00:03.34 | *** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis) |
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00:14.00 | zopsi | Has anyone setup a spectralink 8020 for use with asterisk? |
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00:14.25 | lorsungcu | zopsi |
00:14.28 | lorsungcu | yes |
00:15.12 | zopsi | I have it pulling the files from tftp server, but my config appears to be wrong. I just get 1 of 1 lines unregistered. |
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00:19.19 | *** join/#asterisk volga629 (~volga629@173.209.129.250) |
00:20.54 | volga629 | Hello All, which vendor for ip phone with vpn ( but not openvpn ) recommended for asterisk ??? tnks |
00:23.14 | wonderworld | the first Playback() in a call always stutters for me. later Playbacks are OK. any idea on how to fix that? |
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00:24.19 | jpsharp | volga629: The Cisco SPA525G support SSL VPNs. |
00:27.29 | [TK]D-Fender | Sounds ike the time it takes a jitterbuffer to fully kick in. Not much youre going to be be able to do about that. |
00:31.47 | volga629 | it require ASA firewall or I can use any SSL firewall > |
00:31.51 | volga629 | ? |
00:37.50 | beardy | wonderworld: Play a second or a few milliseconds of silence first? |
00:40.56 | WIMPy | Or use Answer() with the observed time. |
00:41.03 | WIMPy | ... needed. |
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01:06.44 | CrazyTux[m] | Any easy way to do Gotoif(X matches regexp) ? |
01:11.21 | wonderworld | thanks |
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01:15.02 | wonderworld | delay of 3 secs in answer didn't fix it |
01:15.04 | lorsungcu | CrazyTux[m] https://wiki.asterisk.org/wiki/display/AST/Function_REGEX |
01:15.06 | wonderworld | trying silence now |
01:15.41 | wonderworld | is there a premade silence recording available? |
01:16.07 | lorsungcu | wonderworld use playback(silence/<number of seconds>) |
01:19.03 | wonderworld | 2 seconds of silence fix it |
01:19.07 | wonderworld | great |
01:19.57 | volga629 | is anybody was using ATCOM AT-620P ip phone ? |
01:20.50 | [TK]D-Fender | Atcom = cheap Chinese crap |
01:22.12 | lorsungcu_ | damn tk cold blooded. |
01:22.22 | volga629 | Just they claim that l2tp in place |
01:23.27 | wonderworld | i once had a chinese ip phone. it had 20 pages of config in it's menu. and never worked as expected |
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01:25.17 | volga629 | I have android fanvil one and can't update firmware at all and no documentation :-(, looking for something like snom 370 but not openvpn |
01:27.35 | lorsungcu_ | why not openvpn? |
01:28.46 | volga629 | because on distribution layer we have fortinet and cisco 3560 |
01:29.38 | volga629 | ideal will be terminate on fortinet l2tp tunnel and put on separate vlan |
01:30.24 | volga629 | I guest I can find voip gateway and regular phone ? |
01:34.24 | volga629 | any suggestion about small voip gateway ? |
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01:59.38 | Xaviertoor | Hello people |
02:00.13 | Xaviertoor | I installed asterisk 10.7.0 and comand sip no found in CLI |
02:00.31 | Xaviertoor | Any ideia? |
02:01.15 | Xaviertoor | what's wrong? |
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02:54.33 | *** join/#asterisk becca_r (~becca_r@12.25.151.58) |
02:54.49 | becca_r | hello, is anyone on? |
02:56.04 | darthanubis | just throw your question |
02:56.28 | darthanubis | the great thing about irc is that it'll be here when you step away |
02:56.49 | darthanubis | if someone can answer they will eventually |
02:56.54 | becca_r | I'm having an issue with acks being relayed. It appears only in 1 direction. My outbound calls are cutting off at approximately 20 seconds, but inbound calls are working without issue. |
02:57.10 | becca_r | Any help would be great. |
02:58.17 | darthanubis | just keep your chat program running, and some back later. I don't have an answer for you...:( |
02:59.19 | becca_r | 1.7 |
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03:16.04 | archetech | <PROTECTED> |
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03:33.13 | lorsungcu | becca, you get your stuff figured out? |
03:33.19 | becca_r | no, not yet |
03:33.23 | becca_r | found more about my issue. |
03:33.31 | lorsungcu | sounds like nat |
03:34.08 | lorsungcu | can you pastebin your sip.conf? |
03:34.24 | becca_r | I see that the ack comes in with the ruri for the destination as5400 and to uri of opensips. |
03:34.34 | becca_r | this is all nonat and same network |
03:34.42 | lorsungcu | ij |
03:34.43 | lorsungcu | ok |
03:35.04 | lorsungcu | do you have a sip capture of the call failing? |
03:35.08 | becca_r | I see that opensips is not finding a transaction for the ack, and then does not find a pattern match in the rules and routes the ack to the default route |
03:35.11 | lorsungcu | and can you pastebin that? |
03:35.45 | becca_r | I can pastebin opensips debug logs, and pcap |
03:35.47 | lorsungcu | is this issue on an asterisk machine, or opensips? |
03:36.30 | becca_r | opensips I do believe |
03:36.44 | becca_r | crap, wrong channel |
03:36.45 | becca_r | sorry |
03:36.48 | lorsungcu | well, i see you're in #opensips... |
03:36.53 | lorsungcu | :p |
03:37.34 | becca_r | thanks, sorry |
03:37.35 | lorsungcu | if there's an asterisk side to this, i'd be happy to help with that? |
03:37.48 | becca_r | well I think the asterisk side is working fine. |
03:38.01 | lorsungcu | ok |
03:39.08 | lorsungcu | brb taco pizza!@$ |
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03:59.33 | fling | [Aug 8 10:25:19] NOTICE[4746]: chan_sip.c:22674 handle_request_invite: Call from '' (50.56.187.230:5070) to extension '0810972599532957' rejected because extension not found in context 'default'. |
04:00.22 | lorsungcu | getting haxzed |
04:00.30 | fling | lorsungcu: hehe :p |
04:02.18 | fling | lorsungcu: I have this on the router > -A PREROUTING -p udp -m multiport --dports 5060,10000:11122,11124:20000 -j DNAT --to-destination 10.0.1.101 |
04:02.45 | fling | lorsungcu: but they can't call without auth? right? |
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04:21.04 | *** join/#asterisk CRCinAU1 (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
04:21.09 | CRCinAU1 | hmmm |
04:21.11 | CRCinAU1 | so I'm wondering. |
04:21.20 | CRCinAU1 | I have a fax machine hooked to an FXS port on a dahdi card. |
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04:21.48 | CRCinAU1 | and I use _X.,blah to grab what is dialed from the fax machine, then shoot it out over a SIP provider |
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04:22.16 | jofry | Hi folks |
04:22.34 | CRCinAU1 | Now I'd like to try and use t.38 on the asterisk->SIP provider side of things - as in theory, this will give better results (maybe) |
04:23.20 | CRCinAU1 | so, I figure somehow I have to tell the Dial() to try and go for a t.38 |
04:23.44 | CRCinAU1 | however this doesn't seem to happen - nor can I find a reference to t38 in the dial command in asterisk 10.7 |
04:24.41 | CRCinAU1 | so, how do I do this, and is there any advantage to doing it? |
04:24.43 | lorsungcu | https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway |
04:26.04 | CRCinAU1 | gateway would already assume SIP input with fax payload data though wouldn't it? |
04:26.06 | lorsungcu | also |
04:26.06 | lorsungcu | http://en.wikipedia.org/wiki/T.38 |
04:26.40 | CRCinAU1 | as its coming in from a dahdi source, wouldn't it be a T38 originator? |
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04:32.34 | fling | how can I redirect the call? |
04:33.06 | lorsungcu | in the default context, pattern match whatever call is being made, and send it where you wanty |
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04:34.16 | fling | I say "you should talk to the manager" then I press something like *102 (this is manager's number) and call goes to the manager <- this is what I want |
04:34.51 | CRCinAU1 | you want a transfer then, not a redirect. |
04:35.05 | fling | oh! ok… hmm hmm |
04:35.12 | CRCinAU1 | assuming you don't use SIP phones, look in features.conf |
04:35.27 | CRCinAU1 | if you use SIP phones, hopefully your SIP phone has a transfer button |
04:36.19 | fling | CRCinAU1: softphones and a single analog phone connected to dvg-7111s |
04:36.37 | CRCinAU1 | urgh lol |
04:36.54 | lorsungcu | wtf would you do that to yourself lol |
04:36.55 | CRCinAU1 | if its a softphone, hopefully the soft phone talks SIP and knows how to transfer |
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04:37.18 | fling | CRCinAU1: umm? |
04:37.43 | lorsungcu | what softphone |
04:37.59 | fling | ekiga, zoiper… |
04:38.04 | CRCinAU1 | fling: the transfer function is built into the SIP protocol. then its up to the software you use on if they've put a button there for you to press. |
04:38.35 | fling | windows-lovers use 3cx-phone |
04:38.58 | lorsungcu | http://www.zoiper.com/downloads/Zoiper_2.0_Free_Manual.pdf |
04:39.03 | lorsungcu | page 16 |
04:39.03 | fling | lorsungcu: ok |
04:39.07 | fling | CRCinAU1: thanks :p |
04:39.58 | fling | what about dvg-7111s? it should be builtin in the device too? |
04:40.22 | lorsungcu | doesnt need to knwop about the transfer |
04:40.48 | CRCinAU1 | fling: that will probably be handed in features.conf and using the flash button on the analogue phone |
04:41.13 | lorsungcu | the zoiper manual goes into that, actually.. |
04:42.19 | CRCinAU1 | actually, if you're using an ATA, the ATA will have its own feature codes |
04:42.33 | CRCinAU1 | so you'll dial something like: FLASH *70#102 |
04:42.42 | CRCinAU1 | obviously those numbers are made up, but yeah |
04:44.04 | fling | CRCinAU1: transfer button in client works |
04:44.14 | CRCinAU1 | as it should :) |
04:44.15 | fling | CRCinAU1: thanks! |
04:44.36 | fling | CRCinAU1: but flash button on analog phone is not |
04:44.57 | CRCinAU1 | read the book on your ATA |
04:45.03 | CRCinAU1 | it'll hopefully go into it |
04:45.39 | CRCinAU1 | you might have to do some config swindling on the ATA to enable / whatever it |
04:46.35 | fling | but what is ATA? :# |
04:46.54 | CRCinAU1 | what you plug your analogue phone into that then converts it to SIP |
04:47.05 | fling | oh! ok |
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05:23.06 | Sean-Der | Hey are there any web front ends to CEL? If not I am gonna make a simple one and share it if anyone cares |
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05:46.09 | ChannelZ | Hmmm.. so is there a way to make Asterisk's Jabber/gtalk integration show that it's available for voice chat, but not for IMs? I'm realizing that if someone IMs me while I'm offline with a client (Pidgin, my phone..) they're probably still getting sent to Asterisk and just disappear. |
05:51.31 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
05:51.33 | v0lZy | lo |
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05:57.33 | fling | v0lZy: lo |
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05:59.55 | v0lZy | hi |
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06:28.21 | fling | This is what I want: somebody calls my external number and if nobody answers she may press 102 and call goes to internal number 102 |
06:28.52 | v0lZy | extenisons.conf |
06:29.30 | v0lZy | then bind external number to your application |
06:29.36 | fling | v0lZy: http://bpaste.net/show/39005/ |
06:29.42 | fling | hmm hmm |
06:30.16 | v0lZy | theres a time option to dial btw |
06:30.37 | v0lZy | that afte a certain amount of seconds, if noone picks up, it continues the dialplan execusion |
06:30.37 | fling | can you give me an example or a link to read? |
06:30.44 | v0lZy | so then your user doesnt need to press anything |
06:30.56 | fling | no, this is not what I want |
06:31.12 | kaldemar | fling: is it only 102 you want the call to go to? |
06:31.32 | fling | kaldemar: no, I want it to be any local number |
06:31.33 | v0lZy | ah, kaldemar to the rescue |
06:31.38 | fling | v0lZy: :p |
06:32.01 | fling | exten => kaldemar,1,Hello() |
06:32.26 | v0lZy | :D |
06:32.52 | kaldemar | use a timeout in app Dial and then use WaitExten, Read or DISA to get the new destination. |
06:33.44 | kaldemar | but be extremely careful not to open up your box for outbound calls. |
06:35.07 | fling | I have this on the router > -A PREROUTING -p udp -m multiport --dports 5060,10000:11122,11124:20000 -j DNAT --to-destination 10.0.1.101 |
06:35.47 | fling | kaldemar: and a lot of these things in the log > [Aug 8 10:25:19] NOTICE[4746]: chan_sip.c:22674 handle_request_invite: Call from '' (50.56.187.230:5070) to extension '0810972599532957' rejected because extension not found in context 'default'. |
06:37.19 | kaldemar | those two are not related. |
06:37.58 | kaldemar | well, related in the way that the first enables the second to happen. |
06:38.06 | fling | kaldemar: they are trying to call but they can't while unauthorized? right? |
06:39.03 | fling | first is when somebody calls me and then press 972599532957 to call to Israel :P |
06:39.13 | kaldemar | fling: no. you're not even requiring them to authorize. like the message says, you don't have an extension in [default] that would match the dialed number. |
06:39.38 | fling | kaldemar: is this normal? |
06:39.41 | kaldemar | or authenticate rather. |
06:40.10 | kaldemar | fling: if you leave your box open to guest calls like that, then it is normal. |
06:40.17 | fling | hmm hmm |
06:40.31 | fling | what file to edit to prevent this? |
06:40.45 | kaldemar | sip.conf, allowguest=no |
06:41.07 | fling | fixed! thanks |
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06:46.30 | v0lZy | man, this is a bit cryptic for me |
06:46.42 | kaldemar | you might also want to set alwaysauthreject=yes to make bruteforcing harder. |
06:46.52 | v0lZy | im trying to reject a call forwarding setting if there is a loop |
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06:47.54 | v0lZy | what im doing is using DB to create a CF/${EXTEN} and checking if its set to "" or some other extension. |
06:48.25 | kikohnl | expecially when you get a sip attack like I'm getting, crappy 2wire/pace firewall can't null route and I'm getting about 150 registers a second from 182.72.249.66, all for the same extension |
