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| 01:17.18 | jonno11 | How can I use CURL? |
| 01:17.58 | jonno11 | exten => 8,1,Set(foo=${CURL(http://SERVER:3000/log_call?number=1)}) |
| 01:18.02 | jonno11 | doesn't work |
| 01:18.56 | Billiard | "doesn't work" is not very helpful |
| 01:19.07 | Billiard | is anything logged? |
| 01:19.32 | Billiard | or notices/warnings/errors printed in CLI |
| 01:22.02 | jonno11 | Sorry! I get Executing [8@gotoexten:1] Set("SIP/2000-0000018e", "foo=") in new stack |
| 01:22.07 | jonno11 | But that's it |
| 01:22.27 | jonno11 | and the host is set to log to console when a request is made |
| 01:22.33 | jonno11 | Billiard ^ |
| 01:22.47 | Billiard | are you sure that extension is even reached? |
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| 01:30.25 | jonno11 | What do you mean? |
| 01:30.39 | jonno11 | Billiard: It goes on to the next item in that extension list |
| 01:30.49 | jonno11 | And it says "Executing:" |
| 01:31.06 | jonno11 | But just setting the variable to null |
| 01:31.18 | jonno11 | Which should be the contents of the curl request |
| 01:32.01 | Billiard | and this on the same machine prints something?: curl http://SERVER:3000/log_call?number=1 |
| 01:34.05 | jonno11 | Billiard: Yeah |
| 01:34.21 | jonno11 | with SERVER replaced for my server ofc |
| 01:35.07 | Billiard | checked your server's access logs |
| 01:35.14 | Billiard | ? |
| 01:35.22 | jonno11 | as in the web server? |
| 01:35.26 | Billiard | yes |
| 01:35.32 | jonno11 | no such thing, I'm using node. |
| 01:35.43 | jonno11 | so as soon as it should connect |
| 01:35.50 | jonno11 | or makes a get request |
| 01:35.55 | jonno11 | it prints to the console |
| 01:36.07 | jonno11 | But it doesn't |
| 01:36.43 | jonno11 | Billiard: So the request is never made |
| 01:37.33 | Billiard | try CURL(http://google.com) or something known to be working? :p |
| 01:38.00 | jonno11 | No no I mean, curl works |
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| 01:38.13 | jonno11 | It's successful |
| 01:38.34 | jonno11 | from shell, but from asterisk it isnt |
| 01:39.37 | Billiard | you've tried CURL(http://google.com) to see if foo contains something? |
| 01:39.49 | jonno11 | Billiard: Yeah just did, that didn't work :/ |
| 01:39.55 | jonno11 | :S * |
| 01:39.58 | Billiard | you did reload the dialplan? |
| 01:40.04 | jonno11 | yep |
| 01:40.12 | Billiard | how did you determine foo was empty? |
| 01:40.20 | jonno11 | Same output as before |
| 01:40.32 | jonno11 | Executing [8@gotoexten:1] Set("SIP/2000-0000018e", "foo=") |
| 01:41.19 | jonno11 | Billiard: It's odd |
| 01:41.36 | Billiard | iunno, any module loading errors when you start asterisk? |
| 01:42.02 | jonno11 | What module will it be? |
| 01:42.17 | Billiard | just check for any module loading errors |
| 01:42.58 | jonno11 | Hmm how can I do that? |
| 01:43.09 | jonno11 | Sorry, relative n00b here |
| 01:43.13 | jonno11 | :P |
| 01:43.21 | Billiard | check logs probably or run asterisk -v and look I guess |
| 01:44.16 | jonno11 | I can't see anything |
| 01:44.30 | jonno11 | Billiard: Plus ./configure has a path for curl |
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| 03:52.24 | infinity_ | is there a website that will place a test call to an * box accepting anonymous calls? |
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| 04:03.44 | TysonBrooks | Anyone active? |
| 04:03.52 | infinity_ | boo |
| 04:04.06 | TysonBrooks | Google Voice Intergration, is it still working? |
| 04:05.00 | Billiard | TysonBrooks: crappily |
| 04:05.14 | TysonBrooks | can you elaborate? |
| 04:06.33 | Billiard | it works on and off, there are alternatives to using gvoice with asterisk directly, you can read about the troubles on the internet |
| 04:06.49 | TysonBrooks | in the wiki I'm seeing that asterisk has to be compiled with chan_gtalk / chan_motif - Yet its showing in the setup both are depreciated and no longer installed, its showing XXX |
| 04:07.03 | TysonBrooks | I wanted to use Gvoice since its been a free phone number |
| 04:10.03 | TysonBrooks | Billiard: What other alternitives are there? |
| 04:12.04 | Billiard | TysonBrooks: other programs or (free) services using the other programs :p |
| 04:12.36 | Billiard | https://michigantelephone.wordpress.com/2012/01/26/i-no-longer-recommend-using-asterisks-google-voice-support-try-these-methods-instead/ |
| 04:16.30 | TysonBrooks | Guess I'll keep researching, but continue with the install of asterisk 11, he did mention that they have tried to tweak things to make it work better. We'll see how it works. |
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| 05:05.55 | infinity_ | does anyone have documented a moh solution that streams from the local asterisk server as to not always start a person from the beginning of a track? |
| 05:06.23 | infinity_ | preferably it plays wavs and not mp3s. |
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| 06:12.14 | shadar | Hi, please help me with the following problem: |
| 06:12.14 | shadar | I have an asterisk server configured. It's writing all calls coming through into mysql-cdr table. |
| 06:12.14 | shadar | Incoming calls are going through SIP-gate from analogue phone line, with DTMS caller ID, and then through the same SIP-gate into analogue phone. |
| 06:12.15 | shadar | The thing is, that for some reason the letter "D" is added in front of each phone-number, so now the phonebook in analogue phone is not working. |
| 06:12.15 | shadar | How do I erase the D, so that un-D-ed number is written into cdr-table and sent to analogue phone? |
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| 06:37.47 | kaldemar | shadar: Set(CALLERID(num)=${FILTER(0-9,CALLERID(num))}) |
| 06:39.11 | shadar | kaldemar: thank you |
| 06:39.23 | kaldemar | shadar: other approach would be to use func REPLACE to only remove D's from the number. |
| 06:41.13 | shadar | kaldemar: Would that work: ${CALLERID(num):1} ? |
| 06:41.37 | kaldemar | if D is always the first character. |
| 06:42.14 | shadar | kaldemar: it is, thanks again |
| 06:43.02 | shadar | kaldemar: actually I'm going to use FILTER =) |
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| 07:42.25 | bombev | good morning all :) |
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| 07:42.48 | DND | hi guys i wanted to know about SIP trunking. do i still need a PRI card for it? |
| 07:45.06 | kaldemar | short answer: no. |
| 07:46.51 | DND | kaldemar, if i have freepbx, i can just setup SIP trunking with the ip user/pass right? |
| 07:47.35 | DND | do you think its more reliable than ISDN to think SIP trunking will share with the usual internet browsing,email etc |
| 07:49.22 | kaldemar | DND: you should be able to configure an ITSP connection with freepbx. for that you'll find help in #freepbx. reliability depends on your connection and usage, you'll have to test it yourself and be the judge of that. |
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| 08:08.56 | schmidts | good morning |
| 08:12.19 | bombev | good morning :) |
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| 08:31.52 | chris- | hi |
| 08:32.12 | chris- | i know im maybe wrong here for these questions |
| 08:32.30 | chris- | but is there any irc channel for phpagi questions where i could go?:) |
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| 08:49.41 | shadar | kaldemar: Me again, after setting CALLERID(num)=${FILTER(0-9,CALLERID(num)) - the analogue phone defines number as "asterisk" =| |
| 08:51.41 | kaldemar | shadar: what do you see in CLI when making a call? pastebin the output. |
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| 08:53.26 | shadar | kaldemar: http://pastebin.com/4hTnKnce |
| 08:53.54 | slash213 | btw, is it possible to filter cli output by variables? for example, only get info related to ${EXTEN}=100 |
| 08:54.14 | slash213 | i have a lot of calls coming through my box, and debugging can be an itch sometimes |
| 08:54.19 | kaldemar | slash213: no. |
| 08:54.25 | slash213 | that's sad |
| 08:55.07 | kaldemar | shadar: -- Executing [fxs@from-gw:5] Set("SIP/fxo-0000000c", "CALLERID(num)=") <-- the caller id number part gets set to an empty value. what exactly do you have in your dialplan? |
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| 08:56.58 | shadar | kaldemar: http://pastebin.com/NH2MPvkn |
| 08:57.39 | kaldemar | shadar: ${FILTER(0-9,CALLERID(num))} should be ${FILTER(0-9,${CALLERID(num)})}. my bad. |
| 08:58.01 | kaldemar | shadar: the function takes a string, not a variable/function name. |
| 08:58.32 | shadar | kaldemar: ahh.. thanks again |
| 09:00.34 | kaldemar | slash213: you can use grep though. "asterisk -vvvvr | grep \\[100@" for example. |
| 09:01.46 | slash213 | that's what i'm using right now - although simple grep like this one doesn't cover all the occurences in the debug and i have to think a bit every time i need it |
| 09:02.01 | slash213 | i thought maybe i'm missing something |
| 09:02.22 | slash213 | like, a big red button labeled 'MAKE EVERYTHING COMFORTABLE' :) |
| 09:02.31 | kaldemar | you're bettef off grepping a channel instead of an extension or a context. |
| 09:02.35 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.106) |
| 09:02.59 | slash213 | yup |
| 09:07.14 | *** join/#asterisk Kako (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 09:08.39 | Kako | Hi, I would like to set 2 SippHeaders with AMI-Originate, but only the first is been set (http://pastebin.com/XZfpmyA7). Is this not possible or is my way the wrong way? |
| 09:22.29 | Gugge | Kako: you can not set the same variable twice |
| 09:22.49 | Gugge | i expect you use the SIPADDHEADER variable in your dialplan |
| 09:23.01 | Gugge | make another variabel for the seconds header, and use that too |
| 09:27.22 | *** join/#asterisk pietro (~pietro@78.134.27.146) |
| 09:27.28 | pietro | hello |
| 09:27.48 | chris- | is it possible that I cant make two ami requests at the server at the same time? |
| 09:27.52 | pietro | I reproduced this suspended JIRA: https://issues.asterisk.org/jira/browse/ASTERISK-19883 |
| 09:27.59 | pietro | someone can re-open it ? |
| 09:28.28 | chris- | I'm programing an ami interface and wanted to make a call and get the channels all the time over a thread |
| 09:28.37 | pietro | (I already attached more details and a patch to the issue) |
