IRC log for #asterisk on 20100606

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00:55.26*** join/#asterisk sky1975 (~76f3d0db@gateway/web/freenode/x-hrynztbtocaqflze)
00:55.38sky1975Hi
00:56.05sky1975Can anybody help me to set security on skype for asterisk?
00:56.38[TK]D-FenderskyWhat "security"?
00:56.52sky1975I mean authentication
00:57.25sky1975I tried auth_policy=accept:martini
00:57.36sky1975Doesn't work
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01:01.43*** join/#asterisk acxty (~acxty@201.220.136.118)
01:01.45sky1975Shall I put my chan_skype.conf entry here?
01:02.08acxtyHi guys, I define some sip accounts on sip.conf. How can I see which ones are register to the server?
01:02.11sky1975<[TK]D-Fender> ?
01:02.51[TK]D-Fenderacxty: "sip show registry" / "sip show peer [peer]"
01:10.59sky1975<PROTECTED>
01:15.05dinesh___well if the order is the same as for the codecs then yes it is wrong
01:15.18*** join/#asterisk x303 (~x303@20.226.118.70.cfl.res.rr.com)
01:15.46sky1975Ahh. is that? I will change and check
01:16.43dinesh___btw does someone happen to know what codecs the Unified Communication server from Microsoft is using ? no voice is going through the sip calls i'm making from my asterisk
01:16.56drmessanoG711
01:17.04drmessanoYou need SIP TCP
01:17.37dinesh___thanks
01:17.52dinesh___i also think that it only supports TLS (called SIPS)
01:18.29dinesh___which is probably why i couldn't login from x-lite
01:18.36dinesh___(which does have SIP TCP)
01:18.49drmessanoNope
01:18.50drmessanohttp://blogs.technet.com/b/gclark/archive/2010/05/24/3134398.aspx
01:21.30*** join/#asterisk SAIDias (~SAID@97-125-143-112.desm.qwest.net)
01:21.34SAIDiasHowdy
01:21.51SAIDiaswhats the best way of streaming audio to shoutcast
01:21.57SAIDiasfrom an extension
01:23.49sky1975<dinesh___> Will this do the job? [mx-400] secret=xxxxxx  context=from-pstn  auth_policy=deny  auth_policy=accept:martini
01:26.01xheliox~itsp-usa
01:26.19xheliox~itsp
01:26.19infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
01:26.26xheliox~itsplist-us
01:26.27infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
01:30.32[TK]D-FenderSAIDias: Run a softphone on auto-answer.  Wire the speaker-out into Line-In.  Have your shotcast app use that
01:30.57[TK]D-FenderSAIDias: Or actually... you ccould use chan_oss and do it more directly
01:31.19SAIDias[TK]D-Fender: I only want to use a single machine todo it
01:32.29[TK]D-FenderSAIDias: I just gave you 2 ways you can.
01:37.15SAIDiasthx
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04:33.42acxtyHi guys I am getting this error  pbx_spool.c:276 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test.call: Operation not permitted. What this mean?
04:36.33acxtyI am getting this when I place a .call file on outgoing
04:36.42acxtychown asterisk:asterisk
04:36.45acxtyand chmod 777
04:39.16*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
04:41.53acxtymay someone help me with that?
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04:46.22pabelangeracxty: Are you running Asterisk with -U asterisk and -G asterisk?
04:51.15acxtypabelanger, No
04:53.20pabelangeracxty: then why are you chown asterisk:asterisk your call file?
04:56.18acxtyIs I use root:root it says permission denied
04:56.28acxtyI only did an aptitude install asterisk
04:56.36acxtyand only change sip.conf and extension.conf
05:04.18jblackperhaps an overly tight umask
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05:19.25p3nguinAsterisk runs as root on that system?  Yuck.
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05:26.39jblackI think he said he is running asterisk as asterisk, and creating callfiles as root.
05:30.32p3nguin(2346.23) <pabelanger> acxty: Are you running Asterisk with -U asterisk and -G asterisk?
05:30.36p3nguin(2351.15) <acxty> pabelanger, No
05:30.46p3nguinI interpret this as not running it as asterisk.
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05:39.40Dovidgood morning. anyone know what would casue this and if it is an Asterisk thing or a Linux thing ?
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06:34.53*** join/#asterisk P-Chan (~jason@ip70-162-221-117.ph.ph.cox.net)
06:34.59P-Chan'ello!
06:36.56P-ChanI've got a system I just set up that I keep getting an immediate "goodbye" and disconnect when calling in.  I've got a Digium card w/ 2 lines and I think it's an issue with the incoming context being "default" instead of "from-pstn", but I can't seem to change it
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06:38.43P-ChanStarting simple switch on 'DAHDI/2-1'
06:38.44P-ChanExecuting [s@default:1] Playback("DAHDI/2-1", "vm-goodbye") in new stack
06:39.16P-ChanI'm guessing "s@default:1" means it's coming in on the "default" context, correct?
06:39.41P-Chan's' =>            1. Playback(vm-goodbye)                       [pbx_config]
06:39.50P-Chan<PROTECTED>
06:39.57P-Chan<PROTECTED>
06:40.03P-Chanthat's my "default" dialplan
06:40.32P-Chanbut it ignores the "context=from-pstn" line in dahdi_channels.conf
06:41.03P-Chanany ideas on how I can troubleshoot this?  Thanks in advance.
06:42.59P-Chan[ Context 'default' created by 'pbx_config' ]
06:42.59P-Chan<PROTECTED>
06:42.59P-Chan<PROTECTED>
06:42.59P-Chan<PROTECTED>
06:43.19P-Chan<PROTECTED>
06:43.20P-Chan<PROTECTED>
06:43.23P-Chanand this is despite:
06:44.24P-Chan; Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER)
06:44.24P-Chan;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
06:44.24P-Chansignalling=fxs_ks
06:44.24P-Chancallerid=asreceived
06:44.24P-Changroup=0
06:44.24P-Chancontext=from-pstn
06:44.24P-Chanchannel => 1
06:44.25P-Chan;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
06:44.25P-Chansignalling=fxs_ks
06:44.26P-Chancallerid=asreceived
06:44.26P-Changroup=0
06:44.27P-Chancontext=from-pstn
06:44.27P-Chanchannel => 2
06:44.30P-Chan(that's my dahdi_channels.conf
06:44.35P-Chansorry for the spamming
06:45.01P-Chanoops... posted to the same chan twice, the second time it was meant for #freepbx... >>;
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06:53.05DovidP-Chan: please use pb
06:53.07Dovid~pb
06:53.16infobot[~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.
