| 00:02.58 | carrar | moo |
| 00:03.05 | ChannelZ | milks carrar |
| 00:03.41 | carrar | wet outside |
| 00:03.57 | carrar | pouring down rain last night |
| 00:05.03 | *** join/#asterisk k3rmat (~k3rmat@64.122.45.178) |
| 00:05.15 | carrar | must get my morning coffee |
| 00:08.07 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 00:19.55 | atan | How would one send a private call to flowroute? Doesn't seem to play nice with my Private<00000000000> supplied |
| 00:23.39 | sawgood | exten => _1855X.,2,Dial(SIP/flowroute/12345678#${EXTEN}) |
| 00:23.52 | sawgood | that is a working example of how I send some calls to flowroute |
| 00:26.53 | atan | Can you set P-Asserted-Identity inside sip.conf? |
| 00:29.08 | sawgood | flowroute never required that from me, so I do not know (sorry) |
| 00:44.39 | *** join/#asterisk brut- (~brut-@h75-100-39-115.pqlkmn.dsl.dynamic.tds.net) |
| 00:44.57 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
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| 00:47.10 | X-Rob | OK. I think I need an old Tormenta ISA card for a talk I'm doing on asterisk. |
| 00:47.28 | X-Rob | Anyone have one lying around anywhere in a cupboard or a box or something? |
| 00:50.45 | atan | Well color me confused... why does this run every time, despite callerIdPrivacy being 0? |
| 00:50.50 | atan | http://pastebin.com/WTfyEtW5 |
| 00:54.43 | *** join/#asterisk TSM2 (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 01:16.08 | carrar | to start out with |
| 01:16.14 | carrar | you are missing matching brackets |
| 01:19.00 | atan | carrar: here's the latest http://pastie.org/private/tehtghb3dymkvyffnnbnq |
| 01:19.14 | atan | Where do I place the matching bracks? |
| 01:19.31 | carrar | read what I said |
| 01:19.34 | carrar | stil applies |
| 01:19.39 | atan | DOH! |
| 01:19.42 | atan | brb |
| 01:24.06 | atan | Okay added those on there but it's not going to the correct lines. Did I screw up on formatting it again? http://pastie.org/private/gq6btu43zb1b2emaopgta |
| 01:24.11 | atan | s/formatting/syntax/ |
| 01:24.19 | carrar | might need to put parentheses around your string as well |
| 01:25.20 | carrar | looks ok |
| 01:25.27 | atan | Inside sip.conf? I changed it to be "on" and "off" but it still skips right past I'm puzzled |
| 01:25.31 | carrar | now you need to find out what the real value of that string is |
| 01:25.35 | carrar | err variable |
| 01:26.16 | atan | I get ..."Caller ID Private is set to: "off"") in new stack |
| 01:27.11 | atan | Yet GotoIf($["${callerIdPrivacy}" = "off"]?noCallerIdPrivacy) does not send them to (noCallerIdPrivacy) |
| 01:30.03 | [TK]D-Fender | I don't see complete dialplan |
| 01:30.10 | [TK]D-Fender | Those 2 lines prove nothing on their own |
| 01:34.03 | atan | One sec :-) |
| 01:36.33 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 01:37.20 | atan | Please don't laugh: http://pastie.org/private/mtgs2yp8q9itczbclusa |
| 01:42.38 | [TK]D-Fender | atan: Now show me the complete failure |
| 01:44.41 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 01:45.29 | atan | [TK]D-Fender: http://pastie.org/private/ep44caarewe3mwagjfsjbg |
| 01:46.29 | atan | Numbers redacted a bit but the rest unchanged |
| 01:47.08 | [TK]D-Fender | atan: -- Executing [0123456789@outbound:14] NoOp("SIP/1134-0000034d", "Caller ID Private is set to: "on"") in new stack <-- noticae anything here... |
| 01:47.13 | [TK]D-Fender | ? |
| 01:47.33 | [TK]D-Fender | exten => _1NXXNXXXXXX,n,NoOp(Caller ID Private is set to: ${callerIdPrivacy}) |
| 01:47.42 | [TK]D-Fender | compare. Its right there... |
| 01:47.50 | atan | Something about the extra quotes? |
| 01:48.16 | [TK]D-Fender | YES |
| 01:48.22 | [TK]D-Fender | setvar = callerIdPrivacy="on" <--- you put literal quotes here |
| 01:48.27 | [TK]D-Fender | * doesn't do "typed" vars |
| 01:48.39 | atan | Okay so remove from my sip.conf then I do suppose? |
| 01:48.52 | [TK]D-Fender | you added literal chars which your gotoif won't parse out how you'd like |
| 01:48.54 | [TK]D-Fender | yes |
| 01:49.39 | atan | Bwhahaha! Great success. |
| 01:49.47 | atan | Thank you [TK]D-Fender and carrar :-) |
| 01:49.58 | [TK]D-Fender | atan: You're welcome |
| 01:50.11 | atan | The PHP in the back of my head is screaming "You put that in quotes... it's not an int... do it or I die" |
| 01:50.43 | [TK]D-Fender | atan: You could have also added the extra quotes to your gotoif .. but that would have been accmodating something that didn't serve any real purpose |
| 01:59.24 | atan | Yeah no worries that works really well now :-) rock on |
| 02:06.59 | atan | Is it possible to define a var at the start of my extensions.conf file and reuse it in the file? For example a username, password, etc |
| 02:07.09 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:07.45 | atan | Hmm, this [globals] thing looks interesting will try it out. |
| 02:13.14 | [TK]D-Fender | yup |
| 02:13.22 | [TK]D-Fender | ok, I'm off for the night. LAter all |
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| 03:10.13 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:18.51 | dlynes | atan, yes...that's exactly what the globals section is for |
| 03:19.09 | atan | :-) |
| 03:19.36 | dlynes | atan, I don't know if I'd suggest putting usernames and passwords in there, but.... |
| 03:20.05 | atan | Okay so question #2 is when I get an inbound call to my softphone I click on "redirect" and enter a new number, but I see this when I watch the call: Executing [9896543@outbound:22] Dial("Local/9896543@outbound-9c8d;2", "SIP/flowroute/9896543, 160, r") in new stack |
| 03:20.26 | atan | I have no idea where "Local" came from. It's not a context that can be used for calls. Is it a default that is used for stuff and I should set it up? |
| 03:20.36 | dlynes | atan, Local is another channel |
| 03:20.50 | dlynes | atan, Like SIP, or IAX2, or DAHDI |
| 03:21.02 | dlynes | atan, But there's no devices associated with Local channels |
| 03:21.18 | atan | I think I understand. So it's working just like it should be then. |
| 03:21.27 | dlynes | atan, yes |
| 03:21.42 | dlynes | atan, You're using asterisk 1.8 right? |
| 03:21.44 | atan | The call itself is not being transferred out though, instead it's hitting the voicemail |
| 03:22.03 | atan | I am indeed on 1.8 right now. Have been wanting to play with 10 b2 but I'll wait till I fix my bugs :D lol |
| 03:22.25 | dlynes | atan, Yeah...local channels if I remember correctly became more ubiquitous with 1.8 |
| 03:22.42 | dlynes | atan, On 1.6, I don't believe it would have used a Local channel to do that |
| 03:24.07 | SeRi|aFk | well I just went from Asterisk 1.8.4 to 1.8.7.1 |
| 03:24.08 | atan | Can I somehow define how a call is redirected? I see it does somehow hit the outbound context but it's missing a lot of data it needs to complete the call correctly. -- Called SIP/flowroute/9896543 is clearly invalid |
| 03:24.51 | atan | I want the redirected call to take on everything as if it was the SIP peer itself. So all the setvars, accountcode, and such |
| 03:26.10 | dlynes | atan, it could perhaps help if you could pastebin your sip.conf and extensions.conf in their entirety |
| 03:26.29 | dlynes | atan, and a full log of what's happening for the call |
| 03:27.13 | dlynes | atan, without seeing any of that, I'd just be making wild speculations |
| 03:27.58 | atan | dlynes, you bet! I'll just be one moment while I grab the parts of it that are in use :-) |
