IRC log for #asterisk on 20081001

00:08.05ManxPowerscooby2: plug in a loopback plug
00:08.18ManxPoweryou can do a software loopback via zttool
00:08.34scooby2obviously but I would like to go home and not have to sit here waiting for the telco:)
00:08.35ManxPowerbut that is mainly useful only for the telco testing the lines.
00:08.48ManxPowera loopback plug will test the card, port, and config
00:09.42scooby2correct
00:10.01scooby2but sometimes they like it looped back the other way to see if they can see it. If its a line issue or not
00:14.13ManxPowerscooby2: they can loop to the smartjack.
00:14.17ManxPowerthat is enough for line testing
00:14.41*** join/#asterisk SuxieDapper (n=SuxieDap@190.245.108.2)
00:14.44SuxieDapperHello
00:15.23SuxieDapperDoes anybody know if I can send SMS with my Vonage account?
00:15.37drmessanoWUT
00:15.50ManxPowerSuxieDapper: no you cannot do that
00:16.00ManxPowerAsterisk SMS only support the European SMS standards
00:16.56SuxieDapperSomebody told that I could use my Vonage account to send SMS with it. But I wasn't sure if it was using Asterisk, some Vonage site or something like that.
00:17.04drmessanoYeah, Asterisk doesn't support 568A SMS
00:17.09drmessanoonly 568B
00:17.15drmessanoSorry
00:17.37C4colook, g729 question
00:17.54C4cololet's assume I"m not in latvia and I need to pay for the licenses
00:17.56scooby2and speak of the devil. GBLX circuit is now up/up.
00:18.14SuxieDapperThank you guys
00:18.24C4coloI have a sip gateway, a call router, and a registration/feature server
00:18.28C4colothe call hits all three systems
00:18.46C4colo(I have multiples of each, but for the sake of simplicity let's assume I have only these three systems)
00:19.16drmessanoLike in series?
00:19.27C4colodo I need to license g.729 on EVERY system even though the call will be routed through these systems
00:19.29C4coloyea in series
00:19.38drmessanoAre any of them transcoding?
00:19.46C4colofirst and last might
00:19.47drmessanoRemember it's a "transcoding license"
00:19.57drmessanoSo anywhere you're transcoding
00:20.04C4colohmm
00:20.28*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
00:20.36C4coloso sip phone (729) -> asterisk -> asterisk -> asterisk -> PSTN
00:20.56C4colothe last one would need a license to transcode to the PSTN right?
00:21.17drmessanoI believe so, since Asterisk is doing media conversion
00:21.34C4colohow does one configure the other servers to pass g.729 through without needing a license?
00:21.48C4colojust install the codec?
00:22.42drmessanoI know "out of the box" * supports passthru..
00:22.49*** join/#asterisk pcrane (n=pcrane@202.49.106.158)
00:22.52C4colooh?
00:22.55C4colohmm
00:22.57C4colonever tried it
00:23.41drmessanoI dont know how to explain it.. But it will pass G729 but not transcode.. I guess it comes with a Passthru codec (like the early G722 support)
00:24.02*** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com)
00:24.35drmessanoI'm gonna assume the licensed codec is smart enough to count properly
00:24.54C4coloyea
00:25.07C4coloI'll have to do some testing
00:25.29drmessanoA default install will lock you into passthru only.. which actually may be better for testing
00:25.46drmessanoEasier for it not to work than to keep watching the CLI, i guess
00:28.25*** join/#asterisk pixlated (n=not@ool-44c0746a.dyn.optonline.net)
00:33.18*** part/#asterisk jan1607 (n=niewerth@dslb-084-061-121-080.pools.arcor-ip.net)
00:41.40*** join/#asterisk hi365_m (n=hi365@213.151.44.101)
00:57.07*** join/#asterisk Itiliti (n=Itiliti@216.159.238.110)
01:02.57ItilitiI am trying to get faxing working through my PRI. I am running spanDSP spandsp-0.0.4-9.pre15.i386, and app_nv_faxdetect-1.0.6_1.4 and background detect of the same version. But when it detects the fax and kicks it over to the rx_fax app, it crashes asterisk? Any ideas?
01:03.27Itilitithis is only happening when calls coming in on PRI..
01:04.35*** join/#asterisk marc7 (n=marc@S0106001ff33f8523.vc.shawcable.net)
01:04.53Itiliti.join asterisk-dev
01:05.32*** join/#asterisk cnielsen (n=Cole@209.181.98.68)
01:06.39cnielsenCan anyone tell me how to resolve an issue with Asterisk rejecting calls due to codecs on FC7 ?
01:06.54*** join/#asterisk Compy (n=compy@h123.192.18.98.dynamic.ip.windstream.net)
01:06.57cnielsenI'm using the FC7 packages
01:07.34tmckayI'm using ekiga.net for sip and trying to configure IPKall with it.  What do I enter for my SIP proxy?
01:07.45*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
01:07.45*** mode/#asterisk [+o mog] by ChanServ
01:08.05v4mphey guys any idea to this after u get the audio menu options it hangs up rather than w8ing for u to pick an option
01:08.06v4mphttp://pastebin.com/d9d2f949
01:08.34*** part/#asterisk cnielsen (n=Cole@209.181.98.68)
01:12.23*** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net)
01:12.30v4mpthis is my extensions.conf http://pastebin.com/d5bfe77b7
01:13.25l2trace99~softphones
01:13.27l2trace99crap
01:13.29l2trace99there's not jbot response for softphone recommendations ?
01:14.43*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
01:15.40*** join/#asterisk moy (n=moy@189.169.68.109)
01:17.37*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
01:17.37*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev
01:17.44v4mp~softphones
01:17.56v4mpl2trace99, u tyed x-lite ?
01:18.25l2trace99yes
01:18.38subdolusis there a way to send a DTMF tone down the line once a call has been established with asterisk?
01:18.47jeff_smokerIf you receive a gsm based call on a COM port, can you enter AT commands to forward that call once its been picked up?
01:19.02l2trace99i was looking other alternatives that support multiple accounts
01:19.36l2trace99and that I can remotely configure
01:20.20v4mphmm i used to know of 1 but i cant remember name of it no more
01:24.58l2trace99mmmmm  Ekiga for windows
01:26.48*** join/#asterisk mackes (n=root@cpe-76-180-145-138.buffalo.res.rr.com)
01:27.10*** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
01:34.39jeff_smokersubdolus: what are you trying to accomplish?
01:35.51subdolusi want to idle in a teleconference, but the conference kicks people if they dont press a button (send a DTMF tone) within a certain period of time
01:36.17subdolusso i want it to, say, every 2 minutes press a button / send DTMF
01:36.26subdoluspossible?
01:37.22jeff_smokersubdolus: I think it is, but I haven't worked with asterisk enough to know. i think you might be able to write a looping php script to do this.
01:37.53[TK]D-Fendersubdolus: Tricky to set up, but quite doable
01:38.51[TK]D-Fender~softphone
01:38.51jbot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
01:45.39jeff_smokerdoes anyone know of a voip service provider that lets you make a direct socket connection to their voip server?
01:46.28*** join/#asterisk subdolus (n=subdolus@subby.afraid.org)
01:46.59subdoluskicks router
01:47.15subdolusjoker, you still around matey?
01:48.20Compy~xlite
01:48.21jbot[~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/
01:49.28[TK]D-Fenderjeff_smoker: ....HUH?
01:49.47Carlos_PHXUm, yeah, why would you do that?
01:50.51[TK]D-FenderForget "why"... more like "what".  As in "what is that supposed to mean?"
01:52.00LiNeTuXYou know, a plug-in for the socket.  So you can port your XML over SIP for real-time Web 2.0.
01:52.16[TK]D-FenderWith *, SIP & RTP is all UDP.  Thats right, CONNECTIONLESS.  So did he meam to imply some sort of VPN-like connection over which he would alter pass that same traffic?  We may never know....
01:53.17Carlos_PHXHuh, interesting idea actually.  Like tunneling UDP in TCP without the overhead of GRE?
01:53.20[TK]D-FenderJust hearing someone saying "but I'm connected to the provider" makes me look at them funny...
01:53.32Carlos_PHXHeh
01:53.39[TK]D-FenderCarlos_PHX: Possible... not great on reliability, jitter, etc.....
01:53.52LiNeTuXwhy would you want TCP overhead?  <shrug>
01:54.26*** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net)
01:55.02Carlos_PHXYou get back information on the state of your connectivity.  I've only lightly experimented with SIP over TCP but there are some benefits.  I'm not saying it's great, just that there may be reasons.
01:55.05*** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net)
01:55.10jeff_smokerTDK: similar to how you can make a direct socket connection to a POP server, can you make a direct socket connection to a VOIP server and then just enter commands?
01:55.42[TK]D-FenderWith *, SIP & RTP is all UDP.  Thats right, CONNECTIONLESS.  So did he meam to imply some sort of VPN-like connection over which he would alter pass that same traffic?  We may never know....
01:55.46[TK]D-Fenderjeff_smoker: ^^^
01:55.58Compyick, voice over tcp... *quivers*
01:56.00[TK]D-Fenderjeff_smoker: there aren't jsut "commands".
01:56.23[TK]D-Fenderjeff_smoker: Can you try to describe a specific scenario of what it is your looking to do...
01:56.24LiNeTuXYou might imply the commands are in the UDP packets.
01:57.08[TK]D-Fenderjeff_smoker: SIP is not a scripting language, its a call setup and teardown protocol.
01:57.32jeff_smokerSo if you want to make VOIP calls without asterisk, you still need to pay some company that has a voip server connection to a pstn trunk, right?
01:58.02[TK]D-Fenderjeff_smoker: Asterisk has nothing to do with the PSTN in and of itself.
01:58.31[TK]D-Fenderjeff_smoker: ITSP's use their protocols and accounts to allow you access to the PSTN.
01:58.40[TK]D-Fenderjeff_smoker: * is not needed for this.
01:58.49jeff_smokerok, right
01:58.55[TK]D-Fenderjeff_smoker: You could cset up a softphone direct just the same
01:59.03jeff_smokerexactly, ok
01:59.20[TK]D-Fenderjeff_smoker: Again, these are CALLS, not just some kind of arbitrary "command"
01:59.40jeff_smokerSo what to know is if you write your own softphone application, you have to write a connection script to the ISTP yes or no?
01:59.51jeff_smokercorrection: what I want to knw
01:59.53jeff_smokerkonw
01:59.55jeff_smokerknow
02:00.20[TK]D-Fenderjeff_smoker: and "making voip calls" doesn't require anything other than 2 pieces of software communicating with each other.  You can point 1 softphone DIRECTLY at another an bam, thats a "voip call"
02:01.02LiNeTuXjeff: SIP is a protocol.  If you write an application to use SIP, you'd be able to connect to any provider that uses SIP.
02:01.07[TK]D-Fenderjeff_smoker: 1 : there is no CONNECTION.  No such thing.  Period.
02:01.18Carlos_PHXSIP is connectionless.
02:01.23*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
02:01.40jeff_smokerbut if you have to get access to the pstn, you have to make a connection to something right?
02:01.43Carlos_PHXI say "INVITE" and you reply, and I reply, and then we both start shooting voice packets.
02:01.48Carlos_PHXThat's all.
02:01.56Carlos_PHXjeff_smoker:  NO
02:02.04jeff_smoker??
02:02.04[TK]D-Fenderjeff_smoker: You place your call.  They ask for auth.  You give it.  They say OK.  You both set up RTP.  You Talk.  You say "I'm done", and thats the end of it
02:02.18Carlos_PHXYou are shooting packets at a PSTN gateway, but you don't literally "connect" in the networking sense.
02:02.23[TK]D-Fenderjeff_smoker: You need to go read a VoIP protocol primer guide.
02:02.28Carlos_PHXYup
02:02.36jeff_smokerso why use a voip service provider?
02:02.41jeff_smokerwhat role do they play?
02:02.48Carlos_PHXYou send your packets to them to send to the PSTN.
02:02.51[TK]D-Fenderjeff_smoker: because THEY will bridge your call to the PSTN.
02:03.13Carlos_PHXjeff_smoker:  Let me try to say it another way.
02:03.18jeff_smokerOk
02:03.23Carlos_PHX"Connection" in the context of networking means something specific.
02:03.29[TK]D-Fenderjeff_smoker: You can feel free to pass all the VoIP traffic around the world that you want but it won't magically get onto the PSTN unless someone is providing you termination <-
02:03.31Carlos_PHXSIP is connectionless.
02:03.58Carlos_PHX"Connection" in layman's parlance could be what you mean, IE you are negotiating with a PSTN gateway to connect you to the PSTN.
02:04.11Carlos_PHXBut at the networking level there is no connection.
02:04.31jeff_smokerCarlos, are you in Phoenix?
02:04.37Carlos_PHXYes
02:04.44jeff_smokerI'm in Tempe
02:04.49Carlos_PHXSmall world
02:04.57jeff_smokerKnow of any club meetings that have to do with VOIP?
02:05.13Carlos_PHXI've been talking with cohorts about setting something up, but nothing so far.
02:05.31Carlos_PHXHowever a few of us are open to beer and wings to talk VoIP here and there.
02:05.46jeff_smokerWell, I'd be very interested.
02:05.52jeff_smokerI'm at ASU
02:06.14jeff_smokerI just sold my party bus, otherwise I'd throw some girls in and host
02:06.29Carlos_PHXShoot me an e-mail at carlos at televolve.com and I'll drop you a line some time.
02:06.33Carlos_PHXHeh
02:08.55LiNeTuXCarlos_PHX: I'm jealous of you guys... you have The Yard House...
02:09.03jeff_smokerCarlos_PHX: Alright, thank...I just sent you an email.
02:09.08LiNeTuXdoesn't have good beer on tap where he is. :(
02:09.13jeff_smokerthank(s)
02:10.57jeff_smokerAlright, so if you have a softphone you can connect to basically anyone else who has a softphone as long as you point to them. But as far as PSTN call termination goes, that's when you have to pay someone $.019 / minute or whatever. And you guys are saying what protocol does that?
02:11.57[TK]D-Fenderjeff_smoker: the protocol is the carrier.  There are MANY VoIP protocols.  SIP, IAX2, H.323, MCGP, Jabber, Skype, and so on...
02:12.40jeff_smokerSo why choose one over the others?
02:12.41[TK]D-Fenderjeff_smoker: it does not matter WHICH one this "provider" uses.  Naturally though by choosing to acquire their service one would think that they would tell you how to actually USE it.
02:13.20*** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
02:13.30[TK]D-Fenderjeff_smoker: 1 provider over another?  Maybe one provider has cheaper rates to a country you want to call more often.  Maybe one provider is cheap but only lets you have 1 channel at a time when you need to support multiple.
02:13.33UnixDawgso is the new asterisk-now iso out ?
02:13.41jeff_smokerOther than voice quality and reliability, what are other considerations when selecting a provider?
02:13.48jeff_smokerand price of course
02:13.56UnixDawgthe one the had downloadable at astricon ?
02:14.01[TK]D-Fenderjeff_smoker: Maybe some providers utilization rates such and they run out of lines to terminate your calls so you can't even GET to the PSTN
02:14.24[TK]D-Fenderjeff_smoker: Or their servers are in a bad location and jitter, latency etc screw with your calls
02:14.27jeff_smokerWhat providers do you use?
02:14.38jeff_smokerCould you become your own provider?
02:14.46LiNeTuXjeff_smoker: latency is a big key in choosing a provider that works for you.
02:14.46[TK]D-Fenderjeff_smoker: Why are there so many cellphone carriers?  because some suck more than others <-
02:15.28[TK]D-Fenderjeff_smoker: Sure.  You have an analog line?  get a piece of hardware that lets you connect your line and something that will accept VoIP calls and use it <-
02:15.38[TK]D-Fenderjeff_smoker: BTW... that would be ASTERISK
02:15.43jeff_smokerIs there any kind of definitive resource on the net that tracks the average latency of providers? Or is this just something you have to test on your own.
02:15.55[TK]D-Fenderjeff_smoker: Have to test for yourself.
02:15.59CompyAre there any other decentralized carriers out there besides skype?
02:16.02LiNeTuXjeff_smoker: YMMV.
02:16.08Compyhas wondered that
02:16.28[TK]D-Fenderjeff_smoker: How is every testing company going to know how bad YOUR connection will suck?  Do they have computers on evvery network?  In every segment?
02:16.30LiNeTuXjeff_smoker: Latency is a factor of what is between you and them.
02:17.03jeff_smokerLiNeTux: I see.
02:17.06LiNeTuXCompy: of course.  VoicePulse (one example) has at least 4 locations you can 'pick'.
02:17.11[TK]D-Fenderjeff_smoker: common sense answer just about all of these questions.
02:18.34LiNeTuXjeff_smoker: ex: I can get to provider "A" from my house in 80ms.  I can get to the same provider from my colo server in 20ms.  That box is physically only 10 miles from me, but with another (much better) provider that has better perring arrangements than my home ISP.
02:18.37jeff_smoker[TK]D-Fender: So you're saying that providers have literally thousands of PSTN termination numbers?
02:18.42CompyLiNeTuX: hmm, that doesn't exactly strike me as decentralized/mesh...
02:19.10mostyis there a way to do GotoIfTime for a specific timezone that's different to the system timezone?
02:19.11[TK]D-Fenderjeff_smoker: yes
02:19.33LiNeTuXComy: depends on how you do it.  You can use a single DNS entry and do much trickery based on location to route/reroute/failover.  you can also use SIP gateways to help load balance and other such thigns.
02:19.43[TK]D-Fenderjeff_smoker: they are each a small telephone company  so THOUSANDS of calls at a time
02:20.14[TK]D-Fendermosty: AGI.  Have fun :)
02:20.39mosty[TK]D-Fender, yeah i figured
02:21.25CompyLiNeTuX: yeah, i know you can do that, just wondering. Skypes is for the most part fully peer based, just wondering if any others had picked up on that kind of architecture.
02:21.26jeff_smoker[TK]D-Fender: How do they operate the lines so cheap?
02:22.18LiNeTuXCompy: trust me, Skype isn't the first to do this - and remember, they went completely down not too long ago due to some "windows updates" <cough>.  So even the best plans can fail.
02:22.21jeff_smoker[TK]D-Fender: I mean, when it costs you $15 to get a basic phone number...they can have literally thousands of phone numbers for far cheaper, right?
02:22.34[TK]D-Fenderjeff_smoker: Are you really expecting an answer to "Wow, can you explain exactly how phone companies work in detail?"
02:22.39LiNeTuXjeff_smoker: a phone number doesn't mean it's a phone line.
02:22.49LiNeTuXheh
02:23.25[TK]D-Fenderjeff_smoker: Lets say I have 5 lines.  I then take on TWENTY customers on the assumption that no more that 25% of them will actually even be on the phone at a given time <-
02:23.36CompyLiNeTuX: oh ofcourse, I assumed they werent the first, hence I asked if you knew of any others. But yeah, I remember the massive outage when they lost all their supernodes, lol.
02:23.48tzanger[TK]D-Fender: sounds like erlang equations
02:23.58[TK]D-Fenderjeff_smoker: why do you think phone systems lock up in major emergencies?  Because the telcos themselves can't survive huge concurrency
02:24.04*** join/#asterisk carrar (i=tim@osburn.com)
02:24.33LiNeTuXCompy: Skype does do it differently than most... their model is very akin to how Hamachi works if you're familiar with them.
02:24.39[TK]D-Fenderjeff_smoker: its all about percentages.  When a company offers an "unlimited" plan, do you think they do this blindly?
02:24.58[TK]D-Fenderjeff_smoker: this is big math.
02:25.37jeff_smoker[TK]D-Fender: So YOU CONNECT to the PROVIDER via VOIP...the PROVIDER routes your VOIP connection to the PSTN as a normal phone call and still can only put one call through each line at a time, is that right?
02:25.44CompyLiNeTuX: yeah, they use a variant of the fasttrack protocol, they literally route everything over the mesh, even logins.
02:26.28[TK]D-Fenderjeff_smoker: they can accept as manny calls from you at a time as they feel like offering you based on what you're paying for.
02:26.32mostyjeff_smoker, no. digital telephone lines that telephone companies use aren't the same as a single line/phone number you can get at home
02:26.48jeff_smokermosty: how so?
