irclog2html for #asterisk on 20060430

00:14.40*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
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00:36.02*** join/#asterisk kaz0358 (n=kurtzogl@asterisk.telecom.ksu.edu)
00:36.56*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
00:39.42kaz0358when you create an entry in an enum registrar, you can specify a sip uri. is it valid to specify an extension? ie guest@hostname.com/1234
00:41.15gaupekaz0358: http://www.voip-info.org/wiki/view/ENUM+syntax
00:46.43kaz0358gaupe, thanks.. but i'm not for sure that answers my question. if you have several PSTN DIDs listed in e164.arpa and they all point back to the same asterisk server, how do you direct them to the right extension?
00:48.52gaupeENUM is just the registry, the call has to be set up with standard SIP - so you will pass the phone number there
00:49.11gaupethen it's up to you :)
00:50.44kaz0358gaupe, so for instance.. if you want to call 1-234-567-8901 and it comes back with a sip entry of guest@asterisk.com.. the equilavent command in asterisk would be dial(SIP/guest@asterisk.com/12345678901,60)?
00:51.38gaupeyes, I think so - just started looking in to this my self
00:53.42kaz0358gaupe, okay.. we i wasn't a 100% sure.. while registering on e164.org i did a search for anyone registered in my city and found 1 hit.. but when i did a lookup, it wouldn't go through.. it appears that he has incorrectly listed his info b/c the sri contains the pstn number... ie guest@asterisk.com/pstn-num..
00:54.23kaz0358s/we\ i/we/; s/sri/uri/;
00:55.42gaupeseen this? http://www.e164.org/wiki/AsteriskExamples
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00:57.20kaz0358gaupe, no.. i hadn't seen that.. but i was looking at http://www.voip-info.org/wiki/view/RFC+Compliant+ENUM+Macro
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01:00.18kaz0358gaupe, what i'm not getting is why on both the url you gave me and the one i gave you.. you basically find.. Set(DIALSTR=SIP/${ENUM:4}) instead of Set(DIALSTR=SIP/${ENUM:4}/${EXTEN})
01:01.49kaz0358gaupe, that would seem to imply for instance that an enum lookup pointed to an ansterisk server and a default extension.. right?
01:02.37kaz0358gaupe, if the sip uri in the naptr doesn't contain the destination extension.. and it isn't being supplied back in the dial script, then you end up with that situation.. right?
01:06.19gaupeyes, that seems right - SIP/${ENUM:4} should give you the phonenumber too
01:07.46kaz0358but the uri doesn't contain the phone number if you make it "guest@asterisk.com/phonenum".. asterisk will think the hostname is "asterisk.com/phonenum"
01:08.19kaz0358okay that wasn't clear..
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01:23.59*** join/#asterisk talljon84 (n=jonathan@66-188-104-144.dhcp.mdsn.wi.charter.com)
01:24.11talljon84Is anyone aware of a SIP client for Palm OS?
01:24.44*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
01:24.53Drukenevening everyone
01:24.59talljon84evening Druken
01:25.01kaz0358hi druken
01:25.32Drukenso uhmm, yeah... who's got a WORKING wakeup call agi ?? :)
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01:26.22Qwellwhy AGI?
01:27.05Drukendoesn't have to be, just something that works
01:27.35Drukeni guess agi would only be for the phone to set wakeups
01:27.38kaz0358druken, you could just setup a cron job to create a .call file
01:28.12Drukenkaz0358: i'm very awear of that... but i'm lookin for the backend to that
01:28.31Drukeni'm being a lazy fuck and don't want to make my own :)
01:28.55kaz0358druken, ahh. :)
01:30.02DrukenQwell: you got something for me?? :)
01:33.34QwellDruken: $$$
01:33.35Qwell:p
01:33.51Qwell$$$ talks, rather
01:34.58DrukenQwell: my checks are so tight, they would have withstood katrina
01:35.11Drukener.. cheeks i guess
01:35.28Qwelleh?
01:36.29*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
01:39.17wunderkinyou don't want to say that in here :D
01:40.22Drukenwhy not? i give back to the community here...
01:40.43wunderkinas long as you will take it for the team
01:41.01Drukensomeone has to... hehe
01:41.31Drukenhehe
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01:48.03*** part/#asterisk BadPacket (n=root@unaffiliated/badpacket)
01:50.47Drukenis it just me, or shouldn't the call files to changed so we can schedule calls into the future? i can think of alot of placed that would come in handy... fax back services, wakeup calls, etc
01:55.21QwellDruken: You already can.  `touch` it to a specific date/time
01:55.29Qwellbefore moving it
01:57.33Drukenoh... so if the creation date is in the future, asterisk will ignore it?
01:59.22QwellThat's the rumor
01:59.36Drukenahh, excelent...
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02:02.59tomcontr3hello guys,  I have found a problem with asterisk,  that might be a bug or maybe a bad configuration
02:04.08tomcontr3I have 2 trunks cofigured from the same privider,   and I can only use the last trunk that I added.
02:04.21tomcontr3I thoung I might be a problem from the provider
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02:04.57tomcontr3but,  the I configured an X-lite softphone,  with one of the trunks
02:05.11tomcontr3and I was able to make calls using both lines
02:05.22tomcontr3so I must be a problen regarding asterisk,
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02:05.36tomcontr3it
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02:18.23rpmi finally figured out how to get the wgt634u running in client-bridge mode, except the bridge interface keeps going through the spanning-tree recalculations
02:28.42bsdfreakheh
02:30.45*** join/#asterisk RES2 (n=RES@chello213047231029.tirol.surfer.at)
02:31.17RES2hi
02:31.24ManxPower~docs
02:31.25jbotextra, extra, read all about it, docs is 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
02:31.26ManxPower~mailinglist
02:31.28jbotSearch Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm
02:32.41RES2Are there people, who have spandsp on asterisk running?
02:33.43RES2my problem: "loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str"
02:33.48*** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
02:34.52wwalkerAnyone know of a reason that the manager interface limits the listen() backlog to 2?  I want to set it to 40 instead...
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02:42.16*** mode/#asterisk [+o denon] by ChanServ
02:44.36ManxPowerwwalker, try it and see
02:44.59ManxPowerRES2, did you search the mailing list archives for that error message?
02:45.22RES2ManxPower: yes
02:45.50ManxPowerI THINK that error message is either 1) problem with spandsp or 2) a problem with the libtiff libraries, check the spandsp readme and confirm you have a correct version.
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02:47.09RES2ManxPower: I already use the correct version of libtiff and libtiff-devel.   :-(
02:47.13wwalkerManxPower: thx.  I know it will fix my problem.  I just wonder if anyone knows of another problem it will create that might not be obvious in testing and will bite me hard in production under load.
02:47.53ManxPowerRES2, Try using the code from here if you have not already done so: http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre25/
02:48.37RES2ManxPower: I already use this code.
02:48.44ManxPowerwwalker, The manager interface has problems with handling a large number of connections, which is why astmanproxy was created, as talked about on the mailing lists in the past few days.
02:48.57wwalkerAh, thx
02:48.57ManxPowerRES2, I don't know what to suggest then.  You can ask coppice if he's around.
02:49.39ManxPowerwwalker, astmanproxy also does REALLY COOL stuff like provide a translation layer to talk to the manager interface using things like XML and other types of stuff.
02:49.56RES2MaxPower: OK. Thank you.
02:51.25wunderkinManxPower, the manager should not have a problem with a large number of connections anymore
02:51.42ManxPowerrpm, That sounds like a network loop
02:53.00ManxPowerwunderkin, Cool.
02:53.24ManxPowerwunderkin, so why is the connection backlog set to 2?
02:53.39wunderkini only use like 3 manager connections at a time, but i reported some manager issues awhile back, i would think that should have fixed most of that
02:53.42wunderkini don't know what that is
02:53.53ManxPowerwwalker, you can also ask on #asterisk-dev
02:54.04wwalkerthx, will do
02:56.37[hC]anyone here have/used a linksys wip300?
02:56.41[hC]mine seems to be... broken?
02:56.46Qwell[hC]: the new wifi one?
02:56.50[hC]Yeah.
02:56.56QwellI used it VERY briefly at VON
02:57.00Qwelllike, 2 minutes :p
02:57.09[hC]mine doesnt seem to want to detect any wireless networks anymore
02:57.17[hC]site survey returns 'no records' and specifying manually just does... nothing.
02:57.32tainted-broken antenna
02:57.35tainted-probably
02:57.42[hC]like either the radio is asleep, or the firmware is messed up
02:57.45[hC]the antenna is inside the thing
02:57.54[hC]it did work for a while
02:57.55[hC]then it just stopped.
02:58.14tainted-expensive brick
02:58.33*** join/#asterisk johngalt (n=john@eowyn.blacksun.net)
02:59.14RES2cu
02:59.24[hC]looks that way so far
02:59.48[hC]qwell, i think i might have enough time tonight to load trunk on a box here and try your skinny driver with my 7914 FINALLY
03:00.02Qwell[hC]: well, you're in luck.  I made a branch this morning
03:00.05[hC]I had to spend all last week writing a billing interface and phone mgmt interface for a client
03:00.11Qwellteam/north/chan_skinny-fixup
03:00.28[hC]OOooh really.
03:00.38MikeJ[Laptop]go team north!
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03:01.21key2!seen kram
03:01.28Qwellkey2: weeks
03:01.30Qwell~seen kram
03:01.34jbotkram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 26d 23h 54m 55s ago, saying: 'oh most certainly :)'.
03:01.41MikeJ[Laptop]key2, yeah, I've seen him.. .
03:01.48key2lol
03:02.06MikeJ[Laptop]I heard him more recently tho
03:02.28[hC]Qwell: ive never used a team branch, i presume this patches against 1.2.7.1, is that the idea?
03:02.40Qwell[hC]: nope, it's a full pre-patched version of trunk
03:02.50johngaltour asterisk system was working fine earlier today but now I recieve incomming calls fine but get dead air when I try to dial out.  the phones here are all sip based, 3 grandstream bt-101, and a sipoura adapter for a cordless pots phone.  no config changes have been made and the system has been working well for months.  any idea what I should be looking at?
03:03.19xachenjohngalt: * in general
03:03.21[hC]Qwell: oh yeah i see that now that im browsing the tree
03:03.24ManxPowerjohngalt, did you reboot?
03:04.27johngaltyes, I rebooted all the phones, no go...then I rebooted asterisk, then rebooted the phones again. still same issue
03:04.43ManxPowerxachen, how are your calls going out to the PSTN
03:04.53xachenI just use SIP termination
03:05.26ManxPowerxachen, what does the asterisk console say when you dial out?  Use pastebin.ca if it's more than 3 lines
03:05.34johngaltcombination of sip and t1 interface.  the strange thing is that we can recieve incomming calls fine.
03:05.49ManxPowersorry, that was for johngalt
03:05.50johngaltjust can not outdial
03:05.53xachenManxPower: I'm not having problems now
03:06.00xachenI just cracked a funny
03:06.05xachenhah hah laugh
03:06.07xachen:)
03:07.44*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
03:07.51johngaltideas?
03:08.12xachenjohngalt: Have you stopped asterisk and restarted it?
03:08.32justinu|laptopturn on sip debug
03:11.59*** join/#asterisk file (i=jcolp@216.237.114.82)
03:12.32johngaltthe reboot should have done that but I have limited access to the box.  I have a email in to our main admin to get root but I can't exactly call him.  mainly looking for what to look at when I can look further.  I realize that only so much can be done without logging in.
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03:25.34[TK]D-Fender<PROTECTED>
03:26.30Drukenhey tk, your village called, they want their idiot back :) hehe
03:28.48[TK]D-FenderTell them I'm still looking, but its hard to sift though so many here for ours :)
03:29.02Drukenhehe too true
03:30.13Drukenprobably he one in here the otherday looking for a provider that allows 1-900 calls :)
03:31.40[TK]D-FenderDear God the people we get in here.... Every schmuck who insists their POS modem MUST work with *.  And then the semi-enlightened who use all the stuff we divert to other channels for...
03:32.52*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
03:34.01filehrm, a spoon
03:34.04filehow...
03:34.05filespoon like
03:34.13[TK]D-Fenderhow... astute
03:34.45fileyay
03:36.28tainted-what's the proper syntax for setting calleridname and calleridnum in dialplan?
03:37.17[TK]D-FenderSet(CALLERID(num)=1234567890)
03:37.26tainted-what about name
03:37.27[TK]D-FenderSet(CALLERID(name)=Joe Blow)
03:37.36tainted-what variable uses holds name
03:37.41[TK]D-FenderAssuming 1.2.x of course
03:37.43tainted-${CALLERIDNAME} ?
03:38.00[TK]D-Fendertainted- : the function in reverse ${CALLERID(name)}
03:38.16[TK]D-Fendertainted- : Go creck out "asterisk functions" on the WIKI
03:38.43tainted-i lose callername and callerid when i send a call between asterisk boxes
03:38.47X-Robor even better 'show function CALLERID'
03:38.47Qwellspork == school-house shank
03:39.28[TK]D-Fendertainted- : How are you sending them across?
03:39.45[TK]D-FenderQwell : Man.... you outta south-central?
03:39.55Qwell[TK]D-Fender: I am now!
03:41.01tainted-[TK]D-Fender just Dial(IAX2/secondserver/${EXTEN}) from first server
03:41.24[TK]D-Fendertainted- : Shouldn't lose it unless you have a caller-id line for the user end...
03:43.19*** join/#asterisk george____ (n=hanpc14@p161ds3xi.xDSL-1mm.sentex.ca)
03:43.21george____Hi
03:45.43[TK]D-FenderSHHH!!!! You'll wake the crickets!
03:53.44*** join/#asterisk bmg505 (n=leon@dsl-146-24-53.telkomadsl.co.za)
03:55.57[TK]D-FenderOk, I'm baked... outta here.  later all.
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04:18.57rpmbahah, i love this telemarketter torture.. (fwdtel: 712906)
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04:41.31coppiceRES2: don't mix softeware versions
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05:08.16Alystairhmmmmmmmmmmmmm
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05:29.05*** join/#asterisk Flauto (n=zhao@adsl-75-3-170-44.dsl.chcgil.sbcglobal.net)
05:29.18Flautoanyone here tried to use icall with asterisk?
05:32.50tainted-Fellatio: what is icall?
05:40.08*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
05:46.45AlystairIs there a way to do some neat integration between asterisk and our ActiveDirectory server at the office?
05:46.52AlystairEg. when I make a new account it gives them an extension etc
05:49.49Alystairignore the question
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06:01.56blackgeckoanyone has had problems with the tdm2400p ?? im using it but it randomly strips some numbers and the dialed number is incorrect, any idea ?
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06:04.21Flautotainted, it is a free service, check out www.icall.com
06:04.37websae[A]icall
06:04.39websae[A]what is that?
06:04.58*** join/#asterisk kavit (n=kavit@60-240-44-231.static.tpgi.com.au)
06:05.07websaewhat is icall about?
06:05.27Flautoit is a free service
06:05.36Flautothe enables calls to usa
06:05.40Flautofor free
06:06.01websaethat wont last long
06:06.02Flautobut i am trying to figure out how to make it to work with asterisk
06:06.04kavitsay is it possible to use asterisk as the gateway for video calls with h323 driver?
06:06.20websaeh323 sucks
06:06.25websaeespecially on asterisk
06:06.38Flautotried i tried it and it works only sometimes
06:06.43Flautoi mean video
06:07.29Flautowebsae, i got the windows client for icall and it works
06:07.42Flautoi tired a few calls and it sounded okay
06:07.54Flautobut the thing is that i want to use it through asterisk
06:07.59kavitwell i just want to use it as a gateway and pass the call to an mcu
06:08.06coppiceicall claim better sound quality than skype. i wonder what they use
06:08.06Flautoi tired a few things and it did not work
06:08.36Flautocppice, it uses sip
06:08.47blackgeckoanyone has had problems with the tdm2400p ?? im using it but it randomly strips some numbers and the dialed number is incorrect, any idea ?
06:08.51Flautobut i dont' know what codec they use
06:09.41coppiceyou should be able to find something about the codec from the SIP messages
06:10.29Flautowhen i use that client, it pops a notpad file
06:10.37Flautowith some info
06:10.48Flautoso i figured that the server is beta.icall.com
06:10.52Flautoand they are using sip
06:11.10Flautobut for authantication, i am not sure
06:12.11coppicesignup by invitation only. how very exclusive :-)
06:14.06Flautocoppice, use FLAUTO
06:14.15Flautoas invitation
06:21.05*** part/#asterisk blackgecko (i=blackgec@201.152.14.187)
06:21.57Flautocoppice, are you there?
06:22.18coppiceyep
06:23.03Flautodid you get the invitation
06:24.03FlautoApr 30 01:23:37 NOTICE[10595]: chan_sip.c:9548 handle_response_invite: Failed to authenticate on INVITE to '"Zhao Liu" <sip:flauto@beta.icall.com>;tag=as732b7366'
06:24.10Flautoi got this when i tried to call
06:24.29coppiceI didn't try to sign up
06:24.37Flautooh
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06:27.40Flautowould anyone here be interested in figuring out how to make icall to work with asterisk?
06:28.05rdgztI'm having a weird problem, despite my best efforts, I can't register via SIP, asterisk tells me the password is wrong even when I'm sure it isn't.
06:30.41rdgztDoesn't matter what I set as secret in sip.conf or anything.
06:32.55Flautoit does matter
06:33.22rdgztWell, yes, obviously, but I'm sure the secret in sip.conf and the password I'm using in the softphone are the same.
06:33.30rdgztI've tested with two different softphones to be sure.
06:33.38websaeFlauto: by the time you figure out how to connect to icall they will be shut down because of loss
06:33.46rdgztI'm following the example setup in Asterisk: The Future of Telephony.
06:34.23Flautowebsae, hehe, possible, but they must have some way to be able to provide free service
06:34.37rdgztAnd I've used both the xten softphone and ekiga, both give the same result.
06:34.50Flautordgzt, show your sip.conf settings
06:34.54Flautoit might tell something
06:35.42websaesure....they incorporate it as a startup cost.............and then they a) either have to start charging the end user a nominal fee to start making up for overhead, etc b) they shut down, and even if they do a) they could become like nufone and not have the capacity to support paying customers
06:36.12rdgztFlauto: Ok to paste here?
06:36.21Flautonot here
06:36.25Flautopastebin.ca
06:37.02rdgztOk, let me see...
06:38.28rdgzthttp://pastebin.ca/52345
06:38.36Flautois insecure=very still working in asterisk 1.2?
06:38.46rdgztThat has my sip.conf, and also, below the ---, what asterisk says when I try to register.
06:38.54Flautoor it is port,invite, that kind of things now
06:39.11Flautookay
06:39.47rdgztIt's really weird, it's an incredibly basic config, and it just doesn't want to work.
06:39.51rdgztKind of has me stumped.
06:42.42rdgztIt was badly wrapped in pastebin.ca, I updated it now.
06:43.10Flautoi can not open it
06:43.35rdgztNo? Let me see.
06:43.55rdgzthttp://pastebin.ca/52346
06:44.00rdgztThat doesn't work for you?
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06:46.20Flautoit does not
06:46.33rdgztWhat do you get as an error message?
06:49.04tainted-Flauto what's icall all about
06:49.25websaefree
06:49.29tainted-is it softphone?
06:49.33websaebeta voip
06:49.40tainted-SIP?
06:49.50websaefree softphone interface
06:49.52Flautoyes
06:50.03websaeto make long distance PSTN calls fro free
06:50.06tainted-is it locked to icall netowrk?
06:50.26websaedo you know if they even offer sip connections via your own sip client?
