irclog2html for #asterisk on 20060220

00:02.09afrosheennice and quiet here today....
00:03.03asterisk99anyone familiar with zaptel compiles on ubuntu?
00:03.21afrosheennot really but what is it doing
00:04.03asterisk99I am getting a 'Module wctdm missing' when I modprobe wcfxs
00:04.22afrosheenwhat kernel, and are you using udev
00:04.44asterisk99afrosheen: 2.6.12
00:04.54afrosheenso you did the make linux26 for the make command right
00:04.55ManxPowerasterisk99, what zaptel version are you using?
00:05.05*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:05.08Ariel_hello folks
00:05.14asterisk99afrosheen: cd ..
00:05.16afrosheensup Ariel_
00:05.31asterisk99afrosheen: 1.2
00:05.41Ariel_just stopping by to see how everyone is doing.
00:06.27ManxPowerwcfxs and wctdm are the same module
00:06.37afrosheenyep, has been for awhile
00:06.46afrosheenhence it being missing
00:07.02afrosheenasterisk99: did you do a 'make' or a 'make linux26' when you built it
00:07.32asterisk99afrosheen: I did a make linux26
00:08.27afrosheenasterisk99: did the make install throw out any errors at the end?
00:08.47asterisk99afrosheen: NOt that I noticed
00:09.15afrosheenManxPower: how will I know if fxotune worked or not
00:10.05afrosheenasterisk99: what happens when you do a 'modprobe wctdm'
00:10.34ManxPowerafrosheen, should be in dmesg
00:10.37ManxPower-- Setting echo registers:
00:10.37ManxPower-- Set echo registers successfully
00:10.40asterisk99I am getting a 'Module wctdm not found'
00:11.26ManxPowerasterisk99, that would mean that that module was not installed in the right place.
00:11.39ManxPowerI suspect you have a version mismatch between your kernel source and your running kernel
00:11.51afrosheenasterisk99: have a look here http://blog.kirjava.net.nz/posts/76
00:11.53asterisk99afrosheen: The make linux26 ends with 'leaving directory usr/src/linux-headers-2.6.12-386'
00:12.06afrosheenthat should be right
00:12.53afrosheenManxPower: sweet, it worked on reboot :)
00:13.42*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
00:15.24afrosheengah this zaptel dtmf issue is driving me up the wall
00:15.35afrosheenhalf the time it doesn't match the digits on the ivr
00:15.47asterisk99afrosheen: Am digesting link u sen
00:16.21afrosheenasterisk99: ok
00:17.00ManxPowerafrosheen, what interface?
00:17.20ManxPowerafrosheen, in analog ports messed up gains can screw up DTMF.
00:17.30afrosheenhow so
00:17.51afrosheencoz these are fxo ports with incoming business lines jacked in
00:18.07ManxPowerwell if the gain is too high, then the audio will be distorted, if it's too low, the system may not detect all DTMF digits.
00:18.22ManxPoweralso relaxdtmf=yes can mess it up too.
00:18.38afrosheenwell the strange thing is, if you call the ivr from a cellphone, the digits are always perfect
00:18.49afrosheenif you dial it from some landlines, dtmf tones get lost
00:19.06afrosheenso I guess ma bell isn't transmitting as loudly as t-mobile
00:19.17ManxPowerafrosheen, perhaps your cell company transmits DTMF at a higher or lower gain and analog phones.
00:19.40afrosheenhow do I balance that without introducing nasty echo, just bump the rx gain a little? I think it's already at +3
00:19.45ManxPowerincrese your rxgain by 4 then try, if that does not work try decreasing it by 4
00:20.01ManxPowerso if rxgain-0 then try rxgain=4 and rxgain=-4
00:20.36ManxPowerrxgain won't mess with echo, only txgain
00:20.40afrosheenwhew
00:20.45afrosheenthat's what I'm worried about
00:21.09afrosheenafter I make changes should I only reboot *
00:21.30[av]banigaaaaaaaaaaaah
00:21.36[av]baniFuriousGeorge mangled my edits
00:21.47[av]baniFuriousGeorge: "preview" is your friend
00:21.53ManxPowereither of these two ways: 1) stop and start asterisk or 2) unload chan_zap.so then load chan_zap.so
00:22.12ManxPowerreload chan_zap.so might do it, but I don't know.
00:23.14Ariel_ManxPower, don't you have to also restart the zaptel service?
00:24.22ManxPowerAriel_, why?  gains are in /etc/asterisk/zapata.conf not /etc/zaptel.conf
00:25.16Ariel_great I have always stoped and restarted the service as well. I guess I was doing more then what is needed.  Good to know.
00:25.28*** join/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net)
00:25.55themikester60has anyone here used a wrt54g with a third party firmware installed in their network setup for an asterisk server?
00:26.04ManxPowerany changes to /etc/zaptel.conf needs a ztcfg to apply the changes (which will kill all active calls)
00:26.56*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net)
00:28.04Abydos313themikester60 someone in here said they have it running on openwrt
00:28.18ManxPowerthemikester60, I suspect most people here just buy compatable hardware
00:28.34ManxPowerrather then spending 20 hours making something work that you can just buy for under $500
00:28.56themikester60ManxPower, I imagine they would, I just have a slight issue for anyone who uses a wrt54g, I've been asked to try to get the QoS features of it to work as a switch rather than a router but have had no luck so far using the alchemy firmware
00:29.05Abydos313maybe asterisk will be included in future firmware like seavofts
00:29.10MGSsanchothemikester60 i am
00:29.20afrosheenthemikester60: yeah I had trouble with dd-wrt firmware's QoS also
00:29.25MGSsanchowrt54gs
00:29.31ManxPowerAhrimanes, Wireless QoS.
00:29.35Ariel_I have uses openwrt and ti work
00:29.41ManxPowerNext they will be asking you to turn lead into gold.
00:29.42Abydos313i have freemans' basic if you want to try it
00:29.43themikester60MGSsancho, have you found a way to get the router to act as a switch while still utilizing the QoS featuers?
00:29.44orlokwireless QoS is an oxymoron :)
00:29.58MGSsanchothemikester60 no :(
00:30.20ManxPowerWhat most people don't realize is that their network doesn't need QoS
00:30.21afrosheenI'm not looking for wireless QoS, just QoS for the polycom plugged into it
00:30.27themikester60I'm not using any wireless with this router, it is simply being used a switch, QoS works as a router but I'm trying to find a firmware that can make the QoS work as a switch
00:30.41themikester60ManxPower, what makes you say this?
00:30.41ManxPoweror should I say, that their LAN does not need QoS
00:30.47MGSsanchodont use the wan port then <_<
00:31.01orlokhmm
00:31.09orlokthese grandstreams dont wanna register with *
00:31.21ManxPowerthemikester60, have you measured the data thruput of your LAN?
00:31.24Abydos313freemans basic will let you assign port priority :)
00:31.25orlokhas anybody used a GXP-2000 with asterisk?
00:31.29themikester60MGSsancho, the wan port isnt used, I have 3 ports in use, 1 and 2 should have higher priority than 3 and I have configured it so within the QoS page, but no QoS is occuring at all
00:31.32MGSsanchothat will make it a switch. then disable dchp
00:31.38themikester60ManxPower, yes I have
00:31.58ManxPowerEven using ulaw, a call will use something like .008 of a 100Mbps LAN
00:32.06themikester60Abydos313, so does alchemy, but it doesn't seem to be having any effect, have you tested freemans basic's QoS when operating as a switch instead of a router?
00:32.07MGSsanchoonly need QoS over wan when you wanna make calls and your dling "linux distros" on bit torrent
00:32.31themikester60MGSsancho, I have disabled dhcp, its working as a switch like I want it to but I'm not noticing any QoS occurring
00:33.10MGSsanchoi havent been able to get working like that unless you do static routing to get multiple routers working
00:33.12themikester60I agree with all of you on the part that QoS isnt really necessary, but the networks we install these asterisk servers on tend to have high bandwidth usage so for these instances it is needed or else the calls begin to have breakup
00:33.16ManxPowerthemikester60, there's a good chance the hardware does not support QoS between the LAN ports.
00:33.19*** join/#asterisk litage (n=nick@203.220.55.70)
00:33.38MGSsanchothats possible
00:33.41themikester60ManxPower, I had assumed so but was hoping I might find out otherwise, do you think it is because of the hardware though or is it possible a different firmware could control it?
00:33.52ManxPowerthemikester60, *nod*  If they have high bandwidth usage on the local lan, then perhaps QoS would help.
00:34.05ManxPowerOn the other hand, it might be easier to just put each phone on it's own Ethernet port.
00:34.07MGSsanchouseally it means QoS over wan since thats the slowest port
00:34.21afrosheenQoS is critical for this stuff...if the packets don't make it out of the wan to our trunking provider, people cry
00:34.27ManxPowerassuming the phones and the asterisk server are on the same switch.
00:34.40themikester60MGSsancho, I had assumed so, though within the options for the QoS you can specify QoS as either for Wan or Wan&LAN so I assumed that it might work
00:34.42Ariel_for qos to work on a local lan you need to have a good layer 2 or 3 switch in the mix
00:34.50ManxPowerafrosheen, QoS on WAN is important
00:34.53afrosheenand it's necessary for all our remote users with phones at their homes
00:35.02MGSsanchoif you have a lot of users, then you might want to get a more powerfull switch to handle the load
00:35.24themikester60I was considering a more powerful switch, but the modified firmware makes these routers handle QoS beautifully
00:35.31themikester60if only they had QoS as a switch it would be perfect
00:35.39MGSsanchowerd
00:35.41afrosheenI'd like to be able to roll out a router package that guarantees our home users don't have issues with calls regardless of what they're doing on their computers
00:35.56MGSsanchohmm
00:36.04themikester60afrosheen, thats actually pretty similar to what I'm trying to do
00:36.05MGSsanchoi see what you mean
00:36.22afrosheenI thought the wrt54g would be a good candidate so I bought one and played with different firmwares, could never get it to act right
00:36.28ManxPowerIf the Asterisk server is a dedicated machine, and the phones are not sharing their port with a PC, and the phones and the server are on the same switch, I can't imagine QoS helping any.
00:36.41afrosheenManxPower: that's probably true
00:37.05MGSsanchoi agree
00:37.19MGSsanchohave the phones on a powered QoS switch
00:37.22ManxPowersince if the switch is doing it's job, phones<->server won't have to contend for bandwith
00:37.26themikester60ManxPower, I have that setup, but the PCs are peers to the asterisk server, so traffic theyre generating through the switch is killing calls
00:37.29MGSsanchoand use a another one for PCs
00:37.47ManxPowerthemikester60, define "peers"
00:38.41ManxPowerMGSsancho, why a QoS switch
00:38.52MGSsanchogood point
00:39.04MGSsanchoput phones on a POE switch
00:39.05themikester60ManxPower, the two networks converge on one point which is the switch (wrt54g) which I'm trying to get to utilize QoS, each network has its own router with its own static IP, so one dsl line with two static IPs. When traffic gets too demanding on the PC network, it effects the quality of the cals
00:39.15ManxPowerNow, if the phones and the PC are sharing the same port, etc.
00:39.28themikester60MGSsancho, forgive me for sounding stupid, but what exactly is a POE switch? Power over ethernet?
00:39.33MGSsanchoyes
00:39.34ManxPowerthemikester60, of lan calls or wan calls?
00:39.40themikester60wan calls
00:40.03ManxPowerWell, yes, for WAN you would want to do some QoS
00:40.27ManxPowerI thought these phones and the server were all in the same location, since you talked about QoS between the LAN ports
00:40.36themikester60ManxPower, right, and thats why I was trying to get the wrt54g to use QoS when acting as a switch, but so far the QoS only seems to work when its operating as a router
00:40.41MGSsanchomay also have to keep them separate. have the phones on their own dsl line. and pcs on another
00:40.47themikester60oh no, sorry about that I shouldve clarified
00:41.12MGSsanchosince each call is like 10K, and in the US, we only get like 30K up :(
00:41.14*** join/#asterisk jpablo (n=jpablo@201.139.54.46)
00:41.22themikester60MGSsancho, that would make my life much easier, but my boss seems to want to conserve and only use one DSL line, because the voice really doesnt consume much bandwidth, but it does get affected if too much is being used
00:41.25ManxPowerUnless you control the other end of your internet access line, QoS will still be somewhat limited.
00:41.33jpablohey people, any idea what' current chan_bluetooth cvs ?
00:41.38ManxPowerSince you can only apply QoS to TRANSMITTED traffic.
00:42.06themikester60ManxPower, well I have control over it to some degree, enough that when I had QoS working as a router and I had the bandwidth being used to the maximum the calls suffered no loss
00:42.08MGSsanchotrue
00:42.20ManxPowerthemikester60, you are just starting to realize why I don't put calls on a WAN
00:42.39afrosheensomeone needs sdsl
00:42.58MGSsanchotell ur boss that dsl is like $14 a month. and if you want to expand as a buisness, the current dsl can only handle so much traffic
00:43.10ManxPowerAnd why I turn down cheap customers that want me to do a half-assed job.
00:43.18MGSsanchommmm sdsl
00:43.24MGSsanchoor get a T1
00:43.41afrosheenif he's too cheap for sdsl or an upgrade, he sure as hell won't get a t1
00:43.49afrosheenhow many people work there, 4 or 5?
00:43.51ManxPowerMGSsancho, if they won't spend money on a real switch, they are not going to spend money on a T-1/E-1
00:44.03MGSsanchoi know trying to be funny :P
00:45.13Nuggetit's not as effective as outbound, obviously, but it's much more effective than you'd expect.
00:45.43Nuggetfor sites where tcp is the bulk of the traffic, it's quite feasible to have useful inbound queuing
00:46.22Nuggetthat said, I think it's downright foolish to want to do business voip over the internet.
00:47.41ManxPowerCTRL-A DEL  There, all caught up on asterisk-users.   And people wonder why I don't want web based "forums"
00:48.03Nuggetweb forums are the cesspools of the internet.
00:48.30jpablogrr, where's people getting channel bluetooth
00:49.11MGSsanchothis chan is about VOIP systems
00:49.17MGSsanchonot bluetooth sorry
00:49.27Nuggetchan_bluetooth is an asterisk module, you doof.
00:49.29jpablohumm, isn't this channel about asterisk ?
00:49.33Nuggetthat couldn't be more on-topic
00:50.03afrosheenargggh this dtmf stuff is unfixable
00:50.23jpabloafrosheen, what's the problem?
00:50.28themikester60sorry I walked away for a bit, the more expensive connection resolution doesnt really work here because my boss is trying to implement QoS as more of a "Save Our Ass" feature so that the customers never have to complain about call quality issues
00:50.35ManxPowerPerhaps by reading this page: http://www.asterisk.org/developers
00:50.43afrosheenmy zaptel interfaces (fxo) don't register dtmf tones correctly most of the time
00:50.48themikester60thats why I'm trying to get a QoS switch, we've tried more expensive ones in the past but none have ever quite worked well enoug
00:51.00afrosheenI've tried everything from rxgain = -4 to rxgain = 10 and nothing changes
00:51.02themikester60the QoS on the wrt54g worked well enough, but it appears so far to work only when its functioning as a router
00:51.13ManxPowerUm, if you don't want customers to complain about call quality, don't send the calls over the Internet
00:51.13themikester60but one of our installs needs it to work as a switch, thats where I find myself up river without a paddle
00:51.16jpabloafrosheen, analog or {e,t}1?
00:51.27afrosheenjpablo: analog, tdm400 card
00:51.34themikester60ManxPower, yea but the whole focus for us is VOIP
00:51.47themikester60ManxPower, thats how we supply the majority of channels for the users
00:51.51jpabloafrosheen, works fine for me ...
00:52.03ManxPowerthemikester60, What are you going to do if the box simply cannot do LAN QoS
00:52.10afrosheenjpablo: lol thanks buddy
00:52.20*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
00:52.31*** join/#asterisk welles (n=welles@61.150.43.113)
00:52.44themikester60ManxPower, most likely find a different switch solution, something is bound to be able to work as a QoS switch, I'd just rather find a solution with this switch and move on to more important issues
00:52.50welleshi al
00:53.08afrosheenarg this is so retarded, it gets the digits from my cellphone perfect every time, but not from an inbound landline call
00:53.16jpabloafrosheen, doesn't zaptel has something liek relaxdtmf ?
00:53.24afrosheenrelaxdtmf didn't help either :(
00:53.48*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
00:53.51jpabloafrosheen, how many diferent phones have you tested ? maybe the landline phone is b0rked ...
00:53.54afrosheenI need to just get our business lines ported to commpartners and take a big hammer to this tdm400 card
00:54.10ManxPowerSorry, from here: http://www.asterisk.org/download
00:54.11afrosheenjpablo: about 10 different landlines, all sbfc
00:54.18ManxPowerrelaxdtmf selcom helps
00:54.21afrosheen<PROTECTED>
00:54.26welleswhy my asterisk can not receive packets from another asterisk using iax2 protocol?
00:55.11jpablowelles, that's network question not really a * question.
00:55.48robin_zhere, you'll be needing this
00:55.53jpabloafrosheen, does zaptel has a jitter buffer ? maybe disabling it will help
00:56.21ManxPowerzaptel has something called a jitter buffer, but it's not one, in the sense of VoIP
00:56.42*** join/#asterisk wellng (n=welles@61.150.11.54)
00:56.47ManxPowerafrosheen, can you fax over the lines?
00:57.06ManxPowerafrosheen, are you talking about RECEIVING DTMF or SENDING DTMF
00:57.11jpabloafrosheen, is your card missing interrupts somehow ?
00:57.31afrosheenreceiving dtmf
00:57.44glm2kafrosheen: does a verbose setting show the dtmf digits being pressed?
00:57.46afrosheenand it's only having trouble receiving from landlines, we don't use any faxing
00:58.07ManxPowerbecause if you are trying to send DTMF to remote IVRs then that's the classic DTMF length time problem, see digits.h in the zaptel source, change the DTMF length, recompile
00:58.11afrosheenglm2k: it shows a few, i.e. I dial 235 and it sees 255 or 25 or 355
00:58.45ManxPowerafrosheen, classic rxgain probleml.
00:59.01ManxPowerI had to bring my rxgain up to 8 before my asterisk would see DTMF from the PSTN
00:59.08glm2kouch
00:59.37afrosheenManxPower: is this related at all to the settings fxotune writes out
00:59.43ManxPowerglm2k, I believe the audio level coming from the telco is low.
00:59.50afrosheenmine wrote 8,0,0- for each module
00:59.59ManxPowerafrosheen, no.
01:01.49*** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net)
01:03.17*** part/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net)
01:04.17afrosheenlol I turned the rxgain up to 50 and now it won't detect anything
01:06.24glm2ktime to complain to sbc?
01:06.31afrosheenyeah I suppose
01:06.44*** join/#asterisk fjean (n=fjean@201.29.122.10)
01:06.44afrosheenanyway I need to get these numbers ported away from sbc
01:06.55afrosheenI fully trust commpartners now
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01:07.34*** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
01:08.28fjeanHello guys, I need some help regarding DeadAGI, apparently when I dial directly an IAX2 route from extensions.conf I hear the ring tone, but the same route from a DeadAGI would not give me the ring tone...anybody has a hint ?  :- )
01:09.25fjean(I use 1.0.9)
01:10.19websaeis it possible to have sip and iax connections at the same time with an asterisk box?
01:12.00jpablowebsae, yes
01:12.20jpablothat's the whole fun of asterisk
01:13.36websaei need to setup an client asterisk box to connect to mine via sip
01:13.41websaewhat's the best way to do this?
01:14.00*** join/#asterisk welles (n=welles@61.150.60.46)
01:15.36*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
01:15.48jpablocreate friends in both ends, and voila
01:16.29websaeon my end...have a user setup with type=friend
01:16.44websaeand then on his end, setup the same exact friend
01:16.46websaewith same settings
01:16.48websaecorrect?
01:17.01jpablonot same settings, the ip will change
01:17.06websaehost= my ip on my end
01:17.12websaehost=his ip on his end
01:17.19jpablothat should work
01:17.23websaedo i set his ip on my box
01:17.26jpablothen just dial(SIP/peername/bla)
01:17.40jpablojust like you typed
01:18.03websaei am just trying to understand how it knows to peer to my server
01:18.20websaei think i would have to type is ip as host on my server
01:18.22jpabloyou host = bla in his side ?
01:18.31websaeand he'd have to type my ip for host on his
01:18.42*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
01:18.50websaeohh---- host=hisip    on my* server
01:19.05websaeand host=my ip on his*server
01:19.13jpabloyes
01:19.29websaeokay
01:19.37websaedon't need a register command or anything?
01:20.11welleshi all. now i want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but  the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while
01:20.11wellesiaxclient can not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated!
01:21.34websaenat
01:21.34jpablowebsae, you only need to register if you are in a dynamic ip
01:21.46websaehe has a dynamic ip address
01:22.13jpablothen you configure in your vox host = dynamic and he must put a register command
01:22.28jpablos/vox/box
01:22.33websaewhat would hist register command be?
01:22.40websaeregister =>
01:22.43websaewhat the
01:22.46fjeani am not sure if i got it, but I had this experience too where if I use an * box and softfone out of the same public IP to connect to another public IP I would not be able to call back the * box that has the same public IP as my softfone, it would teel me "in a call"
01:22.48websae*then
01:23.25jpablodon't remember search the wiki or see the examples that are already in sip.conf
01:23.45jpablois basically will have a user name (the peer name in your box) a secret, and your ip
01:25.02websaejpablo: any familiarity with iax?
01:25.07jpabloyes
01:25.14websaeis that hard to setup?
01:25.21jpabloalmost the same as sip
01:25.33websaewhat's different
01:25.43jpablohumm, it's called iax :P
01:26.00jpabloso you use dial(iax/peer) :P
01:26.33jpabloreally the configurations are basically the same, the dial command gets more options as it understand more options on the remote box like contexts and stuff
01:26.49jpablobut at a configuration level they are basically the same
01:27.19wellesjpablo, will u help me?
01:27.28websaeiax more stable then sip ?
01:27.47jpablomaybe with especific questions im having my own fun
01:28.03a1fahey
01:28.07jpablohumm, same stability, it uses just one port so it doesn't have all the rtp nat nightmares
01:28.10a1fawhat is that voice mail menu?
01:28.14a1faVoiceMailMenu()?
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01:29.07jpablovoicemailmain
01:29.17a1fauff.. thats what i ment..
01:29.30a1fait wont move me into menu.. it wants an extension
01:29.44a1fai want it to be a menu where you put your extension
01:29.46jpabloyou need to configure voicemail.conf
01:29.57jpablowhat do you mean ?
01:30.04*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-20-52.buckeyecom.net)
01:30.11a1favoicemail.conf is configured
01:30.12*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
01:30.46jpabloa1fa, i didn't understand you, what do you want to do
01:31.27a1faFeb 20 01:31:16 WARNING[11776]: pbx.c:1688 pbx_extension_helper: No application 'VoicemailMain ' for extension (myphones, 28346, 1)
01:31.32a1fathis is the error i get now
01:31.54jpablohumm
01:32.18jpablodo you have /usr/lib/asterisk/modules/app_voicemail.so ?
01:32.19fileyou have a space... in your extensions.conf entry that calls VoicemailMain
01:32.20a1faweerd
01:32.23jpabloah, wait
01:32.27jpablothere a space, corrent
01:32.35a1faits there
01:32.40jpablodelete it
01:32.43a1fabut for some reason it is pwned by root
01:32.52a1fadelete it why?
01:33.01filehe meant delete the space
01:33.06a1faok
01:33.13fileyou have something like:
01:33.18a1faright i see it now
01:33.19a1fa:(
01:33.23fileyeah... fix it
01:35.05a1fainteresting
01:35.08a1fait is not authenticating
01:35.36a1fauff it is
01:35.48a1fareally interesting
01:36.01*** join/#asterisk in-side (n=lowgitek@es-217-129-30-48.netvisao.pt)
01:36.10in-sideHi anybody here works with ser or openser?
01:36.39in-sideI was wondering if anybody knows how to concat two avp strings to only one?
01:36.41FuriousGeorgehmmmmm, so how hard would it be to monitor parking spots from the manager and set dev states based on that?
01:36.45a1faare you guys using md5secret or clear secret?
01:37.01FuriousGeorgecuz im not gonna be able to get it working in the dialplan just right
01:37.09FuriousGeorge(with meetmes)
01:37.39in-sidenobody here use ser? or openser ?
01:39.57fjeanhi, anybody uses AGI->Exec("Dial...  with the ring option ('r') in his script ?
01:42.04in-sidewhy ?
01:42.20in-sideI use it
01:42.25in-sidewazz up?
01:42.34[av]baniFuriousGeorge: "preview" is your friend
01:43.18fjeanin-side, it seems the 'r' option doesn work, we can hear nothing until the person answers...
01:43.53*** join/#asterisk wellng (n=welles@219.144.238.194)
01:44.17fjeanin-side, this how i do it:  $dialstr = "IAX2/$res->{path}/$phone|60|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)|r";
01:44.42*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
01:44.56in-sidesorry i don't work with iax but let me check it..
01:45.08fjeancool
01:45.15in-sidewhy ":60000:30000)
01:45.16filefjean: remove the | in between ) and the r
01:45.34fjeanok..let me test
01:46.05in-sideno man the | is need
01:46.20FuriousGeorge[av]bani: preview?  preview what...  btw, i decided the only way its gonna work right is with an API, so I may as well start reading
01:46.59*** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
01:47.16*** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
01:47.29fjeanok
01:47.47filefjean: did it work?
01:47.55in-sideya...
01:47.59fjeani am switching user, 2 secs
01:48.00in-sidethe | is not need sorry
01:48.03in-sideit is a option
01:48.09in-sideas r
01:48.15in-sideso should be together
01:49.10in-sidewell nobody works here with ser sip proxy server?
01:49.31in-sideor openser ? and knows how to contact two avps to only one?
01:49.32fjeanin-side, nope, both ways don't work...
01:49.43fjeanthis is so trange
01:49.58fileI'd ask why it's strange, but it'll just end up with me getting irritated
01:50.24fjeanhehe
01:50.33in-sidefjean: try to only send r
01:50.34fileso I'll do it anyway
01:50.37filewhy is it strange?
01:50.39in-sideto see if it works
01:50.52fjeanright..let me check
01:51.03fileI could give you a speech about what the difference is...
01:51.53in-sideor try to put r first
01:52.39*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
01:52.55fjeanthis guy did not ring:  $dialstr = "IAX2/$res->{path}/$phone|60|r";
01:53.15fjeanI am using a SIP user, could that be translation problem ?
01:53.25in-sideerr.. anyway r option sux
01:53.34in-sidehave no ideia just use sip on it
01:53.38in-sidemaybe codec
01:53.41fjeani know...
01:54.06fjeanwell, maybe the codec, i ll try a couple
01:54.08filein-side: you're an interesting individual... but anyway
01:54.26filefjean: what exactly is the situation, what's happening that should/shouldn't be happening
01:54.54in-sidefile: thanks for your opnion about myself but I didn't asked it...
01:55.39in-sidefjean: why are you forcing the it ring anyway?
01:55.42[av]baniFuriousGeorge: voip-info. your edit borked my feature request
01:55.58[av]banisnom-360
01:56.32fjeanin-side, because when i dial from the script, the caller cannot hear ringing...but, the funny thing is that if i dial directly from extensions.conf it gives it
01:57.33in-sidehmmm... sorry can't help you more, all my sip signals handling are done in openser
01:57.44fjeancool, no prob
01:59.02*** join/#asterisk Katty (n=angela@64.82.232.54)
01:59.14Kattyhihi
01:59.23Ariel_Katty, how are you?
01:59.33Kattygetting better.
01:59.36Kattyi think.
01:59.47Kattyswelling should peak tonight
02:00.15orlokchan_sip.c: Registration from '<sip:1234@192.168.1.243;user=phone>' failed for '192.168.1.211'
02:00.21in-sidefjean: progressinband ?
02:00.32orlokAny ideas? grandstream gxp-2000 to asterisk
02:00.43in-sideis true for sip client?
02:01.09fjeanin-side: mmmmm, good question, i guess its the default value, letme check
02:01.29in-sidedefault is no
02:01.31Ariel_swelling??? what did you do? Katty
02:01.32in-sideso checkit out
02:01.51in-sidetake care that inband must be also set in sip client
02:01.58fjeanok
02:02.05in-sideand not all codecs supports it properlly
02:02.11KattyAriel_: surgeons cut out my wisdom teeth friday
02:02.25justinufile: what are these two talking about?
02:02.26Ariel_argh sorry... I understand
02:02.31justinublind leading blind?
02:02.38Kattyand file won't come take care of me.
02:02.40Kattysniffle.
02:02.56filejustinu: I like my ignore list
02:02.59justinuhah
02:03.47fjeanin-side, isee it goes in general settings, not in individual blocks
02:04.14in-sideKatty: tas a ver a ideia nao tas... tao prontos.. Vai te lixar o meu caralho!
02:04.32*** join/#asterisk websae_ (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
02:04.47Kattyin-side: i speak english.
02:05.02fjeanin-side, que palavrao feio  ;-)
02:05.10fileI speak I/O talk
02:05.20in-sideKatty: tao o problema é teu! err.. not my problem sucker
02:05.28in-sidefjean: anda aqui po pvt
02:06.00*** join/#asterisk nozey (n=nozey@20150042085.user.veloxzone.com.br)
02:06.35websae_i don't get why my phone sometimes has great sound quality---and then voice quality is gargley--gets lound and quiet, loud and quiet----any ideas anyone?