02:27.04[TK]D-Fenderjeff_smoker: now is the point to take the word "line" and throw it right out the window.  We are in a world of CHANNELS
02:27.23mostyjeff_smoker, look up ISDN on wikipedia
02:27.40LiNeTuXCompy: SIP has other such devices - like SIP gateways (can't recall the name of them at the moment)... plus you can give SIP devices 'backup' registration info as well... so it's very flexable.
02:27.58jeff_smokerOk, so how channels can I use concurrently on my GSM mobile phone, 2? What about if I start sending AT commands to a voice modem...how many channels can I use?
02:28.09jeff_smoker(how many channels)
02:28.15[TK]D-Fenderjeff_smoker: a "line" is a piece of wire.  In the case of an analog phone line in your home it has a ver fixed signalling on it.  It can only support 1 call at a time.  the telco switch can conference in on its end, but coming in to you is 1 audio stream period.
02:28.36[TK]D-Fenderjeff_smoker: leave MODEMS out of this.
02:28.53[TK]D-Fenderjeff_smoker: if you are talking DATA, then that is where voip comes in, and that is based on bandwidth
02:29.13[TK]D-Fenderjeff_smoker: and through your concept of "AT commands out along with.
02:29.57Compyyeah, LiNeTuX. I have and still am considering those options. I guess I will just have to design a protocol to offload this traffic onto commodity hardware-based peers.
02:30.08jeff_smoker[TK]D-Fender: Ok. I'm just wondering why you couldn't put make multiple calls on your cellphone by utilizing a bunch of channels at once? What's the limitation?
02:30.32LiNeTuXCompy: there were some good talks at Astricon around this subject.
02:30.49[TK]D-Fenderjeff_smoker: if you're talking about GSM as a data transmission protocol for raw data, only the lack of hardware designed to talk that way.
02:31.05Compybummer that I missed it
02:31.23LiNeTuX"Why can't I get HDTV on my old TV"?
02:31.42pcranejeff_smoker: are you looking at initiating a call on asterisk, and having it go out via mobile phone?
02:31.43jeff_smoker[TK]D-Fender: But if you're talking about GSM for voice transmission on multiple channels in the same sense that the voip providers use ISDN like mosty says
02:32.15[TK]D-Fenderjeff_smoker: What GSM device do you have that supports multiple simultaneous channels?
02:32.41LiNeTuXCompy: Leif Madsen, who's in here from time to time, knows a lot about that subject.  Well, he helped write "the book"...so he should :)
02:33.31[TK]D-FenderLeif is Leif!  Na na na na na!
02:34.42CompyLiNeTuX: excellent :) I will also scour for some info concerning this topic
02:35.45LiNeTuXhad lunch with Leif at Astricon and didn't even know who he was :)
02:36.05LiNeTuX(at least by sight!)
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02:37.29jeff_smokerpcrane:Good question. I was working with AT commands on my Samsung SGH-t639 mobile phone which the AT commands indicate does not support voice transmission over the modem. I wanted to circumvent asterisk because I don't know linux, and put together my own pbx using php and apache...and receiving voice directly over the COM ports on a server. But now I'm moving away from this and  towards VOIP
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02:38.32[TK]D-Fenderjeff_smoker: Voice is not a web page.  your concepts of "com" don't apply here.
02:38.38jeff_smokerpcrane: Because I can't seem to find any evidence that shows a gsm voice modem can receive and forward more than one voice call at a time
02:38.53pcraneI'd have thought that it would depend on the phone
02:38.56tzangerjeff_smoker: I'm not aware of any that can
02:39.12jeff_smoker[TK]D-Fender: What do you mean?
02:39.22[TK]D-Fenderjeff_smoker: And AT commands are jsut a way of telling a modem to dial.  from there it is jsut a data link.  Not even a packet interface by definition.
02:39.28pcranee.g. I've got a motorolla e770 which has a different feature set (via AT commands) than a nokia 6121 classic
02:40.07[TK]D-Fendertzanger: Plenty of crack floating around here...
02:40.07jeff_smoker[TK]D-Fender: I became curious about this when somebody mentioned chan_bluetooth to me
02:40.08CompyLiNeTuX: thanks a bunch. I'll definitely document my progress on this.
02:40.17tzanger[TK]D-Fender: heh :-)
02:40.56[TK]D-Fenderjeff_smoker: Yes, well Bluetooth IS a packet protocol and gee, I dunno.. there are HEADSETS out there for it for some strange reason, which seems to have the word "voice" hanging over it for some other equally mysterious reason.
02:41.13tzangeryou also don't use AT commands to transfer voice data to bluetooth headsets
02:41.38[TK]D-FenderSomebody is definitely stuck in 1990 in the BBS years.
02:41.48jeff_smokertzanger: No, but it appears you can use them to answer, dial, and join calls.
02:41.52[TK]D-FenderGet back into your Delorean.
02:41.54subdolusjeff_smoker: my router cracked the shits earlier... do you know of a command to send DTMF down a channel?
02:41.54tzangerthey were fun times, I can't balme him :-)
02:42.13[TK]D-Fendersubdolus: "core show application senddtmf" <-
02:42.24tzangerjeff_smoker: again, I am almost positive that the calls are joined at the telco switch and not the cell phone
02:42.48[TK]D-Fendertzanger: Yes, I was a sysop back to the 80's, ran my own, wrote my own, and when the internet came to the common people I LET GO.
02:43.11jeff_smokertzanger: I see.
02:43.24tzanger[TK]D-Fender: agreed, I did too, but there was something very cool about it being local instead of global
02:43.36[TK]D-Fenderjeff_smoker: And for chan_bluetooth, its a single voice channel, just like your headset to your cellphone.  There is no magic there.
02:44.12[TK]D-Fendertzanger:  I know..... I occasionally hit onne of the TW2002 telnet sites from time to time just for nostalgia ;)
02:44.13Nuggettelnet is eeeeeeevil!
02:44.20[TK]D-Fenderpets the nugget-bot
02:44.29tzanger:-)
02:44.35jeff_smokerSo in other words, you guys are saying that you CANNOT just hook up a bunch of mobile phones to your laptop and transfer and dial calls back and forth between them?
02:45.14QwellI don't know about transfer
02:45.23[TK]D-Fenderjeff_smoker: Well if you have a pile of BT receivers on your server the concept is quite possible.
02:45.27Nuggeteyes [TK]D-Fender
02:45.29tzangerjeff_smoker: sure, with bluetooth if the phone has the appropriate profiles, or with a bunch of sound cards and usb cables.
02:45.37[TK]D-Fenderjeff_smoker: And you can get GSM direct cards as well
02:45.45[TK]D-Fenderhands Nugget a spoon.
02:46.20jeff_smokerOk, but you guys are saying you can't split these phones into multiple channels so that they can handle multiple calls at once?
02:46.38Qwelljeff_smoker: no..
02:46.42[TK]D-Fenderjeff_smoker: So aside from all of this field&stream of consciousness are you actually fishing for something specific, or just kinda ramblimg?
02:46.46QwellI've never seen a cell phone that can do multiple calls at once
02:47.15[TK]D-Fenderjeff_smoker: Again, what kind of phone have you ever seen that lets you talk to 3 people INDEPENDANTLY?
02:47.17jayteeField and Stream is good reading for the crapper
02:47.23tzangerheh
02:47.33[TK]D-Fenderjaytee: nice to see you appreciated the mixed metaphor ;)
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02:47.43tzanger[TK]D-Fender: it's one of the new ones; they're made of unobtanium
02:48.02tzangerI've got an iphone and it won't even do that :-p
02:48.03jayteealthough I also am fond of It Pays To Enrich Your Wordpower in Reader's Digest.
02:48.27[TK]D-Fenderjaytee: Now imagine that my normal state of consciousness operations on 2+ tiers more often than not :)  God it sucks trying to communicate with mortals sometimes ;)
02:48.40[TK]D-Fendertzanger: lol!
02:49.06[TK]D-Fendertzanger: on the "period"ic table, its the one that plays "hard to get", isn't it? ;)
02:50.49jaytee[TK]D-Fender, I don't have a background in psychology but I've detected what I believe may be a streak of masochism and martrydom in your psyche. You continue to "suffer fools gladly nite after nite.
02:51.32tzanger[TK]D-Fender: absolutely
02:51.36[TK]D-Fenderjaytee: I won't let it kill me, so martyrdon is out.
02:52.04[TK]D-Fenderjaytee: "Why do we always come here?  I guess we'll never know.  It's like a kind of torture... TO HAVE TO WATCH THIS SHOW!"
02:52.07jayteeAnd I've actually witnessed you working on a few noobs and literally beating the sense into their thick skulls.
02:52.23[TK]D-Fender"ANIMAL'S" OUT
02:52.42tzanger[TK]D-Fender has amazing patience
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02:53.17[TK]D-Fendertzanger / jaytee : Its been a real strain lately and you've seen it run out FAST in some cases.
02:53.50[TK]D-Fendertzanger: It hurt my previous rep and was a wake-up call as it wasn't restricted to jsut in here.
02:54.24tzangerI bet, you are one of a handful of regulars in here who don't seem to give up and lurk after a while... like me, for example.
02:54.59tzangeryou, manxpower (who's not here?!), qwell... those are probably the top three active non-paid digium people here
02:55.05[TK]D-Fendertzanger: No, I don't give up, and thats the problem.  Burn out.
02:55.08Qwelltzanger: ...
02:55.13QwellI AM paid-Digium
02:55.17tzangeroh
02:55.24tzangerI did not know that :-)
02:55.25Qwellhave been for like 2 years :p
02:55.33jayteeManxpower isn't paid Digium
02:55.33tzangerand I am an idiot
02:55.36tzangerI know he isn't
02:55.41tzangerI didn't think qwell was though
02:55.56Carlos_PHXWell, I'm a noob to the channel, not to Asterisk.  But I reserve the right to ask stupid questions anyway.  I'm gonna go have a couple beers and see what I can think up.
02:56.05tzangerso does digium have all their new customer service people hang out in here for 3 months as a training period? :-)
02:56.16Qwelltzanger: good idea :p
02:56.28LiNeTuXis already on his way to the 2+ beer limit.
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03:01.37drmessanoOh lord
03:01.49drmessanoNo more Digium new employees... please
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03:02.06Qwelldrmessano: go test
03:02.16drmessanolol
03:02.18QwellI put up new versions of stuff today
03:02.26drmessanoo.o
03:02.27Qwellnew ISOs too
03:02.30drmessanoO.O
03:02.38Qwelljust updates stuff
03:03.00drmessanoIn the words of Donald duck
03:03.07drmessanoZOMGGGG TORRENT PLZ
03:03.11drmessanohang on
03:03.15tzangerdrmessano: "rabbit season" ?
03:03.16Qwellbtw, if anybody wants a super-exclusive pre-beta of AsteriskNOW, lemme know :p
03:03.38LiNeTuXraises his hand
03:03.56jayteeis it still using that custom distro?
03:03.57drmessanohttp://farm2.static.flickr.com/1174/1477939087_a56d8bdbfa.jpg?v=0  <--- PWNS HARD
03:04.03Qwelljaytee: CentOS
03:04.13Qwelldrmessano: oh, it's got a hawt splash screen too
03:04.15jayteeno way! when did that change?
03:04.17Qwellyou should download the new one just for that
03:04.50Qwelljaytee: technically it was announced last week
03:04.54Qwell"officially"
03:05.02LiNeTuXyeah, where is it? lol!
03:05.03Qwellbut we told TMC like...over a month ago
03:05.26drmessanoZOMG
03:05.27jayteeI'd like to test it
03:05.33voxterQwell: asterisknow++
03:05.36voxterQwell: link me!
03:05.43drmessanoTORRENT PLZ
03:05.44drmessanoha
03:05.51drmessano!!!!ONES!!!11!!!!
03:05.55voxterhaha
03:07.15drmessanoFile size is smaller
03:07.27drmessano++
03:07.40Qwelldrmessano: it actually should be 1 ISO...
03:07.51drmessanoOh, people still burn CDs?
03:08.01QwellI screwed up when I created these ones..  the size was a little larger on my test build of it, then I cleaned it up and rebuilt
03:08.01drmessanodownloads the DVD
03:08.07drmessanoAh
03:08.38drmessanoso it was 700 + extra, and you rebuilt the 700
03:08.49drmessanoBut extra needs to be moved over
03:14.29Carlos_PHXSo you guys at Digium know who (if anyone) is working on the T.38 stuff?
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03:15.13Qwelldrmessano: no, there were duplicate packages on disc2 that I cleaned out
03:15.23Qwellthose were an extra 100mb or so, pushing it over the 650mb
03:15.24drmessanoah
03:15.33drmessanoSo only need CD1?
03:15.39Qwellget the dvd
03:15.49drmessanoBut... but
03:15.52Qwellyou need both CDs if you do it that way
03:16.01QwellCD2 has @development-tools
03:16.04drmessanoIm not using the CDs anyway
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03:16.16drmessanoYoure just making me confused and stuff
03:16.17Qwellright russellb ?
03:16.27Qwelldrmessano: yeah, just get the damn dvd :p
03:16.42drmessanoWhere is CD3 with Skype on it?
03:16.51Qwelldrmessano: refresh
03:17.08LiNeTuXheh
03:17.14drmessanoAh
03:17.15LiNeTuXthat's another 1G file
03:17.15Carlos_PHXrussellb:  Since you happen to be here...do you know who, if anyone, is working on T.38?
03:17.56russellbCarlos_PHX: not currently, no
03:18.00russellbQwell: huh?
03:18.04Qwellrussellb: exactly
03:18.11voxtersoooo why does festival make my asterisk eat 100% cpu and not hang up the call
03:18.18voxterIs there a free license/trial of cepstral out there?
03:18.31russellboic
03:18.41Qwellvoxter: I implemented festival stuff in dialplan once
03:18.44Qwellit was hot
03:18.59voxterIve made it work on another box, but this one is being bitchy.
03:19.08Carlos_PHXBah, I guess I should give up on T.38, but its *so* close to working.
03:19.13Qwellmost of it was System() and Playback()..  had md5 caching of the text and everything
03:19.21Carlos_PHXHey, I can reliably crash Asterisk with a T.38 call, that's something.
03:19.35voxterQwell: oh you mean THAT way. haha. was this before the festival module, or were you just trying to torture yourself? :)
03:19.35QwellCarlos_PHX: is there a bug open for it?
03:19.45Qwellvoxter: it wasn't too hard, really
03:19.52Qwellworked really well too
03:19.58Carlos_PHXYes, and the fix is to get a version of Spandsp that supports a specific call.
03:20.06Carlos_PHXBut...that version doesn't seem to exist.
03:20.15voxterQwell: im sure it was easy! probably easier than using Festival() in my case!
03:21.00Carlos_PHXBut Kevin Fleming says he's seen it working, so someone must have it.
03:21.52coppiceCarlos_PHX: use spandsp-0.0.5pre4
03:22.18Carlos_PHXcoppice: That's what I'm using.
03:22.31Carlos_PHXReinstalled twice, let another Asterisk guy have a go, no dice.
03:22.58QwellCarlos_PHX: what version does it say to use?  coppice is the man who'd know
03:23.11coppicethen the problem is with the * code, or you are doing something else wrong. That version is in heavy use
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03:23.50Carlos_PHXInteresting to hear.  I've found no docs, so I'm winging it, could be something I've done wrong.
03:24.09Carlos_PHXThe symptoms match those in bug 13473.
03:24.24Carlos_PHXSomeone else is reporting the same thing also with 0.0.5pre4.
03:24.53Carlos_PHXcoppice: Are you saying that it's in use with T.38 and *?
03:25.03coppiceI have no idea
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03:26.29coppiceif the system can't find t38_terminal_init, you are not using 0.0.5pre4
03:26.53Carlos_PHXI can't be sure that's the case, it just crashes *.
03:27.21coppiceno. * crashes.
03:27.31Carlos_PHXWell, yeah, sorry.
03:27.38Carlos_PHXI can be sure I have this though: spandsp-0.0.5pre4.tgz
03:27.48Carlos_PHXClean system, no other version possible.
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03:28.11coppicemaybe, but do you have other older versions installed? from the distro itself, maybe?
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03:28.41Carlos_PHXAre you saying Asterisk has a version of Spandsp?  I will have to look then.
03:29.21coppicemany distros install a spandsp RPM or whatever your local package manager might be
03:29.26Steve_J-obshi, anybody familiar with the Mysql command on the dialplan?
03:30.07Carlos_PHXOh, it's apt, I will double check.  Fairly sure not, since menuselect was not showing app_fax as an option until I installed spandsp manually.
03:31.04Carlos_PHXI did use apt to install libspandsp, which was not included with spandsp.
03:32.26coppicethat last line made no sense
03:33.11coppicethe core thing which spandsp does is install libspandsp
03:33.57Carlos_PHXOk, then I had one problem first,and another now.  I installed spandsp from source, and Asterisk said it could not find libspandsp.
03:34.18Carlos_PHXSo then I installed libspandsp via apt, and Asterisk could find it, but crashes.
03:34.50coppicethen you must get rid of the other libspandsp you installed, and tell your system how to find the correct one.
03:35.24Carlos_PHXI was unable to locate libspandsp in the expected locations.
03:35.32Qwellcoppice: does it install to somewhere like /usr/local/ by default?
03:35.32coppiceThis is fully covered in the spandsp FAQ http://www.soft-switch.org/spandsp_faq/ar01s12.htm
03:35.36voxterlame. app_swift is telling me failed to set voice now. TTS is just not meant to be on this box...
03:35.45Carlos_PHXOk, I'll take a look through there.
03:35.49Qwellcoppice: 404?
03:36.09Carlos_PHXcoppice: Thank you very much for your time and attention.
03:36.25coppice./configure type packages normall install in /usr/local by default, and spandsp follows the rules :-)
03:37.02Carlos_PHXYeah, but this fails to find it, so...  find /usr/lib/ -name libspandsp.so\*
03:37.43Carlos_PHXFYI, the URL above is 404, but I can open the main FAQ page.
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03:38.59coppicewhoops. a letter got chopped off - http://www.soft-switch.org/spandsp_faq/ar01s12.html
03:41.54Carlos_PHXHeh, yeah, point taken on that.  Now I just have to figure out how to tell * where the right version is.
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03:42.39ReDNeQsup all
03:45.19coppiceIf you want spandsp to install in /usr/lib use "./configure --prefix=/usr/lib"
03:46.49Carlos_PHXcoppice: Thanks.  I am going to assume I should uninstall/reinstall spandsp followed by *.
03:47.52coppiceThe dumb thing with most distros is they install by default to scan /usr/local/include for headers, but not /usr/local/lib for the associated libraries. this causes endless support hassles
03:48.19phixcoppice: yeah
03:48.35phixcoppice: I say just install under /usr not /usr/local
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03:48.55phixcoppice: apt-get source asterisk if you want to compile it your self
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03:49.31Carlos_PHXGreat info.  At the end of this I'll put the documentation on voip-info so there's at least a starting point for others.
03:49.34phixhey fredonIRC
03:50.46voxteris there seriously no function to streplace in asterisk?
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03:51.06Carlos_PHXcoppice: Are you Steve Underwood?
03:51.25coppiceonly if you have money to pay me
03:51.29Carlos_PHXHeh
03:51.48Carlos_PHXThank you for your contributions, Spandsp has helped me tremendously over the years.
03:52.08roatgiderHey all, I just install *NOW in a Virtual Server to test it out, everything is working great with the exception of the default IVR
03:52.15roatgiderwhen I dial 7000 from any sip phone, and watch on the cli
03:52.20roatgiderthe call is answered, the 1 second wait period passes, and then the call is dropped
03:52.27roatgiderit's weird, because it was working fine the first few times
03:52.31drmessano----> #asterisknow
03:52.42roatgiderno one is responding in there :\
03:52.51drmessanoYou're cross spamming
03:53.11roatgiderI'm restating my problem in a different channel that has more active users
03:53.18roatgiderdidnt mean to spam
03:53.36drmessanoI took offense to it
03:53.42roatgiderapologies
03:54.21drmessanoReally, really, really hurt here
03:54.27drmessanoI feel so... used
03:54.32roatgider<.<
03:54.36drmessanoOk, more than normal
03:54.38Carlos_PHXAnd you like it
03:54.44jayteehehehe
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03:55.48[TK]D-Fenderdrmessano: how long... do the memories last?!
03:56.18drmessanoI still hear the cries of the clowns.. it's so.... tragic..
