IRC log for #asterisk on 20100204

00:03.07*** join/#asterisk jksM (jks@193.189.93.254)
00:16.46*** join/#asterisk jakent (~john@soleil.johnkent.mooo.com)
00:18.42*** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361)
00:31.55*** join/#asterisk tdjacobs (~tiagoj@host002.ht2p.net)
00:32.00*** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri)
00:32.34tdjacobshello guys, does asterisk have the capability to mix voices on the server?
00:33.11tdjacobsI want to integrate that with a existing voice talking solution.. I want to mix all voices on server to reduce bandwith usage...
00:33.30tdjacobswas thinking about using asterisk for that, is it possible?
00:34.00tdjacobsI have RAW audio, want to send it to somewhere (possible asterisk), and get merged/mixed audio in raw back
00:39.43voipmonkmix all voices?
00:39.46voipmonkexplain
00:39.51voipmonkwhere do these voices come from?
00:39.54voipmonkconference?
00:44.55tdjacobsvoipmonk: yes...
00:45.01tdjacobsin fact, its for e-learning
00:45.38tdjacobsI know that we can just understand one people talking at same time.. but I really want to mix that sounds togheter providing a "ROOM" feeling...
00:46.07voipmonkfor recording?
00:46.17voipmonkyou want to record the audio from a conference room?
00:47.03*** join/#asterisk shimizu (~shimizu@li142-120.members.linode.com)
00:48.47shimizuis it possible for asterisk to grab sip user password from external source, say some script backend?
00:48.59tdjacobsno
00:49.10voipmonkwhat do you mean shimizu ?
00:49.18tdjacobsI want to grab it real time and reencode to my existing software...
00:49.28tdjacobsI think asterisk is much bigger than the thing I need
00:49.33*** join/#asterisk Sedorox (brandon@smartserv/cna/Sedorox)
00:49.41tdjacobsdoes it uses a software mixer?
00:49.50voipmonku want to grab a password from where and reencode the password into your existing software?
00:49.53bmoraca_workshimizu: it can get SIP authentication parameters in a database (realtime), but it cannot do it otherwise.
00:50.02shimizui have an database where passwords are stored with salt + sha1
00:50.11tdjacobsI can't use hardware mixer because of many "rooms" at same time on same server
00:50.29shimizuhow can i authenticate user's to that database?
00:50.34voipmonkahhh
00:51.00ChannelZtdjacobs: asterisk does mix in software yes - it's all data
00:51.02bmoraca_workshimizu: you probably cannot.  you would have to code that yourself
00:51.09voipmonkyou can do that with asterisk "REALTIME" functions or the MYSQL application from the asterisk-addons package, or a php script
00:51.26tdjacobsChannelZ: its internal asterisk code or 3rd part?
00:51.29voipmonksalt + sha1 will need to be dealt with before hand tho
00:51.36ChannelZtdjacobs: it's internal
00:51.39tdjacobsChannelZ: any API?
00:51.42tdjacobsor doc?
00:51.48*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
00:51.49shimizuvoipmonk: i have api
00:52.16shimizusay some function check_password(username,plaintextpassword)
00:52.41ChannelZtdjacobs: do you just want to record a conference?
00:53.12shimizuvoipmonk: php script ?
00:53.42bmoraca_workChannelZ: no, he wants asterisk to mix "room" sounds with his audio and then pass it back to another system to go out to the audience
00:53.50tdjacobsChannelZ: no, I am not a asterisk user yet... I have another solution (red5), and I want to MIX the sound of all people in a room (via software).
00:54.45tdjacobsso, I have decoded RAW samples and I want to pass some channels of audio to a Software based Mixer and get the resulting channel back, encode and send to application
00:55.01tdjacobsI am looking for a software based mixer...
00:55.05bmoraca_workshimizu: you cannot authenticate SIP users by default against anything but a plain text document or a SQL database.  you can use MD5 encrypted shared secrets, but if you want SHA1, it's just not going to work.
00:55.18bmoraca_workshimizu: the capability has not been programmed in to asterisk.
00:55.40bmoraca_workshimizu: i suppose you could adapt the realtime modules to add that capability, but you will not be able to do it otherwise.
00:55.59ChannelZwell I suppose you could write your own softphone type app to act as a robot participant in the conference, and then you'd get an RTP stream you could do whatever you wanted with
00:57.01shimizubmoraca_work: hmm, that's bad
00:58.00tdjacobsChannelZ: good idea, its trivial to create a room on asterisk? any howto? I am downloading asteriskNow
00:58.11shimizubmoraca_work: i guess freeradius can help me
00:58.26bmoraca_workshimizu: actually, now that I think about it, you probably can't.  i believe that shared secrets in SIP messages are hashed with MD5.  your SHA1 are one-way hashes, and the SIP hashes are one-way MD5.  they're not compatible
00:58.44bmoraca_workshimizu: no, no it can't, because Asterisk cannot authenticate against a RADIUS server currently either.
00:59.06bmoraca_workshimizu: you could use openSER, though, as your registrar server.  i believe it can authenticate against RADIUS
00:59.21shimizubmoraca_work: yeah, it can
00:59.44*** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net)
01:00.04bmoraca_workshimizu: that must be new (like, today), because last time I checked, asterisk could not authenticate sip peers from a RADIUS server
01:00.34ChannelZtdjacobs: well doing conferences in * is not hard no, but it sounded like you already have a phone system in place?  or are you wanting to replace what you have?
01:00.40*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
01:00.49*** join/#asterisk ManxPower-work (~EWieling@216.186.151.147)
01:01.01tdjacobsChannelZ: I need to go now, will back in 10 minutes
01:01.05tdjacobsthanks for the moment
01:01.11ChannelZoh.. Red5 is just some streaming audio thing
01:01.16ChannelZsure
01:01.23tdjacobsyes, and I am core developper of that ;D
01:01.26tdjacobsttyl
01:01.28*** part/#asterisk tdjacobs (~tiagoj@host002.ht2p.net)
01:01.34*** join/#asterisk jmcdowell (~nooe@173-112-87-255.pools.spcsdns.net)
01:01.39jmcdowellhello all..
01:01.55jmcdowellI am back begging for me help with this polycom.. Although I am getting close..
01:02.05bmoraca_workbest part of the work day = going home
01:02.14ChannelZindeedy
01:02.19ChannelZshuts down his laptop
01:02.25shimizubmoraca_work: i guess I'll just generate new password based on sha1 hashes and give them to users
01:02.52jmcdowelllol
01:03.12shimizubmoraca_work: or some random stuff
01:03.38jmcdowellYou could help this begging lil pesant..
01:03.38jmcdowell:D
01:03.53bmoraca_workshimizu: i'm not sure what you're trying to do, but sha1 hashes are not understood by asterisk.  asterisk can use plain text passwords or MD5 hashes.  neither way helps against brute force attacks.  i usually use a 20-30 character random string.
01:03.56*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
01:04.21bmoraca_workMD5 hashes make it so that you can't understand the password by looking at the file or the database record.
01:04.28bmoraca_workthat's all
01:04.31bmoraca_workanyway...
01:04.35bmoraca_workgoes home
01:04.45shimizubmoraca_work: thank you
01:07.38jmcdowellinteresting, <mac>-phone1.cfg was over written with over rides..
01:10.00ManxPower-workjmcdowell: are you sure it was not <mac>-phone.cfg?
01:10.43ManxPower-workThat's the default overrides file.  <mac>-phone1.cfg is a customary name for the file that contains the system admin settings for the phone (registrations, etc)
01:11.09jmcdowellThey were symbolicly linked
01:11.15jmcdowellso I think that explains it..
01:11.22jmcdowellI was confusered.
01:12.08ManxPower-workjmcdowell: Polycom has a section in the Admin manual and they have a technical paper about how provisioning works.
01:12.16jmcdowellI have read it 10 times
01:12.24jmcdowellI can't seem to get it to work
01:12.26ManxPower-workjmcdowell: the tech paper as well?
01:12.28jmcdowellBut I am getting close..
01:12.38jmcdowellI have read this..
01:13.12ManxPower-workI just finished rebuilding our polycom provisioning server from scratch
01:13.54ManxPower-workIt worked the first time I tried it in production, except for some very old phones.  I did have a test server I used to get it all right first.
01:13.55jmcdowelllol
01:14.01jmcdowellThat makes me wanna cry
01:14.26jmcdowellPerhaps I am just getting ahead of myself, I have a-lot of stress going on right now and am finding it harder and harder to focus..
01:14.31ManxPower-workjmcdowell: I've been using Polycoms and setting up provisioning servers for them since like 2003.
01:14.57jmcdowellIt provisions, it just doesn't do anything from ther.e
01:15.09ManxPower-workIf you think the docs are crappy today......
01:15.21jmcdowellIt says "214" as it should, but dialing produces strange behavior on the server side.
01:15.56p3nguinshimizu: I use apg to create regular, hard-to-guess passwords.
01:16.26*** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id)
01:16.40jmcdowellOn the server it never registers, but it tries to dial and is rejected.
01:16.42jmcdowellIt's strange
01:17.16shimizup3nguin: thx, i still try to lookup smth on google
01:17.19ManxPower-workregistraton has to do with asterisk sending calls TO the phone, not accepting calls from the phone.
01:17.42p3nguinshimizu: If you look for smith on google, you'll have a lot of results.  Try something more relevant, too.
01:18.17jmcdowellHmm.. My softphone registers, and I can see it..
01:23.04*** part/#asterisk paulc (~paulc@unaffiliated/paulc)
01:24.14*** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com)
01:27.49jmcdowellCan you lay on me, what files, and what order.. Real simple..
01:28.08jmcdowellThat's what I can't get out of the white paper, because they say one thing, and then say .."Wait, we changed that.."
01:29.25jmcdowellAnd.. this http://www.polycom.com/global/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf is what I have been reading.
01:35.40*** join/#asterisk jmcdowell (~nooe@173.154.185.70)
01:35.49jmcdowellwooops
01:37.08*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
01:37.08*** mode/#asterisk [+o leifmadsen] by ChanServ
01:37.53jmcdowellholds up a sign.. "
01:38.08jmcdowellholds up a sign.. "Will work for Polycom provisioning tricks.."
01:41.43jmcdowellfolds his ears back, and gets out his big puppy dog eyes...
01:41.52*** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net)
01:43.07leifmadsenjmcdowell: eh?
01:43.15jmcdowellLOL
01:43.26jmcdowellJust begging for Polycom provisioning help and tricks..
01:43.38jmcdowellbut I am reading the white paper again, and seeing it a little easier..
01:43.41*** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002)
01:43.45mattwj2002hi guys
01:43.52jmcdowellI guess I will know if a few minutes if it works..
01:43.54jmcdowell;D
01:43.56jayteejmcdowell, have you gotten a phone to download configs properly
01:44.02*** join/#asterisk coppice (~chatzilla@172.169.232.220.dyn.pacific.net.hk)
01:44.07mattwj2002I am looking to buy a Cisco 7960 for my asterisk server
01:44.21mattwj2002where is the best place to buy used equipment
01:44.23mattwj2002ebay?
01:44.32mattwj2002looking for inexpensive
01:44.33leifmadsenmattwj2002: I have one for sale funny enough
01:44.42jmcdowellonly 5k too..
01:44.43mattwj2002lol
01:44.46mattwj2002:O
01:44.48leifmadsen2.5k!
01:45.16mattwj2002:O
01:45.20leifmadsenI have no idea how much 7960s go for
01:45.33jmcdowellI think it's a switch or a phone but dont know..
01:45.42leifmadsenphone
01:45.54p3nguinWhat would you want for the phone?
01:45.54mattwj2002how about .05k ?
01:46.05jmcdowelllol
01:46.05mattwj2002:P
01:46.08jmcdowell5 bucks?
01:46.16mattwj2002that is actually $50
01:46.18mattwj2002:)
01:46.18jmcdowellI will sell my 2 mitels for 100 bucks all inclusive
01:46.28jmcdowell;
01:46.29jmcdowell?
01:46.38jmcdowellI never claimed to be good @ math.
01:49.02p3nguinshimizu: If you want a good password generator, use the following to create a single password:  apg -a1 -n1 -m13 -x26 -MSNCL -E^[]{}\"? -s
01:51.53leifmadsenjmcdowell: why the 7960 if you don't mind me asking?
01:52.05jmcdowellThat wasn' tme
01:52.11jmcdowellthat was someone else..
01:52.12leifmadsenoops
01:52.19leifmadsenmattwj2002: ^^^
01:52.53jmcdowellHmmmm..
01:53.10jmcdowellHow do I tell the phone the "secret" via the xml file..
01:53.22leifmadsenwhich phone?
01:53.26leifmadsenshould be "password" on polycoms
01:53.36leifmadsenin the <MAC_ADDRESS>-phone.cfg file
01:53.51mattwj2002because I have worked with call manager before in a previous job
01:53.57leifmadsengotcha
01:54.04jmcdowellRight, but there is no xml for it.
01:54.06p3nguinI wouldn't mind replacing my 7940G with a 7960G.
01:54.09mattwj2002and I have always wanted to try to hook it up to an asterisk server
01:54.10mattwj2002:)
01:54.13leifmadsenwas just curious, because polycom's are pretty much the defacto :)
01:54.19mattwj2002yup
01:54.19jmcdowellI have a procurve 2948 for sale..
01:54.20mattwj2002:)
01:54.24leifmadsen7960 is easy -- 7961 not so much
01:54.36mattwj2002yeah that is what my friend said
01:54.45mattwj2002something about java getting in the way?
01:55.30leifmadsenwell, basically the .xml files are normally generated, so they are crazy difficult to modify by hand to work with asterisk
01:55.51p3nguinWhy would a 7961 not cooperate?  It's pretty much the same at 7960, but with different config files for the phone.
01:56.17jmcdowellreg.1.auth.password="123456" <--- Is that it?
01:56.18leifmadsenread what I just said again
01:56.24p3nguinIf you can use a text editor, you can edit the file.
01:56.24leifmadsenjmcdowell: looks like it
01:56.35jmcdowellThe example has a variable in it.
01:56.39jmcdowellreg.1.auth.password="${SECRET}"
01:56.55leifmadsenmy 7970 was a super pain in the ass -- it never read the configurations correctly, and there were pretty much no examples on the internet because they are generated, and I didn't have a CCM to generate a config from
01:57.08leifmadsenjmcdowell: OH -- you're using res_phoneprov
01:57.23leifmadsenjmcdowell: users.conf is where the phones are configured
01:57.31p3nguinHmm, that's interesting.
01:57.37leifmadsenneeds to write that blog post about how to use res_phoneprov
01:57.59jmcdowellI am using no such package
01:58.05jmcdowellI searched the web to get that VAR
01:58.19jmcdowellThe line does not exist in any of my .cfg files..
01:58.28jmcdowellhttp://svn.dd-wrt.com:8000/dd-wrt/browser/src/router/asterisk/phoneprov/polycom.xml?rev=11933
01:58.44leifmadsenphoneprov... ding ding ding
01:58.46p3nguinI wouldn't expect it to be much different from how a 7912 is configured; you have to edit the plain text file and then run Cisco's config app against the text file to generate a binary config that the phone uses.
01:58.54leifmadsenjmcdowell: you ARE using such a thing
01:59.16leifmadsenp3nguin: I've seen the 7912 files -- they 7970s were much more involved from what I remember of it
01:59.31leifmadsenanyways, it wasn't trivial getting it all configured and the phone to accept all the values
01:59.46jmcdowellI am editing them manually
01:59.51jmcdowellI pulled that var from the web
01:59.52*** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net)
02:00.01p3nguinI would like to get a model with a lighted display.
02:00.06leifmadsenp3nguin: amen
02:00.20leifmadsenjmcdowell: how did you install asterisk? you're reading from  a dd-wrt forum
02:00.29leifmadsenwhich sounds like you're using some embedded version
02:00.38coppicep3nguin torches are quite cheap
02:00.40jmcdowellI googled for what I thought what I needed
02:00.47jmcdowellI am using asterisknow
02:00.50leifmadsenwhich is talking about res_phoneprov as you're talking about the res_phoneprov template
02:00.56p3nguinIf I had a spare 7940 display, I would attempt to add backlighting to my phone.  I just need to have a spare display in case I mess up.
02:01.13leifmadsenjmcdowell: normally polycom phones are configured via an FTP or HTTP server with the configurations obtained from polycom.com
02:01.24jmcdowellI am configuring them via FTP
02:01.31jmcdowellWell,I am trying to anyway
02:01.37*** join/#asterisk f0urtyfive (~noone@75.150.130.121)
02:01.41f0urtyfivewaves
02:01.49f0urtyfiveAnyone have asterisk setup with a GSM modem?
02:02.01f0urtyfiveI'm trying to be able to use the same GSM modem for SMS/Phone notifications
02:02.59*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
02:04.06*** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net)
02:06.26leifmadsenjmcdowell: well, the configuration you're looking at is incorrect if you're not using res_phoneprov -- you need to use the configuration from the polycom.com site and then upload it to the FTP server you're using
02:06.44*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
02:07.15jmcdowellI have been trying that, and there is no password statement in it.
02:07.40leifmadsenthere is in the 00000000-phone.cfg file, which you should rename to match the MAC address of your device
02:07.55jmcdowelllet me re-expand the archive and start over
02:08.03jmcdowellperhaps I have screwed something up.
02:08.04leifmadsenruns off to hang out with the g/f
02:08.14leifmadsenit's pretty straight forward :)  make sure you get the right archive for your phone
02:08.56jmcdowellI used their matrix.
02:11.51*** join/#asterisk pawz (~pawz@ppp118-208-82-201.lns20.bne4.internode.on.net)
02:13.10jmcdowellSo the <mac>.cfg is supposed to POINT to all the over config files related to the phone?
02:13.27*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
02:14.13etfonhomeyjmcdowell, yes.
02:14.26*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)
02:14.30p3nguinI know very little about Polycoms, but with Cisco we have a SIPDefault.cfg with config information for all phones, then a SIP<mac>.cfg for all phone-specific settings.  I'm sure Polycoms have a similar setup.
02:15.41etfonhomeyPolycom's have a <MAC>.cfg which points to other related config files for the phone.  I usually use phoneX.cfg and sip.cfg where sip.cfg contains companywide settings and phoneX.cfg has phone specific settings for phone X.
02:15.56jmcdowellSo what I am asking..
02:16.03jmcdowellIs the 003202323.cfg
02:16.11jmcdowelldoesn't contain setting for the phone.
02:16.16jmcdowellbut much like the 00000000000.cfg
02:16.17etfonhomey<mac>-phone.cfg is an overrides file that the phone uploads when you make changes to the settings on the phone itself.
02:16.26jmcdowellit contains pointers to the correct config files.
02:16.35jmcdowellJesus, that's what I missed.
02:16.48jmcdowellI have been trying to use 0000000000.cfg
02:16.56jmcdowellto loadup all the phone... What an ass am i.
02:17.36jayteejmcdowell, no you're not, your just learning a difficult but powerful method of setting up Polycoms
02:17.43jmcdowellOther than the password field, which may show up after I re-expand the archive.. This should be about licked
02:17.55jayteethat will give you more control in the long run than using the web gui and save lots of time
02:20.24jmcdowellhttp://pastebin.ca/1786642
02:20.29jmcdowellis that what you are talking about?
02:20.35*** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com)
02:20.39etfonhomeyjmcdowell, jaytee is right. It took me quite a few tries to figure out the provisioning setup  for Polycoms.
02:21.12*** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002)
02:21.12etfonhomeyjmcdowell, wait 'til a <mac>-phone.cfg overrides file gets in the mix to throw you for a loop.
02:21.41jayteethat's when I do a Format File System on the phone
02:21.45jmcdowellbut is what I pasted in pastebin the correct format for a 00112233445566.cfg ?
02:22.52etfonhomeyjmcdowell, why do you reference sip.cfg and sip_316.profiletech.cfg?
02:23.15jmcdowellI just didn't see them to take them out
02:23.19*** join/#asterisk fofware (~chatzilla@host171.190-30-113.telecom.net.ar)
02:23.23jmcdowellI have been messing with this for a LONG time.
02:23.24jmcdowell:D
02:23.36etfonhomeyjmcdowell, I would simplify it.
02:24.05jmcdowellSo the first line, where it calls out configs, these are the MAIN configs, and the second time are the overriding options per phone, is that correct?
02:25.11jayteejmcdowell, here's my master config file for my phone at work. http://pastebin.ca/1786647
02:25.59etfonhomeyjmcdowell, here's mine:  http://pastebin.ca/1786649  I have everything in the same directory.  I probably wouldn't if I had more than just a few phones.
02:26.03*** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica)
02:26.28jmcdowellSo you are calling them out only once..
02:26.29jmcdowellNICE..
02:26.31jmcdowellhang on
02:26.41jayteejmcdowell, I don't have a second line of "overrides" configs
02:27.38etfonhomeyjmcdowell, I would just try to get only one phone working first and then go from there.
02:27.50jmcdowellThat's what I am doing..
02:27.53etfonhomeyk
02:27.56jmcdowellWhy are you calling sip.cfg ?
02:28.02jmcdowellIs that your "Master" config?
02:28.08etfonhomeyjmcdowell, yes.
02:28.13jmcdowellThe precvious is your "overrides" ?
02:28.16jayteeAll phones need to use these files so they must be listed in the CONFIG_FILES
02:28.16jayteelist. For example, the master configuration file for the phone with Ethernet
02:28.16jayteeaddress of 0004f2000607 is called 0004f2000607.cfg and looks like this:
02:28.48jayteejmcdowell, the following is VERY important to understand
02:28.50jayteeSince the files are processed left to right, any parameter which appears in
02:28.51jaytee0004f2000607-user.cfg will override the same parameter in phone1.cfg.
02:28.51jayteeSimilarly any parameter in local-settings.cfg will override the same parameter
02:28.51jayteein sip.cfg.
02:28.57carrarI wouldn't change the sip.ld
02:29.11carrarthe phone will know what file to get
02:29.12jayteei use the one for my specific model
02:29.23etfonhomeyjmcdowell, what did you mean by "previous is your overrides"?
02:29.50carrar601 will always ask for 345-11605-001.sip.ld
02:29.51jmcdowellMeaning, phone specific settings.
02:29.54carrar2345-11605-001.sip.ld
02:30.00carrarand 2345-11605-001.bootrom.ld
02:30.07jayteethis is the line in my master config file for my phone that loads the appropriate sip.ld file <APPLICATION APP_FILE_PATH="2345-12200-001.sip.ld"
02:30.19jmcdowellUsername, password, extension, display name etc.
02:30.23carrarno
02:30.24jayteethat is for the Polycom330 with SIP firmware 2.2
02:30.30carrarAPP_FILE_PATH="sip.ld"
02:30.36etfonhomeyjmcdowell, phoneX.cfg is where you configure the line appearances for each phone.
02:30.48carrarphone will automaticall add 2345-11605-001. to the front
02:30.54carrarif sip.ld doesn't exist
02:31.06carrar2345-11605-001. is specific to the model of the phone
02:31.07jmcdowellwhat?
02:31.14jmcdowellAnd I thought I was really starting to understand.
02:31.18carrarheh
02:31.33carrarit's in the zip file you got from polycom
02:31.36etfonhomeyjmcdowell, don't worry too much about the sip.ld and bootrom.ld too much.
02:31.44jmcdowellMan, I was really getting excited, I am about to throw these things in the trash.
02:31.48etfonhomeyjmcdowell, at least for now.
02:32.02jmcdowellI already upgraded the firmware and the bootrom on this phone to the max I could go up to.
02:32.19jmcdowellWith the firware came a whole slew of config files etc.
02:32.27jayteejmcdowell, the sip.ld file contains config info for ALL phones. 2345-11605-001.ld is for only the model 601 so it is smaller, uses less memory storage and downloads quicker and loads faster
02:32.29carrarYou need to start with those
02:32.32etfonhomeyjmcdowell, then don't worry about it.  Just make sure you're using the sip.cfg/phone.cfg from the zip file that you got the firmware from.
02:32.47jmcdowellk
02:33.10carrarquicker, faster, better and CHICS DIG IT
02:33.44etfonhomeyjmcdowell, you can get away with not touching the sip.cfg file for your first "hello world" setup.
02:33.55carrarif I had nothing to do I could write a nice long wiki on how I setup polycoms
02:34.06etfonhomeyjmcdowell, you can config everything you need for a basic line registration in the phonex.cfg.
02:34.22jmcdowellok hang on
02:34.29jmcdowellI am trying to boot this things..
02:34.30jmcdowellthing
02:35.17jmcdowellI can bring 1000+ node clusters back from the dead, and birth new ones..
02:35.19*** join/#asterisk jakent (~john@soleil.johnkent.mooo.com)
02:35.24jmcdowellbut I can't setup a polycom./.
02:35.26jmcdowell:/
02:35.37carrarand you learned clustering over night?
02:35.55rnpTo All: if I use asterisk/VICI dial in conjunction with my online crm system, is there anyway I can have my clients login to my crm, and see for instance the list of businesses we are calling for their account and what their responses were?
02:36.38jayteejmcdowell, for comparison purposes here's my "MAC"-phone.cfg file which is actually named 0004f21a0c98-5146.cfg    http://pastebin.ca/1786661
02:37.01jmcdowellclearly I did not
02:37.06jayteemy extension is 5146 and my sip account in sip.conf for asterisk is [5146]
02:37.26etfonhomeyjmcdowell, here is a basic config snippet from my phoneX.cfg with the corresponding sip.conf entry:  http://pastebin.ca/1786660
02:37.49etfonhomeyThat is all I need in order to be able to register "station1" to Asterisk.
02:38.26jmcdowellcool
02:38.26carrarI actually use the whole sip & phone1 file that is sent from polycom
02:38.31jmcdowellalmost there
02:38.38carrarwith all my changes
02:39.07rnpdoes anyone here consult/setup new pbx asterisk based systems?
02:39.15carrareveryone does
02:39.25rnplol
02:39.36carrarask away
02:39.51rnpwell I want to talk to someone priv about what I want and what it will cost to do
02:39.54rnpbasically
02:40.13Sedoroxyou might want to state where you are
02:40.23Kobazanyone know how i make a t1 card bind to a specific cpu
02:40.24rnpUSA
02:40.35Sedoroxstate?
02:40.36Kobazi have a sangoma card that randomly flips it's inturrupts from cpu0 to cpu1
02:40.47rnpI run a remote operation in the philippines and I want to get my own system setup so I control all the calls they make
02:40.54rnpFlorida, I don't need on site help
02:40.59Sedoroxah
02:41.10carrarisn't voip in the philippines to the outside illegal?
02:41.18rnpno
02:41.38Kobazwww.kobaz.net/misc/t1_port1_device_interupts-day.png
02:42.09carrarrnp, should be easy
02:42.11Kobazgreen is cpu0 and blue is cpu1
02:42.23rnpyeah, but there are multiple things that need to be integrated with my online crm
02:42.24etfonhomeycarrar, I use them all as well, just trying to give jmcdowell a "hello world" setup.
02:42.29rnpthat's what is most important to me
02:42.50carrarbbl, dinner
02:43.00jmcdowellConfig file rror is 0x20
02:43.14rnpSedorox, do you do consulting work for this stuff?
02:43.58Sedoroxfor asterisk, rarely, especially what your looking for
02:44.20rnpi see
02:44.40SedoroxI'm more of an end user of asterisk, or can do basic setups
02:44.41jayteeThe most common reported reason for that error, an unescaped ampersand somewhere in the file
02:44.54*** join/#asterisk NicoleMun (~ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net)
02:46.03*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
02:46.34jmcdowellCall..  VMI
02:46.45jmcdowellThat's all I can tell ya, but they aren't all open source.
02:47.00jmcdowellVoice mail incorp
02:48.54jmcdowellAhhh!
02:49.07jmcdowellI figured out the error, got the phone to boot, and it still won't dial
02:49.24jayteejmcdowell, is it registering with asterisk?
02:54.06jmcdowellAnd I thought I was close..
02:55.01jayteemaybe you are close
02:55.06jayteewhat are you trying to dial?
03:09.20*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
03:19.47*** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
03:20.07*** join/#asterisk jblack (~jblack@71.181.248.16)
03:21.14jmcdowellrolls his eyes..
03:21.26jmcdowellthe damn thing hasn't been getting anything this WHOLE time..
03:21.36jmcdowellIt downloads it, but it doesn't use it.
03:27.16*** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com)
03:30.55*** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri)
03:31.29jmcdowellthrows 1 out of 14 polycoms against the wall.
03:31.30jmcdowellnext..
03:32.04ChannelZperhaps there is a syntax error somewhere that causes it to reject the whole thing?
03:34.17jmcdowellCould be but this is from scratch again with the files from the sip firmware upgrade.
