IRC log for #asterisk on 20110513

00:00.41*** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap)
00:06.58*** join/#asterisk K3rmit (~asdf@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com)
00:07.10K3rmitwhat does exten  => a,1,VoicemailMain
00:07.10K3rmit<PROTECTED>
00:07.14K3rmitdo?
00:07.46pdtpatrick_Question: besides extensions.conf .. where else is ivr related information set? im seeing something like this: exten => s,n(ivr),Background(ivr-farheap-hello)
00:10.35pdtpatrick_what does ${ARG1} refer to? I understand it is argument 1 but is it an argument from user input or something else?
00:12.04paulcK3rmit: It says if you press * during voicemail greeting, go run the voicemail login/msg retrieval app
00:12.40paulcpdtpatrick_: ${ARG1} is usually used if you're in a macro and passing variable data in
00:13.54pdtpatrick_i c
00:14.12pdtpatrick_im trying to find out where i can information related to the ivr
00:16.27*** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net)
00:17.30kukuHow do I forward caller ID number when forwarding calls int he dialplan via a trunk ?
00:23.09WIMPykuku: By not changing it.
00:23.19WIMPyi.e. it happens by default.
00:23.47kukubut this dialplan somehow changes it :( What are my options
00:24.05WIMPyRemove the part that changes it.
00:24.36WIMPyOr it gets set in the peer definition.
00:25.42kukuObvious answeres - but they helped. I need to do more tests - thanks !
00:25.42*** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap)
00:25.43*** join/#asterisk docid (~PISSS@s76-9-57-75.nt.northwestel.net)
00:27.14docidanybody got any ideas what would be causing calls between extension on the same system (same switch even, no natting, local server, etc) to have no audio for the first 20 seconds and then work just fine, and yes, every call between internal extentions, calls to and from the outside work fine, all phones and server locked down to ulaw, * ver 1.6
00:27.47paulcpdtpatrick_: You're going to need to be a bit more specific, because none of us know what you're doing with IVR. The dialplan defines a sequence of steps, which can play prompts, collect digits etc..
00:27.57kukucanreinvite = ?
00:28.14docidkuku, that diested at me?
00:28.17dociddirected
00:28.55kukudocid: yes
00:29.00*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
00:29.12docidset to no, trying it with yes
00:29.25*** join/#asterisk mateu (~mateu@missoula.org)
00:30.09docidno change
00:31.08docidit would at least make sense if it was audio, or no audio, but the big delay is confusing.... nothing else on that lan other than phones and asterisk
00:31.37kukuwhat phones
00:32.13*** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net)
00:32.49docidaastra 6731i's and 6757i's
00:33.15docidunfortunantly we dont have any other brands available for testing onsite
00:33.23kukuwhat about x-lite
00:34.00kukucan you show your sip.conf ? ( or at least the definition for you extension(S))
00:34.21pdtpatrick_paulc: here's what i have so far
00:34.21pdtpatrick_http://pastebin.com/mhLQZh2D
00:34.36docidyep, lemme go a diggin
00:34.48pdtpatrick_im trying to follow it to figure out what file is it accessing to play the greeting
00:35.40docidwell, would be in a standard centos install in /var/spool/asterisk/sounds/
00:36.02docidand the name of the playback file with the extention matching the codec of the channel
00:36.03paulcpdtpatrick_: Line 7, (ivr) is just a label. It's the Background app that's playing audio. The file is ivr-farheap-hello. Then you're waiting for 8 seconds for a digit. Which is weird, cos line 6 sets first digit timer to 4 seconds, and line 5 sets a 3 second interdigit timer
00:36.15*** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
00:37.13paulcpdtpatrick_: And if I was going to be anal, I'd say it's shit to answer the line, then simulate ringing for X seconds. Because the customer is paying. I'd do Ringing, then Wait, then Answer - it's more correct from a telco billing perspective.
00:39.38WIMPyWouldn't a Wait() without Answer() imply Ringing()?
00:39.47pdtpatrick_Thanks.. im trying to pick up where someone left off really
00:40.10pdtpatrick_@paulc: how then can i follow where ivr-farheap-hello is set? just search around the same file ?
00:40.31WIMPyIt is the file name.
00:40.59paulcpdtpatrick_: Anything that looks ${LIKETHIS} is a variable. Things like ivr-farheap-hello are probably filenames.
00:41.42paulcso in your example, ${WAITTIME} is probably set somewhere else, either via Set(WAITTIME=5), or maybe just WAITTIME=5 in your [globals] context/block.
00:41.48WIMPyThe name of the sound file that's played.
00:42.11pdtpatrick_i c .. nice it is all starting to make more sense
00:42.28WIMPyHave you tried the book?
00:42.33WIMPy~newbook
00:42.34infobotPlease see ~thebook for more information about Asterisk: The Definitive Guide
00:43.05WIMPyThat's what happens if you haven't been here for a while.
00:43.12WIMPy~thebook
00:43.12infobotAsterisk: 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook
00:43.59pdtpatrick_getting that book now
00:44.02pdtpatrick_thanks guys!
00:46.18pdtpatrick_while i wait for book to arrive. that ivr-farheap-hello file is probably in /var/lib/asterisk/sounds ??
00:46.38WIMPyyes
00:46.47docidkuku, heres sip conf with a couple of the extensions ...  http://pastebin.com/TS9MrXwA
00:48.11pdtpatrick_ha! thanks guys
01:05.41docidkuku, any ideas? im open to random suggestion that might have some effect, or lead me down the path towards possible solutions
01:09.54*** join/#asterisk De_Mon (de_mon@fl-71-49-12-102.dhcp.embarqhsd.net)
01:11.17De_Moni'm looking to build a system that can handle 200 concurrent SIP calls using ulaw for a codec. What sort of minimum should I start testing with?
01:12.02WIMPyImpossible to say. It depends on what you want to do with the calls.
01:12.28De_Monjust passthrough
01:12.46De_Moni figure a P166 is too slow to start with but I really have no idea how high to go
01:13.32De_Monlike, is a a dual quad core xeon over kill? or a good place to start
01:13.34WIMPyIf you use directmedia, you probably don't need much more.
01:14.11WIMPyIf you really don't want to process voice in any way, that overkill.
01:14.40De_Monreally? I was looking at a switchvox system and it said a maximum of 75 concurrent calls so I was a bit puzzled
01:14.56De_Moni guess they assume you're going to use all those features...
01:15.16WIMPyYou often do.
01:15.30WIMPyLike VoiceMail e.g.
01:16.58De_Monin the case where you are using a feature such as whisper or queues those a low cpu hit while things that do transcoding are where the cycles go?
01:17.24De_Monthis is a callcenter scenario where there's not going to be an IVR and just people making outbound calls for the most part.
01:18.02WIMPyIf you want to spy or whisper, you need access to the media stream.
01:18.33WIMPySo you need to shift the data through the box, even if it needs no processing.
01:19.33WIMPySo that's not much more than forwarding of network traffic.
01:19.57De_Monwhich is relitivly light on processing power
01:20.07WIMPyyes
01:20.23*** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee)
01:20.31WIMPyTranscoding is the worst CPU wise.
01:20.37De_Monwell I feel a lot better. a switchvox appliance would probably work just fine in that case.
01:21.12De_Monnow to decide if we need to spend $18,000 for a GUI =)
01:21.21WIMPyI don't know what they are like.
01:21.31WIMPyErr?
01:21.59De_Monwut?
01:22.04WIMPyTake the book and skip the GUI idea.
01:22.26docidive been throughly impressed with the atom 525 boxes weve been building, but never tested them with a large callload
01:22.48De_Monwell, it's between a gui that any idiot can use or taking ownership of the system and managing the configs myself.
01:23.07WIMPy525 is dual core, isn't it?
01:23.15docidtip: any idiot cant take the reins of asterisk and expect it to keep running even with a gui
01:23.19docidyeah wimpy
01:23.33WIMPyindeed
01:24.01docidand when it breaks, youll be very thankfull you know how to run it without one
01:24.11De_Monwell, let me rephrase that and say any windows admin can use it
01:24.15WIMPyyes
01:24.28docidDe_Mon, im not seeing the difference
01:24.31De_Monhahaha
01:24.40De_Monmy windows admins would take offense to that
01:25.10docidthat being said, i do keep freepbx on most my boxes so people can use the web interface for their voicemail, and csr's can make small chanes
01:25.33De_Monis virtualization worth considering? i haven't done any research on that topic in a good 3 years
01:25.58FreeaqingmeDe_Mon, yes, but you cant virtualize fxo/fxs cards so if you're planning on using those it isnt worth it
01:26.03WIMPyIt works for some...
01:26.41docidwell, qemu with the device directly assigned to the vm i would think ya would have a good shot
01:26.42De_Monnope, pure SIP. But last time I tried it there were timing issues
01:26.50docidbut i could see timing issues popping up
01:27.09docidDe_Mon, you can buy timing dongles
01:27.16dociddunno if that would help though
01:27.23Freeaqingmeyou can probably overcome those by assigning dedicated cpu cores to the vm
01:27.48De_Moni doubt it, mapping local hardware from Virtual host to a VM is kinda spotty
01:28.28De_Monwe decided it was problems with ztdummy and a virtualized usb port or somesuch
01:28.28dociddepends what hypervisor ya got behind it
01:28.43Freeaqingmedocid, name me one with which you can do that with pci-e devices?
