IRC log for #asterisk on 20110124

00:10.54*** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com)
00:20.16*** join/#asterisk sourcode (~code@ppp-58-8-43-253.revip2.asianet.co.th)
00:31.27*** join/#asterisk Cain (~Geek@unaffiliated/cain)
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01:01.09drmessanoSo Skype is planning to charge for even USING the Skype Manager
01:01.42drmessanoand since I can only use clients in SFA that have been set up in Skype Manager
01:02.07drmessanoI am going to need to pay a monthly fee to use something I thought I had already paid for..
01:03.28russellb<PROTECTED>
01:03.55drmessanoApparently so
01:06.09drmessanoSounds like they took Digium for a ride
01:06.24drmessanoand the rest of us
01:06.27drmessanoBastards
01:07.58drmessanoWasn't there some old story about making a fair maiden weave baskets to sell, and then an evil witch comes along and enslaves little children in the baskets only to sell them to the land of evil witches
01:08.06drmessanoIf there isn't, THIS is that story
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01:19.16tymanI have some strange behavior with fresh install of fop2.  I have 4 buttons configured defining different registered sip extensions. When i call any of them only my logged in extension rings.  Here's my button.cfg: http://pastebin.com/ijSesq7d
01:19.33tymanAny ideas?
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02:16.52maxagazhi
02:17.22maxagazhow to see the SDP for a RTP stream ?
03:01.08*** join/#asterisk Bidik (~bidik@li267-109.members.linode.com)
03:02.28Bidikany idea why chan-mobile says mobile is ready but on incoming calls i see no reaction from asterisk at all and also when i try discovery of services from mobile phone it says no local services on centos 5.x /
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03:19.44pabelangermaxagaz: *CLI> sip set debug on
03:21.26maxagazpabelanger: thanks!
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03:33.20grkbloodmy MOH isnt working in the queue, i keep getting [Jan 23 22:26:33] DEBUG[4827] res_musiconhold.c: Read 620 bytes of audio while expecting 640
03:33.22grkbloodis this a bug?
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05:45.49sarthorHi. How can i dowload the "Get Started Video" on main http://asterisk.org page,  is the possible to download?
05:47.52*** join/#asterisk pratat (7c9bc453@gateway/web/freenode/ip.124.155.196.83)
05:48.21pratatHi Anyone ard to help ?
05:49.10pratati am having problem with channel not able to hang up after VM Messages have been heard
05:49.12pratat:(
05:53.12ChannelZno
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06:16.21pratatermm
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06:26.01i_heart_asteriskhello
06:26.08i_heart_asterisk~
06:40.53ChannelZo hell
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07:01.43*** join/#asterisk pratat (7c9bc453@gateway/web/freenode/ip.124.155.196.83)
07:02.24pratati am trying to solve DAHDI not detecting caller hangup
07:02.29pratatany one can shed some light ?
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07:20.39shaprpratat: busydetect?
07:21.05pratatDone
07:21.13pratatstill not working
07:21.21pratatdefine under [channels]
07:22.44*** join/#asterisk hehol (~hehol@2001:1438:1009:200:84c8:4053:ee31:fa20)
07:25.30ChannelZwhat kind of dahdi channels
07:29.58*** join/#asterisk DJClean (~djclean@unaffiliated/djclean)
07:32.30maxagazwhen I make a sip phone call between my computer and a sip phone, I can see on my computer that packets are always either sent from the server to my computer or from my computer to the phone. Why are packets sent from the server to my computer, why not straight from the phone to my computer ?
07:33.38ChannelZit depends on your config
07:33.54pratatChannelZ: analog
07:34.05ChannelZThe two end points might not be able to talk to each other, or your Asterisk config is such that you've told it to be in the middle
07:34.52ChannelZpratat: what country are you in?  Sounds like maybe your DAHDI config isn't quite right for the signalling being used
07:38.01pratatChannelZ: i am from Singapore
07:38.21pratatany hints where i should be looking at ?
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07:41.32maxagazChannelZ: my Asterisk config is the default one for Asterisk 1.8.2
07:42.10*** join/#asterisk ndemir (~ndemir@85.105.28.77)
07:42.29ChannelZpratat: sorry no idea what is proper for Singapore.
07:42.59ndemirhello. i am creating a call file with CallerID option. But in my phone, just the numbers are seen. Is there a way to show name?
07:43.13ndemirI mean i want to use alphabetic characters not just numbers.
07:43.31pratatChannelZ: any chance that u know for UK ?
07:43.41pratati believe we are following th UK standards
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07:46.54ndemiris there a way to solve my problem?
07:47.49ChannelZndemir: what is your phone?
07:48.49ndemirandroid samsung. this is not about my phone. i want to say that i want to use alphabetic characters in my call id as callerid.
07:49.05ndemir<PROTECTED>
07:49.31ChannelZpratat: in general it's probably kewlstart but it sounds like your telco maybe has some other wacky signalling maybe.
07:49.36ChannelZhttp://www.voip-info.org/wiki/view/Asterisk+Disconnect+Supervision
07:50.22ChannelZndemir: You are calling a cell phone from Asterisk?
07:50.38ndemir<PROTECTED>
07:51.05ChannelZThe only caller ID you get is the number.  That's just the way it is.
07:51.35maxagazI have unplugged my sip phone but "ip show peers" still shows its ip, why ?
07:52.04ChannelZAt least in the US, no idea if other cell networks are capable of transmitting CNAM
07:52.41ChannelZmaxagaz: has it timed-out yet?
07:53.29ChannelZThe device isn't constantly connected to *, so depending on your settings * might not notice for a time
07:58.13pratatok i got it working to hang up
07:58.27pratatbut still recording the 3 busy tones .. before it hang up
07:59.13ChannelZusing callprogress?
08:01.03pratatermm
08:01.09pratathanguponpolarityswitch=yes
08:01.17pratatin the chan_dahdi
08:02.11ChannelZah
08:02.59pratathmm but how to stop the VM from recording the 3 busy tones ?
08:04.47ChannelZyou can tweak the maxsilence assuming there is a delay between when the person hangs up and you get the tones
08:06.17pratatif  u dont mind , how does the maxsilence works ?
08:06.42ChannelZsee voicemail.conf
08:07.04ChannelZif asterisk hears silence for so many seconds it will automatically end the voicemail record
08:08.45pratatok
08:08.54pratati have the default 5 sec there
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08:26.19maxagazsRTP seems to work on my PBX as I get this [m=audio 5062 RTP/SAVP 0 3 8 101] in the SDP, but I get this error message: SRTP unprotect: authentication failure
08:27.23maxagazalso, when I check the bearer packets, they all seem to be the same in TLS or sRTP, is this normal ?
08:27.30pratatdamn ...
08:27.39pratatthe VM still recording the hang up tone
08:27.41pratat:(
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08:41.56*** join/#asterisk Dovid (~Dovid@213.8.118.62)
08:42.08Dovidis there any way to have asterisk re-invite a call ?
08:42.33Dovidor any other software for that matter to send a re-invite some time in to a call ?
08:44.02*** join/#asterisk krion (~seb@unaffiliated/krion)
08:45.08krionhi
08:45.28krionis a core show locks enough in order to open a bug ?
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08:55.54*** join/#asterisk tamiel (~tamiel@213.30.183.226)
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09:07.36donttrustemHi guy's I no this is not the correct channel but I have a sysmaster switch and it is de-registering clients when it hits 100 connections
09:24.24shaprnever heard of it
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09:33.38maxagazdoes someone know how to create audio file from packets with tcpdump ?
09:35.37krionmaxagaz: should give a try to wireshark
09:36.08maxagazkrion: I'd prefer to do it with tcpdump if it's possible
09:36.51WIMPytcpick
09:39.49maxagazkrion: I just tried wireshark, it works like a charm indeed
09:40.31*** join/#asterisk krokus (~blackdeat@Geometry-1.vmb-service.ru)
09:42.19tamielmaxagaz: use tcpdump with "-s0 -w packets.bin"  and load packets.bin with wireshark
09:43.54maxagaztamiel: okay, I'll try this
09:43.58maxagazthanks all
09:45.11*** join/#asterisk joachim_- (~joachim@post.comvie.no)
09:45.23joachim_-Morning all.
09:45.46joachim_-What can I use for SQL database replication in an Asterisk cluster?
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09:47.37kaiimaxagaz, tamiel: better name the file "foo.pcap" cuz its the pcap format
09:47.57Dovidis there any way to have asterisk re-invite a call ?
09:47.59Dovidor any other software for that matter to send a re-invite some time in to a call ?
09:48.34krokusHi. Can I hold all calls in Mysql and operate them in real time?
09:48.56Dovidkrokus: What do you mean by hold them in MySQL ?
09:49.27hrhrhrgood morning chaps
09:49.30krokusTurn of calls
09:49.56joachim_-What can I use for SQL database replication in an Asterisk cluster? Just need a point in the right direction
09:50.13krokusPut in mysql and operate in real time
09:50.53krokusnot for joachim_-
09:51.28Dovidjoachim_-: Use OpenSipS with load balancing
09:51.40Dovidkrokus: Yes. you can use real time. whats the question ?
