionality: |
| 09:53.22 | tparcina | On incoming phone calls, we play some welcome message. |
| 09:54.36 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:55.04 | tparcina | Then we try to connect the call to some of users. |
| 09:55.35 | tparcina | If none of them is available, we send the call to the voice mail. |
| 09:56.19 | tparcina | If caller waits until it hears voice mail greetings, then we receive the mail with his details. |
| 09:56.28 | tparcina | (phone number, time of the call...) |
| 09:56.59 | tparcina | But, if the caller hangs up some time before, we don't get any notification about that call. |
| 09:57.37 | ipalmer | eirirs: sorted it, we need to do \\\\ |
| 09:57.49 | eirirs | lol |
| 09:57.59 | eirirs | so were were somewhat at correct path, already |
| 09:58.07 | eirirs | we were* |
| 09:58.15 | tparcina | Now, I need to send e-mail for every phone call that wasn't connected to our users, and didn't leave the message in voice mail- |
| 09:58.18 | eirirs | more coffee, bbl |
| 09:58.27 | tparcina | Just I'm not sure how to do that. |
| 09:58.34 | ipalmer | kind of, we needed to use backslash just not sure how many lol |
| 09:58.58 | eirirs | lol yeah |
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| 10:17.21 | stevenm | Is there something (or a product that uses Asterisk) of having mulitple PBX's that can all be configured from one interface? i.e. kind of like a stack/cluster of switches with on UI - but in this case PBX's |
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| 10:54.19 | taplaptap | anyone have an ip 601? when you setup phone<mac>.cfg, does the file have to be lowercase? the phone takes so long to reboot :( |
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| 12:24.26 | pawiecki | Hello! Is there any reason to use so called hunt groups, instead of just using a simple queue? I'm confused about these two. Could someone explain why theese two options are used? |
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| 14:10.41 | *** join/#asterisk t4nk152 (5f5c7904@gateway/web/freenode/ip.95.92.121.4) |
| 14:10.47 | t4nk152 | hello all |
| 14:10.57 | t4nk152 | i have one problem with my asterisk |
| 14:11.35 | t4nk152 | when exist one call between two extension |
| 14:11.53 | t4nk152 | one another try to make a call for one this |
| 14:12.02 | t4nk152 | i don't receive the busy song |
| 14:12.14 | t4nk152 | any ideia for this ? |
| 14:16.10 | Xaviertoor | call limite=1 |
| 14:17.19 | Xaviertoor | call-limit=1 |
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| 14:24.31 | [TK]D-Fender | t4nk152: Ever had call-waiting on an analog lin from a telco before? |
| 14:25.03 | [TK]D-Fender | t4nk152: Ever been on a call on a cell and had a 2nd call ring in that you could answer? |
| 14:26.12 | t4nk152 | @Xaviertoor in extension.conf ? |
| 14:26.14 | [TK]D-Fender | [09:17]Xaviertoorcall-limit=1 <- very bad idea. What if this person needs to check their voicemail while on a call? |
| 14:26.15 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:26.24 | [TK]D-Fender | t4nk152: Not a good approach |
| 14:26.45 | [TK]D-Fender | t4nk152: Check the device state before dialing them. |
| 14:27.02 | t4nk152 | how check this ? |
| 14:27.14 | [TK]D-Fender | "core show function DEVICE_STATE" |
| 14:27.42 | [TK]D-Fender | "core show application chanisavail" |
| 14:27.43 | [TK]D-Fender | etc |
| 14:28.35 | [TK]D-Fender | Still not a good idea in general |
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| 14:46.46 | t4nk152 | exten => s,1,ExecIf($[ ${DEVICE_STATE(SIP/${EXTEN})} = INUSE ]?Busy) |
| 14:47.00 | t4nk152 | i put this line in my extension.conf |
| 14:47.15 | t4nk152 | but the result is the same |
| 14:47.55 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 14:50.06 | [TK]D-Fender | Because there a several mistackes in there |
| 14:50.10 | [TK]D-Fender | mistakes* |
| 14:50.57 | [TK]D-Fender | Wrong application, wrong variable, incomplete expression, etc |
| 14:53.07 | *** join/#asterisk Katty (sid62315@gateway/web/irccloud.com/x-kfztumqspwpldhbw) |
| 14:54.03 | t4nk152 | the default configurations allow this ? |
| 14:57.04 | [TK]D-Fender | ? |
| 14:57.21 | [TK]D-Fender | what "default"? This is the dialplan.. there is no such thing as "default". |
| 14:57.41 | [TK]D-Fender | There is only what you make. |
| 14:59.24 | t4nk152 | the default extension.conf |
| 14:59.53 | [TK]D-Fender | [09:57][TK]D-Fenderwhat "default"? This is the dialplan.. there is no such thing as "default". |
