| 00:21.11 | navaismo | weird stuff I just need to avoid all dtmf on a BRIDGED call only |
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| 07:40.14 | tparcina | When incoming call, over GSM gateway, leaves a voice mail, mail is displaying wrong data. |
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| 07:41.29 | tparcina | In e-mail, there will be "Just wanted to let you know you were just left a 0:05 long message (number 1) in mailbox 777 from gsm_gateway_user_name, on Wednesday..." |
| 07:43.08 | tparcina | What I need is that "gsm_gateway_user_name" is substituded with calling phone number (asterisk knows that number). |
| 07:43.59 | tparcina | Does anyone know how can I accomplish this functionality? |
| 07:45.49 | tparcina | As far as I see it, I would have to modify my dial plan, and substitute "gsm_gateway_user_name" with "phone_number". |
| 07:49.07 | kaldemar | you'll find emailbody defined in voicemail.conf. ${VM_CALLERID} should contain both name and number. |
| 07:49.30 | kaldemar | for number only, use VM_CIDNUM. |
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| 07:57.03 | tparcina | kaldemar: Thank you, I'll check it right now. |
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| 08:27.56 | l0c0 | Hi every body, maybe some one can help ? |
| 08:28.22 | l0c0 | how to force kill a channel to hangup ? |
| 08:28.36 | l0c0 | I've try with soft hangup and the ID |
| 08:29.23 | l0c0 | but nothing append ... I've a message from astersik saying |
| 08:29.26 | l0c0 | Requested Hangup on channel 'Local/0027887548@NoRestrict-8cb7,2' |
| 08:29.41 | l0c0 | but the channel still opened |
| 08:29.59 | l0c0 | I know that if I restart asterik I kill the channel |
| 08:30.18 | l0c0 | but if sombody 've a better id :) |
| 08:38.48 | tparcina | kaldemar: I'm using FreePBX and in voicemail.conf I have "#include vm_email.inc". |
| 08:39.27 | tparcina | In "vm_email.inc" I have different text than I receive in e-mail. |
| 08:40.14 | tparcina | kaldemar: Now I'm searching from what other location could Asterisk get this text that it's sending in e-mail. |
| 08:40.21 | wdoekes | tparcina: app_voicemail doesn't support "overwriting" subject/body |
| 08:40.26 | wdoekes | the first entry is used |
| 08:41.31 | tparcina | wdoekes: There is no overwriting. The lines in voicemail.conf are commented with ; |
| 08:42.18 | wdoekes | in that case you should doublecheck that your #include works as expected |
| 08:42.29 | tparcina | l0c0: Try "core show channels verbose" and then "channel request hangup SIP/..." |
| 08:43.27 | wdoekes | l0c0: if you can't get it to "request hangup" then there's nothing more you can do. note that you're looking at some kind of deadlock/bug there |
| 08:43.36 | wdoekes | what version are you using? |
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| 08:44.11 | l0c0 | tparcina: tks, it is an asterisk 1.2 so the cmd is just show channels.. and I've try the soft hangup channel local/... |
| 08:44.20 | l0c0 | but no way :( |
| 08:44.32 | wdoekes | l0c0: it's probably a bug |
| 08:45.01 | tparcina | wdoekes: It seams that it doesn't work. Is this the correct include statment: "#include vm_email.inc" ? |
| 08:45.26 | wdoekes | I don't use inludes, check the examples in configs/* |
| 08:45.37 | l0c0 | wdoekes: yes indeed it is ... but I've 9 active calls and I can't restart astersik |
| 08:46.04 | wdoekes | l0c0: restart when convenient |
| 08:46.05 | l0c0 | wdoekes: the call is initiated by an API |
| 08:46.15 | wdoekes | although, that won't work |
| 08:46.20 | wdoekes | because the channel is always open :D |
| 08:46.43 | l0c0 | wdoekes: I'll do but I just wanted if some one had a better id ;) |
| 08:47.52 | wdoekes | l0c0: http://paste.ubuntu.com/6013276/ |
| 08:52.35 | tparcina | wdoekes: I have overlooked the ; in included file (vm_email.inc). |
| 08:53.04 | tparcina | wdoekes: So there are no conf files with emailbody line in them. |
| 08:54.24 | tparcina | So, I guess that e-mail that I get is hard coded in Asterisk, and it can be overwriten in voicemail.conf of included files with emailbody line. |
| 08:56.39 | kaldemar | tparcina: the default message in app_voicemail.c indeed uses the caller id name part if such exists. |
| 08:57.01 | kaldemar | if the name is empty, number is tried. if that too is empty "an unknown caller" is used. |
| 08:58.10 | l0c0 | wdoekes: tks ! |
| 08:59.24 | tparcina | kaldemar: Geee, where did you find this? :) |
| 09:00.03 | kaldemar | tparcina: eh, in the source code, in app_voicemail.c. |
| 09:00.49 | tparcina | kaldemar: You are fast. :) |
| 09:00.55 | tparcina | And thank you for your help. |
| 09:01.18 | tparcina | I have to go now. I'll be back later. Bye. |
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| 09:07.12 | kaldemar | tparcina: that sort of thing won't even take a minute if you know where to look. |
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| 10:20.09 | hrolf | Hi #asterisk, we use MixMonitor on a channel, now sometimes we use ChanSpy for whispering (supervisory/consult) and want this voice to be also recorded in the file being recorded? How do I do this? |
| 10:20.58 | hrolf | When I do ChanSpy for whisper, the whisperer's voice doesn't get recorded in the file initiated by MixMonitor |
| 10:26.15 | Greenlight | You're definelty doing the MixMonitor AND ChanSpy to the *same* channel ? |
| 10:26.23 | hrolf | Greenlight: Yes. |
| 10:26.34 | hrolf | Greenlight: First, I start MixMonitor. |
| 10:26.39 | Greenlight | afaik they both use audiohooks |
| 10:26.47 | hrolf | Greenlight: Then depending, if the supervisor wants, he does ChanSpy. |
| 10:27.20 | hrolf | Greenlight: So I believe that it will happen only if I do ChanSpy before MixMonitor, am I right? |
| 10:27.32 | hrolf | Greenlight: But I'm doing it in reverse order? |
| 10:27.42 | hrolf | Greenlight: Is it due to it? |
| 10:27.59 | Greenlight | Yea, it may be to do with order that the audiohooks are taken in; but I don't know enough about how it works to answer with certainty |
| 10:28.06 | Greenlight | However |
| 10:28.11 | Greenlight | You can probably "hack" it |
| 10:28.17 | hrolf | Greenlight: How? |
| 10:28.19 | Greenlight | eg, use an intermediate Local channel |
| 10:28.32 | hrolf | Greenlight: How to use an intermediate local channel? |
| 10:28.57 | Greenlight | Ok, so instead of connecting SIP/<Customer> <--> SIP/<Agent> |
| 10:29.14 | Greenlight | And then doing ChanSpy on SIP/<Agent> and MixMonitor on SIP/<Agent> |
| 10:29.33 | Greenlight | You could try and get a Local channel inbetween |
| 10:29.58 | Greenlight | So SIP/<Customer> <---> Local/1;Local/;2 <---> SIP/<Agent> |
| 10:30.09 | Greenlight | And you could MixMonitor on the agent and chanspy on the Local |
| 10:30.58 | hrolf | Greenlight: How do I get a local channel inbetween? Right now I just put the call in a queue in which the agent resides. |
| 10:31.20 | Greenlight | And you're 100% sure you're doing the mixMon on the AGENT? |
| 10:31.26 | Greenlight | Can I see a call flow in the CLI |
| 10:34.12 | Greenlight | And to answer the question about getting a Local channel inbetween, it depends on your dialplan, but basically say your agent was SIP/200, then instead of putting SIP/200 in as a queue member, you'd put Local/200@default/n as the queue member (assuming 200 @ default would get to that agent). |
| 10:34.18 | hrolf | Greenlight: I'm doing ChanSpy(SIP/<exten>) |
| 10:34.36 | Greenlight | hrolf: Yes, can I see the call flow? |
| 10:34.40 | Greenlight | ~pb |
| 10:34.40 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 11:04.23 | v0lZy | google should have info on that |
| 11:04.34 | v0lZy | wrong win, disregard |
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| 11:28.12 | Greenlight | Whats the default timeout for a sip qualify ? |
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| 11:33.44 | kaldemar | Greenlight: 2 seconds |