06:49.09 | kaldemar | v0lZy: why ""? you should rather delete the value when the forwarding is disabled and use DB_EXISTS to check. |
06:50.05 | kaldemar | kikohnl: that's a time for iptables and -j REJECT or -j DROP for the address, which ever makes their script give up. |
06:50.37 | v0lZy | smarter |
06:51.06 | v0lZy | whats the syntax? DB_EXISTS(entry) ? |
06:51.36 | kikohnl | did that on my asterisk server, an old mac mini with ubuntu, but Hawaiian telcom iptv requres this crappy firewall |
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06:54.57 | kaldemar | v0lZy: "core show function ..." |
06:55.34 | v0lZy | can i show u some code |
06:55.41 | v0lZy | and u tell me if it makes sense |
06:57.22 | kaldemar | write your code and see if it works. if it does not, tell what is wrong, pastebin CLI output and the related dialplan. you don't always have to ask to ask. |
06:59.49 | v0lZy | http://bpaste.net/show/XGllhOvnOmthCRr6TE1M/ |
07:00.20 | v0lZy | im not sure my code is correctly formated... im following some examples... |
07:00.37 | v0lZy | but overall, does this make sense? |
07:00.54 | v0lZy | my concern is if im overlooking something here |
07:00.58 | v0lZy | logic. |
07:01.09 | v0lZy | i have ** as a prefix to call forward. |
07:01.46 | v0lZy | first, i check if the extension is trying to forward to itself... if it is, i skip to cancelforward label and Hangup. |
07:01.55 | v0lZy | (ill add some recoding to it later) |
07:02.23 | v0lZy | on the other hand, if ts not trying to forward to itself, i am checking if the destination it wants to forward to is itself forwarding anywhere. |
07:02.53 | v0lZy | I do that by storing the value in the DESTINATION_CF variable.... if there is no forwarding, it should be "" |
07:03.34 | v0lZy | if its not, that means that the direction im trying to forward to is itself forwarding somewhere... so i set DESTINATION_CF to the destination the number im trying to forward is forwarding to |
07:03.56 | v0lZy | if that number is the number im calling from, i go to tag cancel forward and hangup. |
07:04.56 | kaldemar | "if the destination it wants to forward to is itself forwarding anywhere." <-- anywhere? wouldn't you rather want to check if the destinatino forwards to the particular phone that is trying to enable forwarding? |
07:05.29 | v0lZy | If its not empty, and previously established, not the number im calling from, then i goback to see where its forwarding to.. and i repeat this process until i either encounter my own number from which im calling and hangup, or find a blank spot to set the forwarding |
07:06.09 | v0lZy | kaldemar: no, im concerned about making a forwarding loop involving more than 2 phones |
07:06.38 | v0lZy | A forwards to B, B forwards to C, C forwards to D, D forwards to A or B or C and i have a problem. |
07:07.58 | v0lZy | My reckoning is that if the destination im forwarding to has no forwarding, im safe. If the destination then decides to forward, it wont be allowed to create a forwarding loop where the calls it forwards would reach it back. |
07:08.41 | v0lZy | i dont have a problem with chain forwarding things. |
07:08.47 | v0lZy | as long as they dont create a loop |
07:08.58 | v0lZy | I'm checking this when the user is activating call forwarding |
07:09.07 | v0lZy | so such a loop should never be established. |
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07:09.53 | v0lZy | (i will then check if the destination being forwarded to is an internal number also, and prevent the forwarding if it isnt) |
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07:10.22 | schmidts | good morning |
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07:10.38 | kaldemar | v0lZy: http://pastebin.com/jGLJnjuP |
07:11.14 | kaldemar | v0lZy: you probably don't want to check all possible forwardings. at least not in dialplan. |
07:14.36 | v0lZy | the code you pasted... that only guards from the case where the extension being forwarded to is set to forward calls to the extension thats trying to establish this forwarding |
07:14.55 | kaldemar | yes. |
07:15.06 | v0lZy | it does not take into account a situation where A forwards to B and B forwards to C, and C forwards to A by mistake |
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07:15.29 | v0lZy | (thanks for the same => tip, handy!) |
07:17.27 | v0lZy | well.. is there any way to check for this stuff outside dialplan |
07:17.38 | v0lZy | the thing is such a chain could form at any moment |
07:17.51 | v0lZy | so it has to be prevented before its created. |
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07:20.19 | kaldemar | v0lZy: there are many ways to check it outside the dialplan. e.g. func SHELL, app System, AGI. |
07:20.45 | v0lZy | what would be the advantage of doing this outside dialplan? |
07:20.56 | kaldemar | better tools for the task. |
07:21.49 | v0lZy | dialplan should be able to handle this |
07:21.52 | v0lZy | or asterisk internally. |
07:22.00 | v0lZy | its not that complicated i think |
07:22.09 | v0lZy | let me test my code |
07:23.57 | kaldemar | if we enhance the first thing i pasted a bit: http://pastebin.com/ARYYWiKL |
07:24.35 | kaldemar | that would check forwardings looped until there is no forwarding or the forward is not set to the caller. |
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07:25.17 | kaldemar | in addition to that, you'd have to store all the checked forwardings and check against those too before deciding whether the forward does not make a loop. |
07:27.16 | kaldemar | sure you could use HASH and HASHKEYS functions to store and check, combined with a GoSub for example. |
07:27.23 | kaldemar | but the tools are limited. |
07:27.39 | v0lZy | im trying to understand the code |
07:28.17 | kaldemar | first line checks if the forward is to the caller itself. second line checks if the destination has enabled callforwarding. |
07:28.20 | v0lZy | first, u check if its calling to itself then cancel... ok, makes sense. |
07:28.44 | v0lZy | second, you establish if call forwarding is set on the destination |
07:29.04 | v0lZy | if it doesnt exist, u forward |
07:29.23 | kaldemar | if the destination has enabled call forwarding, start going through the chain. you'd have to replace "same => n,GotoIf($["${DESTINATION}" != "${CALLERID(num)}"]?doforward)" with a better check. |
07:29.24 | v0lZy | if it exists, you check where its forwarding to |
07:29.49 | v0lZy | and if its not forwarding to the caller, then you do forward (dangerous at this point in my opinion) |
07:30.47 | kaldemar | hence my previous comment. |
07:30.56 | v0lZy | ah, sorry, didnt see |
07:31.06 | v0lZy | and whats the one with ?check |
07:31.47 | v0lZy | it checks if the destination is doing call forward all over again.. ok, makes sense |
07:31.57 | kaldemar | oops. syntax errors there i see. |
07:32.07 | v0lZy | why n(loop) there btw? |
07:32.25 | kaldemar | it's invalid. |
07:32.54 | kaldemar | that example is crap and would not work. business as usual. :) |
07:33.41 | kaldemar | i first separated the "forward to self" and "loop" cases, you might want to use different indications before hanging up the call. |
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07:36.10 | youjelly | Can anyone suggest a good ITSP for USA? |
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07:41.01 | v0lZy | kaldemar, i've been looking at this approach, its fundamentally different mainly in the fact that you are using exist to check stuff |
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07:42.30 | fling | I need outlook integration for windows-lovers |
07:42.35 | fling | OutCall is not working |
07:42.40 | fling | what app do I need? |
07:43.06 | dagb | fling: Lync |
07:43.29 | dagb | fling: with the most expensive license |
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07:43.51 | fling | dagb: hmm hmm |
07:45.02 | v0lZy | kaldemar: but basic logic is the same....1. make sure your not forwarding to yourself directly, then check whoever you are forwarding to and loop that until you see if something comes back to you. |
07:46.48 | fling | dagb: will it connect to sip |
07:47.02 | fling | dagb: *to asterisk over sip? |
07:47.58 | din3sh | can a single * box handle +300 concurrent calls? |
07:48.07 | din3sh | alaw |
07:49.44 | dagb | fling: google it. if you want full integration with windows, including presence and whatnot, lync is the way to go. I believe you need some server software as well. Not sure if you can hook lync up to asterisk and be done with it. |
07:50.33 | fling | exten => dagb,n,Hangup() |
07:52.07 | youjelly | ITSP anyone???????? |
07:53.51 | kaldemar | din3sh: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
07:54.01 | kaldemar | ~itsp-us |
07:54.23 | kaldemar | ~itsplist-us |
07:54.23 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
07:55.42 | youjelly | thanks kaldemar <3 |
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08:06.44 | danfromuk | Hi, Ive got a complex dial plan and was wondering if its possible to do this.......... EXECIF(ChannelAnswered,Wait(1)) ? |
08:07.10 | danfromuk | Sorry, i meant to say ExecIf(ChannelNotAnswered,Wait(1)) |
08:07.53 | danfromuk | IE. If the call hasnt been answered elsewhere in the dialplan and is still ringing. |
08:08.48 | kaldemar | a mere Wait is enough. if the call had been answered, there would be no dialplan execution. |
08:09.38 | schmidts | danfromuk maybe you are looking for the M option of the Dial application |
08:10.06 | danfromuk | These are incoming calls, so no Dial command used. |
08:10.36 | danfromuk | I dont want the WAIT to be issued if the incoming call has been answered. This part of the dialplan can have unanswered and also answered calls. |
08:10.54 | kaldemar | all calls are incoming. |
08:12.03 | danfromuk | kaldemar, ok more specifically, no dial cmd is used. |
08:12.08 | schmidts | danfromuk if a call is answered by a peer there is no further dialplan execution, or do you mean the Answer application? |
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08:12.45 | kaldemar | danfromuk: why don't you show exactly what you're doing? |
08:15.21 | danfromuk | When a call comes in, I have a complex dialplan which uses a a few different things such as time to decide what to do. Sometimes the dialplan goes directly to a menu. However sometimes it goes to a cmd BUSY. |
08:15.45 | danfromuk | kaldemar: its a little hard to output as its in a realtime db. |
08:16.02 | danfromuk | Sometimes the call goes to a recorded message, then the menu. |
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08:16.57 | danfromuk | Basically, whenever there is an cmd that causes a unanswered channel to be answered, I want to do WAIT. Otherwise there is no ringing and the client thinks that isnt professional. |
08:17.00 | schmidts | danfromuk then you could easy set a var when you do a playback or answer the call and just check this var later |
08:17.39 | schmidts | or you can use the delay option of the answer application |
08:17.46 | danfromuk | schmidts: thought you might say that. I was wondering if there was a channel status var that showed whether a channel was unanswered. |
08:18.06 | schmidts | maybe let me check it |
08:18.10 | schmidts | which version do you use? |
08:18.13 | danfromuk | 1.8 |
08:21.09 | schmidts | ok there is not really a channel var you can use but the channel state will be UP when the call is answered |
08:21.32 | schmidts | you could maybe use the chanisavail app with the own channel and then check the availstatus |
08:21.42 | danfromuk | Ok. thanks. I'll run some tests. Otherwise your variable suggestion is great. |
08:21.56 | schmidts | forget it sorry |
08:22.27 | schmidts | just use the channel function like this: ExecIf($["CHANNEL(state)"="up"]?wait(1)) |
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08:25.24 | danfromuk | Ok, i'll give that a go and see what happens. |
08:25.34 | danfromuk | Thanks |
08:36.29 | danfromuk | schmidts: looks perfect. |
08:36.33 | danfromuk | thanks again |
08:39.48 | schmidts | your welcome |
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09:20.46 | v0lZy | hm |
09:20.48 | v0lZy | guys |
09:20.51 | v0lZy | anything wrong with this line |
09:21.27 | v0lZy | same => n,Set(FORWARD_TO=${DB(CF/${EXTEN})}) |
09:21.44 | v0lZy | it results in FORWARD_TO= |
09:21.48 | v0lZy | it doesnt set anything... |
09:22.32 | WIMPy | Does the DB contain anything? |
09:23.30 | v0lZy | yes |
09:23.47 | v0lZy | i set it with |
09:24.00 | WIMPy | For that key? |
09:24.11 | v0lZy | argh... |
09:24.13 | v0lZy | hold on |
09:24.15 | v0lZy | maybe it doesnt |
09:24.20 | v0lZy | im doing |
09:24.41 | v0lZy | n,Set(DB(CF/${CALLERID(number)})=${EXTEN:2}) ... |
09:24.52 | v0lZy | which is wrong i think :D |
09:25.07 | v0lZy | should be noop i presume? |
09:25.12 | WIMPy | Are the callerids and extensions the same? |
09:25.21 | v0lZy | nož |
09:25.23 | v0lZy | no* |
09:25.31 | v0lZy | can i run DB directly |
09:25.34 | v0lZy | or must i use noop |
09:25.34 | WIMPy | And it's CALLERID(num). |
09:25.36 | v0lZy | or set or somethign? |
09:25.42 | v0lZy | number works also |
09:25.45 | v0lZy | deprecated? |
09:26.13 | WIMPy | Not sure if it ever existed officially. |
09:26.32 | WIMPy | the *CLI has database ... |
09:28.55 | v0lZy | seems to be the correct syntax though |
09:32.10 | v0lZy | it just doesnt set the variable |
09:33.17 | v0lZy | either that |
09:33.20 | v0lZy | or it is truely empty |
09:33.22 | v0lZy | i dont kno know |
09:33.24 | v0lZy | but im using |
09:33.57 | v0lZy | n,Set(DB(CF/${CALLERID(num)})=${EXTEN:2}) ... |
09:37.28 | kaldemar | v0lZy: and then you try to read ${DB(CF/${EXTEN})} to FORWARD_TO. you haven't shown anything that indicates CF/${EXTEN} having any content. |
09:38.09 | WIMPy | That's the good old topic of using canonicialised numbers. |
09:40.46 | v0lZy | kaldemar: is my syntax in my previous line correct when filling the db? |
09:42.47 | WIMPy | The syntax is fine. |
09:42.57 | WIMPy | But that doesn't mean the data is. |
09:45.34 | kaldemar | v0lZy: you set to CF/${CALLERID(num)} and try to read from CF/${EXTEN}. |
09:46.58 | danfromuk | Is it possible to run multiple commands from one EXECIF? |
09:48.05 | v0lZy | kaldemar: well what im doing is, when i call **16 for example, im setting CF/CALLERID(num) to 16 |