03:34.47jmcdowellWait it is taking some part of it, as I see 3 line keys now
03:35.02jmcdowellbut the phone name is not being taken, the register bits are not being taken.
03:36.25jmcdowellwhen i go into the phone settings on the phone from the phone the sip server is blank
04:00.16*** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru)
04:03.34*** join/#asterisk jmcdowell (~nooe@173.154.185.70)
04:03.58jmcdowellWhat is a channel troll?
04:05.00p3nguinIt's a person who goes in the channel looking to stir up trouble, usually by asking really ridiculous questions that don't really make sense and are completely off topic.
04:05.26*** part/#asterisk hluesea (~hulusikah@88.247.127.66)
04:05.42jmcdowellahhh
04:05.44jmcdowellwell..
04:05.49russellblike ... hey, i heard asterisk is so bad, it puts voice under the IP
04:06.05jmcdowelllol
04:06.07p3nguinLike a common one we get on a Linux channel is a guy saying he needs to uninstall Linux because he needs to use Windows again...
04:06.11jblackDont' run it on a cpu bound system. :P
04:06.59*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
04:07.50jmcdowellok so this damn phone isn't even trying to register..
04:07.54jblacki.e. "cheap bastards will pay the price of being cheap". =)
04:08.00jmcdowelllol
04:08.06p3nguinHe is told that you just delete the partition(s) where it is installed, and then install Windows again.  Then he continues to ask for help installing Windows, which is not a Linux channel topic.  When he can't get help, he keeps bitching that he needs to uninstall Linux.
04:08.31jmcdowellIt's getting an ip taking the config file and labeling the lines..
04:08.34jmcdowellbut that's it..
04:08.45jmcdowellBy the way, my juicer is broken..
04:08.47jmcdowell;)
04:11.34p3nguinjblack: Speaking of trolls, did you know that we still get the "poop guy" every few days?
04:12.12jblackthat sounds familiar.
04:12.22*** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)
04:12.29jblackThat's an actual person, right?
04:12.37p3nguinAfter all this time, he still comes around.
04:12.46jblackHe's been doing that for what? 5 years now?
04:12.59p3nguinYep, he sometimes will interact with people if he doesn't get kicked out promptly.
04:13.13p3nguinProbably around 5 that I can account for.
04:13.14jmcdowellWhy not just ban him for good?
04:13.20jblackHe must have it in his google calendar,  right after "bathe" but right before "take pills"
04:13.50jblackBecause he was harmless. he wanted  to spend a moment each day talking about poop.
04:14.07jmcdowellAh hah!!!
04:14.12jblackother than that 90 seconds each day, he never causes any trouble
04:14.22jmcdowell- No matching peer found
04:14.30jmcdowellIt's trying now and getting rejected!
04:14.34p3nguinWhen you have a reasonable supply of dynamic IP addresses and can choose nearly any nick name you want, it is easier for you to evade permabans.
04:15.24jmcdowellYeah, I could I suppose, mine changes EVERY time I reboot my router and they are way off the charts, not usually anything near the last.
04:15.39jblackYeah. An I'm a jijitsu op.  I believe pushing straight back is pointless. It's better to let people use their own inertia to get themselves out of the way.
04:16.08jmcdowellis there anyway to make asterisk spit out what the phone is trying to log in as?
04:16.19jblackyou can do a sip debug and watch the packets
04:16.23p3nguinHow often does a router really need rebooted?
04:16.49p3nguin<PROTECTED>
04:16.53p3nguinThere's my router.
04:17.06jmcdowellMine doesn't usually, but my aircard dies randomly
04:17.12jmcdowellI think they are acutally kicking me off
04:17.22jmcdowellbecause I use it and abuse it for what it's worth.
04:18.39jblackpeng: o you do facebook?
04:19.23jmcdowellsip debug is showing everything but the failed log ins
04:19.43*** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58)
04:20.32jmcdowellit's getting it
04:20.37jmcdowellbut I don't see the user name
04:21.17jmcdowellOoooohhh..
04:22.39p3nguinjblack: Negative, I don't do social networking.
04:23.15jmcdowellI think I nailed it
04:23.33p3nguinNot twitter, facebook, myspace
04:23.38jmcdowellI had put the pbx address in the field that it apparently and strangely uses as the user ID
04:23.49p3nguinDon't know what else there is, since I don't do those either.
04:24.31jmcdowellSorry, I am thinking out loud
04:24.44jmcdowellin case anyone wants to correct mel;.
04:24.45p3nguindon't worry about that.
04:24.46jmcdowellme
04:24.47jmcdowell:DS
04:25.02p3nguinIRC is like the friend you don't have.
04:25.19jmcdowellso the reg string was pbxaddress.com@pbxaddress.com
04:25.22jmcdowellwould won't work
04:25.23p3nguinYou can talk to it all you want, and if it feels like talking back, it will.
04:25.29jmcdowelllol
04:25.34*** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net)
04:26.12jmcdowellstarts doing the funky chicken dane..
04:26.14jmcdowelldance
04:26.23jmcdowellIt's working.. Sorta.. Now I have dial plan problems..
04:26.51jmcdowellAnyone in here care to help?  It's stripping the 9 off when it send, which causes asterisk to reject the call.
04:27.27p3nguinShow me the dialplan, what you are dialing on the phone, and what you want to happen.
04:28.09jmcdowell<digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3"/>
04:28.12jmcdowellis the current plan
04:28.28dlynes_laptopAnyone know what 'Unable to allocate AST channel structure for SIP channel' means?
04:28.32dlynes_laptoppastebin at:  http://pastebin.com/m3ab852db
04:28.33jmcdowellwhen I dial 913143212222 it strips the 9 and send 13143212222
04:28.41dlynes_laptopI'm guessing it means an out of memory condition?
04:29.03russellbdlynes_laptop: you're likely hitting an open file descriptor limit
04:29.07russellbulimit
04:29.28russellbchan_sip is trying to allocate an Asterisk channel object, and it fails because it can't open a pipe
04:29.36russellbmost common cause is you hit the open fd limit.
04:29.54p3nguin314, as in MO, USA?
04:29.56russellband so then it goes,   :'-(
04:30.07jmcdowellYes
04:30.08jmcdowellas in MO
04:30.24p3nguinI'm about 70 miles from STL.
04:30.30dlynes_laptoprussellb, and those would be listed in 'netstat -anp | grep "^unix"'?
04:30.33jmcdowellI would like to not have to dial the 1 and not have the 9 strip the 9 off..
04:30.40jmcdowellWhat direction?
04:30.44p3nguineast
04:30.45russellbdlynes_laptop: i dunno
04:30.48jmcdowellIL
04:30.51p3nguinyep
04:31.01dlynes_laptoprussellb, and fwiw, my ulimit is unlimited
04:31.01jmcdowellWe'll have to have a nerd party some time..
04:31.02jmcdowell:D
04:31.02p3nguinnear Mt. Vernon
04:31.07russellbdlynes_laptop: orly!
04:31.11jmcdowellI am vaugley familiar with IL
04:31.12russellbwell then.
04:31.12dlynes_laptoprussellb, really
04:31.30russellbany hung channels?  core show channels, sip show channels ...
04:31.40dlynes_laptoprussellb, don't know...I couldn't connect
04:31.47dlynes_laptoprussellb, i had to kill the pid
04:31.58russellbhuh.  it explodified.
04:32.04russellbi suggest a hammer
04:32.05dlynes_laptoprussellb, root      1913  3.2 12.2 142108 125848 ?       Ssl  Jan19 706:49 /usr/sbin/asterisk
04:32.15dlynes_laptoprussellb, using 'ps auxffww'
04:32.17russellband that's the best I can suggest in my near unconcious state, heh
04:32.36p3nguinjmcdowell: But as for the dial plan... what extension does asterisk think you are dialing?
04:32.46jmcdowell9 to get out
04:32.51dlynes_laptoprussellb, if it makes any difference, it's asterisk 1.6.1.8
04:32.53jmcdowelland the extensions are all 1xx
04:33.04jmcdowellthe extensions can change if they need to
04:33.29russellbdlynes_laptop: hmm, lots of fixes since then, even one directly related to a problem that led to symptoms like this
04:33.29p3nguinBut I need to know what * thinks you are dialing, because your phone's dialplan looks normal to me.
04:33.39jmcdowellUhhh
04:33.40russellbyou should consider an upgrade, I suppose.
04:33.40jmcdowellhang on
04:33.47russellbpasses out ... gnight
04:33.47dlynes_laptoprussellb, ok, thanks
04:33.49russellbnp
04:33.52dlynes_laptoprussellb, g'night
04:34.25jmcdowellThe outbound route is 9|.
04:34.31jmcdowellI know stop the laughter
04:34.39jmcdowelli don't understand these dial plans
04:34.39p3nguinI don't even know what an "outbound route" is.
04:34.42p3nguinSounds made up.
04:34.51p3nguinMust be an unsupported FreePBX.
04:35.02jmcdowellNo..
04:35.06jmcdowellThe outbound route
04:35.21jmcdowellIt's the route that gets to the SIP provider
04:35.27jmcdowelland 9|. picks it up
04:35.44p3nguinRight now, the phone is accepting 4-digit extens starting with 2-9.  Also 10 and 11 digit numbers.
04:35.45jmcdowellof anything matches 9|. it goes to the sip provider
04:35.53p3nguinAccording to whom?
04:36.01jmcdowellhang on
04:36.04jmcdowelllet me find something.
04:36.23p3nguinThe phone certainly isn't doing that in its dialplan.
04:36.43jmcdowellhttp://blogs.elastix.org/en/wp-content/uploads/2009/11/melbviamelbpbx.png
04:36.49jmcdowellthat is the screen I see
04:37.08p3nguinDial 13143212222 or 3143212222 and it'll match the dialplan on the phone.
04:37.14jmcdowellThat's the problem, I am dialign 12 digits..
04:37.17p3nguinstop
04:37.31jmcdowellif I dial 1314xxxxxxxx the outbound route will not pick it up.
04:37.35p3nguinDial it correctly.  10 or 11 digits.
04:37.51p3nguinThe "outbound route" will do whatever you tell it to do.
04:38.00jmcdowellYour call cannot be completed as dialed
04:38.03jmcdowell* is telling me that
04:38.12p3nguinThen you haven't configured your extens correctly.
04:38.18*** join/#asterisk pawz (~pawz@ppp118-208-82-201.lns20.bne4.internode.on.net)
04:38.40jmcdowellif I changed the outbound route to accept 1|. it would fly
04:39.13jmcdowelli can dial extension to extension
04:39.43p3nguinYou mean device to device.
04:39.53jmcdowellYes
04:40.10p3nguinexten => _1NXXNXXXXX,1,Dial(SIP/${EXTEN}@yourprovidename)
04:40.17p3nguinThat's for 11 digit dial.
04:40.21jmcdowellSo if I change the dial plan in the trunk to 1|. it gets to my provider but fails to dial because it gets the one stripped off.
04:40.24p3nguinexten => _NXXNXXXXX,1,Dial(SIP/1${EXTEN}@yourprovidename)
04:40.30p3nguinThere's 10 digit dial.
04:40.44dlynes_laptoprussellb, btw...if you're still here...it also resulted in SIP 500's and SIP 403's being sent back to any sip users that were attempting to send calls to the box
04:41.12p3nguinWant 7 digit dial?  exten => _NXXXXXX,1,Dial(SIP/1314${EXTEN}@yourprovidename)
04:41.24p3nguinI think I missed an X in the first two.
04:42.09p3nguinjmcdowell: You really should stop playing with the dialplan on the phone and accept that it was fine the way you showed it to me.
04:42.21jmcdowellThen I have to figure out why I can't get out.
04:42.25jmcdowellhang on
04:42.26p3nguinIt is asterisk's dialplan that needs work.
04:42.35p3nguinextensions.conf
04:42.52p3nguinThe phone dialplan you showed earlier looked fine to me.  Leave it alone.
04:42.59jmcdowellOk.........
04:43.02jmcdowellhmmmmm
04:43.08dlynes_laptopheh...gotta love cunningpike's quit message :)
04:43.19ManxPower-workjmcdowell: dial, then press the DIAL button, that will bypass the phone dialplan by default
04:43.41ManxPower-worki.e. dial digits first
04:43.52p3nguinBut the phone's dialplan is fine, so what's the use?
04:43.55jmcdowellI figured it out..
04:43.58*** join/#asterisk voipmonk (~voipmonk@dsl-67-204-40-42.acanac.net)
04:44.01jmcdowellI have to change the dial plan in *
04:44.08p3nguinNo kidding?
04:44.08jmcdowell1NXXNXXXXXX
04:44.10jmcdowellNXXNXXXXXX
04:44.10jmcdowellNXXXXXX
04:44.15p3nguinDo you read what I type?
04:44.16dlynes_laptopjmcdowell, isn't that what p3nguin just finished telling you?
04:44.28ManxPower-workjmcdowell: A unified, designed, well thought out dialplan will save you countless hours.
04:44.31jmcdowellI was wacking @ this problem..
04:44.36jmcdowellBut yes that is what he basically said
04:44.39Corydon76-digjmcdowell: you're missing the leading underscore
04:44.58p3nguinI did typo my patterns, though,  Missed one X in the first two.
04:45.05jmcdowellHmmmm
04:45.16ManxPower-worksings "Sqlite3, and cookies, and PHP, and Polycom, Oh my!"
04:45.26jmcdowellThe last thing I have left to grasp is the "lines" in the config..
04:45.27dlynes_laptopcringes.
04:45.34Corydon76-digIf it doesn't start with an underscore, it ISN'T a pattern
04:45.35jmcdowellI have a 3 line setup from my sip provider...
04:45.44voipmonkHow does that tune go?
04:46.03jmcdowellThanks for all the help though, it has been very helpful..
04:46.07ManxPower-workLions, tigers, and Republicans, oh my!
04:46.08jmcdowellI am still astarded...
04:46.14jmcdowellI am still astertarded...
04:46.38p3nguinIf you'll just pay more attention here it'll be much easier to receive help HERE.
04:46.47ManxPower-workjmcdowell: this isn't something easy like designing a national data network -- telecom is hard.
04:47.04jmcdowellI feel it.
04:47.58dlynes_laptopIt's definitely more difficult than writing VB code
04:49.38jmcdowelllol
04:55.57*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uaymovvsgaohpjxm)
05:02.53*** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net)
05:03.13*** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2)
05:05.17etfonhomeyjmcdowell, did you finally get that phone registered to Asterisk?
05:08.02*** join/#asterisk pawz (~pawz@ppp118-208-82-201.lns20.bne4.internode.on.net)
05:10.08jmcdowellyes
05:10.15jmcdowellnow I am trying to learn the dialing patterns
05:10.29jmcdowellnot the command line dialing patterns, the freepbx dial patterns.
05:10.41p3nguin~freepbx
05:10.41infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
05:11.11p3nguinDo it in vanilla asterisk, and I'll help.
05:11.39Miccis there a way to not have the h extension jumped to after calling Hangup?
05:11.51MiccLike, should I use Return instead of Hangup?
05:12.24MiccI'm trying to do a .call file and if it fails and jumps to failed, I don't want it to goto h after that.
05:17.10*** join/#asterisk lmsteffan (~laurent@reef.ac-noumea.nc)
05:19.09ManxPower-workMicc: you don't have much choice.
05:19.42ManxPower-workYou can try Return, but I would not expect much.
05:23.32MiccHow can I jump over a bunch of stuff in h? I guess I could do a gotoif
05:25.01dlynes_laptopp3nguin, he's trying to make it easier than vb for himself, but more difficult than acupuncture for you
05:25.35p3nguindialplan, generally speaking, is very easy.
05:27.24p3nguinIt's those quirky things like I am dealing with right now that make me yell and scream.  Using a call file, connect to a channel, then run an extension... auto-outbound,s,1 tries to run and fails twice.
05:28.29*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
05:28.52p3nguinAh, I think I found the error.  The context is auto-outbound, but it is auto-outgoing in the call file!
05:31.49*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
05:32.29p3nguinYep, that solved the issue.
05:34.45*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
05:40.32*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
05:40.47*** join/#asterisk ZxCv47 (~pat@209.58.232.204)
05:41.34ZxCv47anyone know where i can find info on the different error messages produced by fax for asterisk?
05:43.07*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
05:44.51*** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110)
05:45.39*** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
05:46.02*** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath)
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05:47.51sbraththis is probably a dumb question, but I have a call coming in on a DAHDI trunk, and in the CDR logs, I have the Caller-id-name, but when I do a dumpChan on the call caller-id-name is not set,
05:52.10*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
05:54.13jmcdowellHrrrmmm..
05:57.28*** join/#asterisk Slashman (~Slash@ariane.fimasys.com)
05:58.09sbrathI see: CallerIDName=       (N/A)
05:58.16sbrathin my DumpChan(10)
05:59.01sbraththis isn't a complicated setup, just a PRI coming in from the telco that's delivering the Cid-name. Is their a DAHDI paramater that I need to let that name flow thru?
06:02.20*** join/#asterisk brunner (~chris@108.sub-75-235-172.myvzw.com)
06:02.59brunnerIs there any hardware that I can plug into a phone jack in my home that has enough current to power the rest of my home phones?
06:04.03brunnercould I just run a telephone cable from any FXS port to a telephone jack to do that?
06:04.17brunner(assuming AT&T isn't connected at the demarcation point)
06:06.29sbrathI guess as long as the demarc to at&t is disconnected, it should work.
06:06.48sbrathI'm not sure how many phones it will power tho, do you have any FXS card now?
06:18.32*** join/#asterisk pawz (~pawz@ppp118-208-100-34.lns20.bne4.internode.on.net)
06:20.08rnpanyone on now that does asterisk implementation?
06:24.04florzanyone on now that does specific questions?
06:25.16*** join/#asterisk fiddur (~fiddur@192.121.104.121)
06:29.59p3nguinha
06:30.08ChannelZno
06:32.13rnplol
06:32.35*** join/#asterisk viq (~viq@unaffiliated/viq)
06:35.23kaldemarsbrath: do you have usecallerid=yes and callerid=asreceived in chan_dahdi.conf? feel free to pastebin the cli output of a call.
06:37.21rnpDo you think 4k is a lot to get asterisk setup with custom integration with an online crm?
06:37.56carrarYou better be supplying the hardware for that
06:38.29rnpreally
06:39.11rnpcarrar do you setup astrisk for others?
06:39.15carraryeah
06:39.45rnpalright, let me hit you with an im maybe you can help
06:39.59carrarI'm not interested however, but thanks
06:40.26rnpoh, alright
06:40.48carrarWhat CRM do you want it to use?
06:40.55carrarand how
06:41.00rnpa custom one I had made from scratch
06:41.06rnpuses mysql though
06:41.16carrarSo you would be writting that part?
06:42.08carrarYou need to write out a proposal of exactly what functionality to exist in the server
06:42.25carrarotherwise if you expected call parking and didn't get it, well you need to write it out
06:42.38carraras a example
06:43.22rnpok
06:43.30rnplet me outline what is needed
06:43.42rnpI'm looking for an app that runs on top of Asterisk that will query CRM for a client account to work on which includes the following:
06:43.48rnp1.  A set of numbers (with names) to call.
06:43.54carrarWhat is needed from the phones perspective, PBX and what you want the PBX to do with the CRM
06:43.54rnpThe number of appointments to achieve for the client
06:44.24rnpI want these numbers (1 above) to be dialed automatically for call agents residing in the Phillipines.  Then, at the termination of each call, the agent should be able to specify if a successful appointment was achieved (a yes or no).  You would need the app to post back to the CRM the result of the call and then start the process again with a new call.
06:45.55carrarmight be better to build a web app around that
06:46.16rnpwhat kind of cost am I looking at to get something like this setup
06:46.36carraragent might be walking away to the bathroom and you got this automatic dialer going crazy
06:47.15carrardepends how you have to get the numbers from mysql
06:47.44carrarbut that alone is a customized app
06:47.53carrarnot even including you need the PBX part too
06:48.11rnpwell I said 4k and you said better be including the hardware for that too
06:48.22rnpso what kind of price do you think is real for what I want
06:48.57carrarthats one 1 part
06:49.02carrarYou need a complete request
06:49.07sbrathkaldemar: my config for the dahdi is split between users.conf and dahdi-channels.conf
06:49.18carrarYou might want something in the system that isn't customized
06:49.23carrarerr that is
06:49.36rnpI see
06:49.51carrarYou are writting a RFP
06:50.28rnpwhat if the business list was loaded onto asterisk, and then only appointments were transmitted to the crm ?
06:50.31rnpis that a lot easier?
06:50.48carrarbut what you describe is certainly doable
06:51.04rnpwell carrar, what I describe, what do you think it should cost $$ ?
06:51.06kaldemarsbrath: dahdi-channels.conf is something that chan_dahdi.conf includes into itself when the configuration script is used. the PRI should be defined in dahdi-channels.conf then.
06:51.16carrarI think you are leaving a lot out
06:51.24carrarand again you need to write a complete RFP
06:51.34carrarif it's not written out you don't get it
06:51.36nix8n82depends on who you ask and how many hours of custom programming you want
06:52.09sbrathkaldemar: it is also configured in dahdi-channels.conf   I don't have the usecallerid=yes in there, that is listed under the span config in users.conf, but I'll add it to dahdi-channels.
06:52.11rnpwhat's a reasonable $ per hour for this kind of programming?
06:52.32carrardepends who you ask
06:52.39carrar80-$250/hr
06:52.43rnpshew
06:52.45carrarand how good they are
06:53.08carrarmaybe less
06:53.11nix8n82do you have the hardware?
06:53.13rnpbased on the limited info i've given you, how many hours do you think this would take?
06:53.29carrarbased on the limited info I can't say really
06:53.34rnpno hardware, was just going to get a server on slicehost.com
06:53.45kaldemarsbrath: don't mix users.conf and chan_dahdi.conf. actually, don't use users.conf at all for the PRI.
06:54.43sbrathkaldemar: I'm trying to get out of users.conf, but shouldn't the settings work the same?
06:54.52carrarJUST for that single application of dialing numbers by polling a table in mysql you looking at a day
06:55.08carrarat tops
06:55.23carrarbut I suspect you want more out of that app
06:55.27kaldemarsbrath: you don't have the same settings in users.conf and chan_dahdi.conf, and users.conf is not meant for configuring trunks.
06:56.06kaldemarsbrath: do yourself a favor and configure the PRI in dahdi-channels.conf only.
06:56.08rnpwell one concern i have is that once we set the appropriate number of appointments for 1 given area then we need to stop dialing and move on toa  new area
06:56.39carrarif you don't completely spec out what you want someone to write you aren't gonna get what you asked for
06:56.52carrarrepeat, repeat , repeat
06:56.57rnpI understand, spec it out
06:57.11rnpso what do you do carrar?
06:57.19carrareverything
06:57.23rnpbig IT honcho position
06:58.49sbrathkaldemar: I'm pulling it from users.conf, but even thou I have group=2 in dahdi-trunks for the 2nd pri, both pri's are ending up in group 1??
06:59.09kaldemarsbrath: pastebin it
07:00.02sbrathhttp://pastebin.com/m57f36fe5
07:01.15kaldemarsbrath: group definitions must be above channel lines. now you have them both below.
07:01.22sbrathok
07:01.41carrarrnp, 20 years of unix and programing, 15 years of network engineering, 9 years of asterisk
07:02.02kaldemarsbrath: all parameters apply for channel lines below them, until otherwise defined.
07:02.28sbrathkaldemar: very tricky....
07:02.38sbrathok, I have 2 groups now.
07:03.04sbrathstill no caller-id-name
07:03.06kaldemarsbrath: you have a typo here: "callerid=asrecieved"
07:03.33carrar8 years of asterisk, sorry
07:03.40carrar8-9
07:03.43carrarsomewhere in there
07:03.45carrarheh
07:03.47kaldemarsbrath: and the caller id parameters were below "channel => 25-47"
07:04.08kaldemarcarrar: it's the ninth year that makes the difference. :)
07:04.13carrarhaha
07:04.23carrargah, missed it by a hour
07:04.29sbrathso does every paramater in dahdi-channels then behave that way, they all colelct up, and then apply to channels when the channel line hits?
07:04.39kaldemarsbrath: yes
07:05.36*** join/#asterisk dansays (~dansays@cpe-66-108-113-18.nyc.res.rr.com)
07:06.15sbrathwell that would explain most of my dahdi issues then... :)
07:06.20sbrathlet me try it again.
07:07.09sbrathdahdi restart should reload all that, or do I need to do another reload to purge the users.conf stuff.
07:07.09*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
07:07.40*** join/#asterisk benngard (~benngard@213.88.138.230)
07:09.30kaldemarsbrath: i'd restart whole asterisk to be sure. :P
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07:11.50sbrathkaldemar: restarted whole server.
07:12.18sbrathkaldemar: just wondering, are other configs in asterisk order dependant ?
07:12.27*** join/#asterisk pawz (~pawz@ppp118-208-100-34.lns20.bne4.internode.on.net)
07:13.49sbrathcarrar: but the real question is how many programming languages do you know :)
07:14.05kaldemarsbrath: only dahdi, others follow the context structure.
07:14.09*** join/#asterisk dangerdan (~irssi@unaffiliated/dangerdan)
07:14.39sbrathkaldemar: so I'm restarted, and I still see callerid-name in the cdr logs, but still N/A on a DumpChan.
07:15.09*** join/#asterisk soman (~somnath@stargate.starnet.fi)
07:17.07carrar5 'programing langauges'
07:17.14kaldemarsbrath: feel free to pastebin a cli output of a call. add NoOp(${CALLERID(name)}) into the extension aswell.
07:18.10carrarI don't consider shell scripting, cgi, html, SQL etc..programming language
07:18.21*** join/#asterisk lost_soul (~noymfb@cpe-74-71-234-100.twcny.res.rr.com)
07:19.53*** join/#asterisk oej (~olle@ns.webway.se)
07:20.18sbrathcarrar: so C++, Smalltalk, Pascal, Fortran, MASM, etc are "Programming Languages"
07:20.26sbrathonly 5 :)
07:20.29carrarhaha
07:20.58*** join/#asterisk e-jones (~jkastner@nat/redhat/x-ddkyqtytwbluwzye)
07:21.03carrarI wish I knew C++
07:21.37florzSQL is a programming language nontheless ;-)
07:21.43carrarthen 6
07:21.52kaldemarsbrath: what is the name in cdr?
07:22.22florzWell, if you don't consider SQL a programming language, chances are you don't know SQL ;-)
07:22.36sbrathok, I don't have the 9 years, of asterisk, but I'm close on the rest of your numbers :)
07:23.21*** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net)
07:23.43sbrathIf you want to call SQL a language, you need to know more than just Select * from users :)
07:24.05florzthat's basically what I mean :-)
07:25.22sbrathif you can write me a full 20 line "program" in SQL using full ANSI syntax, and inner and outer joins in the same query.... Then you know SQL ..
07:26.16sbrathkaldemar: did you get my msg?
07:26.48florzand some joins with unions of aggregating subqueries, or somesuch, yeah
07:26.50sbrathcarrar: Do you know any other OO languages?
07:27.00carrarno
07:27.15carrarno C++
07:27.25sbrathwe have one SQL query in a reporting app that has 34 pages of SQL!!!!! I wish I could slap the developer.
07:27.38sbrathcarrar: any java or smalltalk?
07:27.41carrarno
07:27.43florzbasically, when you do something that could reasonable be called computation, and not just selection
07:27.52florzerm, reasonably
07:28.13florz34 pages? that'
07:28.18florz34 pages? that's really a lot for SQL :-)
07:28.36sbrathAmazing the DB dosent eat itself when that runs...
07:28.45florzpretty much for any definition of "page" ;-)
07:29.10carrarwith comments you can make a selec count take half a page
07:29.16carrarheh
07:29.22*** join/#asterisk qjb (~qjb@2001:470:1f15:5ad:21f:5bff:fef1:f005)
07:29.23sbrathIt's like about 20 seperate queries for different business metrics data sets all unioned and minus's together, with a crap load of temp-tables...
07:29.56*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
07:30.05sbrath34pages of sql is un-forgivable, when you try to write something in SQL that belonged in Java or the calling app, the developer should be slapped just from the point of Maintenance.
07:30.39florzwell, it really depends on the task, I guess
07:31.07sbrathI can code stored procedures, but when you spend 20 lines of SQL to work around the fact that it's not a "language" it makes the code hard to support when the 3rd person after you inherits it.
07:31.43florzafter all, data aggregation and correlation often can be expressed far more succinctly in SQL
07:31.51sbrathMost of the 34 page SQL is more of a "Hey look how conveluded I can make a SQL to do everything you asked for but nobody else will ever figure this out unless they break it all down line-by-line "
07:32.02sbrathagreed.