| 09:28.42 | Kako | Gugge: SIPADDHEADER is not defined by me but ist the var that is automaticaly interpreted by Asterisk to add a header. But this was the right hint... When I set SIPADDHEADER2 it runs, thank you! |
| 09:28.43 | chris- | if i activate the thread and make a call it crashes sometimes |
| 09:29.50 | Kako | Gugge: this works now (http://pastebin.com/qzxvaNMY) |
| 09:39.09 | kaldemar | pietro: see the second commend of the issue. |
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| 09:43.56 | ea1het | Morning... |
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| 09:44.32 | drucik | ~books |
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| 09:51.21 | T-1000 | Hi there, I have question about cdrs and remote datbases: does anyone know what software can transport asterisk cdrs over MQ broker (RabbitMQ/Redis) into database on remote location? |
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| 10:16.19 | chris- | someone knows a good howto for setup a fastagi server? |
| 10:22.17 | pietro | kaldemar: yes, thanks I asked on #asterisk-bugs |
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| 10:48.42 | x1user | I have asterisk which loads users from external mysql database, but it doesnot use odbc nor res_mysql? I want to find out how does it loads the sipusers? |
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| 10:51.25 | kaldemar | x1user: "module show like sql". how do know that is uses mysql in the first place? also, what exactly do you mean by users? |
| 10:52.29 | x1user | sip users show all sip registration. It is some old asterisk 1.4 and I know it uses mysql because i see the databases records being updated, but the old admin is gone ... |
| 10:52.45 | x1user | module show like sql returns 0 |
| 10:55.55 | schmidts | xluser outside of asterisk (shell) try asterisk -rx"modules show" | grep -i sql |
| 10:56.01 | schmidts | i guess you will see more then |
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| 10:58.15 | kaldemar | 1.4 should have the "module show like" command. |
| 10:58.35 | kaldemar | at least 1.4.40 has it. |
| 10:59.13 | x1user | Still nothing... |
| 10:59.28 | x1user | I am starting to think this is hardcoded somewhere. |
| 10:59.40 | kaldemar | what does your extconfig.conf say? |
| 11:00.21 | x1user | sipusers => odbc,odbc_data,sifriends |
| 11:00.54 | x1user | buy isql -v myodbc3 returns ERR |
| 11:01.12 | x1user | odbc is not running at all |
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| 11:01.31 | kaldemar | "odbc" suggests that res_config_odbc is used. |
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| 11:02.09 | x1user | Yes, but if it is isql -v myodbc3 should drop me to SQL> prompt |
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| 11:02.44 | x1user | [IM002][unixODBC][Driver Manager]Der specified [ISQL]ERROR: Could not SQLConnect |
| 11:02.58 | kaldemar | why would it? |
| 11:03.59 | x1user | i cant connect to odbc withh isql, this is why i think it is not working |
| 11:04.01 | kaldemar | what does your res_odbc.conf have? |
| 11:04.22 | x1user | the dsn is myodbc3 |
| 11:04.28 | x1user | in [odbc_data] |
| 11:04.33 | x1user | all other data is correct |
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| 11:13.10 | jonno11 | What's the correct syntax to include a variable in setting a variable? |
| 11:13.38 | jonno11 | ie. Set(log=${SHELL(curl http://localhost:3000/log_call?number=${CALLERID(all)})}) |
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| 11:16.49 | mandx | Hi |
| 11:17.51 | kaldemar | jonno11: what you have there are functions, not variables. looks syntactically correct however. |
| 11:18.09 | kaldemar | jonno11: is CALLERID(all) really what you want instead of CALLERID(num)? |
| 11:18.10 | mandx | I'm trying to fix a probably quite old installation of asterisk. What could be the reason that external calls through a voip gateway work without a problem but for internal calls, I can only hear the other but the other cannot hear me? |
| 11:18.38 | mandx | Asterisk 1.6.2.9-2+squeeze4 as it seems |
| 11:18.38 | jonno11 | kaldemar: Hmm it doesn't work at all |
| 11:18.58 | jonno11 | The log variable gets set to null |
| 11:19.24 | jonno11 | however without the ${CALLERID(all)}, it works completely fine |
| 11:20.22 | *** part/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 11:20.23 | kaldemar | jonno11: add Verbose(curl http://localhost:3000/log_call?number=${CALLERID(all)}) to see what the command really is. |
| 11:24.10 | jonno11 | kaldemar: Aha, changing all to num worked a tream |
| 11:24.13 | jonno11 | treat* |
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| 11:27.52 | davlefouAMD | Hi, i have install asterisk 1.8. I have probléme with codec. http://pastebin.com/TNe3Hp6m |
| 11:32.11 | davlefouAMD | is it not native to convert ulaw and gsm? |
| 11:33.06 | kaldemar | native would mean that the endpoints speak directly to each other. that is a no go if they don't share the same codec. |
| 11:35.33 | davlefouAMD | So, why it use the slin beetwin? |
| 11:36.29 | kaldemar | disable directmedia and it will. |
| 11:40.42 | davlefouAMD | What is directmedia? |
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| 11:44.13 | kaldemar | davlefouAMD: http://svn.digium.com/svn/asterisk/tags/11.1.0/configs/sip.conf.sample |
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| 12:14.25 | davlefouAMD | is normal to use slin between codec translation? |
| 12:15.01 | kaldemar | that's the way asterisk transcodes. |
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| 12:18.35 | davlefouAMD | ok, so it works ok? |
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| 12:21.23 | chris- | someone already worked with fastagi here? I dont get it to work |
| 12:21.29 | chris- | i used these instructions |
| 12:21.30 | chris- | http://enricosimonetti.com/2009/04/27/asterisk-fastagi-with-php/ |
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| 12:40.38 | ruben231 | hi guys any idea on this error on my asterisk ------------> http://pastebin.com/AagHJPay ------------> unusual happens, over lapping audio, any idea guys |
| 12:42.23 | ruben231 | anyone have idea guys..? |
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| 14:26.59 | KSalem | hello there , I need your help as I Extracted by mistake all the records on the server into the recording directory and now I can't open it so is there any command do delete the records , note that I used the command rm *.mp3 but it gave me error -bash: /bin/rm: Argument list too long |
| 14:29.46 | WIMPy | use xargs or remove them wsing patterns. |
| 14:32.14 | kaldemar | find -maxdepth 1 -name \*.mp3 -delete |
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| 14:33.08 | wotanskrieger | hi folks |
| 14:33.40 | wotanskrieger | please, help me http://forums.digium.com/viewtopic.php?f=1&t=85097&p=181443&hilit=hipath+3800&sid=b0cfe2f025443b9f54ed4a864672cb27#p181443 |
| 14:33.43 | wotanskrieger | thanks in advance |
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| 14:40.52 | kaldemar | wotanskrieger: in /etc/dahdi/system.conf the syntax is span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow]. when the <timing source> is 0, the span will provide timing (master). if it is non-zero, dahdi takes timing from the line. use 1 for the span that is connected to hipath. |
| 14:41.43 | kaldemar | wotanskrieger: if you want to change the signalling from pri_net to pri_cpe, that is done in /etc/asterisk/chan_dahdi.conf. |
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| 14:46.18 | wotanskrieger | kaldemar: thanks for helping me. btw, I'm using a khomp board. Is it a valid setting to a non-digium hardware (according to dahdi settings)? |
| 14:51.02 | kaldemar | wotanskrieger: if the card does what DAHDI tells it to, it should work. |
| 14:54.57 | wotanskrieger | Ok, thanks kaldemar |
| 14:56.00 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 14:56.32 | WIMPy | wotanskrieger: I'd recommend a bit of upgrading first. |
| 14:56.51 | wotanskrieger | Wimpy: which kind of 'upgrading'? |
| 14:57.03 | WIMPy | Asterisk |
| 14:57.47 | saint_ | I have a question. Looking at the logs in my console, I see this : Error in extension logic (missing '}') |
| 14:57.50 | wotanskrieger | I tried to upgrade Asterisk 1.6, but khomp driver isn't compatible to. Unfortunately |
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| 14:58.08 | saint_ | and this is the line it's refering to: same => n,Set(RETURNED_VALUE=${INCOMINGCALLER_CHECK(${CALLERID(num)})} |
| 14:58.15 | WIMPy | Are you using zaptel? |
| 14:58.25 | saint_ | I have exactly the same amount of } and ) that I have ( and { . so what's wrong ? |
| 14:59.01 | WIMPy | saint_: Do you? I see one more ( than ). |
| 14:59.32 | WIMPy | wotanskrieger: Don't try 1.6 use something current. At least 1.8, buit I'd go with 11. |
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| 14:59.46 | saint_ | WIMPy: shame on me |
| 14:59.54 | saint_ | i spent the whole night on that |
| 15:00.41 | saint_ | thanks for the 2nd pair of eyes |
| 15:00.57 | wotanskrieger | Wimpy: If driver isn't compatible to 1.6, I suppose the same to 1.8+ |
| 15:01.45 | wotanskrieger | WIMPy: well, I'll try to contact Khomp. Perhaps I can get a current driver. |
| 15:02.22 | WIMPy | wotanskrieger: The driver is not part of Asterisk. It has to work with dahdi and from your post I see that's a current version. So that will work with all Asterisk versions. |
| 15:03.38 | wotanskrieger | got it |
| 15:07.10 | wotanskrieger | WIMPy: but in case of an interconnection (siemens < - > asterisk ) may I experience problems if khomp driver doesn't support higher asterisk versions? |
| 15:07.38 | saint_ | I am making a sql query on a table. what's the best way to test if it returned something and deal with it ? |
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| 15:08.04 | saint_ | Can I use something like GotoIf(${RETURNED_VALUE}?OK:NOK) ..? |
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| 15:09.35 | p3nguin | Something like it, yes. |
| 15:09.41 | wotanskrieger | WIMPy: anyway I'll try to upgrade asterisk. |
| 15:12.18 | rolandow | if i was going to install a new asterisk, would one go for 1.10 already? because the definitive guide is for 1.8, and i don't know how much has been changed.. ? |