09:54.08joachim_-Dovid: I allready got heartbeat setup. only lack the sql replication
09:54.16krokusDovid, Need use asterisk with crm+mysql, operator should have possibility to receive instant notices on calls and to have their possibility redirect
09:55.04Dovidjoachim_-: For that you need to ask in #mysql how to do mysql replication
09:55.24Dovidkrokus: That should have no affect on Asterisk
09:57.32krokusDovid: ok
09:59.15maxagazwhen I use TLS, wireshark can't detect voip calls anymore
09:59.33Dovidmaxagz: Of course. It's encypted !!!
09:59.40WIMPyThat's the idea
09:59.49maxagazDovid: only the signal is encrypted
09:59.57maxagaznot the bearer
10:00.02Dovidumean the rtp ?
10:00.14Dovidu mean*(
10:00.17Dovidmean**
10:00.40WIMPyYes, but there is no "announcement" of the rtp comming.
10:00.45krokusjoachim_-:  http://www.howtoforge.com/mysql_master_master_replication
10:00.52WIMPySo it's just not automatic.
10:01.19joachim_-krokus: Will have a look at that. Thank u very much!
10:01.29maxagazWIMPy: so, I should use another way than Wireshark to recreate the audio file, right ?
10:02.30WIMPymaxagaz: That sould work just as well, but you will probably have to use a few more clicks.
10:03.04krokusjoachim_-: you are welcome
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10:05.39maxagazWIMPy: I tried with "Telephony" > "VoIP Calls" in Wireshark, what else can I try ?
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10:16.21eMBeegood evening
10:17.14*** join/#asterisk mahiti-irc (~mahiti@122.166.127.74)
10:17.17*** part/#asterisk dlirit (~lirant@80.74.100.10)
10:17.17mahiti-irchi
10:17.32Dovidjoachim_-: You can have a look here to get an idea: http://integrics.com/products/itsp/guides/latest/en/field/install/mysql/replication/
10:17.43Dovidit is for their specific software but it can help over all
10:17.52Dovidmahiti-irc: Hi there
10:18.29mahiti-irci am having asterisk 1.4.38 configured and a PRI with 32 channels
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10:18.45Dovidyou mean 31
10:18.49mahiti-irchow do i define the spanmap in the chan_dahdi.conf??
10:18.55mahiti-ircyup Dovid
10:18.59eMBeeis ordering an ISDN line, and the phone company is asking wether i want specific features on that line. i am not sure what to do with "routing-on-demand"
10:19.21DovideMBee: Ask them for an explination
10:19.30Dovidmahiti-irc: Have a look at the default.
10:19.41Dovidif you give me your details I can try to get a basic config
10:19.58mahiti-ircwhat details?
10:20.13mahiti-irci tried to understand the default
10:20.20mahiti-ircbut confuses me more
10:20.20Dovidmahiti-irc: Do you know what channel the D-Chan is on ?
10:20.24mahiti-ircyup
10:20.25mahiti-irc16
10:20.52WIMPymahiti-irc: spanmap is only used for NFAS.
10:21.10mahiti-irc??
10:21.33WIMPymahiti-irc: If you only have one line, you don't need it.
10:21.54mahiti-ircnot for PRI?
10:22.04WIMPyIt's only used to make multiple lines use a single signalling channel.
10:22.19WIMPyNot for only one.
10:22.38eMBeewhat about "direct dialing in" sounds like making a line connect directly to another number.
10:22.51mahiti-ircu mean mulitple PRI lines connected to single asterisk?
10:23.19mahiti-ircWIMPy, ^^
10:23.28WIMPyeMBee: That's the opposite of MSNs. i.e. a block of numbers vs multiple single numbers.
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10:24.04WIMPymahiti-irc: No I mean only one line. You only need that if you have a trunk that spans multiple lines.
10:24.39mahiti-ircsry WIMPy , i dont get you?
10:25.35WIMPySo what's the question then?
10:27.14mahiti-ircWIMPy, we had configured our PRI line with asterisk 1.4
10:27.20mahiti-ircit works superb
10:27.58mahiti-ircbut at times we get a yellow alarm and a PRI got event: Bad FCS on primary channel
10:28.43mahiti-ircwe googled and saw tht the issue is due to wrong framing mode in zaptel.conf
10:28.58mahiti-ircwhich i think has become chan_dahdi.conf
10:29.19WIMPyYes, and the contents haven't changed much.
10:29.58mahiti-ircso i was going thru chan-dahdi
10:30.46mahiti-ircwhere it seemed to me that i have not yet configured the spanmap and trunkgroup in the chan_dahdi.conf file
10:32.34WIMPyYou have _one_ PRI?
10:32.34mahiti-ircyup
10:32.34WIMPyThen it's back to the beginning: You don't configure them.
10:32.34eMBeeWIMPy: oh: the form is asking for how many blocks i want and the  DDI Range#:
10:32.34WIMPyThose options only make sense for multiple interfaces.
10:32.34eMBeewell, i guess i want one block of 10 numbers
10:32.55WIMPyeMBee: There may be rules allpied by the telecoms regulator on how any numers you can get.
10:33.25eMBeeor actually, do i need that? i eman asterisk can handle extensions, so one number should be enough, and i'd just use the extensions to distinguish calls
10:33.31WIMPyA block of 10 should always be available.
10:33.42eMBeeyes, is charged extra though
10:33.56mahiti-ircWIMPy, one more thing, we spoke with our PRI provider, and he has asked to disable
10:33.57WIMPyThat ARE the extensions as far as calls from that line are concerned.
10:34.08mahiti-ircasked to disable CRC
10:34.33WIMPymahiti-irc: Then remove the ",crc4"
10:34.44mahiti-ircWIMPy, ok cool
10:35.16eMBeeyes, but i mean if i have one base number 12345678 and then configure my local extensions 11 to 99, then people just dial 10 digits to get to each extension, right? so what do i need a block of 10 numbers?
10:37.37mahiti-ircWIMPy, is there way to check if CRC is on or off??
10:37.37WIMPyeMBee: That sounds like a block of 100 (00-99).
10:37.37WIMPymahiti-irc: Not that I know.
10:37.37mahiti-ircWIMPy, ok thx :)
10:38.03eMBeehmm
10:38.21WIMPyeMBee: Depending on your operator youmay be able to actually make the numbers longer yourself. They usually don't care.
10:38.32eMBeethat is what i mean
10:38.43WIMPyYou just end up with longer numbers.
10:38.48eMBeeexactly
10:39.26eMBeeok, so i guess i'll take one block yo be sure, just in case i can not make the numbers longer
10:39.34eMBeethanks
10:40.07eMBeehmm, the form asks for a range, do i put 0-9?
10:40.48WIMPyYou can probably order a shared block, like 00-59 for 60 numbers.
10:41.38WIMPyAs I said before there may be rules on how many numbers you can get depending on the number of channels and/or physical exensions you've got.
10:43.37eMBeewell, their offer says 1 block of 10 numbers costs $10, so i am guessing if i say i want one block i get 10 numbers, so i am confused why they would need me to specify a range on top of that, unless it is only needed if i want more than one block or if i can say something like give me 10-19 or whatever...
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10:44.33WIMPy10-19 doesn't make sens that would be the base number +"1" and a range from 0-9.
10:45.14WIMPys/ns t/nse. T/
10:46.44*** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor)
10:48.42eMBeewell exactly, which is why i am wondering why they are asking for a range
10:49.26WIMPyIf you need a block size that is not a power of 10.
10:50.21WIMPyHere the default block for a PRI would be 000-599.
10:53.43*** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl)
10:53.45niekvlessertquestion: how do i disable the moh completely? So that the RTP is not on Asterisk when putting the call on hold
10:54.29eMBeeWIMPy: yeah, that makes sense. i'll just leave that empty
11:18.56*** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry)
11:20.16ghenryDoes anyone know of any implementations for billing that has heartbeat? i.e. call credit increments every 60 secs not at the end of a call?
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11:23.16ghenrydecrements even
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11:49.48maxagazWhen I make phonecall using sRTP from my sip phone to PhonerLite on windows, it works fine, but when I do the contrary, I get the following error message: We are requesting SRTP, but they responded without it!
11:49.56maxagazWhat's wrong with my config ?
11:50.35shaprSo, what's specific to a non-PRI T1? Does that only mean it uses CAS (channel associated signalling)?
11:58.30maxagazit seems to be a problem with auth tag
11:58.40maxagazcan someone help me about this ?
12:00.21maxagazhow to set auth-tag to 80 ?
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12:17.02maxagazI found the problem
12:17.27maxagazit had to check SAVP in PhonerLite
12:17.34maxagazI thought it came from the sip phone
12:18.55maxagazmy problem now is that I don't see any difference when I'm using srtp or not over tls
12:19.43maxagazthe Data looks the same, something begining with 800... whatever it's in srtp or not
12:20.44maxagazand I still get this error ;essage: SRTP unprotect: authentication failure
12:20.54maxagazmessage
12:22.05maxagazalso, my tls certificate is self-signed, and both the key and the certificate are only on the server, could it be related ?
12:23.08leifmadsenwell TLS is just the signalling
12:23.37leifmadsenhowever,  I haven't setup SRTP and TLS signalling before. There is some stuff documented here though....