07:32.27sbrathbut back to my asterisk problem :)
07:32.30florzwell, yeah, if you start writing loops in SQL despite heving the option to use an application language for that .. then probably something is wrong ;-)
07:32.53sbrathflorz: which flavor of SQL do you do? Oracle/MySql/MSSQL ?
07:33.03florzPg? =:-)
07:33.09sbrathpostgres.
07:33.14florzyeah
07:33.18florzmostly, at least
07:34.09sbrathI've only dabbled at postgres, I had a telco switch once "Broadsoft" that was based in Postgres, and I had to tweak the DB, and do backups. but never "programmed" it... Seems like it's a step up from MySql with stored procedures...
07:34.32sbrathkaldemar: Did I loose you?
07:35.12florzit's ... like ... up a high mountain from MySQL? ;-)
07:35.39sbrathwithout all the "Oracle" ownership behind it :)
07:35.57sbrathat least for the InnoDB stuff.
07:35.58florzwell, Oracle now does endorse Pg, too!
07:36.19florzhttp://postgresql.blogg.se/2010/february/sun-oracle-postgresql.html <- !
07:36.24sbrathbut Oracle Bought InnoDB engine that MySql uses for transactions
07:36.47florzyeah, and with Sun they also bought the endorsement of Pg ;-)
07:37.08florzbut they don't "own" any of it, that's true
07:38.40kaldemarsbrath: strange. the name part must come from somewhere. use PRI debug on an incoming call to see where the information is.
07:41.34*** join/#asterisk oej_ (~olle@ns.webway.se)
07:42.05sbrathI see this: Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
07:42.13sbrathAfter the DumpChan
07:43.12sbrathand a few more lines lower this:  < Facility (len=31, codeset=0) [ [Feb  4 01:40:33] VERBOSE[5029] chan_dahdi.c: 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0F, 'BRATH SHANE
07:43.44sbrathIs the call getting the name info to late?
07:44.21sbrathand then: Received simple calling name 'BRATH SHANE    '
07:44.57sbrathBut all the happened "After" the call was connected to my SIP phone.
07:45.18sbrathso the CDR being written down after the call is done has the "late" caller Name info.
07:47.39*** join/#asterisk e-jones (~jkastner@nat/redhat/x-juljjdpfqsunkynu)
07:49.15kaldemarsbrath: well, you'll have a hard time getting the information to your phone. you could try to Answer() the call before dialing the SIP phone, but that's a bit ugly.
07:50.04sbrathIs this just a case of the telco takes 15 ms to lookup the Cid, and the call is routed to quickly?
07:51.42sbrathI did a wait2, answer, and now I get the name...
07:51.45sbrathGeesh.
07:52.03*** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:147a:5361:a65a:7c98)
07:52.22kaldemarmaybe, don't know. if the progress message is not tied to an answer, a mere Wait might be worth a try.
07:53.01sbrathok, off to bed... night and thanks for the help.
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08:15.54qjbI have a problem with playing custom numbers using sayDigits.
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08:32.49qjbI am using Asterisk 1.6
08:35.18kaldemarqjb: what is the problem? and which 1.6?
08:38.17qjbWe use 1.6.2.0-beta2
08:38.53qjbThe problem is I would like to play custom sound digits with sayDigits. However when I set the language it keeps playing the english version
08:39.13*** join/#asterisk eggers (~eggers@cpe-70-124-59-214.austin.res.rr.com)
08:39.24qjbI put the sound files under /var/lib/asterisk/sounds/za/digits/[0-9].au
08:39.47kaldemarhow do you set the language? and btw, 1.6.2 is on 1.6.2.2 already, that's a really old beta.
08:39.51qjbAnd use Set(language=za) to set the language
08:40.18qjb(Beta is old but I cannot change it at the moment, need to retest everything first)
08:40.33qjbI check the language using: NoOp(LANGUAGE ${LANGUAGE})
08:42.03qjbThat reflects the setting but then it happily using 'digits/9.gsm' (language 'en')
08:48.33kaldemaryou're setting it wrong, use Set(CHANNEL(language)=en)
08:48.56kaldemarlanguage=.. is used in the configuration files.
08:50.02kaldemarand, check it with ${CHANNEL(language)}
08:51.10*** join/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228)
08:51.12kaldemarof course replace "en" in the Set with what you want.
08:52.14qjbChanged it but it makes no difference
08:53.02kaldemarwhat does the language check say? where are your sound files?
08:53.13*** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99)
08:53.47qjbThe check reflects the language I set. The sound files are in /var/lib/asterisk/sounds/za/digits
08:54.08qjbIn /var/lib/asterisk/sounds/za I only have the 10 digits
08:54.20qjbIn .au format
08:54.31c0rnoTaHello, everybody. Can anybody tell me how i can count channels on specific extension at moment. For example, i have extension _9XXXXXXX and i want to know how much channels now are on that extension?
08:54.35*** join/#asterisk mlarsen (~mlarsen@212.37.141.188)
08:55.40plundraHow does penalty on a member in a queue actually work? I mean, what happens with a value of 1? what about 5?
08:55.41kaldemarc0rnoTa: in shell, do asterisk -rx 'core show channels' and then grep it as you wish.
08:56.12plundraI want to see it as kind of a tripping-point, n-tries on lower priority memebers have to be tried first.
08:56.23plundra-e
08:56.26qjb(I actually copied the en digits to za digits to be sure the format was not the problem)
08:59.37*** join/#asterisk pawz (~pawz@ppp118-208-100-34.lns20.bne4.internode.on.net)
08:59.54kaldemarqjb: do you have languageprefix=yes in asterisk.conf?
09:00.19c0rnoTakaldemar: Ok, but 'core show channels' show extens like 91234567 or 97654321 not "_9XXXXXXX". And i couldn't select which of them use "_9XXXXXXX" and which "_97.". Only using analysing process in my mind i could divide it. So, i want to divide it automaticly in AGI script.
09:00.39qjbkaldemar: that line is commented
09:01.03kaldemarqjb: uncomment, restart, try again
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09:02.37*** join/#asterisk markwaters (~markwater@weloveit.info)
09:03.01qjbkaldemar: I used core restart gracefully after the change but it did not fix the issue
09:03.15qjbstill playing Playing 'digits/9.gsm' (language 'en')
09:03.39kaldemarfeel free to show a cli output of the whole call.
09:03.49markwatersanyone got any ideas for getting asterisk to send me messages via irc for things like incoming callerid , i do it with jabber atm but would prefer irc
09:03.51kaldemaruse a pastebin
09:03.53c0rnoTakaldemar: speaking in global of my task, i want to limit abilities of outgoing calls through trunks on each exten.
09:03.59mlarsenCan anyone confirm that the following line is supposed to goto ext-queues,200,1 if the date is within january 5th to febuary 5th, and if the day of week is within monday to thursday and the time of day is with in 08:00 to 18:00, because that ain't happening?
09:04.00mlarsenexten => 82,1,GotoIfTime(08:00-18:00|mon-thu|5|jan-feb?ext-queues,200,1)
09:04.51dandreHello,
09:05.30c0rnoTamlarsen: it will go to ext-queues only in 05.01 or 5.02
09:05.37c0rnoTanot 5.01-5.02
09:06.30c0rnoTai think :)
09:07.51*** join/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net)
09:08.09dandreI am using originate command from the manager interface. Everything works fine but I need toset some variable when the first phone is dial and make this variable available when the party is dialed. I have tried to do something like set(__foo="bar") but ${foo} remains empty on the second leg of the call.
09:08.09dandreHow could I do?
09:09.05kaldemarmlarsen: use two lines. one for january with days as 5-31 and one for february with 1-5.
09:09.14AmorsenIs there an easy way to generate a unique file name for ReceiveFax?
09:09.26mlarsenThank you for your replies
09:09.29AmorsenI can't think of a way which is guaranteed to work without race conditions
09:10.03c0rnoTaAmorsen: ${UNIQUEID} ;)
09:10.25Amorsenc0rnoTa: Clever
09:11.14*** join/#asterisk pentanol (~pentanol@77-35-1-202.pppoe.primorye.net.ru)
09:11.20pentanolhello
09:11.53c0rnoTaAmorsen: it's the easiest why, i think. Anyway, I'm always use this way.
09:13.04*** join/#asterisk lost_soul (~noymfb@cpe-74-71-234-100.twcny.res.rr.com)
09:13.27c0rnoTaAmorsen: you can get epoch timestamp too
09:16.05pentanolif I would use asterisk +cdr, I should do trigger in the database so as to synchronize sip accounts?
09:16.05AmorsenNext question: How do you run System on untrusted data without having problems with people adding e.g. quote characters?
09:18.00c0rnoTaAmorsen: only to use AGi before System to prepare untrusted data (remove quote, or 'commenting' it like \").
09:18.08AmorsenOuch.
09:18.28c0rnoTaonly to work with the string before it sill be passed to System
09:19.20*** part/#asterisk markwaters (~markwater@weloveit.info)
09:19.35c0rnoTaor check it in the field, where it should be typed. For ex., in PHP form
09:20.04Amorsenc0rnoTa: I'm trying to pass REMOTESTATIONID from ReceiveFax
09:20.32AmorsenIt would suck to be 0wned by a fax
09:20.48AmorsenIf somewhat unusual
09:22.02AmorsenSo the solution is to use an AGI instead of System
09:22.36c0rnoTai think so
09:24.49c0rnoTadon't use REMOTESTATIONID as argument for AGI. get it from $AGI->get_variable
09:24.56AmorsenRight
09:25.16AmorsenNext challenge is to ensure that I actually get to the AGI; the sender might hang up on me
09:26.57c0rnoTause AGI on 'h' extension like DeadAGI
09:27.13AmorsenRight
09:27.23c0rnoTaRceiveFax should be last exten in fax receive logic
09:27.58c0rnoTapost processing only on 'h' extension.
09:29.48AmorsenDoes the h extension get called even when my side does a Hangup()?
09:30.25c0rnoTayes
09:30.28TommyBottenHow can i set the callerid for analog lines in asterisk? Callerid(num) does not seem to have any effect.
09:31.14AmorsenTommyBotten: Are you trying to set it towards your service provider or towards your phones?
09:34.26c0rnoTaAmorsen:  "h" exten always be called. it does not matter who makes  Hangup. Another trouble is to destroy channels after call ends, and precess 'h' logic on ZOMBIES channels. If your AGI script takes a long time to work, and bridged channel don't want hangup, you did 'exten => s,n,Hangup', and 'h' logic starts, but channels are connected, and billing seconds still rise.
09:35.20c0rnoTathat's the problem i still can not resolve.
09:36.21TommyBottenAmorsen: Towards my provider... but it seems that it's not the callerid after all. When sending a fax from an analog device via asterisk and sip to my vendor, the fax origin lists as a different number than the callerid
09:36.30TommyBottenIn fact a number that is not in our series at all.
09:37.25c0rnoTaTommyBotten: i think, is telco issue, not your's
09:39.05*** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda)
09:39.30mbrevdacan you set bindaddr on a per trunk/peer basis?
09:41.01kaldemarmbrevda: no
09:41.04*** join/#asterisk soman (~somnath@stargate.starnet.fi)
09:42.07AmorsenTommyBotten: You can't send callerid towards your provider with analog
09:42.10c0rnoTaanother question, can i set one externip for first peer and another for second
09:42.11*** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net)
09:42.17c0rnoTa&
09:42.18c0rnoTa?
09:42.19AmorsenThat's hard coded per analog line
09:42.27mbrevdakaldemar: thnx
09:43.06kaldemarc0rnoTa: no
09:43.40c0rnoTaok, thx
09:50.31*** join/#asterisk krion (~seb@unaffiliated/krion)
09:53.11TommyBottenc0rnoTa: It sounds like it.
10:03.32*** join/#asterisk chasing`Sol (~Ahmed@217.54.177.218)
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10:11.50chasing`Solis there a way to disable call transfer to certain extensions?
10:14.34krionhi
10:14.53krionanyone using sevana product in order to measure MOS voip score  ?
10:16.44*** join/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com)
10:16.54jmlsmorning all
10:16.59jmlsa little question:
10:17.56jmlsthe manager originate command can take any number of variable:foo=baa constructs, and these are available to the dialplan (${foo})
10:18.27jmlscan you add variable:foo=baa to other ami commands, such as  Redirect ?
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10:25.13krionjmls: you could try and see :)
10:26.06*** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net)
10:27.10jmlskrion: yup. I could.
10:27.30jmlsI could also write my own pbx from scratch ;)
10:28.00krion:)
10:28.08jmlsjust hoping someone else had already thought about this and tried.
10:28.17jmlsnm. I'll check it out
10:29.04*** join/#asterisk soman (~somnath@stargate.starnet.fi)
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10:31.10fiddurjmls: Only originate, afaik...  the documentation from "manager show command ..." should be accurate
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10:50.44cavalier_workHi, I've setup an asterisk server (opensuse 11.2) and configured it with the web gui (diguim asterisk) i've added a few users for sip phones. I can call the demo extension and these work. If I call another phone it rings and i can pickup the phone, but I get no audio.
10:51.48*** join/#asterisk albertoandrade (~albertoan@189.58.23.130.dynamic.adsl.gvt.net.br)
10:51.48cavalier_workI enabled rtp set debug on and the last thing it says for rtp is Packet2Packet bridging SIP/6003-0000002b and SIP/6000-0000002c
10:52.37cavalier_workI searched for this but I couldn't find a solution
10:55.42*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)
10:57.36cavalier_workMy bad, it seems to be a problem with the firewall
11:06.21*** join/#asterisk garymc (~garymc@81.138.225.161)
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11:15.19TommyBottenIs it possible to have a .call file do multiple comands - without it having to enter the dialplan
11:17.45AkiraaDoes anyone have some experience with the Cisco SPA9000 IP PBX?
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11:21.00hlueseaHello channel, I try to configure a voicemail for the numbered extensions but i failt. My conf files are here http://pastebin.com/d4f242466. What should i do ?
11:26.32garymcanyone know what I need to set in Asterisk CLI to monitor a Analogue call? As im on PRI do i still use Pri debug span 1 ?
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12:10.19kaldemargarymc: just add verbosity and you'll see the call. if you want more accurate debug, use core set debug.
12:10.52kaldemargarymc: the pri commands only apply for PRI channels, not for analog channels.
12:11.46garymcright
12:12.01garymcso why is my analogue not displaying the DID i want it too?
12:13.06kaldemarbe more specific.
12:14.00kaldemarstart with what kind of channel we're talking about, what you have connected to the channel, what you expect it to do and show a cli output of a call.
12:14.40kaldemarhluesea: how did you fail?
12:15.46*** join/#asterisk cuco (~Diego@local.xorcom.com)
12:16.30garymcok. I have an A200 synced with A101D . The A200 is for FAX through my PRI. E1. I set the extension on my Analogue A200 to display DID 881049 like I do on my sip extensions. But it displays the default number 01514876699 . DEBUG here http://pastebin.mozilla.org/700984
12:18.21kaldemargarymc: i suppose you have already asked folks at #freepbx?
12:18.36garymci have
12:18.52kaldemarin asterisk terms, what do you mean by "set the extension on my Analogue A200 to display DID 881049"?
12:19.29garymcwell thats a freepbx thing i think.
12:19.31kaldemaryou don't have anything like that in the extension you pasted.
12:19.40garymcI know. its not showing it
12:19.52garymcits showing 01514876699 as the outgoing
12:20.32kaldemar"-- Executing [s@macro-outbound-callerid:12] ExecIf("DAHDI/32-1", "1|Set|CALLERID(all)=01514876699") in new stack"
12:20.38garymcyes
12:20.49kaldemarthat's a freepbx issue if it's supposed to do something else.
12:21.07kaldemargo bug them some more.
12:21.09hlueseakaldemar : actually i hear the beep tone and i try to record somethings, but i can't found any records like prompt neither /var/spool/asterisk/voicemail/ nor /var/lib/asterisk/sounds/en
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12:21.40garymcok
12:23.30kaldemarhluesea: this looks really strange: exten => 500,2,Record(prompt:gsm)
12:23.33*** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844)
12:23.35kaldemarhluesea: show a cli output of the call
12:30.43TommyBottenIs there a way to manually set hints?
12:33.47*** part/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com)
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13:07.39krioni want to send a wav directly to a sip account, can you spare me time and tell me an FOSS who does it ?
13:08.13Gido-Ekrion ?
13:08.29Gido-Eyou can spare me time and fund my bankaccount
13:08.40krion:-)
13:08.49krionsorry if my english is not understandable
13:09.16krioni meaned to ask for help nicely, not rudely
13:09.18*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
13:13.23kaldemarkrion: if you mean calling someone and playing a wav file, asterisk can do that for you. you can judge FOSS part yourself. :)
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13:30.12ManxPower-workTommyBotten: As I understand it, in 1.6 you can manually set the device state in the dialplan.
13:31.43leifmadsenDEVICE_STATE() function I think
13:32.42TommyBottenThanks... I'll check it out
13:34.57Kobazhow would i send a hangup to a polycom phone to get rid of a call.... ie: a call comes in, asterisk crashes, and now there's a phone call ringing on the polycom that cannot be picked up
13:36.01Gido-Easterisk crashes?
13:36.04Kobazand this is part of a ringall queue... so now you have to go around and hit the reject button on 25 phones
13:36.07Gido-Ethat souldn't happen
13:36.12*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
13:36.15KobazGido-E: it does... get used to it :P
13:36.25Gido-Eit NEVER! crashes!
13:36.28Kobazheh
13:36.55Gido-EKobaz whe have multi tenant servers, never ever does a production environment crash
13:37.01Kobazso the 90830983405834 crash reports on issues.asterisk.org are figments of my imagination...
13:37.11*** join/#asterisk jmacz (~jmacz@190.144.75.22)
13:37.23Gido-EKobaz go regression test. use sipp
13:37.30KobazGido-E: i have a 1.4 server that's been running for 1 year plus straght
13:37.39KobazGido-E: not good enough... i have my own testing suite
13:37.54Gido-EKobaz wat is nog good enough? SIPP?
13:38.03Gido-EWhat is nog good enough on it?
13:38.06Kobazand i have other servers, running for weeks,months without crashes... and then I have this one system that crashes several times a day
13:38.24KobazGido-E: i need lots of custom code to test my crap
13:38.33Kobazbut anyways
13:38.44Gido-EKobaz on centos?
13:38.51Kobazthe issue here, is that, i need to be able to drop a call on a polycom, without asterisk actually knowing about the call
13:38.54Kobazdebian
13:39.06Kobaz2.6.27.38
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13:39.54Kobazi wonder if i just store the sip call id, and then just fire off a spoofed cancel packet
13:40.58KobazGido-E: i wish i could always reproduce the problem on a consistant basis.. then i can find and fix the bug, or if i can't,.. post the bug and the asterisk guys can fix it
13:41.22Kobazbut asterisk tends to crash when you're using lots of local channels... it'll crash on channel hangup
13:42.20Gido-EKobaz ok. Try to have the same basic LIBC and GCC versions as centos.
13:42.33Kobazi don't think it has anything to do with libc
13:43.29Kobazthere's either memory corruptio or buffer overflow in asterisk somewhere, or a referencing counting problem, or something like that
13:43.45Kobazi need to write more tests
13:44.14Gido-EKobaz ok, you did write your own RTP implementation for testing?
13:45.31Kobazno, i have a whole slew of asterisk call drivers that can simulate just about anything you can do in asterisk
13:46.41Gido-EKobaz? I have a dialplan for it.
13:47.01Gido-EI just use SIPP, to make and recieve the calls           SIPP - Asterisk - SIPP
13:47.27Kobazyeah
13:47.31Kobaznot good enough for my purpose
13:47.46Gido-Ewhat can you test, i can't?
13:47.48Kobazsipp just makes sip calls, and has limited call control
13:48.33Gido-EKobaz ok.
13:48.42Kobazi can wait for events, wait for speech/silence, send dtmf, run any dialplan
13:49.26Gido-EKobaz why  sould that not be able with my setup?       The magic is done by asterisk.
13:49.40Gido-ESIPP is just a stupid caller or callee
13:49.44Kobazi don't need a call blaster, i need scenerio testing
13:49.54Kobazwhich i've already written
13:50.24KobazGido-E: well the point is, you still need asterisk
13:50.36KobazGido-E: sipp alone isn't a solution
13:50.43dlynes_laptopKobaz, I surmise the polycom phones cannot be logged into with a text browser such as lynx or links, or using wget?
13:50.53*** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net)
13:51.10Kobazdlynes_laptop: you can log into a polycom with a web browser, but you cant send it commands like 'hey hang up this call'
13:51.12[TK]D-Fenderdlynes_laptop: You can... but it won't be pretty
13:51.28dlynes_laptopKobaz, but if you reboot it, it'll hang up the call
13:51.37[TK]D-Fenderdlynes_laptop: (links).  WGET has no point.
13:51.40dlynes_laptopKobaz, or you can't reboot a polycom from the web interface?
13:51.42*** part/#asterisk cavalier_work (~cavalier_@ip4daa055d.direct-adsl.nl)
13:51.46Kobazyeah but, hitting the reject button is 100 times faster than rebooting
13:51.56Kobazyou can reboot a polycom with a sip notify
13:52.01Kobazyou dont even have to hit the web interface
13:52.06dlynes_laptopok
13:52.27Kobazi think i'll experiment with spoofing a sip CANCEL
13:53.08dlynes_laptopKobaz, incidentally, which version of asterisk are you running?
13:53.19dlynes_laptopI'm just curious
13:53.25*** join/#asterisk MedicineMan (~jrodgers@75.87.82.200)
13:53.36Kobaz1.6.0.19
13:53.40Kobaz1.6.0.20 crashes even more
13:53.53dlynes_laptopYeah...I've heard the whole 1.6.0 series is quite unstable
13:54.04*** part/#asterisk MedicineMan (~jrodgers@75.87.82.200)
13:54.11Kobazit's great if all your doing is passing in and out sip calls
13:54.14Kobazor t1, or whatever
13:54.23Kobazif you're running lots of dialplan with local channels, not so good
13:54.33dlynes_laptopI've got my own issues with 1.6.1.8....just going to upgrade to 1.6.1.14 right now
13:54.39dlynes_laptopHad a whole machine go down last night
13:54.47Kobazyeah i've had asterisk kill a box
13:54.54Kobazsucks up 100% cpu, and all available io
13:54.54dlynes_laptopwell
13:54.58dlynes_laptoplemme restate that
13:55.01Kobazthe only thing you can do is hit the reset button
13:55.11dlynes_laptopAsterisk locked up...not the whole box
13:55.14Kobazyeah
13:55.16Gido-Edlynes_laptop which distro?
13:55.25[TK]D-Fender1.6.0.22 is out....
13:55.26Kobazi've found that asterisk -r is bad
13:55.30dlynes_laptopGido-E, it doesn't matter...the distro has nothing to do with it
13:55.36Gido-Edlynes_laptop it has
13:55.40Kobaznot using realtime solved a lot of my lockup issues
13:55.41dlynes_laptopGido-E, how so?
13:55.47Gido-EBut i am not going to argue about that.
13:55.49dlynes_laptopKobaz, i'm not using real time
13:55.53Gido-EI am experience on that point.
13:56.07dlynes_laptopGido-E, it's Debian Lenny
13:56.13Gido-Edlynes_laptop ok, try centos
13:56.15dlynes_laptopno
13:56.18Kobazhaha
13:56.19Gido-Ei think it will be more stable.
13:56.36TommyBottenleifmadsen / ManxPower-work: function_devstate does mostly that. Is there a way to manipulate the actual device state, and not a custom one?
13:56.36KobazGido-E: distro has very little bearing
13:56.51KobazGido-E: bugs in asterisk are bugs in asterisk
13:57.06dlynes_laptopGido-E, wtf's your problem?  Centos and Debian are both running the exact same operating system
13:57.08Gido-Eok, and i am on 1.4
13:57.10Kobazyes, if you had a buggy kernel or libc, you'll be having problems, but that's more rare
13:57.27dlynes_laptopGido-E, I trust Debian a lot more than Centos
13:57.33Kobazhah, yeah
13:57.40ManxPower-workdlynes_laptop: Chances they are running different kernel, gcc, etc versions.  You're right, the distro has little bearing on this.
13:57.40Kobazi <3 my debian
13:58.02KobazManxPower-work: i can run centos... and build my own kernel, upgrade gcc, put on a broken libc
13:58.21dlynes_laptopManxPower-work, yeah, but Debian tends to test their libc/gcc into the ground before they release, so they're usually several versions behind Centos
13:58.32KobazManxPower-work: the distro just by itself is almost meaningless.. what matters are the versions of these particular packages that are used as part of the distro
13:59.12Kobazusing "centos" rather than "debian" or "fedora" is just religion
13:59.22Gido-EKobaz you have to learn a lot.
13:59.27KobazGido-E: ?
13:59.28dlynes_laptopGido-E, btw...fwiw, my Debian boxes have less problems than my Centos boxes, but stupid software like FreePBX is very Centos-centric
13:59.37KobazGido-E: i've been in this industry for almost 20 years
13:59.49Gido-EKobaz so?
13:59.58Gido-EMy bank account is bigger!
14:00.13KobazGido-E: from my experience, i have a good idea of what actually matters
14:00.38[TK]D-Fenderdlynes_laptop: How is FreePBX distro-centric?
14:00.55dlynes_laptop[TK]D-Fender, it's completely broken for an install on anything other than Centos
14:01.08dlynes_laptop[TK]D-Fender, You have to run a special script to get it to work properly on Debian
14:01.14[TK]D-Fenderdlynes_laptop: BS... tons of people running it on Ubuntu, Debian, Slackware, etc
14:01.21dlynes_laptop[TK]D-Fender, and even then, the script doesn't work 100%
14:01.31dlynes_laptop[TK]D-Fender, i didn't say it couldn't run on other distros
14:01.37Kobazanyways, off to breakfast
14:02.14dlynes_laptop[TK]D-Fender, Have you tried installing it on Debian?
14:02.18krionkaldemar: yes, i want to do it from a client side, like ekiga or something
14:02.19*** join/#asterisk _omer (~omer@119.152.140.100)
14:02.24ManxPower-workUm. FreePBX doesn't work 100%
14:02.47krionkaldemar: it's in the purpose of measuring MOS score
14:02.49KobazGido-E: i agree there's always more to learn, i do need to learn tons of stuff that I know I don't know... but most likly it's not the things you think
14:02.57_omercompiling asterisk-1.4.29 in CENTOS 5.4  .... error message... http://www.pastebin.org/86220   any help ?
14:03.56[TK]D-Fender_omer: ther is no error message in there.
14:04.09dlynes_laptop_omer, where's the rest?
14:04.43ManxPower-workI thought I was the only one that didn't see an error.
14:04.44AmorsenHmm, ReceiveFax is fine and dandy but unfortunately I can kill Asterisk 1.6.0.21 with it
14:04.49_omerlet me paste once again please
14:05.08krionmaybe if i call someone with ekiga then i run my wav from my computer the called would hear it
14:05.14dlynes_laptopAmorsen, might help to let peeps know if you're using ffa, or the built in fax module
14:05.20_omerthere is not any error message..but compilation is stopped ..
14:05.31*** join/#asterisk benngard (~benngard@213.88.138.230)
14:05.32Amorsendlynes_laptop: Built-in
14:05.50dlynes_laptop_omer, there must be something, because in the pastebin you pasted, it mentions a problem from previous
14:05.54_omerhttp://www.pastebin.org/86229
14:05.59AmorsenI guess I have won another rebuild, this time with lock debugging enabled
14:05.59[TK]D-Fender_omer: maybe its just FINISHED.
14:06.23ManxPower-work_omer: What is the command you are running?
14:06.24[TK]D-Fender_omer: "make clean" != compile.  MAKE = COMPILE
14:06.29*** join/#asterisk muiro (~muiro@unaffiliated/muiro)
14:06.44Gido-E_omer it tels you it went ok
14:06.45dlynes_laptop_omer, what i see in your second pastebin is that it worked, with  no errors
14:06.49_omeroh..
14:06.55Gido-Ewhy do yuo think it went wrong?
14:07.06[TK]D-FenderGido-E: clue[-1]
14:07.09Gido-EYou issue, a make clean
14:07.10dlynes_laptop_omer, i thought those other two lines were output by the makefile, but i guess they were just your comments
14:07.22_omerI think I need to do make install too ... oops first time in linux and asterisk
14:07.39Gido-E_omer dutch?
14:07.57[TK]D-Fender_omer: "make clean", "make", "make install"
14:08.04dlynes_laptop_omer, make clean ; ./configure && make menuconfig && make && make install
14:08.08Gido-Edont forget: make menuselect
14:08.24_omerok
14:08.25dlynes_laptoperm menuselect...getting it mixed up with the kernel
14:08.29[TK]D-Fenderyeah... the ./configure would help...