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06:57.40ChannelZare you sure your chan_dahdi.conf is including that file and not some other version?
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07:27.41ChannelZgets his groove on
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07:42.18joobieguys is there a way i can limit the RTP ports my sip peer uses? i use a paid sip peer to route some calls and they are usinga ll sorts of ports
07:42.27joobiecan teh client somehow specify to the peer that it was the ports limited?
07:43.03ChannelZDepends on the device
07:43.59ChannelZAs a 'client' (a phone) you can request what RTP port the remote end should send to you at.  The reverse is also true (including Asterisk, via rtp.conf)
07:44.22joobieChannelZ, how can this be done? I'll give it acrack
07:44.31joobiei use pennytel, want to get pennytel to use only a certain range
07:44.41joobieright now they are going all over the shop which makes my firewall rules a little less specific
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07:46.01kerx_l/join #freeswitch
07:46.04kerx_err
07:47.56ChannelZAgain it depends on the device.. you'd have to dig around in it's configs
07:48.01ChannelZwhat are you using?
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07:49.26joobiei have asterisk setup to connect to pennytel, not a device
07:50.28ChannelZwell then as I said, you can edit /etc/asterisk/rtp.conf and set the port range Asterisk uses
07:51.05joobieisnt that for what asterisk uses? ie it wont affect pennytel as a sip peer to asterisk?
07:51.06ChannelZYou can't control what the other end wants however, but it's usually not a big problem allowing outbound trafficggg
07:51.25joobieahh
07:51.36joobiethat's what i wanted to do :/ doh
07:51.44ChannelZThere's two RTP streams... one from you to someone else, and one from someone else back to you.
07:52.00joobieya
07:52.06ChannelZYou can only ask the remote site where to send their traffic to you at;  You can't control where they want their traffic sent to
07:52.09ChannelZs/site/side/
07:52.14joobieyer
07:52.16joobiedoh
07:52.24joobiecos thats what i wanted to do
07:52.44joobiei have outbound firewall rules that id have tp loosen if not
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07:54.38ChannelZDo you contact them at random IP addresses?  If not it's pretty trivial to just allow all outgoing traffic only to them if you're paranoid about it (or a large chunk of ports)
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07:55.17ChannelZ16384-16482 apparently
07:56.02joobiena jsut the one ip
07:56.14joobiebut they use very random ports
07:56.21joobieto the point where im thinking to just let all udp out to them
07:56.29joobie.. greater than 1023
07:56.35*** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru)
07:56.52ChannelZyeah... according to them, 16384-16482
07:56.57ChannelZhardly the world
07:58.34joobiethat's bs
07:58.44joobieim getting much more random than that
07:58.56ChannelZWell then maybe you should yell at them, because that's what their FAQ says
07:58.59ChannelZ"You are behind a firewall. In this case you will need to have UDP ports 5060-5061 and 16384-16482 (default RTP ports for Linksys devices) open and forwarded to be able to register and use the service."
07:59.05joobiemaybe it's the AU servers
07:59.11joobieyer
07:59.16joobiethey suck for customer service tho
07:59.25joobiegotta be transfered through to people overseas who barely speak englihs
07:59.34joobiealways a challenge to explain something like this to them
07:59.35joobieall good
07:59.56joobieill wokr around it ChannelZ .. was just hoping the protocol had some prevision to specify a prefered port range or sumthen
08:00.08ChannelZwell you can complain here where nobody can do anything about it or complain to them (or switch providers)
08:00.21ChannelZBut if you only contact them on a single IP, it's hardly a big deal
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08:00.58joobieit's not.. i just dont like to put gay rules like that in my fw
08:01.51ChannelZsuit yoursef
08:01.57ChannelZhave fun not making calls
08:02.48wdoekes2~
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08:08.14ChannelZ^
08:09.29joobieChannelZ, making calls fin
08:09.30joobiee
08:09.40joobiejust opened up the firewall
08:09.47joobieto the insecure way that you're so keen to do
08:14.03ChannelZOH NOES I HOAP PENNYTEL DO NOT HACKS YOU!
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08:17.54joobieobviously your firewall management skills are yet to extend beyond your home network
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08:18.59ChannelZchuckles
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09:07.19k-manjoobie: evening
09:14.34joobiehey k-man
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10:02.41guzahello
10:02.50guzai was download asterisk now 32 version
10:03.15k-mansorry guza i don't understand what you mean
10:03.17guzaafter install it i was got error that say kernel missing
10:03.54guzaasterisknow was not install kernel in /boot
10:04.06guzaso grub can not found it and boot it
10:04.09k-manwhat was it you were donwloading?
10:04.27k-manthat's a linux distro issue not an asterisk issue
10:04.27guzai wass downloading from asterisk.org
10:04.35guzayes it is
10:04.48guzabut this is yours distro that u release it
10:05.15k-manwrong on both counts
10:05.18guzahttp://www.asterisk.org/downloads/asterisknow/i386/asterisknow32.iso this i was downloaded
10:05.42k-manthis is a user support channel, not a channel run by digium as far as i know
10:06.22guzaok i know that
10:06.49guzabut just to know distro on yours offical site dont work
10:07.16guzanow i downloading 64 version
10:07.49k-manguza: on "the" official site not "your" official site
10:09.06guza:)
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10:13.21jblackguza: My suggestion is to find the asterisknow bug tracker and file a bug.
10:13.51jblackThat seems like an awfully severe error that nobody's run into it before. Maybe you accidentally told apt to remove all the kenrels.
10:14.58jblackThat would be like buying a new car only to find out it doesn't have an engine.. It just doesn't happen. You couldn't get even get it home, so to speak. =)
10:15.19ChannelZpoints to #asterisknow
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13:10.07Hoihi, does anyone know whats wrong with the asterisk.org forum? can't register and there doesn't seem to be much activity since May 18 2010!
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13:37.14tzafrir_laptopHoi, well at least you have IRC
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14:56.05cuscohi
14:56.16cuscowhat is wrong with: if (${Produto}=460 | ${Produto}=461)
14:56.20cuscoin a AEL?
