IRC log for #asterisk on 20100505

`00:00.10`	`ChannelZ`	`I can't exactly, I'm not at home...`
`00:00.18`	`TehRabbitt`	`oh lol`
`00:00.23`	`ChannelZ`	`well wait maybe I can do it here`
`00:00.43`	`[TK]D-Fender`	`TehRabbitt: ...`
`00:00.45`	`[TK]D-Fender`	`~sipnat`
`00:00.46`	`infobot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`00:00.48`	`[TK]D-Fender`	`^^^^`
`00:01.00`	`TehRabbitt`	`[TK]D-Fender: I've followed that guide, i've done everything it says, but i'm still having an issue`
`00:01.21`	`ChannelZ`	`well unauthed sip doesn't work :) sec`
`00:01.37`	`TehRabbitt`	`what do you mean unauthed sip? lol`
`00:01.38`	`[TK]D-Fender`	`TehRabbitt: O RLY? I can't vouch for that`
`00:03.10`	`TehRabbitt`	`[TK]D-Fender: wanna see my SIP.Conf so you can see i've specified my WAN IP? wanna see a screenshot of the router's ports being fowarded to my internal LAN ip of the server? i've done everything the guide says... people who register from outside can't even place calls to 411 which I can place internally no problem`
`00:03.26`	`CatLynx`	`tshark is my friend when it comes to issues like that :)`
`00:04.23`	`ChannelZ`	`do your ports in rtp.conf match whatever you forwarded`
`00:04.30`	`TehRabbitt`	`lemme check`
`00:04.37`	`[TK]D-Fender`	`TehRabbitt: yes`
`00:04.44`	`TehRabbitt`	`yes they match`
`00:05.37`	`ChannelZ`	`hmm I'm getting 'congestion'`
`00:06.56`	`drmessano`	`Try Mucinex`
`00:07.02`	`CatLynx`	`heh`
`00:07.08`	`TehRabbitt`	`ChannelZ: what does that mean :-\`
`00:07.09`	`ChannelZ`	`isn't that for the other end?`
`00:07.19`	`drmessano`	`No, that's constipation`
`00:07.25`	`ChannelZ`	`TehRabbitt: ? your system is rejecting me.. what does your console say?`
`00:07.34`	`CatLynx`	`Pepto-Bismol`
`00:07.38`	`ChannelZ`	`drmessano: oh I'm thinking of Mucilix or something`
`00:07.39`	`drmessano`	`You're thinking about immodium`
`00:07.56`	`*** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com)`
`00:08.02`	`ChannelZ`	`Musilex`
`00:08.06`	`drmessano`	`Mueslix works for the other end too, but that's more a long-term solution`
`00:09.18`	`TehRabbitt`	`[TK]D-Fender: as per your request: screenshot, http://www.tenehawk.com/rabbott/untitled.bmp`
`00:09.33`	`TehRabbitt`	`[TK]D-Fender: and the settings in sip.conf: http://pastebin.com/kdyrgeVd`
`00:09.41`	`ChannelZ`	`HAH`
`00:09.41`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`00:09.46`	`ChannelZ`	`Forbidden!`
`00:09.50`	`TehRabbitt`	`hold on...`
`00:10.15`	`TehRabbitt`	`try now`
`00:10.42`	`TehRabbitt`	`http://www.tenehawk.com/rabbott/screenshot.bmp`
`00:10.45`	`TehRabbitt`	`whoops 0_o lol`
`00:10.50`	`TehRabbitt`	`forgot what i called it haha`
`00:11.37`	`TehRabbitt`	`[TK]D-Fender: do you see the ports fowarded?`
`00:11.47`	Systemt`	`hey some one can help me ?`
`00:11.47`	Systemt`	`http://pastebin.com/iBGAaqcV`
`00:12.21`	Systemt`	`how can i see on softphone NOT Anonymous just the number..?`
`00:12.26`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`00:13.07`	`[TK]D-Fender`	Systemt`: set the name to blank
`00:13.19`	`[TK]D-Fender`	`TehRabbitt: Looks ok... check your externIP.`
`00:13.50`	`TehRabbitt`	`[TK]D-Fender: externIP matches the wanIP`
`00:13.52`	`[TK]D-Fender`	`TehRabbitt: pastebin your ITSP peer.`
`00:14.02`	`TehRabbitt`	`where do I find that?`
`00:14.11`	`*** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net)`
`00:14.23`	`[TK]D-Fender`	`TehRabbitt: Actually.. you don't even have one yet, do you.. this is just a test between 2 phones?`
`00:14.31`	`TehRabbitt`	`Yes.... :-\`
`00:14.34`	`[TK]D-Fender`	`TehRabbitt: local devices to the server?`
`00:14.40`	`TehRabbitt`	`No... local devices work great`
`00:14.49`	`[TK]D-Fender`	`TehRabbitt: Check firewalls on both sides (the server itself), and the client`
`00:14.58`	`TehRabbitt`	`when I place a call from Device A (inside the lan) to device B (in the WAN outside NAT) then i have issues`
`00:15.36`	`TehRabbitt`	`[TK]D-Fender: Device A is on the same lan as the * server... Device B has all ports fowarded properly afaik / i've even tried going directly attached to a cable modem on device B`
`00:15.41`	Systemt`	`[TK]D-Fender: how can i set Anonymous calls to blank ?`
`00:16.05`	`TehRabbitt`	`[TK]D-Fender: i'm updating the firmware of my router to support SIP fowarding to see if it fixes anything heh`
`00:16.15`	`ChannelZ`	`I'm just getting "SIP/thoth.tenehawk.com-00000017 is circuit-busy" from you but I don't have time to look at a sip debug right now to see if it's me.. will check when I get home`
`00:17.12`	`TehRabbitt`	`ChannelZ: aightt`
`00:17.23`	`TehRabbitt`	`brb firmware is gonna cause me to lose connection for a few`
`00:17.39`	`[TK]D-Fender`	Systemt`: Change the callerid name
`00:17.55`	`*** join/#asterisk TJ^ (~TJ@193.47.83.49)`
`00:17.57`	Systemt`	`where?`
`00:17.59`	Systemt`	`;\`
`00:18.07`	Systemt`	`it's anonymous call`
`00:18.12`	`[TK]D-Fender`	Systemt`: before you dial obviously
`00:18.28`	Systemt`	`it's incoming call`
`00:18.39`	`[TK]D-Fender`	Systemt`: and it hits dialplan like everything else
`00:18.55`	Systemt`	`from anonymous to my sip`
`00:18.59`	Systemt`	`:\`
`00:19.00`	`[TK]D-Fender`	Systemt`: Change it at whatever point you feel like
`00:19.12`	Systemt`	`dialplan ?`
`00:19.21`	`[TK]D-Fender`	Systemt`: How is that a question?
`00:19.30`	`[TK]D-Fender`	Systemt`: YES, you change it.. in your DIALPLAN
`00:19.42`	Systemt`	`sorry im new man`
`00:20.42`	Systemt`	`this is my DialPlan http://pastebin.com/BskaKkLt`
`00:21.20`	`[TK]D-Fender`	Systemt`: well that little bit doesn't appear to have anything to do with the other bit you showed us.
`00:21.41`	Systemt`	`i showd u from my CLI`
`00:21.42`	`ManxPower`	`Why are you using !`
`00:21.46`	`[TK]D-Fender`	Systemt`: and you are starting with a useless Set and jumping. Guess you didn't want to change it there.
`00:22.01`	Systemt`	`listen`
`00:22.16`	`CatLynx`	`listens`
`00:22.29`	Systemt`	`i want evry budy that call my from anonymous will show is real number`
`00:22.40`	Systemt`	`i can see the real number but just on logs`
`00:22.42`	`ManxPower`	Systemt` Simplify your dialplan to aid in troubleshooting. Of course, if you are using a GUI well you are far, far beyond our help
`00:24.02`	`ManxPower`	Systemt`, Where SPECIFICALLY "in the logs" do you see the "number"?
`00:24.16`	`TehRabbitt`	`ok, it's definatally a NAT issue...`
`00:24.16`	`TehRabbitt`	`on THEIR end...`
`00:24.16`	`TehRabbitt`	`their private IP is the ip being registered with my * server`
`00:24.16`	`TehRabbitt`	`not their WAN ip..`
`00:24.19`	`TehRabbitt`	`but they are using a softphone..`
`00:24.39`	`ManxPower`	`TehRabbitt, nat=yes will make asterisk ignore the reg IP`
`00:24.53`	`ManxPower`	`and use the real IP from the header`
`00:25.08`	Systemt`	`yes`
`00:25.09`	`TehRabbitt`	`ah`
`00:25.09`	`*** join/#asterisk coppice (~chatzilla@191.193.17.210.dyn.pacific.net.hk)`
`00:25.37`	`TehRabbitt`	`nat=yes is already on :(`
`00:25.49`	`p3nguin`	`I bet you I don't have a NAT issue, and I can't register to your box.`
`00:25.53`	`ManxPower`	`TehRabbitt, for that peer or in [general] ?`
`00:26.38`	`TehRabbitt`	`ManxPower: both`
`00:26.49`	`TehRabbitt`	`p3nguin: you're back! lol`
`00:26.56`	`TehRabbitt`	`p3nguin: yea idk what's going on :(`
`00:27.26`	`ManxPower`	`TehRabbitt, maybe the softphone has NAT enabled. That would be bad`
`00:28.20`	`TehRabbitt`	`hm brb restarting my router after firmware flash`
`00:28.27`	`p3nguin`	`Yep, just got in.`
`00:28.42`	`*** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68)`
`00:28.50`	`*** part/#asterisk cosmicwombat (~cosmicwom@69.7.44.68)`
`00:28.54`	`p3nguin`	`I tried with and without NAT traversal enabled, and I still got the same results from his box.`
`00:28.56`	`ManxPower`	<ManxPower> Systemt`, Where SPECIFICALLY "in the logs" do you see the "number"? <Systemt`> yes
`00:29.02`	`*** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68)`
`00:29.02`	`ManxPower`	`sometimes I wonder why I even try`
`00:35.28`	`*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)`
`00:38.54`	`Kyosh`	`bcause u love it`
`00:41.06`	`leifmadsen`	`ManxPower: because you're a masochist`
`00:42.29`	`*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)`
`00:42.42`	`ManxPower`	`leifmadsen, everyone knows that! 8-)`
`00:53.00`	`*** join/#asterisk jeffspeff (~jeffspeff@h180.203.28.71.dynamic.ip.windstream.net)`
`00:54.01`	`*** join/#asterisk diegomad (mad@186.81.147.202)`
`00:54.10`	`jeffspeff`	`i'm thinking about setting up an alarm system and having asterisk manage the triggers from the panel. are there any particular systems that are recommended, or any that work better than others, etc. ?`
`00:55.21`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`01:11.58`	`*** join/#asterisk lanning (~lanning@208.87.235.224)`
`01:18.59`	`*** join/#asterisk sourcode (~code@ppp-58-8-238-176.revip2.asianet.co.th)`
`01:20.07`	`*** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.nj.comcast.net)`
`01:20.10`	`TehRabbitt`	`ok back`
`01:21.22`	`p3nguin`	`And the result is?`
`01:22.22`	`*** join/#asterisk coppice (~chatzilla@m121-202-80-162.smartone-vodafone.com)`
`01:23.27`	`TehRabbitt`	`not sure yet`
`01:23.31`	`TehRabbitt`	`just reflashed with SIP support`
`01:23.44`	`TehRabbitt`	`apparently you can setup SIP fowarding or something`
`01:24.50`	`p3nguin`	`Sounds like what you need.`
`01:24.58`	`*** join/#asterisk orioni (~chatzilla@95.107.225.94)`
`01:25.13`	`TehRabbitt`	`http://forum.milkfish.org/`
`01:25.37`	`*** join/#asterisk voipnoob (shiva@69.31.82.90)`
`01:25.41`	`orioni`	`hi there, im trying to move from an old pabx-gsm to asterisk-gsm`
`01:25.49`	`orioni`	`the gsm has only rj11 ports`
`01:26.06`	`orioni`	`what do i need on the pc to connect to that gsm gateway ?`
`01:26.38`	`orioni`	`a fxo or a fxs card /?`
`01:26.52`	`[TK]D-Fender`	`orioni: Depends what its manual says`
`01:27.16`	`orioni`	`not much , its a chinese product`
`01:27.45`	`orioni`	`http://www.suncomm.com.tw/Product_detail.asp?P_ID=6008450&F_ID=1000344`
`01:27.53`	`orioni`	`this is the product`
`01:28.45`	`orioni`	`on manual it says "Phone portâconnect to a phone "`
`01:29.02`	`orioni`	`so i need a fxo port on asterisk to connect to the gsm ?`
`01:29.04`	`TehRabbitt`	`p3nguin: i'm totally signing up for this: http://www.google.com/tisp/`
`01:29.05`	`TehRabbitt`	`lmfao`
`01:29.06`	`TehRabbitt`	`k`
`01:29.07`	`TehRabbitt`	`jk`
`01:29.22`	`TehRabbitt`	`(i know it's old)`
`01:29.55`	`Andrew_M`	`orioni: What is a GSM?`
`01:30.45`	`voipnoob`	`Hi, I have an asterisk related question. My question is theoritical - i.e. i don't have any actual setup yet, I am just trying to feel my way around VOIP. There is an asterisk server on the internet. There are 5 different customers. These are all customers who have use DSL from an ISP for internet connection. Between the DSL Modem & their switch sits a Linux Box which does Firewall, AntiVirus etc. This Linux Box has Apache, OpenSSL etc installed. N`
`01:31.08`	`Andrew_M`	`PABX connected to mobile network?`
`01:31.13`	`orioni`	`gsm gateway ,`
`01:31.28`	`orioni`	`to connect to the mobile providers with sim cards`
`01:32.01`	`Andrew_M`	`OOh, I never heard of such. Is this common in Asia?`
`01:32.10`	`p3nguin`	`voipnoob: You cut off.`
`01:32.24`	`voipnoob`	`Cut off meaning?`
`01:32.36`	`TehRabbitt`	`p3nguin: try connecting now`
`01:32.40`	`*** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id)`
`01:32.53`	`*** join/#asterisk lanning (~lanning@208.87.235.224)`
`01:32.55`	`p3nguin`	`voipnoob: You typed too much and you client isn't smart enough to make a new line, so your text was truncated.`
`01:33.00`	`voipnoob`	`ok.`
`01:33.09`	`voipnoob`	`let me separate it into mulitple lines`
`01:33.26`	`p3nguin`	`tenehawk, registration succeeded (expires = 3600 seconds)`
`01:34.18`	`ariel_`	`voipnoob: you don't have too, you can do all that`
`01:34.20`	`ariel_`	`yes`
`01:35.27`	`voipnoob`	`Hi, I have an asterisk related question. My question is theoritical - i don't have any actual setup yet,`
`01:35.27`	`voipnoob`	`I am just trying to feel my way around VOIP by design a simple project on paper.`
`01:35.41`	`voipnoob`	`There is an asterisk server on the internet. There are 5 different customers.`
`01:35.41`	`voipnoob`	`These are all customers who have use DSL from an ISP for internet connection.`
`01:35.41`	`voipnoob`	`Between the DSL Modem & their switch sits a Linux Box which does Firewall, AntiVirus etc.`
`01:35.41`	`voipnoob`	`This Linux Box has Apache, OpenSSL etc installed. I control the asterisk server & also these Linux Firewall boxes at these 5 customers.`
`01:35.41`	`voipnoob`	`If I want my Asterisk Server to reject all requests which don't come through these firewalls - is that possible at all?`
`01:36.19`	`Doc`	`yes`
`01:36.54`	`voipnoob`	`Doc - is that yes for me?`
`01:36.59`	`orioni`	`voipnoob : use iptables`
`01:37.15`	`voipnoob`	`IP tables on my asterisk box?`
`01:37.19`	`orioni`	`block all other non-sip traffic`
`01:37.41`	`orioni`	`yes`
`01:37.45`	`voipnoob`	`The IPs of these customer firewalls may change`
`01:37.53`	`voipnoob`	`so wouldn't that be a problem?`
`01:37.54`	`Andrew_M`	`orioni: You should need an FXS cards with 8 port + 4 port, and a server that accepts 2 PCI or PCI Express cards.`
`01:38.05`	`p3nguin`	`voipnoob: Have you read The Book?`
`01:38.05`	`orioni`	`use a dynamic dns client, i.e inadyn`
`01:38.10`	`Doc`	`what orioni said`
`01:38.15`	`*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)`
`01:38.25`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91)`
`01:38.28`	`Doc`	`ohh... the firewalls are dynamic?`
`01:38.42`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`01:38.54`	`orioni`	`nope . but the fqdn of the client will be the same even if the ip changes`
`01:39.19`	`voipnoob`	`ok`
`01:39.58`	`voipnoob`	`p3nguin - my timezone is such that it's around 7:00 am now. I downloaded the book yesterday night 12`
`01:40.01`	`voipnoob`	`i will read the book today`
`01:40.14`	`orioni`	`which book ?`
`01:40.17`	`p3nguin`	`~book`
`01:40.18`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`01:40.23`	`MhaddogM1`	`what book do you talk about?`
`01:40.23`	`p3nguin`	`orioni: This one ^^^^`
`01:40.27`	`Andrew_M`	`voipnoob: Why not use VPN?`
`01:40.28`	`orioni`	`;)`
`01:40.29`	`p3nguin`	`mhaddogm1: That one ^^^^`
`01:40.37`	`MhaddogM1`	`a ok ok`
`01:40.46`	`voipnoob`	`VPN would increase bandwidth consumption, right?`
`01:40.53`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`01:41.07`	`orioni`	`isnt sth more updated than August 2007`
`01:41.34`	`Andrew_M`	`voipnoob: Yes, but would also make the connecton secure and encrypted.`
`01:42.34`	`ariel_`	`now why would vpn increase your b/w?`
`01:42.56`	`voipnoob`	`ariel_ - i was told it would increase bandwidth`
`01:43.00`	`p3nguin`	`a regular phone call as compared to a phone call through a VPN...`
`01:43.09`	`Andrew_M`	`ariel: inclreases consumption`
`01:43.14`	`ariel_`	`if you only have 1.5mb your vpn will only use what it has`
`01:43.42`	`Andrew_M`	`ariel: VPN is overhead`
`01:43.53`	`voipnoob`	`VPN would increase the traffic as compared to non-VPN - so that would have to have an effect`
`01:44.13`	`ariel_`	`yes but it's not an issue since you give priority to the voip which is what is needed`
`01:44.24`	`ariel_`	`you can't use up more then what you have`
`01:44.40`	`*** part/#asterisk stones_ (stones@linuxpro.org)`
`01:44.42`	`Andrew_M`	`ariel: It becomes an issue with limited bandwidth.`
`01:44.45`	`voipnoob`	`in that case it would increase the latency`
`01:44.49`	`voipnoob`	`or degrade quality`
`01:44.53`	`ariel_`	`no`
`01:45.13`	`ariel_`	`it aids with limited b/w, in that you allocate what you need`
`01:45.34`	`Andrew_M`	`ariel: what causes more traffic, people inside cars, or people on bycicles?`
`01:45.37`	`ariel_`	`it's correct that outside the pipe your b/w will be less but over all it's still uses them same`
`01:46.27`	`Andrew_M`	`ariel: If people are the useful data, less of them get through if the vehicle is larger, right?`
`01:46.34`	`ariel_`	`Andrew_M: I fully understand the PPS and how much a call takes with overhead.`
`01:46.38`	`p3nguin`	`Your available bandwidth does not change, so pushing bits+overhead of VPN takes up more of that bandwidth that just the bits of the phone call.`
`01:46.55`	`ariel_`	`p3nguin: now that I agree with`
`01:47.12`	`*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)`
`01:48.51`	`*** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.pa.comcast.net)`
`01:49.08`	`TehRabbitt`	`and a few bugs aside i'm back again 0_o damn firmware settings lol`
`01:49.31`	`voipnoob`	`one last question - in my scenario will running any VOIP related software on my firewall box help anything - security or quality? or should the phones just connect to the Asterisk server directly. Do I need a proxy or something running on the linux boxes?`
`01:50.28`	`TehRabbitt`	`lmfao the results of a cache server running on your network: http://www.speedtest.net/result/804177223.png`
`01:50.29`	`ariel_`	`voipnoob: it all depends how you want it to run, you can use pin wholes, but best voice is via direct connections no nat which is the sound killer`
`01:50.56`	`TehRabbitt`	`anyway...`
`01:51.01`	`voipnoob`	`what is pin wholes?`
`01:51.05`	`TehRabbitt`	`someone wanna try connecting and see if it goes through?`
`01:51.10`	`TehRabbitt`	`300/300 thoth.tenehawk.com?`
`01:51.41`	`p3nguin`	`(2033.25) <p3nguin> tenehawk, registration succeeded (expires = 3600 seconds)`
`01:51.47`	`ariel_`	`voipnoob: pin holes are routes you setup via the fw direct to an inside system. like port forwarding`