01:28.45De_Monlatest vmware
01:28.59docidwouldnt know, havent moved to any pci-e servers
01:29.02docidheheh
01:29.07docidwe use used hardware
01:29.11docidmostly
01:32.49docidhavent really found a good reason to virt an * box
01:34.15Freeaqingmea reason would be to put multiple boxes on the same hardware
01:34.21Freeaqingmeto reduce costs
01:34.28De_Monor just a lower datacenter footprint
01:34.42Freeaqingmethat'd be a result
01:34.50De_Monwe condenced about 3 racks into 10U this year with virtualization
01:34.59docidaye, i wasnt saying there werent good reasons... i just said i hadent found one in my experence
01:35.05docidnice
01:35.07De_Monwith the added benifit of not having to worry about equipment becoming obsolete
01:35.12docidbeen meaning to dig in again
01:35.17docidbeen away for a while...
01:35.35FreeaqingmeDe_Mon, yeah, same for us (approx), but the most important thing is that we're way faster in recovery with hardware issues
01:35.45De_Monit sucks when you have 10 servers with hardware warranties about to expire that you can't renew much less find parts for on ebay =(
01:39.05docidis fantasizing longingly about hardware with warranties
01:40.07De_Monyou have no idea what you're missing!
01:45.35*** join/#asterisk kaushal (~kaushal@115.246.154.6)
01:45.41docidhrmm, so any ideas on why today my phones started doing this odd thing where when ya call another extention on the same system (no nat, seperate network/switch/etc)  will not pass audio for the first 12-20 seconds of the call, calls from outside trunks are fine
01:46.08kaushalwhat programming language is being used to develop Asterisk ?
01:47.17De_MonC# !
01:47.26De_Monwohoo i have a status line in vim again
01:47.42seraphielol
01:47.48seraphiekaushal: C
01:48.11kaushalC or C++ ?
01:48.16kaushalC# ?
01:48.49seraphieC
01:48.53kaushalok
01:49.00De_MonC# was a joke
01:49.04kaushal:)
01:49.11seraphieI believe there are some small pieces in C++, however, I have no experience with those.
01:49.21seraphiethe vast majority at least is C.
01:49.22kaushalok
01:50.26De_Monall you gotta do is look at the source to see if they are c or cpp
01:50.37kaushalok
01:51.25seraphiekaushal: http://svnview.digium.com/svn/asterisk/trunk/main/
01:54.55docidok, so ive traced it down i believe, but it seems to indicate another issue... (issue being calls between extensions have no audio for first 12 seconds) ...   apperently the audio does not start untill the rtpkeepalive timer is hit, but im not natting, so this shouldnt have an effect right?
02:01.04kaushalseraphie: Thanks a lot
02:01.09kaushalDe_Mon: Thanks
02:06.43*** part/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu)
02:07.28kaushalseraphie: is it easy to learn to C :)
02:07.46kaushalC is so powerful still its being used
02:09.04seraphieSure, it's easy to learn C. It's not easy to learn C on the Asterisk codebase.
02:09.18florzno, it's not easy to learn C, quite to the contrary
02:09.30florzas long as you think it is you probably don't know C
02:11.06docidhides his battered dusty scrolls of pascal
02:11.20florz*g*
02:11.59seraphieOK, yeah, so "easy" is a bad word.
02:12.24seraphienevertheless, do not try to learn C by developing Asterisk.
02:16.21seraphieLearning C is a worthwhile pursuit.
02:21.00*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net)
02:25.03kaushalseraphie: yes
02:25.12kaushalI agree to it
02:25.30kaushalflorz: I also feel the same
02:25.40kaushalC is very hard to learn
02:25.45De_Moni learned C while fixing bugs and developing small featuers in asterisk.. alas thats as far as I got.
02:25.48kaushalatleast for me
02:27.53*** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net)
02:33.13*** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net)
02:45.06russellbseraphie: why not learn C on Asterisk?  :-)
02:48.02Maxus2hi people is this valid for a sip connection string in asterisk? SIP/[username]:[password]@[host]:[port]/[extension]
02:48.53russellbcheck the top of sip.conf.sample
02:49.00russellbit documents the valid SIP dial strings
02:49.25Maxus2i have doen that, none appear to include the extension
02:49.34Maxus2is putting the extensiont here valid/
02:49.35Maxus2?
02:50.15Maxus2the only ones that have extension, dont include username, password and host
02:51.35russellbtry it and see what happens
02:51.55Maxus2did, didn't work
02:52.05russellbk, then I guess it's not valid
02:52.09Maxus2but im not sure if im missing somthing else
02:52.23Maxus2well then can you tell me what would be valid?
02:52.34russellbwhat you see in sip.conf is the documentation of what's valid
02:53.07Maxus2then you telling me it is impossible to pass an extension when using a username, password and host?
02:53.43russellbI guess so.
02:53.59Maxus2yeah, im not buying it
02:54.04russellbok.
02:54.08Maxus2if you dont know, just say so.
02:54.50russellbi pointed you to where the documentation was for what is supported.
02:55.04Maxus2thanks.
02:56.54WiretapWorkMaxus2, putting the extension there is valid, so long as the other end expects it
02:57.07Maxus2thanks WiretapWork
02:57.16WiretapWorksipgate use exactly that format
02:57.19Maxus2that is a useful answer.
02:57.32russellbwhat is your problem?
02:58.01Maxus2im dial that string and the other asterisk box is simple returning:
02:58.43Maxus2Got SIP response 503 "Service Unavailable" back from *.*.*.* (ipaddress removed)
02:59.10Maxus2i managed to get it to work once, but not sure if the structure of my dial command is correct
02:59.23WiretapWorkhang on a tick
02:59.31WiretapWorkthe syntax you posted above is for a register string
02:59.35WiretapWorkit is NOT for a dial string
02:59.37Maxus2oh
02:59.49Maxus2that could be my problem :)
02:59.49WiretapWorkthe dial string is SIP/<EXTEN>@<HOST>:<PORT>
03:00.02WiretapWorkyou can't use a username and password in a dial string
03:00.04WiretapWorkno such thing
03:00.17Maxus2what if im dialing to a remote box?
03:00.22russellbyou can, but it doesn't support also including the extension ..... as the documentation says
03:00.23*** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net)
03:00.31WiretapWorkMaxus2, one that requires registration?
03:00.48Maxus2im having to do it without registration
03:00.56WiretapWorkMaxus2, the SIP DIAL format never errs from the syntax I supplied
03:01.01Maxus2just basically dialling and ip with a password and user name
03:01.06WiretapWorkyou can't do that
03:01.34Maxus2so two boxes must be register in order to pass calls?
03:01.38Maxus2registered
03:01.59WiretapWorknope
03:02.05*** join/#asterisk jizzzum6 (jizzzum6@peacekeeperv6.darksideresearch.com)
03:02.16WiretapWorkif they have peer definitions for each other that are _Static_ then no
03:02.21WiretapWorkif theyre _dynamic_ then yes
03:02.28WiretapWorkbut obviously one must be a static host
03:02.35WiretapWorkotherwise they won't be able to find each other at all
03:02.54Maxus2hmmm okay
03:03.12Maxus2will have a play and see how i go, thanks for the help.
03:04.19*** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew)
03:04.41russellbthe syntax says ... SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
03:04.55russellbthe username part is slightly misleading.  that's the user part of the URI, not the authentication username.
03:05.25russellbso SIP/EXTEN:mypassword::myauthenticationusername@ipaddress
03:05.47russellbthat should work.
03:05.50Maxus2cool but still no extension
03:05.56Maxus2oh wait
03:06.01Maxus2sorry missed that at the start
03:06.16russellbyeah, it's the user you are dialing
03:06.23russellbor extension, whatever you want to call it
03:07.19WiretapWorkrussellb, bear in mind that you'll probably be treating that dial as from external, so you'll have to assign a 'VDDI' for the call
03:07.59russellbis off to bed
03:08.01russellbgood luck.
03:08.42Maxus2thanks!
03:14.14Maxus2yeah no luck with that
03:14.40Maxus2im using realtime and have been told i cant do registrations in the database
03:14.55Maxus2so im trying to do it in the dial string, so i dont define things in the sip.conf
03:15.03Maxus2no luck so far.
03:22.01stopeI'm having fax issues: I have a linksys 2102, * 1.8.4 with the digium FFA module and keep getting this: T.38 re-INVITE detected but no fax extension
03:22.32stopefaxing in from the pstn to * works over 711, but faxing from the linksys 2102 to * if iffy
03:22.47stopeanything obvious that I'm missing?
03:25.56WiretapWorkit seems to be looking for a fax extension to send the call to
03:26.00*** join/#asterisk OldGrumpy (Whacko@p5B312AEB.dip.t-dialin.net)
03:35.57stopestupid question coming..... whats a fax extension?