| 03:28.13 | dlynes | atan, 'entirety' :) |
| 03:28.32 | dlynes | atan, you can scrub any passwords or hostnames, though |
| 03:28.46 | p3nguin | 9896543 is apparently the number you entered. |
| 03:28.49 | dlynes | atan, and i would suggest you do |
| 03:29.11 | p3nguin | And I recommend losing the r Dial option. |
| 03:29.32 | p3nguin | And for calls going through another peer, I recommend ditching the timeout as well. |
| 03:29.34 | dlynes | p3nguin, yeah, but perhaps there's some additional translation he needs to do on the number before sending out through flowroute |
| 03:29.55 | dlynes | s/oh/on/ |
| 03:31.32 | p3nguin | If you don't want the call to go to 9896543, dial a different number. |
| 03:32.46 | dlynes | p3nguin, I'm just speculating, considering he said the dial string was clearly wrong |
| 03:32.59 | dlynes | p3nguin, and didn't elaborate |
| 03:33.39 | atan | http://pastie.org/private/zn2uymhvjyd034quslhva |
| 03:34.35 | atan | I'm not sure but I think things are going off around Got SIP response 302 "Moved Temporarily" back |
| 03:35.03 | atan | My SIP peers all have a var called prefix which contains the first part of their 7 digit number but it doesn't seem to come along with the call when it gets redirected |
| 03:35.30 | dlynes | atan, SIP 302, is because you hit transfer on your phone |
| 03:35.53 | atan | Seems to me it's treating the transfer as though it was a call dropped right into the "outbound" context without any of the extra fixins |
| 03:35.58 | dlynes | atan, or you had it set to call forward to somewhere else |
| 03:36.20 | atan | dlynes, is there any way to add in the bits I want the forwarded call to have? Like account code for billing and such |
| 03:36.22 | dlynes | atan, and 'outbound' is the context you've got defined for your bria |
| 03:36.33 | p3nguin | You either need to dial the entire 10-digit number, or make an accommodating extension in the outbound context. |
| 03:36.37 | dlynes | atan, do that in your outbound context |
| 03:37.01 | dlynes | atan, it follows the same rules as any other number you call from your bria |
| 03:37.11 | atan | But it's not |
| 03:37.18 | dlynes | atan, it is |
| 03:37.37 | dlynes | atan, -- Executing [9896543@outbound:1] Set("Local/9896543@outbound-6785;2", "SEVENEXTEN=9896543") in new stack |
| 03:37.38 | atan | When my bria calls out it hits this: exten => _NXXXXXX,1,Set(SEVENEXTEN=${prefix}${EXTEN}) |
| 03:37.54 | atan | But ${prefix} isn't getting set correctly |
| 03:38.04 | atan | Same deal with ${accountcode} for billing and all that |
| 03:38.08 | dlynes | atan, and where is prefix coming from? |
| 03:38.08 | p3nguin | The variable 'prefix' is not defined in the new channel. |
| 03:38.23 | atan | dlynes, it's set on the bria sip peer |
| 03:38.37 | dlynes | atan, remember how i told you to paste everything in its entirety? |
| 03:38.38 | p3nguin | It's on the original channel. It will not inherit unless you define it accordingly, by using one or two underscore(s). |
| 03:38.44 | dlynes | atan, prefix is part of that entirety |
| 03:38.44 | p3nguin | He pasted it. |
| 03:38.58 | p3nguin | Next to last line in the peer entry. |
| 03:39.02 | dlynes | ah...function of sip peers i've never used before |
| 03:39.03 | atan | p3nguin, ahhhh I need the underscores so the child channels get it! I see. |
| 03:39.12 | dlynes | p3nguin, is that new in 1.8? |
| 03:39.16 | p3nguin | no |
| 03:39.27 | dlynes | p3nguin, but newer than 1.4? |
| 03:39.29 | atan | One day you fellas will teach me :-) and I will be helping not seeking help all the time :-) |
| 03:39.33 | p3nguin | no |
| 03:39.44 | p3nguin | one underscore to inherit in one new channel; two underscores to inherit indefinitely. |
| 03:39.49 | dlynes | p3nguin, oh...I guess ya learn something new every day |
| 03:39.53 | p3nguin | _prefix or __prefix |
| 03:40.11 | atan | p3nguin, so I might use __prefix within my sip.conf? :-) |
| 03:40.14 | p3nguin | Set it like that, but refer to it normally. |
| 03:40.16 | p3nguin | Yes. |
| 03:40.33 | p3nguin | setvar=_prefix=1905 |
| 03:40.37 | atan | Can I do similar things to accountcode or is this limited to those within setvar? |
| 03:40.38 | dlynes | p3nguin, thought you could only do those channel vars in the dialplan |
| 03:40.44 | p3nguin | Then it will inherit into one new channel off the main channel. |
| 03:40.59 | p3nguin | It's only for variables,. |
| 03:41.01 | dlynes | atan, accountcode should come from your accountcode field in your sip.conf file |
| 03:41.37 | dlynes | atan, and i'm guessing that's coming through, based on this entry: -- Executing [9896543@outbound:2] MYSQL("Local/9896543@outbound-6785;2", "Connect connid redacted") in new stack |
| 03:42.08 | dlynes | atan, redacted is your accountcode |
| 03:42.15 | atan | dlynes, the accountcode for this SIP peer is "0" |
| 03:42.25 | atan | the redacted part was the MySQL database connection information |
| 03:42.36 | p3nguin | Had you not redacted that, we'd know. |
| 03:42.49 | p3nguin | There's a reason we ask for pastes to hide ONLY PASSWORDS. |
| 03:42.53 | dlynes | [1133] ; Bria 3 SoftPhone |
| 03:42.53 | dlynes | accountcode=redacted |
| 03:42.56 | p3nguin | Nothing else is "private." |
| 03:44.05 | dlynes | atan, no reason to redact the mysql stuff, either, unless for some strange reason you open up your mysql server to the internet |
| 03:47.05 | atan | I fear the unknown. |
| 03:47.58 | atan | So I added __ to my prefix in sip.conf but it's not picking it up |
| 03:48.05 | atan | That was two underscores, right? |
| 03:48.18 | atan | setvar = __prefix=1905 for example |
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| 03:50.27 | p3nguin | If you want inheritance to one new channel, use one underscore. If you want infinite inheritance, use two underscores. |
| 03:50.56 | atan | I feel like I must need to set this on the inbound context because all the other details being used are that of the inbound call. Nothing else exists to tell what SIP peer it's from |
| 03:51.19 | atan | Well, nothing that I notice right now anyway. |
| 03:58.46 | atan | https://issues.asterisk.org/jira/browse/ASTERISK-12817 |
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| 04:10.00 | atan | If a call is a blind transfer is there a way to detect and pull vars from the call that sent it? It's just landing there in the outbound context and setvar = __var isn't passing along to it |
| 04:10.26 | atan | Perhaps I can build some if(var is blank){ pull vars from blind transferer } somehow |
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| 04:42.14 | SeRi|aFk | p3nguin, you in? |
| 04:55.04 | kaldemar | ~ask |
| 04:55.04 | 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. |
| 04:56.28 | ChannelZ | atan: ${BLINDTRANSFER} will tell you the channel the transfer came from, not sure if you can use that to then snarf variables out of it. Their channel basically goes away once they hit transfer though |
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| 05:27.21 | MDesade | hello all. i changed OS's to CentOS today, per digium's recommendation. so, my question for you all is: what is the preferred method for installing on a CentOS-6.0-64bit system?? |
| 05:28.47 | WIMPy | Why could the distro make a difference? |
| 05:29.17 | WIMPy | Or rather how? |
| 05:32.25 | WIMPy | You can try to experiment with preemtion settings. Not sure what influence that might have. |