02:06.48nozeyhi ... im trying to run asterisk here, i can dial my friend, but theres no sound ... i already opened 5060 udp(and tcp) at my router ... what can be wrong?
02:06.49justinuwrite(fd,"english, muthafucka, do you speak it?",53);
02:06.56nozeysorry for my english ... im brazilian #)
02:07.13filejustinu: yes
02:07.15file:P
02:07.16nozeyim using sip
02:07.18justinucool
02:07.50in-sidenozey: nao ligues pa estes gajos sao todos eliteee... err..
02:07.56Ariel_wow google could not translate that
02:08.04Kattyin-side: err, k
02:08.05in-sidefile: ha um bug nisso pa
02:08.32NewSoleon a sip call how can I pass it from asterisk server to outside server
02:08.48justinuquite simply
02:09.24*** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net)
02:09.25in-sidehttp://bugs.digium.com/view.php?id=3695
02:09.27nozeyheh in-side voce é espanhol?
02:09.39in-sidenozey:  nao mane sou brasileiro
02:09.46Kattyi don't have the patience for this
02:09.48mzosomeone give me a sledge-hammer
02:09.54mzoi want to smash this 9800 all in-wonder pro to bits
02:09.57in-sideKatty: really cvall you mum
02:10.01Kattyespecially not after having my widom teeth out.
02:10.06nozeyi can hear the ring, but theres no voice sound
02:10.06Kattyin-side: she's right here, hunny.
02:10.08justinuKatty: i don't blame you... it's tiring even without the mouth pain
02:10.32Kattyjustinu: yeah, the stitches feel insanely tight.
02:10.34in-sideKatty: whatever...
02:10.53mzobut insanely tight is sometimes a good thing
02:11.03Kattymzo: yeah, but not with stitches and swelling
02:11.06justinuhah
02:11.19Kattywell, maybe swelling
02:11.25Kattybut definately /not/ stitches
02:11.44websae_any suggestions why my phone sometimes works great..no problems, but then in the middle of a call on and off it gets loud and soft, and voice becomes gargly????any suggestions, anyone? :) greatly appreciated
02:11.50I-MOD!mp3
02:11.56in-sidefile: are you there???
02:11.59Ariel_websae_, network issues
02:12.03I-MODsry
02:12.04Ariel_crappy phone?
02:12.04I-MODnm
02:12.16websae_Ariel: have a grandstream
02:12.23Ariel_then the 2nd
02:12.25websae_i use to have a sipura---it was even worst!
02:12.41Ariel_sounds like then it's your network what's it like
02:12.42[av]banisipura 841 :D
02:12.59websae_any suggestions
02:13.04*** join/#asterisk fjean (n=fjean@201.29.122.10)
02:13.10Ariel_websae_, more info needed
02:13.19websae_what info?
02:13.27websae_i connect with g729
02:13.33websae_to a server in LA---im in wisconsin
02:13.41websae_the server is on a gig-e connection out there
02:13.43Ariel_are you on cable
02:13.49websae_static ip dsl
02:14.03Ariel_what is your ping times
02:14.10websae_80ms
02:14.14Ariel_also what firewall/nat router are you using?
02:14.27websae_dlink
02:14.31Ariel_try ulaw and see how that one sounds....
02:14.32websae_but the phone registers fine
02:14.49Ariel_sound goes out on rtp ports between 10,000 to 20,000
02:15.06websae_ulaw---what's the speed on that
02:15.23NovceGuru80kbps I think
02:15.24Ariel_it's 80k but it's sound is better no compression needed
02:15.26NovceGuruwith overhead
02:15.45Ariel_if your on a static IP with dsl you should be able to do ulaw
02:15.45nozeyi have no audio using sip ... can someone help?
02:15.54nozeyim behind a router
02:16.01Ariel_nozey, sure have you looked up on the wiki about nat
02:16.04Ariel_~doc's
02:16.19nozeyyep
02:16.19websae_iax---the answer to everything
02:16.29websae_no worries about rtp and nat
02:16.29Ariel_websae_, no it's not
02:16.34nozeywell ... first i need to learn sip
02:16.40nozeythen im going to try iax
02:16.49Ariel_nozey, sip registers on port 5060 or 5061
02:17.03Ariel_they are needed to be open for registation
02:17.16websae_yep
02:17.17nozeyi opened 5060 both udp and tcp
02:17.23Ariel_are the ports forwarded to the asterisk box
02:17.26nozeyi can dial
02:17.27websae_okay---then something with your voice packets
02:17.27NewSoleon a sip call how can I pass it from asterisk server to outside server
02:17.32nozeyyes Ariel_
02:17.40websae_<PROTECTED>
02:17.42websae_with your provider
02:17.43nozeyi can dial ... just can hear the other people speaking
02:17.44in-sidenozey: check pvt
02:17.50Ariel_how about rtp.conf what ports are there? have you forwarded them?
02:18.27Ariel_again sound is via rtp from the range of 10,000 to 20,000 depending on device
02:18.41nozeywell
02:18.46in-sidethe default starts at 8000 i think
02:18.52Ariel_this can be set by editing the rtp.conf file located in the /etc/asterisk directory
02:18.59Ariel_in-side, really
02:19.01*** join/#asterisk tengulre11 (n=tengulre@221.11.5.180)
02:19.04nozeycan i open only the 10.000 port?
02:19.10in-sidenao
02:19.11in-sidenopes
02:19.13nozeyor should i open the range?
02:19.21in-sideyou need a range.. same srange
02:19.25in-sidethat is in rtp.conf
02:19.30in-sideas Ariel_ told you
02:19.30Ariel_10,000 to 11,000 set it up on the rtp.conf for asterisk then forward the ports
02:19.36nozeyok
02:19.37nozeylet me try
02:19.39nozeyone minute
02:19.42Ariel_you don't really need 10,000 ports open
02:19.53in-sideya..
02:20.58Ariel_wow just checked out the first server I setup with asterisk it was back in april 2002.... almost 4 years ago....
02:21.05nozeyudp, tcp or both?
02:21.10Ariel_udp
02:22.24Ariel_you might have to check your device like phone sipura likes in the range of 16,380 and cisco like 8000 range.  (Seems like they don't like getting a standard setup).
02:22.45websae_peering two asterisk servers...what does one have to do
02:23.29Ariel_peering is fairly easy.  if you only have a few channels needed use iax2 if not then more then 8 use sip
02:23.45nozeydidint worked
02:23.57gaupewebsae_: http://www.voip-info.org/wiki-Asterisk+-+dual+servers
02:23.59Ariel_nozey, what devices
02:24.35websae_on server a, make entry for server b and on server b make entry for server a?
02:24.49Ariel_websae_, pretty much
02:25.00Ariel_nozey, which version of asterisk do you have?
02:25.23websae_server b is dynamic----so i have to do a register on server b with the credentials from the entry on server a?
02:25.23nozey1.2.4
02:25.45nozeyi have 5060 and 10000-11000 opened and forwarded to my pc
02:25.57Ariel_your pc
02:25.58nozeymy friend is behind a router too
02:26.06nozeyyes ... im running asterisk here
02:26.10nozeyonly for learn
02:26.24Ariel_then you might have a problem with your friends router
02:26.41Ariel_nozey, setup a free fwd account and do some testing from that
02:26.51nozeylet me try
02:26.54nozeytkz again
02:27.00Ariel_double nat is a bear to fix
02:27.11nozeyi can see that
02:27.15nozey:(
02:27.17Ariel_it can be fixed but takes a bit
02:27.48Ariel_are you running it on vmware?
02:28.24nozeynop
02:28.28nozeyslack 10.2
02:28.39*** join/#asterisk wellng (n=welles@61.150.43.113)
02:28.44nozeyon local machine, not vmware
02:28.55nozeysorry for me english ... i know it sucks :(
02:29.38Ariel_well it's better then my brazillian
02:30.17nozeyheh :P
02:31.02nozeylike what?
02:31.05websae_dropped calls...what's the best way to debug that
02:31.10websae_with 'sip debug'
02:31.14websae_and just screen that
02:31.19*** join/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com)
02:31.20Ariel_ethereal
02:31.36websae_what would i do with ethereal?
02:31.42websae_i think i have that on my box
02:31.46kc5cqmquick question:  I'm ztcfg isn't configuring a channel, but zttool is finding the unconfigured x101p car
02:31.57kc5cqmdebian 3.1
02:32.04nozeyis ethereal a gui for tcpdump?
02:32.22Ariel_ethereal gui hehehe
02:32.31WasPhantomwell, ethereal uses the same pcap library to capture the packets
02:33.03Ariel_or ngrep could help as well
02:33.06mzoi'm being stupidly brave and configuing FWD for asterisk :P
02:33.12mzosomeone call me and see if something good happens?
02:33.17websae_what command do i run with ethereal
02:33.20websae_to do this debuggin?
02:33.25Ariel_number
02:33.33mzo749414
02:33.43mzomaybe it'll blow up :P
02:34.04Ariel_no just busy
02:34.13mzohmm, is that their end or mine?
02:34.18websae_do i just type ethereal at the command line?
02:34.25websae_or do i need to specify something with ethereal?
02:34.31Ariel_webmind, google how to use ethereal
02:34.33mzowhat'd you get?  a busy signal or an asterisk prompt?
02:34.40Ariel_busy
02:34.40websae_webmind?
02:34.42websae_whose that?
02:35.53nozeywell
02:36.03*** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net)
02:36.08mzoim sure i fucked something up :P back soon with the 2x4
02:36.09nozeyi just tested with a friend that is not behind a router, and he can listen to me :-)
02:36.34Ariel_nozey, ok so he can hear you. But you do not hear him
02:36.49Ariel_which means your rtp is not coming back to you outbound is ok
02:37.02nozeyyep
02:37.06nozeyrouter problem
02:37.23Ariel_ngrep and ethereal are programs you run in the command line
02:37.41justinutethereal is cli
02:37.45justinuethereal is gui
02:38.54Ariel_http://www.ethereal.com/docs/man-pages/tethereal.1.html
02:39.44Ariel_nozey, most routers will open and leave the ports open when it goes out
02:42.57mzohmm, iax status: rejected = bad? :P
02:43.29Ariel_mzo, well yes
02:44.11mzohaha, i must have made a typo :P
02:45.22mzoi wonder if there is a lag
02:46.56*** join/#asterisk dmaust (n=dmaust@adsl-67-120-175-165.dsl.lsan03.pacbell.net)
02:46.59dmausthi
02:48.37mzoanyone got an idea? :P
02:48.37dmaustI just set up Asterisk a couple days ago and was trying to connect to FWD.  When I set it to verbose output, it said "IAX2/192.246.69.186:4569/7 is circuit-busy".
02:48.37mzohttp://pastebin.com/563801
02:48.42dmaustWhat does that mean?
02:49.10mzono idea, i get rejected :P
02:49.13mzooh wait
02:49.14mzoduh
02:50.01dmaustafter that message, I get Registration of '749404' rejected: Registration Refused
02:51.16mzoi checked my email i thought maybe i had to confirm, but nothing, same error :P
02:51.18dmaustmzo, what were you trying to do?
02:51.32mzooh just connect using the instructions on the site :P
02:51.51dmaustI see
02:51.58*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
02:51.59dmaustexactly the same as what I was trying to do
02:52.44mzooh well
02:53.29dmaustI just know 55555 is tech support :-).
02:53.39mzoi'll mess with it tommorow
02:53.40dmaustunfortunately, it's rejecting my login
02:53.44dmaustone idea...
02:54.10dmaustthey do have an option to activate IAX in their Extra Features section
02:54.15mzoi'llmess with it later
02:54.44dmaustokay, good luck.
02:54.49mzoi just turned that on
02:54.51mzolet's see
02:55.09dmaustweird...
02:55.12dmaustI just turned it off
02:55.13dmaustand now it works
02:56.06dmaustwell.. not quite, but I did get Registered 'dmaust' (AUTHENTICATED) at 192.168.5.128:4569
02:56.08mzoi dunno.
02:56.10mzohaha.
02:56.12mzolucky you
02:56.41*** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net)
02:57.22dmauststill doesn't work
02:57.33dmaustThat message just got my hopes up for a second.
03:01.39dmaustoh.  That was a totally unrelated message.  Since I turned the verbosity up, now it notifies me when my other IAX2 client re-registers.
03:02.56Abydos313hey guys, i have asterisk already installed and want to start using cvs.. do i need to remove before i setup cvs
03:05.47*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
03:05.48*** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300771.sympatico.ca)
03:07.06asteriskmonkeyi need some help with an echo issue.. the issue is the caller from the pstn side gets echo .. caller => pstn => pri => asterisk box => sip phone
03:07.50wellnghi all. i meet problems. just like this url describle http://lists.digium.com/pipermail/asterisk-users/2003-December/030787.html. anyone would like to help me?
03:08.04*** join/#asterisk TuckerAdel (n=Tucker@58.160.196.17)
03:09.17TuckerAdel<PROTECTED>
03:11.48*** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net)
03:12.04*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net)
03:12.06wellnganyone will help me:my iaxclient receive too many vnak frames.
03:14.41*** join/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2)
03:15.28nozeysip + nat is a pain
03:15.35*** part/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2)
03:15.45*** join/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2)
03:16.02TuckerAdel<PROTECTED>
03:17.05Aum-AumI get the following error starting Asterisk
03:17.09Aum-Aum<PROTECTED>
03:17.22orlokMy grandstream is failing to register - ANy hints?
03:17.25Aum-AumModules load and ztcfg finds ports
03:17.51asteriskmonkeyanyone know of any toronto telco 1004hz test phone numbers?
03:19.18TuckerAdel<PROTECTED>
03:19.58TuckerAdel<PROTECTED>
03:22.00nozeyAriel_, tkz for the help ... i will try agian later
03:22.03nozeysee ya
03:26.26*** part/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com)
03:26.39tengulre11Hi,all
03:26.58tengulre11I want to building a asterisk system without hardware, anybody can give me some tips?
03:27.24TuckerAdel<PROTECTED>
03:30.58Aum-Aumtengulre11 - you mean no Zaptel hardware
03:31.13Abydos313ztdummy needs to be loaded for timing issues
03:31.26Abydos313so you have to install zaptel anyways according to book
03:31.47rtanyone here running asterisk on openwrt?
03:32.08rtjust looking for a thumbs up or thumbs down on it before I go to the effort of trying it out. :-)
03:34.12shido6Aum-Aum 2.6 + ?
03:35.00Aum-AumYes
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03:35.50Aum-AumShido6 -  Yes 2.6.14-1
03:36.12shido6README.udev is your friend
03:36.15Aum-AumShido6 - Software versions
03:36.15Aum-Aum<PROTECTED>
03:36.15Aum-Aum<PROTECTED>
03:36.15Aum-Aum<PROTECTED>
03:36.15Aum-Aum<PROTECTED>
03:36.15Aum-Aum<PROTECTED>
03:36.17Aum-Aum<PROTECTED>
03:37.07shido6in teh zaptel dir you can make config after you make install and run /etc/init.d/zaptel restart to get better results... after you've editted /etc/zaptel.conf of course.
03:46.27Aum-AumShido6 - udev all ready set
03:46.40shido6kewl
03:46.50shido6zaptel all compiled?
03:46.58Aum-Aumyes
03:47.02shido6make install
03:47.04shido6done ?
03:47.11shido6make config  done, too ?
03:47.32Aum-Aummake clean, make linux26, make install, make, make config   all done
03:48.00shido6what kind of zaptel gear do you have installed?
03:48.06orlokwtf
03:48.08Aum-AumUsing init.d/zaptel gives same result
03:48.14Aum-AumTDM2413E
03:48.23shido6oooh, fun!
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03:49.03shido6how many FXS modules and how many FXO modules are REALLy on the board?
03:49.04orlokicmp dest. unreach can be sent as a response to a udp packet, correct?
03:49.39orlokhmm, but theres no udp packets in this dump..
03:49.59Aum-Aumone green and three red (1 FXS and 3 FXO)  FXS in slot one
03:50.25Aum-AumSlot one being closest to bracket
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04:12.51nozeyzael, man ... this is a pain
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04:26.16Mavvieaha. now that makes sense....
04:26.53Mavviethe fact that ^W deleted until the beginning of the line instead of to the next non-whitespace place is an emacs thing
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04:32.25jeebusroxorshaha badass - dual output is amazing
04:33.33websae_how does one setup a peer?
04:33.38[hC]dual output?
04:34.05websae_have the same entry in each server?
04:34.16websae_and then host=the opposite server?
04:34.23websae_type=friend?
04:35.22[hC]voip-info.org has an example labelled 'dual servers'
04:35.23[hC]i think
04:35.34websae_hrm
04:35.53websae_very confusing
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04:47.28websae_how's everybody doing?
04:49.16Mavvielooks like the removal of ChanIsAvail and just pushing everything through zap/g2 has resolved my congestion problems.
04:49.33Mavviequestion of course is, like always: why?
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04:55.55UberbotHi all.
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04:58.29UberbotSome time ago, I heard a rumour that they were going to re-write the voicemail system in *.  Is this true?
05:00.31rajivUberbot: i heard the same thing. there is a wiki page about it
05:01.06UberbotI'll look.  If not, I'm thinking of doing something different as an agi...  Any thoughts?
05:01.48Corydon76-homeSome time ago, the voicemail system WAS rewritten.  We're on version 2.
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05:23.40Ubeyguyhello can someone help me with this: Xlite - 21:19:41.9 Proxy slot #1 () - Failed to register! error-code: 403, msg: 'Forbidden' ?
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05:35.44Mavviehmm... new syntax of CUT is euhm... very unusual
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05:52.07UbeyguyMavvie: ??
05:56.45Mavviewell, if you're used to "Cut(foo=bar,,2)",then "Set(foo=${CUT(bar,,2)})" is something you need to look at once or twice before you understand it.
05:58.13Ubeyguyi did
05:58.19Ubeyguyi cant figure out whats wrong
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05:59.45wellngmy iaxclient send Excessive VNAK frame .anybody know why?
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06:19.32[av]bani...
06:21.53*** join/#asterisk Z-Knight (i=user@cpe-67-10-25-30.houston.res.rr.com)
06:23.24Z-Knightstupid question - What are some motherboards you'd suggest for an * setup (possibly with TDM2400P card) ?? please?
06:24.51MGSsanchoany?
06:25.17Z-Knightanyone here?
06:26.24Z-Knightdon't all of you respond at once!  ;)
06:27.24MGSsanchoany mobo will do
06:27.31MGSsanchoall you need is a PCI slot
06:27.36Z-Knightnot really...it needs the right pci slot
06:27.47MGSsanchoand enough room in your case for that long as board
06:27.50Z-Knightbut tthe TDM2400P has a particular slot...am I wrong?
06:28.05Z-KnightI'm looking at the pictures and it looks like it won't fit most boards
06:28.12Z-Knightthat's what has me confused
06:28.33MGSsanchoits pci 2.2
06:28.38MGSsanchoany new comp will have it
06:28.52MGSsanchojust need to make sure you have clearence
06:28.52Z-Knightok...here is a stupid observation then
06:28.58Z-Knighthttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P
06:29.01Z-Knightlook at that picture
06:29.10MGSsanchoi am
06:29.12Z-Knightthe pci keying is  "small....large....small"
06:29.15Z-Knightsee the bottom of it
06:29.21Z-Knighthttp://www.digium.com/index.php?menu=whatpcislot
06:29.24MGSsanchothats fine
06:29.25Z-Knightnow look at that picture
06:29.29Z-Knightthe slots 3 4 5
06:29.35MGSsanchooh
06:29.35Z-Knightdon't have that little front notch
06:29.40MGSsanchodont matter
06:29.42Z-Knightyou need slot 2
06:29.44MGSsancho345 will do
06:29.48Z-Knightahh
06:29.49Z-Knightok
06:29.49MGSsancho12 are PCIX
06:29.59Z-Knightyeah I misspoke
06:30.06Z-Knightso you think those are fine then?
06:30.08Z-Knightthe 3 4 5
06:30.12MGSsanchono 12 are pci 64bit
06:30.15MGSsanchoyeah
06:30.20Z-Knightyes, ok
06:30.23Z-Knightthank you
06:30.30Z-Knighthave you ever used the TDM2400P ?
06:30.34MGSsanchoi would personaly hold it down with electrical tape or something
06:30.39MGSsanchonope :(
06:30.42Z-Knightahh
06:30.42Z-Knightk
06:30.44MGSsancho<-- poor
06:30.48Z-Knight:)
06:30.55Z-Knightyeah that thing is expensive from what I see
06:31.15Z-KnightI was just wondering how really long it is...I suspect even smaller micro-atx cases should support it
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06:31.50MGSsanchonot sure about it
06:32.10Z-Knightwhat hardware have you used?  any?  or mostly SIP?
06:32.18MGSsanchobetter of getting it. then once you have it in ur hands, goto Frys or something and look to see what mobos and cases will house it
06:32.28Z-Knighttrue
06:32.31MGSsanchouse me router
06:32.37MGSsanchoWRT54GS
06:32.41MGSsancho<_<
06:32.44Z-Knightyou embedded it in there?
06:32.48MGSsanchono
06:32.50Z-Knight* in wrt54gs?
06:32.53MGSsanchoyes
06:33.00MGSsanchoflash the firmware
06:33.02Z-Knightyup
06:33.07Z-Knightwas that easy?
06:33.12MGSsanchoreletivly
06:33.21MGSsanchoall you have to do is update it :)
06:33.22Z-KnightI'm guessing there is no way to get voicemail
06:33.28MGSsanchothen enable the asterisk module
06:33.35MGSsanchothen use sip
06:33.38Z-Knightk
06:33.40MGSsanchoyou can
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06:33.47MGSsanchothe router can
06:33.49Z-Knightreally? how much space does the flash have?
06:33.56Z-KnightI mean the router?
06:33.57MGSsanchobut my router has like 8mb flash so its limited
06:34.08MGSsancho4mb ram 8mb ram
06:34.12MGSsanchosomething like that
06:34.35MGSsanchoi wish routers had usb ports >_<
06:34.36Z-Knightahh...you'd have to do  a hardware mod to add more then....beyond my capability right now
06:34.48MGSsanchoyeah
06:34.49Z-Knightsome do...one port
06:34.55Z-Knightwait
06:34.56Z-Knightnever mind
06:34.57MGSsanchoor the WRT54GL has 32mb flash
06:35.00Z-KnightI'm thinking cable modem
06:35.30Z-Knightso with the router you have support for at least 4 direct SIP lines
06:35.49Z-Knighti'm guessing with a hub, etc  you can then get more
06:36.16MGSsanchoyup
06:36.31MGSsanchobut limited to the bandwith your ISP will give you
06:36.36Z-Knightyeah
06:36.44MGSsanchoand the dinky 200mhz processor in my router
06:36.55Z-Knightthat is really cool....I want to setup a * server that can support an office of 12 people ...that is my goal
06:37.13MGSsanchothen yeah lol you will need a real pc
06:37.16Z-Knightyup
06:37.25Z-Knightthat's why the question of motherboards
06:37.30MGSsanchoand sdl or a good cable connection
06:37.31Z-Knightreally hard to settle on one
06:37.34MGSsancho*sdsl
06:37.35Z-Knightyes
06:37.52Z-Knightit will likely be a mix of SIP and regular lines
06:37.59MGSsanchoyeah
06:38.14MGSsanchoget a mobo that has like 4 or more pci slots
06:38.17MGSsancho<_<   >_>
06:38.30Z-Knightwhy so manY?
06:38.44MGSsanchoadd more lines
06:38.46MGSsancho:)
06:38.48Z-Knightahh
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06:38.59MGSsanchobut 12 isnt that bad
06:39.11MGSsanchoyou can just do the entire phone system over ehternet
06:39.13Mavvie124 lines per PCI slot...
06:39.18MGSsanchoIP phones
06:39.40MGSsanchoand VOIP service from a provider. then just have your PBX do its job
06:39.48MGSsancho124 <_< damn
06:40.11Z-Knightsay I have 4 regular incoming land lines then I just need one of the DIgium cards with the 4 inputs
06:40.34Z-KnightI'd only need the TDM2400 if I want to connect to regular phones (not IP phones)
06:40.47Z-Knightelse I'd need a router for the IP phones
06:40.49Z-Knightright?
06:40.49glm2kMavvie: wouldn't that be dependent on the codec? 124 using ulaw?
06:41.12MGSsanchothe output ones are for umm
06:41.15Mavvieglm2k: no, it's depending on the hardware. A QuadPRI card is 124 lines.
06:41.16MGSsanchomaking calls
06:41.19Mavvie:-)
06:41.23MGSsancholike you connect a regular phone to it
06:41.54glm2kMavvie: egad.
06:42.14glm2kall you really need is 1 slot in most cases
06:42.44MGSsanchohttps://shop.eikonex.net//catalog/product_info.php?products_id=250
06:42.48MGSsanchothat?
06:42.51Z-Knightmy goal is to upgrade an office using regular phones now and move them to VOIP....so actually probably have 2 hard lines coming in and then upgrade most phones to IP phones while keeping a couple that for regular connection
06:43.13MGSsanchoyou can keep the 4 lines
06:43.33MGSsanchocuz you can only use those 4 lines at most (internal is independant
06:43.35Z-Knightyeah, I could, but it would be better to migrate slowly away from land lines
06:43.51MGSsanchoyeah good thinking
06:43.51Z-KnightI'd replace the 4 lines with VOIP eventually
06:43.55MGSsancho:)
06:44.05Z-Knightis that about the right price for the TE411P?  $2400?
06:44.19Z-KnightI thought those would be cheaper, but that's about the price I'm seeing
06:44.25Z-Knightseems expensive as hell for only 4 lines
06:45.04Z-Knightor am I missing something about what this card actually does?
06:45.21X-Robit's 120 lines, not 4.
06:45.30X-Rob4 ports. 1 port == 30 lines (E1)
06:45.34Z-Knightahhh
06:45.42Z-KnightI am missing something then....I need to read more
06:45.57MGSsanchoso you need internal phones?
06:46.00MGSsancho12?
06:46.15X-RobNormal lines use a TDM card, not a TE
06:46.18Z-Knightwell, the office has 12 internal phones now (all regular phones, not IP phones)
06:46.19X-RobTE is ISDN
06:46.22Z-Knightyes
06:46.25Z-Knightk
06:46.32Z-Knightso the TDM2400P
06:46.37Z-Knightfor the current phones
06:47.09Z-KnightI guess I'm confused then how you use this TE411P....what does one of the 4 connections go to? a router?
06:47.10X-RobI stronly suggest you use SIP phones, not analog.
06:47.21X-RobMany more features.
06:47.24Z-Knightyes
06:47.26Z-KnightI agree
06:47.28X-RobZ-Knight, It's an ISDN connection, it comes from your telco.
06:47.46Z-Knightso the ISDN connects to the those ports?
06:47.50Z-Knightthose are all incoming ports?
06:47.57Z-Knightsupport for 4 ISDN lines?
06:48.02X-RobSupport for 4 PRI's
06:48.05X-RobPrimary Rate Interfaces
06:48.06Z-KnightAHH
06:48.08Z-Knightyes
06:48.09Z-Knightthanks
06:48.21X-Robor T1's for the yanks
06:48.25Z-Knight:)
06:48.35MGSsanchommm T1
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06:49.56Z-Knightso your PBX would have the PRI lines coming in to this TE411P, then the PBX connects to a router/switch through a regular ethernet card, and then all other IP phones connect to the router/switch as well.
06:50.01Z-Knightthat is the basics right?
06:50.19X-RobYeah
06:50.23Z-Knightsweet
06:50.32X-Robdoing the cablign is the easy bit though
06:50.37X-Robit's the configuration that's the difficult one.
06:50.42Z-KnightI can see that
06:50.46Z-Knightthere is so much in *
06:50.53Z-Knightfreaking steep learning curve
06:51.05*** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com)
06:51.09Z-KnightI want to write my own interface to it using...make a control panel using Java
06:51.36X-RobThere's a flash operator's panel
06:51.41Z-KnightI know
06:51.47X-Robthat might give you some hints.
06:52.00Z-Knightyes, but that is sort of like an Admin phone receptionist panel
06:52.12Z-KnightI want AMP I think
06:52.19Z-Knighttrying to remember the name...the PHP tool
06:52.26Z-KnightPHP interface to the PBX
06:52.32Z-Knightfor configuration of it
06:52.35X-RobAMP
06:52.37X-RobI'm a developer
06:52.50Z-KnightI guess I was mistaken by saying control panel...I mean config panel
06:52.52Z-Knightthat's cool
06:52.58Z-Knightwhy PHP by the way?
06:53.02X-Rob*shrug*
06:53.16Z-Knightany thought to doing JAVA?  Do you know anyone doing a JAva interface?
06:53.59Z-Knightbasically I'd like to do what Fonality has
06:54.11X-RobWhy java?
06:54.12Z-Knightand what AMP is
06:54.16X-Roball the config has to be done on the server
06:54.19Z-Knight*shrug*
06:54.22Z-Knight:)
06:54.24X-Robso you'll have a java gui that talks to .. what?