14:56.31cuscothe space before and after |
14:56.33cusco?
14:56.43cuscothat is a Logical OR, right?
14:57.01Trixboxertry with ||
14:57.07Trixboxernot sure though
14:57.13cuscothats not it
14:57.20cusco[Jun  6 13:55:54] WARNING[11268] ast_expr2.fl: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
14:57.21Trixboxerok.. I never done that
14:57.38cuscoah
14:57.45cusco${Produto} is empty?!
14:57.51cuscoerr
14:57.56domihow can i make asterisk answer incoming calls directly to its IP without any SIP-Provider?
14:58.29cuscodomi: who's making the call?
14:58.30cuscosip client?
14:58.34domiyes
14:58.44cuscothat client is registered with asterisk?
14:58.46domibut the client should not register to my box
14:58.52domiforeign clients
14:59.38cusconot sure
14:59.38domiso that any IP-Phone can call my box with its IP
14:59.46cuscoisn't that inseure?
15:00.07cuscoit must be possible, I don't know how, tho
15:00.16*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
15:00.21domiwhy? this context can nat dial out, only for calling my internal phones
15:00.26domis/nat/not/
15:00.36domilol nice bot ;)
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15:05.18domiah... will try this: http://lists.digium.com/pipermail/asterisk-users/2007-April/184131.html
15:05.45*** join/#asterisk Benwa (~Benwa@dyn.83-228-160-169.dsl.vtx.ch)
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15:53.25kallisti5is there a way to detect an offline sip extension before dialing in extensions.conf?
15:54.28kallisti5i was thinking GotoIf... but i have no clue on how to use it
15:55.01p3nguinIf you mean an offline device, you can use ChanIsAvail().
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15:57.39kallisti5p3nguin: thanks!  that was just what i was looking for !
16:05.32domihmm i cannot connect to gtalk. get "aji_act_hook: JABBER: encryption failure. possible bad password."  but the credentials are correct
16:05.58domiasterisk 1.6.2.6-1 on debian squeeze
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16:45.18p3nguinDoes this romb thing ever do anything other than join, ping timeout, and change nicks?
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17:11.41[sr]hi guys
17:11.49[sr]spent some time reading about BRI and PRI
17:12.02[sr]so i need to configure a BRI 2B+D
17:12.11[sr]where can I find some good docs about it?
17:12.52WIMPy[sr] What hardware?
17:13.21[sr]openvox, HFC4
17:13.35[sr]i just realized that cant access the machine :(
17:13.59[sr]have to see why its not up tomorow, anyway that the card, or at least part of the description
17:14.36[sr]it has two ports
17:15.02WIMPyDo they supply drivers?
17:15.13[sr]nothing!
17:15.14WIMPyOnly two ports?
17:15.22[sr]but its supported by dahdi, i remember that
17:15.52[sr]yes, two ports only
17:16.07WIMPyIf you only need TE mode, that's fine. NT mode and dahdi doesn't seem to work with all cards.
17:16.46[sr]well, i'll have to use it with the NT..
17:16.50[sr]just hope ir works :|
17:16.55WIMPyI'm using LCR now. All other options seem to be dead ends.
17:17.19[sr]what LCR stands for?
17:17.33WIMPyLinux Call Router
17:18.44*** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)
17:18.45[sr]going to investigate on the net... sec
17:20.57[sr]i see on docs here that the HFC-S (that the correct model i rememebr now) works in NT mode
17:22.03WIMPyThey all do, but dahdi doesn't seem to do it with all cards.
17:22.24[sr]HFC-4S i mean
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17:22.45WIMPySo that would be four ports.
17:23.33[sr]well yes, for connect it only has 2 fisical ports,
17:23.39[sr]thats why i called it 2 prots
17:27.02[sr]i think i may try this LCR
17:27.12[sr]does it has freepbx builtin also?
17:27.33TJNIIOh, physical ports.  I read that as fiscal ports and was quite confused.
17:27.48tzafrir_laptop[sr], which specific HFC-4S card?
17:27.55tzafrir_laptopoh, the openvox one?
17:28.10[sr]tzafrir_laptop: yes
17:28.31tzafrir_laptop. How do can you tell that "NT mode is not supported"? How did you set NT mode?
17:28.48[sr]tzafrir_laptop: i'm just investigating for now
17:29.00[sr]doing some homework
17:29.21tzafrir_laptopAFAIK it should work. Though NT/PtMP will require asterisk trunk (NT PTP should work well)
17:29.51tzafrir_laptopWhat do you connect to it?
17:30.22WIMPy[sr]: LCR connects to Asterisk via chan_lcr.
17:30.22[sr]does asterisk 1.6.2.8 doesn't have that feature yet?
17:30.34tzafrir_laptopsadly, no
17:31.46[sr]ok asterisk trunk wont be a problem
17:32.15[sr]WIMPy: i see, i was checking and the 11MB size of the file couldn't be a distro!!
17:33.07WIMPy[sr]: No, it's an applications. Chose whatever distro, you like.
17:35.25[sr]im going to setup a box to compile asterisk from source
17:35.35[sr]right now i have a trixbox with a 4FXO port card
17:35.41[sr]that confighuration is easy
17:35.45[sr]and simple
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17:37.43[sr]the 1st part i already cleared that, i have a BRI interface, 2B+D
17:37.52[sr]so i have to focus on that conf type
17:37.59[sr]with 10 MSN's
17:40.12[sr]ah one thing i didn't found
17:40.20[sr]what's the type/protocol
17:40.31[sr]for the normal central PBX
17:40.38[sr]for a isdn fone?
17:41.01[sr]i mean, if I want to connect an isdn fone to this card, what's the name of the configuration?
17:42.00WIMPyThat's the NT mode. The mentioned NT-ptmp to be exact.
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17:42.45[sr]hum wait, i'm a bit confused now
17:42.47TrixboxerHi, Is there a way to append the chosen IVR number to recording file ?
17:42.59TrixboxerI mean User call in and presented with an IVR to press 1,2,3,etc
17:43.22[sr]1st scenario: connect from NT to the card, NT=>CARD using BRI
17:43.40[sr]1nd scenario, connect a isdn fone to the card, ISDN phone=>card
17:43.49[sr]that's the same configuration?
17:44.00WIMPyNope
17:44.48WIMPy1) requires TE mode, 2) NT mode
17:45.34WIMPyIf you want to connect phones as well, I'd recommend LCR.