03:59.17voxterIs there no dialplan function to cut all occurrences of, say, @ from a variable?
03:59.36jayteeum CUT?
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04:03.17voxterjaytee: that doesnt exactly remove them so much as give you specific fields around every @
04:04.49[TK]D-Fendervoxter: aND YOU GLUE THOSE BACK TOGETHER AND THERE YOU HAVE IT
04:04.57[TK]D-Fender(along with my capslock)
04:05.10voxterSo then the answer is no :)
04:05.50voxterHmm, there also doesnt seem to be a way without using LEN() first to chop just the LAST character/integer off the end of a variable using :#:#
04:06.01voxteri guess ill ahve to make this code a bit uglier than i'd hoped
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04:27.42[TK]D-Fendervoxter: Well you could do it in 1 line with grep/sed and app_backticks (3rd party app)
04:29.03Qwellfunc_shell
04:29.12voxterquivers
04:29.24voxterI did it nested inside of 3 CUT's
04:29.26voxterugly, but it'll do
04:29.49voxtersomeone asked me for a way to have it read back their SIP URI, so i set up sip:getsipuri@voxter.ca to read it back to them
04:29.57voxterof course app_swift is bitchy about the characters it will read.
04:36.45[TK]D-Fendervoxter: I'm sure it'd have been an easy job to invent a function or app to do what you needed.
04:46.32jayteetime to snooze
04:46.34jayteenite all
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04:51.19voxter[TK]D-Fender: yeah. It would really be nice to see a STR_REPLACE function that understood regex. it's come up more than once.
04:56.14voxterfound a chanisavail bug
04:56.15voxtertime to report
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05:12.45[TK]D-Fendercheckout time.  Later all
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05:20.21CutlassI just installed asterisk 1.6 and I can't find the config files...has anyone seen this before?  I used "/usr/local" as the installation PREFIX...
05:20.43Cutlass...there is nothing in the whole filesystem....
05:25.55mostythey're in the configs directory in the asterisk source aren't they?
05:27.00Cutlassah!!...so they're in the location that I unpack the tar file???  I see them now...doesn't it install these somewhere else?
05:33.15mostythere is a make target that will install the sample configs
05:33.43jameswf-homemake broken
05:34.11jameswf-homemake configs_incompatible
05:34.14jameswf-homeheh
05:34.30mostymake install-useless-default-configs
05:35.05jameswf-homethat is just make configs.. it was much to long before
05:36.19Cutlassthanks!...that help :)
05:36.25Cutlass*helps
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06:20.56jameswf-homeheh http://linuxslut.net/album/linuxsluts/
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06:59.39andrewgodwinthis one has been puzzling me for a while - if my laptop's phone port appears as a second soundcard (which it does
06:59.49andrewgodwinis it possible to get asterisk to use it?
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08:32.56nicoxhi, anyone there?
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08:35.41nicoxhi
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08:37.05Steve_Ollis/me is a n00b .. just watching and learning
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08:45.52anebihi
08:46.35anebii have a problem with one of the trunks and i can't understand how to solve this. i have a sip account from telsome and i have set it to asterisk on our server.
08:47.00anebiit works well some times, but during the day we get a message wrong password and then it disconect and stop to work
08:47.25anebialso we get timed out messages for this trunk, what can it be the reason and how can i solve this problem?
08:47.49anebiwe have installed asterisk with freepbx
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08:49.19Kernel_Corehi all
08:51.39*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
08:51.40anebithe password is correct, i can't understand why this happen
08:52.04anebiafter asterisk restart it works again, register to sip provider
08:52.35*** join/#asterisk op3r (n=op3r@121.96.100.177)
08:54.25defsworkanebi: sounds like something to ask tour provider first I would guess
08:56.17*** join/#asterisk niros (n=nir@89-139-53-119.bb.netvision.net.il)
08:56.21anebidefswork: as my boss told me, the provider told him that our server register every 60 secs
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08:57.47defsworkanebi: right - but someone is telling you your credentials are bad
08:58.05defsworkanebi: that's pretty explicit - not a timeout etc..
08:58.25Kernel_Corehi all
08:58.46nicox<PROTECTED>
08:58.52Kernel_Coreafter installing OSLEC with asterisk , it isnot possible to increase txgain or rxgain in zapta.conf
08:59.14Kernel_CoreI want to increase my txgain
08:59.18Kernel_Corehow do I do it ?!
08:59.37anebidefswork: yes, you are right.
08:59.56defsworkanebi: are they tetchy about registering so often ?
09:00.42anebidefswork: not at all
09:01.02nicoxany idea why a call Transfer is not working with IAX?
09:02.11defsworkdid I read somewhere that aastra are making blf's tri-state so that a ringing call can be collected ?
09:05.21defsworkaah - directed call pickup is already there
09:06.54*** part/#asterisk Steve_Ollis (n=Steve_Ol@cms.hillsong.com)
09:10.27*** join/#asterisk dwagner (n=dwagner@195.202.166.182)
09:10.36dwagnerhello, i would like to talk to murf
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09:27.14Andre101hello.. Is it possible to get a dialplan to match on characters? Trying to match dialing a skype username
09:27.51kaldemaryes
09:28.47Andre101kaldemar: How do I go about doing it?
09:31.20kaldemarexactly the same as with numbers.
09:31.33Pagautasexten => _hello,1,dial(SIP/hello@skype)
09:31.43kaldemar_ isn't needed in that.
09:33.32Pagautasi've got a question about queues
09:33.44Pagautasi had a dialplan like this
09:33.45Pagautashttp://pastebin.ca/1215633
09:33.54Pagautaseverything worked fine
09:34.12Pagautasuntil my boss asked to put an announcement
09:34.15*** part/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg)
09:34.26Pagautasto calling side before
09:34.34Pagautasagent phone begins to ring
09:34.41Pagautasso i've done like this
09:34.46Pagautashttp://pastebin.ca/1215632
09:35.02Pagautasbut then asterisk stopped to record calls
09:35.06Pagautaswhy?
09:38.21nicoxhi, anyone an idea why Call transfer on IAX is not working after upgrading to asterisk 1.6.0-rc6?
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10:14.06*** join/#asterisk Ng_ (n=cmsj@mairukipa.tenshu.net)
10:14.31Ng_if one has a PRI (E1) hooked up via a zaptel device, is there a way to see which channels are actually being used?
10:14.46Ng_the nearest i can find is "sip show channels", but that just shows active SIP calls, which may not be the same thing
10:15.36nicoxzap show channels
10:15.47nicoxor simple show channels verbose
10:15.58nicoxor core show channels verbose
10:16.21nicoxthere you see everything, which channel is connected with which channel and how long....
10:16.36*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
10:16.38Ng_nicox: oh interesting, I must have only run "zap show channels" when nobody was on the phone, I can actually see which extension is using a channel \o/
10:16.48Ng_I don't appear to have "simple" or "core" though
10:17.14Ng_(I get abused for running an old asterisk each time I pop in here with a question ;)
10:17.49Ng_nicox: thanks :)
10:18.05nicoxi'm using 1.2.17 to 1.6.0-rc6 so i have to use everything..
10:18.08nicoxyour welcome
10:19.38Ng_1.2.12 here. I will upgrade it happily when a need arises, but it's been in operation for 18 months with no problems (i had to reset the sangoma driver twice in that time, but that's not asterisk's fault)
10:20.41nicoxon my site sangoma driver runs better then asterisk *g* also ss7box running great
10:21.50Ng_on the whole it's been fine, but every so often the provider drops the line over a weekend, and sometimes the sangoma driver doesn't bring it back properly. calls work, but they sound really distorted. A quick poke of the driver restores normality so I've not really traced it any deeper
10:22.52nicoxhm, try a firmware-upgrade and upgrade of the wanpipe-driver. this should help
10:23.48Ng_yeah, I've had that queued up in my todo list for some months now ;)
10:35.36*** join/#asterisk _Roman (n=roman@87.254.78.150)
10:43.51_RomanHello, I am using asterisk 1.4 (under trixbox), with an X100P based pstn card.  The problem I am having is that when a call comes in, I send it to a queue, I answer one of the internal SIP phones (a grandstream 200) and then hang the internal phone up.  The problem is that the external line does not hangup.  Does anyone have any suggestions?
10:44.47*** part/#asterisk Ng (n=cmsj@mairukipa.tenshu.net)
10:52.01*** join/#asterisk Squeeb (i=squeeb@eggwee.co.uk)
10:52.41SqueebI keep getting an error, and also my Queue has stopped reporting the position status to customers.. This is the error:
10:52.44Squeeb[Oct  1 11:46:07] WARNING[66413]: ast_expr2.fl:407 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input: = 1
10:52.47SqueebAny ideas where it's coming from?
10:56.25*** join/#asterisk padski (n=paddy@5ac311a1.bb.sky.com)
10:56.29SqueebNobody here?
10:57.11kaldemarwhen did you start getting the error? when do you get the error?
10:57.43SqueebWhen someone is in the queue, just before it switches extension, it's in rrmemory stratigy.. I'm not sure when I started getting this error
10:59.16Squeebhmmm .. actually it appears to be whenever the queue tries to call my 6002 extension
11:01.48kaldemarshow the extension
11:02.30Squeebhttp://pastebin.ca/1215690
11:03.08Squeebhmm
11:03.15Squeebit happens if I try and call that extension directly
11:06.27kaldemarthat is not an extension, that is a users.conf entry. an extension is in your dialplan, extensions.conf.
11:06.28SqueebI wish it told me the line number the fault occures
11:08.59Squeebhttp://pastebin.ca/1215694
11:09.41Squeebnot sure what o,1, does
11:11.46kaldemarlooks like you're using some gui.
11:12.10Squeebyea, the digium web panel
11:13.04kaldemarexten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3) might be a problem if ARG1 has spaces in it for some reason.
11:13.35Squeebah
11:13.56kaldemarvariables should be quoted in comparisons like that.
11:14.01Squeebwhat does ARG1 contain?
11:15.04kaldemarno way of knowing without seeing the whole dialplan. you better ask digium about that. that is not direct asterisk stuff.
11:15.23kaldemarbut rather built on asterisk.
11:18.28_RomanHello, I am using asterisk 1.4 (under trixbox), with an X100P based pstn card.  The problem I am having is that when a call comes in, I send it to a queue, I answer one of the internal SIP phones (a grandstream 200) and then hang the internal phone up.  The problem is that the external line does not hangup.  Does anyone have any suggestions?
11:21.33*** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582612.dsl.bell.ca)
11:22.10nicoxhi, anyone expierence with Call transffer on IAX calls, and why they are not working?
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11:25.52gr0mit_Roman, you mentioned trixbox, which is not really supported here
11:26.12gr0mithowever, clearing of analogue channels is somewhat complex
11:26.45gr0mitcan you explain your config in a bit more details pls
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11:31.00InsolentDreamsHey is there a trick to unregister a SIP device and re-register it?  I'm having a issue where someone's phone rings, but is unable to pick it up.  Nothing happens when they pick it up, I think it's a registration/cache issue.
11:32.00kaldemarwhy would you think it's a registration issue?
11:33.31kaldemarregistering just tells asterisk where the phone is. if the phone rings, the phone must either be registered or defined with a static ip. either way, location or registering shouldn't be the problem.
11:38.54*** join/#asterisk mesfet (n=iw3grx@89-96-227-146.ip14.fastwebnet.it)
11:40.14mesfetHi. Just a question: what about a Playback(message|m) to play a message with music on hold background?
11:41.15kaldemarrephrase your question, please.
11:41.52SqueebWhat's the astdb ?
11:42.11mesfetI believe that in interesting feature (not available, now) is the possibility to play a message (a sound file) with music in the background.
11:42.35mesfetThis can be done mixing the sound file with the music-on-hold.
11:42.38kaldemarSqueeb: asterisk's internal database
11:42.44Squeebaah cool
11:43.29mesfetPlaying "welcome" message and "ivr-instructions" with music in the background should be nice for the caller.
11:43.53mesfetI don't know if this feature was evaluated or not.
11:44.01*** join/#asterisk LiNeTuX (n=LiNeTuX@64.132.248.206)
11:44.06InsolentDreamskaldemar: Thanks mate, The phone appeared to be in a messed up state, cheapo polycom's seemed to return a bunch of gibberish when I was sip debugging it
11:44.54kaldemarnp
11:44.59InsolentDreamsI reloaded from firmware defaults and re-set the settings and it's fine  ;)  I asked a bit prematurely before digging in properly.  :)
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12:01.30traxxoshello all
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12:08.18kielstrandhi there, anyone here can help me with sccp and a cisco 7960
12:09.25kielstrandmessage after booting: File Auth Fail: CTLFile.tlv
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12:30.28mort_gibHI
12:30.36SqueebHELLO
12:31.35mort_gibI'm having problems getting SIPPEER(100:curcalls) to work in a macro -Are there any problems with SIPPEER ??
12:33.48kaldemarreplace : with , or |
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12:45.31stonyhi
12:46.07stonyi have a strange problem - out of nowhere the asterisk binary hang and now it produces 100% cpu load on start and isn't booting completly
12:46.16stonyany idea where this could come from ?
12:47.40*** join/#asterisk write_erase (n=Olivier@goodgw.m2m-fleet.com)
12:47.43stonyi haven't changed anything
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12:51.28mort_gibKaldemar: Yeah?
12:52.58padskiI'm fighting with rfc3389 comfort noise packets again today.  I'm being told that they are coming from BT and that BT can't be persuaded to switch them off, but our upstream provider is acting as some kind of proxy between us and it is unclear whether that may impact this problem.  anyone got experience with asterisk talking to BT IPXes in the UK ?
12:54.19SqueebI get the same thing with Sipgate
12:54.26SqueebOct  1 12:58:48] NOTICE[66413]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.10.77.24
12:54.29Squeebet...
12:55.21eric2Squeeb - I get that same problem  :(
12:55.24kaldemarmort_gib: what yeah?
12:55.53mort_gibkaldemar: Was that for me (replace : with , or |)
12:55.58kaldemaryes, it was.
12:56.54mort_gibOkay thanks, but in voip-info.org it states SIPPEER(<peername>[:item])
12:56.56padskiSqueeb, have you spoken to them about it ?
12:57.58Squeebno
12:58.27kaldemarmort_gib: well, voip-info has been known to have errors. core show function SIPPEER uses pipes.
12:58.34[TK]D-Fendermort_gib: SIPPEER(<peername>[|item]) <-
12:58.43padskiSqueeb, one of the fun symptoms I am seeing is music-no-hold that only plays when you talk to it.  have you got that too ?
12:58.49[TK]D-Fendermort_gib: "core show function SIPPEER"
12:58.53kaldemarmort_gib: try it and fix it to the wiki if it's wrong there.
12:59.01[TK]D-Fendermort_gib: You shouldn't be using the WIKI for instructions.
12:59.24v4mp:o
13:00.07kaldemari'd use wiki for instructions, but test every single thing myself before believing it works.
13:00.29mort_gibAs did I, but alas it didn't work, which is why I'm here :-)
13:00.45v4mphey guys i have a problem where as when u call in u get the welcome then the options read to you then when you try pick an option it doesn't wrong right 1 way i had the line it spammed loads of line in the cli i will paste my config and cli output
13:01.15*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:01.28kaldemarmort_gib: use cli helps, docs in the source package and the book as primary docs.
13:01.42mort_gibOk
13:01.43km-padski: This may not be even remotely relevant to that music-on-hold thing, but I have heard that asterisk will not start sending rtp until it receives rtp first.
13:01.48mort_gibThanks
13:02.13km-padski: if the lines have VAD, maybe the handset isn't sending rtp until there's energy on the handset?
13:02.43mort_gibIs incominglimit of ANY use in 1.4 -> ??
13:02.45v4mpthis is the config http://pastebin.com/d5bfe77b7 not too sure about after the welcome part what the line should look like to read what to do... 1 line i did have kind of worked but when u tryed to go to option 1 it said it didn't exist
13:02.50mort_gibin sip.conf
13:02.59[TK]D-FenderCLI is the #1 way to get instruction on app syntax.
13:03.36stonyeven if i start asterisk without any hardware (no modules in the kernel loaded) it still hangs
13:03.38*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:03.49mort_gibI'm beginning to realize that. I have always used .Wiki, Google or similar
13:04.09mort_gibExamples works better for me....
13:04.31padskikm-, there is already rtp in progress. the silence at the asterisk end corresponds to rfc3389 CN packets incoming in the tcpdump.  the first CN packet arrives, if anythin, earlier than I would expect the next rtp packet.
13:04.47v4mp[TK]D-Fender, yes if you know what to look for :p
13:05.01Squeebpadski: I've not tried that
13:05.03SqueebI will check
13:05.10mort_gibv4mp: Yes!
13:05.14SqueebI don't think it's a problem though
13:05.21SqueebIt's only a notice .. not a warning
13:05.24*** join/#asterisk TeamFrrst (n=TeamFrrs@c-98-243-174-131.hsd1.mi.comcast.net)
13:05.41[TK]D-Fenderv4mp: "core show functions" , "core show function [functionname]" , "core show applications" , "core show application [applicationname]"
13:05.51[TK]D-Fenderv4mp: There is EVERYTHING.
13:06.43padskikm-, but I suppose that if the CN packets don't count as rtp, and every packet had to be a reply then this would account for it, but I can't help thinking that would be a nonsense, since then we would drop packets in response to dropped incoming packets ??  The explanation I have seen is that is is timing related.
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13:07.08*** join/#asterisk phpboy (n=shane@196.211.1.45)
13:07.55km-padski: ah.  I don't really have much knowledge of cng to comment on it, sorry.
13:07.59phpboyhey all, I have 2 PRI cards, one for land lines (incoming and outgoing) and one for mobile calls (outbound only) what would be the best way to configure this in zapata.conf?
13:08.12*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:09.00[TK]D-Fenderphpboy: Just about the only thing those might have in common is just sharing a group # for applicable channels.
13:09.49padskikm-, it doesn't seem to me that the actual generation of comfort noise is all that important.  silence would do fine.  I'm seeing issues that go way beyond CNG.  but thanks for your thought, maybe it will turn out to be part of the puzzle :-)
13:09.52phpboyI thought so, that's all I'll have to worry about?
13:10.00*** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net)
13:11.16phpboy[Oct  1 15:10:48] WARNING[23735]: channel.c:3025 ast_request: No channel type registered for 'ZAP'
13:11.16phpboy[Oct  1 15:10:48] WARNING[23735]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented)
13:11.26Hertzy3Does anyone know where I can find instructions on how to install * on a Xen DomU running CentOS? I am having problems installing zaptel
13:11.32phpboyit's giving me those errors now on the outbound mobile pri :(
13:11.52*** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net)
13:11.57phpboyHertzy3: what problems are you experiencing?
13:13.46Hertzy3upon running 'make', I get this error: You do not appear to have the sources for the 2.6.18-92.el5xen kernel installed.
13:13.52phpboy[TK]D-Fender: which paste bin can I use to show you my config?
13:14.07Hertzy3i tried doing yum install xen kernel-xen kernel-xen-devel
13:14.11phpboyHertzy3: easy to fix
13:14.13Hertzy3but its a host
13:14.18phpboydid it install?
13:14.19Hertzy3err, vm
13:14.30Hertzy3they all installed just fine
13:14.33phpboyok
13:14.35phpboythen run
13:14.40phpboyyum update
13:14.42phpboythen reboot
13:14.46phpboyand it should work
13:14.52Hertzy3thank you I will give that a try
13:15.02phpboyI'm pretty certain that will sort you out
13:15.08phpboywould you like the reasoning behind this?
13:15.22[TK]D-Fenderphpboy: Doesn't matter which
13:15.23stonyis there a way to strace the asterisk process ? it always detaches and then the strace isn't working
13:15.35stonyi added the -f switch, but that doesn't stop asterisk from detaching
13:15.38[TK]D-Fenderphpboy: And clearly chan_zap.so is not even loaded
13:15.49Hertzy3yes
13:15.52[TK]D-Fenderphpboy: Sounds like you compiled zaptel after *
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13:16.08phpboy[TK]D-Fender: no no, you'll see now what the problem is, hang ten...
13:16.41awannabehey guys, ever seen where a polycom registerrs can take in/out calls, but if you try to dial *99 to access voicemail (without the button) asterisk says unable to authenticate?!