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| 12:00.28 | taube | hi, im trying to get a caller Id lookup working with mysql. http://pastebin.com/i4kBWW7d This is the code at extensions.conf and the output. The output of the call is taub2, but in my mysql table is no entry with taub2... i dont know whats wrong with it, maybe there is sth wrong with the code? |
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| 12:07.24 | wasanzy | hi |
| 12:07.54 | wasanzy | anyone knows how to patch chan_dongle for asterisk 11.5.0 |
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| 12:12.54 | kaldemar | wasanzy: with what? |
| 12:14.37 | wasanzy | I got this error: make: *** [app.o] Error 1 when compiling chan_dongle so when I searched, they said it is not yet supporting asterisk 11+ but there is a patch, I found the patch here: https://github.com/pld-linux/asterisk-chan_dongle/blob/master/chan_dongle-pin.patch |
| 12:14.45 | wasanzy | but I don't know what to patch |
| 12:16.42 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 12:17.21 | ipalmer | Hi all, I want to dial a Local channel and pass in a couple of parameters with the dial, is there a way of doing this? |
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| 12:18.28 | kaldemar | wasanzy: each diff command output there creates patch that can be applied to the chan_dongle source code with the "patch" command line tool. man patch. |
| 12:21.33 | kaldemar | wasanzy: save https://raw.github.com/pld-linux/asterisk-chan_dongle/master/chan_dongle-pin.patch to your chan_dongle source dir and then use "patch -p4 < chan_dongle-ping.patch" to apply it. |
| 12:22.06 | wasanzy | kaldemar: ok thanks |
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| 12:23.01 | kaldemar | i managed to put a typo in the command, ping where it should have pin. that needs to be the file name that "patch" uses as its input. |
| 12:23.18 | kaldemar | wasanzy: do you understand what that does? |
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| 12:24.19 | wasanzy | not clearly though |
| 12:25.15 | kaldemar | it modifies the source code, using the .patch file as input. the .patch file tells which files and lines to change and with what content. |
| 12:25.50 | wasanzy | oh ok thank you |
| 12:25.58 | kaldemar | after that (assuming no chunk of the patch failed), you need to recompile and reinstall the application. |
| 12:26.22 | kaldemar | if chunks of the patch fail, your source is not compatible with the .patch file. |
| 12:27.11 | wasanzy | ok |
| 12:29.10 | WIMPy | If you google for what you want, you should find it easily. |
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| 12:31.19 | [TK]D-Fender | ipalmer: you have standard variable inheritance rules, that's all |
| 12:33.01 | ipalmer | D-Fender:thought as much, been trying to think of a workaround such as using macros or something, but can find nothing that will work |
| 12:34.13 | wasanzy | WIMPy: I did google that was why I got to know about the patch etc |
| 12:34.20 | [TK]D-Fender | ipalmer: DialipalStore some vars in a DB or something and make what you dial the DB key so it can retrieve them |
| 12:34.24 | [TK]D-Fender | ipalmer: another option. |
| 12:34.35 | [TK]D-Fender | ipalmer: increasingly fugly |
| 12:35.26 | WIMPy | wasanzy: When I searched for "chan dongle asterisk 11" I immediately found a version that works. |
| 12:36.13 | ipalmer | D-Fender: that could work, but you're right it is ugly especially as I have already used a database to dial the Local channels |
| 12:36.17 | wasanzy | can you point me there? is not like I didn't google, I searched base on the error I got |
| 12:37.19 | [TK]D-Fender | heading to work, BBIAB |
| 12:41.47 | WIMPy | Ok, I have to admit it was on page 4 now. |
| 12:41.52 | WIMPy | http://sisyphus.ru/en/srpm/Sisyphus/asterisk11-chan_dongle/sources |
| 12:43.01 | wasanzy | WIMP: thank you |
| 12:43.40 | tzanger | interesting, I have one of those dongles, never thought to use it for voice |
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| 12:45.01 | tzanger | lol, try to download the tarfile and get "source is too large" |
| 12:46.00 | wasanzy | WIMPy: yes source is too large |
| 12:46.27 | cusco | chan dongle moved repo ? |
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| 12:47.11 | WIMPy | What? |
| 12:47.30 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 12:47.56 | tzanger | I found https://github.com/jstasiak/asterisk-chan-dongle which claims to track the svn repo on google |
| 12:48.14 | WIMPy | cusco: No. But tiy find (or found) a working copy there. |
| 12:48.40 | WIMPy | s/tiy/you/ |
| 12:49.04 | tzanger | if you search for asterisk on github there are some neat projects there |
| 12:49.20 | tzanger | https://github.com/zaf/asterisk-speech-recog |
| 12:49.52 | WIMPy | So you can't download that thing? |
| 12:50.30 | tzanger | WIMPy: the sisyphus.ru link does not let me download the tar, no. it says "source is too large" |
| 12:50.38 | WIMPy | Doesn't seem to be the one I found initially, anyway. |
| 12:52.09 | tzanger | I might hav come into the conversation late, I only saw that url |
| 12:53.40 | WIMPy | The only link I find now is broken. |
| 12:55.11 | wasanzy | tzanger: you can check this one: http://packages.altlinux.org/en/Sisyphus/srpms/asterisk11-chan_dongle/sources |
| 12:55.59 | tzanger | wasanzy: I just cloned the git repo or go directly to the google svn source. don't particularly want to stray too far from the origins unless there's a good reason |
| 12:56.07 | WIMPy | http://voice.yeti.dk/chan-dongle-asterisk11.zip Here is the one I downloaded. |
| 12:56.45 | wasanzy | ok |
| 12:57.09 | WIMPy | Ah, that was a working one. |
| 12:59.03 | wasanzy | WIMPy: downloaded yours |
| 12:59.06 | wasanzy | thank you |
| 13:00.09 | WIMPy | You should use tzangers link. It might be more recent. |
| 13:00.18 | wasanzy | wwwowwwww, it worked |
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| 13:05.13 | wasanzy | After installing, I can't find the chan_dongle on the menuselect |
| 13:06.11 | WIMPy | menuselect only shows you the stuff that comes with Asterisk. |
| 13:06.23 | WIMPy | Other add ons don't show up there. |
| 13:07.14 | wasanzy | oh ok |
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| 13:38.39 | wasanzy | under /var/log/asterisk which of the log files shows the real asterisk transactions? |
| 13:38.56 | AllanButton | depends on what you have in your logger.conf |
| 13:39.10 | AllanButton | and also what you mean by real asterisk transactions |
| 13:39.12 | kaldemar | there is no such thing as "asterisk transaction". what are you really after? |
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| 14:01.48 | [TK]D-Fender | What's a "fake" Asterisk Transaction (tm)? |
| 14:07.49 | cusco | tzanger: I built my own googletts script lol |
| 14:08.00 | cusco | but nice to know there is stuff there |
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| 14:25.42 | wasanzy | how do I know which of my ports is data and the other audio? /dev/ttyUSB1 and /dev/ttyUSB0 |
| 14:26.00 | wasanzy | asterisk is not able to connect to the ports |
| 14:26.08 | wasanzy | [dongle0] Trying to connect on /dev/ttyUSB0... |
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| 14:43.30 | zoid_ | Is there any way to qualify a sip provider where no authentication ocurrs? |
| 14:44.43 | WIMPy | Where is the link between thos two properties? |
| 14:45.22 | zoid_ | sorry, I'll ellaborate |
| 14:45.27 | zoid_ | WIMPy: I see here http://www.voip-info.org/wiki/view/Asterisk+sip+type that peers do authenticate |
| 14:45.34 | zoid_ | and I know asterisk monitors peers |
| 14:46.02 | zoid_ | I would like to do the same with a "peer" that doesn't require auth |
| 14:46.42 | zoid_ | the idea is to monitor the remote peer |
| 14:47.39 | WIMPy | And what keeps you from doing so? |
| 14:48.07 | igcewieling | zoid_: qualify uses OPTIONS packets. most devices don't auth those. |
| 14:48.34 | zoid_ | I don't know how should I declare the peer in my config |