03:36.19stopeI just want to get it from my faxing machine to the server and land as a tif
03:40.21*** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com)
03:49.39*** join/#asterisk dinesh___ (~dinesh@46-126-192-144.dynamic.hispeed.ch)
03:50.13dinesh___hey all, I'm setting my asterisk server up again (and migrating from 2.4 to 2.6), everything is working as before, expect 1 little thing
03:50.49dinesh___playback() claims that it is currently playing my .gsm file, but i don't hear anything on the line (SIP)
03:51.07dinesh___voice does go through
03:51.20dinesh___any idea where this could come from?
03:55.04*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cmqmjiixsapxteym)
03:56.44WiretapWorkdinesh___, asterisk hasn't even reached version 2.0 yet???
03:56.46*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
04:01.05dinesh___oh yeah right, i meant 1.6
04:03.08dinesh___well well i don't know what's going on, but the "reload" command in the asterisk -r console is not found anymore
04:05.07WiretapWorkdinesh___, core reload
04:05.13WiretapWorkits not just one reload now
04:05.16WiretapWorkthere are several
04:10.26*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
04:11.20dinesh___actually
04:11.23dinesh___i fixed both issues
04:11.45dinesh___the thing is that i deleted all the files in /etc/asterisk, and left only sip.conf, extensions.conf and modules.conf
04:11.52dinesh___because the other ones i have no clue what they are doing
04:12.12dinesh___this approach was working fine with asterisk 1.4, but it doesn't look like it's similar with 1.6
04:12.34dinesh___now that i put the default files back, and overwrote only sip.conf and extension.conf it's all ok
04:14.22dinesh___hm well no, that was just for the "reload" part
04:14.32dinesh___removing the "noanswer" from my playback fixed the playback issue
04:14.35dinesh___but that's not optimal
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06:01.28ectospasmI have a quick question:  suppose I'm a SIP server (not necessarily Asterisk), and in the normal INVITE process I send a 183 Session Progress message.  Shouldn't I send a 200 OK that should be ACK'd before I send out RTP?  My customer is using Asterisk to send to a remote SIP provider, and I see <183 followed by <RTP, with no <200 OK or >ACK, then the far end sends <403 Forbidden
06:02.21ectospasm'<' meaning coming from the SIP provider, and '>' meaning coming from Asterisk (my customer's machine)
06:06.50jizzzum6my brain hurts
06:11.24kaldemarectospasm: in a case of early media, RTP might begin before the 200 OK.
06:11.54ectospasmkaldemar: yeah, I was thinking that might be the case.
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06:29.01schmidtsgood morning
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06:36.50oejGood morning
06:37.03X-RobHeya oej
06:38.02X-RobI need to have a good look at applerasin in the not too distant future.
06:38.10oejGreat
06:38.30oejThe FreePBX people have done a lot of testing of it lately
06:39.08X-RobYeah, I got them onto it.
06:39.26X-RobI'm not MEANT to be doing any work until 1st July
06:39.34X-Robbecause I'm not getting paid for it
06:39.38schmidtsMorning olle
06:39.43X-Robbut I've been playing with a pile of astribanks
06:40.07X-RobA literal pile: http://hipbx.org/sites/default/files/styles/large/public/field/image/IMAG0258_0.jpg
06:40.18schmidtsoej have you seen the reply from siemens to the problem we had yesterday? issue 19281
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06:40.40oejschmidts: no
06:41.03oejx-rob: Yeah, that's a huge pile
06:41.42X-RobIt's (un)surprising how many things break when you have that many.
06:41.50X-Roblike I can't turn them all on at the same time - half of them crash.
06:41.59schmidtsmaybe you can take a look, i am not sure if they didnt understand the rfc wrong or if maybe asterisk does something wrong ;)
06:42.02X-RobI'm mildly irritated.
06:42.11X-Robbut I'm being nice and have emailed support.
06:44.42X-Robschmidts, I may be stupid, but what's the point of a device sending a media line and then saying 'DON'T USE THIS'.
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06:45.15X-RobI've alway ssaid that SIP is a protocol written by a committee.
06:45.17X-Robheya coppice
06:45.18oejschmidts: One always learn. I need to check if they're right in the assumption of order of preference though. Have never heard that.
06:45.25schmidtsX-Rob dont ask me, thats siemens style ;)
06:45.37oejThe a=sendrecv is definitely wrong, they don't comment on that
06:45.49coppiceSIP is a protocol written by people who had never seen a telephone
06:46.02oejcoppice: What's a "telephone" ?
06:46.17schmidtscoppice so you mean h323 is better than?
06:47.02coppiceH.323 builds on 100 years of learning what works well in telephony. The IETF wanted non of that perversity in their network
06:51.11coppiceSIP is so horrible, they had to write the MGCP specs just to make it look good by comparison
06:54.51X-Robyou know, I'd never actually LOOKED at the MGCP protocol until just then, coppice
06:54.54X-Robnow I need eye bleach.
06:55.02wdoekes2good morning
06:56.04schmidtsmorning walter
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07:06.43jacc0gooooood morning all!!
07:06.45jacc0:)
07:08.58schmidtsmorning jacc0
07:09.19jacc0I've been reading some RFCs last night and i hate to admit that siemens is right
07:09.26jacc0:P
07:09.52jacc0there responce was a copy/past from that rfc
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07:13.04jacc0so if the port is ZERO we MUST NOT use it
07:15.02schmidtsjacc0 which RFC?
07:15.46jacc0RFC3264
07:16.44jacc0end beside that; asterisk is doing something else wrong; the first one should be the preferd
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07:19.44jacc0and ONLY when streams of DIFFERENT types are present it means that it wants both; not when they arew the same type
07:20.50jacc0So I'll try and make a patch that does exactly that
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07:30.17schmidtsjacc0 i dont think this will be solved by a simple patch cause its a deep infrastructure thing
07:30.35schmidtsonly using the first media descriptor will not work, you have first check if asterisk could even handle this
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07:34.42IneluctableI want to install asterisk 1.8 on a CentOS 5.6 fresh install.  I would like to use chan_gtalk.so, and res_jabber.so.  My question is can i install these modules if I install asterisk from the asterisk/digum repo?  Or is the only way to use these modules is to compile from source?
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07:35.26ChannelZthey really need to be compiled from the same source as asterisk was
07:36.07jacc0@Ineluctable: I'm not sure. but what I usualy do do is : install from repo (to get things like the astcanary and dependencies and stuf installed)
07:36.20jacc0then download source en compile:
07:36.30jacc0./configure
07:36.42jacc0make menuselect
07:36.44jacc0make
07:36.46jacc0make install
07:36.57ChannelZthere's not much point in using the packages at that point
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07:38.00jacc0why not? it creates the user/group and installs astcanary en runs asteriks in realtime prio; do you want to do this all manualy?
07:38.04IneluctableWhat about the dep factor as jacc0 pointed out?
07:40.00wdoekes2why realtime/canary? and useradd is not too hard to do.. I'd advise against having an asterisk installed by package next to a manually installed one
07:40.12ChannelZI don't like mixing a package and then building from source yourself, as when the package gets updated it could stomp on or break your current setup
07:40.21wdoekes2exactly
07:40.31ChannelZDo one or the other.
07:41.50ChannelZIf you want to install the asterisk *source* package to let it fetch other development dependencies you might need out of laziness, fine..
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07:48.24IneluctableChannelZ: asterisk18-devel ?
07:48.59ChannelZmaybe, I'm not familiar with CentOS and it's package system and what they might call things
07:49.48ChannelZUsually 'devel' on other distros are headers/libs needed for developers to use whatever it is, and there is a 'src' or 'source' version which is the actual source code for a package.
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07:51.58IneluctableOK. If that is the case why cant I just install asterisk18 from the repo with asterisk18-devel do the ./configure make menuselect deal to add the modules?  Would this not work, or would it recompile everything?
07:53.53jacc0after make menuselect you should recompile: make
07:53.59ChannelZLet's back up for a second - have you looked, are other channel modules separate packages?
07:54.00jacc0and install : make install
07:54.46ChannelZand/or are you sure that particular package doesn't already have the gtalk channel driver?
07:56.39Ineluctableasterisk18 does not cone with it afaik
07:56.42kaldemarat least the .deb packages from digium already have chan_gtalk and res_jabber.
07:56.44Ineluctable*come
07:57.45IneluctableI will give it a try and see.
07:58.45ChannelZThey are included in the base source but aren't built by default - however most people who build binary packages will build separate packages for 'optional' modules like this, if they didn't just build it into the main package (since it's a loadable module in Asterisk anyway, there's no harm in supplying it with the package..)
08:00.18Ineluctablehow?
08:00.39ChannelZhow what?
08:02.13Ineluctablewere you saying if it is not built into the package then build it myself?
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08:04.50ChannelZI'm saying do you even know that it isn't before we go down this road?
08:05.11ChannelZI'm browsing an RPM searcher thing on the net and it looks like it already is.
08:05.59ChannelZthough I don't know exactly what package from what repository you'd actually be getting
08:06.28Ineluctableasterisk / digum repo
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08:11.58ChannelZit looks like those modules are already built in the main package
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08:14.30IneluctableWell after installing asterisk 1.8  asterisk -rx "module show"  only shows  res_adsi                       ADSI Resource                            0
08:14.54ChannelZthat only means it's not loaded probably because it's not configured
08:15.24Ineluctablewhat res_adsi or chan_gtalk?