06:54.28Z-KnightI like java
06:54.30Z-Knightno
06:54.35Z-Knightjava webserver
06:54.38Z-Knightlike PHP server
06:54.41Z-Knightno different
06:54.45Z-Knightnot java gui
06:54.55Z-Knightjust using Java servlets, JSPs
06:55.00Z-Knightsorry should have clarified
06:55.08Z-Knightmysql interface, etc
06:57.16Z-KnightI'm still not decided, but I felt that maybe I can write a Java interface so I can learn it from scratch and really understand the ins/outs of controlling *
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06:57.42camonzhi
06:57.48Z-Knighthey
06:59.57Z-KnightX-Rob....how hard is the AMP code to grasp?  I know that is a nebulous question but I'm curious what your feels are on it?
07:00.18X-RobIt's not all that difficult really - espeically with the v2 stuff (currently in CVS) as it's all modular.
07:00.29Z-Knightthat's good
07:01.25Z-KnightI'll need to go into then, I've not learned PHP yet but can't be any more difficult than ASP,JSP,etc ....plus if the base is there it might be easier to use PHP
07:01.32Z-Knightis a PHP server hard to config?
07:03.14X-Robno, it's usually already enabled in most linux distributions
07:03.39Z-Knightthat's better than Java...geez, I'd have to setup a Tomcat server
07:04.19Z-Knightfor your stuff do you use Asterisk@Home or do you just use your own flavor of Linux and install * on it?
07:04.33Z-Knightor what would you suggest?
07:05.25X-RobIf you're starting out, grab A@H
07:05.42X-RobIt's a nice beginners piece, but it's quite complex under the hood
07:05.57X-Robso you can get the feel of it, and figure out how stuff works by looking at the generated configs.
07:06.17Z-KnightI am starting out, though I do have programming/computer experience and I'd like to get a system that I could sell
07:07.22Z-KnightI'm trying to figure out if for a production server (or whatever you'd call it) I'm not sure I should go with Asterisk@Home, though I really don't have any reason that I shouldn't
07:08.19X-Rob*shrug* give it a shot.
07:08.25Z-Knighthehe
07:08.35X-RobIt's a good distro (CentOS 4.2) which I'd recommend anyway, and it's got all the required bits you need
07:08.45Z-Knightyes
07:08.49X-Robso even if you don't end up using AMP to manage *, it's still a good starting point.
07:08.58Z-KnightI really don't know why I wouldn't go with *@home
07:09.19Z-Knightthanks for all your help/info
07:12.49wellnghi Z-Knight
07:12.52Z-Knighthey
07:12.59camonzif my * box is behind a NAT device i should set the externip value to my pub ip address right?
07:13.18Z-Knighttrying to think...
07:13.21Z-Knightone sec to confirm
07:13.36camonzi've been running some tests with some friends and we're experiencing 1 way audio only
07:13.55Z-Knightthat's what I did
07:14.01Z-Knightdo you have nat=yes?
07:14.14Z-Knightare they using xlite phones by chance?
07:14.17camonznormally the client can listen to me but i'm not getting their rtp packets
07:14.19camonzyep
07:14.28camonzbut nat=yes is when the client is behind a nat
07:14.42camonzi have both cases, the client and the * server are behind different nats
07:15.08Z-Knightdo you have reinvite=no  and canreinvite=no on the sip.conf for them
07:15.15camonznope
07:15.21Z-KnightI think you may need that
07:15.36Z-Knightand also when I had this issue I had to make sure all the portforwarding was set
07:15.45Z-Knightyou not only need the 5060 port
07:16.03Z-Knightbut * uses ports 10000 to 20000 (I think) as well for the SIP (I think again)
07:16.15camonz10k to 20k for rtp audio
07:16.18Z-Knightthe port forwarding was one thing that got me
07:16.42Z-Knightwait...maybe nat=no  ( I don't recall what is should be )
07:17.04camonzwhat i really can't understand as to why i'm not getting the audio packets from the client is that the * server is on my router dmz
07:17.22camonzso i should be getting all outside packets
07:17.37Z-Knightyeah, but you sure your friend is not blocking it
07:17.49Z-Knightcan you get him to remove his router out of the equation
07:18.15camonznope, but  i ran tests this afternoon with his gf wich is on a public ip
07:18.19camonzone way audio as well
07:18.19Z-Knightthe last time I did this config was about 3 weeks ago so I managed to forget my solutions
07:18.25camonzshe can hear me but i cannot hear her
07:18.27Z-Knightok
07:20.08Z-Knightfor my config that worked I had this:
07:20.11Z-Knighttype=friend
07:20.13Z-Knightnat=no
07:20.17Z-Knightusername=...
07:20.25Z-Knightsecret=..
07:20.29Z-Knighthost=dynamic
07:20.36Z-Knightcontext=testcontext
07:20.40Z-Knightreinvite=no
07:20.44Z-Knightcanreinvite=no
07:20.58camonzok
07:20.59Z-Knightalso in [general] I had
07:21.07Z-Knightexternip = my external ip (hehe)
07:21.19Z-Knightlocalnet = 192.168.0.1/255.255.255.0
07:21.26Z-Knightnot sure if those are needed
07:21.46Z-KnightI also accidentally had a nat=yes in the [general] but I'm not sure it even belongs there
07:22.12camonzi'm wondering right now if adding externip value in [general] is going to screw up the set up for my house network
07:22.17Z-Knightthen I totally opened up my router to allow my friend to connect and the PBX to be totally accessible
07:22.42camonzi have 4 interfaces for my lan, as well as 2 extensions outside of it
07:22.44Z-KnightI can't help you with that because I don't quite know
07:23.05camonzno prob :->, gonna change the value and reload to see what happens
07:23.21Z-Knightgood luck....
07:24.03Z-KnightI had my box succeeding 3 weeks back and then I had to take it apart so I need to do that same test soon again....I'm surprised a detailed description of this is not available
07:24.29Z-Knightby the way...do their XLITE phones register successfully?  (does it say logged in on the xlite?)
07:24.58camonzyep,
07:25.17camonzfor all extensions, the in house ones and the outside the nat ones
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07:25.23Z-KnightI'd try the reinvite,canreinvite and see what happens
07:25.27camonzthe prob is with the audio
07:25.32Z-Knightyes I know what you mean
07:25.42Z-KnightI had the exact same thing and then I started tinkering
07:26.04Z-Knightunfortunately between the time it didn't work and the time it did work I changed like 3 things so I can't pinpoint which was the solution
07:27.27Z-KnightI'm really thinking that having this (canreinvite=no)  for each user will make it work
07:27.37Z-Knightoh and nat=no
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07:30.25puzzledmorning
07:30.44Z-Knightgood morning
07:31.01camonzmorning
07:31.09Z-Knightany look, camonz?
07:31.14camonzyep
07:31.18Z-Knightany luck*
07:31.24camonzfor my lan it still works
07:31.26Z-Knightgeez...I can't type
07:31.28camonzaparently
07:31.35camonzwanna run a test
07:31.42Z-Knightsure
07:31.55Z-Knightlet me get xlite
07:31.58Z-Knightand mic
07:31.59X-Robwtf is with this '*' shit? My son does it when he's msn'ing me.
07:32.09camonzok
07:38.49QwellX-Rob: ?
07:40.58X-Rob<Z-Knight> any look, camonz?
07:41.01X-Rob<Z-Knight> any luck*
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07:41.28X-Robif you use regexps, jbot even dixes it for you.
07:41.31X-Robs/dixes/fixes/
07:42.14QwellX-Rob: kids don't know regex :p
07:42.21X-Robthey damn well shoud!
07:42.30Qwellteach your kid :p
07:42.33X-RobShould be taught in primary school along with clicking and double clicking!
07:42.36Qwellmaybe he'll teach others
07:42.43Qwelland it'll spread through the internet
07:42.49Qwelland eventually...we'll never see it again. :D
07:42.54X-Robteh interwebs!
07:43.00Qwellintarweb
07:43.05X-Robthazzit.
07:44.01Qwelldon't msg me
07:44.08X-RobI didn't.
07:44.10Qwellnot you
07:44.13X-RobI know
07:44.15X-Robcoz I didn't!
07:44.15Qwell:p
07:44.29X-Robwho do we get to mock?
07:44.31Qwellwellng:
07:44.58MGSsanchomy mom!!
07:45.02X-Robwellng takes it up the arse, doo dah, doo dah.
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07:52.31wasim<PROTECTED>
07:56.12MavvieZap/99-1             s@SJH-AAPT1:1        Rsrvd   (None)
07:56.18Mavvieheh... wonder who to get rid of these beautifies
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07:56.22Mavviebeauties
07:57.05X-RobI used to work for AAPT
07:57.08X-Robmany moons ago
07:57.14Qwellaapt?
07:57.25X-RobQwell, small telco in .au
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07:57.44X-RobMavvie, I got better!
07:57.50X-Rob(They turned me into a newt!)
07:58.00Mavvie:-P
08:01.20`SauronMavs
08:01.30Mavvienow I have a PRI full of Rsrvd channels, and trunkavail() automatically jumped to the second PRI!
08:01.31Mavvieyay
08:01.34Mavviehi `Sauron
08:01.38`Sauronyou're all bark no bite, buddy :)
08:01.53Mavvie`Sauron: all hissing, no clawing.
08:03.13X-RobBad kitty!
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08:03.26X-Rob*squirt squirt*
08:03.26Mavvieoh oh
08:03.40Mavviedouble oh oh
08:03.47Mavvieit's raining outside again.
08:03.52Mavviethat's two loads of washing getting wet.
08:04.01X-RobBugger.
08:04.13Mavvieis only hanging there now for three days.
08:04.17X-RobGladstone's good for that. We don't get rain here.
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08:05.26robin_zWashing? Outside? ... coo. How very 20th century ;)
08:05.51Mavvierobin_z: much cheaper than using a dryer.
08:06.15robin_zerrr ... well, apart from the fact that its dripping wet, yes.
08:06.34Mavvieit's always dry in the afternoon.
08:09.55robin_zhmmm . this "group" thing is weird ... it seems to hava count 1 higher than I expect
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08:11.31robin_zshrug .. but at least it works, unlike call_limit
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08:27.07jeebusroxorsno dice
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08:44.49wellesmy * send TXREJ and LAGRQ very fast, and my iaxclient all response VNAKs .then the call hangup .what's wrong?
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09:03.30[hC]hmm. i have mixmonitor recording calls, but it seems like quite a lot of the time, even though the cli seems to report proper mixmonitor start/stop, the recordings come out truncated, like half way thru a call
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09:09.29pycsuszHi Everybody!
09:09.35GBLRAhi
09:09.42X-RobHi doctor nick!
09:10.00pycsuszSomedody cab help me, how can I provide sip with asterisk?
09:10.08pycsuszHow can I configure?
09:10.12X-Roboh fuck me.
09:10.24X-Robpycsusz, you've got a pile to learn. start with http://www.voip-info.org
09:10.53pycsuszX-rob thanx, but there where?
09:11.03[hC]and maybe the oreilly asterisk book, but voip-info is a good right-away source.
09:11.08X-Robsearch for sip.conf
09:11.36X-Robhttp://www.voip-info.org/wiki-Asterisk+config+sip.conf
09:11.58pycsuszX-Rob I use Asterisk for one year, but still I didn't set sip trunk
09:12.35pycsuszX-Rob I would lik to create a SIP trunk
09:12.40X-Robwell look at /usr/src/asterisk/configs/sip.conf.sample
09:13.08pycsuszX-Rob Thanx, I will try it!
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09:13.51saftsackhi
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09:16.14webmindgood morning (or your favorite part of the day), I've got a Fritz! isdn card, and trying to get asterisk to work with chan_capi, but it keeps bugging about "cc_init_capi: CAPI not installed, CAPI disabled!" and the fritz driver install page on voip-info seems to be for 2.4 (I'm running 2.6.15)
09:16.23webmindanyone like to help me out ?
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09:19.21Kernel_corehi all
09:19.40Kernel_coreanybody compiled PWLIB 1.9.0 on Debian ?!
09:19.57pycsuszKernel_core yes
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09:20.24Kernel_corepycsusz: which version of GCC do you use ?
09:20.57pycsuszKernel_core 3.4
09:21.11pycsuszKernel_core it don't compile with 4
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09:46.36Winkiehey guys, anyone got any experience with grandstream gxp-2000s? I need to know if it's possible to muck about with their LEDs
09:47.34[av]baniyep
09:47.48[av]baniblf is simple
09:47.49Winkieah excellent, i need to have some sort of agent logged on indication
09:47.58Winkieblf?
09:48.04X-RobAaah. You need devstate
09:48.05[av]banibusy lamp field
09:48.09*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
09:48.10X-Robwhich is part of bristuff
09:48.25[av]baniif you want agent tracking, thats something else... devstate may let you do that
09:48.51Winkienot tracking particularly, just want a simple agent logon procedure that will light up an LED on the phone
09:49.03WinkieX-Rob: is there any documentation on it?
09:49.48[av]banihttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
09:50.09[av]bani~devstate
09:50.10jbotfrom memory, devstate is http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
09:50.59Winkiei should learn to use ~ more often
09:52.27Winkieheh, bristuff is useless for this because we have a PRI it would seem
09:52.31Winkiestill BLF might be enough
09:53.03[av]baniblf wont really do agent logon stuff like that
09:54.38Winkiehmm
09:54.43Winkieit's quite worrying that
09:55.35[av]baniblf without devstate is only good for tracking extension status
09:55.51Winkieyeah, from the look of bristuff i can install it but use just devstate without affecting the PRI?
09:58.40pycsuszX-Rob Can you help me sill a bit?
10:00.10pycsuszX-Rob I have got 2 subnetwork, and in both I have got telephones, but just that can log in whitch is on same subnetwork as asterisk. Why?
10:00.32X-Robpycsusz, firewall or bad routing. DCHP wrong? Plenty of reasons.
10:02.03pycsuszX-Rob thanx
10:03.14*** join/#asterisk RoyK (n=roy@80.239.107.70)
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10:21.56Dabbaany know how to replace this line with a line that will handle a varying number of rows so i dont have to repeat the fetch fifty times
10:22.01Dabbaexten => s,3,MYSQL(Fetch fetchid ${resultid} myteam)
10:22.30Dabbaexten => s,4,MYSQL(Fetch fetchid ${resultid} myteam1)
10:22.31Dabbaetc
10:22.33Dabbaetc
10:24.48Dabbai.e interpolate the variable name with a 1,2,3,4 etc
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10:26.13sherif_i have a question when a phone rings and no one is there to answer i want to ba able to grap the call to my set.
10:26.59glm2ksherif_: you're probably looking for pickupgroup and callgroup
10:27.01Dabbaenable call pickup
10:27.06Dabba:-)
10:27.07Mavviehmmm... which oart of zt_pvt lets me know if I'm in a call or not.
10:27.24glm2kDabba: jinx! :)
10:28.40glm2kDabba: i think you can use a loop for that
10:29.08glm2kbut i'm not so sure about calling mysql in such a way
10:29.44Dabbai use it regular it works well, but is a pain with a row count that varies
10:30.00Dabbaor at least its a pain at the moment
10:30.07glm2khmm, would you be able to capture the row count ahead of time?
10:31.42Dabbaoh i think i see where your going
10:31.45glm2kbefore you reach that part of the dialplan, do you already know how many rows to fetch?
10:32.32FLeiXiuSAsterisk is not allowing me to dial without receiving a busy tone...anyone have any ideas?  Line's are up and configured.  I'm running SCCP.
10:32.36Dabbano i dont i wonder if a GoSUb in a Gosub will kill it all
10:33.07FLeiXiuSI receive a dial tone but I'm unable to push more than 1 button without receiving the busy tone.
10:33.14Dabba> FLeiXius a console output ??
10:33.37FLeiXiuSDabba: I don't see anything...
10:33.52Dabbaverboseity on ?
10:34.45FLeiXiuSYep, nothing relevant.
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10:36.20[av]banifulgas: what model cisco?
10:36.28FLeiXiuSdabba, let me check out the debug, I was wondering about it.
10:38.54FLeiXiuSI still don't see any output which would restrict my dialing to only 1 number before it receives a busy tone.
10:39.23Winkieanyone got a clue whether pgsql support for databases of sip peers etc is in svn or planned at any poing?
10:39.24Winkiepoint*
10:40.40FLeiXiuSDabba:  Care to see the output when a button is pushed?
10:40.54Dabbasure pastebin it
10:41.22FLeiXiuSDabba: http://pastebin.com/564164
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10:42.20X-Genhey freaks
10:46.37DabbaFLeiXiuS as you said looks ok, what are you doing with 9 in the dialplan, i havent ever used sccp so im blind really, my 7960 is running sip firmware
10:47.16FLeiXiuSDabba: 9 does absolutely nothing...I'm just giving an example.  After pushing one number on the keypad it instantly sends the busy tone.
10:47.41RoyKI wonder why they change function and variable names within the 1.2 "stable" track
10:47.49Mavvieeeks... app_trunkisavail.so is loaded before chan_zap.so is loaded, and thus the loader complains about it.
10:48.26*** join/#asterisk zoa (n=kkk@pirus.securax.be)
10:48.49RoyKzoa: hi
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10:48.59zoahey ho
10:49.06_foxfire_hi
10:49.11RoyKzoa: tried the jb patch on 1.2.svn? they have changed function- and variablenames again :(
10:49.26zoanopez, didnt try it yet
10:49.30zoaoh no, not again
10:49.32zoa:(
10:49.39RoyKfscking idiots
10:49.49RoyKit's supposed to be frozen, isn't it?
10:49.58zoadont think so, no
10:50.04RoyK1.2 is
10:50.12zoaah yes
10:50.16RoyKthey won't allow a single line of new code into it
10:50.27RoyKbut changing names is ok, it seems
10:51.49Mavviealso wonder who killed my agi-bin directory!
10:52.33puzzledRoyK: where can I find that patch again?
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10:53.13RoyKmantis
10:53.16RoyKi'll find the bug
10:53.25RoyKand i may as well upload a new 1.2.trunktoday patch
10:53.43RoyK#3854
10:54.54puzzledRoyK: thanks. in case you are interested, I made 1.2.4 compatible patches for #2863 and #5374
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10:56.03_foxfire_i am upgrading an asterisk 1.0.9 to an asterisk 1.2.4 , all seemed well but i have one strange behavior, when i dial a user using "dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r)" and some channels do not exist like the IAX2 channel even if the user ANSWERS the cdr will always show "FAILED" , this didn't happen in 1.0 any idea why ?
10:56.22puzzledRoyK: which one in #3854 should I try on 1.2 svn? ast_jb-1.2.0.patch3?
10:57.28RoyKpuzzled: the one i just uploaded :P
10:57.40RoyK2006-02-20-svn-1.2-rev10558.patch
10:58.17puzzledgot it. thanks
10:58.33RoyKpuzzled: be a good friend and test it and test it and test it :)
10:58.57puzzledRoyK: will include it in my rpm and see how it works
10:59.21puzzledRoyK: any particular conf setting I should avoid/should use?
10:59.38RoyKjust see the ones in sip.conf-sample
10:59.44puzzledok thanks
11:00.02RoyKi don't know if adaptive jitterbuffer is as well tested as the fixed one, but the fixed one works very well
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11:00.27Kernel_corehi all
11:00.32RoyKwe tested on a 704/128kbps adsl link with some 20 concurrent downloads with g.711a and we didn't have a single hichup
11:00.36Kernel_coreanybody compiled PWLIB 1.9.0 on Debian ?!
11:00.40RoyKlatency from the depth of hell, though
11:00.56zoaroyk, depends how much ms buffer you take :)
11:00.56puzzledRoyK: hehe. will try the static forst then
11:01.04zoanormally you have enough with 40ms or so
11:01.15RoyKthe PLC really works well
11:01.28Kernel_coreRoyK: I hate it too ... because after 2days playing hard with it , I couldn't comiple it on DEBIAN !
11:01.38puzzledzoa: is that the jb-max-size option?
11:01.42RoyKwe had the ATAs tuned up with larger JB and PLC turned on as well
11:01.43zoayes
11:01.47puzzledok
11:01.53zoaim off
11:01.54zoafood
11:01.58puzzledenjoy
11:02.27welleshi all,i want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while iaxc
11:02.27welleslient can not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated!
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11:03.12Mavvieis there a way to force the export of a function in a channel-module?
11:05.05welleshi zoa ,what's your opinion about my issue?
11:05.23ChrisUKAnyone know when the new firmware for the Grandstream GXP2000 the v1.2 is getting released officially.?
11:07.55Kernel_coreanybody compiled PWLIB 1.9.0 on Debian ?! :((
11:08.04GBLRAhave they actually fixed the bugs yet?
11:08.29Kernel_coreor is there any Package for debian ?
11:10.47_foxfire_i am upgrading an asterisk 1.0.9 to an asterisk 1.2.4 , all seemed well but i have one strange behavior, when i dial a user using "dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r)" and some channels do not exist like the IAX2 channel even if the user ANSWERS the cdr will always show "FAILED" , this didn't happen in 1.0 any idea why ?
11:22.12kmilitzerHello everyone. Is it possible that there is a Bug in handling the hangupcause in ast_softhangup_nolock ?
11:23.12Mavvieof course there is.
11:23.20*** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt)
11:23.23Mavvie(the possibility that)
11:24.31kmilitzerMavvie: OK, i guess I put it the wrong way. I see strange behavior in passing the hangupcause back to the dial application i am doing some debugging right now to verify that
11:25.09kmilitzerI only have to figure out, if the problem lies within asterisk itself, or if it is a bug of chan_ss7 that i use
11:25.31RoyKcypromis: ping
11:26.38*** join/#asterisk mac7 (n=karsten@c184067.adsl.hansenet.de)
11:28.13remissany suggestions for fun things to do with the dialplan?
11:28.55remissexcept rm'ing it...
11:29.14iDunnothat's not a fun thing to do in the dialplan.
11:29.44remissuh?
11:30.54sherif_hum the defualt value for picking up the call is *8 ;-)
11:31.25mac7I'm currently unable to register with iax.fwdnet.net! any other have this problem?
11:32.26mac7connection to iax.fwdOUT.net is OK
11:34.30*** join/#asterisk Mw3 (n=mw3@daisy.chains.ch)
11:35.05sherif_what dose call parking means ?
11:35.26iDunnomeans that you can park the call
11:35.31iDunnothink 'hold' but different.
11:35.47iDunnobasically, when you park a call you should get given an extension number that it was parked on...
11:36.00iDunnoif you then hang up and dial that extension, you'll pick the call back up
11:36.09iDunno(does that make sense?)
11:36.48remissany way to transfer it instead of parking it?
11:37.29remissblindxfer?
11:38.03iDunnopush the transfer button and transfer to an extension that isn't call parking? ;)
11:38.13remissyes, yes, but...
11:44.09znoGiDunno: but.. in which cases is it useful to do call parking?
11:44.21remiss#1 is transfer
11:44.26remissno.. just sharp
11:44.29znoGapart from parking it then going to another extension and dialing it back up, i can't see much use for it
11:45.33iDunnoznoG: when you're not sure who needs to take the call next? ;)
11:45.45remissno, no, no
11:45.49remissfirst you park the call
11:46.19remissthen you send mail to everyone with the extension and tell someone to pick it up
11:46.19sherif_remiss: yes i can transfare calles
11:46.19iDunnoso you phone round and find out who wants it then you tell them the extension that $person is on ;)
11:46.20iDunnoremiss: *grin* - that's evil ;)
11:46.20remisshehe
11:46.40sherif_remiss: but the problem is i'm new in this PBX things but i know how to transfare it :) u click hold then ext number them transfer or read ur phone manuals ;-)
11:46.49sherif_now i'll play with music on hold :D
11:47.37FLeiXiuSi wish I could play moh without a soundcard.
11:47.42remissasterisk is kind of cool :)
11:47.44FLeiXiuSwith mpg321
11:47.54remissFLeiXiuS: use something else?
11:48.05FLeiXiuSIdeas?
11:48.08remissmplayer?
11:48.24FLeiXiuSwill mplayer play .pls'?
11:48.49remiss.pls..
11:48.57remisswhat is that?
11:49.04FLeiXiuSshoutcast stream..
11:49.10remissoh
11:49.13puzzledI use madplay
11:49.17FLeiXiuSI'll check it out
11:49.17znoGiDunno: in which sort of asterisk setup would that be ... useful?
11:49.22*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
11:49.23remissFLeiXiuS: i don't think so..
11:49.48iDunnoznoG: it's only 3 lines or so of dialplan and enabling it in features, it's probably useful to have anyway.
11:50.06znoGiDunno: yeah i know it's easy to setup, i'm just wondering what sort of business could make use of it
11:50.10*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
11:50.31iDunnolarge places that aren't sure where the call needs to go to, long term hold, etc etc.
11:50.44znoGi see the use when going from one office to another, for example
11:51.28znoGwould be neat if the call park would ring certain extensions when they're free and connect them to the parked ext
11:51.31znoGbut i guess thats what queues are for
11:51.33iDunnoor if the person that you're going to transfer to is engaged, might be easier to park the call first ;)
11:55.19trixterMicrosoft To Offer Free Wireless VoIP http://it.slashdot.org/it/06/02/20/0227200.shtml
11:55.34kmilitzerI found a bug in ast_softhangup_nolock
11:56.41RoyKtrixter: they prolly just want your soul, but that'll be considered 'free' anyway, won't it?
11:57.27trixterthey prolly are just integrating something into office, and its really going to be a wifi->office thing and not pstn
11:58.10trixterbut I havent read the article so I am just guessing
11:58.14kmilitzerThe Hangupcause is not set there, so it is not correctly transmitted to the other channel
11:58.25kmilitzerI inserted a chan->hangupcause = cause; and now it works ...
11:58.49kmilitzerWhat do I have to do to file this bug? Never did something like that before ...
12:04.12*** join/#asterisk oej (n=oej@apollo.webway.se)
12:04.32kmilitzerHelloooo anyone there that could answer my questions?
12:05.25remisshttp://bugs.digium.com/main_page.php <-- probably
12:07.39puzzledkmilitzer: yes post the diff to kmilitzer or send an email to the asterisk-deve mailinglist or mention it in #asterisk-dev
12:07.57puzzledkmilitzer: post the diff to bugs.digium.com off course :)
12:10.53mutilatorcan you turn on/off threeway calling on a zap chan
12:10.56mutilatorper channel
12:11.03mutilatoror is that a global setting
12:11.45puzzledIf not possible in a config file I guess you could limit it via Set(GROUP=...
12:12.01puzzledand check for the number of active calls and limit to one
12:12.30mutilatorya
12:13.46remissanyone got a nice link to configuring musiconhold?
12:14.09_foxfire_i am migrating from 1.0 to 1.2 , everyting worked ok with the exception of one anoying detail. if i dial Dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r) an for example IAX2/foxfire channel does not exist and SIP/foxfire accepts the call the cdr record will return "FAILED" instead of the expected "ANSWERED" , is this a bug or has some option changed ?
12:15.01_foxfire_this does not happen with version 1.0
12:15.57*** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net)
12:20.07sherif_what is the bitrate for the hold on music ?
12:20.22*** join/#asterisk fulgas (n=fulgas@82.102.2.254)
12:22.25*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
12:24.51GBLRA8bit
12:25.02kmilitzerpuzzled: Thanks, went to asterisk-dev now
12:25.22RoyKGBLRA: 8bit is not a bitrate :)
12:25.27puzzledkmilitzer: yup, seen it
12:25.32RoyKsherif_: 8khz
12:26.26*** join/#asterisk Mw3 (n=mw3@national.t-error.hu)
12:26.32*** join/#asterisk Bambr (n=Bambr@213-35-238-17-dsl.end.estpak.ee)
12:26.38GBLRAsorry
12:26.47GBLRAnot thinking
12:29.46*** join/#asterisk l-fy (n=diana@yate/developer/l-fy)
12:30.07*** join/#asterisk fjean (n=fjean@201.29.122.10)
12:31.10l-fydoes anyone knows a library that analizes voice delays and jitters and things like that?
12:31.44fjeanHello everybody, anybody would have a suggestion on where to start looking for SIP phones that don't hear ringing tone on an IAX2 channel that is in progress ?
12:31.45zoathere are some
12:31.49zoabut they work on the network level
12:31.55l-fyhey zoa
12:32.01zoameaning they look at the timestamps only
12:32.08l-fyoooo
12:32.22l-fyi need something that does convolution and corelation of the voice stream
12:32.27l-fysomething not very fancy
12:33.50fjeanit happens for SIP only, not for IAX2 devices...
12:34.01l-fyi want to send a voice to some voip equipment do some echo there and than get the voice back and compare
12:36.12fjeandoes anyone knows if IAX2 send ringing tone inband or outband ?
12:37.20zoaoutband
12:37.26fjeanok
12:37.30zoal-fy not that i know of
12:37.36l-fyok zoa thank you
12:37.44l-fyzoa where are you living this days?
12:37.52zoabulgaria still
12:38.09l-fywow
12:38.18l-fyyou haven't been in .nl?
12:39.35zoaim from .be
12:39.40zoaim going there tomorrow
12:40.27*** join/#asterisk coppice (n=chatzill@19.206.17.210.dyn.pacific.net.hk)
12:40.30fjeanzoa, do you think that my ATAs are listening inband for ringing tone and that would be why they hear nothing ? what are the chances ?  :- )
12:40.48zoahmm, that doesnt matter
12:40.51l-fysorry :(
12:40.54zoaas your ata will be sip
12:40.55l-fyi'm soooo stupid
12:41.00l-fyi've forgot
12:45.38*** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com)
12:47.15*** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com)
12:48.04*** join/#asterisk linville (n=linville@azure.tuxdriver.com)
12:56.02fjeansip.conf:  does progressinband=yes really work ??