13:16.57phpboyHertzy3: Zaptel requires the src of the kernel you are running at the point of compile, which in this case you don't have
13:17.06phpboyso after a new kernel and yum upgrade
13:17.07c4t3lawannabe: check your polycom configs
13:17.09phpboyand a reboot
13:17.12phpboyyou should be fine
13:17.28Hertzy3that does make sense, thank you, i am currently rebooting, hopefully it works
13:17.33*** join/#asterisk tcseke (n=chatzill@217.20.134.239)
13:18.24phpboy[TK]D-Fender: http://pastebin.com/m7f752733 - if I uncomment the commented lines, zap does not work... if it's commented then zap works
13:18.43awannabec4t3l, i have over and over, it only does it for *99 its really weird
13:19.06*** join/#asterisk andrewgodwin (n=andrew@southill.torchboxapps.com)
13:19.19c4t3lawannabe: what about your dialplan?  This works for the mapped button on the phone??
13:19.53*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
13:19.58awannabeon like the 550s the button works fine, but no workie if i just dial *99
13:20.30[TK]D-Fenderphpboy: phpboy>[Oct 1 15:10:48] WARNING[23735]: channel.c:3025 ast_request: No channel type registered for 'ZAP' <--- no chan_zap loaded.  En of story
13:21.09*** part/#asterisk InsolentDreams (n=Insolent@ip68-5-17-7.oc.oc.cox.net)
13:21.11c4t3lawannabe: and your msg.mwi.1.callBack whatever is mapped tp *99??
13:21.33mort_gibI need some advice.... I have ONE out of a bunch of clients that complains about dropped calls. I can turn on sip debug, but it only works while I'm logged in. Any way of making it permanent??
13:21.46phpboy[TK]D-Fender: but it works with the bottom lines commented out
13:22.00phpboySo I think it's safe to assume zap is working
13:22.03mort_gibThey have this issue, maybe once a week or so, some weeks 3-4 times
13:22.11phpboyit's just not happy with those last commented lines :(
13:22.21awannabec4t3l, correct, is that the prob you think somehow?
13:22.36phpboyIt works perfectly without the last comment lines, but I need the last lines for outbound mobile :(
13:22.38c4t3lmort_gib: change your logger.conf settings to debug and grep away at the logs :)
13:22.48mort_gibSIp debug too??
13:22.52awannabeits like the phone is trying to not use the registration when you dial *99, the phones have one registration on them, and it spans to all linekeys
13:24.29phpboy[TK]D-Fender: Any ideas?
13:24.33c4t3lawannabe:  sounds like some parameter is not being passed to you dialplan
13:24.38mort_gibHow do I get sip debug into the log file??
13:24.39c4t3lyour**
13:25.17awannabec4t3l, yeah, thats what i thought to, i can dial any other extension number, but *99 its like star codes are screwed up almost
13:26.07fred-tmftawannabe: do any other star codes work directly?
13:26.10phpboyCan anybody help please, http://pastebin.com/m7f752733 <-- zapata.conf .... the commented out lines at the bottom seem to be breaking zap :(
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13:26.15kaldemarphpboy: does your zaptel have such channels?
13:26.28phpboykaldemar: it does
13:26.35kaldemarwhere have you come up with the channel numbers?
13:26.37Hertzy3phpboy: Now the VM won't boot back up.  Which I will let you know that if something got messed up it is not a big deal, the whole point of this server is for me to learn how Xen works and this is the only VM on there.  But heres the error I receive now.  http://pastebin.com/d97e237f
13:27.15phpboykaldemar: /etc/zaptel.conf
13:27.22phpboyit comes out of there
13:27.23awannabeoh wtf
13:27.29awannabe*90 works lol
13:27.46awannabeoh wait, nevermind no it doesnt
13:28.03awannabeif you hit speakerphone and dial it does, if you hit a line key then it it fails, for both * codes
13:28.20phpboyHertzy3: did yum upgrade run properly?
13:28.38[TK]D-Fenderphpboy: You have not proved that it is loaded, or which channels * should recognize.  You have not shown FULL CLI output for your failed attempt either
13:28.52Hertzy3phpboy: I had planned to run that after the reboot
13:29.00fred-tmftawannabe: this will be within your polycom dialstring... make sure you have *XX allowed
13:29.09phpboy[TK]D-Fender: ZAP is working for group=1
13:29.16kaldemarphpboy: so you have 2 cards with four ports?
13:29.27phpboykaldemar: correct
13:29.27awannabeuri="sip:*@
13:29.30awannabeyeah, bastards
13:29.37[TK]D-Fenderphpboy: you aren't showing me anything of value.  I cannot help you
13:29.39phpboyand group=2 is the first port on the second card
13:29.45phpboy[TK]D-Fender: no problem
13:29.56phpboythanks anyway
13:30.24awannabei bet thats it, rebooting now
13:30.38kaldemarphpboy: show your zaptel.conf, /proc/zaptel/1 and /proc/zaptel/5
13:30.47phpboykaldemar: anyhoo, so, giving my zapata.conf file, if I comment out group=2 and it's channels... it works without a problem... only for the first cards channels of course... problem is, when I wanna bring the second card into the picture... it breaks
13:30.49phpboykaldemar: ok
13:31.38*** join/#asterisk Kobaz (n=kobaz@64.27.7.21)
13:31.57v4mpon my config the extensions 1 4 and 5 are setup correct ?
13:32.39*** join/#asterisk moy (n=moy@nat/ibm/x-6d01573306b2f990)
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13:33.38phpboykaldemar: http://pastebin.com/m11d92c65
13:33.46asteriskmonkeyhas any one had this error before? chan_zap.c:1045 zt_digit_begin: Couldn't dial digit 5
13:33.53phpboyI see one problem, now that I look at that, I left a channel out on span 5
13:34.07phpboyAlthough I don't think that'll damage anything?
13:34.26asteriskmonkeyit seems dtmf isnt passing out or in after a call is established but it is showing and stating it cant dial
13:34.41*** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi)
13:35.36Kattymorning
13:35.42phpboyKatty :D
13:35.50LiNeTuXKatty: yes, it is!
13:36.03Kattyputs on Warning, I am Grumpy sticker
13:36.20tzangerhah
13:36.20phpboyMy business day is almost finished
13:36.27phpboykaldemar: what are your thoughts?
13:36.43tzangeryou don't know grumpy until you've met my wife when she's reconciling the books
13:36.57v4mpi get this with WaitExten   == Auto fallthrough, channel 'SIP/84455707-08837138' status is 'UNKNOWN'
13:37.42kaldemarphpboy: set your dchan for span5 as 140 and bchan=125-139,141-155 and try again.
13:38.17awannabefred-tmft, thanks dude, that was it, added *xxT
13:38.43fred-tmftmy pleasure.
13:39.22phpboykaldemar: otherwise everything looks fine?
13:39.42write_eraseHi, someone know French FranceTelecom RNIS connectivity ? I'm not sure how to disconnect the existing PABX and connect my Asterisk server. thx
13:39.54kaldemarphpboy: with a quick look.
13:40.10SqueebHmm
13:40.14kaldemarphpboy: then you'd want to fix the channels in your zapata.conf too.
13:40.24Squeebhow can I pass caller ID to an AGI script?
13:40.41Squeebit doesn't seem to turn up if i just do AGI(test.agi)
13:40.53Squeebalthough it appears in Master.csv
13:42.31phpboykaldemar: you are too kind, it worked. thanks :D
13:43.02kaldemarphpboy: did you figure out what was wrong?
13:43.20Squeebor is there a better way of using MySQL with asterisk?
13:43.26Squeebto record calls in and out?
13:44.45*** join/#asterisk mog (n=mog@nat/digium/x-662e59cbe2f1b0f4)
13:44.45*** mode/#asterisk [+o mog] by ChanServ
13:46.25phpboykaldemar: I believe it's because I was missing one channel, why it would do this, i don't know
13:46.26phpboybut ja
13:46.28phpboyit's working now
13:46.35phpboythanks a million :D
13:46.46c4t3lKatty:  Good morning!
13:46.53*** join/#asterisk angryuser (n=Miranda@43.252.146.195.dynamic.adsl.abo.nordnet.fr)
13:46.59Kattyc4t3l: hi
13:47.25kaldemarit wasn't because of the missing channel, but your d-channel being set as the 17th channel on the span, asterisk doesn't like any other than 16th.
13:49.02padskiWhat does asterisk do if it misses rtp packets from the other end due to packet drops ?
13:49.45Kattyc4t3l: i'm a bit grumpy today due to being up with riddick half the night )=
13:50.10c4t3lKatty:  How's the pup doin?
13:51.41phpboykaldemar: this I did not know
13:52.07*** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net)
13:52.47phpboyThanks again, I really do appreciate your help
13:55.59*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
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14:10.49*** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-176.usadatanet.com)
14:11.28ctaloihey all - anyone have an example of a rollover type dial plan?  example, call comes in to DIDA, if DIDA busy try DIDB etc...
14:11.29ctaloi?
14:11.57[TK]D-Fenderctaloi: Inbound rollover is ont *'s job, its the telco's <-
14:12.17ctaloi[TK]D-Fender - in this case, I am the telco
14:12.24ctaloiproviding DID trunking
14:12.27[TK]D-Fenders/is ont/isn't/
14:12.46phpboykaldemar: what does it mean if asterisk says that a zap channels status is yel?
14:12.53phpboyyellow I'm guessing
14:13.11UnixDawgthat it has turned chicken
14:13.12[TK]D-Fenderctaloi: if the call comes in on another DID then you do whatever you want with it.
14:13.19coppiceor the gain is too high
14:14.05[TK]D-Fenderctaloi: Sound like you need to keep a rollover DB for the order to attempt channels in.
14:14.07*** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130)
14:14.26[TK]D-Fenderctaloi: start loop, try1, increment, try 2, etc.
14:15.08ctaloiD-Fender - that sounds about right...... do you know of  any examples of that in a dial plan I can refrence?
14:16.01[TK]D-Fenderctaloi: No such thing.  How you do it is up to you and your DB method, etc.  And the mere fact you're asking this and claiming to be a telco scares the hell out of me.
14:16.13*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a1191a1782757a23)
14:16.13*** mode/#asterisk [+o putnopvut] by ChanServ
14:16.46ctaloiD-Fender - No reason to be scared, the call flow isn't what you think it is...
14:16.54[TK]D-Fenderctaloi: Up to the method you handle your peers with.  What tech... only about a few dozen different factors..
14:17.19*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
14:17.52ctaloi[TK]D-Fender - Agreed, was just wondering if anyone had any examples, no biggie
14:18.11[TK]D-Fenderctaloi: Too many factors for there to be a usable sample
14:19.05ctaloiOkay... Thanks
14:19.58[TK]D-Fenderctaloi: If you're looking at a hard-coded method , in order you would check the status of 1st.  If in-use jump to next, otherwise dial it.
14:20.35*** join/#asterisk shriven (n=shriven@rdu.crosscomm.net)
14:21.09phpboywoah, this second span is freaking out
14:21.28ctaloiyah, that's what I'm thinking... I'm not routing based on the DID, I'm routing a 1k block to the Asterisk, and then matching and routing to a username on an Adtran at the customer CPE, the user equals a physical DS0 on the CSR's gear
14:21.40shrivenhello. I am curious about asterisk 1.6.1, I am curious if anyone knows why it was split fro 1.6.0? Especially as 1.6.0 isn't even released?
14:21.40phpboygetting the following for all the channels on the second span on the second card:- [Oct  1 16:19:38] WARNING[9608]: chan_zap.c:1465 zt_disable_ec: Unable to disable echo cancellation on channel 185
14:22.19[TK]D-Fendershriven: Because 1.6.0 RC is feature locked <-
14:22.31shrivenahhh ok thanks
14:23.05adr3nalin3phpboy: I have been getting that too.  '
14:23.31shrivenokay, another question. Is the ldap integration built into 1.6.0 usable?
14:23.40*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:24.01phpboyadr3nalin3: it's got something to do with the dchan :/
14:24.18phpboykaldemar: you in for a little bit of final advice (hopefully)
14:24.33adr3nalin3mine did too.  I kept "losing" my  d-channel
14:24.40phpboybchan=156-170,172-186 \n dchan=171 <--- that looks correct?
14:24.51adr3nalin3Is it a te122p?
14:25.10beekmorning
14:25.23phpboyhi
14:25.29phpboyno
14:25.45[TK]D-Fendershriven: Go try
14:25.50phpboyadr3nalin3: 2 x Digium Quads
14:26.02phpboyI forgot the model number :(
14:26.15shrivenwell I am hoping to, but if someone here already knows that it's jacked, no point in wasting my time. ;)
14:26.38*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000022.dsl.bell.ca)
14:27.12adr3nalin3phpboy: did your log look similar to this?  http://pastebin.com/m45615c9a
14:27.26beek[TK]D-Fender: I just had an interesting conversation with my proprietary system vendor.  I want to put an * box in front of it and have my PSTN PRI connect to the * box, then from * to the Iwatsu.  They asked if * can replicate NI2 (e.g. act like the CO).   I assume yes, but can it?
14:27.40adr3nalin3yes
14:28.01[TK]D-Fenderbeek: Yes
14:28.48beek[TK]D-Fender: That's excellent news.   Thanks very much.
14:28.52phpboyadr3nalin3: mine looks identical
14:28.57phpboyI can help you fix yours though
14:29.08[TK]D-Fenderbeek: "signalling=pri_cpe" <- to telco.  "signalling=pri_net" <- to PBX
14:29.08phpboyin /etc/zaptel.conf
14:29.33[TK]D-Fenderbeek: You would take timing from your telco span, and pass it to your PBX span
14:29.40phpboyadr3nalin3: you using T1 or E1?
14:29.43adr3nalin3t1
14:30.03phpboyadr3nalin3: I don't think your dchan on zaptel.conf should be 24 :(
14:30.25*** join/#asterisk murraytm (n=murraytm@wsip-68-224-219-238.br.no.cox.net)
14:30.31phpboymy country uses E1, so I'm not 100% sure about T1 but I'm pretty sure your dchan shouldn't be 24
14:30.44beek[TK]D-Fender: I really appreciate your help.   Thanks.
14:30.52adr3nalin3I think it should, bchan 1-23 dchan 24
14:31.23*** join/#asterisk cguerrero (n=cuauhtem@200.79.231.94.static.cableonline.com.mx)
14:31.26cguerrerohello
14:32.06cguerrerocan any one help me, I like to know if it is posible to dial a cel phone and get the voicemail mesage
14:32.18adr3nalin3Yes on my other box it is that way working well, but when I checked my /etc/zaptel.conf it has bchan 1-22, which I have fixed already but the damn gui overwrote again
14:32.20[TK]D-Fenderadr3nalin3: It should....
14:32.21cguerrerobecause right now when I dial a cel i get a busy tone
14:32.24phpboyadr3nalin3: Try to confirm that, because I think I read otherwise about T1's and asterisk
14:32.45cguerreroafter 30 seconde
14:32.57phpboyI wonder why my config isn't happy with the dchan :(
14:33.07cguerreroand i f I dial from a landline I get the recording for the voicemail
14:33.20phpboy[TK]D-Fender: Does my config for E1 seem right for the 6th span? bchan=156-170,172-186 \n dchan=171
14:35.34*** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com)
14:35.44murraytmi'm having a problem where when asterisk is started from /etc/init.d/asterisk, my AGI scripts can't load some libraries.  if asterisk is started from /usr/sbin/asterisk, everything works fine.  what's the difference between those two environments?
14:35.51[TK]D-Fenderphpboy: Depends on what cards, what order, etc.  I'm not going to answer based on 1 single line like that.
14:36.25c4t3l<PROTECTED>
14:36.32murraytmcentos 5
14:36.32[TK]D-Fendermurraytm: Because your init script might be running * as non-root.
14:36.44c4t3lgood point!
14:36.58[TK]D-Fendermurraytm: And you shoud already have been fully aware of that.
14:37.07[TK]D-Fendershould*
14:37.37murraytmi figured as much, but can't tell what user it's running as.  i didn't configure it to run as non-root
14:38.15angryuseri need to save all sip debug output of defined peer to a file, any help ?
14:38.49murraytmso then i need to get with some centos / redhat people then?
14:39.05[TK]D-Fendermurraytm: Easy to see.
14:39.12c4t3lmurraytm: did you install via rpm?
14:39.14lowtekangryuser: asterisk -rx "sip show peer XXX" > outputfile
14:39.21[TK]D-Fendermurraytm: How did you install * in the first place?
14:39.37lowtekoh, debug output, sorry
14:39.38murraytminstalled from source
14:39.52tzafrir_laptopphpboy, I just use zapconf / genzaptelconf to generate zaptel.conf without worrying about the exact numbers
14:39.58[TK]D-Fendermurraytm: go look at the running daemon and see what user its running as.
14:41.41murraytmlooks like it's running as root
14:44.03[TK]D-Fendermurraytm: go run System with an open shell script that will confirm things.
14:44.55*** join/#asterisk leif[astricon] (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com)
14:45.20murraytm[TK]D-Fender: "System"?
14:45.32[TK]D-Fendermurraytm: "core show application system"
14:45.55phpboytzafrir_laptop: thanks a lot, it generated it the way i did, which is good news, anyway... asterisk is still giving me issues about the dchan getting yellow alarm(4), any ideas?
14:46.39tzafrir_laptopphpboy, do you use crc4 on that line?
14:46.46tzafrir_laptopis that line ok?
14:49.17*** join/#asterisk seanmh (i=seanmh@216.31.101.24)
14:49.34phpboytzafrir_laptop: yeah, seems fine
14:49.35phpboycrc4, yes
14:49.50phpboyoh, you mean the physical line?
14:50.21tzafrir_laptopYes. Someone along the line has a red alarm
14:50.31tzafrir_laptopAt least in one direction
14:51.09phpboyhmmmm, I think it may be on my telco's side
14:51.25phpboythis is possible, yes?
14:57.59*** part/#asterisk leif[astricon] (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com)
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15:02.16Katty[TK]D-Fender: my boss just discovered sip trunking :<
15:03.07cguerrerohow can I gate ringing tones from asterisk before early media
15:03.46cguerreroget
15:04.47*** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net)
15:05.00*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:05.33Kattyhi james.
15:05.48Kattyseanmh: ping?
15:06.42seanbright~siptrunk
15:06.42jbotNo such thing, my friend.. Like too much salty plum soda.
15:06.54seanbright~seanbright
15:06.55jbotfrom memory, seanbright is a girl with standards
15:06.59seanbrightcorrect!
15:07.03jameswf~me
15:07.03jbotno u
15:07.08Katty~Katty
15:07.08jbotyou are probably the only girl in the channel, so be nice to her
15:07.21Kattyjbot: Katty?
15:07.22jbotyou are probably the only girl in the channel, so be nice to her
15:07.26*** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net)
15:07.26jameswf~jameswf
15:07.27jbotjameswf loves unsolicited technical support, or http://jameswf.info
15:07.29Kattywe need mutiple replies )=
15:07.34[TK]D-FenderKatty: ...
15:07.36[TK]D-Fender~siptrunk
15:07.37jbotNo such thing, my friend.. Like too much salty plum soda.
15:07.40jameswf~porn
15:07.41jbotPorn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type.
15:07.46jameswf~botabuse
15:07.46jbot[botabuse] fun
15:07.52Katty[TK]D-Fender: i'm scared )=
15:07.55tzangerha
15:08.08tzangerit's a good thing the porn link and the siptrunk link aren't related
15:08.09[TK]D-FenderKatty: that's "skeered" ;)
15:08.11Katty[TK]D-Fender: i hear bad things about sip trunks )=
15:08.37jameswfgot junk in the sip trunk hayyyyyy
15:08.44*** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
15:08.46creativxi see dead sip trunks
15:08.54Kattycries.
15:09.04Kattyi just wanna go home and play with my puppy
15:09.07Kattythe world is too skeery today
15:09.09tzangerthe best take on that was "I see dumb people.. they're everywhere.. I work with them."
15:09.16jameswfcreativx: you know some dont even know they are dead
15:09.32gr0mitgaaaa - my asterisk box just died.
15:09.33[TK]D-Fendertzanger: "If I don't work for a circus.... then why am I surrounded by clowns?"
15:09.37*** join/#asterisk diverscuba (n=diverscu@d-72-9-4-157.cpe.metrocast.net)
15:09.39tzanger[TK]D-Fender: that's another good one
15:09.51jameswfEverytime you boot trixbox god kills a puppy.... thats what i hear
15:09.53tzangerthis vhdl is killing me
15:10.22Kattyholds riddick :<
15:10.23creativxjameswf: indeed
15:10.28KattyNOT MY PUPPEH!