17:46.46[sr]nice
17:46.49[sr]going to read the docs!
17:46.54[sr]thank you for now!!
17:48.16[sr]ah wait
17:48.23[sr]for the card list i'm seeing,
17:48.42[sr]an ISDN modem (normaly used for internet connection in the past) can also work here, am i correct?
17:50.16[sr]misdn.org is down, normal?
17:50.38WIMPyLCR uses misdn2 from the kernel which at least supports anything HFC based. Not sure about other stuff.
17:51.06cuscohi...
17:51.08[sr]hum if it's already on the main kernel nice
17:51.31cuscoim fiddling with peer to peer comunication with the canreinvite options
17:51.41cuscoit works OK but...
17:52.30cuscothere are 2 asterisk boxes. If one of them has a queue and then dials, the comunication will always stay between the asterisk queueing and the peer
17:53.34cuscoasterisk1 -> dial -> asterisk2 -> queue, dial -> peer
17:53.54cuscocomunication between peer and asterisk1 is not direct, instead goes from peer to asterisk2
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17:58.19[sr]hey WIMPy, in which part of the kernel are the isdn required modules?
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18:08.57[sr]WIMPy: found it
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18:30.13xEBIxhello
18:30.57xEBIxI'hve got a telefonie problem with my asterisk
18:31.48xEBIxive some telephones on my local asterisk server, most of them are in the same net. One is connected from another net via NAT and the internet
18:33.27xEBIxonly with that one ive some problems, a connections is opening but we can't here each other, but he can hear waitmusic when activate it, after i switch that off i can here him very short
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18:33.52xEBIxwhat kind of problem might that be?
18:33.57cuscoxEBIx: in the peer info set nat=yes
18:34.03TJNII~sipnat
18:34.03infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
18:34.11xEBIxcusco, yes thats is set
18:34.53TJNIIxEBIx: Is the server behind NAT, or does it have a public, static IP?
18:35.13xEBIxthe server is not NAted
18:35.24xEBIxonly the peer telephone
18:35.33TJNIIhmmm... And you have your IP ranges set properly in sip.conf?
18:35.45TJNIIexternip and localnet, iirc.....
18:35.46xEBIxthe server has a static ip
18:36.37xEBIxaa that could be a problem, i didn't think of having telephone out of my net when i set it up
18:36.50cuscoso the IP is listening on port 560 and the udp ports range...
18:36.52xEBIxthanks for that firstofall
18:38.15xEBIxyes it is listening on 0.0.0.0 and the contactpermit rule is commented out
18:38.47xEBIxi think you think about port 5060?
18:38.56cuscoyes sorry
18:40.30xEBIxwhere do i have to set my IP of the server?
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18:41.38cuscosip.conf
18:41.42cuscoexternalip
18:43.47xEBIxcusco, is that the exact directive?
18:44.18cuscohold
18:44.54cuscoexternip=213.63.137.210
18:45.03cuscolocalnet=192.168.2.0/255.255.255.0
18:45.14cuscothats what I have
18:47.06xEBIxis a dns name for externip working also?
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18:49.49cuscoyes
18:50.53xEBIxi set that and relaoded with the command, nothing changed...
18:52.58cuscosip reload
18:55.51WIMPy[sr
18:55.56WIMPyups
18:56.13WIMPy[sr]: drivers>isdn
18:56.23xEBIxhmm no change
18:59.27xEBIxdo you have a working open stun server for me
19:07.58xEBIxnow the echo test is working but, its no change with my telephones
19:09.05xEBIxif i switch music on hold on and off i can here him for a very short time...
19:09.22xEBIxany further idea?
19:15.18TJNIIIs only one external phone not working?  Can you try another phone from outside?
19:16.03xEBIxno i cant, at this time
19:16.34xEBIxcan we try it?
19:17.39TJNIII'm sure you can.  You don't have access to a network behind another public IP?
19:18.02xEBIxno i do nat have
19:18.09xEBIxs/a/o
19:18.46TJNIISo your asterisk server is also your NAT gateway for your private network?
19:18.55xEBIxyes thats right
19:20.25xEBIxTJNII yes right
19:21.38TJNIISo you obviously have a firewall on that box.  Are you sure it is configured properly?  Not blocking RTP traffic on the public port?
19:22.00*** join/#asterisk Tim_Toady (~moi@193.92.244.148.dsl.dyn.forthnet.gr)
19:22.10xEBIxTJNII, no i do not have a firewall active on the public port
19:22.46TJNIIYes, you do.  You have to have a firewall running to have NAT working.
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19:22.55TJNIIiptables on Linux.
19:23.44xEBIxTJNII, yes thats true, but no further rules and policies on ACCEPT
19:24.32TJNIIWell, then I would try a softphone from another network to try and isolate the problem.
19:24.46TJNIINot the network the suspect phone is behind.
19:25.09xEBIxyes i would do that if i could
19:27.15xEBIxdo you have a softphone? would you connect to me, for testing?
19:28.15TJNIII don't want to install a softphone and I don't have a microphone.
19:28.52TJNIIPlus, my attitude is that you should do your own testing.  No offense.
19:29.01xEBIxaah  ok
19:29.17xEBIxso any other idea?
19:29.48TJNIIYou should be able to find another network to test from easily.  Work, a friends house, a coffee shop, a WiFi hotcpot, etc.
19:30.05*** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com)
19:30.31xEBIxok thanks or your time
19:32.25*** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com)
19:33.22TJNIIHaha, I just found a job listing on Craigslist that pays in chickens.
19:37.07*** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es)
19:40.50m_tadeuhi all...I'm trying to use a agi php script but I'm getting a "broken pipe" error
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19:46.16ChannelZTJNII: Is it in Mexico?
19:46.39ChannelZm_tadeu: the app is either not getting called right or is terminating in a not-so-nice way
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19:47.33p3nguinI guess that PAP2T with the authentication problem is a pile of crap, because a softphone on that same LAN works just fine.
19:47.47ChannelZI think the firmware on it was ancient
19:47.54p3nguinEven after a firmware upgrade, the problem still exists.
19:48.02ChannelZreally
19:48.18p3nguinIt was at 3.1.something, now it is using 5.1.6.