12:23.41leifmadsen~newbook
12:23.41infobotHelp review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/  Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/
12:23.42maxagazleifmadsen: yes, so it shouldn't be related
12:23.48leifmadsenright
12:25.23maxagazleifmadsen: but, I don't understand how is used the key and certificate, it's only on the server side, so how can the client decrypt the signaling content ?
12:26.35maxagazit seems that asterisk is encoding the crypting the signaling and telling the client how to decrypt it
12:26.59*** join/#asterisk dacm_work (~dan@host109-156-61-116.range109-156.btcentralplus.com)
12:27.02dacm_workHi guys
12:27.10maxagaz(sorry, remove "encoding the")
12:27.37dacm_workCan anyone recommend a good windows softphone? (Preferably SIP, that supports call transfer.)
12:31.21Tozz_x-lite?
12:33.33mzaharievzoiper
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12:38.46mzaharievhttp://icanblink.com/
12:39.17mzaharievvery nice :) i use it under macos
12:49.16FlashDeluxeHi! I am using asterisk 1.8.1 with dahdi 2.4.0 and i have several problems, e.g. sometimes it appears, that a call is interrupted after dialing the number and i am not quite sure what it could be. Here`s a log of a call which produced this error: http://pastebin.de/14133  Furthermore i have an error which appears quite often an one-way-audio problem, the called person can hear me, but i cannot hear him, for 5 seconds, after this
12:49.17FlashDeluxeperiod i can continue the call without problems. This error often occurs after putting the called party to MOH but it appears also during a call which is not transferred in any way, can somebody help me please?
12:52.43*** join/#asterisk Faustov (user@gentoo/user/faustov)
12:52.57FlashDeluxeI saw different posts about that one way audio problem, does anybody know if there`s a general fix or does anybody know the cause for that error?
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13:21.07FlashDeluxeany suggestions? I would appreciate any hints :)
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13:34.05DovidFlashDeluxe: Any NAT involved ? did you try rtp debug ?
13:36.23FlashDeluxeDovid: every outgoing call is going from sip telephones to asterisk and from there over isdn to the call party at the other end, so there shouldn`t be anything nat-ted, no i didn`t try it
13:36.35FlashDeluxebut i will trun it on now
13:37.40DovidFlashDeluxe: Wasn't sure of your set up which is why i asked about the NAT.
13:37.43FlashDeluxebut it is very very much, on what do i have to have a closer look?
13:37.58Dovidalso i doubt this has to do with it but maybe there is a timing issue
13:38.20FlashDeluxeDovid: how can i find out?
13:38.32DovidFlashDeluxe: rtp set debug on
13:38.37Dovidthats for rtp debug
13:38.49Dovidfor timing you need to look at your set up
13:38.51FlashDeluxeDovid, i already did that, a loooot of output
13:38.59Dovidi would start with rtp debug and go from there
13:39.59FlashDeluxeDovid: ok thanks, how can i find out if theres a problem? its very much, are there any words i can grep for?
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13:42.30dacm_workTozz_: X-lite seemed really buggy when I tried it.
13:43.32dacm_workAnyway, thanks for the suggestions guys.
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14:04.49Tozz_dacm_work: X-Lite works fine here
14:04.55Tozz_whats buggy about it?
14:06.12arekmI have two sip devices (gateways) with call-limit=1. How to make something like single channel from these two but with call-limit=2? So I call, it goes out via first available device
14:07.08arekmUsing two Dial() with 1 and 2 priority doesn't work well. If first gets busy (from cellphone for example) it will still try to dial via second device
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14:09.56bbryantdoes anyone know where to find the postgre driver for odbc on ubuntu? it's not located at /usr/lib/libodbcpsql.so and the only so I can find is libodbcpsqlS.so
14:10.29bbryantafter installing odbc-postgresql that is
14:10.49*** join/#asterisk ihor (~Miranda@194.44.15.90)
14:12.34russellbbbryant: try looking up the contents of the package on packages.ubuntu.com
14:13.07wdoekes2dpkg -L ?
14:13.26russellbor that, heh.
14:14.34bbryantrussellb: thanks
14:14.40bbryantI was able to find a connection template that way
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14:18.57pecenipicekquestion, how can i specify the subnet for a dundi or iax peer that i wish to connect to?
14:19.45russellbwhy do you need to specify that?
14:20.33pecenipicekmatters not, now does it?
14:20.48pecenipicekvpn setup with iax2 trunk if it does.
14:21.11russellbwell i asked because you shouldn't have to specify that anywhere
14:21.49russellbif it's a matter of getting the traffic to go out the vpn interface, that's a routing table issue
14:22.18pecenipicekits a matter of getting the boxes to communicate via iax and dundi.
14:22.25pecenipiceki can ssh into the remote box without problem.
14:22.40pecenipicekits getting the iax trunk through thats the primary problem.
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14:24.31Tozz_arekm: You could fix that with a Goto ${DIALSTATUS}
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14:32.36dacm_workTozz_: The GUI in general. I'd click on something and it wouldn't register. (This is on XP.)
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15:01.04bipHello, I have just installed a asterisk server. Users complains about dropped calls, what can I do to investigate if the fault can be corrected from our side or if it is a telephone company issue ?
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15:24.56leifmadsenmaxagaz: Might want to check this page too:  https://wiki.asterisk.org/wiki/display/~mdavenport/So+you'd+like+to+make+some+secure+calls
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15:26.22pifhi, where do I defined the serveremail when using users.conf for vm?
15:27.19leifmadsenI don't think you do
15:27.27leifmadsenprobably voicemail.conf still
15:28.10pifok, and can I control the envelope FROM when asterisk sends email?
15:28.28pifI'd like to be something else that asterisk@..
15:32.25leifmadsenthat is done in the [general] section of voicemail.conf
15:32.31leifmadsencheck voicemail.conf.sample
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15:33.07leifmadsen(or this page:  http://ofps.oreilly.com/titles/9780596517342/ch08.html#Voicemail_id272814)
15:33.23pifthanks
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15:41.07WIMPyFlashDeluxe: I don't see anything wrong in your trace. Looks like the other end hung up on you after 2s.
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16:09.32bmintI am working on a project that uses .call files to autodial our clients and play a message.   I would like to set multiple variables within the call file.  I was able to set one variable by adding the line Set: variable=value but can't determine the syntax for setting two variables.  Does anyone know how to accomplish this?
16:12.12beekAdd another lnie.
16:12.16beeks/lnie/line/
16:14.06bmintbeek: Simple enough.  It is working.  Seems like I tried this before but the call did not go through.  Thank you.
16:14.20beekbmint:  you're welcome.
16:16.13*** join/#asterisk ChannelZ (channelz@burner.com)
16:19.55*** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
16:19.59drift-i have a linode.com shared server, id like to setup freepbx+google voice , all i seem to see is .iso for centos 5.5 install and freepbx any one know of any tutorials out there to set it up on virtual server? freepbx + google voice , having hard time finding one
16:23.54ChannelZDoes linode give you a means to upload an install source to create your VM from, or do they just want a frozen Xen image?
16:24.17*** join/#asterisk guilhermebr (~Guilherme@200.103.96.98)
16:25.06LinuturkI've noticed that the most recent firmware released by polycom puts the default config files in Config
16:26.13Linuturkdoes the firmware automatically look on <ftproot>/Config when I define values in the config_files="" in the <mac>.cfg file, or do I have to spell it out? ie. config_files="phone1.cfg" vs config_files="/Config/phone1.cfg"
16:27.00drift-chanelnelz, they give you list you choose from
16:28.33ChannelZpick a distro you like and just build Asterisk on it
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16:31.36WIMPyThis is getting serious here. I've got a box that has piled up over 2000 dead SIP dialogs since this morning which is probably the cuse for it locking up sometimes.
16:32.15WIMPyI've had the issue here as well, but it went away after an update. The same update didn;t help on the other box. The dialogs all look like:
16:32.22WIMPy192.168.12.145   (None)           3c365c9dac21-wp  0x0 (nothing)    No       Rx: PUBLISH                <guest>
16:32.26leifmadsenWIMPy: https://issues.asterisk.org/view.php?id=18657  <-- maybe related?
16:33.25WIMPyDoesn't match the PUBLISH, but I think it happens on a per call basis.
16:36.50drift-does asterisk 1.8 come with a gui?
16:38.39*** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23)
16:39.00leifmadsendrift-: no, but you can install a GUI after installing 1.8 (or better yet, install AsteriskNOW and get both at the same time)
16:39.50asteriskATmarmuDI got some problems with dahdi. if I type "dahdi show status" on the CLI, I get "No such command 'dahdi show status'"
16:40.09asteriskATmarmuDbut dahdi is installed correctly (dahdi_tool shows configured devices)
16:40.23leifmadsenasteriskATmarmuD: that means you didn't install chan_dahdi
16:40.32WIMPyasteriskATmarmuD: Then you have problems WITHOUT dahdi.
16:40.34leifmadsenasteriskATmarmuD: you need to run ./configure and 'make install' after installing DAHDI
16:40.46asteriskATmarmuDdid that
16:41.00leifmadsen'modules show like dahdi'
16:41.20leifmadsenI'd start asterisk in the foreground and see what errors and warnings oyu get
16:41.29WIMPyTurn up verbose and debug and try module load chan_dahdi.
16:41.42asteriskATmarmuDdoes /etc/asterisk/modules.conf with "load => chan_dahdi.so" help, I inderted that line, didn't help
16:42.03WIMPyThat should tell you why it doesn't load if it's there.