08:15.49ChannelZgtalk.  Look in /usr/lib/asterisk/modules (hopefully)
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08:17.11Ineluctableno chan_gtalk, or res_jabber
08:17.31ChannelZhmm then I don't know where that package came from
08:17.55ChannelZI just looked at one of the .deb files from digium and it's in there
08:18.56IneluctableYeah nothing for centos. That is why I am not using asterisknow
08:19.51ChannelZthen I'd just remove the package, download the source and build the whole thing.
08:20.33ChannelZThere aren't many dependencies for the base of asterisk besides the development tools (compiler et al).  For gtalk/jabber in particular you need libiksemel-dev
08:21.37Ineluctablealright I will give that a try.
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09:01.02basti1101hello, has anybody a valid configuration for asterisk with sip over tls with client certificate validation?
09:02.26kaldemarbasti1101: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
09:04.45basti1101thx, but i know this page, there the client certification isn't enforced by the server. it's possible to connect without client certificate too
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09:06.51basti1101there is the undocumented option tlsverifyclient for sip.conf. but it doesn't seen to work. with this option my server automaticly restarts on connecting
09:10.52jacc0where can I find the License v3.0 papers to sign if I want to upload patches?
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09:16.12sigmountehi !
09:18.57jacc0okay, I found this: https://issues.asterisk.org/view_license_agreement.php
09:19.09jacc0should I print it and sign it? where do I send it?
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09:19.48StaRetjihello good people
09:20.52jacc0hi
09:21.10StaRetjiI'm testing webphone on a website, so I've made a sip user and it works great. Now, my problem is want to block incoming call to that number
09:21.41StaRetjiat least until i prepare sip users for each account
09:21.59StaRetjiI've looked on google, but I must be blind lol
09:23.25jacc0exten = 200,1,hangup()  ; if the number is 200
09:24.50StaRetjijacc0: thx so much, I found blocking anon calls etc, but couldn't find how to block to a specific sip user, in your example user is 200
09:24.55StaRetjithx :)
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09:31.49jacc0@StaRetji: be carefull if you use: _X. in your dialplan
09:32.06jacc0it might still be possible to call 200
09:32.42jacc0if some sip phone dials: 150&sip/200
09:32.44StaRetjijacc0: oh, I used it
09:33.25StaRetji[a2billing] exten => _X.,1,Answer
09:33.43jacc0if the dialplan looks like dial(sip/${EXTEN})
09:34.18jacc0if ${EXTEN} = 150&sip/200 it will result in : dial(sip/150&sip/200)
09:34.26jacc0;)
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09:35.20StaRetjijacc0: thx for this, must admit I will have to figure out first lol
09:35.39StaRetjiI have exten => _00.,1,Dial(SIP/${EXTEN},60)
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09:37.41jacc0so if a sipphone calls : 00231424234&sip/200
09:37.50jacc0phone 200 will ring
09:42.24jacc0http://www.securiteam.com/securitynews/5BP380K19G.html
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09:45.38jacc0thinks everybody is checking his dialplans now
09:46.10jacc0:P
09:46.36kaldemarfunc FILTER is good for extensions that allow characters.
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09:51.38StaRetjijacc0: will take a risk ;)
09:51.57StaRetjithanks once again, I really appreciate your help
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11:34.36QuantumSchemaGood moring all!
11:34.50LantiziaLo, anyone here use a2billing (or could point me to a more appropriate IRC channel for it?)
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11:35.51Chainsawleifmadsen: Got a patch for you on https://issues.asterisk.org/view.php?id=19192 that allows me to use 1.8.4 now.
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11:42.27jacc0is it my connection or is issues.asterisk.org very slow?
11:42.41QuantumSchemaI'm having a little problem using mpg123 to stream music from ShoutCast... the process starts (viewable from "ps -ax" and the dialplan output in the CLI shows that it is playing the right class... just no music....
11:42.53QuantumSchemaDoes this musiconhold.conf look right? http://pastebin.com/60nRVayw
11:43.47jacc0it's not my connection; I guess issues.asterisk.org is down
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11:49.04coppicewell, it gets depressed being full of all those issues that sit there unfixed for so long
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11:53.23jacc0thats no reason for DDOSing issues.asterisk.org
11:53.30jacc0:p
11:54.07jacc0O, it's working again
11:54.10jacc0:)
11:56.27jacc0@coppice: for that reason I joint asterisk dev. today; maybe you should do the same
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11:56.41jacc0I’ve created a patch that will make asterisk ignore all media streams with port 0 and therefore will fix interoperability between asterisk and the HiPath 3000 V8 M5T SIP Stack/4.0.26.26 and will make asterisk more compliant with RFC3264
11:56.58jacc0you can find it here : https://issues.asterisk.org/view.php?id=19281
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11:58.14QuantumSchemaAny one? Please? :-)
12:00.58ChainsawQuantumSchema: With Asterisk, I found the best way to find out is to try...
12:01.29ChainsawQuantumSchema: It seems daring to me to use a remote MP3 feed as MOH, as it could skip or fail entirely for reasons outside of your control.
12:02.40puzzledQuantumSchema: those " look a bit odd. are you sure that is correct
12:03.01QuantumSchemaThanks Chainsaw... at the moment we're just seeing how feasable using an outside MOH source is.
12:03.24puzzledbad idea if you ask me
12:03.59QuantumSchemaPuzzled, from what I've seen, when you use the -@ switch the prompt is expecting quotes but I'll give that whirl with out them...
12:04.58QuantumSchemaSame effect without quotes.
12:05.38QuantumSchemaI am noticing that if I list processes, it lists  /usr/local/bin mpg123 -q -s --mono -r 8000 -f 8192 -b 2048 http://184.107.159.100:8100... notice the space between bin and mpg123.
12:05.41*** join/#asterisk adnc (~akif@unaffiliated/adnc)
12:06.12QuantumSchemait's the only process listed that seperates the executable from the path.
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12:06.24adnchello, could someone tell me if there is anything wrong with this peace of config http://pastebin.com/d5uWDvin
12:06.36puzzledQuantumSchema: buffering takes too long. there are seconds of silence and then MoH starts. my clients would not find that acceptable
12:08.04QuantumSchemaPuzzled, from what I've read in the docs, the moments of silence are only existant the first time the process starts to buffer. Once the process is up and running, buffering continues and further MOH instances tap into the existing buffer. Is that not true?
12:08.46kaldemaradnc: what do you expect it to do and what is the unexpected behavior?
12:09.15puzzledQuantumSchema: afaik MoH starts every time it is requested. but as Chainsaw said you just need to test it and see how it works
12:09.32QuantumSchemaI'm with you on the testing part...
12:09.38QuantumSchemaI just can't get the stream to ever start.
12:09.41adnckaldemar, I've written a  tool that generates this sort of configs. it should behave like an IVR.  a friend just sait this wouldnt work. for my knowledge it looks good, but I wanted to ask some experts here
12:09.58puzzledQuantumSchema: so afaik it is not start once, buffer and play but start, buffer play to client #1, start buffer, play to client#2 etc.
12:10.11QuantumSchemaHmmm...
12:10.40QuantumSchemaI do know the switches are correct for mpg123, if I execute it from the command line it actually pulls down the stream.
12:11.01kaldemaradnc: "won't work" is not enough to work on. there's nothing wrong with it as is, but there can be many issues depending on the surroundings. you need a CLI output of a call.
12:11.05puzzledyes I tried that too and it works but it takes too long for the MoH to kick in. at least from .nl where I am
12:11.15leifmadsenChainsaw: thanks! marked as a blocker for 1.8.5
12:11.48puzzledQuantumSchema: just use local MoH and if you are worried about I/O just put the files on tmpfs
12:11.58puzzledhi leifmadsen
12:12.02leifmadsenohai
12:12.06jacc0hi :)
12:12.08leifmadsenramdisks ftw! :)
12:12.13puzzledindeed :)
12:12.22Chainsawpuzzled: Can you not mitigate that with cachertclasses=yes in the config though?
12:12.49QuantumSchemaWhat I'm confused about though is the process list..
12:13.01jacc0just signed the v3.0 License but my patch still states: license pending; can you help me with that leif?
12:13.10puzzledChainsaw: not familiar with that option. is that a 1.8 option?
12:13.27leifmadsenjacc0: you just asked in another room and I answered you there
12:13.32adnckaldemar, thank you very much
12:13.35QuantumSchemaThere's a space between /usr/local/bin and mpg123... does that look right?
12:13.45puzzledno
12:13.46Chainsawpuzzled: I thought that worked in 1.6 already; but I might be mistaken.
12:14.14puzzledChainsaw: heh I am not familiar with 1.6 either. have planned to move to 1.8 or trunk in the next two weeks or so
12:14.18Chainsawpuzzled: Basically it only has 1 MOH stream, always keeps it going, and just uses that for clients. So they get patched in halfway somewhere.
12:14.19QuantumSchemadisregard that last one about the process.
12:14.38Chainsawpuzzled: My MOH is a 12 minute or so loop that sounds great even from a random point. So that saves resources.