`01:52.18`	`TehRabbitt`	`p3nguin, can you call ext 500?`
`01:52.36`	`Andrew_M`	`voipnoob: IAX will work through NAT, SIP will not, or hard to make it work through NAT. Only softphones exist that are IAX.`
`01:52.52`	`ariel_`	`sip works via nat`
`01:52.53`	`TehRabbitt`	`went through... "Rob, Ext300"`
`01:53.01`	`TehRabbitt`	`no audio :(`
`01:53.07`	`ariel_`	`just need to do it right and sometimes you need a stun server`
`01:53.17`	`p3nguin`	`tehrabbitt: If you said anything, I never heard it.`
`01:53.26`	`TehRabbitt`	`so no audio 2 way.. nice`
`01:53.27`	`TehRabbitt`	`lol`
`01:53.35`	`Andrew_M`	`ariel: you are correct, but it is much more work than IAX.`
`01:53.39`	`p3nguin`	`Sounds like an RTP problem.`
`01:53.49`	`voipnoob`	`Andrew_M -> I am confused here - most companies have firewalls & NATS. & they use SIP phones, right?`
`01:54.05`	`TehRabbitt`	`sooo how do i fix the RTP problem lmao`
`01:54.08`	`voipnoob`	`I will again say that i am very new at all this -> so i maybe misunderstanding something`
`01:54.23`	`ariel_`	`voipnoob: yes`
`01:54.24`	`p3nguin`	`It's usually a NAT problem that causes no audio.`
`01:54.40`	`Andrew_M`	`voipnoob: companies use firewalls, not NAT or port forwarding. Home routers use NAT, typically.`
`01:54.52`	`p3nguin`	`Companies use NAT, too.`
`01:54.52`	`TehRabbitt`	`p3nguin: sooo, i'm guesssing pastebin the debug again?`
`01:54.54`	`ariel_`	`Andrew_M: not correct`
`01:55.04`	`ariel_`	`basic firewall is a nat server`
`01:55.14`	`p3nguin`	`but a NAT server is not a firewall.`
`01:55.19`	`ariel_`	`correct`
`01:55.25`	`ariel_`	`it's complicated`
`01:55.28`	`voipnoob`	`hmm - in all places i have worked, i think the firewall did NAT. My PC always had in an intranet IP Address`
`01:55.29`	`TehRabbitt`	`try calling again`
`01:55.33`	`voipnoob`	`so I assume there was NAT`
`01:55.47`	`ariel_`	`you can get outbound ports open via most fw`
`01:55.54`	`ariel_`	`that sets up your path`
`01:55.57`	`ariel_`	`for the rtp`
`01:56.25`	`TehRabbitt`	`p3nguin: http://pastebin.com/pAD3yeTR`
`01:56.29`	`TehRabbitt`	`theres the SIP debug`
`01:56.30`	`orioni`	`where to get some infos how the termination transit works ?`
`01:57.42`	`voipnoob`	`ariel_ - so SIP is possible through Firewall, right?`
`01:57.45`	`p3nguin`	`tehrabbitt: That's nothing from my call to you.`
`01:57.51`	`voipnoob`	`how difficult is to get it working?`
`01:57.52`	`ariel_`	`voipnoob: yes`
`01:57.53`	`TehRabbitt`	`thats all that showed... hold on`
`01:57.59`	`TehRabbitt`	`lemme try again dont call yet`
`01:58.01`	`p3nguin`	`tehrabbitt: sip set debug peer 300`
`01:58.06`	`ariel_`	`voipnoob: depends on fw and your know how`
`01:58.29`	`TehRabbitt`	`Go ahead`
`01:58.35`	`voipnoob`	`my fw is Iptables on linux`
`01:58.49`	`ariel_`	`some fw like sonicwalls are bad for voip, for my teast`
`01:58.55`	`ariel_`	`ok`
`01:59.02`	`voipnoob`	`and I have a colleague who is very knowledgable about it - though he doesn't know much about VOIP`
`01:59.12`	`p3nguin`	`voipnoob: For example, a lot of us here run NAT and we almost all use SIP with no troubles. On the other hand, tehrabbitt has been working on it for a week and still can't get calls.`
`01:59.16`	`TehRabbitt`	`p3nguin: this might be more helpful: http://pastebin.com/swHT6rQk`
`01:59.35`	`voipnoob`	`I was told this yesterday also - that it's sonicwall is problem for SIP`
`01:59.53`	`voipnoob`	`not using sonicwall, but just out of curiousity why is sonicwall such a problem`
`02:00.00`	`ariel_`	`sonicwall is overall a problem .`
`02:00.17`	`ariel_`	`they suck in there settings and port triggers`
`02:00.24`	`voipnoob`	`does sonicwall not use some standard Linux IPtable based firewall - is it some propreitary solution?`
`02:00.32`	`ariel_`	`no`
`02:00.47`	`TehRabbitt`	`voipnoob: i think it all comes down to what firewall / NAT you are using...`
`02:00.57`	`p3nguin`	`tehrabbitt: You've got way too many audio codecs configured. Fix it soon, please.`
`02:01.02`	`ariel_`	`IPtables works well`
`02:01.12`	`TehRabbitt`	`p3nguin: how do I fix it lol`
`02:01.15`	`voipnoob`	`tehrabbit - i am using a standard linux box & using iptables to configure stuff`
`02:01.17`	`p3nguin`	`tehrabbitt: Two lines. disallow=all and allow=ulaw`
`02:01.20`	`p3nguin`	`in that order.`
`02:01.22`	`TehRabbitt`	`ah`
`02:01.27`	`p3nguin`	`Take out EVERYTTHING ELSE.`
`02:01.29`	`TehRabbitt`	`not alaw/ulaw?`
`02:01.37`	`TehRabbitt`	`in my sip.conf right?`
`02:01.38`	`p3nguin`	`in the US, we use ulaw.`
`02:01.44`	`TehRabbitt`	`ah`
`02:02.01`	`p3nguin`	`sip.conf general and in every peer.`
`02:02.59`	`p3nguin`	`Make sure my 300 peer has nat=yes and canreinvite=no in it.`
`02:04.59`	`TehRabbitt`	`ok hold on`
`02:05.38`	`TehRabbitt`	`go for it`
`02:06.00`	`*** join/#asterisk micols (~mio@rlogin.dk)`
`02:06.20`	`TehRabbitt`	`http://pastebin.com/ECQBQQrZ`
`02:06.26`	`*** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id)`
`02:07.00`	`TehRabbitt`	`Via: SIP/2.0/UDP 192.168.192.254:5062;rport;branch=z9hG4bKecrwmptz`
`02:07.05`	`TehRabbitt`	`5062???`
`02:07.30`	`p3nguin`	`I should probably change that.`
`02:07.34`	`TehRabbitt`	`Heh`
`02:08.18`	`TehRabbitt`	`could that be why it's not working?`
`02:08.28`	`[TK]D-Fender`	`SonicWall is normally fine. Just disable ALG`
`02:08.31`	`p3nguin`	`no`
`02:08.34`	`TehRabbitt`	`oh lol`
`02:08.46`	`p3nguin`	`That's just my signalling port.`
`02:09.00`	`p3nguin`	`I changed it to 5060 to be conventional.`
`02:09.13`	`TehRabbitt`	`oh lol`
`02:09.23`	`p3nguin`	`I think the problem is your phone.`
`02:09.39`	`p3nguin`	`sccp no rtp`
`02:09.50`	`TehRabbitt`	`hm :-\ how come it works internally then?`
`02:09.57`	`p3nguin`	`I don't know that it does.`
`02:10.11`	`TehRabbitt`	`i can call my dekstop and it goes through / I hear 2 way audio`
`02:10.16`	`TehRabbitt`	`desktop being SIP`
`02:10.19`	`TehRabbitt`	`phone being SCCP`
`02:10.23`	`p3nguin`	`Let me call your SIP phone`
`02:10.31`	`TehRabbitt`	`ok hold on... lemme plug in the headset heh`
`02:11.07`	`TehRabbitt`	`Ext 100`
`02:11.09`	`p3nguin`	`capturing debug?`
`02:11.14`	`TehRabbitt`	`no not yet`
`02:11.22`	`p3nguin`	`say when`
`02:11.33`	`TehRabbitt`	`go`
`02:11.40`	`p3nguin`	`603 Declined`
`02:11.45`	`TehRabbitt`	`??`
`02:11.47`	`TehRabbitt`	`:-\`
`02:12.06`	`p3nguin`	`Hope the debug says what went wrong there.`
`02:12.22`	`TehRabbitt`	`nothing..`
`02:12.32`	`TehRabbitt`	`try dialing 411`
`02:12.36`	`TehRabbitt`	`see if it goes to Goog-411`
`02:12.50`	`p3nguin`	`wait`
`02:13.00`	`p3nguin`	`I was trying to call 100 on my own system.`
`02:13.04`	`TehRabbitt`	`Ohhh`
`02:13.04`	`TehRabbitt`	`lol`
`02:13.29`	`p3nguin`	`411 isn't working well for me.`
`02:13.30`	`TehRabbitt`	`go for it`
`02:13.33`	`TehRabbitt`	`hm`
`02:14.15`	`TehRabbitt`	`http://pastebin.com/hxBQWBXR`
`02:14.19`	`TehRabbitt`	`thats everything i could copy`
`02:16.29`	`p3nguin`	`To me, that looks like a successful call.`
`02:16.54`	`p3nguin`	`Let's try anonymous calls by SIP URI.`
`02:17.14`	`p3nguin`	`In sip.conf, do you have allowguest=yes?`
`02:17.20`	`TehRabbitt`	`lolummm where?`
`02:18.01`	`TehRabbitt`	`under [general]?`
`02:18.24`	`leifmadsen`	`yes`
`02:18.34`	`TehRabbitt`	`it's enabled now`
`02:18.35`	`TehRabbitt`	`go for it`
`02:18.49`	`TehRabbitt`	`Desktop SIP phone == 100, SCCP phone = 500`
`02:19.05`	`TehRabbitt`	`411 == should call out to 1800Goog411`
`02:19.42`	`p3nguin`	`What is your default context?`
`02:19.58`	`TehRabbitt`	`[users]`
`02:20.46`	`p3nguin`	`Hmm, that isn't working.`
`02:20.49`	`TehRabbitt`	`p3nguin: also, with flowroute, do you need to pay those USF fees for each call?`
`02:21.08`	`p3nguin`	`USF is a monthly fee if you have 911 services.`
`02:21.17`	`TehRabbitt`	`ah otherwise you dont have to?`
`02:21.38`	`p3nguin`	`ask them.`
`02:21.52`	`p3nguin`	`VoIP.ms doesn't charge me because I don't have e911 with them.`
`02:22.07`	`TehRabbitt`	`true... VoIP.ms told me my account has to be manually approved :( idk why :(`
`02:22.11`	`TehRabbitt`	`3-4 days`
`02:22.32`	`p3nguin`	`Weird. Mine was instant.`
`02:22.37`	`TehRabbitt`	`yea :-\`
`02:22.38`	`TehRabbitt`	`idk`
`02:22.42`	`p3nguin`	`I can't call in to you via SIP.`
`02:22.54`	`TehRabbitt`	`so does that mean my NAT is still blocking somehow?`
`02:22.56`	`p3nguin`	`Did you sip reload after making changes?`
`02:23.01`	`TehRabbitt`	`yes`
`02:23.04`	`TehRabbitt`	`should i restart asterisk?`
`02:23.17`	`p3nguin`	`You usually don't need to restart asterisk.`
`02:23.23`	`TehRabbitt`	`just restarted it but yea :-\`
`02:23.27`	`p3nguin`	`There are very few components that actually require it.`
`02:23.34`	`TehRabbitt`	`well when you modify sccp you need to lol or was that skinny`
`02:23.35`	`TehRabbitt`	`i forget`
`02:23.49`	`TehRabbitt`	`ringing?`
`02:23.53`	`p3nguin`	`yes`
`02:24.01`	`TehRabbitt`	`anything?`
`02:24.05`	`p3nguin`	`yell`
`02:24.11`	`TehRabbitt`	`i am`
`02:24.16`	`p3nguin`	`I hear nothing.`
`02:24.18`	`p3nguin`	`you?`
`02:24.23`	`TehRabbitt`	`lemme do a txfr to my desktop`
`02:24.24`	`TehRabbitt`	`nothing`
`02:24.54`	`TehRabbitt`	`anything?`
`02:25.53`	`p3nguin`	`I hear nothing.`
`02:26.00`	`TehRabbitt`	`neither do I`
`02:26.13`	`TehRabbitt`	`weird thing is, if you dial in on the 848 number it works...`
`02:26.17`	`p3nguin`	`and I can't call you via SIP, so something is still messed up.`
`02:26.47`	`TehRabbitt`	`like I dont get that... you can call on the 848 DID and it goes through... but you can't directly call in?`
`02:27.09`	`TehRabbitt`	`848 number goes through GV --> Gizmo --> * Server --> SCCP Phone`
`02:27.14`	`p3nguin`	`Dial(SIP/500@thoth.tenehawk.com)`
`02:27.19`	`p3nguin`	`That's what I'm doing.`
`02:27.22`	`TehRabbitt`	`hm...`
`02:27.30`	`TehRabbitt`	`whats it returning?`
`02:27.46`	`p3nguin`	`I get a SIP/2.0 404 Not Found`
`02:27.56`	`TehRabbitt`	`Hm..`
`02:28.02`	`ChannelZ`	`me too`
`02:28.09`	`TehRabbitt`	`71.59.82.2`
`02:28.12`	`TehRabbitt`	`try that`
`02:28.13`	`TehRabbitt`	`as the host`
`02:28.55`	`p3nguin`	`no go`
`02:29.00`	`TehRabbitt`	`try now`
`02:29.06`	`p3nguin`	`same`
`02:29.09`	`TehRabbitt`	`disabled "filter multicast and ping" from my router`
`02:29.09`	`TehRabbitt`	`hm`
`02:29.21`	`TehRabbitt`	`how about... now`
`02:29.27`	`p3nguin`	`nope`
`02:29.33`	`ChannelZ`	`now I'm at least getting no auth`
`02:29.37`	`TehRabbitt`	`wait...`
`02:29.48`	`TehRabbitt`	`Dial(SCCP/500@thoth.tenehawk.com)`
`02:29.56`	`TehRabbitt`	`500 isn't an SIP phone...`
`02:29.57`	`p3nguin`	`Why?`
`02:29.58`	`TehRabbitt`	`or does that not matter`
`02:30.04`	`TehRabbitt`	`500 is the sccp phone`
`02:30.07`	`p3nguin`	`I want to dial your PBX via SIP.`
`02:30.08`	`ChannelZ`	`WTF`
`02:30.09`	`TehRabbitt`	`oh`
`02:30.10`	`TehRabbitt`	`lol`
`02:30.15`	`TehRabbitt`	`confused`
`02:30.22`	`ChannelZ`	`is done helping`
`02:30.27`	`p3nguin`	`Your PBX will call your phone by whatever exten 500 is configured to do.`
`02:30.28`	`TehRabbitt`	`i've completely turned off the SPI firewall 0_o`
`02:30.33`	`TehRabbitt`	`true`
`02:33.26`	`p3nguin`	`I'm thinking that your default context is not actually what you say it is. paste your sip.conf again.`
`02:34.03`	`TehRabbitt`	`hold on...`
`02:34.15`	`TehRabbitt`	`i was just reading the same thing online about the context not matching i'm checking something`
`02:34.30`	`*** join/#asterisk chendy (~chatzilla@204.152.211.137)`
`02:34.48`	`TehRabbitt`	`[users]`
`02:34.48`	`TehRabbitt`	`include => stations`
`02:34.48`	`TehRabbitt`	`include => wireless`
`02:34.48`	`TehRabbitt`	`include => internal`
`02:34.48`	`TehRabbitt`	`include => external`
`02:34.49`	`TehRabbitt`	`include => voicemail`
`02:34.49`	`TehRabbitt`	`include => tollfree`
`02:34.50`	`TehRabbitt`	`include => incoming`
`02:34.53`	`p3nguin`	`stop`
`02:35.22`	`ChannelZ`	`you've wasted what 3 days on this? Post your damn sip.conf and extensions.conf and lets get on to some other topic`
`02:35.26`	`TehRabbitt`	`ok ok`
`02:36.37`	`TehRabbitt`	`extensions.conf: http://pastebin.com/Zzmunnyn`
`02:37.43`	`*** join/#asterisk OrNix (~ornix@178.49.0.149)`
`02:38.07`	`*** join/#asterisk aceio (~c2cbd7f3@gateway/web/freenode/x-plvnornqurogctgd)`
`02:38.50`	`ChannelZ`	`ok I'm not in the room to hear what happened but I'm pretty sure your thing answered, played tt-weasels, and hung up.`
`02:38.51`	`TehRabbitt`	`sip.conf: http://pastebin.com/m05Lp5yJ`
`02:38.53`	`TehRabbitt`	`there you go`
`02:39.02`	`TehRabbitt`	`sorry had to sanatize it (the right way this time) lol`
`02:39.04`	`aceio`	`hi all`
`02:39.16`	`p3nguin`	`context=default`
`02:39.24`	`p3nguin`	`That does not say context users.`
`02:39.32`	`aceio`	`i am trying too setup my isdn2e line with b410p isdn card`
`02:39.43`	`TehRabbitt`	`hm wehre does it say that?`
`02:39.46`	`p3nguin`	`That's why 500 doesn't work via SIP URI... there is no 500 in [default]`
`02:39.50`	`TehRabbitt`	`which one?`
`02:39.53`	`p3nguin`	`line 2`
`02:39.58`	`*** join/#asterisk coppice (~chatzilla@m121-203-236-97.smartone-vodafone.com)`
`02:39.59`	`TehRabbitt`	`sip or extensions i mean`
`02:40.11`	`aceio`	`so far i have plugged in an rj45 cable into one off the ports`
`02:40.22`	`ChannelZ`	`sip.conf`
`02:40.28`	`TehRabbitt`	`got it...`
`02:40.31`	`TehRabbitt`	`hold on lemme reload`
`02:40.38`	`p3nguin`	`If I call s@yourIP, tt-weasels plays.`
`02:40.48`	`aceio`	`but port are flashing red`
`02:40.50`	`TehRabbitt`	`tt-weasels?`
`02:40.52`	`p3nguin`	`So your box DOES accept SIP calls.`
`02:40.59`	`p3nguin`	`exten => s,n,Playback(tt-weasels)`
`02:41.00`	`TJNII`	`And now anyone can bounce calls through his ITSP.....`
`02:41.05`	`TehRabbitt`	`ah lol`
`02:41.13`	`p3nguin`	`He has no ITSP.`
`02:41.23`	`TehRabbitt`	`TJNII: what do you mean anyone can bounce calls through it lol`
`02:41.27`	`TJNII`	`Then what's [futurenine]`
`02:41.33`	`TehRabbitt`	`1800 only lol`
`02:41.39`	`TehRabbitt`	`anyone can use that`
`02:41.46`	`TJNII`	`Yes, any one can.`
`02:41.48`	`TehRabbitt`	`the credentials are on their website`
`02:42.04`	`*** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net)`
`02:42.04`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`02:42.29`	`TehRabbitt`	`TJNII: how would someone bounce a call through though? by signing into my server you mean using ext 300?`
`02:42.35`	`TehRabbitt`	`i'm lost on how someone would do that`
`02:42.50`	`aceio`	`i am now getting a steady light on port one but the light is still red`
`02:42.56`	`TehRabbitt`	`[May 4 22:43:28] NOTICE[1529]: chan_sip.c:15195 handle_request_invite: Call from '' to extension 's' rejected because extension not found.`
`02:43.06`	`TJNII`	`You made users the default and are allowing unauthed access.`
`02:43.13`	`TJNII`	`users contains everything.`
`02:43.30`	`aceio`	`any help would be great`
`02:47.41`	`aceio`	`any one worked with b410p isdn card`
`02:49.56`	`*** join/#asterisk lepine (~leprechau@69-196-161-147.dsl.teksavvy.com)`
`02:56.47`	`ManxPower`	`TJNII, So many unsecure PBX's, so many many people happy to take advantage of them.`
`02:56.58`	`TJNII`	`Indeed.`
`02:57.28`	`TJNII`	`I had someone try and all london through mine once. I still have no idea how he got in.`
`02:57.49`	`TJNII`	`But my contexts were modified, passwords changed, and firewall reinforced so it never happened again.`
`03:04.42`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`03:07.20`	`ChannelZ`	`oh, the quiet.`
`03:08.44`	`p3nguin`	`Oh, the huge manatee!`
`03:14.57`	`*** join/#asterisk DarkRift (~dark@modemcable015.68-200-24.mc.videotron.ca)`
`03:20.48`	`ChannelZ`	`kicks courier-imap to the curb`
`03:21.40`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91)`
`03:35.05`	`*** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net)`
`03:37.09`	`Slugs_`	`MY ASTERISK STOPPED WORKING AFTER A FEW DOZEN ANTS AND FRESH WATER TURTLE FOUND ITS WAY INTO THE CASE!!!!! PLZ HELP`
`03:38.06`	`coppice`	`that sounds like a bug problem`
`03:38.10`	`Slugs_`	`ChannelZ plz yhx`
`03:38.22`	`*** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)`
`03:38.27`	`*** part/#asterisk ManxPower (~manxpower@190.sub-75-216-186.myvzw.com)`
`03:38.50`	`Slugs_`	`burns coppice`
`03:39.41`	`coppice`	`yeah? what are you gonna use? A Pentium P4?`
`03:40.08`	`Slugs_`	`old dell battery`
`03:40.17`	`coppice`	`oh, nasty`
`03:41.07`	`Slugs_`	`yeah ;)`
`03:41.17`	`Slugs_`	`coppice whats ur name mean?`
`03:41.45`	`Slugs_`	`alli found was something about bushes on wiki ;)`
`03:42.04`	`coppice`	`yeah, it means something about bushes`
`03:42.51`	`coppice`	`or something roughly equivalent to my surname`
`03:43.02`	`Slugs_`	`lol`
`03:46.19`	`*** join/#asterisk cesar_CR (~cesar@201.201.41.242)`
`03:48.27`	`*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)`
`03:48.32`	`*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-reoqoqeftprpxixk)`
`03:53.00`	`*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)`
`03:54.10`	`*** join/#asterisk ManxPower (~manxpower@190.sub-75-216-186.myvzw.com)`
`03:59.21`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91)`
`04:00.29`	`boodu`	`goodbye`
`04:15.53`	`*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)`
`04:18.44`	`ChannelZ`	`Slugs_: Humm, maybe try put the fudge on CPU to soke up waters?`
`04:24.30`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91)`
`04:28.10`	`*** join/#asterisk fofware (~fabian@190.7.25.160)`
`04:29.15`	`*** join/#asterisk geoffmcc123 (~Geoff@cpe-72-231-200-14.buffalo.res.rr.com)`
`04:30.28`	`geoffmcc123`	`hi guys/gals - maybe a dumb question, but should i have my phones behind firewall. I am using APF and am trying to decide if i should just trust my one pc or the whole local range`