16:43.49asteriskATmarmuDmodule show like dahdi --> chan_dahdi.so, app_dahdibarge.so, app_dahdiras.so, app_dahdiscan.so, codec_dahdi.so
16:45.04asteriskATmarmuDok, module load chan_dahdi gives me errors
16:45.23asteriskATmarmuDsome are "ok", since I can't connect the E1 line right now
16:45.51asteriskATmarmuDcan't chan_dahdi be loaded at all, if the E1 line is not connected?
16:45.56WIMPyThat wouldn't prevent it from loading.
16:46.13WIMPyIt must be due to comfiguration errors.
16:46.29asteriskATmarmuDI get warnings because of the "red" status (line not connected)
16:47.12asteriskATmarmuDok found some interesting error
16:47.12asteriskATmarmuD<PROTECTED>
16:47.39asteriskATmarmuDany hints? I will look for them myself. but If you got a key word, feel free to tell me
16:48.07*** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
16:48.25WIMPyYour dahdi/system.conf and your asterisk/chan_dahdi.conf mismatch.
16:49.05asteriskATmarmuDok, can't explain where and why they mismatch now, but will check and figure it out...
16:49.17asteriskATmarmuDthanks a lot for the quick help
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16:57.29drift-hrm
16:57.34drift-this asterisk now thing is only in .iso format
16:58.11drift-how do i mount it on centos 5.5? AsteriskNOW-1.7.1-i386.iso
16:58.50asteriskATmarmuDdrift-: I assume you don't want to burn and install it?
17:00.09asteriskATmarmuD<PROTECTED>
17:02.16drift-ah
17:02.18drift-i cant do that
17:02.30drift-i have a centos 5.5 virtual server allready installed with it and running cpanel
17:02.43drift-so install 1.8 from scratch? then install gui?
17:03.20asteriskATmarmuD<PROTECTED>
17:03.26drift-yeah installing that now
17:03.29drift-but how do i install gui?
17:03.42asteriskATmarmuD<PROTECTED>
17:03.55drift-i dunno i just want to connect to like pbx.mydomain.com
17:03.58drift-and see the gui interface
17:04.11asteriskATmarmuD<PROTECTED>
17:04.18drift-compiling 1.8.2.2
17:06.10asteriskATmarmuD<PROTECTED>
17:06.25drift-heh i got asteirk installed now hrm
17:07.40asteriskATmarmuDI cant test my PRI-line until tomorrow. but chan_dahdi is now loaded... all good :)
17:08.25asteriskATmarmuD<PROTECTED>
17:08.47drift-i want to get a phone working with google voice
17:08.53drift-free in and out calls
17:09.44asteriskATmarmuD<PROTECTED>
17:09.57asteriskATmarmuD<PROTECTED>
17:10.24drift-yeah
17:10.30drift-i have... i have it setup in the office here
17:12.02asteriskATmarmuD<PROTECTED>
17:12.48drift-heh hope so
17:14.42*** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23)
17:14.57*** part/#asterisk devilspgd (me@devilspgd.net)
17:18.05Bidikhi anyone here with expirience on chan-mobile ... after pairing phone shows no services on bluetooth but asterisk conectas and says phone is ready but no incoming calls get registered ...
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17:34.45*** join/#asterisk phr0zen (phr0zen@blk-224-135-210.eastlink.ca)
17:34.52phr0zenhello
17:35.08phr0zenso i've been here before about a server crash, nmi unknow code 30 and 21
17:35.14phr0zenanywho, i think i found out why
17:35.31phr0zenor the cause at least and was wondering if anyone else has seen this
17:36.03phr0zenbasically when i have queues setup to record calls, the system will crash.. so far with testing, turning off the recording, i do not see a crash
17:36.17phr0zencould this be due to disk i/o or something that cause the server to lock up?
17:36.44Qwelldepends on how many calls you're recording, but yes, probably
17:36.57*** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr)
17:36.58phr0zenroughly 25-30 calls being recorded at once
17:37.16Qwellthat shouldn't be too bad, unless you've got really slow disks or something
17:37.34phr0zenhowever, it did manage to get through 1200 or so calls one day without crashing, but now it will crash within 20 mins of enabling recording
17:38.11phr0zenrunning 2x seagate 400 gig sata drives in raid 1 with an adaptec 3805 controller
17:39.46phr0zenwhat i am curious about though is why it would lockup the system... shouldnt i just notice a slowdown as opposed to a full halt
17:43.42n3hxs-wkphr0zen, I am not sure of the reason, but how much memory does your system have?
17:44.03phr0zen4 Gb
17:44.28phr0zenintel sr2500
17:44.33citywokif you open top andl ook at the wait # how high is it?
17:44.51citywoki can record 150 calls on an old dual xeon w/ dual 76gb scsi drives
17:45.06citywoksomewhere closer to 200 * segfaulted
17:45.27phr0zeni dont even see a wait number in top
17:45.47citywokwa is i/o wait iirc
17:46.25phr0zennot seeing that
17:46.30citywok"wa -> iowait: Amount of time the CPU has been waiting for I/O to complete." -- if you are running out of IO due to the recording that number should creep up pretty high
17:47.13phr0zenyea when i run top, i dont see an iowait at all
17:47.51citywokwhat os?
17:48.35phr0zencentos 5.5 64 bit
17:49.13Qwellthere is definitely a wa column in that verison of top
17:49.16n3hxs-wkon the line that starts with Cpu(s)  x.x%wa
17:49.20citywoka quick google looks like top should have wa numbers
17:49.37phr0zenahh there is it
17:49.43phr0zen0.0%wa
17:49.48phr0zenwith no recording on
17:49.51n3hxs-wknow turn on recording.
17:49.53citywokyea try it under load :P
17:50.09n3hxs-wkthen duck when they yell at you for the crash.
17:50.16phr0zenyea not gonna do that
17:50.18phr0zenlol
17:50.23phr0zenimma get stabbed
17:50.27citywokhow many concurrent calls?
17:50.33phr0zen35 tops
17:50.41citywokhow much cpu power?
17:50.51phr0zenCpu(s):  5.5%us,  2.4%sy,  0.0%ni, 91.8%id,  0.1%wa,  0.0%hi,  0.2%si,  0.0%st
17:51.06citywoki mena what are the procs? lol
17:51.10phr0zenlol
17:51.12drift-hrm how can i do tcpdump to display 5066 port
17:51.23citywokman tcpdump
17:51.28citywokgoogle "man tcpdump"
17:52.09phr0zendual xeon 5130
17:52.41citywokso 4 cores. lol, that should do a lot more than 35 calls.
17:52.54phr0zenthat what i thought
17:52.55citywokdo you have it compiled with dont optimize &or debug threads?
17:53.07phr0zeni used a pbxinaflash
17:53.21citywokmy old ass dual 3.8 can handle 150-160 just fine, w/ recording
17:53.39phr0zeni also convert those recordings to mp3 (using lame)
17:53.47phr0zenmaybe that is causing the issue?
17:53.54citywokyea, just make sure you nice the lame encoder
17:54.22phr0zenperhaps i can set it to convert to mp3 overnight instead of on the fly
17:54.46phr0zenbut either way, 35 calls wont end at the same time, and wont convert at the same time
17:54.49citywokthat takes way more effort, so i'd try and figure out why it crashes at all
17:55.11phr0zeni've seen 2 crashes so far
17:55.18phr0zennmi unknown error code 30
17:55.28phr0zennmi unknown error code 21
17:56.07phr0zenthe 21 happenend today, the 30's havnet happend in a 3-4 days
17:56.22citywokhttp://www.cyberciti.biz/faq/linux-kernel-uhhuh-nmi-received-for-unknown-reason-30/
17:57.08*** part/#asterisk ihor (~Miranda@194.44.15.90)
17:57.47phr0zeni am using boot options:  nmi_watchdog=0 nohpet acpi=off
17:58.06phr0zenthought about adding:  nomce
17:58.11phr0zennot sure on the effect of that however
18:01.10*** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9)
18:02.59phr0zencant find much on the "nmi received unknown reason 21" though
18:04.30Qwellphr0zen: pastebin the output of `cat /proc/interrupts`
18:04.37Qwell~pb
18:04.37infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:07.12drift-[Jan 24 13:06:44] NOTICE[1244]: chan_sip.c:21355 handle_request_invite: Call from '400' to extension '10' rejected because extension not found in context 'default'
18:07.33Qwelldrift-: does 10@default exist?
18:08.43drift-hrm
18:08.49drift-i figured 10 is default extension for voicemail
18:09.01QwellWhat made you figure that? O.o
18:09.17drift-not sure
18:09.40QwellThere is no such thing as a "default extension" anywhere in Asterisk.
18:09.40phr0zenok, so it isnt the recording
18:09.44Qwell(except parking, but that's cheating)
18:09.47phr0zenserver just crashes again with the nmi 21
18:09.59phr0zen*crashed
18:10.03Qwellphr0zen: see above
18:10.56phr0zenjust getting it back up
18:12.35phr0zenhttp://pastebin.com/q1KUV1ej
18:13.37Qwelldoesn't look terrible.  not sharing an IRQ
18:13.56phr0zeni have no idea why this keeps crashing
18:14.34drift-[Jan 24 13:13:57] ERROR[1244]: chan_sip.c:13814 register_verify: Peer '400' is trying to register, but not configured as host=dynamic [Jan 24 13:13:57] NOTICE[1244]: chan_sip.c:23497 handle_request_register: Registration from '<sip:400@69.164.217.96>' failed for '174.                                                                         48.1.164:5060' - Peer is not supposed to register
18:14.49citywokenable with dont optimize & start asterisk with -g so you can get a dump and backtrace it?