| 05:32.32 | kaldemar | MDesade: there is no single universally preferred method. many people on this channel prefer compiling from source. |
| 05:34.26 | WIMPy | If any possible try to make sure you don't have TDM and bttv on the same IRQ and then use smp_affinity. |
| 05:44.51 | lovetide | hi, is "Asteriskâ¢: The Definitive Guide" the 3rd edition of "Asterisk - The Future of Telephony" ? |
| 05:45.35 | kaldemar | kind of. it is written for 1.8. |
| 05:50.17 | p3nguin | mdesade: Grab the AsteriskNOW ISO image and install it using the "no GUI" option. Enjoy. |
| 05:50.24 | lovetide | kaldemar: thanks! this guide is so useful |
| 05:51.51 | lovetide | MDesade: personally, i prefer the yum install method ( http://www.asterisk.org/downloads/yum ), but there're no build for CentOS-6 yet. |
| 05:52.46 | p3nguin | If you follow my advice, you'll have a working system in 15 minutes or less. |
| 05:57.25 | p3nguin | Or don't. I don't even care how difficult you want to make things on yourself. |
| 05:59.44 | kaldemar | it won't hurt if you actually learn to install asterisk on a system. |
| 06:05.26 | *** join/#asterisk oej (~olle@ANice-257-1-97-170.w92-143.abo.wanadoo.fr) |
| 06:06.32 | WIMPy | Do PRIs have an E channel? |
| 06:13.00 | lovetide | E channel? never heart about it: http://en.wikipedia.org/wiki/Primary_Rate_Interface |
| 06:13.47 | *** join/#asterisk irroot (~irroot@197.168.205.205) |
| 06:15.49 | WIMPy | There are more than just B and D channels. |
| 06:16.37 | WIMPy | The E channel is important for collision detection on ptmp, but there is no ptmp PRI. |
| 06:19.34 | kaldemar | there is the 0 channel for synchronization and some operator functions. |
| 06:25.47 | irroot | no the E channel is not a nick name for a rave era distributor |
| 06:25.58 | WIMPy | :-) |
| 06:28.21 | irroot | the more i look at app_queue the more i want to bang head |
| 06:33.59 | irroot | 10 points for guessing what update_status in app_queue does |
| 06:38.54 | wdoekes2 | haha, I think I've had a similar experience |
| 06:40.13 | lovetide | are you reading the code ? |
| 06:46.13 | lovetide | what knowledges i need to learn to modify asterisk code? |
| 06:47.11 | irroot | lovetide how to use vi ... :P modifying it is easy its getting a usful modification that is tricky |
| 06:48.33 | irroot | wdoekes2 the state stuff in app_queue is sick im busy hacking it into shape |
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| 06:49.27 | irroot | lovetide can follow the reviewboard and see #asterisk-dev |
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| 06:53.01 | lovetide | irroot: thanks, i know vi a little. i just want to modify the "say.c" to fit chinese users's convention (we don't say 10 thousands, we say 1 wà n) |
| 06:53.31 | irroot | ah that will be useful |
| 06:53.32 | lovetide | irroot: thanks, i'll go ask in #askterisk-dev channel |
| 06:53.53 | irroot | -dev is dead on weekend so during week best |
| 06:54.12 | irroot | here in africa wit the currency been in thousands |
| 06:54.12 | lovetide | (O_O) |
| 06:54.20 | irroot | with |
| 06:55.06 | irroot | in zambia 1000K is 1pin so if someone says 5pin thats 5000 it cameabout when mony was "pinned" together in 1000's <- useless fact |
| 06:55.39 | irroot | so its common now to say pin instead of thousand for almost anything there |
| 06:57.08 | irroot | lovetide http://www.asterisk.org/doxygen/trunk/ also the wiki |
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| 07:00.19 | lovetide | hmmm, doxgen page is forbidden |
| 07:05.18 | irroot | great firewall ?? |
| 07:05.32 | irroot | ah wait |
| 07:05.50 | irroot | http://www.asterisk.org/doxygen/trunk/main.html |
| 07:06.56 | irroot | lovetide ping ^^ |
| 07:07.37 | lovetide | lol, not the great firewall. |
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| 08:10.59 | joobie | hey guys.. anyone know if the polycom vvx can use the video on skype? |
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| 10:15.25 | puzzled | hi |
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| 13:38.19 | dandate2 | some hackers/competitor/disgruntled customer is calling one of my DID with 20 phones consecutive. how can i see which DID they are calling exactly? |
| 13:43.48 | asilva | dandate2: like your phones rang at the same time?! |
| 13:44.10 | dandate2 | well we have a queue and its stacked with anonymous callers |
| 13:44.21 | dandate2 | that seem to be from an automated dialer |
| 13:44.37 | dandate2 | good thing we dont pay per minute! |
| 13:45.17 | asilva | in some of my servers yesterday, all phones rang at the same time... |
| 13:45.34 | dandate2 | mabye you have ringall enabled lol |
| 13:45.42 | asilva | i'm supposing to be an aattempt of attack? Cause there's a similiarity to the apache log servers |
| 13:45.56 | asilva | http://pastebin.com/n411jwry |
| 13:46.07 | dandate2 | i've been running a pbx for a few years and we get strange shit all the time |
| 13:46.33 | dandate2 | before these 20 consecitve phone calls it was a bunch of sip ip calls |
| 13:47.28 | asilva | strange? i have approxm (34 servers) over 5k endpoints sip, and this particulary event seemed to be an attack!! I made some upates and firewall changes let see if happens again!! |
| 13:48.27 | asilva | well, i'm off to lunch :) tomorrow i'll be boarding towards denverr for astricon!! :D |
| 13:48.39 | dandate2 | lunch |
| 13:48.46 | dandate2 | wtf are u from russia or something |
| 13:48.58 | asilva | brazil |
| 13:49.01 | asilva | ;) |
| 13:49.10 | dandate2 | ;) |
| 13:49.17 | asilva | where u from!? |
| 13:50.04 | puzzled | asilva: in my experience anything coming from a cheap (free) amazon vm is usually bad news. try blocking their IP addresses in your firewall |
| 13:51.11 | puzzled | amazon is 50.16.0.0/14 |
| 13:51.17 | asilva | puzzled: sure?. did it :) but it seem to be using the last know DoS apache attack, but might be an 0day or something, my apache user has perm to execute asterisk to perform sip reloads and dialplan reloads, cause of the weg mgnt interface. so there might be something related! |
| 13:51.18 | puzzled | at least EC2 is |
| 13:51.47 | puzzled | yup that would be one way. and next is toll-fraud so keep an eye on things |
| 13:53.41 | puzzled | dandate2: is't lunchtime for me too but I'm not even near Brasil. The world is a big place you know with lots of timezones :) |
| 13:54.26 | dandate2 | its only 10am in brazil. how are you going to eat lunh |
| 13:54.40 | puzzled | it's nearly 4pm here |
| 13:55.09 | puzzled | and didn't Brasil have different timezones too? |
| 13:55.09 | dandate2 | naw isnt it the same as eastern standard time USA |
| 13:55.15 | asilva | we are in DST |
| 13:55.20 | asilva | so its 11:55 |
| 13:55.23 | dandate2 | oh your not asilva lol |
| 13:55.27 | dandate2 | 4pm there... |
| 13:55.33 | dandate2 | you must be in ukraine or something |
| 13:55.37 | asilva | l0l |
| 13:55.42 | puzzled | sigh |
| 13:56.00 | puzzled | asilva: I once visited Pipa. lovely place |
| 13:56.03 | asilva | dandate2: wil you be at astricon? |
| 13:56.10 | asilva | puzzled: Pipa ? |
| 13:56.16 | puzzled | near Natal |
| 13:56.21 | dandate2 | actually im stuck here in the philippines with an expired passport |
| 13:56.32 | dandate2 | midas well run a call center |
| 13:56.34 | asilva | oh i see |
| 13:56.38 | asilva | Rio Grande do norte |
| 13:56.43 | puzzled | yup |
| 13:56.46 | asilva | niceee |