12:14.39asterisk-learnerHey guys, how do u think the acquisition of Skype by Microsoft will affect the world of VOIP and asterisk ?
12:14.54puzzledChainsaw: sounds good. thanks for the info
12:14.56Chainsawasterisk-learner: Less income for Digium if Skype for Asterisk disapears.
12:15.06Chainsawasterisk-learner: But more business if Windows users finally get the hang of SIP.
12:15.06leifmadsenI'm sure they'll be fine :)
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12:15.57Chainsawasterisk-learner: Remember that Skype isn't open; it is not using the SIP protocol. So it's an outsider.
12:16.04asterisk-learnerwhat do u mean by "get the hang of SIP." ?
12:16.28asterisk-learneryeah but i heard u could connect a skype account to asterisk or smthg like that ....
12:16.34Chainsawasterisk-learner: That Windows users might finally use the standard like the rest of us, instead of using some weirdo closed protocol client.
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12:16.39asterisk-learnerthere was effort to make them compatible or so ...
12:16.41puzzledasterisk-learner: Skype will die but not before it is renamed Microsoft Unified Office Communication Super Duper Thingy 2011, is integrated into Lync and they chuck it full of security holes
12:16.59asterisk-learnerpuzzled: hahaha
12:17.01leifmadsenChainsaw: https://issues.asterisk.org/view.php?id=19289  <-- same issue?
12:17.19jacc0lol
12:17.29leifmadsenpuzzled: I don't know.... Skype is a pretty good brand name to use
12:17.44Chainsawleifmadsen: I think that's different. "My" bug is a segfault, very definitive.
12:18.09Chainsawleifmadsen: It's likely that I am affected by what they've reported, because on a bad day my SIP stack does go unresponsive and I have to kill -9 asterisk to get things going again.
12:18.09puzzledproblem is that J. Rosenberg of SIP fame (thanks for the NAT handling by the way) is now a Microsoft fellow so it will be interesting to see how that pans out
12:18.13asterisk-learnerChainsaw: so ur saying either Windows users will hate skype and look for an alternative (SIP) or in case Microsoft does it right (probably not) , then asterisk will be weakened ?
12:18.13leifmadsenChainsaw: ok sounds good -- just didn't want you two working on the same thing independently
12:18.39Chainsawasterisk-learner: Asterisk will not be weakened, because Skype has very different goals.
12:18.50leifmadsenasterisk is just another end point
12:19.02Chainsawasterisk-learner: Asterisk is a PBX that speaks SIP. Skype does not speak SIP. It speaks some weirdo super-encrypted secret protocol.
12:19.11puzzledleifmadsen: it's a good brand name but with little traction in the Enterprise, hence the integration into Lync etc.
12:19.30leifmadsenpuzzled: I could see Skype being the consumer division and integration with Lync being Enterprise
12:19.39puzzledyup me too
12:19.43Chainsawpuzzled: The paid video conferencing was actually quite good. We were considering using it instead of Marratech, which was embraced & extinguished by Google.
12:20.00Chainsawpuzzled: Now Microsoft will be embracing & extinguishing Skype... so that makes me sad.
12:20.02asterisk-learnerChainsaw: true but skype is so widespread, I even have it on Android phone
12:22.14Chainsawasterisk-learner: It's still a closed application that a single company has control over. Even if Microsoft does not kill the Android client off immediately, it will die the next upgrade.
12:23.13Chainsawasterisk-learner: This is how we slowly but surely lost our Marratech when Google killed it.
12:23.51coppicesure, skype is widespread today, but ms has the power to change that
12:24.03asterisk-learnerChainsaw: mmm, does this situation compare to mysql / oracle or is this a diffrent story ?>
12:24.25Chainsawasterisk-learner: Very different. MySQL is open, you can fork it if you don't like what the company in charge is doing.
12:24.42*** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani)
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12:25.09asterisk-learnerok but do u feel mysql is sitll being update like it used to be before oracle's acquisition, or is the process slower now ?
12:25.09Chainsawasterisk-learner: Same with Asterisk. If Digium were to be bought out by someone malicious, you could fork the code and run off.
12:25.33Chainsawasterisk-learner: MySQL got forked into MariaDB and now has a competitor. If MySQL slows down, people will jump ship.
12:25.37Chainsawasterisk-learner: This helps to keep Oracle honest.
12:26.02Chainsawasterisk-learner: Same thing happened with OpenOffice. Got forked, a competitor appeared. I think they've already given up on that one actually.
12:26.17filegotta make my mind up... which seat can I take? it's Friday Friday gotta get down on Friday
12:26.32asterisk-learnerMariaDB never heard of it before...
12:27.02asterisk-learnerOpenOffice is not being developed anymore ??
12:27.11jacc0hardly
12:27.34jacc0Ubuntu now includes libereoffice by default
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12:28.00ChainsawAs I predicted, OpenOffice -> LibreOffice was as quick and dramatic as XFree86 -> X.Org
12:28.29asterisk-learnermmmm
12:29.21coppicex.org gave us more development. libreoffice is starved of rsources
12:29.49Chainsawcoppice: Libreoffice dropped the copyright assignment weirdness. This gets them a lot more developers.
12:29.50wdoekes2leifmadsen: my 19289 is not about kill -9'ing or about the deadlock. it's about not being able to bind() after it fails once
12:30.02Chainsawcoppice: Not to mention an immediate merge of "OpenOffice Go".
12:30.38asterisk-learnerSince we are on the subject of Oracle and Open Source .... what is the suit that Oracle have against Google concerning Android / Dalvik ... ?
12:31.16Chainsawasterisk-learner: They feel that Dalvik is an unauthorised Java platform, due to changes that Google have made.
12:31.31coppiceopenoffice needs some hard core development of its spreadsheet, or its going nowhere
12:31.54Chainsawasterisk-learner: More of a software patent issue then anything else. It might kill java. That'd be fun. Better performance everywhere as students are forced to use C++ instead!
12:32.18Chainsawcoppice: A lot is happening on that front, see the commit logs.
12:32.37asterisk-learnerChainsaw: u mean anyone is allowed to write a JVM , but they felt Google made a bad implementation that could hurt java... ? like Microsoft did once ?
12:32.49asterisk-learnerChainsaw: Amen to C++ !!!
12:32.52asterisk-learnermore work to us
12:32.54asterisk-learner:-P
12:33.15eXcAliBuRHey guys I be backs
12:33.20Chainsawasterisk-learner: There are a few "levels" of java that you can implement, which mandate operands that you must implement.
12:33.39Chainsawasterisk-learner: Google picked J2ME, but did not implement a few of the mandatory operands.
12:33.51asterisk-learnercould you elaborate plz ?
12:34.15Chainsawasterisk-learner: Sure. http://en.wikipedia.org/wiki/Dalvik_(software)#Lawsuit
12:34.24coppiceif the openoffice spreadsheet performed like excel 2003, but without the crashes, they'd have a winner
12:35.45asterisk-learneryou mean some parts of the java specifications oblige you to implements some parts in a very specific way ?
12:35.50asterisk-learnerand Google bypassed it ?
12:35.55asterisk-learner(according to Oracle)
12:36.43leifmadsenI've never been able to get past how 1999 OpenOffice looks
12:36.51Chainsawasterisk-learner: Pretty much, yeah.
12:37.29Chainsawasterisk-learner: And if they dislike what you have done, they can sue you for any frivolous software patent they have.
12:38.27asterisk-learnerChainsaw: thx for the clarification
12:38.59asterisk-learnerand if Oracle wins the case, will Andoroid be removed from all cell phones accross the world ?
12:39.01asterisk-learner:-P
12:39.05Chainsaw(Then again, I feel any software patent is frivolous, the US courts disagree quite strongly with that. Take my opinion with a suitable amount of salt)
12:39.19*** part/#asterisk oelewapperke (wapper@eth1.mine.nu)
12:39.20leifmadsenI like salt
12:39.52coppiceI like salt, so I pepper my meals with it
12:40.41leifmadsenclassy!
12:40.52leifmadsenactually when I have food I don't put extra salt on it, just pepper :)
12:41.20coppiceactually I hate salt, but that doesn't make a one liner
12:46.34leifmadsen:)
12:49.34ChainsawVery secretive bugtracker. 19272 is forbidden.
12:50.24jacc0I like salt aspecialy when i digest my messages
12:50.50leifmadsenChainsaw: they thought it might be a security issue
12:51.08Chainsawleifmadsen: Oh, right.
12:51.27Chainsawis not convinced one could remotely trigger this
12:51.56*** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net)
12:54.04eXcAliBuRhey leifmadsen,
12:54.08eXcAliBuRcongrats
12:54.18leifmadsenyes, I'm still alive, and it's glorious
12:54.27eXcAliBuRI saw your pic yesterday
12:54.35eXcAliBuRyour one fine looking dude
12:54.42leifmadseno.O
12:54.45eXcAliBuRO.o
12:54.54eXcAliBuR:}
12:55.04leifmadsendid you just congratulate me on being good looking.... ?
12:55.05eXcAliBuRisn't it nice to get a compliment?