`04:31.18`	`carrar`	`Would you put PC outside your network?`
`04:31.51`	`geoffmcc123`	`?`
`04:32.21`	`carrar`	`Yes your phones should be behind a firewall`
`04:32.22`	`geoffmcc123`	`meaning would there be connection to internet?`
`04:33.05`	`geoffmcc123`	`thats how i had it setup but then got to thinking if that is what should do since i am just a home user and no one really plugs into my chunk of the network`
`04:34.13`	`carrar`	`So your home IP is designated as a no hacking IP and people will leave it alone?`
`04:34.20`	`carrar`	`awesome`
`04:34.23`	`carrar`	`where do I sign up`
`04:35.27`	`*** join/#asterisk voipnoob (shiva@detroit.slack.net)`
`04:36.18`	`geoffmcc123`	`lol, no i just meant i know no one could plug in without me knowing so i was thinking would be safe to trust local network`
`04:36.39`	`geoffmcc123`	`i guess i shoulda known better`
`04:37.10`	`voipnoob`	`Hi - for a single central Asterisk PBX server serving 1000 different users (most of them at different locations using different ISPs), 150 concurrent calls maximum, only Voice - no video - what kind of hardware requirements should I be looking at?`
`04:37.33`	`voipnoob`	`for the server running Asterisk`
`04:37.41`	`geoffmcc123`	`i am just worried about having vsftpd open on the firewall but would like to use it for phones that are in the building, so i thought if i closed it but trusted the phones`
`04:38.29`	`*** join/#asterisk arnotixe (~arnotixe@190.131.124.20)`
`04:39.31`	`voipnoob`	`anyone?`
`04:42.11`	`voipnoob`	`For a single central Asterisk server serving 1000 diff users (most of them at distinct different locations using different ISPs) - 150 concurrent calls maximum -`
`04:42.17`	`voipnoob`	`only voice, no video`
`04:42.30`	`voipnoob`	`what kind of hardware requirements are there for running the asterisk server?`
`04:43.02`	`arnotixe`	`hi all I have a strange problem that I can't figure out: My asterisk server is behind a NAT router, pointing 5060 and 10000-10100 to the server. The clients are out on the net behind a second NAT. Now, SIP signalling works, the channel is taken up and down correctly, but my server sends RTP packets to the wrong address: 10.0.0.3 (which is the client's local ip of course). I sniff this with wireshark. any idea why nat=yes is not respected on these RTP`
`04:43.02`	`arnotixe`	`<PROTECTED>`
`04:43.24`	`[TK]D-Fender`	`~sipnat`
`04:43.25`	`infobot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`04:43.26`	`[TK]D-Fender`	`^^^^`
`04:45.40`	`*** join/#asterisk sourcode (~code@ppp-58-8-238-110.revip2.asianet.co.th)`
`04:49.03`	`*** join/#asterisk joobie (~joobie@mx01.anric.com.au)`
`04:49.22`	`lepine`	`is there a way to trace all contexts through which a incoming call passes through?`
`04:49.34`	`lepine`	`My dialplan isn't working and I can't figure it out.`
`04:50.47`	`joobie`	`guys im setting up a boot server for my polycom 320.. trying to set the "Number Of Line Keys" field.. ive used something like "<userinfo><reg reg.1.displayName="4000" .. reg.1.lineKeys="2"/></userinfo>" .. however the Number of Line Keys is blank in the web config (and the phone still has only 1 line key available`
`04:50.50`	`joobie`	`any ideas?`
`04:51.01`	`Slugs_`	`what logging are u doing lepine`
`04:51.30`	`[TK]D-Fender`	`joobie: your web config overrides the provisioning`
`04:51.36`	`[TK]D-Fender`	`joobie: you need to trash it`
`04:51.37`	`lepine`	`everything, I believe ... current verbosity is set at 11. I dont see a verb. difference from 5 though`
`04:51.53`	`[TK]D-Fender`	`lepine: We don't see the difference either...`
`04:58.17`	`*** join/#asterisk chendy_ (~chatzilla@204.152.211.137)`
`04:59.34`	`*** join/#asterisk ChannelZ (~bobm@burner.com)`
`05:01.13`	`arnotixe`	`[TK]D-Fender, thanks I have all that setup, but the asterisk server still sends RTP to 10.0.0.3 instead of 85.x.x.x. Could it be the remote side has trouble with STUN or something? the asterisk debug says "Peer audio RTP is at port 10.0.0.3:16446". Since asterisk does NAT-translate the SIP traffic, I find it odd it doesn't NAT-translate the RTP traffic?`
`05:04.48`	`[TK]D-Fender`	`arnotixe: I'm not seeing debug and configs...`
`05:04.56`	`[TK]D-Fender`	`~pb`
`05:04.57`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`05:05.13`	`joobie`	`TK, how do i trash the web config?`
`05:06.01`	`*** join/#asterisk Corydon76-dig (eleven@c-69-137-80-31.hsd1.tn.comcast.net)`
`05:06.01`	`*** mode/#asterisk [+o Corydon76-dig] by ChanServ`
`05:06.34`	`arnotixe`	`[TK]D-Fender, arrgh / all right! I had put my public hostname arno.homelinux.org in /etc/hosts with the local IP (trying to be smart and make local web traffic on the server just go inside the server)... Removed that line and voilÃ¡ everything works. Sorry to bug you with this non-asterisk problem...`
`05:07.12`	`arnotixe`	`well it became a problem hehe`
`05:07.58`	`Slugs_`	`.`
`05:14.31`	`joobie`	`lol @ Slugs's quit msg`
`05:14.38`	`voipnoob`	`I was reading this -> http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`05:15.11`	`voipnoob`	`there are those 9 options there`
`05:15.44`	`voipnoob`	`what about a different option - Asterisk as SIP Server outside on the internet, different clients each of the behind their own NAT+Firewall`
`05:15.50`	`voipnoob`	`is that possible?`
`05:15.50`	`*** join/#asterisk nickw (~nickw@c-76-111-107-117.hsd1.md.comcast.net)`
`05:16.22`	`[TK]D-Fender`	`voipnoob: Yes. Try reading the FIRST guide linked`
`05:18.36`	`voipnoob`	`ok - thanks - that has the info`
`05:20.40`	`voipnoob`	`DFender - for a single central Asterisk PBX server serving 1000 different users`
`05:20.40`	`voipnoob`	`most of the 1000 users at different locations using different ISPs - 150 concurrent calls maximum, only Voice - no video -`
`05:20.40`	`voipnoob`	`what kind of hardware requirements should I be looking atfor the server running Asterisk`
`05:21.16`	`[TK]D-Fender`	`voipnoob: Quad-core, 4 gig minimum, raid5+`
`05:21.30`	`voipnoob`	`ok, thanks`
`05:22.13`	`voipnoob`	`and what is the maximum number of concurrent calls this will provide (i.e. my current req is 150, but i want what's the max for this configuration)`
`05:22.23`	`voipnoob`	`i want to know i mean`
`05:22.26`	`joobie`	`TK, is "Reset Local Config" on http://www.sonoracomm.com/support/20-voice-support/70-polycom sufficient to wipe the web config?`
`05:24.55`	`joobie`	`.. if so, that didnt work`
`05:25.32`	`joobie`	`heck, ill do "format filesystem" now that i have the boot stuff going`
`05:26.22`	`[TK]D-Fender`	`checkout time. Later all`
`05:27.38`	`p3nguin`	`~itsp-us`
`05:27.44`	`p3nguin`	`~itsp`
`05:27.44`	`infobot`	`[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.`
`05:27.45`	`*** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.pa.comcast.net)`
`05:27.51`	`p3nguin`	`~itsplist-us`
`05:27.52`	`infobot`	`Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com`
`05:28.09`	`*** join/#asterisk xuser (~xuser@unaffiliated/xuser)`
`05:36.09`	`p3nguin`	`http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=280501905217`
`05:38.42`	`ChannelZ`	`doesn't look very skinny to me..`
`05:38.46`	`p3nguin`	`http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=310215794095`
`05:51.38`	`*** join/#asterisk oej (~olle@ns.webway.se)`
`05:52.12`	`TehRabbitt`	`http://cgi.ebay.com/NEW-CISCO-7914-7915-7916-7916G-Color-Expansion-Module-/220596838053?cmd=ViewItem&pt=LH_DefaultDomain_2&hash=item335c989ea5#ht_3088wt_1165`
`05:56.23`	`enyawix`	`p3nguin what about fax?`
`05:56.38`	`p3nguin`	`enyawix: What about it?`
`05:56.49`	`ChannelZ`	`It needs to die die die`
`05:57.22`	`lepine`	`~itsplist-ca`
`05:57.23`	`infobot`	`[~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca`
`05:57.27`	`enyawix`	`how do you go about it? i am using an analog machine with ATA device`
`05:58.02`	`enyawix`	`p3nguin faxing is a crap shoot at bast for me at this point`
`06:01.09`	`p3nguin`	`http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps8537/prod_white_paper0900aecd806fa57a.html`
`06:05.50`	`*** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com)`
`06:06.46`	`*** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de)`
`06:06.54`	`kruemeltee`	`hey all :-)`
`06:09.54`	`*** join/#asterisk gelo (~gelo@mx01.quobis.com)`
`06:10.58`	`coppice`	`enyawix: which ATA?`
`06:12.33`	`kruemeltee`	`it seems as if my asterisk uses a wrong Request-URI for sending INVITE to a connected telephone system ... how am I able to set the correct Request-URI? (I successfully tried "fromuser" within sip.conf but after that the CallerID does not come through)`
`06:13.23`	`kruemeltee`	`so ... how to set the Request-URI correctly?`
`06:18.01`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`06:23.07`	`*** join/#asterisk MetaMucil (~Omeras@99-2-200-244.lightspeed.milwwi.sbcglobal.net)`
`06:30.13`	`*** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com)`
`06:33.46`	`*** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110)`
`06:38.06`	`*** join/#asterisk oej (~olle@ns.webway.se)`
`06:39.28`	`*** join/#asterisk Da-Geek (~Da-Geek@213-193-154-130.perm.easynet.be)`
`06:40.15`	`TehRabbitt`	`http://freeplaymusic.com/search/category_search.php?t=s&i=48`
`06:44.09`	`coppice`	`maybe the "Feeling" entries should be replaced with "Soulless formula" entries`
`06:48.48`	`*** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no)`
`06:51.08`	`TehRabbitt`	`http://www.cisco.com/cgi-bin/tablebuild.pl/callmgr-33`
`06:54.30`	`TehRabbitt`	`ciscocm-MusicOnHoldSampler`
`06:54.53`	`TehRabbitt`	`ciscocm-MusicOnHoldSampler.3.1.zip`
`06:55.56`	`TehRabbitt`	`ciscocm-MusicOnHold.3-1.zip`
`06:57.34`	`TehRabbitt`	`http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/upgrade/3_1/upgra312.html`
`07:03.13`	`TehRabbitt`	`anyone know where i can find those Cisco Music on Hold samples?`
`07:04.25`	`ChannelZ`	`GIYF`
`07:07.22`	`TehRabbitt`	`ChanServ: huh?`
`07:07.39`	`TehRabbitt`	`ChannelZ: huh?`
`07:07.48`	`ChannelZ`	`Google Is Your Friend`
`07:07.51`	`TehRabbitt`	`can't find it`
`07:07.57`	`TehRabbitt`	`not on the cisco site`
`07:09.20`	`*** join/#asterisk oej (~olle@ns.webway.se)`
`07:11.30`	`TehRabbitt`	`found it... all 300MB of it 0_o`
`07:13.45`	`p3nguin`	`link?`
`07:14.08`	`TehRabbitt`	`it requires a SmartNET contract I think... or so it worked for me hold on`
`07:14.16`	`ChannelZ`	`fucking cisco`
`07:14.26`	`ChannelZ`	`Their website is the suck of all suck`
`07:14.34`	`TehRabbitt`	`lol`
`07:14.37`	`TehRabbitt`	`it was hard to find`
`07:14.39`	`p3nguin`	`Yeah, it's usually broken.`
`07:14.46`	`TehRabbitt`	`http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=/cisco/voice/callmgr/3.3/ciscocm-MusicOnHold.3-1.exe`
`07:14.50`	`TehRabbitt`	`Full set of music and prompts for Cisco CallManager Music on Hold`
`07:14.51`	`ChannelZ`	`Actually, I take that back. GoDaddy's site is the suck of all suck. Cisco is 2nd`
`07:15.01`	`TehRabbitt`	`ChannelZ: agreed`
`07:15.01`	`TehRabbitt`	`lol`
`07:15.11`	`ChannelZ`	`Have to login and click 10 different links just to download some firmware`
`07:15.19`	`*** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net)`
`07:15.26`	`p3nguin`	`I logged in and it told me I needed to login.`
`07:15.37`	`TehRabbitt`	`hahahaha do you haz smartnet :-p`
`07:15.39`	`TehRabbitt`	`thats why :(`
`07:15.51`	`p3nguin`	`nope`
`07:15.57`	`p3nguin`	`Download not permitted (Do not bookmark this URL.)`
`07:16.00`	`TehRabbitt`	`if you dont have smartnet it wont let you download`
`07:16.01`	`TehRabbitt`	`lol`
`07:17.11`	`ChannelZ`	`I logged in and it told me I had to reset my login to login.`
`07:17.18`	`ChannelZ`	`"We made it more good!"`
`07:17.21`	`p3nguin`	`which requires you to login.`
`07:17.24`	`TehRabbitt`	`LOL`
`07:17.31`	`TehRabbitt`	`50% downloaded heh`
`07:17.34`	`TehRabbitt`	`300 MB file :-\`
`07:17.42`	`TehRabbitt`	`better be some good MoH`
`07:17.43`	`TehRabbitt`	`lol`
`07:17.49`	`ChannelZ`	`oooh it's asking me my security questions`
`07:18.25`	`ChannelZ`	`well, it was. Now it's stuck.`
`07:18.28`	`TehRabbitt`	`best ever; http://www.bit-tech.net/news/gaming/2010/04/15/gamestation-we-own-your-soul/1`
`07:19.39`	`ChannelZ`	`hmm I thought that was Apple`
`07:19.45`	`TehRabbitt`	`haahahahhahaa`
`07:19.48`	`TehRabbitt`	`might as well be`
`07:19.59`	`*** join/#asterisk voipnoob (shiva@detroit.slack.net)`
`07:20.03`	`TehRabbitt`	`so once i get this MoH downloaded, who wants to help me fix my MoH issue?`
`07:20.05`	`TehRabbitt`	`lmao`
`07:20.10`	`TehRabbitt`	`"File not found"`
`07:20.14`	`TehRabbitt`	`even though it's there... in WAV format`
`07:20.41`	`TehRabbitt`	`btw whats the status on this: http://www.binrev.com/forums/index.php/topic/36867-ma-bell-pat-fleet-agrees-to-do-asterisk-prompt-set/`
`07:21.51`	`voipnoob`	`I have a question - i am confused about the term "SIP Proxy" - is that the same as SIP server? or is it different?`
`07:25.23`	`TehRabbitt`	`did "ma bell" ever record anything for asterisk?`
`07:26.58`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`07:27.13`	`ChannelZ`	`http://www.flowroute.com/services/voice/`
`07:27.51`	`*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)`
`07:28.12`	`TehRabbitt`	`Dude..... talk about NOC`
`07:28.12`	`TehRabbitt`	`http://www.youtube.com/watch?v=Ud1ysN9bS_0&feature=related`
`07:28.36`	`ChannelZ`	`http://pbxinaflash.net/source/sounds/PatFleet-asterisk-ulaw-1.0.tar.gz`
`07:28.39`	`TehRabbitt`	`Roffle @ :14`
`07:32.03`	`*** join/#asterisk TimeRider (steve@5ac318a3.bb.sky.com)`
`07:37.02`	`ChannelZ`	`wow Pat is so much nicer to listen to`
`07:37.13`	`TehRabbitt`	`lol yep`
`07:37.14`	`TehRabbitt`	`http://www.youtube.com/watch?v=HR1OKBdLK0w&feature=related`
`07:37.34`	`TehRabbitt`	`ChannelZ: thanks for the dl lol`
`07:37.49`	`ChannelZ`	`it was in that thread YOU posted`
`07:38.00`	`TehRabbitt`	`oh haha`
`07:39.01`	`TehRabbitt`	`i like the nixietube displays that the "new" systems use in that vid`
`07:39.56`	`coppice`	`ah, the good old days, when calculators had nicely rounded numbers :-)`
`07:40.30`	`TehRabbitt`	`http://www.youtube.com/watch?v=sz9oiwK3wNo&NR=1`
`07:40.36`	`TehRabbitt`	`now THAT is a relic 0_o`
`07:40.45`	`*** join/#asterisk c0rnoTa (~c0rnoTa@94.25.173.118)`
`07:41.17`	`TehRabbitt`	`p3nguin: this is what I was telling you about earlier: http://www.youtube.com/watch?v=hiei-3YADYY&feature=related`
`07:41.23`	`TehRabbitt`	`Microwave stations like the one in NJ here`
`07:44.06`	`coppice`	`I used to use the world's first ATM machine, and I only found it was the first 25 years later when there was a TV report about it`
`07:46.50`	`*** join/#asterisk tamiel (~tamiel@213.30.183.226)`
`07:47.33`	`TehRabbitt`	`knock knock...`
`07:47.33`	`TehRabbitt`	`http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=new+brunswick+nj+college+ave&sll=40.478162,-74.435978&sspn=0.100412,0.154324&ie=UTF8&hq=&hnear=College+Ave,+New+Brunswick,+Middlesex,+New+Jersey+08901&ll=40.49481,-74.442748&spn=0.001577,0.002411&t=h&z=19&layer=c&cbll=40.494546,-74.44271&panoid=MzFczqRgUhxCIKxT4dU_rQ&cbp=12,304.46,,0,-32.94`
`07:47.34`	`TehRabbitt`	`lmfao`
`07:47.50`	`TehRabbitt`	`thats one of the main call routing buildings in NJ apparently`
`07:47.59`	`TehRabbitt`	`few dozen microwave relay towers on top of it`
`07:48.55`	`coppice`	`all those city centre telecoms towers which are now just relics......`
`07:49.08`	`TehRabbitt`	`yep lol`
`07:49.29`	`TehRabbitt`	`that tower is proabbbly mostly VoIP now 0_o with some copper`
`07:49.30`	`TehRabbitt`	`lol`
`07:50.12`	`coppice`	`why would it still be in use at all?`
`07:50.33`	`TehRabbitt`	`it's one of the main central CO's for NJ`
`07:50.34`	`TehRabbitt`	`lol`
`07:50.43`	`TehRabbitt`	`almost all calls in the 732 area code go through there`
`07:51.01`	`coppice`	`but the tower part?`
`07:51.54`	`TehRabbitt`	`Ah, the tower afaik from when I talked to a tech who works there one day, is only used as a backbone to NYC nowadays... that and staten Island... other than that nothing else goes over it but voice to / from NYC and staten island`
`07:52.29`	`TehRabbitt`	`mind you there's another tower that is used for that in fort-lee NJ as well`
`07:52.44`	`coppice`	`many of those city centre towers are incredibly strong. they were designed to stay up with a nuclear blast a few km away`
`07:53.07`	`TehRabbitt`	`yep hence why they are still used for major links...`
`07:54.19`	`TehRabbitt`	`http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=new+brunswick+nj+college+ave&sll=40.478162,-74.435978&sspn=0.100412,0.154324&ie=UTF8&hq=&hnear=College+Ave,+New+Brunswick,+Middlesex,+New+Jersey+08901&ll=40.49441,-74.443721&spn=0.001577,0.002411&t=h&z=19&layer=c&cbll=40.494394,-74.443841&panoid=O5bBSTVKRWy00o6JEZzTmQ&cbp=12,62.99,,0,-29.45`
`07:54.28`	`TehRabbitt`	`thats a better pic showing some of the feed horns on it`
`07:56.37`	`coppice`	`and a label says "fallout shelter" :-)`
`07:56.37`	`TehRabbitt`	`lol there Was a fallout shelter there... converted to a comic shop lmao thats the funny part`
`07:56.44`	`TehRabbitt`	`even better pic`
`07:56.44`	`TehRabbitt`	`http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=new+brunswick+nj+college+ave&sll=40.478162,-74.435978&sspn=0.100412,0.154324&ie=UTF8&hq=&hnear=College+Ave,+New+Brunswick,+Middlesex,+New+Jersey+08901&ll=40.494357,-74.444094&spn=0.001577,0.002411&t=h&z=19&layer=c&cbll=40.494342,-74.44421&panoid=ANTn2r9S67cVaqj4chrlBA&cbp=12,68.71,,2,-18.17`
`07:57.10`	`TehRabbitt`	`http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=new+brunswick+nj+college+ave&sll=40.478162,-74.435978&sspn=0.100412,0.154324&ie=UTF8&hq=&hnear=College+Ave,+New+Brunswick,+Middlesex,+New+Jersey+08901&ll=40.494357,-74.444094&spn=0.001577,0.002411&t=h&z=19&layer=c&cbll=40.494342,-74.44421&panoid=ANTn2r9S67cVaqj4chrlBA&cbp=12,67.48,,0,-18.02`
`07:57.13`	`TehRabbitt`	`and that one is unzoomed`
`07:57.20`	`TehRabbitt`	`to give an idea how big those things are lol`
`07:57.55`	`coppice`	`the fallout shelter next to the telecoms tower in central london was turned into a document store, so after a nuclear war there will still be no excuse for some people not to file their tax returns`
`07:58.16`	`TehRabbitt`	`LOL wow....`
`07:58.34`	`*** part/#asterisk voipnoob (shiva@detroit.slack.net)`
`07:59.31`	`TehRabbitt`	`"All of the outside openings were able to be sealed with steal blast deflectors using a pneumatic control panel located in the control room. There was a five minute timer, presumably to allow the HVAC units to be secured before the doors where closed. They where heavy gauge steel shutter designed to deflect the pressure wave of a nuclear explosion. This microwave relay site would not have withstood a direct hit from a nuclear warhead`
`07:59.31`	`TehRabbitt`	`"`
`07:59.41`	`TehRabbitt`	`0_o`
`08:00.25`	`coppice`	`a direct hit :-)`
`08:02.07`	`TehRabbitt`	`heh`
`08:02.08`	`TehRabbitt`	`All of this telephone traffic was transmitted on digital data channels un-encrypted. Many have argued that this allowed the government (most notably the NSA or National Security Agency) to intercept and listen to most domestic long distance telephone calls within the US. There is a book called Puzzle palace by James Bamford if you are interested in NSA history. It was written more than 20 years ago, so it doesnt really apply today,`
`08:03.22`	`*** join/#asterisk arnotixe (~arnotixe@190.131.124.20)`
`08:04.14`	`coppice`	`in london military telephony traffic used to go unencrypted through ducts that run along two sides of the USSR's embassy compound :-)`
`08:04.27`	`TehRabbitt`	`LOL`
`08:07.58`	`TehRabbitt`	`http://survivalrealty.com/images/kayentabunker/img1.jpg`
`08:08.08`	`TehRabbitt`	`lmao zombie apocolypse here we come`
`08:08.09`	`TehRabbitt`	`jk`
`08:08.58`	`TehRabbitt`	`http://www.survivalrealty.com/`
`08:10.32`	`creativx`	`"Step outside the front door and you can fire a pistol, throw knives, strip naked, whatever!"`
`08:10.43`	`ChannelZ`	`I'll take it!`
`08:11.31`	`creativx`	`lots of cheap land there`
`08:12.39`	`*** join/#asterisk soman (~somnath@stargate.starnet.fi)`
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`08:12.56`	`creativx`	`lots of funny properties there`
`08:12.58`	`TehRabbitt`	`heh`
`08:13.04`	`TehRabbitt`	`the cave house used to be on there`
`08:13.06`	`TehRabbitt`	`1 Million`
`08:13.08`	`TehRabbitt`	`inside a cave`
`08:13.29`	`TehRabbitt`	`nvm it's gone up since then`
`08:13.29`	`TehRabbitt`	`http://thecavehouse.com/`
`08:13.32`	`TehRabbitt`	`lol`
`08:17.25`	`*** part/#asterisk IronChef (~alloy@tropyx.com)`
`08:18.06`	`xheliox`	`yawns`
`08:20.15`	`TehRabbitt`	`how do u do text like that btw`
`08:20.34`	`xheliox`	`/me says /me`
`08:20.44`	`TehRabbitt`	`ah lol`
`08:20.51`	`TehRabbitt`	`says thanks :-p`
`08:20.59`	`daya`	`Any help, I setup asterisk in public IP, I have 2 pap devices behind the nat in two places connecting the asterisk server, the two can dial each other and ring tone came but no any voice`
`08:23.11`	`*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)`
`08:24.02`	`daya`	`my sip.conf setting is http://pastebin.com/m4GkRG6s`
`08:37.42`	`ChannelZ`	`~sipnat`
`08:37.43`	`infobot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`08:37.52`	`ChannelZ`	`going to bed`
`08:38.12`	`xheliox`	`goodnight`
`08:40.06`	`daya`	`ChannelZ, doesn't work on me`
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`09:10.57`	`Jumpie`	`wow..could not connect on port 6667 for the life of me`
`09:11.14`	`xheliox`	`Well here you are now.`
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`09:47.49`	`evangelion`	`can i use dialplan variables for the dial variable in sip.conf`
`09:47.51`	`evangelion`	`?`
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`10:04.38`	`gelo`	`evangelion: what do you mean with "dial variable"?`
`10:05.53`	`evangelion`	`gelo: http://pastebin.com/YMv4dL6D`
`10:05.57`	`*** join/#asterisk Z_God (~julius@wlan235212.mobiel.utwente.nl)`
`10:06.17`	`evangelion`	`please look at line 13`
`10:09.30`	`*** join/#asterisk Jumpie (~lah@ip68-98-29-177.ph.ph.cox.net)`
`10:09.39`	`Jumpie`	`why the hell am i disconnecting every few minutes`
`10:09.44`	`Jumpie`	`server unstable toniht`
`10:10.26`	`gelo`	`evangelion: I didn't even know that option existed`
`10:11.25`	`gelo`	`is it documented anywhere?`
`10:11.28`	`evangelion`	`gelo: neither i`
`10:12.23`	`evangelion`	`gelo: i found it in a preconfigured sip.conf ( we're currently using trixbox ) ! so i asked here`
`10:12.32`	`garymc`	`Anyone got an IPSec VPN setup on thier asterisk box?`
`10:12.44`	`*** join/#asterisk Tim_Toady (~moi@193.92.246.150.dsl.dyn.forthnet.gr)`
`10:13.43`	`evangelion`	`gelo: maybe it is used by third party AGI`
`10:14.43`	`gelo`	`evangelion: yeah, it must be some trixbox's specific thing`
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`10:44.12`	`jkroon`	`hi all, i'm trying to get T.38 working, but am running into "Unsupported SDP media type in offer: image 20030 udptl t38" warning just before the call drops (the user just goes away)`
`10:44.16`	`jkroon`	`any ideas?`
`10:45.23`	`*** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net)`
`10:47.40`	`coppice`	`try enabling T.38 support`
`10:48.29`	`*** join/#asterisk Jumpie (~lah@ip68-98-29-177.ph.ph.cox.net)`
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`10:53.07`	`jkroon`	`coppice, i'm trying. but clearly I'm missing something.`
`10:56.02`	`coppice`	`look in sip.conf`
`10:56.59`	`jkroon`	`http://pastebin.co.za/16109 <-- says T.38 support : Yes`
`11:00.19`	`*** join/#asterisk hipitihop (~denis@203.132.229.236)`
`11:02.37`	`hipitihop`	`Is there a reson why when I call a busy DID via sip why I don't hear a busy signal ? I get a circuit is busy in the console then this "[May 5 20:57:51] WARNING[4308]: pbx.c:4404 __ast_pbx_run: Don't know what to do with 'SIP/LinksysPAP-abbd2c18'"`
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`12:14.08`	`devmod`	`through the CLI, can I do some sort of AgentLogin?`
`12:16.03`	`*** join/#asterisk BugKhaM (~BugKhaM@125.25.12.116.adsl.dynamic.totbb.net)`
`12:16.52`	`BugKhaM`	`I'm using zaptel without libpri with ISDN-PRI and it works`
`12:16.59`	`BugKhaM`	`is this strange?`
`12:17.30`	`xheliox`	`mmm, yes?`
`12:18.17`	`[TK]D-Fender`	`devmod: AMI`
`12:18.39`	`*** join/#asterisk Andrew_M_ (~1867ea4a@gateway/web/freenode/x-cninircexucibjjy)`
`12:19.07`	`tzafrir_laptop`	`BugKhaM, what version of asterisk?`
`12:19.28`	`devmod`	`[TK]D-Fender, I can't do that through the CLI directly tho right? I will need to write some sort of script to connect to asterisk? was just looking for a quick AgentLogin test`
`12:19.30`	`tzafrir_laptop`	`Do you use chan_zap / chan_dahdi? A different channel driver?`
`12:19.30`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.78)`
`12:21.57`	`BugKhaM`	`tzafrir_laptop: it's 1.2.37`
`12:22.01`	`[TK]D-Fender`	`devmod: then just call an extension`
`12:22.26`	`BugKhaM`	`tzafrir_laptop: I am having a bit of problems though`
`12:23.00`	`tzafrir_laptop`	`BugKhaM, again: what channel driver do you use?`
`12:23.21`	`tzafrir_laptop`	`If you don't know: in what asterisk configu file did you need to configure that ISDN device?`
`12:23.28`	`BugKhaM`	`tzafrir_laptop: wcte11xp,tor2`
`12:23.32`	`tzafrir_laptop`	`(And what device is it)`
`12:23.45`	`tzafrir_laptop`	`So those are Zaptel, indeed`
`12:24.07`	`BugKhaM`	`tzafrir_laptop: Tormenta II and TE110`
`12:24.10`	`tzafrir_laptop`	`Without libpri, I guess you just don't have ISDN support in chan_zap?`
`12:24.56`	`devmod`	`[TK]D-Fender, haha I know just trying to avoid that bc of other users i need to remotely activate to test some stuff. So its either extension or AMI , right?`
`12:25.08`	`BugKhaM`	`tzafrir_laptop: the alarms are not shown correctly when TE110 is loaded before Tormenta II`
`12:25.21`	`BugKhaM`	`the alarms shown as OK for all channels`
`12:26.47`	`BugKhaM`	`I had to restart zaptel until the Tormenta II is started before TE110 to get correct alarms, how does this sound?`
`12:27.19`	`BugKhaM`	`<PROTECTED>`
`12:27.25`	`[TK]D-Fender`	`devmod: go cruise CLI for help. I fail to see how this need even involve any other user.`
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`12:27.36`	`jkroon`	`coppice, turns out grandstream's HT-502 doesn't like me to initiate T38 - it has to initiate. it's still hanging up though.`
`12:27.55`	`jkroon`	`anybody here that's reasonably familiar with t38?`
`12:28.08`	`BugKhaM`	`tzafrir_laptop: anyway, I just install libpri and see how it goes`
`12:29.30`	`*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)`
`12:29.47`	`jkroon`	`trying to get it working under ast 1.6.1.18 with no luck so far, tcpdump shows traffic going in both directions and udptl set debug on also confirms the fact that packets are moving in both directions, however, the call gets terminated with "app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.", and it seems the HT is the one initiating the hangup with BYE`
`12:33.26`	`devmod`	`[TK]D-Fender, all right thanks`
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`12:35.27`	`tzafrir_laptop`	`BugKhaM, hmm... I don't see how libpri support is related to reporting of alarms`
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`12:35.34`	`tzafrir_laptop`	`What version of Zaptel is it?`
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`12:39.33`	`BugKhaM`	`tzafrir_laptop: it's 1.2.27 and now my server stopped working`
`12:40.06`	`BugKhaM`	`tzafrir_laptop: after reinstall zaptel 1.2.27 , libpri 1.2.8 and asterisk 1.2.37`
`12:40.33`	`BugKhaM`	`tzafrir_laptop: can't we install while they're loaded?`
`12:42.08`	`BugKhaM`	`Message from syslogd@ at Wed May 5 19:35:35 2010 ...`
`12:42.08`	`BugKhaM`	`ip-serv-10 kernel: EIP: 0060:[<f8bb1268>] Not tainted VLI`
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`12:43.15`	`devmod`	`Is this wrong? bc it isnt working.. exten => 911,n,SetVar(QUEUE_PRIO=10) I put it right before Queue...`
`12:44.22`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`12:49.41`	`tzafrir_laptop`	`BugKhaM, I must point out that you use a rather old version of Zaptel. It has not been actively maintained for quite some time`
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`12:51.53`	`BugKhaM`	`tzafrir_laptop: yeah, and after installing libpri, the system doesn't work anymore`
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`12:52.47`	`tzafrir_laptop`	`BugKhaM, do you have a more complete trace in the kernel logs? (/var/log/kern.log , /var/log/messages, whatever)`
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`12:54.11`	`BugKhaM`	`tzafrir_laptop: can't access the server now, it's at a remote site. I only have the last errors when I restart zaptel`
`12:55.38`	`*** join/#asterisk Kobaz (~kobaz@its.kobaz.net)`
`12:56.07`	`BugKhaM`	`tzafrir_laptop: ans it's here http://pastebin.com/AFkRf0XG`
`12:56.56`	`BugKhaM`	`tzafrir_laptop: how do I uninstall libpri?`
`12:58.04`	`tzafrir_laptop`	`BugKhaM, Asterisk will not be able to support ISDN (PRI) without libpri`
`12:59.09`	`*** join/#asterisk Baylink1 (~jra@cerberus.vicimarketing.com)`
`12:59.58`	`BugKhaM`	`tzafrir_laptop: I was able to receive calls from zap channels`
`13:00.15`	`BugKhaM`	`tzafrir_laptop: so I had libpri support somehow`
`13:00.37`	`ManxPower`	`are you sure you have a PRI?`
`13:01.02`	`Baylink1`	`ManxPower: you have any luck on the VPM front?`
`13:01.14`	`ManxPower`	`Baylink1, no,`
`13:03.26`	`*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)`
`13:03.26`	`*** mode/#asterisk [+o malcolmd] by ChanServ`
`13:05.43`	`*** join/#asterisk brad_mssw (~brad@shop.monetra.com)`
`13:13.46`	`*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)`
`13:17.10`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`13:20.20`	`*** join/#asterisk garymc (~chatzilla@host81-139-76-8.in-addr.btopenworld.com)`
`13:20.54`	`*** join/#asterisk rgsteele (~rgsteele@207.106.239.81)`
`13:22.26`	`*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)`
`13:26.19`	`garymc`	`I need a VPN specialist to help me OUTTTT!!!`
`13:26.27`	`garymc`	`:S`
`13:27.08`	`florz`	`no, you just need somebody with some basic knowledge about networking stuff`
`13:27.23`	`*** join/#asterisk Akiraa (~Akira@92.81.197.101)`
`13:30.14`	`*** join/#asterisk hluesea (~hluesea@88.247.127.66)`
`13:30.21`	`garymc`	`you know anything? florz?`
`13:31.01`	`ManxPower`	`I imagine you would find more help on a channel where VPN is on-topic`
`13:31.14`	`garymc`	`i cant find a channel`
`13:31.26`	`ManxPower`	`garymc, then just pick a random channel that is not this one.`
`13:31.35`	`garymc`	`ok bye`
`13:31.47`	`ManxPower`	`come back when you have asterisk questions`
`13:33.07`	`hluesea`	`hello channel`
`13:33.30`	`hluesea`	`i have a problem about AMI`
`13:34.27`	`hluesea`	`actually i can reload asterisk via ami but when i try to add and remove queue ami commands i can't success, i looked up the ami commands on cli and i can see the QUEUEREMOVE and QUEUEADD commands`
`13:35.14`	`hluesea`	`Is anybody have a sample AMI for the add and remove queue or how can i do that ?`
`13:40.30`	`*** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com)`
`13:44.27`	`*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)`
`13:44.35`	`leifmadsen`	`has anyone seen this when trying to use cdr_odbc via a freetds connection on asterisk 1.4.24? > cdr_odbc: Error in PREPARE -1`
`13:44.36`	`leifmadsen`	`<PROTECTED>`
`13:46.31`	`BugKhaM`	`my lsmod shows "zaptel 177316 385 zttranscode,wcte11xp,tor2"`
`13:46.53`	`BugKhaM`	`does it mean that wcte11xp is loaded before tor2?`
`13:47.25`	`ManxPower`	`BugKhaM, I doubt it. What is your specific issue?`
`13:48.24`	`BugKhaM`	`ManxPower: I placed MODULES="tor2 wcte11xp" in my /etc/sysconfig/zaptel`
`13:48.48`	`BugKhaM`	`and wanted tor2 to be loaded first`
`13:48.59`	`ManxPower`	`Why?`
`13:49.15`	`ManxPower`	`What card are you using that uses the tor2 driver?`
`13:49.34`	`ManxPower`	`the tor2 cards were old when I started using Asterisk...in 2001`
`13:50.08`	`BugKhaM`	`ManxPower: I get incorrect alarms like "OK" even the cable are not connected if wcte11xp is loaded first`
`13:50.38`	`*** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman)`
`13:50.48`	`ManxPower`	`BugKhaM, Ah, OK. I wish you the best of luck.`
`13:50.49`	`BugKhaM`	`ManxPower: looks like modules have to be loaded in particular orders`
`13:50.57`	`ManxPower`	`BugKhaM, they don't.`
`13:51.08`	`BugKhaM`	`ManxPower: had to restart zaptel every time I reboot`
`13:51.34`	`ManxPower`	`if zaptel is not starting then you have an os related init script issue`
`13:51.40`	`BugKhaM`	`ManxPower: u know how zaptel is loaded in Fedora?`
`13:52.02`	`ManxPower`	`BugKhaM, exactly the same as any other RPM based distro,.`
`13:52.10`	`ManxPower`	`Do you even have it set to start on boot?`
`13:52.15`	`BugKhaM`	`ManxPower: I did "chkconfig zaptel off" but zaptel is still loaded`
`13:52.37`	`ManxPower`	`that tells zaptel to not start on boot. It has nothing to do witht he running zaptel`
`13:52.44`	`BugKhaM`	`ManxPower: normally I did that, start zaptel and then asterisk on boot`
`13:53.24`	`ManxPower`	`No, it did not.`
`13:53.25`	`BugKhaM`	`ManxPower: I'm using ISDN PRI and will need it to load`
`13:53.48`	`BugKhaM`	`ManxPower: TE110 and Tor 2`
`13:53.55`	`ManxPower`	`when you do "chkconfig zaptel off" it disables zaptel on boot.`
`13:54.42`	`ManxPower`	`<ManxPower> What card are you using that uses the tor2 driver?`
`13:54.48`	`BugKhaM`	`ManxPower: yeah, it should but zaptel is still loaded. I had to do rmmod wcte11xp tor2 zaptel and restart zaptel`
`13:55.15`	`BugKhaM`	`ManxPower: I am using Tormenta II for that`
`13:55.50`	`ManxPower`	`BugKhaM, You need to learn your distro.`
`13:56.07`	`ManxPower`	`chkconfig sets the BOOT stuff. It DOES NOT LOAD OR UNLOAD DRIVERS OUTSIDE OF BOOTING`
`13:56.27`	`ManxPower`	`perhaps you are just incredibly confused and mean to type "service zaptel stop"??`
`13:56.42`	`ManxPower`	`because THAT will stop and unload zaptel`
`13:57.10`	`BugKhaM`	`ManxPower: OK, I got what you mean`
`13:57.38`	`BugKhaM`	`ManxPower: I wanted zaptel to be loaded on boot temporarily`
`13:58.05`	`BugKhaM`	`ManxPower: but confused when it was still loaded on boot`
`13:58.23`	`BugKhaM`	`ManxPower: sorry , I wanted zaptel NOT to be loaded on boot temporarily`
`13:59.00`	`ManxPower`	`if "chkconfig zaptel off" is run and zaptel is still loading on boot then you have some other problem.`
`13:59.24`	`BugKhaM`	`ManxPower: hmm, i see`
`13:59.39`	`BugKhaM`	`ManxPower: i'm on init 3`
`13:59.58`	`ManxPower`	`BugKhaM, feel free to read /etc/rc.d/init.d/zaptel`
`14:00.33`	`BugKhaM`	`ManxPower: and /etc/rc3.d still has zaptel and asterisk on it`
`14:01.33`	`ManxPower`	`BugKhaM, then you have an operating system issue.`
`14:02.04`	`BugKhaM`	`ManxPower: I better remove the links from /etc/rc3.d`
`14:02.16`	`BugKhaM`	`ManxPower: ok`
`14:02.21`	`ManxPower`	`BugKhaM, chkconfig should have done that for you. you have an operating system problem`
`14:03.21`	`BugKhaM`	`ManxPower: yeah, it should`
`14:04.03`	`ManxPower`	`sounds to me like you may have installed zaptel and/or Asterisk from an RPM and not from source,`
`14:05.38`	`*** join/#asterisk frk2 (~faraz@zivios/member/fkhan)`
`14:06.55`	`BugKhaM`	`ManxPower: I did install them from sources`
`14:07.25`	`BugKhaM`	`ManxPower: from those in http://downloads.asterisk.org/pub/telephony/`
`14:07.52`	`BugKhaM`	`ManxPower: but I did install asterisk 1.4 and then changed to 1.2`
`14:09.49`	`*** part/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)`
`14:11.00`	`ManxPower`	`no wonder nothing is working`
`14:11.40`	`ManxPower`	`either downgrade your zaptel to 1.2 or upgrade Asterisk to 1,4`
`14:11.52`	`ManxPower`	`then rebuild your Asterisk. Same applies to libpri`
`14:11.56`	`*** part/#asterisk ManxPower (~manxpower@193.sub-75-254-10.myvzw.com)`
`14:14.47`	`*** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca)`
`14:15.48`	`*** join/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net)`
`14:16.37`	`gelo`	`Marquis42: would you like some debug, or just SIP trace?`
`14:17.31`	`Marquis42`	`both! :) If it's not too much to ask, anyway. I can figure it out on a SIP trace alone, but a debug in addition would help.`
`14:21.32`	`gelo`	`Marquis42: just found out something I hadn't notice`
`14:22.32`	`gelo`	`the problem appears when in Asterisk I make a playback, which is my ordinary test`
`14:22.59`	`gelo`	`I have just tried calling to a SIP extension, and the transfer worked`
`14:23.08`	`gelo`	`I'm a bit confused now`
`14:23.41`	`gelo`	`I'll make the pastebin for both, if you don't mind`
`14:23.53`	`*** join/#asterisk cmsouza (~chatzilla@c9067bea.static.spo.virtua.com.br)`
`14:25.09`	`*** join/#asterisk kyosh (whoa@pool-96-246-230-27.nycmny.fios.verizon.net)`
`14:26.13`	`*** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net)`
`14:26.13`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`14:32.23`	`*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)`
`14:32.23`	`*** mode/#asterisk [+o putnopvut] by ChanServ`
`14:32.29`	`*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)`
`14:33.21`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`14:34.23`	`gelo`	`Marquis42: here not working transfer: http://pastebin.com/bUdFjWRX`
`14:34.40`	`*** join/#asterisk ManxPower (~manxpower@193.sub-75-254-10.myvzw.com)`
`14:34.55`	`gelo`	`and here the one to the SIP extension: http://pastebin.com/8DguDaKp`
`14:35.56`	`gelo`	`sorry for all the realtime crap :-S`
`14:37.51`	`garymc`	`Anyone know where I can go for help. I have my VPN setup and my Iphone connects to it, but I dont know if my softphone is using my asterisk server through VPN`
`14:38.04`	`garymc`	`I cant get safari to work with VPN on`
`14:38.11`	`*** part/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman)`
`14:38.20`	`garymc`	`so im guessing I got some IP settings wrong in my pptpd.conf maybe`
`14:38.44`	`fenrus`	`does the connection come up correctly, are you able to see ip-addreses etc`
`14:38.45`	`*** join/#asterisk TehRabbitt (~rabbott56@18263008.cst.lightpath.net)`
`14:38.50`	`TehRabbitt`	`p3nguin: ya there? lol`
`14:38.52`	`fenrus`	`what does the log of the pptpd say ?`
`14:40.28`	`garymc`	`says its fine by the looks of it`
`14:41.56`	`*** join/#asterisk Tim_Toady (~moi@193.92.246.150.dsl.dyn.forthnet.gr)`
`14:42.03`	`chuckf`	`garymc: if your softphone is configured to use the internal ip of the * server over the vpn and its working it is going through the vpn. If your softphone is configured for the public ip of your * server it may or may not be through your vpn depending on the configuration of your network connections`
`14:42.17`	`TehRabbitt`	`can anyone here help me with a NAT issue? i'm still having it... i'm actually outside my home network (at school) and can test different configs now since i'm ssh'ed into the server`
`14:43.02`	`garymc`	`It looks like PPTPD is using port 500 would that be blocked by my router?`
`14:43.03`	`cmsouza`	`TehRabbitt: what exactly is going on?`
`14:43.59`	`TehRabbitt`	`cmsouza: i've followed the ~sipnat instructions, my router allows inbound SIP calls (direct sip works) but if I register using a client such as X-Lite and try to call an extension, or even make an outbound call, it rings then it connects, but there is no audio, 2-way`
`14:44.11`	`garymc`	`ok so my softphone should be configured to the internal IP to my asterisk box`
`14:44.30`	`TehRabbitt`	`ports 5060 and 10000-20000 are open for UDP on my router and fowarded properly... sip.conf has the correct external IP...`
`14:44.41`	`garymc`	`yes mine too`
`14:44.59`	`garymc`	`but I want to go through VPN when abroad and use my softphone that way`
`14:45.30`	`TehRabbitt`	`basically i need to use my softphone on my laptop wherever I am, especially at school`
`14:47.14`	`cmsouza`	`TehRabbitt: in sip.conf did you set the extension to be natted? using nat=yes?`
`14:47.21`	`TehRabbitt`	`cmsouza: I was able to have a reliable 3 hour long phone call from p3nguin last night when he actually used Dial() and directly dialed my extension using SIP bypassing registering to my * server (using his internal extention to dial mine as a guest)`
`14:47.35`	`TehRabbitt`	`cmsouza: yes, they are all set to "yes" that are external`
`14:48.01`	`cmsouza`	`TehRabbitt: have you also configured localnet and externip in sip.conf?`
`14:48.29`	`TehRabbitt`	`yep, localnet is defined 192.168.1.1/255.255.255.0`
`14:48.39`	`TehRabbitt`	`and externip is set to my proper IP from my ISP`
`14:48.54`	`TehRabbitt`	`confirmed using whatismyip.com as well as ipchicken.com`
`14:48.58`	`*** join/#asterisk af_ (~getsmart@88-149-240-123.dynamic.ngi.it)`
`14:49.03`	`cmsouza`	`TehRabbitt: can you post the results from sip show peers from CLI?`
`14:49.08`	`TehRabbitt`	`ok hold on`
`14:50.02`	`*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)`
`14:50.03`	`*** join/#asterisk Kobaz (~kobaz@its.kobaz.net)`
`14:50.03`	`*** join/#asterisk arnotixe (~arnotixe@190.131.124.20)`
`14:50.03`	`*** join/#asterisk saisoma (~saisoma@client72.jdcc.edu)`
`14:50.03`	`*** join/#asterisk nitram (foo@superblob.com)`
`14:50.03`	`*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)`
`14:50.03`	`*** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net)`
`14:50.03`	`*** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net)`
`14:50.03`	`*** join/#asterisk s4msung (~s4msung@dice.s4msung.de)`
`14:50.05`	`TehRabbitt`	`http://pastebin.com/7bBBAyz6`
`14:50.11`	`TehRabbitt`	`i'm currently connected to ext 300`
`14:50.18`	`TehRabbitt`	`well name/un 300`
`14:51.21`	`*** join/#asterisk eliel (~eliels@201.234.94.226)`
`14:52.08`	`TehRabbitt`	`same issue with users 200 and 100... they all have the same settings but i'm currently using 300 for testing purposes`
`14:53.29`	`*** join/#asterisk Akiraa (~Akiraaaa@79.112.44.66)`
`14:53.45`	`cmsouza`	`TehRabbitt: It also shows that you have set qualify=yes, so no use asking that. since you registered your extention successfully i believe your problem may reside in the actual RTP connection`
`14:54.14`	`*** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs)`
`14:54.26`	`*** join/#asterisk dymaxion (~dymaxion@203080238194.static.ctinets.com)`
`14:54.29`	`*** part/#asterisk hluesea (~hluesea@88.247.127.66)`
`14:54.39`	`TehRabbitt`	`hm`
`14:54.43`	`TehRabbitt`	`what would that mean?`
`14:54.44`	`cmsouza`	`do you have any extensions available right now, a brother, roomate that is inside you local network and can pick up an extension? we cam make a test call via originate CLI`
`14:54.57`	`TehRabbitt`	`411 should work in theory`
`14:55.12`	`TehRabbitt`	`since it's just an outbound call using an 1800 termination`
`14:55.18`	`TehRabbitt`	`(goog-411)`
`14:55.53`	`TehRabbitt`	`what do you mean originate CLI?`
`14:55.58`	`*** join/#asterisk cards1007 (~cards@64-83-215-202.dhcp.stcd.mn.charter.com)`
`14:58.22`	`*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)`
`14:59.16`	`ManxPower`	`sounds like you might not have canreinvite=no`
`14:59.37`	`cmsouza`	`TehRabbitt: originate is a CLI command that starts a call and then bridge the call to the spcified extension`
`14:59.58`	`TehRabbitt`	`ManxPower: i've go that in there... (`
`15:00.18`	`TehRabbitt`	`cmsouza: hm... could i bridge the call to 411 (goog411)`
`15:01.07`	`*** join/#asterisk kartik (~koolkarti@117.199.127.147)`
`15:02.05`	`cmsouza`	`yes`
`15:02.50`	`cmsouza`	`something like originate DAHDI/g0/4664411 extension 300@internal if internal is your extension context`
`15:02.50`	`*** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net)`
`15:03.22`	`TehRabbitt`	`well everythign works internally... it's whenver I try to connect on the outside of my LAN`
`15:03.35`	`cmsouza`	`it should call 4664411 and then ring your softphone`
`15:04.14`	`TehRabbitt`	`hm aight`
`15:04.17`	`TehRabbitt`	`wait what is DAHDI?`
`15:04.20`	`TehRabbitt`	`and g0`
`15:04.30`	`[TK]D-Fender`	`~dahdi`
`15:04.30`	`infobot`	`[~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav`
`15:04.41`	`cmsouza`	`this bot is wonderful`
`15:06.21`	`cmsouza`	`dahdi connects your * to POTS, so you can call landline phones`
`15:06.38`	`[TK]D-Fender`	`...`
`15:06.49`	`TehRabbitt`	`ah`
`15:06.51`	`[TK]D-Fender`	`cmsouza: presumptive`
`15:07.01`	`TehRabbitt`	`well i'm using SIP outgoing though...`
`15:07.12`	`cmsouza`	`if you do not have it configured i assume futurenine is a SIP provider...`
`15:07.16`	`*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)`
`15:07.20`	`TehRabbitt`	`yes it is`
`15:07.23`	`cmsouza`	`it also works`
`15:07.31`	`TehRabbitt`	`right now internally if I dial "411" it uses futurenine to place the call`
`15:07.37`	`TehRabbitt`	`and it goes through perfect`
`15:07.39`	`TehRabbitt`	`dmtf everything`
`15:07.47`	`[TK]D-Fender`	`TehRabbitt: DAHDI provides a timer/mixer for things like MeetMe & IAX2 Trunk Mode as well as supporting PCI-type cards from Digium, etc`
`15:08.21`	`TehRabbitt`	`ah`
`15:09.10`	`cmsouza`	`originate SIP/futurenine/4664411 extension 300@internal should to the trick`
`15:09.17`	`TehRabbitt`	`ok... I just dialed my cell phone from the softphone, it showed up that the CID was "300" and then there was no audio coming through :-\ eithe rway`
`15:09.27`	`TehRabbitt`	`ah aight lemme try that`
`15:09.41`	`TehRabbitt`	`300@internal?`
`15:09.59`	`cmsouza`	`i assume internal is your default internal extensions context`
`15:10.07`	`*** join/#asterisk cesar_CR (~cesar@201.192.86.30)`
`15:10.23`	`TehRabbitt`	`nope lol`
`15:10.25`	`*** join/#asterisk Ad-Hoc (~nimbus@62.1.238.196.dsl.dyn.forthnet.gr)`
`15:10.56`	`TehRabbitt`	`ok it dialed my extention still no audio`
`15:11.09`	`cmsouza`	`but your softphone rang`
`15:11.25`	`TehRabbitt`	`yes`
`15:11.48`	`cmsouza`	`ok, then your problem is either codecs or rtp stream`
`15:11.49`	`TehRabbitt`	`showed up as CID: asterisk@71.59.82.2 (my WAN for the * server)`
`15:12.05`	`TehRabbitt`	`cmsouza: if it works if i'm inside my NAT, would it stil be codecs?`
`15:12.08`	`cmsouza`	`what sofphone are you using?`
`15:12.14`	`TehRabbitt`	`x-lite`
`15:12.21`	`TehRabbitt`	`i've also tired droidsip`
`15:12.44`	`TehRabbitt`	`well my friend did`
`15:12.44`	`cmsouza`	`the same notebook, connected to your LAN works?`
`15:12.49`	`TehRabbitt`	`yes`
`15:12.57`	`TehRabbitt`	`if I log onto my wifi at home. i can dial in / out`
`15:13.08`	`TehRabbitt`	`as soon as i connect outside my LAN... nothing`
`15:13.09`	`cmsouza`	`so codecs are ruled out`
`15:13.18`	`TehRabbitt`	`:-\ so it's RTP?`
`15:13.22`	`cmsouza`	`your problem is rtp stream`
`15:13.28`	`TehRabbitt`	`how do I fix that heh`
`15:13.34`	`TehRabbitt`	`can it be fixed :-\`
`15:13.37`	`cmsouza`	`what kind of router/firewall do you have?`
`15:13.41`	`cmsouza`	`of course it can`
`15:13.43`	`cmsouza`	`haha`
`15:13.48`	`[TK]D-Fender`	`TehRabbitt: ...`
`15:13.53`	`TehRabbitt`	`Linksys WRT160N running DD-WRT v23`
`15:13.54`	`[TK]D-Fender`	`~sipnat`
`15:13.55`	`infobot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`15:13.56`	`[TK]D-Fender`	`^^^^^^^`
`15:14.05`	`[TK]D-Fender`	`TehRabbitt: trash DD-WRT`
`15:14.05`	`cmsouza`	`holy mother`
`15:14.15`	`[TK]D-Fender`	`TehRabbitt: Been a known offender`
`15:14.21`	`TehRabbitt`	`:-\ seriously?`
`15:14.25`	`cmsouza`	`dd-wrt is something i never used in my entire life`
`15:14.41`	`TehRabbitt`	`I can't really trash it as it's acting as a point to point network`
`15:14.50`	`TehRabbitt`	`there's no workaround? :-\`
`15:14.51`	`cmsouza`	`flash this kid back to its original firmware`
`15:15.10`	`TehRabbitt`	`if I flash it back, I lose all connectivity to all my equipment`
`15:15.38`	`ManxPower`	`TehRabbitt, DISABLE any ALG or SIP specific features of your router`
`15:15.40`	`cmsouza`	`you can use netcat to pass packets to and from your lan and see if they are properly forwarded`
`15:15.40`	`[TK]D-Fender`	`TehRabbitt: Test another router. Ensure there is no ALG or SIP transform. Ensure that proper ports are forwarded. Validate yoru configs. Verify the SIP debug of a failed call`
`15:15.48`	`TehRabbitt`	`the router (upstairs) is broadcasting a signal to a receiving client-bridge which then proides internet to my entire network downstairs as i cannot run wire and ethernet over powerline isn't an option either`
`15:15.58`	`TehRabbitt`	`ALG or SIP transform?`
`15:16.13`	`ManxPower`	`TehRabbitt, both.`
`15:16.18`	`TehRabbitt`	`what are they though?`
`15:16.42`	`ManxPower`	`TehRabbitt, they are "SIP BREAKAGE FEATURES"`
`15:17.00`	`TehRabbitt`	`ah... :-\`
`15:17.07`	`TehRabbitt`	`whats the point of them heh`
`15:17.17`	`cmsouza`	`ROFL here`
`15:17.25`	`cmsouza`	`sip breakage features`
`15:17.32`	`ManxPower`	`TehRabbitt, Google is your friend`
`15:18.14`	`cmsouza`	`do you have linux running in school or notebook or anywhere else outside your lan?`
`15:18.32`	`cards1007`	`I'm having an issue with a remote phone that frequently becomes unreachable (chan_sip.c: Peer '555' is now UNREACHABLE! Last qualify: 29). My first thought was to increase the qualify from "yes" to "10000" to allow more time for a response before marking it unreachable, to account for potential network latency, but that definitely hasn't resolved the problem. Am I understanding the log entry and the qualify setting correctly?`
`15:18.39`	`gelo`	`Marquis42: r u there?`
`15:18.52`	`TehRabbitt`	`no linux in school... linux only on my server`
`15:18.54`	`ManxPower`	`cards1007, disable qualify`
`15:19.37`	`*** join/#asterisk drift- (~47c4759e@gateway/web/freenode/x-asycjkrchwsaiyre)`
`15:19.47`	`drift-`	`anyone fimilar with voip.ms?`
`15:19.56`	`cmsouza`	`TehRabbitt: this sure does limit your troubleshooting options`
`15:20.05`	`[TK]D-Fender`	`drift-: They are an ITSP.`
`15:20.12`	`TehRabbitt`	`again I can ssh into the server though`
`15:20.26`	`cards1007`	`ManxPower: Ok, I'll give that a try. Any info (link?) where I can get a better understanding of what the qualify setting is for and how it works?`
`15:20.26`	`[TK]D-Fender`	`[11:18]<TehRabbitt>no linux in school... linux only on my server <- so?`
`15:21.02`	`TehRabbitt`	`hm... apparently the latest version of DDWRT allows you to disbale SIP ALG`
`15:21.09`	`TehRabbitt`	`under "P2P settings"`
`15:21.12`	`TehRabbitt`	`weird place to put it...`
`15:21.39`	`cmsouza`	`get yourself netcat for windows, it is fairly easy to use, send packets to port 10000 and up and set netcat in your server to listen to these ports whatever you send by netcat should appear on your server`
`15:21.52`	`cmsouza`	`ou can shutdown * server`
`15:21.54`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`15:22.00`	`*** join/#asterisk bent_screwdriver (~socain00@74.255.249.66)`
`15:22.15`	`[TK]D-Fender`	`cmsouza: Windows can do networking? WTF?!`
`15:22.19`	`TehRabbitt`	`LOL`
`15:22.20`	`drift-`	`ive got a problem , I have 4 phone numbers , and 2 of them seem to have issue, when person calls its 50/50% chance it will go thru then hang up :(`
`15:22.28`	`cards1007`	`ManxPower: Well, the phone was marked as reachable as soon as I reloaded the config with that change. What's the potential downside? Seems like off would be default if it meant things just work better`
`15:22.58`	`Marquis42`	`gelo: I had to step awayfor a bit (and will be). I will definitely have a look at your pastebins later today, though, and get back to you.`
`15:22.59`	`drift-`	`i have 11 extensions , 6 use 2 phone numbers , other 4 use 1 phone number and 1 extension uses 1 number`
`15:23.16`	`drift-`	`the 3600 and 3900 line have a hang up issue :( on iax2`
`15:23.22`	`drift-`	`voip.ms not sure what to do i'm lost`
`15:23.38`	`*** join/#asterisk corretico (~laguilar@201.201.46.106)`
`15:24.16`	`TehRabbitt`	`just VPNed into my desktop at home, gonna try to disable that SIP ALG thing`
`15:24.55`	`gelo`	`Marquis42: well, it's getting a little late here, so maybe we can talk tomorrow. Thank you!!!`
`15:25.03`	`cmsouza`	`google netcat and learn it`
`15:25.21`	`cmsouza`	`it's the most useful network troubleshooting tool in the world`
`15:25.34`	`cmsouza`	`i'll be back after lunch 40 minutes`
`15:26.04`	`p3nguin`	`tehrabbitt: Yes, I'm here.`
`15:26.29`	`ManxPower`	`TehRabbitt, that is one of the first things people should have suggested`
`15:26.44`	`TehRabbitt`	`ManxPower: what is?`
`15:27.03`	`ManxPower`	`TehRabbitt, SIP NAT or ALG on your router`
`15:27.15`	`TehRabbitt`	`what if I disable the Firewall /SPI all together?`
`15:27.37`	`ManxPower`	`TehRabbitt, that is your call. We only care about ALG and SIP NAT support in the router`
`15:28.23`	`TehRabbitt`	`http://www.dd-wrt.com/wiki/index.php/Siproxd`
`15:28.34`	`[TK]D-Fender`	`TehRabbitt: NO`
`15:28.45`	`[TK]D-Fender`	`TehRabbitt: stop fucking with sip`
`15:28.49`	`ManxPower`	`If you want me to hold your hand I expect dinner, drinks, and a credit card first.`
`15:28.58`	`TehRabbitt`	`LOL`
`15:29.02`	`Baylink_work`	`heh`
`15:29.24`	`TehRabbitt`	`i've already got DD-WRT v23sp1voip installed... it comes with a SIP proxy built in... just gotta figure it out apparently`
`15:29.33`	`ManxPower`	`TehRabbitt, no!`
`15:29.38`	`ManxPower`	`you need to figure out how to DISABLE IT.`
`15:30.04`	`TehRabbitt`	`can't in DDWRT`
`15:30.09`	`ManxPower`	`What part of this is so hard to understand????? SIP NAT in routers and phones don't work with Asterisk. This isn't fricking rocket science`
`15:30.19`	`TehRabbitt`	`has "milkfish SIP router" built in`
`15:30.27`	`ManxPower`	`TehRabbitt, then don't expect to have it work with Asterisk`
`15:30.34`	`Baylink_work`	`Waitaminnit.`
`15:30.49`	`Baylink_work`	`If the router has a SIP proxy in it, then you don't need to NAT that end, right?`
`15:30.57`	`*** part/#asterisk gelo (~gelo@mx01.quobis.com)`
`15:31.14`	`ManxPower`	`Baylink_work, maybe, maybe not, but it is far beyond the scope of this channel.`
`15:31.51`	`Baylink_work`	`Setting up a SIP proxy at the edge of a 1918 network? Is off-topic for #asterisk? Really?`
`15:31.57`	`Baylink_work`	`Cause that's where I'd go...`
`15:32.34`	`ManxPower`	`Baylink_work, just edit the /etc/asterisk/sip.conf for your proxy....`
`15:33.42`	`p3nguin`	`drift-: Did you have a question about VoIP.ms?`
`15:35.05`	`ManxPower`	`Baylink_work, I'd go to the support channel for the SIP proxy I'm using.`
`15:35.16`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`15:35.48`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`15:36.52`	`Baylink_work`	`Once he finds that out, yes.`
`15:37.11`	`p3nguin`	`If you put a SIP proxy at the edge of your network, that takes asterisk out of the connection. There's no reason to expect support from #asterisk when asterisk isn't in the equation.`
`15:37.51`	`TehRabbitt`	`p3nguin: remind me to never mess with the router while remotely connected lmao`
`15:38.12`	`p3nguin`	`hmm`
`15:38.26`	`p3nguin`	`It _can_ be a bad idea, but it doesn't have to be.`
`15:38.27`	`TehRabbitt`	`just booted myself off :(`
`15:38.43`	`TehRabbitt`	`lost complete connectivity 0_o`
`15:38.44`	`p3nguin`	`I'm comfortable working on routers remotely.`
`15:39.28`	`TehRabbitt`	`nvm got it back haha`
`15:39.48`	`ManxPower`	`Baylink_work, I don't see how now knowing what SIP proxy you are using is a problem for this channel.`
`15:40.07`	`ManxPower`	`s/now/not`
`15:40.42`	`Baylink_work`	`Be ye not overly annoying, nor too easily annoyed. Have we forgotten the Fidonet creed already?`
`15:41.03`	`TehRabbitt`	`p3nguin: reason I wanna get SIP working is because I wanna be able to get those cisco desk phones eventually`
`15:41.34`	`ManxPower`	`Baylink_work, stick around for a while, you'll forget it tool.`
`15:41.37`	`ManxPower`	`too.`
`15:41.41`	`p3nguin`	`As I mentioned before, get a known-to-work router and put inline in place of yours.`
`15:41.48`	`Baylink_work`	`Been doing this for 24 years now.`
`15:41.59`	`p3nguin`	`If RTP magically works, your router is at fault.`
`15:42.07`	`ManxPower`	`24 years in #asterisk?`
`15:42.16`	`Baylink_work`	`When I don't have the patience for newbies (or eeediots), I just stop talking. // 24 years on the net.`
`15:42.21`	`TehRabbitt`	`p3nguin: can't really do that though with my setup :( both ends need to be DD-WRT to reliablly work with my wireless setup`
`15:42.29`	`Baylink_work`	`I don't think #a is quite that old. :-)`
`15:42.54`	`TehRabbitt`	`i've tried using linksys firmware on one end, DD-WRT on the other and sadly, it just can't see the Linksys router (Txmt power is set to 21 instead of 71mW)`
`15:43.22`	`p3nguin`	`You need to take out variables to effectively troubleshoot.`
`15:43.47`	`TehRabbitt`	`true :-\`
`15:44.02`	`*** join/#asterisk pkecastillo (~pirruar@190.113.141.122)`
`15:44.07`	`p3nguin`	`In your situation, what I would do is put my * box between my modem and my wireless bridge.`
`15:44.08`	`TehRabbitt`	`hm... i'ts almost 12PM here... i'mma go to that computer recycling place I told you about p3nguin... i wanna see what they have for "sale" lol`
`15:44.08`	`ManxPower`	`You'll eventually see that people seem to think of #asterisk as a dumping ground for people that can't get support for whatever PoS software that might possible be related to VoIP that they are using. All the way from how to pirate the Cisco SIP firmware to OpenSER to FreePBX to (as you can see) routers.`
`15:44.19`	`*** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk)`
`15:44.21`	`pkecastillo`	`hello guys!`
`15:44.51`	`TehRabbitt`	`p3nguin: not an option, can't put the server in my bedroom too loud lol mind you it's pretty huge (and very very heavy)`
`15:45.02`	`ManxPower`	`This channel is about ASTERISK. there are other channels for non-Asterisk related stuff.`
`15:45.07`	`p3nguin`	`tehrabbitt: Let me know. I may have to get you to pick up some equipment and send to me.`
`15:45.16`	`TehRabbitt`	`other solution, run about 120 feet of cat5`
`15:45.21`	`p3nguin`	`tehrabbitt: Use two cables and leave it where it is.`
`15:45.46`	`TehRabbitt`	`ManxPower: how does this not have to do with *?`
`15:45.59`	`TehRabbitt`	`p3nguin, true... might have to try that... just gotta get the cable lol`
`15:46.07`	`ManxPower`	`TehRabbitt, your problem is your router.`
`15:46.17`	`p3nguin`	`You've got to do something since it doesn't work right currently.`
`15:46.31`	`TehRabbitt`	`ManxPower: so you're saying if I reflash to Linksys firmware it should work? :-\`
`15:47.01`	`p3nguin`	`I simply do not understand how I can make a call to you via SIP URI and it works perfectly, but configuring a client as a peer to you doesn't pass audio.`
`15:47.08`	`p3nguin`	`does not make sense at all to me.`
`15:47.12`	`TehRabbitt`	`that's what has me confused as well`
`15:47.16`	`ManxPower`	`TehRabbitt, as long as you disable any ALG or SIP Nat YES.`
`15:47.17`	`Baylink_work`	`ManxPower: What channel, though, is there for 20,000ft general VoIP questions?`
`15:47.48`	`Baylink_work`	`Just checked #voip. I'm the op. ;-)`
`15:47.54`	`p3nguin`	`lol`
`15:47.58`	`TehRabbitt`	`lol`
`15:48.06`	`ManxPower`	`Baylink_work, I don't know., Just like I don't know of a place to get help with my ford Truck. However I know that it is NOT the Nisson dealership.`
`15:48.38`	`TehRabbitt`	`p3nguin: anyway i'mma go for a lil while, gonna go to that place and see if they have any good stuff for sale...`
`15:48.44`	`Baylink_work`	`Sure. But I'm not sure that's really a valid analogy here. You're trying to make the car work. There are lots of independent non-dealer mechanics...`
`15:48.45`	`TehRabbitt`	`i'll let you know what they have`
`15:48.51`	`TehRabbitt`	`what models were they again?`
`15:48.52`	`p3nguin`	`tehrabbitt: alright`
`15:48.58`	`TehRabbitt`	`7960?`
`15:49.09`	`TehRabbitt`	`Cisco 7961?`
`15:49.27`	`p3nguin`	`tehrabbitt: Most of the 7900 series phones: 7940/7960, 7941/7961`
`15:49.31`	`TehRabbitt`	`aightt...`
`15:49.46`	`TehRabbitt`	`also, do u think you could send me that ZIP / TAR of the ringtones? lol`
`15:49.51`	`p3nguin`	`Also the other 796x or 797x with color screens would be nice.`
`15:50.02`	`p3nguin`	`I'll get it today for you.`
`15:50.04`	`TehRabbitt`	`p3nguin: yea i'd like a color screen (or three) lol`
`15:50.05`	`TehRabbitt`	`aightt`
`15:50.14`	`TehRabbitt`	`my email is Rabbott@tenehawk.com if it helps`
`15:50.23`	`p3nguin`	`okay`
`15:50.37`	`Baylink_work`	`It is, after all, ManxPower, not like he was trying to get freeswitch running with his router...`
`15:50.43`	`TehRabbitt`	`anyway i'll bbl for now i'm off to the computer recylcing place in search for IP phones lol`
`15:51.38`	`pkecastillo`	`iam a problem with a conection in "E1 + Provider", this provider, tells me I need to send when connecting the call signal "IMI-ACM-REL"... any idea?`
`15:52.11`	`TehRabbitt`	`ManxPower: btw, all I am trying to do is figure out if there is a way to get * to work with my current setup without having to drastically rework a network that has worked perfectly fine for over 2 years... so techncailly me asking questions about my router / NAT setup has direct ties to what * can and can't do as I try to determine ways to get * to work...`
`15:52.33`	`TehRabbitt`	`anyway, i'll bbl...`
`15:52.40`	`TehRabbitt`	`p3nguin: i'll let you know what I find lol`
`15:52.47`	`[TK]D-Fender`	`TehRabbitt: * can't stop you from fucking up your networking.`
`15:52.51`	`[TK]D-Fender`	`TehRabbitt: There.`
`15:53.46`	`pkecastillo`	`because calls are cut just 1 minute, and it is because it generates the time out for not having sent the signal "anm"...`
`15:54.17`	`TehRabbitt`	`[TK]D-Fender: i know that... but i'm just trying to figure out if there was a way of getting * to work without having to rework the way I have the network "designed"... not softwarewise but hardwarewise / having cable run down a staircase / across a livingroom floor`
`15:55.22`	`TehRabbitt`	`and PoE isn't an option as my "server room" is technically my garage and there is only one power outlet out there which is for the garage door opener.... the rest of the power is off a high-gauge extension cord run to an outlet in the kitchen which is on it's own circuit`
`15:55.27`	`*** join/#asterisk dlublink (~david@206-248-163-245.dsl.teksavvy.com)`
`15:55.34`	`TehRabbitt`	`err not PoE i meant to say powerline ethernet`
`15:55.45`	`dlublink`	`I have used 'faxdetect=both' in chan_dadhi.conf, is there a command I can add to the dialplan for certain extensions to disable fax detection ?`
`15:55.53`	`[TK]D-Fender`	`TehRabbitt: * can't work around the fact your networking is screwweed?`
`15:56.16`	`[TK]D-Fender`	`TehRabbitt: How does Ford fix the giant hole in the highway that stops you from driving your truck past it?`
`15:56.35`	`TehRabbitt`	`[TK]D-Fender: the networking really isn't... AP1 broadcasts SSID to AP2 which receives it... it's a Point to Point network similar to how businesses transfer voice / data across long distances without wires`
`15:56.47`	`TehRabbitt`	`they have one antenna pointed to another, usually yagi's`
`15:57.39`	`TehRabbitt`	`anyway i gotta go so i can see what the computer place has... i'll bbl`
`16:02.53`	`*** part/#asterisk dlublink (~david@206-248-163-245.dsl.teksavvy.com)`
`16:04.53`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`16:05.09`	`pkecastillo`	`guys!, iam a problem with a conection in "E1 + Provider", this provider, tells me I need to send when connecting the call signal "IMI-ACM-REL"... any idea?.`
`16:05.36`	`pkecastillo`	`...because calls are cut just 1 minute, and it is because it generates the time out for not having sent the signal "anm" ¿?`
`16:09.11`	`*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)`
`16:09.52`	`bent_screwdriver`	`anyone had an issue that when you "reload" at asterisk cli the active calls breakup? (1.6.2). Reload runs noticeably slower too. Only about 4 active calls on an ML-350 G5 xeon 3Ghz.`
`16:10.35`	`bent_screwdriver`	`top shows almost no cpu being used and only about 2% while reloading`
`16:12.05`	`*** join/#asterisk imox1234 (~imox1234@p4FC5C509.dip0.t-ipconnect.de)`
`16:16.08`	`*** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl)`
`16:17.26`	`p3nguin`	`barthezz: Don't run "reload" on Asterisk.`
`16:17.40`	`p3nguin`	`dammit`
`16:17.48`	`p3nguin`	`bent_screwdriver: Don't run "reload" on Asterisk.`
`16:18.00`	`BarthezZ`	`lolwut? :p`
`16:18.08`	`p3nguin`	`barthezz: disregard`
`16:18.09`	`BarthezZ`	`i can run reload whenever I want! :p`
`16:18.22`	`p3nguin`	`barthezz: You shouldn't.`
`16:18.39`	`BarthezZ`	`i know, but I can ^^`
`16:18.42`	`*** part/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl)`
`16:18.47`	`bent_screwdriver`	`p3nguin: why not? run the mudules seperateley?`
`16:18.48`	`*** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl)`
`16:18.59`	`p3nguin`	`bent_screwdriver: That's what you're supposed to do.`
`16:19.02`	`BarthezZ`	`my key combinations are giving me a hard time today`
`16:19.17`	`p3nguin`	`bent_screwdriver: There should even be a message printed on the CLI telling you to NOT run reload.`
`16:19.43`	`p3nguin`	`bent_screwdriver: What was your reason to run "reload" in the first place?`
`16:19.59`	`bent_screwdriver`	`p3nguin: gotcha. still doesn't explain explain why it is slow and the identical standby box is fast while running reload.`
`16:20.09`	`bent_screwdriver`	`p3nguin: made sip changes and was lazy`
`16:20.15`	`p3nguin`	`"sip reload"`
`16:20.18`	`p3nguin`	`Not too hard.`
`16:20.49`	`bent_screwdriver`	`p3nguin: old habits are hard to break. even if it means just doing nothing ;)`
`16:21.45`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`16:21.49`	`*** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net)`
`16:22.24`	`bent_screwdriver`	`p3nguin: could multiple reloads cause the slowness over several days or something? seems to have gotten progressivly worse in the last few days....`
`16:22.45`	`p3nguin`	`No clue, since I don't run "reload" ever.`
`16:23.39`	`bent_screwdriver`	`p3nguin: okay, i'll restart * tonight, then never use reload, and see if any issues come up. Thanks for the help!`
`16:24.17`	`jkroon`	`hi guys, t38 negotiation in asterisk 1.6.2.6 - how reliable is it generally? because depending on where I wack in wait() statements around Ringing(); Answer(); and SendFAX(); I get way different results.`
`16:24.19`	`p3nguin`	`"retart gracefully" or "restart when convenient" are good commands for a busy system.`
`16:24.29`	`p3nguin`	`s/retart/restart/`
`16:24.50`	`bent_screwdriver`	`10-4 thx`
`16:27.23`	`*** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-qydrwfdepodsgbxo)`
`16:28.07`	`*** join/#asterisk Knightfal (~j@mailer.1callres.com)`
`16:28.28`	`*** join/#asterisk viq (~viq@unaffiliated/viq)`
`16:29.43`	`CatLynx`	`hmm just got back to work, and I did not relized I forgot to close IRC window when I went home`
`16:30.03`	`p3nguin`	`Now you know what it's like to idle.`
`16:30.09`	`cmsouza`	`back from lunch`
`16:30.09`	`CatLynx`	`heh`
`16:30.26`	`CatLynx`	`just in a rush to go home from work`
`16:30.39`	`p3nguin`	`I don't ever voluntarily disconnect from IRC or IM services.`
`16:30.48`	`*** join/#asterisk logyati (~pauloarru@201.17.59.67)`
`16:30.52`	`logyati`	`hello guys`
`16:30.53`	`*** join/#asterisk Tim_Toady (~moi@62.1.102.159.dsl.dyn.forthnet.gr)`
`16:30.59`	`logyati`	`what do you sugest for mediaproxy?`
`16:30.59`	`*** join/#asterisk trelane (~trelane@funtoo/staff/trelane)`
`16:31.06`	`logyati`	`i mean, which software?`
`16:31.28`	`logyati`	`can asterisk act only as mediaproxy?`
`16:33.26`	`*** join/#asterisk orioni (~chatzilla@pc.tring.al)`
`16:33.28`	`orioni`	`~book`
`16:33.29`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`16:34.47`	`*** join/#asterisk garymc (~chatzilla@host86-135-203-71.range86-135.btcentralplus.com)`
`16:34.53`	`logyati`	`orioni`
`16:35.02`	`orioni`	`yep`
`16:36.39`	`CatLynx`	`my coworker kept asking me basic Asterisk quetions, so I e-mail him that PDF since then, he never ask me a quetion any more :)`
`16:37.34`	`logyati`	`orioni, can asterisk act as media proxy only?`
`16:39.59`	`[TK]D-Fender`	`logyati: * is a B2BUA, not a proxy`
`16:40.30`	`florz`	`a media proxy kindof has to be a b2bua, too`
`16:40.53`	`florz`	`still, * isn't particularly suited as a general purpose sip media proxy`
`16:41.14`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`16:41.29`	`logyati`	`hello d-fender! how long!`
`16:41.48`	`*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)`
`16:41.53`	`*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)`
`16:41.58`	`logyati`	`so guys, what do you recommend to use as media proxy? http://mediaproxy.ag-projects.com/ ?`
`16:42.21`	`[TK]D-Fender`	`logyati: Why do you think you need one?`
`16:43.03`	`*** join/#asterisk Devon_ (~chatzilla@63.214.236.169)`
`16:44.23`	`logyati`	`well, i have a friend that uses a proprietary voip solution. He asked my help to build a media proxy, cos their clients are behind nat. So, i thought about comming here and ask you guys, that have a big experience on voip`
`16:44.58`	`[TK]D-Fender`	`logyati: * is not the solution for you. Go look at...`
`16:45.00`	`[TK]D-Fender`	`~ser`
`16:45.00`	`infobot`	`somebody said ser was [~ser] Sip Express Router - see http://www.iptel.org/ser/, or at #ser`
`16:45.07`	`[TK]D-Fender`	`~openser`
`16:45.08`	`infobot`	`openser is probably an open source GPL project that aims to develop a robust and scalable SIP server. It is spawned from FhG FOKUS SIP Express Router (SER) and it promotes a development strategy open for contributors and contributions. From project's website http://www.voip-info.org/wiki/view/About+OpenSER, or now known as Kamailio`
`16:45.21`	`[TK]D-Fender`	`~Kamailio`
`16:45.21`	`logyati`	`my first question is, can asterisk act as media proxy? you said no. so, i would like to know which mediaproxy solution did you work with, and which do you recommend?`
`16:45.31`	`[TK]D-Fender`	`logyati: No... it is a NOT a proxy`
`16:45.47`	`[TK]D-Fender`	`logyati: and I just linked you.`
`16:46.18`	`orioni`	`is there a good tutorial to integrate a2bill to freepbx`
`16:46.22`	`*** join/#asterisk deadpigeon (~kvirc@office.xpressamerica.net)`
`16:46.33`	`orioni`	`so the extensions that i add on freepbx are created / billed on a2bill ?`
`16:47.41`	`logyati`	`whats the difference between media proxy and SIP proxy?`
`16:47.43`	`[TK]D-Fender`	`orioni: Neither are supported in this channel`