| 13:57.00 | asilva | puzzled: will you be at astricon? |
| 13:57.18 | dandate2 | nevermind astricon. asterisk 1.4 is all you ever need |
| 13:57.24 | dandate2 | dont need no asterisk 1.8 |
| 13:57.30 | puzzled | asilva: nope no time. got deadlines |
| 13:57.39 | asilva | :( |
| 13:57.50 | asilva | 1.4 dont hae ipv6 ;) |
| 13:57.58 | puzzled | well looking at the differences between 1.4 and 1.8 it makes sense to move to 1.8 (or 1.10) |
| 13:58.00 | dandate2 | ipv6 is fraud |
| 13:58.09 | asilva | lol |
| 13:58.14 | puzzled | TLS/SRTP to start with |
| 13:58.17 | asilva | leets see you say that in next 3 to 4 years |
| 13:58.25 | asilva | theres that too ;) TLS SRTP |
| 13:58.28 | asilva | well, i'm leavingg now.. bye!! :D |
| 13:58.31 | dandate2 | yeah i know the differences "look nice" but for proprietary purposes 1.4 does fine |
| 13:58.31 | puzzled | bye |
| 13:58.58 | dandate2 | once you get into these newest upgrades you lose money |
| 13:58.58 | puzzled | if you don't bump into the bugs 1.4 has then it makes sense to use |
| 13:59.41 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 13:59.57 | dandate2 | i use 1.4 and the only problem i have is no chanspy with whisper and a non-functioning attended transfer |
| 14:00.27 | puzzled | if you can live with that and the ret works then good for you |
| 14:00.31 | puzzled | rest |
| 14:00.41 | dandate2 | what more could people need/ |
| 14:00.55 | puzzled | TLS/SRTP, IPv6 and GTalk :) |
| 14:01.08 | dandate2 | whats TLS/SRTP stand for |
| 14:01.23 | puzzled | it's like httpS, the secure one. so encryption |
| 14:01.36 | puzzled | and google is your friend :) |
| 14:01.42 | dandate2 | encryption for what? |
| 14:01.49 | puzzled | the voice link |
| 14:01.58 | dandate2 | so people dont listen in? |
| 14:02.00 | dandate2 | hackers i mean |
| 14:02.03 | puzzled | yup |
| 14:02.08 | dandate2 | let them |
| 14:02.10 | puzzled | and the NSA, etc |
| 14:02.19 | dandate2 | we sound damn good over here |
| 14:02.29 | dandate2 | can't hate if someone wants to listen |
| 14:03.00 | dandate2 | on the otherhand |
| 14:03.21 | dandate2 | if someone can bear to listen through hours of g729 to cell phone conversations, let there be fraud |
| 14:04.51 | dandate2 | if someones using a cell phone speaking to an agent on g729 the sound quality is so bad they simply should not be exposing sensitive details at all |
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| 14:47.30 | leifmadsen | puzzled: s/1.10/10/ |
| 14:48.04 | puzzled | hi leif. yes you are right :) |
| 14:48.08 | leifmadsen | :) |
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| 15:19.54 | irroot | leifmadsen yo there ltns |
| 15:20.06 | leifmadsen | irroot: :) |
| 15:24.13 | lovetide | hmmm, SayNumber only accecpt int number? can say float/double number? |
| 15:47.17 | leifmadsen | lovetide: what version of Asterisk? |
| 15:47.49 | lovetide | leifmadsen: 1.8.7.1 |
| 15:48.05 | leifmadsen | huh thought they all supported floats now, guess not |
| 15:49.01 | lovetide | the parameter in source code is "int num" |
| 15:49.15 | leifmadsen | there you go :) |
| 15:50.00 | [TK]D-Fender | half a dozen line of dialplan to do this |
| 15:50.19 | [TK]D-Fender | (as a nice macro) |
| 15:50.42 | leifmadsen | that'd be a good recipe actually |
| 15:50.46 | leifmadsen | thanks for the idea |
| 15:50.53 | leifmadsen | (I'll forget it in like 10 seconds though) |
| 15:51.29 | lovetide | lol |
| 15:51.34 | leifmadsen | Actually that's really easy to do with CUT() and SayNumber() heh |
| 15:51.39 | leifmadsen | I just thought of how to write it in my head |
| 15:51.42 | *** join/#asterisk navaismo (~Airo@201.123.42.167) |
| 15:51.58 | leifmadsen | maybe a blog post on how to do it will inspire me to work on my astricon presentations |
| 15:52.12 | [TK]D-Fender | a GotoIf and point.ulaw |
| 15:52.36 | [TK]D-Fender | This is really so 101... not writing it alone should be inspiration .... |
| 15:52.55 | [TK]D-Fender | Gotta give budding programmers a soft-ball to run with |
| 15:52.56 | navaismo | Good morning all, i have a quick question, in the unicall.conf is there a way to set the option national_suscriber? |
| 16:04.10 | [TK]D-Fender | navaismo: Quick statistical thought for your issue. Almost the only people that use UNISTIM are business users who desperately had to hold only Nortel leftovers. We're on a weekend the actual nuber of UNISTIM users if incredibly small as it is let alone those who come here. I suspect you'll get an answer quicker if you put this out on the mailing lists. |
| 16:08.06 | navaismo | potassium |
| 16:08.52 | navaismo | ill try this http://lists.digium.com/pipermail/asterisk-dev/2007-July/028430.html thanks anyway |
| 16:09.50 | [TK]D-Fender | no... Nortel is a lemon.. that makes it citrus, not potassium. That'd be for potatoes, and Nortel is nowhere near anything "hearty":p |
| 16:10.04 | navaismo | lol |
| 16:10.46 | navaismo | time to got have a great weekend! |
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| 17:10.16 | MDesade | good morning all... |
| 17:11.15 | MDesade | i sorta asked last night, but i have more time this morning to ask beter questions right now: ok, does anyone have a link to a site with good install instructions for asterisk on CentOS-6.0-64bit? |
| 17:12.22 | MDesade | (i was using Kubuntu, but after some time with tech support at Digium on the phone this week, their recommendation was/is use CentOS, as supposedly it has better "realtime" resource management) |
| 17:13.32 | WIMPy | That doesn't make sense. |
| 17:13.46 | MDesade | so, i am curious also if any of you know what they meant by that (specifically) as in: is there some configuration files, or granularity that CentOS is offering, that Kubuntu doesn't? |
| 17:14.11 | WIMPy | That's kernel stuff. |
| 17:14.15 | MDesade | WIMPy - i thats why i am asking |
| 17:14.40 | MDesade | seemed like a broad statement for them to say, without really saying WHY |
| 17:15.19 | WIMPy | IIRC the only issue you have is with IRQs. |
| 17:15.52 | WIMPy | So things like cgroups or anything shouldn't even matter. |
| 17:16.02 | MDesade | oh! another thing i thought that was interesting that they said is: "it is preferred to NOT have a GUI like KDE installed |
| 17:16.49 | WIMPy | The X server can use quite some resources, yes. |
| 17:16.50 | MDesade | which? i am ok with, once it is up, running, and configured properly, but? having a GUI makes that configuration process a LOT easier, right? |
| 17:17.30 | WIMPy | I find it much easier without. |
| 17:18.11 | MDesade | to me, it would make more sense to have a runlevel setup to boot into KDE, config what i have to config, web browse, all that... then make the default runlevel CLI only, but should i have to go back? just init the gui runlevel |
| 17:19.03 | WIMPy | I never install X on a server. That's just not appropriate. |
| 17:19.18 | MDesade | WIMPy - yes, the issue that they were saying ABOUT KDE was the resources, that the PCIe graphics card uses... well, ok? so i can use the onboard graphics, but? |
| 17:20.07 | WIMPy | And don't use frambuffers. |
| 17:20.35 | MDesade | luckily, i spent enough years using slackware in the 90s that i am OK with CLI only, but? like i said? editing files, writing bash scripts IS easier in eclipse or emacs, than say? nano, or VI/VIM |
| 17:21.12 | WIMPy | Since when do you need X for emacs? |
| 17:21.28 | MDesade | you don't, just a preferance |