09:48.37 | v0lZy | so that i know that the number that dialed **16 is to be redirected to 16 |
09:49.23 | v0lZy | when the call comes in, im checking the extension that was called (31) and seeing what the CF/31 is |
09:49.32 | v0lZy | hopefully it should give me 16, and use that to dial 16 instead of 31. |
09:51.28 | WIMPy | danfromuk: GosubIf? |
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09:52.58 | kaldemar | v0lZy: useless explanations. show the CLI output of a call and "database show". |
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09:54.44 | youjelly | all ITSPs offer SIP Trunks? |
09:55.05 | youjelly | kaldemar: |
09:55.49 | v0lZy | kaldemar: i dont see it in database show |
09:56.18 | v0lZy | just /SIP/Registry/ stuff |
09:56.24 | v0lZy | so... its not even getting set :| |
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09:56.39 | kladze | hello, i'm having a issue with asterisk 1.8.15 - I'm trying to use the danish voice prompts instead of the default english once.. I've copyed the "da" folder into /var/lib/asterisk/sounds/da - and changed sip.conf, with language=da - yet when i try to dial a test extension.. it says the following issue: http://pastebin.com/FsZ9iu7n - hope somebody can help me, and go easy on me, as im trying |
09:56.39 | kladze | to learn asterisk :) |
09:57.21 | kaldemar | youjelly: not necessarily. that's up to you to find out what they sell. |
09:57.55 | plundra | kladze: Does the file 1-and.ulaw exist? |
09:58.10 | kladze | its a .gsm |
09:58.36 | kladze | but yes the file exist |
09:58.50 | plundra | Try converting them to ulaw or do the call using gsm instead. |
09:59.04 | plundra | Using gsm as the codec that is. |
09:59.18 | bulkorok | shouldn't asterisk convert from gsm to ulaw when necessary!? |
10:00.31 | WIMPy | Could be a digits/da vs da/digits thing. |
10:02.39 | kladze | i noticed that the english sounds is .gsm files aswell and that works.. ? |
10:02.57 | plundra | Okay. Well then a path-thing probably :) |
10:03.55 | plundra | (I know I convert all prompts before hand, and I think it was due to it not transcoding on the fly for me :-) Also, it really seemed stupid doing that over and over again for all calls, all the time) |
10:04.09 | v0lZy | Any idea why its not setting the entry into the database? |
10:04.15 | kaldemar | kladze: if you do a Set(CHANNEL(language)=da) before the playback, does it work? |
10:04.28 | v0lZy | or better yet, can u give me an example of how to set something in the databasE? |
10:05.35 | WIMPy | v0lZy: Your syntax is still correct. |
10:05.50 | v0lZy | then i dont get it... |
10:05.59 | v0lZy | why isnt it setting? |
10:06.04 | WIMPy | Apart from the num vs number thing. |
10:06.09 | v0lZy | i fixed that |
10:06.12 | v0lZy | its all (num) now |
10:06.14 | WIMPy | What does CALLERID(num) contain? |
10:06.34 | kaldemar | the syntax written HERE was correct. |
10:06.54 | v0lZy | let me paste then |
10:07.43 | v0lZy | http://bpaste.net/show/MHmjkBCbByv457XqNNTo/ |
10:09.45 | WIMPy | It looks to me as if the line setting the DB entry would never be reaced. |
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10:10.05 | kaldemar | v0lZy: where's the CLI output? |
10:10.15 | v0lZy | moment |
10:10.33 | WIMPy | Is it supposed to be before the Goto(done)? |
10:11.49 | v0lZy | http://bpaste.net/show/XKraHPf0JaqkZc23s9Q1/ |
10:12.16 | v0lZy | ah, doh! |
10:12.31 | WIMPy | See. It never happens. |
10:12.49 | v0lZy | not before |
10:12.53 | v0lZy | its supposed to be done. |
10:13.26 | kladze | kaldemar - http://pastebin.com/xfkwWiYR see line 17, i guess is that what you wanted me to do correct? if that's thats the case.. then it still dont work :( |
10:13.38 | v0lZy | thanks guys |
10:13.46 | WIMPy | Or there. |
10:14.55 | kaldemar | kladze: do a "dialplan reload" before trying again. i think you forgot that since the Set is not getting executed. |
10:14.59 | *** join/#asterisk din3sh (~din3sh@41.212.202.249) |
10:15.59 | kladze | still the same, even after a dialplan reload |
10:16.39 | kaldemar | kladze: then you edited the wrong extension. |
10:18.14 | WIMPy | kladze: Try to check the path as I suggested. That oder has changed some time ago. |
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10:20.43 | kladze | http://pastebin.com/xfkwWiYR I added it here, line 17 - and output from asterisk: http://pastebin.com/gx1JR9Ju |
10:22.09 | v0lZy | hm |
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10:22.11 | kaldemar | kladze: still does not show up in the output between priorities 1 and 2. |
10:22.18 | v0lZy | ok, now its not doing something else its supposed to do, eheh. |
10:22.24 | v0lZy | have to recheck the logic behind this thing |
10:22.50 | WIMPy | v0lZy: You can spend lots of time on that topic. |
10:23.31 | WIMPy | e.g. I try to at least check that there's an extension for the destination. |
10:27.02 | v0lZy | i will add that check too |
10:27.14 | v0lZy | but right now im just trying for basic functionality and then upgrading |
10:27.48 | kaldemar | using dialplan like that (throwing all unknowns into _X!) is asking for trouble. |
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10:30.58 | v0lZy | kaldemar: at this point i suppose |
10:31.02 | v0lZy | but the way i have it set up now |
10:31.12 | v0lZy | its just a matter of changing _X! to _XX ... or _XXX etc |
10:32.48 | kaldemar | nearly as bad. |
10:33.52 | v0lZy | ...what should it be like then? |
10:33.55 | v0lZy | hardcoded? |
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10:35.54 | kaldemar | v0lZy: so that you don't have extensions that match dialed numbers but have non-existant destinations. |
10:36.55 | v0lZy | so.. hardcoded? |
10:38.02 | kaldemar | if that's what you like to call it. |
10:38.24 | v0lZy | hardcoded as in ... instead of _XX .. i use _00, _01, etc etc |
10:38.25 | v0lZy | ? |
10:40.07 | kaldemar | or _0[1-9] if you really have 01-09. XX and 100 separate lines are not two only options. |
10:41.14 | v0lZy | ah, gotcha |
10:42.36 | polomolo747 | hello, is there a way to cause sip rtp traffic to go through asterisk instead of p2p? |
10:45.47 | kaldemar | polomolo747: directmedia=no in sip.conf |
10:46.21 | v0lZy | hm |
10:46.23 | v0lZy | question |
10:47.02 | v0lZy | exten => _**X!,1,GotoIf($["${EXTEN:2}" = "${CALLERID(num)}"]?cancelforward) |
10:47.04 | v0lZy | <PROTECTED> |
10:47.05 | v0lZy | <PROTECTED> |
10:47.07 | v0lZy | <PROTECTED> |
10:47.46 | v0lZy | if ${DB(CF/${EXTEN:2})} is not set and has no value stored.... |
10:48.24 | v0lZy | then my gotoif there is useless.... i guess. |
10:48.47 | v0lZy | never mind, think i got it |
10:49.42 | v0lZy | damn, no, lost it... had it in my head and bunf, gone. |
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10:53.54 | polomolo747 | kaldemar, thanks, it kinda worked, now i can see the traffic goes to the asterisk and back to the devices but i suddenly i can't hear anything on either of them. |
10:54.57 | kaldemar | polomolo747: is NAT involved? |
10:55.09 | polomolo747 | no |
10:55.17 | polomolo747 | but eventually yes |
10:55.41 | v0lZy | damnation. |
10:56.02 | v0lZy | i think this line is wrong same => n(loop),ExecIf($["${DESTINATION_CF}" != ""]?Set(DESTINATION_CF=${DB(CF/${DESTINATION_CF})}) |
10:56.11 | kaldemar | "sip set debug on" in CLI and make a call that has sound issues. pastebin all output. |
10:59.17 | polomolo747 | kaldemar, http://pastebin.com/gC6CJNfn |
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11:05.56 | kaldemar | polomolo747: with a quick glance i can't see anything wrong in there. |
11:08.19 | polomolo747 | kaldemar, thanks. |
11:12.13 | v0lZy | funny |
11:12.24 | v0lZy | asterisk seems to handle these call forward loops by itself |
11:12.38 | v0lZy | right now i have 16 forwarding to 31, 64 forwarding to 16 |
11:12.46 | v0lZy | i call 64 from 31, and 16 rings |
11:13.06 | *** join/#asterisk b0ot (~Administr@147.177.60.174) |
11:13.19 | b0ot | why wouldn't my asterisk/freepbx install have the reload option in the CLI |
11:14.20 | kaldemar | b0ot: because the CLI alias is not configured in cli_aliases.conf. the real command is "module reload". |
11:15.28 | b0ot | thanks |
11:15.54 | v0lZy | kaldemar: does asterisk even allow for chaining callforwards? |
11:16.25 | v0lZy | or once u use a dial command, thats it, ur dialing, no more forwarding or anything |
11:16.41 | WIMPy | Asterisk doesn't do CF, it's your dialplan. |
11:17.46 | kaldemar | v0lZy: that is just dialplan, there is nothing to allow or disallow. |
11:18.09 | v0lZy | do context fold back on themselves? |
11:18.11 | v0lZy | like |
11:18.39 | v0lZy | I have n,Dial(SIP/${FORWARD_TO},to) |
11:18.53 | v0lZy | now if all phones are set to forward to eachother |
11:19.00 | v0lZy | hm |
11:19.06 | v0lZy | i guess i need another phone to test this |
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11:19.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
11:20.46 | b0ot | when i do a dundi show peers I get: 1 dundi peers [0 online,0 offline, 1 unmonitored] |
11:21.01 | b0ot | the guide i'm following shows it should be online |
11:21.19 | b0ot | any idea what could cause it to be unmonitored vs online |
11:23.18 | polomolo747 | kaldemar, i found the problem |
11:23.31 | polomolo747 | i binded the udp on 127.0.0.1 |
11:23.44 | kaldemar | b0ot: "qualify=yes" in dundi.conf |
11:23.48 | polomolo747 | how can i force sip to use tls? |
11:26.21 | b0ot | woot! |
11:26.37 | b0ot | kaldemar, thanks |
11:26.53 | b0ot | i had accidently typed it as quality |
11:30.28 | *** join/#asterisk os_Florent (~chatzilla@62.244.88.2) |
11:32.01 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
11:32.41 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
11:33.53 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
11:34.09 | WIMPy | polomolo747 |
11:34.11 | WIMPy | oops |
11:34.16 | WIMPy | polomolo747: transport=tls |
11:34.36 | polomolo747 | i want asterisk to not even listen on the sip port. |
11:35.27 | kaldemar | use tlsbindaddr only. |
11:35.31 | WIMPy | Yu can set it in the global section. |
11:35.55 | WIMPy | Wouldn;t it just use defaults for udp then? |
11:36.32 | WIMPy | But udpbindaddr=127.0.0.1 would at least make it unjavailable for others. |
11:37.03 | polomolo747 | the 127.0.0.1 will disable rtp |
11:37.25 | b0ot | I have to servers (A,B) where I am attempting the DUNDi setup. Both show each other as online with "dundi show peers" |
11:37.32 | polomolo747 | tlsbindaddr without idpbindaddr defaults to 0.0.0.0:5060 |
11:37.33 | v0lZy | obviously my call forwarding code doesnt allow for chained forwarding |
11:38.02 | v0lZy | I have 64 forwarded to 16 and 16 forwarded to 31 |
11:38.07 | v0lZy | when i call from 50 |
11:38.09 | b0ot | however I get "DUNDi Query EIT returned no results" when I do dundi query <mac server B> from server A |
11:38.18 | b0ot | any thoughts? |
11:38.30 | v0lZy | it should go 50 => 64 => 16 => 31 (ring) |
11:38.44 | v0lZy | Instead ... i get 16 ringing... which means it doesnt hop to 31 |
11:39.18 | v0lZy | and this is because i have Dial |
11:39.36 | v0lZy | i thought that dialing a channel would repeat the whole process in the internal context |
11:39.51 | v0lZy | will have to work on that too :| |
11:40.04 | kaldemar | if udpbindaddr is not configured and transport=tls is along tlsbindaddr, then it won't listen on any UDP port. |
11:40.23 | WIMPy | polomolo747: Can you elaborate on the disable rtp bit? That shounds wrong. |
11:41.23 | polomolo747 | if i set the bindaddr to 127.0.0.1 all the rtp ports will bind to the localhost, and i won't be able to get any rtp packets in |
11:41.38 | WIMPy | v0lZy: Dial a local channel. |
11:41.50 | polomolo747 | udpaddrbind is default at 0.0.0.0:5060 so i can't disable it. |
11:41.55 | v0lZy | WIMPy: how? |
11:42.13 | *** join/#asterisk wonderworld (~ww@dsdf-4db5d510.pool.mediaWays.net) |
11:42.18 | polomolo747 | there isn't an option to disable udpsip but leave rtpupd open, is there? |
11:42.33 | WIMPy | v0lZy: When doing to CF, dial the destinations extension via a local channel instead of a device. |
11:43.16 | WIMPy | polomolo747: I don;t know any, but rtp using the udpbindaddress sounds like a bug to me. |
11:43.32 | v0lZy | ... dial(LOCAL/16) instead of SIP there? |
11:43.41 | polomolo747 | there isn't a rtp bind addr is there? |
11:44.16 | WIMPy | polomolo747: Not that I know. |
11:44.41 | WIMPy | v0lZy: local/exten@context |
11:44.43 | polomolo747 | well, i can't disable it in the firewall level, so i guess it's still ok |
11:44.54 | polomolo747 | is srtp/zrtp is on udp? |
11:45.28 | WIMPy | polomolo747: yes |
11:45.47 | b0ot | How can I see each other with "dundi show peers" as online but fail to query one another |
11:46.11 | WIMPy | b0ot: What do you query? |
11:46.31 | kaldemar | b0ot: because of your mappings, if the peers are working. |
11:47.02 | polomolo747 | kaldemar, WIMPy , thanks. |
11:48.10 | b0ot | From Server A, I query Server B so "dundi query 00:00:00:00:00:0B" |
11:49.26 | WIMPy | b0ot: If you query for nothing, you will probably get nothing. |
11:49.33 | WIMPy | Add the right context |
11:50.13 | os_Florent | Hi, |
11:50.15 | os_Florent | I have a problem with an old version of asterisk (1.4.26.1) |
11:50.17 | os_Florent | My SS7 gateway sends call to my asterisk using SIP, and sometimes One-Way audio problems occur. |
11:50.18 | os_Florent | When such a problem occur, I can see those logs : |
11:50.20 | os_Florent | [Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring. |
11:50.31 | os_Florent | [Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring. |
11:50.35 | os_Florent | [Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring. |
11:50.37 | os_Florent | I can certify there is more than 64 headers in this case because of the source code : |
11:50.38 | os_Florent | #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */ |
11:50.40 | os_Florent | [..] |
11:50.41 | WIMPy | Don't flood the channel! |
11:50.42 | os_Florent | if (f >= SIP_MAX_HEADERS - 1) { ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n"); } |
11:50.43 | os_Florent | Is it possible to modify the source code to increase the 64 SIP headers limit ? |
11:50.45 | os_Florent | Upgrading Asterisk to another branch is currently not an option, because I need to find a fix quickly and doing so would require a lot of time. |