06:50.41websaecan terminate to PSNT lines free
06:50.46Flautotained, if you want to register, you can use FLAUTO as invitation code
06:51.04Flautothat is the part i don't know
06:51.13Flautoi just see the connection they use is sip
06:54.11kavitDoes asterisk work well with ISDN <---> H323 for video conferencing? Can't seem to find a definitive answer
06:54.54websaeH323 is a pian to get going on asaterisk to begin with
06:55.00websaeso i would not recommend that at all
06:55.25kavitwell can it be done, pain aside?
06:55.49websaedo you like pain where you feel like you're at the point of death?
06:56.27kavitno but if I have to endure it to get something working
06:59.03kavitbasically I want to know if I can get asterisk to relay ISDN video calls as H323 to a gatekeeper
07:00.47bmg505isn't a video call just a straight data connection that u can handle with v4l?
07:01.28bmg505s/v4l/i4l/
07:01.44bmg505:)
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07:03.10distortionok, im reduced to asking- what is the substring function in c?
07:03.47distortioni want to effectively do this: ${EXTEN:4} in an asterisk application
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07:40.34bmg505distortion: memove could do the job in straight ansi C
07:40.45bmg505memmove
07:41.29bmg505be mindful of the strcpy command as it is know to not do what u expect
07:42.01distortionmemmove(temp, temp+4, 11); -- is what i was playing with, cool
07:42.07distortionthx bmg
07:47.12distortionsweet that worked!!
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08:59.02Alystairare polycoms really good phones?
09:00.17AlystairI mean is the price justified
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09:05.38Oloboladudes, can't I just bypass a provider and recieve phone calls in some magical way? I can't wait around for this shit!
09:09.09tainted-Olobola what do u mean?
09:09.34evilbunyOlobola: you mean having the call go directly to your SIP device?
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09:22.25Olobolaoh I dunno -- I used to go through a webhost, now I host my own site and consequently never have to worry about downtime. I wish I could do the same with incoming toll free service.
09:25.35Alystairoutsource to india :D
09:27.38evilbunyOlobola: you technically can
09:27.38evilbunyalthough you are still dependent on carriers to route the call
09:33.57Alystairwhy isn't there a nice wiki which has reviews of all these ip phones
09:35.02kristalinoAlystair, i agree
09:35.35coppiceAlystair: start one :-)
09:35.55Alystaironly if I get free phones ;)
09:37.10coppicea sort of tomshardwareoverip, where you get more than just free phones - you get kickbacks :-)
09:37.53evilbunyOlobola: http://voip.wikispaces.com/IPDialling
09:38.27Alystairheh
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11:08.12SheriF_WorKi have diguim card with 2 moduls / FXS and FXO only. what additional modules should i load for zaptel to work ?
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11:16.42key2none
11:23.11SheriF_WorKkey2: sure ?
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11:28.24robin_szAlystair, there cant be a wiki about the IP phones for legal reasons
11:29.28robin_szbecause eventually, someone will review the GXP2000 and then we'll be in trouble with the ISP for bad language and obscenities
11:30.04poisonerhrhr
11:30.21poisonerHas anyone experiences with chan_sccp?
11:31.20poisonerI run into problems if I call a meetme-number from a skinny phone
11:33.04viperdudeukpoisoner: there is a chan_sccp mailing list you might be better asking on there. However I think there is a issue with chan_sccp and MeetMe
11:33.19poisoneryap
11:33.27poisonerThe question is who to blame for it
11:33.48robin_szhe is
11:33.51viperdudeukwell there is a guy called Sergio that maintains the code
11:34.23viperdudeukbut as its open source and essentially born out of good will I wouldn't think complaining will get you anywhere
11:34.43poisonerthe chan_sccp developers for letting chan->type be NULL or the developers of app_meetme.c for not putting a check before stcapcmp'ing chan->tpye...
11:34.51robin_szI had lots of problems with chan_sccp and some DECt->sccp adaptors
11:35.06robin_szthe biggest problem being  the docs :(
11:35.11poisoneryap
11:35.23viperdudeukchan_sccp is not officially part of * which is the main reason for the issue
11:36.00robin_szit was assumed you knew how it works and great chunks of "what to do to get it going" were not mentioned at all
11:36.10poisonerviperdudeuk: but not checking before comparing a string... isn't this a generic fault?
11:36.39poisonerchan_skinny perhaps crashes asterisks as soon as my 7940 want's to register
11:36.54viperdudeuki agree but if you feel strongly about it sumbit a patch
11:37.18poisonerfor app_meetme.c ?
11:37.25viperdudeukwe have stopped using chan_sccp with our 7960 due to crashes
11:37.45viperdudeukwhatever fixes it
11:38.02poisonerviperdudeuk: What are you using now? chan_skinny oder SIP-Firmware von 7960?
11:38.06poisoners/von/on
11:38.09viperdudeukyou might want to read the chan_sccp mailing list for more details
11:38.17viperdudeukSIP firmware
11:38.42robin_sz...
11:38.46viperdudeukwe only tried sccp as we wanted to use 7914's, however in the end we went with FOP
11:38.50robin_szhas it happened yet?
11:38.58poisonera coworker thinks that the SIP-firmware has missing some features...
11:39.06poisonerFOP?
11:39.07viperdudeuksuch as?
11:39.20viperdudeukFlash Operator Panel
11:39.38poisonerthe "conference" thing
11:39.46viperdudeukgoogle it
11:39.47poisonerbut in 7.5.something it worked
11:40.07viperdudeuk3 party conf works on 7.5 on 7940's I use it at work
11:40.37poisoneryap.. This is wat I discovered.
11:40.50poisoneronly 3party? or also more?
11:41.03viperdudeukwhile we are talking SIP does anyone know how to save the debug to a file?
11:41.32viperdudeuk3 party only on the phone
11:42.02poisonerat work, with our CCM an sccp firmware I think it could be more...
11:42.03viperdudeuk7940's at least not tried on a SIP 7960
11:42.14viperdudeukok
11:42.17poisonerok... thx for the infos.
11:51.53poisoneriiiek...
11:52.21poisonerviperdudeuk FOP uses swf as the name says, right?
12:05.26viperdudeukyes
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13:16.32stoffell_hhm, is "avoiding initial deadlock for sip/..." a bad thing?
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13:20.19gaupestoffell_h: I'm seeing it when a phone does a redirect, at the same time I got error messages all over the console
13:21.45stoffell_hgaupe, ah, weird. I have this happening on an asterisk 1.2.7 box.. could be related to this maybe: http://bugs.digium.com/view.php?id=7004&nbn=6
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13:23.45gaupestoffell_h: have seen that one with call parking and unattended transfers, but that bug is supposed to be fixed at 1.2.7
13:24.41gaupethis is what I'm seeing - http://bugs.digium.com/view.php?id=4101
13:27.15stoffell_hoh, that's a different one i guess, yep
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13:31.20frk2whatsup people
13:31.28frk2whos working on a sunday? :)
13:32.01frk2guess its just me :)
13:32.57frk2i just wanted to tell everybody of my revived sudden faith in pa1688 based phones
13:33.16lesouvageI added exten => 300,1,MeetMe(,MD,)  to my configuration. The idea is to be able to open a conference room with a number of choice and be prompted to also add a pincode to the conference.
13:33.39lesouvageThe problem is that I'm not prompted to add a pin code to the new conference room.
13:34.45lesouvageIs this supposed to work (starting new conference room if added number does not exist and be asked to add a pin on the new conference room)
13:38.25frk2is anybody using the atcom's and has this sudden 'silence' in the middle problem?
13:38.33frk2the phone just stops for like a second
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13:41.23frk2it dont happen in the older firmware
13:41.30frk2the newer firmwares seem to have this issue though
13:42.32Zeeekfrk2 see the yahoo group on PA1688
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13:43.08frk2i did zeeek
13:43.14frk2seems like a unresolved issue
13:43.23Zeeekgo back to older firmw
13:43.27frk2i hate these yahoo groups.. such bad navigation
13:43.31Zeeekyeah
13:43.42Zeeekbut that one's so quiet it don't matter so much
13:43.44frk2zeeek i cant find the older firmware anyways
13:43.54frk2do you know where i could get it from?
13:43.59Zeeekheh you should have been saving them all
13:44.01frk2aredfox only has the latest one
13:44.03frk2hahah
13:44.06frk2which one do you use?
13:44.11Zeeekwhich do you need?
13:44.22ZeeekI think .49 but I don't have the phone here
13:44.30frk21.49 has issues too
13:44.37frk2the SIP version atleast
13:44.43ZeeekI don't use the phones much (I have three of them)
13:44.53ZeeekSIP? Who uses SIP on those IAX phones?
13:45.06Zeeekshit if I want SIP I'll use a real phone :)
13:45.12frk2hahah
13:45.14frk2hmmm
13:45.22frk2so you saying their iax stuff is better than sip?
13:45.25ZeeekI think I tried SIp once to see if it worked
13:45.32Zeeekno it's IAX though
13:45.44ZeeekI never used SIP with those things. The main advantage is they do IAX
13:46.01frk2true
13:46.06frk2the phones actually arent that bad
13:46.21frk2the voice quality is okkkk.. nothing great. but what do you expect for $40
13:46.28frk2they dont crash much
13:46.43Zeeekdepends on the phone. But the idea of compilable firmware is cool
13:46.44frk2and look decent
13:46.59Zeeekall PA1688 are not the same
13:47.14frk2the atcom's have a pa1688S
13:47.25frk2im guessing all pa1688S's are essentially the same
13:47.27frk2no?
13:47.32Zeeekthe white atcom?
13:47.36frk2black atcom
13:47.42Zeeekof course they're not!
13:47.50Zeeekdifferent keypads and speakers
13:47.53frk2oh yeah
13:47.57Zeeekand plastic case
13:47.57frk2well other than that i mean
13:48.04frk2functionality wise
13:48.12Zeeekthose are the most important features of most phones
13:48.28frk2none of my clients mind the quality of the black atcom much
13:48.30Zeeekthe atcoms I have, white ones aren't the greatest but they work
13:48.44frk2the white ones look too chinese
13:48.45frk2:)
13:48.46Zeeekthe black which I think is a yixin
13:48.49Zeeekis cool
13:49.05Zeeekthe women prefer the white ones... that's what cracked me up
13:49.10Zeeekat the office
13:49.13frk2hahahah
13:49.15frk2really?
13:49.18frk2i hate the white ones
13:49.20ZeeekI like the black one
13:49.29Zeeekwhat is the model number of your black phone?
13:49.35frk2if i could only solve the 'voice gap' issue on these phones.. i think they're a good bargain
13:49.38frk2AT-320
13:49.42frk2i think you have the AT-323
13:49.48ZeeekI think I have the same in white
13:50.00frk2no no- hang on
13:50.10Zeeekactually mine arenet branded atcom but I guess there about the ame
13:50.19frk2'oh yeah
13:50.24frk2did you get it for $40
13:50.32frk2thats the price atcom sells t
13:50.33frk2at
13:50.37frk2$35 in wholesale
13:50.45ZeeekI've had it for a long time. At that time they were $50
13:50.48frk2ive been able to get them down to $32
13:50.53Zeeek$60 with a HUB*
13:51.03frk2oh this is $32 with a hub
13:51.09ZeeekI don't need a bunch of cheap phones at the moment
13:51.11frk2i ordered 150 of them
13:51.33ZeeekI wonder what 150 Sipuras would cost?
13:51.37frk2exactly
13:51.41frk2thats the problem
13:51.50frk2in a third world country, the cost of phones is unbearable
13:51.53Zeeekwhat? Double? They're 5 times better
13:51.55frk2i need to find a cheap ass phone
13:52.04frk2Sipura phones?
13:52.15ZeeekI think it's more that no one gives a shit about whether the user is comfortable
13:52.37Zeeekanyway, working for $20/month, how comfortable ya need to be?
13:52.56frk2you mean $200/month
13:53.06Zeeeknot that much difference
13:53.12frk2lol
13:53.24Zeeekunless an appartment is $25/month
13:53.39frk2hmm
13:53.40Zeeekanyway, those are good starter phones but that's it IMO
13:53.48frk2i know
13:53.53frk2you using 841s?
13:53.59Zeeeklots of fun to mess with too
13:54.03*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
13:54.18ZeeekI have one 941 - it's very good speaker sound
13:54.27Zeeekboots in about 10 seconds
13:54.39frk2see my prob is that 941 is like $130?
13:54.46frk2for that price i'll just get a aastra
13:54.48Zeeek150 of them might be a lot less
13:54.57Zeeekaastra is supposed to be decent
13:55.05frk2aastra is awesome
13:55.10ZeeekGrandstreams are "deprecated" now
13:55.24Zeeekthe first one I had was decent though, worked fine
13:55.26frk2ive had too many issues with the GXPs
13:55.39ZeeekI'm talkin BT100
13:55.43frk2The 102 is okay
13:55.51frk2but then its not too much better than the atcom
13:55.57frk2similar build/voice quality
13:56.29Zeeekis anyone using IAX providers ?
13:56.37robin_szsip phoens will be like $30 within 12 months
13:56.37frk2i use voicepulse
13:56.49frk2robin i hope so
13:56.50Zeeekyeah me too, works ok
13:56.55frk2that'll jump start the industry
13:57.07Zeeeklook at the price of cellphones!
13:57.08frk2the main problem is the cost of a phone
13:57.11robin_szfrk2, you just have to watch whats happening with the Big Players ...
13:57.27frk2zeeek cell phones are personal items
13:57.31robin_szeven BT are pushing VOIP now ...
13:57.33frk2ip phones are usually corporate items
13:57.55Zeeekthe point is, sell the service and give the phone away. Coming soon to all providers
13:57.55robin_sznah, ip phoens are now corportae, but by 12 months will be domestic
13:58.07frk2oh definitely
13:58.13Zeeekmany people have them in the us along with voip routers
13:58.14frk2i would wanna give a low cost phone away :)
13:58.18robin_szevery broadband user will have some
13:58.36frk2yes
13:58.38frk2death of the PSTN
13:58.45frk2sad demise of the POTS network
13:58.47Zeeekthe best use of those atcoms is actually to take it in your suitcase and plug it in at the hotel
13:58.49frk2i would be so freakin happy
13:58.54Alystairhmmmm
13:59.08AlystairSo what's a good brand to go with? Polycom?
13:59.13Zeeekwhat about junction networks (for IAX providers) ?
13:59.14frk2how long do you think it'll be before the PSTN dies off completely
13:59.24Cybertoyzeeek, yeah ... except in most hotels you have to go through some web-page before you actually have access to the internet.
13:59.28frk2Polycom is the safest bet
13:59.30Cybertoyhow do you get around that?
13:59.32ZeeekPolycom is good but getting up in the price scale
13:59.35frk2aastra is damn good too
13:59.40ZeeekCybertoy
13:59.53frk2yeah
13:59.54ZeeekCybertoy I didn't have that problem in Madrid
14:00.01frk2and some hotel idiots dont even NAT
14:00.14ZeeekNAT has nothing to do with IAX
14:00.14frk2or give good bandwidth
14:00.15robin_szgxp2000 has been a big disappointment for me .. the firmware upgrade is a disaster
14:00.23frk2NAT needs to work man
14:00.32Cybertoyzeeek, I had that experience in many hotels in Zurich, London, Paris, Tokyo, Singapore, Orlando ...
14:00.37frk2im sure the hotel aint providing you a public IP
14:00.38Zeeekthere were 200 people talking into phones in tat Madrid hotel on the same connection!
14:00.40Cybertoyjust to name a few where I stayed in the last 6 months.
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14:01.00frk2robin
14:01.01Zeeekyour don't need a public ip
14:01.02frk2it is
14:01.04Cybertoyin USA many hotels now stopped charging...
14:01.11Cybertoybut you still have to go through a web-page ...
14:01.13frk2my GXP's crash all the time if the call load increases
14:01.23frk2yes zeek, then you need NAT
14:01.26Cybertoyin Europe many want to charge... through ipass or credit-card.
14:01.34frk2some hotels ive been to WONT nat.. they'll want you to use a proxy
14:01.49Zeeekfrk2 I have used this phone with no problem with NAT ever
14:02.22frk2zeeek yes- no issues.. but you NEED NAT
14:02.59ZeeekI see what you're saying. I've been lucky I guess
14:03.13Cybertoyalso many WiFi hotspots are like that...
14:03.26Zeeekwifi is a whole n'other subject
14:03.26Cybertoyso a small phone with some small built-in browser would be nice.
14:03.32frk2the worst thing is they wanted to charge $25/hr in Zurich airport :)
14:04.07Cybertoyfrk2, yes... but if you get close enough to the business class lounge you can pick up their signal ... and there it's free.. :D
14:04.19frk2dude i was in the business class lounge!!!
14:04.28Cybertoyhmm.. then they changed it.
14:04.37Cybertoyyear ago that was the case.
14:05.15frk2but who ever pays that much?
14:05.39CybertoyI wouldn't... but it seems like people do.
14:05.48Alystairmy wifi connected printer prints out money!
14:06.00Cybertoysomeone mentioned the junxion box?
14:06.09Zeeekno junction networks
14:06.47Cybertoyah ... I am looking at getting the junxion box for my car...
14:06.50Cybertoyand then a wifi phone.
14:06.55tzafrirCybertoy, how much would a minimal PC cost?
14:07.17Cybertoytrafrir, for what?
14:07.23Cybertoytzafrir that is
14:07.24ZeeekI thought a junction box was a thing with screws that put two fat wires together
14:07.44Cybertoyno ... www.junxionbox.com ... it's for wireless high speed broadband internet.
14:08.08Zeeekso cellphone in the left hand and keyboard in the right? None on the wheel?
14:08.16Cybertoylol
14:08.35Zeeekwhat'you, Britney Spears?
14:08.43Cybertoylatin blood... :)
14:08.54ZeeekCybertoy kinda says it all
14:09.52frk2hahahahah
14:10.41poisonerhmmm
14:10.43frk2dude... when would GPRS be low-latency enough for VOIP?
14:10.53Zeeekanyone in the USA at the moment?
14:11.01CybertoyEV-DO or EDGE ... no gprs...
14:11.03poisonerfrk2: try UMTS with HSDPA
14:11.09CybertoyZeeek, I am.
14:11.12Zeeekfunny, I was just about to ask a GPRS question
14:11.26ZeeekDo any cell companies do GPRS now?
14:11.36Cybertoyyeah ....
14:11.36frk2zeek we share the same chain of thoughts.. atcom, gprs :) lol
14:11.38ZeeekI just bought a quad-band phone
14:11.39*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
14:11.43frk2in pakiland they do.. all of then
14:11.51Zeeekin the USA?
14:11.54Cybertoyyes.
14:12.08Zeeekbecause I've seen some web pages that say few providers do GPRS
14:12.22CybertoyCingular has it ... that's the one I'm on ...
14:12.24ZeeekI need to buy a SIM when we're over there in the summer
14:12.30frk2im sure last time i was in the US cingual/tmobile had it
14:12.31ZeeekHow are they?
14:12.32CybertoyI would guess that Sprint does it as well.
14:12.36Cybertoyand T-Mobile
14:12.44CybertoyI'm quite happy with Cingular.
14:12.45ZeeekSprint? The used the okld crap system
14:12.51frk2sprint uses something else
14:12.55frk2not gprs
14:12.57ZeeekI wonder if Cingular sells prepaid SIM?
14:13.02Cybertoythey do
14:13.03frk2its cdma 1xRTT i think
14:13.09ZeeekPCS is the old system right?
14:13.14frk2since they aint gsm phones
14:13.28ZeeekI'll have a look at Cingular
14:13.37frk2man.. why cant be just have a BIG HUGE 802.11b network
14:13.41frk2that would be quite awesome
14:13.43Cybertoyzeeek, did you look at www.united-mobile.com if you travel a lot?
14:13.51Zeeekno but I will, thx
14:13.58Cybertoyno roaming cost to many countries.