| 17:21.58 | WIMPy | Edit your files from another box. |
| 17:22.14 | MDesade | IE. another thing i like about KDE is called "apt-url" so, you are browsing in mozilla, find your package (in kubuntu) click on it, and bam... its installed |
| 17:22.34 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:22.42 | MDesade | WIMPy - ironically, that was my plan... edit stuff, and just copy them over the network |
| 17:23.09 | MDesade | FYI - when i installed CentOS last night, i didn't install a GUI |
| 17:23.27 | WIMPy | I do most stuff via NFS, even on my home box. |
| 17:24.00 | MDesade | did a rather minimal install, only adding packages that i know asterisk is gonna want IE. mysql, apache2 (for freepbx) and kept it at that |
| 17:24.07 | *** join/#asterisk wonderworld (~ww@port-92-201-98-136.dynamic.qsc.de) |
| 17:25.31 | WIMPy | In your situation I'd try to keep everythign as simple as possible, i.e. no mysql or apache. |
| 17:25.35 | wonderworld | hi, i am trying to test asterisk 10 ConfBridge. When using it in the dialplan, ConfBridge exits instantly and no error message is given. How could be wrong? |
| 17:26.43 | WIMPy | wonderworld: Not yet, but that seems to happen to me as well. |
| 17:27.09 | WIMPy | Hmm. New syntax? |
| 17:27.46 | wonderworld | i used the syntax from the docs. exten=>6666,n,ConfBridge(1234) |
| 17:28.22 | MDesade | WIMPy - i am keeping it simple, but apache2 is for both zoneminder and freepbx/FOP and mysql is used by both freepbx and zm as well |
| 17:29.02 | WIMPy | wonderworld: 'core show application confbridge' |
| 17:29.28 | MDesade | the heart of the matter is the IRQ steering issues with having so many PCI cards... so? if its gonna work at all? then mysql or apache2 isn't gonna be the problem |
| 17:30.00 | WIMPy | You could run into trouble with disk I/O as well. |
| 17:30.00 | *** join/#asterisk johaidi (~johaidi@94.30.149.10) |
| 17:30.05 | wonderworld | displays a template for packageinfo |
| 17:30.22 | MDesade | and if it DOESN'T work at all? well? worst case is building another box, ONLY for asterisk (which i know is preferred) but, i thought it would be really cool to have them together (if possible) |
| 17:30.40 | WIMPy | But as I said in the morning: Try to avoid havin both bttv and tdm cards on the same IRQ if any possible. |
| 17:30.53 | WIMPy | Then use smp affinity. |
| 17:30.56 | MDesade | WIMPy - of course |
| 17:31.15 | wonderworld | cards just cause trouble |
| 17:31.26 | MDesade | back in the day, you could set some of that in the BIOS of the motherboard, but that doesn't exist anymore |
| 17:31.36 | WIMPy | Ideally tdm cards should have an irq of their own, each. |
| 17:31.40 | MDesade | right |
| 17:32.12 | WIMPy | But if you can get the tdm cards on one irq and te bttv on another that might help, depending on how good the drivers are. |
| 17:32.45 | *** join/#asterisk Fritz09 (~Adium@pop1-2199.catv.wtnet.de) |
| 17:32.46 | MDesade | i even asked digium, if IE. would it be ideal, if i were to but the 24x port card, and just populate only half of it, but the point being is: only 1x card, not two |
| 17:32.53 | MDesade | and they didn't that that would matter |
| 17:33.04 | WIMPy | But having both kinds on the same irq ist most probably the show stopper. |
| 17:33.48 | MDesade | right, i am aware that the BTTV cards AND the TDM cards fighting for IRQs is the show stopper |
| 17:33.57 | WIMPy | If you can get one card with an unshared riq, yo should be finde with that, I guess. |
| 17:34.14 | wonderworld | i had so much trouble with pci hardware and euro E1s. i stopped using them completely. |
| 17:34.24 | WIMPy | Not for irqs, for the SAME irq. |
| 17:34.38 | MDesade | WIMPy - i get it |
| 17:35.38 | WIMPy | Yes, I know the issue as well. My test box is "only" a dual PIII. That didn't like more than 4 E1. |
| 17:35.53 | MDesade | WIMPy - ill give this a try, and? like i said? worst case? i have to build another box |
| 17:35.58 | *** join/#asterisk Sorcier_FXK (~Sorcier_F@unaffiliated/sorcierfxk) |
| 17:36.00 | WIMPy | But I'm not sure if that's a general issue or just a dahdi one. |
| 17:36.51 | MDesade | other than the money and power? it's not a dealbreaker to have a separate box... its just disappointing, considering half the point of what i wanted to accomplish was combining the two in one box |
| 17:37.41 | WIMPy | I have heard that Sangoma is less picky with irqs, but I don't know if there's a real difference. |
| 17:38.11 | MDesade | no matter what? the solution, ill be running zoneminder (cause its awesome with the cameras) and ill be running asterisk (cause its awesome for phones, intercoms, and my business) |
| 17:38.34 | MDesade | WIMPy - a few people have said that recently about my issue as well |
| 17:39.01 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:41.40 | WIMPy | Or as I said yesterday: Go pci-e. |
| 17:42.25 | wonderworld | if you want to invest, get a media gateway. stress free. |
| 17:43.31 | WIMPy | In that respect, yes. They may have other issues. |
| 17:43.42 | MDesade | WIMPy - that was one of my questions to digium, about just getting a TDM2400 and using 2x FXO (4-port) modules, and 1x (4-port) FXS in a PCIe card |
| 17:43.50 | WIMPy | Just like you could use IP cameras. |
| 17:44.31 | WIMPy | pci-e is ptp, so you always have an irq on your own. |
| 17:44.36 | MDesade | well, just like using a bunch of ATAs, that costs MORE money than a separate box, replacing the dozen NTSC cameras i already have installed |
| 17:46.35 | MDesade | its cheaper to build in a new box a core2quad or core i7 for 500-600 bucks, and populate it with my cards, then to replace a dozen cams at 150 each? |
| 17:46.49 | MDesade | or 8x ATAs for phones, at 100 ea |
| 17:47.36 | WIMPy | You should get away with a much smaller box. |
| 17:48.28 | MDesade | so, the solution is either fix the IRQs via either what i am starting this weekend (new OS, flashed mobo, disabled everything not in use (serial, parallel, USB) OR a PCIe card |
| 17:48.33 | MDesade | OR build another box |
| 17:49.02 | ChannelZ | or find a new career/hobby (as the case may be) |
| 17:49.16 | WIMPy | Changing distro won't make a difference. |
| 17:49.31 | WIMPy | ChannelZ: :-) |
| 17:50.33 | WIMPy | Find out which card produces which irq in which slot. And if you have mapped that out, you know what to do. |
| 17:51.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:52.50 | wonderworld | i want to test asterisk 10 ConfBridge. when using it in the dialplan, ConfBridge exits instantly and no error message is given. what might be wrong? |
| 17:53.21 | p3nguin | Pastebin the configuration. |
| 17:53.57 | MDesade | WIMPy - totally agree, but? i figured i'd just try it to see what they were talking about, since my OS got hosed after flashing the mobo anyway? |
| 17:54.39 | p3nguin | You aren't trying to make a link between flashing the BIOS and hosing an OS, are you? |
| 17:55.05 | wonderworld | p3nguin: standard dist file. |
| 17:55.21 | p3nguin | I don't know what that means. Pastebin your configurations. |
| 17:56.35 | irroot | who is brave and wants to try a app_queue with devstate stuff redon |
| 17:56.42 | irroot | s/redon/redone/ |
| 17:57.06 | *** join/#asterisk Fritz09 (~Adium@pop1-2199.catv.wtnet.de) |
| 17:57.49 | MDesade | WIMPy - i prefer Kubuntu, and all the debian way of stuff... however? like i said? since i have the opportunity after flashing the mobo? what the hell? give it a whirl? and if i don't like it? well? nothing lost |