12:56.23GBLRAi found setting it to no stopped the external and internal ringing on my Aastras
12:57.03fjeanah, ok
12:57.30fjeanyou know any other settings that affect ringing for sip devices ?  I can't manage to get one
12:57.58GBLRAwhat a ring?
12:58.56fjeanmm?
13:00.28fjeanyes, a ring
13:00.38fjeansorry..
13:00.41*** part/#asterisk l-fy (n=diana@yate/developer/l-fy)
13:01.53welleszoa ,there?
13:01.57zoayes
13:02.03wellessee my question?
13:02.09zoawhere ?
13:02.28wellesok i post  again
13:02.38wellesi want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while iaxclient c
13:02.38wellesan not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated!
13:03.04zoaah dunno
13:03.09zoathat seens to work fine for me
13:03.31*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:03.37wellesit works fine when using public ip
13:03.51wellesbut when use private ip it can not work
13:04.35wellesand myiaxclient receive too many TXREJ and send too many VNAK
13:04.37fjeanyou have one server on the same public IP as your softfone ?
13:05.00wellesno
13:05.50wellessoftphone use private ip and two aterisk use different public ip
13:06.48wellesif softphone use public ip. it works fine
13:08.17mexuar-timwelles, what is the 'cause' in the TXREJ ?
13:09.01wellesif asterisk can not transfer successfully .why not use relay mode. the asterisk seem to try too many times to transfer
13:09.17wellesmexuar-tim,  let me have a check
13:11.29*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:14.43*** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br)
13:15.16_Paulo_Hi all...
13:15.35wellesmexuar-tim, i dunno why it transfer fail.maybe my softphone behind a nat which aterisk can not access
13:15.36*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
13:16.25wellesthe debug info only show 'unable to transfre ...''
13:17.39mexuar-timWho ever sends the TXREJ must include a 'cause' field in the TXREJ packet.
13:17.57wellesok
13:18.00mexuar-timI was hoping that might give a clue.
13:18.04_Paulo_I got a TDM400P. I want to connect an external modem for faxing.
13:18.29_Paulo_I heard that TDM400P is not reliable for fax.
13:18.33_Paulo_Is that true?
13:20.04_Paulo_I want to use the external modem because none of the "pure software" solutions (app_txfax and iaxmodem) are working.
13:20.42*** part/#asterisk exonic (n=exonic@209.172.11.54)
13:21.27*** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net)
13:22.39wellesthe cause code is 27
13:23.17holmehHow would you guys go by shaping the two last bytes of a string?
13:23.47*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.229)
13:23.53Kernel_corehi all
13:23.58Kernel_coreanybody compiled PWLIB 1.9.0 on Debian ?! :((
13:24.24holmeherr
13:24.35mexuar-timwelles, 27 means "Destination out of order   ". Sorry that isn't much of a clue.
13:24.36holmehI am in the wrong channel trolling =)
13:26.38coppice_Paulo_ if iaxmodem and spandsp don't work the external modem won't work either
13:32.22_Paulo_coppice, Im receiving well with spandsp. Sending is the problem.
13:33.10coppicea few people complain about that, but nobody follows up well enough to find why it doesn't work for everyone
13:34.40_Paulo_Im not sure the problem lies in spandsp, so I decided to try an oldfashioned modem.
13:34.54wellesmexuar-tim, sorry. i check the packet carefully. the it show the data field of TXREJ  is all zero.
13:35.02coppicethat's just asking for more trouble
13:35.07*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com)
13:35.51fugitivocoppice: do you have any record of r2 working with asterisk connected to another pbx?
13:36.27fugitivoi mean asterisk --- R2 --- some pbx
13:37.03coppicefugitivo: lots of people do that. However, because there are lots of variants, and many implementations are buggy there is no guarantee any particular setup will work
13:37.48wellesmexuar-tim, TXREJ 's subclass is 0x1c but ethreal shows that it is ox1b. what's wrong?
13:37.53fjeanwe do have it working here in brazil...
13:38.04fugitivofjean: what pbx are you using?
13:38.13fugitivoi need to do that with a siemens hipath 3550
13:38.44fjeansorry, i would have to ask the guys at the other office, but i know its there and working  :- )
13:39.00coppicefugitivo: it will depend on configuration and software versions, as well as the model of PBX
13:39.40fugitivocoppice: software versions?
13:39.44*** join/#asterisk pengyong (n=lala@218.19.188.13)
13:40.29coppiceof the software in the PBX
13:40.41_Paulo_coppice, Why the analog modem is worst than the soft ones?
13:40.55fugitivocoppice: ugh, i can't control that
13:41.25coppice_Paulo_ because the existing analogue port options, like the TDM400, are mostly flaky
13:42.11wellesany ideas about my issueïĵŸ
13:42.16_Paulo_coppice, I heard something in that line... :-(
13:43.08_Paulo_coppice, I has hoping they have somewath improved...
13:43.09fugitivocoppice: does the pbx need any special config or it's just luck with the software version?
13:43.28*** join/#asterisk alerios (n=alerios@201.244.246.58)
13:43.29_Paulo_fugitivo, you will have to try
13:43.45mexuar-timi would have to see the ethereal packet dump to understand that.
13:43.49_Paulo_fugitivo, I think its a model by model issue.
13:45.23_Paulo_fugitivo, you will offer solutions for people who already have a PBX?
13:45.28fugitivoyes
13:45.53_Paulo_fugitivo, If so, I think you canot claim to support every PBX out there.
13:46.15fugitivoi don't
13:46.35fugitivoi told the customer that it could work or not
13:46.47_Paulo_fugitivo, After some time, you will have a few PBX validated.
13:48.13fugitivodo you have any?
13:52.15_Paulo_I have none, you should ask the maillist
13:54.13_Paulo_coppice, How can i help diagnosing the issue with app_txfax?
13:54.52*** join/#asterisk nozey (n=nozey@20150042008.user.veloxzone.com.br)
13:55.29nozeyhi ... im having problems with sip and nat. can someone help?
13:58.47docelm0nozey, try stun
13:59.03fugitivonozey: describe your scenario
13:59.22_foxfire_i am migrating from 1.0 to 1.2 , everyting worked ok with the exception of one anoying detail. if i dial Dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r) an for example IAX2/foxfire channel does not exist and SIP/foxfire accepts the call the cdr record will return "FAILED" instead of the expected "ANSWERED" , is this a bug or has some option changed ?
13:59.26*** join/#asterisk zael (n=zael@20150034147.user.veloxzone.com.br)
13:59.59nozeyi instaled asterisk here on my machine(thats behind a router) ... its running ... i can dial to my friends. If my friend is not behind a router(like me) he can hear me .... but i cant hear him
14:00.13nozeyif hes behind a router too ... none of us can hear each other
14:00.24nozeysorry for my english ... im brazilian #)
14:00.40fugitivonozey: try stun like docelm0 said
14:00.47docelm0Its natting..   You need to put your asterisk box on the outside of your router.  If its on the inside it makes life VERY hard
14:00.59RoyKhm. may sip/udp packet ever fragment?
14:01.12nozeywell hes inside
14:01.12RoyKor will all consist of a single udp packet?
14:01.16*** join/#asterisk Tuttle_ (n=Tuttle@kelinat210.keli.cz)
14:01.31nozeyand unfortunally i need to make it work inside a nat
14:01.36Tuttle_What is the favorite SW phone for Linux?
14:01.41nozeybut let me try stun ... thanks for the help
14:01.43docelm0RoyK, if there is an inconsistancy of a udp packet it will never frag..  it will just be dropped
14:01.43zoaidefisk! :p
14:01.44_foxfire_nozey : are you using IAX2 or SIP ?
14:01.48nozeysip
14:01.56RoyKgood...
14:01.56zoa(its mine so im not objective)
14:02.15nozeyi need to use sip :( ... but i learned that iax is much simplier to tranverse nats
14:02.18docelm0RoyK, one of the drawbacks to UDP..   They are never rebuilt and sent again like TCP
14:02.21_foxfire_sip is the best but when nat comes in the middle .... ikes
14:02.34nozeyi can see that
14:02.37zaelhey guys, i'm with nozey in this problem... we need to set an asterisk server using sip through a nat
14:02.52_foxfire_did u forward any ports in your router ?
14:02.57nozey5060
14:03.03nozeythe rtp ports too
14:03.05docelm0zael does your router have the option for DMZ and also is your IP static?
14:03.32_foxfire_did u use the ports explained in the asterisk site ?
14:03.40*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)
14:03.47zaeli dunno, mine is a d-link 502g... i cannot access the router page through firefox
14:03.58zaelis there any way to config this using telnet?
14:04.05docelm0ya.. they suck..   You have to use IE
14:04.14nozeyno way
14:04.17zaelbut how? :)
14:04.23zaeli'm using slack
14:04.26zaellol
14:04.27nozeyi can use dmz at my router
14:04.42nozeybut i dont thinks thtas the problem, cause i already forwarded all the ports that sip uses
14:04.49TagorI've the following problem with asterisk; I am using a sip provider for my calls. When I make an external call everything works fine. But when I try to call the asterisk server using a normal phone, then it just says; 'trying to connect'. When I try 'sip debug' on the CLI, I see nothing. Does anyone know how to find out what's going wrong?
14:04.52fugitivolast firmware of dlink routers works with ie only :)
14:05.00nozeyzael, is the one that im dialing to
14:05.16docelm0But you can set a DMZ..   What I did at my house was to static assign my internal IP to something like 200..   Then DMZ 200 and make asterisk forward on the public IP as its own..
14:05.31docelm0but it has to be static..  dynamic will work for the most part till it changes
14:05.42zaelmy internal IP is static
14:05.50docelm0nozey you must not understand how SIP works
14:06.01zaelbut i can't change these DMZ settings in the router
14:06.04docelm0zael, what bout external?
14:06.10zaelexternal is dynamic
14:06.14docelm0Good luck
14:06.16nozeymine too
14:06.16*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:06.21nozeydocelm0, what do u mean?
14:06.26_foxfire_nozey try http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules for more help
14:06.38TagorNobody? :(
14:06.49docelm0Tagor, wait your turn or leave..
14:07.13nozeyi made this already _foxfire_
14:07.17nozeyim logging that port too
14:07.49_foxfire_tagor : probable you don't have an extension assosiated with th eincoming number
14:07.59nozeyi can see the connections ... but no audio ... i think audio uses the rtp ports described in rtp.conf, or am i worng?
14:08.05docelm0nozey, zael, When a sip packet comes in it works cause of the invite being setup on the originating location.  When you send out from the nat it broadcasts the private IP block of information which is unroutable.  This is bad..   You need to tell it to use the public IP over the private
14:08.25*** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com)
14:08.47zaelyeah, we know this
14:08.48Tagor_foxfire_ >> The extension in extensions.conf is the same as the extension name in sip.conf and the context in sip.conf
14:08.53zaelbut we dont know HOW to do this
14:08.54nozeythats the problem
14:09.17*** join/#asterisk oej (n=oej@apollo.webway.se)
14:09.18_foxfire_nnozey youi can limit the range of the rtp ports on asterisk and on all major software and hardware clients
14:09.19SplasPoodit'd be nice if all the sipbroker sip providers would list their DIDs in the ENUM db
14:09.41docelm0So when your RTP tries to sync up it tries to use the private..   Either A..  Use stun which asterisk to my knowledge doesnt fully support..   OR manually push your public IP from the asterisk box
14:10.12_foxfire_tagor maybe i didn't understand your problem right can you be a bit more detailed
14:10.16nozeylet me try stun
14:10.20docelm0_foxfire_, thats all fine and dandy..  but how are they gonna change the information in the SIP Header Packet?
14:10.28SplasPoodOr if I could otherwise get a list of all their assigned DIDs somewhere
14:10.35nozeythanks for the help guys ... we are going to try it ...
14:10.39*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
14:11.29_foxfire_docelm : iup thats a prob ... , another one is if he has an dynamic ip
14:11.47*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
14:12.01docelm0Good lord what have I been saying
14:12.08docelm0They both have dynamic..
14:12.09docelm0geesh
14:12.19Tagor_foxfire_ >> I use a sip provider to handle outgoing and incomming calls. When I call internally everything works fine. Also I can make calls to external phone numbers. But when an external phone tries to call the asterisk server then nothing happens. The phone tries to connect but after a few times it says that the network is busy. When I try 'sip debug' to see if there is any signal from the voip provider, I see nothing
14:13.07*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:13.44_foxfire_Tagor hmm did you tried using tcpdump to see if any connections are conecting to the server ?
14:13.59docelm0Tagor, are you natted also or public ip?
14:14.03_foxfire_Tagor also did you "register" on your remote server ?
14:14.45synthetiqwhy would message fowarding work only half the time
14:15.36Tagor_foxfire_ >> No, let me try that. Yes, it registers in sip.conf. Though I am not sure if it is possible to put the register line in front of the extension
14:15.46*** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com)
14:16.03Tagordocelm0 >> natted, but all routers have static ip's and use DMZ. If I try x-lite it's no problem to get incoming calls
14:17.14Tagor_foxfire_ >> Is there a specific port orso which I can monitor with tcpdump? Else I get too much output
14:18.43_foxfire_hmm choose only udp trafic
14:19.01_foxfire_yoiu can also give the src host if you know its IP
14:19.27docelm0Where does Xlite sit in reference to your * box?
14:19.34docelm0same side of router or opposite sides
14:20.03docelm0Xlite also supports Stun
14:20.11Tagor_foxfire_ >> If I try 'tcpdump udp' and call asterisk then it does nothing
14:20.34Tagordocelm0 >> I 'replaced' the server with the xlite computer
14:21.00docelm0You have a config issue on your ast box.
14:21.20_foxfire_hmm hold on tagor use the src host then
14:21.27docelm0thats all it can be then  but as I said..  Xten does use STUN as asterisk doesnt
14:21.44*** join/#asterisk _deg_ (n=deg@200.150.147.29)
14:22.47_foxfire_tagor tcpdump sees it as IP
14:23.40*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
14:23.59Tagor_foxfire_ >> I tried 'tcpdump src host ip.address.of.sip-provider' and this also doesn't output anything
14:24.29Tagordocelm0 >> I understand that there is some setting wrong. But I can't find out which one since 'sip debug' doesn't show anything
14:25.11Tagordocelm0 >> Note it has worked before with the current router settings. So I don't think it's a nat problem
14:25.23docelm0I dont know where to even tell you begin..   do a sip show settings  and make sure its bound to all the correct IP's
14:25.25*** join/#asterisk macaco (n=tiago@joltid-gw.joltid.org)
14:25.30macacohey
14:25.37fjeanhello, anybody uses a grandstream 286 here ?
14:25.42*** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net)
14:25.47docelm0Good lord no
14:25.53fjean:- )
14:26.00macacohas anyone been using the svn asterisk? there's a huge memory leak when establishing zap/sip channels
14:26.07Tagordocelm0 >> As far as I can see the ip's are correct. Else I also wouldn't be able to make calls
14:26.15macaco132 bytes to be precise, corresponding to the pool string allocation
14:26.18macacoanyone noticed this?
14:26.23_foxfire_fjean i use an grandstream gxp2000 and likes it
14:26.26docelm0macaco, check the bugs
14:26.38*** join/#asterisk coppice (n=chatzill@221.162.17.210.dyn.pacific.net.hk)
14:26.47docelm0_foxfire_, I have 75 GXP2000's deployed in 3 countries
14:26.59fjeanfoxfire: you had no ring tone with it at some point while you were using it ?
14:27.37docelm0Tagor, thats right..  forgot your not using your linux box as a route
14:27.39docelm0router
14:27.40_foxfire_nope i tftpd a few ring tones of my own , but never had problems with it
14:27.46docelm0which is my setup
14:28.11sherif_where can i get hold on music or asterisk ?
14:29.26macacodocelmo: there's no bug about it
14:29.34docelm0macaco, submit one
14:29.36xachenFreeplaymsic.com sherif_
14:29.42docelm0sherif_, any MP3 works
14:29.52macacook... i'd just like to know if someone had already noticed that :)
14:29.56xachenyeah but Freeplaymusic.com is royalty free :)
14:30.26docelm0macaco not me.. then again I use ZAP/SIP @ my office only
14:30.44sherif_docelm0: no not all is working :-s i put some MP3 which is not working.;
14:30.47docelm0macaco, what did you use to find the leak?
14:30.50sherif_i need like 8khz music
14:31.01xachenit'll transcode it
14:31.03xachento 8khz
14:31.15docelm0Im using full blown MP3's and they work fine
14:31.17xachenI usually take mp3s and convert them to raw format though. saves lots of cpu
14:31.24*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
14:31.49xachendocelm0: yeah, that works but its way easier to just transcode it to raw first and watch your CPU loads drop so they can be used for other important tasks
14:32.15macacodocelm0: i just connect a sip client and terminate asterisk
14:32.44Tagor_foxfire_ >> Any other suggestion?
14:32.52docelm0macaco, I feel ya..
14:32.58docelm0Tagor, Call Digum
14:33.03docelm0err Digium
14:33.19*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:33.19*** mode/#asterisk [+o anthm] by ChanServ
14:33.20macacodocelm0: my guess is that the pool isn't freed when the channel is freed...
14:33.38_foxfire_tagor what does and "sip show peers" show by the way ?
14:33.39docelm0macaco what are you using to find this leak?
14:34.11macacodocelm0: ah sorry :) I'm using valgrind
14:34.21sherif_xachen: can i see ur onholdmusic.conf ?
14:34.30xachensure :p
14:34.56xachen[default]
14:34.56xachenmode=files
14:34.56xachendirectory=/home/telecom/root/var/lib/asterisk/moh
14:34.58xachenthats it
14:35.48Tagor_foxfire_ >> It shows: 12connect/**MY-PHONE-NUMBER**      **IP-OF-THE-PROVIDER**         N      5060     Unmonitored
14:36.24Tagordocelm0 >> Why should I call them?
14:36.30sherif_xachen: where the command to run it :D?
14:36.30*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
14:37.04xachenerm
14:37.05xachenthere is none
14:37.08xachenits * 1.2
14:37.12xachenit has its own builtin mp3 player
14:37.39_foxfire_tagor and "sip show registry" ?
14:37.58docelm0Tagor, your asked for more help
14:37.59Tagor_foxfire_ >> All registered
14:38.09fjean--- hi anyone would know how to force a ring tone on * for SIP devices ?
14:38.41sherif_xachen: mine is Asterisk CVS-v1-0-03/29/05-08:24:40 built by root@localhost on a i686 running Linux
14:38.47xachenerm
14:38.52xachenUpdate :)
14:38.58xachenthats real old
14:39.05sherif_xachen: seems too ;-) i don't why why i have this one already !!
14:39.05xachenyour going to have nothing but problems with that version
14:39.20sherif_i'm not the guy who installed it.
14:40.03*** join/#asterisk Sebb (n=sebastia@einstein.f0o.de)
14:40.25GBLRAdont you have to pay freeplay.com for usage?
14:40.40_foxfire_Tagor it's not easy to debug like this , are you shure you put the correct local extension ate the end of the register line
14:41.16Sebbhi.. is it possible that dial gives a busy, if any of the called channels ("ZAP/g1/11&ZAP/g1/12&SIP/13") is busy? that would make sense if there is only one person and some phones.. ;)
14:41.32*** join/#asterisk lthnnpwr (n=sdf9sfo3@195.166.60.12)
14:41.53Tagor_foxfire_ >> The register line is just like this: register => **MY-PHONE-NUMBER**@**IP-OF-THE-SIP-PROVIDER**
14:42.10lthnnpwrhi, may I ask you a question guys? you might know the answer, hopefully
14:42.26_foxfire_hmm my sip register lines are a bit more comples
14:42.38Tagor_foxfire_ >> Or should I use something like this: register => **MY-PHONE-NUMBER**@**IP-OF-THE-SIP-PROVIDER**/contextname
14:42.47Tagor_foxfire_ >> Can you give me an example?
14:43.03_foxfire_;register => xxxxx:yyyyy@proxy01.sipphone.com/1001
14:43.14_foxfire_where 1001 is my local extension
14:43.26_foxfire_at least thios worked for me
14:44.03_foxfire_Sebb i am having a similar problem
14:44.54Sebb_foxfire_: perhaps that is possible with chanisavail, but i'm not sure if that would work with zap-channels (zaphfc), too
14:45.05_foxfire_Sebb i did an upgrade from asterisk 1.0 to 1.2.4
14:46.21lthnnpwrHi. I am trying to install pwlib152 - the updated one, and the compilation fails with theses errors: http://pastebin.com/564375
14:46.39_foxfire_when i try to call SIP/foxfire&SIP/3001&IAX2/foxfire and one channel does exist the CDR always gives me FAILED even if it is answered .
14:47.08_foxfire_Sebb does this happen to you too ?
14:47.23Tagor_foxfire_ >> Just tried that too, but I don't get any incoming call :(
14:47.26*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
14:47.27fugitivo_foxfire_: why don't you create an extension and call that extension using Local/xxx
14:48.08*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:48.25Sebb_foxfire_: no. i don't have problems with unexpected behaviour, i just want to extend functionality.. so, if i already use one phone, other people who try to call my get a busy, because i can't use two phones at the same time
14:48.48_foxfire_fugitivo , hmm i wanted to ring several phones simultaneosly
14:49.14fjeancan we setup call progress indication with ztdummy ?
14:49.24*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.201)
14:49.25fugitivo_foxfire_: create the extension with multiple devices, then call that extension using Local, don't call the extension directly
14:49.29Kernel_corehi there ...
14:49.51Kernel_coreis ASterisk 1.2.4 Debian Package availabe?
14:50.13_foxfire_Sebb my useres have 2 logins  ext and login and i want to ring both at the same time
14:50.13razuis it normal that RDNIS info doesn't go forward in IAX trunk ?
14:50.47_foxfire_Sebb and it works but the CDR record is incorect
14:50.51lthnnpwrhas anyone else hot any problems compiling pwlib152?
14:50.59Sebb_foxfire_: sorry, i don't know..
14:51.16fugitivo_foxfire_: why don't you try what i told you?
14:51.40_foxfire_fugiutivo i  will can you give me an example
14:52.38_foxfire_Tagor sorry i am running out of ideas
14:52.53_foxfire_did you try this with an other provider
14:53.07_foxfire_like free world diallup for example
14:53.22ketanpTagor: what provider are you using?
14:53.25_foxfire_maybe it's not you problem
14:53.26TagorNo haven't tried that yet, _foxfire_. Thanks for your help
14:53.35Tagorketanp >> 12connect (astate)
14:53.49iCEBrkryo yo yo
14:54.33ketanpTagor: oh ok, sorry can't help then... try it with a free provider and see what happens... otherwise, go get some coffee and try again later, it may become obvious then
14:58.36*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
14:59.02*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
15:01.32*** join/#asterisk ramtha (n=ramtha@195.14.234.162)
15:01.34ramthapeace
15:01.58ramthahmm asterisk overlapdial=yes is not recognized by asterisk 1.2.4
15:02.05ramthawhy is that?
15:02.06ramthaany hint
15:02.08ramtha?
15:03.05tzafrirramtha, no idea here, but it would help if you mentioned where it has last worked for you
15:03.25*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
15:04.04mutilatoror hell even the imail imapd chokes on that much
15:04.06MRH2hi anyone forsee a problem using both monitor and mixmonitor in the dialplan to have 2 recording files
15:04.12mutilatorwtf
15:04.13mutilatoranyone know if linksys wrt54g's can do WARP?
15:05.35ramthahm no one uses overlapdial funktion?
15:05.37MRH2like:  4,MixMonitor(/tmp/MM/${CALLFILE}.g729|bv(1)V(1))   ...   5,Monitor(gsm,${CALLFILE},bm)
15:06.54*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
15:07.30nextimeramtha : i use overlapdial on a PRI
15:07.58lthnnpwrany idea why the pwlib152 update fails to compile? anyone encountered any problems with that?
15:08.09ramthanextime: ok, mei pstn switch dials only the firs 4 digits and pushes the call
15:08.27ramthareload from asterisk says me: setup_zap: Ignoring overlapdial
15:08.42ramthadoes this option go away?
15:08.58ramthai have a wildcard in the mashine
15:10.15*** join/#asterisk pnviking (n=pnviking@c83-248-7-150.bredband.comhem.se)
15:13.32lthnnpwraw. has ANYone encountered any problems while upgrading to pwlib152?
15:13.34*** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu)
15:13.41pycsuszhi Everybody!
15:13.54pycsuszI have got a question
15:14.14pycsuszFeb 20 16:11:37 NOTICE[2115]: chan_sip.c:10915 handle_request_register: Registration from '5004<sip:5004@80.95.69.180>' failed for '80.95.69.180' - Not a local SIP domain
15:14.30pycsuszsomebody can tell me something for this?
15:19.29*** join/#asterisk lodeon (n=not4u@h119n5c1o1023.bredband.skanova.com)
15:21.42*** join/#asterisk peanuter (n=saasdf@216.176.177.138)
15:23.31*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
15:25.01*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
15:28.43*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
15:32.49*** join/#asterisk coppice (n=chatzill@189.201.17.210.dyn.pacific.net.hk)
15:34.33*** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com)
15:34.57SeyrAnyone know how to set the time on Polycom phones? The servers time is NOT what is displayed on it
15:35.57ChrisUKneed a NTP service running on your server
15:37.04Seyrbesides that :-)
15:37.10SeyrI dont *have* to have one
15:38.13SeyrI was looking more for: "Seyr: The Polycom phones get their time setting from DHCP, so make sure your DHCP server is dishing out the correct NTP server and time offset"
15:38.55_Sam--if you know so much, what is the point of asking a question
15:39.24Seyrbecause I just stumbled across the web page with that info, thank you very much
15:41.15[TK]D-FenderSeyr : What is providing DHCP for your network?
15:43.19*** join/#asterisk marv[work] (n=timr@64.89.118.139)
15:43.40*** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net)
15:43.46paulhuynhplease help me asap
15:43.55paulhuynhi have major problem this morning
15:44.21*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc04v.dialup.mindspring.com)
15:44.32paulhuynhi did a asterisk upgrade overnight and this morning non of the device is registering to the server
15:44.47paulhuynhserver seem to be working
15:44.48*** part/#asterisk mhnoyes (n=mhnoyes@user-38lc04v.dialup.mindspring.com)
15:45.00Hmmhesays[TK]D-Fender: how goes it?
15:46.23paulhuynhplease help me
15:46.23paulhuynhanyone here
15:46.23[TK]D-FenderHmmhesays : Getting by.  recovering from a cold, but my * consulting projects are really picking up and I'm getting a job offer because of it from a digital based co going VoIP
15:46.40Hmmhesayscool, cool
15:46.44[TK]D-Fenderdefinately..
15:46.51*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
15:46.52HmmhesaysI tanked the intro to "my own worst enemy" in front of 300 people last night
15:46.58Hmmhesays:D
15:47.01fileevery time you need me you know I will be there
15:47.04fileyou know I really care
15:47.07filesay you will, say you dare
15:47.17filehot'n'sexy Hmmhesays!!!
15:47.21Hmmhesayshey file
15:47.24filehiya
15:47.26[TK]D-FenderGF situation is flakey but I'm attracting some attention and who knows what may develop so life is "improving" slowly.  Gotta go Appt hunting soon though...
15:47.36paulhuynhplease can anyone help me
15:47.48paulhuynhmy offic is completely dead in the water
15:47.51*** join/#asterisk danzig (n=chatzill@130.226.173.92)
15:47.51paulhuynhno phone service
15:47.53[TK]D-FenderHmmhesays : "Tanked" =good?
15:48.07danzigEHLO * gurus :-)
15:48.21Seyr[TK]D-Fender: Not sure. It is at a customers site in another state and they have a seperate network team that works on their network
15:48.23Hmmhesaysi went up and played a telecaster, never done it in my life, i completely bombed the 1st bar
15:48.27filepaulhuynh: that's why you keep backups... and revert if it fails
15:48.31Hmmhesaysas in screwed it up bad
15:48.36nozeycan someone help me with stun?
15:48.38[TK]D-FenderSeyr : are you provisioning the phones?
15:48.39Seyr[TK]D-Fender: I am trying to figure out if I can set the NTP servers via a config file, so I can do it remote
15:48.41fileHmmhesays: did they notice?
15:48.53*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
15:48.55NovceGurupaulhuynh, state your problem
15:48.57Hmmhesayshahah hell yeah they did i stopped smiled and said "lets try that again"
15:49.00Seyr[TK]D-Fender: All I have found so far is setting the ntp via dhcp
15:49.01[TK]D-FenderHmmhesays : OH...
15:49.07paulhuynhnothing is register to the *
15:49.08fileHmmhesays: ah yes
15:49.27Hmmhesaysthen I got more people on the dance floor than any of the other jammers that night
15:49.28filepaulhuynh: are you using trunk? if so, don't. it can be broken at any given time
15:49.30Hmmhesayswith pictures
15:49.31[TK]D-FenderSeyr : You can have the phones pick it up through DHCP or with SIP 1.6.3 (maybe .4 or .5) you can have it in the provisioning files.
15:49.51Seyr[TK]D-Fender: You know a link that shows all the config file params?
15:50.16paulhuynhno i use trunk just for sip acrrier and ext for the user
15:50.16[TK]D-FenderSeyr : its in the changelog.
15:50.29paulhuynhmy * was from asterisk@home
15:50.47paulhuynhand i did a * to 1.2.4 from 1.2.1
15:50.52Seyr[TK]D-Fender: kk, thanks :-) thats all I needed
15:51.14NovceGuruMy first experience was with asterisk@home, it wasn't a good one
15:51.17paulhuynhsystem status is showing everything is working
15:51.24Hmmhesaysahh the "if its not broke, don't fix it" theory comes into play
15:51.32paulhuynhbut no deviced or network can register to my *
15:51.45[TK]D-Fenderfile: !!!