14:14.05ZeeekI have seen a few sites that have a bunch of info
14:14.24Zeeekit was really co,nfusing at first with all those different bands and standards
14:14.35Zeeekway worse thazn voIP :)
14:15.00Cybertoyjunxion box and wifi voip phone...
14:15.03Cybertoyway to go... :)
14:15.13Zeeekgoogle should habe a swith "NOT ebay"
14:15.25frk2junxion
14:15.29frk2let me check that shit
14:17.22frk2yo cybertoy
14:17.34frk2what speeds can you get?
14:17.41frk2what latency to the first router?
14:17.49Cybertoyno idea... I'm trying to find out more about it as well ...
14:18.01frk2damn
14:18.12Cybertoybut sure sounds sexy...
14:18.17frk2even if it gives 64/128k with less than 100ms its pretty usable for voi
14:18.18frk2voip
14:18.28Zeeekunited-mobile calls are very expensive
14:18.30frk2however
14:18.44blitzragemorngin all
14:18.47ZeeekUS to US nearly $2/minute?
14:18.48blitzragemorning even
14:18.48frk2coolest thing is nokia is coming out with phones, ordinary phones, that would have WiFi and a SIP phone built in
14:18.51Zeeekhi blitz
14:18.54Cybertoyzeeek, frigg that...
14:18.58blitzrageZeeek: hey ho!
14:19.03ZeeekI don't get it
14:19.23blitzragefrk2: yah --- any idea if it comes with a mini-browser too?
14:19.25Zeeekwhat's the point? I need to just buy a US SIM
14:19.26Cybertoyzeeek, I think that only makes sense if you receive many calls... but not make them.
14:19.37Cybertoyzeeek, and USA is not in the free roaming list.
14:19.38blitzragewifi phones with no mini-browser are useless imho
14:19.43Zeeekin Europe recd calls are free anyway
14:19.51ZeeekCybertoy ok I see now
14:19.53Cybertoyzeeek, that's right... but not in the usa.
14:20.03frk2blitz.. dont matter.. if I can just use a cell phone to register to asterisk and make gsm calls too thats kick ass
14:20.10frk2eliminates the need for separate IP phones
14:20.18Zeeekso, folks, who sells USA SIM cards as prepaid with no locked phone?
14:20.32Cybertoyebay
14:20.37ZeeekCingular doesn't apper to by the way
14:20.38blitzragefrk2: yah, but if you'r ein an airport or public wifi and want to save on the cell costs, a mini-browser is necessary to login
14:20.53frk2oh true
14:21.01Cybertoyzeeek, they do .. I have friends that use cingular... but they don't sell an unlocked phone with it.
14:21.03frk2but dude... im just looking at benefits in the office
14:21.09frk2even those are amazing
14:21.15ZeeekI want the phone to do GPRS because it has a built in email client. That was the whole point of using it
14:21.19frk2employees have their own cell phone + sip phone
14:21.45ZeeekCybertoy ? you mean you buy a phone and put the SIM in your international phoen?
14:21.47*** join/#asterisk Dimitripietro (i=Wut@modemcable017.237-202-24.mc.videotron.ca)
14:22.17CybertoyZeeek, you can't just buy a sim card??
14:22.34Zeeekthat's what I'm asking about. I don't see the possibility on their site
14:23.08DimitripietroI'm using a TDM400P and i'M getting some background noise when using it. Anyone could gice me some trick too look at ?
14:23.24Zeeektake a good look at cables and phones first
14:23.46DimitripietroPolycom IP phone
14:24.05DimitripietroIf i'm doing a SIP TO SIP, no problem
14:24.32Zeeekwhat kind of noise?
14:24.38Zeeekcrackling?
14:24.55Zeeekcould be IRQ conflicts, bad cable or wall connector
14:24.56DimitripietroNot cracking, constant noise
14:25.02Zeeekhummmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm?
14:25.19Zeeekapplause?
14:25.28[TK]D-FenderDimitripietro : Could be an impedence issue or grounding.
14:25.38Zeeekyes, or card placement
14:25.54Zeeeksometimes video cards interfere with others on the bus
14:25.57DimitripietroIs there a way in Linux to manually mange the IRQ ?
14:26.05[TK]D-FenderYeah, general outside interference is a possibilty
14:26.16Zeeekit's done in the BIOS
14:26.18[TK]D-FenderDimitripietro : Not really.  Check your BIOS
14:26.37DimitripietroThere is no options in the Bios ...
14:26.37[TK]D-FenderDimitripietro : Do a "cat /proc/interrupts" and pastebin it.
14:26.38[TK]D-Fender~pb
14:26.40jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
14:26.42*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
14:26.43coppiceconstant noise sounds like my kids
14:26.50coppiceor my mother in law
14:27.22Cybertoyzeeek, I can't see it on the web-page either.. but just go to one of their stores...
14:27.30ZeeekI FOUND IT YAY!
14:27.43Zeeekonly in California, but that's where I'll be! YES!!!
14:27.52Zeeekhttp://www.cingularwireless.com/download/Prepaid%20Brochure%20(2%20MB).pdf
14:28.01Cybertoywhat a scam
14:28.11Dimitripietro<PROTECTED>
14:28.20Zeeekshit, no that's prepaid wireless
14:28.25Zeeekdamn
14:28.32DimitripietroThe tdm is not sharring IRQ
14:28.47Zeeektry another slot if you can
14:28.55DimitripietroAlready tried
14:28.59Dimitripietrosame noise
14:29.03[TK]D-FenderDimitripietro ; hmm, made sure your loadzone and so on are rigtht ro your area?
14:29.05ZeeekI'm out of suggestions
14:29.19Dimitripietroloadzone zapata ?
14:29.32Cybertoynedd to get breakfast... see ya all.
14:29.46Zeeekprepaid wireless is a cellphone? What is this marketing giberish?
14:29.49Zeeekbye
14:29.52[TK]D-FenderDimitripietro : zaptel
14:30.07Cybertoyzeeek, ah .. yeah .. they call mobile phone telephony here wireless... don't confuse it with wifi
14:30.23Zeeekwell the news is good then despite the terminology
14:30.27Zeeekbon appétit
14:30.30Dimitripietroloadzone=us and i'm in canada
14:30.32Cybertoymerci
14:30.34Dimitripietroshould be fine
14:30.52Zeeekloadzone=timbuktu
14:31.10Zeeekworks fine but calls are very expensive!
14:31.33[TK]D-FenderDimitripietro : You on DSL?  Got it filtered before entering the card?
14:31.48Zeeekgood call; that'd do it
14:32.13[TK]D-FenderNo, he's on cable...
14:32.23[TK]D-Fenderhmm
14:32.30Zeeekheh well, strike five for the debug team :)
14:32.38[TK]D-FenderI'm not done yet...
14:32.52Dimitripietro<[TK]D-Fender> I can send you a wav file
14:33.02[TK]D-FenderDimitripietro : Try plugging in-line with a powerbar telephone surge suppressor
14:33.14[TK]D-FenderDimitripietro : Its jsut a loud hum right?
14:33.25Dimitripietroyep
14:33.52[TK]D-FenderDimitripietro : Yuo can also try changing the module's position on the card.  Have seen a few that flaked out because of where it was.
14:34.12[TK]D-FenderMine was DOA till i swapped the FXS & FXO modules for some unknown reason.
14:34.46DimitripietroOk, I,M goona try the power telephone filter (have one) first
14:35.24Dimitripietro<[TK]D-Fender> May I send you a recording of the problem ?
14:36.02[TK]D-FenderDimitripietro : if you wish
14:37.00DimitripietroHave you received the send request ?
14:39.45[TK]D-Fenderwell.. that failed :)
14:39.47[TK]D-FenderPM
14:40.26rpmkudos to whoever wrote that telemarketter script, now i just need to find a way to get a list of telemarketters numbers.
14:41.13Dimitripietrosent
14:41.38Alystairyay RDC to VMWare to CentOS
14:46.00[TK]D-FenderDimitripietro : Thats just your breathing normally during a call with both sides mutually quiet?
14:47.46Dimitripietroyep
14:48.07[TK]D-FenderDimitripietro : Whats your rxgain at currently?
14:48.12[TK]D-Fenderin zapata
14:48.24Dimitripietro8
14:48.27Dimitripietrotx: 0
14:48.41[TK]D-Fenderlower to 0.0 across the board
14:49.11DimitripietroWhen the gain is at 0.0, the sound is to low
14:49.14Dimitripietrotoo
14:50.21saftsackare any chan_capi users here?
14:50.52[TK]D-FenderDimitripietro : try to scale it back a bit then and raise the default volumes on your phones
14:51.46*** join/#asterisk esculapio_ (n=ESCulapi@66.98.18.236)
14:52.11*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
14:52.34DimitripietroK, give me 5 minutes, I just changed the order of the FX0 modules
14:59.20*** join/#asterisk \home\Carloz (n=gnunix@unaffiliated/packet)
14:59.31*** part/#asterisk \home\Carloz (n=gnunix@unaffiliated/packet)
14:59.48ManxPower[TK]D-Fender, Be careful or you will become known as "the nice version of ManxPowr"
15:00.12*** join/#asterisk isaiah (n=test@208-187-93-4.br1.hnv.mi.frontiernet.net)
15:00.51[TK]D-FenderManxPower : Yin to your yang :)
15:01.30Dimitripietro[TK]D-Fender : rx:0 and swaped the order of modules. The sound is still there but lower. I will now try to raise the sound in the polycom configuration files
15:01.39*** join/#asterisk ramo (n=ramo@59.92.141.2)
15:02.12[TK]D-FenderDimitripietro : Better than doing it per-call.  It amps better on the phone than doing it on the card.
15:02.44*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.150.Dial1.SanJose1.Level3.net)
15:04.46tekatiexten => _*7.,1,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${ARG1:2},60)
15:05.07frk2guys
15:05.31frk2why do i have to adjust my txgain/rxgain to ridiculously negative numbers for proper echo cancellation on PRIs?
15:05.32tekatiWhy when I dial *7 things appear to work right but when I dial *7*?????? as soon as I hit that second * it gives me a busy in the phone?
15:05.54frk2tekati some phones treat * as 'send' or 'dial'
15:06.14frk2sorry thats hash
15:06.22frk2ignore what i just said
15:06.23frk2:)
15:06.54frk2i mean is a txgain of -15 and rxgain of -14 normal?
15:07.05frk2or is my telco doing funny stuff
15:07.10Zeeektekati what phone?
15:07.39*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.150.Dial1.SanJose1.Level3.net)
15:07.46tekatiI use a cordless phone with a LinkSys PAP2
15:08.01Zeeekthe phone it self may use service codes
15:08.28tekatiInteresting.  Let me look at the PAP2 config.  See if there is anything in there.
15:08.37Zeeektry looking at the web page of the linksys (I assume it show a page?)
15:08.51Zeeekmany SIP phones have a table of service codes
15:09.55[TK]D-Fendertekati : You are using ARG1 where you should be using EXTEN.  Looks like you ripped that off some guy's macro...
15:10.24tekatiSorry I changed it using my own macro to make sense here.  I do use Exten and Arg in the right places my bad.
15:10.47tekatiI think I found the issue in the PAP2 config.  It has a Dial Plan string: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
15:10.50[TK]D-Fendertekati : Pastebin the whole thing and everything related to it (except the constant definitions
15:10.53[TK]D-Fender~pb
15:10.56jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
15:11.18*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
15:11.47[TK]D-Fendertekati : Try the much simpler : (*.T,#.T,x.T)
15:12.10CunningPiketekati: We have a PRI with a gain of -10dB - if the numbers you are using work for you, use them :)
15:12.39tekatiCunningPike: that was meant for frk2 I think.
15:12.45CunningPikeOops - sorry
15:12.47[TK]D-FenderGain should be 0.0 across the board.  Anything else is typically a sad last attempt to alleviate echo...
15:13.12tekatiFender: Is that for the dial plan for the PAP2?
15:13.22[TK]D-Fendertekati : yup
15:13.38[TK]D-Fenderbasically "take anything I feel like dialing and jsut do it"
15:14.06frk2Fender thats what i thought
15:14.14[TK]D-FenderReal dialplans on en-points can be an adminstrative PITA.
15:14.25frk2but unless txgain/rxgain are BOTH less than -14, echo happens
15:14.26Dimitripietro[TK]D-Fender : with gain 0.0 there is a huge differnce in the volume of a sip to sip call and a sip to pstn call throught TDM
15:14.34[TK]D-FenderJust let the phone do "whatever" and control it on the * level
15:15.02[TK]D-FenderDimitripietro : hmmm yeah... sip - sip would need to be adjusted. work something out in between.
15:15.32tekatiI agree.  I am still having the same issue after switching both Line1 and Line2 to the (*.T,#.T,x.T) for the Dial Plan.  Must be another setting somewhere.
15:15.37[TK]D-Fenderfrk2 : Try all the other settings before ever touching gains....
15:15.51frk2Fender there arent much settings
15:16.04frk2echo is being created on the PRI box
15:16.16frk2i need to cancel it there
15:16.24frk2MG2 helps, but not without gain adjustment
15:16.24[TK]D-Fenderfrk2 : echocancel=[taps], echotraining, recompile Zaptel with several other options...
15:16.44[TK]D-Fenderfrk2 : Maybe its time you get a better PRI card.
15:16.46frk2yes tried that.. what other options? I've tried MG2
15:16.59frk2I got a digium TE 110P
15:17.01[TK]D-Fenderfrk2 : MG2 is DEFAULT isn't it?
15:17.07frk2no KB1 is default
15:17.10frk2MG2 is the new one
15:17.17frk2supposed to be more effective on PRI cards
15:17.19ghenryHi, is it a Tele provider or Asterisk dialplan that allows someone to dial 01358 279644, when the internal extension is 44 and the main line is say 01358 279600
15:17.36frk2and it is
15:17.41frk2but not without gain adjustment
15:17.52ghenrySo each internal extension can be reached from outside the company
15:17.57[TK]D-Fenderfrk2 : Oh the joys of software echo cancellation....
15:18.13frk2heheh
15:18.20frk2maybe my telco is amplifying
15:18.29frk2cuz at gain = 0, the voice is REALLY loud
15:18.35frk2to the point that MOH hurts
15:18.44[TK]D-Fenderfrk2 : Could be.....
15:19.08frk2I say damn the PSTN to hell
15:19.11CunningPikefrk2: We dicked around with software settings for months before finally purchasing a Ditech echo can
15:19.15CunningPikeNo more ecbho
15:19.25frk2ditech echo can?
15:19.26CunningPikeExcept when I type :S
15:19.41CunningPikeDitech make carrier-grade echo cancellation boxes
15:19.45ManxPowerCunningPike, We got Tellabs EC off eBay.  Works great.
15:19.52[TK]D-FenderI did the same for about a month before switching things at my company as well.
15:20.12CunningPikeManxPower: We have 4 PRIs, so the Ditech is actually cost-effective at that scale
15:20.14[TK]D-FenderManxPower : A bit of a PITA to set up, but works decent right?
15:20.44frk2Pike the new quad digium hardware echo can cards are really good
15:21.28[TK]D-Fenderfrk2 : I've heard plenty of mixed reviews at least about their 1st gen.  Not as much "in the know" about the newer VPM though.
15:21.39CunningPikeNot in our experience, I'm afraid - we found them to be worse than MG2, and they introduced audio drop outs
15:21.43ManxPower[TK]D-Fender, Tellabs?  Yes.  About $250 - $300, for the shelf and 20 or so EC cards and a -48V power supply.  The docs SUCK, but it works very well.
15:22.13ghenryAnyone? ;-)
15:22.17frk2thats weird man
15:22.24[TK]D-FenderManxPower : Yeah I think I read a few things on the WIKI about that... looked scary.
15:22.24frk2we got the newer tdm400 and te411p
15:22.28frk2they are awesome
15:22.38frk2tdm2400
15:22.40frk2sorry
15:22.48frk2the echo cancellation was pretty decent
15:22.49ManxPower[TK]D-Fender, I have the official tellabs docs in PDF format.....
15:23.43frk2so basically there are others running very negative gains
15:23.47Dimitripietrofrk2 : Are you using hardphone with your tdm2400 ?
15:23.59frk2yup
15:24.03CunningPikeghenry: It is your provider - you need to purchase DIDs from them
15:24.07frk2basically the tdm2400p is a failover for the PRI
15:24.35*** join/#asterisk miguel3239 (n=miguel32@ns1.nashuacs.com)
15:24.37Dimitripietrodid you needed to turn software echo canceller on even with the hardware echo canceller ?
15:24.49*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
15:24.58frk2hmmm
15:25.09frk2it is actually on now that i think of it
15:25.18frk2will try turning it off
15:25.18CunningPikeDimitripietro: The VPM disables the software EC
15:25.25frk2however echo training refuses to work
15:25.49CunningPikefrk2: That's because the software EC is disabled
15:25.55frk2yeah probably
15:26.01frk2but the echo cancellation was pretty good
15:26.43DimitripietroWith my tdm2400p and hardware echo canceller, if I turn in zapata echocancel=no then I'm getting echo in my polycom phone
15:27.05frk2hmm
15:27.14DimitripietroI needed to turn echocancel=yes to eliminate the echo
15:27.16frk2ill double check
15:27.41[TK]D-FenderManxPower : An eve those suck huh?
15:27.55ManxPower[TK]D-Fender, correct.
15:28.01[TK]D-FenderDimitripietro : Thats the NORMAL thing to do regardless.
15:28.03*** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
15:28.31DimitripietroOk
15:28.46*** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
15:28.59*** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
15:29.02DimitripietroEven with an hardware echo can, we need to turn the echocancel=yes
15:29.08*** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
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15:29.57*** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
15:33.54*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
15:34.16ManxPowerDimitripietro, if you need to do echocancel=yes with external EC then the external EC is not working correctly.
15:35.04[TK]D-FenderManxPower : I believe for the hardware he's using "echochancel=yes" just actives the HWEC, not soft
15:35.33frk2thats what i thought
15:35.39frk2but theres no way to check i guess
15:36.27ManxPower[TK]D-Fender, so he is using the VPM addon board for the Digium card?  If so, then echocancel=yes enables the EC on the daughter card.  If he's using an actual external box, then my previous statement stands.
15:36.57ManxPowerI just don't trust Digium's EC.
15:37.28[TK]D-FenderManxPower : Careful... the "Man" is watching! ;)
15:37.37frk2hahahah
15:39.51*** part/#asterisk bmonty (n=bmontgom@ubuntu/member/bmonty)
15:46.47*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
15:50.04*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:50.38poisonerhas anyone linked succesfully asterisk mit an older Cisco CM?
15:50.48DoktorGregarrghhh
15:50.54DoktorGregsip phone...
15:50.58DoktorGregno matter what i do
15:51.03DoktorGregif i let it sit for while
15:51.06DoktorGregor reboot it
15:51.13DoktorGregfirst call
15:51.17DoktorGregcant hear anything
15:51.25DoktorGregbut can talk
15:51.28DoktorGregafter that
15:51.30DoktorGregworks fine
15:51.32AlystairDoktorGreg: I'm doing the unthinkable and trying a VM setup ;)
15:51.33Dimitripietro[TK]D-Fender : If I use the zapbarge fuction on one of my fxo, i'm hearing the same noise even if my line isn't plugged in the fxo
15:51.46DoktorGregcant hurt to try
15:51.49Alystairyep
15:51.59DoktorGregI thought you found an old smp p3 system?
15:52.00DimitripietroSo it's sure the noise is coming from my box
15:52.10AlystairI did but this monster machine is going on unused :(
15:52.42DoktorGregits not going unused
15:52.47DoktorGregit is your domain server
15:53.11Alystairso, nothing important :)
15:53.28tekatiWow LinkSys guards the Admin Guide to PAP2 like it is their pot of GOLD!!!
15:53.32AlystairOh, also our profile server and central backup
15:53.34DoktorGregarnt you using roaming desktops?