11:50.46 | os_Florent | Thank you ! |
11:50.48 | os_Florent | sorry |
11:50.57 | b0ot | drowned |
11:51.45 | b0ot | So i thought if you did the dundi query of just the mac it should return the EID WIMPy |
11:51.47 | WIMPy | Change that define and see if it's all it takes. |
11:51.53 | kaldemar | os_Florent: open source, dude. just modify it and recompile. |
11:51.57 | v0lZy | ha, i think it works now WIMPy, it hops to the next in line |
11:52.19 | WIMPy | b0ot: You need to provide a context. |
11:52.40 | kaldemar | the dundi query uses e164 by default if no context is provided. |
11:53.26 | WIMPy | Ok, but that needs to exist then. |
11:53.53 | v0lZy | btw |
11:53.54 | v0lZy | ringtones |
11:54.01 | v0lZy | phone ringtones |
11:54.10 | v0lZy | thats up to each device, or can asterisk push something to them' |
11:54.14 | os_Florent | kaldemar : ok thank you a lot. Do you know if this modification can impact asterisk negatively ? |
11:54.26 | WIMPy | v0lZy: yes :-) |
11:54.34 | v0lZy | lol |
11:54.35 | v0lZy | yes to which one |
11:54.52 | WIMPy | v0lZy: Yes, it's device specific, but some devices will accept the URL of a ringtone on your server. |
11:56.04 | WIMPy | os_Florent: You will have to find out. It looks unlikely, but my guess is as good as yours. |
11:56.53 | os_Florent | WIMPy: ok thank you |
11:57.13 | v0lZy | excellent |
11:57.13 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-79-252.bois.qwest.net) |
11:57.26 | v0lZy | thanks WIMPy/kaldemar, i got call forwarding working the way i wanted to |
11:57.36 | v0lZy | btw, can u explain the difference between dialing a device and a local chan? |
11:57.38 | kladze | Hi, when trying to do a playback of a file, etc of the number 1.gsm it can't be found.. it's danish/da voice prompts.. however if i move the 1.gsm from digits/1.gsm to /da main folder.. it can be played... isnt asterisk suppose to figure out that it shold look in the digits folder.. ? |
11:58.08 | b0ot | WIMPy, kaldemar I double checked the MAC's and they were correct. I also checked the dundi.conf file and server A has [MAC B] defined and vise versa |
11:58.14 | b0ot | not sure what the issue could be |
11:58.43 | WIMPy | v0lZy: The difference would have been obvious if you didn't use the extension numbers as device names. |
11:58.59 | WIMPy | v0lZy: A local cahnnel loops back to your dialplan. |
11:59.20 | kaldemar | b0ot: include, permit and mappings. |
11:59.29 | v0lZy | and device just shoots in the direction of whateve ru tell it to, got it |
11:59.47 | WIMPy | kladze: Did you note my two replies earlier? |
12:00.09 | WIMPy | b0ot: What contexts do you have defined for dundi? |
12:00.21 | rolandow | kladze: are the permissions in the folder danish/da correct? is the folder readable by asterisk? |
12:00.49 | v0lZy | btw, sounds i can play upon doing something |
12:00.50 | kaldemar | kladze: did you get the CHANNEL(language)=da in your extension? |
12:01.04 | v0lZy | can it be a wav |
12:01.13 | v0lZy | or must it be something specifc' |
12:01.26 | WIMPy | v0lZy: Ringtones? |
12:01.32 | v0lZy | no |
12:01.34 | v0lZy | like voicemail |
12:01.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
12:01.38 | v0lZy | stuff thats thrown together |
12:01.40 | v0lZy | i want it to say |
12:01.42 | WIMPy | That would obviousely be device specific. |
12:01.58 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
12:02.14 | b0ot | WIMPy, in IAX_extensions.conf |
12:02.15 | v0lZy | 'Call forwarding activated' and 'call forwarding deactiated' and 'error' and stuff like that. |
12:02.22 | b0ot | I defined [priv] |
12:02.32 | kladze | WIMPy hmm, i dont think so :\ |
12:02.36 | b0ot | with context=incomingdundi |
12:02.46 | kladze | rolandow they should be... drwxr-xr-x 2 root root 4096 Aug 8 13:53 digits |
12:03.14 | WIMPy | b0ot: What's that file? Where is it used? |
12:03.28 | v0lZy | i dont have the stuff i want it to say in my var lib asterisk sounds end dir |
12:03.31 | rolandow | kladze: i think you should do Playback(digits/1) |
12:03.33 | v0lZy | en* |
12:03.39 | kladze | kaldemar ye.. but it could not find the file in main "da" folder... i need to move the file out of "da/digits" to "da" before it was working |
12:03.57 | b0ot | As I understand it it, that is used to setup the IAX2 channel between the two servers |
12:03.58 | rolandow | kladze: it does check for the correct language, but when you want to go a folder deeper, you'd have to specify it. |
12:04.00 | b0ot | for communication |
12:04.12 | b0ot | WIMPy, following this guide: http://www.voip-info.org/storage/users/813/47813/images/1654/DUNDi_So_Easy.pdf |
12:04.36 | kladze | rocksfrow, oh okay.. i didnt know that.. thanks.. |
12:04.43 | WIMPy | kladze: Could be a digits/da vs da/digits thing. / Try to check the path as I suggested. That oder has changed some time ago. |
12:04.47 | kaldemar | kladze: asterisk does not automatically know that you mean a file in under the digits directory. |
12:05.02 | b0ot | lol although according to that the IAX2 channel isn't used for queries |
12:05.10 | kladze | so i always need to specify the subfolder aswell |
12:05.19 | kaldemar | b0ot: and it is not. |
12:05.20 | kladze | etc when i want to play a digit |
12:05.33 | rolandow | kladze: yes... and you could use your own created subfolders as well |
12:05.43 | WIMPy | b0ot: That's not a file Asterisk uses. So uinless you included that somewhere it won;t get used. |
12:05.49 | kladze | alright, thanks :) |
12:05.51 | rolandow | kladze: the part that asterisk automates, is to use the correct language (based on your settings) |
12:06.01 | rolandow | or at least that's how i understand it :) |
12:06.09 | kladze | alright :> |
12:06.16 | kladze | thanks |
12:06.19 | b0ot | so in the dundi file I have the [general] |
12:06.24 | b0ot | with my info in it |
12:06.31 | WIMPy | Oh, I thought it was happening when saying numbers/digits. |
12:06.35 | kaldemar | b0ot: the [priv] is only used when actually dialing the other box. you need to configure mappings in dundi.conf. |
12:06.48 | rolandow | hm.. |
12:07.01 | rolandow | good point WIMPy |
12:07.10 | rolandow | kladze: what command do you use to playback the digit? |
12:07.32 | rolandow | kladze: i think WIMPy is right when you use Digits() .. it probably takes the files from digits automatically .. |
12:07.46 | kladze | oh |
12:07.48 | kladze | hm |
12:07.48 | kladze | exten => 1000,n,Playback(1) |
12:07.53 | kladze | for number 1 |
12:08.12 | kladze | and there is also one named 1-and |
12:08.14 | WIMPy | kladze: Then it's as rolandow said. |
12:08.17 | b0ot | kaldemar, I have the dundi.conf file. It has the [general] section with organizational info. I have the [mapping] section with the priv => dundiextens,0.... and below that I have [mac other server] with model,host,inkey,outkey,include,permit,qualify,order defined |
12:08.28 | WIMPy | was thinking about SayDigits or SayNumber |
12:08.36 | rolandow | kladze: right.. with playback you have to specify the subdir |
12:08.51 | kladze | Okay :) |
12:08.53 | kaldemar | b0ot: and does it permit "priv"? |
12:08.55 | b0ot | I have already generated my keys and copied both the pub/priv key from server A where I generated to server B in the /var/lib/asterisk/keys |
12:09.02 | b0ot | yes |
12:09.05 | b0ot | permit = priv |
12:09.15 | rolandow | so you probably should use SayDigits or SayNumber :) |
12:09.41 | kaldemar | b0ot: "dundi query 00:00:00:00:00:0B@priv" |
12:10.01 | kladze | well thanks WIMPy, rocksfrow, kaldemar my da did work as intended before.. just me unaware of some playback and folder issues :D |
12:10.11 | kladze | rocksfrow, will try that :) |
12:10.15 | kladze | wops |
12:10.22 | kladze | rolandow, will that :) |
12:10.28 | kladze | try* |
12:11.31 | b0ot | kaldemar, that works! |
12:11.35 | *** join/#asterisk Jasnejac (~EnorMouse@2001:b70:500:2:645b:c5f6:428d:bfe0) |
12:12.05 | b0ot | well it originally didn't but i didn't have the inkey/outkey correct, but it gave me an error that was obvious ennough to fix it |
12:12.08 | b0ot | and then it worked! |
12:14.23 | b0ot | Sweet... now I just need to figure out what to do next |
12:14.25 | b0ot | lol |
12:15.02 | WIMPy | Pray |
12:16.16 | b0ot | How would I set it up for Dundi to first check my local extensions |
12:16.21 | b0ot | and then if not use Dundi |
12:16.49 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
12:17.32 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
12:17.32 | Ice_Strike | Hi |
12:17.33 | WIMPy | That happens automatically if you use a dundi switch. |
12:19.21 | Ice_Strike | Why I am getting a lot of error |
12:19.26 | Ice_Strike | sometime it does make calls |
12:19.27 | Ice_Strike | http://pastebin.com/LBw8bLZD |
12:19.41 | Ice_Strike | I moved asterisk to new server and now getting this error |
12:19.47 | Ice_Strike | I have not seen this before |
12:20.45 | WIMPy | That peer checks your IP? |
12:22.37 | Ice_Strike | 7x.129.2x3.52 my server ip |
12:23.31 | WIMPy | You talk to yourself? DNS troubles? |
12:23.57 | WIMPy | No, that's just the contents. |
12:24.45 | Ice_Strike | how do I check? |
12:24.56 | Ice_Strike | Is this something to do with RTP? |
12:25.16 | WIMPy | Use sip debug to find out where it's comming from. |
12:25.31 | WIMPy | Or use wireshark. |
12:28.44 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
12:29.13 | polomolo747 | Is there anyone here with sip text message experience? |
12:31.14 | *** join/#asterisk Blueneon (~Blueneon@196-210-194-169.dynamic.isadsl.co.za) |
12:32.13 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
12:32.27 | *** join/#asterisk aurs (~aurs@84.49.69.109) |
12:34.20 | Blueneon | Hi, I have an asterisk server setup and working perfectly, it also already has our office hours setup and going directly to an "after hours" message and voice mail when the office if closed. However, on some occasions the office would be closed under what would normally be standard office hours. During these times/days the boss wants to be able to set a temp message stating that the office |
12:34.20 | Blueneon | is closed and have asterisk direct incoming calls to voicemail. I know I can do this programatically in extensions.conf each time we need to do this. Though the boss is not very technical and wants a simple solution instead. |
12:34.26 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
12:34.32 | Blueneon | Is there something built into asterisk that will allow this? |
12:35.57 | Blueneon | Basically like the temporary message in the voicemail system, but rather than just for that extension, the entire system would not play the standard welcome greeting and dial the extension for X time and then get the temp msg, but instead just play the temp greeting and go direct to voicemail |
12:37.04 | v0lZy | WIMPy: can i playback a wav file with the playback command? |
12:37.16 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-tgumtrtxlfvcgnqo) |
12:37.16 | *** mode/#asterisk [+o newtonr] by ChanServ |
12:37.29 | [TK]D-Fender | Blueneon, Use a flag as a forced-state indicator as to your status and check that when the call comes in. Make an exten to toggle it on demand |
12:37.44 | [TK]D-Fender | Blueneon, "core show function DB" |
12:37.47 | [TK]D-Fender | ^^^ |
12:38.09 | DrDamnit | v0lZy: Yes. It has to be encoded properly, though. http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk |
12:38.26 | Blueneon | but that would still have asterisk play the standard greeting .. then go to extension, check flag, play msg and go to voicemail |
12:38.47 | [TK]D-Fender | Blueneon, sure... it's your dialplan.. put that check wherever YOU want it to be |
12:39.00 | Blueneon | oh ok im with you |
12:39.16 | rolandow | Blueneon: you could also write a dialplan that records a message |
12:39.32 | Blueneon | check flag first, then play temp "office closed" and go to voicemail, OR if flag is ok, resume normal ops? |
12:40.14 | Blueneon | ok what function is used to set/check flag? |
12:40.33 | v0lZy | 8kbit ad al that? |
12:40.53 | *** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis) |
12:41.06 | [TK]D-Fender | <PROTECTED> |
12:41.31 | DrDamnit | Yep. |
12:42.16 | Roelt | afaik the asterisk book has an example on that.. including an blf light for the status |
12:42.19 | Blueneon | Ah ;) |
12:43.26 | Blueneon | ok so, create exten, say "123" which gets user to record a message in say "/tmp/rec", then sets flag in DB and disconnects... then in dial plan have asterisk check flag, if set, play /tmp/rec and go to voicemail, otherwise resume normal functions? |
12:43.34 | Blueneon | thanks TK :) |
12:44.41 | [TK]D-Fender | yup |
12:44.57 | rolandow | Blueneon: http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/IVR_id247054.html <-- recording example |
12:44.59 | WIMPy | v0lZy: Sure. But depending on your Asterisk version there are some restrictions on the format. |
12:45.39 | rolandow | Blueneon: little bug in there, the mv command on line three from the bottom should have the full path as well, as shown at the rm command |
12:46.15 | rolandow | anyways.. you could asjust that to your needs |
12:47.27 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
12:48.42 | Blueneon | thanks |
12:49.08 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
12:52.26 | b0ot | what pastebin should we use here |
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12:52.59 | WIMPy | any |
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12:53.44 | rolandow | the registered version |
12:55.06 | b0ot | WIMPy, kaldemar This is my setup. I think there might be something wrong the way I setup my extensions_customs.conf |
12:55.07 | b0ot | http://paste2.org/p/2098445 |
12:55.15 | b0ot | It is about as minimalist as you can get |
12:55.15 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
12:55.26 | b0ot | the calls don't seem to try to use dundi |
12:56.43 | *** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis) |
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12:58.35 | kaldemar | b0ot: you're lacking "dundiextens" in extensions.conf. |
12:59.19 | kaldemar | b0ot: you've configured that in the priv mapping. so that context is what the priv mapping knows, and those extensions are used in DUNDi responses. |
13:00.00 | kaldemar | b0ot: exten => _2XXX |
13:00.07 | kaldemar | b0ot: ^ an invalid line |
