IRC log for #asterisk on 20120808

00:03.34*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
00:05.43*** join/#asterisk zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189)
00:10.23*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
00:14.00zopsiHas anyone setup a spectralink 8020 for use with asterisk?
00:14.18*** join/#asterisk ChannelZ (channelz@burner.com)
00:14.25lorsungcuzopsi
00:14.28lorsungcuyes
00:15.12zopsiI have it pulling the files from tftp server, but my config appears to be wrong. I just get 1 of 1 lines unregistered.
00:19.12*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
00:19.19*** join/#asterisk volga629 (~volga629@173.209.129.250)
00:20.54volga629Hello All, which vendor for ip phone with vpn ( but not openvpn ) recommended for asterisk ??? tnks
00:23.14wonderworldthe first Playback() in a call always stutters for me. later Playbacks are OK. any idea on how to fix that?
00:23.34*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
00:24.19jpsharpvolga629: The Cisco SPA525G support SSL VPNs.
00:27.29[TK]D-FenderSounds ike the time it takes a jitterbuffer to fully kick in.  Not much youre going to be be able to do about that.
00:31.47volga629it require ASA firewall or I can use any SSL firewall >
00:31.51volga629?
00:37.50beardywonderworld: Play a second or a few milliseconds of silence first?
00:40.56WIMPyOr use Answer() with the observed time.
00:41.03WIMPy... needed.
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01:02.54*** join/#asterisk lorsungcu__ (~anonymous@65.103.31.37)
01:03.43*** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37)
01:06.44CrazyTux[m]Any easy way to do Gotoif(X matches regexp) ?
01:11.21wonderworldthanks
01:11.57*** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net)
01:15.02wonderworlddelay of 3 secs in answer didn't fix it
01:15.04lorsungcuCrazyTux[m] https://wiki.asterisk.org/wiki/display/AST/Function_REGEX
01:15.06wonderworldtrying silence now
01:15.41wonderworldis there a premade silence recording available?
01:16.07lorsungcuwonderworld use playback(silence/<number of seconds>)
01:19.03wonderworld2 seconds of silence fix it
01:19.07wonderworldgreat
01:19.57volga629is anybody was using ATCOM AT-620P ip phone ?
01:20.50[TK]D-FenderAtcom = cheap Chinese crap
01:22.12lorsungcu_damn tk cold blooded.
01:22.22volga629Just they claim that l2tp in place
01:23.27wonderworldi once had a chinese ip phone. it had 20 pages of config in it's menu. and never worked as expected
01:24.16*** join/#asterisk djinni (~djinni@li125-242.members.linode.com)
01:25.17volga629I have android fanvil one and can't update firmware at all and no documentation :-(, looking for something like snom 370 but not openvpn
01:27.35lorsungcu_why not openvpn?
01:28.46volga629because on distribution layer we have fortinet and cisco 3560
01:29.38volga629ideal will be terminate on fortinet l2tp tunnel and put on separate vlan
01:30.24volga629I guest I can find voip gateway and regular phone ?
01:34.24volga629any suggestion about small voip gateway ?
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01:59.31*** join/#asterisk Xaviertoor (~Xaviertoo@186.213.213.10)
01:59.38XaviertoorHello people
02:00.13XaviertoorI installed asterisk 10.7.0 and comand sip no found in CLI
02:00.31XaviertoorAny ideia?
02:01.15Xaviertoorwhat's wrong?
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02:54.33*** join/#asterisk becca_r (~becca_r@12.25.151.58)
02:54.49becca_rhello, is anyone on?
02:56.04darthanubisjust throw your question
02:56.28darthanubisthe great thing about irc is that it'll be here when you step away
02:56.49darthanubisif someone can answer they will eventually
02:56.54becca_rI'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.
02:57.10becca_rAny help would be great.
02:58.17darthanubisjust keep your chat program running, and some back later. I don't have an answer for you...:(
02:59.19becca_r1.7
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03:16.04archetech<PROTECTED>
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03:33.13lorsungcubecca, you get your stuff figured out?
03:33.19becca_rno, not yet
03:33.23becca_rfound more about my issue.
03:33.31lorsungcusounds like nat
03:34.08lorsungcucan you pastebin your sip.conf?
03:34.24becca_rI see that the ack comes in with the ruri for the destination as5400 and to uri of opensips.
03:34.34becca_rthis is all nonat and same network
03:34.42lorsungcuij
03:34.43lorsungcuok
03:35.04lorsungcudo you have a sip capture of the call failing?
03:35.08becca_rI 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
03:35.11lorsungcuand can you pastebin that?
03:35.45becca_rI can pastebin opensips debug logs, and pcap
03:35.47lorsungcuis this issue on an asterisk machine, or opensips?
03:36.30becca_ropensips I do believe
03:36.44becca_rcrap, wrong channel
03:36.45becca_rsorry
03:36.48lorsungcuwell, i see you're in #opensips...
03:36.53lorsungcu:p
03:37.34becca_rthanks, sorry
03:37.35lorsungcuif there's an asterisk side to this, i'd be happy to help with that?
03:37.48becca_rwell I think the asterisk side is working fine.
03:38.01lorsungcuok
03:39.08lorsungcubrb taco pizza!@$
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03:59.33fling[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'.
04:00.22lorsungcugetting haxzed
04:00.30flinglorsungcu: hehe :p
04:02.18flinglorsungcu: 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
04:02.45flinglorsungcu: 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)
04:21.09CRCinAU1hmmm
04:21.11CRCinAU1so I'm wondering.
04:21.20CRCinAU1I have a fax machine hooked to an FXS port on a dahdi card.
04:21.22*** join/#asterisk hebber (~hebber@118.175.66.17)
04:21.48CRCinAU1and I use _X.,blah to grab what is dialed from the fax machine, then shoot it out over a SIP provider
04:22.01*** join/#asterisk jofry (~jofry@port-212-202-54-40.dynamic.qsc.de)
04:22.16jofryHi folks
04:22.34CRCinAU1Now 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)
04:23.20CRCinAU1so, I figure somehow I have to tell the Dial() to try and go for a t.38
04:23.44CRCinAU1however this doesn't seem to happen - nor can I find a reference to t38 in the dial command in asterisk 10.7
04:24.41CRCinAU1so, how do I do this, and is there any advantage to doing it?
04:24.43lorsungcuhttps://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
04:26.04CRCinAU1gateway would already assume SIP input with fax payload data though wouldn't it?
04:26.06lorsungcualso
04:26.06lorsungcuhttp://en.wikipedia.org/wiki/T.38
04:26.40CRCinAU1as its coming in from a dahdi source, wouldn't it be a T38 originator?
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04:32.34flinghow can I redirect the call?
04:33.06lorsungcuin the default context, pattern match whatever call is being made, and send it where you wanty
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04:34.16flingI 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
04:34.51CRCinAU1you want a transfer then, not a redirect.
04:35.05flingoh! ok… hmm hmm
04:35.12CRCinAU1assuming you don't use SIP phones, look in features.conf
04:35.27CRCinAU1if you use SIP phones, hopefully your SIP phone has a transfer button
04:36.19flingCRCinAU1: softphones and a single analog phone connected to dvg-7111s
04:36.37CRCinAU1urgh lol
04:36.54lorsungcuwtf would you do that to yourself lol
04:36.55CRCinAU1if its a softphone, hopefully the soft phone talks SIP and knows how to transfer
04:37.09*** part/#asterisk Bullmoose (~Bullmoose@75-174-79-252.bois.qwest.net)
04:37.18flingCRCinAU1: umm?
04:37.43lorsungcuwhat softphone
04:37.59flingekiga, zoiper…
04:38.04CRCinAU1fling: 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.
04:38.35flingwindows-lovers use 3cx-phone
04:38.58lorsungcuhttp://www.zoiper.com/downloads/Zoiper_2.0_Free_Manual.pdf
04:39.03lorsungcupage 16
04:39.03flinglorsungcu: ok
04:39.07flingCRCinAU1: thanks :p
04:39.58flingwhat about dvg-7111s? it should be builtin in the device too?
04:40.22lorsungcudoesnt need to knwop about the transfer
04:40.48CRCinAU1fling: that will probably be handed in features.conf and using the flash button on the analogue phone
04:41.13lorsungcuthe zoiper manual goes into that, actually..
04:42.19CRCinAU1actually, if you're using an ATA, the ATA will have its own feature codes
04:42.33CRCinAU1so you'll dial something like: FLASH *70#102
04:42.42CRCinAU1obviously those numbers are made up, but yeah
04:44.04flingCRCinAU1: transfer button in client works
04:44.14CRCinAU1as it should :)
04:44.15flingCRCinAU1: thanks!
04:44.36flingCRCinAU1: but flash button on analog phone is not
04:44.57CRCinAU1read the book on your ATA
04:45.03CRCinAU1it'll hopefully go into it
04:45.39CRCinAU1you might have to do some config swindling on the ATA to enable / whatever it
04:46.35flingbut what is ATA? :#
04:46.54CRCinAU1what you plug your analogue phone into that then converts it to SIP
04:47.05flingoh! ok
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05:23.06Sean-DerHey 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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05:46.09ChannelZHmmm.. 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.
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05:51.33v0lZylo
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05:57.33flingv0lZy: lo
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05:59.55v0lZyhi
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06:28.21flingThis is what I want: somebody calls my external number and if nobody answers she may press 102 and call goes to internal number 102
06:28.52v0lZyextenisons.conf
06:29.30v0lZythen bind external number to your application
06:29.36flingv0lZy: http://bpaste.net/show/39005/
06:29.42flinghmm hmm
06:30.16v0lZytheres a time option to dial btw
06:30.37v0lZythat afte a certain amount of seconds, if noone picks up, it continues the dialplan execusion
06:30.37flingcan you give me an example or a link to read?
06:30.44v0lZyso then your user doesnt need to press anything
06:30.56flingno, this is not what I want
06:31.12kaldemarfling: is it only 102 you want the call to go to?
06:31.32flingkaldemar: no, I want it to be any local number
06:31.33v0lZyah, kaldemar to the rescue
06:31.38flingv0lZy: :p
06:32.01flingexten => kaldemar,1,Hello()
06:32.26v0lZy:D
06:32.52kaldemaruse a timeout in app Dial and then use WaitExten, Read or DISA to get the new destination.
06:33.44kaldemarbut be extremely careful not to open up your box for outbound calls.
06:35.07flingI 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
06:35.47flingkaldemar: 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'.
06:37.19kaldemarthose two are not related.
06:37.58kaldemarwell, related in the way that the first enables the second to happen.
06:38.06flingkaldemar: they are trying to call but they can't while unauthorized? right?
06:39.03flingfirst is when somebody calls me and then press 972599532957 to call to Israel :P
06:39.13kaldemarfling: 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.
06:39.38flingkaldemar: is this normal?
06:39.41kaldemaror authenticate rather.
06:40.10kaldemarfling: if you leave your box open to guest calls like that, then it is normal.
06:40.17flinghmm hmm
06:40.31flingwhat file to edit to prevent this?
06:40.45kaldemarsip.conf, allowguest=no
06:41.07flingfixed! thanks
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06:46.30v0lZyman, this is a bit cryptic for me
06:46.42kaldemaryou might also want to set alwaysauthreject=yes to make bruteforcing harder.
06:46.52v0lZyim trying to reject a call forwarding setting if there is a loop
06:47.30*** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net)
06:47.54v0lZywhat im doing is using DB to create a CF/${EXTEN} and checking if its set to "" or some other extension.