15:53.40tekatiDoes anyone have the Admin Guide for the PAP2-NA?
15:53.52AlystairDoktorGreg: yeah this machine is basically the heart of the company's office hah
15:54.04[TK]D-FenderDimitripietro You might jstu have a defective FXO module which does happen... call up Digium, explain everything you've tried and they may take you througha few more tests, but ultimately I believe thy my change your card.
15:54.24DoktorGregwell again, if all your are useing is its samba functions, nuke the os and put samba on it
15:54.48DoktorGregsamba works better as windows fileserver than windows does anyhow
15:55.17DimitripietroOn your side, the sound using a analog card is clear ?
15:55.37AlystairDoktorGreg: including security settings?
15:55.42Alystairand rights management?
15:55.42DoktorGregyah
15:55.44Alystairhrm
15:55.48DoktorGregread about samba
15:55.55DoktorGregalso
15:56.00[TK]D-Fender<PROTECTED>
15:56.01DoktorGregno CAL's in future
15:56.13DoktorGregyou can also set samba up as domain servers
15:56.14DimitripietroOk
15:56.26ghenryCunningPike: Thanks. Will look into it ;-)
15:56.46CunningPikeghenry: np
15:56.59ghenryCunningPike: What does it stand for?
15:57.20CunningPikeDID? Direct Inward Dial
15:57.39ghenryah, yes. Thanks
15:58.24ghenryIsn't it DDI for Europe/UK?
15:59.01CunningPikeIt may well be
15:59.07ghenrythought so ;-)
15:59.14ghenryMaybe be a bad idea anyway for security
15:59.14*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
15:59.24CunningPikeBad for security? How?
15:59.29ghenrybetter going through IVR or Receptionist
15:59.30tekatiAnyone have a Admin Guide for the PAP2?
15:59.32ghenrybbl
16:03.21*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
16:03.21*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
16:04.08brif8Reviewing log files I see this SIP OPTIONS (404 not found) asterisk@my.voip.provider to s@my.ip.address:5060,  this is not the register as I see that also ?
16:04.27ManxPowerDID and DDI are just different terms for the same thing
16:07.22brif8any ideas
16:09.01ManxPowerbrif8, that means that for some reason your provider is trying to send an options packet to extension "s" on your server.
16:09.21brif8ManxPower: what is an options packet ?
16:09.43*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
16:10.08ManxPowerbrif8, it can be used for a zillion things.  Asterisk uses them when you use qualify in sip.conf
16:10.48brif8ok I have qualift = yes to my voip provider, would it occur roughly every minute
16:10.59brif8qualift => qualify
16:11.13ManxPowerbrif8, I have no idea
16:11.21brif8ok
16:12.35brif8thanks anyways
16:16.20*** join/#asterisk Seyr (n=Seyr@grant254.grantgeo.com)
16:17.18SeyrI have an * server with 25 7960 phones using SIP and they want one to have a 7914 module. From what I read, the 7914 requires SCCP?
16:17.37QwellSeyr: yes
16:17.48SeyrAny problems having 24 7960 use SIP and one use SCCP for the 7914?
16:17.52Qwellno
16:18.07Seyrany "gotchas" i need to look out for?
16:18.54Seyrshould I seperate the TFTP for the SCCP from the SIP? or will the configs work side by side?
16:19.05Seyrthe xml and loads and stuff?
16:24.17tekatiIn the PAP2 dial plan Fender sent me (*.T,#.T,x.T) What does the T do does anyone know?
16:25.18ManxPowertekati, Timeout
16:25.20DimitripietroTimeout
16:26.28viperdudeukSeyr: we have had SIP and SCCP configs on the same TFTP before now without a issue
16:26.47*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:29.05DoktorGreghttp://video.google.com/videoplay?docid=109587088318033773&q
16:29.56Seyrthanks
16:31.13DoktorGregyargghhh
16:31.25DoktorGregok, ATA, sipura 1001
16:31.31DoktorGreglet it sit for while
16:31.37DoktorGregpick up phone, dial
16:31.42DoktorGregcant hear anything
16:31.46DoktorGregbut can talk ok
16:31.50DoktorGreghang up phone
16:31.55DoktorGregdial again
16:31.58DoktorGregall works good
16:32.04DoktorGregwait 10 minutes or so
16:32.06DoktorGregtry it again
16:32.11DoktorGregsame thing
16:32.11*** join/#asterisk ToTo (n=ToTo@host235-158.pool875.interbusiness.it)
16:32.20DoktorGregany idea where to look?
16:32.56ManxPowerDoktorGreg, sounds like you have NAT involved and are not using qualify=yes
16:33.17DoktorGregin sip.conf?
16:33.23ManxPowerof course.
16:33.41ManxPowerusually if you have nat=yes you want qualify=yes too
16:33.51DoktorGregi have qualify=200
16:34.00ManxPowerthat will also work
16:34.32DoktorGregso its something else then...
16:34.32ManxPoweryou're not doing any port forwarding on the NAT router on the remote side, right?
16:34.41DoktorGreglemme double check
16:34.45ManxPowerI assume Asterisk is on a public IP and the SIP client is behind NAT
16:34.50DoktorGregyup
16:35.00DoktorGregwell i take that back
16:35.08DoktorGregasterisk is on a one to one nat
16:35.27DoktorGregasterisk has its own internal ip address
16:35.32DoktorGreger
16:35.33ManxPowerAh.  I don't really feel like giving you a tutorial on running Asterisk behind nat.  Check the Wiki
16:35.52Zeeekvas ist das problem?
16:35.54DoktorGregevery other phone works no prob though... also behind nat
16:38.05*** join/#asterisk gursikh (n=guriskh1@adsl-69-151-246-132.dsl.hstntx.swbell.net)
16:41.42*** part/#asterisk Seyr (n=Seyr@grant254.grantgeo.com)
16:42.38*** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net)
16:57.54*** join/#asterisk Assid (n=assid@203.115.64.12)
16:58.03Assidheya
16:58.09Assidhows everybuddy doing?
16:58.29*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
16:58.40Zeeekwe are micromanaging the current adversity, thank you
16:58.40QwellAssid: poorly, since gaim got popular
16:59.31Assidhehe
16:59.44Assidi was waiting for someone to notice that
16:59.44*** join/#asterisk blackgecko (n=blackgec@201.152.14.187)
16:59.47Qwellin fact, it's dead
17:00.34blackgeckohi, anyone with a tdm2400p ??
17:01.38Assidso what you been upto ?
17:01.41*** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
17:02.21*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
17:04.47*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
17:04.54Zeeekwhat is the parameter that holds the timeout time for a sip register?
17:05.54Dimitripietro<blackgecko> me
17:06.15rdgztI have a weird SIP auth problem on register. With various softphones, I can't register, Asterisk says the password is wrong, although it's clearly correct.
17:06.23rdgztAnyone have any ideas?
17:07.07ManxPowerPASTE the error message.
17:07.22rdgztApr 30 02:06:04 NOTICE[14457]: chan_sip.c:10817 handle_request_register: Registration from '<sip:john@10.0.254.35>' failed for '10.0.2.2' - Wrong password
17:07.28*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
17:07.40ManxPowerAnd you have a [john] section in sip.conf?
17:07.45rdgztI'm following the examples from the Asterisk: The Future of Telephony book.
17:07.45rdgztYes.
17:08.03ManxPowerput the [john] section on pastebin.ca
17:08.41VoicePulseAnyone know if Asterisk::LCR in widespread use?
17:08.46filedo a lil' dance, get down tonight
17:08.47blackgeckodimitripetro; have you had any problem with it ??
17:09.33*** join/#asterisk rdgzt (n=joakim@201.137.86.15)
17:09.34blackgeckodimitripietro: have you had any problem with it ??
17:09.38rdgztSorry, got disconnected.
17:09.50rdgztI'll put the section on pastebin, hold on.
17:10.39Dimitripietroblackgecko : what kind of problem ?
17:11.15blackgeckostriped numbers from the dialed ones, worng number dialed, etc
17:11.37DimitripietroNo
17:12.06ManxPowerblackgecko, sounds like your telco does not start accepting DTMF fast enough.  Try prepending w or ww to your dial string to add a .5 or 1 second wait before sending digits
17:12.09blackgeckodimitripietro: cause i get random errors from my telco "the number you have dialed is incorrect"
17:12.14DimitripietroIs yor card sharing an irq with something else ?
17:12.24ManxPowerclassic problme, talked about at least once per month on the mailing lists.
17:12.40rdgztManxPower: http://pastebin.ca/52402
17:12.54blackgeckodimitripietro: no irq sharing, ive already added www to the dialed number, without luck
17:13.08ManxPowerblackgecko, paste the Dial line
17:13.09rdgztManxPower: It's my whole sip.conf, since it's very small, in case that's helpful.
17:13.39*** join/#asterisk ramo (n=ramo@59.92.141.2)
17:13.42blackgeckothe dial string is Zap/g0/wwwwww54234569
17:14.17blackgeckothe problem is the error isnt constant, it happens randomnly
17:14.40blackgeckodimitripietro: can you share your zapata.conf ??
17:14.56ManxPowerblackgecko, no, that is only part of your dial line.
17:15.21ManxPowerI need the actual dial line from your extensions.conf.
17:15.32ManxPowerrdgzt, looks good to me, other than you are not allowing any codecs.
17:16.30Strom_Cno no no, the answer is "dead hookers"
17:16.37blackgeckomanxpower using freepbx for the config, is it bad ?
17:16.39rdgztManxPower: I get the default set if I don't specify any, don't I? Besides, the error message pretty explicitly says that it's "Wrong Password"
17:17.05ManxPowerblackgecko, I cannot help you with FreePBX.  Did you not look at the /topic of this channel?
17:17.10rdgztAs I said, following the instructions from the Asterisk book, and I've tried this with two different softphones, both do the same thing.
17:17.14rdgztSo I'm a bit stumped here.
17:17.30*** join/#asterisk Nugget (i=nugget@dazed.slacker.com)
17:17.42Zeeekrdgzt are you cutting and pasting the password or username?
17:17.53ManxPowerrdgzt, what happens if you comment out the secret= line and do a reload?
17:18.01ManxPowerand not touch the config on the SIP device
17:18.36rdgztZeeek: The password is "welcome", and even though I'm not cutting and pasting it, I'm typing it very carefully, and have several times.
17:18.40rdgztManxPower: Let me see...
17:18.40blackgeckomanxpower, yeah but mi problem isnt with freepbx, ive done it without a gui, and with amp, and is the same problem
17:19.25ManxPowerblackgecko, Yeah, but FreePBX uses such complicated config files that nobody wants to spend the day or so working with you to understand your setup.
17:19.32ManxPowerAs I said, I cannot help you.
17:19.59blackgeckois this correct ??   exten => _ZXXXXXXX,1,Dial(Zap/g0/www{EXTEN})
17:20.53ManxPowerblackgecko, not even close.
17:20.53Strom_Cit's ${EXTEN} silly
17:21.13ManxPowerStrom_C, Obvious that he did not even paste the line.
17:21.23rdgztManxPower: If I remove the secret line, I don't get that error message, I get some sip debug output in asterisk, but not that exact error message, and then the softphone says "registration failed: timeout".
17:21.33Strom_CManxPower: well yeah
17:21.37ManxPoweralso that exten line will only match an 8 digit number
17:21.48Strom_Cthere's never a line that simple in freepbx
17:21.50ManxPowerrdgzt, you have me stumped.
17:21.54Strom_Cit's like [code_vomit]
17:21.55rdgztManxPower: Will putting that SIP debug output in a pastebin help?
17:22.17rdgztManxPower: Yeah, I'm stumped myself, it's had me scratching my head since yesterday.
17:22.21ManxPowerrdgzt, only if 1) I wanted to think hard on a sunday.  Someone else may be able to help you.
17:22.46Zeeekwhat the heck, go for it, there's a big audience out there
17:22.52blackgeckomanxpower: im willing to try it on a no gui asterisk but i think the problem isnt the gui
17:23.13rdgztAt this point I'm considering building and installing Asterisk from source, since I'm currently using the Ubuntu packages, in case there's something weird with the defaults or something in the Ubuntu packages.
17:23.23rdgztBut that's kind of grasping at straws.
17:23.50ManxPowerblackgecko, The line you pasted says Match an 8-digit number, the first digit matcing 1-9, dial out Zap group 0, wait 1.5 seconds, then dial the DTMF digits { E X T E N }  of which, of course there are no such DTMF digits.
17:24.08ManxPowerblackgecko, if you continue to try to convince me to help you with this problem I will put you on /ignore.
17:24.11ZeeekManxPower, I want to benefit from your long, hard earned wisdom (or anyone who can answer this) :
17:24.30Strom_CZeeek: the answer is "dead hookers"
17:24.34rdgzthttp://pastebin.ca/52408 is the SIP debug output, if anyone's interested.
17:24.44blackgeckomanxpower: sorry to bother you
17:24.53rdgztFrom when I removed the secret line in the config.
17:25.28ManxPowerrdgzt, be sure to mention what SIP device you use, it might be important
17:25.40ZeeekI thought is was 42. I'll have change that part of sip.conf
17:25.59rdgztManxPower: I've tried with the Xten softphone and Ekiga, both get the same result. Currently testing with Ekiga, since the Xten interface is horrible.
17:26.02stoffell_hit seems avoiding initial deadlock is not a "good thing" ? I have (http://pastebin.ca/52409) a LOT of those errors, just before my * server stopped doing anything.. any idea on how to troubleshoot ?
17:26.05Strom_C42 is The Answer.  different than just the answer :)
17:26.08*** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net)
17:26.21Zeeekand THe ansWer ?
17:26.41Strom_Cthat's "They spinnin' yo"
17:27.12Zeeekincidentally, X-Ten is great for testing SIP, but remember out there, it automatically uses STUN so it won't be the same on asterisk
17:27.36Zeeekunless you disable STUN in the X-Ten of course
17:27.48*** join/#asterisk IceManRISK (n=kart@201.14.2.169)
17:30.22rdgztThe softphone and the server are on the same network with nothing inbetween, shouldn't that make STUN irrelevant?
17:31.01Zeeekindeed
17:31.11*** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk)
17:31.18Zeeekthe comment is a general avertissement
17:31.57rdgztZeeek: Right, thanks. I guess that's why the Asterisk book recommends it. Although I insist that the interface is horrible.
17:32.48Zeeekit works well when you get used to all those clicks
17:33.22rdgztSoftware that tries to look like a physical object is generally a bad idea, but never mind, I'm sure that's not related to my problem. :)
17:33.53Zeeekaiming at the youth element
17:33.56Zeeek"skins"
17:34.04rdgztI'm in a bind here, I have a bunch of hardphones that I'm supposed to set up with this thing, but that seems like a tall order if I can't even get a simple softphone working. :)
17:34.11ZeeekI have seen the password issue somewhere
17:34.16*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
17:34.19PakiPenguinevening
17:34.21rdgztZeeek: Really?
17:34.26Zeeekbut where?
17:34.30rdgztZeeek: I tried to google, but I didn't have much luck.
17:34.39Zeeekyou've obviously googled it to death
17:34.39Zeeekheh
17:34.48Zeeekand looked on the mailing lists
17:35.46rdgztI haven't explicitly looked on the mailing lists, mostly because I assume that Google would search their archives, but...
17:37.25ZeeekI'd just look for password in the ml
17:37.25Strom_Crdgzt: what version of asterisk are you running?
17:37.25rdgztStrom_C: My package is 1.2.1.dfsg-4ubuntu1
17:37.27QwellStrom_C: I uploaded the code into my chan_skinny branch
17:37.28key2for what purpose people use asterisk with postgres ?
17:37.30Strom_Cok, time to upgrade.  we're on 1.2.7.1 now, you know
17:37.35Qwellteam/north/chan_skinny-fixup/
17:37.44Strom_CQwell: oooh
17:37.54Strom_Ci will totally have to play with it when I get out of bed
17:37.55rdgztStrom_C: Yeah, I was actually considering building asterisk from the latest source instead of using packages.
17:38.00Qwell...
17:38.05Strom_Crdgzt: yes, that would be a good idea
17:38.11rdgztStrom_C: In case there's a bug that's been fixed or something. Although I generally prefer using the package system.
17:38.14Qwellnote to self: make sure to read things IN CONTEXT
17:38.17rdgztI'm going to test that now, I think.
17:38.41Strom_Crdgzt: with asterisk, packages are almost always a bad idea
17:38.53Qwells/almost //
17:39.13rdgztStrom_C: I noticed a tendency to that, actually, why is it that distributions don't package the zaptel drivers, for instance?
17:39.19*** join/#asterisk tomcontr3 (n=gcontrer@247-79-246-201.adsl.terra.cl)
17:39.31Strom_Cbecause they blow donkeys for nickels?
17:39.36tomcontr3hi dows ony one know about this? http://www.voip-forum.com/news.php?p=187
17:39.40rdgztThe drivers or the distributions?
17:40.03Strom_Cthe distros, obviously
17:40.23Strom_Czaptel is good stuff
17:40.56rdgztBuilding kernel drivers from source is generally a pretty bad idea, since they might break arbitrarily if you upgrade the kernel through the package system. Which is one of the reasons I generally prefer the package system.
17:41.23Strom_Cthats why you rebuild zaptel after you upgrade the kernel.
17:41.25Strom_Cduh
17:42.32rdgztStrom_C: Yes, well, that's why packages would be good, since the distro would just install zaptel drivers built against the new kernel.
17:42.37*** join/#asterisk budmang (i=budman@12.206.134.162)
17:43.25rdgztBuilding stuff from source is so 1999. :)
17:44.10Zeeekonly true for WIndows
17:44.21Zeeekassembler is best for that
17:45.57*** join/#asterisk mutilator (i=WebChat@65.111.201.122)
17:46.18mutilatordo the zaptel drivers control the hardware echo can?
17:46.22Zeeekwhy is there no distro called 'Kernel Klink' ?
17:46.23mutilatori have a sangoma a104d
17:55.46tomcontr3Im having problems installing an SVN version of asterisk
17:55.56tomcontr3gcc -g -o menuselect ../strcompat.o menuselect.o menuselect_curses.o ../mxml/libmxml.a -lncurses
17:55.56tomcontr3gcc: ../strcompat.o: No such file or directory
17:55.59tomcontr3any idea?
17:56.34Strom_Ctomcontr3: you downloaded the stable version from SVN, right?
17:58.50tomcontr3its a version that a guy name OEJ is building
17:58.55tomcontr3to fix a SIP problem
17:59.02tomcontr3http://svn.digium.com/view/asterisk/team/oej/sipregister/
17:59.17tomcontr3http://www.voip-forum.com/news.php?p=187
17:59.58Strom_Cyou've got the ncurses stuff installed, right?
18:00.15tomcontr3dont know what that is... lolol
18:00.29*** join/#asterisk esculapio_ (n=ESCulapi@187stb68.codetel.net.do)
18:00.31*** join/#asterisk tparcina (n=tparcina@83-131-143-11.adsl.net.t-com.hr)
18:01.22Strom_Ctomcontr3: uh, yeah.  stick to installing stable releases for now.
18:01.46tomcontr3hmmm,  but I need to fix that problem,
18:01.56tomcontr3and that installation it seems to be the only way
18:05.43*** join/#asterisk darby_t (i=darby_t@aaoy166.neoplus.adsl.tpnet.pl)
18:09.41tomcontr3yes I have installed  the ncurses-devel-5.4-17
18:16.37*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
18:19.36blackgeckomanxpower: now its working, it was tx rx gain problem, the txgain was to high. thanks
18:23.27*** join/#asterisk bmg505 (n=leon@dsl-146-56-106.telkomadsl.co.za)
18:24.25*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
18:25.18SwK_misfire
18:26.57*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
18:29.00*** join/#asterisk frk2 (n=kvirc@202.141.251.102)
18:29.09frk2dude
18:29.23frk2have you guys ever sat and wondered how cool asterisk really is? :)
18:29.47frk2i mean damn man. mark spencer has caused a semi-revolution at the age of 28
18:29.53frk2or 29
18:30.25MikeJ[Laptop]isn't that a beatles song?