19:48.31p3nguinUser-Agent: Linksys/PAP2T-5.1.6(LS)
19:48.37m_tadeuChannelZ: I'm sure it's getting called...gonna check the other situation
19:48.41TJNIIChannelZ: No here in Colorado.  I'll let you make the obligatory immigrant jokes.
19:48.42ChannelZI suppose it could be something in the config but I don't know what -- it just seemed to be totally ignoring the digest auth request
19:49.12ChannelZwonders if chickens are taxable income
19:49.16TJNII"natural fram" or something similar.  So hippies.
19:49.28p3nguinTwinkle works fine using the same peer information.  Ekiga was working when he went direct to VoIP.ms, too.
19:50.27bluOxigenis now known as TheBird
19:50.31*** join/#asterisk linkd (~switch@unaffiliated/linkd)
19:51.01ChannelZWell plenty of other people seem to have it working so it has to be some config or another.
19:51.10*** join/#asterisk wayne (~wayne@ool-ad03ce08.dyn.optonline.net)
19:53.14b14ckYo.
19:53.21p3nguinIf there was a known-good PAP2T that he could plug in and make work, that would prove that his ATA is borked.
19:53.24b14ckHow often do the developers touch code in pbx/?
19:53.36b14ckI've been doing some code review, there are several things that could be updated there.
19:53.44b14ckI was wondering if that is on any sort of development schedule or not.
19:54.32p3nguinIf you have a patch, you can submit it for inclusion.
19:54.33ChannelZsubmit patches
19:54.36b14ckFor example, in pbx_spool.c there are numerous hard-coded constants, and routines which do thinks like trim whitespace. They just seem to be copy+pasted around. But the include/asterisk/strings.h contains functions that should be used for that stuff :x
19:55.11ChannelZprobably better to take it up in #asterisk-dev
19:57.12m_tadeuChannelZ: ok, no broken pipe anymore...but I'm outputing some garbage...shouldn't it show up in the console?
20:01.03ChannelZoutputting how?  stdout is for sending commands to Asterisk, it doesn't wind up on the console
20:01.21ChannelZyou can turn on AGI debug and see what commands it's doing IIRC
20:02.47*** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein)
20:04.28*** part/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr)
20:07.00m_tadeuChannelZ: ah ok...I thought I would see some error or something
20:08.41m_tadeuit's working :)
20:08.43ChannelZwith debug off all you'll see is the script being called, and its exit code when it's done
20:10.08m_tadeuChannelZ: Thanx  a lot :)
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20:20.31seanjohnI finally decided to use Cepstral and have compiled it manually to get swift() application. Anyone know how to slow this B@tch down?
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20:42.55[sr]people, WIMPy, when compiling dahdi-linux from trunk, or last 2.3.0.1 i get this: /usr/local/src/asterisk/dahdi-linux/drivers/dahdi/voicebus/GpakCust.h:114: error: field 'sem' has incomplete type
20:43.03[sr]GCC 4.4
20:43.53Chainsaw[sr]: Ah yes, you need to talk to tzafrir_laptop.
20:43.58Chainsaw[sr]: He said that bug doesn't exist.
20:44.10[sr]it does exist :P
20:44.19[sr]tzafrir_laptop: can you help? ;)
20:44.30Chainsaw[sr]: Or wait, I may have a bug open about that. There are about 3.
20:44.32Chainsaw[sr]: Sec.
20:44.41WIMPy[sr]: I have eliminated dahdi
20:45.26Chainsaw[sr]: Here you go: https://issues.asterisk.org/view.php?id=17382
20:45.43tzafrir_laptophttps://issues.asterisk.org/view.php?id=17382 ? (initial patch there by Chainsaw )
20:45.49Chainsaw[sr]: I need to redo that patch as it breaks kernels below 2.6.26
20:45.51[sr]WIMPy: only LCR?
20:45.54Chainsaw[sr]: But it should work for you.
20:46.19WIMPy[sr]: LCR with Asterisk
20:47.48[sr]ChanServ: tzafrir_laptop, hum a missing include
20:47.53[sr]works OK now
20:48.27[sr]WIMPy: I'll do that also, going to do my 1st test with dahdi
20:49.37Chainsawtzafrir_laptop: https://issues.asterisk.org/view.php?id=17383 should be safe to apply now, we're at RC2.
20:49.43m_tadeuhow can I make the agi_calleridname known?
20:50.07Chainsawtzafrir_laptop: I need to wrap that include in an ifdef, which I know you dislike. But I can't think of a portable way that is compatible with <2.6.26 otherwise.
20:50.29tzafrir_laptopChainsaw, I'll try to look at it tommorow
20:50.39Chainsawtzafrir_laptop: Thanks, I'll do my best to have them both ready for you.
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20:59.41[sr]WIMPy: who maints LCR? doesn't compile :S
21:02.34WIMPyJolly
21:03.30ChannelZm_tadeu: what do you mean?
21:03.40*** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com)
21:04.56[sr]ok emailed him
21:05.11m_tadeuChannelZ: the vars that are passed inside an agi script...one of them is agi_calleridname, and it's value is "unknown". I was expecting the caller's username, or something
21:05.45ChannelZwell it only knows what Asterisk knows.. are you sure you're getting callerID over the channel in question?
21:06.30ChannelZIE put a NoOp(${CALLERID(all)}) in your dialplan and see what you're getting
21:08.29*** join/#asterisk boodu (~antoine@175.158.129.128)
21:09.12booduhello
21:10.00ChannelZohell
21:10.34m_tadeuChannelZ: I get the name of the peer I'm calling...not the caller
21:12.58ChannelZpastebin the console output of a complete call
21:13.04ChannelZcore set verbose 4
21:13.25*** join/#asterisk MiserySoft (~LND@89.193.239.100)
21:14.29m_tadeuChannelZ: http://pastebin.com/Y57gaevz
21:16.02*** join/#asterisk gnude (~andre@muedsl-82-207-249-193.citykom.de)
21:16.17ChannelZis s,2 the NoOp(${CALLERID(all)}) ?  (right before the AGI)
21:16.51*** join/#asterisk cesar_CR (~cesar@201.199.168.170)
21:17.05ChannelZand where is this call coming from, just another local device?
21:18.21m_tadeuyes...right before the agi call, and also yes, local device, registered on sip.conf,I mean
21:19.13ChannelZand does that device have a "callerid=Foo <123>" in sip.conf?