13:00.25 | b0ot | sorry I think my paste got messed up, I'm fixing it there |
13:00.48 | kaldemar | b0ot: and you don't have a DUNDi switch or any manual lookups in extensions.conf. |
13:01.43 | b0ot | http://paste2.org/p/2098448 |
13:01.51 | b0ot | corrected extensions_custom.conf |
13:03.24 | b0ot | "All circuits are busy now" when I try to call |
13:04.11 | WIMPy | I don;t have any idea what a dundi trunk means. Might be a good idea to take the whole thing to #freepbx. |
13:04.48 | kaldemar | there is no such thing as a DUNDi trunk. |
13:04.55 | [TK]D-Fender | FreePBX <------------ |
13:05.07 | kaldemar | b0ot: you have DUNDi configured but your dialplan does not use it in any way. that's your issue. |
13:05.34 | *** join/#asterisk bchia (~Adium@nat/digium/x-yqxcksvcqxzaxmcg) |
13:05.39 | b0ot | kaldemar, the DUNDi trunk is configured via freepbx |
13:07.19 | Blueneon | TK: your plan worked like a charm |
13:07.29 | Blueneon | just wanted to say thank you so much :) |
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13:13.02 | bobb_WU | could someone examine this console output and offer advice on how to improve the transfer to voicemail (x2600 with a diversion header)? http://pastebin.com/z41dw1Ja |
13:13.12 | b0ot | what do you mean by a dundi switch |
13:13.16 | b0ot | kaldemar, |
13:13.45 | bobb_WU | and maybe explain what the warnings mean... |
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13:15.59 | WIMPy | b0ot: You need a 'switch => dundi/context' somewhere in your dialplan. |
13:19.16 | WIMPy | bobb_WU: I don't know what waitid thing is, but retransmissions are either networking issues or serious protocoll errors. |
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13:33.23 | Katty | hello my asterisk does not work at all how to fix??? answe rplz. |
13:34.18 | rolandow | do you have a macro for that katty? |
13:34.28 | Katty | what is macro??? |
13:34.37 | rolandow | &!%@& |
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13:39.28 | eXcAliBuR | when i call a number from my sip phone through asterisk, it doesn't allow me to enter the extension number... the dialplan i have for the number i'm calling is exten=> 306,1,Dial(Sip/306,10) -- same => n,Hangup() |
13:40.08 | eXcAliBuR | i can dial 306, but then when connected, any thing else I dial is ignored |
13:40.10 | eXcAliBuR | :( |
13:41.02 | eXcAliBuR | I don't know where my big book is from leifmadsen :< I think i left it at a site i was working at |
13:41.22 | rolandow | why would you dial something when you're connected ? |
13:41.34 | eXcAliBuR | because when connected, i have to dial a extension |
13:41.35 | rolandow | http://astbook.asteriskdocs.org/ |
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13:42.27 | rolandow | how did you configure dtmfmode ? |
13:42.31 | b0ot | I have one way calling working with dundi |
13:42.33 | b0ot | lol |
13:42.40 | b0ot | don't know how it's possilbe the configs are identical |
13:43.30 | rolandow | eXcAliBuR: you should use directmedia=no in sip.conf .. and probably dtmfmode=rfc2883 .. but i think that depends on the phone you're using. |
13:43.35 | WIMPy | eXcAliBuR: And what do you expect to happen with any keypresses after you have been connected? |
13:43.58 | eXcAliBuR | well it should pass them to the device i'm connected too |
13:44.04 | rolandow | assuming that you're using WaitExten or something like that in your dialplan after you connected to 306 |
13:44.52 | WIMPy | Ok, so that has nothing to do with dialling. And yes, check your DTMF modes. |
13:45.09 | eXcAliBuR | my dtmf = rfc2833 |
13:45.15 | eXcAliBuR | should i change to 2883 ? |
13:45.38 | eXcAliBuR | i have the digium phone |
13:45.42 | eXcAliBuR | :) |
13:45.43 | rolandow | eh no |
13:45.47 | rolandow | typo |
13:46.04 | WIMPy | RFC2833 is usually the best choice. |
13:46.05 | rolandow | rfc2833 is ok |
13:46.14 | WIMPy | And what about the peers configuration? |
13:46.15 | v0lZy | WIMPy: how to check if an extension is a valid sip peer? (as in, not application or external number9 |
13:46.16 | eXcAliBuR | maybe i don't need the Hangup() line? |
13:46.33 | rolandow | eXcAliBuR: pastebin your dialplan |
13:46.42 | rolandow | for 306 |
13:46.57 | WIMPy | v0lZy: You can't You can only check for existance. Use different contexts. |
13:46.58 | rolandow | you're calling 306, then you type an extension and you want something to happen?? |
13:47.18 | eXcAliBuR | when 306 answers, it gives me options press this,,, press that |
13:47.24 | [TK]D-Fender | An extension is an extension. |
13:47.29 | rolandow | eXcAliBuR: so what's your dialplan for 306? |
13:47.33 | [TK]D-Fender | An extension is not a peer |
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13:48.28 | v0lZy | WIMPy: uhm... seriously... no way to do that? |
13:48.34 | eXcAliBuR | http://asterisk.pastebin.ca/2178478 |
13:48.43 | WIMPy | v0lZy: no |
13:48.55 | eXcAliBuR | thats my extensions.conf |
13:48.57 | v0lZy | what about sip.conf |
13:49.02 | eXcAliBuR | i didn't leave anything out |
13:49.11 | WIMPy | v0lZy: The only relation between extensions and devices is what you write in your dialplan. |
13:49.36 | v0lZy | what about stuff registered in sip.conf |
13:49.54 | eXcAliBuR | http://asterisk.pastebin.ca/2178479 |
13:49.57 | WIMPy | v0lZy: That has nothing to do with extensions at all. |
13:50.14 | v0lZy | well |
13:50.18 | [TK]D-Fender | eXcAliBuR, i can dial 306, but then when connected, any thing else I dial is ignored <----- when you're connected, you're IN A CALL. * doesn't keep processing stuf you do in the background |
13:50.25 | [TK]D-Fender | eXcAliBuR, Dialplan is linear |
13:50.25 | v0lZy | i can check if the dialed extension also resides in sip.conf |
13:50.43 | WIMPy | v0lZy: Off course if you happen to name them the same as your extensions (which is not a good idea for security reasons) you could make some assumptions. |
13:50.58 | v0lZy | i name them like |
13:50.59 | [TK]D-Fender | v0lZy, "core show function SIPPEER" |
13:50.59 | v0lZy | SIP16 |
13:51.14 | [TK]D-Fender | v0lZy, And stop saying an "extension" is in "sip.conf" |
13:51.32 | v0lZy | thanks Fender |
13:52.01 | v0lZy | thanks for all the help guys |
13:52.01 | v0lZy | gona scoot now |
13:52.02 | eXcAliBuR | Tk fender: when I dial 9, phone number, i can dial stuff when connected |
13:52.11 | eXcAliBuR | :{ |
13:52.12 | v0lZy | have a nice day |
13:52.14 | v0lZy | bye |
13:52.23 | rolandow | eXcAliBuR: the dialplan you show here, doesn't have any "press 1 for this, press 2 for that" stuff |
13:52.26 | [TK]D-Fender | eXcAliBuR, Not inside the rest of your dialplan you can't |
13:52.42 | rolandow | eXcAliBuR: i would expect you'd Playback() some audio, wand use WaitExten or something for input |
13:53.37 | b0ot | I really don't understand how this can work one way, but not the other |
13:53.44 | b0ot | the configs are identical |
13:54.42 | eXcAliBuR | i'm going to try something |
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13:57.41 | eXcAliBuR | could it be the dialplan in the phone messing me up? |
13:58.26 | WIMPy | eXcAliBuR: No. You are not dialling. |
13:59.46 | eXcAliBuR | ok when i dial 306 i hear ringing, then 306 answers and says please enter conference id... I enter the ID but it's not getting there... 306 continues to ask me for ID |
14:00.28 | WIMPy | Check the DTMF modes ON BOTH ENDS. |
14:01.11 | WIMPy | Or all 3 even. |
14:01.34 | [TK]D-Fender | eXcAliBuR, That;s a little clearer... your initial description was very vague |
14:01.50 | [TK]D-Fender | eXcAliBuR, and you have a DTMF MODE problem as WIMPy has mentioned |
14:01.53 | rolandow | so 306 is a conference device? |
14:01.59 | eXcAliBuR | yes |
14:02.13 | eXcAliBuR | that has sip capabilites |
14:09.36 | rolandow | i'd say dtmf mode |
14:10.02 | rolandow | what codec do you use?? you could try inband as well if you use alaw/ulaw |
14:11.06 | rolandow | hm .. i also read something on this channel last week from some guy who raised the rx or txgain .. something like that.. |
14:11.14 | rolandow | seemed the dtmf tones weren't loud enough ? |
14:12.54 | eXcAliBuR | i have a rmx 1515 |
14:12.58 | eXcAliBuR | polycom |
14:13.28 | eXcAliBuR | i haven't found dtmfmode on it yet |
14:13.29 | eXcAliBuR | :/ |
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14:15.11 | [TK]D-Fender | eXcAliBuR, So go through them ALL one at a time |
14:15.15 | b0ot | Anyone know what this could be from: channel.c: Prodding channel 'SIP/2003-0000006' failed |
14:15.29 | rolandow | eXcAliBuR: dtmfmode is in your sip.conf |
14:15.53 | [TK]D-Fender | b0ot, Stick to #freepbx and don't post singular messages like that. Show them the ENTIRE call, not tiny little crumbs |
14:18.37 | Katty | dude. |
14:19.01 | mirela666 | eXcAliBuR: Do You have a webUI for polycom phones? |
14:20.31 | eXcAliBuR | the phone is digium |
14:21.07 | eXcAliBuR | it's a nice phone, i'd recommand everyone to buy it |
14:21.09 | mirela666 | On my Linksys I can change many SIP, codecs and dtmf mods but only trough WebUI |
14:21.09 | eXcAliBuR | :) |
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14:40.45 | b0ot | why won't you PROD channel |
14:40.53 | b0ot | don't you want to be prodded |
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14:51.39 | eXcAliBuR | YAY it's working |
14:51.46 | eXcAliBuR | it was a problem in the polycom :) |
14:52.25 | lorsungcu_ | what? |
14:52.50 | lorsungcu_ | didn't you just say it was a digium phone? |
14:53.32 | [TK]D-Fender | <eXcAliBuR> i have a rmx 1515 |
14:53.32 | [TK]D-Fender | <PROTECTED> |
14:53.48 | [TK]D-Fender | it's still on my screen... didn't even have to scroll.... |
14:54.00 | rolandow | maybe you have a very large screen |
14:54.07 | lorsungcu_ | <eXcAliBuR> the phone is digium |
14:54.23 | [TK]D-Fender | Yes, and the PHONE wasn't the issue |
14:54.26 | [TK]D-Fender | it was the otehr end |
14:54.29 | [TK]D-Fender | other* |
14:54.33 | lorsungcu_ | right on |
14:54.34 | lorsungcu_ | missed that |
14:54.35 | rolandow | so what was the solution ? |
14:59.36 | Ice_Strike | How to copy 100 files from a dir to another dir? recently created file |
15:00.44 | lorsungcu_ | generally i have two windows open |
15:00.48 | lorsungcu_ | so i can type faster. |
15:01.09 | lorsungcu_ | then just open each file and type the contents into the other window. |
15:01.23 | rolandow | quit |
15:01.26 | rolandow | wrong window |
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15:06.08 | b0ot | busy |
15:06.16 | b0ot | tones make me |
15:06.20 | b0ot | want to throw |
15:06.22 | b0ot | phones |
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15:08.14 | geeknation | hey |
15:08.22 | lorsungcu_ | hi |
15:08.24 | geeknation | I've got an issue with the ruby-agi gem |
15:08.58 | geeknation | Was working on Rackspace, but just moved over to EC2 and its throwing errors |
15:09.59 | geeknation | any idea if it needs to be configured differently on Amazon? |
15:11.37 | lorsungcu_ | no idea. I've been meaning to mess with that, though |
15:12.32 | geeknation | testing the requires in rib throws this |
15:12.34 | geeknation | LoadError: no such file to load -- ruby-agi/rs/ |
15:12.52 | geeknation | rib=irb ;) |
15:14.04 | geeknation | whats the best way to debug? |
15:14.17 | b0ot | lorsungcu you have any freepbx-fu |
15:14.35 | lorsungcu_ | way you need b0ot |
15:14.58 | lorsungcu_ | geeknation, id start by checking that that file exists... |
15:15.14 | b0ot | I have a freepbx setup as basic as possible. 2 servers each with 3 phones |
15:15.23 | b0ot | I have attempted to setup DUNDi |
15:15.29 | b0ot | I have one way calling working |
15:15.47 | geeknation | I'm looking in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi/rs |
15:15.56 | navaismo | lorsungcu, Qwell finally its working the agi script \o/ http://youtu.be/CRpyLRtmszM |
15:16.07 | b0ot | lorsungcu_, my setup: http://paste2.org/p/2098542 log of call that doesnt' work: http://paste2.org/p/2098547 |
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15:17.17 | geeknation | Which file deals with the require? |
15:17.30 | Qwell | navaismo: Was I right about the problem? |
15:17.39 | FinboySlick | While this isn't 100% asterisk related. Are any of you aware of firmware/hardware differences between SPA2102-R1 and SPA2102-NA? |
15:17.55 | lorsungcu_ | awesome navaismo |
15:18.01 | lorsungcu_ | what was the issue |
15:18.10 | FinboySlick | I figure some people here might have experience with relatively large deployments of those. |
15:18.27 | Qwell | lorsungcu: I'm almost certain is was agi-bin/ not being g+r or o+r |
15:18.40 | lorsungcu_ | yes thats what it looked like, qwell |
15:19.12 | Qwell | I didn't bother checking how ls shows permissions of .., and he never gave me what I actually asked for. |
15:19.45 | Qwell | anyways, he'll never say what the problem actually was, so any discussion is moot. |
15:20.00 | lorsungcu_ | navaismo: dammit what was the problem :D |
15:20.04 | navaismo | Qwell, lorsungcu actually the issue was with php and memory, I testes a simple agi script (stream file) in my machine with same permission |
15:20.30 | navaismo | and it works so i start from stcratch again and recompile asterisk without many unused modules |
15:20.35 | navaismo | create a bigger swap |
15:20.43 | Qwell | start from scratch = fixed permissions |
15:20.50 | lorsungcu_ | yeah i kind of agree |
15:20.53 | navaismo | after 3 hours copy again the script and work aout of the box |
15:21.04 | navaismo | nope i dont change the permissions |
15:21.11 | lorsungcu_ | post permissions for ago-bin and ../agi-bin |
15:21.13 | navaismo | on scripts or agi dir |
15:21.19 | navaismo | I just copied again |
15:21.48 | Qwell | ls -ld /var/lib/asterisk/agi-bin/ |
15:21.50 | Qwell | Show us what that says. |
15:21.55 | navaismo | yep 1 sec |
15:23.29 | b0ot | :( |
15:24.27 | navaismo | here -> http://pastebin.com/Rr0KM8zj |
15:24.41 | Qwell | o+r |
15:24.47 | lorsungcu_ | yeah its different |
15:24.57 | navaismo | :S |
15:25.00 | geeknation | hmmm, weird -- my amazon installation didn't create a /var/lib/asterisk/agi-bin |
15:25.01 | Qwell | You should have listened to what you were told, before going off any reinstalling everything. |
15:25.02 | lorsungcu_ | :p |
15:25.07 | Qwell | This was an incredibly simple fix. |
15:25.09 | lorsungcu_ | but |
15:25.10 | geeknation | maybe thats my problem….? |