| 15:12.39 | WIMPy | wotanskrieger: Wrong department. The driver uses dahdi and your dahdi version is a current one so it's definitely usable with Asterisk 11. |
| 15:12.40 | rolandow | i mean 11.1.0 |
| 15:13.33 | WIMPy | rolandow: Not that much. ConfBridge is probably the biggest change. |
| 15:14.03 | rolandow | ok .. is there a good URL / doc where i can catch up? |
| 15:14.25 | WIMPy | UPGRADE.txt in the source. |
| 15:14.42 | rolandow | does that explain what ConfBridge is and how it works? |
| 15:14.49 | saint_ | On a ressource point of view, is it better to do GotoIf(xxxx?AAA:BBB) with AAA and BBB in the same extension ? Or making AAA and BBB Macros or Subs ? |
| 15:14.51 | WIMPy | Apart from a few new features the book is still valid. |
| 15:15.18 | WIMPy | No. Look at the help for confbridge and it's config file. |
| 15:15.53 | wotanskrieger | WIMPy: thanks for the clarification. I think I need to study how it works a little bit |
| 15:17.22 | p3nguin | saint_: Those are supposed to be labels. Labels are used within the current extension. |
| 15:17.55 | p3nguin | But, in the past, it was always possible to use another extension and priority in place of the label. |
| 15:18.19 | saint_ | p3nguin: does it change anything regarding ressource consumption ? |
| 15:18.32 | p3nguin | I doubt it. |
| 15:18.54 | WIMPy | One stack entry. |
| 15:18.56 | p3nguin | It's a Goto any way you look at it. |
| 15:18.57 | saint_ | p3nguin: okay. i'm learning asterisk, mysql, and all that goes around live right now. never done that. we're going to use it for our fire dept. |
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| 15:20.37 | kaldemar | p3nguin: it still is really [[context,]extension,]priority. |
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| 15:21.22 | WIMPy | It has even been documented. |
| 15:21.32 | p3nguin | The app info indicates label, though. |
| 15:21.37 | p3nguin | GotoIf(condition?[labeliftrue][:labeliffalse]) |
| 15:22.18 | WIMPy | But in the [Arguments] section the meaning/options are explained. |
| 15:22.40 | saint_ | and the big difference with macro and sub is that sub can return a value ? |
| 15:22.46 | kaldemar | and below is the addition that "label" actually means [[context,]extension,]priority in this case and not only label as in a named priority. |
| 15:23.09 | p3nguin | It doesn't say that. |
| 15:23.11 | kaldemar | saint_: and Macro has been deprecated for ages. |
| 15:23.32 | saint_ | kaldemar: ha.. ok, cause im reading the 3rd edition of the asterisk book and it does not say it. |
| 15:23.32 | WIMPy | p3nguin: Need to upgrade? :-) |
| 15:23.36 | saint_ | or i missed it. |
| 15:23.38 | kaldemar | p3nguin: that's what it means. |
| 15:24.18 | p3nguin | I don't feel like I need to upgrade since I'm using a currently developed branch and what I'm using is working. |
| 15:24.19 | kaldemar | saint_: macros are so widely used that the app still remains. |
| 15:24.44 | saint_ | kaldemar: ok, good to know. i'll focus on sub then to see if it makes sens to me to use that. |
| 15:24.46 | saint_ | thanks |
| 15:35.02 | Sean-Der | I am used to mxml, and it isn't that much different. I am trying to use the ast_xml_* functions But this is giving me a null http://paste.debian.net/215780/ |
| 15:35.15 | Sean-Der | I think I am using read_memory wrong, but I can only find one other example of it in the 1.8 code base |
| 15:40.22 | saint_ | can someone tell me why Verbose( ${RETURNED_VALUE} ) gives me this: '********** 1' is not a verboser number |
| 15:40.38 | p3nguin | Yes. |
| 15:40.44 | saint_ | knowing that $RETURNED_VALUE has a value of 1,xxx,yyy,zzzz |
| 15:40.49 | saint_ | does it need to be in between " ? |
| 15:40.56 | p3nguin | Because you didn't specify the verbose value. |
| 15:41.08 | p3nguin | Verbose(level,stuff to print) |
| 15:41.26 | saint_ | hhaaa... damn it.. thanks |
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| 15:41.36 | carrar | HAPPY 121212 |
| 15:41.42 | p3nguin | If you want it to print on all levels, use Verbose(0,${stuff}) |
| 15:41.51 | saint_ | p3nguin: thank you |
| 15:41.57 | p3nguin | Only 9 more days until the end of the world! |
| 15:42.54 | carrar | yeah |
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| 16:06.06 | saint_ | can someone tell me what s wrong with this small dialplan: http://pastebin.com/NS9LdctM |
| 16:06.16 | saint_ | that's the result of my logs: http://pastebin.com/stYRKmWX |
| 16:06.43 | saint_ | in the subs, I have a Verbose command that does not show up, so I believe the dialplan is not calling those subs or can t reach them ? |
| 16:07.56 | blitzrage | saint_: don't use Goto() |
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| 16:08.03 | blitzrage | you need to use ExecIf() |
| 16:08.11 | blitzrage | GoSub() is an application, not a destination |
| 16:08.28 | blitzrage | basically, exact same statement, but replace GotoIf() with ExecIf() |
| 16:08.31 | blitzrage | then it should work |
| 16:09.17 | p3nguin | GotoIf(${RETURNED_VALUE}?GoSub(subCallerOK,start,1(${RETURNED_VALUE}):GoSub(subCallerNOK,start,1(${RETURNED_VALUE})) <--- all wrong |
| 16:09.28 | kaldemar | or GoSubIf |
| 16:09.34 | blitzrage | kaldemar: +1 |
| 16:09.38 | p3nguin | I think that would be more appropriate. |
| 16:09.39 | blitzrage | that actually makes more sense :) |
| 16:10.09 | blitzrage | GoSubIf($[${RETURNED_VALUE}]?subCallerOK,start,1:subCallerNOK,start,1) |
| 16:10.49 | saint_ | ha.. gosubif is not in the book. |
| 16:11.11 | p3nguin | Not every detail can be in the book. |
| 16:11.38 | blitzrage | it is expected that you might actually look for an application once you know how to do things :) |
| 16:12.47 | p3nguin | core show applications like sub |
| 16:13.15 | saint_ | why do you use $[${RETURNED_VALUE}] instead of ${RETURNED_VALUE} by itself ? |
| 16:13.27 | p3nguin | Evaluate it rather than use the actual value. |
| 16:14.20 | p3nguin | If you wrap it with $[], you'll return the more appropriate 0 or 1 that the GosubIf app expects. |
| 16:14.24 | blitzrage | in this case, probably not needed if you know you're going to return 0 or 1 |
| 16:14.40 | saint_ | hhaaaaa, ok |
| 16:14.48 | blitzrage | didn't look carefully enough to determine what your RETURNED_VALUE contained |
| 16:14.59 | p3nguin | Any time you are looking for true or false, it seems using $[] is always a good idea. |
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| 16:15.27 | saint_ | p3nguin: gotcha, thanks |
| 16:17.22 | jonno11 | Hi, it's me again. When calling this extension http://pastebin.com/hJTYT1Kv the call ends with "Auto fallthrough, channel 'SIP/sipgate-000001bd' status is 'UNKNOWN'" Why is this? |
| 16:18.14 | p3nguin | Wrong syntax. |
| 16:18.25 | p3nguin | same => n,App() |
| 16:18.44 | jonno11 | So I shouldn't use 2,3? |
| 16:18.48 | p3nguin | Actually, I don't know that... |
| 16:18.51 | jonno11 | When using same |
| 16:18.58 | p3nguin | I have never tried using same and numbered priorities. |
| 16:19.08 | jonno11 | Hmm I'll see if it fixed the issue |
| 16:19.10 | p3nguin | I spoke on that before I thought about it. |
| 16:19.21 | saint_ | same => n,Hangup() maybe ..? |
| 16:19.31 | p3nguin | And your hangup line is lacking priority. |
| 16:20.28 | jonno11 | Sorry I actually wrote that in paste bin :P |
| 16:20.47 | p3nguin | You almost never need numbered priorities, though. It's an old style and it complicates making complex changes later. |
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| 16:21.11 | jonno11 | p3nguin: Yeah I do see that, they were actually only numbered when I was debugging this issue |
| 16:21.11 | navaismo | same => n,app() its ok |
| 16:21.23 | jonno11 | to see if that was the issue |
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| 16:21.43 | saint_ | p3nguin: with your GosubIf , how to you pass arguments ? |
| 16:21.59 | jonno11 | but dialplan show says everything is fine |
| 16:22.06 | blitzrage | you can use same => and numbered priorities, but it doesn't really make sense :) |
| 16:22.16 | blitzrage | if you're using same => n, then just use priority labels |
| 16:22.20 | jonno11 | Either way, my issue is still occurring ;( |
| 16:22.37 | blitzrage | saint_: what do you mean "pass arguments" |
| 16:22.53 | blitzrage | GoSubIf($[...]?subRoutine,start,1(foo,bar)) |
| 16:23.02 | blitzrage | should just act like that I believe |
| 16:24.06 | saint_ | yes, just found it. |
| 16:24.58 | jonno11 | I'm getting this: http://pastebin.com/fTeue2qW When calling this: http://pastebin.com/hJTYT1Kv |
| 16:26.48 | blitzrage | jonno11: ${test} contains no data |
| 16:27.10 | saint_ | mmhh.. now i have this funky error: ast_yyerror(): syntax error: syntax error, unexpected ',', expecting $end; Input: |
| 16:27.30 | jonno11 | blitzrage : Yeah, it should contain the data entered by the caller, but it hangs up straight after playing hello-world |
| 16:27.51 | jonno11 | Well that's what I'd like it to do... |
| 16:28.25 | blitzrage | saint_: typo |
| 16:28.35 | blitzrage | you have an extra comma somewhere asterisk isn't expecting it |
| 16:31.00 | saint_ | blitzrage: in : GoSubIf($[${RETURNED_VALUE}]?subCallerOK,start,1(${RETURNED_VALUE}):subCallerNOK,start,1(${RETURNED_VALUE})) , my $RETURNED_VALUE has fields separated with comas... |
| 16:31.20 | wdoekes | saint_: bantime = 86400 |
| 16:31.20 | wdoekes | findtime = 86400 |
| 16:31.23 | wdoekes | drat |
| 16:31.34 | blitzrage | saint_: then you need them to be escaped |
| 16:31.36 | dr0ck | jonno11 are you hitting # |
| 16:32.05 | saint_ | blitzrage: I can't do something like put " around it ? |
| 16:32.20 | blitzrage | no, because " is just a literal character like A |
| 16:32.30 | blitzrage | foo\,bar\,baz |
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| 16:32.41 | saint_ | so i need to create a function for that ? |
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| 16:32.52 | blitzrage | if returned value is just text, it will always be true |
| 16:33.03 | wdoekes | unless it is "0" |
| 16:33.04 | blitzrage | because $[ ] will just see "hey i got text" and return 1 |
| 16:33.14 | blitzrage | wdoekes: yes :) |
| 16:33.31 | saint_ | mmhh.. ok.. i will see if I can work with a mysql qery that return only 1 field from the table instead of all.. |
| 16:33.37 | saint_ | thanks for the explanation |
| 16:33.49 | blitzrage | saint_: there are substitute functions that you could probably change , to \, |