| 14:59.59 | [TK]D-Fender | this is YOUR dialplan. |
| 15:00.01 | [TK]D-Fender | YOU make it. |
| 15:01.18 | [TK]D-Fender | If you're referring to the SAMPLE CONFIG, that is just a syntax guide, and as "inspiration". It is effectively garbage that is not meant to be used. |
| 15:01.39 | *** join/#asterisk albercuba (~albercuba@HSI-KBW-46-237-192-160.hsi.kabel-badenwuerttemberg.de) |
| 15:01.58 | albercuba | hello everyone. Can i set up two phones with the same extension??? |
| 15:02.45 | albercuba | hello everyone. Can i set up two phones with the same extension??? if i want to have a desk phone and a wireless phone and they both answer to the same number |
| 15:03.21 | [TK]D-Fender | only chan_pjsip supports multiple registrations. |
| 15:03.41 | [TK]D-Fender | Otherwise you'll have to set them up as separate devices and simply dial both in your Dial() command |
| 15:03.56 | albercuba | no, the idea is to have the same number on both |
| 15:04.26 | [TK]D-Fender | both work |
| 15:04.37 | [TK]D-Fender | "have same number" doesn't necessarily mean anything |
| 15:04.46 | albercuba | ok let me explain myself |
| 15:05.02 | [TK]D-Fender | If you want to call 2 at the same time... you can dial 2 at the same time. |
| 15:05.26 | [TK]D-Fender | If you need the SAME ACCOUNT for login purposes.. then that is only supported unduer chan_pjsip |
| 15:05.34 | albercuba | i want that if i have a desk phone and a wireless phone, and someone calls my number, they both answer to that number |
| 15:05.48 | [TK]D-Fender | Either can show as the same callerid and have access to dial the same things |
| 15:05.53 | albercuba | ok yes they both need to answer to the same account |
| 15:05.58 | [TK]D-Fender | either can work |
| 15:06.17 | albercuba | ok i will take a look at chan_pjsip |
| 15:06.19 | albercuba | thanks |
| 15:06.42 | [TK]D-Fender | [10:05]albercubaok yes they both need to answer to the same account <- No. Call processing is DIALPLAN. having the same account on both is NOT a requirement. Dial() can call MULTIPLE devices simultaneously regardless of type |
| 15:07.25 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-110-231.tubit.tu-berlin.de) |
| 15:07.51 | [TK]D-Fender | It's a lot less work to simply create another peer than to implement an entirely new channel-type if you aren't using it already |
| 15:07.54 | albercuba | sorry i am a noob with asterisk so i don't really understand too much about it yet |
| 15:07.58 | [TK]D-Fender | Also pjsip is only 12+ |
| 15:08.23 | albercuba | what do you mean with 12* |
| 15:08.25 | albercuba | 12+ |
| 15:08.32 | [TK]D-Fender | Asterisk 12 and higher<------- |
| 15:08.36 | albercuba | a ok |
| 15:09.07 | albercuba | thanks |
| 15:10.09 | [TK]D-Fender | "core show application dial" |
| 15:10.15 | [TK]D-Fender | ~book |
| 15:10.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:10.17 | [TK]D-Fender | ^^^ |
| 15:19.12 | t4nk152 | @[TK]D-Fender when i show the dialplan my macro dial |
| 15:19.26 | t4nk152 | <PROTECTED> |
| 15:19.32 | t4nk152 | i see this |
| 15:20.12 | [TK]D-Fender | That is some pretty broken code to start with |
| 15:20.42 | [TK]D-Fender | But go on... |
| 15:21.07 | t4nk152 | what is the better solution for this case ? |
| 15:21.42 | t4nk152 | thanks for your help |
| 15:22.38 | t4nk152 | when i solve this problem i join you to drink one beer :D |
| 15:24.05 | [TK]D-Fender | Before thinking "better solution" you should ask yourself "what is my code really doing? What is that desing meant to accomodate? Why was it written that way?". |
| 15:24.36 | [TK]D-Fender | So far I'm getting a feeling like you have just copies some other little pieces over and are running on recycled code you don't fully understand |
| 15:25.35 | t4nk152 | actually I did not. I 'm trying to fix what was done |
| 15:25.42 | [TK]D-Fender | And so far .... we don't have an actual "problem". |
| 15:25.56 | [TK]D-Fender | I just see some sloppy code |
| 15:26.08 | [TK]D-Fender | So go try and proper integrate what I told you into it. |
| 15:26.16 | [TK]D-Fender | propoerly* |
| 15:26.24 | [TK]D-Fender | just can't type these days... |
| 15:28.04 | t4nk152 | my problem is that the asterisk not signals when user is in Call |
| 15:28.43 | [TK]D-Fender | t4nk152: because it doesn't care and neither does the device you're calling |
| 15:29.01 | [TK]D-Fender | t4nk152: I gave you the commands to LOOK at what it's doing so you can coose to act DIFFERNTLY <- |