18:18.11phr0zencitywok is that for me or drift?
18:18.18drift-heh
18:18.23drift-not for me
18:18.32drift-i dont think
18:18.39drift-asterisk -g does not work... for me 1.8.2.2 what i'm running
18:19.01citywokdrift-: that was for phr0zen
18:19.11drift-okay
18:19.20phr0zenthe thing is, is that the entire system crashes
18:19.22phr0zennot just asterisk
18:19.30drift-where do i configure voicemail box extension?
18:19.33drift-in extensions
18:19.40citywokdrift-: voicemail.conf
18:19.46citywokphr0zen: sounds to me like your system is not stable
18:19.59phr0zenbut u see, i had it up 2 months without an issue
18:20.08phr0zenbut this last ~10 days this shit creeps up
18:20.48citywokif the entire system locks up that is either a REALLY bad lock in asterisk, or more likely system issues
18:20.54*** join/#asterisk peep (637c543e@gateway/web/freenode/ip.99.124.84.62)
18:21.25peepAnyone around that could shed a little light on the AMI action Atxfer?
18:21.39phr0zenwhat piece of hardware should i look at with an "nmi received unknown reason 21"
18:21.51citywokno idea
18:23.05phr0zenif i had "noapic" to the kernel boot arguments, would that cause any negative effects? would it potentially alleviate this issue?
18:23.06peepphr0zen: Thats usually a timing issue. Depending on your setup you may be able to tune/tweak that in the BIOS
18:23.48phr0zentiming issue, with respect to RAM
18:23.51phr0zen?
18:23.52*** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
18:24.14peepwith respect to the CPU
18:24.32peepnoapic may or may not help - I know that was awesome help ;)
18:25.30phr0zeni updated the bios the other day, and set it to optimized defaults... to make sure it wasn't something i ddi
18:25.43phr0zen(bios updated after crashes started)
18:26.47phr0zenso if it is the cpu timing, is there anything specific i should look for/change?
18:26.52*** part/#asterisk arekm (arekm@pld-linux/arekm)
18:28.32peeppowermanager, APCI, and APIC settings in your BIOS should all be off if you can set them
18:29.57peepinteresting
18:30.10peepif all else fails it looks like you can actually disable the NMI watchdog - http://www.cyberciti.biz/faq/linux-kernel-uhhuh-nmi-received-for-unknown-reason-30/
18:30.30phr0zenyea i did that peep (the disable watchdog)
18:30.48phr0zeni believe i turned off apci/apic/powerstuff etc
18:30.50phr0zenin bios
18:31.26peep:( I had issues with a machine doing this and (luckily) it was all fine after upgrading the BIOS and resetting to default settings.
18:31.48phr0zendid that
18:31.59peepSounds like you've tried all of the obvious stuff though, you may need the help of a low-level hardware superhero
18:32.02Kattypeeks in
18:33.06drift-Name/username              Host                                    Dyn Forcerport ACL Port     Status 400/400                    172.16.1.4                               D          5060     Unmonitored
18:33.11drift-hrm why does that ip say local ip
18:33.15drift-it should show my internets ip
18:33.20*** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn)
18:33.47*** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com)
18:35.06*** join/#asterisk mahlon (mahlon@martini.nu)
18:36.00drift-this is interesting
18:36.07drift-i programmed my voicemail box for "dailpad 10"
18:36.16drift-doesnt show anything in asterisk screen
18:36.37*** join/#asterisk m_tadeu (~quassel@89.180.176.52)
18:36.43phr0zendanng.. i need to fix this bug! lol
18:36.49phr0zenfrustrating when nothing fixes it
18:39.19carrarbug are better off squished!
18:39.22carrarbugs
18:40.15*** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor)
18:40.27*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
18:42.22*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
18:44.48phr0zentrying to squish the bug
18:44.56phr0zenit is quite an elusive bug
18:48.26*** join/#asterisk cusco (~tralala@49.192.54.77.rev.vodafone.pt)
18:48.28cuscohi
18:49.05cuscowhere can I find a list of compatible cheap fxo cards?
18:51.29peepcusco: Unless you hate yourself/your customers, its probably better to spend the money on a Digium/Rhino/Sangoma card.
18:51.54Qwells/\/.*//
18:52.57cuscopeep: I have several digium cards for our bizness.. however this is only for testing setting up several solutions based on hardware like a LAN fax gateway
18:58.56*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
19:01.08KattyHAI QWELL
19:01.16Kattyruns, jump, glomps Qwell
19:01.35Naikroveklooks up "glomp"
19:01.50Naikrovekah nice
19:01.59Kattyyou've never been glomped?
19:02.07Naikroveki have, i just never knew what it was called
19:02.11Kattyoh, right.
19:02.16phr0zenok, so just asked intel about the sr2500 with an adpatec 3805 ... told me it is untested and that i should remove it
19:03.05phr0zenand that it might be the cause of my errors
19:03.21phr0zenalthough all of that was vague and stuff
19:03.31KattyNaikrovek: i was afraid iw as going to have to make a special trip just to glomp you
19:04.02Naikrovekaw thansk, but glomping happens every day when i get home.  my daughter has a spaz attack when her favorite toy gets home
19:04.40KattyNaikrovek: most excellent ^_^
19:05.00Kattyi have great memories of being dragged about by my mom's ankle
19:05.11Kattyof course i was probably /four/ at the time
19:05.21carrarPICS!!
19:05.31Kattyhmm. i might have a picture of me at 4
19:05.32carrarOR IT NEVER HAPPEN
19:05.43phr0zenso anywho, i checked teh 3805 compatability from adaptec.. says sr2500 is tested... my god... i swear this error is killing me
19:06.14KattyNaikrovek: you should save empty boxes of groceries
19:06.21KattyNaikrovek: mac n cheese boxes, cereal, soda bottles
19:06.32KattyNaikrovek: then set them all up in a room somewhere so she can go grocery shopping
19:06.39KattyNaikrovek: Best. Game. Ever
19:06.46Naikroveknot a bad idea
19:06.52Naikrovekshe's really into video games though
19:06.59Naikrovekgot Little Big Planet 2 for her birthday
19:07.01Naikrovekshe's 6
19:07.02Kattyah :/
19:07.08Kattyk, so a bit too old for her then
19:07.10Naikrovekgot all the way to the last boss by herself in like 6 hours
19:10.29*** join/#asterisk lanning (~lanning@208.87.235.224)
19:10.51*** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net)
19:15.44*** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net)
19:16.02omanihi, where do I have to configure the caller picture for snom phones?
19:16.07omanion the phone itself or asterisk?
19:16.27omanibecause, openldap provides a user picture
19:16.56omaniI got it work for me to set the caller identification through ldap
19:17.05omanion incoming calls
19:17.22omanibut I didnt get it work to get a caller picture
19:18.13phr0zenjust curious, which snom phone?
19:18.13Kattyi don't know of anything about defining a picture in asterisk
19:18.21Kattyi would guess that's part of the phone config
19:19.43*** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn)
19:20.54n3hxs-wkHad an LOL from helpdesk today.
19:21.00n3hxs-wk<PROTECTED>
19:21.36Kattygiggles
19:21.59n3hxs-wkalmost fell out of his chair.
19:22.02*** join/#asterisk tasca (~tasca@189.4.104.162)
19:22.13Kattyn3hxs-wk: may i facebook that?
19:22.19n3hxs-wkSure
19:22.33n3hxs-wkI don't have U on my Facebook.
19:22.34*** join/#asterisk vinhdizzo (~vinh@dhcp-v007-165.mobile.uci.edu)
19:22.46Kattywell we will just have to fix that now won't we.
19:22.54n3hxs-wk;)
19:24.55beekhugs Katty
19:25.05Kattyhuggiths the beekith
19:26.22beekgrins
19:26.39beekSo how art thou Katty?
19:27.02Kattyhmm.
19:27.07Kattypretty good. my hand hurts tho
19:27.18Kattybumped the oven while pulling lunch out :<
19:27.23Kattyotherwise, i'm just peachy! how're you dear?
19:27.42beekJust fine... trying to stay warm.
19:28.44Kattyis it cold out your way?
19:28.56beekWhen I went to the car this morning to head to work it was -4
19:29.27Kattyeeeeeek!
19:29.37Naikrovekheh yeah
19:29.43beekYesterday morning (a relatively balmy 9) I found out that I needed a new car battery.
19:30.14Kattyyick :<
19:31.11beekSo that's the news from here.
19:32.05Kattyapplies blankets to beek
19:32.12beekThanks!
19:32.21beekMaybe a little brandy too?
19:35.00KattyNugget: telnet
19:35.00Nuggettelnet is eeeeeeevil!
19:35.15Nuggetsilly katty
19:35.22Kattyi got today's!
19:35.25Kattyhugs Nugget
19:35.27Nugget:D
19:35.39Kattybeek: at work?!