06:48.25kikohnlexpecially 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
06:49.09kaldemarv0lZy: why ""? you should rather delete the value when the forwarding is disabled and use DB_EXISTS to check.
06:50.05kaldemarkikohnl: that's a time for iptables and -j REJECT or -j DROP for the address, which ever makes their script give up.
06:50.37v0lZysmarter
06:51.06v0lZywhats the syntax? DB_EXISTS(entry) ?
06:51.36kikohnldid that on my asterisk server, an old mac mini with ubuntu, but Hawaiian telcom iptv requres this crappy firewall
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06:54.57kaldemarv0lZy: "core show function ..."
06:55.34v0lZycan i show u some code
06:55.41v0lZyand u tell me if it makes sense
06:57.22kaldemarwrite 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.
06:59.49v0lZyhttp://bpaste.net/show/XGllhOvnOmthCRr6TE1M/
07:00.20v0lZyim not sure my code is correctly formated... im following some examples...
07:00.37v0lZybut overall,  does this make sense?
07:00.54v0lZymy concern is if im overlooking something here
07:00.58v0lZylogic.
07:01.09v0lZyi have ** as a prefix to call forward.
07:01.46v0lZyfirst, i check if the extension is trying to forward to itself... if it is, i skip to cancelforward label and Hangup.
07:01.55v0lZy(ill add some recoding to it later)
07:02.23v0lZyon 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.
07:02.53v0lZyI do that by storing the value in the DESTINATION_CF variable.... if there is no forwarding, it should be ""
07:03.34v0lZyif 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
07:03.56v0lZyif that number is the number im calling from, i go to tag cancel forward and hangup.
07:04.56kaldemar"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?
07:05.29v0lZyIf 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
07:06.09v0lZykaldemar: no, im concerned about making a forwarding loop involving more than 2 phones
07:06.38v0lZyA forwards to B, B forwards to C, C forwards to D, D forwards to A or B or C and i have a problem.
07:07.58v0lZyMy 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.
07:08.41v0lZyi dont have a problem with chain forwarding things.
07:08.47v0lZyas long as they dont create a loop
07:08.58v0lZyI'm checking this when the user is activating call forwarding
07:09.07v0lZyso such a loop should never be established.
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07:09.53v0lZy(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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07:10.22schmidtsgood morning
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07:10.38kaldemarv0lZy: http://pastebin.com/jGLJnjuP
07:11.14kaldemarv0lZy: you probably don't want to check all possible forwardings. at least not in dialplan.
07:14.36v0lZythe 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
07:14.55kaldemaryes.
07:15.06v0lZyit does not take into account a situation where A forwards to B and B forwards to C, and C forwards to A by mistake
07:15.09*** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au)
07:15.29v0lZy(thanks for the same => tip, handy!)
07:17.27v0lZywell.. is there any way to check for this stuff outside dialplan
07:17.38v0lZythe thing is such a chain could form at any moment
07:17.51v0lZyso it has to be prevented before its created.
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07:20.19kaldemarv0lZy: there are many ways to check it outside the dialplan. e.g. func SHELL, app System, AGI.
07:20.45v0lZywhat would be the advantage of doing this outside dialplan?
07:20.56kaldemarbetter tools for the task.
07:21.49v0lZydialplan should be able to handle this
07:21.52v0lZyor asterisk internally.
07:22.00v0lZyits not that complicated i think
07:22.09v0lZylet me test my code
07:23.57kaldemarif we enhance the first thing i pasted a bit: http://pastebin.com/ARYYWiKL
07:24.35kaldemarthat would check forwardings looped until there is no forwarding or the forward is not set to the caller.
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07:25.17kaldemarin 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.
07:27.16kaldemarsure you could use HASH and HASHKEYS functions to store and check, combined with a GoSub for example.
07:27.23kaldemarbut the tools are limited.
07:27.39v0lZyim trying to understand the code
07:28.17kaldemarfirst line checks if the forward is to the caller itself. second line checks if the destination has enabled callforwarding.
07:28.20v0lZyfirst, u check if its calling to itself then cancel... ok, makes sense.
07:28.44v0lZysecond, you establish if call forwarding is set on the destination
07:29.04v0lZyif it doesnt exist, u forward
07:29.23kaldemarif 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.
07:29.24v0lZyif it exists, you check where its forwarding to
07:29.49v0lZyand if its not forwarding to the caller, then you do forward (dangerous at this point in my opinion)
07:30.47kaldemarhence my previous comment.
07:30.56v0lZyah, sorry, didnt see
07:31.06v0lZyand whats the one with ?check
07:31.47v0lZyit checks if the destination is doing call forward all over again.. ok, makes sense
07:31.57kaldemaroops. syntax errors there i see.
07:32.07v0lZywhy n(loop) there btw?
07:32.25kaldemarit's invalid.
07:32.54kaldemarthat example is crap and would not work. business as usual. :)
07:33.41kaldemari 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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07:36.10youjellyCan anyone suggest a good ITSP for USA?
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07:41.01v0lZykaldemar, 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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07:42.30flingI need outlook integration for windows-lovers
07:42.35flingOutCall is not working
07:42.40flingwhat app do I need?
07:43.06dagbfling: Lync
07:43.29dagbfling: with the most expensive license
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07:43.42*** mode/#asterisk [+o file] by ChanServ
07:43.51flingdagb: hmm hmm
07:45.02v0lZykaldemar: 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.
07:46.48flingdagb: will it connect to sip
07:47.02flingdagb: *to asterisk over sip?
07:47.58din3shcan a single * box handle +300 concurrent calls?
07:48.07din3shalaw
07:49.44dagbfling: 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.
07:50.33flingexten => dagb,n,Hangup()
07:52.07youjellyITSP anyone????????
07:53.51kaldemardin3sh: http://www.voip-info.org/wiki/view/Asterisk+dimensioning
07:54.01kaldemar~itsp-us
07:54.23kaldemar~itsplist-us
07:54.23infobotHere 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
07:55.42youjellythanks kaldemar <3
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08:06.44danfromukHi, Ive got a complex dial plan and was wondering if its possible to do this.......... EXECIF(ChannelAnswered,Wait(1)) ?
08:07.10danfromukSorry, i meant to say ExecIf(ChannelNotAnswered,Wait(1))
08:07.53danfromukIE. If the call hasnt been answered elsewhere in the dialplan and is still ringing.
08:08.48kaldemara mere Wait is enough. if the call had been answered, there would be no dialplan execution.
08:09.38schmidtsdanfromuk maybe you are looking for the M option of the Dial application
08:10.06danfromukThese are incoming calls, so no Dial command used.
08:10.36danfromukI 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.
08:10.54kaldemarall calls are incoming.
08:12.03danfromukkaldemar, ok more specifically, no dial cmd is used.
08:12.08schmidtsdanfromuk 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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08:12.45kaldemardanfromuk: why don't you show exactly what you're doing?
08:15.21danfromukWhen 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.
08:15.45danfromukkaldemar: its a little hard to output as its in a realtime db.
08:16.02danfromukSometimes the call goes to a recorded message, then the menu.
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08:16.57danfromukBasically, 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.
08:17.00schmidtsdanfromuk then you could easy set a var when you do a playback or answer the call and just check this var later
08:17.39schmidtsor you can use the delay option of the answer application
08:17.46danfromukschmidts: thought you might say that. I was wondering if there was a channel status var that showed whether a channel was unanswered.
08:18.06schmidtsmaybe let me check it
08:18.10schmidtswhich version do you use?
08:18.13danfromuk1.8
08:21.09schmidtsok there is not really a channel var you can use but the channel state will be UP when the call is answered
08:21.32schmidtsyou could maybe use the chanisavail app with the own channel and then check the availstatus
08:21.42danfromukOk. thanks. I'll run some tests. Otherwise your variable suggestion is great.
08:21.56schmidtsforget it sorry
08:22.27schmidtsjust use the channel function like this: ExecIf($["CHANNEL(state)"="up"]?wait(1))
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08:25.24danfromukOk, i'll give that a go and see what happens.
08:25.34danfromukThanks
08:36.29danfromukschmidts: looks perfect.
08:36.33danfromukthanks again
08:39.48schmidtsyour welcome
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09:20.46v0lZyhm
09:20.48v0lZyguys
09:20.51v0lZyanything wrong with this line
09:21.27v0lZysame => n,Set(FORWARD_TO=${DB(CF/${EXTEN})})
09:21.44v0lZyit results in FORWARD_TO=
09:21.48v0lZyit doesnt set anything...
09:22.32WIMPyDoes the DB contain anything?
09:23.30v0lZyyes
09:23.47v0lZyi set it with
09:24.00WIMPyFor that key?
09:24.11v0lZyargh...
09:24.13v0lZyhold on
09:24.15v0lZymaybe it doesnt
09:24.20v0lZyim doing
09:24.41v0lZyn,Set(DB(CF/${CALLERID(number)})=${EXTEN:2}) ...
09:24.52v0lZywhich is wrong i think :D
09:25.07v0lZyshould be noop i presume?
09:25.12WIMPyAre the callerids and extensions the same?
09:25.21v0lZynož
09:25.23v0lZyno*
09:25.31v0lZycan i run DB directly
09:25.34v0lZyor must i use noop
09:25.34WIMPyAnd it's CALLERID(num).
09:25.36v0lZyor set or somethign?
09:25.42v0lZynumber works also
09:25.45v0lZydeprecated?
09:26.13WIMPyNot sure if it ever existed officially.
09:26.32WIMPythe *CLI has database ...
09:28.55v0lZyseems to be the correct syntax though
09:32.10v0lZyit just doesnt set the variable
09:33.17v0lZyeither that
09:33.20v0lZyor it is truely empty
09:33.22v0lZyi dont kno know
09:33.24v0lZybut im using
09:33.57v0lZyn,Set(DB(CF/${CALLERID(num)})=${EXTEN:2}) ...
09:37.28kaldemarv0lZy: and then you try to read ${DB(CF/${EXTEN})} to FORWARD_TO. you haven't shown anything that indicates CF/${EXTEN} having any content.
09:38.09WIMPyThat's the good old topic of using canonicialised numbers.
09:40.46v0lZykaldemar: is my syntax in my previous line correct when filling the db?
09:42.47WIMPyThe syntax is fine.
09:42.57WIMPyBut that doesn't mean the data is.
09:45.34kaldemarv0lZy: you set to CF/${CALLERID(num)} and try to read from CF/${EXTEN}.
09:46.58danfromukIs it possible to run multiple commands from one EXECIF?
09:48.05v0lZykaldemar: well what im doing is, when i call **16 for example, im setting CF/CALLERID(num) to 16
09:48.37v0lZyso that i know that the number that dialed **16 is to be redirected to 16
09:49.23v0lZywhen the call comes in, im checking the extension that was called (31) and seeing what the CF/31 is
09:49.32v0lZyhopefully it should give me 16, and use that to dial 16 instead of 31.
09:51.28WIMPydanfromuk: GosubIf?
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09:52.58kaldemarv0lZy: useless explanations. show the CLI output of a call and "database show".
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09:54.44youjellyall ITSPs offer SIP Trunks?
09:55.05youjellykaldemar:
09:55.49v0lZykaldemar: i dont see it in database show
09:56.18v0lZyjust /SIP/Registry/ stuff
09:56.24v0lZyso... its not even getting set :|
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09:56.39kladzehello, 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
09:56.39kladzeto learn asterisk :)
09:57.21kaldemaryoujelly: not necessarily. that's up to you to find out what they sell.