15:51.50file[TK]D-Fender: !!!
15:51.55NovceGurupastebin your sip.conf (remove passwds)
15:52.00SeyrSeyr: !!!
15:52.16*** join/#asterisk MikeJ[Laptop] (n=vircuser@71-36-209-237.dlth.qwest.net)
15:52.21Seyrhey file, invoice is HTML, can I view it online?
15:52.33fileSeyr: can you? dunno, but I can
15:52.34[TK]D-Fender* has been kind to me.  Last friday's complete overhaul was virtually perfect, and I'm getting consulting jobs coming out all over the place now...
15:52.48Seyrfile: if you can view it, go ahead and pay for it as well :-)
15:52.53[TK]D-FenderAnd now a job offer...
15:52.54fileSeyr: I can have it resent as well
15:53.11paulhuynhtk d-fender
15:53.19paulhuynhcan you please help me
15:53.23Seyrfile: naw, it auto-debits, i was just curious how much this new client ran
15:53.29fileSeyr: ah I can go look
15:53.33fileSeyr: which account?
15:53.38HmmhesaysI could use some more asterisk jobs, mine are running a bit dry
15:53.40Seyrhrm.. no clue
15:53.57Seyrfile: i'll check and get back with you in #asterlink later
15:54.02fileSeyr: great
15:54.04danzigIf I download asterisk 1.2 and build from source, will conferencing work straight off? Last time I looked one needed to install zaptel and make a fake loopback adaptor... Is this still the case?
15:54.15generalhanhey guys can i get a little help ?? My sip phones are unregistering ALL THE TIME. sometimes they will stay registered, once i restart the phone, for an hour, sometimes an entire day. this all happened after i switched over from a VoIP provider to a PRI, and upgraded to 1.2.1 ... this is my sip.conf and a show peers if anyone could take a look at it and give me some suggestions :: http://generalhan.pastebin.ca/42213
15:54.21Seyrdanzig: yeh
15:54.48danzigseyr: yeh it will work or yeh I need zaptel? :-)
15:54.52Seyrdanzig: just read the README in zaptel dir and make sure you do all the needed stuff for ztdummy
15:55.09danzigok, thanx.
15:55.21Seyrdanzig: its flawless if you follow the instructions :-)
15:55.28filezaptel pseudo channels are cool...
15:55.29shido6got any zaptel interfaces or usb interfaces , generalhan ?
15:55.50generalhan<PROTECTED>
15:56.18paulhuynhcan someone please tell me how can i restore it from backup?
15:56.30generalhanall my phones are sip phones directly into the network that the asterisk server is on. the PRI line comes into that server and then the nic connects it to the rest of the phones on the network
15:57.32generalhanwhen the phone registers i get this " -- Registered SIP '7106' at 192.168.0.106 port 5060 expires 120" is that 120 minutes ? or what? maybe i have to set that way higher some how ??
15:57.39file120 seconds
15:57.42*** join/#asterisk buutymalapico (n=jorgito@82.113.32.241)
15:57.43buutymalapicohi
15:57.44generalhanhaha
15:57.54generalhanso that would explain it huh !!
15:58.07generalhanwhat conf file can i set that to something else ?
15:58.28fileit's on 'da phone
15:58.33generalhanhmm
15:58.33buutymalapicohave problem with festival, i can hear just half of sentence ..
15:59.19paulhuynhanyone here can help please
15:59.40Seyr[TK]D-Fender: congrats on the job offer!
15:59.41paulhuynhif $$ is involed please PM me
16:01.11paulhuynhHELP ME
16:02.38*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
16:02.39*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
16:03.32wunderkinlol, paulhuynh, did you make any config file changes after the upgrade? were you running a release? just reinstall the old version if you dont have a backup and deal with it later
16:03.32Hmmhesaysshe was on google talk earlier
16:06.44*** join/#asterisk lalito (n=erg@201.137.152.125)
16:09.17TagorWhich port does asterisk by default use for incoming calls?
16:10.03*** join/#asterisk kpettit (n=keith@69.15.174.114)
16:10.33lunaphytehello
16:12.17docelm0Depends on technology
16:12.31fugitivoTagor: port for what?
16:12.37fugitivoTagor: sip?
16:12.46TagorYeah
16:13.02fugitivo5060 udp, 10k-20k udp
16:13.04*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
16:13.26buutymalapicosomebody using festival here ?
16:13.38*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
16:13.43nozeyhey fubster i changed the rtp ports to 10000-10010 ... is this a problem?
16:14.16nozeyfugitivo ... not fubster ... sorry #)
16:14.21iCEBrkrha
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16:14.59fugitivonozey: if you have a lot of sip connections it will, 10k-20k are for rtp packets
16:16.19lunaphyteis it true that i need a certain firmware image file to get an older 12sp+ to work with asterisk?
16:19.04buutymalapicosomebody using festival here ?
16:19.36puzzledno I like my ears
16:19.48fugitivobuutymalapico: www.cepstral.com
16:20.15*** join/#asterisk robb_ (n=robb@kapow.vm.bytemark.co.uk)
16:20.33buutymalapicofugitivo, well i need help about festival..
16:21.04paulhuynhso no taker on my problem
16:22.24puzzledlet's see if I can find a taker for my mysterious problem...
16:22.27puzzledHELP
16:22.38*** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F41EA.dip0.t-ipconnect.de)
16:22.41buutymalapicopuzzled, what s the problem _
16:22.54puzzledbuutymalapico: that was not supposed to work :)
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16:23.03puzzledand I don't have a problem either actually
16:23.23zoayou rang my lord ?
16:23.29_foxfire_fugitivo : i tried what you suguested and dial a local extension , i receive a valid CDR now , but also the invalid CDR so i end up with 2 .
16:23.37puzzledzoa: hehe
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16:29.22fugitivo_foxfire_: hmmm
16:29.37fugitivo_foxfire_: isn't that better than only only invalid record? :)
16:29.57danzigCan anyone explain the logic of when CDR started/ended/written? Im messing with ForkCDR and ResetCDR(w), and niether do exaclty what I would expect... I need to do things like user comes in, gets dialtone, dials new number [start new CDR]
16:33.09danzigI'm getting error "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line 723" - but my sip.conf has 100 included files. Any clever tricks to figure out which file the error is in?
16:33.47fugitivo100 included files???
16:33.54fugitivogod
16:33.58fugitivowhy did you do that?
16:34.04peanuteri live in ny looking to play with iax and asterisk.  any suggestions as to where to get cheap service?
16:34.20danzig100 users :-) They can only muck up their own config...
16:34.40fugitivothere're better ways to do that
16:34.46fugitivo100 include files is impossible to admin
16:34.53danzigfug: how? :-)
16:35.04fugitivodanzig: a web page and a database?
16:36.10*** join/#asterisk mkrufky (n=mk@68.160.103.77)
16:36.11danzigThey do it themselves with a web interface. Works fine. Until now, there must be an error in our web interface that let someone wirte something wrong... Could put it in a DB, but would then have to chuck it into files anyway...
16:36.24iCEBrkr100 includes?!?!
16:36.26iCEBrkrWTF
16:36.40fugitivodanzig: into a single file
16:36.54fugitivodanzig: you don't need one include file for each user
16:37.28_foxfire_fugitivo : yeah , but it messes up my scripts, ;-) i will stay with 1.0.9 for now .
16:38.03danzigwell, actually you just do #include "sip-conf.d/trunk-*.conf"
16:39.39Crontibswow
16:39.45Crontibsthats pretty intense dan
16:40.27danzigit solves a concurrency problems - 1 user=1file=only 1 person trying to alter file at a time. which is the only thing a database has to offer in this case Works great. Have been in production 8 months.
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16:43.57mikefooiCEBrkr: sup sup
16:45.55iCEBrkryo
16:46.02iCEBrkrMon Feb 20 11:45:57 EST 2006
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16:50.40peanuteranyone else having compiling problems with zaptel on fbsd 5.4?
16:51.16*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
16:51.28mikefoopeanuter: you have zaptel card or emulating it?
16:51.36peanuteremulating
16:51.45tmccraryIs there any functionality in asterisk to "barge into" a converstaton between two people? Like forcing a conference call?
16:52.26trixtertmccrary: you can always transfer people into a meetme
16:52.46trixterdepending on system speed it might be anything from noticable to barely perceptable
16:53.07trixterI think thgere is a barge app though not sure
16:53.21tmccraryoh ok, thanks
16:53.40mikefootrixter: hey sup..
16:53.55trixternot much
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16:54.24mikefooany other decdent books on asterisk besides the voip-info one?
16:54.32mikefooneed to get some reading material at b&n today.
16:54.37trixter~docs
16:54.38jbotsomebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
16:54.53[TK]D-Fendermikefoo : the only one I could suggest would be TFOT
16:54.58[TK]D-Fender~thebook
16:55.00jboti guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
16:55.05trixteroh that way, no not really ,there is another asterisk related voip book, but there are many non asterisk voip books that are good to read
16:55.14tmccraryalso, is there a way to "barge in" but only have one side of the conversation hear you? Say for training so you can talk to YOUR rep but not have the "customer" hear what you're saying (for like coaching lets say)
16:55.16*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
16:55.26trixterthey tend to cover different aspects of voip in general, some in great detail about the technology itself
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16:58.32develhey all.  anybody make a decent wall mount sip phone?
16:59.03[TK]D-Fenderdevel : What does it need to do?
16:59.44develnothing special, just hang on a wall.  everything i have here (sipura, grandstream, polycom. snom) won't mount on a wall...
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17:00.31mutilatoruse an ata?
17:00.39mutilatorand a wall mount phone..
17:00.54*** join/#asterisk fjean (n=fjean@201.29.122.10)
17:01.00[TK]D-Fenderdevel : Well.... its not a great phone but I bought Uniden UIP-200's for wall mounting in public places (less $ risk if vandalized, and they support PoE
17:01.07_Paulo_devel, try connecting a standarrd wall phone to an ATA,
17:01.11[TK]D-FenderThough I wouldn't suggest them for anything else....
17:01.26[TK]D-Fenderdevel : Actually yeah... ATA would be the best way....
17:01.32[TK]D-Fendercheapest too.
17:01.52develyeah, they're for use in a "shop area", and i didn't really want to go the ATA way.  but from a cheapness standpoint, i guess i can consider it.
17:01.59tmccraryis there a way to "barge in" but only have one side of the conversation hear you? Say for training so you can talk to YOUR rep but not have the "customer" hear what you're saying (for like coaching lets say)
17:02.09develthanks, all.
17:02.10mutilatortmccrary: no
17:02.16mutilatorso stop asking
17:02.55mutilatorif they're for use in a 'shop' i'de say do it anyway
17:03.02mutilatordon't want a $100 phone hangin on the wall
17:03.06mutilatorrather a $10
17:03.15mutilatoru can put the ata in a closet somewhere
17:04.09develyeah, i guess that does make more sense, just put the ATA in the wiring cabinet.
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17:12.48mutilator=\
17:13.01mutilatori just noticed, the lil bumps on my f and j keys are gone
17:13.28*** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com)
17:14.34[TK]D-Fenderdevel : Whats wrong with ATA?  It mean you can put a phone that will get abused there...
17:15.02mutilatori think we already knocked sense into him
17:15.04mutilatoro_O
17:15.06mikefoo[TK]D-Fender: thanks for the book advice, I will download the pdf today along with the other docs  :)
17:15.28[TK]D-Fendermikefoo : I duplex'd it here a few times for backup for when the WIKI goes down :)
17:15.54mikefoo:)
17:17.31mikefoocould someone point me in the right direction if I wanted to run a incoming line with ani ability, what I would need to do
17:19.50t0keanyone know if I can to have enable musiconhold if I havent soundcard on server?
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17:20.30[TK]D-Fendert0ke : Yes
17:20.44GoRKt0ke: mpg123 does not require a sound card, it decodes to stdout where asterisk reads the stream. Native moh will also work
17:20.57[TK]D-Fendermikefoo : ANI?  As in "CALLED" # (like DID)?
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17:21.18mikefoo[TK]D-Fender: yup
17:21.19t0keok, thnks..then will be error mine in extensions.conf
17:21.33mikefoohttp://en.wikipedia.org/wiki/Automatic_Number_Identification
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17:22.11[TK]D-Fendermikefoo : Your choices typically look like PRI/T1 or a VoIP provider that does the same.
17:22.16[TK]D-FenderWhat scale of solution?
17:22.25GoRKmikefoo: as a customer it is very difficult to get ANI. typically you can only get it on things like 900 numbers but sometimes your provider will give it to you on 800 numbers.. do you have a line with ANI already?
17:22.48danzigHa - found error. Was not actually from the web interface that edits the include files, was manual edit by the other sysadmin :-(
17:23.02GoRKmikefoo: in any case you can only read ANI over ISDN/PRI D channels
17:23.09danzigCan anyone explain the logic of when CDR started/ended/written? Im messing with ForkCDR and ResetCDR(w), and niether do exaclty what I would expect... I need to do things like user comes in, gets dialtone, dials new number [start new CDR] continue with new CDR
17:23.23mikefooGoRK: no line with ANI as of now.
17:23.37mikefooso I need to find a voip provider that is offering ani, you are saying?
17:23.54GoRKmikefoo: why do you need ANI instead of just caller id?
17:24.30mikefoomy clients don't want to see "blocked call" on records.
17:24.59[TK]D-Fendermikefoo : I take it you wnt a number of differnt DID's from them?
17:24.59mikefooits for lead generation so they can call back who ever calls them.
17:25.17mikefoo[TK]D-Fender: what do you mean?
17:25.20GoRKmikefoo: well i will tell you now that telling that to anyone capable of providing you ANI service will not be a sufficient reason to sell it to you
17:25.52develyeah, [TK]D-Fender, i don't really know what i was thinking there...  kind of a "all ethernet" mindset.  i have indeed had sense knocked in to me :)  thanks
17:25.53[TK]D-Fendermikefoo : as in you want a # of differnt phone #'s targeting your PBX and for your outbound calls to identify as coming from the specific # that will lead to the agent that called them?
17:25.59GoRKmikefoo: you can sometimes get ANI on 800# service though so that's probably your best bet
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17:26.26[TK]D-Fenderdevel : I run ATA's in my switching room, and plug the el-cheapo phones into the RJ45 jack direct.  works great...
17:27.10mikefooGoRK: but I was reading and came across: The Federal Communications Commission allows anyone who is paying for a phone call to know who is calling them.
17:27.39GoRKmikefoo: that is why you can sometimes get it on 800 number service and why you can on 900/976 service
17:27.44Hmmhesaysfreaking autofall through
17:28.09mikefooAhh ok..
17:28.29mutilatoranyone wanna buy a benchmade billabong butterfly knife? $150
17:28.55mutilatorhttp://www.icegn.net/knife/
17:29.13mikefoo[TK]D-Fender: oh, well no its other way around, I have numbers pointing to PBX, client calls into PBX with *67 for instance, my client is charged for the call, but doesn't know who called, so thats why they are bitching.
17:29.26GoRKmikefoo: but if your caller is not calling into your 800/900/976 number, then no ANI.. also the only way to actually receive the ANI is via ISDN/PRI -- so you'd need an ISDN/PRI or a T1/PRI at the very least.. or buy an 800# from a voip provider that will relay the ANI to you instead of the normal callerid
17:29.32paulhuynhdoes anyone here work on asterisk@home
17:29.35paulhuynhcan help me
17:29.39paulhuynhI broke my
17:30.12paulhuynhasterisk will not register any device anymore after upgrade centos + asterisl1.2.1 to 1.2.4
17:30.15mikefooGoRK: yah I was thinking of getting 800# from voiup provider that can possibly relay ANI information to me.
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17:30.54GoRKmikefoo: it would be a specialized sort of deal.. afaik no protocols have provisions for both callerid and ANI at the same time, so they'd have to replace callerid with the ANI on calls coming to you
17:31.51GoRKmikefoo: but i dont see any reason they couldn't do it from a technical perspective.. they would of course have to be able to receive and capture the ANI with whatever equipment they have
17:32.06mikefoooh yah? ahh ok then maybe I need to not charge clients for blocked calls I guess, I will see how that goes.
17:32.29mexuar-timGoRK: actually IAX has an info element for ANI. I don't know if any one supports it.
17:32.46GoRKmexuar-tim: ah didnt know that/ guess i should read more code :)
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17:34.11mexuar-timGoRK: I'm deep into debugging some IAX code, so I have the spec open all the time :-)
17:34.55GoRKmikefoo: anyway normally the ANI is simply used behind the scenes by whoever is providing the 800# service for billing purposes.. so it's not strictly necessary to pass it to the customer WITH the call.. so you will have to negotiate with a provider to do this for you.. even if you are dealing with a CLEC or something, they may (and usually are) buing 800# service from someone else so they may not even be getting ANI themselves
17:34.59mikefoowell I need to buy some dids anyway.. anyone have experince with virtualphoneline?
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17:37.38filewe do get ANI at Asterlink, but we don't pass it to the customer... like GoRK said we use it for rare billing instances
17:37.48fileie: US cellphone customer roaming in Canada, gets billed at Canada inbound rate
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17:39.52mockerDoes asterisk have a drop folder where I can make it call by placing a file in that directory?
17:40.05wasimmocker: /var/spool/asterisk/outgoing
17:40.23mockerwasim: Awesome, thanks.
17:41.41GoRKmocker: it's very greedy about .call files though; make sure you create them somewhere else on the FS then use mv to put them in the spool directory.. it doesnt do any kind of checking to make sure it's got a complete file not open by other processes before it reads it in.. if you use cp or write directly to the dir sometimes asterisk will get at the file before you have finished creating it
17:42.47GoRKmocker: also be wary about too many files in the spool dir at once.. try to serialize them a bit if possible with a slight dealy between -- 500ms to 1s should do
17:43.33fileif god had a name what would it be and would you call it to his face if you were faced with him in all his glory
17:43.42filewhat would you ask if you had just one question?
17:44.26Nivexfile: are you feeling alright? (that was directed at you, not God)
17:44.34filealways
17:45.18GoRKi'd ask him who wants to buy a benchmade butterfly knife? WTF is it with all the offtopic crap? we might as well all start with the Chuck Norris stuff
17:46.14fileGoRK: it's very very... calm today
17:46.51GoRKChuck Norris while having a calm day at home was watching television when he became furious and began walking down the street punching every kid he saw and screamed "Trix are for Chuck Norris"
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17:51.09filedang nabbit my printer is broken
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17:51.57GerbilWrkCan you have hardcoded agents in a queue, plus agents that can call in and add themselves to it?
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17:56.02[TK]D-FenderGerbilWrk : yup
17:56.45robb_i'm having a problem getting pstn->voip to work. i have an asterisk box running on fedora kitted out with an x100p. when i ring the line from the pstn the recorded greeting sounds fine, but when i try and dial an extension to a sip/iax client there's a lot of hiss. does anyone know why that might be the case?
18:00.53mikefoofile: so I am not passing it to customer, I am just including it in billing statements. so my clients don't see a bill for a blocked number.
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18:04.10[TK]D-Fenderrobb_ : Is your X100 on its own IRQ?  Running any other nasty processes like X?
18:04.16mikefoofile: I also am thinking about going with asterlink, I heard some good things.
18:04.47robb_no, i've killed X and other nasty things like that. it's on its own irq
18:04.52robb_irq 10
18:05.02robb_getting 1000 interrupts a seconds
18:05.16filemikefoo: well you're more then welcome
18:05.19[TK]D-Fenderhmm... set up an IVR option for an echo test and see if it sounds clean while it remains within *.
18:05.44robb_ok. good idea
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18:06.14Mothergreetings
18:06.30Motheris there a way to get asterisk to answer SDP payloads?
18:06.59Motheror if not, any pointers as to where in code I could start prodding?
18:07.15zoarobb_: thats normal
18:07.20austinnichols101~seen opsys
18:07.25jbotopsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 7d 12h 26m 23s ago, saying: 'betaboi" true'.
18:07.25zoa1000 interrupts per second are ok
18:08.18MotherI have a bunch of a= records coming from a client, but asterisk only sends back a bare 200 reply
18:09.11robb_zoa: yeah i meant that its not an irq issue because it's getting the 1000 interrupts a second
18:09.41GerbilWrkanyone have experience making snom 360's not show a missed call if the missed call goes to a queue?
18:13.04[TK]D-FenderGerbilWrk : Sorry... if a SIP phone misses a call it'll have no idea what the server will do with it after in order to make a decision....
18:13.04robb_[TK]D-Fender: during the echo test there was a low frequency noise even when there was no audio being input
18:13.37[TK]D-Fenderrobb_ : Plug a real phone into the jack you use for the X100.  Does it sound crappy as well?
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18:14.08robb_[TK]D-Fender: tried that already. the line sounds ok
18:14.41[TK]D-Fenderrobb_ : Hmmm.... played with the gain/loadzone any?
18:15.00[TK]D-Fenderis the noise a stable constant hum or variable?
18:15.20robb_constant hum. played a bit with the gain a while ago
18:15.39robb_hum becomes distortion when i speak into the line
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18:17.37Motherso, no takers on the SDP issue?
18:18.52[TK]D-Fenderrobb_ : Hmmm... maybe an impedance thing or other electrical interference....
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18:20.35robb_[TK]D-Fender: sounds plausible alright.
18:20.51robb_[TK]D-Fender: might try moving it. cheers
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18:22.54[TK]D-Fenderrobb_ : Glad if it helps...
18:22.59rene-hello
18:23.27rene-is the tdm400  5v or 3.3v?
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18:24.19[TK]D-Fenderrene- : dual IIRC
18:25.15rene-[TK]D-Fender: well it seems that it would fit better in the pci-x slots of the server than in the 5v 32bit rated ones
18:25.44rene-the tdm400 has an extra rib not found in the pci socket but well it did went all the way so
18:26.04[av]bani\o/
18:26.25iCEBrkrrene-: I believe it's auto-sensing.  I can only make this assumption from the picture.
18:26.47iCEBrkrrene-: It's slotted for both
18:28.26rene-i see, well i need to turn the server on and see if it works,
18:28.55rene-thanks, otoh is there anyone from signate in here? i have some questions regarding their product
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18:36.24QbYdoes anyone know how a MOS score can be calculated automatically for each call?
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18:42.55trixteryou can automatically calculate it
18:43.08trixterthere are 2 basic ways of doing MOS scores
18:43.23trixterone is you have a group of people rate the call 1-5 average em together there is your score
18:43.27trixterthe 2nd is twice as good
18:43.30trixterthey rate it 1-10
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18:44.55Abydos313good morning everyone
18:44.59trixterI have asked digium about a year ago if they had any MOS scores and under what environment that was done, they told me (at that time) they hadnt done any such testing.  and in reality the system, network, endpoints all add into that so you may end up with a lower mos score just because you were using a worse phone or had network congestion
18:45.26Qwell[]or can't figure out how to fix echo properly
18:45.31Abydos313i'm having an issue with AMP. all works except the 'extensions' link
18:45.35mutilatoror doing it over 9 mile wifi links
18:45.36Qwell[]~amp
18:45.37jbotmethinks amp is NOT supported here! people using it should join #amportal
18:45.48Qwell[]mutilator: or 11 in ManxPower's case
18:45.55mog_workhey qwell
18:45.56trixteryou can try to guess what the score might be if you look for high jitter, dropped packets, and run sphinx or something looking for the word 'what'
18:46.00Qwell[]mog_work: y0
18:46.01mutilatoryea, we have 1 20 mile link
18:46.08Qwell[]mog_work: Why isn't today a holiday?
18:46.14mog_workyou have been working on chan_skinny a lot lately
18:46.18Qwell[]or, rather...why am _I_ at work on this holiday? :(
18:46.18mog_workwe dont work for the gov
18:46.21mutilatortry not to do more than 10 tho
18:46.30Qwell[]mog_work: I work for a bank...95% of the people here are off. :(
18:46.38Qwell[]it's so lonely here, heh
18:46.39*** part/#asterisk rene- (n=rene-@201.127.101.127)
18:46.46trixterwhat holiday?
18:46.52Qwell[]presidents day or something
18:46.57mog_worklol
18:47.03Qwell[]mog_work: never touched chan_skinny...only chan_sccp
18:47.09mog_workthats what i meant
18:47.17mog_worki need some help writing rtp handlers
18:47.21mog_workfor the channel im writing
18:47.33Qwell[]I've not gotten that far into it...
18:47.37mog_workahh okies
18:48.02Qwell[]chan_skinny is probably unbloated in that regard though
18:48.11Qwell[]might be a good place to look for advice
18:48.26Qwell[](the real chan_skinny...)
18:48.48mog_workahh
18:48.54*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
18:48.56mog_workreading chan_sip gave me a headache
18:50.01ZipKidhi, simple question, is it possible to use Asterisk with a simple external serial modem for the incoming telephone line?
18:50.25Qwell[]ZipKid: and do what with the modem?
18:50.33Qwell[]use it as a modem, over voip?
18:50.37cypromisd beep beep beep
18:51.00ZipKidrecieve incoming calls and sent them on to an internal softphone for instance
18:51.06Qwell[]no
18:51.08ZipKidno, pots
18:51.11Qwell[]you need an fxo
18:51.18Qwell[]modem != fxo
18:51.43trixteror someone to press the bare phone wires on their tounge and is really good with a morse code keyer so they can put the bits on the ethernet wire in realtime
18:51.45mog_workwell it would if someone wrote a driver for it, and it was a capable modem
18:51.46ZipKidah, ok....
18:51.49trixterbut watch for jitter on that setup
18:51.55Qwell[]mog_work: well, sure
18:52.09Qwell[]but, in that case...you could use anything, really
18:52.19ZipKidQwell[]: thanks, that is not clearly defined anywhere i can find..
18:52.25mog_workZipKid, you can get an x100p off of ebay for 10 bucks or a digium tdm01b for like a hundred
18:52.27Qwell[]rj11 over serial port, rj11 over ethernet (hey, if there are drivers, it could work, right?)
18:52.35*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
18:52.53ZipKidmog_work: i have seen thos things but i wanna play now ! :-)
18:52.55Qwell[]actually, rj11 over serial would be kinda cool, heh
18:53.16Qwell[]just get an el-cheapo adapter for like $1.50
18:53.33mog_workwell get one overnighted
18:53.35Qwell[]you don't need a dsp, do you?
18:53.52mog_workdo it in software qwell
18:53.58Qwell[]yeah, that's what I'm thinking
18:54.07Qwell[]there are already thinsg that do that, right?
18:54.10Qwell[]in asterisk, that is
18:54.14mog_workthats what i was telling marko
18:54.19mog_workwe should stop selling t1 cards
18:54.22Qwell[]heh
18:54.26mog_workand just do ethernet
18:54.27Qwell[]just do it over ethernet?
18:54.30Qwell[]yeah, totally
18:54.32ZipKidTDM01B : 1 FXO port  124.22EUR
18:54.36mog_workand do all framing signalling garbage in software
18:54.42ZipKidno shipping included
18:54.52Qwell[]mog_work: good idea, except the whole paying Digium's bills thing :p
18:54.55mog_workthats a bit of a markup we sell it state side for around 100 i thought
18:55.13mog_workwell and it would be immensally difficult to do in real life
18:55.29Qwell[]Kevin could knock it out in 4 hours!
18:55.30Qwell[]heh
18:55.40Abydos313g4 kevin?
18:55.46mog_workgetting kevin for 4 hours is incredibly difficult though
18:55.54Qwell[]mog_work: indeed, heh
18:56.07Qwell[]the other day, Kevin said "anything in asterisk can be done in 4 hours"
18:56.16Qwell[]"or...that's what the sale guys tell everybody"
18:56.20Qwell[];]
18:56.30mog_worksounds like sales
18:56.40denonthats what the tech guys always say too
18:56.50ZipKidQwell[]: but one of those crappy winmodem cards would do the trick? Or have i again terribly misread things...?
18:56.59mog_worklol
18:57.02Qwell[]ZipKid: You've terribly misread things...
18:57.07Qwell[]~fxofxs
18:57.18jbotmethinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
18:57.22Qwell[]thank you
18:57.34denon~fwap Qwell
18:57.47Qwell[]ZipKid: If you wrote zap drivers for the modem, you could use it
18:57.51Qwell[]but...good luck
18:57.58ZipKidhum.. no thanks
18:58.18denonnot to mention, not all modems would duplex well, as I understand it ..
18:58.20Qwell[]denon: My standard answer is "a few days"
18:58.34denoner, FD
18:58.46Qwell[]"Oh, you need a typo fixed?  That'll take me a few days"
18:59.31denon"But ... but .. it's just on the wiki!"
18:59.52Qwell[]"Yes, but there are change control processes you're forcing me to follow.  So...it'll take me a few days."
19:01.16*** join/#asterisk teleniek0 (n=marc@167.Red-80-35-144.staticIP.rima-tde.net)
19:01.44*** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-61-29.dialin.kamp-dsl.de)
19:01.56teleniek0hi ppl. would you give me some clues on getting a TE110P working on Spain ? when I plug the card to the PRI box a nice "ALM" led blights on it ;(
19:02.09*** join/#asterisk clint_ (n=clint@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
19:02.12Qwell[]teleniek0: got the right kind of cable?