| 16:34.03 | blitzrage | saint_: if this is returning from the database: |
| 16:34.20 | wdoekes | but what would $[a,b,c] mean? |
| 16:34.20 | blitzrage | Set(ARRAY(field1,field2)=${ODBC_GET_DATA(whatever)}) |
| 16:34.34 | blitzrage | ya, you're not evaluating anything with that string |
| 16:34.44 | blitzrage | youre just saying "is there data" basically |
| 16:34.59 | wdoekes | but then he wouldn't have to pass 1(${RETURNED_VALUE}) in the fail case |
| 16:35.04 | wdoekes | since it'd be "" |
| 16:35.27 | blitzrage | wdoekes: i think the whole logic is weird and needs to be re-evaluated |
| 16:35.47 | saint_ | Oh, you know what I will do .. I will call the subs without arguments |
| 16:36.00 | saint_ | I already know when I call them if there is something in the database or not |
| 16:36.17 | saint_ | the subs only need to take keyboard input from the phone, and update the DB after |
| 16:36.22 | saint_ | it should work like that.. thanks blitzrage |
| 16:39.13 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 16:40.33 | p3nguin | I'm not sure you even need a sub at all for that. |
| 16:43.18 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 16:44.13 | jonno11 | dr0ck: Nope, it quits instantly after playing hello-world |
| 16:44.26 | jonno11 | I don't even get the chance to enter a number |
| 16:44.33 | saint_ | I still have the error when I call with a valid number. Error is at http://pastebin.com/5NCyAh1e , dialplan is at http://pastebin.com/nUCeavSm |
| 16:44.51 | p3nguin | jonno11: Did you pastebin the verbose output on that? |
| 16:45.32 | jonno11 | p3nguin: Yeah, I'm getting this: http://pastebin.com/fTeue2qW When calling this: http://pastebin.com/hJTYT1Kv |
| 16:46.31 | p3nguin | Is '4' the file name that you intend to play in the Read()? |
| 16:46.38 | p3nguin | I'd guess it isn't. |
| 16:46.40 | jonno11 | Ah no |
| 16:46.48 | p3nguin | What is the 4? |
| 16:46.54 | jonno11 | dr0ck just pm'd me with that suggestion |
| 16:46.59 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 16:47.01 | jonno11 | I'm an idiot |
| 16:47.16 | p3nguin | PMs aren't helpful to others who are in the channel trying to learn things. |
| 16:47.26 | jonno11 | Yeah, It's supposed to be the required number of digits |
| 16:47.48 | p3nguin | Then you probably want Read(test,,4) |
| 16:48.29 | p3nguin | You can eliminate that Playback() by using Read(test,hello-world,4). |
| 16:48.46 | jonno11 | Awesome. Thanks p3nguin + dr0ck, v v helpful! |
| 16:48.57 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:51.53 | *** join/#asterisk dorphalsig (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 16:51.59 | dorphalsig | Hello |
| 16:52.34 | p3nguin | saint_: What is this INCOMINGCALLER_CHECK() thing? |
| 16:53.21 | saint_ | p3nguin: http://pastebin.com/dtGjBBqw , it is in func_odbc.conf |
| 16:53.31 | dorphalsig | Im trying to build debian packages for asterisk-11, dahdi-2.6 and libpri |
| 16:54.00 | p3nguin | So it is a registered function, then? |
| 16:54.38 | saint_ | i don t know what is a registered function.. sorry i m new at this. it s a sql query that is in the odbc.conf file, as in the example in the book. |
| 16:55.14 | dorphalsig | so I downloaded the debian svn version of all the debian package files |
| 16:55.24 | dorphalsig | but it breaks when compiling |
| 16:55.52 | dorphalsig | patching file drivers/dahdi/Kbuild Hunk #1 FAILED at 31. 1 out of 1 hunk FAILED -- saving rejects to file drivers/dahdi/Kbuild.rej |
| 16:56.00 | dorphalsig | any ideas what can it be? |
| 16:56.30 | p3nguin | saint_: If the number is in the DB, what value does your INCOMINGCALLER_CHECK() function return? And what if the number does not exist in the DB? |
| 16:57.04 | p3nguin | dorphalsig: The patch isn't valid for that version of dahdi. |
| 16:57.13 | drucik | d |
| 16:57.39 | saint_ | p3nguin: it return the whole table about this number (index,phone_number,last_call_in,estimated_time_of_arrival,number_of_time_caller_called) |
| 16:57.56 | saint_ | but since I put it in $[ ] like you guys just showed me, it return 1 if it has something in it |
| 16:58.03 | saint_ | or return NULL if is has nothing. |
| 16:58.20 | saint_ | that works.. if I have a valud number, i see in the logs that I go in the subCallerOK |
| 16:58.30 | saint_ | and if it is not in the DB, I go to the subCallerNOK |
| 16:58.44 | p3nguin | You're setting the variable to null and then evaluating the null variable? |
| 16:58.46 | saint_ | the only thing is that if the number exists, then i get the error message |
| 16:58.46 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 16:58.49 | dorphalsig | p3nguin: what do you think could be easier... to build everything from scratch or use the debian svn thingy? |
| 16:58.59 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 16:59.07 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 16:59.30 | saint_ | p3nguin: I am not setting a null variable, I am evaluating a variable. if it has something in it, then it goes to subCallerOK . if not, it goes to subCallernOK |
| 17:00.01 | saint_ | I mean yeah, through the SET I set it. but that is working. |
| 17:00.10 | saint_ | the error is when a variable exists in the table |
| 17:00.11 | p3nguin | I see what you're doing, but I don't understand why you're doing it. |
| 17:00.52 | p3nguin | Set(RETURNED_VALUE=$[${INCOMINGCALLER_CHECK(${CALLERID(num)})}]) Here, you are setting RETURNED_VALUE to either 1 or 0, right? |
| 17:00.59 | saint_ | right |
| 17:01.09 | saint_ | and that part is working |
| 17:01.37 | saint_ | 1, his number exists in the database ; 0, his number does not |
| 17:01.59 | p3nguin | And then in the next line, you are again evaluating the variable to 1 or 0. 0 is still a value. |
| 17:02.12 | p3nguin | So if the variable's value is 0, that is a value... and then you evaluate it to either 1 or 0 again, and it will always be 1. |
| 17:02.25 | saint_ | ha.. good point.. |
| 17:02.37 | saint_ | so I need to remove the [ ] on the 2nd line then.. i guess ..? |
| 17:02.46 | p3nguin | I would not evaluate it on the Set(). |
| 17:02.50 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 17:03.01 | p3nguin | Set it to a value or set it to null. |
| 17:03.07 | saint_ | ok, let me change it and test it |
| 17:03.16 | *** join/#asterisk navaismo (~Administr@189.144.128.152) |
| 17:03.20 | p3nguin | Then the next line can evaluate it to see if it contains something or nothing. |
| 17:05.04 | saint_ | i still get the error if a value is in it |
| 17:05.20 | p3nguin | Did anyone ever make a SetIf() or does it still need to be done with Set(IF()) or ExecIf(...?Set())? |
| 17:07.36 | saint_ | ok so I manually tried to SET a value, and it looks like it does not like if there is a coma in it, or a space |
| 17:07.43 | blitzrage | p3nguin: one of those options is the approach, yes |
| 17:08.20 | p3nguin | saint_: I'm thinking about how to do it with EXISTS() and/or ISNULL() to eliminate that. |
| 17:08.44 | saint_ | confirmed: it works if I do Set(XXX="this is a test") , but will give me an error if I do Set(XXX=this is a test) |
| 17:08.45 | blitzrage | saint_: you need to escape it if you want to literally write it |
| 17:09.04 | p3nguin | Quotes are in error, yes. |
| 17:10.00 | p3nguin | Try using EXISTS(${INCOMINGCALLER_CHECK(${CALLERID(num)})}) somehow. |
| 17:10.11 | *** join/#asterisk navaismo (~Administr@189.144.128.152) |
| 17:10.15 | saint_ | what about if I do Set(RETURNED_VALUE=$[${INCOMINGCALLER_CHECK(${CALLERID(num)})}]) |
| 17:10.17 | p3nguin | That will check if the value or your INCOMINGCALLER_CHECK is true or false. |
| 17:10.25 | saint_ | and then test it with GoSubIf(1?subCallerOK,start,1:subCallerNOK,start,1) |
| 17:10.33 | blitzrage | saint_: that will just set 0 or 1 |
| 17:10.52 | saint_ | yeah, but then the gosubif is right now, right ? if i test with 1 before the ? |
| 17:11.02 | blitzrage | why aren't you just using EXISTS() ? |
| 17:11.10 | p3nguin | The GosubIf I saw looks okay. |
| 17:11.32 | saint_ | damn it. |
| 17:11.43 | p3nguin | The problem lies within your usages of it and the previous Set(). |
| 17:11.54 | p3nguin | But I already went over that part. |
| 17:12.03 | saint_ | the $[${xxx}} expression does not like the spaces and comas again in xxx |
| 17:12.27 | saint_ | I thought that if I added [ ] , the set would see a 1 or 0 , and would not care about what is being evaluated |
| 17:12.36 | p3nguin | Show me what happens if you use EXISTS(${INCOMINGCALLER_CHECK(${CALLERID(num)})}) in your Set. |
| 17:13.01 | p3nguin | $[] causes the evaluation of something or nothing and returns 1 or 0. |
| 17:13.26 | p3nguin | So if you Set(foo=$[]), it will ALWAYS CONTAIN A VALUE, even if the value is 0. |
| 17:13.44 | p3nguin | It will always contain either 1 or 0, depending on what you just evaluated. |
| 17:13.56 | p3nguin | So your variable will always be 1 or be 0. |
| 17:14.13 | p3nguin | And then if you evaluate the variable, it will always evaluate true because both 1 and 0 are values. |
| 17:14.24 | saint_ | i understand that |
| 17:14.40 | p3nguin | So don't do that! |
| 17:14.40 | saint_ | but it looks like the issue is the space and comas in what i evaluate |
| 17:14.55 | saint_ | so i thought if I put [ ] around this variable , it will only look at the 1 or 0 |
| 17:14.57 | p3nguin | It probably is. Why aren't you trying EXISTS() like I said twice already? |
| 17:15.13 | saint_ | but the fact that i still have , and space in what's evaluated, asterisk does not like it |
| 17:15.18 | saint_ | i m trying it now |
| 17:15.20 | saint_ | stand by |
| 17:16.07 | saint_ | same thing |
| 17:16.10 | p3nguin | I don't think EXISTS() will care if you have commas or spaces. If it has value, it exists; otherwise it does not exist. |
| 17:16.33 | saint_ | this is what you want me to try right: Set(RETURNED_VALUE=EXISTS($[${INCOMINGCALLER_CHECK(${CALLERID(num)})}])) |
| 17:16.44 | saint_ | oh hold on |
| 17:16.51 | saint_ | i saw the [ ] arestill here |
| 17:16.52 | p3nguin | No. |
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| 17:17.19 | p3nguin | Set(RETURNED_VALUE=${EXISTS(${INCOMINGCALLER_CHECK(${CALLERID(num)})})}) |
| 17:17.22 | p3nguin | or |
| 17:17.34 | p3nguin | Forget the set entirely and do it all in the GotoIf. |
| 17:17.55 | p3nguin | or GosubIf, rather. |
| 17:18.15 | saint_ | p3nguin: you are the man |
| 17:18.24 | saint_ | this does not give me an error anymore |