19:36.20beekKatty: why not?
19:36.48Naikrovekare you kidding?  work is the place where alcohol is MOST appropirate
19:37.01Kattypfft
19:37.02Qwells/appropriate/necessary/
19:37.07Naikrovek^^^ that!  yes
19:37.11Naikrovekstands corrected
19:37.22beek+1 Qwell
19:37.24Qwellcounter-glomps Katty
19:37.49p3nguinIt would be appropriate if you work in a bar.
19:38.08Qwellcorrect
19:39.11*** join/#asterisk boch (c829e45a@gateway/web/freenode/ip.200.41.228.90)
19:39.16Kattyhugskwishes Qwell
19:39.17bochhello
19:39.19omanidoes anybody know how to set up caller pictures on snom phones?
19:39.34omaniis this thing handled by asterisk or the phone?
19:39.36Qwellcallerid pictures?  that's new..
19:40.01bochanyone knows why SendURL() is not returning? The problem is over an IAX channel, and the 'w' options is NOT being used
19:40.03omaniI got a snom phone here. with openldap and asterisk as a voip pbx
19:40.03Naikrovekkind of a neat idea
19:40.09Naikroveklike xface way way way back when
19:40.09QwellNaikrovek: yeah..
19:40.16Qwellor like every cellphone since 1982
19:40.28omaniin openldap there is a picture assigned to a adressbook entry
19:40.31Naikroveki hear you but that kind of thing on a business phone isn't common
19:40.32p3nguinchortles
19:40.32Qwell(and by 1982, I mean like 2005)
19:40.52omaniand when this person calls, the phone shows his caller id like name surname etc.
19:40.54Naikrovekand that's the kind of thing that sells your phone over your competitor's
19:40.59omanibut no picture. although it is set
19:41.02Qwellsurprised nobody else has done that, tbh
19:42.13n3hxs-wkthe only picture on my phone is held there with Scotch tape.
19:42.26Kattyhehe
19:48.03*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80)
19:50.13*** join/#asterisk tclark (~Administr@S0106001310ead738.du.shawcable.net)
19:51.05_Corey_Must be some demand for that feature, Switchvox does it...
19:51.22_Corey_(looks better on a color LCD though)
20:01.28KattyI"M IN THE MOOD FOR LOVE, SIMPLY BECAUSE YOU"RE NEAR ME
20:01.49p3nguinwoohoo
20:01.59Qwellblinks
20:02.05KattyQwell: it's a song
20:02.10Qwellsuuuuuuuure
20:02.15*** join/#asterisk donttrustem (~Trustem@188.127.169.192)
20:02.22Kattystands by Qwell
20:02.27Qwelleep!
20:02.31beekis jealous
20:02.34Kattybwuahahaha.
20:04.26n3hxs-wkThat song is older than I am.
20:04.36Kattyyou're not old dear.
20:05.25n3hxs-wkRemind me of that in the morning.
20:05.35n3hxs-wker... that didn't sound quite right...
20:05.54Kattykay.
20:05.58bochanyone knows why SendURL() is not returning? The problem is over an IAX channel, and the 'w' options is NOT being used
20:06.01*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)
20:06.04n3hxs-wkDon't let my profile pix fool you.
20:06.11raden_workNaikrovek, What Up Bro !
20:06.12Kattyhugs raden_work
20:06.19Kattyraden_work: hey....bro...
20:06.38raden_workgives Katty Huge Hugs
20:06.44raden_workHey :P
20:07.01Kattyn3hxs-wk: whatever do you mean?
20:07.55Kattyraden_work: i gotta work on my streettalk
20:08.08raden_workLMAO too cute
20:10.19Kattyyou gots to get them benjamins so you cingit dat blingbling for da cutes. Mmmmm you know it!!
20:10.59Kattyyeah. i think i'll just stick with Bland White English.
20:13.44Naikroveksupdude
20:15.22Kattyduddeee
20:15.43Kattyi love how dude is the universe word for all definitions and meanings
20:15.54Kattys/universe/universal
20:16.08Naikrovekyep
20:16.10Naikroveki mean dude
20:16.18Kattynods. dude.
20:20.23raden_workLmao
20:20.31p3nguinda kine
20:20.33raden_workNaikrovek, how goes it ?
20:20.37Naikrovekraden_work: good.
20:20.45raden_workstaying busy
20:21.12Naikrovekraden_work: you ever play duck hunt as a kid?
20:21.14Naikrovekyes busy
20:21.17*** join/#asterisk drift-_ (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
20:21.30Naikrovekhad someone to help with IT stuff but he got shoved into a new position with a client right quick
20:21.48Naikrovekso back to grunt work, down from architecture issues and decision making
20:21.49Naikrovek:(
20:21.53Kattylol i remember duck hunt.
20:21.55Naikrovekhttp://soundcloud.com/sonicwalter/duck-hunt\
20:21.58Naikroveklisten to that
20:21.59Kattyi used to stand up by the tv with the gun ;)
20:21.59Naikrovekhttp://soundcloud.com/sonicwalter/duck-hunt
20:22.02Naikrovekminus the \
20:22.21Kattylistens
20:22.45Kattygrooves
20:22.54Naikrovekyeah it's good
20:23.01Kattyquack quack boom? lol
20:23.05raden_worklol
20:23.21raden_workI need a gaming console
20:23.26Naikrovekps3
20:23.26raden_workto much work need to play more
20:23.32Kattyi have an extra snes
20:23.37Naikrovekextra?!
20:23.40raden_workNaikrovek, keep debating between ps3 and Xbox
20:23.47Naikrovekps3 > xbox all day
20:23.56Kattyyes'r. i need a power supply. found a snes at a yard sale, with the controllers and games and such for 10 bucks.
20:24.04Kattys/need/needed/
20:24.10*** join/#asterisk [Outcast] (~anonymous@64.202.62.5)
20:24.18raden_worklol
20:24.32raden_workNaikrovek, im thinking of going that route
20:24.32Kattyso ya, extra console.
20:24.50KattyNaikrovek: mind if i fb that?
20:25.06Naikrovekthe duck hunt thing?  go for it
20:25.10Naikrovekthere are many more.
20:25.19Naikrovekgoogle "LP of Devastation"
20:25.22Naikrovekfor the whole album
20:25.38Kattyomg there's a whole ALBUM
20:25.41Kattysquees!
20:27.00KattyYESYESYES they are on grooveshark!!!
20:27.11Kattyhugskwishes Naikrovek to bits
20:27.44Naikrovekdragon warrior, metroid, ninja gaiden, duck hunt, metroid, mega man 2, mario 1 & 2, Zelda 1 & 2.  all good
20:29.55Naikrovekdude is actually a very capable rapper if you can call it rap
20:29.58Naikroveki guess you can
20:30.08Naikrovekdudes, i should say
20:30.11Naikrovekthere's two of them
20:30.14Naikrovekanyway
20:30.21Naikrovekcan't get these songs out of my mind
20:30.32Kattydudddeee
20:30.53Naikrovekdragon warrior is my fave atm
20:31.38Kattyso far the tetris one is my fav
20:32.41Naikrovekyeah that one starts out real lame, like untalented, then it goes into high gear and you're like "whoa.. dude!"
20:35.55*** join/#asterisk Godfather_ (~godfather@217.Red-83-58-86.dynamicIP.rima-tde.net)
20:40.01*** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com)
20:45.20*** join/#asterisk Janos (~Janos@190.10.52.113)
20:49.05*** join/#asterisk vinhdizzo (~vinh@dhcp-v007-165.mobile.uci.edu)
20:51.24*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
20:52.12p3nguinAnyone here have any good experiences with Western Digital Black hard drives?
20:53.01Naikrovekturns his head and looks to his right.
20:53.14Naikrovekall i have is a seagate external usb drive and it sucks big butt
20:53.26Naikrovek4 months old, failing
20:53.40Tozz_omg! i'm lost!
20:53.43Tozz_i thought this was #asterisk
20:53.49*** join/#asterisk bipolar (~bipolar@offsitesysadmin.com)
20:53.55Naikrovekyeah i know
20:54.03Naikrovekwe're offtopic when there's no asterisk talk going on
20:54.09Tozz_ah k ;)
20:54.10Naikrovekwe stop when people start asking questions
20:54.13p3nguintozz_: Asterisk needs to run on computers with file systems of some sorts.
20:54.16Tozz_but no, no (good) experience
20:54.48p3nguinI've had pretty good luck with Seagate disks, but they are pretty old... back when they made good stuff.  When I heard Seagate started making crap (a few years ago), I stop buying Seagate.
20:55.13*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
20:55.18p3nguinI went Samsung and Hitachi.  Now that I need another disk, I was considering WD black.
20:55.19Tozz_no matter what brand you buy, they all die
20:55.27Tozz_I replaced a Samsung today
20:55.30Naikrovekyeah, but within 4 months?
20:55.39Tozz_can happen if u have a bad batch
20:55.40Naikrovekso tired of my boss and his stupid "here is your back" crap
20:55.47Naikroveks/back/backup/
20:56.03Tozz_lol
20:56.23Tozz_or maybe your controller is incompatible
20:56.32Naikrovekwill NOT buy a tape drive or another SAN or enough bandwidth to backup online
20:56.43JanosHey there, i'm getting some errors on a Zap channel and calls are not going through it:
20:56.45Janos[Jan 24 14:38:17] NOTICE[23758] chan_zap.c: Got event 4 (Alarm)...