| 15:29.05 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 15:29.34 | BKhan | Hi . There is an issue in call transfer need help about it |
| 15:30.59 | BKhan | inbound call (person P) ---> receive by caller A ---(transfer to)----> Caller B ( issue is caller B and person P can not hear voice) |
| 15:31.15 | t4nk152 | I have multiple extensions. do you think I should set up one by one ? |
| 15:31.56 | [TK]D-Fender | t4nk152: Doesn't matter so long as it does what you want. |
| 15:41.14 | *** join/#asterisk WHiZZi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 15:41.44 | WHiZZi | goodday everyone |
| 15:42.24 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-bxqfzkzptkdbqcdv) |
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| 15:44.28 | WHiZZi | I have a challenge here with sip peers who are acting weird as far as we can tell |
| 15:45.34 | WHiZZi | sip peer = user-500 |
| 15:45.39 | WHiZZi | type=friend |
| 15:45.53 | WHiZZi | nothing fancy else, canreinvite=yes, secret set and a call-limit |
| 15:45.59 | WHiZZi | user can log in perfectly fine |
| 15:46.11 | WHiZZi | but, when I do a 'sip show peer user-500' |
| 15:46.27 | WHiZZi | * Name : user-500 |
| 15:46.59 | WHiZZi | Reg. Contact : sip:user-500@ip:5060;transport=udp |
| 15:47.22 | WHiZZi | Def. Username: banana-123 |
| 15:47.27 | *** join/#asterisk Xaviertoor (~jerson.ju@177.99.205.154) |
| 15:48.04 | WHiZZi | why is the def. username not matching the sip-peer name/reg. contact ? |
| 15:49.03 | WHiZZi | and it doesn't matter if there's a [banana-123], even without this account, the Def. Username doesn't match the sip-peers name |
| 15:50.23 | dan_j | Hi, Where can I find the definition of Status in this AMI event? |
| 15:50.24 | dan_j | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ExtensionStatus |
| 15:50.33 | dan_j | IE, what does Status mean? |
| 15:50.52 | [TK]D-Fender | Usual. INUSE, RINGING, ETC |
| 15:51.08 | [TK]D-Fender | Shoud be same values as for presence |
| 15:51.42 | dan_j | Looking at sample data, Status is a digit. |
| 15:51.58 | dan_j | Can I assume it means the same as v13? |
| 15:51.58 | dan_j | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_ExtensionStatus |
| 15:54.01 | [TK]D-Fender | Should be |
| 15:55.39 | WHiZZi | ok, before I have to read the Asterisk source perhaps somebody can tell me where Asterisk stores it's Def. Username value since it's not coming from a config-file whatsoever? |
| 15:55.56 | *** part/#asterisk mjordan (mjordan@nat/digium/x-mmexdtelivsljofo) |
| 15:57.04 | [TK]D-Fender | WHiZZi: Show us the actual config and peer dump |
| 15:57.13 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 15:58.22 | WHiZZi | sure, hold on |
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| 16:00.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:00.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:02.26 | WHiZZi | [TK]D-Fender: http://pastebin.com/fUDiJL2R |
| 16:03.22 | WHiZZi | bananax2d-202 used to exist on the server, but doesn't exist anymore (is deleted from all config-files) |
| 16:05.36 | [TK]D-Fender | * can keep stuff cached... |
| 16:05.41 | [TK]D-Fender | Have you restarted * since? |
| 16:06.04 | WHiZZi | 2 weeks at this moment |
| 16:06.32 | WHiZZi | problem was there before, restarted the whole server and even upgraded to a later version of Asterisk. Problem still existed afterwards |
| 16:07.28 | WHiZZi | same extensions, there are multiple extensions with this problem showing currently around 4 different 'Def. Usernames' |
| 16:08.23 | *** join/#asterisk kj22594 (32f52289@gateway/web/freenode/ip.50.245.34.137) |
| 16:08.41 | WHiZZi | Peers are registered on different ip's (different ISP's even) and there's no relation between the customers |
| 16:09.14 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 16:09.24 | WHiZZi | happens with different clients as well (phones, softclients) |
| 16:09.47 | WHiZZi | the only thing I can imagine at this point, is that the internal db of Asterisk is corrupted in some way |
| 16:10.14 | WHiZZi | I do see a lot of duplicate entries when I do a 'database show' |
| 16:10.49 | *** part/#asterisk kj22594 (32f52289@gateway/web/freenode/ip.50.245.34.137) |
| 16:11.12 | WHiZZi | for example, 81 entries of: |
| 16:11.22 | WHiZZi | SIP/Registry/hmgv2c-106 : 89.255.54.64:39118:3600:intersportveldhuis-212:sip:hmgv2c-106@192.168.12.20:5070 |
| 16:12.17 | WHiZZi | hmgv2c-106 has a correct Def. Username though |
| 16:13.04 | WHiZZi | 106 database entries of hmgv2c-106 |