14:08.31*** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com)
14:08.37AmorsenMy my, it appears I'm hitting AST-2010-001
14:09.18_omer**** The configure script must be executed before running 'make'.
14:09.18_omer****               Please run "./configure".
14:09.23dlynes_laptopAmorsen, 1.6.0.22 is out...don't know if it solves that issue
14:09.25_omeri did  ./configure
14:09.30dlynes_laptop_omer, what did I tell you?
14:09.31_omerand then make and make install...
14:09.34_omerreturn remains the same
14:09.46dlynes_laptop_omer, you didn't do './configure', or it wouldn't be telling you there
14:09.48dlynes_laptop_omer, you didn't do './configure', or it wouldn't be telling you that
14:09.51kaldemarkrion: or just make an extension that originates a new call
14:10.34dlynes_laptop_omer, perhaps you're trying to do make before configure?
14:10.39dlynes_laptop_omer, not the other way around?
14:11.10Gido-E_omer, after running  "make clean" you can start over again.
14:11.27Amorsendlynes_laptop: It says it does, I'm compiling now
14:11.31_omerhere is the result of ./configure  ... http://www.pastebin.org/86232
14:12.15Gido-E_omer on centos?
14:12.23_omeryes  Centos 5.4
14:12.26Gido-Eyum install gcc
14:12.36Gido-Eyum install ncurses-devel
14:12.55_omerPackage gcc-4.1.2-46.el5_4.2.i386 already installed and latest version
14:12.55_omerNothing to do
14:13.16_omerPackage ncurses-devel-5.5-24.20060715.i386 already installed and latest version
14:13.16_omerNothing to do
14:13.19[TK]D-Fender_omer: G++ <-------------
14:13.38[TK]D-Fender_omer: Go read the INSTructiONS which tell you the pile of packages you'll need...
14:13.42Kattyhi.
14:13.44[TK]D-Fender_omer: You don't seem to have done this
14:13.44dlynes_laptop[TK]D-Fender, cpp is the c preprocessor, not the c++ compiler
14:13.54_omerG++ is not a package..
14:14.42Kattyyour mom's a package
14:14.42Gido-Eyum install gcc-c++
14:14.42Amorsengcc-c++...
14:14.42dlynes_laptopKatty, aren't you chipper this morning? :)
14:14.42Kattymostly asleep
14:14.42Gido-Emaybe, i am not into centos that much
14:14.43Kattyno caffeine today
14:15.10_omergcc-c++   .. updating something
14:15.20Gido-Etry again...
14:15.30dlynes_laptopWhy is it calling '/lib/cpp' a c++ preprocessor?  very weird....
14:15.52Warp4dlynes_laptop, because it is
14:16.03Gido-Eit needs to :-)
14:16.13Warp4IIRC , g++ handles the actual C++ compiling
14:16.13dlynes_laptopWarp4, hrm...could've sworn it was a C preprocessor in gcc 2.95
14:16.39Warp4dlynes_laptop, things change with newer versions, i suppose :)
14:16.59dlynes_laptopWarp4, so asterisk actually has c++ code now?
14:17.09Gido-EYep, seems to...
14:17.18Gido-Eor just one of the wrappers for easy install...
14:17.48Kattyzonks out
14:18.04Gido-E_omer there is repo for centos asterisk, check out asterisk.org
14:18.19*** join/#asterisk brad_mssw (~brad@shop.monetra.com)
14:18.25_omerrepo?
14:18.34_omerok let me check
14:19.28dlynes_laptopKatty, you've been up all night?
14:19.40*** join/#asterisk knctrnl (~aembrey@76.164.169.130)
14:20.12Gido-E_omer http://www.asterisk.org/downloads/yum
14:21.06Kattyno )=
14:21.15Kattyi slept rather well.
14:21.27_omerok
14:21.56*** join/#asterisk minotaur01 (~minotaur0@24.215.3.50)
14:22.09dlynes_laptopKatty, ah...I guess you just need your timmy's, dunkin' donuts, starbucks, or whatever it is you drink in missouri :)
14:22.39*** join/#asterisk andres833 (~andres833@190.144.75.22)
14:22.40Kattyehhh well
14:22.43Kattyi'm giving up soda, again
14:22.44_omerGido-E .. thanks ..
14:22.50Kattywhich is what i usually have in the AM
14:23.00dlynes_laptopKatty, ah...soda's not good....way too much sugar
14:23.07Kattynods
14:23.09_omerbut I think  yum install gcc-c++  solved the problem
14:23.15dlynes_laptopKatty, or that nasty aspartame crap
14:23.23Kattyugah, i can't drink that stuff
14:23.28Kattyit makes me hungryhungryhungry
14:23.29_omerso what is the sequence?  ./configure  make   make clean ?
14:23.50*** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net)
14:23.52_omer./configure  finished WELL .... gave me ASTERISK BIG LOGO
14:24.06[TK]D-Fender_omer: read the INSTRUCTIONS int he tarball ----------
14:24.07dlynes_laptopKatty, i won't even touch it...it fools your body into thinking it's sugar, your body processes it like sugar (or at least attempts to do so), and you get mutated cells as a result
14:24.10Gido-E./configure && make menuselect && make && make install
14:24.19dlynes_laptopKatty, it shouldn't even be legal to sell it to anyone but diabetics
14:24.19_omerok....thanks
14:24.28_omerI will read the instructions now...
14:24.29Kattydlynes_laptop: ehh i used to drink it back in the day
14:24.31_omerthanks all
14:24.33Gido-Emake clean, whill just make a clean environment, as if you just downloaded it.
14:24.35Kattydlynes_laptop: and i drank it for years and years and years
14:24.45Kattydlynes_laptop: and wondered why i was starving all the time
14:24.52Kattydlynes_laptop: course i cooked with splenda.
14:25.02Kattydlynes_laptop: and bought all sugar-free items when i grocery shopped
14:25.05dlynes_laptopKatty, splenda's way better for you than splenda
14:25.10dlynes_laptoperm nutrasweet i mean
14:25.11Kattydlynes_laptop: orly
14:25.15Kattydlynes_laptop: is that so ;>
14:25.21Kattyidk what nutrasweet is
14:25.29Kattyaspartame? saccarin?
14:25.31dlynes_laptopKatty, splenda is sucralose, which is just sugar with a molecule removed
14:25.45Kattystill makes me hungry
14:25.58dlynes_laptopKatty, aspartame and saccharine are both chemicals, and both quite nasty at that
14:25.58Kattyhow would you like to feel hungry all day long
14:26.11*** join/#asterisk basty (~basty@212.218.65.131)
14:26.13bastyHi
14:26.14Kattyyou could eat a bigmac combo meal
14:26.19Kattyand then an hour later, be hungry
14:26.24dlynes_laptopKatty, there's also stevia (a natural sweetener), but apparently it's not very healthy either
14:26.25*** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com)
14:26.35Kattyi'll stick with sugar cane
14:26.45*** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net)
14:26.55Kattyi think i'd rather have that then shovel creepy processed food items in me
14:27.02dlynes_laptopKatty, we try to buy everything with cane sugar, or honey, or chinese herbal sweeteners
14:27.05Kattyhello daddy voipmonk
14:27.14Kattymmmm, honey
14:27.18Kattygood with pb :>
14:27.24dlynes_laptopheh
14:27.24dlynes_laptopyeah
14:27.30*** part/#asterisk Zeeek (~Zeeek@pdpc/supporter/active/zeeek)
14:27.50dlynes_laptopKatty, if you go into a chinese supply store, pretty much all the sugar in there is cane sugar
14:27.58dlynes_laptopKatty, not as sweet as white sugar, and less processed
14:28.01bastyI have a strange problem. I just installed asterisk 1.4.29 on an ubuntu 64bit system. Everything works well. After I restarted the machine, I noticed that the CPU Load for the Asterisk is 100%. So I stopped the daemon and started the asterisk manual. The cpu load was gone. So I stopped it again..and ran the init script again...and boom - asterisk uses 100% again. So there must be something weirdo with the init script I guess....anyone knows of that "problem"
14:28.13Kattydlynes_laptop: good stuff.
14:28.29dlynes_laptopKatty, but they also use chinese dates to sweeten the soup
14:28.45coppiceI dated a few Chinese
14:28.56AmorsenNope 1.6.0.22 didn't fix it. Damn, I'm back to Hylafax for now.
14:29.09dlynes_laptopKatty, and some other thing (it looks like a brown ball, very light, feels like it's hollow, and sold in a plastic globe), that's a natural sweetener
14:29.12Kattydon't think i've ever had a chinese date
14:29.24dlynes_laptopKatty, it's quite similar to a honey date
14:29.27Kattybut i will put it on my grocery list
14:29.40dlynes_laptopKatty, they're always sold dried, not fresh
14:29.40coppiceKatty: I think he means red dates
14:29.42Kattyi found a recipe for this amazing looking SOUP
14:30.08dlynes_laptopcoppice, they're red in color, yes...but afaik, they're only called chinese dates
14:30.09Kattymust aquire groceriees to make it :>
14:30.29Kattydigs through TOH website
14:30.32dlynes_laptopcoppice, do you know what those brown balls are called, offhand?
14:30.33coppicedlynes_laptop: here in china we call them red dates
14:30.38dlynes_laptopah
14:30.50coppicethey make nice tea
14:30.58dlynes_laptopcoppice, they're some kind of a fruit that's grown on trees, and they're used for sweetening soup
14:31.12coppiceand tea
14:31.13dlynes_laptopcoppice, they can also be processed to produce sugar
14:31.39dlynes_laptopbut i guess you don't know the name
14:31.48dlynes_laptopnever heard of anyone using them in tea, though
14:32.51coppiceIn Shenzhen Science Park there is an excellent SiChuan restaurant that serves red date tea to everyone
14:33.36coppiceand they fill your cup in the really traditional way, with a kettle that has a super long spout
14:34.32dlynes_laptopcoppice, oh...thought you were talking about the chinese honey dates
14:34.40dlynes_laptopcoppice, not the brown balls
14:36.15dlynes_laptopcoppice, sounds like those tea pots that you see in the old chinese dynasty films
14:37.18Kattyhttp://www.recipezaar.com/Crock-Pot-Taco-Soup-40022 <- tell me that don't sound good
14:38.20coppicedlynes_laptop: that's the kind of thing. a brass kettle with a metre long spout
14:38.22creativxcrack pot?
14:38.44coppiceyou have to store your crack somewhere
14:39.33dlynes_laptopKatty, now why would anyone drain the juices?  that sounds just plain weird
14:39.35[TK]D-Fenderwould never do a drug named after a part of his ass
14:39.43Kattydlynes_laptop: some of us don't like grease
14:39.46*** join/#asterisk brezular (~brezular@adsl-dyn-104.95-102-247.t-com.sk)
14:40.09coppiceKatty: you must hate Shanghai food :-)
14:40.15Kattynever had it.
14:40.23Kattybut if it's overly greasy, probably make me sick
14:40.24[TK]D-Fender^ Tragically white
14:40.52Kattysome dishes aren't liquidy
14:40.54[TK]D-FenderKatty: Hot damn girl..... get some colour in ya!
14:41.05eppigyMONRING
14:41.13Kattyhugs eppigy
14:41.53eppigyhuggles Katty
14:42.19Kattyeppigy: so did you have steak n potato for dinner last night
14:42.33eppigyyesh
14:42.39Kattydid you cook it yourself?
14:42.42eppigyyesh
14:42.45eppigyrare
14:42.45Katty:>>>
14:42.48eppigyRARE
14:42.49Kattyapplauds
14:42.52Kattymooing?
14:42.56eppigyyes
14:42.59Kattyk
14:42.59eppigymooing for mercy
14:43.08Kattyi like mine medium
14:43.13Kattyand not oozing red stuff
14:43.23AmorsenAh, I'm hitting issue 16374
14:43.26ManxPower-work<-- not a fan of eating dead animals
14:43.37Kattyoh nice, some person just went in my yard
14:43.43Kattyhe looks...suspiciously religious
14:43.43dlynes_laptopKatty, what does draining the juices have to do with grease?
14:44.00Kattydlynes_laptop: err
14:44.04Kattydlynes_laptop: when you cook ground beef
14:44.09Kattydlynes_laptop: all of the fat 'melts' into the pan
14:44.16Kattydlynes_laptop: and that's called grease
14:44.20dlynes_laptopKatty, they're talking about draining the juices out of canned tomatoes and the like
14:44.29Kattywhat?
14:44.31Kattylooks at recipe
14:44.34dlynes_laptopKatty, in your recipe
14:44.49Katty4 - do not drain cans
14:44.55dlynes_laptopKatty, exactly
14:45.09Kattycan you repeat the question
14:45.11Kattyrubs eyes
14:45.21dlynes_laptopKatty, now why would anyone drain the juices?  that sounds just plain weird
14:45.32Kattyit's a recipe
14:45.34[TK]D-FenderKatty: ... What fat is there to drain exactly?
14:45.38Kattysome people don't know how to cook
14:45.42*** join/#asterisk jaytee (~jforde@unaffiliated/jaytee)
14:45.48Kattyhi jaytee
14:45.52jayteehi Katty
14:45.53Katty[TK]D-Fender: your mom.
14:45.56Kattyhugs jaytee
14:46.06coppicedlynes_laptop: surely you don't want any liquid in your soup
14:46.09[TK]D-FenderKatty: Well?
14:46.13Katty[TK]D-Fender: Deep Subject
14:46.16dlynes_laptopcoppice, no...that would make no sense at all
14:46.19[TK]D-FenderKatty: Deep dish
14:46.24Katty[TK]D-Fender: pizza?
14:46.28Kattyperks up
14:46.28[TK]D-FenderKatty: So what fat are we talking about there?
14:46.37Katty[TK]D-Fender: we're talking about fat?
14:46.38[TK]D-FenderKatty: I fail to see how it applies in this case
14:46.42*** part/#asterisk ManxPower-work (~EWieling@216.186.151.147)
14:46.43Katty[TK]D-Fender: i thought we were talking about draining vegetables
14:46.48dlynes_laptopwell...besides
14:47.07dlynes_laptopif you buy extra lean ground beef, there shouldn't be much fat, anyways
14:47.08Kattydlynes_laptop: i would probably drain and rinse the kidney beans
14:47.14[TK]D-FenderKatty: You brought up drining the juices to cut grease
14:47.18[TK]D-Fenderdraining*
14:47.20Kattydlynes_laptop: just because the stuff they pack it in is pretty .......ick
14:47.36dlynes_laptopKatty, they pack it in water
14:47.47Kattyyeah but have you seen the water they're sitting in?
14:47.51Kattyit looks awful
14:47.56dlynes_laptopKatty, yeah..same water it was cooked in
14:48.00Kattyeww eww eww
14:48.15dlynes_laptopKatty, well, you're eating the kidney beans
14:48.25Kattyyeah but they look normal :P
14:48.26dlynes_laptopKatty, and it was cooked in that water
14:48.32Kattynot greyish brown liquid
14:48.48Kattythat just looks nasty
14:48.49dlynes_laptopKatty, oh...it's not greyish brown liquid here
14:48.53Kattyit is here
14:48.54*** join/#asterisk fofware (~chatzilla@host171.190-30-113.telecom.net.ar)
14:48.59dlynes_laptopit's a much paler color than the color of the beans
14:49.14Kattymaybe i'll try another brand then
14:50.27Kattycrock pot soup is the best.
14:50.41Kattyi loves it.
14:50.54eppigysleeple
14:51.03Kattyme too
14:51.04eppigyi want more caffeine
14:51.07eppigybut I shouldnt
14:51.24Kattywhy
14:51.42dlynes_laptopKatty, stokely van camp's is usually pretty good
14:51.44Kattyyou have the metabolism of a miniture greyhound
14:52.09Kattydlynes_laptop: k
14:52.29Kattydarn religious people in my yard
14:52.31Kattyscarin my critters
14:52.33dlynes_laptopKatty, the cheap brands are usually quite nasty
14:52.54Kattydlynes_laptop: yeah and they always put these crazy added ingredients in stuff
14:53.29KattyMajor General is awake!
14:54.27Kattyhe nests in the big tree across the street :>
14:55.31*** join/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com)
14:56.03*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
14:56.03*** mode/#asterisk [+o putnopvut] by ChanServ
14:56.23Kattyhi putnopvut
14:56.34putnopvutHi Katty. What's happening?
14:56.49Kattynot sure, but when i wake up i'll let ya know
14:57.22Kattyso far, i have swapped around backup harddrives
14:57.46putnopvutawesome
14:57.51putnopvutI'm headed to Brussels today!
14:58.52Kattytravel safely
14:59.01Kattyand don't let the crazies get you
15:01.45Kattymajor general is stuffin his chompers this morning
15:01.57Kattyp3nguin: idk if you heard, but they're callin for ice tonight
15:02.56Kattyp3nguin: and then snow this weekend
15:06.19*** join/#asterisk soman (~somnath@stargate.starnet.fi)
15:06.38dlynes_laptopIs it normal to have a channel id of '00000000'?
15:07.29dlynes_laptopeg.   SIP/navigata-gw-out-00000001 is making progress passing it to SIP/MorganSwitch-00000000
15:07.30*** join/#asterisk Cuz (~plastik@mail.gradeatechs.com)
15:07.48dlynes_laptopIn this case, both channel ids look abnormal to me
15:08.15dlynes_laptopIt always seems to be a random 8 digit hex number, but in this case, they seem to be sequential
15:09.38CuzCould anyone help me with a custom CDR problem I'm having? As I read it, all I need to do is set an accountcode= for extensions that i'd like to do custom outbound logging for, setting an "accountcode" (in my case i use "support" to designate the support queue), it makes a support.csv file in /var/log/asterisk/cdr-csv, for whatever reason my support.csv remains at 0 bytes consistently
15:10.08Cuzi mean the file gets created when i specify accountcode, but nothing seems to be getting logged to it, is there a switch i missed turning on in some conf file somewhere?
15:10.30dlynes_laptopCuz, it should be creating it in /var/log/asterisk/cdr-custom/Master.csv
15:10.38Cuzi'm trying to create these custom CDR files to parse out for a queue statistics generator
15:11.02Cuzdlynes_laptop: no, thats where things go if you don't specify "accountcode", Master.csv is the default CDR log file
15:11.23dlynes_laptopHrm....did asterisk 1.6.1.14 change everything over so that channel identifiers are now sequential?
15:11.36dlynes_laptopCuz, well, you were saying you were creating a custom cdr
15:11.50Cuzyes, which is "support"
15:11.50dlynes_laptopCuz, adding account code does not make the cdr custom
15:11.51Kattyyour mom's a custom cdr.
15:12.15Cuzdlynes_laptop: oh, how i read this, it sounded like it did
15:12.20[TK]D-FenderCuz: accoundcode is a FIELD in the amster.csv
15:12.25Cuz2. All outgoing calls are stored in CDR files. The files are placed in /var/log/asterisk/cdr-csv/ by default. The Queue Statistics application does not parse the Master.csv file, because in some cases it might be too large and heavy. That is why you have to create your own *.csv files for the groups you need. You may add 'accountcode' value for the accounts you use in sip.conf or iax.conf.
15:12.46dlynes_laptopCuz, accountcode and userfield are both normal fields....they just aren't normally in the csv file unless you use them
15:12.49[TK]D-FenderCuz: what "Queue Statistics application"?
15:12.56Cuzso I've done that, how do i populate the data?
15:13.03Cuz[TK]D-Fender: http://www.asteriskguru.com/tutorials/installation_guide.html
15:13.34[TK]D-FenderCuz: Oh.. you men 3rd party stuff we don't directly support!
15:13.36[TK]D-Fendermean*
15:13.49Cuzi'm not asking about support on 3rd party products
15:13.56Cuzi'm asking how to create a custom cdr file :)
15:14.06[TK]D-FenderCuz: Well accountcode is jsut a field in the main
15:14.39Cuzwell, why does it create a 0 byte <accountcodename>.csv when i give any extension an accountcode then?
15:14.54[TK]D-Fender'cuzThere is a CONF FILE for you to make a custom format for different purposes... but this is completely different
15:14.55Cuzin /var/log/asterisk/cdr-csv/
15:15.24dlynes_laptopCuz, what it's saying is that it'll still log to Master.csv, but when you use account codes, asterisk will automatically create an 'accountcode.csv' file for every account code used
15:15.32Cuzas i read it, i just need to create custom cdr files for all extensions i have in the queue for this thing to parse, so that's all I'm trying to do.
15:15.41dlynes_laptopCuz, they're not custom
15:15.55Cuzsorry, not custom
15:16.24Cuzjust so the data for the accounts specified either dumps logs into both Master and support.csv or just support.csv, is this possible?
15:16.42dlynes_laptopso say you've got a sip peer with an account code of 'joeblow', you'll get an additional file called 'joeblow.csv' in your cdr-csv directory
15:17.00Cuzi still don't follow
15:17.00dlynes_laptopall the calls for joeblow will end up in both joeblow.csv and Master.csv
15:17.11Cuzyes, that's what i want
15:17.25*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
15:17.27Cuzthat's not what's happening now
15:17.30dlynes_laptopAnd it does it that way so that your parsing engine only has to parse joeblow.csv
15:17.39Cuzthe "joeblow.csv" is staying at 0 bytes
15:17.52dlynes_laptopCuz, are your calls showing up in Master.csv for your accountcode?
15:18.12dlynes_laptopCuz, if not, it's because you haven't made any calls that were assigned to that accountcode yet
15:18.50dlynes_laptopCuz, the way you're currently doing it, it'll only log outbound calls, using the accountcode, not inbound
15:20.04Cuzoh no, i think there's a bigger problem here actually
15:20.20CuzI just realized Master.csv hasn't grown in size since November
15:20.23dlynes_laptopCuz, accountcode= only affects the sip user
15:20.27Cuzso that's probably the issue
15:20.33dlynes_laptopCuz, there ya go
15:20.55dlynes_laptopCuz, check to make sure cdr_csv.so is loaded
15:21.48TommyBottenIs it so, that there is no way to retain the original caller id on attended transfer?
15:22.03dlynes_laptopTommyBotten, correct
15:22.08Cuzbut the funny part is, November was when i started using these "accountcode" lines, and was when i originally started trying to get these queue stats to work
15:22.10dlynes_laptopTommyBotten, other than in the logs
15:22.28dlynes_laptopCuz, that's not the reason it stopped working
15:22.37dlynes_laptopCuz, you probably made other changes that affected it
15:22.39Cuzdlynes_laptop: how do i check to see if that module is loaded?
15:23.11*** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net)
15:23.20dlynes_laptopCuz, module show
15:23.44Cuzew 185 modules
15:23.46dlynes_laptopCuz, anything that's listed is loaded
15:23.47Cuzhow do i grep it? :P
15:24.02dlynes_laptop/usr/sbin/asterisk -rx "module show" | grep cdr_csv
15:24.11Cuzoh yah, that would work
15:25.38*** join/#asterisk zorp75ck (~zorp75ck@pool-209-158-23-39.altnpa.east.verizon.net)
15:26.58*** join/#asterisk NicoleMun (~ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net)
15:27.27Cuzcdr_csv.so Comma Separated Values CDR Backend 0
15:27.32Cuzokay, so it's loaded
15:27.41Cuzwhat else would cause the CDR's to stop?
15:28.43Kattythinks about making a comment
15:29.55Cuzwhoa
15:29.58Cuzokay
15:30.05Cuzso i'm missing the cdr.conf file in /etc/asterisk
15:30.10Cuzthough cdr.conf.sample is there
15:30.21Cuzthat ain't normal, is it? :P
15:30.44dlynes_laptopCuz, that ain't normal, cuz you're not normal :)
15:31.05dlynes_laptopCuz, rename the cdr.conf.sample to cdr.conf and use a text editor and configure it
15:31.06Cuzis it, or isn't it?
15:31.17Cuzyea, and do a reload? or an amportal restart?
15:31.58Kattyyou do the hokeypokey
15:32.10Kattycause that's what it's all about
15:32.20Cuzso helpful
15:32.24Kattyi know, aren't i?
15:32.32Cuzrolls his eyes
15:32.33Kattyhave a nice day.
15:32.52dlynes_laptopCuz, ummmm
15:32.56dlynes_laptop~gui
15:32.57infobotgui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png.  Of course Real Programmers use the command line interface.  See cli
15:33.00dlynes_laptop~freepbx
15:33.01infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
15:33.17dlynes_laptopCuz, freepbx has all kinds of issues...
15:33.19Cuzyea?
15:33.21Kattydlynes_laptop: the hokeypokey!
15:33.24Kattydlynes_laptop: come on then!
15:33.36Kattyalso, my workstation keeps rebooting
15:33.37Kattyand irritating me
15:33.41Kattyand it is not the power supply
15:33.44Kattygrumps, throws things
15:33.54Cuzwhat about freepbx?
15:33.58Cuzi never said i was using freepbx
15:34.03Cuzwhy would you make that assumption?
15:34.13Kattyare you denying it?
15:34.21Gido-E:-)
15:34.26Cuzsure
15:34.45Kattythat is not an appropriate response.
15:34.50Kattyan appopriate response is Yes or No
15:34.58*** join/#asterisk basty (~basty@212.218.65.131)
15:35.00bastyhi
15:35.00Cuzdoes it matter?
15:35.12Kattybasty: hi
15:35.13[TK]D-Fender[10:31]<Cuz>yea, and do a reload? or an amportal restart? <--- AMPORTAL = FreePBX
15:35.24Cuzkoay fine
15:35.28[TK]D-FenderCuz: Don't play us for dumb
15:35.40*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
15:35.42Kattyplays [TK]D-Fender for dumn
15:35.43bastyis there any way to get app_rxfax.c and app_txfax.c compiled on asterisk 1.4.29 ? We moved from 1.2.X to the latest 1.4. And we used that app to recive fax with spandsp.
15:35.44Kattyoh
15:35.46Cuzso do i do a /etc/rc.d/init.d/asterisk restart?
15:35.51Kattyplays [TK]D-Fender for DUN DUN DUN DUN
15:35.54Cuz...
15:35.56Kattyhi tony
15:36.05[TK]D-Fendercurestart * however appropriate to how you're running your install
15:36.11[TK]D-FenderCuz: restart * however appropriate to how you're running your install
15:36.56Cuzi know how borked freepbx is, i still have to use it, because i can't afford to train 3 other guys on how to use asterisk console
15:37.27[TK]D-FenderCuz: Less story, more action
15:38.28*** join/#asterisk guax (~guax@unaffiliated/guaxinim)
15:38.55Cuzdoes the CDR module really require the .conf file to be sitting in /etc/asterisk though? i mean most of the lines in that file are commented out anyways
15:39.02Cuzin the sample
15:39.14Cuzand the defaults are fine for what i want to do
15:39.24guaxjust remove and test it
15:39.55Kattyguax: he's using freepbx. there's no telling what his system is doing
15:39.56Cuzi'm trying
15:40.00Cuzi run 23 lines off this box
15:40.04[TK]D-FenderCuz: You want custom CDR then you have to DEFINE IT
15:40.05Cuzi need to wait till everyone's done their calls
15:40.14guaxyou must have a backend, so prehaps the conf file just need to say that cdr is active, the rest stays on cdr_odbc and so on
15:40.21guaxKatty, oh
15:40.38Cuz[TK]D-Fender: i don't think i want a "custom cdr" per-se, i just want to take the data that normally gets logged in Master.csv and put it in support.csv
15:40.39Cuzor both
15:40.58[TK]D-FenderCuz: Symlink it then.
15:41.17Cuzno no, i mean for specific extensions with the accountcode "support"
15:41.30guaxno support for that on cdr tough
15:42.58Cuzwhat do you mean?
15:43.41guaxdo you want to keep the rest of data on Master.csv and just record for suport?
15:43.49guaxsupport*
15:43.52Cuzyes
15:44.14Cuzmy problem right now is though nothing is going to either log file
15:44.21guaxa pipe on cron can help you but i dont see native support in asterisk for that
15:45.15[TK]D-FenderCuz: I see no mechanism for * to split  where it saves CDR records to.  Custom CDR only selects the line formatting, not the selection criteria
15:45.45[TK]D-FenderCuz: perhaps you could make a script to filter the master into sub-files.
15:45.55Cuzwhy does it create a new .csv file when an accountcode is specified for an extension then?
15:45.58[TK]D-FenderCuz: You'd have to trigger this on some regular basis
15:46.13russellbor use a database, and a simple SELECT will get you only the data you want.
15:46.54russellbi would much prefer that over text file mangling, personally.