15:10.42gr0mitsneezes
15:10.44jameswfKatty: put down the lime green box
15:10.47Kattyi haven't had a chance to nom on his ears yet )=
15:10.58murraytm[TK]D-Fender: i must not understand what an open shell script is or how i should be using system.  i put it in the dialplan and ran it, but came up with nothing useful.
15:11.05Kattyjameswf: i stay farrrr away from trixbox
15:11.40jameswfLime should be reserved for alchoholic beverages
15:11.54jblackAnd concrete
15:11.59*** join/#asterisk diverscuba (n=diverscu@d-72-9-4-157.cpe.metrocast.net)
15:12.19gr0mitand cheesecake
15:12.23*** join/#asterisk logicwrath (n=no@c-68-42-253-39.hsd1.mi.comcast.net)
15:12.24jameswfI thought that was lye... also good for disposing of bodies
15:12.27Kattyi like cheesecake.
15:12.57*** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net)
15:13.32jblackjameswf: Absolutely true. Just a few days ago, I saw a pile of dead puppies, and figured "someone must have rebooted their trixbox server a couple dozen times".
15:13.36jameswfI am here today to talk about the effects of ADD/ADHD in the business world....... oh crap is that the new ipod
15:14.07murraytmdid i join 4chan or something when i wasn't looking? :)
15:14.10creativxlye? lutefisk? si si
15:17.36[TK]D-Fendermurraytm: #!/bin/bash      whoami > someplace&file
15:18.03[TK]D-Fendermurraytm: No, this is clearly SomethingAwful
15:18.21jameswfAwful is relitive
15:20.08murraytmthanks, it's definitely running as root
15:20.26jameswfheh http://houghi.org/shots/vim001.gif
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15:21.21tzangerjameswf: haha
15:21.24tzangerold but fun
15:21.38murraytmjameswf: awesome
15:21.59jameswfI use to have one... it appears you are trying to write a suicide note....
15:22.05tzangeryeah
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15:27.22[TK]D-Fenderjameswf: I have that one at home.
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15:46.59murraytm[TK]D-Fender: any other suggestions?  if it's running as root, what else could be different?
15:47.33[TK]D-Fendermurraytm: You never actually showed your problem or any of your backup so I have nothing to comment on.
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15:49.34murraytmsorry, i thought i had.  the problem is that my AGI scripts can't load some libraries when it was started by /etc/init.d/asterisk (as opposed to just /sbin/asterisk).  what else do you need?
15:49.46[TK]D-Fender~pb
15:49.47jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:49.49[TK]D-Fender^^^^
15:50.24[TK]D-Fendermurraytm: I would need to see the problem, not your description of it.
15:52.24murraytmi'll see if i can put together a simple reproduction case and come back later
15:52.27murraytmthanks for your help
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15:56.29[TK]D-FenderAmazing... ask to actually see a problem and *poof*, out they go.
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16:22.16hardwirehai'
16:25.49hardwireI mean it
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16:26.15hardwireblah
16:27.21jameswfhas the sky fallen yet?
16:30.21bkw_jameswf: yep.. I'm sweeping it up as I type
16:30.29bkw_jameswf: do you want that in a bag or a box?
16:30.58jameswfummmm is the bag recycled?
16:31.24theharpaper or plastic?
16:31.30theharor did you bring your own bag with you this am?
16:31.46c4t3lgood lord what is with ppl today!!!???
16:31.57c4t3lnot you guys of course
16:32.03c4t3lyou guys are cool
16:32.06jameswfi feel like a crazy peson bringing my own bags
16:32.11jameswfI need to go hug a tree
16:32.35jayteeif it's a giant redwood, better bring alot of friends
16:33.01bkw_haha
16:33.03jameswfIf its a gialnt redwood i am going to bring a chain saw psh
16:33.07c4t3lKatty:  You still grumpy?
16:33.09theharthe checkers at whole foods look down on you when you don't bring a bag and ask for plastic
16:34.22jameswfI look at em and say F___ off i use linux i am a better hippy than you will ever be
16:35.04tzangerhaha
16:35.13jameswfevery year billions of trees are killed so microsoft and apple can compete..... tree killers
16:35.33hardwirethose bastards
16:35.40theharthe signature in my company's announcements is "save a tree use email!"
16:35.42thehartender, i know.
16:35.45hardwirejameswf: I put my hippy foot out the other day
16:35.59hardwireSun was proclaiming being eco friendly by using x86 in their proprietary chassis
16:36.09hardwire"Can I replace the mainboard with any ATX mainboard?"
16:36.10hardwireno..
16:36.18jameswfStop global warming kill a cow
16:36.22hardwire"Can I do anything to it that would use off the shelf parts?"
16:36.25*** join/#asterisk oej (n=olle@ns.webway.se)
16:36.25hardwirenot really.. no
16:36.35*** part/#asterisk mesfet (n=iw3grx@89-96-227-146.ip14.fastwebnet.it)
16:36.37theharwe use 100% reuseable energy in our full colocation and our offices. http://www.xmission.com/community/green/
16:36.40hardwire"So you guys made yet another chassis we have to melt down and recycle?"
16:36.44hardwireyes.. yes we did.
16:37.02hardwirethehar: you work for xmisson eh?
16:37.04theharyes
16:37.07hardwireGODO 4 U
16:37.10theharoh?
16:37.12hardwireerr
16:37.12hardwiregood
16:37.13hardwireoh
16:37.20hardwireis emailing all the people he met @ astricon today
16:37.20*** part/#asterisk jsolis (n=jimmy@190.232.168.56)
16:37.24hardwirea bit late.. better late than never.
16:38.02theharhow do you know about xm?
16:38.06theharif you say maddox i'll laff
16:38.06hardwiredude
16:38.09hardwireI get around
16:38.11hardwireand yes.
16:38.21theharbest page in the universe
16:38.21hardwirenot directly
16:38.38hardwireI'm the occasional onlooker
16:38.54theharhehe
16:39.00theharyeah XM rocks.
16:39.06hardwirejobplz?
16:39.17hardwirekthx.
16:39.32theharyou want to live in the big slc, eh?
16:39.43hardwireI'm not holy enough for SLC
16:39.56jjshoeslc?
16:40.03theharif you live in the metro area you will be just fine.. outside of downtown you're a big sinner
16:40.08theharsalt lake CITY, utard
16:40.22hardwireway to wave the politically correct flag :)
16:40.32hardwireI love pointing my finger at PC violators
16:40.35hardwireit gives me a rush
16:40.40thehar:D
16:40.44hardwireexcept I think I said retarded at least 15 times yesterday
16:40.51hardwireand I inspired somebody else to say it
16:41.03theharno that's the word for utahns. it's really the perfect definition.
16:41.10jameswfRetarded is PC
16:41.13hardwireI said "not like this is all that PC.. but.. this cement is retarded.. sorry, that's just how I feel"
16:41.22hardwirewhich made the person I was talking to somehow say it 6 more times.
16:41.42hardwirejameswf: anything derogatory isn't PC. :)
16:41.48jameswfthe support groups are all caller "arc" XXXX associatonn for retarded citizins
16:42.52hardwirelike.. you never here "and lets give it up for all the retards here at the special olympics.. bang up job guys."
16:43.00hardwirehear
16:43.01*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
16:43.01hardwireblah
16:43.04hardwireI'm a bit out of whack today.
16:43.10bkw_TODAY only?
16:43.14bkw_I think ALWAYS
16:43.18hardwiredo I know you?
16:43.38bkw_no clue... but apparently you haven't a sense of humor  :P
16:43.50Cutlassdumb question...I just upgraded to 1.6 from some pre-1.2 version....isn't there a CLI command to show all the SIP users?
16:43.51jameswfYOu are all differently abled
16:43.58hardwirebkw_: I never said you were wrong.
16:44.02theharwat is a sense of humor, precious?
16:44.05hardwireI just never know who's looking over my shoulder lately.
16:44.06bkw_hardwire: hehe
16:44.14bkw_hardwire: I know I'm totally freakin nutz
16:45.08hardwirebkw_: that's a matter or perspective.
16:45.13bkw_could be
16:46.13hardwirehugs his repaired laptop
16:46.14theharsubmits order for Asterisk: The Future of Telephony, 2nd Edition
16:46.17hardwireI missed it soooo
16:46.22hardwireslowest laptop evar
16:46.23hardwirebut I love it
16:46.56jameswfthanks for the reminder i need to order a couple books for give aways
16:48.36thehari have the lovely pdf but a physical book makes life so much easier
16:49.05bkw_thehar: I use mine as a monitor stand!
16:49.58thehari have lovely three 22" LG lcds.. i don't need monitor stands :)
16:50.36bkw_I have a single 24 inch iMac
16:50.36jameswfI find consoles look funny when you put it on big screens
16:50.47bkw_jameswf: I love it.. my terminal is 160x61
16:50.52bkw_at 14 pt
16:50.52jameswfooh washington DC calling
16:51.09theharmine is 93 x 28
16:51.17theharat like 8 pt
16:51.24bkw_you must not love your eyes
16:51.25bkw_:P
16:51.54thehari have a lot of terminals to use and windows! i need space
16:52.07[TK]D-FenderMine is 80x25 @ 1.2 IPC ;)
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16:58.09theharhow easily asterisk-users takes over my mailbox
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17:04.33logicwrathim using all SIP, with softphones and a 1 or 2 desk phones.  I have no digium hardware in my server.  What kind of EC solutions can I use?  I have soft echo but I think it required a digium card.  I would buy a cheap card if that would work but all the channels will be using SIP and skipping teh card.  I was not sure if that would work.  Can someone advise?
17:04.47*** join/#asterisk hardwire` (n=hardwire@rdbk-11447.mtaonline.net)
17:04.49logicwrath*I have seen soft echo..
17:05.09angryusercan someone add a tag to ~siemens like this 'C675ip 2 phones OK: C650ip 2 phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00, : C450ip 2 phones OK'
17:05.18Greek-Boyany call center admins here?
17:06.44[TK]D-Fenderlogicwrath: If you've got echo with an ITSP, its their fault & problem.
17:07.03[TK]D-Fenderlogicwrath: There is no EC over VoIP, its done at the PSTN level.
17:08.08logicwrathis ITSP internet telephone service provider?
17:08.15*** join/#asterisk docelmo (n=docelmo@206.248.239.194)
17:08.37J0n5555i have, I hope, a simple question... I'm wanting to setup a voip pbx for 4 of my family members and me... we call each other alot, but still need to call out as well.... If i get to voip trunks for 5 houses, i know I can save upwards of $200-300 a month (done the math already)... however, is there any way to report different addresses for 911 service? three of us are disabled and are concerned
17:09.08docelmoe911 is attached to your DID
17:09.14[TK]D-Fender~seimens
17:09.27docelmoso if every house has a seperate DID then the e911 info will be registered to this house.
17:09.47[TK]D-Fender~siemens
17:09.48jbot[~seimens] Seimens C675IP, C650IP, C450IP phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00
17:09.57docelmoYour outbound routing will need to be setup as such the correct callerid is presented to the outbound provider correctly for the caller
17:10.02J0n5555docelmo: hmm... I was hopeing to save money by just having 1 did and using extensions... but if that's the way it needs to be I can go read up on it
17:10.04angryuserthank you
17:10.34docelmoj0n5555 DID's are fairly inexpensive $2-10 per DID in most cases
17:11.10docelmothen just purchase pay as you go outbound this way your MRC is only the DID's it will end up being maybe ~60 a month for 5 house holds at worst case scenario
17:11.15J0n5555docelmo: great, yea, i remember seeing i on the list... I'm looking at bandwidth.com, but haven't shopped around much yet... just getting my feet warm
17:11.36docelmoj0n check www.aretta.net  they have hosted pbx's also
17:12.08J0n5555docelmo: ty!
17:12.13docelmoall good
17:12.27[TK]D-Fenderlogicwrath: Yes
17:12.57docelmosay anyone done any load testing with meetme on asterisk to find the max number of conferences that can be brought up under ztdummy or dahdidummy?
17:12.59[TK]D-Fender~siemens
17:12.59jbot[~siemens] Siemens C675IP, C650IP, C450IP phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00
17:13.06[TK]D-FenderThere, fixed the spelling
17:14.25angryuser[TK]D-Fender : you need to specify the number of phones, siemens is able to support 6 for some bases, and the rumor says the instability is more important with more phones, but i cant confirm or deny that, i have tested with 2 only
17:15.03docelmo[tk]d-fender any knowledge of meetme?
17:15.28[TK]D-Fenderdocelmo: Not to answer your previous question with.
17:15.43docelmowonder if there is anything on google..
17:16.06docelmoI guess we shall see or I could use SIPp and just bombard a box with ulaw and see where it explodes
17:16.32[TK]D-Fender~siemens
17:16.32jbot[~siemens] Siemens C675IP, C650IP, C450IP phones OK using 2 phones. C470ip -> bad firmware for * V02123: 021230000000 / 041.00
17:19.01angryusernice phones, on the paper ;)
17:19.03*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
17:19.58[TK]D-Fenderangryuser: Same can be said of Grandstream
17:20.49*** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net)
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17:22.00UnixDawganyone know when the new asterisk-now iso is going to be released ?
17:22.24[TK]D-FenderUnixDawg: "when its done"
17:22.58UnixDawgwell I thought it was as they had a dlsite atastricon some one said.
17:23.04UnixDawgwas it only a beta
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17:23.57scooby2offtopic kinda: Anyone know if you can make an 800# failover? We have two pri's from different providers. If one provider goes down, is there any way to make the 800's flip to the other?
17:24.10scooby2or any other way to make it more robust
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17:25.38[TK]D-FenderUnixDawg: I'll ammend that with "When its no longer beta" :)
17:26.44[TK]D-Fenderscooby2: Yes, thats the telco's job.
17:28.33scooby2I was hoping someone here had experience with that. We have two different providers so we have a little redundancy but it would be nice if the numbers could roll over some how
17:28.36*** join/#asterisk hardwire` (n=hardwire@srv001.gandi.brutetech.com)
17:28.44*** join/#asterisk gsiener (n=gsiener@209.169.48.66)
17:29.35[TK]D-Fenderscooby2: Nothing for us to have experience with.  Its the TELCO's job
17:30.20gsienerRecommendations for US iax trunk provider?
17:31.01[TK]D-Fendergsiener: IAX is generally unrecommended unless necessary.  Otherwise, Teliax
17:31.17*** join/#asterisk chigital (n=chigital@tmo-122-1.customers.d1-online.com)
17:31.21gsienerI guess I meant voip trunk provider
17:31.22jksMwhy is it that IAX is not recommended?
17:31.33[TK]D-FenderjksM: Stability & quality
17:31.58jksMhmm, I'm using IAX2 for trunking right now... should I consider switching to SIP? - what kind of stability and quality problems can I run into?
17:32.02jksMare we talking sound quality or?
17:32.14jameswfWhere do i get a voip trunk google says i cant have one
17:32.17hardwiresigh
17:32.33jameswf:)
17:32.53gsiener<PROTECTED>
17:33.11[TK]D-FenderjksM: If you are running fine, then leave it.
17:33.49[TK]D-Fendergsiener: what is "looping"?  What is "logging in" supposed to mean?
17:34.09jksMwell, I have a weird problem with audio not functioning properly when the machine is under high load (problem = parties cannot hear each other, even though both sound streams are available when doing a network sniff at the asterisk server)
17:34.38gsiener[TK]D-Fender: when I log into asterisk-gui, it enters a loop of parsing the config files repeatedly, instead of logging in
17:35.13[TK]D-Fendergsiener: Definitely a problem for their channel then.  This is not 2nd level GUI support
17:35.29[TK]D-FenderjksM: any transcoding?
17:35.34jksMno transcoding
17:35.42gsiener[TK]D-Fender: fair enough, just wondering if anyone had seen it before
17:36.28[TK]D-FenderjksM: Try using SIP and see if it improves.
17:37.28*** join/#asterisk mltlnx (n=mltlnx@207-237-36-133.c3-0.nyw-ubr3.nyr-nyw.ny.static.cable.rcn.com)
17:37.50jksMhmm, I would like to be more sure that it is IAX that is the culprit before switching... it's hardcoded a lot of places
17:37.56mltlnxHello, I have the need to block inbound calls from a certain area code....What is the best way to do this?
17:38.03jksMand it's very hard to debug as it seems to happen only once a week or something like that
17:38.39*** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com)
17:39.04Kattyis there a way to do something /after/ a caller hangs up
17:39.23*** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com)
17:39.23Kattyexten => foo,1,Dial - exten => foo,2,if user hangs up do this
17:39.35nny_2Is anyone here well versed with Tellabs ECs? I am in a bind here.
17:41.19[TK]D-FenderKatty: read up on your Asterisk Standard Extensions....
17:41.52*** join/#asterisk Talirk (i=434e2716@gateway/web/ajax/mibbit.com/x-241dfef0ddd48b2f)
17:42.02Kattyi need something more specific than that.
17:42.09Kattynot all the time
17:42.13angryuserKatty: : g option in dial or use 'n' in priority
17:42.15Kattyjust when certain numbers are dialed.
17:43.10Kattyooh, g will be useful
17:43.10TalirkDoes  anyone here use FreePBX, I have a question on apply a custom  dialrule , but I want to do it thru the gui verus directly editing the conf files. Is it possible
17:43.26jayteemltlnx, you could use a GotoIf where if the area code of the callerid matches the unwanted area code it would jump to a priority that hangs up the call or plays a message and hangs up and if it doesn't match just continues in the dialplan.
17:43.39[TK]D-FenderKatty: "h" <-
17:43.54[TK]D-FenderTalirk: Ask in #freepbx . It is not supported here
17:43.54mltlnxthanx
17:43.59Kattyh?
17:44.01Kattylooks at h
17:44.13Katty[TK]D-Fender: that's not what i want.
17:44.22Kattyangryuser: g is what i want.
17:44.24Kattyangryuser: thanks
17:46.14zacwolfeANNOUNCEMENT: Visual call-flow editor and stand-alone server now in Beta and available for free download from http://www.safisystems.com
17:46.58Kattyangryuser: i'll pastebin what i'm doing if it works
17:47.12jameswfzacwolfe: for linux?
17:47.41jjshoezacwolfe nice, based on eclipse rcp?
17:47.54*** join/#asterisk tkbeat (n=tk@p54B9474A.dip.t-dialin.net)
17:47.59zacwolfeyeah its an RCP-based app
17:48.04Kattydoes anyone know where rm is?
17:48.16*** join/#asterisk exothermc (n=miles@74.85.89.146)
17:48.19zacwolfethe server runs on Linux but designer is only avail for Windows at this time
17:48.21jameswfwhich rm
17:48.28KattySystem,1,(rm  /this/file)
17:48.48jameswfjames@james-laptop:~$ which rm
17:48.48jameswf/bin/rm
17:49.03Kattycheers.
17:49.39jameswfdoesnt own a windoze box :(
17:50.13Kattyyay it works
17:51.19Kattyangryuser: http://pastebin.ca/1215999
17:51.40angryuserh is good too, but not like option to dial(), at hangup dial searches for h  in context where dial is and executes it's code. I am not sure if you need to add another option to dial() to make it happen
17:51.53Kattyneeds a few changes
17:51.54angryuserexecuted*
17:51.56Kattybut basically accomplishes what i want
17:53.15zacwolfejameswf: yeah there were some minor graphical glitches in the Linux version of the designer but that was around last Christmas and I know there were some fixes that came out for the graphical lib we're using. I'll try making a linux build again and see how it looks.
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17:54.28nny_2can someone explain the difference between a -48 vdc power supply and a 48vdc power supply?
17:55.22nny_2I am in a greta bind here. Had someone sell me a cage and EC and now that I have them all I realize this setup is absolutely not mean for 1 card, and the person I was talking to is not available
17:55.35angryusernny_2 : alternative or continiuos power supply (not sure if it sound like this in english ;)
17:55.42nny_2AC
17:57.02Kattymmm, variable power signals!