| 17:18.24 | saint_ | exten => ${EFIRE},1,Set(RETURNED_VALUE=${EXISTS(${INCOMINGCALLER_CHECK(${CALLERID(num)})})}) |
| 17:18.45 | p3nguin | GosubIf($[${EXISTS(${INCOMINGCALLER_CHECK(${CALLERID(num)})})}]?labeliftrue:labeliffalse) |
| 17:19.08 | saint_ | why do you put ${ before EXISTS ? |
| 17:19.23 | p3nguin | It's a function and you have to treat it like a variable. |
| 17:19.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:19.30 | saint_ | mmhh.ok |
| 17:20.10 | p3nguin | But serious, this doesn't need a sub at all. |
| 17:20.42 | p3nguin | This can be evaluated and executed based on the outcome in the same extension without subroutines. |
| 17:21.38 | p3nguin | Your subs never return, so there isn't a lot of point in them. |
| 17:21.51 | p3nguin | Subroutines are meant to be returned back to where they came from. |
| 17:22.46 | saint_ | i just thought it would make the extensions.conf file more clear |
| 17:22.59 | saint_ | since I have to do some database handling in both cases |
| 17:23.24 | p3nguin | It doesn't. |
| 17:24.11 | p3nguin | What happens when a caller is determined to be NOK? |
| 17:24.48 | saint_ | a specific voice guide is going to play, and will give the possibility to leave a voicemail |
| 17:24.59 | saint_ | hold on, BRB.. need to take my dog out |
| 17:27.31 | p3nguin | saint_: http://pastebin.com/R7Wy5tRf |
| 17:28.04 | p3nguin | No subroutines needed. |
| 17:29.56 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 17:30.45 | saint_ | p3nguin: whoa, i like yours way better. how long have you been working with asterisk ? |
| 17:31.17 | p3nguin | I'm not sure. Maybe five years. |
| 17:34.04 | saint_ | the only thing i like about subs is that if the code needs to extends in the future, it will keep things clear no ? I mean, your code is freaking neat.. but granted it does not grow up too much... |
| 17:34.52 | p3nguin | Here's another variation of the same thing: http://pastebin.com/JbExw7mh |
| 17:35.31 | p3nguin | Subs are meant to return after the subroutine is completed. What you were doing was starting a subrouting and then killing it before you returned it. |
| 17:36.01 | p3nguin | s/subrouting/subroutine/ |
| 17:36.23 | p3nguin | If you're going to kill it without returning, don't use Gosub, just use Goto. |
| 17:36.46 | p3nguin | There's no problem with using Goto and sending the call to another location such as those two other contexts that you created. |
| 17:37.37 | p3nguin | I'll show you a variation using other contexts. |
| 17:39.29 | p3nguin | http://pastebin.com/ecVvuPgt |
| 17:40.46 | p3nguin | ... just to reinforce what I said about using Goto and not Gosub, this uses GotoIf rather than GosubIf. |
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| 17:43.07 | saint_ | p3nguin: that's super clear. |
| 17:43.11 | saint_ | p3nguin: thanks a ton .. |
| 17:44.06 | p3nguin | So now you have at least three variations to try out and see which way feels the best. |
| 17:44.53 | p3nguin | I definitely understand your concern with scaling the single extension method. |
| 17:47.23 | p3nguin | You could also use subroutines in the case that much of your dialplan would be duplicated for both conditions. |
| 17:47.35 | saint_ | I'll keep your last example, it's the best one. |
| 17:47.46 | saint_ | Now.. I have another question... this is a SIP trunk that I have |
| 17:47.49 | *** join/#asterisk minotaur01 (~minotaur@bas8-hamilton14-3096451379.dsl.bell.ca) |
| 17:47.52 | p3nguin | There is no SIP trunk. |
| 17:47.58 | saint_ | hu ? |
| 17:48.05 | p3nguin | ~trunk |
| 17:48.05 | infobot | rumour has it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 17:48.21 | p3nguin | SIP has no trunk capability. |
| 17:48.43 | saint_ | so what's the terminology for it ? |
| 17:48.52 | p3nguin | Depends on what you mean. |
| 17:50.04 | saint_ | in isdn work, you call a T1 a trunk . |
| 17:50.22 | saint_ | I thought 25 sip channels going to a PBX would be considered as a ttrunk too |
| 17:50.29 | p3nguin | If you're trying to describe a peer, it is safe to call it a peer. |
| 17:51.01 | saint_ | a trunk to me is a circuit connecting telephone equipments. a circuit being made of channels. |
| 17:51.08 | saint_ | i guess you call that a peer then ? |
| 17:51.38 | p3nguin | If those 25 channels were carried over a single pipe, I might call it a trunk. In SIP, there is no trunking; every channel is the same as every other channel. |
| 17:52.22 | p3nguin | IAX2 does trunking. There is a single "pipe" between you and the peer where multiple calls take place. |
| 17:52.41 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 17:52.47 | *** join/#asterisk minotaur01 (~minotaur@HMTNON14-1176243898.sdsl.bell.ca) |
| 17:52.55 | saint_ | in the digital world, 1 channel = 1 call only |
| 17:53.07 | p3nguin | The trunk is established with the first call, and subsequent calls between you and the peer are trunked within. |
| 17:53.13 | saint_ | granted they are in the same pipe, but it's 23 different channels |
| 17:53.21 | p3nguin | Exactly. In sip, one channel carries 1 call. |
| 17:53.34 | saint_ | so if I have 25 sip channels, that does not make a sip trunk ? |
| 17:53.38 | p3nguin | Nope. |
| 17:53.38 | Qwell | saint_: So you're suggesting that all your HTTP traffic is trunked? |
| 17:53.58 | p3nguin | It means that you have 25 AVAILABLE channels that the peer allows you to use. |
| 17:54.01 | saint_ | Qwell: mmhh.. i never looked at it this way.. |
| 17:54.12 | p3nguin | It isn't a bundle of 25 channels. |
| 17:54.22 | p3nguin | It's 25 individual channels existing exclusively from each other. |
| 17:55.15 | saint_ | damn . all carriers are calling those sip pipes SIP TRUNKS . |
| 17:55.31 | p3nguin | They say a lot of things that aren't true. |
| 17:55.38 | saint_ | p3nguin: yeah, i just realized . |
| 17:55.40 | Qwell | "unlimited" |
| 17:56.10 | saint_ | they do the trend though. if you go on the field, end users, customers, carriers , pbx manufacturers call those a sip trunk |
| 17:56.16 | saint_ | so what's the good terminology ? |
| 17:56.31 | Qwell | call, channel, peer, pick several |
| 17:56.35 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.106) |
| 17:56.40 | Qwell | account |
| 17:57.01 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:58.07 | saint_ | so anyway.. |
| 17:58.16 | saint_ | I have a couple of sip channels ... :D |
| 17:58.47 | saint_ | when i call my asterisk from an analog line , i get into the callerOK sub, which is good. |
| 17:59.02 | saint_ | in it, I have a Wait(3), SayDigits(1), and Hangup() |
| 17:59.12 | [TK]D-Fender | Acutally a channel is an actual established path.... |
| 17:59.27 | saint_ | mmh.. hold on, let me try something before I say it |
| 17:59.30 | [TK]D-Fender | Technically a call uses channels. They are not reserved however |
| 17:59.42 | [TK]D-Fender | So probbaly best to just say "peer account", etc |
| 18:00.23 | saint_ | [TK]D-Fender: i'll go for it from now on, but it s going to be a hard work to change the telephony guys from sip trunk to this. |
| 18:01.20 | blitzrage | people get all hung up on "sip trunks" not being a thing, and they're not, but we all know what they mean |
| 18:01.48 | *** join/#asterisk navaismo (~Administr@189.144.128.152) |
| 18:02.54 | jacekowski | Qwell: but those channels use same SIP "connection" for control data |
| 18:02.58 | jacekowski | Qwell: so that's trunked |
| 18:04.06 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:06.05 | [TK]D-Fender | jacekowski, Depends on your idea of "connection. That vast majority of SIP is UDP which is stateless. Each channel's signalling is unique and not tied to any other channel. That breaks the idea of "turnk" right up. |
| 18:06.44 | saint_ | well.. |
| 18:06.56 | saint_ | I think when you talk about trunk too you are thinking about a pipe |
| 18:07.05 | saint_ | whether the channels depends on each other or not |
| 18:07.11 | saint_ | like for a sip trunk, in my mind |
| 18:07.19 | saint_ | it's a pipe that goes from the carrier to my pbx |
| 18:07.26 | saint_ | with 25 channels in my case |
| 18:07.37 | saint_ | granted they are independant and in the internet by themselves |
| 18:07.51 | [TK]D-Fender | channel isn't in the same pipe. there is no real container. |
| 18:07.53 | saint_ | it's still 25 simlutaneous connections that i can have from my pbx to the carrier |
| 18:08.02 | saint_ | i understand that |
| 18:08.21 | saint_ | but the way i see it from the digital telephony world, applying it to the sip world |
| 18:08.26 | saint_ | it's 25 channels from the carrier to me |
| 18:08.42 | saint_ | whether they take the same route, come in the same router or not |
| 18:08.45 | [TK]D-Fender | saint_, It is as blitzrage mentioned, a nearly worthless distinction to make and nothing to neurose about when dealing with ITSPs, etc |
| 18:08.57 | saint_ | all i can have is 25 simultaneous , and thos 25 together make a trunk to my eyes |
| 18:09.06 | saint_ | which is where i might be wrong, but that is the way i used to see it |
| 18:09.21 | saint_ | agreed |
| 18:09.47 | [TK]D-Fender | saint_, I don't really care much for the conceptual difference in calling SIP as a "trunk". IAX however has a special meaning for the term so I like to ensure the term is used right there. |
| 18:09.49 | p3nguin | jacekowski: If it is a different call, it does not use the same SIP "connection" for the control data... because it isn't trunked. IAX2, on the other hand, does behave that way because of trunking. |
| 18:11.09 | [TK]D-Fender | p3nguin, a PRI is a trunk in the sense that the channel paths are there, utilized or not. with VoIP being allocated dynamically, and perhaps unsuccessfuly depending on BW and your point of view, when you have no calls in progress.... is IAX2 still considered a tryunk? :) |
| 18:11.26 | [TK]D-Fender | p3nguin, No calls = no pipe, empty or otherwise |
| 18:11.46 | p3nguin | If you have more than one calls in progress, the trunk is literal. |
| 18:12.23 | p3nguin | At zero or one calls, it's not yet an actual trunk. |
| 18:13.44 | [TK]D-Fender | Schroedinger's Trunk :p |
| 18:13.53 | Snivets | lol |
| 18:14.03 | p3nguin | I'm almost afraid to look up that. |
| 18:16.06 | p3nguin | saint_: Was there any more part to your issue that you were having? |
| 18:17.04 | citywok | is there a way to disable events in the AMI stream from manager.conf, rather than specifying events: off? |