09:57.55plundrakladze: Does the file 1-and.ulaw exist?
09:58.10kladzeits a .gsm
09:58.36kladzebut yes the file exist
09:58.50plundraTry converting them to ulaw or do the call using gsm instead.
09:59.04plundraUsing gsm as the codec that is.
09:59.18bulkorokshouldn't asterisk convert from gsm to ulaw when necessary!?
10:00.31WIMPyCould be a digits/da vs da/digits thing.
10:02.39kladzei noticed that the english sounds is .gsm files aswell and that works.. ?
10:02.57plundraOkay. Well then a path-thing probably :)
10:03.55plundra(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)
10:04.09v0lZyAny idea why its not setting the entry into the database?
10:04.15kaldemarkladze: if you do a Set(CHANNEL(language)=da) before the playback, does it work?
10:04.28v0lZyor better yet, can u give me an example of how to set something in the databasE?
10:05.35WIMPyv0lZy: Your syntax is still correct.
10:05.50v0lZythen i dont get it...
10:05.59v0lZywhy isnt it setting?
10:06.04WIMPyApart from the num vs number thing.
10:06.09v0lZyi fixed that
10:06.12v0lZyits all (num) now
10:06.14WIMPyWhat does CALLERID(num) contain?
10:06.34kaldemarthe syntax written HERE was correct.
10:06.54v0lZylet me paste then
10:07.43v0lZyhttp://bpaste.net/show/MHmjkBCbByv457XqNNTo/
10:09.45WIMPyIt looks to me as if the line setting the DB entry would never be reaced.
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10:10.05kaldemarv0lZy: where's the CLI output?
10:10.15v0lZymoment
10:10.33WIMPyIs it supposed to be before the Goto(done)?
10:11.49v0lZyhttp://bpaste.net/show/XKraHPf0JaqkZc23s9Q1/
10:12.16v0lZyah, doh!
10:12.31WIMPySee. It never happens.
10:12.49v0lZynot before
10:12.53v0lZyits supposed to be done.
10:13.26kladzekaldemar - 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 :(
10:13.38v0lZythanks guys
10:13.46WIMPyOr there.
10:14.55kaldemarkladze: 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)
10:15.59kladzestill the same, even after a dialplan reload
10:16.39kaldemarkladze: then you edited the wrong extension.
10:18.14WIMPykladze: Try to check the path as I suggested. That oder has changed some time ago.
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10:20.43kladzehttp://pastebin.com/xfkwWiYR I added it here, line 17 - and output from asterisk: http://pastebin.com/gx1JR9Ju
10:22.09v0lZyhm
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10:22.11kaldemarkladze: still does not show up in the output between priorities 1 and 2.
10:22.18v0lZyok, now its not doing something else its supposed to do, eheh.
10:22.24v0lZyhave to recheck the logic behind this thing
10:22.50WIMPyv0lZy: You can spend lots of time on that topic.
10:23.31WIMPye.g. I try to at least check that there's an extension for the destination.
10:27.02v0lZyi will add that check too
10:27.14v0lZybut right now im just trying for basic functionality and then upgrading
10:27.48kaldemarusing dialplan like that (throwing all unknowns into _X!) is asking for trouble.
10:28.16*** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com)
10:30.58v0lZykaldemar: at this point i suppose
10:31.02v0lZybut the way i have it set up now
10:31.12v0lZyits just a matter of changing _X! to _XX ... or _XXX etc
10:32.48kaldemarnearly as bad.
10:33.52v0lZy...what should it be like then?
10:33.55v0lZyhardcoded?
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10:35.54kaldemarv0lZy: so that you don't have extensions that match dialed numbers but have non-existant destinations.
10:36.55v0lZyso.. hardcoded?
10:38.02kaldemarif that's what you like to call it.
10:38.24v0lZyhardcoded as in ... instead of _XX .. i use _00, _01, etc etc
10:38.25v0lZy?
10:40.07kaldemaror _0[1-9] if you really have 01-09. XX and 100 separate lines are not two only options.
10:41.14v0lZyah, gotcha
10:42.36polomolo747hello, is there a way to cause sip rtp traffic to go through asterisk instead of p2p?
10:45.47kaldemarpolomolo747: directmedia=no in sip.conf
10:46.21v0lZyhm
10:46.23v0lZyquestion
10:47.02v0lZyexten => _**X!,1,GotoIf($["${EXTEN:2}" = "${CALLERID(num)}"]?cancelforward)
10:47.04v0lZy<PROTECTED>
10:47.05v0lZy<PROTECTED>
10:47.07v0lZy<PROTECTED>
10:47.46v0lZyif ${DB(CF/${EXTEN:2})} is not set and has no value stored....
10:48.24v0lZythen my gotoif there is useless.... i guess.
10:48.47v0lZynever mind, think i got it
10:49.42v0lZydamn, no, lost it... had it in my head and bunf, gone.
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10:53.54polomolo747kaldemar, 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.
10:54.57kaldemarpolomolo747: is NAT involved?
10:55.09polomolo747no
10:55.17polomolo747but eventually yes
10:55.41v0lZydamnation.
10:56.02v0lZyi think this line is wrong same =>       n(loop),ExecIf($["${DESTINATION_CF}" != ""]?Set(DESTINATION_CF=${DB(CF/${DESTINATION_CF})})
10:56.11kaldemar"sip set debug on" in CLI and make a call that has sound issues. pastebin all output.
10:59.17polomolo747kaldemar,  http://pastebin.com/gC6CJNfn
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11:05.56kaldemarpolomolo747: with a quick glance i can't see anything wrong in there.
11:08.19polomolo747kaldemar, thanks.
11:12.13v0lZyfunny
11:12.24v0lZyasterisk seems to handle these call forward loops by itself
11:12.38v0lZyright now i have 16 forwarding to 31, 64 forwarding to 16
11:12.46v0lZyi call 64 from 31, and 16 rings
11:13.06*** join/#asterisk b0ot (~Administr@147.177.60.174)
11:13.19b0otwhy wouldn't my asterisk/freepbx install have the reload option in the CLI
11:14.20kaldemarb0ot: because the CLI alias is not configured in cli_aliases.conf. the real command is "module reload".
11:15.28b0otthanks
11:15.54v0lZykaldemar: does asterisk even allow for chaining callforwards?
11:16.25v0lZyor once u use a dial command, thats it, ur dialing, no more forwarding or anything
11:16.41WIMPyAsterisk doesn't do CF, it's your dialplan.
11:17.46kaldemarv0lZy: that is just dialplan, there is nothing to allow or disallow.
11:18.09v0lZydo context fold back on themselves?
11:18.11v0lZylike
11:18.39v0lZyI have n,Dial(SIP/${FORWARD_TO},to)
11:18.53v0lZynow if all phones are set to forward to eachother
11:19.00v0lZyhm
11:19.06v0lZyi guess i need another phone to test this
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11:20.46b0otwhen i do a dundi show peers I get: 1 dundi peers [0 online,0 offline, 1 unmonitored]
11:21.01b0otthe guide i'm following shows it should be online
11:21.19b0otany idea what could cause it to be unmonitored vs online
11:23.18polomolo747kaldemar, i found the problem
11:23.31polomolo747i binded the udp on 127.0.0.1
11:23.44kaldemarb0ot: "qualify=yes" in dundi.conf
11:23.48polomolo747how can i force sip to use tls?
11:26.21b0otwoot!
11:26.37b0otkaldemar, thanks
11:26.53b0oti had accidently typed it as quality
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11:34.09WIMPypolomolo747
11:34.11WIMPyoops
11:34.16WIMPypolomolo747: transport=tls
11:34.36polomolo747i want asterisk to not even listen on the sip port.
11:35.27kaldemaruse tlsbindaddr only.
11:35.31WIMPyYu can set it in the global section.
11:35.55WIMPyWouldn;t it just use defaults for udp then?
11:36.32WIMPyBut udpbindaddr=127.0.0.1 would at least make it unjavailable for others.
11:37.03polomolo747the 127.0.0.1 will disable rtp
11:37.25b0otI have to servers (A,B) where I am attempting the DUNDi setup. Both show each other as online with "dundi show peers"
11:37.32polomolo747tlsbindaddr without idpbindaddr defaults to 0.0.0.0:5060
11:37.33v0lZyobviously my call forwarding code doesnt allow for chained forwarding
11:38.02v0lZyI have 64 forwarded to 16 and 16 forwarded to 31
11:38.07v0lZywhen i call from 50
11:38.09b0othowever I get "DUNDi Query EIT returned no results" when I do dundi query <mac server B> from server A
11:38.18b0otany thoughts?
11:38.30v0lZyit should go 50 => 64 => 16 => 31 (ring)
11:38.44v0lZyInstead ... i get 16 ringing... which means it doesnt hop to 31
11:39.18v0lZyand this is because i have Dial
11:39.36v0lZyi thought that dialing a channel would repeat the whole process in the internal context
11:39.51v0lZywill have to work on that too :|
11:40.04kaldemarif udpbindaddr is not configured and transport=tls is along tlsbindaddr, then it won't listen on any UDP port.
11:40.23WIMPypolomolo747: Can you elaborate on the disable rtp bit? That shounds wrong.
11:41.23polomolo747if 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
11:41.38WIMPyv0lZy: Dial a local channel.
11:41.50polomolo747udpaddrbind is default at 0.0.0.0:5060 so i can't disable it.
11:41.55v0lZyWIMPy: how?
11:42.13*** join/#asterisk wonderworld (~ww@dsdf-4db5d510.pool.mediaWays.net)
11:42.18polomolo747there isn't an option to disable udpsip but leave rtpupd open, is there?
11:42.33WIMPyv0lZy: When doing to CF, dial the destinations extension via a local channel instead of a device.
11:43.16WIMPypolomolo747: I don;t know any, but rtp using the udpbindaddress sounds like a bug to me.
11:43.32v0lZy... dial(LOCAL/16) instead of SIP there?
11:43.41polomolo747there isn't a rtp bind addr is there?
11:44.16WIMPypolomolo747: Not that I know.
11:44.41WIMPyv0lZy: local/exten@context
11:44.43polomolo747well, i can't disable it in the firewall level, so i guess it's still ok
11:44.54polomolo747is srtp/zrtp is on udp?
11:45.28WIMPypolomolo747: yes
11:45.47b0otHow can I see each other with "dundi show peers" as online but fail to query one another
11:46.11WIMPyb0ot: What do you query?
11:46.31kaldemarb0ot: because of your mappings, if the peers are working.
11:47.02polomolo747kaldemar, WIMPy , thanks.
11:48.10b0otFrom Server A, I query Server B so "dundi query 00:00:00:00:00:0B"
11:49.26WIMPyb0ot: If you query for nothing, you will probably get nothing.
11:49.33WIMPyAdd the right context
11:50.13os_FlorentHi,
11:50.15os_FlorentI have a problem with an old version of asterisk (1.4.26.1)
11:50.17os_FlorentMy SS7 gateway sends call to my asterisk using SIP, and sometimes One-Way audio problems occur.
11:50.18os_FlorentWhen such a problem occur, I can see those logs :
11:50.20os_Florent[Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring.
11:50.31os_Florent[Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring.
11:50.35os_Florent[Aug 8 13:03:34] WARNING[9524] chan_sip.c: Too many SIP headers. Ignoring.
11:50.37os_FlorentI can certify there is more than 64 headers in this case because of the source code :
11:50.38os_Florent#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
11:50.40os_Florent[..]
11:50.41WIMPyDon't flood the channel!