15:46.59guaxcat Master.csv | grep "support" > support.csv on crontab :)
15:47.23guaxrussellb, then you loose the style :D
15:47.29russellbsends a bunch of calls with callerid name of "support"
15:47.31[TK]D-Fenderrussellb: I'mt hinking his 3rd party reporting tool isn't flexible for that
15:47.42Cuzi understand what you guys are saying
15:47.55Cuz[TK]D-Fender: do you use queues?
15:47.59[TK]D-FenderCuz: Yes
15:48.04russellbYou know, I wrote a CDR module, cdr_python a long time ago that would have been nice for this
15:48.08Cuzdo you generate statistics for said queues?
15:48.14russellbit just gave you CDRs as a python object that you could do whatever with
15:48.16KattyFOR SAID QUEUES
15:48.18[TK]D-FenderCuz: No, ASTERISk generates them :)
15:48.20Kattysaid queues.
15:48.26KattyQs.
15:48.30russellbit would be a 2 line script ... if accountcode == "support": writeitoverhere...
15:48.32KattyFor, Said Qs.
15:48.44russellbshould get that merged.
15:48.46guaxrussellb, you are cheating, the suggested kludge works here, hehe.
15:48.50Cuzyes, i understand how to manipulate text data on a cronjob
15:48.55KattyKludge?
15:48.55Cuzi'm pretty good with awk, sed, grep
15:49.02Kattyguax: define Kludge.
15:49.06[TK]D-Fenderthat's what she sed :p
15:49.12Cuzi'm wondering if asterisk has any build in mechanisms to accomplish my goal
15:49.23[TK]D-FenderCuz: Clearly looking like "no".
15:49.24guaxKatty, http://en.wikipedia.org/wiki/Kludge
15:49.39Cuz[TK]D-Fender: asterisk does not generate anything usable, such as pie charts/graphs/etc for the suits
15:49.41[TK]D-FenderCuz: Shitty CDR tool to use CSV where every other tool uses SQL
15:49.47n3hxsKludge... a software Rube Goldberg?
15:49.48Cuzi mean the raw data, yes
15:49.58Cuz[TK]D-Fender: okay
15:50.05Cuzso maybe i started out with the wrong question
15:50.06[TK]D-FenderCuz: * isn't a GUi reporting tool, its a bunch of engine parts <-
15:50.16[TK]D-FenderCuz: Get out of your Happy-Happy GUI FreePBX world.
15:50.22Kattyguax: ohisee, i will have to add that to my fun list of words.
15:50.24[TK]D-FenderCuz: You don't seem to realize where you are and what * is
15:50.25Cuzwhat would you guys use to generate queue statistics that you can give to suits so they can go oooooh ahhh
15:50.50[TK]D-FenderCuz: Your 3rd party tool was based on a dumb idea.  Complain to them.  My tools work GREAT
15:51.03Cuzright, what tool are you using then?
15:51.13[TK]D-FenderCuz: ScopServ
15:51.15Cuzthis should have been my question to begin with
15:51.20guaxKatty, thats all the theory behind it: klumsy, lame, ugly, dumb, but good enough
15:51.32Cuzoh if Katty's trying to talk to me, sorry...
15:51.34*** join/#asterisk ManxPower-work (~EWieling@216.186.151.147)
15:51.38Cuzcan't see
15:51.45[TK]D-FenderCuz: No, asking to find a way to use what you have better is a good start.  Realizing that yours is a DEAD END is a good thing too.
15:51.50Kattyguax: there's a similiar word.
15:51.55Cuzyep
15:51.56Kattyguax: MonkeyRigging
15:52.00Kattyguax: but it has racist undertones
15:52.02Kattyguax: :<<<
15:52.05Cuzi hated this piece of crap the last time i tried to set it up
15:52.27Cuzi found a good commericalware package, but the boss wants me to test out free ones first
15:52.29*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)
15:52.34[TK]D-Fendersets the PR team for "Primate Equality" on Katty
15:53.05KattyWhat the razzleberries are you talking about
15:53.16ManxPower-workWe're here!  We're primates!  Get used to it!
15:53.26*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:53.26*** mode/#asterisk [+o leifmadsen] by ChanServ
15:53.38Kattyhji mister blitzrage
15:53.40[TK]D-Fendermonkey see, monkey doo-doo... flungat your face!
15:53.41Katty-j.
15:53.51leifmadsenKatty: yo!
15:53.57[TK]D-Fenderleifmadsen: I don't want to meet your mom!
15:54.01leifmadsenI just want...
15:54.03[TK]D-Fenderleifmadsen: ! ! !
15:54.05[TK]D-Fender\o/
15:54.08Kattyleifmadsen: cougar
15:54.09leifmadsenI sure do :)
15:54.10Kattyoh
15:54.12Kattywrong person
15:54.14Katty[TK]D-Fender: cougar
15:54.15leifmadsenhaha
15:54.16Katty[TK]D-Fender: bate
15:54.22Katty[TK]D-Fender: in soviet russia...
15:54.30[TK]D-Fender"bait" <- and yes.... this was my previous demographic
15:54.40Cuz[TK]D-Fender: i have this working right now, and it's SQL based
15:54.41Cuzhttp://www.asternic.biz/
15:54.46Cuzbut it costs $$$
15:54.51Kattyoh noes, it costs $$$
15:54.57KattyEND OF THE WORLD
15:55.09Kattyhow dare my lunch cost money
15:55.15Kattyit is unacceptable
15:55.16[TK]D-FenderCuz: Don't go expecting that you are going to find a free tool you will like for every business goal you have <-
15:55.28Kattyi demand lunch for free, on a silver platter, with everything i want.
15:55.33Cuzno, i'm 100% aware of this
15:55.39KattyNO SUBSTITUTES ACCEPTABLE
15:55.40[TK]D-FenderCuz: Life sucks, but rarely swallows
15:55.45ManxPower-workTelecom wants ot be free!
15:55.57KattyManxPower-work: freeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee
15:55.59KattyManxPower-work: as a bird
15:56.07*** part/#asterisk benngard (~benngard@213.88.138.230)
15:56.14Cuzso do you know the name of any 1/2 decent queue stats parser i can look into, since this asteriskguru one is complete ass
15:56.37Cuzhave this one going too
15:56.37Cuzhttp://www.orderlyq.com/
15:56.44Cuzbut again, it's commerical
15:56.50Gido-ECuz :-)
15:56.51Kattyi know one.
15:56.59Kattyit's called WriteYerOwn, Inc.
15:57.01ManxPower-workMost queue and cdr stat applications will cost money.
15:57.24Kattydoes asterisk-stat have queue stuffs
15:57.33Kattydon't think it does
15:57.45Gido-EAlmost the latest Linux traffic shaping patches ready. In a short while you can see traffic per rule in the traffic shaper.
15:58.09CuzManxPower-work: Yea, that's becomming more and more obvious the harder i look
15:58.38Cuzbut my question is, is anyone in here generating queue stats with any free software they could recommend to me :P
15:58.50ManxPower-workCuz: Most people seem to write their own.
15:59.05Cuzuh yea
15:59.09*** join/#asterisk Deeewayne (~dwayne@75.76.254.162)
15:59.10*** mode/#asterisk [+o Deeewayne] by ChanServ
15:59.10Gido-ECuz push it to snmp? :-)
15:59.16Cuzlol
16:02.02*** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com)
16:02.16Kattyhugs Deeewayne
16:02.17*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
16:02.29KattyDeeewayne: youj're missing some awesome squirrely action this morning
16:02.37DeeewayneO.O
16:02.41DeeewayneKatty, good morning!
16:02.47Deeewaynechecks the squirrels
16:04.13GeminizerHello all.  Is there an exclusion operator that can be used with trunk settings (e.g. basically allow any number of the form NXXNXXXXXX except 110NXXXXXX)?
16:04.48[TK]D-FenderGeminizer: No.
16:05.11Gido-EGeminizer but asterisk wil do the exact match for the other one
16:05.20[TK]D-FenderGeminizer: Check the # in that extension, or make another exten with the other pattern so that it catches it instead
16:05.20Gido-EYou can add a noop or hangup
16:05.22DeeewayneKatty, awwww
16:07.04Gido-EDeeewayne it is also 5pm here end of the normal workday
16:07.22Gido-Es/ also//
16:07.37DeeewayneGido-E, yay
16:07.37KattyDeeewayne: much face stuffing going on.
16:10.47Geminizerthanks
16:11.05*** join/#asterisk Alfio (~amunoz@75.112.88.200.m.sta.codetel.net.do)
16:13.05*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
16:13.23*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
16:14.49*** join/#asterisk IBC_JKENNEY (~jkenney@ip65-44-169-66.z169-44-65.customer.algx.net)
16:15.40IBC_JKENNEYHey has anyone had any issues where you cannot dial a specific number.  We are using asterisk with PRI cards on XO IP-FLEX service
16:15.51IBC_JKENNEYthey say we are using non-standard signaling
16:16.09IBC_JKENNEYhas anyone else had this problem
16:16.16ManxPower-workIBC_JKENNEY: That non-standard signalling might be?
16:16.34ManxPower-workIBC_JKENNEY: you understand that PRIs don't play busy messages or number disconnected messages, right?
16:16.49IBC_JKENNEYYes
16:17.00*** join/#asterisk atis_work (~atis_work@193.238.212.171)
16:17.01IBC_JKENNEYAnd i get a message
16:17.02ManxPower-workIBC_JKENNEY: what is the HANGUPCAUSE of the failed call?
16:17.09IBC_JKENNEYthe call doesn't fail
16:17.17IBC_JKENNEYthe call is a ATT conference bridge
16:17.34IBC_JKENNEYwhen i call from traditional pots (outside XO) or my cell it works fine
16:17.54ManxPower-workSo you CAN call that number.
16:17.55*** join/#asterisk moy (~moy@74.12.123.169)
16:18.00IBC_JKENNEYif i call from IP-flex it rings 5 or six times and states that it could not access the bridge
16:18.09IBC_JKENNEYYes but not from my asterisk pbx
16:18.11IBC_JKENNEYor PRI's
16:18.18IBC_JKENNEYon the IP Flex
16:18.37IBC_JKENNEYwe have another PRI that is broke out traditionally that works fine
16:18.45ManxPower-workIBC_JKENNEY: IF it's a PRI we don't care what the underlying transport is (except for cases of fax and modem)
16:18.46IBC_JKENNEYfor our modem pools so i stole a line from there and it works
16:18.57ManxPower-workMaybe you are sending out bad callerid?
16:19.17IBC_JKENNEYI thought of that however XO has us set if our callerid is invalid or wrong we can't call out
16:19.32ManxPower-workIBC_JKENNEY: then I have no further suggestions
16:19.37[TK]D-FenderIBC_JKENNEY: got debug to show us?
16:19.40thansenanyone have experience with nortel NTDU91 and asterisk (sip)? or nortel in general?
16:19.48n3hxs[away]is now away - Reason : Auto-Away after 30 minutes
16:19.51[TK]D-FenderIBC_JKENNEY: because so far you haven't given us any details.
16:20.40IBC_JKENNEYnot for today i am running testing tonight once business closes and call traffic is slower
16:21.08IBC_JKENNEYhowever when XO plugged the PRI into a test set the call worked but they plugged it in before it went into the cisco router
16:21.55IBC_JKENNEYso there are 3 T1's that plug into the cisco and 2 of them plug into the asterisk pbx on a digium PRI card
16:21.56*** join/#asterisk wcselby (~wcselby@216.110.88.194)
16:21.56Kattyconsiders some lunch
16:21.59wcselbyo/
16:22.04Kattyhi wcselby
16:22.05IBC_JKENNEYafter the router
16:22.14wcselbyhowdy
16:22.21*** part/#asterisk guax (~guax@unaffiliated/guaxinim)
16:22.49IBC_JKENNEYI just wanted to know if asterisk could even be the kulprit.  Since if i was having problems wouldn't i have problems on all numbers.
16:23.46c0rnoTathansen: asterisk has specific channel for nortel phones
16:24.46*** join/#asterisk cuco (~cuco@local.xorcom.com)
16:24.47c0rnoTathansen: chan_unistim
16:24.57c0rnoTause google for more information :))
16:25.35thansenc0rnoTa: ok, I am...this particular model looks like it has sip.  I'm just looking for general attitude toward nortel phones though
16:25.54*** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi)
16:26.19*** part/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com)
16:26.32c0rnoTai have seen in real life  that Nortel phones works with asterisk :)
16:26.56Kattyhmm. lunch.
16:27.00Kattysuggestions? recommendations?
16:27.27c0rnoTaTea + cake
16:27.44*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
16:28.00Kattyhow about something with protein
16:28.04wcselbypizza buffet
16:28.08wcselbychinese buffet
16:28.18wcselbythai food
16:28.29c0rnoTahey, it's just a lunch
16:28.30wcselby^^ all the places around today's client
16:28.42c0rnoTawcselby: :))
16:28.43wcselbyooh, and tex-mex
16:28.55Kattyokay how about something HEALTHY
16:28.57Kattywith protein
16:29.04Kattywhich totally cuts out buffets >.<
16:29.15wcselbyi think there's a salad bar nearby in the grocery stor
16:29.18wcselbystore*
16:29.26wcselbywhich is sort of like a buffet :P
16:29.47[TK]D-FenderKatty: Shish-taouk
16:29.49wcselbyso you didn't bring anything with you today?
16:29.55[TK]D-FenderKatty: clearly time for Lebanese
16:30.34c0rnoTaO_o it's just a lunch, guys. Eat something and go for a work
16:30.38c0rnoTa:)))))
16:30.51Kattyooooh, salad
16:30.53KattyOoOOooh
16:30.56Kattyplots
16:31.44[TK]D-FenderKatty: Lots of great little Lebanese places here... $10 feast...
16:31.46*** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com)
16:32.34c0rnoTaand i'm waiting for a dinner
16:32.52c0rnoTait's 7:32PM
16:33.16Kattywaiting for a dinner?
16:33.30c0rnoTayeah..
16:33.31Kattyyou mean you're waiting for it to get done cooking?
16:33.51Kattywhat are you cooking?
16:34.16*** join/#asterisk lanning (~lanning@208.87.235.224)
16:34.17c0rnoTano. just wait for a time, when i'll be at home and eat something :)
16:34.39Kattyahh
16:35.59c0rnoTaall what i'm cooking now is just a SESSION cookie
16:38.46c0rnoTaSo, can anybody now answer on my question? :) 7 hours ago there was no answer couse you all sleeped
16:38.57c0rnoTaan anybody tell me how i can count channels on specific extension at moment. For example, i have extension _9XXXXXXX and i want to know how much channels now are on that extension?
16:38.59wcselbywhat was teh question?
16:39.25c0rnoTa'core show channels' show extens like 91234567 or 97654321 not "_9XXXXXXX". And i couldn't select which of them use "_9XXXXXXX" and which "_97.". Only using analysing process in my mind i could divide it. So, i want to divide it automaticly in AGI script.
16:39.40c0rnoTaspeaking in global of my task, i want to limit abilities of outgoing calls through trunks on each exten.
16:40.17wcselbyuse COUNT()?
16:41.20wcselbysorry, I think I mean GROUP
16:41.42*** join/#asterisk cod3hax0r (~Boyperwis@121.97.56.48)
16:41.45cod3hax0rhi guys
16:41.45wcselbyman
16:41.48wcselbymy brain isn't work yet
16:41.50cod3hax0rgood day to all
16:42.00wcselbyi'll look c0rnoTa, I think I know what you need
16:42.27cod3hax0ri need help...  im getting SIP/2.0 401 Unauthorized
16:42.35[TK]D-Fenderc0rnoTa: "core show function GROUP_COUNT"
16:42.36c0rnoTawcselby: so, i can set group in each channel dialed throu exten, then group_count before dial this exten again, right?
16:43.28c0rnoTa[TK]D-Fender, wcselby! Thanks, guys!
16:43.31*** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
16:43.39c0rnoTait should work :)
16:43.51wcselbyc0rnoTa, yeah, create a process that first checks if the group_count is below your threshhold, if it is, allow the call and add to the group, then if it's above your threshhold, deny the call.
16:44.10c0rnoTawcselby: thanks a lot :))
16:44.28c0rnoTaat the end of the day, i'v found an anser :)
16:44.31c0rnoTaanswer
16:45.12c0rnoTaand i have no ideas before, either global vars :)
16:46.50*** join/#asterisk paulc (~paulc@unaffiliated/paulc)
16:47.02Kattystares at clock
16:47.21ecranehttp://www.theregister.co.uk/2010/02/03/voip_hacker_guilty/
16:49.08eppigystares at Katty
16:49.38*** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001)
16:50.05*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)
16:52.41*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
16:58.29wcselbyi love how the guy made millions, and only paid his accomplice 20,000
16:58.57wcselbyreferring to ecrane's register article
16:59.32*** join/#asterisk soman (~somnath@stargate.starnet.fi)
17:02.46*** join/#asterisk comradeb14ck (~comradeb1@72.37.252.50)
17:03.51*** join/#asterisk kalib (~lkhlui@osiris.aspec.com.br)
17:04.28kalibWhat does it means when I receive in my CLI: "Remote UNIX connection | Remote UNIX disconnected
17:04.47kalib?
17:04.57Chainsawkalib: It means a command was run using asterisk -rx (in most circumstances).
17:05.10kaliboh.. thanks ;]
17:05.15Chainsawkalib: So if you have bash scripts or other magic running in the background, that's probably it :)
17:06.22*** join/#asterisk tbenson (~tbenson@c-67-174-228-93.hsd1.ca.comcast.net)
17:07.08*** join/#asterisk corretico (~laguilar@201.201.46.106)
17:09.19tbensonIs there a simple way to spawn a new channel ID from within the h extension of a context?  I am attempting to dial out after a call hangs up.  I had been using a call file created from within the dialplan but am not sure how to identify the channel asterisk creates for the call file so i can examine the CHANNELSTATUS to handle exceptions.
17:11.41*** join/#asterisk jameswf (~james@unaffiliated/jameswf-home)
17:12.28ManxPower-worktbenson: you should not do dialing out in your 'h' exten.
17:12.47voipmonkheheh
17:12.48ManxPower-worktbenson: if you want to post process after the callEE has hung up see the "g" option to dial.
17:13.12ManxPower-workThere's no point in dialing after the callER hangs up.
17:14.02*** join/#asterisk yahh (~root@122.169.77.74)
17:15.08yahhhello guys..
17:15.13yahhi have a query
17:15.56comradeb14cksup
17:16.13Corydon76-dig~ask
17:16.14infobotQuestions 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.
17:17.01yahhi am using 2 - quad processor
17:17.06yahhasterisk is running on it
17:17.10blackok
17:17.26yahhright now cpu usage is going upto 60%
17:17.35blackok
17:17.38yahhfor asterisk
17:17.39*** join/#asterisk bklang (~bklang@tesla.alkaloid.net)
17:17.41c0rnoTait's ok
17:17.53c0rnoTaflood of qualify messages ?
17:17.55yahhso can i go to up to 800% as it is 2 - quad
17:18.07blackyahh, what operating system are you using?
17:18.14yahhcentos
17:18.18black32 or 64 bit?
17:18.22yahh64
17:18.40blackyou won't get 800% cpu, asterisk doesnt support multiple cores (afaik)
17:18.56p3nguinWas there a question involved in this story?
17:19.44yahhohh so max i can go upto 100% only is it?
17:20.36Corydon76-digblack: Uh, it doesn't?
17:20.43c0rnoTablack: is it only on 64bit systems
17:20.43c0rnoTa?
17:20.48cod3hax0rhello
17:20.59cod3hax0rcan anyone help me with sip unauthorized errors?
17:21.00p3nguincorydon76-dig: Is asterisk a multi-threaded application?
17:21.14Corydon76-digp3nguin: Yes, it most certainly is
17:21.31p3nguinThen I guess he can have 800% CPU usage!
17:21.51voipmonkuse the right password/username combination cod3hax0r  :)  - you have some debug to show ( use pastebin.ca )
17:22.09yahhblack: pls give your advise
17:22.13yahhi mean input
17:22.20blackCorydon76-dig knows way more than I do.
17:22.23blackHe is a developer.
17:22.36p3nguinI used to run it on a quad core, but I never remember checking threads.  I since moved to a single core, since I have such a low load -- no since in wasting a quad core system.
17:22.38blackI didn't think that asterisk ran on multiple cores, but I could be wrong.
17:22.50blackAlso, just because somethig is multithreaded doesn't mean it will run on multiple cores.
17:22.59Corydon76-digThere is one place that I can recall where Asterisk won't use more than one core:  a fairly old version of FreeBSD, but I'm fairly certain they've fixed that since
17:23.26yahhokay
17:23.36p3nguinMaybe that FreeBSD didn't have an SMP kernel.  :)
17:23.51Corydon76-digBut that's a limitation of the OS kernel, not Asterisk
17:24.04Qwellblack: it's a multi-threaded application..  why wouldn't it?
17:24.06c0rnoTacod3hax0r: what's your problem?
17:24.15Corydon76-digp3nguin: No, it was that FreeBSD did not allow the threads of a single process to execute on more than one core concurrently
17:24.35Corydon76-digp3nguin: you could run multiple processes concurrently, just not the threads of a single process
17:24.46p3nguinThat's kind of an interesting feature(?) to have, I guess.
17:24.47Qwellblack: sounds like nonsense your former employer fed people. :)
17:25.02blackQwell, I said I didn't know.
17:25.04blackNobody told me this.
17:25.16Qwellfair enough ;)
17:25.20blackAwesome if it does though, good work.
17:25.25*** join/#asterisk ch1nawhyte (~dliu@aker.tobi-office.com)
17:25.27Qwellbut yeah, any app that has multiple threads will run on multiple cores
17:25.42yahhcorrect
17:25.45Corydon76-digblack: Thank the kernel developers.  Wasn't anything we did, other than to use proper mutexes
17:25.48Qwellif it didn't, we wouldn't need any locking
17:25.48yahhmake sence
17:26.11Corydon76-digQwell: that's not completely true, either
17:26.24QwellCorydon76-dig: generalization
17:26.36Corydon76-digWe wouldn't need locking if the kernel used cooperative multitasking, instead of preemptive multitasking
17:27.01cod3hax0r<c0rnoTa> cod3hax0r: what's your problem? --> im getting SIP 2.0/unathorized when i do sip debug
17:27.28c0rnoTabut you sent correct username/secret is it?
17:27.40yahhso i think summary is... asterisk can go upto 800% for 2 quad
17:27.41c0rnoTacod3hax0r: but you sent correct username/secret is it?
17:27.44yahhright?
17:28.01Corydon76-digyahh: I think you mean 8.0 load average, but yet
17:28.04Corydon76-digyes
17:28.07yahhyes
17:28.12c0rnoTayes
17:28.49bmoraca_work8.0 load average?  that's ridiculously high
17:29.04cod3hax0r<c0rnoTa> cod3hax0r: but you sent correct username/secret is it? --> yes sir did
17:29.08bmoraca_workthat's 8 processes waiting every second for a chance to run their work...that's too much
17:29.09*** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844)
17:29.11c0rnoTacod3hax0r: try to set pedantic=yes in general section of sip.conf.
17:29.15yahhif i look top.,  there is %CPU for asterisk
17:29.18Corydon76-digbmoraca_work: that's maximum for 8 cores, without delaying any thread execution
17:29.30yahhthat could reach upto 800% , isn't it
17:29.48bmoraca_workCorydon76-dig: ahhh...8 cores.  that's somewhat ok, i suppose.
17:30.24cod3hax0r<c0rnoTa> cod3hax0r: try to set pedantic=yes in general section of sip.conf. --> just set it right now
17:30.36Corydon76-digbmoraca_work: read up.  :-)
17:30.47cod3hax0r<c0rnoTa> cod3hax0r: try to set pedantic=yes in general section of sip.conf. --> whats next?
17:31.06c0rnoTacod3hax0r: sip reload in cli and go on
17:31.10c0rnoTatry to auth
17:31.23bmoraca_workCorydon76-dig: yeah, i suppose coming in to the middle of a conv. isn't exactly enlightening :P
17:31.46p3nguinWhy would asterisk cause an 8.0 load average?  If asterisk is using 800% CPU and you have 100 processes waiting to use the processor, the load average is going to be more than 8.0.
17:32.04yahhCorydon76-dig: is there any difference for 8.0 load and %CPU for asterisk process
17:32.06yahh?
17:32.12c0rnoTacod3hax0r:  either i have no fast idea what it could be. need to see sip debug and look on SIP pockets exchange
17:32.27yahhit is similar to the 800% cpu right?
17:32.34bmoraca_workyahh: load average is impossible to equate directly to % cpu usage
17:32.35Corydon76-digyahh: Yeah, one is an average over a period of time, and the other is a snapshot in time
17:33.15Corydon76-digWell, % cpu usage is kind of a misnomer.
17:33.22tzafrir_laptopyahh, yeah, there is
17:33.57tzafrir_laptopalso, %cpu is for a shorter period. load avarage is for the last minute
17:33.58MadkissI have a problem with asterisk 1.6. Somebody is coming inside from an official IP, and actually I see the packets flying in, but: asterisk doesn't care at all.
17:34.01*** join/#asterisk albertoandrade (~albertoan@189.58.23.130.dynamic.adsl.gvt.net.br)
17:34.28Corydon76-digCPUs are only either in use or not... You can't really use 50% of a CPU...
17:35.01yahhhmm..
17:35.07p3nguinYou can certainly use the CPU 50% of the time, though.
17:35.08*** join/#asterisk gelpg (~chatzilla@94-21-99-19.pool.digikabel.hu)
17:35.11Madkissasterisk definetely listens on the corresponding port.
17:35.45yahhokay.. so in other words
17:36.11yahhasterisk can use 800% time with 2 quad
17:36.24bmoraca_workMadkiss: perhaps if you had a debug log of the failed call...maybe we could help you.  but from your description of the problem, there's nothing we can do, because the description of the problem doesn't make any sense
17:36.35cod3hax0rstill the same
17:36.49mort_gibMadkiss: /etc/init.d/iptables stop
17:36.56Madkissbmoraca_work: it's not about the call. it's about registering. registering fails.
17:37.05Madkissmort_gib: iptables it not even installed on this machine.
17:37.11bmoraca_workMadkiss: ok, so show a failed registration debug.
17:37.14p3nguinYou need not register to make a call.
17:37.37mort_gibMadkiss: 99% of those probs are related to a firewall....
17:37.44MadkissSIP/2.0 401 Unauthorized
17:38.02Madkisshu.
17:38.08bmoraca_workMadkiss: that's not a debug.  pastebin an ENTIRE debug from the failed registration.
17:38.16yahhasterisk can use 800% time with 2 quad (if we assume that no other porcess is not running)if am not wrong
17:38.31Warp4anyone have any luck doing any php scripting to communicate with the AMI?
17:38.31yahh?
17:38.51bmoraca_workyahh: it can use 100% of each core.  that's not 800%, though.
17:39.10mort_gibMadkiss: Then you might want to rephrase the "But asterisk dosen't care at all"
17:39.10p3nguin8 cores, some "monitor" apps will report 800%.
17:39.30mort_gibMadkiss: Asterisk cares, that phone/user is just not authenticated....
17:39.35gelpghi
17:39.37gelpgI need some help to install the asterisk with misdn driver
17:39.42gelpgi already installed the mISDN and mISDNuser drivers and tools
17:39.44yahhokay
17:39.50gelpgthe misdn_info shows the card (Digium B410P)
17:39.55gelpgbut I cannot configure the asterisk to enable the misdn channel
17:40.05mort_gibgelpg: Why misdn??
17:40.19gelpgbecause with dahdi it doesn't work
17:40.19*** part/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228)
17:40.28*** join/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228)
17:40.35gelpgit says the bri signalling doesn't supported
17:40.36mort_gibgelpg: Not correct
17:40.44c0rnoTacod3hax0r: use sip debug to find answer
17:40.56mort_gibgelpg: That's different, you need to install libpri
17:41.05mort_gibbefore dahdi
17:41.19gelpgI already did it
17:41.44mort_gibOk, I have it running with 2 different clients....
17:41.52mort_gibSo
17:41.56MadkissVia: SIP/2.0/UDP 192.168.92.55:2 <= can somebody tell me where that 192-IP is actually coming from?
17:42.25mort_gibpoint-to-multipoint
17:42.32cod3hax0rstill getting unauthorized using sip debug
17:42.40cod3hax0rim behind a cisco 800 router
17:43.33bmoraca_workMadkiss: it's probably the internal IP address of the device trying to register with you, BUT WITHOUT A FULL SIP DEBUG, IT IS IMPOSSIBLE TO SAY.