20:56.45Naikrovektakes a week to fill this drive
20:56.47Janos[Jan 24 14:38:24] WARNING[23758] chan_zap.c: CallerID returned with error on channel 'Zap/9-
20:56.48p3nguinI've decided against WD green and blue, but black has pretty good performance ratings.
20:56.49Janos1'
20:56.53Janosany idea what this might be ?
20:56.54Naikrovekplease no pasting
21:01.03Naikrovekbut no i have no idea
21:01.10Naikrovekdoes the caller id work on the call this error comes from
21:01.44Chainsawp3nguin: RE4-GP 2TB here, they've been good to me. I'm sure black is even better.
21:02.18ChainsawJanos: Alarm suggests no battery voltage. This is normally a bad thing.
21:02.34ChainsawJanos: What does that face? An actual telco, some PABX hardware of your own?
21:02.49p3nguinThey cost a little more than their counterparts from other manufacturers, so I figured they might be better.  I've found that I often get what I pay for.
21:03.08*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)
21:04.01Chainsawp3nguin: Having the spindle secured at both ends sounds like a good plan, yes.
21:04.05Chainsawp3nguin: Bit surprised others don't do that yet.
21:04.15peepAnyone around that could shed a little light on the AMI action Atxfer?
21:04.40ChainsawAttended transfer?
21:05.15Janosthis channel is in a AEX800 which has multiple analog lines connected to them, so far i have only see the alarm on channel 9 and not on any other channel
21:05.25Linuturkany polycom central config guys in here?
21:05.34LinuturkI want to make sure I understand the process correctly
21:05.43Naikrovekwhat's your question
21:05.46kn0xLinuturk: whats your quesiton
21:06.04LinuturkI've got a range of soundpoint IP's to support
21:06.05Naikrovekwhere's the ambiguity
21:06.09Naikrovekokay go on
21:06.11Linuturkfrom 300's to 650's
21:06.40ChainsawJanos: So where is channel 9 connected to at the *other* end?
21:06.52Naikrovekoh yeah you have two generall classes of polycoms: legacy and .. uh.. non legacy
21:06.56JanosChainsaw: analog line from a telco
21:07.07Linuturkbased on my research, I'm going to need the 4.1.x 4.2.x and 4.3.x bootroms on my server. I'll also need the 2.1.x 3.1.x 3.2.x and 3.3.x sip firmwares
21:07.25ChainsawJanos: Does the fault move if you swap channel connections?
21:07.27NaikrovekLinuturk: what are all the models you are supporting
21:07.32Naikrovekall the polycom model numberws
21:07.46JanosChainsaw: will try that next, be back, thanks
21:07.50Linuturk300/500/301/501/600/601/430/650
21:08.05*** join/#asterisk dustybin (~dustybin@78-86-171-176.zone2.bethere.co.uk)
21:08.19dustybinanybody remember the sox command to convert a .wav file ?
21:08.31Naikrovekdustybin where the hell have you been dude
21:08.39LinuturkNaikrovek: ^
21:08.44Naikrovekwhat's youre new site
21:08.46NaikrovekLinuturk: whoa
21:08.47Naikrovekokay
21:09.22dustybinoh sorry!
21:09.24dustybinits here
21:09.26dustybinwizbox.net
21:09.29Naikrovekthought so
21:09.38dustybinhttp://wizbox.net/index.cgi/debian_asterisk%3A2010-09-01%3ADebian
21:09.45dustybini been doing other stuff
21:10.09Naikrovekit's okay
21:10.11Naikrovekjust curious
21:10.15Kattydances with Naikrovek
21:10.22Naikrovekwhich song you like best katty
21:10.31NaikrovekLinuturk: so you probably need all those yeah
21:10.36LinuturkI've got several issues here though. First, all the phone's bootrom's are at various versions, so I need to get them all on the same page. It should be as simple as unzipping the various bootrom versions, oldest to newest, in my ftproot, right? second, I will have to update the config files before I update the sip versions, so nothing wonky happens.
21:10.37Naikrovekall those different firmwares and bootroms
21:11.04Linuturkat least, that's what I've been led to believe via the admin guide for 3.3.0
21:11.12NaikrovekLinuturk: that may very well be true
21:11.19LinuturkI don't see an admin guide for the bootrom stuff on polycom's site though
21:11.39Linuturkthe crux of my question is basically a sanity check on what I'm going ;p
21:11.42Linuturkdoing*
21:11.43Naikrovekbest option would be to pull the specific files you need from the different firmwares and bootroms and assembling what you want
21:11.49peepChainsaw: Yes attended transffer. When you use it on a channel it just returns "Atxfer successfully queued" and has no effect on the call
21:12.34LinuturkNaikrovek: so, basically, I can update the bootroms without any config file changes. just "drop" them in my ftp root.
21:12.43Naikrovekthe phone, once booted, can pull configs based on model and so on.
21:12.58p3nguinchainsaw: The black disks also have 5 years warranty.  Others are only 2 or 3 years.  That can save some cash in the event that one actually does crap out.
21:13.00Naikrovekthe bootroms don't have an associated config file to meddle with so yeah.
21:13.05LinuturkNaikrovek: then, the pain of config file changes will hit, where all my e1000.cfg become phone<mac>.cfg
21:13.10Chainsawp3nguin: My RE4-GP has 5 years warranty.
21:13.31Chainsawp3nguin: And so very silent. You should hear them seek. It sounds like rain hitting a window.
21:13.31p3nguinMaybe all the new WDs have 5.
21:13.34NaikrovekLinuturk: split up your configuration into per-MODEL configs.
21:13.42Chainsawp3nguin: http://www.vroon.org/drives.mov
21:13.50NaikrovekLinuturk: PM me your email address and I'll send you something that may explain it more
21:13.52Chainsawp3nguin: That's 8 of them as loud as they will get.
21:14.08LinuturkNaikrovek: right, phone1<model>.cfg or whatever
21:14.29LinuturkNaikrovek: with the individual extenion information setup as phone<mac>.cfg
21:14.32Naikrovekyeah something like that.  also sip-650.conf if you want
21:14.45Naikrovekright the individual mac.cfg files will mention the files required for that model
21:14.55Naikroveki think you can use substitution in there so the phone pics them automatically
21:15.10Linuturkwell, the official stance from polycom is to get rid of the individual <mac>.cfg files
21:15.18Linuturkhave them all point to the default 0000.cfg
21:15.19Naikrovekeh don't
21:15.41Naikrovekhow are they to get individual login details if they all point to the same file
21:15.48Linuturkthat would use the new options in the 4.0.0+ branch of the bootrom to load up a phone<mac>.cfg file
21:15.59Naikrovekis there a bulletin ID on that document that you can give me
21:16.01Linuturkin the 00000
21:16.12Linuturkyou put a variable PHONE_MAC_ADDRESS
21:16.36Naikrovek... so get rid of the mac.cfg so you can use phone-mac.cfg instead?  that saves you zero effort
21:16.37Naikrovekwtf
21:16.55Naikroveki generate all my configs from a script so it makes no sense to me but whatever
21:16.58Linuturkthat dynamically replaces it, so you could say CONFIG_FILES="phone[PHONE_MAC_ADDRESS].cfg, etc etc
21:17.17Naikrovekyeah i gotcha, but you still gotta have those phone0004f2abcdef.cfg files all over the place
21:17.26Linuturkright
21:17.45Naikrovekso it does nothing but abstract configuration one level
21:17.51Naikrovekyou still gotta maintain all those specific files
21:18.03Naikrovekpointless..
21:18.04Naikrovekanyway
21:18.15Linuturkthat's correct, but it allows you to abstract out so you can support legacy phones
21:18.22Naikrovekyou can do that anyway
21:18.47Naikrovekso i guess both approaches are the same.  pick the one you prefer
21:19.27Naikrovekmy phones don't find 000000000000.cfg and use their own mac.cfg and that points to the specific individual files they need
21:19.40Naikrovekmy generation script plugs in the proper values in those [mac].cfg files
21:20.06Linuturkso, bootrom's just get dropped in the root of the ftp? what can I expect when upgrading a bunch of unknown bootroms to the latest version? long reboot times I'm sure
21:20.37Naikroveknot really; they'll reboot and notice a new bootrom, download it, verify it , then install it and reboot again
21:21.03*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
21:21.05Naikrovekthen same with SIP firmware
21:21.14Naikrovektotal time for my phones is about 10 minutes
21:21.27Naikrovekbut mine are newer than yours
21:22.06Naikrovekwe jsut talked about everything i was going to say in my email so i probably wont' send anything now
21:22.41*** join/#asterisk anonymus (~yaa@178.176.71.175)
21:22.47Naikroveksnag someone's phone and use it for testing
21:22.53Naikrovekwhen your'e ready
21:22.58anonymushi
21:23.04*** join/#asterisk IsUp (~nocturne@unaffiliated/isup)
21:25.58Linuturkthanks Naikrovek
21:26.09Linuturkupdating bootrom firmware on so many phones makes me nervous
21:33.14grkbloodmy MOH isnt working in the queue, i keep getting DEBUG[3396] res_musiconhold.c: Read 620 bytes of audio while expecting 640
21:33.25grkbloodmy musiconhold doesnt work when in queue
21:33.47p3nguinIt works when not in a queue?