19:02.22Qwell[]I know zero about T1, so...yeah
19:02.50teleniek0Qwell[] the cable is a straight cat5?
19:02.58Qwell[]got me
19:03.18teleniek0wasn't sure but hope I guessed hehe
19:03.31teleniek0it's E1 not T1 anyway :))
19:03.41*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
19:03.42*** join/#asterisk PaulHuynh (n=paulhuyn@c-68-37-18-82.hsd1.de.comcast.net)
19:03.43Qwell[]should be the same
19:03.49PaulHuynhhello everyone
19:03.59PaulHuynhi'm rebuilt the asterisk@home
19:04.07teleniek0maybe, anyway it's red red red all the time ughh x"D :)
19:04.10Qwell[]teleniek0: tried calling Digium?  You get free installation support for buying a card
19:04.14PaulHuynhi'm going to recreate all the ext +  config again
19:04.21Qwell[]They're there to help...and they're really good I hear
19:04.24teleniek0Qwell[] didn't know! hehehe thanks :)
19:04.29PaulHuynhbut i wonder how can i keep my old CDR
19:04.54MRH2hi anyone forsee a problem using both monitor and mixmonitor in the same extension  to create 2 recording files
19:05.16Qwell[]MRH2: besides that it'd be pointless?
19:05.29MRH2yeah besides that
19:05.36Qwell[]probably.  try it
19:05.41MRH2something like 4,MixMonitor(/tmp/MM/${CALLFILE}.g729|bv(1)V(1))   ...   5,Monitor(gsm,${CALLFILE},bm)
19:05.52*** join/#asterisk dpolitech (n=Owner@207.224.48.130)
19:07.03Qwell[]seems kinda silly to me, to write 3 streams to disk at a time when one will do
19:07.26MRH2mix monitor let me down earlier this year so i am trying to ease into it
19:08.00*** join/#asterisk trelane_ (n=trelane@209.43.90.13)
19:08.05*** join/#asterisk Cresl1n (n=matt@146.229.177.231)
19:09.20jontowwoo, fixed another issue with the VM machine.. longstanding one too :)
19:09.37*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
19:09.41Qwell[]VM machine?
19:09.50Qwell[]that kinda like an ATM machine?
19:10.19jontowvoicemail for our CLEC ;)
19:10.36Qwell[]your CLEC carrier?
19:10.39jontowit sits behind a coppercom softswitch, and gets calls thrown at it via forward-on-busy-or-unavailable
19:10.44jontowwe are the CLEC
19:10.44jontow:)
19:10.53Qwell[]You don't get it yet, do you? :p
19:10.57jontowof course not!
19:10.58Qwell[]too subtle, I guess
19:11.09Qwell[]VM machine is redundant :P
19:11.24jontowhow so?
19:11.30Qwell[]M = machine
19:11.35jontowM = mail
19:11.35Qwell[]oh, haha...voicemail
19:11.40jontow;)
19:11.44trixterbut its not EM
19:11.46trixterwonder why
19:11.50Qwell[]EM?
19:11.53trixteremail
19:11.53Qwell[]email?
19:11.56Qwell[]hmm
19:11.57jontowtrixter: because the E is the only important letter!
19:11.58trixteras in electronic mail
19:12.01Qwell[]yeah
19:12.08jontowmail has happened forever.. but *E*mail, now thats something new and exciting!
19:12.09Qwell[]though, some people do refer to it as vMail
19:12.11trixterand its not SM for snail mail
19:12.18Qwell[]no, but SM + squirrelmail
19:12.23trixteralthough content filters may have a problem with people tossing letters like S&M around
19:12.36Qwell[]s/+/=/
19:13.07*** join/#asterisk bjohnson_ (n=bjohnson@66.11.164.106)
19:13.15jontowbut, aside the semantics of the issue.. we had a longstanding problem because we had to swap the fields that were being shoved into the voicemail system, so it would ring into * as a DID, though none of the numbers were actually destined for it..  creates a small problem when you also have a generic "dial X number from outside our network to check your voicemail from anywhere!"
19:13.39jontowso i had to play some magic with a perl script named 'ismbox.pl' and a SQL query or two..
19:13.57jontowfelt like an ass once i'd done the proof-of-concept deployment on my dev. box and realized it was super-easy to write
19:14.01Motheryo
19:14.01jontow:P
19:14.12Mothertrixter....as in the tscm-l trixter?
19:14.13Qwell[]brb
19:14.21jontowtook 25mins to write it.. mostly because i don't know perl
19:14.23*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:14.25trixteryes
19:14.29*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
19:14.30Motherwell greetings
19:14.38trixterhi
19:14.44trixterwhich list?  there are so many
19:14.53MotherJMA's one
19:15.00trixteroh the sucky one
19:15.02Mother*the* TSCM list
19:15.04MotherLOL!
19:15.23Motherthe Uhrigh-bashing one
19:15.30trixterits all lame jokes and random spam from him, since moving I noticed he is no longer using it to harass someone (against yahoos rules and most likely why they canceled the group)
19:15.50Motheryeah, was the group actually cancelled by yahoo?
19:16.03MotherI was rather busy at that time and have some 200 backlogged messages
19:16.15*** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net)
19:16.17trixteryeah after 6 months of posting personal information and using it to harass someone...  he claims its becuase he posted information that is publicly available on any ham radio site
19:16.37trixter200?  I get that many in a day from all the lists especially asterisk-users
19:16.43Motherwtf...
19:16.54Motheroh yeah, the asterisk one is quite an exercise :D
19:17.11Motherat one stage I had 20k unread on that one
19:17.14trixterI think asterisk-users is averaging about 150/day now
19:17.22PaulHuynhanyone?
19:17.36Motherjeez
19:17.46PaulHuynhi need to tranfer some cdr info over from my dead asterisk to my working one
19:17.53Motherany talk of PoC? I'm trying to get it working with the Nokia client
19:17.56*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
19:18.16Motherbut there's a bunch of crap on the SDP portion of the register message, to which * doesn't answer
19:19.07MotherI thought of writing a proxy that took care of that and translated it asterisk-friendly, but I'd rather have it do it natively
19:19.53*** part/#asterisk godfree2 (n=user@hlfxns0148w-142167238144.ns.aliant.net)
19:19.56Abydos313would be nice if asterisk only used one or two static ports for everything. would help with fireall issues
19:19.57filea REGISTER shouldn't/needn't contain SDP
19:20.24fileand last I checked the Nokia client used a codec that Asterisk didn't/couldn't support due to licensing
19:20.51MotherAbydos313: it's called IAX
19:21.15remisseeks
19:21.24Motherfile: yes, it's AMR, but I'm not sure on the licensin issue - you can download C source from the 3GPP site
19:21.29Mother*licensing
19:22.00[av]banibleh
19:22.03Abydos313Mother do you have a preference for softphones for windows that use iax? or hardware adapter
19:22.04Motherfile: besides, I don't want it doing translation right now, all it needs to do is act as marshall and gateway, so-to-speak
19:22.17remissAbydos313: idefisk works...
19:22.45Abydos313i have that one but haven't tried it yet. i really like xlite. wish it were iax
19:22.47fileMother: Asterisk acts as B2BUA, ie: two separate call legs... not as a proxy
19:23.40Motherfile: agreed, but it can just simply forward the RTP right? there is no need for it to support the codec, as with G729, where it has to be licensed if you want to translate between it and other codecs
19:24.00fileit needs to know about the codec
19:24.09fileas it doesn't simply forward the RTP to start with
19:24.43Motherhmmmm define *know* in regards of G729 when it's not licensed on a particular asterisk server
19:24.56Qwell[]Mother: the codec has to be defined in *
19:24.58fileinternally in the core Asterisk still knows about G729
19:25.01Qwell[]at the very least
19:25.11fileit knows it exists, it knows it's SDP information
19:25.16MotherQwell[]: agreed
19:25.53MotherOK, that's a starting point - could I then define the extra headers that the Nokia client sends, associated to AMR/8000 for example?
19:26.03fileummm hold
19:26.13filehttp://quark.file-radio.com/asterisk/amr_passthru.diff
19:26.30filethat's for 16000, but you can uh... use it as a base
19:26.45Mothergreat stuff, thanks!
19:27.00Mothernow I need to implement floor control and all the other crap :)
19:29.09Motheroh...and...."convergence"....seemed to be *the* buzzword at 3GSM
19:29.33*** join/#asterisk sch19 (n=sch19@adsl-9-107-161.mia.bellsouth.net)
19:29.35fileAsterisk - Now with SpeedyConvergence(tm) technology!
19:29.51Motherlol you could make millions with that
19:30.28_Sam--<PROTECTED>
19:30.35_Sam--<PROTECTED>
19:30.37*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
19:30.43Motherthere was a stand that all it had was a platform, and half-naked women, and they sold something 'convergent'
19:30.58_Sam--when you were gone on friday, switch-04 decided to not work
19:31.03_Sam--*sigh*
19:31.12file_Sam--: bkw and MikeJ also work for Asterlink
19:31.14fileso you can bug them too
19:31.25_Sam--im done bugging people i just want stuff to work!
19:31.39_Sam--i have 10.4% loss at the last hop before switch-*
19:33.20filebkw_: poke
19:33.39trixterMother file: yes, it's AMR, but I'm not sure on the licensin issue - you can download C source from the 3GPP site
19:33.44trixterAMR is patented and not free
19:33.54trixteranything that uses it (like GSM/AMR) requires a license
19:34.34*** join/#asterisk BladeRunner05 (n=feelme@adsl-111-54.38-151.net24.it)
19:35.38file_Sam--: AIM.
19:36.02*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
19:38.37websaei am curious--is it possible to setup such a network where you have two WAN connections, internally you have an asterisk server  and sip phones as well as machines surfing the internet,---is it possible to hook up one of WAN connections to the asterisk box (with 2 lan cards) so that all sip trafic goes through one connection
19:38.52Qwell[]websae: Sure
19:39.05Qwell[]asterisk can listen on a single interface
19:39.12Qwell[]or, rather, on an IP
19:39.44trixter_Sam--: do you have network problems to other sites?  maybe try www.yahoo.com and www.google.com
19:39.51*** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com)
19:39.52trixterit may be a general network problem and not any one specifically
19:40.01trixterthat should at least tell you if its closer to you or not
19:40.40*** join/#asterisk backblue (n=moo@87-196-12-97.net.novis.pt)
19:41.26Mothertrixter: you surely mean in the US....
19:42.19Motherwe haven't invented software patents yet
19:42.38*** join/#asterisk lazzarello (n=lee@dsl254-077-209.nyc1.dsl.speakeasy.net)
19:43.27trixterjust becuase the implementation is in software doesnt mean the patent is only software
19:43.44websaewhat type of server hardware would one suggest for an asterisk box supporting 30 end users
19:43.45websae?
19:43.57trixterremember you cant really patent an implementation as such,  that is what copyrights are for, you can patent the process indifferent to the implementation
19:44.00Motherer...well...if there was a *chip* doing AMR then yes, but this is hardly the case
19:44.29*** join/#asterisk breakdecks (n=breakdec@adsl-065-006-209-021.sip.mem.bellsouth.net)
19:44.32trixterbut it may not matter depending on things..  if the patent is on a process and the process is implemented in software that doesnt mean its a software patent per se
19:44.38backblueanyone knows any way to get TEI fixed with zaphfc?
19:44.43Motheragreed
19:44.47trixterit might, it depends on a variety of factors
19:45.42Motherbut in the case of codecs, I don't think that applies, at least if it was so no doubt there'd be a flurry of lawsuits in EU in a matter of minutes
19:45.42trixterAMR has to do with voice coding and odds are they wrote it in such a way to cover both hardware as well as software implementations sine its likely to be both
19:45.42breakdecksI am having an issue registering with FWD
19:45.42trixterAMR isnt a codec per se
19:45.52[av]baniilbc 4-evar
19:45.56*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:46.07trixterits a way of doing parts that can be used in a codec.  such as GSM/AMR is the GSM codec but with the AME stuff..  as opposed to the GSM/FE which is free
19:46.25_Sam--is above.net any better than cogent?
19:46.26trixterdoesnt ilbc use as much cpu as g7239 but more bandwidth?
19:46.30websaeany suggestions for what requirements i would need to setup a asterisk machine supporting 30 end users...anyone?
19:46.38*** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net)
19:46.45trixterwebmind: more information would be a start
19:46.51breakdecksI put in my correct register => Number:Password@iax2.fwdnet.net in iax.conf, but my console says it can't log in
19:46.57Motherhttp://www.dsprelated.com/showmessage/17150/1.php
19:47.04Motherthat's some bit of info
19:47.05trixterwhat are those 30 people going to be doing?  how many concurrent calls?  transcoding?  hardware?
19:47.06trixteretc
19:47.33[av]baniaccording to show translations, ilbc takes slightly less cpu than g729
19:47.35[av]banion x86_64
19:47.35austinnichols101sam: you lookin for a provider?
19:47.42*** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca)
19:47.46websaei am wondering what hardware i would need.....it would only transcode for voicemail....and prolly 10-15 concurrent calls
19:47.55_Sam--austinnichols101 :  no , just looking for a better route to some.
19:47.57*** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com)
19:48.05Qwell[]websae: you could get away with about 1ghz
19:48.11_Sam--wondering if above.net is any better than cogent
19:48.15Qwell[]so...any recent machine would do the job nicely, for only 15 calls
19:48.27[av]banig729 uses 8kbps, ilbc uses 13
19:48.34iCEBrkr_Sam--: IT RUNS! IT RUNS!
19:48.43_Sam--[av]bani:  above or cogent?
19:48.47austinnichols101sam: in terms of reliability or performance?
19:48.47websaecogent
19:48.51[av]bani_Sam--: yes
19:48.54_Sam--lol
19:49.01websaecogent---reliability and performance
19:49.04[av]bani_Sam--: um, neither?
19:49.12[av]baniwebsae: reliability? lollerskates.
19:49.14denonavoid cogent ..
19:49.18denonat all costs :)
19:49.21_Sam--if you could get data to a host...and that host was on cogent or above...which way would you go
19:49.22Qwell[]cogent = pissy ex-girlfriend
19:49.56[av]baniboth abovenet and cogent are poo
19:49.56denonI know they're offering you 20Gbps for $9.95/mo ..
19:49.56denonbut its not worth it
19:49.56[av]banithey both are very spam-friendly and get blacklisted often
19:49.56[av]baniand cogent especially is subject to lots of ddos
19:50.09[av]banibecause any script kiddie can buy hosting on there and setup irc server and warez and bots...
19:50.22_Sam--seems whichever way i set up my remote gateway, its an every other day affair...one day its fine, the next it isnt.
19:50.36_Sam--and i am doing nothing but chasing my own tail day after day
19:50.42Mothertrixter: it looks like there are companies developing AMR codecs for licensing, not just one, so I guess the copyright is on the implementation, and not the purpose, so-to-speak
19:51.01MotherIMHO you could develop your own version/implementation, and be free from IP issues
19:51.26austinnichols101IMO, interNap is the network choice
19:51.30*** join/#asterisk MikeJ[Laptop] (n=vircuser@71-36-209-237.dlth.qwest.net)
19:51.51[av]baniaustinnichols101: internap is also very spammer-friendly ("bullet proof hosting")
19:51.57AndyCapMother: copyright and patents are not the same.
19:52.19austinnichols101[av]bani: sounds like you're thinking of Interland.
19:52.30*** join/#asterisk jtodd (n=jtodd@mccpool-2.ci.monterey.ca.us)
19:52.33trixteryou c ant patent the implementation only the process, the implementation would be copyrighted, and in effect you can end up oweing to two people one who holds the copyright and the other the patent
19:52.35austinnichols101[av]bani: InterNap does't put up with spam at all
19:52.54MotherAndyCap: I'm aware of that
19:52.59[av]baniaustinnichols101: no. pnap.net. definitely.
19:53.06[av]baniaustinnichols101: they DO NOT respond to complaints.
19:53.28_Sam--[av]bani:  what backbone does your VOIP traffic take to your remote gw for terminating calls?
19:53.33denonaustinnichols101: what's your experience with internap these days? Ive talked to a few carriers who have dropped them in the couple years because they were "slipping"
19:53.46AndyCapMother: so how do you see people developing an AMR implementation free from patent issues?
19:54.02austinnichols101[av]bani: I've had a completely different experience.  I get excellent response from them.
19:54.05sevarddenon: were you the one who made the weather script using Alison Smith?
19:54.23MotherAndyCap: who holds the patent to start with? I don't think there is a patent on AMR - Ericsson claimed to have one but it was rejected
19:54.30[av]baniaustinnichols101: are you a spammer =)
19:54.42austinnichols101[av]bani: but I've never worked with them as a non-customer...
19:54.51Motherand now 3GPP has standarised it as the mandatory codec for 3G, so I can hardly see *everyone* having to pay royalties to 3GPP for it
19:54.57_Sam--[av]bani :  how sure are you that BLF works fine with GXP2000 / * 1.2.*?
19:55.01[av]baniaustinnichols101: pnap were hosting vibe direct media for ages. (NET-64-94-51-128-1). unrelenting spammers.
19:55.09_Sam--because ive done some more tests and can break it pretty much at will
19:55.09[av]baniaustinnichols101: and pnap took like 8 months before they wree booted
19:55.11AndyCapMother: I guess voiceage claims to hold the licensing rights to the "patent pool"
19:55.37[av]bani_Sam--: i havent used it heavily, but it goes blinky blinky in my tests
19:55.50austinnichols101denon: I get a 100% SLA from internap as they're really just running the seven largest us carriers across their flow-control-platform hardware.
19:55.52_Sam--the extensions you are hinting, are gxps also?
19:55.52iCEBrkrBlinky lights are the essense of technology
19:55.56[av]bani_Sam--: yes
19:55.59AndyCapMother: that may of course be FUD or they have a legitimate patent. :P
19:56.00_Sam--how many?
19:56.04*** join/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com)
19:56.06MotherAndyCap: well, possibly, but I don't think they could get away with it if challenged - people using their codec will pay because they've done the implementation work
19:56.09*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
19:56.17austinnichols101denon: and I've never had a second of downtime in a little over two years as a customer
19:56.32DagMollerhi... is possible to utilize 2 TDM04B in one machine?
19:56.39austinnichols101[av]bani: no, but I get a few hundred a day...
19:56.43[av]bani_Sam--: 5 so far
19:56.46denonaustinnichols101: but with regards to their latency and peering
19:56.46AndyCapMother: and what's the 3gpp's policy on patents in the standards? some allow it and others don't
19:57.03_Sam--there is a note in older firmware about it not working right
19:57.09_Sam--and apparently its still not fixed
19:57.16[av]baniaustinnichols101: its possible > 5 makes it break. shrug..
19:57.24[av]banis/austinnichols101/_Sam--/
19:57.46_Sam--no, i got it to break with 3 or 4
19:57.54_Sam--and the hint extensions werent even gxps
19:58.02_Sam--this is part of it:
19:58.04_Sam--"NOTE: (Nov29/05) When receiving the "''Incoming call: Got SIP response 415 "Unacceptable Content-Type" error, the BLF doesn't seem to work - looks like the phones are refusing the BLF messages all together. - conexim
19:58.05_Sam--"
19:58.36_Sam--i got those all over the console before things really start to break
19:58.40austinnichols101denon: check out interpulse.net and set focus 'from internap / to internap'
19:58.49MotherAndyCap: I see your point, maybe it's true that they have nominated VoiceAge, Nokia and Ericsson as the "administrators" of the patent rights
19:58.50AndyCapMother: of course they're not saying outright what patents you would be paying for. :-P http://www.voiceage.com/amr_faqs.php scum
19:59.05denonaustinnichols101: well, that doesnt really tell us jitter over time
19:59.09Motherhehe indeed
19:59.24austinnichols101denon: I'm getting that now...
19:59.25AndyCapMother: and it seems you only need to license it for "non-wireless" use
19:59.36[av]baniaustinnichols101: http://www.spamhaus.org/sbl/sbl.lasso?query=SBL14726
19:59.48[av]baniaustinnichols101: http://www.spamhaus.org/sbl/listings.lasso?isp=internap.com  23 active listings
19:59.56[av]baniaustinnichols101: including some still active from 2004 ...
19:59.59DagMollerhi... is possible to utilize 2 TDM04B in one machine?
20:00.30[TK]D-FenderDagMoller : yup
20:00.43DagMoller[TK]D-Fender, ???
20:00.54austinnichols101[av]bani: can't speak to the spam issue other than they've been very responsive when we've complained to them
20:00.57AndyCapMother: nah, misinterpreted that. If you want wireless use you have to beg the competition for a license (both ericsson and nokia)
20:01.16[av]baniaustinnichols101: if internap was indeed intolerant of spam, that list would be empty. not stuffed with active listings up to 2 years old
20:01.18austinnichols101[av]bani: but thanks for the link - hadn't seen that one before
20:01.41MotherAndyCap: yep...however a curious point is that it says you need a license if you will use it one product only...how about many products...it's a bit unclear
20:01.45[av]baniaustinnichols101: you might want to ask them why they host and protect so many spammers. they'll pribably ignore you though
20:01.53Mothersorry, not product, application
20:02.05*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
20:02.15DagMoller[TK]D-Fender, yes or no???
20:02.58AndyCapMother: so in essence they sell you a pig in a poke.
20:03.01AndyCap:P
20:03.23trixterbetter a pig in a poke than to poke a pig
20:03.27trixtercause no one likes a pig poker
20:03.35[TK]D-FenderDagMoller : I said yes already....
20:03.43Seldon1975i do
20:03.52DagMoller[TK]D-Fender, no, you say 'yup'
20:04.08[av]baniaustinnichols101: do let me know their response. i'll be interested to know it.
20:04.08MotherAndyCap: LOL yes
20:04.14DagMoller[TK]D-Fender, thanks...
20:04.24Motherwell, gotta get going, cheers all
20:04.39*** part/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com)
20:04.46*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
20:05.06AndyCapI can't believe that the actual patents should be so hard to find
20:06.05PakiPenguin_evening
20:06.18iCEBrkrPakiPenguin_: G'afternoon
20:06.21iCEBrkr:)
20:07.33austinnichols101denon: Jan 2006 Stats: N. America Latency 37.3143ms Packet Loss  0.0359% Network Availability 100% Jitter  0.01164ms
20:08.25austinnichols101denon: N. America-Europe 70.3305ms 0.0256% 100% 0.01206ms for the same time period
20:08.35Hmmhesaysi'm sorry i'm bad, i'm sorry i'm blue, i'm sorry about all the things I said to you
20:08.45lazzarelloI recently heard rumors about ISPs in the states introducing jitter into consumer broadband connections. anyone else hear the same?
20:09.20Hmmhesaysthat would be a good way to piss a lot of people off
20:09.28lazzarellosure would
20:09.38_Sam--it would only piss them off if they knew about it.
20:09.43PakiPenguin_hey iCEBrkr
20:09.46Hmmhesaysjitter while i'm playing quake would piss me off
20:10.03lazzarelloor convince them to "use cablevision's digital phone service" because it's "better".
20:10.06Abydos313why? is regular phones being discarded for voip too quickly ?
20:10.17_Sam--lazzarello:  i heard the same rumors about comcast.net
20:10.29_Sam--and that they messed with voip traffic...i think its just a rumor personally
20:10.37iCEBrkrMangle QoS settings mid-stream.. Man, that'd piss me off
20:11.13_Sam--i wouldnt be terribly surprised if they did things that did interfere with voip traffic to make their own services seem better
20:11.15austinnichols101[av]bani: just sent support an inquiry regarding the spam listing.  Let see what they say...
20:11.18lazzarelloverizon's FTTP service blocks incoming TCP traffic on port 80. they don't advertise /that/ anywhere.
20:11.38_Sam--lazzarello :  yes they do, they specifically tell you you cant run servers
20:11.44_Sam--usually, anythig that runs on port 80 is a server
20:12.03iCEBrkrBrighthouse/RoadRunner doesn't block any inbound ports.
20:12.04_Sam--i just read the FIOS terms of service
20:12.06_Sam--it says right there
20:12.32*** join/#asterisk lalito (n=erg@201.154.202.110)
20:13.10lazzarelloooooh, watch out for the "servers"!
20:13.13_Sam--erisk ("..quit..")
20:13.17_Sam--Verizon FiOS Internet Service consumer packages include 10 MB of personal Web space. The consumer offers do not permit customers to host any type of server, personal or commercial.
20:13.25Seldon1975if the voicemail.conf entry format is: 1234 =>{password},Some User,email@address.com how can I let users change their own passowrd?
20:13.28*** join/#asterisk saftsack (n=saftsack@p54A7E210.dip.t-dialin.net)
20:13.47_Sam--you are the one saying they dont tell you that...
20:13.49lazzarellosweet. so they give you 1.5 megs inbound and block inbound traffic!
20:13.52_Sam--but in fact they do, that is my only point
20:13.57iCEBrkrSeldon1975: phpsuexec + PHP and a voicemail.conf class :P
20:13.59_Sam--you have to pick your argument
20:14.05_Sam--if you are arguing they dont tell, they do.
20:14.11_Sam--if you are complaining that it sucks, that is different
20:14.20lazzarelloI didn't read the fine print.
20:14.27Seldon1975iCEBrkr: no, really
20:14.31PaulHuynhhello
20:14.35austinnichols101[av]bani: the first link you sent me (rokso / datran media) is in the same CoLo facility as we are (http://www.napoftheamericas.net)
20:14.43iCEBrkrSeldon1975: Really.
20:14.49PaulHuynhwhere is the custom voice promt located on the asterisk server?
20:14.59iCEBrkrSeldon1975: Actually, doesn't Asterisk allow users to dial in and change their passwords?
20:15.12Seldon1975iCEBrkr: I hope so
20:15.13iCEBrkrPakiPenguin_: Custom voice prompt??
20:15.15[av]baniaustinnichols101: nice
20:15.15Seldon1975iCEBrkr: but how?
20:15.24iCEBrkrSeldon1975: VoiceMailMain()?
20:15.33PakiPenguin_no
20:15.37Seldon1975users have said they've tried that and the password didnt change
20:15.44Seldon1975iCEBrkr: users have said they've tried that and the password didnt change
20:15.52GerbilWrkanyone know what could cause this to start coming up, Feb 20 14:15:14 NOTICE[1567]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
20:16.01iCEBrkrSeldon1975: File permissions are different from user:group Asterisk is running as?
20:16.02austinnichols101[av]bani: yeah - I wonder where their cabinet is.  I might be able to solve this problem myself :-)
20:16.13iCEBrkrPaulHuynh: Huh? Voice prompt?
20:17.10Seldon1975iCEBrkr: hmm, ill have a look'
20:18.03Seldon1975iCEBrkr: thx
20:19.28*** join/#asterisk breakdecks (n=breakdec@adsl-065-006-209-021.sip.mem.bellsouth.net)
20:19.30[av]baniaustinnichols101: i'd be interested in hearing internap's response. $50 says it's a non-response
20:19.44[av]baniaustinnichols101: eg a response with no content, just doublespeak
20:20.33breakdecksI have a DVG-1120S VoIP gateway, but when I dial digits in a menu, I get the error 'codec_ilbc.c:144 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?'
20:20.36breakdeckswhat do I do?
20:21.16|omni|I think I'm going to write a windows based asterisk manager interface
20:21.27iCEBrkr|omni|: Yea, we all say that
20:21.43austinnichols101[av]bani: $50 not necessary - I'll do it just out of pure curiosity.  And I'll definitely share the results, doublespeak or no.
20:21.53|omni|was just looking at what's available and I've got some things I need handled
20:22.03iCEBrkr|omni|: me too
20:22.14iCEBrkr|omni|: Personally all the desktop call managers out there suck ass
20:22.17|omni|just did a system for a real-estate office and could really do some cool things with the manager interface
20:22.20|omni|yea they do, so far
20:22.40|omni|did you write the callerid popup?
20:22.40iCEBrkr|omni|: One of the many.
20:22.46|omni|ya, just looked at yours a bit ago
20:22.58iCEBrkrVery basic.. Kind of a trial I guess..
20:23.00rayvdI am not a midget!
20:23.12iCEBrkr|omni|: I have bigger/better plans.
20:23.20|omni|cool
20:23.27[av]banibreakdecks: dvg-1120s doesnt support ilbc
20:23.47iCEBrkrMy callerID thing is pretty shitty itself :)  But like I said, it was just a start and of course I wanted to see what kind of interest was out there
20:23.53|omni|I was just messing with quick and dirty commands via a little Delphi app...can do some neat things with the manager interface
20:24.02|omni|ya definitely
20:24.03iCEBrkr|omni|: yup
20:24.34iCEBrkr|omni|: I'm not sure how true it is today (v1.2.x), but there were some scalability issues with the manager port/interface
20:24.35NivexI'm trying to figure out the best way to have astersk query an HTTP server for a call URL and then Dial it.  Any pointers on where to start?
20:24.52iCEBrkrNivex: CURL()?
20:24.56|omni|scalability issues using a proxy or no? or either?
20:25.10iCEBrkr|omni|: Without a proxy and having all the clients connect directly to Asterisk
20:25.13*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)
20:25.35|omni|hmm..should play with that on a test box
20:25.41|omni|would be interesting to see how much better it does with a proxy
20:25.46|omni|if at all
20:25.50|omni|still has to pass out a lot of info
20:26.04iCEBrkr|omni|: I think it'd do better for the machine in general as there's less socket management
20:26.06austinnichols101[av]bani: sending you a copy of the message in a separate window
20:26.14|omni|could be
20:26.19mockerIs there a variable for the dialed number?