15:25.13 | navaismo | Nope |
15:25.14 | lorsungcu_ | now he knows :D |
15:25.25 | navaismo | that was not easy |
15:25.28 | Qwell | lorsungcu_: You overestimate things. |
15:26.11 | Qwell | geeknation: well, if Asterisk can't see a file, then it can't execute it. I haven't seen your problem description though. |
15:26.52 | navaismo | before compilation o+r doesnt work but important thig is working now |
15:28.00 | geeknation | Qwell ruby-agi is throwing an error on require |
15:28.08 | Qwell | geeknation: pastebin |
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15:29.09 | geeknation | http://pastebin.com/3XzQ4k5V |
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15:38.51 | navaismo | geeknation, and this exist ruby-agi/rs/ |
15:39.43 | geeknation | I have a bunch of rb files in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi/rs |
15:40.26 | geeknation | but ruby-agi.rb is in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib |
15:40.31 | geeknation | should i move it to /rs |
15:40.33 | geeknation | ? |
15:41.46 | navaismo | or in your include you set the complete path |
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15:43.49 | geeknation | getting the same error hardcoding the path |
15:44.54 | geeknation | did the following - require '/var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi.rb' |
15:45.07 | lorsungcu_ | check permissions? |
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15:45.18 | geeknation | yah - did that as well :( |
15:46.03 | geeknation | ruby-agi.rb has -rw-rw-r-- |
15:46.09 | Qwell | no x? |
15:46.15 | Qwell | Do you need +x to be included? |
15:47.15 | geeknation | same permission were working on rackspace |
15:47.31 | lorsungcu_ | try with 777 and revert if it doesn't change |
15:47.47 | geeknation | there's also a folder called 'ruby-agi' with the following permissions: drwxr-xr-x |
15:47.58 | navaismo | jhehe sounds me familiar |
15:48.03 | lorsungcu_ | :p |
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15:50.21 | geeknation | just changed the permissions to 777 |
15:50.23 | geeknation | it's working |
15:50.25 | geeknation | :D |
15:50.41 | geeknation | u think it's safe if i switch it to 755? |
15:51.03 | Qwell | geeknation: depends on ownership |
15:51.13 | Qwell | actually, I lied |
15:51.15 | geeknation | well - thanks for all the help |
15:51.17 | Qwell | 755 should be fine |
15:51.22 | geeknation | thanks |
15:51.29 | geeknation | qwell - i appreciate all the debugging |
15:51.36 | geeknation | hopefully i can reciprocate one day :) |
15:51.38 | Qwell | geeknation: That'll be $299.95 |
15:51.47 | geeknation | ha |
15:52.00 | geeknation | checks in the post |
15:52.06 | geeknation | :) |
15:52.20 | Qwell | I should stop saying that. Somebody might actually take me seriously one day... |
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16:07.50 | lorsungcu_ | swell, you know anything about swift? |
16:07.56 | lorsungcu_ | wtf autocorrect |
16:08.11 | lorsungcu_ | Qwell: you know anything about swift? |
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16:28.35 | geeknation | hey guys… I'm having a new issue… none of my asterisk sound files are playing - even 'hello-world' |
16:28.54 | geeknation | i overwrote all my files in the asterisk/sounds/en (with a set from another server) |
16:29.01 | geeknation | and i can't get them to play |
16:29.12 | geeknation | i've tried changing the owner to ubuntu, asterisk, and root |
16:29.20 | geeknation | and all the files are set to chmod 777.... |
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16:31.45 | lorsungcu_ | ps aux | grep asterisk geeknation |
16:32.48 | lorsungcu_ | get that dynamic context thing working, CrazyTux[m] ? |
16:33.17 | geeknation | lorsungcu - when i type 'aux | grep asterisk' i get no command 'aux' found |
16:33.35 | lorsungcu_ | ps aux |
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16:35.10 | geeknation | lorsungcu - thanks -srry i didn't get it. here are my results - http://pastebin.com/ppqFqurY |
16:35.26 | CrazyTux[m] | lorsungcu I just rewrote the extensions |
16:39.00 | geeknation | all of my sound files are under these permissions and ownership: |
16:39.01 | geeknation | -rwxr-xr-x 1 777 asterisk 9174 2012-08-05 17:12 vm-intro.gsm |
16:39.32 | geeknation | i'm thinking maybe it should read 'asterisk asterisk' instead of just a single 'asterisk' |
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16:54.14 | lorsungcu_ | geeknation: did you try setting asterisk:asterisk? |
16:54.24 | geeknation | yes |
16:54.26 | geeknation | with no success |
16:54.58 | geeknation | i now have all my sound files as |
16:54.59 | geeknation | -rwxrwxrwx 1 asterisk asterisk 1056 2012-08-05 17:12 vm-Work.gsm |
16:55.21 | geeknation | but I'm getting the following in my debug 'file.c:644 ast_openstream_full: File '/var/lib/asterisk/sounds/en/hello-world' does not exist in any format' |
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16:58.41 | b0ot | Anyone good with DUNDi around? |
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17:14.21 | [TK]D-Fender | geeknation, then go show us exactly where you put them, and "core show settings" <------------------------------------------ |
17:14.23 | [TK]D-Fender | ^^ |
17:14.50 | [TK]D-Fender | geeknation, give a FULL dump, including the ls -la of the folder itself. No more puny fragments. |
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17:20.49 | geeknation | drwxrwxrwx 16 asterisk asterisk 16384 2012-08-05 17:12 en |
17:21.28 | geeknation | and this is one of the files in the en folder: -rwxrwxrwx 1 asterisk asterisk 1188 2012-08-05 17:11 vm-first.gsm |
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17:29.17 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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17:31.34 | lorsungcu_ | geeknation: read TKDs last message |
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17:32.58 | geeknation | sorry for fragments… will pastebin |
17:33.28 | geeknation | core show setting: http://pastebin.com/5b5sE7RJ |
17:35.02 | geeknation | ls -la of both the en directory and its contents: http://pastebin.com/NZJsQg7K |
17:35.30 | Katty | i'm having a bad day :< |
17:36.28 | lorsungcu_ | not as bad as geeknation |
17:36.39 | Katty | idk about that. |
17:36.48 | geeknation | :D |
17:37.53 | lorsungcu_ | http://www.youtube.com/watch?v=sGF6bOi1NfA |
17:38.25 | geeknation | thanks... |
17:38.29 | lorsungcu_ | geeknation: thought all files were asterisk:asterisk |
17:38.43 | geeknation | ya, been playing around with different owners |
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17:38.53 | geeknation | i can change them back right now |
17:38.57 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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17:39.43 | lorsungcu_ | can you check permissions of /var/lib/asterisk and its subdir? |
17:39.46 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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17:42.53 | geeknation | all the ls -la of directories (i also changed the chown back to asterisk:asterisk) - http://pastebin.com/A6TuARcW |
17:42.56 | geeknation | thanks for helping |
17:43.52 | lorsungcu_ | so now playback of a file fails with the same error |
17:43.54 | lorsungcu_ | does not exit |
17:43.56 | lorsungcu_ | exist |
17:46.06 | lorsungcu_ | (I'm asking if thats still true) |
17:46.33 | geeknation | ya. same error: WARNING[3016]: file.c:644 ast_openstream_full: File '/var/lib/asterisk/sounds/en/hello-world' does not exist in any format |
17:47.28 | lorsungcu_ | you core reload? |
17:47.48 | dwayne | anyone try grandstream phones lately and want to comment on their quality? |
17:47.57 | lorsungcu_ | not a fan |
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17:48.33 | jofry | hi folks |
17:48.39 | lorsungcu_ | hi |
17:48.43 | geeknation | yah - just tried core reload - same error |
17:49.05 | dwayne | lorsungcu, thanks. Same here, but I haven't tried them in years and I was just in a meeting w/ someone who wants to recommend them |
17:50.39 | dwayne | we are probably going to get a couple grandstreams, digium, and snoms to compare against polycoms. Anyone recommend any other phones? |
17:51.13 | lorsungcu_ | depends on what you need |
17:51.59 | dwayne | sip, not many line keys. for call center agents that don't want to use a softphone |
17:52.27 | [TK]D-Fender | dwayne, What do you need? |
17:52.49 | [TK]D-Fender | "Call center" can scale different ways. |
17:56.11 | [TK]D-Fender | dwayne, My call center people got top of the ilne Aastra 6739i's with Plantronic CS500 series headsets with EHS cables |
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17:56.21 | dwayne | [TK]D-Fender, yeah, we have a hosted platform and have a few thousand agents logged in at any time for call centers of various sizes. |
17:56.59 | [TK]D-Fender | Live queue stats on the phone, tons of coloured BLF speed-dials for their coowrkers to know who's on the line, SD's for login/out, etc |
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17:58.14 | dwayne | [TK]D-Fender, thanks I'll add Aastra to the list of phones to compare. I forgot about them. |
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18:20.07 | Ice_Strike | Is it possible to return data back from AGI to dailplan? |
18:20.52 | citrusfizz | using asterisk 1.8 is it possible to have multiple sip phones register under the same extension? and ring all phones properly? |
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18:22.33 | [TK]D-Fender | Ice_Strike, End your AGI |
18:22.56 | [TK]D-Fender | citrusfizz, No |
18:25.31 | citrusfizz | i'm no dev by any means, but you'd think more people would want that functionality than just me |
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18:25.43 | *** mode/#asterisk [+o newtonr] by ChanServ |
18:25.51 | lorsungcu | citrusfizz: what are you trying to do? |
18:26.09 | [TK]D-Fender | citrusfizz, We do. It's NOT happening. |
18:26.34 | [TK]D-Fender | lorsungcu, he JUST told us what he wants to do. 4 lines ago. |
18:27.39 | citrusfizz | must be pretty complex or some kinda security issue for it to NOT ever happen. |
18:27.58 | citrusfizz | oh well, i will find another way around it. |
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18:31.06 | [TK]D-Fender | set up peers for each of them. have the reg to their respective peers. DIAL() them together. |
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18:36.30 | Qwell | citrusfizz: The point of registering is to tell the system where you are. |
18:36.56 | Qwell | Registering from multiple places simultaneously makes no sense. |
18:37.07 | geeknation | Setting up a new Amazon instance to see what permissions asterisk defaults to on sounds/en |
18:37.27 | citrusfizz | Qwell: yeah but couldn't asterisk store multiple IP addresses and just ring all and send data to which ever responds to a pickup? |
18:37.45 | citrusfizz | Qwell: theoretically i mean |
18:38.11 | Qwell | No, that would be silly. |
18:39.48 | citrusfizz | good counter point |
18:39.56 | citrusfizz | lol |
18:40.02 | Qwell | You have a screwdriver. You don't need a hammer. |
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18:41.55 | citrusfizz | clearly |
18:42.52 | [TK]D-Fender | It isn't "silly", and does have it's place. Other SIP proxies & registrar's handle this |
18:43.10 | [TK]D-Fender | However "generally unnecessary" about sums it up |
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19:09.45 | VultureZ | I am trying to setup Realtime to MySQL on Asterisk 10, does that now require the use of ODBC to connect to MySQL instead of app_mysql? If so is ODBC built automatically or do I need to manually build that? |
19:10.55 | jpsharp | You can use ODBC or native MySQL, but to do native mysql, you have to install the addons packages to get it. |
19:11.39 | VultureZ | Okay, strange I installed the app_mysql (depricated) and I am still getting the error " Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available" |
19:12.15 | VultureZ | I do have extconfig.conf and res_mysql.conf configured, I will post the config on pastebin real quick |
19:13.18 | VultureZ | http://pastebin.com/8qV8xdEM |
19:13.39 | VultureZ | pretty streight forward, don't understand why the errors. This is the first time in a while I have attempted to use Realtime |
19:14.05 | VultureZ | oh wow... |
19:14.14 | VultureZ | /etc/asterisk/res_config_mysql.conf is the new editing file.... |
19:14.25 | b0ot | have DUNDi working!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
19:14.28 | VultureZ | well thanks for the ear jpsharp :P |
19:14.42 | jpsharp | No worries. That's all it takes sometimes. |
19:15.07 | b0ot | ahhhhhhhhhhhhhhhhhhhhhhh. nope. apply hadn't fully happened |
19:15.10 | b0ot | still broken |
19:15.25 | mjordan | VultureZ: you need res_config_mysql |
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19:18.53 | VultureZ | still giving the same error... |
19:20.04 | mjordan | k. Regardless, the res_config_mysql module provides the realtime engine, not app_mysql. |
19:22.16 | VultureZ | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
19:22.25 | VultureZ | According to this it is res_mysql.conf in 1.6.x |
19:23.10 | Qwell | voip-info.org is wrong about something? Shocking. |
19:23.15 | VultureZ | lol |
19:23.44 | VultureZ | on the confluence wiki I was unable to find configuration information for Realtime 10, only information on the app |
19:25.44 | b0ot | keys need special permissions/ownership? |
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19:27.49 | VultureZ | According to https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration it is res_mysql.conf not res_config_mysql.conf |
19:27.59 | VultureZ | unless the driver is called "config_mysql" |
19:30.04 | lorsungcu | dammit i am so sunburnt :/ |
19:31.04 | drmessano | Configure Asterisk inside |
19:31.13 | lorsungcu | i guess. |
19:31.13 | b0ot | "ALL CIRCUITS ARE BUSY" |
19:31.16 | b0ot | now |
19:31.18 | b0ot | sorry caps |
19:32.29 | lorsungcu | b0ot: you've really got to paste bin cli captures or something |
19:32.34 | lorsungcu | that could be absolutely anything |
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19:34.21 | VultureZ | okay, needed to build in support for res_config_mysql for realtime, which wasn't so that was my issue, fyi |