11:50.42os_Florentif (f >= SIP_MAX_HEADERS - 1) { ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n"); }
11:50.43os_FlorentIs it possible to modify the source code to increase the 64 SIP headers limit ?
11:50.45os_FlorentUpgrading 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.
11:50.46os_FlorentThank you !
11:50.48os_Florentsorry
11:50.57b0otdrowned
11:51.45b0otSo i thought if you did the dundi query of just the mac it should return the EID WIMPy
11:51.47WIMPyChange that define and see if it's all it takes.
11:51.53kaldemaros_Florent: open source, dude. just modify it and recompile.
11:51.57v0lZyha, i think it works now WIMPy, it hops to the next in line
11:52.19WIMPyb0ot: You need to provide a context.
11:52.40kaldemarthe dundi query uses e164 by default if no context is provided.
11:53.26WIMPyOk, but that needs to exist then.
11:53.53v0lZybtw
11:53.54v0lZyringtones
11:54.01v0lZyphone ringtones
11:54.10v0lZythats up to each device, or can asterisk push something to them'
11:54.14os_Florentkaldemar : ok thank you a lot. Do you know if this modification can impact asterisk negatively ?
11:54.26WIMPyv0lZy: yes :-)
11:54.34v0lZylol
11:54.35v0lZyyes to which one
11:54.52WIMPyv0lZy: Yes, it's device specific, but some devices will accept the URL of a ringtone on your server.
11:56.04WIMPyos_Florent: You will have to find out. It looks unlikely, but my guess is as good as yours.
11:56.53os_FlorentWIMPy: ok thank you
11:57.13v0lZyexcellent
11:57.13*** join/#asterisk Bullmoose (~Bullmoose@75-174-79-252.bois.qwest.net)
11:57.26v0lZythanks WIMPy/kaldemar, i got call forwarding working the way i wanted to
11:57.36v0lZybtw, can u explain the difference between dialing a device and a local chan?
11:57.38kladzeHi, 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.. ?
11:58.08b0otWIMPy, 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
11:58.14b0otnot sure what the issue could be
11:58.43WIMPyv0lZy: The difference would have been obvious if you didn't use the extension numbers as device names.
11:58.59WIMPyv0lZy: A local cahnnel loops back to your dialplan.
11:59.20kaldemarb0ot: include, permit and mappings.
11:59.29v0lZyand device just shoots in the direction of whateve ru tell it to, got it
11:59.47WIMPykladze: Did you note my two replies earlier?
12:00.09WIMPyb0ot: What contexts do you have defined for dundi?
12:00.21rolandowkladze: are the permissions in the folder danish/da correct? is the folder readable by asterisk?
12:00.49v0lZybtw, sounds i can play upon doing something
12:00.50kaldemarkladze: did you get the CHANNEL(language)=da in your extension?
12:01.04v0lZycan it be a wav
12:01.13v0lZyor must it be something specifc'
12:01.26WIMPyv0lZy: Ringtones?
12:01.32v0lZyno
12:01.34v0lZylike voicemail
12:01.34*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
12:01.38v0lZystuff thats thrown together
12:01.40v0lZyi want it to say
12:01.42WIMPyThat would obviousely be device specific.
12:01.58*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
12:02.14b0otWIMPy, in IAX_extensions.conf
12:02.15v0lZy'Call forwarding activated' and 'call forwarding deactiated' and 'error' and stuff like that.
12:02.22b0otI defined [priv]
12:02.32kladzeWIMPy hmm, i dont think so :\
12:02.36b0otwith context=incomingdundi
12:02.46kladzerolandow they should be... drwxr-xr-x 2 root root  4096 Aug  8 13:53 digits
12:03.14WIMPyb0ot: What's that file? Where is it used?
12:03.28v0lZyi dont have the stuff i want it to say in my var lib asterisk sounds end dir
12:03.31rolandowkladze: i think you should do Playback(digits/1)
12:03.33v0lZyen*
12:03.39kladzekaldemar 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
12:03.57b0otAs I understand it it, that is used to setup the IAX2 channel between the two servers
12:03.58rolandowkladze: it does check for the correct language, but when you want to go a folder deeper, you'd have to specify it.
12:04.00b0otfor communication
12:04.12b0otWIMPy, following this guide: http://www.voip-info.org/storage/users/813/47813/images/1654/DUNDi_So_Easy.pdf
12:04.36kladzerocksfrow, oh okay.. i didnt know that.. thanks..
12:04.43WIMPykladze: Could be a digits/da vs da/digits thing. / Try to check the path as I suggested. That oder has changed some time ago.
12:04.47kaldemarkladze: asterisk does not automatically know that you mean a file in under the digits directory.
12:05.02b0otlol although according to that the IAX2 channel isn't used for queries
12:05.10kladzeso i always need to specify the subfolder aswell
12:05.19kaldemarb0ot: and it is not.
12:05.20kladzeetc when i want to play a digit
12:05.33rolandowkladze: yes... and you could use your own created subfolders as well
12:05.43WIMPyb0ot: That's not a file Asterisk uses. So uinless you included that somewhere it won;t get used.
12:05.49kladzealright, thanks :)
12:05.51rolandowkladze: the part that asterisk automates, is to use the correct language (based on your settings)
12:06.01rolandowor at least that's how i understand it :)
12:06.09kladzealright :>
12:06.16kladzethanks
12:06.19b0otso in the dundi file I have the [general]
12:06.24b0otwith my info in it
12:06.31WIMPyOh, I thought it was happening when saying numbers/digits.
12:06.35kaldemarb0ot: the [priv] is only used when actually dialing the other box. you need to configure mappings in dundi.conf.
12:06.48rolandowhm..
12:07.01rolandowgood point WIMPy
12:07.10rolandowkladze: what command do you use to playback the digit?
12:07.32rolandowkladze: i think WIMPy is right when you use Digits() .. it probably takes the files from digits automatically ..
12:07.46kladzeoh
12:07.48kladzehm
12:07.48kladzeexten => 1000,n,Playback(1)
12:07.53kladzefor number 1
12:08.12kladzeand there is also one named 1-and
12:08.14WIMPykladze: Then it's as rolandow said.
12:08.17b0otkaldemar, 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
12:08.28WIMPywas thinking about SayDigits or SayNumber
12:08.36rolandowkladze: right.. with playback you have to specify the subdir
12:08.51kladzeOkay :)
12:08.53kaldemarb0ot: and does it permit "priv"?
12:08.55b0otI 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
12:09.02b0otyes
12:09.05b0otpermit = priv
12:09.15rolandowso you probably should use SayDigits or SayNumber :)
12:09.41kaldemarb0ot: "dundi query 00:00:00:00:00:0B@priv"
12:10.01kladzewell thanks WIMPy, rocksfrow, kaldemar my da did work as intended before.. just me unaware of some playback and folder issues :D
12:10.11kladzerocksfrow, will try that :)
12:10.15kladzewops
12:10.22kladzerolandow, will that :)
12:10.28kladzetry*
12:11.31b0otkaldemar, that works!
12:11.35*** join/#asterisk Jasnejac (~EnorMouse@2001:b70:500:2:645b:c5f6:428d:bfe0)
12:12.05b0otwell 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
12:12.08b0otand then it worked!
12:14.23b0otSweet... now I just need to figure out what to do next
12:14.25b0otlol
12:15.02WIMPyPray
12:16.16b0otHow would I set it up for Dundi to first check my local extensions
12:16.21b0otand then if not use Dundi
12:16.49*** join/#asterisk Neptu (~Neptu@mail.avtech.aero)
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12:17.32Ice_StrikeHi
12:17.33WIMPyThat happens automatically if you use a dundi switch.
12:19.21Ice_StrikeWhy I am getting a lot of error
12:19.26Ice_Strikesometime it does make calls
12:19.27Ice_Strikehttp://pastebin.com/LBw8bLZD
12:19.41Ice_StrikeI moved asterisk to new server and now getting this error
12:19.47Ice_StrikeI have not seen this before
12:20.45WIMPyThat peer checks your IP?
12:22.37Ice_Strike7x.129.2x3.52 my server ip
12:23.31WIMPyYou talk to yourself? DNS troubles?
12:23.57WIMPyNo, that's just the contents.
12:24.45Ice_Strikehow do I check?
12:24.56Ice_StrikeIs this something to do with RTP?
12:25.16WIMPyUse sip debug to find out where it's comming from.
12:25.31WIMPyOr use wireshark.
12:28.44*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
12:29.13polomolo747Is there anyone here with sip text message experience?
12:31.14*** join/#asterisk Blueneon (~Blueneon@196-210-194-169.dynamic.isadsl.co.za)
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12:34.20BlueneonHi, 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
12:34.20Blueneonis 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)
12:34.32BlueneonIs there something built into asterisk that will allow this?
12:35.57BlueneonBasically 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
12:37.04v0lZyWIMPy: can i playback a wav file with the playback command?
12:37.16*** join/#asterisk newtonr (~newtonr@nat/digium/x-tgumtrtxlfvcgnqo)
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12:37.29[TK]D-FenderBlueneon, 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-FenderBlueneon, "core show function DB"
12:37.47[TK]D-Fender^^^
12:38.09DrDamnitv0lZy: Yes. It has to be encoded properly, though. http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
12:38.26Blueneonbut 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-FenderBlueneon, sure... it's your dialplan.. put that check wherever YOU want it to be
12:39.00Blueneonoh ok im with you
12:39.16rolandowBlueneon: you could also write a dialplan that records a message
12:39.32Blueneoncheck flag first, then play temp "office closed" and go to voicemail, OR if flag is ok, resume normal ops?
12:40.14Blueneonok what function is used to set/check flag?
12:40.33v0lZy8kbit ad al that?
12:40.53*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
12:41.06[TK]D-Fender<PROTECTED>
12:41.31DrDamnitYep.
12:42.16Roeltafaik the asterisk book has an example on that.. including an blf light for the status
12:42.19BlueneonAh ;)
12:43.26Blueneonok 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?
12:43.34Blueneonthanks TK :)
12:44.41[TK]D-Fenderyup
12:44.57rolandowBlueneon: http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/IVR_id247054.html <-- recording example
12:44.59WIMPyv0lZy: Sure. But depending on your Asterisk version there are some restrictions on the format.
12:45.39rolandowBlueneon: 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
12:46.15rolandowanyways.. you could asjust that to your needs
12:47.27*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
12:48.42Blueneonthanks
12:49.08*** join/#asterisk fergus (~fergus@178.124.149.113)
12:52.26b0otwhat pastebin should we use here
12:52.30*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
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12:52.59WIMPyany
12:53.31*** join/#asterisk danfromuk (~IceChat77@2.27.27.15)
12:53.44rolandowthe registered version
12:55.06b0otWIMPy, kaldemar This is my setup. I think there might be something wrong the way I setup my extensions_customs.conf
12:55.07b0othttp://paste2.org/p/2098445
12:55.15b0otIt is about as minimalist as you can get
12:55.15*** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg)
12:55.26b0otthe calls don't seem to try to use dundi
12:56.43*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
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12:58.35kaldemarb0ot: you're lacking "dundiextens" in extensions.conf.
12:59.19kaldemarb0ot: 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.
13:00.00kaldemarb0ot: exten => _2XXX
13:00.07kaldemarb0ot: ^ an invalid line
13:00.25b0otsorry I think my paste got messed up, I'm fixing it there
13:00.48kaldemarb0ot: and you don't have a DUNDi switch or any manual lookups in extensions.conf.