18:30.37frk2revolution?
18:30.39frk2hell yes
18:31.15frk2i wonder if mark spencer had the digium idea in mind before he started asterisk
18:31.50frk2cuz if he did not - that makes it even more noble
18:32.17frk2i guess nobody cares :)
18:35.10Nuggetnoble?  it's software.
18:35.23*** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net)
18:35.23Nuggetit's not like asterisk is  curing cancer.
18:35.38dpryoEver wondered about how neat Apache really is?
18:35.40dpryo;)
18:35.41filein the end, it's only software
18:35.51*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
18:35.57dpryoor any other piece of open source software..
18:36.34dpryoFirefox, postfix, .. gimp, gnome, kde.. etc
18:36.46frk2yeah man
18:36.48frk2it is neat
18:37.01frk2specially shit like KDE
18:37.09frk2i mean damn thats a lot of code
18:37.20fileyessir
18:37.44frk2But man asterisk is screwing a lot of larger companies over
18:38.12dpryoYeah, and I still have to use a sucky Avaya-system.
18:38.13NuggetIt's likely that none of that open source software would exist were it not for a robust and thriving commercial software industry.
18:38.18*** part/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com)
18:38.23*** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com)
18:38.30frk2this cisco biz dev was complaining about asterisk
18:38.58frk2whatdya mean nugget?
18:39.27fileI like being part of Asterisk... it's nice to see people use it
18:39.35Nuggetwell, for one, much of the open source software that exists is just re-implementing the innovations that come from the closed-source world.  The open source community hasn't shown itself to be very innovative at all.
18:39.57fileNugget: slacker.
18:40.01frk2oh come on i dont think so
18:40.11frk2maybe stuff like KDE, etc
18:40.18Nuggetand secondly, a staggering majority of the "donated" time, energy, and code is coming from people who are gainfully employed as programmers, for companies who release closed-source products.
18:40.24stoffell_hNugget, uh, asterisk is the example that the OS communicate IS innovative !
18:40.34stoffell_hcommunicate=community :)
18:40.42Nuggetif it wasn't possible to earn a living coding, there would be a lot fewer volunteers with the expertise and disposable income to donate their time
18:41.08Nuggetwith the possible exception of dundi, asterisk isn't very innovative at all.
18:41.10Nuggetit's just free.
18:41.15frk2i dont know if ALL of these coders are employed by companies who make money selling code
18:41.28Nuggetwhat's asterisk do that people haven't been doing for decades with commercial solutions?
18:41.45frk2Nugget, the ability to customize
18:41.52Nuggetthat's not unique either
18:41.56stoffell_hsoftware pbx....
18:41.59frk2thats pretty unique
18:42.11filefight fight fight!
18:42.14frk2hahahah
18:42.16ZeeekI think you're all missing the religious connatations here
18:42.20stoffell_hno fight! :d
18:42.22Nuggetopen source is a great way to commoditize innovation that comes from the commercial world, but it hasn't really demonstrated that it's a good way to actually innovate directlyu
18:42.34stoffell_hgotta go catch a beer (hope it's free :p )
18:42.45Nuggetgimp isn't innovative.  gimp is a free clone of photoshop (or, at least, tries to be)
18:42.49frk2dude
18:42.57frk2KDE did in like 3 years what took windows 10
18:43.09Nuggetbut only after windows did it first.
18:43.22frk2and i think by kde 4 it would have innovated past windows
18:43.33frk2gotta give them time man
18:43.34Nuggetand kde is still years behind the current state of the art when it comes to desktop interfaces.
18:43.53dpryoNugget: What do you consider "state of the art desktop"?
18:43.54Nuggetwhat is kde working on, then, that's not already present in windows or os x?
18:43.58frk2oh come on. kde is pretty damn good - many people i know compare it to windows
18:44.16NuggetI didn't say it wasn't good.  I said it wasn't innovative.
18:44.17frk2nugget check their site out... they are gonna go pretty crazy
18:44.29frk2with kde 4
18:44.36Nuggetwhy do you assume I am not familiar with kde?
18:45.23dpryoAll those nice things like you can remove window borders, have different styles on different windows etc.. That's pretty cool.
18:45.28frk2cuz if you were, you wouldn't say many things werent innovative
18:45.37frk2yeah man - STYLES
18:45.40frk2thats unknown to windows
18:45.44dpryoIt sure has been available in enlightenment for 10 years, but I haven't seen it in windows ;)
18:45.57SwK_nugget is just a irc troll he has never done anything important or well known
18:46.03frk2lol
18:46.04DoktorGregare the debian update servers only down for me?
18:46.23Nuggetskins are your idea of innovation?  wow.
18:46.27dpryoDoktorGreg: Those are distributed around the world, I don't think they all go down at once
18:46.45frk2dude its a desktop!!!! yes in that scenario skins are innovative
18:46.49SwK_nugget trolling again eh?
18:46.49DoktorGregum, i cant get on ... any right now...
18:47.04DoktorGregskins?
18:47.26DoktorGregall i wana know, is can i set the default windows 2000 'skin'
18:47.44NuggetI'd rather use an environment where the clipboard works between applications and for data other than plain 7bit clean text.
18:47.57Nuggetor an environment that's making good use of 3d acceleration for common desktop tasks.
18:48.27Nuggetbut if the epitome of an environment for you is to be able to put rounded corners on windows, then I guess I can see where you'd be impressed by kde.
18:48.39frk2hahah
18:48.48frk2i see your point
18:49.24filesomeone guess what Nugget uses for an OS...
18:49.30NuggetI use them all.
18:49.39frk2but you must have one primary one
18:49.41SwK_nugget uses DOS
18:49.48Nuggetok, I don't use dos.  :)
18:49.49dpryoThe abillity to have more users using a single computer is pretty innovative.. Connect more keyboards, displays etc and run more desktops on the very same computer.
18:50.00DoktorGregDOS rocks!
18:50.01Nuggetthat's not an open source innovation.
18:50.02blitzrageDOS rox!
18:50.08dpryoNugget: Sure it is.
18:50.13Nuggetno, it plainly is not.
18:50.28SwK_dpryo no its not
18:50.29dpryoIt wouldn't work in windows, nor OSX.
18:50.29Nuggetpeople did that with dos, with proprietary software, in the early '80s
18:50.33DoktorGregFreeDos has expanded DOS intil a 32 bit multitasking operating system
18:50.35dpryoIt works with X.org
18:51.14frk2i guess its impossible to say whether something is an opensource innovation or a closed source one anyways.
18:51.25Nuggethow so?
18:51.36Nuggetyou just have to look to see who first came up with the idea and implemented it.
18:51.40SwK_doktorgreg and thats not even innovative... I ran majorBBS back in the day that was dos based 32bit multitasking and could handle 100s of users on 386s
18:51.44frk2nugget
18:51.57frk2how do you decide who invented the wheel for example?
18:52.05frk2the 'idea' of the wheel
18:52.20Nuggetthat's hardly a fair comparison.
18:52.32frk2no man
18:52.32SwK_its not the inventing that you are pointing out here, its all the reinvention of old ideas
18:53.03VoicePulseNugget is correct in saying that most of the things you guys are mentioning existed a long time ago, developed by companies like IBM, Xerox and AT&T -- long before the Internet or open-source communities as large as current ones existed.
18:53.20*** join/#asterisk zyth (n=Anon5017@66.244.197.93)
18:53.36VoicePulsewww.uspto.gov is where you look to see who came up with it first
18:54.32SwK_and dont forget people like QuarterDeck that extended DOS into a multitaking environment with QuarterDesk, and allowed it to access more then 1M of ram w/ QEMM, and Pharlaps which made it even easier to access 32bit based memory
18:54.35DoktorGreglol, thats right, nothing has been invented since 1968
18:55.00frk2hehe
18:55.20frk2i dont know whats counted as innovation
18:55.37VoicePulseDoktorGreg: That's why he asked you to name one to support your argument...
18:55.45SwK_i mean how innovative is a telephone really? yeah we can make it do new tricks, but you can still hook up A.G.Bell's original phone to a modern pots like and make a call with it
18:55.47DoktorGregok, voip
18:55.51DoktorGregvoip is new
18:55.57SwK_VoIP is not new
18:56.09VoicePulseI believe the discussion is if open-source invented XYZ before closed-source.
18:56.54DoktorGreghttp://www.voipreview.org/news.details.aspx?nid=51
18:56.54Nuggetopen source has a poor track record for actually producing innovative software and features.
18:56.55SwK_Packet Switched Networks and pushing realtime (or near realtime) data feeds have been around forever... see TymeNet and x.25
18:57.07DoktorGregthere you go, voip is new
18:57.18DoktorGregwell except for voip
18:57.39frk2:)
18:57.49Nuggetto my eyes, open source excels mainly at solving the challenge of "I want that cool software, but I don't want to pay what they charge for it"
18:58.06DoktorGreglike Apache???
18:58.15DoktorGregphp
18:58.18Nuggetor, to be fair, "I want that cool software but the terms of use for it are ridiculous"
18:58.27DoktorGregdatabases
18:59.01frk2well see doktor what nugget is saying is that 'mysql' is not innovative, which i can understand, because databases have been around for many years
18:59.03DoktorGregbittorrent
18:59.16Nuggetbittorrent is an excellent example of open source innovation, yes.
18:59.29DoktorGregso is voip
18:59.46frk2voip isnt dude
18:59.48SwK_VoIP != OpenSource innovation
18:59.49Nuggetso cisco callmanager came after asterisk?
18:59.53frk2voip s been around forever
18:59.57DoktorGregsince 95
19:00.01SwK_CCM != innovation
19:00.08DoktorGreg95 != forever
19:00.20DoktorGregor even a long time
19:00.31SwK_cisco didnt innovate shit in VoIP they bought it
19:00.35Nuggetwhat is an example of an open source voip solution that came prior to 1995, then?
19:00.50Nuggetthat first, innovative voip implementation that was open source.
19:01.53DoktorGregits maybe the first product that is open soice
19:02.06DoktorGregbut the foundations of voip were opensource
19:02.16Nuggetwhy do you say that?
19:02.31Nuggetand how does that make voip, then, an open source innovation?
19:03.10zythI have 16 rooms in a hotel I need to provide phone service to - basically, the call from telco comes to a main #, I then want people to be able to punch in the room # and talk to whomever. Is VoIP a viable solution to that?
19:03.19zythw/o it costing much..
19:03.41SwK_zyth: I have asterisk in a 200 room hotel
19:03.48Nuggetheh
19:04.08zythSwK_: any appreciable cost outlay I'd need to be aware of before trying to implement such a thing?
19:04.24SwK_channel banks T1 Cards etc...
19:04.26justinu|laptopprobably around 5 grand
19:04.28justinu|laptoptops
19:04.34SwK_its just another PBX
19:04.51tomcontr3how can I doenload this SVN version?
19:04.52tomcontr3http://svn.digium.com/view/asterisk/team/oej/sipregister
19:04.59DoktorGregunless you have someone install it for you, in which case it is about 30k
19:05.07zythis that apart from our internet link?
19:05.28zythDoktorGreg: yeah, for that we could lay regular phones to the building.
19:05.43kaz0358i have done a bit of searching and i cannot find what all the valid characters are for an iax2 password.. i know letters and numbers.. but what symbols are also valid?
19:05.45SwK_zyth you dont necesarily need as IP link to the internet to use asterisk
19:06.04SwK_it can replace a legacy Avaya, nortel or siemens hospitality solution
19:06.18frk2definitely
19:06.35frk2but do you HAVE to go pots?
19:06.37Zeeekzyth you'd usually be safe with RADIX50 characters
19:06.38SwK_you can do 16 stations for 5 to 10K USD
19:06.45zythhmm ok
19:07.09Zeeeknah, that' can't be right
19:08.34tomcontr3can any one help me to install this SVN version?
19:08.34tomcontr3<PROTECTED>
19:09.01Qwelltomcontr3: svn co http://svn.digium.com/view/asterisk/team/oej/sipregister asterisk-sipregister
19:09.02*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
19:09.09Qwellcd asterisk-sipregister && make install
19:09.25tomcontr3just make install
19:09.26tomcontr3?
19:09.31kaz0358radix50.. do you think & and ! are valid characters?
19:09.35Qwellyes
19:09.37tomcontr3ok
19:09.45Qwellunless it bitches about needing to restart make...then just run it again
19:10.17tomcontr3Generating the configure script ...
19:10.29tomcontr3its running a config script ,  is that normal?
19:11.07SwK_yes
19:11.13*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-170-22.bflony.east.verizon.net)
19:11.17hwti'm having some serious problems getting anything but xten to send dtmf to asterisk
19:11.24hwtor rather getting asterisk to understand dtmf
19:11.26hwte.g. in meetme.
19:11.28hwtany tips?
19:11.41hwti suspect it's trying to send inband, and that's not very smart when using gsm.
19:12.13*** join/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net)
19:12.25tomcontr3Qwell,  it didn worked
19:12.37tomcontr3it seems  that this needs to be configured first
19:12.48tomcontr3and now it says
19:12.48tomcontr3gcc -g -o menuselect ../strcompat.o menuselect.o menuselect_curses.o ../mxml/libmxml.a -lncurses
19:12.49tomcontr3gcc: ../strcompat.o: No such file or directory
19:12.57hwttried rfc, auto and inband
19:12.59hwtno luck
19:13.41*** part/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net)
19:13.52SuPrSluGi'm having a similar issue. but it's w/ broadvoice. When I dial in through voicepulse I can get to the meetme 22. when I got through broadvoice it dials 2 in my ivr and never waits for the second 2
19:13.53kaz0358hwt, what do you have dtmfmode set to in sip.conf?
19:14.41kaz0358hwt, the default is rfc2833 unless you have changed it.
19:15.21hwtkaz0358: everything. auto, rfc and inband
19:15.29Zeeekwith reloads each time?
19:15.37hwtyes, of course.
19:15.48Zeeekdid you change the setting in X-Lite
19:15.51Zeeek?
19:16.02hwtZeeek: i haven't done anything for it to work in x-lite.
19:16.10hwtbut i have tried calling in from my gsm
19:16.12hwtand no luck
19:16.15Zeeekthere is a setting that forces inband
19:16.23*** join/#asterisk trimi` (i=Whatt@62.162.243.194)
19:16.25hwtyeah, but that won't work with gsm
19:16.31trimi`hey pla any on can tell me what does this error mean   Apr 30 20:33:10 NOTICE[13712]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
19:16.35trimi`<PROTECTED>
19:16.37Zeeekexactly, you'd want to make sure it wasn't set
19:17.02hwtyeah, but i've tried setting the option globally.
19:17.03SwK_trimi that means the devices is most likely not registered or asterisk doesnt know that endpoints IP
19:17.08Zeeektrimi` use pastebin to show the dial command
19:17.30trimi`its working if i use directly
19:17.43trimi`but if i register my asterisk in other user
19:17.47trimi`not dialing
19:17.53Zeeekok
19:18.12trimi`btw how do i see which codec its used while a sip its calling
19:18.26Zeeekhwt what did yiou mean calling from your gsm?
19:18.29robin_szsip show channels
19:18.37trimi`thnx
19:18.37hwtZeeek: yes, calling from my gsm
19:19.03Zeeekwhat gsm?
19:19.08PakiPenguincellphone
19:19.11PakiPenguin:p
19:19.16Zeeekgsm doesn't mean cellphone
19:19.23robin_sz.me nods
19:19.25kaz0358trimi, not following too closely.. but asterisk does tell you the codec used if you are on the console and gave -vvvv
19:19.33Zeeeknor does a cellphone put gsm to asterisk
19:19.42trimi`<kaz0358> if i use iax yes
19:19.45trimi`if i use sip no
19:20.00PakiPenguin:)
19:20.25Zeeekhwt plus cellphones are so bad they often can't send tones to regular lines let alone asterisk
19:20.31*** join/#asterisk gr0mit (n=Gr0mit@extrt.txrx.org.uk)
19:20.45PakiPenguinZeeek, mine works alright :)
19:20.56Zeeekit's connection dependent
19:21.05Zeeekanyway
19:21.11robin_szdtmf over a gsm compressed link can be dodgy
19:21.22Zeeekeggsakly
19:21.30PakiPenguinyup
19:21.42Zeeeksomeone has a nice little test out there
19:23.24robin_szi'll tell you what I want, what I really, really, want,
19:24.00robin_szI wanna, I wanna ... someone to send me thevery latest, not on the wiki anymore, GXP2000 firmware
19:24.00[hC]proper polycom BLF.
19:24.21[hC]or fixit instructions for my linksys wip300!
19:24.31SuPrSluGanyone know why asterisk dials exten 2 immediately instead of waiting for 22? only in one context. the other inbound line works fine. dtmf?
19:24.50ZeeekBanana-Rama?
19:24.58Zeeekthat's dredging pretty low!
19:25.04robin_szwell, yes
19:25.12robin_szbut we are talking grandstream here
19:25.19robin_szI had to go as low as I could
19:25.20Zeeekheh
19:25.34[hC]you got a 2 for 1 there with spice girls, too.
19:25.37Zeeekto think I *almost* bought one of those
19:25.52hwtZeeek: well, it doesn't work from snom phones or another ata in the network
19:25.55hwtZeeek: either.
19:26.15ZeeekDTMF? no where?
19:26.25robin_szahh yes, the Spice Girls. a group so .. so ... unsullied by talent
19:26.38hwtZeeek: yes, the dtmf doesn't work from anywhere except my x-lite.
19:26.43Zeeekwell the talent was only visible to managerial elements prolly
19:27.07robin_szyou mean your boss liked them?
19:27.19Zeeekno their managers prolly saw "talen"
19:27.31robin_szahh
19:27.46robin_sztheir managers probably saw $$$ ...
19:27.51ZeeekOne famous actor years ago said "talent? Yeah it's spelled s.u.c.k."
19:28.03Zeeekbut we digress
19:28.08robin_szwe do ...
19:28.16robin_szso, the latest firmware?
19:28.21Zeeekso SIP, gsm, no dtmf - a geeks' dream
19:28.31Zeeekfind a GS geek forum
19:28.56Zeeekor... try the google related time machine thingie
19:29.03Zeeekor the google cache
19:29.12robin_szahh ... point.
19:29.19robin_szwayback
19:29.20Zeeeksometimes the shit is there and they just changed the links :)
19:29.41Zeeekbut before you get all wet, I doubt that'll work
19:30.05Zeeekit's just a pacifier I sent along so yopur crying in the middle of the night doesn't keep us awake
19:30.46Zeeekdamn, well I solved a few of my little problems tonight. 2 down one to go
19:31.21Zeeekhave you ever walked in a store and asked for a SIM card?
19:31.22gr0mitrobin_sz, what sw do you want for the GS?
19:31.48Zeeekhe's looking for foimware
19:34.37gr0mityes which voision?
19:35.59Zeeekthe "foim" part indicates a recent, but not the most recent version. Whereas "firware" are wooden trousers
19:36.33Zeeekit's either time to go to bed or time for a beer.
19:36.44gr0mitboth prolly
19:37.03Zeeeknaw, I don't like balancing it in bed
19:39.50hwthmm,  pbx.c:1406 ast_func_write: Function LANGUAGE not registered
19:39.52hwtwhat does that mean?
19:39.58hwtwhat module am i missing?
19:40.08Qwellfunc_language, would be my guess
19:40.25hwtQwell: nah.
19:40.42hwtperhaps pbx_functions.so
19:40.46Qwellfunc_language.so               Channel language dialplan function
19:40.54QwellNo, func_language
19:41.37hwtuhm, i don't have that module.