21:20.51*** join/#asterisk Squeeb (~squirt@eggwee.co.uk)
21:21.08m_tadeunop...gonna take care of that
21:21.18SqueebHi, Quick question about AEL: How can I assign the output of an application (System() in this case) to a variable.
21:21.24*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
21:21.57ChannelZSystem() as an application might not give you app output
21:22.05Squeebah.
21:22.54*** join/#asterisk Benwa (~Benwa@dyn.83-228-160-169.dsl.vtx.ch)
21:22.55SqueebHmm, it's executing a small bash script that spits out a number.
21:23.02SqueebI need to take that value.. and pass it to another Application
21:23.09ChannelZyou'll porbably have to write an AGI in some other language to do what you need.. problem is a program could output all sorts of crazy things
21:23.31SqueebWell I started with the bash script being called by AGI
21:23.43Squeebbut then I had the same problem, how do I pass the return of the AGI to another App?
21:24.29ChannelZIf it's an app in the dialplan, you'd just have to set a channel variable from the AGI
21:25.01SqueebAha.. of course.
21:25.02SqueebThanks :)
21:25.35ChannelZsho thang
21:25.58[sr]brb
21:26.15Squeebmeh, I should stop trying to write AGI scripts in bash tbh :/
21:26.58m_tadeuChannelZ: same thing...didn't change...I should have something like "agent1" calling "agent_pbx"
21:27.38*** part/#asterisk gnude (~andre@muedsl-82-207-249-193.citykom.de)
21:28.15ChannelZdoes 'sip show peer xxx' show the Callerid you set for the peer?  (xxx being whatever its name was.. agent1 or something based on what you just said maybe)
21:29.35m_tadeuyes
21:29.54m_tadeuthe same string I set in sip.conf
21:29.59*** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com)
21:31.18ChannelZand that is the device you were calling _from_ and it's still showing up totally blank?
21:33.26m_tadeuyup...still showing "unknown"
21:34.25booduI need help with sccp, my device can't register on the server. In the phone, I can found in DeviceConfiguration>UnifiedCMConfiguration "Unified CM 1 : unavail." but it's the right ip address.
21:34.39m_tadeubut I think the caller name should be in agi_calleridname, which I'm saying is unknown
21:35.02booduif you have an idea
21:35.04ChannelZhmmm no makey sense
21:44.41*** part/#asterisk Ole_ (ole@54b.pl)
21:44.44m_tadeuI'm out of ideas...gotta read more
21:45.41ChannelZwell I dunno why it's being erased in the dialplan
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21:47.24m_tadeumaybe tomorrow something will popup :) thanx for everything
21:47.30MangoWord of advice.
21:47.32Squeeb*sigh* .. I wish Cisco had better documentation for their XML configs :(
21:47.38MangoDon't do exten => h,n,NoOp(${CHANNEL(peername)}) - it crashes Asterisk.
21:47.40Mangoer
21:47.47Mangowhat I meant to paste was: exten => h,n,NoOp(${CHANNEL()})
21:48.11cdahmedehhello, i want to setup a simple voip getaway.. just want to access my home phone from the internet anywhere.. currently looking for the proper hardware... looking for the cheapest usb to rj11 adapter possible.. can anyone help ?
21:48.45MangoSangoma has one though I'm not sure of the cost.   http://www.sangoma.com/products/hardware_products/analog_telephony/usb_fxo.html
21:48.54MangoI don't believe it has hardware echo cancellation.
21:49.08ChainsawThat's the politically correct version of "this is going to suck".
21:49.17cdahmedehoh ok
21:49.28Mangogiggles at Chainsaw
21:49.45cdahmedehlike i'm planning to buy used or something
21:50.48Chainsawcdahmedeh: It can be done cheap, but I'd recommend that you use a VoIP gateway. You can get those for quite reasonable prices. It'll do echo cancellation and... you won't have to leave a PC on.
21:51.53cdahmedehlike i just want to setup my own server.. and this is only for a couple of months
21:52.14MangoChainsaw, what would you use for that?  I've heard bad things about the SPA3102
21:52.40ChainsawMango: I wouldn't use cheap kit anywhere, but my reliability concerns and budget are completely different.
21:52.42Squeebcdahmedeh: If it's just to make calls over the internet for a month or so, why not buy a USB handset and use Skype?
21:52.59cdahmedehi want to use the home phone line
21:53.00Squeeb£10 from InsertCheapVendorHere
21:53.34ChainsawMango: The prospect of having to log into something remotely, and using cheap USB hardware... I'm not going to lie to you. It fills me with dread. And nightmares of having to send people in to "fix" stuff.
21:54.11MangoAye.
21:54.29SqueebSo let me get this straight, you want to call through your standard land line, from your computer?
21:54.35SqueebI believe the term is "Modem"
21:54.37MangoI haven't heard anyone who claims they have a reliable, cheap way of doing that.
21:55.10*** join/#asterisk Lantizia (~Lantizia@93-97-23-110.zone5.bethere.co.uk)
21:55.36LantiziaCheap but good single or dual SIP to FXS adapters that arn't a PAP2T :) Hit me :P
21:56.02LantiziaI've seen Pattons range and they're stuuuupidly priced, any better ideas would be most appreciated and thanked :)
21:56.04SqueebLantizia: http://www.byfarthecheapest.com/products/Cisco-Small-Business-Linksys-VoIP-Adaptor-%252d-1-FXS,-1-FXO-Ports,-2-Ethernet-with-NAT.html ?
21:56.31ChainsawLantizia: Yes, Patton is telco-grade. It'll cost ya.
21:56.32Lantiziaumm thats basically an SPA3102
21:56.34cdahmedehmaybe something like that
21:56.41SqueebLantizia: pretty much
21:56.49cdahmedehjust very basic.. all i need is just plug in my phone line into the computer..
21:57.01cdahmedehso i can use the phone line via internet
21:57.04cdahmedehvia an sip client
21:57.08cdahmedehjust looking for the cheapest hardware
21:57.34LantiziaChainsaw, well I'm considering Patton for BRI and PRI gateways... but thats only because they're well known and the cheapest.  If they're telco-grade, who isn't telco-grade and doing BRI/PRI gateways :P
21:57.34Squeebcdahmedeh: that link I posted is probably also what you're looking for
21:58.02cdahmedehok.. so it's called a pstn gateway ?