17:44.03gelpgmort_gib: yes, that's what I need
17:44.14Madkissbmoraca_work: take it easy on me. I am handling two totally crazy chicken right now yelling at me because they can not phone, thinking they are way smarter than me, I have more than enough shit to deal with ... ;)
17:44.36mort_gibMadkiss: He is right tho
17:44.43MadkissI know he is
17:44.54mort_gib:-)
17:44.55bmoraca_workMadkiss: that's fine, but until you provide a FULL SIP debug of the failed registration, there is no possible way I can help you
17:45.32mort_gibMadkiss: Check very CAREFULLY usernames and passwords
17:46.21Madkisshttp://paste.debian.net/58720/
17:46.23Madkissthere you go.
17:46.29ManxPower-workcod3hax0r: you'll always get unauthorized, then the client will retry with auth info
17:46.44bmoraca_workMadkiss: I suspect that the issue is one of three things:  1) There is an ALG on the user's NAT router that is improperly configured to work with asterisk and the phone; 2) Asterisk is not set to look for a natted phone; 3) the phone is configured to look for asterisk as if it were not a NAT.
17:46.44mort_gibgelpg: So, latest version of Dahdi, and install libpri BEFORE dahi
17:46.49bmoraca_workMadkiss: what model is the phone and the user's firewall?
17:48.19Madkisshold on for a second.
17:49.48bmoraca_workMadkiss: additionally, you did not show Asterisk's failure.  you didn't give me the entire debug and you obfuscated certain parts that are pretty useful in debugging this.
17:51.13bmoraca_workMadkiss: other things which might be helpful would be the sip peer definition from sip.conf (with ONLY the secret obfuscated)
17:54.06cod3hax0rits a funny thing because i can register to another sip provider but not to my asterisk server
17:54.30*** join/#asterisk xpot-mobile (~xpot@173-14-232-121-Utah.hfc.comcastbusiness.net)
17:55.28*** join/#asterisk mnick86 (~mnick86@95-90-246-116-dynip.superkabel.de)
17:56.50*** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net)
17:57.00wcselbyMadkiss - what kind of phones, what's the setup like?
17:57.39ManxPower-workMaybe Madkiss should come back when he has time to concentrate on the problem he is having.
17:57.49wcselbyi.e - phones at people's homes behind home nat / routers, asterisk at work behind firewall or on a public IP, etc?
17:58.11wcselbyi may have missed all that, I came back in the middle of his issues
17:58.36cod3hax0rheres my sip debug output http://pastebin.com/m55d54ef1
17:58.59wcselbycod3hax0r - that's only one transmission
17:59.08wcselbythere's more in a failed auth attempt
17:59.28wcselbyeverytime you register a phone with a secret, you'll always get a 401 unauth first, then the phone is supposed to resubmit with the proper auth
18:00.12*** join/#asterisk Ad-Hoc (~nimbus@62.1.172.179.dsl.dyn.forthnet.gr)
18:00.46cod3hax0rwhat will i need?
18:00.47Ad-Hochi ppl
18:00.50wcselbyif it's not sending the second auth with the correct auth info, either it's entered incorrectly in the phone itself, or it isn't receiving the 401 unauth message from asterisk (which can happen because of many issues)
18:00.56ManxPower-workThe way digest auth (both SIP and in HTTP) works.  Client connects, gets rejected with an auth required and is sent a key (called salt) to use to encrypt the response along with the password.
18:02.12wcselbycod3hax0r - you need to show us everything, from the first REGISTER statement (from the phone -> asterisk), the UNAUTH response from asterisk -> phone, and any response the phone may be sending back to asterisk.
18:02.44wcselbyalso, what's your network setup?  is the phone or the server behind a firewall, on separate networks, the same, etc?
18:03.22cod3hax0rhere is the other http://pastebin.com/m3c1877bc
18:03.29cod3hax0rmy server is collocated
18:03.38cod3hax0rmy zoiper softphone is behind a cisco 800 router
18:04.18wcselbywhat comes after the last sip message?
18:04.34*** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com)
18:04.40wcselbyyou've only showed the beginning of the sip negotiation
18:04.49Kattyhi
18:04.57wcselbyo/ kaldemar
18:04.59wcselbyerm
18:05.01wcselbyo/ Katty
18:05.02wcselby:)
18:05.10Kattythat's what you get for tab completing
18:05.22wcselbyit is indeed
18:05.30Katty;>
18:05.43ManxPower-workcod3hax0r: have you done a "service iptables stop"
18:05.51Kattyi acquired sparkling water on the way back
18:05.59tbensonManxPower-Work firgure out a workaround (had it typed in not to bother, but took a call).  just into lots of code for this application so was getting lost for a moment, just making my AGI create the call file dial from an alternate context that the hangup code and channelstatus will be accessible for the call in dialplan, rather then try to pull it back into the current dialplan.  This h exten is also not in default, its a custom c
18:06.00tbensonontext for alert notification to engineers.  Thanks all.
18:06.42Kattytries sparkling water for the first time.
18:06.51wcselbyi've never liked sparkling water
18:06.52Kattyhmm.
18:06.56Kattyneeds ice. brb
18:06.58wcselbyi don't mind sparkling grapejuice
18:07.08wcselbyor sparkling wine, for that matter
18:07.27MadkissLooking for 3946 in from-sip (domain 10.9.11.244) -- erm. thing is ... 3946 is in another domain, actually
18:07.40cod3hax0ryes ive done that service iptables stop
18:07.54cod3hax0rfunny thing is i can connect from my other branch in canada
18:07.57Madkisscan I tell asterisk in sip.conf that 3948 is in another domain?
18:07.58cod3hax0rbut i cant connect from here
18:08.48Kattytries again, with ice.
18:08.52wcselbycod3hax0r - did you understand what I meant about the last pastebin being incomplete?
18:08.54[TK]D-FenderMadkiss: * is NOT a PROXY
18:09.28Kattywcselby: not too shabby
18:09.36Kattywcselby: this has lemony flavors added
18:09.49wcselbyahh
18:09.54wcselbyso more like a sprite or 7up?
18:10.00Kattynot really
18:10.04wcselbylol
18:10.04Kattynot that strong
18:10.09Kattyjust, lightly lemmonied
18:10.10wcselbyi dunno, i may have to try it
18:10.18Kattyit's called Perrier Lemon
18:10.23wcselbyahh
18:10.26wcselbynever like perrier
18:10.38wcselbybut then, i never tried their flavored versions either
18:11.02Kattyit's different
18:11.22Madkissuh. I see the problem.
18:11.40Kattychecks on the critters
18:11.43wcselbywas annouce-position added to queues.conf in 1.6.x?
18:12.01wcselbyor is it supposed to be available in 1.4.x?
18:12.56ManxPower-workwcselby: what does queues.conf.sample say?
18:13.23ManxPower-workFrom YOUR Asterisk, not from some web site like voip-info.org
18:13.31wcselbyi don't see it in there, i just see it on the wiki (don't shoot me please.... ;)  ).  that's why I thought I'd ask.
18:13.38wcselbyhaha
18:13.46*** join/#asterisk lanning (~lanning@208.87.235.224)
18:13.46wcselbyyeah, it's not listed in my queues.conf.sample
18:13.58wcselbywhich is a shame
18:14.00Kattyeveryone sure is hostile today
18:14.43[TK]D-FenderKatty: LIES.
18:14.44ManxPower-work"The Internet crashed today, plunging the nation into productivity."  <-- The Onion
18:15.19Katty^- hostility
18:15.19[TK]D-FenderKatty: And I'll &#^$ING kill anyone who says otherwise
18:15.19wcselbylol
18:15.19Katty^- reeks of hostility
18:15.37ManxPower-workthat's not a nice think to say about wcselby
18:15.51wcselby;)
18:16.04*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
18:16.07Kattyputs up No Free Radicals Allowed sign
18:16.37Kattyeven my lunch is being hostile
18:16.43Katty:<
18:17.23bmoraca_workmmm...lunch...too bad it's a distant 105 minutes away
18:17.24Kattysomething which feels similiar to heartburn
18:17.34wcselbymy morning coffee is being pretty hostile to my stomach at the moment
18:17.42*** join/#asterisk atis_work (~atis_work@193.238.212.171)
18:18.24Katty:<
18:18.46Kattylet's write today off as a bad idea and go home
18:18.47bmoraca_workway OT, but has anyone here ever done those edible arrangements?
18:18.53Kattyyes
18:19.06Kattyi got the chocolate dipped strawberry plant
18:19.11cod3hax0rhere is the full log http://pastebin.com/m3b7d1dd5
18:19.11bmoraca_workany good?
18:19.15Kattymhmm
18:19.16Kattyspensive tho
18:19.55Kattyi'm getting ryan a case of beer for valentines day
18:20.39bmoraca_worki'm looking for something for my wife...already cooking her filet mignon, but she'll want flowers or something...and i'm thinking edible arrangements could be both pretty and useful for other reasons
18:20.40wcselbycod3hax0r - looks like your phone isn't receiving the 401 Unauth response
18:21.07wcselbycod3hax0r - what's the network setup between * and phone?
18:21.16KattyFor Other Reasons
18:21.17ManxPower-work<--- Transmitting (NAT) to 61.5.156.146:28659 --->
18:21.18Katty(tm)
18:21.20[TK]D-Fendercod3hax0r: I don't see configs to match...
18:21.21ManxPower-workThat concerns me
18:21.28[TK]D-FenderManxPower-work: It shouldn't
18:21.33KattyManxPower-work: you concern me.
18:21.40KattyManxPower-work: how are you feeling?
18:21.42bmoraca_workbad thing is that they don't deliver to where we are
18:21.51Kattybmoraca_work: they allow pickup
18:21.57ManxPower-workKatty: coming from Squirrel Girl.....
18:21.59bmoraca_workyeah...12 miles away
18:22.18KattyManxPower-work: razzleberries.
18:22.36Kattybmoraca_work: ahhhh
18:22.45Kattybmoraca_work: you could always make your own
18:23.22bmoraca_workKatty: when everyone else went to the ARTISTIC class in elementary school, I went to the AUTISTIC class.  i can't even draw my name on a peice of paper :P
18:23.32cod3hax0rits zoiper softphone
18:23.40cod3hax0rnetwork is behind a cisco 800 router
18:23.42cod3hax0rbasic nat
18:23.59Kattybmoraca_work: if you can do steak on a stick
18:24.04Kattybmoraca_work: you can do edible arrangements
18:24.12wcselbyhmmm, maybe i'm not seeing something obvious, but - is there a way to play a sound file to the caller when they enter a queue?  before any hold music or anything like that?  Or at the very least, let them know their position in queue and potential hold time?
18:24.55Kattyhttp://www.ediblearrangements.com/fruit-bouquet-detail.aspx?ArrangementID=318&StoreID=0&OrderType=&SelectedDate=&AreaName=&set=true <- bmoraca_work
18:25.16wcselbycod3hax0r - what's the network like on the * side?  is it behind a firewall / router?  does it have a public IP address?
18:25.32cod3hax0rit has a public ip address
18:25.33bmoraca_workcod3hax0r: ManxPower-work is right...your phone is never receiving the 401 Unauthorized.  Because of that, it never tries to register again with credentials (or it's not configured to).  check the config of your softphone.  it's not a NAT problem because it's not being retransmitted.
18:26.08cod3hax0rthe server is on a colocated public ip address in the internet
18:26.23Kattybmoraca_work: just get some big long wooden skewer things and put a bunch of fruit chunks on it
18:26.24bmoraca_workkatty:  I kind of liked this one: http://www.ediblearrangements.com/fruit-bouquet-detail.aspx?ArrangementID=316&StoreID=579&OrderType=2&SelectedDate=02/14/2010&AreaName=95382&set=true
18:26.24cod3hax0rand the workstation is behind a t1 with a public ip address
18:27.01Kattyit's still fruit on a stick
18:27.07bmoraca_worklol
18:27.13Kattytake skewer
18:27.14Kattyapply fruit
18:27.21Kattyplace in pretty colorful mug
18:27.28Kattyor bowl or vase, whatever
18:27.35Kattydrizzle with chocolate, chill in fridge
18:27.42bmoraca_workKatty: I also kindof liked this: http://products.proflowers.com/flowers/HugsandKisses-5395?viewpos=4&trackingpgroup=vca&ref=HomeNoRef&pagesplit=
18:28.08Kattyhmm
18:28.10bmoraca_worki suspect she wants roses though...being that she is a girl and my wife
18:28.23Kattyryan's never gotten me flowers.
18:28.25Kattypouts
18:30.10*** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com)
18:30.29bmoraca_workscrew it...a dozen roses it is.
18:30.34Naikroveki got my wife flowers once
18:30.38Naikrovekcat ate them and died
18:30.45Naikrovek"you don't ever buy me flowers again" she said
18:30.47Naikrovekokay...
18:30.47bmoraca_worklol
18:30.53bmoraca_workeasy way out
18:30.56Kattybmoraca_work: a reasonable choice
18:31.17Kattyi'll have yet another year of....no valentines day
18:31.18wcselbyNaikrovek - my wife has never, and will never, tell me anything even remotely like that
18:31.44benngardnow i am lost... shouldnt ${CALLERID(DNID)} be set to callee number when u execute a "channel originate ..." command?
18:31.46wcselbythe closest may be if I just simply get the wrong kind of flowers, she may say, don't get me THOSE kind of flowers again.  but really, you can't go wrong with roses, at least for my wife
18:32.30Naikrovekyeah wives are all different, just like like regular people
18:32.55Kattymaybe i'll send myself flowers this year
18:32.58wcselbylol
18:33.01Naikrovekaww
18:33.01benngardits empty both from cli and ami, and cdr post dst is empty to
18:33.28correticohello people
18:33.33Kattyand THEN i'll take myself out to dinner :>
18:33.54Kattyhave a few drinks, strike up a conversation
18:35.09*** join/#asterisk brad_mssw (~brad@shop.monetra.com)
18:35.09Naikrovekkinda like Tyler Durden.  forced against your own nature so much your Id becomes its own personality
18:35.34wcselbybut hopefully without all the mass murder and psychosis and stuff
18:35.41Naikrovekhe never killed anyone
18:35.49Naikrovekone guy died, but he was shot by cops
18:35.58Naikrovekbut plenty of psychosis
18:35.58wcselbyhe planned on blowing up a bunch of buildings and stuff though
18:36.15wcselbyi've read teh book, but it's been a LONG time....
18:36.16Naikrovekwcselby: yes, and he did, but security and maintenance were all in on it, and made sure the buildings were celar
18:36.27Naikroveks/celar/clear/
18:36.35Kattywhat an awful job
18:36.48Kattyget up in the morning, have a cup of coffee
18:36.51Kattydisarm a building
18:36.55Kattylunch
18:36.58Naikrovekthe book and movie are a bit different
18:37.01*** join/#asterisk _omer (~omer@119.152.140.100)
18:37.02Naikroveki recall Marla's mother in the book
18:37.05Naikroveknot in the movie
18:37.15Kattycan you imagine hitler's diary?
18:37.22Naikroveki can't imagine him keeping one
18:37.24Kattydeath death death death death
18:37.26n3hxsis no longer away : Gone for 2 hours 47 minutes 37 seconds
18:37.28Kattylunch
18:37.30_omerhi
18:37.31Kattydeath death death death death
18:37.34Kattyshower
18:37.39Nuggethugs hugs hugs hugs hugs
18:37.40jayteeI saw the recent Internet Hitler rant about the iPad. funny as usual
18:37.42Kattydeath death death death death, painting
18:37.43Naikrovek1pm: Manic.
18:37.46dlynes_laptopKatty, you're quite morbid and depressed this year?
18:37.49Naikrovek2pm: depressed
18:37.52Naikrovek3pm: manic
18:37.53Naikroveketc.
18:37.55dlynes_laptopKatty, why aren't you having valentine's day?
18:37.57KattyNaikrovek: yesh
18:38.06Kattydlynes_laptop: because ryan doesn't want to
18:38.14dlynes_laptopKatty, that's a dumb reason
18:38.15Kattydlynes_laptop: he thinks it's all commericalized and stuff
18:38.25jayteehe's such a romantic
18:38.28dlynes_laptopno kidding
18:38.30Kattyyeah, totally
18:38.41dlynes_laptopKatty, give him a kick :)
18:39.10Kattyheh
18:39.21dlynes_laptopmy sister-in-law can't pull that one off
18:39.22Kattythat man does do a single thing he doesn't wanna do
18:39.26wcselbyKatty - you need to explain to him that it doesn't matter what he thinks about Valentine's Day - it's what matters to YOU.
18:39.30Kattys/does/doesn't/
18:39.40[TK]D-FenderKatty: He's right.  Is he at least acceptably romantic acceptably often enougha s it is?
18:39.49dlynes_laptopShe was dumb enough to get married on Valentine's day :0
18:39.50Katty[TK]D-Fender: yes.
18:40.01wcselbyi feel the same way, but that doesn't mean I can get away with not doing anything for my wife
18:40.14Kattywcselby: well i'm not your wife either
18:40.16wcselbyalthough we usually schedule something on a day OTHER than vday
18:40.18[TK]D-FenderKatty: then good for that.  He should still do a little SOMETHING... but that doesn't imply any extra special expense.
18:41.16benngardcan some smrter than me explain whats wrong here: http://pastebin.com/d797a8309 ${CALLERID(DNID)} is empty but it dials my mobile
18:41.46_omerasterisk-addons help ... http://www.pastebin.org/86356
18:41.51theharComcast to buy NGT. :(
18:41.57Kattywha'ts ngt
18:42.11theharNew global telecom
18:42.14theharhttp://www.xchangemag.com/hotnews/exclusive-comcast-to-buy-ngt.html
18:42.21thehara large large wholesaler
18:42.29Kattywhat's this mean
18:42.59theharmeans comcast is on it's way to become the largest telcom provider
18:43.19Kattyand this is somehow, bad, right?
18:43.23theharcomcast == devil
18:43.24theharyes
18:43.31Kattyahh i see
18:45.28[TK]D-Fenderbenngard: exten => s,n,ExecIf($["${CALLERID(num)}"="0317998975"]?Set(CALLERID(all)=Magnus Benngard <317998975>))
18:45.39[TK]D-Fenderbenngard: "all" != "dnid"
18:45.58_omerasterisk-addons help ... http://www.pastebin.org/86356
18:46.02_omer?
18:46.34[TK]D-Fender_omer: Clearly
18:46.44wcselby_omer did you install mysql after you ran ./configure?
18:46.54wcselbybut before you ran make menuselect?
18:47.20_omermysql was installed before asterisk-addons
18:47.45_omereven mysql was installed before I downloaded asterisk-addons
18:48.24Kattythat's odd.
18:48.29KattyMajor General is just sitting there.
18:48.31Kattystaring.
18:48.41Kattyand he's been sitting there, staring, for the better part of 5 minutes
18:48.55*** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844)
18:49.07wcselby_omer - did you install from source or using a package manager?
18:49.19titterIs there a way to force users to change their voicemail password during their first login?
18:49.24_omerI did   yum install mysql
18:49.36_omerin CentOS 5.4
18:49.39wcselby_omer - you also installed mysql-server and mysql-devel?
18:49.50_omeryes
18:49.58_omerbut let me check mysql-devel
18:50.30_omerPackage mysql-devel.i386 0:5.0.77-4.el5_4.1 set to be updated
18:50.42_omeryum is updating mysql-devel now
18:50.51benngard[TK]D-Fender: that not the problem, check what "exten => s,1,NoOp(${CALLERID(DNID)})" procudśe for output
18:51.02benngardproduce*
18:51.10[TK]D-Fenderbenngard: that is looking at NUM, not DNID.  NOT ThE SAME
18:51.38[TK]D-Fenderexten => s,n,ExecIf($["${CALLERID(num)}"="0317998975"]?Set(CALLERID(all)=Magnus Benngard <317998975>)) <-- NOT DNID.
18:51.51*** join/#asterisk davix (~davix@82.166.170.193)
18:52.05wcselby_omer - do a make distclean, make clean, then redo ./configure in your asterisk-addons source directory
18:52.07*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
18:52.16_omerok
18:52.20wcselby_omer - after you've updated your mysql-devel, of course :)
18:52.27_omeryep :)
18:52.30benngardoki , will make it more simple and show u what i mean sec
18:53.07[TK]D-Fenderbenngard: You're talking about DNID, and the only ExecIf that matches isn't LOOKING at DNID
18:56.15_omermake distclean   =  DONE ,   make clean =  DONE ,  ./configure = DONE
18:56.26_omernow ?
18:56.32_omermake menuselect?
18:57.08wcselby_omer - yes
18:57.16wcselbyand let's see what it has to say about res_config_mysql
18:57.20_omerok
18:57.40benngardhttp://pastebin.com/d3816f268
18:57.51_omerIt works!!! options are available
18:57.56wcselby:)
18:58.01_omerthanks alot !! :)
18:58.02wcselbyglad I could help
18:58.13_omer:-D
18:59.04*** join/#asterisk davix (~davix@82.166.170.193)
18:59.07Alfiohi, anyone here knows if there is some SIP or IAX2 cleints for blackberry?
18:59.23wcselbyI know of sip clients for iPhone
18:59.29wcselbyi'm pretty sure there are sip clients for bb
19:00.19[TK]D-Fenderbenngard: Still meaningless.  This has nothingt o do with the other pastebin
19:00.43knctrnlThere are no SiP or IAX2 clients for BB as of the last time I looked.
19:00.50knctrnlthere are for Iphone and android
19:00.56Alfiook
19:01.02*** join/#asterisk opticy (~opticy@that.violates.us)
19:01.22Alfioi have two days looking into google and i cant see anything related to it
19:01.30opticyis it possible to route incoming calls based upon DFMTs entered by the calling party?
19:02.11wcselbyopticy - you mean like an IVR?
19:02.35benngard[TK]D-Fender: i did "clean out" so it was easier to see the error
19:03.09[TK]D-Fenderbenngard: You are looking at DNID in one, and NUM in the other
19:03.14[TK]D-Fenderbenngard: NOT THE SAME
19:03.20benngardand the follow up is that if u originate a call from cli, u will get an emtp dst in cdr
19:03.28opticywcselby: basically i only use asterisk for outgoing SIP/POTS calls; never for incoming POTS calls..i would like a wway that i could call my phone line from the outside via pots and have asterisk route the call only if it hears a sequence of DFMTs
19:03.38[TK]D-Fenderbeek: What don't you get?  You show me a call where you PROVE what you are comparing just before doing the compare.
19:03.58opticykinda like how it can detect a fax
19:03.59*** join/#asterisk voxter (~hardcore@S01060015f238f531.vn.shawcable.net)
19:04.55wcselbyopticy - once the call comes in, the users enter a DTMF code of some sort to route the call?
19:04.58benngard[TK]D-Fender: skip pastebin 1, look at pastebin 2, 2 calls 1 from cli, 1 from phone and they behave differnt
19:05.07wcselbyroute it within the system or back out on another line?
19:05.38benngardbut both works, call from cli will not produce correct cdr, call from phone will do
19:06.11[TK]D-Fenderbenngard: Why are you even looking at DNID?
19:06.31wcselbyopticy - if you're looking to password protect, check the Authenticate() app.  If you're looking to route calls based on a menu presented to the user, check into making IVR's (Google).  If you're looking to route a call back out on another line based on the DTMF input, check into DISA.
19:06.31opticywcselby: basically if i were to call the line, and enter say "*22" while it was ringing, asterisk would answer the call and route it to somewhere (like to an IVR or something)
19:06.33[TK]D-Fenderbenngard: And an ORIGINATE-D channel is not a an inbound call
19:06.46[TK]D-Fenderbenngard: They didn't call IN... Asterisk called OUT
19:07.00[TK]D-Fenderbenngard: Apples 7 oranges
19:07.05opticywcselby: authenticate sounds about right :) thx for the leads, i'll check them out
19:07.11wcselby_omer - opticy - asterisk has to answer the call first before it can handle the DTMF....unless you can do something with early media....but I've never worked with that
19:07.20[TK]D-Fenderbenngard: Feel free to update your records to correct this AFTER The FACT
19:07.25wcselbyyou know who I meant :)
19:07.27n3hxs[away]is now away - Reason : Auto-Away after 30 minutes
19:07.40opticywcselby: on faxes, for instance, it can not answer it unless it detects a fax..right?
19:08.01wcselbyopticy - my understanding is it won't detect that it's a fax until it's been answered.
19:08.08opticyah ok
19:08.09wcselbyso it has to answer it before it detects it's a fax.
19:08.21opticyperhaps that's where i'm a bit confused
19:08.26wcselbyat least, that's how I've always done it.
19:08.53*** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)
19:09.07wcselbylike I said, I haven't ever played with early media.  Or FAX over POTS, for that matter...  FAX over PRI though, the call comes into specific numbers and is routed to fax destinations.  we don't do asterisk's built-in fax detect.
19:09.23wcselbyor I should say, none of my clients do built-in fax detection.  they just setup extensions for fax.
19:09.30wcselbyerm, DID's for fax.
19:10.44benngard[TK]D-Fender: i can make a call to another sip phone but the same error will occur, i cant trust the cdr.dst because dnid is not set
19:11.07[TK]D-Fenderbenngard: How can you have a Dialed Number ID when..... they didn't DIAL A NUMBER?
19:12.16Kattythey used brain powers.
19:12.19benngardexplain for me plz how cdr.dst will be empty, i did try to follow the code backward
19:12.43*** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net)
19:12.53[TK]D-Fenderbenngard: there is no DESTINATION.  The call didn't come IN.  * called OUT
19:13.11[TK]D-Fenderbenngard: there are no corners in a round room.
19:13.14wcselbyis he looking for rdnis?
19:13.43benngardyes i se that in cdr, i see my 0317998975 as src as an empty dst
19:14.15Kattycatches up on the news
19:14.18[TK]D-Fenderbenngard: What you are looking for does not exist.  Nobody dialed that number to start the call.  * jsut called out on its own.
19:14.27rnpWill a propperly provisioned asterisk system hooked up to an online crm to provide it leads / account information need maintenance?  Or is it something that will just work?
19:14.48[TK]D-Fenderrnp: HUH?
19:15.24rnpFender: what I mean is if I have an asterisk system setup with custom scripting to work with my crm, will I have to worry about the asterisk server or will it just work without issue?
19:15.40jksMrnp, if you ask like that, you will have to worry
19:16.05[TK]D-Fenderrnp: Its an app.  Apps can lock-up, crash, fill your filesystem up with logs, have vulnerabilities that can be exploited, etc.
19:16.06benngardso it means if a wirite a small web app that let people make calls from ami they will not get recorded properly in cdr because of "nobody called" :(
19:16.12[TK]D-Fenderrnp: Just like every other app out there.
19:16.50[TK]D-Fender[14:07]<[TK]D-Fender>benngard: Feel free to update your records to correct this AFTER The FACT <--------------------------
19:17.18rnpFender:  most apps can run for months and years without having issues.  I'm just worried that I'm getting into something too complex by having my own asterisk server
19:17.26*** join/#asterisk atis_work (~atis_work@193.238.212.171)
19:17.38[TK]D-Fenderrnp: as opposed to?
19:18.07Kattyrnp: if you're that concerned, outsource it
19:18.34Kattyrnp: let someone qualified manage it
19:19.08rnpkatty: I wasn't concerned until someone trying to sell me the outsourced version struck fear into me
19:19.51*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
19:19.51Kattyrnp: keyword being Qualified
19:19.52*** join/#asterisk [netman] (~netman@147.Red-83-63-246.staticIP.rima-tde.net)
19:20.43rnpKatty: in your experience how much would it cost to have someone check in on an asterisk server once a week or so?
19:20.53[TK]D-Fenderrnp: Well you haven't described what tasks * will be performing.  What resources it will have, what devices it will talk to, etc
19:21.30[TK]D-Fenderrnp: certaint hings require more monitoring than others.  there are security considerations,e tc
19:23.28rnpFender:  it will work directly with my crm, getting leads from the crm to dial, then transmitting back successfull call data to the crm.  for voip it will use a service like www.vitelity.net   - as for the actual server i was going to use a slicehost.com virtual xen server
19:23.34rnpis that enough data?
19:24.25*** join/#asterisk ebroad (~ebroad@72.11.213.194)
19:25.03Kattysighs.
19:25.46*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
19:26.03[TK]D-Fenderrnp: How many simultaneous calls?  You will have keep up with updates on the software,etc.
19:27.03wcselbyrnp - are you going to do the install / linkup with your CRM?  or are you going to need someone else to do this for you?  are you going to need anyone do custom dialplan programming, etc?
19:27.06*** join/#asterisk Ta^3 (~tacvbo@189.146.190.22)
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19:27.16rnpFender: maximum of 20 simultanious calls
19:27.39wcselbywhen you say someone should "check on it" once a week or so, what do you need them to check on?  if it's working?  if it needs updating?  resource load, etc?
19:27.46rnpwcselby: I'm absolutely going to hire someone to set it up, do the link to the crm and custom programming.