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| 14:48.49 | igcewieling | zoid_: declare it as your normally would, then add qualify=yes |
| 14:49.02 | zoid_ | courrently, what I'm doing is writing .call files to somenumber@someip and hoping the ip is up |
| 14:49.24 | zoid_ | I would like to check if it is really up before writing a call file that will fail |
| 14:49.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:49.51 | zoid_ | igcewieling: should I write it like a normal peer? |
| 14:49.58 | igcewieling | zoid_: YES! |
| 14:50.11 | zoid_ | thanks, I'll try that! |
| 14:50.14 | igcewieling | zoid_: now you have changed your question |
| 14:50.30 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.214) |
| 14:51.08 | zoid_ | yes, I was assuming that peers authentication was mandatory |
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| 14:52.45 | zoid_ | igcewieling: declaring the peer with just type, host and qualify worked perfectly :) |
| 14:53.08 | zoid_ | thank you very much, to you too WIMPy |
| 14:55.25 | igcewieling | zoid_: depends on what you call "auth". With only host= you are using host auth |
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| 14:55.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:55.59 | zoid_ | you're right, what I meant by auth was id/password |
| 14:57.47 | igcewieling | then you should have said that. |
| 14:59.11 | zoid_ | next time I will |
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| 15:20.37 | wasanzy | am trying to connect asterisk to my dongle but is not connecting |
| 15:20.53 | wasanzy | any one has success doing this on centos |
| 15:21.23 | AllanButton | sounds painful. |
| 15:22.09 | wasanzy | AllanButton: yea can you help? |
| 15:22.21 | igcewieling | you're holding it wrong. |
| 15:22.44 | wasanzy | <igcewieling: me? |
| 15:22.55 | igcewieling | wasanzy: so few people use chan_dongle that you might have better luck asking on the Asterisk mailing lists or searching the mailing list archives. |
| 15:23.21 | igcewieling | ~mailinglist |
| 15:23.22 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 15:23.39 | WIMPy | Is the dongle itself working? |
| 15:23.54 | wasanzy | yes WIMPy |
| 15:24.18 | WIMPy | Do you have the correct devices? |
| 15:24.38 | wasanzy | is a vodafone modem |
| 15:24.46 | WIMPy | How many devices do you get? |
| 15:24.53 | WIMPy | Have yo tried other combinations? |
| 15:25.11 | wasanzy | lsusb gives me this: http://paste.scsys.co.uk/266199 |
| 15:25.55 | WIMPy | ls /dev/ttyUSB* |
| 15:26.39 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
| 15:27.27 | wasanzy | /dev/ttyUSB0 /dev/ttyUSB1 |
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| 15:28.06 | WIMPy | Ok, only two makes it easy. |
| 15:28.30 | WIMPy | And what's happening? |
| 15:29.50 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 15:29.55 | wasanzy | asterisk can't seem to be able to connect to it |
| 15:30.23 | wasanzy | I gets plenty of this: -- [dongle0] Trying to connect on /dev/ttyUSB0... |
| 15:31.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:31.50 | WIMPy | Nothing else? |
| 15:32.51 | wasanzy | yes, that is what I see when I reload the chan_dongle |
| 15:33.41 | WIMPy | That doesn't say much. Do you have verbose and debug cranked up? |
| 15:34.20 | *** join/#asterisk Ninja0n3 (18250fb6@gateway/web/freenode/ip.24.37.15.182) |
| 15:34.49 | Ninja0n3 | hello, is it possible to transfer a call using sip and release the sip line once the call is transferred? |
| 15:34.59 | wasanzy | verbose 5 is what I set |
| 15:35.10 | wasanzy | let me set it to 10 and debug to 3 |
| 15:35.45 | WIMPy | Ninja0n3: What is a sip line? |
| 15:35.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:36.09 | Ninja0n3 | what you use for voip |
| 15:36.12 | WIMPy | If you transfer a call, your're out. |
| 15:36.17 | Greenlight | Ninja0n3: I think you might be able to do that with a Transfer |
| 15:36.48 | Greenlight | But if you're trying to get around a situation where you've a limited number of channels from your provider, then this won't help |
| 15:37.18 | Ninja0n3 | my question is whether the line will be freed and the number can be called again while the original party is still in line with the transferee or whther once the sip line has been occupied it remains used through the duration of the call |
| 15:37.42 | Ninja0n3 | so it would all come down to the number of channels on the sip line then |
| 15:37.53 | WIMPy | Depends on te limits of your account. |
| 15:38.00 | Greenlight | Indeed |
| 15:38.03 | WIMPy | Stop saying sip line. |
| 15:38.11 | Ninja0n3 | what should I be saying? |
| 15:38.18 | wasanzy | WIMPy: this is my dongle.conf: http://paste.scsys.co.uk/266202 |
| 15:38.25 | WIMPy | You can use sip or lines or both, but "sip line" doesn;t make sanse. |
| 15:38.40 | Ninja0n3 | fair enough, sip then |
| 15:40.25 | WIMPy | wasanzy: You can try to swap the audio and data devices if you don't get anyting. But with debug enabled you should see what it's trying to do and where it's failing. |
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| 15:41.07 | wasanzy | WIMPy: I did swap several times with no luck |
| 15:41.21 | wasanzy | debug is on but the same error kept coming |
| 15:42.20 | WIMPy | Do you have debug enabled in your logger.conf? |
| 15:43.07 | wasanzy | I think there is a permission problem with the logger.conf, let me fix that |
| 15:43.09 | Ninja0n3 | are there any good tutorials on setting up asterisk? |
| 15:43.29 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:43.34 | WIMPy | ~book |
| 15:43.34 | 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 |
| 15:43.50 | Ninja0n3 | thank you |
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| 15:46.06 | SuperNull | very good book too. |
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| 15:48.36 | wasanzy | WIMPy: I see this log now: WARNING[14863]: chan_dongle.c:218 opentty: unable to open /dev/ttyUSB1: Permission denied |
| 15:49.03 | wasanzy | should asterisk user own that file? |
| 15:49.27 | WIMPy | It needs to have both read and write permissions. |
| 15:49.44 | WIMPy | To both devices, off course. |
| 15:50.32 | wasanzy | just did chmod o+rw on that file |
| 15:52.00 | wasanzy | hmm am now getting : WARNING[14863]: chan_dongle.c:218 opentty: unable to open /dev/ttyUSB0: Device or resource busy |
| 15:52.21 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 15:52.37 | WIMPy | That would mean something else is using it. |
| 15:52.53 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:53.15 | wasanzy | nothing is using it to my knowledge |
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| 15:57.31 | skrusty | does anyone know if there's ever the intention to add the ability to stream audio files over FastAGI? |
| 15:57.35 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
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| 16:25.17 | Greenlight | I've a 4 port Digium PRI card. Looking at the card from the back of the server, would channels 1-31 be the port at the left, or the right? |
| 16:25.36 | WIMPy | The top. |
| 16:25.59 | WIMPy | Didn't the have small numbers in the bracket? |
| 16:26.17 | Greenlight | Not sure -- it's a few hundred miles away and am dealing with IT onsite |
| 16:26.22 | WIMPy | (that you maybe can't see when built in) |
| 16:26.48 | Greenlight | So, the side that connects to the motherboard is the port 4, and furtherst away is port 1 ? |
| 16:27.07 | *** join/#asterisk nam3l3zz (~quassel@78-60-118-31.static.zebra.lt) |
| 16:27.14 | WIMPy | is pretty confident. |
| 16:27.22 | Greenlight | Thank you ;) |
| 16:27.28 | WIMPy | Doesn't the manual tell? |
| 16:27.45 | Greenlight | Good point .. |
| 16:27.48 | wasanzy | am getting this error now: Function CALLERID not registered |
| 16:27.56 | Greenlight | goes to get the manaul on Google |
| 16:28.05 | Greenlight | wasanzy: Try CALLERID(num) |
| 16:28.49 | wasanzy | Greenlight: in which config file? |
| 16:28.59 | Greenlight | .... |
| 16:29.07 | Greenlight | The one where you're using CALLERID and it's giving an error.. |