21:58.18SqueebPlug your phone line into the FXO port, then connct to it over the internet (obviously after configuring your network)
21:58.38cdahmedehis it possible to directly plug it into the computer ?
21:58.41cdahmedehserver i mean
21:59.07ChannelZUSB, ethernet, what difference does it make?
21:59.10SqueebYou don't need the server, but you can attach it to a PBX
21:59.36LantiziaI always wondered if old internal dial-up modems can be used for FXS/FXO SIP use
21:59.45cdahmedehin fact.. can they ?
22:00.03Squeebhttp://voip.weblogsinc.com/2005/07/14/use-a-v92-modem-as-an-fxo-card-on-asterisk/
22:00.06Squeebsome can
22:00.08Mangocdahmedeh: You could consider a FXO card that you install in the computer.  It'd be MUCH better quality than the USB ones, but not cheap.
22:00.09LantiziaThey have a port lol, and you can generate noise down it
22:01.12cdahmedehso for the ethernet one.. i connected to the network.. and have the server detect it ?
22:01.20Squeebwhat? no
22:01.30SqueebWhat's with this "Server" what's the "Server" doing?
22:01.30cdahmedehso what do i do ?
22:01.32cdahmedehok
22:01.37cdahmedehthe server will be running asterisk
22:01.45cdahmedehand the devices will connect to the server via sip
22:01.57cdahmedehand i want the server to connect to the phone line
22:02.18Squeebright, well that's one option, if you're server has PCI slots then I strongly suggest purchasing a dedicated FXO card.
22:02.26Squeebwhich will provide physical PSTN sockets
22:02.31Squeebthat you can plug your land line in
22:02.39Squeeband, to be quite honest, possibly the easiest to configure
22:02.50cdahmedehhmm.. ok
22:02.51ChannelZ$2xx
22:02.59cdahmedehcan i get those cheap ?
22:03.02Squeebmeh, there's some Digium clones on eBay
22:03.38Squeebhttp://cgi.ebay.co.uk/Authentic-X100P-SE-FXO-PCI-Digium-Asterisk-VoIP-PBX-/130319896399?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item1e57abcf4f#ht_3808wt_1139
22:03.42SqueebThese are terrible
22:03.44Squeebbut cheap
22:03.57cdahmedehterrible in what sense ?
22:03.59drmessanoOpenVOX single port is better than that crap
22:04.05Squeebdrmessano: true
22:04.09Squeebcdahmedeh: really bad echo cancellation
22:04.10Squeebstatic
22:04.10cdahmedehall i need is one port
22:04.16SqueebPeople sounding the terminator
22:04.17Squeebetc..
22:04.19drmessanoThey're poor quality
22:04.19cdahmedehok
22:04.23LantiziaWhy not just a standard 56k/V90 modem?
22:04.28Squeebbasically, ring Dell customer support.. it'll sound like that
22:04.34SqueebLantizia: depends on what modem
22:04.36MangoLOL
22:04.37cdahmedehok makes sense
22:04.42LantiziaSqueeb, why does it have to?
22:04.47LantiziaI mean I don't get it why someone hasn't made some software to use them
22:05.02drmessanoLantizia: Because they're crap hardware.
22:05.05SqueebI don't know to be honest.. but it's only some that support it
22:05.08LantiziaSomething in userland not alternative drivers
22:05.28Squeebsomething along the lines of vgetty may have to, I don't know about latency though
22:05.30drmessanoLantizia: Someone made software for the MODEM chipset that is the X100P, and you see the end result.  Crap
22:05.40*** join/#asterisk Gos (~jhoekman@ip154-92-210-87.adsl2.static.versatel.nl)
22:05.42Gos#welkommentilnederlandersCSS
22:05.48SqueebSod off.
22:05.52drmessanoModems do not make decent voice cards.. Different design goal
22:06.04SqueebThat and they're designed for a completely different use :P
22:06.15drmessano^^^
22:06.21Lantiziadrmessano, if they're accurate enough for modem use - surely voice quality should be fine
22:06.22SqueebUse the right tool for the right job
22:06.24Squeebis the golden motto
22:06.29Gosspam worlds best canned ham
22:06.49drmessanoLantizia: Negative.. That makes no sense.  They're made for passing data.. Not transcoding audio
22:06.55Squeebwell..
22:07.01Gosspam worlds best canned ham
22:07.11Squeeb*ACTUALLY*.. that's exactly what they're designed for .. just not in a vocal sense
22:07.13Lantiziadrmessano, they don't need to transcode audio - software can do that it just needs to listen/send audio to the modem
22:07.25LantiziaJust like any "phone dialer" software can place calls with a dial-up modem
22:07.27Squeebmore in a 'working out ones and zeros' kind of way
22:07.31Lantiziaand that call quality is fine
22:07.49cdahmedehi'm planning to use some old laptop.. and has a dial-up modem.. if it helps that's fine
22:07.53cdahmedehi'm not looking for perfect quality
22:08.02drmessanoLantizia:  The call quality is not fine, which is what you run into here
22:08.09LantiziaI'm not sold on the whole, "the quality is bad" argument
22:08.11Chainsawcdahmedeh: You're just looking to save some money with a soldering iron and some old modems?
22:08.15Gosspam worlds best canned ham
22:08.23drmessanoLantizia, go buy one and come back in a week
22:08.36LantiziaI've got plenty of old modems lying about
22:08.52cdahmedehlike if can find usb to rj11 to connect into the pc for cheap.. i'm all good
22:08.59drmessanoLantizia: So go for it.. make it happen.  Prove us wrong.. We've only been doing this 24 hours too
22:08.59cdahmedehcan't echo cancellation be done with asterisk ?
22:09.23drmessano~x100p
22:09.31infobotwell, x100p is an obsolete card.  You don't want to bother trying to make it (or any of the "digium compatible" clones) work.  Get a TDM01B, and you will save your sanity, your hair, and countless other things.
22:09.31drmessano~x100
22:09.32drmessano:(
22:09.32Squeebcdahmedeh: not quickly
22:09.57Lantiziadrmessano, I don't *need* it I'm just putting the case forward that it shouldn't be that hard/problematic.  I mean why did Windows come with a "Phone Dialer" for all those years that could be used for placing calls using your internal modem!
22:09.59Squeebcertain codecs support it
22:10.15Gosspam worlds best canned ham
22:10.15SqueebLantizia: that was for precisely that, Dialing
22:10.24Squeebyou'd hook your handset up to the "Phone" port on the modem
22:10.25Squeeband boom
22:10.30Lantiziabut they supported headsets!