13:01.43b0othttp://paste2.org/p/2098448
13:01.51b0otcorrected extensions_custom.conf
13:03.24b0ot"All circuits are busy now" when I try to call
13:04.11WIMPyI don;t have any idea what a dundi trunk means. Might be a good idea to take the whole thing to #freepbx.
13:04.48kaldemarthere is no such thing as a DUNDi trunk.
13:04.55[TK]D-FenderFreePBX <------------
13:05.07kaldemarb0ot: 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)
13:05.39b0otkaldemar, the DUNDi trunk is configured via freepbx
13:07.19BlueneonTK: your plan worked like a charm
13:07.29Blueneonjust wanted to say thank you so much :)
13:08.41*** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell)
13:09.35*** join/#asterisk jfeldmar (~asteriskp@kikinini.enst.fr)
13:10.27*** join/#asterisk bobb_WU (~bobb@206.74.211.7)
13:13.02bobb_WUcould 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
13:13.12b0otwhat do you mean by a dundi switch
13:13.16b0otkaldemar,
13:13.45bobb_WUand maybe explain what the warnings mean...
13:14.42*** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se)
13:15.59WIMPyb0ot: You need a 'switch => dundi/context' somewhere in your dialplan.
13:19.16WIMPybobb_WU: I don't know what waitid thing is, but retransmissions are either networking issues or serious protocoll errors.
13:21.02*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
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13:33.23Kattyhello my asterisk does not work at all how to fix??? answe rplz.
13:34.18rolandowdo you have a macro for that katty?
13:34.28Kattywhat is macro???
13:34.37rolandow&!%@&
13:36.06*** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6)
13:37.39*** join/#asterisk darthanubis (~anubis@unaffiliated/darthanubis)
13:39.28eXcAliBuRwhen 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()
13:40.08eXcAliBuRi can dial 306, but then when connected, any thing else I dial is ignored
13:40.10eXcAliBuR:(
13:41.02eXcAliBuRI don't know where my big book is from leifmadsen :< I think i left it at a site i was working at
13:41.22rolandowwhy would you dial something when you're connected ?
13:41.34eXcAliBuRbecause when connected, i have to dial a extension
13:41.35rolandowhttp://astbook.asteriskdocs.org/
13:42.12*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
13:42.27rolandowhow did you configure dtmfmode ?
13:42.31b0otI have one way calling working with dundi
13:42.33b0otlol
13:42.40b0otdon't know how it's possilbe the configs are identical
13:43.30rolandoweXcAliBuR: you should use directmedia=no in sip.conf .. and probably dtmfmode=rfc2883 .. but i think that depends on the phone you're using.
13:43.35WIMPyeXcAliBuR: And what do you expect to happen with any keypresses after you have been connected?
13:43.58eXcAliBuRwell it should pass them to the device i'm connected too
13:44.04rolandowassuming that you're using WaitExten or something like that in your dialplan after you connected to 306
13:44.52WIMPyOk, so that has nothing to do with dialling. And yes, check your DTMF modes.
13:45.09eXcAliBuRmy dtmf = rfc2833
13:45.15eXcAliBuRshould i change to 2883 ?
13:45.38eXcAliBuRi have the digium phone
13:45.42eXcAliBuR:)
13:45.43rolandoweh no
13:45.47rolandowtypo
13:46.04WIMPyRFC2833 is usually the best choice.
13:46.05rolandowrfc2833 is ok
13:46.14WIMPyAnd what about the peers configuration?
13:46.15v0lZyWIMPy: how to check if an extension is a valid sip peer? (as in, not application or external number9
13:46.16eXcAliBuRmaybe i don't need the Hangup() line?
13:46.33rolandoweXcAliBuR: pastebin your dialplan
13:46.42rolandowfor 306
13:46.57WIMPyv0lZy: You can't You can only check for existance. Use different contexts.
13:46.58rolandowyou're calling 306, then you type an extension and you want something to happen??
13:47.18eXcAliBuRwhen 306 answers, it gives me options press this,,, press that
13:47.24[TK]D-FenderAn extension is an extension.
13:47.29rolandoweXcAliBuR: so what's your dialplan for 306?
13:47.33[TK]D-FenderAn extension is not a peer
13:48.18*** join/#asterisk mirela666 (~mirko@212.200.146.253)
13:48.28v0lZyWIMPy: uhm... seriously... no way to do that?
13:48.34eXcAliBuRhttp://asterisk.pastebin.ca/2178478
13:48.43WIMPyv0lZy: no
13:48.55eXcAliBuRthats my extensions.conf
13:48.57v0lZywhat about sip.conf
13:49.02eXcAliBuRi didn't leave anything out
13:49.11WIMPyv0lZy: The only relation between extensions and devices is what you write in your dialplan.
13:49.36v0lZywhat about stuff registered in sip.conf
13:49.54eXcAliBuRhttp://asterisk.pastebin.ca/2178479
13:49.57WIMPyv0lZy: That has nothing to do with extensions at all.
13:50.14v0lZywell
13:50.18[TK]D-FendereXcAliBuR,  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-FendereXcAliBuR, Dialplan is linear
13:50.25v0lZyi can check if the dialed extension also resides in sip.conf
13:50.43WIMPyv0lZy: 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.
13:50.58v0lZyi name them like
13:50.59[TK]D-Fenderv0lZy, "core show function SIPPEER"
13:50.59v0lZySIP16
13:51.14[TK]D-Fenderv0lZy, And stop saying an "extension" is in "sip.conf"
13:51.32v0lZythanks Fender
13:52.01v0lZythanks for all the help guys
13:52.01v0lZygona scoot now
13:52.02eXcAliBuRTk fender: when I dial 9, phone number, i can dial stuff when connected
13:52.11eXcAliBuR:{
13:52.12v0lZyhave a nice day
13:52.14v0lZybye
13:52.23rolandoweXcAliBuR: the dialplan you show here, doesn't have any "press 1 for this, press 2 for that" stuff
13:52.26[TK]D-FendereXcAliBuR, Not inside the rest of your dialplan you can't
13:52.42rolandoweXcAliBuR: i would expect you'd Playback() some audio, wand use WaitExten or something for input
13:53.37b0otI really don't understand how this can work one way, but not the other
13:53.44b0otthe configs are identical
13:54.42eXcAliBuRi'm going to try something
13:54.46*** join/#asterisk seraphie (~erin@75.76.38.159)
13:57.41eXcAliBuRcould it be the dialplan in the phone messing me up?
13:58.26WIMPyeXcAliBuR: No. You are not dialling.
13:59.46eXcAliBuRok 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
14:00.28WIMPyCheck the DTMF modes ON BOTH ENDS.
14:01.11WIMPyOr all 3 even.
14:01.34[TK]D-FendereXcAliBuR, That;s a little clearer... your initial description was very vague
14:01.50[TK]D-FendereXcAliBuR, and you have a DTMF MODE problem as WIMPy has mentioned
14:01.53rolandowso 306 is a conference device?
14:01.59eXcAliBuRyes
14:02.13eXcAliBuRthat has sip capabilites
14:09.36rolandowi'd say dtmf mode
14:10.02rolandowwhat codec do you use?? you could try inband as well if you use alaw/ulaw
14:11.06rolandowhm .. i also read something on this channel last week from some guy who raised the rx or txgain .. something like that..
14:11.14rolandowseemed the dtmf tones weren't loud enough ?
14:12.54eXcAliBuRi have a rmx 1515
14:12.58eXcAliBuRpolycom
14:13.28eXcAliBuRi haven't found dtmfmode on it yet
14:13.29eXcAliBuR:/
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14:15.11[TK]D-FendereXcAliBuR, So go through them ALL one at a time
14:15.15b0otAnyone know what this could be from: channel.c: Prodding channel 'SIP/2003-0000006' failed
14:15.29rolandoweXcAliBuR: dtmfmode is in your sip.conf
14:15.53[TK]D-Fenderb0ot, Stick to #freepbx and don't post singular messages like that.  Show them the ENTIRE call, not tiny little crumbs
14:18.37Kattydude.
14:19.01mirela666eXcAliBuR: Do You have a webUI for polycom phones?
14:20.31eXcAliBuRthe phone is digium
14:21.07eXcAliBuRit's a nice phone, i'd recommand everyone to buy it
14:21.09mirela666On my Linksys I can change many SIP, codecs and dtmf mods but only trough WebUI
14:21.09eXcAliBuR:)
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14:40.45b0otwhy won't you PROD channel
14:40.53b0otdon't you want to be prodded
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14:51.39eXcAliBuRYAY it's working
14:51.46eXcAliBuRit was a problem in the polycom :)
14:52.25lorsungcu_what?
14:52.50lorsungcu_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-Fenderit's still on my screen... didn't even have to scroll....
14:54.00rolandowmaybe you have a very large screen
14:54.07lorsungcu_<eXcAliBuR> the phone is digium
14:54.23[TK]D-FenderYes, and the PHONE wasn't the issue
14:54.26[TK]D-Fenderit was the otehr end
14:54.29[TK]D-Fenderother*
14:54.33lorsungcu_right on
14:54.34lorsungcu_missed that
14:54.35rolandowso what was the solution ?
14:59.36Ice_StrikeHow to copy 100 files from a dir to another dir? recently created file
15:00.44lorsungcu_generally i have two windows open
15:00.48lorsungcu_so i can type faster.
15:01.09lorsungcu_then just open each file and type the contents into the other window.
15:01.23rolandowquit
15:01.26rolandowwrong window
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15:06.08b0otbusy
15:06.16b0ottones make me
15:06.20b0otwant to throw
15:06.22b0otphones
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15:08.14geeknationhey
15:08.22lorsungcu_hi
15:08.24geeknationI've got an issue with the ruby-agi gem
15:08.58geeknationWas working on Rackspace, but just moved over to EC2 and its throwing errors
15:09.59geeknationany idea if it needs to be configured differently on Amazon?
15:11.37lorsungcu_no idea.  I've been meaning to mess with that, though
15:12.32geeknationtesting the requires in rib throws this
15:12.34geeknationLoadError: no such file to load -- ruby-agi/rs/
15:12.52geeknationrib=irb ;)
15:14.04geeknationwhats the best way to debug?
15:14.17b0otlorsungcu you have any freepbx-fu
15:14.35lorsungcu_way you need b0ot
15:14.58lorsungcu_geeknation, id start by checking that that file exists...
15:15.14b0otI have a freepbx setup as basic as possible. 2 servers each with 3 phones
15:15.23b0otI have attempted to setup DUNDi
15:15.29b0otI have one way calling working
15:15.47geeknationI'm looking in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi/rs
15:15.56navaismolorsungcu, Qwell finally its working the agi script \o/  http://youtu.be/CRpyLRtmszM
15:16.07b0otlorsungcu_, my setup: http://paste2.org/p/2098542 log of call that doesnt' work: http://paste2.org/p/2098547
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15:17.17geeknationWhich file deals with the require?
15:17.30Qwellnavaismo: Was I right about the problem?
15:17.39FinboySlickWhile this isn't 100% asterisk related.  Are any of you aware of firmware/hardware differences between SPA2102-R1 and SPA2102-NA?
15:17.55lorsungcu_awesome navaismo
15:18.01lorsungcu_what was the issue
15:18.10FinboySlickI figure some people here might have experience with relatively large deployments of those.
15:18.27Qwelllorsungcu: I'm almost certain is was agi-bin/ not being g+r or o+r
15:18.40lorsungcu_yes thats what it looked like, qwell
15:19.12QwellI didn't bother checking how ls shows permissions of .., and he never gave me what I actually asked for.