| 16:29.24 | WIMPy | But I think it really doesn;t exist. |
| 16:29.44 | Greenlight | Oh |
| 16:29.46 | Greenlight | That's odd |
| 16:30.13 | Greenlight | core show function callerid |
| 16:30.26 | Greenlight | ^^ What does that say |
| 16:30.30 | Greenlight | (At the CLI) |
| 16:30.36 | WIMPy | function names are case sensitive. |
| 16:30.46 | WIMPy | That also applies to core show... |
| 16:31.02 | WIMPy | So it has to be 'core show function CALLERID' |
| 16:31.19 | Greenlight | Oh, good catch! |
| 16:31.30 | Greenlight | has got lazy with tabbed auto-complete |
| 16:32.21 | Greenlight | Yup you were right about the port numbering; just checked in the manual :) |
| 16:32.59 | wasanzy | ERROR[16699]: pbx.c:4135 ast_func_write: Function CALLERID not registered |
| 16:32.59 | wasanzy | is the full error |
| 16:32.59 | wasanzy | No function by that name registered. |
| 16:33.50 | WIMPy | You must have disabled it in menuselect or in modules.conf. |
| 16:34.33 | Greenlight | Yup sounds like something very odds happened with your install. |
| 16:34.37 | WIMPy | nd diesabled in modules conf can also mean neither autoload nor explicitely loading th module. |
| 16:34.39 | *** join/#asterisk lukerobi (~lukerobi@rrcs-97-79-163-146.sw.biz.rr.com) |
| 16:34.55 | Greenlight | You might also want to check for errors when Asterisk is starting (start directly with -cvvvvvvv) |
| 16:35.11 | wasanzy | ok |
| 16:36.33 | wasanzy | autoload=yes |
| 16:36.40 | Kobaz | errors are bad |
| 16:36.41 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:37.09 | lukerobi | Quick question... Does anyone have any idea why occasionally when making ONLY outgoing calls the receiving party is unable to hear us? |
| 16:37.53 | *** join/#asterisk magicrhesus (~magicrhes@2001:41d0:1:9c01:0:ff:fee7:406a) |
| 16:38.48 | Greenlight | Usually such issues are network or NAT related. |
| 16:38.49 | *** join/#asterisk HolTech (~IceChat77@pool-108-40-39-10.bltmmd.fios.verizon.net) |
| 16:38.59 | Greenlight | Without more information we're just speculating. |
| 16:39.25 | wasanzy | I have func_callerid enabled in menuselect |
| 16:39.45 | Greenlight | wasanzy: Then check modules.conf, and also for startup errors. |
| 16:40.07 | Kobaz | if it's sip... media ports might be conflicting with something in your firewall |
| 16:40.17 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:40.32 | Kobaz | lukerobi: it may be a quick question, but not a quick answer |
| 16:41.30 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-xcmtduulpndbakrv) |
| 16:41.30 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:41.32 | Greenlight | And that's us presuming that this is SIP. Is it? |
| 16:42.02 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:42.13 | lukerobi | It seems to only happen with certain receiving phones..... I believe it is an outgoing SIP connection |
| 16:42.18 | WIMPy | Well, if it fails, it's likely to be sip, isn't it? |
| 16:42.32 | Greenlight | ^^ |
| 16:42.55 | Greenlight | is currently tracking down an ISDN error. |
| 16:43.42 | lukerobi | seems strange it only happens when calling 2 or 3 phone numbers, and we have 0 issues with the rest. |
| 16:44.00 | Greenlight | Ahh so it's specific to the number you're calling. |
| 16:44.01 | WIMPy | BTW: Does anyone have an iea how to debug one-way transcoding? |
| 16:44.31 | WIMPy | Or idea even. |
| 16:44.46 | lukerobi | Indeed... we can make 100 calls successfully, but when we call 2 or 3 numbers they can't hear us. :/ Very frustrating... no errors in the CLI |
| 16:44.51 | igcewieling | WIMPy: maybe the two directions are using different codecs? What is the actual issue? |
| 16:45.24 | WIMPy | igcewieling: one-way audio on calls where Asterisk has to transcode. |
| 16:45.30 | lukerobi | Our sip trunk is flowroute |
| 16:45.36 | igcewieling | WIMPy: which codecs? |
| 16:45.50 | WIMPy | G.722 and G.711 |
| 16:45.55 | Greenlight | lukerobi: You'll need to enable SIP debug and check for anything unusual on the calls which fail. |
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| 16:51.17 | wasanzy | I have this too already: exten => s,1,GoSub(cleanup,start,1(${CALLERID(num)})) |
| 16:51.26 | wasanzy | in extensions.conf |
| 16:51.56 | HolTech | Can someone point me in the right direction. I would like to understand if it is possible to use two analog lines from the telco with *... each connected to its own ATA, and their own extension in * , is it possible for * to answer the call on line 1, and hand it off to line 2, then answer another call on line 1? |
| 16:52.33 | HolTech | clearly such things are possible as big call centers do this, but what is the underlying technology going on ? |
| 16:52.38 | WIMPy | analog is evil |
| 16:52.47 | [TK]D-Fender | HolTech: No |
| 16:53.06 | WIMPy | You want to transfer a call and have the line free again? Or what exactely is your question? |
| 16:53.36 | igcewieling | HolTech: no, you cannot do that with analog |
| 16:53.49 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:54.31 | [TK]D-Fender | This is not possible period |
| 16:54.43 | igcewieling | [TK]D-Fender: 2BCT |
| 16:54.52 | [TK]D-Fender | not on analog |
| 16:54.58 | WIMPy | horrible misnomer |
| 16:55.01 | HolTech | Drag. So the idea being outside phone calls in on phone number, then * answers, and transfers to some other extension, and is then available to take another call. Basically what a call center does. where you call 1-800-xxx... and that call goes to some operator, and then its free to answer another call |
| 16:55.01 | igcewieling | that is what I said 8-) |
| 16:55.07 | [TK]D-Fender | And yes, 2BCT is the good 'ole PRI option |
| 16:55.25 | igcewieling | HolTech: 1) a call center does not use analog 2) a call center does not use ATAs. |
| 16:55.49 | [TK]D-Fender | HolTech: Don't say "call center". It isn't the business that has the functionality, it's the LINK TECHNOLOGY |
| 16:55.55 | igcewieling | 3) the same line is not used for the 2nd and 3rd, etc calls. |
| 16:56.49 | HolTech | Isnt the line in analog at some point though ? |
| 16:57.02 | [TK]D-Fender | HolTech: the line is you to the TELCO |
| 16:57.06 | WIMPy | no |
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| 16:58.16 | igcewieling | HolTech: you will get the same answer no matter how you phrase the question. |
| 16:58.22 | HolTech | Then how does the telco give it to the "call center"... they are paying the telco for that # and line right, or is it something far more advanced than a normal phone line. ie T1 or something? |
| 16:58.32 | [TK]D-Fender | PRI <-------------------- |
| 16:58.35 | [TK]D-Fender | ~pri |
| 16:58.35 | infobot | hmm... pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 16:58.38 | HolTech | lol, thanks guys, your help is always appreciated |
| 16:58.39 | igcewieling | HolTech: call centers do not use analog. |
| 16:58.40 | [TK]D-Fender | NOT analog |
| 16:59.00 | HolTech | ( waving my newbie flag ) |
| 16:59.06 | HolTech | thanks |
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| 17:08.44 | Mormaii | When I try to make a call using SIP and the SPEEX codec my asterisk crashes |
| 17:10.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:10.22 | Greenlight | Which Asterisk version ? |
| 17:10.32 | navaismo | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 17:11.40 | wasanzy | Hi am getting this message, what could be wrong? does it have to do with some context? http://paste.scsys.co.uk/266208 |
| 17:13.05 | [TK]D-Fender | wasanzy: your aren'tpaying attention to the PARAMETERS you are passing to Gosub |
| 17:13.20 | [TK]D-Fender | wasanzy: Go read the instructions fo rit again an look at the precise items you are passing to it. |
| 17:13.52 | wasanzy | this is not a system I worked on myself so is confusing the more |
| 17:13.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.239) |
| 17:14.52 | [TK]D-Fender | wasanzy: Nothing to be confused about . Read the syntax. Look at what you are passing Gosub. Look at your dilaplan. Realize there isn't a match |