21:33.54grkbloodcorrect
21:34.01grkbloodif i manually put someone on hold it works
21:35.59grkbloodthat log is from /var/log/asterisk/full
21:36.24grkbloodim tailing full right now and its constanting outputting that with no activity o nthe lines
21:37.37*** join/#asterisk clintc (~clintc@n128-227-78-187.xlate.ufl.edu)
21:39.34grkbloodany clues?
21:40.19*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
21:40.50*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
21:43.21drift-http://blog.nguyenvq.com/2010/10/30/google-voice-on-asterisk-with-an-auto-attendant-and-free-calls/ < i did all that
21:43.36drift-<PROTECTED>
21:44.01Qwelldoes that extension exist in tntts?
21:44.31drift-[tnttsp] exten => s,1,Answer() ;;exten => s,n,Wait(10) exten => s,n,Wait(1) exten => s,n,SendDTMF(1) ;;needed for google voice; otherwise, only call to computer in gmail will work and not calls made to google voice exten => s,n,Playback(hello-world)  ;; call exten => _1NXXNXXXXXX,1,Dial(Gtalk/tnttspJabber/+${EXTEN}@voice.google.com) << my extensions.conf
21:44.43QwellSo, no.
21:45.10drift-_hrm
21:45.11Qwellreplace your s exten.  That guy doesn't know what he's talking about
21:45.34drift-_replace it with what?
21:47.31drift-i cut pasted * verbatim
21:47.34LinuturkNaikrovek: odd, I've updated a few of the phones, but my oldest phones are running bootrom 2.6 and they aren't pulling a new bootrom :(
21:47.35drift-just changed my l/p for google
21:47.52NaikrovekLinuturk: what version are you going to on those phones
21:48.00Naikrovekyou may need to go to v3 before you go to v4
21:48.03Linuturk4.1.x
21:48.08*** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net)
21:48.20Linuturkah, ok, thanks Naikrovek, seems I have to rebuild my bootrom dir
21:49.33dustybinblist
21:49.42*** part/#asterisk clintc (~clintc@n128-227-78-187.xlate.ufl.edu)
21:53.21*** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net)
21:54.14grkbloodthis should play the default MOH when in queue 1010 correct?
21:54.18grkbloodhttp://pastebin.com/XxBgqmSu
21:54.34p3nguindrift-: You've indicated that you are trying to call extension 954........ but you've shown us extension s instead.
21:54.41retentiveboyTinkering with 1.8 but it appears it's not loading my extensions.conf file.  Has something changed from 1.6 that would affect this?  Figure I'd ask before I started strace'ing and more digging.
21:55.28drift-calling a phone #
21:55.29drift-954
21:55.30drift-xxxxx
21:55.34drift-thats area code
21:55.36p3nguinprecisely
21:56.27drift-<PROTECTED>
21:56.28drift-my users.conf
21:56.30drift-thats all it has in it
21:56.40p3nguin*shrug*
21:56.51p3nguinusers.conf can suck an egg for all I care.
21:56.58drift-heh?
21:57.01drift-i need 1 extension tho 400
21:57.03drift-i made it 400
21:57.15drift-so phone connects to my linode asterisk server
21:57.19dustybinWTF
21:57.22p3nguinIt's not really pertinent that I can tell.
21:57.30Kattypeeks in
21:57.34dustybindrift-: ?
21:57.36drift-kicks dustybin in tha nuts
21:57.38dustybinLOL
21:57.41Kattygoodness.
21:57.48Kattyi'm gone for 30 minutes and we start kicking folks?
21:57.55Kattymakes a girl want to pout
21:57.58Kattypouts.
21:58.10carrarPut on a happy face!
21:58.32Kattykay
21:58.33Katty:>
21:58.36carrarheh
22:00.19drift-_lol
22:03.13*** part/#asterisk dustybin (~dustybin@78-86-171-176.zone2.bethere.co.uk)
22:04.43*** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net)
22:05.49grkbloodcan some please help me figure out why my queue MOH doesnt work? ive been messing with it for like a week
22:08.26Kattywhat have you tried so far
22:09.09grkbloodbasically everything, i know my caller is going to queue
22:09.18grkbloodthe MOH works if i manually put the caller on hold
22:09.20Kattycould you elaborate on the Everything
22:09.48Kattywhat else have you tinkered with in your troubleshooting
22:10.13grkbloodive tweeked my musiconhold*.conf files, created new confs, viewed the panel in freepbx and askterisk -r to make sure the caller is going to conf
22:10.25grkbloodmessed with sound cards
22:10.33Kattycould you define...tweeked?
22:10.34grkbloodviewed logs
22:11.14grkbloodwell, i found that when i made changes in freepbx to MOH settings they didnt update the files in /etc/asterisk so i did it manually
22:11.27grkbloodnow MOH works when i place a caller on hold
22:11.31grkbloodnot the queue though
22:11.52grkbloodand im getting a debug error 24/7 concerning res_musiconhold.conf
22:11.54Kattyso what does your queue moh play? nothing?
22:12.01grkbloodyea, it plays nothing
22:12.04grkbloodits silent
22:12.08Qwella "debug error"?
22:12.27Kattyde bug is in a cocoon error
22:12.48grkblood[Jan 24 17:09:06] DEBUG[4400] res_musiconhold.c: Read 620 bytes of audio while expecting 640
22:12.57grkbloodi dont know if that has anything to do with anything
22:13.27Kattyhave you tried listening to one of the audio files using playback()?
22:13.45grkbloodits not an audio file
22:13.49grkbloodits an application
22:14.02grkbloodits streaming the mic in port with a script
22:14.16Kattyohisee.
22:14.34grkbloodwhich works find when i do it manually liek i said
22:14.37Kattywell that's out of my league. perhaps someone else can help tho (=
22:14.38grkbloodjust not in queue
22:15.03grkbloodKatty, i can change the queue MOH to an audio file, it still doesnt work
22:16.36*** join/#asterisk ccesario (~ccesario@201-42-148-53.dsl.telesp.net.br)
22:20.52grkbloodKatty, ok, i have it set up to play an mp3 now
22:20.57grkbloodstill nothign in queue
22:29.47grkbloodwhere are inbound routes save at?
22:29.56grkbloodim guessing somewhere in /etc/asterisk
22:30.04p3nguinDefine "routes."
22:30.38carrarthe lines on wood made by routers!
22:31.10p3nguinIn that case, they are stored on the edges of the workpiece.
22:31.24grkbloodinround routes
22:31.28grkbloodinbound*
22:31.35p3nguinAgain, define "routes."
22:32.06grkbloodthats what its called in freepbx, where you put the setting for phone numbers and define where their destinations go
22:32.33grkbloodyou set the DID number and destination in it
22:32.35p3nguinThat would be extensions.
22:32.47p3nguin/etc/asterisk/extensions.conf
22:38.12jayteejust built me a mini-itx box with an Intel D510MO board and installed Asterisk 1.6.2.16.1 on it.
22:38.33p3nguinSounds like overkill.
22:38.46jayteefor a small office?
22:38.52p3nguinHow is it on power consumption?
22:39.03jayteenot sure yet
22:39.12p3nguinNo Kill-a-Watt?
22:39.29jayteehuh?
22:39.49p3nguinYou don't use a Kill-a-Watt to measure power usage on appliances?
22:40.10jayteenope
22:40.15jayteeI should get one
22:40.31p3nguin$20 or so... it's a great tool.
22:41.17jayteeyeah, I'll pick one up. I know this box isn't a power hog though.
22:42.57p3nguinDo you happen to know the processor number?
22:43.12carrar5
22:43.28Bidikokey ... 2 days triuing to fige out why chan-mobile says mobile is ready but mobile is not working on incoming or outgoing calls ... no error given
22:43.55p3nguin5, huh?  An Intel Atom 5?
22:44.01carrarno, just 5
22:44.06jayteeit's a D510 Atom. I'd have to connect it up and power up to get to the BIOS for any thing more specific
22:44.56jayteeit's dual core 1.66ghz with hyperthreading
22:45.00p3nguinOh, the D510 is the only processor that works on that board?
22:45.23jayteeif you do a cat /proc/cpuinfo CentOS thinks it has 4 cores :-)
22:45.32p3nguin64-Bits, 13W
22:46.17jayteep3nguin, yeah, it's an embedded chip board but you can get mini-itx boards that will take a Core 2 Mobile Penryn cpu
22:47.04p3nguinI guess I didn't realize they weren't interchangeable.
22:47.06jayteeI'm also considering getting a Jetway board with a 3 port Intel NIC daughter board.
22:47.37jayteebut first I have to test this one thoroughly
22:47.45jayteebrb
22:48.46p3nguinThat one should provide plenty of processing power for Asterisk.  I run on far less.
23:09.13*** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com)
23:31.57p3nguinjaytee: How much do you spend to get the board into a working unit?  case, PSU, RAM, board (with embedded processor), no hard drive
23:35.06p3nguinjaytee: I've sent you a /notice that you might be interested in, especially if it's more than $200.
23:36.30IsUp#hardware
23:36.34IsUp:p
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23:47.16*** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
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