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19:37.22 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com) |
19:37.52 | Sean-Der | Hey does anyone know of a web based front end for CEL? I am gonna start to write one this weekend if there is not |
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19:38.44 | Sean-Der | Just something really basic in PHP and a few nice things with jQuery just for the hell of it |
19:39.04 | lorsungcu | sounds like a good idea |
19:39.22 | VultureZ | Sean-Der, you should consider node if you are looking to display it in realtime |
19:39.24 | lorsungcu | i've been meaning to do something similar, but a bit more involved |
19:39.59 | din3sh | do i need kamailio as sip server with asterisk as dialer for a system of 600extensions and 300concurrent calls? |
19:40.12 | VultureZ | we have been doing pretty impessive stuff with NodeJS here for quick development. Especially where you need realtime data on the screen or multiple people viewing the same thing (collaborative) |
19:40.43 | Sean-Der | VultureZ: I am worried that if I move away from just a few PHP scripts it will scare people away |
19:40.48 | din3sh | or can a single 2 x quad , 12gb ram memory server handle 600 registrations and 300 concurrent calls? |
19:40.59 | Qwell | din3sh: it depends |
19:41.03 | Sean-Der | VultureZ: I have been meaning to look at socket.io though |
19:41.17 | din3sh | @Qwell: depends on? |
19:41.21 | b0ot | [2004-09-23 07:13:05] NOTICE[3352]: pbx_dundi.c:1326 update_key: No such key 'box3' for signing RSA encrypted shared key for '00:0b:97:be:d2:ac'! |
19:41.22 | Qwell | din3sh: Everything. |
19:41.23 | VultureZ | ah true, NodeJS is so new, people don't want to start installing the whole package on their servers when they are already running apache/mysql/php |
19:41.40 | din3sh | no transcoding , g711 calls |
19:41.53 | b0ot | [root@freepbx asterisk]# cd /var/lib/asterisk/keys/ |
19:41.54 | b0ot | [root@freepbx keys]# ls |
19:41.54 | b0ot | 10.1.2.100.key box2.key box3.pub priv.pub |
19:41.54 | b0ot | 10.1.2.100.pub box2.pub priv.key |
19:41.56 | Sean-Der | VultureZ: great now I wanna play with new tech because of you -_- haha |
19:42.04 | b0ot | it has box3.pub |
19:42.16 | b0ot | why is it saying no such key? |
19:42.19 | VultureZ | :) we just wrote a media streamer that takes the agen't screens a records them into the system for QA purposes |
19:42.41 | drmessano | b0ot: Check permissions on the keys |
19:42.49 | VultureZ | combines the call file from * into the video recording and then strips out the "Privacy" areas |
19:42.58 | b0ot | drmessano, I made them +x |
19:43.03 | drmessano | b0ot: Check permissions on the keys |
19:43.09 | drmessano | +x is meaningless |
19:43.15 | Sean-Der | I think I am just gonna do the standard Ajax endpoint style, it may not be as clean as your idea, but it is more common. |
19:43.18 | VultureZ | Sean-Der, I would recommend using express too for setting up the UI |
19:43.20 | Sean-Der | VultureZ: thats pretty cool |
19:43.32 | b0ot | drmessano, chmod +x gives them the ability to execute |
19:43.44 | din3sh | @Qwell: the server will have a quad E1 , 300 concurrent calls (g711), 600 sip registrations |
19:43.46 | VultureZ | All the large scale Queue guys do video/screen caps so it became a requirement |
19:43.51 | drmessano | They dont need to be executed.. they need to be read.. and ASTERISK needs to be able to read them |
19:44.00 | b0ot | drmessano, -rwxrwxr-x 1 asterisk asterisk 272 Sep 23 06:40 box3.pub |
19:44.01 | drmessano | Christ.. Primer on +x.. really? |
19:44.14 | din3sh | should i keep all on the same box or split and use kamailio in front? |
19:45.28 | din3sh | if one box, i was thinking about HP dl380, 2 x quad processor, 16 gb memory |
19:45.46 | drmessano | b0ot: http://www.asteriskguru.com/archives/image-vp68776.html |
19:45.46 | Sean-Der | VultureZ: Well the good thing for me is all my CEL is dumped into one DB via odbc connection so I could just have a random Debian wheezy box and do this on |
19:46.02 | Sean-Der | idk though. Thanks for the advice though! I am gonna pull this stuff in and play with it |
19:46.21 | VultureZ | Sean-Der, are you looking for it to just parse from multiple servers and feed back to a console for your review? |
19:47.12 | din3sh | Sean-Der: have you been able to use CEL for proper billing purposes? (tracking attended xfers) |
19:47.25 | Sean-Der | VultureZ: All of my servers dump into one central MySQL table. So I don't have to worry about that |
19:47.59 | Sean-Der | din3sh: I wrote a system that a couple hotels use for their service staff. Its a whole stack solution. The reason I use CEL is so I can see the dialplan flow |
19:48.04 | VultureZ | okay then you just output that to your php app/script is your plan? |
19:48.23 | Sean-Der | din3sh: So in short sorry I don't know :/ |
19:48.26 | VultureZ | nice, did you write a UI for them too? |
19:48.35 | b0ot | drmessano, I didn't add the key extensions |
19:48.35 | din3sh | lol nvm |
19:48.36 | din3sh | :) |
19:49.09 | Sean-Der | VultureZ: Yep. I got picked up by a Microsoft shop that had a shitty non-free solution and we just decided to rewrite everything from scratch! |
19:50.08 | Sean-Der | I am really happy with how it turned out :) Most of the real work is me accessing APIs of hardware based stuff that the hotel has from us (water pressure, alarms etc...) |
19:50.10 | VultureZ | so sad, there is so much effort on making good UI and they are all closed source... so is ours... |
19:50.32 | VultureZ | its just sad that we can't put together a great UI... FreePBX tried but man that thing is full of holes |
19:50.34 | Sean-Der | VultureZ: I am hoping to make this CEL thing GPL3, but yea everything else is closed :( |
19:50.40 | drmessano | b0ot: I am not google.. Google for the error, and see if you can find something that matches your scenario. This helplessness is getting old. |
19:51.01 | VultureZ | google helps those who help themselves :P |
19:51.15 | drmessano | Indeed! |
19:51.18 | drmessano | Preach! |
19:51.44 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
19:51.46 | Sean-Der | Or just change your nick and ask again :) haha |
19:51.50 | drmessano | LOL |
19:51.54 | din3sh | google helps those who help themselves! amen to that |
19:52.26 | drmessano | so /nick something else, up arrow twice, then enter? |
19:52.26 | drmessano | Brilliant |
19:52.26 | Sean-Der | I swear that is what happens in the general programming channels in #php and #python it is so bad |
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19:53.20 | jaytee | I'm gonna start saving the IRC chat logs and run a script to count the average number of weekly occurences where someone asks about Asterisk for Windows. |
19:53.22 | drmessano | It all equal PDIFM, via proxy .. |
19:54.03 | VultureZ | I would bet a lot in PHP since you have such a mix of coders. PHP is so open it attracts all different crowds. Too bad the big guys always scoff at PHP |
19:54.28 | VultureZ | then cry when they wonder why their Java project is still in QA 2 years after the deadline :) |
19:54.41 | drmessano | The big guys scoff at PHP because the world is being taken over from below by LAMP |
19:55.17 | Sean-Der | Its also annoying how people pull the whole 'lolphp' garbage |
19:55.36 | Sean-Der | If you are gonna bitch about something, make your own damn DSL for the web. |
19:56.03 | drmessano | PHP and perl are the languages of the nerdy nerd nerds who we really want to hire because we had the first losing quarter in our history, but yeah, they suck |
19:56.17 | jaytee | ain't gonna bitch about Java but I will bitch about Starbucks. That coffee is bitter, overpriced swill! They should rename the company Fourbucks. |
19:56.32 | VultureZ | hahaha |
19:57.07 | drmessano | Microsoft Linux 2015 <--- It's coming |
19:57.17 | lorsungcu | nice. |
19:57.18 | jaytee | :-) |
19:57.20 | drmessano | Actually, it will have a cooler name |
19:57.20 | VultureZ | no joke MS jumped on the NodeJS bandwagon too |
19:57.26 | Sean-Der | YEAR OF LINUX ON THE DESKTOP |
19:57.30 | Sean-Der | ITS FINALLY HERE |
19:57.44 | jaytee | and the paperless office is right behind it! |
19:58.18 | VultureZ | I actually like Windows for my environment, I have no problem with it. I just hate the kernel they chose.. and some of their user enforcement policies they have added over the years |
19:58.19 | Sean-Der | Netcraft has confirmed it :D I am so glad I ditched Windows though. When I see people using PuTTY and Cygwin it makes me cringe |
19:58.24 | drmessano | Like "Microsoft Windows 9 Open Edition" |
19:58.25 | Sean-Der | I feel bad for them |
19:59.00 | _Corey_ | Sean-Der: Don't. They had every opportunity not to screw themselves over |
19:59.02 | Sean-Der | VultureZ: Its such a pain to dev in! You cant beat vim+gcc, I love working on low level stuff it just feels so simple and pure |
19:59.02 | VultureZ | I use Putty so I can look cool and have green text flying by my screen at starbucks... with my $4 coffee |
19:59.07 | drmessano | "Windows Clarity" |
19:59.39 | VultureZ | drmessano, sounds like a gastro pub |
20:00.10 | drmessano | Microsoft believes they can code the entire ecosystem.. Except now, they kinda don't, and also ... Windows 8 won't be $299.99, but like 40 bucks.. So yeah, umm |
20:00.58 | VultureZ | I don't have a lot of hope for Windows 8 |
20:01.06 | drmessano | Windows 9 will be $19.99, sold on TV, and include an Ubuntu CD "in case you don't like what we did there" |
20:01.17 | VultureZ | it is cursed by the "mid" release that Microsoft is plauged by |
20:01.24 | drmessano | Yep |
20:01.44 | navaismo | Anyone here use the googleTTS.agi? |
20:01.47 | drmessano | Windows 8 Vista Millenium edition |
20:01.50 | VultureZ | Win 95 - (Win ME) - Win XP - (Win Vista) - Win 7 - (Win 8) |
20:01.53 | [TK]D-Fender | Also known as the "Odd Numbered Start Trek Movie Curse" |
20:01.58 | [TK]D-Fender | Star* |
20:01.59 | VultureZ | they should just not even try on those releases |
20:02.06 | VultureZ | hahaha |
20:02.26 | VultureZ | drmessano, ROFL |
20:02.30 | drmessano | The odd numbered Rocky movie curse |
20:02.30 | jaytee | I feel bad for all the devs who got sucked into .NET and Silverlight and now it's looking more and more like .NET and Silverlight are gonna be phased out in favor of C++ again and HTML5 |
20:02.42 | drmessano | Also, the odd numbered Police Academy Curse |
20:02.44 | VultureZ | oh gawd Silverlight... |
20:02.57 | drmessano | Silverlight! |
20:03.05 | Hive | The every one after the first Starship troopers curse |
20:03.15 | drmessano | Silverlight is going to be on EVERYONE.. except for, well, everything |
20:03.20 | jaytee | Matrix 2 and Matrix 3 |
20:03.35 | drmessano | Die Hard past 2 |
20:04.11 | jaytee | only part I liked about Matrix 2 was the Merovingian when he said, "I love the French language. It's excellent to curse in. Like wiping your ass with silk!" |
20:04.30 | jaytee | but yeah, all the odd number Trek movies sucked |
20:04.45 | jpsharp | Starship Troopers made me want to chew out my own brain. |
20:05.09 | drmessano | Windows 8 is only going to suck because it will be the last non-linux version. COME ON NOW, I CAN HOPE |
20:05.22 | jaytee | the novel by Heinlein is a good read but the movie sucked. |
20:05.22 | jpsharp | 2013 will be the year of Linux on the desktop? |
20:05.46 | drmessano | jpsharp: ${YEAR} + 1 is the year of the Linux desktop!!! |
20:05.59 | Sean-Der | Every year is the year of Linux on the desktop! |
20:06.02 | jaytee | Microsoft will call theirs WinNix |
20:06.08 | drmessano | lol |
20:06.09 | Sean-Der | We are almost at 2% on some websites.... |
20:06.17 | Sean-Der | Microsoft better watch out |
20:06.36 | drmessano | Windows X <--- There we go |
20:06.38 | jpsharp | But then again, any movie that has Denise Richards in...makes me want to hit her with a baseball bat. |
20:07.16 | b0ot | works now |
20:07.17 | b0ot | nbd |
20:07.21 | drmessano | Windows X.. Feature Wine for legacy emulation.. "Because Wine does Win32 better than we do. Fuck" |
20:07.50 | Sean-Der | I am so glad I never touched the Windows ecosystems. It scares me how are win devs work. Hey we found this dll on this random forumn and it fixed the problem 0_0 |
20:07.57 | Sean-Der | our* |
20:07.58 | lorsungcu | drmessano: wine is not an emulator :/ |
20:08.10 | drmessano | True |
20:08.15 | drmessano | Technically, it's not |
20:08.28 | jaytee | MS is also dropping the Metro moniker for Win 8 supposedly due to trademark infringement. Until I read it I'd completely forgotten about Lotus Development Corp's Metro application back in the late 80's. |
20:08.44 | lorsungcu | they replaced metro with something like |
20:08.53 | lorsungcu | windows 8 new user interface version 1.0 |
20:08.59 | lorsungcu | or something equally ridiculous |
20:09.00 | [TK]D-Fender | Wine has been deprecated in favour of Bitch |
20:09.09 | drmessano | They are dropping Metro from it because metro reminds you of fast moving subway cars.. and well... |
20:09.10 | Qwell | [TK]D-Fender: I was hoping for moan |
20:09.30 | [TK]D-Fender | Finger is aplugin for that already |
20:09.40 | drmessano | Wine is being replaced by Cheese |
20:09.42 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
20:09.43 | VultureZ | drmessano, its true just 10 years ago no one would have thought IE would have been removed from the top of the list and replaced with FF/Chrome |
20:09.56 | drmessano | Yeah |
20:09.57 | jaytee | I hate IE9 |
20:10.11 | drmessano | I feel like I am just too close to love you |
20:10.17 | drmessano | So IE9, uninstall |
20:10.39 | drmessano | I hate those commercials |
20:10.54 | VultureZ | can you uninsteall IE on US versions of Win7? |
20:10.58 | VultureZ | *uninstall |
20:10.59 | drmessano | "We now have an AWESOME BROWSER that looks like Chrome, which we know you love.. and we are cool now." |
20:11.03 | lorsungcu | yeah seems like many people are saying ie9 is good |
20:11.06 | lorsungcu | i just don't see it |
20:11.11 | Sean-Der | drmessano: It was a bait and hook for me lol |
20:11.14 | drmessano | Reminds me of Dad wanting to take me to see Guns N Roses. No thanks |
20:11.21 | VultureZ | lol |
20:11.25 | lorsungcu | id totally see gnu with my dad |
20:11.30 | lorsungcu | gnr lel |