17:57.02nny_2Says INPUT AC 100-240v 2.0A OUTPUT 48v 1.3A The card is reported to need "-48vdc power"
17:57.25nny_2Nearly all telephone central offices, where these cards were meant to live, are powered from -48. The shelf, ground, and the positive terminal of the power supply are all connected together. The "hot" side of the supply is negative with respect to ground. Why? Because any leakage current on a telephone line will cause metal to build up on something that is negatively charged with respect to its surroundings (earth) and will cause the metal to dissolve into
17:57.28nny_2from the wiki
17:57.50[TK]D-FenderKatty: "h" was what you asked for.  You said when the caller hangs up.  If you want the caller to continue when the CALLEE hangs up, then thats "g".  And if you don't do "g", BOTH will end up in "h" anyways
17:57.59nny_2I was told to order this power supply, but not clear if it is the proper one or not, as it says NOTHING about -48vdc
17:58.20Katty[TK]D-Fender: just be happy it works (=
17:58.26Katty[TK]D-Fender: stop making a fuss over the little things in life.
17:59.14Kattynny_2: it's AC
17:59.53nny_2well crap this power supply I just blew cash on says output DC
17:59.58Kattynny_2: you'd need something to convert the dual AC voltage into DC negative and positive
18:00.09nny_2I swear screw echo
18:00.12nny_2this is such bullshit
18:00.15Kattylike a diode bridge or something
18:00.19hardwirenny_2: :(
18:00.28scooby2Wierd issue with 11.4.21.2. Occasionally an Agent will login, get one call, then Asterisk kicks the agent out. No mention of anything in warnings or verbose.
18:00.28nny_2fuck that, I just want my god damn 1000 dollar echo canceler to work
18:00.30Kattywhich would make sure the ouput would remain the same directions, regardless of the input polarity
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18:00.44nny_2<-- pissed, brb
18:00.47Kattyi'm not sure your average power adapter is that smart ^_-
18:00.55Kattybut what do i know (=
18:00.56*** part/#asterisk killfill (n=killfill@200.63.96.244)
18:01.02Kattyi just like puppehs
18:05.35*** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com)
18:06.42bipolarDoes anyone know of a way to track all CDR entries for a call from ring to hangup in mysql?
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18:07.17bipolarwonders if that question makes any sense
18:07.57nny_2bipolar: http://www.voip-info.org/wiki-Asterisk+cdr+mysql
18:08.42v4mpdoes this look right exten => 84455707,WaitExten() as i dont see any viable option i would need there from the cli
18:08.48bipolarnny_2: yes, it's installed and running. Afaik that page doesn't cover what I'm asking about.
18:09.10bipolarmaybe there's something in the cdr config I'm missing
18:10.05nny_2bipolar: it seems to track all of them in our implementation, not sure what you are missing
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18:10.52bipolarnny_2: the issue is that if a call comes in and gets passed around between extentions each transfer makes a new entry in the cdr table.
18:10.54nny_2Katty: thanks for trying to help, i dunno. I feel like I am building a bridge to get over a puddle right now
18:11.03nny_2bipolar: ahh
18:11.09nny_2bipolar: yeah mine does that too
18:11.20bipolark. I'm not insane then :P
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18:11.41Kattynny_2: i know that feeling all too well
18:11.48Kattynny_2: sometimes i feel like a rag that just cleans up messes
18:12.06jameswfNeat: http://ocw.mit.edu/OcwWeb/Electrical-Engineering-and-Computer-Science
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18:12.40nny_2jameswf: i saw that, looks cool. Just need to find the time
18:13.30coppicejameswf: that stuff has been there several years
18:13.36nny_2Katty: yup. Installed a system. Dialplan and underlying is solid, but the T1 echo has been a nightmare
18:14.13jameswfis slow...
18:14.56jameswfalways ammused by the frantic subject " I got hacked"
18:15.02coppicethere's some really good stuff there for comms, DSP and other fun topics
18:15.03jameswf:)) dont do that
18:16.50*** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net)
18:17.20FruitBasketuhm... help... when I perform a transfer, ${CDR(accountcode)} ends up blank. How can I fix that? I'd prefer the account code of the incoming call be preserved through a transfer...
18:18.01QwellJerJer: ping
18:18.04bipolarthe CDR webscripts are able to track the CID as it gets passed around, but I don't know to do it in mysql. time to dig into the CDR webscript code.
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18:21.43jjshoeQwell: No Route to Host
18:21.53Qwelljjshoe: see link on efnet
18:21.58Qwellthey responded
18:22.11citywoki have a question about sip bandwidth.  i'm using ulaw (g711), and each call is generating 160kbps of bandwidth.  10 calls / 1.5mbit.  why is it so high?  i thought g711 was supposed to be a bit less than half of that?
18:22.24Qwellcitywok: each direction
18:23.05citywokoh really.  if the provider (voicepulse) doesnt do g729, is that my bst option then?
18:23.22[TK]D-Fendercitywok: 80kbps per DIRECTION
18:23.38citywokyea, i understood that when qwell said it
18:23.58[TK]D-Fendercitywok: Yeah, I was typing and then distracted as he answered...
18:24.01jjshoeQwell everything but the last response was good too, shocking.
18:24.13citywokoh haha, no worries then :-)
18:24.13Qwellgood?  it's terrible.
18:24.49nny_2citywok: http://www.asteriskguru.com/tools/bandwidth_calculator.php
18:24.58Qwelljjshoe: This means that hundreds (thousands?) of Asterisk users are in violation of Ciscos licensing terms
18:25.10*** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
18:25.17Qwelland finding that out, was exactly the point of this whole ordeal
18:25.51citywokthank you nny_2 i was trying to compare bandwidth usage looking up codecs on voip-info
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18:27.39d33p5nthat's interesting, i ought to be able to pump 8 concurrent calls over a 1.5mbps pipe with room to spare
18:27.53citywok10 was hitting 100% or so for me d33p5n
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18:28.14FruitBasketDoes anyone know how I can make Asterisk keep the accountcode when I do a transfer?...
18:28.20d33p5nADSL for me means do your max calculations based on 70-80% of your peak
18:28.25d33p5nto be safe
18:29.02d33p5nFruitBasket can you assign it to a var and then pick up that var after the transfer?
18:29.04citywoki want to know per 1.5mbit, testing with a solitary T1, soon to be a full ds3
18:29.34[TK]D-Fenderd33p5n: Really?  Using what codec?  What's your upstream?
18:29.36jjshoeQwell what's your legal think?
18:29.42FruitBasketdeep: dunno, but that was the next step.. even if I can, though, it's still problematic as I can't guarantee where they will be going.
18:30.01Qwelljjshoe: we don't use Cisco phones.  It wouldn't affect us :p
18:30.27[TK]D-Fendercitywok: IAX2 trunking can save you 16-20 kbps / call from the 2nd onwards.
18:30.27citywokso i set the codec to be GSM, i see it set channel to write format gsm, and if i do sip show channel it says GSM for a few seocnds, then switches to 0x4 ulaw, any idea why?
18:30.33*** join/#asterisk denon (i=denon@synapse.subneural.net)
18:30.34*** mode/#asterisk [+o denon] by ChanServ
18:30.49jjshoeQwell ah
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18:32.36lmadsen~ciscolicense
18:33.07Qwelljbot: ciscolicense is unless you gave Cisco your first born, you probably aren't legally authorized to use their phones.
18:33.08jbotokay, Qwell
18:33.09coppiceQwell: don't cisco ensure that all their licencing is so arcane every user is in breach of it?
18:33.16Qwellcoppice: http://www.ntbox.com/cisco-openletter.html
18:33.20Qwellcoppice: I got official answers.
18:33.38Qwelljbot: ciscolicense is also see http://www.ntbox.com/cisco-openletter.html
18:33.39jbotQwell: okay
18:33.46lmadsenbeat me too it :)
18:33.58lmadsen[TK]D-Fender: ^^^ remember that for Cisco license questions :)
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18:34.08lmadsen~ciscolicense
18:34.09jbotciscolicense is probably unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html
18:34.18*** join/#asterisk hardhatpat (n=hardhatp@c-67-189-23-114.hsd1.or.comcast.net)
18:34.24QwellJeremy wanted to blog about it last week..  I said to hold off.  I emailed him to open the flood gates.
18:34.33lmadsengo JerJer!
18:34.44[TK]D-Fenderlmadsen: I would, but thats a pipe-dream appended to a cheap shot :)
18:35.00hardhatpati am trying to use the web-based voicemail client and i am getting this error: Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 152.
18:35.20Qwellhardhatpat: Do you have a (readable) /etc/asterisk/voicemail.conf?
18:35.27hardhatpatQwell: yes
18:35.37jjshoehardhatpat readable by your apache user?
18:35.39hardhatpati even tried 777'ing it to see if that would fix the problem
18:35.50hardhatpati owned it to the apache group
18:36.03jjshoehardhatpat put this vmail.cgi on pastebin.com or something
18:36.13jjshoeso we can see wtf line 152 is
18:36.25coppiceQwell: that someone needs to ask these questions would in a sane world make cisco's licencing invalid
18:36.37Qwellcoppice: *shrug*
18:37.07Qwellsadly, I've been trying to figure that out for like 3 years
18:37.12jjshoeQwell invalidated all opporutnities for any cisco usage for even asking questions probably :P
18:37.31hardhatpathttp://pastebin.ca/1216048
18:37.32QwellYou can see why it took them nearly 2 weeks to respond to me
18:37.50FruitBasket... no, I can't set a channel variable to store the account code. Chan vars don't appear to persist through a transfer.
18:37.53[TK]D-Fenderjjshoe: I'm sure it was duly filed in the "Ask me if I give a shit" drawer ;)
18:38.05coppiceQwell: I would think that in many countries they are claiming control which is illegal
18:38.16[TK]D-FenderQwell: And do we get to see the response?
18:38.19FruitBasketso I'm getting desparate. How can I make an account code persist through a transfer.
18:38.19jjshoehardhatpat change open(VMAIL, "<$filename") || die("Bleh, no $filename"); to open(VMAIL, "<$filename") || die("Bleh, no $filename: " $! . "|" . $@);
18:38.22Qwell~ciscolicensing
18:38.25jjshoehardhatpat and give us the error again
18:38.33Qwell~ciscolicense
18:38.33jbotsomebody said ciscolicense was unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html
18:38.38[TK]D-FenderQwell: And a crying shame you weren't able to get my advice prior...
18:38.38QwellI can't believe I already forgot the entry
18:38.50jjshoeQwell shows how much you care ;)
18:38.51Qwell[TK]D-Fender: bottom of the link
18:39.37hardhatpatjjshoe: syntax error at /var/www/cgi-bin/vmail.cgi line 152, near ""Bleh, no $filename: " $! "
18:39.37hardhatpatExecution of /var/www/cgi-bin/vmail.cgi aborted due to compilation errors.
18:39.57jjshoehardhatpat add, missed a .
18:40.09jjshoehardhatpat open(VMAIL, "<$filename") || die("Bleh, no $filename: " . $! . "|" . $@);
18:40.23hardhatpatjjshoe: yeah i just caught it
18:40.27coppiceeven with the current consumer hostile attitude of most governments, this kind of "we sell you nothing. everything is licenced at our whim" attitude is surely heading for trouble
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18:40.50hardhatpatit says permission denied ... but /etc/asterisk/voicemail.conf is 777
18:41.22[TK]D-FenderQwell: Asterisk : CAN HAZ FUR-WARE & SUPER-T?  Cisco : NO KTHXBIBI
18:41.22jjshoehardhatpat :)
18:41.33coppicewhy exactly would anyone *want* to use a cisco phone these days?
18:41.35hardhatpatjjshoe: what?
18:41.39jjshoehardhatpat I presume it's a parent directory permission issue
18:42.22jjshoereally stupid cgi, since it doesn't check for such a common issue and give hints on how to correct, or even look for where the issue is
18:42.46jaytee"Dear customer (he who we sodomize at our whim), we, the executives of Cisco (insert list 'o dickwads here) have determined that you have only the rights we deem you worthy of and only after excessive price gouging has commenced."
18:43.04jjshoejaytee and to that, we drink.
18:43.09hardhatpatdrwxrwx--T ... waht does that T mean?
18:44.32hardhatpatok im an idiot who cant type
18:44.38jjshoehardhatpat ?
18:45.01hardhatpathold on ... i changed the permissions of /etc/apache instead of /etc/asterisk
18:45.06jjshoeroffle
18:45.14jjshoe777? swift :P
18:45.19jjshoewhat's the url to your website? :)
18:45.31hardhatpatjjshoe: just chown, not chmod :)
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18:45.38Specialist1hi everyone
18:45.46jayteehi
18:45.48FruitBasketanyone... is there a way to persist channel variables or account code through a transfer? I don't really want to write the data to a database for each ID for each call, that seems slow and.. bad.. especially since they never go away.
18:46.32jayteeFruitBasket, read channelvariables.txt in the docs folder of your asterisk source
18:46.53hardhatpatnone of my messages are showing up, however
18:46.55Specialist1anyone using MERA here?
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18:48.30FruitBasketjaytee: thanks, will read.
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18:49.05jayteeFruitBasket, adding a single or double underscore to the beginning of the variable name allows inheritance to another channel
18:50.40FruitBasketoooohhhhh... does that apply to the account code, by chance?
18:51.01FruitBasketactually, if someone calls in and gets transferred somewhere else.. does the billing period end at the transfer, or does it continue?
18:51.14FruitBasketand if it continues... why the hell would the account code ever go away?!?
18:54.57citywokwhatever protocol i set sip to use (gsm, g726), it uses for 10 seconds in the call, and then falls back to ulaw.  is it dtmf or something causing it to fall back? why would it do this?
18:56.42CrazyTux[TK]D-Fender: quick question, say I'm using Asterisk Manager, and Action: Originate, to send calls, about 30/per second, I'll stop sending the asterisk manager calls, but it'll still have calls to send for a while, is there a way to view/clear this asterisk manager queue, and see which calls have been "ringing" / "called", and which calls are still "queued" ?
19:02.20smthSomething weird about inband dtmf. it only works with 'backgroud' . 'waitexten' or 'read ' did recognized the inband digits on the sip inconming call. I use 1.4.20.1 . and set ulaw codec in sip channel.  Can anyone help figure out?
19:02.20jayteeholy shit! netsplit!
19:02.20jaytee"Cap'n!!!! The engines!!!!!"
19:02.20smthsorry did not recognize
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19:02.38[TK]D-Fender*b00m*
19:03.34keith4wow... my other nick got split off. that hasn't happened before
19:04.15v4mpcan anyone see what i'
19:04.24keith4no
19:05.49v4mpve done wrong here its giving SIP Response message for INCOMING dialog BYE arrived on waitexten this is my conf
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19:05.53v4mphttp://pastebin.com/d60b95b34
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19:06.19[TK]D-FenderCrazyTux: Go look at your active channels
19:07.09smthany idea about inband digits working?
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19:09.54v4mp[TK]D-Fender, can u see whats wrong with my config?
19:10.06CrazyTux[TK]D-Fender: they are not all active
19:10.13CrazyTux[TK]D-Fender: is that what you wanted to point out?
19:10.41[TK]D-FenderCrazyTux: If you want to know if you are clear to feed more, then go look at how many are in progress
19:10.58[TK]D-Fenderv4mp: Who cares wha your config looks like when I can't see the CALL.
19:11.06*** part/#asterisk spaetzle (n=spaetzle@gw02.easterngraphics.com)
19:11.42Alan_HicksHowdy folks.  I'm troubleshooting a terribly odd echo problem that is only affecting a subset of my phones.  Basically, when a caller dials in, asterisk rings the receptionists.  When one of them answers, there is a terrible echo on the line until the caller answers.
19:11.44v4mp[TK]D-Fender, i did say further up that it gives this
19:11.44v4mpSIP Response message for INCOMING dialog BYE arrived on waitexten
19:12.01v4mpeverything above that works thats when it hangs up
19:12.06v4mpthat with debug on
19:12.20jjshoehardhatpat get your shit working?
19:12.27Alan_HicksThese users are on Polycom Soundpoint IP 650 phones.  If instead they use a Polycom 320 phone, there is no echo.  Is there some setting in the polycom configuration that might cause this?
19:12.31[TK]D-Fenderv4mp: And do i feel like guessing where I should be looking in the call?  Never waste time with anything less than complete CLI output.
19:12.37hardhatpatjjshoe: halfway
19:12.42Alan_HicksI've ruled out cabling as a culprit entirely.
19:12.42hardhatpatno errors ... but no voicemails show up
19:13.07[TK]D-FenderAlan_Hicks: Sidetone on the 650.  there are base gains you can adjust and IIRC the stock firmware values weren't great and caused this.
19:13.08jjshoehardhatpat hot
19:13.13v4mp[TK]D-Fender, i dont see how it would make a difference as i told u where it gets to on config :/
19:13.16v4mpbut ok 1 sec
19:13.19hardhatpatjjshoe: any ideas?
19:13.21*** join/#asterisk Ironhand (i=x@xyx.nl)
19:13.26jjshoehardhatpat link to the pastebin again?
19:13.36hardhatpathttp://pastebin.ca/1216048
19:13.39Alan_Hicks[TK]D-Fender: Thank you.  Do you have a link to a discussion of this issue or can tell me what to google/grep for please?
19:14.07[TK]D-FenderAlan_Hicks: sip.cfg
19:14.09Alan_HicksOr should I try to contact Polycom?
19:14.21[TK]D-FenderAlan_Hicks: Just fix your provisioning
19:14.47Alan_Hicks[TK]D-Fender: Thank you.  Do you know what values might need to be changed?
19:15.02Alan_HicksI'll grep the polycom manual in a moment for sidetone.
19:15.10jjshoehardhatpat did you set the context?
19:15.15hardhatpatjjshoe: i can log in with my mailbox and password, but NO vmails show up whatsoever
19:15.24hardhatpatjjshoe: what context?
19:15.31[TK]D-FenderAlan_Hicks: Go look a bit.  I'm sure you'll gind it.
19:15.38Alan_HicksThanks.
19:15.40jjshoehardhatpat perhaps you need to read the cgi?
19:16.02*** part/#asterisk Ironhand (i=x@xyx.nl)
19:16.27jjshoehardhatpat perhaps at least to line $24
19:16.32hardhatpatjjshoe: im reading it
19:16.35hardhatpatwhat do i set it as?
19:16.46jjshoehardhatpat what's the context for your voicemail?
19:16.50v4mp[TK]D-Fender, this is full debug of call http://pastebin.com/d3d0ab5f5
19:17.12jjshoehardhatpat ls /var/spool/asterisk/voicemail/
19:17.36hardhatpatoh
19:17.43hardhatpatdefault
19:18.00jjshoeso add that to the cgi
19:18.10[TK]D-Fenderv4mp: Go look at your dialplan.  The error is quite obvious.
19:18.29v4mpneeds a goto ?
19:18.44[TK]D-Fenderv4mp: No.
19:18.44hardhatpatdo i add just 'default'? or the entire path?
19:18.52smth[TK]D-Fender, something weird about inband dtmf. it only works with 'backgroud' . 'waitexten' or 'read ' did not recognize the inband digits on the sip incoming call. I use 1.4.20.1 . and set ulaw codec in sip channel.  any idea?
19:19.20v4mpthen i have no clue because the lines below it are ; out
19:19.28[TK]D-Fenderv4mp>ve done wrong here its giving SIP Response message for INCOMING dialog BYE arrived on waitexten this is my conf <--- Wrong
19:19.43v4mphmm
19:19.47[TK]D-Fenderv4mp: waitexten isn't even CALLED.  Pay attention to your OUTPUT
19:19.54jjshoehardhatpat you add the context.
19:20.11[TK]D-Fenderv4mp: So go stare at your dialplan untill you see the rather clear mistake.
19:20.53[TK]D-Fendersmth: Aside from "read" "waitexten", etc, where should * care about dtmf?
19:21.11[TK]D-Fendersmth: Do feel free to show me an actual problem.
19:21.30hardhatpatjjshoe: yeah i made it 'default'
19:21.52Kattyanyone work with bandwidth.com?
19:21.55v4mpwould i need AGI or is that just an extra ?
19:22.07[TK]D-Fenderv4mp: No.
19:22.15v4mpok
19:22.39[TK]D-Fenderv4mp: There is an extremely clear reason why Waitexten isn't being called.... keep staring at that line and the one that preceeds it.
19:23.00v4mpi noticed [TK]D-Fender which is why im on next part :)
19:25.35*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
19:26.28*** join/#asterisk l337ingDisorder (n=chatzill@66.183.159.54)
19:27.24bipolarDoes anyone know any way to put a value in the userfield column of the CDR table to track a call on a zap channel from begining to end? Is there another way to do that?