| 17:58.44 | MDesade | once again, my question WAS: is there a good site, that outlines the install instructions for asterisk/freepbx/FOP specific to CentOS??? |
| 18:00.01 | p3nguin | Short FOP, I gave you the easy solution for Asterisk on CentOS yesterday. |
| 18:00.26 | p3nguin | If you want FreePBX, choose FreePBX during install rather than the no GUI option I suggested. |
| 18:00.37 | wonderworld | p3nguin: http://pastebin.com/beGy7YnL |
| 18:01.05 | WIMPy | Yes, that shouold be the most straight forward way. |
| 18:01.08 | MDesade | p3nguin - the AsteriskNOW sidea? |
| 18:01.11 | MDesade | idea? |
| 18:02.02 | p3nguin | That's right. AsteriskNOW will give you CentOS with Asterisk, and an optional FreePBX if you insist on using it. |
| 18:02.04 | MDesade | yeah, i was thinking about that. and (im not sure) but, trixbox is based on CentOS, as well, and ALSO is just a drop the DVD in and off ya go.... |
| 18:02.30 | MDesade | is "AsteriskNOW" related to or affiliated with Trixbox?? |
| 18:02.50 | p3nguin | Except no one here can support Trixbox, but we can support Asterisk, and the FreePBX channel or the AsteriskNOW channel can help with the FreePBX part. |
| 18:02.52 | WIMPy | ~trixbox |
| 18:02.53 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 18:03.07 | MDesade | gotcha... |
| 18:03.32 | p3nguin | There is no relation between Trixbox and AsteriskNOW... other than Asterisk is in them both (kind of). |
| 18:03.32 | *** part/#asterisk Fritz09 (~Adium@pop1-2199.catv.wtnet.de) |
| 18:03.33 | MDesade | i thought they were affiliated... i was NOT going to go the trixbox route... years ago? they were open |
| 18:03.33 | WIMPy | Not that FreePBX per se is much better concerning the support aspect. |
| 18:03.45 | MDesade | yeah, they kinda suck |
| 18:04.21 | MDesade | what distro is AsteriskNOW based on anyway?? |
| 18:04.31 | p3nguin | <p3nguin> That's right. AsteriskNOW will give you CentOS with Asterisk, and an optional FreePBX if you insist on using it. |
| 18:05.10 | MDesade | p3nguin - ok, i missed that yesterday, but scrolled up and saw it (obviously, since we are talking about it>?) |
| 18:05.32 | p3nguin | I said it three minutes ago. |
| 18:05.46 | MDesade | p3nguin - cool, ill give that a whirl, since its the easiest to isolate my problem with a known good working distro |
| 18:06.10 | wonderworld | p3nguin: i fixed my problem, never mind. |
| 18:06.28 | p3nguin | Don't leave me hanging! |
| 18:06.29 | MDesade | i can give or take FOP... easy enough to install, or not.... |
| 18:06.52 | MDesade | i don't care much for anything flash |
| 18:07.38 | p3nguin | I don't see the need for any crap like that anyway. |
| 18:08.08 | MDesade | its useful for like one thing "at a glance, is the phone off the hook" and thats about it |
| 18:08.15 | WIMPy | Well, it might be nice if it worked. |
| 18:08.26 | p3nguin | I can do that without any "panel." |
| 18:08.56 | p3nguin | In the amount of time you have to load the page in your browser and click around to find it, I can type the necessary commands on the asterisk cli. |
| 18:09.18 | MDesade | its just the presentation of the info.. but, again? FOP is not a dealbreaker, its more annoying than helpful |
| 18:09.48 | WIMPy | Get a phone with enough buttons or a side car. |
| 18:10.31 | WIMPy | ... for the channels where Asterisk supports it. |
| 18:10.32 | MDesade | well, thank you guys... |
| 18:11.00 | MDesade | i gotta get dressed and hit this place called "makerbench" that is offering a class on monday about building your own CNC |
| 18:11.19 | MDesade | ill start the download now of asteriskNOW, and load it up when i get home |
| 18:12.36 | WIMPy | is sure we will read about that |
| 18:15.56 | *** join/#asterisk neurosys (~neurosys@c-174-48-142-160.hsd1.fl.comcast.net) |
| 18:20.46 | MDesade | WIMPy - uh? thanks for the left-handed commentary? |
| 18:21.13 | WIMPy | writes with both hands |
| 18:21.28 | WIMPy | But you use quite a lot of question marks. |
| 18:23.23 | MDesade | we talked about this... i use them for inflection |
| 18:23.40 | *** join/#asterisk timahvo1 (~rogue@41.80.1.205) |
| 18:31.24 | p3nguin | I thought question marks were supposed to be used when asking a question. |
| 18:31.59 | p3nguin | So you were asking him for thanks for the left-handed commentary, I guess. |
| 18:45.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:53.51 | *** join/#asterisk gxdssoft (~gxdssoft@190.236.111.135) |
| 19:11.30 | *** join/#asterisk ajkaanbal (~ajkaanbal@201.102.57.14) |
| 19:43.19 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 19:43.44 | dijib | guys whats an authenticate() commandds password file format look like? i cant find an exMPLE |
| 19:46.27 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 19:47.19 | p3nguin | 762:6ea9bc9447688102f56e819d2ab3ddd2 |
| 19:47.39 | p3nguin | <account code>:<md5 hashed password> |
| 19:50.36 | [TK]D-Fender | dijib: a number |
| 19:50.59 | [TK]D-Fender | 1 per line for every valid answer |
| 19:51.05 | [TK]D-Fender | Which is kinda worthless |
| 19:51.08 | dijib | so how to i make the hash? |
| 19:51.18 | [TK]D-Fender | dijib: There is no "hash" |
| 19:51.23 | p3nguin | echo -n 1234|md5sum |
| 19:51.24 | [TK]D-Fender | digits dumb text |
| 19:51.27 | p3nguin | where 1234 is the PIN |
| 19:51.59 | dijib | im hoping i can get it to based on the pin entered determin what the email address of that user is like voicemail does |
| 19:52.27 | atan | Is it possible to force a command to run just before a blind transfer? |
| 19:52.31 | [TK]D-Fender | huh? |
| 19:52.31 | dijib | trying to have incomming calls, set CID for the user themselves, record & callout, then based on the pin email the correct user |
| 19:52.58 | [TK]D-Fender | dijib: If you want to auth a user then VMAuthenticate is a better choice if they have a VM box |
| 19:53.19 | [TK]D-Fender | atan: not without seriously nasty coding |
| 19:53.41 | dijib | ok then. so i can use the authentication the same eh? |
| 19:53.58 | [TK]D-Fender | dijib: Please reword that |
| 19:54.40 | dijib | vmauthenticate? |
| 19:55.44 | [TK]D-Fender | <PROTECTED> |
| 19:55.59 | [TK]D-Fender | Please rephrase your complete question |
| 19:57.50 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 20:12.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:15.55 | dijib | how do i make an entry for ${DB(email/fax-manager)} |
| 20:16.23 | p3nguin | database add <family> <key> <value> |
| 20:16.49 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 20:16.49 | p3nguin | wait |
| 20:16.56 | p3nguin | Maybe it's put, not add. |
| 20:17.09 | dijib | yes put and get |
| 20:17.13 | p3nguin | Either way, database TABKEY and figure it out yourself. |
| 20:17.18 | BlackBishop | anyone can tell me what I'm doing wrong ? http://pastebin.com/U82zDL75 .. trying to auth using realtime ( mysql backend ) |
| 20:17.52 | BlackBishop | the asterisk is in the same network I am ( nated for outside ) .. 1.8.4.4 ( debian wheezy ) |
| 20:19.26 | dijib | so far this is what i have on my plan. |
| 20:19.28 | dijib | http://pastebin.com/C18Rc0Pq |
| 20:19.58 | dijib | can i use the DB to authenticate, and then know the email address? |
| 20:21.18 | p3nguin | Why do you insist on using SayAlpha() for saying digits when there is a perfectly good SayDigits() app? |