20:26.19TagorHmm, I give it another try; I have a problem with incoming calls. Outgoing calls work fine. But for some reason I don't get any signal from my voip provider. When I try tcpdump or sip debug, I just don't see anything when I try to call my number. Anyone an idea how to get more information about why it isn't working?
20:26.32|omni|mocker ${EXTEN}
20:26.44iCEBrkr|omni|: The real key factor in the proxy is mangling the data to/from the client.  So you can do some extra custom stuff.
20:26.45Nivex~iCEBrkr++
20:26.56mocker|omni|: Thanks, I'll try that.
20:27.04iCEBrkrNivex: Think that'll work for ya?
20:27.13*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:27.24iCEBrkrmocker: ${EXTEN} isn't always the dialout number
20:27.33|omni|yea that might be cool... actually with a proxy you could use TCP between proxy <-> asterisk and then make it a bit lighter from the proxy -> client using UDP messages..almost like an IM app
20:27.44|omni|at least for some things.
20:28.03NivexiCEBrkr: should be just what the doctor ordered.
20:28.04iCEBrkr|omni|: Yea, use the proxy to actually process the data, and then spew UDP to the client
20:28.10iCEBrkrNivex: cool stuff
20:28.27mocker|omni|: Hrm, that just says 's'.  What I'm doing is dialing out w/ the drop folder, and I'm using a system command to send an email of what number was dialed.
20:28.41iCEBrkrmocker: You're using a call file?
20:28.48Nivexbasically I'm going to write a script on the server that will return either a call URL or UNAVAIL and then the Asterisk dialplan can react accordingly
20:28.51iCEBrkrmocker: Then most likely you won't be able to get the dialed number.
20:29.15|omni|are you using an AGI to create the call file ?
20:29.31|omni|you could pass the ${EXTEN} to your AGI
20:29.43mockeriCEBrkr: Yeah, I'm using a call file.
20:29.47*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
20:29.49iCEBrkr|omni|: ${EXTEN} doesn't mean the dialed number
20:29.57harryvvmoker, what are using the call file for
20:30.10NivexiCEBrkr: I'm reading the func_curl page on the wiki, but I don't see anything about handling error conditions.
20:30.12|omni|within your extensions it will return the dialed number, no?
20:30.23mockerharryvv: Basically call someone, when they press 2 to acknowledge, send an email saying they acknowledged.
20:30.24iCEBrkr...and since ${DIALEDPEERNUMBER} is broken along with ${DNIS}
20:30.42mockerBut trying to get who it called into the email is my roadblock. ;)
20:30.45harryvvsending a email from whom? the called party?
20:30.48iCEBrkrmocker: Your best bet is to: SetVar: _dialednumber='123-123-1234' in your callfile..
20:31.02mockeriCEBrkr: Oh, well that would solve everything.
20:31.06iCEBrkr|omni|: It returns the dialed 'extension' which isn't always the number you've called.
20:31.10mockerI didn't know I could set variables like that. :)
20:31.18|omni|okay well..fair enough
20:31.29|omni|I guess it returns in the dialed number in my implementation
20:32.14iCEBrkr|omni|: if NXXNXXXXXXX is your extension, sure..
20:32.30|omni|it will be for certain contexts
20:32.34iCEBrkr|omni|: if you land in 's' then ${EXTEN} is going to equal 's'
20:32.35|omni|and I can grab and stuff into a var at that point
20:32.54|omni|I see what you're saying
20:32.57iCEBrkrmocker: I'm having the same problems now, but I'm too far along in the project to start making changes.
20:33.05*** join/#asterisk Skid (i=cm@unaffiliated/skid)
20:33.11*** join/#asterisk fjean (n=fjean@201.29.122.10)
20:34.03fjeanhey guys, how are you..
20:34.06|omni|in most contexts I match the incoming (or outgoing) exten with the full number just so I know what's going on..at least the first time it hits before jumping to another context
20:34.31iCEBrkr|omni|: then ${EXTEN} is gonna change throughout the dialplan :)
20:34.53harryvvmocker, how many calls do u want your astrisk box to make at one time/
20:34.54harryvv?
20:35.00|omni|right, but it will be the dialed number the first time it's grabbed in any matching context
20:35.08fjean-- can someone help me with an AGI script that dials a string ?
20:35.15iCEBrkr|omni|: sure, depending on your pattern matching and what-not
20:35.22|omni|guess it really depends on how you organize everything.. mine's pretty simple.
20:35.37mockeriCEBrkr: That worked like a charm.
20:35.51mockeriCEBrkr: Any way to get it to not have the ' ' around the variable output?
20:36.01iCEBrkrmocker: Then you just NoOp(${dialednumber})?
20:36.04mockerOh, that's in what I'm setting.
20:36.29iCEBrkrYeah, it's habit to quote values..
20:36.45*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:38.14*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:38.42*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
20:38.43fjeani have this in an AGI script:  $dialstr = "IAX2/okscm/559132316200|60";  which does not give me ring back, but this line will give it to me:  exten => _0XX32316200,1,Dial(IAX2/okscm/559132316200,60);
20:39.02fjean-- anybody knows why i would get it in the AGI ?
20:39.04asterisk99anyone here have Asterisk running on UBUNTU?
20:40.12asterisk99iefbr14
20:40.35lazzarelloasterisk99, what's your question?
20:41.17asterisk99lazzarello:   Did you have any problems with the zaptel modules on ubuntu?
20:41.30PaulHuynhhelp
20:41.34PaulHuynhi got this error
20:41.35PaulHuynhNo such file or directory in /var/www/html/admin/ivr.php on line 112
20:41.53lazzarelloI don't run ubuntu but that's should make much difference. did you compile yourself?
20:42.03*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
20:42.11asterisk99lazzarello:   I did the make clean; make linux26; make install .... but when I try modprobing 'em, I get errors
20:42.18*** join/#asterisk Inkubot (n=inkubot@200.74.186.41)
20:42.22Inkubothi
20:42.28lazzarelloasterisk99, what kind of errors?
20:42.50Inkuboti've got a Asterisk Server behind NAT, working with sip.. i can't register from the internet.. i've got something like this
20:43.02InkubotFeb 20 16:01:55 DEBUG[2090] chan_sip.c: Target address 200.74.186.41 is not local, substituting externip
20:43.03*** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
20:43.19Inkuboti set externip and localnet y sip.conf
20:43.22Inkubotalso nat=yes
20:43.49Inkubotany idea on what could be the problem ?
20:43.52Hmmhesaysyeah good luck with that
20:43.56Seldon1975when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist?
20:44.07lazzarellothe fact that SIP behind NAT is /HELL/ could be the problem.
20:44.20Inkubothaha
20:44.23Inkubotdamn
20:44.35lazzarelloInkubot, install ngrep and start capturing headers for a ingle faild session. read the headers and figure out your problem.
20:45.22dudesasterisk99 - did you modprobe zaptel first? if not, try that, otherwise, create "misc" in /lib/modules/(your kernel)/
20:45.31austinnichols101lnkubot: what do you have set on the remote phone?
20:45.38dudesthen goto zaptel ... make clean && make linux26 && make install
20:45.40asterisk99lazzarello:   modprove wctdm gets me FATAL ERROR: module wctdm not found
20:45.59Inkubotthe ip address of the Router->Asterisk Server
20:46.02lazzarellohe he. I guess the module isn't found. what's modinfo wctdm output?
20:46.04Inkubotpublic ip address
20:46.26dudesDebian/Ubuntu do this something (something to do with Udev)
20:46.27austinnichols101lnkubot: is the remote side behind nat too?
20:46.41asterisk99lazzarello:   modprobe zaptel ... same thing
20:46.43lazzarellodudes, Debian and ubuntu are not the same thing.
20:46.54Inkubotaustinnichols101 yes.. i forget that
20:47.00*** part/#asterisk PaulHuynh (n=paulhuyn@c-68-37-18-82.hsd1.de.comcast.net)
20:47.06asterisk99lazzarello:   isn't ubunu a derivative of debian?
20:47.10lazzarelloasterisk99, I'm running Debian stable. when I run 'modinfo wctdm' it outputs this:
20:47.10Inkubotlet me see if NAT is activated in the client configuration
20:47.11Seldon1975when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist?
20:47.12lazzarellolummox:~# modinfo wctdm
20:47.12lazzarellofilename:       /lib/modules/2.6.8-2-386/extra/wctdm.ko
20:47.26lazzarelloasterisk99, what does your system say?
20:47.28asterisk99lazzarello:   1 sec
20:47.31austinnichols101lnkubot: can you move the remote phone into the DMZ on the remote router for testing?
20:47.37dudeslazzarello - DUDE, I use Debian and Ubuntu and get this issue every once in awhile
20:47.39lazzarelloasterisk99, don't paste the whole thing.
20:47.50*** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net)
20:47.58asterisk99lazzarello:   not found
20:48.01lazzarellodudes, NOT THE SAME! you run testing or unstable then.
20:48.02justinud00d
20:48.17lazzarelloasterisk99, then you didn't install the modules for your running kernel. what's in /lib/modules?
20:48.17Inkubotaustinnichols101 i can't.. i'm in another place..
20:48.21*** join/#asterisk areski (n=areski@162.Red-83-34-10.dynamicIP.rima-tde.net)
20:48.39dudeslazzarello - I never said they were, they same ... They still are similiar though.
20:48.39austinnichols101lnkubot: what type of device are you trying to register?
20:48.47lazzarelloasterisk99, does ubuntu have a package for all the asterisk 1.2 stuff.
20:48.48asterisk99lazzarello: (perosnally, I prefer gentoo, but it doesn't like my NIC)
20:48.49Inkubotyeaaaaaaaa
20:48.53Inkubotaustinnichols101
20:48.59Inkuboti forgot nat=yes in the clients :D
20:49.03dudesasterisk99 - I told you how to fix it
20:49.05asterisk99lazzarello: no package
20:49.14lazzarellois there a backport of some kind?
20:49.19lazzarellocause debian stable has a package for 1.2.1
20:49.39dudesIt is Rocket Science to fix ... which I have already mentioned
20:49.46lazzarelloin fact, I just finished building the zaptel-modules for the 386 and 686 kernel in sarge.
20:50.08austinnichols101lnkubot: now let the clients idle for a couple of minutes and check 'sip show peers' on the server and verify that they maintain the registration (I'm assuming that you have qualify = yes on the server)
20:50.11asterisk99lazzarello: I wonder if I should go with debian
20:50.16lazzarelloI wonder...
20:50.26asterisk99lazzarello: I ain;t married to ubuntu
20:50.26lazzarellooh yeah, and the base system is only 140 megs
20:50.33lazzarellothat's pretty cool too
20:50.36dudesI've had this issue on Stable, testing, Unstable, ubuntu, and kubuntu ...
20:50.41lazzarelloubuntu is like, 1.5 gigs, right?
20:50.51asterisk99dudes: I will look in messages above for your fix
20:51.08lazzarelloasterisk99, you didn't tell me what's in /lib/modules
20:51.10asterisk99lazzarello: You can install a 'server' version of ubuntu
20:51.29asterisk99lazzarello: Hang on :) I have 1 monitor on 2 machines
20:51.36asterisk99lazzarello: Cheeeeeep!
20:52.20asterisk99lazzarello: /lib/modules  has 2.6.12-10-386 directory
20:52.27asterisk99lazzarello: plus a few more
20:52.32lazzarellookay! what's the few more?
20:52.47Seldon1975when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist?
20:52.51*** join/#asterisk Inkubot (n=inkubot@200.74.183.55)
20:52.55Inkuboti kick the adsl..
20:52.56Inkubot:D
20:53.01Inkubotaustinnichols101 everything works now
20:53.18lunaphytewhy might asterisk segfault when a skinny phone answers a call from a sip phone?
20:53.29asterisk99lazzarello: 2.6.12 , 2.6.12-10-386 , 2.6.12-10-686 , and 2.6.12-9-386
20:53.31austinnichols101lnkubot: party on!
20:53.51lazzarelloasterisk99, now what does the output of 'uname -a' say?
20:55.10asterisk99lazzarello: Linux mymachine 2.6.12-10-386 # date time ...
20:55.28*** join/#asterisk pengyong (n=lala@218.19.188.13)
20:55.48asterisk99lazzarello: mymachine is my name of the kernel
20:55.51dudesmkdir /lib/modules/`uname -r`/misc && cd /usr/src/zaptel && make clean && make linux26 && make install
20:55.58lazzarelloasterisk99, now... 'find /lib/modules -name 'wctdm.ko''
20:56.21lazzarelloget rid of the first set of quotes.
20:56.29*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
20:56.38hensemahi, is "spawn extension (...) exited non-zero on 'channel name'" a problem, or do all dialplans end that way?
20:57.17dudesonce complete ... cp *.ko /lib/modules/`uname -r`/misc && make clean && make linux26 && make install && modprobe zaptel && modprobe wctdm
20:57.34lazzarellodudes, that's what the makefile is for
20:57.39fjeanis there a difference between a Dial from extensions.conf and a Dial from an AGI, should we do answer() before ?
20:57.44asterisk99lazzarello: /lib/modules/2.6.12/misc/wctdm.ko
20:58.01Qwell[]asterisk99: debian?
20:58.08lazzarelloasterisk99, there you go. that's your problem. your running kernel is different then where the modules were installed.
20:58.10dudeslazzarello - I think I know how to resolve this issue, but since you're the smart guy
20:58.15lazzarelloreboot into the 2.6.12 kernel.
20:58.17asterisk99lazzarello: aye carumba
20:58.21Qwell[]no, no, no
20:58.34Qwell[]You just need to tell the Makefile where the modules should go
20:58.39asterisk99lazzarello: I didn't compile the kernel
20:58.56lazzarellodudes, it's not resolved if you have to type fancy shell scripts. the installer package should do that automatically or it's a bug.
20:59.13lazzarelloasterisk99, then why do you have a module directory for 2.6.12?
20:59.21dudeslazzarello - it's once they haven't fixed in a very long time
20:59.28Qwell[]KVERS=`uname -r` make install
20:59.32Qwell[]simple as that..
20:59.32asterisk99lazzarello: I ast-install'ed the kernel-source
20:59.47lazzarellothen reboot into that kernel and your problem's solved.
20:59.49dudesasterisk99 - do as I said and see the magic
20:59.51Qwell[]Debian defs KVERS
20:59.57Qwell[]lazzarello: That kernel doesn't exist
21:00.14Qwell[]My solution will work, or your money back - guaranteed. ;]
21:00.19asterisk99lazzarello: I think it's because the kernel comes precompiled
21:00.29asterisk99lazzarello: So the trick is to get the right source
21:00.34Qwell[]ho hum
21:00.34lazzarelloI don't compile kernels either. something really broken with ubuntu and zaptel...
21:00.39Qwell[]I talk, but nobody listens...
21:00.50Qwell[]ifndef KVERS
21:00.50Qwell[]KVERS:=$(shell uname -r)
21:00.50Qwell[]endif
21:00.56asterisk99Qwell[]: sorry ... Too many messages ... let me catch up
21:00.57Qwell[]Like I said...heh
21:01.03Qwell[]Debian does funkiness with KVERS
21:01.08lazzarelloif there's a bug somewhere it should be fixed, not just hacked around with some shell scripting via IRC.
21:01.13dudesQwell[] - I said my solution ... But since this guy doesn't seem to want to listen to me
21:01.24Qwell[]lazzarello: Go talk to the debian guys about it.  It's their problem
21:01.35lazzarelloQwell, I just installed zaptel on debian stable and didn't have that problem.
21:01.37*** join/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl)
21:01.44lazzarelloso maybe not?
21:01.46asterisk99Qwell[]: do I put KVERS:=$(shell uname -r) in Make?
21:01.51*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
21:01.51asterisk99Qwell[]: do I put KVERS:=$(shell uname -r) in Makefile?
21:01.52Qwell[]asterisk99: no
21:01.58Qwell[]KVERS=`uname -r` make install
21:02.03asterisk99Qwell[]: that
21:02.09asterisk99that's a command???
21:02.13dudesif you ran the two commands above it'd resolve it too
21:02.13Qwell[]debian won't def KVERS if it's already def'd - just as zaptel won't
21:02.23Qwell[]so, if you explicitly def it...neither will chang eit
21:02.40shmaltzanybody else having trouble with the list? or is it G again?
21:03.05Qwell[]dudes: Your way will work, but it isn't the "proper" way.  The stuff would be compiled against 2.6.x, and not 2.6.x-blah
21:03.22asterisk99Qwell[]: bash hates KVERS
21:03.31Qwell[]asterisk99: type it exactly as I did
21:03.32asterisk99Qwell[]: command not found
21:03.33lazzarelloQwell[], or you can download zaptel-source and use module-assistant.
21:03.35Qwell[]KVERS=`uname -r` make install
21:03.41lazzarellobut I don't know if ubuntu has modass
21:03.45*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
21:03.56asterisk99Qwell[]: aha!!! no spaces!!!!
21:04.05dudesQwell - Debian makes a symlink from the sources to the dir anyway, so it really doesn't matter
21:04.15Qwell[]dudes: yeah, but the modules
21:04.21Qwell[]it's just...funky how they do it
21:04.38Qwell[]hensema: All the time
21:04.42Seldon1975when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist?
21:04.48hensemaQwell[]: is it a problem?
21:04.55Qwell[]Seldon1975: allow writes to voicemail.conf
21:05.02Qwell[]I believe there is an option at the top
21:05.14Qwell[]hensema: don't think so
21:05.16asterisk99Qwell[]: after KVERS=   shouldn't I do a make clean; make linux26; make  install ???
21:05.18*** join/#asterisk Speeder (n=root@a81-84-241-120.cpe.netcabo.pt)
21:05.25Qwell[]asterisk99: no need
21:05.38Qwell[]you can do a make clean first, if you'd like though
21:05.39asterisk99Qwell[]: just make install ?
21:05.42Qwell[]linux26 is detected
21:05.50Qwell[]no...type it EXACTLY as I did...one line
21:05.56lazzarelloasterisk99, what does apt-cache search module-assistant output?
21:06.28SpeederHi. I'm having a problem with asterisk & zaphfc. For outgoing call's it appears can't create channel zap, for incomming call's i'm getting busy tone
21:06.32asterisk99lazzarello: 1 min please
21:07.40asterisk99Qwell[]: still modules not found
21:07.57Qwell[]pastebin the output
21:08.02Qwell[]the full output - including the command you typed
21:08.21Qwell[]you probably have to mkdir the misc dir.
21:09.01asterisk99lazzarello: apt-cache search module-assistant       no output
21:09.04Seldon1975Qwell: thx
21:09.21*** join/#asterisk NotFreak (n=extmail@cp12193-e.tilbu1.nb.home.nl)
21:09.26asterisk99Qwell[]: paste from what to what?
21:09.36asterisk99Qwell[]: the whole make output?
21:09.42Qwell[]yes
21:09.48lazzarelloasterisk99, switch to sarge. all this stuff is apready in packages and installable through apt.
21:10.02Qwell[]I'll actually save you the trouble though
21:10.13Qwell[]mkdir /lib/modules/`uname -r`/misc
21:10.13asterisk99Qwell[]: sarge is better?
21:10.20[av]baniahahaha. just got pictures of a customer computer
21:10.26[av]banithey're a cigar smoker.
21:10.28Qwell[]do that, then do the make install again (with the KVERS)
21:10.32[av]baniit is  N A S T Y
21:10.52Qwell[]there is more funkiness with install -D that I've seen
21:12.17NotFreakis there some one around here that has some knowledge about setting and reading registers on the X100P card?
21:13.03Seldon1975Qwell: it doesn't look as if there is a setting in Voicemail.conf to allow writes - do you know where you saw such a thing?
21:13.22[av]baniew
21:13.25asterisk99Qwell[]: is that mkdir work? it gives me an error .... it thinks uname -r is part of the name
21:13.31[av]bani_Sam--: OMG ILLEGAL
21:13.33[av]bani_Sam--: :)
21:13.40_Sam--lol
21:13.43Qwell[]no, I don't recall...I'm probably thinking extensions.conf
21:13.48Qwell[]asterisk99: with the backticks?
21:13.48Qwell[]`
21:13.50_Sam--living close to the canadia border must have its benefits
21:13.56_Sam--s/canadia/canadian/
21:13.57iCEBrkr_Sam--: Man, did you hear me?! It runs! It Runs!
21:14.00iCEBrkr:)
21:14.05Seldon1975Qwell: I think so
21:14.06[av]bani_Sam--: there's probably an entire fidel castro inside this PC
21:14.12_Sam--lol!
21:14.18_Sam--iCEBrkr :  did you ride the thing?
21:14.20Qwell[]Seldon1975: It should "jsut work".  Do you get any errors/warnings?
21:14.24iCEBrkr_Sam--: Yea, Saturday night
21:14.55asterisk99Qwell[]: no have backtick on tis ^%$^% keyboard
21:15.01Qwell[]umm
21:15.05Qwell[]okay, do this
21:15.06_Sam--iCEBrkr :  so what are you buying next for it? :)
21:15.15Qwell[]mkdir /lib/modules/$(uname -r)/misc
21:15.23asterisk99Qwell[]: how about I just do uname -r and paste result in the dir name
21:15.27iCEBrkr_Sam--: Clutch does this weird thumping thing when it's in neutral and I have it all the way out.. I think it just needs some adjusting.
21:15.30Qwell[]that would work too, yeah
21:15.35Qwell[]the second command should also work though
21:15.42iCEBrkr_Sam--: Light smoked windscreen :P
21:15.59filehow personal
21:16.20[av]bani_Sam--: http://bani.anime.net/cigar_smoker/
21:16.21Qwell[]too much rockstar...heart is racing...ugh
21:16.54_Sam--what, ALL your pc's dont look like that?
21:17.07asterisk99Qwell[]: Still didn;t work
21:17.14shmaltzInteresting:
21:17.16shmaltzhttp://news.yahoo.com/s/cmp/20060218/tc_cmp/180203910;_ylt=AgRQNsJD3E9GRhi7Elz.3pCor7oF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA--
21:17.21Qwell[]asterisk99: if backtick doesn't work on that keyboard...how are you setting KVERS?
21:17.33Qwell[]that also needs backticks
21:17.38|omni|heh..that's awesome... I've been without a bike forever now
21:17.38_Sam--[av]bani bow old is that pc?
21:18.14Seldon1975Qwell: no errors or warnings; as I say the password change does work,.. but only until Asterisk is restarted
21:18.17[av]bani_Sam--: dunno, a year maybe
21:18.29[av]baniits on a table across the room and i can smell it from here
21:18.31Qwell[]Seldon1975: Yeah, it should be saving...  Does * run as root?
21:18.41_Sam--i had one that looked like that after about 5 years of being in an office with 6 cigarrette smokers
21:18.53asterisk99Qwell[]: nopers
21:19.04[av]baniif thats what ends up in a pc, think of what goes in your lungs
21:19.05Seldon1975Qwell: yyes
21:19.08Qwell[]asterisk99: see above
21:19.29_Sam--my lungs have other things to worry about :)
21:19.32asterisk99Qwell[]: I used a regular single apostophie   '
21:19.37Qwell[]backticks :P
21:19.44*** part/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de)
21:19.45Qwell[]I said *exactly* as I typed it, heh
21:20.05asterisk99Qwell[]: OK - so I can;t find backtick on this keyboard
21:20.10Qwell[]replace the backticks with the output of uname -r, or $(uname -r)
21:20.52iCEBrkrasterisk99: backtick is to the left of the number 1 key. below ESC and above TAB.. Well, typically, anyhow
21:21.06Qwell[]iCEBrkr: Silly foreign keyboards. ;)
21:21.08iCEBrkrasterisk99: :D
21:21.16iCEBrkroh well, there IS that problem :)
21:21.57asterisk99Qwell[]: I FOUND it !!!!!!   - thanks iCEBrkr
21:22.21dpolitech```!
21:22.23*** join/#asterisk fjean (n=fjean@201.29.122.10)
21:22.26*** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com)
21:22.32Qwell[]asterisk99: So, it's working then?
21:23.24Inkuboti need an outbound proxy
21:23.26Inkubotfree..
21:23.46Inkubotmaybe you know some ?
21:24.13Qwell[]Inkubot: usually, the outbound proxy is the box you want to call through.  Is that not the case?
21:24.14asterisk99Qwell[]: somthing differed ... error running install prgram for wctdm
21:24.23Qwell[]asterisk99: there you go
21:24.28Qwell[]now you just need to fix your configs
21:24.35asterisk99Qwell[]: wctdm module not found
21:24.41*** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net)
21:24.50Qwell[]or not
21:25.05Qwell[]asterisk99: ls -l /lib/modules/`uname -r`/misc
21:25.10Qwell[]Do you see the zap modules in there now?
21:25.18iCEBrkr:)
21:25.21FLeiXiuSI'm having a problem with dialing my extensions, It's not correctly dialing my extensions at all.  For example, this is the line I have in my dial plane: "exten => _100x,1,Dial(SCCP/100x,15,tT)"
21:25.39Qwell[]FLeiXiuS: X.  I believe case matters
21:25.44iCEBrkrFLeiXiuS: That's because.. what Qwell[] said.
21:25.47MavvieFLeiXiuS: use ${EXTENSION} instead of 100x in teh Dial string
21:25.47Qwell[]sorry, not X.  just X
21:26.07Qwell[]Mavvie: ${EXTEN}
21:26.08puzzledFLeiXiuS: exten => _100x,1,Dial(SCCP/${EXTEN},15,tT)
21:26.22iCEBrkrand there is the who SCCP/100x part that's the problem.. puzzled gave you the correct formatting
21:26.37puzzledexcept I missed the lowercase "x"
21:26.43puzzledFLeiXiuS: exten => _100X,1,Dial(SCCP/${EXTEN},15,tT)
21:26.43asterisk99Qwell[]: nothing in that directory
21:26.52MavvieQwell[]: true. was too enthousiast :-)
21:26.57Qwell[]asterisk99: shame.  pastebin the output of te make install
21:27.02Qwell[]including the command you typed
21:27.24asterisk99ok
21:27.36FLeiXiuSAhh, I was wondering what the variable was, thankyou puzzled Qwell[] iCEBrkr
21:27.40asterisk99Qwell[]: ok - now I have to figure oute pastebin
21:28.19puzzledFLeiXiuS: read the document README.variables that's included in the source. Explains all the variables
21:28.28Qwell[]pastebin.com
21:28.34Qwell[]asterisk99: paste the stuff there, then give us the link
21:28.42FLeiXiuSpuzzled: Will do, I'm also looking on voip-info for a wiki page, I'm sure it's on there some where!
21:29.15puzzledFLeiXiuS: yeah sometimes stuff is a bit difficult to find. Just use the search function
21:30.44NotFreakhey guys, i'm doing some hardware hacking on X100P so it can support other CallerID standards but i need some help on reading and setting the registers of the X100P chip that is used, tiger320/md3200
21:31.00iCEBrkrNotFreak: Wrong place :P
21:31.02Qwell[]NotFreak: That's a little out of scope of this channel
21:31.11iCEBrkrNotFreak: But I wish you luck
21:31.24NotFreakok
21:31.28NotFreakthanks for the info
21:32.01NotFreakthis is more about the general configuration of asterisk ?
21:32.03asterisk99Qwell[]: http://pastebin.ca/42329
21:32.16Qwell[]NotFreak: Pretty much
21:32.19NotFreakok
21:32.31Qwell[]grr :p
21:32.33NotFreakso i better should try the mailing list i presume?
21:32.34Qwell[]asterisk99: ONE LINE
21:32.36alephcomDoes anybody see a problem with this as a timeout?   Invalid timeout specified: 'L(100000:60000:30000)'
21:32.41Qwell[]KVERS=`uname -r` make install
21:32.44Qwell[]EXACTLY like that
21:32.57asterisk99Qwell[]: 1 min
21:33.04Qwell[]If it was supposed to be two lines, I would have typed it on two lines. :)
21:33.41iCEBrkrha
21:35.01iCEBrkralephcom: That's at the end of your Dial() statement?
21:36.05asterisk99Qwell[]: http://pastebin.ca/42234
21:36.24Qwell[]eh?
21:37.05Qwell[]You don't need to (re)compile your kernel or anything...just type what I said
21:37.34asterisk99Qwell[]: I think I did that ... one one line
21:37.39Qwell[]no, you did two lines
21:37.45Qwell[]according to your pastebin
21:37.51asterisk99Qwell[]: I'll do it again
21:39.17*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
21:39.43asterisk99Qwell[]: http://pastebin.ca/42239
21:39.56alephcomiCEBrkr: yes, that's at the end of it.  I can post the line if you want.
21:39.58Qwell[]umm
21:40.13iCEBrkralephcom: Post just that line..
21:40.15Qwell[]I have absolutely no clue what that says
21:40.51Seldon1975Qwell: I have set the file permission on Voicemail.conf to -rw-rw-rw but it still doesn't get edited by comedian mail
21:41.06alephcom<PROTECTED>
21:41.13Qwell[]asterisk99: Was that the right link?
21:41.37Qwell[]there we go
21:41.46asterisk99Qwell[]: http://pastebin.ca/42339    (sorry :( )
21:41.57Qwell[]much better
21:42.22iCEBrkralephcom: I meant the line from the dialplan, not console :P
21:42.26asterisk99Qwell[]: I should get IRC running on ubunto so I don;t have to keep switching
21:43.00Abydos313apt-get install xchat2
21:43.06*** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
21:43.11Qwell[]asterisk99: What does uname -r say?