19:27.57Kattyrnp needs to sit down with a support company and have a bid worked out
19:28.17rnpwcselby: honestly, I don't even know if it's something that needs to be looked at once a week.  My thought was someone should just check it out to make sure no problems were cropping up.
19:28.28wcselbywrnp - what I'm getting at is there a lot of factors involved, and no one can really give you an honest ball-park figure for what you're asking
19:28.52wcselbyuntil you sit down with them and (as Katty said) work out a bid
19:29.29rnpWcselby: can you give me a proper ballpark on this question, if the system is properly setup by a professional, should it need maintenance, or is it like a unix pc that can run for years without issue?
19:29.46Kattyrnp: find a support company
19:29.48Kattyrnp: request a bid
19:30.40*** join/#asterisk [netman] (~netman@131.Red-88-25-139.staticIP.rima-tde.net)
19:31.12Kattywasn't there an irc support channel for this?
19:31.17Kattyasterisk consultants or somethin
19:32.21wcselbyas [TK]D-Fender said earlier, it will be like any other application - issues can crop up, security patches would need to be applied, etc.  likely you'd work out a support contract guarenteeing x number of hours per month for a flat fee, with a negotiated rate for anything above that.  i don't know what other people tend to charge for these kinds of things.
19:32.30wcselbyKatty - I think there's an asterisk-biz mailing list
19:33.01[TK]D-Fenderrnp: Have you calculate the bandwidth requireents, etc?
19:34.29rnpI figure 100k for 20 callers is enough
19:35.40[TK]D-Fenderrnp: Unit of measure please.
19:36.15rnpKilobyte
19:36.21rnpis that what you mean?
19:36.29wcselbylunch time
19:36.29wcselby:)
19:37.10[TK]D-Fenderrnp: Try an say that in a more common UOM / multiple combination...
19:39.05[TK]D-Fenderrnp: One hundred thousand kilo-bytes?  Per?
19:40.03[TK]D-Fenderwishes he had such a 100 MEGABYTE link (is that bi-directional?)
19:40.43rnpFender: I'm not sure what you mean by VOM, but what I mean is each caller would use 5kilobytes of bandwith per second, and with 20 callers that would mean a throughput of 100kilobytes per second (about t1 speed if I'm not mistaken), that would be about 3gigs of bandwith for a 8 hour call shift
19:41.47[TK]D-Fenderrnp: What codec?
19:42.07[TK]D-Fenderrnp: is that 20 calls to the outside?  Who is talking to who?
19:43.12rnpI don't know what codec, the 20 calls are to the outside, to businesses in the usa
19:43.33[TK]D-Fenderrnp: Whats on the other side of the call?
19:47.06rnpa land line
19:47.18rnpwell
19:47.23rnpwe call businesses they have a land line
19:47.26rnpour end is just a computer with mic
19:47.37rnptransmitting to the asterisk server which uses voip
19:49.07[TK]D-Fenderrnp: Local soft-phone on PC?  Fine (Acutaqlly this is a SHIT expeience for the user, but i suspect you don't care)
19:49.26[TK]D-Fenderrnp: Calculate about 85kbps PER CALL.
19:51.01rnpok
19:51.19rnp85 X 50 Calls an Hour X 8 Hours a Day X 20 Callers
19:51.39rnp700megs a day
19:51.40rnpnot bad
19:52.17rnpalright, i've got to run, thanks for the info
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19:52.57Kattyscowls
19:58.26gelpgmort_gib: thanks the idea, know I can user the bri_cpe_pzmp signalling
19:58.35p3nguinHow did he arrive at 700M per day?
19:58.48nix8n82he thought each call last a second
19:58.52raden_worki was just thinking same thing
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19:59.42raden_workwow
19:59.43[TK]D-FenderAssumed call duration....
19:59.48[TK]D-FenderHorrible math
19:59.56[TK]D-FenderSOSO
20:00.19raden_worklol
20:00.20*** join/#asterisk oej (~olle@ns.webway.se)
20:00.41[TK]D-Fender~soso
20:00.42infobot[~soso] Shoot-On-Sight Offense
20:00.52p3nguinI was coming up with like 300GB per day.
20:01.29nix8n82I was wondering what that was
20:02.30nix8n82I think I was the one that scared him last night.
20:06.44*** join/#asterisk Victor_Yure_ (~victor@unaffiliated/victoryure/x-837844)
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20:10.37*** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl)
20:12.28p3nguinAfter recalculating, I'm thinking it's more like 3000GB per day.
20:14.37raden_workany opinion what might be more stable with asterisk 1.6 opensuse 11.2 or ubuntu server 8.04 LTS ?
20:14.55raden_work85*50*360*8*20
20:15.01p3nguinI would probably go with the LTS.
20:15.11Corydon76-digraden_work: I don't think it makes much difference
20:15.18*** join/#asterisk rtp4me (~447db8c1@gateway/web/freenode/x-pyeneglihnfkpech)
20:15.29raden_workCorydon76-dig, they are colo i want the least problems LOL
20:15.39rtp4mehey everyone
20:16.08raden_work244gb a day ???
20:16.08Corydon76-digraden_work: as long as you aren't using VM in production...
20:16.09raden_workwoa
20:16.14niekvlessertraden: don't think it'll make much of a difference
20:16.20raden_workok thanks
20:16.24raden_workil lstick with what i know then
20:16.29niekvlessertwhen you compile yourselve, dunno about the default package quality
20:16.38*** join/#asterisk pietro1 (~pietro@88-149-225-76.dynamic.ngi.it)
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20:16.48p3nguinraden_work: I'm coming up with 2391GB.
20:16.54Corydon76-digIf you use the MySQL ODBC connector, though, compile from source
20:17.03*** join/#asterisk ktwilight (~keliew@254.63-240-81.adsl-dyn.isp.belgacom.be)
20:17.07Corydon76-digYou'll save yourself a TON of grief
20:17.47niekvlesserti like bri stuff... compiles very easy
20:18.22rtp4meI'm experiencing a possible bug in 1.4.26 x86_64 with app_queue.  I have a queue set to ringall and autofill=no, but I am seeing autofill behavior (non FIFO), with latter callers ringing through before the oldest caller has been answered.  Thoughts?
20:19.26DotComStuhello, is there a function that returns the last number to call an extension (similar to 1471 in uk) or should i add some astdb magic to my dialplan to save last caller id to an extension?
20:19.51[TK]D-FenderDotComStu: latter <-
20:20.18rtp4meDotComStu: I think the last number dialed is also stored in the asterisk database, you should be able to find a key for it
20:20.20raden_worki was wrong on my math
20:20.29raden_work85 X 50 Calls an Hour X 8 Hours a Day X 20 Callers
20:20.38raden_workwhat does 50 calls a hour have todo with it
20:20.48raden_work50 hours of calls a hour ?
20:20.52rtp4meRegarding my previous question, where should I head with it?  Submit a bug?
20:21.00DotComStu[TK]D-Fender/rtp4me: ok thanks
20:21.07p3nguindotcomstu: Set(DB(callerid/last)=${CALLERID(num)})
20:21.10rtp4meDotComStu: no problem
20:21.21[TK]D-Fenderraden_work: SHHH!!!
20:21.34[TK]D-Fenderraden_work: Next you'll be expecting themt o be able to SPELL as well!
20:21.42raden_workLMAO
20:21.43p3nguindotcomstu: SayDigits(${DB(callerid/last)})
20:22.17rtp4mehello?
20:22.32DotComStup3nguin: k, but i want to store last caller id per exten so i guess ${DB(callerid/last-${EXTEN})}
20:23.00p3nguindotcomstu: I gave you the basic concept, so I know you'll be able to figure out the rest.
20:23.08[TK]D-FenderDotComStu: Better
20:23.13DotComStuthanks
20:23.45p3nguindotcomstu: I just use it at home to check the last calling number and I even have the option to blacklist it.
20:23.57rtp4meWhen you guys encounter problems that seem to be bugs, do you submit a bug report?
20:24.04rtp4meIs anyone willing to respond to me on here?
20:24.41[TK]D-Fenderrtp4me: We usually check it out with someone else first so we don't pst silly accidents as bugs
20:25.08rtp4me[TK]D-Fender: ok, totally agree.  Who would you recommend I check it out with.  It doesn't seem as though anyone here has any feedback.
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20:25.27p3nguindotcomstu: http://pastebin.com/d1bc37e3e
20:25.34niekvlessertrtp4me: have you tried other strategies? just to make the strategy is the problem
20:25.52niekvlessert* sure
20:26.47DotComStup3nguin: nice - didnt realise you could use *xx as an extension number :-)
20:27.03rtp4meniekvlessert: I haven't.  That's going to be my next step.  Either way, I need to get the ringall working though.  It's like it's using autofill, but it's completely disabled.
20:28.03gelpgI try to dial out via a B410P
20:28.27gelpgexten => 1004,n,Dial(DAHDI/21/06204661111)
20:28.36*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)
20:28.44gelpgbut my error answer in asterisk is the following
20:28.46niekvlessertrtp4me: weird stuff...
20:28.49gelpghttp://pastebin.com/d18e72926
20:29.34gelpgany hints, what did I setup wrong?
20:29.41p3nguindotcomstu: Alternatively, I also have a blacklisting extension in a different context:  http://pastebin.com/d7e1df6b8
20:30.00rtp4meniekvlessert:  Agreed.  I've never seen it happen on any other systems.  Of course, this one has enough volume for it to happen much more often.  I dug through the logs and don't seen any strange errors or timeouts reached... it's acting like its on autofill for all I can tell.
20:30.22rtp4meniekvlessert:  I suppose the course of action is to just submit a bug and change ring strategy in the meantime.
20:30.47*** join/#asterisk bsdmail (~dig@67.228.177.47)
20:30.56niekvlessertgelpg: the word dahdi is not in your log
20:31.02niekvlessertis this the correct log?
20:31.31niekvlessertrtp4me: why don't you try to get the queue smaller?
20:31.45rtp4meniekvlessert:  what do you mean>?
20:32.15gelpgniekvlessert: I think so, at least that's what asterisk shows when I try to call exten 1004 via a sip phone
20:32.16DotComStup3nguin: thank you - got that working nicely, *++
20:32.28wcselbyahh
20:32.30wcselbytex-mex
20:32.33wcselbychips and salsa
20:32.36wcselbyyummy
20:32.53p3nguindotcomstu: I guess the US's *69 is UK's 1471?
20:32.58*** join/#asterisk rtp4me_ (~tcboyko@adsl-68-125-184-193.dsl.irvnca.pacbell.net)
20:33.00niekvlessertrtp4me: well just remove people from the queue to see if all the phones ring then. :)
20:33.01DotComStu<p3nguin> yes
20:33.14niekvlessertcan be done from CLI
20:33.24rtp4mewell, it's not a problem with some of the phones not ringing
20:33.38Kattypeeks in
20:33.45rtp4meits a problem with newer callers ringing on the phones before the oldest one has been answered
20:33.46*** join/#asterisk Amorsen (~Amorsen@94.127.50.7)
20:34.03DotComStuis it poss to provide a "broken" dial tone to indicate stuff like message waiting etc for fones without a msg waiting lamp
20:34.20niekvlessertgelpg: try dialing voicemail or something with your phone first to see if it works
20:34.25dlynes_laptopDotComStu, it's done automatically
20:34.29rtp4meDotComStu: I believe you want to look into a playtones function
20:34.36dlynes_laptopDotComStu, it's called a 'stutter tone'
20:34.57DotComStuok
20:35.06dlynes_laptopDotComStu, all phones I'm aware of have that enabled by default (Aastra, Linksys, Aastra, Mediatrix)
20:35.11niekvlessertrtp4me: ah sorry. that's weird, i've seen that behaviour in the past (we have customers using queues a lot) but that was asterisk 1.2
20:35.16gelpgniekvlessert: I can call another softphone
20:35.21niekvlessertand the load had to be high
20:35.28dlynes_laptopDotComStu, but if it's not enabled by default, it is enablable
20:35.47*** join/#asterisk atis_work (~atis_work@193.238.212.171)
20:35.53DotComStu<dlynes_laptop> will look into that
20:35.54rtp4meniekvlessert: no problem, thanks for the help.  time for a bug submission!
20:35.58niekvlessertgelpg: context problem then maybe? u're not getting to the dahdi part at all
20:35.59dlynes_laptopDotComStu, oops...second aastra should have been sipura
20:36.37sbrathIf I have a call come in on a PRI from another internal Switch I'm rewriting the CALLERID(all)="555-1212", but the CALLERID(num) was an actual number in our office, CDR is being layed down as the rewritten CID, how can I change the CID out to the PSTN, but still write the real originating # to CDR?
20:37.40dlynes_laptopsbrath, write the real originating # as the 'accountcode'?
20:38.18dlynes_laptopsbrath, Set(ACCOUNTCODE=555-1212)
20:39.01dlynes_laptopsbrath, accountcode will take alphanumeric, too, so you can give it an English name as well...doesn't have to be a phone number
20:41.57sbrathThanks :)
20:42.18bsdmailcan someone help-me with a little issue, basically i want to receive a call on my cellphone (with a call-out script) and then dial 9 + externalnumber, and put the dialed number on the conversation, and play a music for both when I press 1. whats wrong? http://pastebin.com/d3a7a678f
20:43.13sbrathHmm, didn't work.
20:43.32p3nguinsbrath: After you set the accountcode, make sure you use CALLERID(num) to set the number before you Dial() out.
20:43.48sbrathThanks p3nguin!
20:43.58*** join/#asterisk Wildy (~simba@89.222.134.42)
20:44.02Kattyi have a cabbage craving
20:44.32Kattycabbage, garlic, mushrooms, and crushed red peppers
20:44.43Kattyand maybe garlic bread
20:44.44p3nguinspews all over the channel
20:45.00Kattyweirdo.
20:45.06sbrathawesome! it works.
20:45.14gelpgniekvlessert: what do you mean by that it does not get to the dahdi part?
20:45.18p3nguinThe only cabbage I will eat is purple cabbage in a salad.
20:45.25niekvlessertthinks katty is always hungry..... she should bring bread from home
20:45.35Kattyi am always hungry
20:45.37p3nguinMushrooms are a no-no.  I don't care for fungus that much.
20:45.44Kattybut if i ate all the time, i'd be huge probably
20:46.17niekvlessertgelpg: well, I don't see any dahdi stuff in the log, so i guess that line in the dialplan is never reached
20:47.10niekvlessert[Feb  4 21:27:19] WARNING[8154]: pbx.c:3675 pbx_extension_helper: No application '' for extension (phones, 1004, 1): not good gelpg
20:47.43niekvlessertdunno what that says, maybe someone in here does... :)
20:47.49bsdmailcan someone help?
20:48.17niekvlessertbsdmail: what does work and what doesn't?
20:48.17gelpgniekvlessert: what do i misconfigure? what kind of output or config would help?
20:48.22dlynes_laptopbsdmail, state your problem...we're not mindreaders
20:48.28bsdmaildlynes_laptop
20:48.29Kattydlynes_laptop: speak for yourself
20:48.29p3nguinWhat does phones,1004,1 have on it for a command?
20:48.32bsdmailcan someone help-me with a little issue, basically i want to receive a call on my cellphone (with a call-out script) and then dial 9 + externalnumber, and put the dialed number on the conversation, and play a music for both when I press 1. whats wrong? http://pastebin.com/d3a7a678f
20:48.34Kattydlynes_laptop: personally, i'm all sort of telepathic
20:48.38[TK]D-Fenderniekvlessert: No Application as it says...
20:49.06bsdmailbefore i dial the 9 extension, i'm able to dial extension 1 and hear the music, but i want to play the music for the other person
20:49.11niekvlessertyeah i know but WHY doesn't it know....
20:49.45p3nguin(1448.29) <p3nguin> What does phones,1004,1 have on it for a command?
20:50.06p3nguinShow us that, it'll likely show the problem.
20:50.12bsdmailniekvlessert did u understood?
20:50.28gelpgp3nguin: how can I do that?
20:50.39p3nguingelpg: You can look at the dialplan.
20:50.43niekvlessertbsdmail, yeah, but u still not told me what does work and what doesn't
20:50.54dlynes_laptopbsdmail, you're talking about hitting '1' both before the 9xxx call _AND_ after it?
20:50.56p3nguingelpg: dialplan show phones  or visually look in extensions.conf
20:50.58niekvlessertis the music on hold the problem?
20:51.16*** part/#asterisk ch1nawhyte (~dliu@aker.tobi-office.com)
20:51.34bsdmailyes, because i cant trigger the music on my side, and other side can't trigger too...
20:51.44bsdmailso the music doesn't starts
20:51.45dlynes_laptopbsdmail, also your pastebin does not illustrate what your problem is...it only illustrates that you don't have anywhere near enough dialplan to implement what it is you're talking about
20:52.09gelpgp3nguin: http://pastebin.com/d3c74ffac
20:52.37bsdmailso can u tell-me a site or tuto with the thing that i want to do?
20:52.39niekvlessert-2?? what is that
20:52.40p3nguingelpg: extension 1004 is broken.
20:53.08p3nguingelpg: Go look in extensions.conf and fix it.  If you aren't sure what is wrong with it, paste it in the pastebin and I'll look at it.
20:53.40niekvlessertbsdmail: once again, where are you at now?? can you recieve the call on your mobile??
20:53.53bsdmailyes
20:54.05bsdmaillet me explain, step by step
20:54.35gelpgp3nguin: http://pastebin.com/d1ad78faa
20:55.18p3nguin;exten => 1004,1,Verbose(1, Extension 1004)
20:55.23niekvlessertexten => 1004,n,Dial(DAHDI/21/06204661905)
20:55.24p3nguinexten => 1004,n,Dial(DAHDI/21/06204661905)
20:55.32niekvlessertthis should be 1,Dial etc
20:55.33p3nguinYOu have no priority 1 for 1004.
20:56.00niekvlessertwoohoo, my dcap is useful for something. :)
20:56.05bsdmaili have a sip softphone waiting for a call that is made with a asterisk call-out script, when i run the script, my sip phone rings and i answer. and i dial the extension: 9mycellphone , so my cellphone rings, and i answer, the next step is that i have to press 1 on my sip softphone to play a music to my cellphone hear and softphone itself too. but the problme is that the music doesn't plays.
20:56.33p3nguingelpg: So you can either uncomment the first line or change the second line from n to 1.
20:56.37niekvlessertbsdmail: does asterisk register the 1 key?
20:56.45bsdmailyes
20:56.46p3nguingelpg: then save/exit, then 'dialplan reload'
20:56.52bsdmailbut, nothing happens
20:57.15gelpgp3nguin: I did, nothing has changed
20:57.27p3nguingelpg: Then you did it wrong.
20:57.33niekvlessertgelpg: did you do reload?
20:57.33bsdmaili think that is some kind of linear problem
20:57.49gelpgyes, I did
20:58.01wcselbyon asterisk 1.4 - if a call-limit is set to 1, and there's a call in progress on that sip channel, and another call comes in, why does asterisk throw an error message stating: ERROR[6011]: chan_sip.c:3358 update_call_counter: Call to peer '2625' rejected due to usage limit of 1  ?  I mean, why is it considered an error, and not just the way it's supposed to be?
20:58.01p3nguingelpg: Which did you do, change n to 1 or uncomment the first line?
20:58.18niekvlessertcan you show us the show dialplan rule for 1004 again?
20:58.30gelpgp3nguin: uncomment the first line
20:58.35*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
20:58.35niekvlessertbsdmail: no idea, I'm not that good yet
20:58.43p3nguin"dialplan show phones"  again
20:59.05*** join/#asterisk ruben23 (~AGENT@122.55.48.243)
20:59.18gelpghttp://pastebin.com/d649cd5c1
20:59.37niekvlessertand now paste the debuglog
21:00.01*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
21:00.38gelpghttp://pastebin.com/d69efede2
21:01.44niekvlessertrule 47 in your paste says it all
21:01.51p3nguinUnable to create channel of type 'DAHDI'
21:02.03gelpgyes
21:02.24niekvlessertdoes dahdi_tool show you the card is up and stuff?
21:02.25gelpgbut dahdi show channels shows
21:02.39niekvlessertgelpg: tried channel1?
21:02.47niekvlessertor a group of channels?
21:02.59gelpghttp://pastebin.com/d5f853cfd
21:03.22gelpgchannel 1 is my first analog phone
21:03.34niekvlessertok, i mean the first isdn 30 channel
21:03.47niekvlessertis your jumper in the right setting? :)
21:04.11niekvlesserttakes a show first
21:04.14niekvlessert*shower
21:04.24p3nguinshower show?
21:04.28bsdmailniekvlessert and there is other problem, i can't hear on softphone what is said from my cellphone, i think is that asterisk isn't putting the calls togheter
21:04.35gelpgthe card lights green
21:04.36p3nguinhopes you're a girl, otherwise will not be watching
21:05.22bsdmailthere is a command to put after the answer to put the calls in a "conference"
21:05.22gelpgit is a simple NT
21:05.23bsdmail?
21:05.35gelpgwith 2 channels
21:05.38p3nguinMeetMe()
21:06.13gelpgthe card is TE mode
21:06.21DotComStuhow do you tell Dial() to use the dialplan instead of directly dialing the channel? eg Dial(SIP/123) is not reaching the macro for extension 123 even though its been included in the context
21:06.43p3nguindotcomstu: local channels -- Dial(Local/123@context)
21:06.46bmoraca_workDotComStu: use the local channel
21:07.00DotComStuthanks - thats logical
21:08.08*** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
21:08.40wcselbyhmmm
21:08.52Wildyhi there :) anyone tried to use res_snmp w/Zabbix?
21:12.36bsdmailapp_meetme.c:774 build_conf: Unable to open pseudo device
21:13.26ManxPower-workbsdmail: you need zaptel/dahdi installed to use MeetMe
21:14.59gelpgp3nguin: here is my dahdi_cfg -vv output:  http://pastebin.com/d659fdc78
21:15.44p3nguingelpg: Hopefully someone else can lend a hand here, as I do not use dahdi channels to know what they should look like.
21:16.09gelpgp3nguin: thanks anyway
21:16.19ManxPower-workwe would need your dahdi config FILES as well
21:20.15niekvlessertbsdmail: ztdummy is enough
21:20.18gelpgmy system.conf : http://pastebin.com/d2e21ffcb
21:21.07niekvlessertspan=3,0,0,ccs,ami
21:21.11niekvlessertno timing?
21:22.02gelpgi read it from a sample config
21:22.21gelpgwhat shall I change?
21:22.38ManxPower-workgelpg: other than lack of echo canceler it looks good.
21:23.08gelpgmy cha_dahdi.conf : http://pastebin.com/d1ab7d117
21:23.16benngardhow do u say in english? pest or kolera?
21:23.52ManxPower-workgelpg: what specific problem are you having?
21:24.26gelpgi try to dial oupt my B410P card
21:24.47titterWhats the best way to force users to change their voicemail password the first time they access their voicemail?
21:24.56gelpgbut I cannot and I'm trying to find out what is the problem
21:25.02ManxPower-worktitter: voicemail.conf.sample should have info on that.
21:25.10titterManxPower-work: thanks
21:25.11ManxPower-workgelpg: pastebin the CLI output of a failed call
21:25.36ManxPower-worktitter: you set their password to be the same as the voicemailbox and then enable some option in voicemail.conf
21:26.02gelpghttp://pastebin.com/d22f3c64c
21:26.08ManxPower-workgelpg: also pastebin the output of "dahdi show channels" in the CLI
21:26.38ManxPower-workThis indicates you have a DAHDI issue.  [Feb  4 22:25:27] WARNING[8451]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
21:26.44gelpghttp://pastebin.com/d1b30eb3e
21:27.15gelpgManxPower-work: yes, that's my problem
21:27.34gelpgbut I have no clue what do I wrong
21:27.45niekvlessertgelpg: span=3.1.0,ccs,ami
21:27.54ManxPower-workgelpg: I would contact Digium support.  Give all the configs you gave us.
21:27.55niekvlessertspan=3,1,0,ccs,ami
21:28.12ManxPower-workniekvlessert: commas, not .
21:28.30niekvlessertManxPower-work: /me is kinda tired :)
21:29.19gelpgniekvlessert: I tried but nothing has changed
21:29.30n3hxsis no longer away : Gone for 2 hours 52 minutes 3 seconds
21:29.46niekvlesserthmm
21:30.15niekvlesserttest
21:30.20niekvlessertah ;
21:30.23niekvlessert:)
21:30.34niekvlesserti shouldn't start a line with /var/log/syslog
21:30.42niekvlessertcheck that gelpg
21:30.52niekvlessertor messages
21:31.19dlynes_laptopHas asterisk always used sequential channel names, and I just never noticed?
21:31.38ecraneAny ideas who this VoIP equipment provider is that tries to use 1.1.1.1? http://labs.ripe.net/content/pollution-18
21:32.03*** join/#asterisk superbeef (~lanej@74.84.194.4)
21:32.45raden_workrunning asterisk as a stand alone server with 300 extensions is there anything  ishould do special for partitioning ?
21:32.47superbeefare there any CLI tricks for disabling forwarding on an extension?
21:33.24nix8n82only if you want to do something special
21:33.33superbeefhow special
21:33.41*** join/#asterisk uqlev (~yuriy@91.184.221.31)
21:33.44ManxPower-worksuperbeef: You must be using a GUI
21:33.47nix8n82that is a good question
21:34.14Warp4hi all, writing a short checking script in php for asterisk (at http://pastebin.mozilla.org/701058) and i am wanting it to pull back the status of a call.  what do I need to add to this to get this to happen?
21:34.51superbeefManxPower-work: yeah AMP or Freepbx manages my extensions..... I tried doing database show and looking for a flag on my ext but didnt see anything to set
21:34.56gelpgniekvlessert:   http://pastebin.com/d395e23ea
21:34.56niekvlessertsuperbeef: use agi to get to a script that disables the forwarding by changing a field in de the db
21:35.00ManxPower-worksuperbeef: since Asterisk does not actually have a Call Forward feature.
21:35.04ManxPower-work~freepbx
21:35.05infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
21:35.18gelpgniekvlessert:   I mean I checked but nothing has happened
21:35.39niekvlessertgelpg: weird... can you give me shell access?
21:36.05niekvlessertgelpg: we can use screen so that you can see i won't pollute anything :)
21:36.07p3nguinn3hxs: I really hope you're going to stop doing that soon.
21:36.12DotComStuwhy doesnt BackGround(digits/12345) work?
21:36.26p3nguinn3hxs: You'll notice that none of the other 242 people here do that.
21:37.06gelpgniekvlessert:   it's a littlebit difficult but I try to organize
21:37.16niekvlessertssh portfowarding ftw
21:37.31niekvlessertdo you guys know ssh -w any:any ?
21:37.45nix8n82Warp4, I would recommend looking at phpAGI it has a manager class that makes things easier
21:37.47niekvlessertone can tunnel sip + rtp through an ssh tunnel
21:37.49niekvlessertwith it
21:37.53dlynes_laptopecrane, probably misconfigured...it's the example ip address usually given in Cisco examples
21:38.00Warp4nix8n82, got a current URL for it?
21:38.14DotComStuis it possible to saydigits() in the style of Background()
21:38.22raden_worknix8n82, mainly just wondering how full mailboxes could get etc....
21:38.53n3hxsp3nguin do what?
21:39.27ManxPower-workDotComStu: not that I'm aware of, but if you find out let me know
21:39.40nix8n82google phpAGI
21:39.51Warp4~phpAGI
21:39.52infoboti heard phpagi is http://phpagi.sourceforge.net/
21:39.54dlynes_laptopecrane, it also seems to be an example ip address used by Huawei, too
21:39.57DotComStuManxPower: http://lists.digium.com/pipermail/asterisk-users/2006-January/136692.html
21:40.35niekvlesserti will ask the question again i also asked yesterday
21:40.58ManxPower-workDotComStu: Ah.  Clever.
21:41.04p3nguinn3hxs: I'm talking about the away/back announcements.
21:41.27ManxPower-workIt won't really work for my needs, but it's still clever.
21:41.32n3hxsOH, sorry, I will fix that.
21:41.42p3nguinn3hxs: Thanks!
21:41.42*** join/#asterisk Greek-Boy (~Monching@41.188.154.137)
21:41.52*** join/#asterisk zafar_ (~IceChat7@91.144.34.220)
21:41.56niekvlessertis someone in here capable and willing to change the asterisk source so that sip headers can be sent by sip after a call has started? So i can change the display from a phone when doing direct call pickup for example... I'm willing to pay for it
21:42.03*** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au)
21:42.11ManxPower-workDotComStu: you can also just merge the individual sound files if it's something that will be the same often
21:42.36nix8n82raden_work, yeah I could see that as an issue, play it cautious man, maybe you could write a short cron job to alert you to when your mail is at critical size
21:42.39*** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be)
21:42.41*** join/#asterisk lupine_85 (~lupine_85@unaffiliated/lupine-85/x-7392152)
21:42.57lupine_85scratches his head at chan_mobile
21:43.01dlynes_laptopniekvlessert, Are you using Polycom or Aastra phones?