| 17:15.00 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:15.23 | wasanzy | that is in extension.conf right? |
| 17:15.31 | [TK]D-Fender | wasanzy: Of course |
| 17:15.38 | Mormaii | Greenlight: Asterisk 1.6.2.0-beta4 |
| 17:15.42 | [TK]D-Fender | wasanzy: You are GOSUB-ing to a place there is no match for |
| 17:16.01 | Mormaii | I've even disabled all codecs and only allowed ulaw in sip.conf but It still crashes |
| 17:17.08 | Greenlight | ~asterisk upgrade |
| 17:17.08 | Mormaii | -Using SIP RTP CoS mark 5 |
| 17:17.08 | Mormaii | -Segmentation fault |
| 17:17.14 | wasanzy | there is no Gosub in the extensions.conf |
| 17:17.18 | Greenlight | Use a recent version of asterisk. |
| 17:17.34 | Greenlight | And not an ancient beta. |
| 17:17.34 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 17:17.36 | WIMPy | ~upgrade asterisk |
| 17:17.37 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:17.45 | Greenlight | WIMPy: That's the one! |
| 17:18.07 | WIMPy | And far too long. |
| 17:18.10 | Mormaii | I see, problem is that this are production servers with lots of users. Upgrading them isn't just that simple |
| 17:18.36 | [TK]D-Fender | Mormaii: There is little reason you can't at least upgrade WITHIN your current branch |
| 17:18.48 | [TK]D-Fender | Mormaii: And your branch is not supported at all anymore. |
| 17:19.08 | [TK]D-Fender | Mormaii: So you need to decide if not having a functional speex is worth it or not. |
| 17:19.42 | Greenlight | Why are you running a beta in production anyway |
| 17:19.58 | Mormaii | Because it's stable and we haven't had any problems in months |
| 17:20.06 | Mormaii | *It's stable for us* |
| 17:20.12 | Greenlight | Evidentely |
| 17:20.14 | WIMPy | months? |
| 17:20.21 | Greenlight | [06:17pm] <Mormaii> -Segmentation fault <-- Looks stable to me. |
| 17:20.24 | WIMPy | IT HAS BEEN HORRIBLY OUTDATED FOR YEARS. |
| 17:20.25 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:20.28 | WIMPy | oos |
| 17:20.31 | WIMPy | oops |
| 17:20.37 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 17:20.43 | Mormaii | Well I've just discovered that while testing something Greenlight |
| 17:20.50 | Mormaii | Else it works flawlessly |
| 17:20.55 | Greenlight | yea, upgrading will likely fix your issue |
| 17:20.58 | Mormaii | Yeah I know I have outdated asterisk >.< |
| 17:21.06 | Mormaii | I'll check if this happens on the updated one |
| 17:21.08 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:21.15 | Greenlight | 11.5 is good :) |
| 17:21.21 | Mormaii | I WANT to upgrade, but it's not my call |
| 17:21.28 | Mormaii | You know, I have a boss |
| 17:21.30 | Mormaii | :/ |
| 17:21.35 | Greenlight | If I had a penny... |
| 17:21.40 | [TK]D-Fender | Mormaii: Ask your boss if he wants speex then |
| 17:22.02 | Mormaii | He goes by the policy: Don't touch anything if it's working |
| 17:22.16 | WIMPy | IF |
| 17:22.35 | Greenlight | So, wait *till* you get hacked, before making it secure. Sounds sensible. |
| 17:22.49 | Mormaii | Yeah I know, not very security concious |
| 17:23.12 | Mormaii | He had disabled alwaysauthreject |
| 17:23.28 | Mormaii | No type of login failure banning like fail2ban or anything |
| 17:23.31 | WIMPy | Heh, if you don't get hacked, security is just a waste of time, isn't it? |
| 17:23.42 | Greenlight | Yup, why bother! |
| 17:23.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:23.57 | Mormaii | Who cares about people making free phone calls and you loosing money |
| 17:24.04 | Mormaii | Security is just an illusion anyways |
| 17:24.14 | WIMPy | True story |
| 17:24.20 | Mormaii | Ok this one is more recent: Asterisk 1.8.15-cert2 |
| 17:24.30 | Greenlight | Not nearly as recent as 11.5 |
| 17:24.33 | WIMPy | Use Asterisk 11. |
| 17:24.36 | Mormaii | Still very old? |
| 17:24.47 | Greenlight | 1.8 is still supported, *just* |
| 17:24.58 | Greenlight | But why not use the most recent |
| 17:25.11 | WIMPy | How soon do you want todo the next upgrade? |
| 17:25.46 | [TK]D-Fender | [13:24]MormaiiSecurity is just an illusion anyways <- no, security is a PROCESS |
| 17:26.40 | Greenlight | I think he forgot his <sarcasm> </sarcasm> tags |
| 17:26.49 | Mormaii | Yes Greenlight |
| 17:26.53 | Mormaii | I forgot my tags |
| 17:26.54 | WIMPy | Who? |
| 17:27.06 | Greenlight | Mormaii |
| 17:27.10 | Mormaii | It was sarcasm, I enable alwaysauthreject and installed fail2ban |
| 17:27.20 | Mormaii | anyway, on 1.8.15 this crash doesn't happen |
| 17:27.48 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:27.49 | WIMPy | alwaysauthreject leaked information in older versions. |
| 17:27.50 | [TK]D-Fender | Mormaii: We're at 1.8.23 BTW.... |
| 17:27.52 | Greenlight | [06:27pm] <Mormaii> It was sarcasm, I enable alwaysauthreject and installed fail2ban <-- That *doesnt* make you anywhere near secure |
| 17:28.12 | [TK]D-Fender | Mormaii: so don't go thinking of upgrading to another outdated release as your "fix". |
| 17:28.15 | WIMPy | +still |
| 17:28.21 | Greenlight | Why not upgrade to 11? |
| 17:28.32 | Mormaii | I know, I want to use the latest but it doesn't depend on me |
| 17:28.37 | igcewieling | setting allowguest=no will give you more security than fail2ban and alwaysauthreject |
| 17:28.38 | Mormaii | You need to understand that ... |
| 17:28.55 | [TK]D-Fender | Mormaii: Ok, well you've got our answer. |
| 17:29.03 | [TK]D-Fender | Mormaii: go Ask your boss what he wants |
| 17:29.06 | Greenlight | So, you boss will upgrade to 1.8 (not the most recent one), but NOT 11. |
| 17:29.10 | WIMPy | thinks someone else needs to understand something. |
| 17:29.22 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:29.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:29.29 | [TK]D-Fender | Mormaii: Anything else we can answer for you? |
| 17:29.29 | drmessano | "This is broken and it's 5 years old and I refuse to upgrade, but I need a fix now" |
| 17:29.30 | igcewieling | I see no major reason to upgrade to Asterisk 11 |
| 17:29.34 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 17:29.37 | Mormaii | Yeah I know, I just tested it and an upgrade fixes this issue |
| 17:29.39 | igcewieling | You should at least upgrade to the latest 1.8.x |
| 17:29.47 | Greenlight | igcewieling: Why *not* |
| 17:29.50 | Mormaii | I'll try to convince him, but it's no easy task |
| 17:29.53 | Greenlight | He's going to upgrade *ANYWAY* |
| 17:29.55 | drmessano | [13:29:37] <Mormaii> Yeah I know, I just tested it and an upgrade fixes this issue <---- ZIIIING |
| 17:30.01 | [TK]D-Fender | igcewieling: lots of good one. hangup handlers, T.38 termination, vastly superior confbridge to name a few |
| 17:30.08 | WIMPy | I see quite some reasons for 11. |
| 17:30.23 | igcewieling | Greenlight: because 11 is a major release and I never recommend upgrading to a new major release without a real reason |
| 17:30.24 | drmessano | 11 makes 1.8 look like 1.6.0 |
| 17:30.33 | Greenlight | He's on 1.6 |
| 17:30.42 | Greenlight | 1.8 is a major release upgarde anyway |
| 17:31.00 | Mormaii | If you could give me some more security advices then I'd be glad too |
| 17:31.09 | Greenlight | We did. Upgrade. |
| 17:31.22 | Mormaii | Anything else besides an upgrade? |
| 17:31.39 | WIMPy | There's a readme-seriousely-bestp[ractices or the like. Do what it says. |
| 17:31.55 | drmessano | Yep |
| 17:31.58 | Greenlight | The basics, block IP's that you don't need to allow access etc etc |
| 17:32.00 | drmessano | It's all caps too, IIRC |
| 17:32.25 | drmessano | LEIK-SRSLY-DUH-READTHISNOWK-BESTEST-PRACTICES |
| 17:32.26 | WIMPy | But using a version that doesn't allow guessing of sip accounts is surely a major one. |
| 17:32.30 | drmessano | It's really all that |
| 17:32.31 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:32.53 | Mormaii | Yup, did that too Greenlight |
| 17:33.06 | Mormaii | Just wanted to know if I'm missing something far too obvious for you but not for me |