19:41.51QwellThen...sounds like you've got problems
19:42.01Qwellsuch as using a version of asterisk that's too old
19:42.33hwtpbx_functions.so did the trick
19:42.42hwtQwell: maybe func_language.so is deprecated?
19:42.46Qwellno
19:43.23Qwelloh, heh
19:43.32QwellDeprecated. Use CHANNEL(language) instead.
19:44.09robin_szdang, going through the tiki-wiki page history, it seems that the latest firmware never made it to the wiki ... bummer.
19:45.15robin_szgr0mit, I would like any version that makes my display work :)
19:45.40gr0mitwhich version do you have?
19:45.46robin_sz1.1.0.1
19:46.00gr0mithmmm i think i have 1.1.0.4
19:46.03gr0mitlet me check
19:46.08robin_sz1.1.0.4 wold be lovely :)
19:46.30robin_szor at lease, stands a better chance of getting my display working
19:46.44robin_szit goes blank the moment after the phone boots :(
19:47.01robin_szapparently I have an older phone.
19:49.36gr0mitneed to got upstairs and check
19:49.47robin_szjust access the phone web onteface
19:51.54*** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-75-9-244-217.dsl.sfldmi.sbcglobal.net)
19:54.48*** join/#asterisk SparFux (n=player@e182022054.adsl.alicedsl.de)
19:55.27gr0mitsorry but i have the same 1.1.0.1 as you
19:55.43*** join/#asterisk darby_t (i=darby_t@aaph204.neoplus.adsl.tpnet.pl)
19:55.53gr0mitthe phone crashed when i logged in t o the web interface.  not entirely stable
19:56.11*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
19:56.31robin_szahh, ok
19:56.35robin_szthanks for looking
19:59.34*** part/#asterisk darby_t (i=darby_t@aaph204.neoplus.adsl.tpnet.pl)
20:01.11SparFuxI would like to have an application app_streamsound(program,parameters) which would use <program> to play a sound to stdout and stream that to an asterisk extension. Like MP3Player does it with mpg123, but just an arbitrary program, not only mpg123.
20:01.22*** join/#asterisk dlynes_ (i=1000@S010600c09f9a0fc4.vc.shawcable.net)
20:03.10*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
20:03.39*** join/#asterisk Cardoe (n=Cardoe@gentoo/developer/Cardoe)
20:03.57Cardoehow do you guys feel about freePBX as a viable system for a home Asterisk setup?
20:04.19Cardoeanything business related is bad with *@Home
20:04.50Hmmhesaysi disagree
20:04.50gr0mitCardoe - just use raw asterisk.
20:05.19gr0mitonce you get the hang of config files you can do anything from vi
20:05.29SparFuxgromit: ack.
20:07.48dlynes_Cardoe: Well, there's one advantage of AMP
20:07.57dlynes_Cardoe: Telephone techs can understand how to use it
20:09.49gr0mitprolly fine but not very flexible
20:10.30CardoeI'm debating if I should just toss in freePBX on top of my Asterisk box here at home.
20:10.33Cardoeor just do it myself.
20:10.39dlynes_telephone techs only need something that can do a very finite set of things, anyways
20:10.40gr0mitjust do it yourself
20:11.00Cardoedlynes: what does it include that telephone techs want?
20:11.14dlynes_Cardoe: ease of configuring
20:11.15CardoeI also don't have a POTS line anymore. Gonna go through SIP purely.
20:11.31dlynes_Cardoe: most telephone techs don't know computers well enough to log into linux, fire up vi, and edit some text files
20:12.03Cardoewhat telephone techs would be messing with my system?
20:12.14dlynes_Cardoe: so for them, it's a gui such as amp or something similar, or nothign
20:12.27dlynes_Cardoe: none...I'm just saying for offices, not for homes
20:12.58CardoeWell I was talking about my home machine
20:13.07dlynes_telephone techs working for interconnects are the ones that traditionally install office pbxes, and maintain them
20:13.13Cardoeat the office we have a propritary system developed by some company.
20:13.16gr0mitjust keep it as raw text.  if you give them a gui you are doing yourself out of a job ;-)
20:13.18Cardoethat provides a web interface to asterisk
20:13.36dlynes_Cardoe: Yeah...I was just saying for the office, amp is somewhat useful...for the home, it's total overkill
20:14.01Cardoeany features you guys find particularly useful?
20:14.09CardoeI played with asterisk like 3 years ago
20:14.09dlynes_Dial()
20:14.16*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:14.21SparFuxAha! I get a feeling that eagi is just what I am searching for. I can use fd 3 to pipe any sound from any application to asterisk, right? :-)
20:14.22NuggetAren't you glad you used Dial()?
20:14.27dlynes_Yep
20:14.34Strom_CDon't you wish everybody did?
20:14.37dlynes_It's the most powerful tool in my whole arsenal :)
20:14.52*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
20:15.07dlynes_Good morning, Dr-Linux
20:15.16Cardoeany recommendations for outgoing SIP/IAX?
20:16.35Dr-Linuxdlynes_: thanks friend
20:16.41Dr-Linuxit's night though heere
20:16.53dlynes_Dr-Linux: oh...thought it'd be like 1 or 2 am
20:17.45Dr-Linuxdlynes_: yah it is
20:17.56*** join/#asterisk Lord_Drachenblut (n=Lord@12.210.117.62)
20:18.07dlynes_well then, it's morning :)
20:18.24dlynes_hanji?
20:23.10*** join/#asterisk ozverenm20 (n=ozverenm@162.27.103-84.rev.gaoland.net)
20:23.44ozverenm20someone expert in ISDN card ?
20:27.01dlynes_Cardoe: you mean providers?
20:27.05*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
20:27.39dlynes_Cardoe: Try www.calltermination.com....there's a whole slew of them on there
20:28.13Cardoethx
20:28.36dlynes_a good number of them will do voip escrow, too
20:31.13*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
20:35.13*** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com)
20:38.47*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram)
20:38.48*** mode/#asterisk [+o kram] by ChanServ
20:48.06ozverenm20i need help about DACS and audio buffer hooking
20:48.32ozverenm20i want to make asterisk between my PBX and my telco
20:48.56ozverenm20i also need to record B channels traffic without decrease audio quality
20:49.03ozverenm20how can i do it ?
20:59.54*** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net)
21:00.08*** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net)
21:03.24tainted-ozverenm20 PRI?
21:04.09ozverenm20yes
21:04.13ozverenm20yes pri
21:04.45tainted-u'd need some zaptel cards
21:04.53ozverenm20wich ?
21:04.56tainted-i'd recommend sangoma
21:05.00RoyK~ser
21:05.02jbothmm... ser is Sip Express Router - see http://www.iptel.org/ser/
21:05.03ozverenm20sangoma ?
21:05.20tainted-how many B channels
21:05.20krami'd recommend digium ;-)
21:05.35tainted-sangoma works better though
21:05.36RoyKsangoma has good cards, but they have a higher cpu overhead than digium's due to the wanrouter stuff
21:06.06tainted-how much higer
21:06.08tainted-higher
21:06.10DoktorGregit is really hard to recommend anything but digium, because that is the only thing i have used
21:06.17RoyKabout the double for what i can see
21:06.28kramnot to mention we actually support the project
21:06.49ozverenm20i am searching for a card that do hardware switching ( bridging ) and sent all audio buffer to me
21:06.51RoyKsangoma's stuff can be used to do other stuff, though, like ss7 with ss7box
21:06.51DoktorGregwell that too, digium actually makes asterisk
21:07.18RoyKbut I beleive I'd recommend digium's E1/T1 cards for plain asterisk PRI works
21:07.50tainted-RoyK do u have screencaps of cpu usage?
21:08.26kramtainted: what makes you recommend sangoma out of curiosity
21:08.36*** join/#asterisk Assid (n=assid@203.115.64.12)
21:08.39ozverenm20digium are reputed to be not stables , no ?
21:08.46ozverenm20digium cards
21:08.46kramthe digium card does support on-card switching
21:08.51DoktorGregwhy would you want to route old slow single megabit connections anyhow?
21:08.57kramhow would we be less reliable?
21:09.08ozverenm20sangoma support on card switching ?
21:09.11krami don't know
21:09.34DoktorGregive had zero stability problem with the two digium cards ive had
21:10.01RoyKkram: the old cards have lots of bad habits. dunno about the new ones, though. we still run zaptel 1.0.7 since we can't upgrade to newer zaptel since it doesn't work with them, and we can't take the cards out of production to have them upgraded :P
21:10.04DoktorGregand i was using 8 channels at a time last week
21:10.07krami should clarify the quad T1/E1 supports on-card switching not the single
21:10.18RoyKozverenm20: sangomas doesn't support oncard switching either
21:10.28DoktorGregwhats on channel switching?
21:10.34DoktorGreger oncard
21:10.55tainted-kram ever since 1.0.4 we've had various issues with digium cards
21:11.03*** join/#asterisk brookshire (i=mbrooks@hijacked.us)
21:11.03krammeans that when a call starts and ends on a card, it doesn't have to pass all the way up to the zaptel layer
21:11.05RoyKDoktorGreg: tell card to attach channel n to channel m and forget about it
21:11.09kramit shaves a millisecond or two off of the latency
21:11.24kramit also means that no matter what happens in interrupt land there is no possibility of missing a sample
21:11.25RoyKit also saves cpu load....
21:11.30tainted-recently after discovering sangoma, we've realized that is JUST WORKS
21:11.53DoktorGregthe native bridging thing that digium cards do when a call just goes in and out of the pri ports?
21:11.59Assidis there any way of adding gsm support to polycom?
21:12.01tainted-but it could be zaptel like RoyK was saying
21:12.13kramtainted: what was the situation in which your digium cards did not work, if i may ask
21:12.30RoyKtainted-: with a single te410p in a couple of single cpu boxes, they use something like 40% cpu with full load, whereas a newer dual cpu box gives about the same load with a sangoma
21:12.42RoyKmeaning about double or a little more cpu
21:12.51RoyKper channel, no transcoding
21:13.03RoyKsimply more abstraction layers etc
21:13.05kramour 2nd gen firmware includes some special SMP optimizations that allow the performance to be improved significantly
21:13.27RoyKkram: but i still have to mail you the cards to get the new firmware :P
21:13.40Assidkram: you work for digium?
21:13.45RoyKlol
21:13.47I-MODlol
21:14.10RoyKhm. i might have one a card spare. kram, how long does it take to get one upgraded and mailed up here?
21:14.12Assidwhatd i say so funny?
21:14.23brookshireassid: kram is the creator of asterisk
21:14.23RoyKAssid: ever heard the name Mark Spencer?
21:14.28kramroyk: i tell you what roy, i'll talk to malcolm and if you've been staying off the trolling i'll see what we can work out
21:14.30Assidoh thats him?!?!!?
21:14.33Assiddamn.. i didnt know
21:14.36SwK_lol
21:14.38RoyKkram: thanks
21:14.55Assidshit.. sorry man.. didnt know
21:15.01RoyK:)
21:15.47ozverenm20junghanns cards does support on card switching ?
21:15.53kramtainted: can you clarify a bit your issues?
21:16.10kramtainted: if you're having real issues, i'd like to try to make sure they're things that have been resolved
21:16.44kramoz: i don't know much about his e1 cads
21:16.46kramcards
21:16.55tzangerwhoa, kram is actually on IRC?
21:17.00filedon't everyone attack kram
21:17.11Strom_Cit's kram!
21:17.14filetzanger: oh goody you're here... I'm doing some chan_iax2 work right now... that I'll want you to test
21:17.15Assidhehe
21:17.33tzangerfile: I can try but I've gotta head out very very shortly
21:17.38Assidman.. i feel like everyones doing something important except me
21:17.47ozverenm20only eicon support on card switching ?
21:17.47filetzanger: it can wait till tomorrow or whatever
21:17.53tzangerok
21:17.54NivexAssid: don't worry, I just lurk here.
21:17.59kramyou mean for BRI?
21:18.32tainted-kram there were audio problems, asterisk would segfault, IRQ incompatibilities
21:18.45kramyou seriously think an asterisk segfault would be related to hardware?
21:18.52kramtell me about the audio problems and irq issues though
21:19.09Assidhrmm.. im segfault.. check ram..?
21:19.35ozverenm20if segfault do 112 call ? :)
21:19.49tainted-kram well audio issues turned out to be problems dealing with MMX CPU
21:20.00kramokay
21:20.05kramso what were the issues with the cards then
21:20.11AssidNivex: i just dont wanna be sitting around doing nothing.. lets see.. what can i do
21:20.16tzangertainted-: did you get my messages?
21:21.32NivexAssid: spread the word!
21:21.59Assidof?
21:22.34tzangerAssid: the word is legs!
21:22.37NivexA
21:22.37*** join/#asterisk faljse (n=martin@83-65-243-10.dynamic.xdsl-line.inode.at)
21:22.45kramwork on the bug tracker :)
21:22.50kramhelp us fix issues in asterisk etc
21:22.51tainted-kram all i'm saying is that for me, sangoma was drop and play
21:23.01faljseis there a limit for agi parameter length(100 chars maybe?)
21:23.04krambut wait, i don't understand yet
21:23.10tainted-u don't have to
21:23.13ozverenm20does digium cards support on chip card + audio buffer recording ?
21:23.21kramyou changed general zaptel parameters that don't have anything to do with the hardware
21:23.22Assidkram: dunno C .. want php ?
21:23.29tzangertainted-: not for me.  I have an A104D that has zero audio... D channel works great but no audio :-)
21:23.32krami'm trying to understand what actually was the hardware issue
21:23.48krami want to be sure we're doing well for people
21:23.55*** join/#asterisk cr_0 (n=y@toronto-HSE-ppp4334596.sympatico.ca)
21:24.03tzangerthe only hardware issues I know of are the VPMs that trigger DTMF on (generally) female voices
21:24.05krami mean if you just want sangoma that's fine but if you're actually having issues with digium hardware i'd like to know about it
21:24.22Assidhell.. you guys are.. i love *.. i'd love to find a way for it to be a source of income one day for me
21:24.23Assidhehe
21:24.30*** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-75-9-244-217.dsl.sfldmi.sbcglobal.net)
21:24.36kramassid: where are you located?
21:24.36cr_0i don't need cdr_odbc.conf so i removed it, but now asterisk warns about disabling some odbc module when restarting.  how do i tell asterisk not to try to load the module?
21:24.43Assidindia
21:25.00tzangerspeaking of VPMs...  kram, is there any thought on a VPM with a heavier (i.e. Tellabs-quality, long tail, bidirectional) echo cancel?
21:25.09fileIndia? you can be imported! just like me.
21:25.29kram:)
21:25.38brookshirefile: you haven't been imported yet
21:25.51filepesky country
21:25.51Assidnah.. my lifes wayyyy tooo complicated.. would take forever for my papers to go through.. one of the reasons i dont live with my dad.. he fucked my life up
21:25.56DaminSigh..
21:26.08ozverenm20no response for my problem ?
21:26.18tainted-kram i'm sure the irq issues were fixed, and the mmx thing wasn't exactly a digium card problem
21:26.20DaminFirst day that I get a break in two weeks and I have to put together a bike for my 4 year old..
21:26.24brookshireoz: what's your problem?
21:26.39kramassid: are you a web coder? :)
21:26.43Assidyep
21:26.51Assidphp mostly..
21:26.54kramjavascript?
21:27.04brookshirehah
21:27.06Assidnot that well.. but decent enough to get my work done
21:27.10brookshireajam!
21:27.11brookshire:D
21:27.13Assidmostly just php
21:27.34Assidwrote a call logger to integrate with asterisk's cdr
21:27.48DaminAssid: I need a fully Multitasking Kernel written in Javascript that can run inside of IE. Can you do it? ;)
21:27.50Assidpretty simple shit
21:28.45brookshiredamin: i need a virtual machine that runs inside of java, that can run firefox
21:28.49tainted-kram also i received better phone support from sangoma
21:29.58brookshirei serious doubt that
21:30.03brookshireseriously
21:30.17Assidim planning to maybe create a web interface similar to AMP.. but from a business angle.. where users can make their own pbx and stuff
21:30.27Daminbrookshire: Hehehe.. While it doesn't run inside Java the Vmware Player Browser Appliance.. http://www.vmware.com/vmtn/appliances/directory/browserapp.html
21:30.35kramtainted: can you explain what you didn't like from our phonoe support?
21:30.41ozverenm20tainted: its sure sangoma support on-card switching ?
21:31.46*** join/#asterisk Eggplants (i=No@dsl-201.cascadeaccess.com)
21:32.11krami think tainted is just a fan of sangoma and either can't or doesn't want to express a real beef with digium
21:32.26kramwhich is unfortunate since, on the whole, it's digium's code and not sangomas that he's using ;-)
21:32.42tainted-well now that's not fair
21:32.58tainted-i've supported digium many times
21:33.02Assidkram: why whatd you have in mind?
21:33.06kramokay then express your beef in something that i can take to my management in order to try to get your issues resolved
21:33.17tainted-i've purchased g729 licenses, hardware, supported new users
21:34.16kramassid: just generally looking for people that want to explore the ajam stuff i've been working on
21:34.16kramweb enabling asterisk
21:34.16Assidoh..
21:34.16Assidrealtime database ?
21:34.16brookshiremanager in xml
21:34.17Assidnot bad.. seems to have good potential
21:34.29kramthere's an ajamdemo.html that lets you demo it a little with transfer and hangup
21:34.42Assidcould also use it for interfacing dundi if its on xml
21:34.59kramtainted-: okay so please do me the favor of telling me what issues you had with digium so i can try to improve for future customers
21:35.09Assidkram: using the manager ?
21:35.33kramtainted-: i'm not trying to convince you to buy digium, it's obvious you want to stick with sangoma but i'd like to at least try to improve upon our processes if there is anything we actually did wrong to push you that way
21:35.35brookshireassid: no, it is manager
21:35.43brookshireonly not manager ;)
21:36.03Assidisnt the manager just a port open... and running commands through it
21:36.25*** join/#asterisk BladeRunner05 (n=feelme@adsl-182-210.37-151.net24.it)
21:36.29brookshireyeah.. but the ajam stuff doesn't use manager
21:36.30Assidi havent done much readup on the manager.. so maybe im wrong
21:36.39Assidright. prolly just an api does the same
21:36.44brookshireit is a direct xml interface to asterisk
21:36.50Assidnice
21:37.09Assidmaybe i can work on it once im done with this project i have in hand
21:37.09brookshireit uses an internal webserver
21:37.14Assidon my free time
21:37.17tainted-kram i don't think u understand
21:37.42tainted-kram i recommended sangoma based on my past experiences with digium hardware and my current experiences with sangoma
21:37.46Assidi think im losing it .. wheres it on the site?
21:38.03kramtainted-: so what about your past experience with digium is the issue
21:38.16brookshireassid: is so fresh.. there is not much to go on
21:38.29brookshireassid: but you can find some docs/readme in trunk
21:38.34Assidaiee
21:38.41tainted-there's no issue
21:38.47tainted-lol
21:38.56kramtainted-: i mean seriously, in the absense of any other differentiation, i would hope that you would support digium since we support the project and even the channel you're chatting in here, recommending our competition
21:39.00Assidfor some strange reason.. if  i use svn.. on debian.. it screws up.. coz of libapr.. then my apache gets messed up if im using source
21:39.10brookshireassid: http://svn.digium.com/view/asterisk/trunk/doc/ajam.txt?view=markup
21:39.15tainted-i do support digium!
21:39.18kramso presumably there must be *some* reason that you would choose not to support us but instead to recommend our competitor
21:39.28tainted-i'm in here every day helping folks with dialplans etc
21:39.38kramand i'm trying to understand what that reason is, concretely, so that i can be sure that we can improve
21:39.50kramand yet you either cannot or choose not to express any such reason whatsoever
21:39.59kramand i cannot fathom any reason why
21:40.10kramare they your employer?