20:11.33 | drmessano | lol |
20:11.43 | Sean-Der | In all honesty do you THINK anyone would be convinced by those commercials |
20:11.57 | drmessano | My Dad would be asking random people for "grass"... and then someone would remind him they haven't called it that in 30 years |
20:12.52 | jaytee | Anyone who pays attention to commercials of ANY kind is deluded. Have you ever in your life had a burger from any fast food franchise that even remotely resembled the picture of said burger in the commercial? |
20:12.56 | drmessano | I think Windows Update forcing a browser upgrade is all the convincing most people are going to get.. The rest is penis waving. |
20:13.20 | VultureZ | jaytee, tried... but it was made of plastic and wax... |
20:13.25 | jaytee | lol |
20:13.32 | drmessano | Really.. "WE MUST DOWNLOAD IE9 NOW BECAUSE ZOMG THE BOOKFACE LOOKS SO MUCH CLEANER" |
20:13.33 | Katty | hi john |
20:13.37 | drmessano | Uh no |
20:13.49 | Katty | hi danny |
20:13.49 | VultureZ | BookFace! |
20:13.57 | VultureZ | aka FaceHarvester |
20:13.59 | drmessano | Hi Angiebell |
20:14.28 | Sean-Der | aka Company that is dying on the public market |
20:14.37 | VultureZ | Its a sad day when a trust a company who's goal is online ads more than a company who "goal" is to bring people together |
20:14.38 | Katty | there was an article on reddit that said some therapists will rate people as not having a fb as having pyschopathic tendencies |
20:14.46 | jaytee | I used to be into photography and I read about all the tricks that food photographers use like using a mist sprayer of light oil on hamburger buns to make the shine a little. |
20:15.05 | drmessano | Social networking is a great way to not have to be social or network at all |
20:15.06 | Katty | jaytee: there's actually a mcdonalds video on youtube about exactly how they do that. |
20:15.11 | Sean-Der | Katty: I saw that also... but I wouldn't deny that I am social outcast either haha |
20:15.17 | Katty | jaytee: it's very straight forward on how they do that as well. |
20:15.23 | drmessano | I am starting a new website called "Outside" where people go and network |
20:15.31 | jaytee | meet ya there |
20:15.36 | lorsungcu | how many people here have never had a fb account? |
20:15.38 | Katty | Sean-Der: a lot of people don't have fb accounts, and they're perfectly normal. |
20:15.49 | Sean-Der | Thats what you think.... |
20:15.56 | Katty | my mother is normal. |
20:16.00 | Katty | very, very normal. |
20:16.03 | lorsungcu | i am normal :> |
20:16.04 | Qwell | lots of psychopaths using facebook too |
20:16.10 | Katty | Qwell: true story |
20:16.10 | Qwell | like me. |
20:16.15 | Sean-Der | I haven't had a bookface, yourspace and what ever and I am fine without it |
20:16.16 | VultureZ | obviously she is pyschopathic |
20:16.30 | Sean-Der | My girlfriend wastes so much time on that stuff |
20:16.37 | VultureZ | along with most social outcasts (developers) |
20:16.42 | jaytee | I don't believe normal exists |
20:16.51 | drmessano | I didn't have a Facebook account until I started killing kittens and stapling pieces of paper with captions to their heads. Now, I have 120 friends and captioned kittens on my FB wall. |
20:16.55 | VultureZ | isn't normal what you precieve as the baseline? |
20:17.22 | VultureZ | drmessano, well that is perfectly normal since you do it on Facebook |
20:17.26 | jpsharp | Sean-Der: My stepmother created half a dozen accounts so she could play all the gambling games. |
20:17.28 | drmessano | Absolutely |
20:17.28 | Sean-Der | I live in the United States.... I am ok not being 'normal' by my countries standard |
20:17.49 | Sean-Der | jpsharp: That makes me believe that you humanity is gonna be alright :') |
20:18.09 | jpsharp | Accounts for her, my dad, each of the two dogs, their cat, and i think a fish. |
20:18.09 | VultureZ | why do you need 6 accounts to gamble on FB? |
20:18.17 | VultureZ | oh they get free $$? |
20:18.20 | Sean-Der | HAHAHAHAH |
20:18.23 | Sean-Der | That is awesome |
20:18.44 | VultureZ | the more "noise" FB gets the more people will dislike it... go go gadget chinese bot army |
20:18.46 | [TK]D-Fender | "Think of how stupid the average person is, and realize half of them are stupider than that." - George Carlin |
20:18.53 | VultureZ | ;) |
20:19.04 | jaytee | I miss George |
20:19.08 | Sean-Der | Carlin and Hicks are gods among men |
20:19.09 | jaytee | I love his take on God |
20:19.33 | [TK]D-Fender | Sean-Der, And if they're wrong ... men amongst God ;) |
20:19.49 | drmessano | Online gambling.. Jeez. I almost had a coup when I blew out Windows on my Dads machine and installed Ubuntu and found one of his gambling sites wouldn't work because their browser malware only ran on IE. |
20:20.10 | jpsharp | I played MafiaWars for about a week when it all first came out. Then I felt like a truckstop hooker whoring myself out for energy points or whatever they used in the game. |
20:20.16 | jaytee | "An invisible man who lives in the sky and there are 10 things he doesn't want you to do and if you do any of them you'll be cast into a lake of fire for all eternity.....but He loves you! He loves you and He needs your money!" |
20:20.19 | jpsharp | Never doing that again. |
20:20.27 | drmessano | GamblingPlugin 1.0 by FreeSmileysAndSpam4All <-- Sounds legit to me |
20:20.33 | Sean-Der | [TK]D-Fender: Hahah that is awesome. |
20:21.17 | Katty | Sean-Der: i waste a lot of time on fb too. |
20:21.19 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
20:21.23 | Katty | Sean-Der: some people are just more sociable than others. |
20:21.29 | geeknation | I'm hitting an error on require 'ruby-agi' in irb |
20:21.31 | geeknation | |
20:21.38 | Katty | Sean-Der: i don't really consider it a 'waste' though. i very much enjoy it |
20:22.03 | lorsungcu | anyone have a good method for checking if a file exists before playing it? |
20:22.34 | [TK]D-Fender | "core show function STAT" |
20:22.36 | [TK]D-Fender | ^ |
20:23.08 | lorsungcu | thanks |
20:23.40 | b0ot | it was the keys |
20:23.41 | b0ot | all along |
20:23.55 | Sean-Der | Katty: I am sociable also, but I don't enjoy small talk. I zone off when people talk about boring stuff. That is what FB is most of the time. |
20:23.57 | [TK]D-Fender | checkout time, BBL |
20:24.05 | lorsungcu | any idea on how much cpu/io it takes to use the function, TK? |
20:24.09 | lorsungcu | dammit. |
20:24.11 | lorsungcu | anyone? |
20:24.55 | lorsungcu | customer is worried that checking whether a file exists will destroy their disk bandwidth |
20:25.11 | lorsungcu | i don't really have any evidence that it won't, except that there are so few checks, relatively |
20:25.12 | b0ot | what are you scripting in? |
20:25.16 | lorsungcu | dialplan :p |
20:25.18 | Sean-Der | STAT is not that expensive, don't know how Asterisk implements it though |
20:26.57 | b0ot | lorsungcu, http://www.voip-info.org/wiki/view/Asterisk+tips+fileexistance |
20:27.05 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
20:27.12 | lorsungcu | thanks |
20:27.38 | lorsungcu | yeah I'm using a system() call now. |
20:27.53 | b0ot | http://www.voip-info.org/wiki/index.php?page_id=2981&comments_page=1 |
20:34.27 | thecardsmith | lorsungcu: I'm guessing that stat is not going to be the bottle neck, at scale... it's going to be reading those files that you're playing off the disc |
20:34.46 | thecardsmith | notes that this is gut instinct |
20:34.48 | lorsungcu | yeah its only checking ~50 files |
20:36.11 | thecardsmith | how often is it changing that they could or couldn't be there? |
20:36.42 | thecardsmith | cause, you could check 'em and put it in astDB, every... say "now and again". Instead of say, "with each and every call" |
20:37.00 | *** join/#asterisk jakent (~jakent@ip-64-134-71-83.public.wayport.net) |
20:37.22 | thecardsmith | but, i don't know what other requirements are tied to the existance/non-existance of those files |
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20:49.46 | VultureZ | Realtime, why do you forsaken me! |
20:54.14 | *** join/#asterisk jakent (~jakent@ip-64-134-71-83.public.wayport.net) |
20:54.33 | lorsungcu | I'm working on a project now that i wish i had done using realtime |
20:54.35 | lorsungcu | suuuucks |
20:55.18 | jpsharp | I did that years ago. I constantly had to regenerate sip.conf and extensions.conf. |
20:55.36 | lorsungcu | what do you mean? |
20:56.32 | jpsharp | I built it using static entries in sip.conf & extensions.conf, but then when the number of phones went past 15 and I was adding a few each day, it became a clusterflop. |
20:56.46 | lorsungcu | ah |
20:57.58 | jpsharp | Took entries out of a database, regenerated sip.conf, extensions.conf, and TFTP config files for 600 Cisco 7940s. |
20:58.09 | lorsungcu | ick |
20:58.58 | VultureZ | well the extensions is still going to be .conf but queues/vm/peers are all moving to realtime |
20:59.48 | jpsharp | I finally broke down and converted *everything* to a database back end. Asterisk, BIND, TACACS, and RADIUS. |
20:59.53 | jpsharp | That was a long week. |
21:00.02 | lorsungcu | yeah i have a feeling that will be happening to me. |
21:00.03 | VultureZ | I bet |
21:00.18 | VultureZ | though its also a good feeling it means the biz is growing :) |
21:00.32 | lorsungcu | i suppose |
21:00.36 | lorsungcu | just wish there were 3 of me |
21:00.37 | lorsungcu | heh |
21:00.41 | Hive | lol |
21:00.51 | jpsharp | I quit two months and 5 bounced paychecks later. |
21:00.57 | lorsungcu | :p |
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22:40.55 | geeknation | Hey -- Anybody know why Amazon might block agi.wait_for_digit |
22:41.39 | [TK]D-Fender | ... pardon? |
22:43.28 | geeknation | Sorry. I've got a ruby agi script that uses wait_for_digit and Asterisk is waiting for something, but it doesn't read any DTMF input |
22:43.52 | [TK]D-Fender | And you have just claimed AMAZON is "blocking" it...... |
22:44.08 | geeknation | Exactly the same script worked fine on Rackspace |
22:44.43 | geeknation | it might not be amazon…just kinda new… wanted to get your opinion on what it might be |
22:44.50 | [TK]D-Fender | maybe * versions aren't the same. Maybe that script depends on something that isn't there on both environments. Maybe a dependency isn't reacting the same as it did on the other |
22:45.39 | [TK]D-Fender | Since you haven't shown us anything, there is no opinion to give except awe at the one leap that was presented. |
22:47.04 | geeknation | excuse me |
22:47.06 | geeknation | lets start over |
22:47.29 | geeknation | I'm running a ruby-agi script via my dial-plan and am looking to wait indefinitely for digits |
22:47.48 | geeknation | when i call into my number everything goes through fine, but my key presses aren't going into the system |
22:47.51 | [TK]D-Fender | Ok, if we're starting over.... first SHOW, then "tell" |
22:47.54 | [TK]D-Fender | ~pb |
22:47.54 | 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. |
22:47.55 | geeknation | my agi debug is : http://pastebin.com/Ax8SA51w |
22:47.56 | [TK]D-Fender | ^^^^^ |
22:48.22 | [TK]D-Fender | show us what version of *, show us the script. Show us your calling it with AGI debug enabled and verbose 10. |
22:48.54 | [TK]D-Fender | Show us the call that validates that any DTMF work OUTSIDE of that AGI in the first place. |
22:51.18 | [TK]D-Fender | [Aug 8 22:36:04] WARNING[10164]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission 03e8c05020e7177b73f5fe85146caa27@66.54.140.46 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display AST/SIP+Retransmissions |
22:51.20 | [TK]D-Fender | Packet timed out after 32000ms with no response |
22:51.22 | [TK]D-Fender | [Aug 8 22:36:04] WARNING[10164]: chan_sip.c:3580 retrans_pkt: Hanging up call 03e8c05020e7177b73f5fe85146caa27@66.54.140.46 - no reply to our critical packet (see htt |
22:51.27 | [TK]D-Fender | Your entire call is getting packet timeouts |
22:51.32 | [TK]D-Fender | It is screwed |
22:51.51 | [TK]D-Fender | this has nothing to do with call processing.. you can't even keep the packets for a CALL up in any capacity |
22:52.22 | [TK]D-Fender | You have netowrking issues |
22:54.00 | geeknation | here's my agi-script - sorry for the delay: http://pastebin.com/1yHuxKAw |
22:54.05 | geeknation | what do you mean networking issues? |
22:56.04 | *** join/#asterisk cyborg-one (1000@212-178-0-89.broadband.tenet.odessa.ua) |
22:56.08 | lorsungcu | geeknation: have you read https://wiki.asterisk.org/wiki/display AST/SIP+Retransmissions ? |
22:57.14 | [TK]D-Fender | geeknation: * can't reliably talk to ipkall |
22:57.25 | [TK]D-Fender | Packets. Aren't. Making. It. |
22:58.55 | geeknation | i've glanced over the post, but i'll read it more thoroughly now. [TK]D-Fender - thanks for the insight… Do you have any suggestions to making a reliable connection to ipKall? |
22:59.24 | lorsungcu | check the firewall on your instance |
23:03.03 | geeknation | ahhhh…thanks for the tip. I've already opened ports 0 - 65535 (UDP). and specifically the TCP port 22 (ssh), and 80(HTTP). Is there a specific TCP port I also need to open to receive DTMF? Port 5060 or 5061 perhaps? |
23:05.17 | [TK]D-Fender | this isn't just DTMF |
23:05.38 | [TK]D-Fender | prove what is and is not being blocked through the ENTIRE path |
23:05.51 | [TK]D-Fender | double check EVERY setting you have any direct influence over |
23:06.52 | geeknation | ok. i can check all ports, but i'm not sure what i'm looking to open in order to get around the potential firewall issue |
23:10.06 | *** join/#asterisk General_Z0d (~z0d@184.178.240.121) |
23:10.20 | [TK]D-Fender | Verify if you left any SI setting in place from old configs like externIP, etc.... |
23:10.24 | General_Z0d | leifmadsen, what is the link to your tutorial again i forgot |
23:10.26 | [TK]D-Fender | anything that might not add up" |
23:15.17 | [TK]D-Fender | ~book |
23:15.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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23:49.57 | General_Z0d | is away: taking a small break then i will resume |
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23:51.16 | geeknation | lorsungcu Thanks all for the advice. It is an IPKall issue as my sip gate number is going through no problem |
23:55.02 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |