IRC log for #asterisk on 20070902

00:01.52zapp-braniganhi the trunk=yes is only por the g729 ?
00:02.33famiconDaviey ehehehe
00:02.34famiconsure
00:04.21Qwellfamicon: When you get them, send them here so I can send out a bill ;)
00:05.16famiconDaviey yeah just gimme your details plz
00:06.03Davieyfamicon: have you given up with ztdummy then?
00:07.05*** join/#asterisk Strom_C (n=strom@netblock-208-127-172-112.dslextreme.com)
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00:16.09Davieyfamicon: ?
00:16.18famiconhush
00:16.22famiconim working
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02:18.12TSCHAKdoes anyone know of a good speakerphone mic that attaches to a soundcard ?
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02:29.44l3jjI need a good provider that will let me call us48
02:29.58l3jjfor a "flat" fee, using asterisk, one channel only
02:30.53coppicelots of people will do it for a fat fee.... oh, flat. can't help. sorry
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02:40.01grandpapaHi all.  I think network solutions is having an outage or something.  Can any of you resolve domains registered there (that aren't cached)?
02:40.17grandpapaI can't even browse to www.networksolutions.com
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02:41.11coppiceI can't browse www.networksolutions.com either. I'm not sure I would consider that a problem though. :-)
02:41.48grandpapaWow.  That sucks bigtime.
02:42.09heelioscoppice: haha. i though the very same while between sentences in your message. :P
02:42.09coppicewho network solutions? yep. they do
02:42.25grandpapaYea, beyond that, theyir dns servers have been at least reliable.
02:44.07coppicewell, they call themselves network *solutions*. maybe they finally dissolved
02:45.13justdaveLevel3 and gblx appear to be having issues on their backbones at the moment
02:45.21justdaveaccording to internetpulse.net
02:45.36heelioswell thats not news. level3 is always having issues with their backbone. <_<
02:47.13*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
02:48.38justdaveMy phone system at home (with 3 extensions on it) is running a Centos4-based Asterisk@Home (with Asterisk 1.2.x on it).  It's beyond time to upgrade it...  Since my asterisk setup isn't very complicated and would be easy to reproduce again, I'm thinking to just wipe it out and start over.
02:49.04justdaveI've heard a lot of mixed comments in here about trixbox (which would be the obvious upgrade path to keep the similar stuff to what I have)
02:49.27justdaveShould I do trixbox or just throw CentOS 5 on there and a vanilla asterisk? :)
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02:55.27Qwellanybody happen to know acceptable power levels for cable internet?  rx and tx (I'm most interested in the tx - I think my signal is too hot)
02:55.30Qwellcoppice: you, maybe? :D
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03:03.17tzafrir_homejustdave, you can use freepbx independenly of trixbox, if you like it so much
03:04.35EchinosI don't suppose anyone here has any experience unlocking VTAs with the CYT tool?
03:05.17kiscokidwhat is a VTA?
03:05.38Echinosvoice terminal adapter
03:05.46Echinoslike what you get from a voip provider
03:06.03kiscokidok, like an ATA
03:07.09Echinosyeah
03:07.29Echinossorry, it's the model of the box is dlink-vta
03:07.33Echinosata, yes.
03:09.33*** part/#asterisk Nuitari (n=Nuitari@mail.nuitari.net)
03:10.17coppiceQwell: can't help. sorry. I've never worked with cable modems. I don't know how sensitive their receivers usually are
03:12.00Qwell83 packets transmitted, 43 packets received, 48% packet loss
03:12.01Qwellgotta love comcast
03:12.13coppiceQwell: I doubt your tx would be too high, though, unless someone has tampered with the modem's software. they usually need to limit this for approvals
03:12.32EchinosQwell: http://www.dslreports.com/faq/5862
03:12.34Qwellwell, earlier today when it was working okay, the tx was 37dBmV, and now it's like 42
03:12.36Echinosany help?
03:13.45EchinosI also see "levels that vary more than 3db in 24h usually indicate a problem"
03:14.02coppiceQwell: I beleive its adaptive. each end tells the other how strong a signal it is receiving, and they send the minimum necessary. this reduces pollution, and is likely to vary by a few dB, depending what else is on the cable at that time
03:14.37Qwellso, 42 should be just fine
03:15.01Qwellof course, comcast locks me out of the other pages, so I can't see snr
03:18.00coppicehas sangoma released drivers that will work with linux 2.6.22 yet?
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03:27.05CCFL_Man2Qwell: go dslxtreme
03:27.26CCFL_Man2i gots myself a cisco router with adsl wic, i'm happy
03:27.58tzafrir_homeQwell, I hope you now learn to appreciate ssh and irc :-(
03:28.39CCFL_Man2the way cumcast does things makes me sick
03:28.46CCFL_Man2it's pathetic
03:30.57coppicewe put the con in convergence :-)
03:31.10CCFL_Man2heh
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03:31.24CCFL_Man2tis true
03:31.37CCFL_Man2anyone have cisco VCM or HCM cards?
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03:36.47CCFL_Man2carrier access be back up
03:37.38CCFL_Man2http://72.164.241.3/support/diagnostics/documents/adit_600_9_4_user_manual.pdf
03:40.06CCFL_Man2thank strom for that :)
03:41.37networkjediHas anyone come forward to buy Carrier Access?
03:43.26CCFL_Man2networkjedi: what you mean?
03:43.47CCFL_Man2some big telco supplier wants to buy carrier access
03:43.56networkjediFrom what my CAC Sales rep told me CAC is on the chopping block
03:43.57Qwelltzafrir_home: ssh is unuable
03:43.59Qwellunusable
03:45.33networkjediCCFL_Man2 - Tellabs was supposidly close to closing on buying CAC but the last news I heard was start of August
03:45.36CCFL_Man2networkjedi: can you get me the latest firmware? :P
03:45.44CCFL_Man2ahh
03:46.04CCFL_Man2i need adit 600 firmware :P
03:46.17networkjedifor what cards?
03:46.34networkjedithe firmware is an arm and a leg!!
03:46.39CCFL_Man2the tdm controller, fxs, and isdn bri
03:46.58CCFL_Man2maybe fxs cards don't get a firmware upgrade
03:47.07networkjedihmm.....never used the isdn bri cards, I don't think fxs cards get firmware
03:47.15*** part/#asterisk kiscokid (n=ron@208.106.35.66)
03:47.27CCFL_Man2tdm firmware would be greatly appreciated :P
03:48.00CCFL_Man2if you already have it
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03:55.41coppicethe firmware for most things costs and arm and a leg, but if you negotiate well you can get to keep your arm up to the elbow.
03:57.46networkjedivery true
03:58.04networkjedilately we've been getting free firmware for fixes to bugs we've found
03:58.12networkjedimostly related to the CMG cards
03:58.33coppicethere needs to be legislation about that crap. it should be illegal to charge for bug fixes
03:58.50coppiceand illegal not to make a great effort to provide them
03:59.00networkjediI agree!
03:59.51networkjediCAC is pretty good about that though, anytime I've called in they get me the fix free, we buy quite a bit from them though so I figure they owe us
04:03.13justdavefor the people who were asking about Network Solutions earlier, I see SANS is reporting them as down now also.
04:03.47networkjediwow, that's nice!!
04:04.57networkjedisomeone must have turned off the one DNS server that Network Solutions has I guess
04:05.51coppicepink slip them, and teach them the true meaning of redundancy
04:06.41networkjedihehe.....interesting to see what the actual issue was....they are pretty big to just be offline
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04:44.29darkgamer20can someone help me configure asterisk to make phone calls at a specified time and start to play a message when the other end picks up
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05:01.27WilliamKsounds like someone has dreams of making an autodialer
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05:13.54Sweeperdarkgamer20: check out call files. but if you create a phone spamming system, the asterisk daemons will come out and do interesting things to your toenails
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05:21.20Teln1100ADoes anyone know how this can be accomplished: Initiate a call from a web interface linked to asterisk that calls a fixed number, then allows that caller to specify what number to conference to, in essence dial out without paying long distance?
05:22.10jqlsimple matter of using a call file
05:22.13heeliosTeln1100A: sounds like something you should be able to do with call files and dialplan magickz.
05:22.37heelioshell not even magickz. <_<
05:23.23jqlHell, a context with just WaitExten and Dial would probably be enough, if you were sufficiently lazy
05:23.43heelioswho isnt?
05:24.00Teln1100AI am trying to set this up on a Centos vps to try and save money on cellphone charges
05:24.03jqlOh, I know people who are insufficiently lazy. It makes me sad
05:24.45Teln1100Aas of now asterisk seems to be installed, but gives the following error: Starting asterisk: Cannot find your TTY (9)
05:27.10Teln1100Adoes the yum install method work for asterisk on Centos 5 or does it need to be compiled from source?
05:29.10jqlhmm... never even occured to me to try that
05:29.24jqlI always compile from source. dunno
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06:20.57CCFL_Man2hah, i have an original fxs card for the adit 600
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11:13.17WildPikachuhrmmm, my gxp2000 has firmwareA.bin , but the firmware on grandstream site is firmwareB.bin/
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12:02.35saftsackhi are there any handys which are capable in doing voip over umts?
12:03.18mvanbaakwith a codec that has little bandwidth it should be possible
12:04.57Mavvielike the what's it's name again....
12:05.04Mavviedit-dit-dash-dash-dit-dash one.
12:05.06MavvieMORSE!
12:05.08*** join/#asterisk dexteruk (n=dexteruk@89.253.168.92)
12:05.17mvanbaaklol
12:05.24dexterukProblem with Asterisk 1.2 with realtime mysql access can anyone help mysql is working with the CDR table but not the realtime
12:07.15dexterukI have upgraded asterisk to the latest version SVN-branch-1.2-r78370M
12:07.31tzafrirhttp://tools.ietf.org/id/draft-bryan-sipping-midi-01.txt
12:07.42tzafrirmidi seems to be a rather efficient codec
12:08.12*** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-217.range81-152.btcentralplus.com)
12:08.19tzafrirwell, until someone actually talks
12:09.15tzafrirdexteruk, I really know nothing about this
12:09.50tzafrirbut pastebin the ralevant parts of your config, and try to convince us you configured your system properly
12:10.51dexterukhttp://pastie.caboo.se/92694
12:11.17mvanbaakMySQL RealTime: Failed to connect database server asterisk on localhost (err 2002)
12:11.17dexterukits as if asterisk is not reading the res_mysql.conf
12:11.26mvanbaakthat's the error !
12:11.41dexterukyes but if you read down i looked at the debug
12:12.22dexterukin the debug asterisk is not reading the username or password from the res_mysql.conf however in the cdr it is
12:13.33mvanbaakwhy did you specify both a socket and a port ?
12:14.11dexterukthat was just the config i found
12:17.26mvanbaakthis is trunk ?
12:18.03dexteruktrunk version yes
12:20.19mvanbaakmeh
12:20.27mvanbaaktrunk is not compiling here ;)
12:22.33dexterukI dont understant what you mean?
12:23.12dexteruksorry im using branches
12:23.29dexteruksvn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
12:23.43dexteruksvn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
12:24.08dexteruksvn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2
12:26.32mvanbaakah, 1.2
12:26.52Wonkaisn't 1.2 quite out of date now?
12:27.00mvanbaakit's not maintained anymore
12:27.05mvanbaakonly security updates
12:27.30dexterukso your saying i should switch to 1.4
12:28.16mvanbaakprobably the best yeah
12:28.48dexterukare the config file 100% compatible?
12:29.25mvanbaakno
12:29.33mvanbaakread the UPGRADE.txt
12:30.51mvanbaakbut thanks for having me look into res_config_mysql in trunk
12:30.57mvanbaakit was borked
12:34.11knarflyanyone else use * on FreeBSD-AMD64?
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12:40.52mvanbaakdexteruk: both the 1.4 and trunk version of res_config_mysql are working
12:41.04mvanbaakfor trunk you need a small patch at this moment to make it compile
12:41.28mvanbaakhttp://bugs.digium.com/view.php?id=10628
12:41.29mvanbaakthat one
12:45.43dexterukok great thanks i will have a look, do you know if there are any problems with spandsp on 1.4 or is it not needed anymore
12:46.01mvanbaakI have no idea. I dont use spandsp
12:52.04dexterukwhat do you use for handling faxes?
12:52.16dexterukjust incase there is something better that i dont know about :-)
12:55.04mvanbaakI let my ITSP do that
12:55.14mvanbaakthey do fax2mail and mail2fax for us
12:56.14dexterukok, so should i use trunk or branches 1.4
12:56.26mvanbaak1.4
12:58.34dexterukthe branches isn't working and tested version where trunk is developing versions
13:03.18*** join/#asterisk QbY (n=Kelvin@208.36.224.228.ptr.us.xo.net)
13:03.33mvanbaakdexteruk: eh ?
13:04.27QbYI'm trying to upgrade to 1.4; currently I use res_odbc for voicemail, etc. I can get ./configure to recognize my postgres but not unixodbc which is on the machine--could i continue with just postgres and it use my existing database for voicemail?
13:04.34dexterukSorry when you download asterisk via SVN there are two versions that you can download the trunk or branches
13:04.44*** join/#asterisk kaigoh (i=kaigoh@82.133.70.150)
13:05.01mvanbaakdexteruk: indeed
13:05.16dexterukbranches  /* the working area; fixing bugs in existing major releases */
13:05.35dexteruktrunk /* newest version of svn code */
13:05.39kaigohhi there guys
13:05.47mvanbaakhi kaigoh
13:06.04dexterukso if you want the most stable version then it should be branches?  Correct?
13:06.09mvanbaakQbY: I think so
13:06.15mvanbaakdexteruk: yes
13:06.37dexterukok great just wanted to make sure as thats always the version i have been using :-)
13:06.43kaigohjust wondering if anyone can give me a clue what "Re-invite to non-existing call leg on other UA" means
13:07.18mvanbaakdexteruk: in trunk are all the new goodies
13:07.32mvanbaakthey might work, but they may also break the system
13:08.07Davieytrunk seems pretty stable atm
13:08.08mvanbaaklike the res_config_mysql in addons now
13:08.18mvanbaakDaviey: it is stable on my systems :)
13:08.51Daviey'systems'.. are you nuts?
13:09.01kaigohjust wondering if anyone can give me a clue what "Re-invite to non-existing call leg on other UA" means
13:09.09mvanbaakDaviey: eh ?
13:09.24Davieymvanbaak: you are using it on production system_s_
13:09.31Daviey?
13:09.45mvanbaakyes, but not for customers
13:09.54mvanbaakcustomers run on 1.4 svn
13:09.55Davieyoh ok
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13:10.26mvanbaakI have a setup with a box at my home office, a box at a friend and a box at work that run trunk
13:10.38kaigohso, can anyone help me? please? I've been pulling my hair out for hours now!
13:10.51mvanbaakkaigoh: I have no idea. did you try google ?
13:11.02kaigohyeah, not luck there :(
13:11.31mvanbaakkaigoh: looks like asterisk sends a reinvite but the user agent tells you there's no call
13:11.34mvanbaakor something like that
13:11.42mvanbaakthat's how I would translate that message
13:12.06kaigohyeah, makes sense. Is it an issue with X-Lite and asterisk?
13:12.40mvanbaakI have never seen that message. And we have a lot of x-lite users
13:13.32kaigohit only seems to happen with a user who is on NAT. He can call in but cannot be called. We did a VPN link between our machines and managed to get two way calling
13:14.04kaigoheven put his machine in DMZ and completely disaled any firewalls, still same problem. Would STUN resolve the issue?
13:14.23mvanbaakif the user is behind nat you should set canreinvite=no
13:14.43mvanbaakin the sip.conf entry for this user
13:14.55kaigohyeah, done that
13:15.18mvanbaakthen why does it do a reinvite ?
13:16.13kaigohnot sure!
13:16.19*** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net)
13:18.31kaigohthis is the output from CLI>
13:18.32kaigoh[Sep  2 15:17:23] WARNING[4343]: chan_sip.c:12033 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '5e457cb43368c6c7499ac0fa6c9ef6ae@82.133.70.146'. Giving up.
13:18.33kaigoh<PROTECTED>
13:19.02kaigohsorry,     -- SIP/700500-09198188 is circuit-busy
13:19.11QbYis module embedding in 1.4 recommended?
13:23.48chemikki need connect mobilem phone siemens E10 to pc via COM port and asterisk, where is find example configuration, sorry for my english
13:24.25*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
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13:47.23YonahWcan anyone help me figure out why a sip phone can not register when the other sip phones can
13:47.54YonahWit would seem like asterisk does not even receive the request but i believe the traffic is allowed in my iptables
13:48.31*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
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13:55.10roxy_I have created conf room number 600 and I have a user SIP/john . Using the manager, I want to connect the room to john (then the room to other people). What would be a proper call ?
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14:13.56*** part/#asterisk QbY (n=Kelvin@208.36.224.228.ptr.us.xo.net)
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14:16.45wiseguy_hello
14:17.27wiseguy_anybody using RemoveQueueMember? Doesn't work for me - outputs no such queue, but AddQueueMember works ok
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14:50.46NotLarryok, 15 seconds into experimenting with voip and I have made a call to digium with my bt headset, where to I read up on how to make this so I can call my wife from work (where my cell phone does not work:)
14:51.39_x86_NotLarry: you'll need a SIP <--> PSTN gateway
14:51.44_x86_(or IAX2)
14:52.00Corydon76-digNotLarry: you either need a voip provider to translate the call to PSTN or an Asterisk machine at your wife's house
14:52.06_x86_NotLarry: the gateway takes the IP call and puts it on the regular telephone network for you
14:52.15*** join/#asterisk t3rror (n=t3rror@adsl-065-005-255-180.sip.owb.bellsouth.net)
14:52.15Corydon76-digor a SIP/IAX phone, if you have a public IP
14:52.45t3rrorcould anyone tell me where to find information about unlocking a ATA?
14:53.07Corydon76-digt3rror: http://asterisk.druncoder.com/hacks/ats-config/
14:53.17Corydon76-digIt's not an ATA, but it is a phone
14:53.53Corydon76-digand it beats the pants off any ATA
14:54.45t3rrorwell, i have a PAP2 v2 that i am trying to get working with teliax
14:55.09Corydon76-digt3rror: you need to call Vonage to get the unlock code
14:55.19Corydon76-dig$20 fee IIRC
14:55.46_x86_vonage is still running?
14:55.56Corydon76-digFor now, yes
14:55.57_x86_i thought they got their pants sued off by verizon
14:56.07t3rrori fugured that someone would have figured that out by now
14:56.11Corydon76-digThe ruling is under appeal
14:56.17QwellCorydon76-dig: hey, could you shoot me a wifi signal?
14:56.21t3rrori already paid $50 for the device
14:56.31Corydon76-digQwell: I am shooting a wifi signal
14:56.43Qwellcould you make it go another 90 miles or so?
14:56.43_x86_t3rror: should have paid $55 and got the unlocked version ;)
14:56.53Corydon76-digI'm just not broadcasting the SSID
14:57.03Corydon76-digQwell: heh
14:57.08Qwell50% packet loss is pretty much the least fun thing ever
14:57.08t3rrori have been unable to find the unlocked version
14:57.16t3rrori will take this back to BB and look for something else
14:57.38_x86_t3rror: http://voipsupply.com/
14:58.03Corydon76-digt3rror: that phone I linked is quite possibly the best deal you'll find
14:58.16_x86_http://www.voipsupply.com/product_info.php?products_id=1630&osCsid=f07b9ca60917750c551dbcd4d3fefd90&searchid=373829
14:58.18Corydon76-digcordless phone with a SIP base
14:58.19filet3rror: did you see that the chinese have already gone after it?
14:58.21Corydon76-dig$70
14:58.55_x86_Corydon76-dig: doesn't load here
14:59.08_x86_ah
14:59.15_x86_you missed the "k" in drunk ;)
14:59.20Corydon76-digOh, sorry
14:59.21t3rroryeah
14:59.26t3rrori figured it out
14:59.31Corydon76-dighttp://asterisk.drunkcoder.com/hacks/ats-config/
15:00.01Corydon76-digFirst link is "Obtain phone"
15:00.27QwellCorydon76-dig: did you ever end up finding the extra stations?
15:01.10Corydon76-digQwell: nope, but I'm going to go find another DECT phone and try pairing today
15:01.22fileCorydon76-dig: ooh tell me how that turns out
15:02.08_x86_wtf... only staples has it?
15:02.14Corydon76-digCorrect
15:02.26_x86_first line item "Requires VOIP service through Lingo"
15:02.30_x86_heh
15:02.35Qwellthere's got to be another name for that phone..
15:03.52Corydon76-dig_x86_: which is why I have that page... unlock codes for the phone
15:04.12*** join/#asterisk YoYo (n=chatzill@pool-141-152-82-158.roa.east.verizon.net)
15:04.53YoYowhat do I need to look at when there are no spans in /proc/zaptel/ ?  the modules load without complaint, but no spans... so ztcfg errors out
15:05.15t3rrori don't see any unlock codes there
15:05.23t3rrorjust the user/pass
15:05.25Corydon76-dig_x86_: it's aka the JIN501
15:05.32Corydon76-digt3rror: that's the unlock key
15:05.40t3rrorgotcha
15:05.45Corydon76-digt3rror: without that login, it won't let you change the SIP settings
15:06.17t3rrordo you have to worry about lingo trying to provision it or anything?
15:06.28Corydon76-digNope
15:06.43Corydon76-digThere's an autoprovisioning mechanism, but it's not enabled by default
15:06.50t3rrorthat sure is one ugly phone
15:07.07*** join/#asterisk saftsack (n=saftsack@p57A757C5.dip.t-dialin.net)
15:07.26filebut DECT is a standard, so you should be able to use your own phones
15:07.29filein theory.
15:09.52YoYoanyone?  what would prevent spans from being created when loading the tor2 module?
15:11.38Corydon76-digLack of the board?
15:11.44Corydon76-digBoard not detected?
15:11.47Corydon76-digBoard fried?
15:12.55coppicedeep fried or sauteed?
15:13.00YoYoit's there.  at least according to lspci
15:13.12YoYothough it doesn't seem to know what it is:
15:13.23*** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net)
15:13.23*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)
15:13.24YoYo01:09.0 Bridge: Unknown device 00b5:d00d (rev 01)
15:14.30chemikkit will be function when i connect mobile phone to pc via COM port and asterisk?
15:19.06tzafrirYoYo, pciradio?
15:19.19*** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
15:19.19YoYopciradio?
15:19.22*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
15:20.34tzafrirYoYo, is that a zaptel card?
15:20.51YoYoyes, it's the tor2 quad T1/PRI card
15:20.55wwalkerI do entirely VoIP stuff with asterisk, no TDM, no POTS, so I'm looking for a little feedback.  A friend needs to run all his incoming/outgoing calls at home through asterisk, without buying new voip phones.
15:21.24wwalkerso, I need an FXS and an FXO port.  Hardware recommendations?
15:21.31QwellDigium TDM400p
15:21.53YoYowwalker: I've used the TDM4xx card... some success, but not satisfied with it... kept locking up, requiring me to unload/reload the zaptel drivers and restarting asterisk
15:22.09QwellYoYo: Did you call support?
15:22.26YoYoIMO, it's not electrically sound
15:22.57YoYoQwell, 2 people @ digium told me they never heard of such a problem before
15:23.15Qwellokay, was it support?  Did you actually call them?
15:23.24QwellIf I were to say I've never seen that problem, it would mean absolutely nothing
15:23.30YoYogranted, I bought the board and modules when it was brand new
15:23.33YoYomight have a fix by now
15:24.02YoYoQwell, it was via IRC... but they were digium ppl
15:24.10*** join/#asterisk famicon (n=pastry@c51447ddc.cable.wanadoo.nl)
15:24.12Qwellbut it wasn't support
15:24.31[TK]D-Fenderwwalker, Linksys SPA-3102.
15:24.45QwellYoYo: Developers aren't gonna know crap about hardware.  Like I said, me saying "I've never seen that before" means nothing
15:24.46[TK]D-Fenderwwalker, $70 and far more flexible
15:25.17YoYoQwell, it doesn't matter anymore... that card was recycled over a year ago
15:25.58YoYobut I would expect developers to know a hell of a lot more than support people
15:25.59Qwellit does matter - if your only opinion about hardware is based on something you never even attempted to get fixed...
15:26.32YoYoand, if anyone @ digium responded to my inquiry (no matter if IRC, email, or phone), I expect that answer to be accurate
15:26.49wwalker[TK]D-Fender: thanks, got one of those at the office.  seems to work well, changed the SIP nat setting once.
15:27.05YoYoso, since digium had never heard of such a problem, I assumed that the hardware was crap and dumped it
15:27.21QwellYoYo: The *developers* you talked to hadn't heard of it.
15:27.38wwalkerYoYo: or it was defective and you should have made them replace it, unless you were running it in Dell hardware.
15:28.20YoYoQwell, well, if unloading/reloading the module provided a temporary fix to the problem, then it was obviously a problem with teh code... if the developers weren't aware of the problem, then somthing was seriously screwed up
15:28.43QwellYoYo: So, a developer for asterisk should know all about kernel driver modules?
15:28.47Qwellwhy?
15:29.03mvanbaakif the developers weren't aware of the problem,
15:29.03YoYodeveloper for zaptel
15:29.04mvanbaak<PROTECTED>
15:29.06YoYoyes
15:29.09mvanbaakthat's called: a bug
15:29.11YoYozaptel from digium
15:29.22YoYodigium developer not aware of problem and not willing to look at it
15:29.30QwellYoYo: So, you're saying that every developer for Digium should know how to write kernel code?
15:29.54mvanbaakdamn, there goes my changes to ever become a digium dev
15:30.00Qwellmine too
15:30.05YoYoQwell, I expect every digium developer to be familiar with their products
15:30.06Qwelloh, wait
15:30.16YoYoif they're not, then they shouldn't have made a comment
15:30.20QwellWhy?  We don't deal with hardware.  It's not why we work there.
15:30.36coppicewell I think everyne there should knwo the TDM400 has had a rather troubled history. if they don't perhaps they aren't getting enough caffeine to keep them awake
15:30.47YoYobtw, the developer in question was Mark... and yes, I expect him to be fully familiar with teh zaptel drivers... or to at least acknowledge the problem
15:30.58YoYobut, as I said... it's history now.. I dumped the card weeks after buying it
15:31.05QwellMark should have referred you to support. :)
15:31.43YoYoQwell, you're funny... why would mark refer me to support?  Just so support could escalate the problem back to Mark?
15:31.59QwellYes.
15:32.04QwellIf it needs to be
15:32.08YoYotoo funny
15:32.29YoYoanyways, if I can't get this tor2 card running, what 2+ port T1 card is currently best of breed?
15:32.32mvanbaakYoYo: ever been part of a company with different divisions ?
15:32.55YoYomvanbaak: ever been part of a company where owners/principles take personal responsibility?
15:33.34mvanbaakyeah, but they never handled customer reports directly
15:33.49YoYomvanbaak: yes, I have... I lasted for about 4 months.  Between the president, 6 VP's, dozens of managers, not one person would accept personal responsibility
15:33.51YoYoso I left
15:33.53mvanbaaksupport can help. if not, they know where to forward the issue
15:34.24mvanbaakthat's how customer support works when there's more then 1 type of coworker in a company
15:34.40QwellIf it was truly a bug in zaptel, there should have been a bug report created.  Was there a bug report created?
15:34.49YoYoQwell, I have no idea
15:35.06YoYobut, I'm tired of you pushing for a full blown flame war
15:35.17*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
15:35.22YoYoIMO, the TDM400 is wacky hardware, and my opinion won't change
15:35.34YoYofirst impressions are hard to overcome
15:35.53YoYoand my first impression of you, is that you're a corporate flunky, and that opinion won't change either
15:35.54QwellHow can you possibly say that your problems were both because of a bug in zaptel, and bad hardware?
15:36.19JTtake cake
15:36.22JTeat too
15:37.01Corydon76-digQwell is far from a corporate flunky
15:37.16YoYoCorydon, he's sure acting like one
15:37.31JTYoYo: you're acting like a whingeing spoilt brat
15:37.42Corydon76-digYoYo: you have no idea what's wrong and you're passing judgement?
15:37.57Corydon76-digBecause he's saying something you don't want to hear?
15:38.34Corydon76-digDigium doesn't have anything under warranty that even USES the tor2 driver, so why would there even BE a corporate line?
15:38.43YoYono, because he's singing the usual corporate hymns
15:38.49QwellCorydon76-dig: his problems were with a tdm400
15:39.03Corydon76-digQwell: he was using the tor2 driver earlier
15:39.08Qwelldifferent problem
15:39.10YoYotor2 worked fine when I had it in service... now it's not
15:39.32YoYotdm400 /never/ worked reliably
15:39.36YoYotwo different situations
15:39.56Corydon76-digwhich board revision are you using?
15:40.07tzafrirYoYo, you can waste time arguming or actually provide details of your problem
15:40.46Corydon76-digIf it's anything earlier than Revision H, I can understand it not working
15:40.51YoYotzafrir: already provided details... tor2 driver loads, but the spans in /proc/zaptel/ aren't created
15:41.16tzafrirYoYo, so card appears in lspci
15:41.19YoYoas for the TDM400, I already stated that I bought it when it was brand new and dumped it a few weeks later
15:41.24tzafrirmodinfo tor2
15:41.34YoYotzafrir: yes, but it's not identified
15:41.42tzafrirShould that driver identify the card with the same PCI IDs
15:41.57tzafrirLook at the "aliases"
15:42.23YoYohttp://www.pastebin.org/1586
15:42.45*** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com)
15:43.32tzafrirthat's 10b5 as a vendor ID, right. I recall you reported up there just "b5" as a vendor ID. Was that a typo?
15:44.08YoYofrom lcpci:  01:09.0 Bridge: Unknown device 00b5:d00d (rev 01)
15:44.30tzafrirAre you sure that this is it?
15:45.16tzafrirwell, you can try patching tor2.c for that different vendor ID and hope for the best
15:46.08coppiceYoYo: where did you get that card? I think there are some RoHS compliant versions around that need a slightly different driver
15:46.20YoYocoppice: from digium... 2 or 3 years ago
15:46.35YoYoit was in use for about a year, then sat in a static bag since then
15:46.44coppicethen forget what I said
15:46.53Qwellthe newer cards are far better than the tor2
15:47.21YoYoI have a T100 card (also from Digium) that's generating a ton of errors that's causing the span to drop... needs to replace it
15:47.46YoYoQwell, yup, but I'd rather get this $1600 card working if I can :)
15:47.57coppicebut the driver for the tor2 has things the newer drivers lack. supporting a remote tor2 is easier
15:48.13Qwellcoppice: like what?
15:48.38coppicethe newer drivers don't report any of the error information from the chipset,. the tor2 does
15:48.47Qwellhmm
15:48.56Qwelllike in dmesg or something?
15:49.14*** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk)
15:49.16coppicelike in /proc/zaptel and in asterisk itself
15:49.20Qwelloh
15:50.43*** join/#asterisk kkn088 (n=kikoun@84.4.216.243)
15:51.07YoYounfortunately, I get no errors at all, except from ztcfg telling me there's no such device or address
15:51.24YoYooh well, guess I order a sangoma card
15:51.54tzafrirYoYo, again, are you *sure* you see the card as having vendor ID b5?
15:52.23YoYotzafrir: that's the only card that's not identified... I copied/pasted directly from lspci
15:52.26tzafrirIf so: the tor2 driver won't probe for it. But I don't see why this would have happened
15:52.54tzafrirso patch tor2-hw.h to include that pci id as well
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16:16.45_x86_YoYo: sangoma++
16:17.22_x86_too many problems with digium cards, and rhino cards were never worth even looking at in the first place ;)
16:17.58*** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu)
16:18.02[TK]D-Fender_x86_, Dunno about Rhino... they have been int he business a good while now and their channel banks seem decent.
16:18.14[TK]D-Fender_x86_, I'm waiting for some quality feedback on them
16:18.17_x86_[TK]D-Fender: use a lot of their channel banks still
16:18.21WildPikachuhrmm ... when dialing into an fxo card, I get put into the queue, i then put down the phone, but the queue still rings through to the members, is this right?
16:18.36_x86_[TK]D-Fender: i just would never consider a rhino T1 card
16:19.03_x86_WildPikachu: hangup detection?
16:19.11WildPikachuhow do I enable that?
16:19.44[TK]D-Fender_x86_, again, I'd wait for someone else to report in.  their 4-port analog HWEC is VERY competitively priced
16:20.05Qwell[TK]D-Fender: So are chinese clones
16:20.07Qwell:p
16:20.12_x86_hah
16:20.44_x86_[TK]D-Fender: yeah I hardly ever deal with analog anymore... i'm looking for 4-port T1 w/ HWEC
16:20.57_x86_some locations, 8 port
16:21.56mvanbaakA104d :)
16:21.56*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
16:22.25_x86_A104DX is what i've been using
16:22.36_x86_PCIe version of the A104D
16:22.37YoYoso, for sangoma users... how about sangoma + asterisk on FreeBSD?
16:22.54QwellYou still need zaptel
16:23.02_x86_YoYo: Asterisk was designed to run on Linux...
16:23.05mvanbaakI have no idea about the zaptel status on freebsd
16:23.18mvanbaakthey claim it's working
16:23.23_x86_YoYo: you can drive a car in reverse too, but it's much faster to put the bitch in drive and do it right ;)
16:23.39Qwell_x86_: see pm :D
16:23.41YoYox86: yeah, that's the only reason I've tolerated linux for the last 4-5 years
16:24.16mvanbaakasterisk runs fine on BSD
16:24.36mvanbaakit's the zaptel part that works on linux only (as far as the original project concerns)
16:29.08*** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net)
16:33.56_x86_mvanbaak: it's only commercially supported on Linux, afaik
16:34.01mvanbaakyup
16:34.24mvanbaakbut it works great on BSD out of the box
16:34.41_x86_ok, well it's only legal to drive forward a distance down the street, doesn't mean i'm going to break the rules and drive in reverse just because i can ;)
16:34.53mvanbaaklol
16:35.04mvanbaakbtw, it wont work on hppa linux ;)
16:35.21Qwellmvanbaak: it does, you just have to break the binary up into 7 pieces
16:35.33mvanbaakhahahahahaha
16:36.30mvanbaakbut I did save my finger !
16:37.08mvanbaakMDB2 Error: no such field
16:37.10mvanbaaknice !
16:37.15mvanbaak_WHAT_ field
16:37.22mvanbaakI hate php errors
16:37.37WildPikachuhrmmmm
16:37.47WildPikachudoesn't appear my telecom sends polarity switch when i hangup
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17:15.01[X-tp]would it be possible to do something like "exten => _+46.,1,Dial(SIP/0${EXTEN:3}@outbound)" to rewrite +46 to just 0 when dialing a number?
17:15.47[TK]D-Fender[X-tp], sure
17:15.52mvanbaakof course
17:16.00[X-tp]is that correctly written?
17:16.20mvanbaakuhhuh
17:16.32*** join/#asterisk implicit_ (n=implicit@vc240232.vpn.uci.edu)
17:17.09[X-tp]so all I have to do now is figure out why that doesnt work... thanks for confirming...
17:17.45[TK]D-Fender[X-tp], You'll want to verify the number coming in.
17:18.12mvanbaaktry something like: exten => _.,1,Verbose(1, Extension to use is ${EXTEN})
17:18.37[X-tp]ok
17:18.46[X-tp]exten => _46.,1,Dial(SIP/0${EXTEN:2}@outbound) works fine for me...
17:18.48mvanbaakI have that as comment in almost all my contexts. comes in handy when debugging stuff
17:19.01[X-tp]sound like a good idea
17:19.11[TK]D-Fender[X-tp], debug your inbound channel to make sure the "+" is interpreted properly
17:21.32*** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu)
17:21.44[X-tp]thanks for the tip... the "+" wasnt sent...
17:23.57WildPikachuhrmmm, how do I get asterisk to stop complaining about in-use on my phone?
17:24.46mvanbaakhangup ?
17:24.49mvanbaaklol
17:24.52mvanbaaksorry
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17:42.20roxy_what is : /dev/zap/pseudo , I am trying to use a room but I get an error: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory . (I use only sip atm)
17:42.46ManxPowerroxy_: you need zaptel running, even if it is just ztdummy
17:42.57roxy_ManxPower: thanks
17:44.37WildPikachumvanbaak, its a phone that can handle more than one sip call, so asterisk is complaining that the phone "should be" inuse
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18:02.11mmlj4hey ManxPower
18:02.37mmlj4have you gotten ormond up and running yet?
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18:16.07bminishhere's a weird one. incoming IAX calls will only ring to my SIp extentions not my Zap ones. incoming zap calls will ring to all extentions. the string to call the extensions is the same for both contexts
18:16.22bminishit all used to work but recently I went to a newer zaptel and newer Asterisk
18:16.25bminishversion
18:17.07roxy_when I use the CLI to use the command meetme I get a table of the used conf rooms with more info. When I use asterisk-java, the only answer I get is "Follows". In what situation does the cli answer "Follows" ?
18:17.34mvanbaakroxy_: there comes more output
18:18.00roxy_mvanbaak: thanks
18:24.04bminishExecuting [91444114@incoming:1] Dial("IAX2/blueface2-out-5", "SIP/brendan&Zap/2&Zap/1&SIP/bmwifi|30|two") in new stack
18:24.18bminishbut only the SIP devices ring
18:24.37bminishExecuting [s@incoming:1] Dial("Zap/4-1", "SIP/brendan&Zap/2&Zap/1&SIP/bmwifi|30|two") in new stack
18:24.49bminisheverything rings
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18:35.18rbdhey guys, can someone give me a pointer to a sip softphone that I can use to directly call a sip endpoint (without requiring some kind of sip provider...this is just for development work)..e.g. have a dial string like sip:123@10.1.1.5:5060
18:35.36rbdthis would be for windows btw
18:36.51fakhirhttp://zoiper.com
18:37.10roxy_mvanbaak: I found the asterisk-java way of getting the result. thanks again
18:41.00Qwell0x98F2D153
18:41.04Qwell^ pointer
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18:47.01rbdfakhir: thanks
18:47.32t3rrorok, i returned the pap2v2 and i picked up a pap2v1
18:47.36t3rrornow i need to unlock it
18:50.24QwellCall Vonage and ask them for the unlock code
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19:07.43[TK]D-Fendert3rror, And while your're at it, call up Cadbury and aske them about their Caramilk bar.....
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19:13.27WildPikachuwhat is an average voice quality bitrate in mp3 format?
19:13.40WildPikachucurrently i've set my monitoring to 64kbps
19:13.44WildPikachuvariable bitrate
19:21.13Strom_CWildPikachu: how does mp3 figure into it
19:21.23WildPikachuI use lame  :)
19:21.35WildPikachulame to convert the .wav
19:21.41WildPikachubut it appears very big :(
19:21.52Strom_Cfor best quality, you want to do the following:
19:21.56WildPikachutaking into account I have at least one channel open 10 hours a day
19:22.26Strom_Cfirst, upsample the wav to 22,050 Hz
19:22.39Strom_Cthen, compress it at about 40kbps
19:23.05Strom_Cbut, really, you're better off just using a codec which is designed for this stuff, like gsm or g729
19:23.13Strom_Cyour gsm files will be 13kbps
19:23.31WildPikachuso better to save in .gsm format?  ... problem there is I can't copy it to my pc and listen to it easily
19:24.00Strom_Cwav49 is wav-encapsulated gsm
19:24.10Strom_Cand IIRC, windows can handle that
19:24.23denonthere are gsm players for windows
19:24.26denoneven plugins for winamp
19:24.30Strom_Cyup
19:25.29denonhmm, I would think videolan would also play pretty much anything
19:25.33denonon any os
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21:26.19kombiI wonder, can you make * direct all it's traffic over eth1, while other services listen eth0?
21:26.28kombi*on
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21:28.07BSD_TechI found a issue in trunk
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21:28.25BSD_Tech<PROTECTED>
21:28.25BSD_Tech/usr/src/zaptel-head/tor2.c: In function ?tor2_remove?:
21:28.25BSD_Tech/usr/src/zaptel-head/tor2.c:603: warning: asm operand 1 probably doesn?t match constraints
21:28.25BSD_Tech/usr/src/zaptel-head/tor2.c:603: error: impossible constraint in ?asm?
21:28.25BSD_Techmake[3]: *** [/usr/src/zaptel-head/tor2.o] Error 1
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21:29.54*** join/#asterisk WileyEggplant (n=Hawk@rrcs-70-63-20-194.central.biz.rr.com)
21:30.41kombithe idea is to use a dial up line fot rtp and a fixed ip one for all else over two nics, anyone done that here yet?
21:31.35JTrtp over dialup?!
21:31.45fujinnasty
21:31.47fujinwhy
21:32.06kombiJT: cheap bandwidth
21:32.09JT...
21:32.12JT~cheap
21:32.13jbotcheap is, like, a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
21:32.17WileyEggplantAnyone familiar with Perl/AGI that would be willing to answer a couple of questions? :)
21:32.19JTcheap quality
21:32.20JTas in
21:32.24JTabsolute shithouse
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21:32.58fujinlol @ rtp over dialup
21:33.00fujinthat's pure legend
21:33.04JTindeed
21:33.48kombiJT: hmm, maybe elaborate a little on that, sip itself does not need a fixed address to my understanding, dsl bandwidth here is at 10Mbit upstream for little money, which problems do you see?
21:33.53asdxcan i try asterisk without pbx cards?
21:34.00kombido!
21:34.04JTkombi: you said dialup, not adsl
21:34.20JTasdx: yes
21:34.35kombisorry, but dsl dials, doesn
21:34.41kombi't it?
21:34.44JTno.
21:34.47WileyEggplantNot really... PPPoE isn't dialing
21:34.48JTmy god
21:34.50asdxJT: what can i do without pbx cards
21:34.54JTread wikipedia or some shit, kombi
21:34.56kombianything!
21:34.58JTyou have no idea how dsl works
21:35.01WileyEggplantIt is just authenticating and allocating a "virtual channel"
21:35.11JTit's connected to the dslam by jumpering at the exchange
21:35.26kombiJT: anyhow, i just used the wrong term then, what about what I meant?
21:36.21kombiasdx: connect with softphones and try stuff out
21:36.32asdxkombi: ok i will do that
21:37.43kombiJT: what I'm wondering is how one could direct *'s traffic to a separate nic in order to use a dsl line
21:39.44kombiquietness there..;) everyone jumps on me because I said dsl dials (frankly don't give a damn, never ever had adsl, just sdsl..;)
21:40.06WileyEggplantWell, kombi, SIP has a reinvite feature...
21:40.22kombiWileyEggplant: so I understand
21:40.37WileyEggplantBut how to get Asterisk to reinvite to another interface, I couldn't say honestly.
21:41.04WileyEggplantMaybe if you could set the NAT IP in the settings to the one on the other interface
21:41.09WileyEggplantand have it listen on both
21:41.17kombiWileyEggplant: good point..
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21:41.17asterisknerds2<PROTECTED>
21:41.34WileyEggplantBut not certain how that would work out without actually testing it
21:42.01kombiso far I got by whithout ever natting rtp..
21:42.46WileyEggplantWhat are you connecting to, or rather what is connecting to you?
21:43.16[TK]D-Fenderkombi, Frankly I don't see * as being able to handle this quite the way you want.  go look at SER
21:43.24kombicombined webserver, webradio, database and * on one box
21:43.48WileyEggplantBut do you have phones connecting to your * box?
21:43.56WileyEggplantor is it another * box on the other end?
21:43.59kombisip gateway
21:44.23WileyEggplantDo they require registration?
21:44.32kombithe other idea I had was to make the router direct rtp ports over the cheap connection
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21:45.07WileyEggplantCan you use DynDNS and your dynamic IP for SIP?
21:45.46kombiWileyEggplant: good point too, I don't like to rely on dyndns for production web servers though
21:45.58WileyEggplantOr heck, most SIP providers I have seen work with Dynamic IPs just by registering
21:46.09WileyEggplantI am just talking about the SIP traffic
21:46.10kombithat was my thought too..
21:46.13WileyEggplantand RTP
21:47.26kombiI think a beefed up router might be the solution, it decides what traffic to send over which line. that needs to be fairly intelligent though
21:47.44WileyEggplantHow are your 2 connections set up currently?
21:48.12kombiso far there is only one and I'm thinking of getting a second one
21:48.17WileyEggplantIf you have 2 different routers on the network, you can modify the routing tables on the asterisk box
21:48.27WileyEggplantSo both connections go to the 1 router?
21:49.16kombithat's what I was thinking, but you're right too, one could route on * box too
21:49.52kombione router handling two separate connections needs to be fairly advanced
21:49.56WileyEggplantYou could get the IP of the SIP provider's server you are communicating with... and direct the box with * on it to route all traffic to that address through a certain gateway
21:49.59WileyEggplantor interface...
21:50.40WileyEggplantI am doing something very similar right now in order to route through a VPN to a public IP to bypass IP restrictions...
21:51.45WileyEggplantso if you have 2 NICs...
21:51.47WileyEggplantroute add -host IP.IP.IP.IP eth1
21:52.14WileyEggplantA command like that would route it to the default gateway on the interface specified, just for that 1 host
21:52.51kombithat's right, thing is though, several services on the same host (or box)
21:53.16WileyEggplantShould not matter... Because the only thing communicating with the provider's SIP server will be Asterisk, right?
21:53.27kombicorrect..
21:53.43WileyEggplantSo it will only route stuff destined for THAT 1 IP over the other interface
21:53.50kombihow do you tell * traffic from the rest though?
21:54.01kombiok...!
21:54.04WileyEggplantYou have to know where the * traffic is going
21:54.16kombigot ya!
21:54.17WileyEggplantBut if you are using a provider, it usually all goes back to the provider's server first and out from there
21:54.20WileyEggplant:)
21:54.23WileyEggplantok
21:54.43kombigood idea indeed! and simple too, me likeum!
21:55.15WileyEggplantYes, and asterisk doesn't even have to know. It can remain clueless to the change
21:55.32kombiwill both directions travel over that line though?
21:55.58WileyEggplantThe connection is typically initiated by your * box registering with the ISP... so the ISP will send it back to the originating IP
21:55.58kombiI mean up/down-stream
21:56.09WileyEggplantSo yeah
21:56.36WileyEggplantand by ISP I mean SIP provider
21:56.37kombieven if that is dynamic, right! splendid indeed
21:58.02kombi50Mbit downstream & 10Mbit upstream at some $90 a month
21:58.10kombiordered!
21:58.11WileyEggplantwow
21:58.16WileyEggplantI wish we had that here
21:58.33kombithey have "no call centers" in there fine print somewhere..,)
21:58.43kombibut who cares!
21:58.56WileyEggplantlol... a call center?
21:58.57kombiWileyEggplant: where're you at?
21:59.10WileyEggplantI have VoIP in callcenter application... nothing but problems
21:59.16WileyEggplantOhio, US
21:59.23kombiok..
21:59.30WileyEggplantI am a Telecom admin at a large callcenter company... so I've used VoIP there
21:59.50WileyEggplantMostly Cisco voice gateways though
22:00.10kombiso you don't use * in there?
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22:00.27asterisknerdscom<PROTECTED>
22:00.38WileyEggplantWe do... for the systems at local sites, and inter-site communication now
22:01.03WileyEggplantBut for handling of client calls, it is usually T1s or Cisco virtual T1s
22:01.04kombiI see
22:01.22*** part/#asterisk pbxtech (n=Miranda@208.254.183.84)
22:02.28kombiWileyEggplant: What kind of problems did occur? sound quality? lost connections?
22:02.55*** join/#asterisk pbxtech (n=Miranda@208.254.183.84)
22:03.08WileyEggplantEverything from ISPs "re-prioritizing" the traffic to network latency causing dropped calls and noise
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22:04.14kombiWileyEggplant: not particularly pleasent..