12:55.11leifmadsenyes I suppose so :)
12:55.47eXcAliBuRhave you had much experience with getting ruby on rails to play nice with asterisk?
12:56.06leifmadsenI've never used RoR but perhaps someone else here has :)
12:56.39*** join/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net)
12:57.10eXcAliBuRPerhaps you would know a better solution than what i'm trying to make... I want to have asterisk call of list of students to alert them of school being cancled.
12:57.38leifmadseneasy enough to do with the language of choice and AMI
12:57.45eXcAliBuRI was gonna have RoR create a .call file and put it in the outgoing folder
12:58.04leifmadsenuse AMI for that kind of thing
12:58.11leifmadsen.call files are not really designed for what you want to do
12:58.21eXcAliBuRoh
13:00.24eXcAliBuRdo u know of any ready made apps that might do it?
13:00.34leifmadsennot off the top of my head, but they may exist
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13:01.56eXcAliBuRi don't think my ami is working... i enabled it, but it won't connect with telnet
13:01.56Nuggettelnet is eeeeeeevil!
13:02.02eXcAliBuRtelnet
13:02.07eXcAliBuRhmmm
13:02.18eXcAliBuRkicks nuggets
13:02.23eXcAliBuRis that a bot?
13:03.27*** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net)
13:03.43*** part/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net)
13:04.21leifmadsennugget isn't, but i think that response is :)
13:04.25leifmadsenI <3 telnet!@
13:07.25*** join/#asterisk wonderworld (~ww@port-92-201-34-220.dynamic.qsc.de)
13:07.31eXcAliBuRtelnet is refusing my connections
13:07.40leifmadsendid you enable AMI?
13:08.01eXcAliBuRyes, I followed page 457 and page 458
13:08.25leifmadseneXcAliBuR: http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html
13:09.08eXcAliBuRhmmm it seems to work if i telnet from the same box
13:09.13eXcAliBuRwhy not external machine
13:12.59*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
13:13.50ruyoeXcAliBuR, maybe you have bindaddr=127.0.0.1.
13:13.52kaldemareXcAliBuR: it's probably listening to 127.0.0.1 only.
13:14.19eXcAliBuRi have all 0's
13:14.29mockerWoo, got on the Google Music beta.
13:14.40eXcAliBuRso it should let all ip's work
13:14.44eXcAliBuRi would think
13:14.58leifmadsenWe're sorry. Music Beta is currently only available in the United States
13:15.07leifmadsenshakes his Canadian fist wildly
13:15.40ruyoeXcAliBuR, then you may have deny/permit configured to 127.0.0.1 only.
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13:43.13eXcAliBuRleifmadsen where abouts do you live?
13:44.25leifmadsenCanada
13:45.06eXcAliBuR;D
13:45.08eXcAliBuRme too
13:45.11eXcAliBuRi'm in quebec
13:45.23mockerleifmadsen: Land of no distribution rights. :(
13:45.51leifmadsenmocker: eh?
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13:48.25mockerleifmadsen: I'm assuming that's why you can't get music.google.com
13:48.37leifmadsenmocker: oh ya
13:50.05eXcAliBuRleifmadsen: how did you become so proficient in asterisk?
13:50.25eXcAliBuRdid u just sit for hours using it?
13:50.29leifmadseneXcAliBuR: many hours of using it :)
13:50.34leifmadsenI've been using it for 7+ years
13:50.49eXcAliBuRi have trouble completing stuff... i get excited at first... then it fades
13:50.54mockerleifmadsen: Teaching classes probably helped.
13:51.00mockerlearns best when teaching others.
13:51.18eXcAliBuRhow do u stay motivated?
13:51.33leifmadsenmocker: indeed, I know all I know from hanging out with people like oej, ssokol, jsmith, russellb, jvanmeggelen, etc....
13:51.38leifmadseneXcAliBuR: it's hard sometimes ;)
13:51.53leifmadsenI have a reasonably short attention span, so I do lots of things
13:52.03leifmadsensometimes not to the benefit of getting things done as quickly as I should
13:52.27jkroonleifmadsen, similar problem here.  task switching/distraction is a major issue.
13:52.48jkroonand yea, using the software is probably the best way to learn it.  3+ years myself, and still discovering new things every other day.
13:53.19leifmadsenjkroon: ya and I'm not good about going back to issues after switching which is the problem -- I'm in the process of offloading customer support elsewhere so I don't get random, "OMG IT'S ON FIRE" calls
13:58.29Lantiziatzafrir, hey you about?
14:02.31QuantumSchemaAlrighty... I ditched streaming MoH.
14:02.40mockerleifmadsen: Guh, I hate that.
14:02.49mockerI have some legacy systems I setup and those calls suuuuuck.
14:02.59leifmadsenmocker: I have one running ABE 2.4.2 :)
14:03.10QuantumSchemaHere's a good one.... is there a way to triger an action if an agent misses a call (ring no answer) from a queue?
14:03.35QuantumSchemaI'd like the agent to be paused if the miss a call (ring no answer) but I'm not quite sure how to sneak my way into that event.
14:04.24leifmadsenQuantumSchema: you could be using Local channels and then evaluate it after the Dial() inside the Local channel to see if it just rang through, then do PauseQueueMember()
14:04.41mocker; Autopause will pause a queue member if they fail to answer a call
14:04.49QuantumSchemaOoooo autopause!!! :-D
14:04.49mocker;autopause=yes
14:04.55QuantumSchematime to look... :-D
14:05.58QuantumSchemaThanks a ton mocker!
14:13.23mockerQuantumSchema: np.
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14:28.26Jcook_5xDataI have a problem i have this context http://pastebin.com/2YARrB9t I when I place a call to 6169888392 instead of going to [voice-custom-5] it is going to the catch all. how do I make respect the number
14:30.05jacc0there is no goto(voice-custom-5)
14:30.07kaldemarJcook_5xData: don't have exten = 6169889392 in the same context.
14:30.09*** join/#asterisk cerberus_za (~coert@196.215.151.194)
14:30.26jacc0so it will never goto [voice-custom-5]
14:30.31kaldemarJcook_5xData: a better match always wins
14:31.58Jcook_5xDataOK thanks for the help I will try that
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14:37.38Jcook_5xDataSorry it was my own fault typo:P
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15:03.31jacc0have a nice weekend all! bye!!
15:04.36DeeewayneeXcAliBuR, see pm
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15:20.58DerkKoHello guys
15:21.22leifmadsenohai
15:24.25*** part/#asterisk AndyML (~alauppe@unaffiliated/andymli7)
15:24.51DerkKoI have a weird problem and looking for troubleshooting pointers.... I have an asterisk box that i use to record calls, my calls are sip to sip. At random asterisk introduces a garble sound in the call. I have not been able to put my finger on why it is happening, i turned up as much verbosity and trying to collect data now. But has this happen to anyone? It is weird.. BTW my codec is g711u on both channels no codec changes
15:24.51DerkKoby asterisk.... any pointers here
15:28.24_Corey_DerkKo: Check your box for unusual activity when you're hearing these effects.  Sometimes excessive load or things like this will impact quality
15:28.39DerkKolow is minimal
15:28.53DerkKoi tried both asterisk 1.8 and 1.6.2
15:29.10DerkKono hardware level errors either
15:29.51DerkKooh and i can reproduce in different hardware
15:29.55_Corey_I had one once where a customer was complaining of bad quality on a box with normal load averages, but ever hour for ONE minute at exactly :44 mins past the hour sendmail was choking on 40k messages it couldn't deliver
15:31.05DerkKommmm ... how did you figure it out?
15:31.19DerkKoI been thinking mayber its because o dont have zaptel or ztdummy
15:31.47DerkKobut i dont use meetme or anything else that would require clock source, the only thing i use is monitor and chan_sip
15:32.00leifmadsenshouldn't be necessary then
15:32.59_Corey_DerkKo: We had a script that monitored process load every 15s and after a few hours it was obvious.  Some things are hard to see with 5m averages
15:35.01*** join/#asterisk phoenixsampras (~phoenixsa@static-190-181-38-121.acelerate.net)
15:35.19phoenixsamprashelp!! IVR menu doesnt respond to keys? just rebooting fixes the problems, and happens every 10 mins
15:35.45leifmadsen~31337
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15:36.30*** join/#asterisk Janos (~Janos@186.32.53.52)
15:36.56leifmadseninfobot: 31337 is <reply> 31337 is the public conference room for #asterisk people to hang out. It's not a support room however. You can connect to the conference room at sip:31337@shifteight.org or ISN: 31337*1273
15:36.56infobotokay, leifmadsen
15:37.01leifmadsen~31337
15:37.01infobot31337 is the public conference room for #asterisk people to hang out. It's not a support room however. You can connect to the conference room at sip:31337@shifteight.org or ISN: 31337*1273
15:37.18Janoshello, simple question, is the Dial command always suppose to return after the timeout ? or is there circumstances that could make it "break" ?
15:37.28phoenixsamprashelp!
15:38.03_Corey_Janos: "break"?