15:19.45Qwellanyways, he'll never say what the problem actually was, so any discussion is moot.
15:20.00lorsungcu_navaismo: dammit what was the problem :D
15:20.04navaismoQwell, lorsungcu actually the issue was with php and memory, I testes a simple agi script (stream file) in my machine with same permission
15:20.30navaismoand it works so i start from stcratch again and recompile asterisk without many unused modules
15:20.35navaismocreate a bigger swap
15:20.43Qwellstart from scratch = fixed permissions
15:20.50lorsungcu_yeah i kind of agree
15:20.53navaismoafter 3 hours copy again the script and work aout of the box
15:21.04navaismonope i dont change the permissions
15:21.11lorsungcu_post permissions for ago-bin and ../agi-bin
15:21.13navaismoon scripts or agi dir
15:21.19navaismoI just copied again
15:21.48Qwellls -ld /var/lib/asterisk/agi-bin/
15:21.50QwellShow us what that says.
15:21.55navaismoyep 1 sec
15:23.29b0ot:(
15:24.27navaismohere -> http://pastebin.com/Rr0KM8zj
15:24.41Qwello+r
15:24.47lorsungcu_yeah its different
15:24.57navaismo:S
15:25.00geeknationhmmm, weird -- my amazon installation didn't create a /var/lib/asterisk/agi-bin
15:25.01QwellYou should have listened to what you were told, before going off any reinstalling everything.
15:25.02lorsungcu_:p
15:25.07QwellThis was an incredibly simple fix.
15:25.09lorsungcu_but
15:25.10geeknationmaybe thats my problem….?
15:25.13navaismoNope
15:25.14lorsungcu_now he knows :D
15:25.25navaismothat was not easy
15:25.28Qwelllorsungcu_: You overestimate things.
15:26.11Qwellgeeknation: well, if Asterisk can't see a file, then it can't execute it.  I haven't seen your problem description though.
15:26.52navaismobefore compilation o+r doesnt work but important thig is working now
15:28.00geeknationQwell ruby-agi is throwing an error on require
15:28.08Qwellgeeknation: pastebin
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15:29.09geeknationhttp://pastebin.com/3XzQ4k5V
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15:38.51navaismogeeknation, and this exist ruby-agi/rs/
15:39.43geeknationI have a bunch of rb files in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi/rs
15:40.26geeknationbut ruby-agi.rb is in /var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib
15:40.31geeknationshould i move it to /rs
15:40.33geeknation?
15:41.46navaismoor in your include you set the complete path
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15:43.49geeknationgetting the same error hardcoding the path
15:44.54geeknationdid the following - require '/var/lib/gems/1.8/gems/ruby-agi-2.0.0/lib/ruby-agi.rb'
15:45.07lorsungcu_check permissions?
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15:45.18geeknationyah - did that as well :(
15:46.03geeknationruby-agi.rb has -rw-rw-r--
15:46.09Qwellno x?
15:46.15QwellDo you need +x to be included?
15:47.15geeknationsame permission were working on rackspace
15:47.31lorsungcu_try with 777 and revert if it doesn't change
15:47.47geeknationthere's also a folder called 'ruby-agi' with the following permissions: drwxr-xr-x
15:47.58navaismojhehe sounds me familiar
15:48.03lorsungcu_:p
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15:50.21geeknationjust changed the permissions to 777
15:50.23geeknationit's working
15:50.25geeknation:D
15:50.41geeknationu think it's safe if i switch it to 755?
15:51.03Qwellgeeknation: depends on ownership
15:51.13Qwellactually, I lied
15:51.15geeknationwell - thanks for all the help
15:51.17Qwell755 should be fine
15:51.22geeknationthanks
15:51.29geeknationqwell - i appreciate all the debugging
15:51.36geeknationhopefully i can reciprocate one day :)
15:51.38Qwellgeeknation: That'll be $299.95
15:51.47geeknationha
15:52.00geeknationchecks in the post
15:52.06geeknation:)
15:52.20QwellI should stop saying that.  Somebody might actually take me seriously one day...
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16:07.50lorsungcu_swell, you know anything about swift?
16:07.56lorsungcu_wtf autocorrect
16:08.11lorsungcu_Qwell: you know anything about swift?
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16:28.35geeknationhey guys… I'm having a new issue… none of my asterisk sound files are playing - even 'hello-world'
16:28.54geeknationi overwrote all my files in the asterisk/sounds/en (with a set from another server)
16:29.01geeknationand i can't get them to play
16:29.12geeknationi've tried changing the owner to ubuntu, asterisk, and root
16:29.20geeknationand all the files are set to chmod 777....
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16:31.45lorsungcu_ps aux | grep asterisk geeknation
16:32.48lorsungcu_get that dynamic context thing working, CrazyTux[m] ?
16:33.17geeknationlorsungcu - when i type 'aux | grep asterisk' i get no command 'aux' found
16:33.35lorsungcu_ps aux
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16:35.10geeknationlorsungcu - thanks -srry i didn't get it.  here are my results - http://pastebin.com/ppqFqurY
16:35.26CrazyTux[m]lorsungcu I just rewrote the extensions
16:39.00geeknationall of my sound files are under these permissions and ownership:
16:39.01geeknation-rwxr-xr-x 1 777 asterisk   9174 2012-08-05 17:12 vm-intro.gsm
16:39.32geeknationi'm thinking maybe it should read 'asterisk asterisk' instead of just a single 'asterisk'
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16:54.14lorsungcu_geeknation: did you try setting asterisk:asterisk?
16:54.24geeknationyes
16:54.26geeknationwith no success
16:54.58geeknationi now have all my sound files as
16:54.59geeknation-rwxrwxrwx 1 asterisk asterisk   1056 2012-08-05 17:12 vm-Work.gsm
16:55.21geeknationbut 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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16:58.41b0otAnyone good with DUNDi around?
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17:14.21[TK]D-Fendergeeknation,  then go show us exactly where you put them, and "core show settings" <------------------------------------------
17:14.23[TK]D-Fender^^
17:14.50[TK]D-Fendergeeknation, give a FULL dump, including the ls -la of the folder itself.  No more puny fragments.
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17:20.49geeknationdrwxrwxrwx 16 asterisk asterisk 16384 2012-08-05 17:12 en
17:21.28geeknationand 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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17:31.34lorsungcu_geeknation: read TKDs last message
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17:32.58geeknationsorry for fragments… will pastebin
17:33.28geeknationcore show setting: http://pastebin.com/5b5sE7RJ
17:35.02geeknationls -la of both the en directory and its contents: http://pastebin.com/NZJsQg7K
17:35.30Kattyi'm having a bad day :<
17:36.28lorsungcu_not as bad as geeknation
17:36.39Kattyidk about that.
17:36.48geeknation:D
17:37.53lorsungcu_http://www.youtube.com/watch?v=sGF6bOi1NfA
17:38.25geeknationthanks...
17:38.29lorsungcu_geeknation: thought all files were asterisk:asterisk
17:38.43geeknationya, been playing around with different owners
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17:38.53geeknationi can change them back right now
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17:39.43lorsungcu_can you check permissions of /var/lib/asterisk and its subdir?
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17:42.53geeknationall the ls -la of directories (i also changed the chown back to asterisk:asterisk) - http://pastebin.com/A6TuARcW
17:42.56geeknationthanks for helping
17:43.52lorsungcu_so now playback of a file fails with the same error
17:43.54lorsungcu_does not exit
17:43.56lorsungcu_exist
17:46.06lorsungcu_(I'm asking if thats still true)
17:46.33geeknationya. same error: WARNING[3016]: file.c:644 ast_openstream_full: File '/var/lib/asterisk/sounds/en/hello-world' does not exist in any format
17:47.28lorsungcu_you core reload?
17:47.48dwayneanyone try grandstream phones lately and want to comment on their quality?
17:47.57lorsungcu_not a fan
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17:48.33jofryhi folks
17:48.39lorsungcu_hi
17:48.43geeknationyah - just tried core reload - same error
17:49.05dwaynelorsungcu, 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
17:50.39dwaynewe are probably going to get a couple grandstreams, digium, and snoms to compare against polycoms.  Anyone recommend any other phones?
17:51.13lorsungcu_depends on what you need
17:51.59dwaynesip, not many line keys.  for call center agents that don't want to use a softphone
17:52.27[TK]D-Fenderdwayne, What do you need?
17:52.49[TK]D-Fender"Call center" can scale different ways.
17:56.11[TK]D-Fenderdwayne,  My call center people got top of the ilne Aastra 6739i's with Plantronic CS500 series headsets with EHS cables
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17:56.21dwayne[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-FenderLive 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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17:58.14dwayne[TK]D-Fender, thanks I'll add Aastra to the list of phones to compare.  I forgot about them.
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18:20.07Ice_StrikeIs it possible to return data back from AGI to dailplan?
18:20.52citrusfizzusing 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-FenderIce_Strike, End your AGI
18:22.56[TK]D-Fendercitrusfizz, No
18:25.31citrusfizzi'm no dev by any means, but you'd think more people would want that functionality than just me
18:25.43*** join/#asterisk newtonr (~newtonr@nat/digium/x-tpjpxgassgdbbokt)
18:25.43*** mode/#asterisk [+o newtonr] by ChanServ
18:25.51lorsungcucitrusfizz: what are you trying to do?
18:26.09[TK]D-Fendercitrusfizz, We do.  It's NOT happening.
18:26.34[TK]D-Fenderlorsungcu, he JUST told us what he wants to do.  4 lines ago.
18:27.39citrusfizzmust be pretty complex or some kinda security issue for it to NOT ever happen.
18:27.58citrusfizzoh well,  i will find another way around it.
18:30.44*** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com)
18:31.06[TK]D-Fenderset up peers for each of them.  have the reg to their respective peers.  DIAL() them together.
18:31.13*** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee)
18:36.30Qwellcitrusfizz: The point of registering is to tell the system where you are.
18:36.56QwellRegistering from multiple places simultaneously makes no sense.
18:37.07geeknationSetting up a new Amazon instance to see what permissions asterisk defaults to on sounds/en
18:37.27citrusfizzQwell: yeah but couldn't asterisk store multiple IP addresses and just ring all and send data to which ever responds to a pickup?
18:37.45citrusfizzQwell: theoretically i mean
18:38.11QwellNo, that would be silly.
18:39.48citrusfizzgood counter point
18:39.56citrusfizzlol
18:40.02QwellYou have a screwdriver.  You don't need a hammer.
18:41.49*** join/#asterisk CrazyTux[m] (~Brandon@wsip-68-15-80-34.oc.oc.cox.net)
18:41.55citrusfizzclearly
18:42.52[TK]D-FenderIt isn't "silly", and does have it's place.  Other SIP  proxies & registrar's handle this
18:43.10[TK]D-FenderHowever "generally unnecessary" about sums it up
18:43.39*** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee)
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19:07.24*** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de)
19:08.48*** join/#asterisk VultureZ (~Vulture@173-165-205-1-jacksonville.hfc.comcastbusiness.net)
19:09.45VultureZI 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?
19:10.55jpsharpYou can use ODBC or native MySQL, but to do native mysql, you have to install the addons packages to get it.
19:11.39VultureZOkay, 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"
19:12.15VultureZI do have extconfig.conf and res_mysql.conf configured, I will post the config on pastebin real quick
19:13.18VultureZhttp://pastebin.com/8qV8xdEM
19:13.39VultureZpretty streight forward, don't understand why the errors. This is the first time in a while I have attempted to use Realtime
19:14.05VultureZoh wow...