19:28.12l337ingDisorderhey folks, just installed Asterisk and gastman on a fresh Ubuntu 8.10 install. When I run gastman I can connect to localhost but it asks for a login and password... I wasn't prompted to enter an admin login/password to use during the install process and I can't seem to find the default asterisk login info anywhere.. can anyone tell me how to log in?
19:28.16codefreeze-lapbipolar: what do you mean, "track a call on a zap channel from begining to end" ?
19:28.35v4mpok does anyone see a problem with this because i dont but cli says its meant to have '='
19:29.18v4mpexten #,1,Playback(sai-thanks)
19:29.23bipolarcodefreeze-lap: if someone gets calls in, the CDR system tracks the call in multiple segments. If the call gets transfered I get a new record in the CDR but no way to tie that recored to the previous segment.
19:29.36bipolarat least I havn't found a way...
19:29.44codefreeze-lapl337ingDisorder: check you manager.conf file and set yourself up an account there.
19:29.51scooby2v4mp: your missing an important part
19:30.11scooby2exten   =>      invalid-format,1,Playback(prompts/dial-first)
19:31.06v4mpi dont get it o_O
19:32.57bipolarI need a way to track an incoming or outgoing call and see every part of the converstation, no matter how many times the call was transfered. doesn't seem to be a way with the existing cdr data. :\
19:33.03codefreeze-lapbipolar: Not yet. If you look at bmd's work on using CEL, you'll see that we keep a channel field for an id, that is viral when two channels interact. All the legs of call xfers can be tied together this way.
19:33.16scooby2bipolar: track using the zap channel
19:33.22smth[TK]D-Fender,  I have a 'background' called before waitexten in dialplan, the inband digit could be known during 'background' a sound file. but if sending inband digits within the period 'waitexten(5)' when the sound file finished playing ,they were not being recognized until time out.   if same case used on rfc2833 or info . both work fine.
19:33.31Kobazhow would i make a custom feature code in [applicationmap] that is active both for caller and calle on the same feature code
19:34.04bipolarscooby2: If I can insert the zap channel into the userfield of the cdr table that would work, but i don't know how to do that.
19:34.18bipolarat least it would be a start.
19:35.26[TK]D-Fendersmth: Show me a failed call with full debug...
19:35.48bipolarcodefreeze-lap: hmmm.....
19:36.31smth[TK]D-Fender, where I paste  them to show you the details
19:36.41[TK]D-Fender~pb
19:36.42jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:36.42bipolarcodefreeze-lap: So it looks like the dstchannel becomes the channel field of the recieving extention. is that correct?
19:37.03smthgot it
19:37.15[TK]D-Fenderv4mp: exten => 84455707,6,Background(sai-choose)  .....  exten #,1,Playback(sai-thanks)
19:37.22[TK]D-Fenderv4mp: Notice anything MISSING?
19:38.06v4mpi hate help sites that dont tell u much info at all about things :/
19:38.19*** join/#asterisk mindCrime (n=chatzill@216.27.62.2)
19:38.31[TK]D-Fenderv4mp: you meant he part where the "=>" after the word EXTENS is NECESSARY?
19:38.48[TK]D-Fenderv4mp: You must be completely blind to how you even formatted all your other lines.
19:38.50*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:38.54*** join/#asterisk theHub (n=theHub@69.177.93.21)
19:39.08v4mpgood point :|
19:39.27v4mpwas doing that at 5am this morning ¬_¬ lol
19:39.40[TK]D-Fenderv4mp: What's your excuse for right now?
19:40.15v4mpi haven't slept much for last week or sumthin
19:40.55[TK]D-Fendergoes off to do something productive
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19:46.15angryuseri have 3 outgoing peers from different providers, they serve for the same destinations, i would like to develep a script 1: Change them each time the destination is called(astdb&groupcount) i do know how to do that, and 2: Verify the registry status BEFORE call so if the provider 1 is dead, callout with another one + round up them, i hope it was clear ;)
19:47.29Kobazhmmmm
19:47.58Kobazi can't seem to get [applicationmap] feature codes to be available to the callee
19:49.47*** join/#asterisk gewuerzwiesel (n=gewuerzw@unaffiliated/gewuerzwiesel)
19:49.49*** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net)
19:49.51gewuerzwieselhello
19:50.54[TK]D-Fenderangryuser: Register status is just another value in AstDB.
19:50.57jeff_smokerDoes anyone know if a complete mid-call handoff is possible with SIP REFER?
19:52.16[TK]D-Fenderjeff_smoker: Sounds like a basic transfer to me...
19:52.21gewuerzwieselI just want to ask you some little questions to start :) what version should I use? install from ubuntu reps or better compile, and then 1.4 or 1.6?
19:52.41nny_2what does  chan_dahdi.c:1442 dahdi_enable_ec: Echo cancellation already on in console mean? If you had to guess at what the situation is that would cause that, what would you say?
19:52.49[TK]D-Fendergewuerzwiesel: 1.6 is not yet a full elease and should e notably less stable, documented, and able to be supported
19:52.53angryuser[TK]D-Fender : database show give me the register status of my local peers and agents not the outgoing ones ;( , maybe i need to enable some option ?
19:52.57[TK]D-Fendergewuerzwiesel: Your call.
19:53.00nny_2I mean it's obvious, but is this the software EC reporting that hardware EC is active?
19:53.07[TK]D-Fenderangryuser: Yes, it DOES.
19:53.48[TK]D-Fenderangryuser: Last i checked anyways
19:53.49gewuerzwiesel[TK]D-Fender: ok, so if I compile it by myself, what do I need to get my fritzcard pci working? do I need zaptel?
19:54.01*** join/#asterisk Assid (n=assid@unaffiliated/assid)
19:54.08jeff_smoker[TK]D-Fender: I mean to say being there is a difference between a conference call and call forwarding. When you forward a call using SIP refer and then terminate the call...will the referred parties remain connected?
19:54.10[TK]D-Fenderangryuser: For everything else there's SIPPEER
19:54.23[TK]D-Fendergewuerzwiesel: Can't answer BRI questiosn, sorry.
19:54.26voxterhas anyone put up what came of astridevcon?
19:54.35gewuerzwieseltkbeat: ok, no prob. thx
19:54.40[TK]D-Fenderjeff_smoker: thats the point
19:54.42angryuser[TK]D-Fender : http://www.pastebin.ca/1216119 .....
19:54.45implicitvoxter, we are too tiredd :)
19:55.02implicitvoxter, i'm sure some stuff will be going up soon. we had 4 groups
19:55.19[TK]D-Fenderangryuser: I see, please refer to the function I jsut mentioned
19:55.32implicitmedia negotiation, cli stuff, new architecture for dev & web stuff, and other stuff
19:56.09angryuser[TK]D-Fender : nice
19:56.15Kobazso, would anyone here happen to be a ninja jedi master of [applicationmap] in features.conf?
19:56.46jeff_smokerSo is a mid-call handoff possible with SIP, in the sense that PARTY1 forwards inbound call PARTY2 to an outside line PARTY3, then terminates the call and is no longer billed by the provider...but PARTY2 and PARTY3 are now independently connected?
19:56.55[TK]D-FenderKobaz: Funny... I don't see a pastebin from you anywhere...
19:57.04voxterimplicit: man i wish i could have stayed for it. Even though my main focus is not dev, i follow it all closely
19:57.21Kobaz[TK]D-Fender: well it's really just two lines
19:57.33[TK]D-FenderKobaz: No, it should be a lot more than that.
19:57.54*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
19:58.23lowtekHi all.  How does one set the default Polycom ring tone via the XML file?  I thought it was with one of the rt.x lines but apparently not.
19:58.33implicitjeff_smoker, it is possible but in asterisk it is a b2bua so it can't leave the signaling stream
19:58.44implicitjeff_smoker: but in sip, proxies can leave both the media and signaling stream
19:59.38Kattyso. if i wanted to Set(Email=foo), and then have h do h,1,Gotoif ( email isn't empty)
19:59.41Kattyhow would i word that?
20:00.04Kattythe gotoif part
20:00.47Kobaz[TK]D-Fender: okay well, it's more than two
20:00.49Kobaz[TK]D-Fender: http://pastebin.ca/1216128
20:00.54Kobaz:)
20:01.19[netman]$["${Email}" != ""] I think
20:01.59[TK]D-FenderKobaz: Go prove that devices DTMF even works.
20:02.13jeff_smokerimplicit: So if I [1] receive an inbound SIP stream on a softphone and then [2] I turn around and refer that stream back to my provider and then [3] the inbound call I received is now connected to the outbound line that I just connected it to and [4] I terminate the call, what would I need to do to make sure the connected parties remain connected ?
20:02.16Katty[netman]: okay, now how do i tell it where to go :/
20:02.36[TK]D-FenderKobaz: and I don't see the call being ANSWERED, nor do I see any debug level that indicated DTMF being detected
20:02.42lowtekHey [TK]D-Fender: Do you know which XML setting sets the default ring-type for Polycom phones?  I tried the <DEFAULT rt.1 ...> but that's apparently only used for alertInfo ...
20:02.54KattyGotoIf($["${Email}" != ""],s,1)?
20:02.54implicitjeff_smoker: REFER is more difficult
20:03.02implicitcause it has to be processed by a UA
20:03.02[TK]D-FenderlowGo override it, and look at the overrides.
20:03.05implicitlike an endpoint
20:03.07implicitor a B2BUA
20:03.09lowtekThanks!
20:03.10seanbrightKatty: core show application GotoIf
20:03.25jeff_smokerimplicit: what method would you use then, other than refer?
20:03.27implicitso it is still two dialogs
20:03.29implicitin SIP
20:03.33impliciteven if the audio is connected
20:04.01implicitjeff_smoker: no method other than refer can be used after 200OK is received for initial INVITE
20:04.13Kattyseanbright: i already did that. it was not helpful.
20:04.16implicityou can only send 3xx replies before dialog is established
20:04.22Kattyseanbright: the wiki page was more helpful, but doesn't show an extension jump anywhere
20:04.24seanbrightKatty: how was it not helpful?
20:04.36seanbrightKatty: it's helpful if you read it.
20:04.42seanbrightKatty: consume words and such.
20:04.48implicityou can do after 180/183, but not after 200
20:04.51Kattyseanbright: i got 90% of what i needed, yeah
20:04.57implicitafter 200 REFER needs to go to the endpoint
20:05.00implicitasterisk is an endpoint though
20:05.02implicitsince it's a b2bua
20:05.08seanbrightKatty: GotoIf(condition?[labeliftrue]:[labeliffalse])
20:05.17angryuser[TK]D-Fender : i see if i do SIPPEER(peer|status) where  '- status       Status (if qualify=yes)' but it's the qualify status ? not the registration status? (i got qualify set to no)
20:05.46jeff_smokerimplicit: Would you argue that you can change endpoints or that you cannot change endpoints?
20:05.49[TK]D-Fenderangryuser: look at "sip show peer [peer]" and see if you're inspired to look at something else.
20:05.58implicitok
20:06.03implicitlets use more accurate terminology
20:06.03angryuseror maybe i need to check EXISTS(SIPPEER(peer|regexten) ?
20:06.05[TK]D-Fenderjeff_smoker: "core show application transfer"
20:06.07seanbrightKatty: if e-mail is blank, where do you want to go?
20:06.11*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-0f27561830b960e8)
20:06.11*** mode/#asterisk [+o Deeewayne] by ChanServ
20:06.18[TK]D-Fenderangryuser: Nope.
20:06.19implicitjeff_smoker UAC and UAS cannot be changed after dialog is established (200OK on initial INVITE)
20:06.28implicitunless REFER is sent
20:06.30implicitby one of them
20:06.32implicitto the other
20:06.36[TK]D-Fenderangryuser: that implies your provider is registering to YOU
20:06.40implicitand it ends up being a new dialog
20:06.48Kattyseanbright: GotoIf($["${Email}" != ""]?foo,s,1) i think
20:06.53implicitbefore 200OK
20:07.00implicityou can have parallel/serial forking
20:07.03implicitand tons of stuff w/ early media
20:07.04Kattyseanbright: if false, do nothing
20:07.07implicitor provisional replies
20:07.11seanbrightKatty: priority 1 of extension s in the foo context?
20:07.17implicitand be dealing with many UAs
20:07.18Kattyseanbright: yes.
20:07.22seanbrightKatty: looks right to me
20:07.28implicitjeff_smoker: i gotta head out but send me a message
20:07.37jeff_smokerimplicit: ok thanks
20:07.39Kattyseanbright: i just gotta see things.
20:07.52Kattyseanbright: the examples are good and all, but it doesn't always sink it like it hsould )=
20:08.21seanbrightKatty: i forgive you
20:09.12angryuser[TK]D-Fender : i can set the qualify and check it, but is there any other way ?
20:09.28[TK]D-Fenderjeff_smoker: the first refer goes from your endpoint to *.  from there you call TRANSER.
20:09.49[TK]D-Fenderangryuser: I've already said what else to look at
20:10.07*** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw)
20:10.28angryusercal me blind  Subscriptions: Yes ?
20:10.42Kattyseanbright: katty learn by do, not read
20:10.58seanbrightKatty: well do more doing and less asking :)
20:10.58[TK]D-FenderangLook at the entire dump.  I'm sure something sueful will stand out.  Just go try for a while.
20:11.04Kattyseanbright: psh.
20:11.20seanbrighti know... it's crazy talk.
20:11.22[TK]D-Fenderseanbright: Yeah... that'll work...
20:11.25*** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum)
20:11.30Kattyhugs [TK]D-Fender
20:11.36Katty[TK]D-Fender: i know you love me.
20:12.29angryuserexpire .........
20:12.34seanbrightKatty gets special treatment cuz she's a gurl.
20:12.37jeff_smoker[TK]D-Fender: I'm not sure if I'm being clear about what I aim to do. I'd like to receive an INBOUND VOIP call and then use a SIP method to send that call out to an EXTERNAL PARTY, and then TERMINATE the connection that I have with my VOIP PROVIDER. I think I understand what you're saying is that asterisk can transfer or forward the call, but asterisk in this case will continue to host the...
20:12.39jeff_smoker...call. I do not want to continue to host the call.
20:12.40seanbrights'not fair
20:12.52Qwellsnot fair?
20:12.57Kattyare you sure it's not because i ask politely?
20:13.12*** join/#asterisk boch (n=fran@customer191-9.iplannetworks.net)
20:13.15KattyQwell: what do you think?
20:13.16seanbrightKatty: most definitely
20:13.18[TK]D-Fenderjeff_smoker: I understand perfectly and have handed you the answer.
20:13.23seanbrightKatty: it's cuz you're a chick
20:13.25QwellKatty: I wasn't paying attention
20:13.34KattyQwell: you have a habbit of that.
20:13.40Qwellof what?
20:13.43seanbrightohhh
20:13.44seanbrighttouche!
20:13.46Qwellsorry, dozed off a second there
20:13.50jeff_smoker[TK]D-Fender: Excellent. i love answers. i will review it exhaustively. Thank you.
20:14.12KattyQwell: seanbright here thinks that people help me and tolerate my questions because i'm a female.
20:14.12bochcould you recommendme a good iax client for linux/gtk ?
20:14.16KattyQwell: what's your viewpoint on this?
20:14.25QwellKatty: seanbright is a wise person
20:14.30Alan_Hicks[TK]D-Fender: Ok.  I'm stuck.  I'm not sure that I'm even modifying the right setting.  I've raised voice.handset.s
20:14.34Alan_Hicksidetone.adjust.IP_650
20:14.35Qwellbut, no, you've been around forever :P
20:14.44seanbrightKatty: tolerated is a strong word
20:14.45Alan_HicksOops...
20:15.03Kattyforever is a strong word, if you ask me.
20:15.06seanbrightKatty: anyone else asking how GotoIf worked would get the following response from you-know-who...
20:15.08Kattylet's go with Years.
20:15.11seanbrightKatty: go READ the DOCUMENTATION.
20:15.32Alan_Hicks[TK]D-Fender: Ok.  I'm stuck.  I'm not sure that I'm even modifying the right setting.  I've raised voice.handset.sidetone.adjust.IP_650 from "-3" to as high as "9" with no perceivable change.  Should I be looking elsewhere to solve this echo problem?
20:15.33KattyQwell: i think seanbright might have a point :<
20:15.35seanbrightam i right or am i right?  or am i right?
20:15.47*** join/#asterisk shaw22dog (n=shaw@pacman.oaklandcorp.com)
20:15.48seanbrightKatty <3
20:15.59Kattymy nagging persistance pays off
20:16.04shaw22dogHello.
20:16.07gewuerzwieseldo I need libpri and zaptel? there are tutorials compiling just asterisk and asterisk-sounds?
20:16.07[TK]D-FenderAlan_Hicks: base gains as well.
20:16.11Kattyanswer me, or i shall bug you every 5 minutes for the next YEAR!
20:16.41Kattyseanbright: it also has a downside.
20:16.49Alan_Hicks[TK]D-Fender: voice.gain.tx.analog.chassis.IP_650  correct?
20:16.52*** join/#asterisk mltlnx (n=mltlnx@32.136.47.207)
20:16.54Kattyseanbright: when you mentioned i was a female, i instantly got 5 /queries from random people :/
20:17.13[TK]D-FenderASL?
20:17.15gewuerzwieselKatty: lol :)
20:17.20Katty[TK]D-Fender: close.
20:17.23shaw22dogI'm running Asterisk 1.2, with a Sangoma card. When I park calls, and then pick them back up, I get an echo on my side for approx. 20. Although there was no echo on the initial connection, any ideas on what is causing this?
20:17.30seanbrightKatty: my bad.
20:17.37Kattyseanbright: FOR SHAME!
20:17.39shaw22dog*approx 20 sec.
20:17.51Alan_HicksASL?  I'm afraid I don't know what that means.
20:17.54seanbrightKatty: /nick BigManTypePerson
20:17.57Kattyseanbright: you may be right, i don't know. never thought about it before.
20:18.04*** part/#asterisk nny_2 (n=scott@64.203.244.146)
20:18.12seanbrightKatty: i'm just giving you a hard time
20:18.14Alan_HicksOh wait.... Age, Sex, location.  I'm a dumb-ass.
20:18.22Kattyseanbright: i've just always tried to ask my questions politely and wait patiently for an answer. and let the person know i appreicate their help.
20:18.30Kattyseanbright: i think it's worked okay for the last 5 years or so (=
20:18.48Kattyseanbright: SO BUZZ OFF ;>
20:18.48seanbright5 years?
20:18.54seanbrightand you don't know how GotoIf works?
20:18.57seanbrightFOR SHAME!
20:19.03Kattyof course i know how it works, kinda.
20:19.15Kattyjust nothing overly complicated ;)
20:19.18[TK]D-Fenderseanbright: the term is "reciprocal lamprey"
20:19.25*** join/#asterisk gabegundy (n=gabe@nat.parentlink.net)
20:19.27Corydon76-digAlan_Hicks: and the answer to "ASL" for you was?
20:19.34Kattyit'd be different if i used all these fancy features!
20:19.54[TK]D-FenderKatty: Next challenge... app_dial!  World beware!
20:20.05Katty[TK]D-Fender: hush dear ;)
20:20.12seanbrightinteresting...
20:20.37*** join/#asterisk sheri_rao (n=rao@203.99.179.244)
20:20.43Kattyseanbright: most of the stuff i set up is pretty simple. (=
20:20.50[TK]D-Fenderseanbright: It's all about obfuscative selection ;)
20:21.17[TK]D-FenderKatty needs no IFs, ANDs, or BUTs !
20:21.24seanbrightKatty: well from now on when someone has a GotoIf question, i'll expect you to be the goto person (ba dum dum)
20:21.37Kattyseanbright: oh look at you mister funnypants.
20:21.38Alan_HicksCorydon76-dig: 0x1C, XY, Lizella.
20:22.14Corydon76-digAlan_Hicks: hawt, though still older than my BF
20:22.36Alan_HicksCorydon76-dig: 1C is much younger that BF.
20:22.42seanbright11100?
20:22.45seanbrightthat's old
20:22.47Corydon76-digAlan_Hicks: the BF is 0x18
20:23.17seanbrighti am 11101 though
20:23.21seanbrightso that makes me ancient.