| 17:33.15 | Mormaii | BESIDES UPGRADING |
| 17:33.20 | Greenlight | ;) |
| 17:33.42 | [TK]D-Fender | Mormaii: Nooneis going to debug this ancient version |
| 17:33.45 | WIMPy | Read that file. |
| 17:33.46 | Greenlight | Well, do you have the same upgrade policies for the rest of your stuff? |
| 17:34.06 | Greenlight | Are you running an outdated, and security hole ridden kernel ? |
| 17:34.12 | Mormaii | No, just for everything telephony (I think it's called like that) |
| 17:34.30 | Mormaii | Most of our clients believe that they need telephony 24/7 |
| 17:34.38 | Mormaii | So you can't even turn it off for 5 minutes to upgrade |
| 17:34.40 | drmessano | Keep in mind that while security updates handle exploits, they are often not the same architecture changes that create a more secure product in general. A major new version often features these kinds of changes. Security updates on older versions are just "silly putty shoved in the hole" |
| 17:35.02 | drmessano | You don't need 5 minutes to upgrade |
| 17:35.12 | [TK]D-Fender | drmessano: More like cut-up newspaper then silly-putty |
| 17:35.16 | Greenlight | "core restart when convenient" |
| 17:35.17 | drmessano | Configure, make install, restart |
| 17:35.23 | Mormaii | It's just a way of saying |
| 17:35.32 | Mormaii | We once restarted to turn alwaysauthreject |
| 17:35.33 | drmessano | So more like 30 seconds of downtime |
| 17:35.37 | Mormaii | Got a call from the client |
| 17:35.38 | drmessano | Thats one dropped call |
| 17:35.39 | Mormaii | Instantly |
| 17:35.46 | drmessano | Ok, so? |
| 17:35.54 | Mormaii | We got told to f*** off |
| 17:35.58 | Mormaii | and not touch anything else |
| 17:36.06 | drmessano | Everyone has downtime. 30 seconds, tell them to call back when its something important. 5 minutes? Thats YOUR bad |
| 17:36.07 | Greenlight | You should probably tell them first... |
| 17:36.21 | Greenlight | ... if we just cut stuff off, our clients would not be happy. |
| 17:36.44 | Greenlight | BUt if you *tell* them that you need 2 minutes downtime, and they need to say when is best.. |
| 17:36.54 | *** join/#asterisk italoros_ (~italoross@187.60.66.11) |
| 17:37.12 | Mormaii | Sometimes they say that never is best |
| 17:37.13 | WIMPy | Man, if major telcos cut you of for hours regularly because the do soem "maintenance", you can do it too. |
| 17:37.23 | WIMPy | And if you do it at the same time, noone will notice. |
| 17:37.28 | [TK]D-Fender | Mormaii: Just so you're aware of the sum of your situation. Your setup does not work. Noone is going to hand-patch the base you have. You apparently aren't going to get cleared to change anything. Yet you want your problem to change. |
| 17:37.29 | Mormaii | When you depend on your clients to tell you what you can do, you have a bad time |
| 17:37.45 | drmessano | Theres a big difference between restarting an Asterisk instance for an upgrade and rebooting a server, which is cause for a maintenance window to be established with users |
| 17:37.52 | igcewieling | Mormaii: not a bad time, you have an inability to do your job |
| 17:37.56 | drmessano | Upgrading to 11, if planned corrected, is the former |
| 17:38.06 | Mormaii | nono [TK]D-Fender, I'm grateful that now I know what causes the issue |
| 17:38.10 | [TK]D-Fender | Mormaii: Tell them their option and if they want the situation to change they're just going to have to let you upgrade |
| 17:38.14 | Mormaii | Now it's my "problem" to see how I upgrade |
| 17:38.36 | drmessano | configure, make install, restart asterisk <--- seems pretty simple |
| 17:38.48 | Mormaii | Happens so often on my job, whenever we need to change firewall rules or upgrade MPD5 or anything of the sort we get so many blocks |
| 17:38.56 | Greenlight | "core restart when convenient" <-- They'd never notice :) |
| 17:39.28 | Mormaii | Some clients are more flexible and tell us that it's ok to upgrade but do it on a weird hour (3AM~) |
| 17:39.31 | igcewieling | Greenlight: unless you have an unexpected issue caused by upgrading to release 2 major upgrades from what you are currently running |
| 17:39.38 | drmessano | Mormaii, letting customers or users bully you into not providing a reliable infrastructure is a failure on your part to serve them. |
| 17:39.56 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:39.58 | Greenlight | igcewieling: I'd have expected a test on dev machine first though |
| 17:40.17 | igcewieling | like, oh, I don't know, needing to install libuuid and not realizing it until too late |
| 17:40.37 | Mormaii | drmessano: Sometimes that is the issue. They don't want us to provide better service for them and then they complaint about problems. |
| 17:40.41 | drmessano | If I let PD's tell me I couldn't shut a station off for 20 seconds during the day to restart a process and we go off the air for 15 minutes, it's ON ME... not them for "Not letting me do my job" |
| 17:41.02 | drmessano | Mormaii, again, that's a failure on your part.. not theirs |
| 17:41.11 | Greenlight | igcewieling: Lol, that's *never* caught *anyone* out.... |
| 17:41.53 | Mormaii | drmessano: How is that a failure on my part when it's my boss orders? |
| 17:42.04 | Mormaii | drmessano: Should I just go against him and have a chance of being fired? |
| 17:42.18 | drmessano | Mormaii, does he own the company? |
| 17:42.28 | Mormaii | drmessano: He does |
| 17:42.53 | Mormaii | He owns the company and he is my boss so whatever he says has to be done |
| 17:42.54 | igcewieling | I disagree. If I refuse treatment, is it the doctor's fault? |
| 17:43.11 | Mormaii | I'd say it's your fault |
| 17:43.17 | Mormaii | Same situation applies here |
| 17:43.20 | igcewieling | if the doctor does the treatment anyway, who is in the wrong? |
| 17:43.42 | igcewieling | if a customer refuses treatment, it is not my fault. |
| 17:43.43 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:43.49 | Greenlight | Depends, the doctor hasn't make it clear of the effect of NOT having treatment done, then it's the doctors fault. |
| 17:44.05 | igcewieling | Greenlight: indeed. |
| 17:44.09 | drmessano | igcewieling, if a customer is paying me for service and they demand that I never take anything down to upgrade or update because it's a bother, and something fails or something is compromised due to my inaction, I am totally and completely at fault |
| 17:44.24 | igcewieling | drmessano: I disagree. |
| 17:44.32 | Mormaii | I disagree too |
| 17:44.39 | igcewieling | something failed because of the CUSTOMER'S inaction. |
| 17:44.50 | drmessano | igcewieling, that's a lack of taking ownership for the services you provide then |
| 17:44.53 | Mormaii | Sometimes even by telling them the effect of NOT having the treatment, they still refuse. |
| 17:45.13 | Mormaii | It has happened to me, I've told a costumer that his server was swapping due to an old version of mpd that contained a memory leak. |
| 17:45.30 | Mormaii | Told him I needed to shut it down for 2 minutes and disconnect everyone for the upgrade |
| 17:45.36 | drmessano | igcewieling, no, there was no inaction on the customers part.. Just yours. You let them dictate a zero length maintence window when maintenance is a necessity. That's inaction on your part |
| 17:45.37 | igcewieling | drmessano: or it is a "customer is always right" policy. Or it could be a policy of "I have more important shit to do than provide counseling to a customer intent on failure" |
| 17:45.37 | Greenlight | If you make it *crystal clear* that it would cause *more problems* then it's customers fault. |
| 17:46.04 | igcewieling | drmessano: no matter what you do they will be former customers anyway. |
| 17:46.19 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:46.49 | drmessano | Greenlight, I wouldn't even go as far as saying it's an issue of notification. The fact that YOU know the consequences and still let the customer dictate that there is no downtime is inaction. |