21:40.10tainted-hey listen
21:40.25kramgive you free hardware?
21:40.25tainted-if u don't want me recommending competitors' products, just flat out say it
21:40.37tainted-i'll keep my mouth shut
21:40.39krami'm not saying that, i'm saying i want to understand why
21:40.54kramafter all, if that's truly how you feel, it's not my place to tell you to go against your feelings
21:41.00Assidbrookshire: any chance thats running php ?
21:41.02kramyour opinions expressed here are obviously and should be your own
21:41.05brookshireassid: zero :)
21:41.06krami'm simply trying to understand the background
21:41.10Assidhehe
21:41.11tainted-i told u
21:41.12kramand you're not helping me with that much
21:41.14brookshireassid: it was written in c
21:41.27Assidmy c knowledge sucks..  :(
21:41.37brookshireassid: yah, but it outputs xml
21:41.39tainted-audio issues, problems with irq, a few frustrating phone calls with support
21:41.52brookshireso.. think of a webservice :)
21:41.52kramyou said your audio issues were MMX related
21:41.53tainted-and i haven't touched digium cards since then
21:41.55kramwhich is not specific to us
21:42.09kramand you haven't specified what the irq or phone call issues were
21:42.15Assidright.. so what part of it needs development?
21:42.24krambut i'm very eager to find out if you'll explain, again so that i can try to improve in the future
21:42.42Assidtainted-: help him help you!
21:42.48tainted-i just want the voip/zaptel platform to WORK so i can focus on writing value adding apps
21:42.59Assidtainted-: explain "WORK"
21:43.05Assidwhats wrong EXACTLY
21:43.16tainted-right now, there's nothing wrong
21:43.17Assidwhats the symptoms
21:43.22kramokay so help me do that by sharing your experience more concretely
21:43.24tainted-i'm just explaining my preferences
21:43.37kramwell at least what was the phone issue?
21:43.44*** join/#asterisk jql (n=jql@ip68-6-153-27.sd.sd.cox.net)
21:43.46Assidtainted-: how do you prefer 1 product over another, if there is no pros/cons
21:43.50kramand when you say nothing is wrong now do you mean with your digium h/w?
21:44.12tainted-i don't own anymore digium hardware
21:44.12Assidtainted-: do you like apples/oranges ?
21:44.15Nivexrandom side question: has anyone here had luck getting Zaptel to compile and function as perscribed under Xen?
21:44.18kaz0358is there something special you have to do if you want to be able to dial *12345 if you are punching it in with dtmf? because as soon as you hit * followed by two numbers, you automatically get a busy/congested tone..
21:44.40Assidkaz0358: transfer ?
21:44.42kramokay so you had issues with the digium h/w
21:44.51kramdo you have your support ticket numbers from when you contacted tech support?
21:45.07tainted-probably somewhere
21:45.22kramdid support resolve your issues?
21:45.29kaz0358assid, no i was wanting to use sipbroker which allows you to dial another provider with a *XXX-enduser-num
21:46.10ozverenm20tainted: what do you think about cologne chipsets ??
21:46.13tainted-no, i recall scouring lists and trying different mobos
21:47.07kramtained: if you can send me your ticket numbers, i'd like to research them and see what happened
21:47.33tainted-ok
21:48.22tainted-BUY DIGIUM
21:48.34kramlol
21:48.43tainted-but switch to freeswitch
21:48.44DaminNo.. vote for pedro!
21:48.48tainted-lol
21:48.49*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
21:48.52kramtainted: seriously i'm asking you to help me try to make our product better
21:49.15tainted-since i got into asterisk i have been
21:49.15kramtainted: i'm not asking you to switch, just to help me understand your experience
21:49.22tainted-i am not a hard core c developer
21:49.43brookshirek
21:49.51krammaybe tainted just likes trolling :)
21:50.04russellbkram: that's what i think
21:50.14Strom_Ctainted-: it's very simple.  tell kram, in a reasonable amount of detail, what happened and why you were unsatisfied
21:50.14kramoh well, if you're willing to, e-mail me your ticket numbers, markster@digium.com
21:50.23krami'd be interested in seeing it
21:50.29krami'm gonna get back to coding :)
21:50.38krami think i remember now why i'm so rarely on irc lol
21:51.10*** join/#asterisk swytch (n=ezcall@d83-179-214-255.cust.tele2.fr)
21:51.23Assidhahaa
21:51.42tainted-lol
21:51.46tainted-he really got worked up
21:52.07swytchYou dont have a true random source in your computer?  Then use the h323-disconnect-cause.
21:52.14russellbworked up when you were not able to provide any reasoning, yes.
21:52.36russellbwhich means you're just a troll
21:52.43blitzragewow.... kram is on irc?
21:52.46blitzrage:)
21:52.49fileblitzrage: scary eh?
21:52.52blitzragequite
21:53.08ariel_kram, your always welcome here....
21:53.10filesoooo IAXtel *should* be operational
21:53.12Strom_Cwell either that or sangoma's really putting some high-quality crack into their kool-aid these days
21:53.53xachenhaha
21:53.56xachenyou mean it finally is?
21:54.00tainted-russellb the issues with digium hardware are long past. i made a recommendation based on my current positive experiences with sangoma
21:54.05filexachen: it is.
21:54.31tainted-but i haven't seen/heard about the cpu overhead issues royk was talking about
21:55.23tainted-i can see why my public recommendation for a competitors product would peeve so many in this channel
21:55.46ManxPowertainted-, I'll prolly go with Sangoma when I do my personal telco stuff
21:55.53tainted-after all, most of the respondants work for digium
21:56.02tainted-but the personal attacks should stop
21:56.13X-Robsangoma makes good hardware.
21:56.16tainted-it makes digium look like a bunch of tards
21:56.18X-RobThere's no issue with that
21:56.33*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
21:56.37X-Roband, c'mon, the TDM4xx is a pile of puss.
21:56.55tainted-X-Rob u'd better explain yourself with support tickets
21:56.56X-Robbut I like the TE110, that's a nice little card. Cheap, cost effective, and does what you want.
21:57.07ManxPowertainted-, I suspect that it's less an issue of hardware design, and more of kernel driver design.
21:57.33X-Robtainted-, heh
21:58.20Strom_CI think it's time for a bike ride
21:59.06X-RobHowever, I am pissed at digium's tech support at the moment.
21:59.12ManxPowerThings like HDLC Abort problems and random TDM400P FXS modules lockups are the main reason I'm not 100% pro Digium cards.
21:59.41X-Robrussellb, you reading this?
21:59.45ozverenm20anyone know junghanns cards here ?
21:59.54krammanx: have you tried hardware hdlc?
22:00.02kramwhat sort of FXS lockups?
22:00.18X-RobI've never heard of FXS lockups.
22:00.25ManxPowerX-Rob, my issues with the TDM400P cards only seem to happen on production machines, and so are a miserable horror to diagnose.
22:00.30kramand do you have any support ticket numbers?
22:00.31ManxPowerkram, no dialtone.
22:00.32X-Robbut, I don't use TDM00's 8)
22:00.50krammanx: what driver version?  There was a known issue with this with older zaptels
22:01.01tainted-kram u'd better go code, u'll be fielding digium hardware problems all day
22:01.03X-RobHey, kram. Why are digium tech support sending people who buy tdm400's to #freepbx, purely because they're using A@H or freePBX?
22:01.19ManxPowerkram, I could look them up.  We fixed the problem by rebooting the server every monday morning.
22:01.30X-Robthere's no magic zaptel differences that we make.
22:01.35kramas i understand it from tech support A@H is very difficult to support outside of the GUI
22:01.40krambut i have no first hand experience
22:01.42ManxPowerkram, what qualifies as "older zaptel"
22:01.47tainted-it's macro madness
22:01.47kramx-rob are you involved with a@h?
22:01.47robin_szX-Rob,  *@h != *
22:02.00X-Robkram, no. They want help setting up their TDM400's. All they want is the FXS and FXO ports set up in zapata.conf
22:02.03krami think most of the 1.0 series, i'd have to look for sure
22:02.10X-Robkram, I'm a freePBX developer, which is the gui that A@H uses
22:02.28X-RobfreePBX does everything else for them, except for zapata.conf
22:02.31X-Roband zaptel.conf
22:02.44kramx-rob: any reason you guys don't do that?
22:02.48X-Robtoo hard? 8)
22:02.51kramx-rob: maybe we can setup a call or something
22:02.58ManxPowerkram, we gave up before 1.2 was released and reboot the server every monday.  We seldom have problems now.  The odd FXS module just stops working, but I assume that's just fried.  We are moving to SIP phones.
22:03.02kramif you're interested just send an e-mail
22:03.05tainted-X-Rob exactly
22:03.20X-Robkram, to which email?
22:03.26krammarkster@digium.com
22:03.31X-Robcoolo
22:03.59kramif you know the a@h people we might involve them too down the road
22:04.03X-Robnah
22:04.09kramhave they considered changing their name yet btw?
22:04.10X-Robthe a@h guy pretty much keeps to himself
22:04.24kramit's really causing us a lot more confusion than i would have originally thought
22:04.41X-RobHe doesn't get on IRC, and barely responds via email.
22:04.46kram*nods*
22:04.49ariel_kram, asterisk@home uses freepbx but it's just an ISO which puts things together as a boot ISO for install.
22:04.54kramyah
22:05.58ariel_name change for asterisk@home is something we all have email him about. But he never replys.
22:06.19X-Robwhy do you want him to change the name?
22:06.41ariel_@home
22:07.00X-Robwe've changed from AMP to freePBX because we're also gunna be supporting softswitch/openpbx etc if they ever get to a stage where they need a seperate dialplan maker.
22:07.10ozverenm20kram: does digium offer me a solution to on card switch ?
22:07.37tainted-X-Rob sweet!
22:07.38file'round the world... are you travelling
22:08.14*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net)
22:08.28tainted-X-Rob i think it's great that u cater to the 'i just want shit to work' market
22:08.46tainted-X-Rob it's much needed
22:08.47*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
22:09.01X-Robtainted-, it's not just that - we've also got hooks everywhere for people who want to futz around with stuff
22:09.13X-Robadmittedly, people have to actually read the .conf files to see that the hooks are there
22:09.42X-Roband yes, the extensions.conf _is_ insanely complex, but we've cut that down a hell of a lot since the AMP days
22:10.00tainted-well it's understandable
22:10.48tainted-i prefer to duct tape asterisk to agi apps to minimize dialplan logic
22:10.51kaz0358do you know an easy way to make _*X. work while entering the number in over dial tone? :)
22:11.04kramoz: the TE410P/TE405P and families do support on card switching yes
22:11.57X-Robtainted-, we had a big discussion about that a year ago, and we decided to do as much as we could in dialplan logic, pretty much 'coz it was there' 8)
22:11.59tainted-pulls biz logic out of voip box so i can migrate to freeswitch or whatever whenever they get a pbx module together
22:12.16kaz0358x-rob, it seems that asterisk is looking for *67 and such and prevents you from dialing something like *373612
22:12.44X-Robkaz0358, I don't know why you're telling me that.
22:12.53tainted-i've heard the argument that dialplan logic is X times faster than agi
22:12.54X-RobI know that the zaptel channels have some built-in feature codes.
22:13.00Assidanyone know a decent place for cheap DID's
22:13.08Assidlike 2-3 bucks
22:13.09Assidtops
22:13.09kaz0358x-rob, you mentioned that extensions.conf is much easier than it use to be. heheh.. i still haven't figured out how to make that work yet
22:13.47X-Robkaz0358, the extensions.conf that's supplied with freePBX is less complex than the extensions.conf supplied with AMP
22:14.02tainted-but to me, hardware is cheap, and time to market is much more important
22:14.27tainted-X-Rob is srvlookup = yes by default in freepbx?
22:14.50X-Robnope
22:15.13X-Robpurely because I've got zero experience with it
22:15.40kaz0358AMP? freePBX? i just compiled and downloaded the source.. i am happy with the way i have my extensions.conf setup. i'd just like to be able to dial *373612 for instance to use sip broker as the lookup for the uri of the provider
22:15.41tainted-some providers don't have SRV records, so it's best to set = no
22:16.03tainted-i've had to support a few freepbx users who couldn't get up and running b/c of that
22:16.55X-Robtainted-, we're using trac as a bugtracker now - if you think it's an issue, raise a ticket. But when I looked at it last time it caused more issues than it solved.
22:17.04X-Robwww.freepbx.org/newticket I think
22:17.06*** join/#asterisk avilv (i=justacas@Real.IRC.Masters.Use.Bitch-X.us)
22:17.09tainted-k
22:17.33*** join/#asterisk anthm (n=anthm@000-450-899.area4.spcsdns.net)
22:17.33*** mode/#asterisk [+o anthm] by ChanServ
22:17.36avilvwhats up everybody?
22:18.55*** join/#asterisk rigid (n=The@port-212-202-73-9.dynamic.qsc.de)
22:19.00rigidre
22:19.24avilvits.... quiet :(
22:20.22rigidi have a asterisk configured to forward 2 sip-accounts to the 2 lines of a sip/analog box... i see calls to <mynumber> (which work) and i see calls to <mynumber>0 (that don't work) when using "sip debug"
22:20.44*** join/#asterisk ozverenm21 (n=ozverenm@162.27.103-84.rev.gaoland.net)
22:20.44rigidthe destination number is then <mynumber>0 ... how can i configure this number in asterisk?
22:21.08ManxPowerrigid, just like you set it up in extensions.conf
22:21.13rigidto be exact, i want to MP3Player() a file when it's called...
22:21.30ManxPowerrigid, are you using AMP/FreePBX/Asterisk@Home
22:21.57rigidManxPower, no... i'm using a compiled packet (gentoo ebuild)
22:22.24*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
22:22.44rigidjust adding: "exten => 123450,0,MP3Player(/var/lib/asterisk/mohmp3/song.mp3)" doesn't work
22:22.54rigidManxPower, (if my number is 12345)
22:23.05ManxPowerrigid, priorities start at 1 not at 0
22:23.05avilvCould somebody tell me why my 2 month old A104 works perfectly but the Digium Wildcard TE411P doesn't?
22:23.12rigidwhile calls to 12345 work (using Ringing)
22:23.14avilvwhich I just got earlier today
22:23.16rigidManxPower, ahh
22:23.30b4kasangoma > *
22:23.33*** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu)
22:23.42avilvasterisk just segfaults when I try to make a zap/ call using the wildcard
22:23.52avilvand i recompiled the whole thing
22:23.58tainted-i heard there are asterisk advisory council trading cards
22:24.09kramavilv: have you placed a trouble ticket with tech support?
22:24.10tainted-is that true?
22:24.23avilvtainted-: this wildcard I picked up must jsut be one
22:24.24krami haven't heard any such thing
22:24.25avilvcompletely useless
22:24.43fileyou get technical support with the card, so use it...
22:24.51rigidManxPower, doesn't work either :(
22:24.51ManxPoweravilv, what specific model of Digium card do you have?
22:24.53kramwhat was completely useless?
22:24.57kramdo you have a ticket number?
22:24.58rigidManxPower, with priority 1
22:25.12ManxPowerrigid, did you do a reload after changing it?
22:25.20rigidManxPower, ;) ofcoz
22:25.45rigidManxPower, ok... s.o. that starts priorities with 0 could also forget to reload i admit :)
22:26.29avilvi got the card on ebay
22:26.48ManxPoweravilv, Then I strongly doubt that is an actual Digium card.
22:27.26avilvhttp://cgi.ebay.ca/Digium-Wildcard-TE411P-Quad-T1-ASterisk-VoIP-PCI-Card_W0QQitemZ9718822238QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
22:27.26kramavilv: so what kind of card is it?
22:27.28Strom_Ceither that or the card got destroyed by an inept user who tried to pawn it off on ebay
22:27.41kramavilv: okay so did you call or e-mail tech support yet about your problem?
22:27.57ManxPowerStrom_C, I saw someone recently that got a "Genuine X100P Clone", complete with hologram sticker.
22:28.09Strom_Chahahahahahahahahahahahahahahaha
22:28.11Strom_Coh god
22:28.13Strom_Cthat tickles
22:28.24rigidManxPower, the incoming call is to uri="sip:123450@asterisk-server" ... don't i have to add a new context or something?
22:28.54avilvkram: no. but I just got it to work bridging to an IAX call and it drifts more more when I dial dtmf :(
22:29.06Strom_Cavilv: what version of zaptel and asterisk are you running, out of curiosity?
22:29.20avilvthe latest
22:29.23budmangis FREEpbx nice?
22:29.38Strom_Cavilv: by "the latest" do you mean "stable" or do you mean "svn trunk"?
22:29.39avilv1.2.7 for asterisk
22:29.39mds2anybody have regular stability issues with WCTDM24XXP cards using 2.4.32?
22:29.43avilv1.2.5 for zaptel
22:29.52kramavilv: if you didn't contact tech support then how are we supposed to help you?
22:30.11Splasanyone know if it's at all possible to play the voicemail messages asterisk can attach to the notification emails on a blackberry.. specially an 8700
22:30.37tainted-where's your fucking support ticket asshole
22:30.48filetainted-: okay that was just rude
22:30.56Strom_Ctainted-: chill, for fuck's sake
22:31.04avilvit should just work. when its in dual opteron with a nice supermicro mobo and the latest fedora with the latest asterisk and zaptel
22:31.06avilvit should just work!
22:31.23fileavilv: but it's not, so you contact support so they can troubleshoot and try to solve the problem
22:31.32Strom_Cavilv: was the card used or brand new?
22:31.36ManxPoweravilv, Why do you expect a non-name fake card to work with Asterisk?
22:31.38avilvbrand new
22:31.38Strom_Cwoohoo, I'm a training card
22:31.42Strom_Cer
22:31.43Strom_Ctrading card
22:31.47avilvI got it off voip supply on ebay
22:31.52avilvso i hope its not a dud
22:31.55Strom_Cyes, I'm thinking, really I am
22:32.36kramit should be good
22:32.40ManxPoweravilv, Digium has not sold X100P/X101P cards in at least 2 years.
22:32.42kramyou can contact either of us for support
22:32.48kramthis is a TE411P
22:33.04ManxPowerAh!  nevermind
22:33.07avilvthis is fucking bullshit. when I want to benchmark and move to all digium cards and I don't even have a good testing platform (because asterisk segfaults or gives bad quality) then what am I supposed to do?
22:33.11kram:)
22:33.28kramyou're supposed to contact tech support so they can figure out what's wrong with your asterisk installation
22:33.34kramsupport@digium.com
22:33.43kramif you haven't had a happy resolution in 3 days you can drop me an e-mail
22:33.47ManxPoweravilv, You have 2 choices.  1) return the card, or 2) contact tech support.
22:33.48kraminclude your support ticket number
22:33.54ManxPowerI recommend option 2
22:34.05Strom_Cavilv: calm down, man.  there may be a minor problem or the card may have gotten damaged in shipment or something.  bitching about it won't help, but talking to support will.
22:34.07krami'd like to point out that option 1 is the dumb one since changing hardware won't change an asterisk segfault
22:34.30tainted-can i start trolling about asterisk yet?
22:34.31avilvok i will do that
22:34.35russellbtainted-: no.
22:34.42ManxPowerkram, option 1 is dumb, but this guy is screaming about problems without evening having contacted support.....
22:34.46tainted-i have some beef with scaling
22:34.46avilvkram: are you mark?
22:34.51budmangI am looking for an easy install of asterisk with a web interface. Any suggestions?
22:35.04ManxPowerbudmang, try #freePBX
22:35.06Strom_Cbudmang: switchvox ;)
22:35.17avilvwhile i'm hear i'm just curious
22:35.27russellbavilv: yes
22:35.35avilvwhy does the console say avoiding deadlok on every iax call i make?