| 18:17.16 | saint_ | I'm good for now, thank you very much. I am going to continue reading the book, and see if I can figure the rest out by myself. my biggest stuff was to query the database, and react depending on if a number was present or not. i need to learn about mysql now, and php or javascript ... do you do freelance work by anychance ? |
| 18:17.26 | citywok | alternatively, does anybody know of an option in Asterisk.NET that can allow you to specify that? I'm fixing somebody elses really awful code, and that owuld make life easier. |
| 18:17.30 | blitzrage | citywok: i think you can control that with the permit and deny functionality? |
| 18:18.07 | citywok | blitzrage: probably, but i was trying to avoid having to make permissions changes that could have unintended effects. granted events: off could too... lol |
| 18:18.31 | p3nguin | I don't know jack about javascript and very little about PHP. I hear rentacoder has plenty of people looking to do work, though. |
| 18:18.36 | blitzrage | citywok: in 1.8 there is also eventfilter |
| 18:18.53 | saint_ | p3nguin: i was more thinking about asterisk |
| 18:19.08 | blitzrage | citywok: try the eventfilter stuff -- i think that'll do what you want |
| 18:19.28 | blitzrage | also, i meant read/write, not permit/deny I guess :) |
| 18:19.36 | blitzrage | but I think those are for actions |
| 18:19.36 | p3nguin | I definitely work with asterisk. Depends on what you want done if I have time or knowledge to do it. |
| 18:19.36 | blitzrage | not events |
| 18:19.48 | saint_ | p3nguin: can I PM you ? |
| 18:19.56 | p3nguin | Yes |
| 18:20.30 | SuperNull | p3nguin can i talk in the channel with you.. possibly lite conversations about packets.. and that kinda jazz. |
| 18:20.46 | p3nguin | I suppose. |
| 18:21.21 | SuperNull | sweet. im to busy sorry man maybe we can talk later. |
| 18:21.55 | p3nguin | Other people know about packets, as well. Just ask your questions when you have time. |
| 18:22.12 | SuperNull | twas more of a joke :) .. |
| 18:22.15 | SuperNull | im on irc all day. |
| 18:22.19 | SuperNull | we know i have plenty of time. |
| 18:22.33 | SuperNull | except for this 50meg microwave that went down. |
| 18:25.54 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
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| 18:41.31 | saint_ | why when i receive a phone call over a sip channel, if I do SayDigits(123) then Hangup() , I dont hear anything , but if I do Playback(xxx) SayDigits(123) Hangup() , then I hear the playback and the digits ? It looks like if there is nothing before the Saydigits , i do not hear anything .. |
| 18:42.06 | p3nguin | You need to bring the channel "Up" first. Bring it up, as in answer it. |
| 18:42.46 | p3nguin | Answer() before the SayDigits() would do it. Playback() has an implicit answer, so that's why it worked with a Playback() first. |
| 18:44.22 | saint_ | i wish those stuff were in the book. thanks again |
| 18:44.53 | *** join/#asterisk lukejt (~luke@149.241.199.189) |
| 18:45.09 | p3nguin | I admit that I have never read the books in whole. |
| 18:47.04 | p3nguin | I usually use them for reference if I can't find the information within asterisk itself, though. |
| 18:49.47 | saint_ | where can i find the list of all the commands / application / macro , etc that is available for asterisk ? |
| 18:49.50 | saint_ | voip-info ? |
| 18:50.05 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 18:50.31 | p3nguin | core show applications |
| 18:50.35 | p3nguin | core show functions |
| 18:50.49 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 18:51.16 | p3nguin | ~asteriskwiki |
| 18:51.16 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
| 18:51.48 | p3nguin | voip-info.org is mostly outdated, so you have to already know what you're looking at to know if it is pertinent in many cases. |
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| 18:54.36 | saint_ | p3nguin: i m trying to play with read() , can you tell me why this is not working ? |
| 18:54.37 | saint_ | http://pastebin.com/eZGk6U32 |
| 18:55.01 | saint_ | In the logs, i have the 2 verboses, playback, and hangup |
| 18:55.07 | saint_ | i never see the read being called |
| 18:55.25 | p3nguin | Did you run dialplan reload? |
| 18:56.00 | [TK]D-Fender | saint_, same => Answer(1000) <- no PRIORITY |
| 18:56.09 | [TK]D-Fender | 4 lines like that there |
| 18:56.24 | [TK]D-Fender | Pay attention to your consistency |
| 18:56.30 | saint_ | damn it damn it damn it |
| 18:56.34 | p3nguin | Yeah, that would pose a problem. |
| 18:56.37 | saint_ | [TK]D-Fender: thanks |
| 18:57.48 | saint_ | is there anything to do for asterisk to be able to "read" dtmf ? |
| 18:58.02 | saint_ | my Saydigits in the log shows "" instead of the "5" |
| 18:58.17 | saint_ | and I pushed 5 after I saw the "read" being called |
| 19:01.59 | lukejt | generally speaking, if a termination provider bills based on peak/offpeak, would it be calculated based on the time the call was initiated, or the answered time? |
| 19:03.24 | lukejt | e.g. you call someone at 5:59pm and 55 secs, the person answers at 6:00pm and 10 seconds.. |
| 19:04.19 | saint_ | lukejt: I would say at the time the call is established with the party you are calling |
| 19:05.02 | saint_ | but it would be better to ask them |
| 19:06.40 | saint_ | if I receive only incoming call into my sip channel, do I need to do a peer connection, or is a register suffisant enough ? |
| 19:12.11 | *** join/#asterisk navaismo (~Administr@189.144.211.20) |
| 19:13.28 | lukejt | saint_: the register will get calls pointed at your asterisk box, whether or not you need a peer depends on what you want to do with the incoming calls. if you don't have a peer set up to identify them, they will go into your default context |
| 19:13.50 | lukejt | afaik |
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| 19:22.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 19:38.45 | p3nguin | lukejt: You shouldn't be billed for unanswered time. |
| 19:40.23 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 19:40.32 | p3nguin | saint_: You use a register statement to tell another device how to reach you. You use a peer definition to handle calls from the other device. |
| 19:44.10 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:48.42 | danfromuk | Is there a channel for discussing voip related business? |
| 19:48.53 | lukejt | p3nguin: what I mean is, if 8am-6pm is peak and 6pm-8am is offpeak, you dial in peak but the call is answered in off-peak, which would apply |
| 19:49.10 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:50.24 | p3nguin | You'd have to ask the ITSP. |
| 19:51.23 | danfromuk | Has anyone worked with DIDX? |
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| 20:39.59 | LedZeplin | I'm working on a channel spy thinggy of which I would like to have a user enter an extension (got that part) but I also only want it to accept the extension if it's in a list, or a ring group, or something. |
| 20:40.45 | LedZeplin | I'm not sure how to go about restricting the allowed to be spied upon channels. |
| 20:41.13 | p3nguin | core show application ChanSpy |
| 20:41.57 | p3nguin | It won't work to see if the extension is in a list; that's what the dial plan does. |
| 20:42.13 | p3nguin | But you can pick a channel from a group. |
| 20:44.42 | p3nguin | Also, if you'd rather spy based on an extension rather than a channel, consider ExtenSpy instead of ChanSpy. |
| 20:45.09 | *** join/#asterisk MarkMMMM (~MarkMM@69.197.65.222) |
| 20:46.24 | MarkMMMM | I got a quick qustion involving alsa, and sound output thrut he soundcard for a pa system. |
| 20:48.00 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 20:48.08 | WIMPy | ~ask |
| 20:48.08 | 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. |
| 20:49.46 | MarkMMMM | I am using alsa to put paging thru my PA system. All configured. Works. Just the sound output sounds like it is talking thru a fan. If i use aplay to play a wav file it is crystal clear. Is there any ideas on this? |
| 20:49.56 | LedZeplin | p3nguin: thanks |
| 20:50.39 | p3nguin | Could be a codec issue. |
| 20:51.08 | WIMPy | I have had issues in older versions. But I can;t remember the details. |
| 20:51.17 | MarkMMMM | ok. didn't think of codec |
| 20:59.56 | lukejt | Does asterisk allow for md5 salt? |
| 21:00.34 | p3nguin | In what capacity? |
| 21:03.07 | *** join/#asterisk cyborg-one (~cyborg-on@79.140.7.247) |
| 21:03.19 | lukejt | p3nguin: for peer authentication, storing plaintext passwords is a no-go but there's the option of providing an md5 hash instead for the password. |
| 21:03.50 | lukejt | a plain, unsalted md5 is almost as bad as storing it plaintext |
| 21:04.10 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.117) |
| 21:04.13 | p3nguin | As far as I know, it only accepts a regular md5 hash. |
| 21:06.04 | SuperNull | so im a complete retard. lets pretend (or not) .. If im in the us and i wanted to do a test international call to 'tesco wine by the case' # what would i dial verbatim: http://www.tesco.com/help/contact/ |
| 21:06.38 | SuperNull | we attempted 0110845 677 5577 and 0845 677 5577 and 011845 677 5577 |
| 21:08.24 | p3nguin | Are they in Scotland? |
| 21:09.26 | p3nguin | I would dial 00 44 845 677 5577. |
| 21:09.39 | p3nguin | You seem to use 011, so use 011 instead of 00. |
| 21:10.13 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 21:10.14 | p3nguin | I actually just called it to confirm. "Thanks for calling Tesco Wine by the Case." |
| 21:10.14 | paulc | Yes.. 011 44 845 677 5577 (drop the 0 from the UK full form area code) |
| 21:10.24 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 21:10.31 | paulc | Ah Tesco.. fond memories of home.. |
| 21:11.03 | SuperNull | i only know of tesco from 'Ramseys kitchen nightmares' lol |
| 21:11.05 | SuperNull | go BBC |
| 21:11.50 | lukejt | I live in the UK, Scotland is in the UK. p3nguin is correct - 44 is the national code. Why are you calling 011? Dial out code or something? |
| 21:12.03 | SuperNull | yah 011 is international dialout code. |
| 21:12.11 | p3nguin | From the US, we typically dial 00 or 011 for an international call. |
| 21:12.11 | SuperNull | at least for us .. not sure if it is a US standard. |
| 21:12.37 | p3nguin | Think of it like an access code to get out of the country. |