22:10.33*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
22:10.35Squeebyes, the modem did
22:10.37Squeebnot the computer
22:10.41Lantiziaah as a passthrough?
22:10.44Squeebexactly
22:10.52Lantiziahmm ok, see I never tried it :)
22:11.03SqueebHandy when you had an address book
22:11.07Lantiziarighty
22:11.09Squeebclick, dial, connect
22:11.15Corydon76-digAdditionally, many of the internal modems were half-duplex
22:11.40SqueebBut now we're not in the 90's :D So thankfully we don't have to deal with that stuff any more
22:11.43Squeebyay
22:11.47ChainsawCorydon76-dig: Would you mind taking care of Gos btw?
22:11.51Squeebyea that too
22:12.17Corydon76-digChainsaw: no idea what you're talking about
22:12.29SqueebCorydon76-dig: there's a spam bot in here called Gos
22:12.54Corydon76-digOh
22:13.09cdahmedehok.. so let's go back to my original call. .some old laptop converted to an old server.. i want to connect to the server a phone line.. and use that phone line via the internet
22:13.22Squeebwait a second..
22:13.24Squeeb"Laptop" ?
22:13.28cdahmedehyes
22:13.30cdahmedehno pci
22:13.31*** mode/#asterisk [+b *!*@ip154-92-210-87.adsl2.static.versatel.nl] by Corydon76-dig
22:13.31Gosis is not bot i just like ht ham
22:13.44ChainsawCorydon76-dig: Much appreciated.
22:14.18cdahmedehso i'm gonna have to connect a ethernet to fxs (i think it's fxs) converter to the next
22:14.20cdahmedehnetwork*
22:14.21cdahmedehright ?
22:14.26SqueebFXO and yes.
22:14.44cdahmedehso i'm looking for a basic fxo converter that connects to the network
22:14.47cdahmedehjust one port for the phone line
22:14.47SqueebLaptop <--> VoIP / FXO convertor <----> PSTN
22:14.54*** part/#asterisk Gos (~jhoekman@ip154-92-210-87.adsl2.static.versatel.nl)
22:14.57cdahmedehyes !
22:15.02cdahmedehexactly
22:15.14cdahmedehso all i need is one pstn/rj11 port, and one ethernet port for that thing
22:15.14*** mode/#asterisk [-b *!*@ip154-92-210-87.adsl2.static.versatel.nl] by Corydon76-dig
22:15.22cdahmedehand looking for a good cheap option for that hardware
22:15.24Squeebhttp://www.voipuser.org/review_8.html
22:15.26Squeeb^^ CHEAP
22:15.29Squeeband very very shit quality
22:15.55Squeebphone line in, ethernet in .. bish bash bosh
22:16.11SqueebSet the thing up as a trunk in users.conf or sip.conf
22:16.13Squeeband you're away
22:16.17cdahmedehsounds good
22:16.32cdahmedehthe sipura 3000 is discontinued ?
22:16.35Squeebea
22:16.37Squeebyea
22:16.38Squeebhella old
22:16.46LantiziaDoes anything even rival the Sipura/Linksys/CSB range of ATA's in terms of price and popularity?
22:17.01cdahmedehthat's why i can only find one on ebay
22:17.01LantiziaSomething less... Linksys lol
22:17.13Squeebcdahmedeh: here's an almost relevant HOWTO
22:17.13Squeebhttp://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx
22:17.55cdahmedehok there we go
22:17.55cdahmedehfound the 3000
22:17.58cdahmedehcheap stuff
22:18.10cdahmedehhttp://shop.ebay.ca/i.html?_nkw=SPA-3000&_sacat=0&_sop=2&_odkw=SPA-3102&_osacat=0&bkBtn=&_trksid=m270
22:18.12cdahmedehthis is it ?
22:19.27Lantiziacdahmedeh, any old Sipura/Linksys/CSB (been rebranded alot) SPAxxxx/PAPxx type device with FXS will do fine :)
22:19.29Squeebthat's the newer one
22:19.35cdahmedehperfect
22:19.42Squeebwait no.. that's THE one
22:19.45Squeebnot the newone one
22:19.48Squeebmisread it
22:19.54Squeebbut yes, that will work
22:19.59Lantiziaan old Sipura SPA3000
22:20.10Squeebtbh, if it's just for one connection at a time.. you probably don't even have to use asterisk :P
22:20.27cdahmedehoh i can connect directly to it ?
22:20.28Squeebbut still, you can if you want to, which I think is the main point
22:20.33Squeebyea, just point your sip client at it
22:20.39cdahmedehoh wow
22:20.40cdahmedehthat's very nice
22:21.10SqueebNot sure what they mean by "UNLOCKED" in the description though
22:21.14Squeeb*shrug*
22:22.13TJNIISqueeb: You're talking about an "UNLOCKED" ATA?
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22:25.12cdahmedehok.. i found some local store in canada that sells this type of stuff
22:25.20cdahmedehwould the spa-2102 work ?
22:26.21*** part/#asterisk kallisti5 (~kallisti5@kallisti5-2-pt.tunnel.tserv9.chi1.ipv6.he.net)
22:27.14MangoDoes anyone here use a carrier in Canada called ISP Telecom?  If so, have you had any termination problems over the past week?
22:36.32cdahmedehok.. so in the end.. i have decided for a Cisco SPA2102
22:36.37cdahmedehis that what i am looking for ?
22:37.59Mangocdahmedeh, that doesn't have an FXO port.  You want a SPA3102.
22:38.37cdahmedehwhat's the difference between fxo and fxs ?
22:38.44Squeebgoogle is your friend
22:39.07jayteefxo is for POTS lines from the telco, FXS is for lines to phones
22:39.10Chainsawcdahmedeh: FXS ports are for plugging phones into. FXO ports are for plugging into a phone network (like BT, KPN, etc).
22:39.32cdahmedehoh ok
22:40.23Squeebwhat you're looking for is an FXO gateway
22:40.28cdahmedehhttp://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=012331
22:40.30cdahmedehok this like this
22:40.30Squeebhense the SPA3xxx
22:40.42Squeebyes
22:40.55cdahmedehwow .. not too expensive
22:41.09SqueebWe have sort of being saying this for about an hour
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