22:36.01rigidManxPower, can i somehow debug how asterisk decides when walking through extensions.conf?
22:36.03ManxPoweravilv, your name isn't RoyK is it?
22:36.09tainted-b/c ur supporse to use SIP for calls dude
22:36.16tainted-IAX is just for NAT traversal
22:36.23avilvno my name is jason
22:36.30ManxPowerrigid, ask me on a week day.
22:36.38rigidManxPower, :)
22:36.53ManxPowerrigid, I'm too tired to do complicated dialplan debugging right now.
22:36.57blitzrageManxPower: yo
22:37.24ManxPowerhiya, royk
22:37.33ManxPowerblitzrage, 'sup?
22:37.48rigidManxPower, hmm... how to make asterisk extension-stuff more verbose, you don't know by heart?
22:37.55rigidManxPower, i can debug it then myself
22:37.56avilvno answer?! THIS IS FUCKING BULLSHIT! I'M BUYING ALL SANGOMA CARDS ON MONDAY AND AM WAITING TIL THAT FREESWITCH COMES OUT
22:38.07ManxPoweravilv, go for it
22:38.10russellbavilv: what are you talking about?
22:38.18ManxPowerrigid, asterisk -rvvv
22:38.29blitzrageManxPower: not to much, just listening to tune
22:38.31blitzragetunes*
22:38.52tainted-damn
22:38.54Strom_Cnote to the short-tempered: you can get wall padding professionally installed now.
22:39.00rigidManxPower, tnx
22:39.53Assidanyone here using sipbroker?
22:40.16RoyKManxPower: wrong guy :)
22:40.21ozverenm21does it plained to asterisk to work as stateless sip proxy ?
22:41.02RoyKozverenm21: asterisk != proxy
22:41.19RoyKozverenm21: what you're looking for is something like SER
22:41.31ManxPowerMaybe avilv is The Person Formerly Known as Timecop....
22:42.00tainted-ozverenm21 u need to run SER/asterisk
22:42.03ManxPowerI'm REALLY REALLY REALLY starting to hate working with DVD video.
22:42.10tainted-ser handle SIP headers, asterisk handle media
22:42.48ozverenm21maybe but asterisk b2bua is not natural and break somes CDR functionnalities because callid change
22:43.01xachen:O
22:43.06xachenthat avilv has a short temper
22:43.48X-Robok, that was funny.
22:44.04tainted-ozverenm21 if u need to do billing, ur in for a ride
22:44.57tainted-ozverenm21 how soon are u going to production?
22:45.31tainted-ozverenm21 or are u already on asterisk life support
22:45.44ozverenm21i am using asterisk in production, just  for doing testing on voip and isdn ...
22:47.03ozverenm21do you plain to add alcatel UA protocol support in a* ?
22:47.35tainted-ozverenm21 try #asterisk-dev
22:48.05russellbany UA should work.
22:48.07*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
22:48.08russellbthat's not a dev question.
22:48.08surfduehi
22:48.14surfdueI cant connect to asterisk with my linksys pap2?
22:48.17surfduecan anyone help
22:48.29Strom_Csurfdue: are they both on the same network segment?
22:48.36surfduewhat do you mean?
22:48.39xachensurfdue: #freepbx
22:48.44surfduehuh
22:48.48xachenyour using FreePBX?
22:48.54surfduemaybe no.
22:48.59RoyKtainted-: asterisk cannot handle media itself, it needs to do both
22:49.05anthmsure blame the segment it's segment's fault!
22:49.21Strom_Csurfdue: are the asterisk box and the PAP2 on the same subnet?
22:50.04surfdueno
22:50.12surfduestorm_c its not local
22:50.22Strom_Csurfdue: is the asterisk box behind NAT?
22:50.25*** join/#asterisk talljon84 (n=jonathan@66-188-104-144.dhcp.mdsn.wi.charter.com)
22:50.34Strom_Cand is the PAP2 also behind NAT?
22:50.36surfdueStrom_C, its on a dedicated server
22:50.43xachenso the server wouldn't b
22:50.45xachenI hope
22:50.52RoyKa dedicated server may well be behind nat, surfdue
22:50.52surfdueStrom_C, the pap2 may be i did setup port forwarding though
22:50.52Strom_Csurfdue: dedicated or not, is there NAT involved?
22:51.07talljon84Is anyone aware of a SIP client for Palm OS? If not, any suggestions where I could post a bounty for one?
22:51.08surfdueStrom_C, i dont think so
22:51.19xachenrather
22:51.20Strom_Csurfdue: is it behind a firewall?
22:51.24RoyKtalljon84: there are a few for wince
22:51.24xachenis your PAP2 behind a firewall
22:51.37RoyKfirewalls are for pussies :)
22:51.37talljon84RoyK: Yes, but I have a Treo 650 so that won't work.
22:51.42*** join/#asterisk wenko (n=wenko@142.232.8.200)
22:51.48wenkohey there
22:51.50surfduemy pap2 is but the ports unblocked
22:51.51surfdueAND
22:51.55wenkoanyone here good with iaxi?
22:51.55surfduemy other line works
22:51.56surfdue:|
22:52.23xachenRoyK: no, the only firewalls to use are pf and ipf
22:52.25surfdueits from another provider though
22:52.25Strom_Cwhich "other line"?
22:52.26surfdueit has 2 lines
22:52.26Strom_Coh, on the PAP2 itself
22:52.26RoyKtalljon84: http://www.google.com/search?client=opera&rls=en&q=sip+palm&sourceid=opera&ie=utf-8&oe=utf-8
22:52.29wenkoi need to do a password recouvery on my iaxi, is that possible??
22:52.49surfdueStrom_C, is there a test i can run to see if its behind a NAT the server?
22:52.53RoyKwenko: google for 'hammer'
22:52.59wenkohammer?
22:53.10Strom_Csurfdue: um, log in and then run ifconfig? :)
22:53.25RoyKwenko: a good tool by Steel or Iron
22:53.33talljon84RoyK: That was my first idea. The first heading doesni't really exist (hoax it appears). The rest are all talking about the lack-of-a-client.
22:53.38surfduewhat am i looking for Strom_C ?
22:53.42wenko...that means im fucked right??
22:54.04RoyKwenko: sorry. don't know. contact digium :)
22:54.13Strom_Csurfdue: whether or not the IP address of the ethernet interface is the same one that faces the public network
22:54.16wenko:S
22:54.25wenkoi get an error when trying to telnet
22:54.25Nuggettelnet is eeeeeeevil.
22:54.38wenkoit tells me that there is no password set
22:55.02Dr-Linuxsurfdue: what does say >>   ifconfig
22:55.07surfdueStrom_C, it does
22:55.10surfdueit is
22:55.27Strom_Csurfdue: is the PAP2 failing to register?
22:55.29xachensurfdue: Is this server the one I was on earlier?
22:55.39*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
22:55.41surfduexachen, yes
22:55.44xachenoh
22:55.49xachenthere is no NAT on that machine then
22:55.55xachenLayeredtech doesn't NAT servers
22:55.58talljon84Is anyone aware of a deb package that will provide the libstdc++.so.2.8 lib that's required for some games?
22:56.03RoyKwenko: http://www.google.com/search?hs=dze&hl=en&lr=&ie=UTF-8&oe=UTF-8&client=opera&rls=en&q=iaxy+reset+password&btnG=Search
22:56.08talljon84err. damnit. wrong room, sorry
22:56.18Strom_Csurfdue: is the PAP2 failing to register?
22:56.23surfdueyes
22:56.31Strom_Cis asterisk spitting out an error?
22:56.32surfdueRegistration State:Can't connect to login server
22:56.41surfdueStrom_C, whereis hte log file again?
22:56.53Strom_Cjust connect to the console and see if there's an error there
22:57.34Dr-Linuxsurfdue: you are using A@home?
22:57.42surfduewhats the link
22:57.44surfdueno
22:57.48*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net)
22:58.02Dr-Linuxsurfdue: type asterisk -r
22:58.13surfdueoh
22:58.13Dr-Linuxif asterisk is already running
22:58.14surfduelol
22:58.17Strom_Csurfdue: what link?  log into the box and connect to the asterisk console after you're whatever user asterisk is running as
22:58.17surfdueit is
22:58.55Dr-Linuxsurfdue: the try debug
22:59.04surfdueim guessing you mean type debug?
22:59.06Dr-Linuxthen*
22:59.07Strom_Cnot yet
22:59.10Strom_Cset verbosity to 10
22:59.13surfduek
22:59.19Strom_Csee whether asterisk is kicking out an error
22:59.30surfduewhats the command
22:59.30surfduelol
22:59.35Strom_Cset verbose 10
22:59.51surfduek
23:02.18Strom_Canything yet?
23:02.28surfdueno ?
23:02.42Strom_Ctry rebooting the PAP2 and then see what happens
23:04.03*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
23:04.04surfduek
23:05.21orlokHey, has anybody here ever used the Sail asterisk manager?
23:05.56surfduelink me orlok
23:06.01surfdueStrom_C, no luck nothing errors?
23:06.30Strom_Csurfdue: well, OK, then it sounds like it's a PAP2 issue
23:06.41surfdueStrom_C, heres whats odd
23:07.00surfduei use asterlink and I have it connected directly to asterlink on line 1
23:07.03orloksurfdue: k, it runs on top of sme server
23:07.08surfdueand on line 2 asterisk which is linked to asterlink
23:07.13orlokwhich is based on centos
23:07.59Strom_Csurfdue: do a sanity check and make sure you have all the settings correct on the PAP2
23:08.03orloksurfdue: http://contribs.org/modules/phpwiki/index.php/SME7Contribs has the link to the addon fo smeserver
23:08.21orlokit seems to work, just that i get errors trying to dial out
23:08.28surfdueStrom_C, sanity check hmm ?
23:08.41Strom_Cyes
23:08.45Strom_Csanity checks are good things
23:08.55wenkoi dont understad how to provision an iaxi, anyone able to help me out?
23:08.57fileinsanity check.
23:09.01xachenhaha
23:09.11Strom_Ci like those too ;)
23:09.11xachen100% in my case
23:09.42surfdueStrom_C, is there a port check i can do to make sure i can talk to my pap2 through my firewall?
23:10.18Strom_Csurfdue: you just said that you're able to talk SIP through your firewall direct to asterlink, so that rules out firewall issues in my book
23:10.31surfdueya
23:10.32surfduejust making sure
23:10.33Strom_Csurfdue: like I said, do a sanity check and make sure your settings are correct
23:10.53*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net)
23:10.57filedid you do a sip debug on your personal server and then restart the PAP2?
23:13.00surfdueStrom_C, http://img244.imageshack.us/my.php?image=untitled1nm.jpg
23:13.08*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net)
23:13.15surfduefile i dont know how to do sipdebug
23:13.37fileyou type sip debug
23:13.40Strom_Csurfdue: that doesn't help me at all.  I don't know the name of your server
23:13.43fileand then watch SIP traffic appear
23:14.31Strom_Csurfdue: for your asterlink config do you have a domain name or an IP address in the "SIP Proxy" field?
23:14.35xachenYou know your a nerd when you watch SIP traffic flow through :P
23:14.54filehehe
23:14.56Strom_Coh baby oh baby oh baby SIP me harder
23:14.57surfduestrom i have the name
23:15.14Strom_Csurfdue: your asterisk box is just "host41.com"?
23:15.33*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
23:15.44surfduewell its an ip
23:15.47surfdueya
23:15.48surfduelol..
23:15.54surfduehost41 directs to the ip
23:16.01Strom_Cand host41.com resolves to the correct IP?
23:16.20surfdueyep
23:17.35Strom_C72.232.27.132 right?
23:17.39surfduelook at sip debug http://pastebin.com/691309
23:17.45surfdues aparently it works ya Strom_C  :)
23:17.57surfduethe sip debug looks fine but not logged in still?
23:18.27Strom_CSIP/2.0 404 Not found
23:18.30Strom_Cthats no good
23:18.31surfduewhats that mean :P
23:18.43fileset nat=yes
23:18.51surfduein the pap2?
23:18.56Strom_Cshow me the relevant section of sip.conf
23:18.57fileno, sip.conf
23:19.08surfduek
23:19.17fileand do whatever Strom says
23:19.27Strom_Cdo the hokey pokey
23:20.02surfduenat isnt in sip.conf ?
23:20.28fileput it in general, nat=yes and do a sip reload
23:20.38surfduek
23:20.42surfdueat the top?
23:20.43surfdueim guessing
23:20.44fileand get the PAP2 to register again... and see what happens
23:20.51filein the general section, somewhere
23:20.51surfduenvm
23:20.52surfduei see it
23:21.25Strom_Chey, speaking of which: is there an IAX provider that offers local DIDs in the los angeles area with unlimited inbound and no restrictions on concurrent calls?
23:21.35surfduereregistering
23:21.52*** join/#asterisk msg43 (n=msg43@adsl-68-255-187-222.dsl.bcvloh.ameritech.net)
23:22.03msg43surfdue, did you realize what you did
23:22.17msg43surfdue, anyone who is from my area that wants to join the irc channel can't
23:22.35surfduemsg43, please dont spam this room about other rooms matters.
23:22.37msg43surfdue, great job on getting rid of costomers
23:22.54msg43Nah I'm just point out what I moran you are :)
23:23.09Strom_Cmsg43: take the drama elsewhere, please.
23:23.13surfdue.
23:23.15*** join/#asterisk kietlak (n=kietlak@apn-99-84.gprspla.plusgsm.pl)
23:23.38msg43Strom_C, I just like pissing of surfdud
23:23.57surfdueplease stop this msg..
23:24.00surfduethats why your banned.
23:24.24surfdueStrom_C, sorry anyways I am still getting cant register its odd
23:24.27msg43surfdud, we can take to a personal level in pm
23:24.33surfdueStrom_C, i dont see sip404 anymore though?
23:24.40surfduemsg43, dont pm me.
23:24.49Strom_Csurfdue: show me your sip.conf
23:25.02msg43surfdue, woops already did
23:25.05russellbsurfdue: /ignore works wonders
23:25.06msg43surfdue, what you gonna do
23:25.16russellbmsg43: please take this elsewhere ...
23:25.16surfduearg, who has ops in here?
23:25.27surfdueStrom_C, It has passwords lol let me take them out
23:26.32russellbgutes?
23:26.38Strom_Cwhat the hell are gutes?
23:26.51surfdueStrom_C, http://pastebin.com/691325
23:26.55russellbStrom_C: i hate your gutes so much!
23:27.10Strom_Crussellb: i hate your gutes even more!
23:27.27surfduelol..
23:27.33ihateyourgutessurfdue, your so cute
23:27.38ihateyourgutesbanning everyone :)
23:27.39ihateyourgutesahh
23:27.45Strom_Csurfdue: so, uh, where is the entry for your PAP2?
23:28.06surfdueStrom_C, is tehre suppose to be ? :P
23:28.10Strom_Cduh
23:28.25surfduewhat do i need to add
23:29.24Strom_Cone sec
23:30.38Strom_Chttp://pastebin.com/691335  is a sample entry for one of my PAP2 SIP lines
23:32.15Strom_Cwelcome back, kram
23:32.19kramthank ye
23:35.01Strom_Ci really need to bring an ethernet connection here into my bedroom so I can get my bedside phone as an extension off my pbx...
23:35.19orlokhmm..
23:35.23orloki'm having dialplan issues
23:35.55orloki think - is there any way to debug why a call doesnt go through?
23:36.03*** join/#asterisk esculapio_ (n=ESCulapi@145stb68.codetel.net.do)
23:36.06Strom_Corlok: what is your specific problem?
23:36.35orlokStrom_C: asterisk server is also the net gateway. phones behind asterisk ca dial each other, but get a fast busy signal trying to dial outbound
23:36.49Strom_Corlok: paste extensions.conf please
23:36.51Strom_Cer
23:36.53Strom_Cpastebin
23:37.15Strom_Cand btw, that fast busy signal is called a reorder
23:38.39*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr)
23:38.46orlokhttp://pastebin.ca/52450
23:39.24Strom_Cwell theres your problem
23:39.40Strom_Cyou cant wildcard match if you dont put an underscore first
23:40.00ms34surfdue = http://z.about.com/d/politicalhumor/1/0/n/U/moran.jpg
23:40.01ms34surfdue = http://z.about.com/d/politicalhumor/1/0/n/U/moran.jpg
23:40.22Strom_Corlok: exten => XXXXXXXXX,1,agi(selintra,OutRoute,Outgoing) is the problem line
23:40.34surfdue:|
23:40.41surfduestop spamming channels!
23:40.50*** mode/#asterisk [+b %ms34!*@*] by russellb
23:41.28orlokheh
23:41.30orlok#teenlinux
23:41.48*** kick/#asterisk [ms34!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (bye!)
23:41.57orlokStrom_C: XXXXXXXXX is specifiable, what should it be?
23:42.08Strom_Cwhat do you mean "specifiable"
23:42.13Strom_Cyou're doing wildcard matching, right?
23:42.48orlokStrom_C: I am setting it up using sail, which is an asterisk config manager using the e-smith templating system
23:43.01Strom_Cto me, that sounds like "blah blah blah blah"
23:43.03orlokyes, i am
23:43.23orlokStrom_C: Well, what specifically about that line is the problem?
23:43.26Strom_Corlok: therefore, if you are doing wildcard matching, you must prefix the wildcard characters with an underscore
23:43.30orlokahh
23:43.34orlokcool
23:43.39Strom_Cwhich is what I already said
23:43.48orloki always saw the underscore, but the docs never seemed to mention it, only XXX and whitespace
23:43.57orlokahh, yeah
23:44.04orlokup before the flood from ms34
23:44.08Strom_Cyes
23:44.55orlokok, now its _XXXXXXXXX, same issue
23:45.01Strom_Cyou did a reload, right?
23:45.34orlokyup
23:45.46Strom_Cwhat is the console saying?
23:46.36orloknothing when i try to dial
23:46.42Strom_Cset verbose 10
23:46.44Strom_Cdial again
23:46.55*** join/#asterisk msg43 (n=msg43@adsl-68-255-187-222.dsl.bcvloh.ameritech.net)
23:47.12ms89rule 1, never ban a nickname
23:47.16ms89:)
23:47.19*** mode/#asterisk [+b *!*n=msg43@*.dsl.bcvloh.ameritech.net] by russellb
23:47.19*** kick/#asterisk [ms89!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb)
23:47.41surfduerussellb, atleast there is an op in here..
23:48.41russellbheh, you're welcome...
23:48.47russellbi'm trying to let you guys talk in peace
23:52.23orlokHmm
23:52.38orlokshould the sip povider asterisk is registering with show up in sip show peers?
23:52.53Strom_Corlok: what.  does.  the.  console.  say?
23:53.54orloknothing!
23:54.04Strom_Cis your AGI even working?
23:54.25Strom_C(speaking of which, why in god's name are you using an AGI to dial when Dial() will work much more cleanly?)
23:55.04*** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net)
23:55.26orlokStrom_C: Only time i've ever gotten asterisk working was using Asterisk@Home
23:55.32Strom_Cugh
23:55.35Strom_CI hate guis
23:55.36orlok<PROTECTED>
23:55.40orlokyeah, so do i
23:55.48Strom_C!thebook
23:55.49Strom_Cer
23:55.52Strom_C~thebook
23:55.53jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
23:55.59*** join/#asterisk Isaiah (n=Isaiah@208-187-93-4.br1.hnv.mi.frontiernet.net)
23:56.04orlokgot it printed out next to m
23:56.27Strom_Corlok: you should try to set up a system without any guis or prototyping tools or anything
23:56.32Strom_Cjust you, asterisk, and vim
23:56.42orlokyeah
23:56.57orlokcos i dont know what the gui/framework system i'm using now is or isnt doing right
23:57.07orlokand its a layer of abstraction that pisses off the people that do know

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