21:43.18Qwell[]2.6.12-10-386?
21:43.29alephcomSorry, I'll try again.
21:43.53asterisk99Qwell[]: yes
21:44.04Qwell[]and what does it say when you modprobe wctdm?
21:44.19Qwell[]and/or ls -l /lib/modules/`uname -r`/misc/
21:44.49asterisk99Qwell[]: modprobe says module wctdm not found
21:45.03*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
21:45.36asterisk99Qwell[]: I do the modprobe in /usr/src/zaptel-1.2  right?
21:45.45alephcomiCEBrkr: Here it is:  exten => _X.,1,Dial(SIP/17808473033@atlasvoice|L(100000:60000:30000))
21:45.50Qwell[]anywhere.  it doesn't matter
21:45.58Qwell[]you're just typing "modprobe wctdm" ?
21:46.04asterisk99Qwell[]: yes
21:46.09Qwell[]no .ko or anything?
21:46.25asterisk99Qwell[]: nope --- look at bootom of pastebin
21:46.51Qwell[]you're typing "modprobe wcfxs"
21:47.09Qwell[]That file was removed
21:47.14filealephcom: you're putting options into the place where you put in the call timeout
21:47.50iCEBrkralephcom: Yea, you're missing a parameter
21:48.23Qwell[]asterisk99: Honestly, if you aren't going to type EXACTLY what I say...please don't ask me for help...
21:48.26alephcomPlease tell me which one???  :-)  This used to work and I'm almost banging my head against a wall. :-(
21:48.33fileadd another |
21:48.39asterisk99Qwell[]: what did I do wrong???
21:48.41filebeside the existing |
21:48.45Qwell[]wcfxs != wctdm
21:48.50alephcomI see, I see, tks.
21:48.53Qwell[]I asked if you were modprobe'ing wctdm, and you said you were.  You aren't
21:49.31alephcomyou just make my day!!!  Tks
21:49.37alephcomor maybe saved my sanity
21:49.47asterisk99Qwell[]: modprobe wcfxs gives me module wctdm not found
21:49.47alephcomIf I had any to start with
21:49.47Qwell[]alephcom: Sanity?  Can I borrow some?
21:49.56Qwell[]asterisk99: Then don't type "modprobe wcfxs"
21:50.00Qwell[]type what I said
21:50.09asterisk99Qwell[]: gotcha
21:50.20fileI need a weapon...
21:50.23filesomething sharp
21:50.55dpolitechbroken broom stick handle?
21:50.55*** join/#asterisk gambolputty (n=root@64.74.225.135)
21:51.06generalhanCan anyone tell me where i define my offset on my ntpd server? all my phones think we are EST, when we are PST right now.
21:51.15gambolputtyhi
21:51.17generalhanthen i have to figure out how to turn it off, so that DST doesnt mess us up
21:51.35lazzarelloasterisk99, install debian stable and use the zaptel-source package. you think less, you make less mistakes, the hard part is taken care of.
21:51.57gambolputtyI have an analog card that won't recognize caller ID.  I wait 2 seconds, then answer, and no caller ID result.  Any ideas?
21:52.00Qwell[]lazzarello: It still won't work, if he doesn't modprobe the right driver
21:52.24lazzarelloQwell[],  a reboot loads the driver automatically.
21:54.01asterisk99Qwell[]: I'm going to switch IRC on the * machine... that way I can cut and paste
21:54.01Qwell[]same thing if you compile from source and use the init script
21:54.15_Sam--Qwell:  when did you become so patient
21:54.29Qwell[]_Sam--: I had a large energy drink this morning
21:54.34asterisk99I'll be bacaaack
21:54.58_Sam--hmm maybe i need to try drinking them at my work
21:55.05_Sam--patience is not a virtue i currently possess
21:55.43Qwell[]yeah...he's never gonna get it
21:56.16_Sam--you could just have him download tzafrir's debian zaptel packages
21:56.18_Sam--they work fine
21:56.29_Sam--i dont know what kernel that guy was on
21:56.37_Sam--<didnt pay that much attn>
21:57.43Qwell[]brb
21:57.51*** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca)
21:57.56Qwell[]or not
21:57.58_Sam--lo
21:58.14asterisk99Quell[]: I'm back now on 1 machine
21:58.16_Sam--this place has tzafrir's modules:
21:58.16_Sam--http://rapid.dotsrc.org/unstable/
21:58.31Qwell[]asterisk99: good, now type modprobe wctdm
21:58.33Qwell[]and nothing else
21:58.33_Sam--simple as downloading then 'dpkg -i somethingorother-zap.deb
21:59.32_Sam--plus if it doesnt work, then you can make tzafrir do the support ;)
21:59.38asterisk99Quell[]: root@konnex:/usr/src/zaptel-1.2# modprobe wctdm
21:59.38asterisk99FATAL: Module wctdm not found.
21:59.38asterisk99<PROTECTED>
21:59.48Qwell[]better
21:59.57asterisk99Quell[]: :)
22:00.05Qwell[]now ls -l /lib/modules/`uname -r`/
22:00.17Qwell[]if stuff isn't there...something is broken.  I saw the make install put them there
22:00.30lazzarellosheesh, there's even packages on backports.org for a 1.2.1 zaptel stuff.
22:01.06Qwell[]erm
22:01.09Qwell[]now ls -l /lib/modules/`uname -r`/misc/
22:01.12Qwell[]that one
22:01.51asterisk99Quell: It's big --- u want me to pastebin that?
22:01.58Qwell[]are there files there?
22:02.04Qwell[]wctdm.ko in particular
22:02.26Qwell[]hurry up, I need to pee :p
22:02.35asterisk99Quell[]: root@konnex:/usr/src/zaptel-1.2# ls -l /lib/modules/`uname -r`/
22:02.35asterisk99total 1268
22:02.35asterisk99lrwxrwxrwx   1 root root     36 2006-02-18 13:32 build -> /usr/src/linux-headers-2.6.12-10-386
22:02.35asterisk99drwxr-xr-x   2 root root   4096 2006-02-17 12:27 initrd
22:02.35asterisk99drwxr-xr-x  11 root root   4096 2006-02-17 12:26 kernel
22:02.36asterisk99drwxr-xr-x   2 root root   4096 2006-02-17 12:27 madwifi
22:02.38asterisk99drwxr-xr-x   2 root root   4096 2006-02-20 16:15 misc
22:02.40asterisk99-rw-r--r--   1 root root 244258 2006-02-20 16:38 modules.alias
22:02.43FLeiXiuSGRRR
22:02.44asterisk99-rw-r--r--   1 root root     69 2006-02-20 16:38 modules.ccwmap
22:02.45FLeiXiuSpastebin.
22:02.46asterisk99-rw-r--r--   1 root root 298510 2006-02-20 16:38 modules.dep
22:02.48asterisk99-rw-r--r--   1 root root    813 2006-02-20 16:38 modules.ieee1394map
22:02.50asterisk99-rw-r--r--   1 root root   1141 2006-02-20 16:38 modules.inputmap
22:02.51Qwell[]meh
22:02.52FLeiXiuSWOW..
22:02.52asterisk99-rw-r--r--   1 root root  21256 2006-02-20 16:38 modules.isapnpmap
22:02.53FLeiXiuSlmao
22:02.54asterisk99-rw-r--r--   1 root root 226143 2006-02-20 16:38 modules.pcimap
22:02.56asterisk99-rw-r--r--   1 root root   1135 2006-02-20 16:38 modules.seriomap
22:02.58asterisk99-rw-r--r--   1 root root 123227 2006-02-20 16:38 modules.symbols
22:03.00asterisk99-rw-r--r--   1 root root 315491 2006-02-20 16:38 modules.usbmap
22:03.02asterisk99drwxr-xr-x   2 root root    360 2006-02-18 15:56 volatile
22:03.06asterisk99Sorry... he needs to pee
22:03.11mockerhttp://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message
22:03.14Qwell[]I said the misc dir :P
22:03.20dpolitechuhh
22:03.20Qwell[]nevermind, I'm done
22:03.32mockerIn this example is says that it would be possible to reschedule the call, can someone give me a rough idea of how to do that?
22:03.38Qwell[]_Sam--: He's all yours
22:03.39puzzledasterisk99: use pastebin.ca if you want to paste more than 4 lines and post the link here
22:03.46asterisk99Quell[]: 0 files in .misc
22:03.51Qwell[].misc?
22:03.56puzzledmaybe they live in extra
22:04.04Qwell[]puzzled: no, I saw the make install put them in misc
22:04.25puzzledmaybe he's been haz0red then :)
22:04.26Qwell[]asterisk99: I'll be back in like 10 minutes.  If you give me ssh access to your machine, I'll have it fixed in about 2
22:04.33_Sam--how about "updatedb"  then "find wctdm.ko" ?
22:04.39_Sam--er locate wctdm.ko
22:04.46Qwell[]_Sam--: was gonna say...
22:04.49*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
22:05.03Qwell[]asterisk99: okay?
22:05.05asterisk99Quell[]: I could give u ssh if needed
22:05.12Qwell[]alright, brb then...get it ready for me
22:05.16*** join/#asterisk darby_t (n=tom@abcf71.neoplus.adsl.tpnet.pl)
22:05.17*** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
22:05.46*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)
22:05.49_Sam--asterisk99 :  that is a nice offer.  hope you buy that guy a case of good beer or something.
22:06.42puzzledthe real stuff like Grolsch :)
22:06.53dpolitechwhich reminds me
22:06.58dpolitechlooks like beer-thirty
22:07.02dpolitechquittin time!
22:07.07_Sam--lol
22:07.08dpolitechlater
22:07.22_Sam--drink a few back for us
22:07.31dpolitechwill do
22:10.15asterisk99_Sam--: I'll buy a round for everyone if this works!!!
22:10.40lazzarellobeer o'clock!
22:10.43_Sam--be careful what you say...we will take you up on that, and it could be an expensive round :)
22:10.46*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:10.48asterisk99_Sam--: But I don;t think PayPal accepts alcohol deposits
22:11.13_Sam--we have some creative souls here...im sure if beer and/or other alcohol is involved for free, we will find some way
22:11.14Qwell[]okay, back
22:11.15austinnichols101asterisk99: not true - just select the 'other' type of payment from the listing
22:11.45asterisk99_Sam--: And I'm talking good 'ol 10%-by-volume Quebec Beer - better than that Canadian pee
22:12.05Qwell[]awesome, my proxy server isn't letting me connect to my other server
22:12.16Qwell[]there we go D:
22:12.23Qwell[]asterisk99: msg me the ip/password
22:12.38*** join/#asterisk rob- (n=robbie@haylott.plus.com)
22:13.19*** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de)
22:14.33lunaphytewhy might asterisk segfault when a skinny phone answers a call from a sip phone?
22:14.52jontowskinny is an ugly, ugly, horrible thing.
22:15.24_Sam--someone else today was talking about chan_skinny, probably qwell / mog
22:16.23lunaphytei gather..  :)
22:16.49lunaphytethe phone (12sp) rings, but when i pick it up, asterisk seg faults.
22:19.23lunaphyteand when i try to call x-lite from the 12sp, asterisk doesn't complete the call, seeming to be waiting for more digits.
22:19.51*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
22:20.14*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
22:25.51cypromisw 8
22:27.54[TK]D-Fenderasterisk99 : Thats right... American beer is fit only for children and the elderly :D
22:28.46*** join/#asterisk rene- (n=rene-@201.127.101.127)
22:29.02jontowlunaphyte: i had "ok" luck with a 30VIP/12SP+
22:29.39puzzledlunaphyte: the stuff from chan-sccp.berlios.de seems to work ok
22:29.57lunaphyteactually, that 12sp i mentioned is a 12sp+
22:30.28lunaphytepuzzled: thanks, i'll give it a shot
22:31.56rene-im using a centos based asterisk distro from signate, zaptel isnt working, signate is telling me to upgrade to centos 4.2, i have tried compiling asterisk from source and it aint working, im getting an invalid module format error when i try to modprobe zaptel modules, would compiling kernel from source be an option for me?
22:32.11puzzledlunaphyte: and if you have an issue with it, subscribe to the list with your problerm. the author is quite responsive
22:32.37*** part/#asterisk devel (n=devel@wiggum.digitalcoven.com)
22:33.29*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
22:35.19cypromisrene-: no
22:35.28cypromisbut we have a shitload of servers with centos
22:35.35cypromisand no issues with them
22:36.38rayvdshitload = 17 :(
22:37.12|omni|I don't like CentOS
22:37.18|omni|not really a fan of RedHat though
22:38.02freezeri love debian
22:38.08rene-cypromis: what are you choosing when you first install them? server, custom? and, is udev an issue?
22:38.35FLeiXiuSIs there a function which will send an incoming call to all extensions?
22:38.50rene-i love debian too, i only like centos because it knows about scsi controllers and its free unlike redhat
22:39.09FLeiXiuSrene-: redhats free version is now known as fedora
22:39.14|omni|I'm a slackware dude... on the linux side anyway... FreeBSD is my fav but asterisk likes slack better
22:39.50freezerwhats better at freebsd?
22:40.07rene-i have had success with fedora 4 and scsi controllers, but the only thing i could load on  a dell 2650 was white box enterprise
22:40.11|omni|better? ..just preference
22:40.31|omni|I've been using FreeBSD longer and all of our webservers and mapservers run BSD
22:40.51|omni|but early on asterisk had issues with BSD and my partner is  a slack guy so I started messing with that for the phone stuff
22:41.49cypromisrene: minimal :)
22:41.52*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)
22:41.55TagorI have a DSL connection which I want to use for several computers. I also want to connect my Asterisk server to the internet, to make external calls. Is there a router which doesn't have NAT problems? Or is this impossible?
22:41.55cypromisand than yum it to what I want it to be
22:42.15X-Rob_<PROTECTED>
22:42.24rene-cypromis: i dont think it has minimal, this signate centos distro
22:42.48remissrene-: it has, but choose something else or you'll be spending your days resolving dependencies
22:42.54TagorX-Rob_ >> Hmm, the problem when I do that, is that the connection gets very slow
22:43.07TagorX-Rob >> Or are there decent programs that can handle that?
22:43.53remissrene-: you probably want workstation or server... :)
22:44.46rene-hehe
22:44.49Seldon1975when i upgrade from * 1.2.1 top 1.2.4 do I have to recompile the wctdm module?
22:45.33jarrodis there a way to set caller-id with ser when matching source account
22:45.34jarrod?
22:49.48mzobleh i broke my asterisk trying to do FWD.  Every number i dial is 404 now. YAY :)
22:50.32jarrodsounds like all your numbers are matching a blackhole inbound ext
22:50.34Hmmhesayscool
22:50.46Hmmhesaysi'll fix it for a cool $40 bucks
22:51.02fileHmmhesays: does that involve putting the cash in the freezer?
22:51.08Hmmhesaysfile: yes
22:51.12Hmmhesayswet cash
22:51.15fileexcellent
22:51.24Hmmhesaysugh, in 7 hours I turn 24
22:51.30remissdon't brake it...
22:51.57fileHmmhesays: you're O-L-D
22:52.03HmmhesaysI know!
22:52.32Hmmhesaysi can feel the aching in my joints
22:52.36X-Rob_Tagor, most people use a linux box as their router, they just don't know it.
22:52.42Abydos313Happy Bday Hmmhesays!!!
22:52.47Hmmhesaysthanks
22:52.53remissHmmhesays: i do that as well and i'm only 20...
22:52.58Hmmhesaysi can't believe i've been hanging out in this channel for 2 years
22:53.02Hmmhesayssweet geebus
22:53.17*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
22:53.18fileI've been here longer then... that's more scary
22:53.22Abydos313i bet you've learned alot
22:53.36HmmhesaysI come in here to BS a lot
22:53.42Hmmhesayshelp out every once in awhile
22:54.00remisscan you help me? i need to fall asleep...
22:54.00Hmmhesaysas a pleasant distraction
22:54.03TagorX-Rob_ >> But can this be handled by a 500mhz 128mb ram computer? Can this computer run as router (for 4 pc's), Asterisk server and CUPS server?
22:54.17Qwell[]I don't know how to save a screen session
22:54.18*** join/#asterisk jtodd (n=jtodd@mccpool-2.ci.monterey.ca.us)
22:54.25Qwell[]erm
22:54.31Hmmhesays128mb of ram would be  cutting it a little short
22:54.31*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
22:54.38X-Rob_Tagor, yes. Most ADSL routers are tiny little 50mhz MIPS machines that easily handle full ADSL speeds
22:54.41fileHmmhesays: hot'n'sexy Hmmhesays
22:55.00Hmmhesaysyes file?
22:55.02TagorOk, thanks a lot X-Rob_!
22:55.04fileHmmhesays: how goes
22:55.12Hmmhesaysstill living
22:55.16remissQwell[]: save a screen session?
22:55.21Qwell[]nothing, ignore that
22:55.23filemy ADSL modem is actually a 48MHz Motorola chip
22:55.23Hmmhesaysgot up and rocked the house last night at jam night
22:55.29filewith 4MB of RAM and 1MB of flash
22:56.25Hmmhesaysmy nintendo was a 4mhz chip
22:56.44b0efany reason I cannot transfer a call to voicemail? - I can transfer all other calls
22:56.44MavvieHmmhesays: don't think you can do cups on that :-)
22:56.53HmmhesaysMavvie You never know
22:57.01Mavvie(that's not strictly true, there were print-servers on XTs in the past :-)
22:57.09fileHmmhesays: damn, you are O-L-D
22:57.26HmmhesaysI know it
22:57.37fileHmmhesays: so what'cha gonna do to celebrate?
22:58.13b0efis it just me or can anybody else transfer calls to voicemail?
22:58.24fileb0ef: see, there's a problem with that question
22:58.45fileb0ef: you write how the dialplan works, so while other people may write their dialplan so it works fine and dandy, you may not have
22:59.53fileyou tell Asterisk what to do, so if you don't tell Asterisk a method of transferring calls directly to voicemail well then it won't do it
23:07.36Qwell[]ugh
23:09.42*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167041112.pppoe-dynamic.nb.aliant.net)
23:09.43rene-im getting while modprobing zaptel. zaptel: version magic '2.6.9-22.0.1.ELsmp SMP 686 REGPARM 4KSTACKS gcc-3.4' should be '2.6.9-22.0.1.ELsmp SMP 586 REGPARM 4KSTACKS gcc-3.4' fatal: error inserting zaptel.... invalid module format, how can i fix it...
23:10.22Juggiestop using pre compiled versions of asterisk
23:10.25Juggiecompile it yourself
23:12.22Netgeekswhy oh why do people write AGI programs to do stuff asterisk does inherently?
23:12.29SplasPoodwell that may not be his/her issue
23:13.29mzojarrod, probably, you mean my problem all my numbers matching a blackhole inbound extension? :P
23:13.34rene-Juggie: iam compiling zaptel from source
23:13.36SplasPoodNetgeeks: I've been tempted to before.. simply cause the dialplan syntax (even with AEL) sorta sucks
23:13.56SplasPoodrene: you're just not compiling it using the same compiler settings as your kernel was originally compiled for
23:14.17SplasPoodrene: note the 686 vs 586
23:14.52mzowhich is 686, pentium III or better?
23:15.17SplasPoodyes, I do believe
23:15.21*** join/#asterisk paxr0 (n=paxr0@216-155-79-99.bk1-dsl.surnet.cl)
23:15.30NetgeeksI was just trying to figure out why A@H has an AGI for the directory function... especially when the AGI doesn't work
23:15.39mzoi think i was drunk when i recompiled 1.2.3 so I don't know when that happened.
23:15.44SplasPoodNetgeeks: heh..
23:15.54mzoNetgeeks, i broke my AAH install quite nicely.  Someday it'll actually be working ;)
23:16.13*** join/#asterisk Shido (n=shido6@d221-68-216.commercial.cgocable.net)
23:16.22*** join/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk)
23:16.23SplasPoodhey, can someone DCC me a stock A@H dialplan (extensions.conf)
23:16.45NetgeeksI refuse to use A@H myself
23:17.08rene-SplasPood: yes i notice that, would compiling a new kernel for this machine would be of any help?
23:17.09Netgeekshowever I get alot of 'please help me fix this I can't get it to work' calls
23:17.13mzowhat's wrong with it? :P
23:17.21mzoi mean i break it at least once a week. ;)
23:17.44SplasPoodrene: chances are if you compile a new kernel it'd compile using the same settings zaptel does, however you may cause yourself other problems if you don't do it right..
23:18.32rene-I once compiled a kernel for vserver usage, so i am no expert but i have done it, the only tricky part i have seen is getting the support for the scsi card and the netcard  right
23:18.49mzoactually somewhere along the way
23:18.50rene-and the SMP part
23:18.54mzoAAH added support for SMP kernals
23:18.59mzoit picked up my SMP out of the box
23:19.22paxr0anybody know softphone (for palm OS) ?
23:19.40TamariskCan anyone recommend a lowend SIP phone I could look to use on a home network to learn *. I want to remove my slow computer issues when trying with softfones. TIA
23:19.44rene-i guess the only way to know is trying
23:19.45mzoso who wants to laugh at me a lot and figure out why i broke FWD :)
23:20.54*** join/#asterisk kremin (n=kremin@dslb-084-063-106-035.pools.arcor-ip.net)
23:21.06rene-i am only interested in the first part but im quite busy right now, so i would have to pass on your offer
23:21.22rene-jk
23:21.25mzoi'm going to fix asterisk by -rm -rf :P
23:21.52De_Monwhats -rm -rf do?
23:22.04Qwell[]De_Mon: Gives a command not found error
23:22.12kreminis this the right place for questions about asterisk configuration?
23:22.15mzodon't try it :P
23:22.20De_Monmzo I don't think that's gona fix asterisk
23:22.21Qwell[]kremin: yes
23:22.33mzoDe_Mon i know it won't :P i broke it bad
23:22.51De_Monmzo try rm -rf /etc/asterisk and rebuild the examples
23:22.51kreminok ... i guess most of you guys are from the us?
23:22.56kreminUS=
23:22.59kreminUS?
23:23.02De_MonI'm not
23:23.03rene-mx here
23:23.16mzoDe_Mon, it works, i broke it when i added FWD, I can jsut remove that and try again
23:24.03kreminanyway .. i need to know if i must use a crossover cable to connect two HFC-S cards to two german "Anlagenanschlüsse" .. that are sprecial ISDN connections.
23:24.08De_Monmzo naaa you broke it worse than that, I suggest going with the rm -rf
23:24.17kreminahh ... it was to long ... agaiN:
23:24.23Netgeeksbut then again I have AGI with a red circle / slash tatood on my forhead
23:24.42luckyduckkremin: depends on the cards you use
23:24.56kremini have two HFC-S cards
23:24.59luckyduckkremin: in the case of quadbri cards, you dont have to
23:25.01luckyducksingle port?
23:25.08kreminyes single port
23:25.15kremintwo of them
23:25.23luckyduckin that case, you need crossover isdn cables for the nt modus
23:25.44luckyduckif you want to connect a tka to the cards
23:25.56kreminwhats a "tka" ?
23:26.06luckyduckif you only want to connect to the ntba, you need the te modus
23:26.07b0effile: well, that's not really my problem: when I call my voicemail to read messages, I want to transfer it to a conference room so that other people can listen whil I go through my voicemail messages
23:26.08mzoDe_Mon you think?  lemme try the rm -rf :P
23:26.11luckyduckand no crossover
23:26.19luckyduckkremin: something like a eumax from the t-com
23:26.22luckyduckand so on
23:26.25mzo~wiki
23:26.33luckyduckor eumex
23:26.36puzzledanyone know how I fix this error compiling asterisk-1.2.4? cli.c:49:30: error: asterisk/version.h: No such file or directory
23:26.39kremini want to connect to the NTBA
23:26.44luckyducka hardware based, quite cheap pbx
23:26.52luckyduckthen, you dont need a crossover
23:27.08kremini want to get rid of the pbx and exchange it with asterisk
23:27.17luckyduckit's only needed to the nt modus, i.e. in the case where you have to provide an nt port
23:27.20De_Monpuzzled figure out why version.h wasn't in your source
23:27.34luckyduckkremin: =)
23:27.53puzzledDe_Mon: problem only occurs on ppc. on x86 it compiles fine
23:28.36kreminso i wired the two HFC-S to two NTBAs ... but crossover. in zapata.conf i said "signalling=bri_net"
23:28.41kreminthat ok isnt it?
23:29.04De_Monppc? there, i can't help.
23:29.35luckyduckkremin: sorry, never played around with the zapata.conf so far.
23:29.49luckyduckhowever, have to go to bed. already late
23:29.52luckyducknight
23:29.55kreminnight
23:30.42kreminanybidy else using HFC-S cards connected to NTBAs?
23:33.28TamariskCan anyone let me know what they recommend as a basic SIP phone on their asterisk server?
23:33.58austinnichols101Tamarisk: price range?
23:34.38TamariskI do not know at moment, I guess less then £50 each this is for home/hobby use so 2nd from ebay would do
23:34.41|omni|handset or ata type?
23:35.01Tamarisklooking for more of a telephone style
23:35.25TamariskI want to stop using softphones
23:35.30|omni|not a big fan of the cheap handsets...cause they feel cheap...but apparently some of the lower end Snom's are pretty good
23:35.42|omni|but..you could always  just get a Sipura ATA (like a 2000)
23:35.46|omni|and use any regular phone you want
23:35.49austinnichols101I definitely didn't like the sipura 841
23:36.07TamariskOK I am writing these down so I can search for them later
23:36.13austinnichols101but the low-end aastra (9112, I think) was good
23:36.22*** join/#asterisk exonic (n=exonic@209.172.11.54)
23:36.24mzostupid question what's your default [your-main-context-whatever-it-may-be] if i'm using aah?
23:36.53exonicI have a problem with caller id not being set on outgoing calls. It only shows one number (that's not even in my configuration!? )
23:37.29Tamariskmy asterisk is purely IP based PC so no cards fitted
23:37.33|omni|even when you use SetCallerID() ?
23:37.50|omni|Tamarisk: : http://www.sipura.com/products/spa2000.htm
23:38.08|omni|2 lines, cheap, no cards needed (SIP) and you can use any standard POTS phone you want
23:38.10TamariskCheers
23:38.29TamariskIs that one of the adapters
23:38.36|omni|ATA, ya
23:39.02TamariskOK catching on slowly
23:39.22FLeiXiuSAny idea why "exten => 500,2,VoiceMailMain(${CALLERID(num)})" is not picking up the CID number.
23:39.32|omni|on the desktop I really like the Cisco phones.. like 7940 or 7960s...you can find them on ebay for a couple hundred bucks
23:39.59FLeiXiuS|omni|: Plus, their SCCP support is quite nice ;-)
23:40.12austinnichols101tamarisk: I would have to agree with omni - ATA is better than any of the phones that are below USD $100.
23:40.22|omni|ya actually I just switched my 7960 to the SIP image
23:40.24rene-how much time does a kernel compile should take on a p4 3.4?
23:40.43austinnichols101FleiXiuS: why do you like the sccp?
23:40.43|omni|I was using chan_sccp until a couple days ago
23:41.01rene-id like to see sccp realtime
23:41.14|omni|but the sip image has more usable options (CfwdAll, Conferencing on the phone, etc.)
23:41.15Qwell[]rene-: chan_sccp?
23:41.19|omni|rene-: , apparently they're working on it
23:41.27Qwell[]working on - and finished
23:41.34|omni|oh really?
23:41.37Qwell[]now...go yell at Sergio to implement my patch
23:42.29*** join/#asterisk JonR800 (i=jon@p1mp.org)
23:42.29rene-great news, i was told that the feature set on sccp was superior to sip on cisco phones
23:42.33mzoshould i be using IAX or SIP for FWD?
23:43.20TamariskOK thanks for the tips all, at least I can go search ebay UK or worldwide.  My softphones work sometimes sometimes not so need to remove one type of issue
23:43.22rene-i need to get realtime, well im still waiting on the firmware for my phones so no rush
23:44.50*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)
23:46.53*** part/#asterisk fjean (n=fjean@201.29.122.10)
23:47.25TamariskSorry another question on ATA's what do they mean by "Unlocked"?
23:47.45Qwell[]Tamarisk: not locked to a provider - like Vonage
23:48.43TamariskAHHH  Ok just found an sipora Grandstream and wondered.
23:48.50*** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-233-136.dsl.snantx.swbell.net)
23:49.04TamariskOK I will leave you all at the hard work thanks for assistance night
23:49.06Qwell[]DarthClue: Eww...it's...you
23:49.16_Sam--[av]bani :  you are Dan on the * users mailing list?
23:49.48[av]bani_Sam--: ?
23:50.00*** part/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk)
23:50.12_Sam--i see some guy active in the gxp discussion..with domain anime.net...
23:50.18_Sam--i thought it was you
23:50.26_Sam--asterisk-users
23:50.57_Sam--<_Sam--> asterisk-users
23:50.58_Sam--er
23:54.58_Sam--[av]bani :  your isp is visp?
23:55.41[av]baniwhatever gives you that idea?
23:55.48_Sam--just speculation
23:55.51[av]baninosy :)
23:57.18_Sam--this guy sounds crafty:  "Patent Pending Permissions Technology"
23:57.26_Sam--and even authored the patent
23:58.09rene-but its flaky on os x
23:58.21rene-sp: flacky
23:58.41_Sam--WasPhantom:  flakey
23:58.43_Sam--er
23:58.55rene-thanks
23:58.57_Sam--flakey like a nice fresh croissant

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