| 20:22.42 | [TK]D-Fender | dijib: No, there is no way to get the box VM address direct in the dialplan. You will have to use some external script |
| 20:23.35 | p3nguin | I also don't see why you first record a file using the full path, and then set a new variable to the full path of that file. There's a perfectly good variable already in existence after you record a file. |
| 20:23.49 | dijib | or can i do an authenticate against the DB and the DB also have the associated email address? |
| 20:24.01 | [TK]D-Fender | dijib: No, there is no way to get the box VM address direct in the dialplan. You will have to use some external script <---------- |
| 20:24.36 | p3nguin | And after the Dial() is finished, the call will die, so you'll never send the email that you're trying to send. |
| 20:24.51 | dijib | then hows that done? |
| 20:24.56 | dijib | external agi? |
| 20:25.13 | [TK]D-Fender | dijib: Howver you want outside of * dialplan |
| 20:25.15 | p3nguin | You should have paid attention the other half dozen times I wrote dial plan for you. |
| 20:25.45 | dijib | did you guys do that conference line thing last night again? |
| 20:26.05 | p3nguin | I didn't. |
| 20:28.01 | dijib | p3nguin, your talking about that fax-in context you wrote right. |
| 20:28.02 | dijib | ? |
| 20:28.59 | p3nguin | I've given you hundreds of lines of dial plan. |
| 20:29.34 | p3nguin | voice, fax, fax to email, etc. |
| 20:29.45 | BlackBishop | http://pastebin.com/PnrP7xHR .. more debug .. ( if anyone cares ) |
| 20:34.22 | dijib | yeah but i killed the fax to email with the system rebuild |
| 20:34.35 | dijib | and i havent been able to get res_digium_fax back in |
| 20:34.45 | dijib | nor did i replace the DB lines we added |
| 20:36.00 | dijib | and p3nguin what was the perfectly good variable in that i could have used? |
| 20:38.19 | *** join/#asterisk god-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 20:39.41 | p3nguin | ${MIXMONITOR_FILENAME} |
| 20:40.35 | p3nguin | MixMonitor(filename.WAV,ab) |
| 20:40.39 | p3nguin | Verbose(${MIXMONITOR_FILENAME}) |
| 20:40.41 | p3nguin | You'll see. |
| 20:41.13 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 20:53.48 | dijib | how do i initiate the email then? or the external script? |
| 20:55.04 | BlackBishop | this is weird. |
| 20:55.23 | p3nguin | Emailing is as easy as what I provided before. |
| 21:03.41 | BlackBishop | any more debug output as of why would the auth fail ? :| |
| 21:16.09 | *** join/#asterisk gxdssoft (~gxdssoft@190.236.111.135) |
| 21:29.23 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 21:34.22 | dijib | where do i put agi scripts? |
| 21:34.51 | p3nguin | Typically you would put them in asterisk's agi-bin directory. |
| 21:35.13 | dijib | which is where? |
| 21:35.54 | p3nguin | under asterisk's home directory, obviously. |
| 21:36.46 | pabelanger | dijib: asterisk.conf defines the path |
| 21:38.47 | p3nguin | getent passwd asterisk would show the homedir, too. That could be a clue. |
| 21:39.36 | p3nguin | but grep agi /etc/asterisk/asterisk.conf should be definitive. |
| 22:01.15 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:18.52 | *** join/#asterisk DanFromUK (~DanFromUK@2.27.7.180) |
| 22:20.41 | DanFromUK | hi all. is there any way to SET a variable in a macro and then access the variable back in the dial plan? |
| 22:20.50 | DanFromUK | Ive tried __ but that doesnt seem to work. |
| 22:26.35 | p3nguin | What happens when the macro ends? My guess is that the call ends and the channel dies. |
| 22:27.30 | DanFromUK | the macro is used just before a call is connected. |
| 22:27.40 | DanFromUK | its a screening macro |
| 22:27.44 | p3nguin | You have two options. 1: Don't use Macro(), but instead use Gosub(); or 2: Set a global variable in the macro. |
| 22:28.00 | DanFromUK | up till now, the operator can press 1 to answer the call, or 2 to reject and send to voicemail |
| 22:28.15 | DanFromUK | i want to a 3 to forward the call to a different number |
| 22:30.01 | DanFromUK | As far as i'm aware, a macro is the only option. |
| 22:31.55 | p3nguin | Can you show me the dial plan? |
| 22:32.44 | DanFromUK | sure. 1sec. i'll pb it |
| 22:37.06 | DanFromUK | http://pastebin.com/XuxZwnRC |
| 22:39.59 | p3nguin | So your phone rings and it plays the "call from" thing to you? |
| 22:40.25 | DanFromUK | yep. currently it says "press 1 to answer the call, press 2 to transfer to voicemail" |
| 22:43.02 | p3nguin | And you want SCREENRESPONSE to be carried back to the other part of the dialplan. |
| 22:43.18 | DanFromUK | Correct, so that i can find out whats been pressed. |
| 22:45.03 | p3nguin | I'm surprised setting it normally doesn't work. |
| 22:45.18 | DanFromUK | the variable just blank |
| 22:45.26 | DanFromUK | the variable is just blank |
| 22:45.26 | p3nguin | I guess you'll have to set it as a global variable or enter it into the astdb. |
| 22:45.33 | DanFromUK | just going to test with a noop. |
| 22:45.36 | DanFromUK | 1sec |
| 22:47.02 | DanFromUK | NoOp(${SCREENRESPONSE}) returned nothing |
| 22:48.22 | *** join/#asterisk _N1x (~n1x@95.104.13.145) |
| 22:49.49 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 22:50.38 | _N1x | hi guys, i have installed mysql on my machine , anyone's can show me simple mysql query string ? when i press to key 5 playing any sound file from mysql db |
| 22:51.11 | _N1x | i already have connected to db my asterisk |
| 22:52.56 | DanFromUK | p3nguin: do you read a global variable in the same way as a local one? |
| 22:53.22 | p3nguin | Yes. The only difference is how you set it. |
| 22:53.38 | _N1x | p3nguin: i think u can help me :P |
| 22:53.45 | DanFromUK | thanks |
| 22:53.52 | p3nguin | I don't help people who can't spell "you." |
| 22:54.14 | _N1x | p3nguin: okey you :) |
| 22:55.18 | dijib | i dont understand how i can call an agi script and have it do what i want after the dial ends |
| 23:01.04 | p3nguin | When the call is deal, you should probably use DeadAGI(). |
| 23:01.09 | p3nguin | s/deal/dead/ |
| 23:01.56 | p3nguin | But if you're just trying to email a recording, use the dial plan that I wrote for you. |
| 23:05.16 | _N1x | anyone help me? |
| 23:10.50 | *** join/#asterisk The-Kernel (~thomas@unaffiliated/the-kernel) |
| 23:12.07 | The-Kernel | I'm not seeing the asterisk trying to register out to the SIP Server, I have added this line into the sip.conf file and when I restart asterisk there's no attempt....register => 5551212:5551212@test.net:5060/1000 |
| 23:13.31 | _N1x | sip show regisry |
| 23:13.52 | The-Kernel | Host dnsmgr Username Refresh State Reg.Time |
| 23:13.52 | The-Kernel | 0 SIP registrations. |
| 23:15.10 | _N1x | register => user:password@sip/100 |
| 23:15.28 | The-Kernel | that's basicallt what I have |
| 23:15.30 | The-Kernel | no port? |
| 23:15.34 | _N1x | nah |
| 23:15.37 | p3nguin | Did you put your register statement in the general section, before any peer definitions? |
| 23:15.48 | p3nguin | You only need to specify the port if it isn't 5060. |
| 23:15.50 | The-Kernel | let me check |
| 23:16.11 | p3nguin | It needs to be before any peers and before the authentication section if it exists. |
| 23:16.39 | p3nguin | And don't forget to run "sip reload" after saving any changes. |
| 23:16.54 | _N1x | p3nguin: "you" probably should help me :) |
| 23:17.56 | The-Kernel | that did iy |
| 23:18.00 | The-Kernel | it |
| 23:18.08 | The-Kernel | I had it last :/ |