| 17:46.53 | Greenlight | It does seem a silly arrangement; a customer thinking they have 100% uptime, for a service which isn't designed that way. |
| 17:47.21 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:47.33 | igcewieling | Greenlight: I blame sales. |
| 17:47.37 | Greenlight | For example, say 911/999 emergency services. They, rightly so, demand 100% uptime. And pay a premium for redundancy and whatnot. |
| 17:47.47 | Greenlight | igcewieling: I agree! |
| 17:48.47 | Mormaii | drmessano: But at the end of the day, He's the one that always is right. |
| 17:49.05 | Greenlight | Who, the boss ? |
| 17:49.08 | Mormaii | I'm just technical support, I can tell him the issues/benefits and risks of upgrading and not upgrading but the final decision is their |
| 17:49.16 | Mormaii | The boss/Clients of the boss |
| 17:50.08 | Greenlight | It's a difficult line to walk. At that point I would start to get an email chain going, when i get them to accept that eventaully they WILL have UNEXPECTED and PROLONGED downtime if they don't take the scheduled short downtime. |
| 17:50.18 | drmessano | Mormaii, that's not true at all. There is a big difference between "the boss/customer is never wrong" and "the boss/customer is always right". You're not right here, nor are they |
| 17:50.46 | Greenlight | You can then play the "I told you so card" when something *does* go wrong. |
| 17:50.54 | Mormaii | I did it the other day |
| 17:50.56 | Greenlight | Trust me, after that, they start to listen. |
| 17:51.12 | Mormaii | The boss told me to upgrade to something I knew that had errors and didn't work well |
| 17:51.20 | Mormaii | I did so on a client and notified them of the upgrade |
| 17:51.33 | Mormaii | The client tried to use it and broke everything. 4000 people lost their internet connection |
| 17:51.45 | Mormaii | And then he blamed me for telling the customer I had upgraded |
| 17:52.03 | Mormaii | When I clearly told the boss that what I was being asked to install had bugs. |
| 17:52.12 | Greenlight | A customer refused to get a proper internet connection in cause of cost. I said that ADSL WILL FAIL, and would take HOURS OR DAYS to fix, he brushed it off and said he'd send his staff home. It took over 18 months, but it DID fail. |
| 17:53.18 | Greenlight | He came on the phone screaming,till I reminded him, at which point he swiftly uturned and ordered fibre in... |
| 17:54.12 | Mormaii | hahah, you were lucky they recognized that you've told them it would fail |
| 17:54.23 | Mormaii | my boss still managed to blame it on me.. |
| 17:54.26 | mjordan | skrusty: define 'stream audio files over FastAGI' |
| 17:54.43 | igcewieling | Mormaii: this is why you need an e-mail trail |
| 17:54.48 | Greenlight | +1 |
| 17:54.54 | drmessano | In my environment, every minute of downtime translates to immediate lost revenue. There's always a cost associated with downtime, and a cost associated with lack of proper maintenance. Upper management insists on ZERO downtime. I insist on minimizing downtime and revenue losses. I always win, because zero downtime is not possible. Something does fail, and not just in theory. |
| 17:55.06 | Mormaii | That's what we somtimes do with our ticket system igcewieling with the clients |
| 17:55.07 | Greenlight | And make sure you repeat it over and over again, in a tone that says "I think this is stupid" |
| 17:55.11 | Mormaii | The problem is that this was inside the office |
| 17:55.27 | drmessano | 30 seconds of downtime during the day always trumps 45 minutes to an hour of it. |
| 17:56.20 | Mormaii | Totally agree with you drmessano |
| 17:56.33 | Mormaii | If the boss had listened to me he wouldn't have had 7 hours of downtime |
| 17:56.41 | pabelanger | Q: What does *68 on a cisco ATA do? |
| 17:56.46 | Mormaii | Everything went to hell that day |
| 17:59.48 | drmessano | Mormaii, you can't agree and still argue that "the customer won't let us..." ... It's one thing to be bound to what the boss/owner dictates, but the attitude that end users or customers have any sort of hold over the steps you need to take to provide service is failing them entirely |
| 17:59.53 | [TK]D-Fender | pabelanger: Depends on which model and how it was configured |
| 18:00.36 | igcewieling | pabelanger: translation: question too vague, ask again |
| 18:01.02 | Mormaii | drmessano: We are told that we have to do what the customers dictates |
| 18:01.07 | drmessano | Failing to provide support due to a customer/users lack of desire to work with us is a failure on our part. If we don't own what we do, then what is the point in doing it. |
| 18:01.33 | wasanzy | ok, I have created a different extension.conf : http://paste.scsys.co.uk/266216 and my dongle.conf: http://paste.scsys.co.uk/266202 when I call the dongle number, the phone should have answerd automatically, but some times, the call goes through without auto answer. |
| 18:01.41 | wasanzy | what am I doing wrong? |
| 18:02.05 | [TK]D-Fender | wasanzy: You are not looking at the call. |
| 18:02.07 | [TK]D-Fender | That is what's wrong |
| 18:03.30 | igcewieling | [TK]D-Fender: not given up on wasanzy yet? |
| 18:04.49 | [TK]D-Fender | igcewieling: Today's quoa is on it's way to being filled |
| 18:04.54 | [TK]D-Fender | quota* |
| 18:05.25 | wasanzy | I get this error now: http://paste.scsys.co.uk/266217 |
| 18:05.25 | igcewieling | daily quota? you are very nice. I use a weekly or monthly quota. |
| 18:05.46 | wasanzy | igcewieling: I don't understand |
| 18:06.06 | [TK]D-Fender | wasanzy: he wasn't talking to you |
| 18:06.30 | wasanzy | ok |
| 18:06.36 | [TK]D-Fender | wasanzy: [Aug 22 18:03:07] WARNING[17674][C-00000008]: pbx.c:6390 __ast_pbx_run: Channel 'Dongle/dongle0-0100000008' sent to invalid extension but no invalid handler: context,exten,priority=dongle,+1234567890,1 |
| 18:06.43 | [TK]D-Fender | wasanzy: there is no MATCH for that in your dialplan |
| 18:06.50 | [TK]D-Fender | wasanzy: What part of that error isn't clear? |
| 18:07.11 | [TK]D-Fender | wasanzy: It's like me telling you the forks are in the top shelf... and they ARENT'T there |
| 18:07.39 | [TK]D-Fender | wasanzy: context,exten,priority=dongle,+1234567890,1 <---- there is no match for this in your extensions.conf |
| 18:07.59 | [TK]D-Fender | wasanzy: YOU told it to go somewhere. You did NOT make an exten properly that matches what you see right there |
| 18:08.14 | wasanzy | [TK]D-Fender: I understand your point, but am just thinking of how to define those in the extensions.conf |
| 18:08.28 | [TK]D-Fender | wasanzy: make a pattern that MATCHES it |
| 18:08.40 | [TK]D-Fender | wasanzy: This very obviously does not. |
| 18:08.47 | wasanzy | ok |
| 18:08.50 | [TK]D-Fender | wasanzy: What line do you think SHOULD have matched it? |
| 18:08.56 | igcewieling | wasanzy: just like everything else. [dongle] exten => +1234567890,1,Noop(I got a dongle!) |
| 18:09.18 | igcewieling | wasanzy: I see you've still not read The Book. |
| 18:09.54 | igcewieling | The book shows you how to create extensions and all sorts of other useful stuff. |
| 18:10.25 | wasanzy | igcewieling: the thing is, when you are ask to do something urgent, going to read the book delays and you will not beat your deadline |
| 18:10.29 | wasanzy | that is the problem |
| 18:10.37 | wasanzy | am reading the book but very slow |
| 18:10.54 | [TK]D-Fender | wasanzy: Do you expect to do a job without understanding the job at all? |
| 18:11.00 | igcewieling | wasanzy: I don't care. We are not your personal tech support lackeys. Do some work on your own. |
| 18:11.15 | drmessano | wasanzy, you don't need to read the whole book at once. You can use it as a research tool. It is broken into sections |
| 18:11.27 | drmessano | wasanzy, at least read relevant sections |
| 18:11.28 | [TK]D-Fender | wasanzy: Do you think your boss will rehire for your position immediately when you fail your deadlines? |
| 18:11.35 | [TK]D-Fender | drmessano: I think I see an opening.... |
| 18:11.55 | igcewieling | I only see someone who wants us to do all the work. |