15:38.21Janos_Corey_, does not continue and exit
15:38.55_Corey_I've never seen it not return some kind of DIALSTATUS
15:39.33_Corey_it may stop before the ring timeout because of a telco response or whatever
15:39.40leifmadsenagreed
15:39.58leifmadsenif the other end "answered" the call and immediately hung up or dropped, that would appear as an answered call
15:40.02leifmadsenand would not continue in the dialplan
15:40.26phoenixsamprashow to reset all asterisk by command line?
15:40.35leifmadsenyou could check the status of ${DIALSTATUS} in the 'h' extension
15:40.40leifmadsenphoenixsampras: core restart now
15:40.48leifmadsenit'll drop all active calls though
15:41.30JonathanRoseThere's also 'gracefully' and 'when convenient'
15:41.42JonathanRoseThose won't drop your calls.
15:41.52JonathanRoseBut they'll make you wait.
15:41.52*** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net)
15:42.05JonathanRosegracefully doesn't accept new calls until it's restarted
15:42.20JonathanRosewhen convenient just restarts once there are no more calls.
15:42.24leifmadsenwhen convenient -- just waits until there are no active calls (could be a long time)
15:42.36leifmadsenI don't necessarily like using those because you never know "when" the restart is going to happen
15:42.46_Corey_use gracefully carefully, it won't place calls either so if you're on a busy system with people navigating IVRs, ugly things can happen :)
15:42.47leifmadsenI usually just keep monitoring and/or schedule a particular time to restart
15:42.55*** join/#asterisk m_tadeu (~quassel@89-180-97-202.net.novis.pt)
15:43.02m4xxcan you nest expressions? ie: GotoIf($[ $[${x} < 26| ${x} > 28] | ${x} = 32]?PRIORITY_LABEL)
15:43.31leifmadsenm4xx: yes
15:43.38pabelangerAnybody ever used res_http_post.so?
15:43.43leifmadsenbut surround each one
15:43.55leifmadsenyou didn't quite do it right in your example
15:44.43leifmadsenGotoIf($[$[20 < 21] | $[foo = bar] | $[${paul} = awesome]]?somewhere
15:44.45leifmadsen)
15:44.56phoenixsamprasleifmadsen: that command will reset everything? for example for some reason the IVR menu doesnt respond to the keys, and incoming calls are halted
15:45.04leifmadsenphoenixsampras: it restarts asterisk
15:45.22phoenixsamprasleifmadsen: how can i fix those 2 problems?
15:46.01m4xxleifmadsen: so i'm only missing the $[] around "${x} = 32" correct?
15:46.11leifmadsenm4xx: not correct
15:46.26leifmadsenyou're kinda doing it funny....
15:46.40leifmadsenGotoIf($[ $[${x} < 26 | ${x} > 28] | $[${x} = 32]]?PRIORITY_LABEL)
15:46.45leifmadsenya I guess that's valid
15:46.59m_tadeuhi, I'm using memberdelay in queues.conf to warn the caller that the call is going to be answered. I was expecting that the caller would receive the regular calling tone instead of MOH within that time. How can I make this happen?
15:47.31leifmadsenm_tadeu: make a MoH class that plays ringing?
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15:47.39JonathanRoseI was about to make that joke :<
15:48.20leifmadsenI wasn't joking :)
15:48.33JonathanRoseWell, funny solution.
15:48.36m_tadeuleifmadsen: and how does it change from the regular MOH for the ringing one?
15:48.49m4xxactually i'm looking for != 32
15:48.52leifmadsenm_tadeu: you set it in the dialplan?
15:49.10JonathanRoseIs there a ringing MoH profile in the default configs?
15:49.16JonathanRoseI never bothered to look at MoH.
15:49.58JonathanRoseI'm guessing no.
15:51.57m_tadeuleifmadsen: please give me some direction. when the call goes into a queue, the dialplan is not calling the shots. how can the dialplan say that when the agent picks the call, to change to the ringing moh?
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15:53.26leifmadsenm_tadeu: I misunderstood, you can't
15:53.34DerkKoUGH!
15:53.37leifmadsenI've never run into that issue as I don't use memberdelay
15:53.42leifmadsenDerkKo: ARGH!
15:54.03DerkKothere is not evidence of anything causing this garble on these calls
15:54.37DerkKotrying to paste the output of my full verbosity debug but its too large for pastebin
15:54.38DerkKo:-)
15:56.11m_tadeu:) so my problem is this. since the caller may be waiting for several minutes listening the moh, I want to warn the caller for 5 secs that the call is going to be picked by an agent. I'm figuring that a regular calling tone is the best choice. Any sugestions for such an aproach?
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16:02.08DerkKojust for a pease of mind, ztdummy is not needed for monitor and mixmonitor correct?
16:07.44*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
16:08.52kaiihm. how can i disable SRTP in SIP entirely? when asterisk is compiled with SRTP support, it automatically offers SRTP. encryption=no in the global section of sip.conf seems to not do the job.
16:09.14kaiiaddition: i use mysql realtime peers, the table structure does not contain a "encryption" field.
16:09.29kaiii am using 1.8.4-rc3
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16:14.29kaiii had three segfaults in res_srtp yesterday
16:16.37leifmadsenunload res_srtp?
16:18.58Janosmm, ok this is what i'm trying to achieve, i'm receiving calls on my DAHDI interface, all this calls are sent to my operator '2000', my operator will then transfer calls to the right extensions, some of these extensions are sip channels on the same asterisk and other are sip channels on another asterisk
16:19.38Janosnow this is my operator context and dialing macro, http://pastebin.com/f7qhrkWY
16:20.30m_tadeuhow "heavy" are AM scripts? I mean, does it make sence to run a script for each call(say about 10 simultaneous calls)? or will this really affect asterisk's job?
16:20.58Janosas you can see the macro is very simple, call the extension and then call the operator again after the timeout
16:21.49Janosthis works when the extension dialed by the operator is a local sip channel, channel rings and if not picked up after the timeout the call return the my operator
16:22.51Janosbut when the same is done on an extension on the remote asterisk the first dial exits with a non-zero error and the second one never gets executed, now this used to work in 1.4
16:23.06Janosi just recently upgraded to 1.6 and it's not working anymore
16:23.18_Corey_Janos: is the other end "answering" the channel somehow?
16:23.25Janosyes
16:23.31Janosand it rings
16:23.32leifmadsenthen that's the way it should work
16:23.44leifmadsenwhat does the SIP trace say?
16:23.49leifmadsenis it 180 Ringing?
16:23.55_Corey_Well, it needs to NOT answer...
16:24.02leifmadsenringing != indicate answer
16:24.18leifmadsenunless it is answering then sending back audio that sounds like ringing or something
16:24.20_Corey_If it gets 180 or 183 all will work
16:24.24Janosok ok, not the extension on the other asterisk never answer
16:24.38Janosi mean, the end user never picks up
16:24.40leifmadsenthe other asterisk is probably answering the call then
16:24.43_Corey_Yeah, but does the dial plan on the other asterisk "answer" even though the caller doesn't pick up
16:24.48leifmadsenI bet it does
16:24.49Janosyes
16:24.51Janosit does
16:25.07Janosverbose does show the call as answer
16:25.12_Corey_yeah, so your local asterisk can't distinguish between that and a person answering so...
16:25.26Janoskk that makes sense
16:25.32Janoshow to avoid it ? :P
16:25.48leifmadsenif the other end answers, there is nothing you can do about it
16:25.59_Corey_the other asterisk needs to just "Dial()"
16:26.00leifmadsenyou could look into the flags in the Dial() app that tells it to continue in the dialplan
16:26.11leifmadsenbut that's more of a hack -- asterisk is doing as it should
16:26.28Janosi control both asterisk so i can change anything on the remote asterisk as well
16:26.44Janoslet me get some info on what the other asterisk is doing
16:28.49Janosit just does 'exten => _3XXX,1,Dial(SIP/${EXTEN},20,tTrdwW)' and 'exten => _3XXX,n,Hangup', may that r flag is causing it to answer ?
16:29.36leifmadsenI'd guess the tT and wW flags first actually
16:29.42_Corey_r shouldn't
16:29.48Janoskk let me try removing them
16:29.51leifmadsenit causes the audio path to be required to go through asterisk and might require asterisk to answer the call
16:30.34_Corey_my money is on the 'Ww'
16:31.11kaiileifmadsen: unloading / noload'ing res_srtp sounds pragmatic
16:37.23Janoswell, removed tTwW from both sides, but still had to remove d from the second site, r flag does seem to work, so right now it only has r flag on both sides, might just had been the d flag, will test on that later
16:37.49kaiii do not understand this part of the documentation regarding SIP and SRTP encryption: "Due to limitations of SDP, encryption=yes in sip.conf results in a call with only a secure media offer, therefor forceencrypt=yes would be redundant in sip.conf."
16:38.10Janosthanks a lot guys, really appreciate the help, this had been driving me crazy for a while now
16:38.39kaiii do not understand which limitations are refered to .. other vendors can of course offer AVP _and_ SAVP in the same SDP
16:44.29kaldemarkaii: maybe a poor choice of words. "limitations of asterisk's implementation of SDP" maybe?
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16:46.02m_tadeuwhere can I check the full list of AM commands and events  for asterisk1.8?
16:48.12serafiem_tadeu: https://wiki.asterisk.org/wiki/display/AST/AMI+Actions <-- Is this what you mean?