| 16:13.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 16:23.34 | WHiZZi | so I guess I'll remove the astdb? What do you think [TK]D-Fender |
| 16:23.58 | [TK]D-Fender | That's an idea... |
| 16:24.14 | WHiZZi | I love to hear other ideas ;) |
| 16:25.44 | *** join/#asterisk mjordan (mjordan@nat/digium/x-nzafaokrabzefzzq) |
| 16:25.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:29.50 | WHiZZi | well, I'll try that. Thanks for reading and confirming |
| 16:30.02 | *** part/#asterisk WHiZZi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
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| 16:56.51 | echo083 | hello |
| 16:56.55 | echo083 | res_fax_spandsp.c: In function spandsp_v21_new: |
| 16:57.00 | echo083 | res_fax_spandsp.c:487: error: MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE undeclared (first use in this function) |
| 16:57.48 | echo083 | how to build res_fax ? |
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| 17:10.29 | BKhan | inbound call (person P) ---> receive by caller A ---(transfer to)----> Caller B ( issue is caller B and person P can not hear voice) |
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| 17:10.42 | BKhan | is any solution of it |
| 17:12.18 | [TK]D-Fender | Stop allowing reinvites |
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| 17:28.24 | BKhan | Thanks D-Fender actually it is previously stop. canreivnit=no |
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| 17:29.40 | [TK]D-Fender | BKhan: pastebin your actual peers |
| 17:31.34 | file | falls over |
| 17:32.39 | BKhan | http://pastebin.com/cqJUqZ23 |
| 17:33.08 | BKhan | D-Fender: Please check it is for local peer |
| 17:33.51 | [TK]D-Fender | canreivnit=no <- SPELLING |
| 17:34.49 | file | Fun fact: In chan_pjsip that wouldn't have allowed the endpoint to be created. |
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| 17:35.13 | [TK]D-Fender | file: awesome. |
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| 17:36.09 | BKhan | D-Fender: Ohhh sorry let me check :) |
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| 18:01.06 | BKhan | D-Fender: I made change but still same. Actually on our server we have two interfaces one with public ip and second with private |
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| 18:02.30 | AnonGirl | hi. |
| 18:03.19 | BKhan | I observed with wireshare. when call tranfer to agent C , his voice come to server but not come to person who is dialing inbound |
| 18:05.02 | [TK]D-Fender | I'm not seeing your "fix" |
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| 18:37.27 | DivideBy0 | When setting a channel variable through the ARI, I can immediately retrieve it, but when I try to get it later, in a hangup request event, I cant see it. Shouldn't the variable last until the channel is destroyed or the var is explicitly unset? http://pastebin.com/r7zeMMaF |
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| 18:39.57 | DivideBy0 | actually, it looks like the channel is already gone in the ChannelHangupRequest event? |
| 18:40.45 | [TK]D-Fender | uSE A HANGUP HANDLER |
| 18:40.59 | [TK]D-Fender | Becuase yes.. that can be an issue |
| 18:41.37 | DivideBy0 | [TK]D-Fender: Thanks I have a handler on the ChannelHangupRequest event through the ARI. Where is the right place I should have one? |
| 18:41.49 | [TK]D-Fender | Not sure about ARI |
| 18:42.00 | [TK]D-Fender | Was thinking raw dialplan terms there for a sec |
| 18:42.20 | DivideBy0 | ok. I'll figure it out. Thanks as always. |
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| 18:46.29 | DivideBy0 | also file said "it's documented in the API - the script which turns that into wiki documentation just may not be smart enough to reflect it" so I thought that meant http://doxygen.asterisk.org/trunk/AstAPI.html but I don't see what we were talking about - it was where the docs for the list of playback controls were. Does anyone know where the right doc for the API he's talking about? (I |
| 18:46.30 | DivideBy0 | haven't seen him online) |
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| 19:03.10 | marceloamorim | guys, I'm trying to install dahdi but I'm getting this messages - Can't read private key - anyone solve this problem? |
| 19:09.19 | WIMPy | What key? |
| 19:09.52 | marceloamorim | <PROTECTED> |
| 19:09.53 | marceloamorim | Can't read private key like this |
| 19:11.26 | WIMPy | I can remember having disabled that part in the past. |