| 18:12.01 | wasanzy | I think this part is the problem: exten => 1234567890,1,Answer() |
| 18:12.07 | drmessano | [TK]D-Fender, remember when I used to ask for the e-mail address of the HR person at their job? Good times |
| 18:12.42 | drmessano | [TK]D-Fender, "Do they accept PDFs of resumes at your work? I want to go ahead and submit mine for your job" |
| 18:12.49 | [TK]D-Fender | wasanzy: If you don't understand your dialplan basics then forget, it's GAME OVER time. Asterisk is all about the dialplan. that is 90%+ of any system |
| 18:12.56 | igcewieling | wasanzy: in what universe is the string 1234567890 the same as the string +1234567890 |
| 18:13.34 | wasanzy | how do I escape "+"? |
| 18:13.40 | [TK]D-Fender | you don't |
| 18:13.42 | drmessano | I'd like to think i've mellowed out. Now I just look for them on LinkedIn |
| 18:13.43 | igcewieling | why would you need to? |
| 18:13.46 | [TK]D-Fender | and you don't HAVE TO |
| 18:13.48 | igcewieling | these are not regular expressions |
| 18:13.55 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 18:13.58 | drmessano | \+ |
| 18:14.02 | drmessano | lol |
| 18:14.31 | wasanzy | drmessano: why will you need my HR's email? just because I asked for a help? |
| 18:14.50 | drmessano | wasanzy, I wasn't talking about you |
| 18:15.02 | wasanzy | I know \ is escape character |
| 18:15.27 | igcewieling | wasanzy: + is just a normal character in asterisk, you do not need to escape it. |
| 18:15.44 | wasanzy | alright sorry |
| 18:15.59 | drmessano | wasanzy, i'm not interested in this circular debacle. This is basic dialplan, which is addressed as a chapter in the book and in many how-to's. I think what others are trying to emphasize is that you're failing to show any initiative, which is making it difficult |
| 18:16.11 | drmessano | As for myself, I am eating popcorn |
| 18:16.28 | drmessano | MMMM POPCORN |
| 18:16.58 | wasanzy | drmessano: I love popcorn so much, should I come for some? |
| 18:17.09 | drmessano | wasanzy, I doubt you will |
| 18:19.51 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 18:20.27 | wasanzy | drmessano: oh why, you are my good friend ooo |
| 18:20.30 | wasanzy | lol |
| 18:20.49 | wasanzy | am reading the dialplan basics now |
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| 18:41.21 | wasanzy | thank you guys for pushing me to read the book, am enjoying what am understanding from it so far |
| 18:46.36 | boom^time | They just do that because they get a kickback every time someone buys the free book. |
| 18:47.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.105) |
| 18:47.18 | [TK]D-Fender | And if you don't like the free book we'll give you DOUBLE your money back |
| 18:47.49 | wasanzy | TK]D-Fender: I don't understand |
| 18:48.15 | [TK]D-Fender | facepalms |
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| 18:57.28 | drmessano | lol |
| 19:04.22 | boom^time | Does anyone have any clue why AMD will not detect any audio when I call my cell? |
| 19:04.49 | boom^time | I've turned the silence_threshold down to 32 as desperate as that is. That that level a flea can sneeze and it'l register. |
| 19:06.34 | boom^time | nm ignore me. |
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| 19:18.59 | boom^time | Seriously it is really touchy with cells |
| 19:20.15 | WIMPy | hasn't tried for a long time, but in the time when the PSTN still worked, you'd get a redirection notifier when a call went to network supplied VM. |
| 19:24.23 | boom^time | Hmm that's an interesting factoid |
| 19:24.31 | boom^time | Thanks |
| 19:27.17 | boom^time | You'd still have to know you were calling a cell that had a carrier which sent that before the call |
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| 19:35.53 | mastermonk | anyone here using google voice motif? |
| 19:37.42 | mastermonk | I have it configured but I am having a hard time with inbound calls, I am able to make external calls, |
| 19:37.48 | mastermonk | anyone played around with it? |
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| 19:46.28 | skrusty | mjordan: sorry for late reply, i.e. send audio over fastagi (tcp) so that the audio can be creatred on a remote machine and played into the channel |
| 19:47.12 | skrusty | i noticed there is a patch floating about on github, but nothing official |
| 19:48.49 | mjordan | skrusty: highly unlikely |
| 19:48.57 | mjordan | there's no plans at least to do that. |
| 19:49.41 | mjordan | Much more likely would be to specify a URI to a file to be played |
| 19:49.54 | mjordan | where that URI may not be a resource local to the Asterisk server |
| 19:50.31 | skrusty | yeah :/ shame really - i understand why. but it would be such a handy feature |
| 19:50.38 | skrusty | i wish i knew c better! :) |
| 19:50.59 | skrusty | i coordinate the asternet (previously asterisk.net) project, and we've been talking about it quite a bit |
| 19:51.40 | skrusty | at the moment people have to use shares or shared storage, save the audio to that shared path and then use streamfile etc |
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| 20:04.31 | mjordan | skrusty: the less quirky way to do that would be to specify a URI. |
| 20:04.53 | mjordan | skrusty: Asterisk, upon seeing a URI that doesn't resolve to a local resource, would CURL it down to a temporary location and play it back |
| 20:16.34 | skrusty | mjordan: yeah |
| 20:17.41 | igcewieling | eagi? |
| 20:19.07 | skrusty | fastagi doesn't support eagi though |
| 20:21.07 | igcewieling | ah. |
| 20:21.14 | igcewieling | so top using fast AGI |
| 20:21.21 | igcewieling | so stop using fast agi |
| 20:21.54 | skrusty | heh |
| 20:22.28 | igcewieling | in the standard fast agi examples it doesn't give you THAT much advantage to run it remotely. |
| 20:22.48 | igcewieling | you still have to fork/exec from the fastagi server process |
| 20:23.55 | skrusty | for a lot of people however, fastagi works for their requirements, so it's rather frivolous to assume they can just stop using it :) |
| 20:24.20 | skrusty | plus, if an AGI expends lots of resources, why would you want it running on your asterisk box? |
| 20:25.20 | igcewieling | skrusty: but for you fast agi isn't working for your requirements |
| 20:25.56 | igcewieling | skrusty: how many calls per second do you need to handle? |
| 20:26.11 | skrusty | it's not volume, it's what's done on the call - anyway, im not asking for me |
| 20:26.18 | skrusty | im asking for the asternet user base |
| 20:27.37 | skrusty | one option we have is to directly support AGI running on mono as an option, at the moment we only support FastAGI and AMI in the framework |
| 20:28.26 | skrusty | personally i like mjordan's idea of URI based resources |
| 20:31.45 | skrusty | URI would make using services like http://tts-api.com/ nice and easy too |
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| 22:46.16 | kemmler | hi |
| 22:46.26 | kemmler | i like sip |
| 22:46.32 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 22:47.12 | Katty | NUGGET |
| 22:47.15 | Katty | hugs le Nugget |
| 22:48.08 | navaismo | i like icecream |
| 22:48.23 | Katty | me too |
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| 22:51.34 | Nugget | la la la |
| 22:52.32 | Katty | how're you le Nugget |
| 22:56.11 | Nugget | busy week, but otherwise good |
| 22:57.07 | Katty | yay |
| 23:04.46 | kemmler | anyone know how to check what the p-asserted-identity is in a dialplan |
| 23:05.28 | WIMPy | see function SIP_HEADER |
| 23:06.23 | kemmler | gotcha, thanks |
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| 23:08.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:14.21 | WIMPy | goes back to sleep |
| 23:14.48 | WIMPy | So here the giveaway: I do it like |
| 23:14.55 | WIMPy | same => n,Set(CALLERID(ANI)=${CUT(CUT(SIP_HEADER(P-Asserted-Identity):,@,1),:,2)}) |
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| 23:21.06 | kemmler | awesome thanks WIMPy |
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