| 23:18.10 | p3nguin | You had it in the wrong spot, or you forgot to sip reload? |
| 23:18.12 | p3nguin | Oh. |
| 23:18.16 | p3nguin | yeah, that's wrong. |
| 23:18.19 | p3nguin | Common mistake. |
| 23:19.03 | p3nguin | I think there should be a special section for registrations, which you could place anywhere. |
| 23:19.21 | The-Kernel | now I have a new problem |
| 23:19.27 | The-Kernel | the from is that of the phones extension.... |
| 23:19.39 | p3nguin | What does that mean? |
| 23:19.43 | _N1x | The-Kernel: i already know ur problem :D |
| 23:20.04 | WIMPy | Or replace your register statements by entries in your peer definitions. |
| 23:20.11 | WIMPy | (if it works for you) |
| 23:20.40 | p3nguin | That would also be acceptable, but I think a special [registrations] section would work well. |
| 23:21.24 | WIMPy | Who knows if simple registrations will be supported in the future. |
| 23:21.37 | p3nguin | I'm sure it will always be supported. |
| 23:21.57 | p3nguin | It's how we tell a peer how to reach us. |
| 23:21.58 | WIMPy | would not bet on that one. |
| 23:22.00 | _N1x | which gui is recommended on debian? |
| 23:22.14 | p3nguin | GTK+ |
| 23:22.29 | _N1x | p3nguin: i mean asterisk gui :) |
| 23:22.32 | The-Kernel | Wow...and in the invite the INVITE sip:551212@test.net SIP/2.0 it cut off the first 5!!! |
| 23:22.47 | p3nguin | GUIs for asterisk are not recommended. |
| 23:22.53 | The-Kernel | oh wait |
| 23:23.00 | The-Kernel | maybe if I dialed the numebr right |
| 23:23.15 | _N1x | p3nguin: i know , but i'm networking engineer and don't havy many times to learn everything |
| 23:23.22 | _N1x | :) |
| 23:23.37 | p3nguin | Then you're in the wrong industry. |
| 23:24.02 | _N1x | p3nguin: nah , i'm using unix and good have good knowledge why i "hate" guis |
| 23:24.03 | _N1x | ;) |
| 23:24.06 | [TK]D-Fender | The-Kernel: your register is in the wrong place if you see nothing at all |
| 23:24.08 | The-Kernel | nope its cutting off the first number of what I dial... |
| 23:24.37 | p3nguin | I'd have to see the sip debug. |
| 23:24.55 | _N1x | The-Kernel: i know ur problem |
| 23:26.06 | The-Kernel | here ya go -- http://pastebin.ca/2092654 |
| 23:26.06 | [TK]D-Fender | _N1x: * config GUI's have no advantage on one distro vs another. |
| 23:27.08 | _N1x | just do, exten => any,1,Answer() |
| 23:27.08 | _N1x | <PROTECTED> |
| 23:27.33 | _N1x | before ur calls goes to siptrunk |
| 23:28.43 | p3nguin | the-kernel: I'd say your dial plan is offsetting the extension. Pastebin your dial plan if you don't see where it's going wrong. |
| 23:29.23 | p3nguin | Probably a bogus ${EXTEN:1} for an unknown reason. |
| 23:31.51 | p3nguin | You usually want something like _NXXXXXX,1,Dial(SIP/peer/1408${EXTEN}) for 7-digit dial. |
| 23:32.24 | The-Kernel | like this? http://pastebin.ca/2092655 |
| 23:32.27 | p3nguin | or my preferred way, _NXXXXXX,1,Goto(1408${EXTEN},1) |
| 23:32.49 | p3nguin | line 19 |
| 23:33.07 | The-Kernel | aaahhhh |
| 23:33.09 | p3nguin | It is stripping the first digit, for no good reason that I can see. |
| 23:33.37 | p3nguin | Is this for NANP? |
| 23:33.51 | _N1x | The-Kernel: nah i'll send u correct con |
| 23:33.52 | _N1x | f |
| 23:33.59 | p3nguin | I'm not familiar with onvoip to know where you are. |
| 23:34.09 | [TK]D-Fender | no such thing as "any" unselle the invite is directly to the word "any" |
| 23:34.23 | [TK]D-Fender | unless* |
| 23:35.28 | _N1x | The-Kernel: http://pastebin.com/fzzhj2Zm |
| 23:35.54 | p3nguin | Don't Answer() before you Dial(). |
| 23:36.11 | p3nguin | the-kernel: Don't take advice from this guy. He can't even spell "you" or "your." |
| 23:36.32 | _N1x | p3nguin: it's not ur problem ;) |
| 23:36.58 | _N1x | i'm never sayng i know something and i'm GOD like u :d |
| 23:39.41 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 23:40.11 | dijib | p3nguin, what dialplan did u make that sent the recording by email? |
| 23:43.53 | The-Kernel | ALMOST THERE! but the from still has "1000" in it |
| 23:44.09 | The-Kernel | http://pastebin.com/jcEJcpSm |
| 23:44.43 | p3nguin | You're calling from a phone by the name of 1000, right? |
| 23:44.57 | The-Kernel | right and I just saw an ")" that should be there |
| 23:45.03 | The-Kernel | that might be it |
| 23:45.12 | _N1x | The-Kernel: what us "ur" number? |
| 23:46.24 | p3nguin | Yeah, what us you are number. :/ |
| 23:46.34 | The-Kernel | the phones number is 1000, the actual number that the PBX uses to register with and call out with is 483517293...I am calling 4085001414 |
| 23:47.23 | p3nguin | I guess I don't understand the problem. |
| 23:47.25 | _N1x | Set(CALLERID(num)=483517293)) |
| 23:47.45 | The-Kernel | _N1x, thats not the problem I got the formatted correctly |
| 23:47.56 | p3nguin | Watch for syntax failure if you pay attention to this guy. |
| 23:48.24 | _N1x | The-Kernel: you have problem , whe you trying to call out to siptrunk , your number is 1000 |
| 23:49.54 | p3nguin | Can you just describe what you want, so I can compare it to what you already have? |
| 23:50.10 | The-Kernel | how do I change From: "1000" <sip:4083517293@68.167.143.51>;tag=as524bbb61 into From: "4083517293" <sip:4083517293@68.167.143.51>;tag=as524bbb61 |
| 23:50.48 | _N1x | The-Kernel: what is your public number? |
| 23:50.57 | _N1x | provided from sip provider |
| 23:51.02 | The-Kernel | 408351729 |
| 23:51.06 | The-Kernel | 4083517293 |
| 23:51.25 | _N1x | The-Kernel: register => user:pass@sip/4083517293 |
| 23:51.36 | _N1x | Set(CALLERID(num)=4083517293) |
| 23:51.47 | _N1x | replace this string , reload sip and go rock |
| 23:52.10 | The-Kernel | that's how its set up as |
| 23:52.16 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 23:52.48 | [TK]D-Fender | You should be fixing your peer |
| 23:53.10 | [TK]D-Fender | No messing with caller-id |
| 23:53.11 | p3nguin | Is that a call from your phone or from your ITSP to your asterisk? |
| 23:53.59 | The-Kernel | yeah its a call from the 1000 phone, I don't think the "1000" matter now that I look at it...I'm getting a 604 back from the provider. I do register though |
| 23:54.15 | *** part/#asterisk _N1x (~n1x@95.104.13.145) |
| 23:55.30 | *** join/#asterisk leftist (~leftist@173.160.65.209) |
| 23:56.02 | leftist | can someone suggest a good product to backup a fresh asterisk build? it's been up for a month however i know i need to back the entire server up. |
| 23:56.19 | [TK]D-Fender | "register" has nothing to dow ith sending a mis-authed call |
| 23:56.26 | p3nguin | rsync, tar, dump... |
| 23:56.28 | The-Kernel | yeah i think I see the problem |
| 23:56.46 | The-Kernel | the FROM has the IP of the PBX, how can I change it to the IP of the Sip Server? |
| 23:57.02 | p3nguin | Why are you worried about that? |
| 23:57.56 | The-Kernel | p3nguin, I'm comparing it to a trace of a phone that registers directly and calls out |
| 23:58.15 | [TK]D-Fender | The-Kernel: * is a B2BUA, or course it is swending it's own IP |
| 23:58.19 | [TK]D-Fender | Your peer is bad. |
| 23:58.23 | p3nguin | Stop concentrating on the wrong things. |
| 23:58.28 | [TK]D-Fender | Stop wasting time on callerid and pb it |
| 23:58.33 | *** join/#asterisk gxdssoft (~gxdssoft@190.236.111.135) |
| 23:58.36 | p3nguin | its own |
| 23:58.43 | [TK]D-Fender | yes, that. |
| 23:58.51 | [TK]D-Fender | Too tired to type right.... |
| 23:59.03 | p3nguin | Fix the phone peer. Fix the ITSP peer. Done. |
| 23:59.11 | p3nguin | Make calls; enjoy. |