`16:47.52`	`logyati`	`talking about SIP Voip`
`16:48.03`	`[TK]D-Fender`	`logyati: They handle separate halves of the call`
`16:48.23`	`[TK]D-Fender`	`logyati: Media = SIP, Signalling=SIP`
`16:48.39`	`p3nguin`	`?`
`16:49.03`	`p3nguin`	`thinks [tk]d-fender needs a break.`
`16:49.24`	`*** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net)`
`16:50.06`	`logyati`	`i used openser several years ago. I found its so hard to configure, and it works as SIP server, with authentication and so on. I thought that i could find an easier solution for media proxing`
`16:50.21`	`*** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net)`
`16:51.13`	`[TK]D-Fender`	`logyati: * is not a proxy. End of discussion.`
`16:51.51`	`p3nguin`	`Oh, c'mon... you know that's not the end of it. ;)`
`16:56.33`	`*** join/#asterisk acxty (~acxty@201.220.136.118)`
`16:57.43`	`acxty`	`Hi guys, I am trying to do a dial() and I am getting this error Auto fallthrough, channel 'SIP/116-081ce918' status is 'CONGESTION'. It doesn't matter to which peer I am dialing I always get it`
`16:58.03`	`acxty`	`I have 6 peers configure on asterisk`
`16:58.26`	`acxty`	`What I am trying to do is that when I receive a call on peer1 to make a dial to a different number using peer6`
`16:59.17`	`p3nguin`	`What does your Dial command look like?`
`16:59.22`	`cmsouza`	`~cmsouza`
`17:03.35`	`[TK]D-Fender`	`~pb`
`17:03.35`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`17:03.37`	`[TK]D-Fender`	`acxty: ^^^^^^^6`
`17:08.03`	`*** join/#asterisk hugorebelo (~hugo@p5B27586F.dip.t-dialin.net)`
`17:14.40`	`acxty`	`http://dpaste.com/191250/ there it is`
`17:17.22`	`*** join/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org)`
`17:18.08`	`*** part/#asterisk bsaxon (~bsaxon@pool-71-98-135-63.tampfl.dsl-w.verizon.net)`
`17:19.34`	`[TK]D-Fender`	`acxty: Where is teh FAILED CALL to look at?`
`17:19.56`	`*** join/#asterisk fofware (~fabian@190.7.25.160)`
`17:22.08`	`*** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no)`
`17:22.53`	`iscario`	`hi, i'm looking forward to install odbc support for my asterisk server (running on opennsd 4.5). I first installed iodbc..... from packages, but i do not know what to do after that... I suppose i have to fill the odbcinst.ini but i need a driver for mysql first. And i do not find it on the mysql site (o do not find my version of mysql which is 5.0.77)`
`17:24.38`	`bent_screwdriver`	`the active calls still get garbled/breakup while doing "dialplan reload" and the "dialplan reload" is still even after an asterisk restart. any ideas?`
`17:26.19`	`[TK]D-Fender`	`iscario: * also supports MySQL rather directly. Any reason not to do that?`
`17:27.40`	`acxty`	`http://dpaste.com/191253/`
`17:29.08`	`iscario`	`mmmh, no real reason maybe [TK]D-Fender... do you know if i would need it in order to interface OpenSIPS with Asterisk ? Otherwise, that was just to learn...`
`17:30.09`	`acxty`	`What I am trying to do is that when someone dial on the peers I got configure on the asterisk to automatically dial 104 which is a phone connected to the pbx. I want to use peer 116 for outgoing`
`17:31.12`	`p3nguin`	`bent_screwdriver: dialplan reload shouldn't interfere with an active call.`
`17:32.04`	`[TK]D-Fender`	`iscario: I don't see what a separate SIP server, a database, and Asterisk have to do with each other`
`17:32.22`	`bent_screwdriver`	`p3nguin: the calls start to get garbled, like the system is over loaded, while reloading. and the reload output to the console runs slower than I'm used to on ther boxes. I've restarted * but not the box yet.`
`17:32.26`	`acxty`	`[TK]D-Fender, I attach the failed call on the start http://dpaste.com/191253/`
`17:32.34`	`[TK]D-Fender`	`acxty: Every device seems to ahve the same IP. And you are not showing us a FAILED CALL.`
`17:32.45`	`p3nguin`	`bent_screwdriver: What's the load average on the server?`
`17:33.10`	`acxty`	`what you mean with failed call?`
`17:33.10`	`[TK]D-Fender`	`acxty: Got SIP response 480 "Temporarily Unavailable" back from 192.168.0.101 <--- that's their response. * works. What more are you expecting of *?`
`17:33.11`	`iscario`	`they just have to share a database. So I suppose i do not need odbc support [TK]D-Fender`
`17:33.32`	`bent_screwdriver`	`p3nguin: 0.55, 0.51, 0.65`
`17:33.41`	`[TK]D-Fender`	`iscario: Well perhaps both can make use of a database, but there is nothing they "share" there`
`17:34.09`	`p3nguin`	`All six of the peers have the same IP address. Why would anyone have six softphones running on his computer?`
`17:34.21`	`*** join/#asterisk Tim_Toady (~moi@188.4.66.69.dsl.dyn.forthnet.gr)`
`17:35.37`	`iscario`	`It was what it was told in the tutorial from OpenSIPS : they will have common data apparently [TK]D-Fender`
`17:36.03`	`*** join/#asterisk rare1980_ (~rare1980@12.25.228.67)`
`17:36.12`	`[TK]D-Fender`	`iscario: Feel free to point to some specifics you think you should be caring about.`
`17:37.05`	`iscario`	`[TK]D-Fender: so apparently, you think that i won't need odbc right ?`
`17:37.48`	`acxty`	`[TK]D-Fender, From asterisk I cannot make any call only receive, but if I configure one of does account to xlite I can make calls normally`
`17:38.03`	`[TK]D-Fender`	`iscario: Well you aren't showing anything that requires it yet.`
`17:38.10`	`*** join/#asterisk nachoguy (~boster@nacho-tested.hrapproved.com)`
`17:38.27`	`[TK]D-Fender`	`acxty: the other side refused the call.`
`17:38.34`	`[TK]D-Fender`	`acxty: Go look on that box.`
`17:38.58`	`*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)`
`17:39.00`	`acxty`	`[TK]D-Fender, so if not an asterisk problem? will check the pbx then`
`17:39.03`	`acxty`	`thanks`
`17:39.15`	`nachoguy`	`How do the timeout and retry parameters work for a queue that is staffed by agentlogin (not agent callback login) users?`
`17:41.31`	`iscario`	`[TK]D-Fender: here is my tuto (http://www.opensips.org/Resources/DocsTutAsterisk) : it is just said (1.5) 'You need to have the voicemail support with DB support, so take care and install the unixodbc support in Asterisk'`
`17:41.48`	`devmod`	`When dialing a single digit extension, asterisk replies with a 484 address incomplete... Any pointers?`
`17:42.06`	`iscario`	`[TK]D-Fender: and they are using obdc after`
`17:42.19`	`iscario`	`[TK]D-Fender: so i would prefer to install it`
`17:42.30`	`*** join/#asterisk kartikm (~koolkarti@117.199.127.147)`
`17:43.38`	`[TK]D-Fender`	`devmod: Apparently you don't have a match`
`17:43.58`	`*** join/#asterisk kartik (~koolkarti@117.199.127.147)`
`17:44.44`	`devmod`	`I have "_X."`
`17:45.41`	`nachoguy`	`devmod: it may be in the wrong context`
`17:46.05`	`devmod`	`its inside the default context and it works for any number that is not a single digit`
`17:46.29`	`p3nguin`	`devmod: That's two digits.`
`17:46.50`	`p3nguin`	`X is one, . is one or more = two or more`
`17:47.11`	`p3nguin`	`It's no wonder one-digit extensions won't work.`
`17:47.18`	`devmod`	`oh my bad, I was thinking like it was a reg expression the modifier applied to the previous exp`
`17:47.35`	`Corydon76-dig`	`No, it's a short-circuit operator`
`17:47.38`	`devmod`	`so how do u say 1 digit or more?`
`17:47.44`	`p3nguin`	`_X!`
`17:47.56`	`devmod`	`ohh ok got it, thanks`
`17:48.00`	`p3nguin`	`! = 0 or more`
`17:48.26`	`florz`	`no, ! also has different timing behaviour (IIRC)`
`17:48.36`	`ManxPower`	`um, no, ! means early dial`
`17:49.04`	`ManxPower`	`asterisk as no "0 or more" dialplan pattern`
`17:49.14`	`p3nguin`	`(1247.38) <devmod> so how do [non-English compilation of characters (possibly "you")] say 1 digit or more?`
`17:49.17`	`p3nguin`	`(1247.44) <p3nguin> _X!`
`17:49.22`	`p3nguin`	`You want to try to convince me this is wrong?`
`17:49.46`	`ManxPower`	`p3nguin, Yes! Would you like me to polint out the docs for the "." pattern?`
`17:49.46`	`*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)`
`17:49.55`	`devmod`	`isnt that a digit followed of (0 or more characters) ?`
`17:50.14`	`ManxPower`	`there is a lot of difference between "1 or more" and "0 or more"`
`17:50.34`	`*** join/#asterisk superbeef (~lanej@74.84.194.4)`
`17:50.41`	`superbeef`	`~book`
`17:50.42`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`17:51.00`	`florz`	`p3nguin: it's not "wrong", it's just that there is also a timing component to it`
`17:51.28`	`ManxPower`	`florz, I imagine ! would simply not work for some types of interfaces`
`17:51.44`	`superbeef`	`Will the Asterisk Book address programming syntax in 1.6.x.x?`
`17:51.49`	`ManxPower`	`superbeef, no.`
`17:51.59`	`florz`	`ManxPower: well, that totally depends on what you want to achieve`
`17:52.06`	`devmod`	`uhm, so any ideas how to say "One or more digits"`
`17:52.19`	`ManxPower`	`Actaully, "that depends on which programming syntax you are looking for"`
`17:52.33`	`devmod`	`I am just using extensions.conf`
`17:52.36`	`florz`	`ManxPower: hu?`
`17:52.47`	`ManxPower`	`florz, that was for superbeef`
`17:52.54`	`raden_work`	`can someone explain to me why everytime our internet provider goes down for a few min or i restart our router I have to restart asterisk cause the destination to everywhere seems to be unreachable ?`
`17:52.55`	`raden_work`	`http://pastebin.com/AtriRDuH`
`17:52.56`	`florz`	`oh, ic ;-)`
`17:52.57`	`superbeef`	`ManxPower: bummage`
`17:53.37`	`acxty`	`I have 6 ip address and 6 peers. How can I assing what ip address each peer will use?`
`17:53.38`	`ManxPower`	`superbeef, nod Once you know what "programming syntax" you are asking about, we will be able to provide better answers.`
`17:53.50`	`raden_work`	`[TK]D-Fender, ?`
`17:54.10`	`superbeef`	`I upgraded a 1.4 box to 1.6 and I had some code for pausing and unpausing agents in a queue`
`17:54.17`	`acxty`	`I try using bindaddr on each pear configuration`
`17:54.20`	`acxty`	`peear`
`17:54.52`	`ManxPower`	`superbeef, you didn't find anything useful in the extensive UPGRADE*.txt files included in the Asterisk source code tarball?`
`17:55.06`	`[TK]D-Fender`	`raden_work: disable qualify for those peers`
`17:55.19`	`raden_work`	`and it will work ?`
`17:55.32`	`devmod`	`I took a peek at the book, I don't think it shows a way to do "One or more digits"`
`17:55.39`	`p3nguin`	`devmod: I already told you. _X!`
`17:56.21`	`ManxPower`	`heh! devmod is the only person currently on my /ignire list.`
`17:56.33`	`*** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org)`
`17:56.48`	`acxty`	`The idea is that each peer register with a different ip address to the pbx`
`17:56.57`	`p3nguin`	`devmod: You don't want to use _. like was being discussed above. That'll match ALL things of one character or more, include the standard extensions.`
`17:57.14`	`ManxPower`	`acxty, the idea is a bad one and I doubt it is supported by asterisk`
`17:58.03`	`acxty`	`ManxPower, I am getting the sip accounts from a panasonic pbx, and I need to register each account using different ip for it to work`
`17:58.09`	`p3nguin`	`devmod: As I already mentioned, in the "_X!" pattern, the X is one and the ! is 0 or more, which equals ONE OR MORE digits.`
`17:59.09`	`[TK]D-Fender`	`[13:56]<acxty>The idea is that each peer register with a different ip address to the pbx <- * cannot do this`
`17:59.20`	`superbeef`	`ManxPower: what? I'm supposed to read stuff now?`
`17:59.45`	`ManxPower`	`From the official docs: ; ! - wildcard, causes the matching process to complete as soon as it can unambiguously determine that no other matches are possible`
`18:00.02`	`raden_work`	`[TK]D-Fender, in your words can u please explain to me what qualify does ??? that did seem to fix the issue`
`18:00.09`	`*** join/#asterisk kartik (~koolkarti@117.199.127.147)`
`18:00.36`	`ManxPower`	`raden_work, Qualify sends a SIP OPTIONS packet to the peer and measures the round trip response time.`
`18:00.38`	`[TK]D-Fender`	`raden_work: its sends a keep-alive for NAT purposes and also uses as a way to determine if a host is up. Once it gives up it won't even try to call.`
`18:00.56`	`[TK]D-Fender`	`raden_work: Until it tries to to it again and succeeds.`
`18:01.06`	`raden_work`	`[TK]D-Fender, so something simple like a router going down for 30 seconds just makes it all crazy`
`18:01.09`	`[TK]D-Fender`	`raden_work: disabling this means it will send the call regardless`
`18:01.15`	`raden_work`	`gotcha`
`18:01.16`	`[TK]D-Fender`	`raden_work: yes`
`18:01.40`	`raden_work`	`thank you`
`18:03.19`	`raden_work`	`only thing I dont understand is even when it keeps trying to qualify or register ( whatever its doing ) itll say attempt #10 , attempt #11 and it just never comes back`
`18:03.23`	`raden_work`	`is that normal`
`18:03.44`	`raden_work`	`Im just not going to qualify cause that seems to be proper way to do it , just trying to make sense of it is all`
`18:04.05`	`ManxPower`	`Raden, for some values of "normal", of course. especially if you are doing something silly like using externhost= or something like that`
`18:04.29`	`iscario`	`do you think i can use this driver for mysql-odbc knowing that i am using openBSD (can't test right now, otherwise i would try) ? http://dev.mysql.com/downloads/connector/odbc/#downloads (second one)`
`18:04.50`	`pabelanger`	`question: app_voicemail, when listening to old voicemails, is there a way for make asterisk playback the newest message first? Not oldest?`
`18:04.56`	`ManxPower`	`raden_work, Asterisk was never designed to, nor should it be expected to work when your internet connection is flaky`
`18:05.06`	`raden_work`	`ManxPower, don't have that set anywhere in sip.conf`
`18:05.25`	`*** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com)`
`18:05.42`	`drift-`	`[May 5 09:21:07] NOTICE[16134]: chan_sip.c:14847 handle_request_invite: Call from '211' to extension '****' rejected because extension not found.`
`18:05.44`	`drift-`	`why does that happen?`
`18:05.44`	`raden_work`	`ManxPower, when our internet comes back on asterisk will not work till i restart core unless qualify is disabled`
`18:05.56`	`raden_work`	`it will just keep saying host unreachable even though i can ping thoose hosts`
`18:06.02`	`ManxPower`	`drift-, because extension **** was not found in the expected context`
`18:06.12`	`raden_work`	`example > http://pastebin.com/AtriRDuH`
`18:06.52`	`ManxPower`	`Raden, you are welcome to spend the rest of your life trying to figure this out or you can accept that this is the way things are with Asterisk. It does not work well on bad internet connections`
`18:07.23`	`raden_work`	`ManxPower, I understand , Just trying to understand qualify`
`18:07.53`	`raden_work`	`ManxPower, and honestly everything works really good and now that qualify is off i can unplug router and plug back in and everything works peachy :)`
`18:09.36`	`superbeef`	`does anybody have a favorite asteirsk 1.6 book?`
`18:09.47`	`ManxPower`	`superbeef, one does not exist`
`18:09.48`	`[TK]D-Fender`	`SuP/doc`
`18:10.02`	`superbeef`	`ManxPower: i see 3 on amazon when i search for asterisk 1.6`
`18:10.03`	`ManxPower`	`all significant changes are documented in UPGRADE*.txt`
`18:10.22`	`p3nguin`	`Eventually, a new book will be compiled.`
`18:10.35`	`ManxPower`	`superbeef, write up a review of them.`
`18:10.38`	`superbeef`	`ManxPower: well here's the deal.... I never learned how to program asterisk in the first place, but I have legacy junk from 1.2 days, and I've been slowing upgrading from 1.4 to 1.6 and i keep on having hurdles`
`18:11.07`	`superbeef`	`ManxPower: notes abourt 1 function in the upgrade-1.6.txt isnt really enough`
`18:11.25`	`ManxPower`	`superbeef, nod read ALL the upgrade*.txt files.`
`18:11.29`	`superbeef`	`lol`
`18:11.40`	`ManxPower`	`something that was deprecated in 1.4 and removed in 1.6 won't be mentioned in the 1.6 file.`
`18:11.58`	`raden_work`	`superbeef, I like Practical Asterisk 1.4 & 1.6`
`18:12.08`	`ManxPower`	`because it was mentioned in the 1.4 file as being deprecated and should not be used in 1.4 even thought the features/funcrion/syntax has changed`
`18:12.20`	`superbeef`	`raden_work: thanks--and I see it has 6 stars on amazon`
`18:12.27`	`ManxPower`	`~book`
`18:12.28`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`18:12.35`	`raden_work`	`superbeef, I like it better than the oreilly POS`
`18:12.36`	`ManxPower`	`as you can see there is only one book we recommend here.`
`18:12.52`	`raden_work`	`superbeef, more real world examples and simple explenations`
`18:12.53`	`superbeef`	`all *nix books must have animals on covers`
`18:13.01`	`raden_work`	`LOL`
`18:13.29`	`superbeef`	`that way you can refer to the book by species to show l33tness`
`18:13.42`	`superbeef`	`i have the first edition of the orielly.. it's charming`
`18:13.49`	`raden_work`	`superbeef, I have both and they both suite there purpose i find myself in the practical asterisk book alot more though`
`18:14.13`	`ManxPower`	`Raden, they can't be that good if they don't even mention what qualify does.`
`18:14.25`	`devmod`	`p3nguin, well according to the book you guys referred me to, the ! matches zero or more Characters which is a superset of Digits...`
`18:14.42`	`raden_work`	`ManxPower, neither book really explains what its doing at its core to tell you the truth`
`18:14.58`	`superbeef`	`raden_work: thanks i think i'm gonna order it`
`18:15.00`	`raden_work`	`I really want to know what makes qualify tick and I'm not going to figure that out so I will just live with that`
`18:15.25`	`ManxPower`	`Raden, a simple sip debug will show you the data that is being sent`
`18:15.42`	`superbeef`	`I think its confusing because asterisk is actually built on top of libpurple`