19:14.14VultureZ/etc/asterisk/res_config_mysql.conf is the new editing file....
19:14.25b0othave DUNDi working!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
19:14.28VultureZwell thanks for the ear jpsharp :P
19:14.42jpsharpNo worries.  That's all it takes sometimes.
19:15.07b0otahhhhhhhhhhhhhhhhhhhhhhh. nope. apply hadn't fully happened
19:15.10b0otstill broken
19:15.25mjordanVultureZ: you need res_config_mysql
19:15.41*** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com)
19:18.53VultureZstill giving the same error...
19:20.04mjordank.  Regardless, the res_config_mysql module provides the realtime engine, not app_mysql.
19:22.16VultureZhttp://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
19:22.25VultureZAccording to this it is res_mysql.conf in 1.6.x
19:23.10Qwellvoip-info.org is wrong about something?  Shocking.
19:23.15VultureZlol
19:23.44VultureZon the confluence wiki I was unable to find configuration information for Realtime 10, only information on the app
19:25.44b0otkeys need special permissions/ownership?
19:26.26*** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net)
19:27.49VultureZAccording to https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration it is res_mysql.conf not res_config_mysql.conf
19:27.59VultureZunless the driver is called "config_mysql"
19:30.04lorsungcudammit i am so sunburnt :/
19:31.04drmessanoConfigure Asterisk inside
19:31.13lorsungcui guess.
19:31.13b0ot"ALL CIRCUITS ARE BUSY"
19:31.16b0otnow
19:31.18b0otsorry caps
19:32.29lorsungcub0ot: you've really got to paste bin cli captures or something
19:32.34lorsungcuthat could be absolutely anything
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19:34.21VultureZokay, needed to build in support for res_config_mysql for realtime, which wasn't so that was my issue, fyi
19:36.17*** join/#asterisk ravnx (~mpalmer@dash.thecci.net)
19:37.22*** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com)
19:37.52Sean-DerHey does anyone know of a web based front end for CEL? I am gonna start to write one this weekend if there is not
19:38.39*** join/#asterisk din3sh (~din3sh@41.136.87.222)
19:38.44Sean-DerJust something really basic in PHP and a few nice things with jQuery just for the hell of it
19:39.04lorsungcusounds like a good idea
19:39.22VultureZSean-Der, you should consider node if you are looking to display it in realtime
19:39.24lorsungcui've been meaning to do something similar, but a bit more involved
19:39.59din3shdo i need kamailio as sip server with asterisk as dialer for a system of 600extensions and 300concurrent calls?
19:40.12VultureZwe 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)
19:40.43Sean-DerVultureZ: I am worried that if I move away from just a few PHP scripts it will scare people away
19:40.48din3shor can a single 2 x quad , 12gb ram memory server handle 600 registrations and 300 concurrent calls?
19:40.59Qwelldin3sh: it depends
19:41.03Sean-DerVultureZ: I have been meaning to look at socket.io though
19:41.17din3sh@Qwell: depends on?
19:41.21b0ot[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'!
19:41.22Qwelldin3sh: Everything.
19:41.23VultureZah 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
19:41.40din3shno transcoding , g711 calls
19:41.53b0ot[root@freepbx asterisk]# cd /var/lib/asterisk/keys/
19:41.54b0ot[root@freepbx keys]# ls
19:41.54b0ot10.1.2.100.key  box2.key  box3.pub  priv.pub
19:41.54b0ot10.1.2.100.pub  box2.pub  priv.key
19:41.56Sean-DerVultureZ: great now I wanna play with new tech because of you -_- haha
19:42.04b0otit has box3.pub
19:42.16b0otwhy is it saying no such key?
19:42.19VultureZ:) we just wrote a media streamer that takes the agen't screens a records them into the system for QA purposes
19:42.41drmessanob0ot:  Check permissions on the keys
19:42.49VultureZcombines the call file from * into the video recording and then strips out the "Privacy" areas
19:42.58b0otdrmessano, I made them +x
19:43.03drmessanob0ot:  Check permissions on the keys
19:43.09drmessano+x is meaningless
19:43.15Sean-DerI 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.
19:43.18VultureZSean-Der, I would recommend using express too for setting up the UI
19:43.20Sean-DerVultureZ: thats pretty cool
19:43.32b0otdrmessano, chmod +x gives them the ability to execute
19:43.44din3sh@Qwell: the server will have a quad E1 , 300 concurrent calls (g711), 600 sip registrations
19:43.46VultureZAll the large scale Queue guys do video/screen caps so it became a requirement
19:43.51drmessanoThey dont need to be executed.. they need to be read.. and ASTERISK needs to be able to read them
19:44.00b0otdrmessano, -rwxrwxr-x 1 asterisk asterisk 272 Sep 23 06:40 box3.pub
19:44.01drmessanoChrist.. Primer on +x.. really?
19:44.14din3shshould i keep all on the same box or split and use kamailio in front?
19:45.28din3shif one box, i was thinking about HP dl380, 2 x quad processor, 16 gb memory
19:45.46drmessanob0ot:  http://www.asteriskguru.com/archives/image-vp68776.html
19:45.46Sean-DerVultureZ: 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
19:46.02Sean-Deridk though. Thanks for the advice though! I am gonna pull this stuff in and play with it
19:46.21VultureZSean-Der,  are you looking for it to just parse from multiple servers and feed back to a console for your review?
19:47.12din3shSean-Der: have you been able to use CEL for proper billing purposes? (tracking attended xfers)
19:47.25Sean-DerVultureZ: All of my servers dump into one central MySQL table. So I don't have to worry about that
19:47.59Sean-Derdin3sh: 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
19:48.04VultureZokay then you just output that to your php app/script is your plan?
19:48.23Sean-Derdin3sh: So in short sorry I don't know :/
19:48.26VultureZnice, did you write a UI for them too?
19:48.35b0otdrmessano, I didn't add the key extensions
19:48.35din3shlol nvm
19:48.36din3sh:)
19:49.09Sean-DerVultureZ: 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!
19:50.08Sean-DerI 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...)
19:50.10VultureZso sad, there is so much effort on making good UI and they are all closed source... so is ours...
19:50.32VultureZits just sad that we can't put together a great UI... FreePBX tried but man that thing is full of holes
19:50.34Sean-DerVultureZ: I am hoping to make this CEL thing GPL3, but yea everything else is closed :(
19:50.40drmessanob0ot:  I am not google.. Google for the error, and see if you can find something that matches your scenario.  This helplessness is getting old.
19:51.01VultureZgoogle helps those who help themselves :P
19:51.15drmessanoIndeed!
19:51.18drmessanoPreach!
19:51.44*** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163)
19:51.46Sean-DerOr just change your nick and ask again :) haha
19:51.50drmessanoLOL
19:51.54din3shgoogle helps those who help themselves! amen to that
19:52.26drmessanoso /nick something else, up arrow twice, then enter?
19:52.26drmessanoBrilliant
19:52.26Sean-DerI swear that is what happens in the general programming channels in #php and #python it is so bad
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19:53.20jayteeI'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.
19:53.22drmessanoIt all equal PDIFM, via proxy ..
19:54.03VultureZI 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
19:54.28VultureZthen cry when they wonder why their Java project is still in QA 2 years after the deadline :)
19:54.41drmessanoThe big guys scoff at PHP because the world is being taken over from below by LAMP
19:55.17Sean-DerIts also annoying how people pull the whole 'lolphp' garbage
19:55.36Sean-DerIf you are gonna bitch about something, make your own damn DSL for the web.
19:56.03drmessanoPHP 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
19:56.17jayteeain't gonna bitch about Java but I will bitch about Starbucks. That coffee is bitter, overpriced swill! They should rename the company Fourbucks.
19:56.32VultureZhahaha
19:57.07drmessanoMicrosoft Linux 2015 <--- It's coming
19:57.17lorsungcunice.
19:57.18jaytee:-)
19:57.20drmessanoActually, it will have a cooler name
19:57.20VultureZno joke MS jumped on the NodeJS bandwagon too
19:57.26Sean-DerYEAR OF LINUX ON THE DESKTOP
19:57.30Sean-DerITS FINALLY HERE
19:57.44jayteeand the paperless office is right behind it!
19:58.18VultureZI 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
19:58.19Sean-DerNetcraft has confirmed it :D I am so glad I ditched Windows though. When I see people using PuTTY and Cygwin it makes me cringe
19:58.24drmessanoLike "Microsoft Windows 9 Open Edition"
19:58.25Sean-DerI feel bad for them
19:59.00_Corey_Sean-Der: Don't.  They had every opportunity not to screw themselves over
19:59.02Sean-DerVultureZ: 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
19:59.02VultureZI use Putty so I can look cool and have green text flying by my screen at starbucks... with my $4 coffee
19:59.07drmessano"Windows Clarity"
19:59.39VultureZdrmessano, sounds like a gastro pub
20:00.10drmessanoMicrosoft 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
20:00.58VultureZI don't have a lot of hope for Windows 8
20:01.06drmessanoWindows 9 will be $19.99, sold on TV, and include an Ubuntu CD "in case you don't like what we did there"
20:01.17VultureZit is cursed by the "mid" release that Microsoft is plauged by
20:01.24drmessanoYep
20:01.44navaismoAnyone here use the googleTTS.agi?
20:01.47drmessanoWindows 8 Vista Millenium edition
20:01.50VultureZWin 95 - (Win ME) - Win XP - (Win Vista) - Win 7 - (Win 8)
20:01.53[TK]D-FenderAlso known as the "Odd Numbered Start Trek Movie Curse"
20:01.58[TK]D-FenderStar*
20:01.59VultureZthey should just not even try on those releases
20:02.06VultureZhahaha
20:02.26VultureZdrmessano, ROFL
20:02.30drmessanoThe odd numbered Rocky movie curse
20:02.30jayteeI 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
20:02.42drmessanoAlso, the odd numbered Police Academy Curse
20:02.44VultureZoh gawd Silverlight...
20:02.57drmessanoSilverlight!
20:03.05HiveThe every one after the first Starship troopers curse
20:03.15drmessanoSilverlight is going to be on EVERYONE.. except for, well, everything
20:03.20jayteeMatrix 2 and Matrix 3
20:03.35drmessanoDie Hard past 2
20:04.11jayteeonly 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!"
20:04.30jayteebut yeah, all the odd number Trek movies sucked
20:04.45jpsharpStarship Troopers made me want to chew out my own brain.
20:05.09drmessanoWindows 8 is only going to suck because it will be the last non-linux version.  COME ON NOW, I CAN HOPE
20:05.22jayteethe novel by Heinlein is a good read but the movie sucked.
20:05.22jpsharp2013 will be the year of Linux on the desktop?
20:05.46drmessanojpsharp:  ${YEAR} + 1 is the year of the Linux desktop!!!
20:05.59Sean-DerEvery year is the year of Linux on the desktop!
20:06.02jayteeMicrosoft will call theirs WinNix
20:06.08drmessanolol
20:06.09Sean-DerWe are almost at 2% on some websites....
20:06.17Sean-DerMicrosoft better watch out
20:06.36drmessanoWindows X <--- There we go
20:06.38jpsharpBut then again, any movie that has Denise Richards in...makes me want to hit her with a baseball bat.