| 21:13.01 | SuperNull | yeah. thats how i think of it.. |
| 21:13.22 | lukejt | hmm.. here we just dial the intl number.. i.e. 0035 for Ireland, 001 for USA |
| 21:13.23 | SuperNull | just get confused i honestly thought 011 was country code and uk was inside of it some how. damn me for not knowing international calling. |
| 21:13.25 | p3nguin | So you were pretty close. You just forgot to dial the area code after your 011. |
| 21:13.50 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 21:14.14 | p3nguin | lukejt: I use the 00 as well. For the UK, 00 44 followed by the phone number. |
| 21:14.25 | p3nguin | I actually support both 00 and 011 in my dial plans. |
| 21:16.36 | lukejt | I just support 00, all my providers require e164 anyway so I just strip the 00 |
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| 21:19.10 | SuperNull | p3nguin you said you got through right? |
| 21:19.14 | SuperNull | to tesco.. |
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| 21:19.59 | p3nguin | Correct. |
| 21:20.22 | SuperNull | k thank you much. |
| 21:21.10 | SuperNull | Where are ya located if ya dont mind me asking ? |
| 21:21.16 | p3nguin | IL USA |
| 21:21.32 | SuperNull | obama country ;) |
| 21:21.44 | p3nguin | Sort of. |
| 21:21.57 | p3nguin | He's from Chicago, we're 300 miles south of that. |
| 21:22.51 | SuperNull | isnt obamas ex-right hand man mayor of Chicago ? |
| 21:23.25 | p3nguin | He was a senator. I don't know about before that. |
| 21:23.44 | p3nguin | He went from the senate to the presidency. |
| 21:23.58 | SuperNull | Rahm Emanuel .. |
| 21:26.45 | dr0ck | donno about 'right hand man', he was chief of staff for minute |
| 21:27.11 | SuperNull | prolly right.. holder is right hand man. The man who knows 'the law' heh. |
| 21:27.45 | SuperNull | i can say this.. im def not for Lawyers being officials .. they will create laws for gain. |
| 21:28.31 | SuperNull | kind of a .. awkward day. |
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| 21:28.40 | shido6 | list |
| 21:28.56 | *** join/#asterisk voipnet-tech (~voipnette@66.63.72.130) |
| 21:28.58 | voipnet-tech | hello friends, quick question. I have a pbx where ExtensionState is 2 (BUSY) for all extensions, atleast this is what Dialparties.agi says. I can call the extension because they have call waiting on, however if the user is in a queue they do not get calls. I checked hints and they all says Idle. Not sure if that's the right place to look. How can I see and/or reset the ExtensionState ? |
| 21:29.31 | SuperNull | i had to blow up my direct boss.. he went to far with things.. stealing company equipment using company resources for gain and competition personally. and then throwing all of his crew 'under the bus'. He just was relieved of his duties to manager our technical supporrt department due to harassment of other employees.. hes all but shit canned. |
| 21:29.51 | SuperNull | oh yeah and aparently hes a heroin addict now. |
| 21:29.56 | SuperNull | details ;) |
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| 21:54.45 | saint_ | . |
| 21:54.49 | saint_ | ~mysql |
| 21:54.49 | infobot | SQL (Structured Query Language) database server. URL: http://www.mysql.com/ |
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| 22:04.38 | SuperNull | A network admin and i came up with a new wine.. |
| 22:04.41 | SuperNull | 'Port 5060' |
| 22:05.28 | WIMPy | How much of it did you consume before you found the name? |
| 22:05.44 | *** join/#asterisk philippel_mac (~p_lindhei@199.59.105.144) |
| 22:06.06 | philippel_mac | Qwell: ping you around? |
| 22:08.04 | SuperNull | yeahhhh uhm. |
| 22:08.06 | SuperNull | none. |
| 22:08.09 | SuperNull | i dont like booze. |
| 22:08.18 | SuperNull | kind of a smoker.. but lately i have been ill so.. even that displeases me. |
| 22:08.27 | SuperNull | vaporizer :-/ |
| 22:09.24 | philippel_mac | question though this may be more of a #asterisk-dev question: |
| 22:09.45 | philippel_mac | Once I have set CALLERID(all) my CDR(clid) is set properly |
| 22:10.06 | SuperNull | im a paranoid type we set CDR() directly. |
| 22:10.15 | philippel_mac | if I change either CALLERID(all) or CALLERID(name) then the CNAM part of CDR(clid) gets appropriate changed. |
| 22:10.41 | philippel_mac | however, if I change CALLERID(number) the CNUM does not change and remains what it was first set to |
| 22:11.20 | WIMPy | It sould be "num", not "number". |
| 22:11.33 | philippel_mac | now if I turn around and set the CALLERID(ANI-num) that changes the CDR(clid) number part appropriately at that point |
| 22:11.50 | philippel_mac | however that also changes CDR(src) |
| 22:12.02 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
| 22:12.22 | philippel_mac | um on the num vs. number both have the same affect and in fact on 10, let me check but I believe the help says num only even though it has always been number |
| 22:12.35 | Nugget | s/affect/effect/ |
| 22:12.43 | dijib | SeRi: are you around? |
| 22:12.57 | philippel_mac | yup core show function CALLERID shows only 'num' these days |
| 22:13.04 | WIMPy | If I had found the time to do a little more testiong, I would have opened an isse fro the ani part. It seems to leak into a forwarding ID. |
| 22:13.56 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-187.phlapa.fios.verizon.net) |
| 22:14.05 | philippel_mac | so question … is this really expected behavior, seems like a bit of a 'bug' but I don't want to jump to conclusions too quickly and I have not looked into the code yet |
| 22:14.49 | WIMPy | I fear that's all open to interpretation as there doesn't seem to be any documentation on what is supposed to be what. |
| 22:15.06 | philippel_mac | WIMPy: was that back to me? |
| 22:15.18 | WIMPy | yes |
| 22:15.31 | philippel_mac | WIMPy: well I guess there are a couple of fairly solid views |
| 22:15.57 | philippel_mac | first one would be, CDR(clid) should reflect the clid that was used at the time the call was placed, I don't think there is too much interpretation there |
| 22:16.18 | saint_ | is it possible to update a value in a mysql database from a dialplan ? |
| 22:16.20 | philippel_mac | secondly the cnam part changes correctly but the cnum part seems to be fixed after the first setting of it |
| 22:16.35 | saint_ | i tried to use writesql in func_odbc.conf without success. |
| 22:16.52 | WIMPy | I would think so too, but it's not written anywhere. |
| 22:16.56 | philippel_mac | so, yes, there is always some level of interpretation but there is also some level of expected consistency which is what I am checking into |
| 22:17.41 | philippel_mac | WIMPy: well the beauty of a project like this is to find where things seem a bit off and come to concuss, though sometimes there is a good reason for things … thus the initial discussion before drilling deeper into it :) |
| 22:18.20 | WIMPy | But I don't see all of the CDR field you mention. |
| 22:18.38 | philippel_mac | clid and src are the only two I mentioned |
| 22:19.08 | WIMPy | Yes. Sorry. Just had to read that again. |
| 22:19.10 | philippel_mac | core show function CDR |
| 22:19.56 | WIMPy | src - Source. Doesn't say much about what it could be :-( |
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| 22:21.13 | WIMPy | But yes, that looks like another inconsistancy. |
| 22:21.17 | philippel_mac | WIMPy: thanks for your engagement but I'm trying to get more at one of the devs or someone else who is intimately familiar with CDR/CID stuff to help assess the situation as well as potential history since I don't know if this has changed or not, I have not done thorough cross version testing at this point |
| 22:27.08 | WIMPy | Interesting. So you get CALLERID(ani-num) in to CDR(src) if you set it, but at the beginning it contains CALLERID(num)? |
| 22:27.42 | philippel_mac | yes an that also successfully resets the CALLERID(num) |
| 22:28.05 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 22:28.14 | philippel_mac | and of all the tests I did, seems to be the only way for me to reset it at the point where I am trying to do such as it appears once set it isn't letting me change it |
| 22:28.18 | WIMPy | Err. If you set ani, num gets changed? |
| 22:29.01 | philippel_mac | CDR(clid) changes, I don't know if it changes CALLERID(num), I didn't check that as it was not what I was interested in |
| 22:29.10 | philippel_mac | plus, what I was doing in my test was: |
| 22:29.18 | WIMPy | I'm pretty sure I have changed all of them. What Asterisk Version are you on? |
| 22:29.20 | WIMPy | Ok. |
| 22:29.34 | philippel_mac | CALLERID(ani-num)=CALLERD(num) to try to force it down the CDR's throat |
| 22:29.35 | WIMPy | But that surely seems rather strange. |
| 22:30.41 | philippel_mac | this testing was on 10.9.0 but I am pretty sure the CALLERID(all) not resetting the number part has existed for a long time |
| 22:31.38 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 22:33.37 | WIMPy | There have been changes in that area. So I wouldn't be sure it was like that in 1.8. |
| 22:33.42 | WIMPy | Or still is in 11. |
| 22:34.31 | philippel_mac | I'm saying it mostly from a long history and what I'm pretty sure I've watched for a long time, that it has been this way |
| 22:35.18 | philippel_mac | so any developers know off hand what this does: |
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| 22:35.28 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:35.31 | philippel_mac | num = S_COR(c->caller.ani.number.valid, c->caller.ani.number.str, S_COR(c->caller.id.number.valid, c->caller.id.number.str, NULL)); |
| 22:36.27 | WIMPy | Use ani if valid, otherwise num if valid, otherwise nothing. |
| 22:36.29 | philippel_mac | is that basically saying if their's already a valid ANI use it, otherwise use the cid number? |
| 22:36.38 | philippel_mac | k that is what I thought |
| 22:36.39 | _Corey_ | philippel_mac: If you haven |
| 22:36.42 | _Corey_ | gah |
| 22:36.57 | _Corey_ | if you haven't found #asterisk-dev yet, I'd look in there |
| 22:36.57 | philippel_mac | _Corey_: ? |
| 22:37.03 | WIMPy | It does seem to make sense that far. |
| 22:37.13 | philippel_mac | _Corey_: yeah I figured I should start migrating to there at this point |
| 22:37.36 | _Corey_ | If you have code-level questions, I'd go there |
| 22:37.56 | WIMPy | The trouble only starts when it cahnges mid call and only for the name or only for the number part. |