16:48.36kaldemarm_tadeu: in CLI, "manager show commands", or https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI)
16:52.47m_tadeuserafie, kaldemar, thanx...that is exactly what I needed
17:03.02QuantumSchemaAlrighty.... I'm think this one through... Instead of queue having a timeout for a call and then failing over, is there away that if all the agents are busy it rolls over to the fail over destination?
17:03.23QuantumSchema(wow I'm tired. my grammar just took a nose dive)
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17:03.54*** mode/#asterisk [+o malcolmd] by ChanServ
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17:08.05DerkKoim a little confuse... for asterisk 18 i can still use ztdummy correct ?
17:08.12paulcQuantumSchema: isn't that just a timeout of zero?
17:08.27paulcDerkKo: Wouldn't you rather switch to DAHDI?
17:08.47DerkKoyea i downloaded dahdi-firmware
17:08.53QuantumSchemaWell, that would make sure that the call doesn't time out and stays in the queue right?
17:09.07DerkKobut modprobe ztdummy fails
17:09.14QuantumSchemaI'd like it to stay in the queue but fail over if all agents are busy/unavailable.
17:09.32paulcztdummy=zaptel, you're now using DAHDI? it's one or the other?
17:10.11paulcQuantumSchema: So you send the call into the queue, but if all agents are busy and the call can't be serviced,  you want to send the call somewhere else? Trying to understand what you're trying to accomplish?
17:11.48QuantumSchemapaulc: You're right.
17:11.58QuantumSchemaThat's what I'm looking to do.
17:13.10DerkKoso whats the equivalent of ztdummy for dahdi
17:13.30DerkKosorry for the stupid question and delay, trying to read trough documentation at the same time
17:14.28paulcDerkKo: a dummy timing device equivalent is built into DAHDI I think
17:14.33kaldemarDerkKo: there is no separate module for the dummy functionality anymore. the core dahdi module includes it. there was dahdi_dummy in older dahdi versions.
17:14.52m_tadeuserafie, kaldemar: sorry, I'm missing the events as in here http://www.voip-info.org/wiki/view/asterisk+manager+events but this is from 2004. where can I find an up to date version of this?
17:16.46paulcQuantumSchema: Not sure how to do what you want (ie "immediately"). You may have to balance between a timeout that is acceptable to callers versus allows agents to answer
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17:27.06_Corey_m_tadeu: wiki.asterisk.org
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17:32.00m_tadeu_Corey_: thanx...but I can't find the AMI events there....only AGI commands and AMI actions
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17:36.55_Corey_hmm good point
17:37.22_Corey_seems like this is close: https://wiki.asterisk.org/wiki/display/AST/Some+Standard+AMI+Headers
17:37.28_Corey_but not exactly what you want
17:37.39_Corey_they're spread around in the sources too
17:38.26_Corey_you could always do a "grep manager_event -r ." in your source folder i guess ;)
17:38.28kaldemarm_tadeu: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/doc/manager_1_1.txt
17:40.02kaldemarmight be the same as this: https://wiki.asterisk.org/wiki/display/AST/AMI+1.1+Changes
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17:45.38m_tadeu_Corey_, kaldemar, thanx guys....I think I'll make a small manager and dump the events for the flow I'm using...might be simpler, 'cos I'll find out which ones I'm supposed to care about :)
17:47.13QuantumSchemapaulc: sorry, I was thinking it through and had work get in the way.
17:47.42paulcQuantumSchema: No worries. I'm the same at work :-)
17:48.06QuantumSchemapaulc: LoL If only asterisk was the primary concern...
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17:48.22QuantumSchemapaulc: What I was thinking was kind of like having the distant end being a kind of like an agent but not able to be hit untill all other agents had been tried.
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17:48.47ruben23hi guys where i can find teh voicemail directory..?
17:48.54paulcQuantumSchema: I hear ya! Asterisk all day would be sweet, compared to some of the other stuff I do.
17:49.25paulcQuantumSchema: So.. hmm.. what about if you had a local channel as an agent, with a priority lower than all others? Hmm.. but.. not sure - maybe that would only handle one call at a time
17:49.29QuantumSchemaruben23:/var/spool/asterisk/voicemail/
17:49.30paulckludgey workaround
17:51.07QuantumSchemapaulc: Hmmm... how would that work if the distant end had the highest priority...
17:51.17drdruI'm looking for a SIP host that will allow me to connect my users directly to their service from their mobile devices, allow them to receive incoming and make outgoing telephone calls - can someone tell me how to find such a provider?
17:51.49QuantumSchemapaulc: by the way, it the big idea about this is that the distant end is an outsourced call center to catch over flow for a sales queue we have.
17:52.30paulcQuantumSchema:  I was thinking give your real agents a higher priority, and a dummy/local agent a lower priority.. calls would always go to real agents if available, or hit the lower priority agent if not.. where you could "do something" with the call.
17:52.49QuantumSchemahmmmm....
17:53.04paulcdrdru: I would have thought any providers would be fine with this - the real issue is getting a SIP client running on your users' devices, no?
17:53.37paulcQuantumSchema: depending on how the far end is set up, you could have their agents log in to your system.. or jsut have a bunch of virtual agents that send calls their way
17:53.40QuantumSchemadrdru & paulc: the iPhone and Android devices have sip clients...
17:53.53QuantumSchemaadd a hosted SIP/IPPBX provider and you should be good to go.
17:54.04QuantumSchemaI've played with both clients on the different devices....
17:54.29drdruQuantumSchema: yeah, I am looking for a SIP host - I already have my own SIP app :)
17:54.37drdruright now I'm using sipgate.com
17:54.43drdrubut want to know of alternatives
17:55.03drdruI thought I would be able to make it work with Vonage, but had some technical problem with my app making calls on their network
17:55.07drdruprobably had the wrong encoding
17:55.15paulc~istp-us
17:55.20paulc~itsp-us
17:55.31paulcfrowns - what am I missing?
17:55.36paulcinfobot: itsp-us
17:56.11paulcdrdru: Yeah, probably not Vonage.. but there's a bunch of others.. vitelity.com, voip.ms, flowroute.com
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17:56.26drdruthank you
17:56.30drdruis there a way to search for them?
17:56.38drdruI have tried googling a lot, but not found anything
17:57.31paulcdrdru: seriously? ;-)   http://www.voip-info.org/wiki/view/VOIP+Service+Providers
17:57.55paulcor http://www.voip-info.org/wiki/view/DID+Service+Providers
17:58.36drdruawesome
17:58.37drdruthanks
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18:03.59docidlol, jackasses... http://i.imgur.com/RgHXv.jpg
18:04.42carrarahah 15 years ago
18:04.58docidnice, new holes in the fukishima reactors, fuel leaking into ocean... no sushi for a while peoplez... http://blog.alexanderhiggins.com/2011/05/13/tepco-holes-fukushima-nuclear-reactor-1-3-discovered-radiation-leaking-ocean-22704/
18:05.04docidohh geeze
18:05.05docidsorry
18:05.13docidwhrong channel, apologies
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18:08.39Qwelldocid: That letter is not passive aggressive enough.
18:09.12docidhaha, lol, good point... but it was ment for another network/channel
18:09.56docidahh, yer a good source of info, gunna repost a question if ya got a min to think about it....
18:10.02docid<PROTECTED>
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18:10.35docidbut audio was immediate from stuff coming in on the analog card
18:16.58ack_synhey do I need to install oss modules to make a auto dial out call ??
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18:36.50fish-bulback_syn: Nope. You do mean like, with a call file right?
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18:40.29ack_synfish-bulb: yes I made it
18:40.30ack_synthank you
18:40.43fish-bulbnp
18:40.45ack_synI just moved a formated file to my spool/asterisk/outgoing
18:41.36fish-bulbthe oss module is for playing audio through an OSS device, like your sound card
18:41.44fish-bulbgroovy
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18:57.52ack_syngot it, thank u fish-bulb
18:58.26fish-bulbno problem
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19:08.03m4xxleifmadsen, i've found a few mistakes typeos if you're interested
19:08.17m4xxmistakes/typeos
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20:06.39pdtpatrick_Question -- in Asterisk, tonezone, there's bunch of numbers, im guessing those are all various sounds? how can one tell what a number does or how it relates to a sound
20:09.04QwellThey are frequency pairs.
20:09.59Qwell1000+1200/2000,0/4000 = 1000Hz and 1200Hz played for 2000 milliseconds, then silence for 4000 milliseconds.
20:17.09russellbsyntax should be discussed somewhere ... tonezone.conf.sample maybe
20:23.02pdtpatrick_oh i c
20:23.04pdtpatrick_thanks again
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21:00.01SyrexAnyone here usin Polycom?
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21:10.07pabelangeryes
21:10.10pabelangerand
21:10.11pabelanger~ask
21:10.11infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
21:15.19SyrexHaving a problem with SIP 3.3.1... BLF directed call pickup doesn't work as the phone doesn't append extension number after pickup code... Anyone here using 3.3.1?
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21:23.40atanDoes Plantronics offer a headset with two buttons, one for a bluetooth cellphone and the other to activate a lifter for a desk phone?
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