21:43.03raden_worknix8n82, thanks ;)
21:43.06DotComStuManxPower: its for random inbound callerids so i'mm sticking with the loop way of doing it
21:44.05lupine_85so the module is failing to load, but not giving me any decent output as to why. that's with core set debug (big) and core set verbose (big), then module load chan_mobile. Anyone know how to coerce it into telling me what its problem is?
21:44.11niekvlessertdlynes_laptop: aastra
21:44.29dlynes_laptopniekvlessert, why not use xml?
21:44.40niekvlessertdlynes_laptop: slow and ugly
21:45.10niekvlesserti have to check through ami if a call has started and then decide what to push
21:45.23niekvlessertif the call ends i have to push it empty again
21:45.59raden_workbrb
21:46.54niekvlessertdlynes_laptop: any opinions about it? it's an anoying problem.... my hopes are on sipsak right now
21:47.30dlynes_laptopniekvlessert, no idea...I don't know sip on a low enough level
21:47.58niekvlessertdlynes_laptop: same here... :( i'm willing to pay!! find me someone to fix it :)
21:49.26Deeewayneanyone here do any asterisk-java development?
21:49.54ManxPower-workHeh, Just got notification from one of our carriers that there will be an outage tonight to "clean the fiber" (I'm assuming fiber ends).
21:50.17Deeewayneoffers ManxPower-work a bran muffin
21:50.29dlynes_laptopManxPower-work, if it's inside of an airtight connection, why should they even need to clean it?
21:51.09ManxPower-workdlynes_laptop: You shouldn't, but that card is taking errors so they are looking at things like light levels, etc as well
21:51.17dlynes_laptopah
21:52.02zafar_how much r u people ready to pay for this, i think i have people who can do it
21:52.29zafar_i need to talk to the guys though
21:52.43*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)
21:52.57niekvlessertzafar_: talking to me?
21:53.06*** join/#asterisk came0 (~came0@rrcs-71-42-53-211.se.biz.rr.com)
21:53.07niekvlessertwell we can discuss hourly rates
21:53.14came0polycom 335 woot! :)
21:53.20niekvlessertand then make a guess on how much hours it will be
21:53.31zafar_yes
21:53.42zafar_i need to know the estimate so i can talk to them
21:54.57*** join/#asterisk unspin (~unspin@96.49.129.159)
21:55.07niekvlessertestimate for what?
21:55.22niekvlesserti can write down what needs to be done?
21:55.30niekvlessertmaybe they can decide how much hours that would be
21:55.40zafar_like how much time can u give them and how much per hour
21:58.46Kattyjeebus
21:58.49Kattyi fell asleep
21:58.50unspinis it possible to apply a simple add on top of an existing integer value without first reading it in?
21:59.02Kattyand i do believe that is the fastest i have ever darted awake >.<
21:59.02unspinfor example:  update partner set balance=add(5) where iid=1;
21:59.02unspini'm running mysql 5.1
21:59.09Qwellunspin: #mysql ?
21:59.25unspinaw crap
21:59.25unspinwrong window
21:59.25unspin:)
21:59.36Kattyanother cup of tea
21:59.38Kattymove down move down
22:00.47Kattyeppigy: what's fur dinner tonight?
22:00.58Deeewaynehugs Katty
22:01.04Kattyhugs Deeewayne
22:01.25KattyDeeewayne: you must some major cuteness.
22:01.31eppigyKatty: STEAK AND BAKED POTATO
22:01.35eppigy8D
22:01.36Kattyeppigy: again?!
22:01.38eppigyYESH
22:01.44Kattyeppigy: leftovers?
22:01.46eppigyI am on a diet
22:01.49eppigynegative
22:01.52Kattyoh
22:01.59eppigythe easiest way for me to do diets
22:02.02Kattywell you're not gonna gain much weight on steak and potato, dear
22:02.04eppigyis just eat the same thing every day
22:02.07DotComStu<unspin> update abc set balance=balance+1 where x=y
22:02.09*** join/#asterisk mmj_nix (~mmj069@c-76-27-116-95.hsd1.ut.comcast.net)
22:02.14eppigyI eat six meals a day
22:02.20Kattyokay well that might work
22:02.32eppigyyesh
22:02.34Kattyput some cheese on that potato
22:02.38eppigychicken and baked tatoes
22:02.43eppigyand avocados
22:02.48Kattynow i want a baked potato
22:02.57Kattyit sounds DELISH
22:03.02eppigyyesh
22:03.25Kattywhy do i always get hungry at 4? :<
22:03.36eppigyyou are a masochist?
22:03.57Kattyno
22:04.10Kattywhat's that got do with getting hungry at 4
22:04.24lupine_85oooh, another step along
22:04.25eppigyI am not sure :<
22:04.34Kattymmmmmmmmmisee. k
22:04.48niekvlessertzafar_ you here?
22:04.50*** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil)
22:05.04Kattyeppigy: how many calories are you shooting for per day
22:05.43mmj_nixwhats the easiest  way to do incoming CID pattern matching w/o freePBX or the like?
22:05.50Kattyinfobot: freepbx
22:05.50infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
22:06.27niekvlessertzafar_: plz!!
22:06.28eppigyKatty: like three thousand something
22:06.28mmj_nixnot using a GUI is what I meant
22:06.48Kattyeppigy: that's a lot.
22:06.51eppigyIts been 2 1/2 months
22:06.57Kattyeppigy: how much have you gained?
22:07.00eppigyand I have put on like 27 lbs of muscle
22:07.05Kattyhot.
22:07.05eppigy5 pounds fat
22:07.08eppigyyesh
22:07.11Kattyi wanna see.
22:07.28eppigypic incoming
22:07.41eppigywith silly expression
22:07.45Kattycheers.
22:08.09Kattyomg. that's hilarious
22:08.20zafar_sorry budy i was getting a cup of tea :)
22:08.28eppigyyesh
22:09.45p3nguinmmj_nix: You can use two ways.  Use the old callerid matching method of exten => 123/432,1,Stuff() where 432 is the caller id number to match against, or you can use the newer method if GotoIf([${CALLERID(num)} = 432]?placetogo)
22:10.12*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
22:12.24niekvlessertaaah zafar_ is back with tea :)
22:13.01Kattyi want some tea.
22:13.47Kattyand some steak.
22:13.49mmj_nixp3nguin: tried to use DID/CCID, stuff  in trunk include, seems not to find it though
22:13.54Kattyno, i don't want steak.
22:13.58Kattyi want....
22:14.01KattySoup
22:14.15niekvlessert[Feb  4 23:13:36] WARNING[9437]: channel.c:4003 ast_request: No channel type registered for 'DAHDI'
22:14.15niekvlessert[Feb  4 23:13:36] WARNING[9437]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)
22:14.19niekvlessertwhat does this mean?
22:14.20benngardhad onion soup for dinner
22:14.26p3nguinmmj_nix: I don't even know what "in trunk include" means.  We don't support FreePBX here.  I'm just telling you how it is done in extensions.conf.
22:14.28Kattybenngard: do you have a recipe?
22:14.43benngardKatty: no, but my wife has
22:15.03Warp4hmm
22:15.43Deeewaynemmmm.... onion soup .....
22:15.43Warp4ok having an issue with the phpAGI thingy
22:15.43benngardKatty: i ask her to write it down and then i pass it to u
22:15.43Kattybenngard: :>>>>>>>>>>
22:15.43Kattybenngard: tank you berry much.
22:15.43KattyDeeewayne: well i was thinking more like colby corn chowder, personally
22:15.48Warp4not quite sure what this means:  PHP Parse error:  syntax error, unexpected T_DOUBLE_ARROW in /root/scripts/test_rw.php on line 15
22:16.17DeeewayneI don't even know what I want for dinner :-(
22:16.36QwellDeeewayne: the souls of the infidels
22:16.41Qwellwith a little white wine sauce
22:16.58KattyDeeewayne: when you get hungry, it'll come to you
22:16.59niekvlessertniekvlessert: [Feb  4 23:13:36] WARNING[9437]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)
22:17.03DeeewayneQwell, people would be walking into automatic doors everywhere
22:17.04mmj_nixp3nguin: not using freepbx, just editing extensions.conf with DID/CCID, stuff from an included context
22:17.16Warp4http://pastebin.mozilla.org/701059
22:17.22DeeewayneQwell, I'm guessing you got the simpsons reference
22:17.26Qwellindeed
22:17.34KattyDeeewayne: http://farm4.static.flickr.com/3501/4035529177_2707198336_o.jpg
22:17.43Qwellwould have come up with a witty response, if I wasn't responding to an sms
22:17.52DeeewayneKatty, is that bacon ?
22:17.57KattyDeeewayne: why yes, it is
22:18.02Deeewayneyay!
22:18.07KattyDeeewayne: is there a law that says soup can't have bacon?
22:18.38Deeewaynenot sure, but I heard that there is a law in Washington state forbidding the sale of lollipops
22:18.46Kattymost unfortunate
22:19.18KattyDeeewayne: what's your opinion of great northern beans?
22:19.34Deeewaynejava beans ?
22:20.06Deeewaynehonestly, I fear most non-green beans unless they are mixed in with something else
22:20.26Kattydigs up an old recipe
22:21.02KattyDeeewayne: http://42ndrecipestreet.blogspot.com/2009/07/ryans-country-beans.html
22:21.10KattyDeeewayne: ^- uses 2lbs of bacon
22:21.19Deeewaynewoot!
22:21.31p3nguinmmj_nix: Here's an example of how to use it:  http://pastebin.com/d3abd8d07
22:21.34KattyDeeewayne: enjoy.
22:21.41Qwelleww
22:21.41Deeewaynerumor has it, it is Miller time at my office
22:21.43p3nguinmmj_nix: Pay close attention to lines 4, 5, and 6!
22:22.21DotComStui would like to handle the calling of a sip extension thats not online, CHANUNAVAIL look like the DIALSTATUS to check - yet the response from dial is "exited non-zero" and it doesnt goto s-CHANUNAVAIL
22:22.43p3nguinsip extension, eh?
22:22.48DotComStuanyes
22:23.04p3nguinExtension discrimination!
22:23.10DotComStulol
22:23.21DotComStuis that a known feature
22:23.23KattyDeeewayne: it was naptime in my office earlier
22:23.24p3nguinYou know that devices are not extensions, yes?
22:23.51DotComStuyes but how do i tell if its registered
22:23.59Kattyaka, I don't care.
22:24.10DotComStuthe docs say "CHANUNAVAIL: Channel unavailable. On SIP, peer may not be registered. "
22:24.20DotComStuguess MAY is the operative word
22:24.35p3nguinIf the device is not registered, it cannot receive a call.
22:24.39Kattyp3nguin: we're supposed to get more weathers here :<
22:24.46Kattyp3nguin: of the bad variety
22:24.49p3nguinI heard we should get snow tomorrow.
22:26.11Kattychecks weather
22:26.30Kattywell it was calling for ice
22:26.56Kattyit's been changed to rain/snow now
22:27.29Kattyah, and snow all weekend. joy.
22:27.49niekvlessert<PROTECTED>
22:27.50niekvlessertargl!!
22:28.13*** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca)
22:31.40p3nguinI was just told that we should expect rain to start around 6pm.
22:32.37niekvlesserthttp://www.voip-info.org/wiki/view/Bristuff
22:32.47niekvlessertsorry wrong window :)
22:36.58*** join/#asterisk slima (slima@unaffiliated/slima)
22:37.24*** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754708.dsl.bell.ca)
22:37.36*** join/#asterisk mykhyggz (~col@evolone.org)
22:41.27*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
22:43.37*** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net)
22:44.32lupine_85does anyone know offhand which bluetooth profile is required for sending SMS with chan_mobile? Looks like SPP?
22:45.05NuggetGet down with SPP (yeah you know me!)
22:46.05Qwelllupine_85: HFP supports SMS, I believe
22:46.16mmj_nixdialplan show 801.......@DID_trunk_1_default - shows a match from _NXXZXXXXXX, but still choosing 's' instead on the inbound call
22:46.36Qwellbut yeah, HFP uses SPP,so..
22:47.08p3nguinI decided _NXXZXXXXXX should really be _NXXNXXXXXX to meet NANP's specifications.
22:47.34mmj_nixtrying
22:47.45random_mikegreetings
22:48.18p3nguinmmj_nix: That change won't affect your result in this case, though.
22:48.20dlynes_laptopIs a TURN server a new alternative to STUN?
22:48.39p3nguinmmj_nix: Paste your debug of the failed call into pastebin.com
22:49.03lupine_85Qwell: thanks. I'm trying to convince a windows mobile phone to provide the correct profile...
22:49.11lupine_85as with any sort of windows thing, is Hard(tm)
22:50.59Kattyeppigy: hmm. the hungries went away
22:51.06*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
22:51.10p3nguinYeah, now I got 'em!
22:51.11Kattyhi jaytee
22:51.18Kattyp3nguin: oh? did you steal my hungries?
22:51.29Kattyp3nguin: so kind of you ;>
22:51.30p3nguinI assumed you gave them to me.
22:51.42Kattycould be, could be
22:52.05jayteehi Katty
22:54.08p3nguinI have some fancy chips made of rice that I am considering devouring.
22:54.58Kattyaren't you going home in 5 minutes?
22:55.31p3nguinriceworks sweet chili
22:55.46*** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus)
22:55.49Kattyyou're gonna ruin your dinner
22:56.24p3nguinAh, good point.  I'm making tacos for supper.
22:56.34p3nguinerr, assembling tacos, really.
22:56.57*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
22:56.59p3nguinstore-bought stuff, just have to cook the meat and put everything together.
22:57.13Katty^_-
22:57.19Kattyi don't get it.
22:57.22Kattythat's all there is to tacos
22:57.28Kattymeat, tortillas...and assorted chopped veggies
22:57.32mmj_nixp3nguin: http://pastebin.com/d5441307d
22:57.43dlynes_laptopKatty, yeah...no point to buying a 'mix', is there?
22:57.49p3nguinWell, I'm not making my own shells, not using my own meat seasoning, et cetera.
22:58.06bmoraca_workit's easy
22:58.17p3nguinI'll use Taco Bell brand seasoning, Old El Paso shells, Ortega sauce.
22:58.34jayteethe best of all worlds
22:58.36p3nguinmmj_nix: http://pastebin.com/d38bbdf00
22:58.59p3nguinmmj_nix: Pay close attention to lines 4, 5, and 6.
22:59.13bmoraca_workwindows XP is a peice of shit
22:59.14Kattyp3nguin: ah, right
22:59.15p3nguinmmj_nix: CLOSE attention.
22:59.33jayteecan't remember if it's Old El Paso or Taco Bell but one of em makes a seasoning pack that is chipotle flavoring. it's pretty good
22:59.41bmoraca_worki can't run Dreamweaver, Photoshop and SQL Server Manager
23:00.09p3nguinI will shred my own cheese, though.  Kraft block cheese is cheaper than buying it shredded alrady.
23:00.16bmoraca_worki have to shut one of them down before i can use another
23:00.29p3nguinjaytee: Taco Bell.  That is the exact one I use.
23:01.02*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
23:01.02bmoraca_workp3nguin: i like the Lowrie's taco seasoning
23:01.47p3nguinjaytee: "chipotle flavor taco seasoning" is what's on the front.
23:01.51Kattyhometime
23:01.55Kattylater gaters
23:01.56jayteeyep
23:01.58jayteelater
23:02.33p3nguinI don't know why one brand can't have all the products I like, but I have to get all the different brands to be satisfied.
23:03.27p3nguinOh, and I like store brand sour cream better than the national brand.  :/
23:04.26*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
23:05.27p3nguinmmj_nix: Are you understanding the significance of those lines that I say are important?
23:05.27jayteeI like the Stand and Stuff shells from Old El Paso
23:05.38p3nguinsquare bottoms?
23:05.41jayteeyep
23:05.49p3nguinDo they have those in super stuffer size?
23:05.52jayteeI make mine toasted
23:06.26p3nguinI prefer the super stuffer shells because I don't have to make and eat as many tacos.
23:06.47jayteesuper stuffer size? don't think there's much difference between the Stand and Stuff from them and the Super Stuffer ones because they're wider than the regular shells so you can really fill em up
23:06.54Qwellpfft, shells.  those aren't real tacos
23:07.07p3nguinMexican-American!
23:07.21jayteeQwell, I take it you're a soft taco man?
23:07.31Qwelljaytee: I'm a real taco man
23:07.47p3nguinReal tacos don't use tortilla?
23:07.52jayteeQwell, please elaborate on what you think makes a real taco
23:08.02Qwellhttp://soundbites.typepad.com/photos/uncategorized/tacos1.jpg
23:08.03mmj_nixp3nguin: those are the CID matching lines (4 & 5) the NoOp line is the key?
23:08.04Qwellthat
23:08.40p3nguinmmj_nix: The NoOp line, which also has the SAME PRIORITY as the CID matches allows non-matching calls to continue down the dialplan.
23:08.47Qwelljaytee: *completely* different
23:08.53Qwellground beef.  pfft!
23:09.00*** join/#asterisk unspin (~unspin@96.49.129.159)
23:09.05Qwelland...cheese?!
23:09.09jayteethose look like soft flour tortillas
23:09.10p3nguinqwell: Yummy!  Is that what you're making for me tonight?
23:09.16Qwelljaytee: they are corn
23:09.25Qwellwell, okay, those might be flour
23:09.40Qwell(they should be corn though)
23:09.42jayteemight be masa flour
23:09.50*** join/#asterisk hhkahya (~hulusikah@88.247.127.66)
23:09.55jayteelooks tasty
23:10.32jayteethere's a place near me that makes authentic tamales
23:10.38p3nguinI should do real homemade tacos some day soon.
23:10.53jayteeand it's real shredded beef or chicken
23:11.17QwellI used to live in the ghetto.  We had a tamale guy that would walk around the neighborhood with his cart.
23:11.25p3nguinI have two restaurants here:  Tequila and El Rancherito
23:11.26Qwell$1/each.  pretty good
23:11.43p3nguinThey both make authentic food.
23:12.45QwellI miss the corn man and his van. :(
23:13.21Qwellcorn on the cob, buttered, put on a bunch of like parmessan cheese, and chili powder
23:13.26Qwellmmm
23:13.39*** join/#asterisk garymc (~chatzilla@host86-158-86-203.range86-158.btcentralplus.com)
23:13.40jayteehad a guy like that in Oklahoma City. 4 for a buck back in the 70's. really good homemade tamales
23:14.05Qwellhttp://www.saveur.com/article/Recipes/Mexican-Corn-on-the-Cob-1000075465
23:14.13QwellI guess it was mayo and not butter
23:14.23dlynes_laptopdamn....why don't they have food that cheap here? :(
23:14.24p3nguinI don't know how to shred the beef correctly, so mine might not be authentic when I'm finished.  :/
23:14.32ruben23hi anyone used eyebeam softphones for attended transfer on asterisk.
23:14.37garymcHi peeps, anyone know if or how I can get into a polycoms software in the office to fix the phone for the boss? The Server IP is opent to the web and i know the ip for the phone on the subnet is 192.168.0.34
23:14.39DeeewayneQwell, my in-laws in georgia get their potatoes from a guy who fills his car with potatoes and drives from building to building trying to off load them
23:14.46garymcim at home
23:14.51Qwellpotatoes?
23:14.54Deeewayneyes
23:15.00Qwelllike, just regular old potatoes?
23:15.05ruben23i have requirements on it but dont know how to setup the attended transfer..
23:15.13Deeewayneyup.  a tiny old russian car full of potatoes
23:15.19Qwellare they like not very common there?
23:15.35DeeewayneI don't know
23:15.39Qwellhow odd
23:15.40garymcis it possible to do via net or putty etc?
23:15.57p3nguinHow do you normally admin the phones?
23:16.07garymcat the office
23:16.15p3nguinThat's not HOW.
23:16.24p3nguinThat is WHERE.
23:16.25garymci log into them at the office with their ip
23:16.28p3nguinHOW
23:16.32garymcon the internet browser
23:16.36garymc;)
23:16.42p3nguinokay, now we're getting somewhere.
23:16.43*** join/#asterisk fofware (~chatzilla@190.7.25.160)
23:16.46garymc:)
23:16.51Qwelloh, man, and the breading stuff with lemon juice and chili..  I forget what that was called :(
23:17.03*** join/#asterisk hachi (hachi@shego.kuiki.net)
23:17.04Qwellwow.  I miss living in the ghetto.  wtf?
23:17.10garymcis it possible to do ? like quickly?
23:17.14dlynes_laptopgarymc, install squid to bind to localhost on the server, then do ssh -L 3128:localhost:3128 serveraddress, and then fire up firefox, and tell it use a proxy of 127.0.0.1:3128
23:17.16p3nguinYou can use ssh (use PuTTY) to create a tunnel (socks proxy) from your home computer to your server in the office.  It will be similar to a VPN.
23:17.24p3nguingarymc: ^^
23:17.40garymcdo you know what command i need to use?
23:17.48dlynes_laptopgarymc, i just gave it to you up above
23:18.16garymcfuk it hell have to wait :S
23:18.21garymche will*
23:18.23dlynes_laptopgarymc, you need to have squid installed, however...I suggested binding to localhost though, so you're not running it as an open proxy server
23:18.24p3nguinThat's a lot of unnecessary work.
23:18.26p3nguingarymc: Just wait.
23:18.38p3nguingarymc: You don't need squid.  Just wait while I get the putty command.
23:18.43garymcthanks
23:18.51dlynes_laptopp3nguin, oh...there's another way?
23:19.04hachiI used to use voip-info (a while ago) for all my asterisk dialplan command documentation needs, is there a new source of this information? because the http://www.voip-info.org/wiki/view/Asterisk+cmd+Record page isn't even showing the comma syntax that is apparently what asterisk wants
23:19.25jayteehachi, what version of *?
23:19.38dlynes_laptophachi, replace all pipe (|) symbols with commas (,)
23:19.46hachiyes, but the docs don't even show it
23:19.53hachi1.6.2.0, from debian... laziness
23:20.05hachiI'm looking for the docs on the 'options' field, and things like that
23:20.22jayteebest docs are on the CLI, type core show application dial (or another application)
23:20.23dlynes_laptophachi, Record(filename:format,silence,maxduration,option)
23:20.24*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
23:20.37jayteeso core show application record
23:20.40hachiyes, thank you, but the docs also say .format
23:20.49p3nguingarymc: PuTTY.EXE -C -D 443 -P 22 -l gary -ssh -N server.domainname.com
23:20.53dlynes_laptophachi, eg Record(myrecording:WAV,10,120,${OPTIONS})
23:21.02p3nguingarymc: That will create a socks proxy to your server.
23:21.18garymcok then what?@
23:21.27p3nguingarymc: Then you need to open your browser's proxy settings and set SOCKS proxy to localhost port 443.
23:21.52p3nguingarymc: Now you have a proxy that takes you into the server and LAN in the office.
23:21.53garymccommand not found
23:22.07dlynes_laptopgarymc, are you on linux, or windows?
23:22.09p3nguinYou need to get putty, obviously.
23:22.17garymcim on a windows laptop
23:22.25garymcthe server is linux
23:22.43garymcbash -c not found
23:22.50garymcbash -C not found
23:22.58p3nguinWhat are you doing?
23:23.07p3nguinOn your Windows client....
23:23.13garymcim in putty
23:23.13dlynes_laptopp3nguin, he's trying ot run your command on the linux server
23:23.14p3nguinopen CMD.EXE
23:23.19hachiwhat can I use to record and keep the data if the user hangs up?
23:23.19garymcahh ok
23:23.43hachiRecord() actually says in the internal docs that all data will be lost, so this is what I experience, but I would like to record till the hangup
23:23.47p3nguinUse the path\to\putty.exe with the options I gave you.
23:25.24p3nguinI use a bat file on my USB flash drive so I don't ever have to type the commands.
23:25.43p3nguinI also use pageant so I don't have to enter my password.
23:26.58p3nguin"Program Files\PuTTY\PAGEANT.EXE" "Program Files\PuTTY\myserver.com.ppk" -c "Program Files\PuTTY\PuTTY.EXE" -C -D 443 -P 22 -l rob -ssh ns1.myserver.com
23:27.24p3nguinThe more things I need to tunnel to the server, the more -D <port>s I put in.
23:27.33garymcok that opened putty
23:27.33p3nguinWant to tunnel xmpp, add in -D 5223.
23:27.55p3nguingarymc: Good.  Leave it alone.  That is your client side of the tunnel.
23:28.04p3nguingarymc: Now adjust the browser's proxy settings.
23:28.24p3nguingarymc: SOCKS type, localhost, port 443
23:28.46p3nguingarymc: Then you can put the phone's IP address into the browser just like if you were at work.
23:28.52mmj_nix<PROTECTED>
23:29.01garymcok just need to find proxy settings
23:29.08p3nguingarymc: Which browser?
23:29.15garymcfirefox
23:29.35p3nguinWhat's the menu next to Help?
23:29.38p3nguinTools?
23:30.02garymcyep
23:30.11p3nguinIt's the bottom selection on that menu.
23:30.43p3nguinI can't remember which item in the settings, but it should be like the second from the right side.
23:30.50p3nguinThere is a tab for Networking.
23:31.26garymcok in options
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23:31.40*** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002)
23:31.46mattwj2002well guys I did it
23:31.57garymcmanual prox settings or auto?
23:32.02p3nguinmanual
23:32.07mattwj2002I order a Cisco 7960
23:32.14mattwj2002*ordered
23:32.24p3nguingarymc: Leave all the boxes empty except for the socks one.
23:32.32niekvlessertcu guys
23:33.06garymcBINGO LAAAA!!!!
23:34.17p3nguinopenssh's sshd is a pretty good socks proxy.  It's great for browsing securely through your server at home or work when you are in the opposite place.
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23:35.36garymccool
23:36.05garymcafter all that though I cant do what i need to through that. Ill have to do it in the phone lol but hey what a tool for future usage :P
23:36.17p3nguinheh
23:36.28p3nguinBut you did reach the phone's web interface, right?
23:37.07mattwj2002anyone want to configure my asterisk server for my new Cisco 7960 phone?
23:37.23blackwhy wouldyou buy a cisco 7960?
23:37.24p3nguinHow much does the job pay?
23:37.24black~phones
23:37.25infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones.  Do not consider Cisco phones.  Ever.
23:37.26black~phone
23:37.26infobotphone is, like, warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha.
23:37.34black"Do not consider Cisco phones.  Ever."
23:37.42black^
23:37.47QwellWho changed that?
23:37.51p3nguinWhy would someone write that?
23:37.58mattwj2002because I like Cisco
23:38.02p3nguinqwell: That's what I was wondering.
23:38.04blackThen use a cisco pbx
23:38.05black=p
23:38.10mattwj2002lol
23:38.12mattwj2002:P
23:38.16jaytee~gs
23:38.17infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
23:38.28jaytee~grandstream
23:38.29infobotfrom memory, grandstream is the Yugo of VoIP hardware.  Run.  Run away now..  Though therealcircut says that they're not that bad
23:38.31p3nguinThat's fairly recent.
23:38.47p3nguinmattwj2002: So how much was the job paying?
23:38.51jayteethe Cisco one is new to me but the other two have been around for awhile
23:39.00Kobazwhat's crazy about grandstream... is that the speakerphone is higher quality than the handset
23:39.05mattwj2002why do you ask?
23:39.07Kobazthe handsets are all staticy and crackly
23:39.16Kobazdoesnt happen with the speakerphone
23:39.29p3nguinjaytee: Someone recently added that statement about Cisco.
23:39.42Qwellp3nguin: it's gone
23:39.42garymcThanks P3nguin what a star :P
23:39.47garymcgood night
23:40.30mattwj2002p3nguin: what do you ask?
23:40.45jayteegiven Cisco's attitude about licensing I kind of agree with that statement
23:43.00dlynes_laptopp3nguin, your solution..how do you get it working with firefox?
23:43.25dlynes_laptopp3nguin, firefox doesn't give me the option to tell it to use socks, specifically
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23:45.40hachithanks folks, got this all sorted now
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23:48.51mattwj2002hey guys
23:48.54mattwj2002I have a question
23:50.02dlynes_laptopp3nguin, nvm...figured it out...thanks for the tip
23:50.33mattwj2002never mind I got it
23:50.47mattwj2002well supper time
23:50.49mattwj2002bye all!
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23:58.00bmoraca_workOM NOM NOM!
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