20:23.33Corydon76-digseanbright: mega
20:23.46Corydon76-dighands seanbright his cane
20:24.03seanbrightewww... that is *not* a cane!
20:24.03seanbrightheh
20:24.12thehareh?
20:24.22seanbrightwha?
20:24.35Corydon76-digWith your eyesight, I'm surprised you noticed  :-P
20:24.46seanbrightwell played.
20:25.20angryuser[TK]D-Fender : well i have compared the dump from registered and not registered (same) outgoing peer, i can use only status with SIPPEER, i dont see more, see by yourself http://www.pastebin.ca/1216151
20:27.22[TK]D-Fenderangryuser: I suppose you could also script something to look at "sip show registry" easily enough.  Messy but more than doable
20:27.54[TK]D-Fenderok, heading home.  Later all
20:30.30*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
20:30.30*** mode/#asterisk [+o lmadsen] by ChanServ
20:33.00angryuserwell i will enable qualify for 2 days, wait&see what happens, if my provider dont cut the qualify packets to reduce traffic i will use that, lazy solution but easyer of coding all stuff with bash
20:37.57codefreeze-laptomas-- you around?
20:38.41Kattyseanbright: http://pastebin.ca/1216172 (=
20:39.00Kattyseanbright: i will now have more spam than i can realistically handle!
20:40.30Qwellspams Katty
20:41.48*** part/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com)
20:44.01KattyQwell: it all start with a company who had analog lines, who needed to know the person who was transfered to their cellphones.
20:44.13KattyQwell: and then i decided to put an attachment on the email
20:44.41KattyQwell: and then i got the bright idea of always recording certain extensions in our call center, to a quality control email address...that could have a few audio files listened to, and the rest dumped.
20:46.20*** join/#asterisk bijit (n=benji@190.241.15.48)
20:47.56gewuerzwieseldo I need zaptel or is it only for digium hardware?
20:49.23*** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com)
20:49.27Kobazawww
20:49.27seanbrightKatty: pretty sure you can leave out the ,s,1 part for both of those
20:49.34Kobazthe fender went byebye
20:49.34seanbrightKatty: but not 100%
20:49.50Kattyseanbright: it works. i'm happy with the working.
20:50.03seanbrightKatty: working schmirking.
20:50.16Kobazokay so
20:50.21Kobazmaybe someone else can shed some light
20:50.31Kobazhttp://pastebin.ca/1216128
20:50.40*** part/#asterisk Talirk (i=434e2716@gateway/web/ajax/mibbit.com/x-241dfef0ddd48b2f)
20:50.41Kobazhaving an issue with features.conf and custom feature codes
20:51.48smthseanbright,http://pastebin.com/m79b7d310  I use this test inband dtmf . it does not work on the incoming call. (ulaw/1.4.20.1) any idea?
20:52.00seanbrightsmth: no
20:52.30seanbrightsmth: i don't really help here.  i just put people down, usually.
20:52.30v4mphow do you create the queues ?
20:52.54Kobazseanbright: damn right
20:52.57seanbrightv4mp: queues.conf
20:53.00smthwho takes care dtmf stuff . any help? ;)
20:53.31angryuserKobaz : add W option to your dial, see what happens
20:53.41gewuerzwieseldo I need zaptel or is it only for digium hardware? :)
20:54.04angryuserKobaz : ah no
20:55.13bijitI am having problems registering my sip trunk.
20:55.35*** join/#asterisk tkbeat (n=tk@p54B9474A.dip.t-dialin.net)
20:55.35bijitI tried alot of stuff and I can't get it to work.
20:55.45*** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net)
20:55.54angryuserKobaz : what kind of client is 2605 and 2608 ? same hardware ?
20:55.58bijitanyone have any idea what could it be?
20:56.07*** join/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com)
20:56.18Maliutabijit: firewall, your settings, the visp's settings ....
20:56.23angryuserbijit : pastebin cli messages
20:56.34*** join/#asterisk s0lid (n=s0lid@122.53.107.186)
20:56.35Maliutabijit: the list is longer than my schwong
20:56.44Maliutaand that long
20:57.22hardwireanything better than a2b in the free software world?
20:57.54seanbrighthardwire: apache?  open office?
20:58.02hardwireasterisk2billing
20:58.05seanbrightright
20:58.11hardwireright
20:58.12seanbrightapache and open office are both better
20:58.19hardwireI refuse
20:58.25*** part/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com)
20:58.30MaliutaOpenBSD has to make that list
20:58.36angryuserno spectrum zx webserver is the best ;)
20:58.37hardwiredon't forget teh hurd
20:58.44seanbrightGNU/Hurd, thank you
20:58.58seanbrightstallman would kill you for not using "GNU" in front of it
20:59.06Maliutagcc?
20:59.13hardwirestallmans sword is weak
20:59.13seanbrightindeed
20:59.20seanbrighti think the point is
20:59.23Maliutawhere would we be without the g
20:59.36seanbrightwhen it comes to free software, many things are better than a2b
20:59.38hardwireMaliuta: in a world with less grubby g keys
20:59.44seanbrightglad we got that cleared up.
20:59.58Maliutahardwire: you could live in such a world?
21:00.22*** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com)
21:00.25Maliutaaccuses hardwire of being typograph-ist
21:00.53QwellMaliuta: he already lives in Alaska!  can't get much worse.
21:01.17hardwireQwell: you said you wouldn't tell
21:01.19Maliutaalsaka eh?
21:01.20bijithere is pastebin http://pastebin.ca/1216195
21:01.24Qwellhardwire: Where's my million dollars?
21:01.32hardwireQwell: where's my fuzzy peach?
21:01.48Qwellwe threw the core away
21:01.51Qwellpeaches have cores, right?
21:01.52hardwireaww
21:02.00hardwirethey have pits
21:02.00seanbrightQwell: pits
21:02.04Qwellthat then
21:02.09QwellDeeewayne ate it :(
21:02.15hardwireDeeewayne: !
21:02.19MaliutaI have a million monkeys at a million typewriters ... one bunch has some shakespeare thing, another has a functioning kernel
21:02.38Qwella million typewriters per monkey, or a million typewriters total?
21:02.45*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:02.47hardwirehow many fingers does each monkey have?
21:02.53hardwiredo toes count?
21:02.55Qwellhardwire: I'm guessing 10 million
21:03.14Kobazmr fender!
21:03.21Kobaz:)
21:03.32bijit[Oct  1 15:02:46] WARNING[18589]: chan_sip.c:1950 retrans_pkt: Cancelling retransmit of OPTIONs (call id 03a7769506945f6359343dd760ad358f@190.241.15.48)
21:03.35bijitis that bad?
21:03.37Alan_Hicks[TK]D-Fender: Welcome back.
21:03.39Deeewaynehardwire: I'm de-fuzzing the peach
21:03.45MaliutaQwell: well the number of typewriters is constant, the per monkey thing changes as they split into groups to do "writers stuff" or jerk off to pr0n
21:03.56bijitsup Alan_Hicks
21:04.05hardwireDeeewayne: like how you decorn corn?
21:04.07Alan_Hicks[TK]D-Fender: When you told me I must change the default gains as well, did you mean this?  voice.gain.tx.analog.chassis.IP_650
21:04.11Kobaz[TK]D-Fender: so anyways... dtmf works on both phones.... if i dial from 2605->2608, 2605 can use the feature codes... if i dial from 2608->2605, 2608 can use the feature codes
21:04.25Kobaz[TK]D-Fender: i'll make you some cookies
21:04.26Alan_Hicksbijit: Troubleshooting a damn irritating echo problem on a subset of my phones.
21:04.32Maliutabijit: there is a firewall between you and the sip server?
21:05.26bijitMaliuta: nope. It was registering very well till yesterday all of a sudden it stopped. If I connect a ATA works fine.
21:05.45bijitAlan_Hicks: Me struggling with sip trunk register. :-)
21:06.03*** join/#asterisk franck (n=franck@tikiwiki/franck)
21:06.09*** join/#asterisk pluggo (n=bburns@mookie.synset.net)
21:06.57v4mpwhich option should i be using to name queues as i couldn't see that part in queues.conf
21:06.58franckHi all
21:07.02pluggoalright so... i am trying to set up an asterisk server with a te122p and dahdi... everything seems to be right, the module is loaded and everything, but i don't see the channel show up when i type dahdi show channels
21:07.13franckI'm looking for info on how to interface asterisk with cisco pbx
21:07.51Kobazcrisco
21:07.59pluggoi see the pseudo channel but no numbered ones
21:08.06pluggoand it shows up ok in dahdi_scan
21:08.42*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
21:09.13pluggoany ideas anyone?
21:09.27pluggo(apparently dadhi = the new name for zaptel)
21:10.09*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
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21:19.34*** part/#asterisk FengShui (n=cabeen@milli.chem.ucsb.edu)
21:23.14pluggo*poke poke*
21:23.39Cutlassdahdi = zaptel as of 1.6 ....legal issues
21:23.57pluggoyeah... i figured that out
21:24.16pluggoi actually got it where its showing up as a channel in dahdi show channels
21:24.24Maliuta"legal issues" very much depends on which part of the world you live in :)
21:24.47Cutlasstrue...
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21:24.51pluggonow, if i select pri_net signalling it says it thinks it's the network and we think we're the network... ditto for cpe
21:27.25pluggoaccording to our globalcom, we need a straight through cable to connect the te122p to the adtran... they didnt seem very sure tho
21:27.56pluggois it possible this thing is looped up in software somehow?
21:29.06pluggobtw... dahdi is also zaptel as of 1.4.22
21:29.32Cutlassok
21:30.15*** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net)
21:30.47pluggoalright
21:30.52pluggoy'all have been right helpful
21:32.17Cutlassquestion:  I installed 1.6 with prefix=/usr/local/src ...now I can't seem to find the config files.  There are some under the configs directory in the root of where I built asterisk, but I don't find any in the /usr/local/etc/asterisk directory (isn't that where it should be??)....The ones that I did find don't seem to affect the operation of asterisk
21:32.43Cutlass...sorry pluggo, I don't know the answer to your quesiton...I'm waiting for a response so I can learn too :)
21:33.26Kobaz[TK]D-Fender: yaaaaaaaaaaay
21:33.41Kobazangryuser: and a yaaaaaaay
21:34.00Kobazangryuser: wW doesn't work if i dial from an iax extension to an iax trunk
21:34.10angryuserCutlass : and if you start asterisk -vvvvvv does it say where he cant find conf files ?
21:34.29Kobazangryuser: i used M(macro) in my Dial() to turn on the DYNAMIC_FEATURES for the called channel as well
21:34.42angryuserKobaz : answer my last question please
21:34.58Kobazangryuser: 2605 is iax, 2608 is sip
21:35.19Kobazangryuser: i'll show you the stuff that works
21:35.34angryuserKobaz : and you have dtmfmode set ?
21:35.44Kobazyeah it's nothing to do with dtmf
21:35.57Kobazit's based on DYNAMIC_FEATURES not being set for the called channel
21:36.10Kobazexten => _91NXXXXXXXXX,n,Dial(SIP/2608,600,rM(foo))
21:36.23Kobaz[macro-foo]
21:36.24Kobazexten => s,1,Set(DYNAMIC_FEATURES=toggleRecorda#toggleRecordb)
21:36.29Kobazwell
21:36.39Kobazthat works for setting the features for the called phone
21:36.49Kobazput it before the Dial, and then you get it on both ends
21:36.56Cutlassit says that modules.conf and features.conf could not be found...does not specify where it is looking
21:37.48angryuserCutlass : it's a new install ?
21:37.54Cutlassyes
21:38.12angryuserCutlass : so do make samples and find them
21:38.23CutlassI didn't do make samples
21:38.29Cutlassjust make, make install
21:38.55Cutlassis that the suggested method?....make samples?
21:39.46angryuserCutlass : it will generate you the sample config files, you can wipe them if you want after
21:39.54Cutlassactually...I think I may have done that...I have all the sample files in the configs directory
21:40.16[TK]D-FenderCutlass: If you are working off a 100% fresh install then do "make samples"
21:40.26Cutlassok
21:40.42Cutlassand then rename all the files to remove the ".samples" extension?
21:41.07angryuserCutlass : so "find / | grep extensions.conf" and here you are
21:41.17*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
21:41.17[TK]D-FenderCutlass: it will do everything
21:42.47Kobaz[TK]D-Fender: i figured it out.... the Dial(M(macro)) feature is amazing
21:43.07Kobazthis will solve like,   a bazillion problems
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21:47.10Cutlassthanks guys...I think it may have worked...I now see config files under /usr/local/etc/asterisk
21:48.03v4mphow would i setup a queue with a name as i didn't see how to in queues.conf
21:48.33hardwire[queueoftheawesome]
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21:48.47hardwirequeue(queueoftheawesome,...)
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21:54.57v4mpthat go in queues.conf yea ?
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22:04.39AndyMLasterisk 1.4.19.1 - how/where would I enable callwaiting?
22:05.16AndyMLactually - I think this is a freepbx question - i bet its this dialparties.agi causing the problem. nevermind!
22:05.35*** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net)
22:06.31MaliutaAndyMillar: look at call parking
22:11.54jmaczHi everyone, I'm trying 1.6 with TLS and I'm getting a "SSL cert error" when starting *, but no aditional information (asterisk logs shows nothing). What may be causing this?
22:19.48angryusercan someone help me with GROUP_MATCH_COUNT i need to count the total amount of channels of groups (group1,group2,group3) i dont see how to use the pattern, syntax is here http://www.pastebin.ca/1216252 , maybe something like this Gotoif($[${GROUP_MATCH_COUNT(group?)} >= 5]?true:false)
22:21.00*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
22:21.27nicoxi'm sorrx what do you mean with tls, iax/tls?
22:26.19angryuserGotoif($[${GROUP_MATCH_COUNT(^group[1-3])} >= 5]?true:false) that must be it.. testing ;)
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22:29.13*** join/#asterisk sircco (n=sircco@dh207-102-138.xnet.hr)
22:29.32sirccowhat's the difference in dialing queue with Dial() and with Queue() ?
22:30.12[TK]D-Fendersircco: "Dial" is not "Queue"
22:30.20sirccoi know..
22:30.41*** join/#asterisk pcrane (n=pcrane@202.49.106.158)
22:30.46sirccobut lets say i have queue 1000 I can go to it with Dial(1000@context) or with Queue(1000)
22:31.01[TK]D-Fendersircco: Dial just calls the devices you list for a single timeout.  Queue will dial multiple devices with weighted priorities over MULTIPLE attempts,e tcx
22:31.28[TK]D-Fendersircco: Dial(1000@context)  <- invalid syntax
22:31.34sirccoso going to queue with Dial() is a bad idea?
22:31.59[TK]D-Fendersircco: So far your "plan B" is invalid from the start.
22:32.17sirccowell that's why i came here to ask :)
22:32.53[TK]D-Fendersircco: Dial(1000@context)  <- simply not valid.  What more is there to say?  You want the call to fall into a Queue then go ahead an call Queue.
22:33.27sirccohere is exact one Dial(Local/${NLABQUEUE}@from-internal)
22:33.47[TK]D-Fendersircco: No point to doing a dial there that I can see.
22:34.17[TK]D-Fendersircco: Nothing you can't do just by doing a straight Goto, or jsut calling the same apps as that exten would lead to.
22:34.51sirccothis is some trixbox box im trying to extend with queues from mysql
22:34.57[TK]D-Fendersircco: And of course this new sample doesn't prove to me what it will execute even upon attempt.
22:35.16[TK]D-Fendersircco: Well Trixbox & FreePBX are not supported here.  They have their own channels
22:35.51sirccoyes i know but im making my own dialplan, nothing to do with clicking and web :)
22:49.46citywok[TK]D-Fender, i learned today that having 4 codecs in allow statements in sip.conf means that the other end just might pick any of those codecs it likes, not in preference order.
22:49.58citywokcontrary to all of the documentation online
22:50.52[TK]D-Fendercitywok: Its a negotiation of preference.
22:51.07[TK]D-Fendercitywok: Doesn't mean * will win the fight in its order
22:51.25citywokhah, yea, thats what i learned from the tech at voicepulse.  well, actually, we learned this together.
22:52.08citywokgsm is a very, very cpu light codec. 15 concurrent calls and a p3 1.13 isn't even breaking a sweat
22:53.21[TK]D-Fendercitywok: Unless * is transcoding there is nothing to load your system
22:53.36[TK]D-Fendercitywok: And you should be picking ONE codec per peer
22:53.50citywokit's doing the encoding to gsm, from a zap channel (ulaw right?)
22:54.25citywokand yea, that was there fuckup for putting all supported codecs in their pre-configged sip.conf file, and my fucup for using it.
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23:12.07briana bit offtopic but does anyone know how those free teleconferencing  /  telephone partyline services make money?
23:12.36citywokthey generally provide free local numbers, but charge you if you want to provide a TFN
23:13.14brianThat's not what I meant
23:14.18citywoksorry, i wasnt clear.  my thought on the matter is they dont make money at all, except for when people pay per minute to be able to provide a TFN to their clients/customers/etc. -- they then make money by charging 8cents/minute or something retarded for that
23:14.56brianHow do people get numbers in Minnesota or Iowa and get kickbacks?
23:15.14citywoki'm not sure what you mean
23:15.49brianLike, whenever a long distance call is made to the number, the phone company gets paid for it, and they give you a fraction of what they get.
23:16.55_ShrikEbrian: I think you are talking about cabs
23:17.00briancabs?
23:17.22brianI don't know what the proper word for it is
23:19.16brianI think you're messing with me though...I'm being serious...
23:21.09angryuserwhy when i do Noop(The quantity of calls is "${GROUP_MATCH_COUNT(group[1-3])}") i got  "The quantity of calls is "0"") in new stack     why "0"" with 2 "" at the end ?
23:22.21angryuseroh found it nevermind
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23:37.20v4mphow would i setup a queue with a name as i didn't see how to in queues.conf
23:37.43angryusermy script is over, accepting any critics ;) http://www.pastebin.ca/1216317 ;)
23:37.55angryuserv4mp : what ?
23:38.12v4mpto load in new voicemail settings from voicemail.conf do i need to restart *
23:38.15hardwireangryuser: hah hah.. line 20 amuses me...
23:38.18hardwireand 21
23:38.21hardwireand 32
23:38.22v4mpangryuser, want to setup a queue for callers
23:38.36hardwireactually. I didn't look at it
23:38.37Carlos_PHXv4mp: reload app_voicemail
23:39.01v4mpty Carlos_PHX
23:39.39v4mpsay a queue for 'sales' queue for 'support' etc
23:39.42Carlos_PHXv4mp: forgot .so at the end.  I just press tab after app_v
23:39.46angryuserv4mp : so create a queue with a name you want [thenameofqueue)
23:40.07*** join/#asterisk subdolus (n=subdolus@subby.afraid.org)
23:40.17angryuserv4mp : no just do reload in CLI
23:40.26theharor reload app_voicemail
23:40.34*** join/#asterisk RB2 (n=RB2@pool-71-125-95-246.nwrknj.east.verizon.net)
23:40.48*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:40.51*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
23:44.29v4mpangryuser, what conf does that go in and how do i lay it out or know of somewhere to help ?
23:46.12angryuserv4mp : hm, i am no sure if i understood the question, so you want to setu a queue right ?
23:46.33angryuserv4mp : how would you like to name it ?
23:47.02v4mpyes want to setup queues for certain extens when theres noone available to take the call
23:47.06v4mpyes would like to name it
23:50.48*** join/#asterisk ManxPower (n=manxpowe@c-76-105-105-153.hsd1.ga.comcast.net)
23:55.08v4mpif i want to allow wav should it be allow=wav ?
23:55.39[TK]D-Fenderv4mp: wav is not a call codec
23:55.40angryuserv4mp : wait a sec i need to do something
23:55.59v4mpok
23:56.34v4mp[TK]D-Fender, wasn't sure because it uses allow=gsm and uses gsm for some of the recordings so weren't sure :)
23:56.53seanbrightthere are gsm encoded wavs
23:57.22ManxPowerv4mp: The codecs installed on your system are the ones listed in "core show translations"  (I think that's the command.
23:58.48v4mpnope
23:58.59v4mpwithout the s works
23:59.08ManxPowerthat would be it then.
23:59.35ManxPowerdo a help on the command.  adding the documented recalc option to that command can make the numbers more accurate.

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