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| 19:38.07 | DivideBy0 | if I originate a channel (out to the pstn via sip) into my ARI/stasis app's bridge, is there a way to hear ringing/busy/etc, or do I have to do that myself somehow? |
| 19:40.08 | file | there is currently no ability for ARI originated channels to be bridged before they are answered |
| 19:41.54 | DivideBy0 | I add the channel to the bridge right after originating it, and I can hear once they pickup, but you're saying I'm unable to hear anything before that, right? |
| 19:42.41 | DivideBy0 | also, they other day, you answered my question about playbacks saying "its in the api, but the wiki script hasn't picked it up yet" - where is the api documented? I thought it was doxygen.asterisk.something? |
| 19:43.22 | file | you can't add an originated channel to a bridge until it has been answered, if that is working for you then the channel is answered immediately and it should work |
| 19:44.17 | file | ARI is documented using an older version of http://swagger.io/, you can explore the API using swagger-ui - for example head to http://ari.asterisk.org/ and use http://neutron.jcn-labs.net:8088/ari/api-docs/resources.json as the URL at the top and test:test as the API key |
| 19:45.04 | DivideBy0 | wow. Thanks! I would have never found that myself. |
| 19:45.41 | file | https://wiki.asterisk.org/wiki/display/AST/Using+Swagger+to+Drive+ARI |
| 19:45.48 | file | it's useful for easily playing about |
| 19:46.04 | file | that's also what is used to create the documentation on the wiki |
| 19:46.25 | DivideBy0 | yeah. it's really neat and fills in the holes for sure |
| 19:46.34 | AnonGirl | file, hi |
| 19:46.40 | file | hi |
| 19:47.34 | DivideBy0 | last sorta related question - when the ARI event ChannelHangupRequest is emitted, the channel seems to be already gone. is that a bug or a feature? I was trying to read the channel details or a channel var |
| 19:49.14 | file | as that event is sent in an asyncronous fashion it's entirely possible for the channel to already be gone by the time you get it and try to act on it |
| 19:50.59 | file | it should go ChannelHangupRequest -> ChannelDestroyed, but even then it can happen so fast that it's gone |
| 19:52.31 | DivideBy0 | aha. that makes complete sense. thanks again! (I'm digging the swagger) |
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| 20:38.33 | talntid | I have an odbc while loop. from inside that while loop, i am calling another odbc function. when that second one runs, it is resetting ${ODBCROWS} and the initial loop gets broken... solution? |
| 20:38.47 | talntid | will making the internal function into a macro break it out of that scope? |
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| 21:30.43 | echo083 | what is the trunk dahdi/g0 please ? |
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| 21:31.40 | echo083 | it was automatically created |
| 21:31.46 | echo083 | should i remove it ? |
| 21:32.07 | rrittgarn1 | ~book |
| 21:32.07 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:34.20 | echo083 | me ? |
| 21:34.27 | echo083 | you are providing these information for me ? |
| 21:34.59 | echo083 | i partially know what is dahdi g0 trunk but i wanted to know if i should remove it or keep it ? |
| 21:37.18 | WIMPy | It's just a variable definition. |
| 21:37.40 | echo083 | WIMPy, should i remove or keep dahdi/go trunk which was automatically created ? |
| 21:37.48 | echo083 | g0 |
| 21:37.52 | WIMPy | And the sample configs are more documentation that meant to be used. |
| 21:38.13 | WIMPy | that means, that you should create your own file. |
| 21:39.35 | echo083 | WIMPy, ok then i'll remove it |
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| 21:40.14 | echo083 | WIMPy, for conferencing should i use app_meet or app_confbridge ? |
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| 21:48.35 | jmetro | Do you guys know if Digium phones have bluetooth? my silly spec sheet is way basic |
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| 21:57.32 | WIMPy | echo083: ConfBridge is the new one with all the new features. |
| 21:57.59 | WIMPy | jmetro: No. I think it was thought of as an option, but I don't know if that exists, yet. |
| 21:59.13 | newtonr | jmetro, nope, no bluetooth. |
| 21:59.43 | jmetro | WIMPy: Looks like it isnt, atleast according to the documentation on the digium wiki. Now looking at the Phone Compatibility list.. im surprised theres only 3 plantronics hones on the EHS Support. |
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