22:04.37WileyEggplantnot at all
22:04.53WileyEggplantWe use VoIP between Cisco gatways over our MPLS between centers
22:05.32WileyEggplantThe ISPs were detecting the CoS on the packets as being Voice, and they would prioritize the first 384KB or so of traffic, then the rest would get squashed
22:05.47kombithat sucks..
22:05.55WileyEggplantWe had to remove or alter the CoS on the packets leaving the Voice gateway, so the ISP didn't know it was voice traffic
22:06.03kombiwhy do they do that, i wonder..
22:06.14kombismart!
22:06.26kombicould the ciscos cope though?
22:06.56WileyEggplantWell, it is a "feature" of their service, they prioritize traffic, and you pay for more than 384k.
22:07.01WileyEggplantYeah...
22:07.17WileyEggplant3M MPLS lines... Usually around 6 Virtual T1s
22:07.18kombiwho's the carrier?
22:07.34WileyEggplantSprint on some, Qwest on others
22:08.01kombiIt just occurs to me that might be the case on the above offer as well..
22:08.40kombiprevent excesive voip use by that token
22:09.08WileyEggplantI *think* Asterisk may have some CoS settings though.
22:09.19kombiso do you handle that kind of load with * internally?
22:09.19WileyEggplantBut then this was a Business VoIP line
22:09.26kombiough..
22:09.36WileyEggplantnot much load on the * boxes
22:09.48WileyEggplantActually, not Business VoIP, just Business MPLS
22:10.06kombiwhat's mpls stand for?
22:10.45WileyEggplantMulti Protocol Label Switching
22:10.58kombithank you! another one to remember..
22:11.02WileyEggplantIt is like Site-Multisite VPN
22:11.12WileyEggplantBut it is at the ISP level so they handle all the routing for you
22:11.50kombisounds nice at the beginning it seems
22:11.56WileyEggplantYou can say, connect 6 different sites with 1.5 or 3M connections to some network cloud so they can talk to each other... maybe have a main site with a 15M pipe
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22:13.00kombinot bad a concept, so you could send voip, samba, domain controllers what ever
22:13.07WileyEggplantYep
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22:13.24WileyEggplantIt may go over internet backbones too... but the routers don't let the traffic mix because it is labeled
22:13.50kombiagain supported by the ciscos I assume
22:14.04WileyEggplantWell, the ISP Ciscos... ours don't have to know about the labels
22:14.30kombipretty good concept, if only it wasn't for said priotization
22:14.33WileyEggplantPackets get labeled on ingress at the ISP... and labels get removed at egress from the ISP cloud
22:14.48WileyEggplantWell, we had one ISP remove the prioritization completely
22:15.35kombiso voip should be running a little better then
22:15.44kombi..over that one
22:16.10WileyEggplantYes... but what you have to make sure of is that you can prioritize your network traffic properly
22:16.11khronosHmm, how does mpls differe from something atm?
22:16.34khronosAh, soemthing like atm?
22:16.36WileyEggplantand reserve enough bandwidth for your VoIP
22:16.38WileyEggplantWell...
22:16.45kombithe tunneling I would guess?
22:16.48WileyEggplantMPLS is like ATM, but at a higher protocol layer
22:16.51*** part/#asterisk javar (n=javar@69.79.134.24)
22:17.13WileyEggplantSo you can have MPLS going over any kind of line
22:17.21WileyEggplantAnything that supports TCP/IP
22:17.55WileyEggplantYou could have it run over Frame relay, ISDN T1s, OC-48s, ATM circuits, anything
22:18.13khronosDoes mpls operate at the same layer as tcp/ip or a different?
22:18.17WileyEggplantAnd because the labels are at a higher protocol layer, once it reaches the other end, it makes no difference
22:18.44WileyEggplantIt just adds another header
22:18.47WileyEggplantto each packet
22:19.15kombianyways, I've got to split here, my daily dose of thinking in java is waiting.. thanks Wiley for that very good idea!
22:19.17WileyEggplantThem removes it at the egress point so the 2 networks being connected are unaware
22:19.25WileyEggplantnp, good luck
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22:20.13khronosCool.
22:20.29WileyEggplantHi b00gz
22:21.52b00gz:)
22:22.13WileyEggplantHey b00gz, are you real familiar with Perl/AGI scripts?
22:22.28b00gzof course not.
22:22.31b00gzwhy whats the question
22:23.43WileyEggplantI have this script I am working on for some crazy tyrannical ahole... and whenever I issue a Noop after a Stream_file command, it cancels the Stream_file
22:24.03b00gzsounds like a personal problem
22:24.15b00gzI am watching some good law and order if that makes it any better.
22:24.24WileyEggplantNot really
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23:12.27JTalso, VoIPoI for a callcentre, dum dum dum dum dum
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23:17.41fujinsdsl?
23:19.28JTit's symmetrical dsl
23:19.36JTusually uses 4 wires
23:19.40fujinah right, we don't have that here
23:19.43fujinonly g.shdsl and adsl
23:20.19JTi think we have almost all dsl acronyms here
23:20.26JTexcept vdsl is still being implemented
23:20.34fujinsweet
23:20.40fujinthey're rolling out adsl2+ to the exchanges now
23:20.47fujinjust makes us get to the ISP bottlenecks faster, though :|
23:21.07JTyeah, vdsl2+ is the next step after adsl2+
23:21.25JTup to 60Mbit/s downstream or something
23:21.28fujinah
23:21.33fujinnice
23:21.47JTif you're really close to the exchange, that is
23:22.46JTwe have PIPE Networks here, a lot less bottlenecks and a lot cheaper data because of them
23:24.00fujinupstream providers are shitting all over the enduser here
23:25.01JTpipe runs peering domestically between most major isps except for telstra
23:25.07JTand have huge fibre networks
23:25.48fujinawesome, owned by who?
23:25.50fujingovernment?
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23:26.08JTnope, it's a seperate entity
23:26.24JTi think technically it's for profit
23:26.43JTbut the guys who runs it does what's right for the industry, not for profits alone
23:27.10JTthey signed a memorandum of understanding with VNSL a few months back
23:27.18JTVNSL being one of the world's biggest carriers
23:27.28JTto put in a fibre between australia and guam
23:27.35JTguam connects to the usa and asia
23:27.49JTthe fibre would initially be 640Gbit/s, upgradeable to 8Tbit/s
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23:52.11fujinheh
23:52.12fujinawesome
23:52.15fujin640gbit -_-
23:52.22fujinare they not using the souther cross?
23:52.37fujinthe sc fibre from nz goes direct to west coast usa I think
23:52.39fujini forget
23:53.16JTone leg goes direct to hawaii then california
23:53.29JTthe other goes to fiji then hawaii then oregon
23:53.42JTAJC is already 640Gbit/s
23:53.48JTthe Australia Japan Cable
23:53.52JT2 legs
23:53.57JTto Guam then Japan
23:54.42JTSCC is 240Gbit/s, being upgraded to around 2Tbit/s iirc
23:55.30JTfujin: at the moment pipe is domestic, they want to bring down the price of international transit too
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