20:07.16b0otworks now
20:07.17b0otnbd
20:07.21drmessanoWindows X.. Feature Wine for legacy emulation.. "Because Wine does Win32 better than we do.  Fuck"
20:07.50Sean-DerI 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
20:07.57Sean-Derour*
20:07.58lorsungcudrmessano: wine is not an emulator :/
20:08.10drmessanoTrue
20:08.15drmessanoTechnically, it's not
20:08.28jayteeMS 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.
20:08.44lorsungcuthey replaced metro with something like
20:08.53lorsungcuwindows 8 new user interface version 1.0
20:08.59lorsungcuor something equally ridiculous
20:09.00[TK]D-FenderWine has been deprecated in favour of Bitch
20:09.09drmessanoThey are dropping Metro from it because metro reminds you of fast moving subway cars.. and well...
20:09.10Qwell[TK]D-Fender: I was hoping for moan
20:09.30[TK]D-FenderFinger is aplugin for that already
20:09.40drmessanoWine is being replaced by Cheese
20:09.42*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:09.43VultureZdrmessano, 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
20:09.56drmessanoYeah
20:09.57jayteeI hate IE9
20:10.11drmessanoI feel like I am just too close to love you
20:10.17drmessanoSo IE9, uninstall
20:10.39drmessanoI hate those commercials
20:10.54VultureZcan you uninsteall IE on US versions of Win7?
20:10.58VultureZ*uninstall
20:10.59drmessano"We now have an AWESOME BROWSER that looks like Chrome, which we know you love.. and we are cool now."
20:11.03lorsungcuyeah seems like many people are saying ie9 is good
20:11.06lorsungcui just don't see it
20:11.11Sean-Derdrmessano: It was a bait and hook for me lol
20:11.14drmessanoReminds me of Dad wanting to take me to see Guns N Roses.  No thanks
20:11.21VultureZlol
20:11.25lorsungcuid totally see gnu with my dad
20:11.30lorsungcugnr lel
20:11.33drmessanolol
20:11.43Sean-DerIn all honesty do you THINK anyone would be convinced by those commercials
20:11.57drmessanoMy Dad would be asking random people for "grass"... and then someone would remind him they haven't called it that in 30 years
20:12.52jayteeAnyone 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?
20:12.56drmessanoI think Windows Update forcing a browser upgrade is all the convincing most people are going to get.. The rest is penis waving.
20:13.20VultureZjaytee, tried... but it was made of plastic and wax...
20:13.25jayteelol
20:13.32drmessanoReally.. "WE MUST DOWNLOAD IE9 NOW BECAUSE ZOMG THE BOOKFACE LOOKS SO MUCH CLEANER"
20:13.33Kattyhi john
20:13.37drmessanoUh no
20:13.49Kattyhi danny
20:13.49VultureZBookFace!
20:13.57VultureZaka FaceHarvester
20:13.59drmessanoHi Angiebell
20:14.28Sean-Deraka Company that is dying on the public market
20:14.37VultureZIts 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
20:14.38Kattythere was an article on reddit that said some therapists will rate people as not having a fb as having pyschopathic tendencies
20:14.46jayteeI 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.
20:15.05drmessanoSocial networking is a great way to not have to be social or network at all
20:15.06Kattyjaytee: there's actually a mcdonalds video on youtube about exactly how they do that.
20:15.11Sean-DerKatty: I saw that also... but I wouldn't deny that I am social outcast either haha
20:15.17Kattyjaytee: it's very straight forward on how they do that as well.
20:15.23drmessanoI am starting a new website called "Outside" where people go and network
20:15.31jayteemeet ya there
20:15.36lorsungcuhow many people here have never had a fb account?
20:15.38KattySean-Der: a lot of people don't have fb accounts, and they're perfectly normal.
20:15.49Sean-DerThats what you think....
20:15.56Kattymy mother is normal.
20:16.00Kattyvery, very normal.
20:16.03lorsungcui am normal :>
20:16.04Qwelllots of psychopaths using facebook too
20:16.10KattyQwell: true story
20:16.10Qwelllike me.
20:16.15Sean-DerI haven't had a bookface, yourspace and what ever and I am fine without it
20:16.16VultureZobviously she is pyschopathic
20:16.30Sean-DerMy girlfriend wastes so much time on that stuff
20:16.37VultureZalong with most social outcasts (developers)
20:16.42jayteeI don't believe normal exists
20:16.51drmessanoI 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.
20:16.55VultureZisn't normal what you precieve as the baseline?
20:17.22VultureZdrmessano, well that is perfectly normal since you do it on Facebook
20:17.26jpsharpSean-Der: My stepmother created half a dozen accounts so she could play all the gambling games.
20:17.28drmessanoAbsolutely
20:17.28Sean-DerI live in the United States.... I am ok not being 'normal' by my countries standard
20:17.49Sean-Derjpsharp: That makes me believe that you humanity is gonna be alright :')
20:18.09jpsharpAccounts for her, my dad, each of the two dogs, their cat, and i think a fish.
20:18.09VultureZwhy do you need 6 accounts to gamble on FB?
20:18.17VultureZoh they get free $$?
20:18.20Sean-DerHAHAHAHAH
20:18.23Sean-DerThat is awesome
20:18.44VultureZthe 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
20:18.53VultureZ;)
20:19.04jayteeI miss George
20:19.08Sean-DerCarlin and Hicks are gods among men
20:19.09jayteeI love his take on God
20:19.33[TK]D-FenderSean-Der, And if they're wrong ... men amongst God ;)
20:19.49drmessanoOnline 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.
20:20.10jpsharpI 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.
20:20.16jaytee"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!"
20:20.19jpsharpNever doing that again.
20:20.27drmessanoGamblingPlugin 1.0 by FreeSmileysAndSpam4All <-- Sounds legit to me
20:20.33Sean-Der[TK]D-Fender: Hahah that is awesome.
20:21.17KattySean-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)
20:21.23KattySean-Der: some people are just more sociable than others.
20:21.29geeknationI'm hitting an error on require 'ruby-agi'  in irb
20:21.31geeknation
20:21.38KattySean-Der: i don't really consider it a 'waste' though. i very much enjoy it
20:22.03lorsungcuanyone 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^
20:23.08lorsungcuthanks
20:23.40b0otit was the keys
20:23.41b0otall along
20:23.55Sean-DerKatty: 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-Fendercheckout time, BBL
20:24.05lorsungcuany idea on how much cpu/io it takes to use the function, TK?
20:24.09lorsungcudammit.
20:24.11lorsungcuanyone?
20:24.55lorsungcucustomer is worried that checking whether a file exists will destroy their disk bandwidth
20:25.11lorsungcui don't really have any evidence that it won't, except that there are so few checks, relatively
20:25.12b0otwhat are you scripting in?
20:25.16lorsungcudialplan :p
20:25.18Sean-DerSTAT is not that expensive, don't know how Asterisk implements it though
20:26.57b0otlorsungcu, http://www.voip-info.org/wiki/view/Asterisk+tips+fileexistance
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20:27.12lorsungcuthanks
20:27.38lorsungcuyeah I'm using a system() call now.
20:27.53b0othttp://www.voip-info.org/wiki/index.php?page_id=2981&comments_page=1
20:34.27thecardsmithlorsungcu: 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
20:34.46thecardsmithnotes that this is gut instinct
20:34.48lorsungcuyeah its only checking ~50 files
20:36.11thecardsmithhow often is it changing that they could or couldn't be there?
20:36.42thecardsmithcause, you could check 'em and put it in astDB, every... say "now and again". Instead of say, "with each and every call"
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20:37.22thecardsmithbut, i don't know what other requirements are tied to the existance/non-existance of those files
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20:49.46VultureZRealtime, why do you forsaken me!
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20:54.33lorsungcuI'm working on a project now that i wish i had done using realtime
20:54.35lorsungcusuuuucks
20:55.18jpsharpI did that years ago.  I constantly had to regenerate sip.conf and extensions.conf.
20:55.36lorsungcuwhat do you mean?
20:56.32jpsharpI 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.
20:56.46lorsungcuah
20:57.58jpsharpTook entries out of a database, regenerated sip.conf, extensions.conf, and TFTP config files for 600 Cisco 7940s.
20:58.09lorsungcuick
20:58.58VultureZwell the extensions is still going to be .conf but queues/vm/peers are all moving to realtime
20:59.48jpsharpI finally broke down and converted *everything* to a database back end.  Asterisk, BIND, TACACS, and RADIUS.
20:59.53jpsharpThat was a long week.
21:00.02lorsungcuyeah i have a feeling that will be happening to me.
21:00.03VultureZI bet
21:00.18VultureZthough its also a good feeling it means the biz is growing :)
21:00.32lorsungcui suppose
21:00.36lorsungcujust wish there were 3 of me
21:00.37lorsungcuheh
21:00.41Hivelol
21:00.51jpsharpI quit two months and 5 bounced paychecks later.
21:00.57lorsungcu:p
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22:40.55geeknationHey -- Anybody know why Amazon might block agi.wait_for_digit
22:41.39[TK]D-Fender... pardon?
22:43.28geeknationSorry. 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-FenderAnd you have just claimed AMAZON is "blocking" it......
22:44.08geeknationExactly the same script worked fine on Rackspace
22:44.43geeknationit might not be amazon…just kinda new… wanted to get your opinion on what it might be
22:44.50[TK]D-Fendermaybe * 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-FenderSince you haven't shown us anything, there is no opinion to give except awe at the one leap that was presented.
22:47.04geeknationexcuse me
22:47.06geeknationlets start over
22:47.29geeknationI'm running a ruby-agi script via my dial-plan and am looking to wait indefinitely for digits
22:47.48geeknationwhen i call into my number everything goes through fine, but my key presses aren't going into the system
22:47.51[TK]D-FenderOk, if we're starting over.... first SHOW, then "tell"
22:47.54[TK]D-Fender~pb
22:47.54infobotA "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.
22:47.55geeknationmy agi debug is : http://pastebin.com/Ax8SA51w
22:47.56[TK]D-Fender^^^^^
22:48.22[TK]D-Fendershow 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-FenderShow 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-FenderPacket 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-FenderYour entire call is getting packet timeouts
22:51.32[TK]D-FenderIt is screwed
22:51.51[TK]D-Fenderthis 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-FenderYou have netowrking issues
22:54.00geeknationhere's my agi-script - sorry for the delay: http://pastebin.com/1yHuxKAw
22:54.05geeknationwhat do you mean networking issues?
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22:56.08lorsungcugeeknation: have you read https://wiki.asterisk.org/wiki/display AST/SIP+Retransmissions ?
22:57.14[TK]D-Fendergeeknation: * can't reliably talk to ipkall
22:57.25[TK]D-FenderPackets.  Aren't.  Making.  It.
22:58.55geeknationi'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?
22:59.24lorsungcucheck the firewall on your instance
23:03.03geeknationahhhh…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-Fenderthis isn't just DTMF
23:05.38[TK]D-Fenderprove what is and is not being blocked through the ENTIRE path
23:05.51[TK]D-Fenderdouble check EVERY setting you have any direct influence over
23:06.52geeknationok. 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
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23:10.20[TK]D-FenderVerify if you left any SI setting in place from old configs like externIP, etc....
23:10.24General_Z0dleifmadsen, what is the link to your tutorial again i forgot
23:10.26[TK]D-Fenderanything that might not add up"
23:15.17[TK]D-Fender~book
23:15.17infobotAsterisk: 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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23:49.57General_Z0dis away: taking a small break then i will resume
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23:51.16geeknationlorsungcu Thanks all for the advice. It is an IPKall issue as my sip gate number is going through no problem
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