| 22:39.04 | philippel_mac | number can be changed no problem in my testing, but I am going to move this to #asterisk-dev at this point |
| 22:40.10 | WIMPy | So the name part doesn't have the same logic? Or is ani-name just unset? |
| 22:42.52 | *** join/#asterisk wasabi1 (~wasabi@ubuntu/member/wasabi) |
| 22:43.11 | wasabi1 | Hi. How does one get Asterisk to work with full SIP URIs, instead of extensions? |
| 22:43.47 | [TK]D-Fender | wasabi1: You don't the dialplan is "it" |
| 22:43.59 | WIMPy | Where? When? What does that mean? |
| 22:44.20 | p3nguin | A SIP URI contains an extension and a host. |
| 22:44.30 | wasabi1 | Yeah. I want to base my routing on the host. |
| 22:44.38 | p3nguin | sip.conf |
| 22:44.42 | wasabi1 | Or, match characters instead of exteensions. |
| 22:44.45 | wasabi1 | numbers, i mean |
| 22:44.52 | p3nguin | You will still use extensions. |
| 22:44.57 | p3nguin | That's what extensions are. |
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| 22:45.25 | wasabi1 | I see. |
| 22:45.50 | wasabi1 | Not really though. |
| 22:46.10 | jpsharp | An extension doesn't have to be numbers. |
| 22:46.12 | wasabi1 | I'd like all calls coming from a certain host, with the domain ending in @somedomain.com, to be bridged to another host... |
| 22:46.47 | p3nguin | If you want to "route" a call, you have two options. You can either create a peer entry for the calling host and set the context, or you can allow guest calls without having a peer entry and determine what to do with the call purely in the extensions. |
| 22:46.51 | jpsharp | exten => wakkawakkawakka,1,Playback(fozzie) is just as valid as exten => 8675309,1,Playback(jenni) |
| 22:47.07 | wasabi1 | Hmm. |
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| 22:47.36 | p3nguin | In both cases, you will use extensions or you will not process the call at all. |
| 22:47.47 | wasabi1 | Okay. I understand that. I have extensiosn for my peers. |
| 22:47.54 | wasabi1 | What I mean is just that I don't want to use numbers for anything. |
| 22:48.05 | wasabi1 | jpsharp: Can you do a wildcard on the domain portion like that? |
| 22:48.40 | p3nguin | You have extensions for your peers? That doesn't really make any sense to me. |
| 22:48.44 | WIMPy | The domain part is not usually used. |
| 22:49.03 | wasabi1 | It's part of what I need to look at to determine the destination, though. |
| 22:49.11 | p3nguin | You don't have to use numbers at all. You can have extension steven just as easily as you could have extension 54321. |
| 22:49.42 | wasabi1 | Okay, then let's forget extensions. Those are just for peers. Whether users or other PBXs. I actually only have three of those: three PBXes. |
| 22:50.05 | p3nguin | then let's forget extensions. Those are just for peers. <------ This does not make any sense at all. |
| 22:50.07 | wasabi1 | But none of them can talk properly to each other. One only supports udp. The other only supports tcp. You see. Asterisk fits nicely in the middle. |
| 22:50.35 | WIMPy | And where do the domains come in to play? |
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| 22:50.38 | p3nguin | EVERY call will use an extension. |
| 22:50.42 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 22:50.44 | p3nguin | This is how asterisk works. |
| 22:50.54 | p3nguin | No extensions, no calls get processed. |
| 22:51.13 | WIMPy | So far I see 3 peers. |
| 22:51.19 | kaldemar | wasabi1: see domain support in the sample sip.conf |
| 22:53.46 | wasabi1 | I get it. What I'm saying is that "extensions" in my language wasn't the same as asterisk. They seem to just be for peering hosts. |
| 22:54.00 | wasabi1 | Whether those hosts are phones, or other PBXes. |
| 22:54.04 | p3nguin | In asterisk, extensions are the calling rules. |
| 22:54.14 | wasabi1 | Or are those contexts? |
| 22:54.15 | p3nguin | And to process calls, will be required. |
| 22:54.24 | p3nguin | Contexts are containers for extensions. |
| 22:54.25 | wasabi1 | What do you call the items placed in sip.conf? :) |
| 22:54.30 | p3nguin | peers |
| 22:54.46 | wasabi1 | Ahh. exten => is considered an "extension", because it's called exten. |
| 22:54.47 | p3nguin | peer definitions, peer entries, peers |
| 22:54.58 | wasabi1 | I was just thinking it was a rule of some sort. |
| 22:55.08 | WIMPy | More the other way round. |
| 22:55.11 | p3nguin | extensions.conf contains... extensions. |
| 22:55.32 | WIMPy | Yes. extensions are the call processing rules. |
| 22:56.04 | wasabi1 | Okay. So, here's the thing. The PBXes I'm using don't very often (unless one is making an outbound PSTN call) use numbers. They use SIP URIs. |
| 22:56.10 | p3nguin | Extensions don't have to send calls to devices. Extensions can play back sound files, perform system commands, any lots of other things. |
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| 22:56.28 | wasabi1 | So I need to bridge between them, OR to the PSTN, by making some rules that check the destination, and then Dial through a peer. |
| 22:56.52 | WIMPy | Yes. That's the extensions. |
| 22:57.08 | wasabi1 | Alrighty. That first part of exten =>, doesn't seem to have the domain portion, as ya'll mentioned. I need it. |
| 22:57.18 | mogra | is this a better place, Qwell ? |
| 22:59.05 | wasabi1 | PBX #1 sends calls with user@domain1.com on the end, PBX#2 sends calls with user@domain2.com at the end. If the user is dialing the PSTN, then yeah, I don't care about the domain. |
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| 23:08.05 | wasabi1 | Might have found my answer. |
| 23:09.10 | saint_ | Can someone tell me why is this not updating my mysql db ? -> writesql=UPDATE FIREHOUSES SET timecalled=('${ARG1}') WHERE number=('${VAL1}') |
| 23:09.50 | saint_ | just after that, when I read the table that was supposed to be updated, it still has the previous value |
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| 23:13.35 | kaldemar | saint_: how are you using the function? |
| 23:13.54 | saint_ | same => n,Set(INCOMINGCALLER_SETTIME(${CALLERID(num)}=${STRFTIME(,,%X)})) |
| 23:14.45 | p3nguin | There's a syntax error in there. |
| 23:14.50 | saint_ | ha.. |
| 23:14.54 | saint_ | what is it ? |
| 23:15.13 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 23:15.32 | p3nguin | Are you trying to set INCOMINGCALLER_SETTIME(${CALLERID(num)}) to a time value? |
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| 23:15.57 | saint_ | I am trying to update one of the fields of the tables to a time value |
| 23:16.18 | saint_ | callerid(num) being the variable i m using to find out the correct table in the database |
| 23:16.20 | p3nguin | Is the field the caller id number field? |
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| 23:16.35 | saint_ | no, the field is timecalled |
| 23:16.44 | p3nguin | What value do you want to set to it? |
| 23:16.55 | p3nguin | An example of a value will be fine. |
| 23:17.02 | saint_ | to whatever STRFTIME(,,%X) gives me |
| 23:17.06 | p3nguin | exampel |
| 23:17.16 | saint_ | like 18:07:36 |
| 23:17.17 | p3nguin | s/el/le/ |
| 23:17.34 | saint_ | which is what STRFTIME(xx%X) returns |
| 23:17.36 | p3nguin | What is the purpose of the CALLERID(num) in there? |
| 23:17.49 | saint_ | CALLERID(num) is the caller ID of the caller |
| 23:17.55 | p3nguin | I know what it is. |
| 23:17.56 | saint_ | his number is in a database |
| 23:17.57 | kaldemar | val and arg should be the other way |
| 23:18.01 | saint_ | to his number is attached a time field |
| 23:18.23 | saint_ | so i am trying to update the time (timecalled) from CALLERID(num) |
| 23:18.46 | p3nguin | So if the INCOMINGCALLER_SETTIME() function is written correctly, there will be an arg for the function of the caller id number, in which you will set a time value. |
| 23:19.03 | kaldemar | and s/=/)=/ |
| 23:19.16 | p3nguin | You have same => n,Set(INCOMINGCALLER_SETTIME(${CALLERID(num)}=${STRFTIME(,,%X)})) |
| 23:19.19 | p3nguin | You need same => n,Set(INCOMINGCALLER_SETTIME(${CALLERID(num)})=${STRFTIME(,,%X)}) |
| 23:19.23 | saint_ | kaldemar: if I look at the verbose log, I have INCOMINGCALLER_SETTIME(xxxxxx6465=18:07:36) which is what I want |
| 23:19.57 | p3nguin | I don't understand your function, so that's my reason for asking questions about it. |
| 23:19.59 | kaldemar | saint_: that is a syntax screw-up. |
| 23:20.32 | kaldemar | func(arg)=value |
| 23:20.52 | p3nguin | If CALLERID(num) is a valid arg to the function, the syntax error was the only problem. |
| 23:21.03 | p3nguin | And we've both told you how to fix it. |
| 23:21.59 | kaldemar | arg and val are still wrong |
| 23:22.01 | saint_ | p3nguin: i changed it to what you suggested, and it still does not keep the value in the database |
| 23:22.15 | saint_ | kaldemar: even though the log is correct ? |
| 23:22.32 | p3nguin | example: Set(FUNCTION(arg)=value) |
| 23:22.34 | kaldemar | saint_: yed |
| 23:22.40 | saint_ | kaldemar: I'll switch them for the heck of it.. but I can see xxxxxxx=18:00:00 , which is what I am looking for *i think* |
| 23:22.48 | p3nguin | You had: Set(FUNCTION(arg=value)) |
| 23:23.16 | saint_ | p3nguin: i did this change, but it is still not saving the value in the DB |
| 23:23.21 | kaldemar | saint_: yes. you need to modify the function in func_odbc.conf |
| 23:23.48 | p3nguin | Ah, so the function is broken in addition to the syntax error in the app call. |
| 23:23.56 | kaldemar | saint_: change the places of arg and val |
| 23:24.08 | saint_ | kaldemar: thanks a ton .. |
| 23:24.29 | saint_ | that worked.. |
| 23:25.01 | kaldemar | saint_: do you understand the difference? |
| 23:25.21 | wasabi1 | Can an asterisk peer be set to authenticate by a SRV record? |
| 23:26.07 | saint_ | I think: the value of the function is $VAL1 and the =XXX is the $VAR1 |
| 23:26.29 | saint_ | value is VAL1 and the variable I want it to is ARG1 , i m sorry |
| 23:26.48 | saint_ | it makes sens now.. |
| 23:26.54 | kaldemar | func(arg)=val |
| 23:27.38 | saint_ | yes.. learning the hard way since it took me hours to try to troubleshoot that, but I was totally away from reviewing this. i was focusing on what i had in extensions.conf only and not in the odbc |
| 23:28.04 | saint_ | i have to go to training. will be back in about 3 / 4 hours ... thanks kaldemar / p3nguin |
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