`18:16.04`	`ManxPower`	`superbeef, You have just made it to my /ignore list.`
`18:16.10`	`superbeef`	`hahaha`
`18:16.17`	`raden_work`	`ManxPower, yes I understand that sir, the issue is for some reason Asterisk cannot find a route to the host after the internet goes down if qualify is on ( which makes no sense )`
`18:16.34`	`raden_work`	`I let it sit there over a hour this morning just to see if it eventually would it didnt`
`18:16.42`	`raden_work`	`had to restart asterisk`
`18:16.52`	`raden_work`	`then our net went down this afternoon`
`18:17.10`	`raden_work`	`and doing qualify=no on peers and a sip reload fixed everything`
`18:17.13`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`18:17.16`	`ManxPower`	`Raden, um, asterisk just sends the packet out`
`18:17.31`	`raden_work`	`example > http://pastebin.com/AtriRDuH`
`18:18.03`	`ManxPower`	`Raden, you need to do a tcpdump or wireshark or sip debug or SOMETHING`
`18:18.12`	`[TK]D-Fender`	`[14:16]<raden_work>ManxPower, yes I understand that sir, the issue is for some reason Asterisk cannot find a route to the host after the internet goes down if qualify is on ( which makes no sense ) <- sure it does`
`18:18.15`	`ManxPower`	`ping is not going to cut it.`
`18:18.26`	`[TK]D-Fender`	`raden_work: can't find a route.. IF YOUR INTERNET IS DOWN.`
`18:18.37`	`raden_work`	`[TK]D-Fender, Look at the pings below !!!`
`18:18.50`	`raden_work`	`Internet was up over 20 min at that point`
`18:18.51`	`[TK]D-Fender`	`raden_work: yes.. it was down when it LAST CHECKED.`
`18:19.03`	`[TK]D-Fender`	`raden_work: Who said that status was verified every second?`
`18:19.21`	`p3nguin`	`devmod: Are you still stuck on this? You asked to match 1 or more digits and I provided you with the pattern.`
`18:19.26`	`raden_work`	`[TK]D-Fender, that was done approximately 20 min after internet came back up`
`18:19.39`	`raden_work`	`it will sit there for an entire 24 hours if i let it and just keep doing that`
`18:19.40`	`[TK]D-Fender`	`raden_work: Not sure what would prompt * to retry`
`18:19.55`	`raden_work`	`it said it was on try #10`
`18:20.09`	`*** part/#asterisk ManxPower (~manxpower@193.sub-75-254-10.myvzw.com)`
`18:20.12`	`raden_work`	`that one day i came on the try number was in the 1000's was down for entire weekend`
`18:21.46`	`raden_work`	`does that make sense ?`
`18:22.11`	`[TK]D-Fender`	`raden_work: was internet service up while those failures were still flying?`
`18:22.56`	`raden_work`	`yes sir`
`18:23.17`	`raden_work`	`we had internet back at failure #3`
`18:23.25`	`raden_work`	`that pastebin was done on #10`
`18:24.13`	`[TK]D-Fender`	`raden_work: what are these #'s exactly?`
`18:24.18`	`raden_work`	`and I was ssh'ed into our co-lo's at vitelity the whole time so if the route was down to vitelity itself from here I dont know how i could be accessing our co-lo's there`
`18:24.41`	`raden_work`	`[TK]D-Fender, registration failure trying again attempt #10`
`18:24.58`	`raden_work`	`with qualify=no and SIP reload came right back up within seconds`
`18:25.09`	`raden_work`	`Grrr buffer wont scroll back to them`
`18:25.22`	`raden_work`	`there a way to log console ?`
`18:25.33`	`[TK]D-Fender`	`raden_work: I'd have liked to have seen that... also with a tcpdump on it`
`18:25.38`	`*** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk)`
`18:25.58`	`raden_work`	`lemee psatebin everything i do have`
`18:26.44`	`raden_work`	`http://pastebin.com/0gG0aeKt`
`18:26.52`	`raden_work`	`not much but all i have left from it`
`18:27.17`	`raden_work`	`line #52 [TK]D-Fender`
`18:28.49`	`raden_work`	`interesting post on same issue http://www.freepbx.org/forum/freepbx/installation/sipstation-dropping-registration`
`18:28.59`	`[TK]D-Fender`	`raden_work: that's a reg timeout, nothing to do with qualify. Something is still amiss with your networking`
`18:29.06`	`[TK]D-Fender`	`raden_work: and that didn't include SIP DEBUG`
`18:29.23`	`raden_work`	`didnt have it on at that moment`
`18:29.40`	`raden_work`	`but the fact is the only time it will do that is when the internet goes down and comes back up`
`18:29.44`	`devmod`	`p3nguin, k nevermind thanks`
`18:29.54`	`raden_work`	`and with qualify = no ( the problem went away )`
`18:30.04`	`raden_work`	`before I could unplug router for 30 seconds recreate the problem`
`18:30.11`	`raden_work`	`now everything is peachy`
`18:31.17`	`raden_work`	`Also Vitelity sends calls to our IP we dont have to be registered and when vitelity is unreachable i see the calls come in but asterisk rejects them and then failover kicks in`
`18:31.18`	`drift-`	`[May 5 09:46:40] WARNING[16690]: app_macro.c:256 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'`
`18:31.20`	`drift-`	`whats that mean?`
`18:31.41`	`raden_work`	`No such context 'macro-stdexten'`
`18:31.53`	`[TK]D-Fender`	`^^^`
`18:32.20`	`[TK]D-Fender`	`Doctor, doctor!!! It hurts when I raise my arm like this!`
`18:33.19`	`*** join/#asterisk MiserySoft (~lnd@89.193.103.41)`
`18:33.43`	`*** join/#asterisk theHub (~theHub@69.177.93.21)`
`18:34.07`	`raden_work`	`[TK]D-Fender, so here is my conclusion when qualify=yes , and the internet goes down , asterisk cannot find a route to any host until a core restart is done , which is why it cannot register .`
`18:34.44`	`raden_work`	`when qualify=no there is no problems internet goes down and comes back up all is good`
`18:35.17`	`[TK]D-Fender`	`raden_work: that should not change your reg's failing as you showed`
`18:35.29`	`[TK]D-Fender`	`raden_work: there is no link between the reg & peer`
`18:35.35`	`raden_work`	`[TK]D-Fender, correct they failed`
`18:35.40`	`drift-`	`chan_sip.c:14780 handle_request_invite: Failed to authenticate user "#####" <sip:##@ip>;tag=as7c4da8bc`
`18:35.42`	`drift-`	`why does that happen?`
`18:35.52`	`[TK]D-Fender`	`drift-: beacuse the auth is wrong`
`18:36.01`	`raden_work`	`but they only will fail when peers are unreachable, I know seems illogical but I have seen it happen 10+ times in last 6 months`
`18:36.40`	`raden_work`	`drift-, here is your sign "Failed to authenticate user "`
`18:36.59`	`drift-`	`yeah i can read thanks`
`18:37.04`	`drift-`	`i dont know where to edit this and fix this`
`18:37.46`	`[TK]D-Fender`	`drift-: sip.conf , [#####]`
`18:38.06`	`[TK]D-Fender`	`drift-: Or the device that's actually trying to connect`
`18:38.16`	`raden_work`	`BRB 30 MIN`
`18:38.20`	`[TK]D-Fender`	`drift-: Noone said * was the one who should be changed`
`18:40.51`	`*** join/#asterisk kartik (~koolkarti@117.199.127.147)`
`18:42.57`	`*** join/#asterisk bsaxon (~bsaxon@174.47.44.19)`
`18:47.16`	`*** join/#asterisk pkecastillo (~pirruar@190.113.141.122)`
`18:50.22`	`hardwire`	`anybody use deny/permit in sip.conf?`
`18:50.22`	`*** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com)`
`18:50.36`	`*** join/#asterisk kartikm (~koolkarti@117.199.127.147)`
`18:51.44`	`*** join/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com)`
`18:51.50`	`jmls`	`hey y'all`
`18:52.02`	`jmls`	`using the group function in 1.4`
`18:52.08`	`jmls`	`should I use`
`18:52.12`	`jmls`	`a) Set(GROUP(inuse)=${EXTEN})`
`18:52.16`	`p3nguin`	`are you`
`18:52.18`	`p3nguin`	`really`
`18:52.20`	`p3nguin`	`going to`
`18:52.20`	`jmls`	`b) Set(GROUP()=${EXTEN}@inuse)`
`18:52.22`	`p3nguin`	`talk like`
`18:52.28`	`p3nguin`	`this`
`18:52.30`	`p3nguin`	`the whole time`
`18:52.31`	`jmls`	`p3nguin: yup :)`
`18:52.32`	`p3nguin`	`you are`
`18:52.33`	`p3nguin`	`here?`
`18:52.47`	`garymc`	`Anyone know how I can see if port 5060 is getting used on my router at home?`
`18:53.08`	`jmls`	`no, actually just wanted to put the two examples on separate lines, got carried away ;)`
`18:53.21`	`jmls`	`p3nguin: sorry for offending you`
`18:55.49`	`garymc`	`ive now got my Iphone using softphone through a VPN`
`18:56.02`	`garymc`	`so wanted to see if I can see port 5060 being used`
`18:56.15`	`p3nguin`	`Define "being used"`
`18:57.06`	`garymc`	`im using udp 5060`
`18:57.17`	`garymc`	`or I suspect someone is using port 5060 how would I know?`
`18:57.44`	`garymc`	`perhaps someone is making voip calls through my wireless connection when i specify people not too`
`18:57.52`	`garymc`	`:)`
`18:58.13`	`garymc`	`Is there a way of checking my Router log files to see if thats the case?`
`18:59.10`	`p3nguin`	`Only you would know, since we aren't your netadmins.`
`19:00.06`	`kartik`	`I am an asterisk newbie, I have configured asterisk to call in the LAN using xlite softphone but the problem is that the softphone is not detecting speakers and mike, Is there any way to figure this out`
`19:01.04`	`fenrus`	`check the softphone config`
`19:01.12`	`beefpastry`	`xlite for windows or Linux?`
`19:01.23`	`kartik`	`beefpastry: xlite for linux`
`19:01.27`	`kartik`	`using fedora`
`19:01.41`	`garymc`	`p3nguin : Yes im not sure how to check this or can it be checked?`
`19:01.57`	`beefpastry`	`audacity works ok?`
`19:01.58`	`p3nguin`	`kartik: Try using twinkle instead of xlite.`
`19:02.25`	`beefpastry`	`I second that...`
`19:02.36`	`kartik`	`fenrus: tried that but it is not detecting, i don't have any other hardware apart from the default speaker or mike, does it require other specific hardware?`
`19:02.40`	`p3nguin`	`X-Lite barely works on Windows.`
`19:02.42`	`*** join/#asterisk paparent (~paparent@unaffiliated/paparent)`
`19:02.51`	`p3nguin`	`I can't imagine how terribly it must be on Linux.`
`19:03.12`	`kartik`	`p3nguin: is twinkle free?`
`19:03.15`	`p3nguin`	`yes`
`19:03.19`	`Knightfal`	`imagine Horrible`
`19:03.20`	`beefpastry`	`it's also only an old version`
`19:03.20`	`paparent`	`Hi! I've a home setup for asterisk, and dunno why, after a 30mins call, the connection is stopped`
`19:03.30`	`beefpastry`	`should be able to yum it`
`19:03.47`	`beefpastry`	`(twinkle is in the fedora repo)`
`19:03.53`	`kartik`	`beefpastry: thanks`
`19:07.50`	`hardwire`	`sigh.`
`19:08.18`	`kartik`	`and calling from linux to windows and vice versa is also possible via twinkle? it sucks using xlite`
`19:08.49`	`beefpastry`	`?`
`19:09.24`	`p3nguin`	`kartik: It's a softphone using the same protocol as another softphone... why would the platform be an issue?`
`19:09.24`	`beefpastry`	`I used it to call my wife on xlite for windows without a hitch...just SIP`
`19:10.18`	`beefpastry`	`The only way that it wouldn't work is if Bill Gates did something strange...but you`
`19:10.24`	`beefpastry`	`would have to call him for that.`
`19:11.59`	`nachoguy`	`How do the timeout and retry parameters work for a queue that is staffed by agentlogin (not agent callback login) users?`
`19:12.49`	`p3nguin`	`nachoguy: When the timeout is reached, it ends.`
`19:13.33`	`nachoguy`	`sorry, the timeout in the queues.conf, not in the dialplan`
`19:13.38`	`devmod`	`how can I redirect an existing SIP call into a new ConfBridge?`
`19:14.25`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`19:14.57`	`nachoguy`	`it's supposed to be how long asterisk will try to call an available agent (using agent callback login) type of queues. With agent login type of queues, you don't have to worry about a rep not picking up.`
`19:14.58`	`p3nguin`	`nachoguy: I guess I don't understand where your confusion comes from.`
`19:15.04`	`*** join/#asterisk kartikm (~koolkarti@117.199.127.147)`
`19:15.44`	`p3nguin`	`The timeout and retry settings are not concerned with how the members are connected with the queue.`
`19:16.55`	`kartik`	`beefpastry: it just didn't work in windows, wasn't able to register to a linux server, I was implementing this in local lan`
`19:17.36`	`kartik`	`beefpastry p3nguin I'll try twinkle and will get back to you if there will be some doubts :)`
`19:17.50`	`*** join/#asterisk Alagar (~Administr@122.164.35.157)`
`19:17.56`	`beefpastry`	`kartik: probably a lan/firewall or config issue...os has nothing to do with it.`
`19:18.30`	`*** join/#asterisk Alii (~alii@nat/digium/x-diysitwllsndzxdo)`
`19:20.50`	`[TK]D-Fender`	`devredirect from where?`
`19:21.00`	`[TK]D-Fender`	`devmod: redirect from where?`
`19:21.03`	`*** join/#asterisk kartikm (~koolkarti@117.199.127.147)`
`19:21.11`	`raden_work`	`BACK`
`19:22.38`	`devmod`	`[TK]D-Fender, A in a call with B. Move them into a ConfBridge`
`19:23.22`	`[TK]D-Fender`	`devmod: Last I checked there wasn't a way to send both parties somewhere. Usually you could target ONE, but the other got dropped off the planet`
`19:23.36`	`[TK]D-Fender`	`devmod: You'd need to check * 1.6.2 AMI actions like "redirect"`
`19:24.27`	`*** join/#asterisk lanning (~lanning@208.87.235.224)`
`19:24.54`	`*** join/#asterisk TehRabbitt (~rabbott56@18263008.cst.lightpath.net)`
`19:24.58`	`TehRabbitt`	`i'm back`
`19:25.07`	`devmod`	`[TK]D-Fender, ohh :/ that sucks. I think I was talking to you about this the other day. Adding a third person to an existing call. Still haven't found a way to do it`
`19:25.46`	`TehRabbitt`	`p3nguin: they had one 7961 (6 extention) that's it :-\ a couple of the 7940's though one is a G one isn't.... only downside is one is missing the keypad / buttons are missing`
`19:25.56`	`TehRabbitt`	`now I did pick up 2 other phones that looked "cool" lol`
`19:26.10`	`TehRabbitt`	`Congruency IPicasso 6000`
`19:26.23`	`TehRabbitt`	`gotta figure out if they'll work with asterisk though`
`19:26.29`	`p3nguin`	`I'll pretend like you said 6 lines and not 6 extensions.`
`19:26.40`	`TehRabbitt`	`p3nguin: i meant lines ;)`
`19:26.41`	`TehRabbitt`	`lo`
`19:26.42`	`TehRabbitt`	`lol`
`19:27.06`	`[TK]D-Fender`	`devmod: do a 3-way call from a phone`
`19:27.06`	`TehRabbitt`	`They had a TON of avaya stuff`
`19:27.10`	`TehRabbitt`	`and polycom`
`19:27.11`	`*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)`
`19:27.11`	`p3nguin`	`How much did he want for the phones that were not broken?`
`19:27.23`	`TehRabbitt`	`i think he said 50 for the 6 line one`
`19:27.28`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`19:27.29`	`TehRabbitt`	`though it doesnt come with the PoE adapter`
`19:27.39`	`devmod`	`[TK]D-Fender, I can't use the endpoint for the mixing. They are all softphones.`
`19:27.39`	`p3nguin`	`none do`
`19:27.43`	`*** join/#asterisk kartikm (~koolkarti@117.199.127.147)`
`19:27.52`	`p3nguin`	`You either get a power cube or use a PoE switch.`
`19:27.52`	`[TK]D-Fender`	`devmod: So?`
`19:27.53`	`TehRabbitt`	`lots of avaya stuff though, LOTS of it 0-o lol`
`19:27.56`	`[TK]D-Fender`	`devmod: Why not?`
`19:28.10`	`TehRabbitt`	`several of those polycom confrence speakerphone thigns too`
`19:28.24`	`TehRabbitt`	`the triangular ones`
`19:28.25`	`devmod`	`[TK]D-Fender, well for once not all of them have that functionality and also I don't want the person to be put on hold while the other party is being called`
`19:28.34`	`TehRabbitt`	`p3nguin: this is what i picked up though: http://www.pcb-vd.atworkweb.com/sw/swchannel/productcatalogcf_v2/internet/model.asp/defTabName/details/ProductMasterID/13124/RequestType/orderquote/PARENTID/28965/Refer/ProductCatalog%7C%7C/SWID/1`
`19:28.36`	`TehRabbitt`	`2 of those`
`19:28.46`	`*** join/#asterisk moos3 (~rgenthner@216.52.121.66)`
`19:28.47`	`p3nguin`	`$50 for a 7961G without a power brick is pretty much a regular ebay price.`
`19:28.56`	`moos3`	`what ports do sip clients us?`
`19:28.56`	`TehRabbitt`	`i could prob talk him down though lol`
`19:29.02`	`[TK]D-Fender`	`devmod: Go look then`
`19:29.26`	`p3nguin`	`I'd have to get it down to like $20 'cause then I'd have to pay for shipping.`
`19:29.30`	`TehRabbitt`	`p3nguin: http://www.corporategroup.co.uk/Telrad/ipicasso.html`
`19:29.52`	`devmod`	`[TK]D-Fender, yeah I have been for a while, cant find a way yet so I thought I would ask here see if anyone had any ideas. thanks`
`19:29.56`	`TehRabbitt`	`well he said I could 'sell' stuff for him / give him a % of whatever i sell it for..... soooooo i tell him i sold it for 50 bucks, i give him $10 lol :-p`
`19:30.02`	`TehRabbitt`	`might be an option`
`19:30.08`	`TehRabbitt`	`that is 20% lol`
`19:30.17`	`TehRabbitt`	`hell even 50% would be 25 for the phone`
`19:30.51`	`p3nguin`	`If you can get a 7961G in good working condition for $25, I'll take it.`
`19:30.59`	`TehRabbitt`	`aight`
`19:31.06`	`TehRabbitt`	`btw did u see the link i sent?`
`19:31.11`	`p3nguin`	`which?`
`19:31.13`	`TehRabbitt`	`http://www.corporategroup.co.uk/Telrad/ipicasso.html`
`19:31.16`	`p3nguin`	`yes`
`19:31.17`	`TehRabbitt`	`any good? i got 2 of them`
`19:31.21`	`p3nguin`	`no idea`
`19:31.23`	`TehRabbitt`	`with the power bricks for them :-\`
`19:31.31`	`TehRabbitt`	`i turn them on they say "checking for firmware updates"`
`19:31.37`	`TehRabbitt`	`color screens`
`19:32.05`	`p3nguin`	`If they're SIP phones, they should work fine with asterisk.`
`19:32.24`	`TehRabbitt`	`http://www.signalvoiceanddata.com/brochures/Connegy_iPicasso_05.pdf`
`19:33.43`	`TehRabbitt`	`apparently it's XML driven...`
`19:33.46`	`p3nguin`	`When do you think you'll go back up there?`
`19:34.00`	`TehRabbitt`	`proabbly next week... he gets more in like every week`
`19:34.08`	`TehRabbitt`	`I have to say though, the avaya phones he has, are VERY nice 0_o`
`19:34.18`	`TehRabbitt`	`some had full color screens with 20 lines built in`
`19:34.26`	`TehRabbitt`	`even has the addon blocks for them 0_o`
`19:34.34`	`TehRabbitt`	`look kinda "boxy" though some of them`
`19:35.07`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`19:35.30`	`p3nguin`	`A 7971G would be a nice find for me.`
`19:36.14`	`p3nguin`	`I'd settle on a 7961G, though. ;)`
`19:36.17`	`TehRabbitt`	`lol`
`19:36.23`	`*** join/#asterisk jpcansa (~jpbenavid@201.196.95.200)`
`19:36.33`	`TehRabbitt`	`they had a TON of avaya video ip phones which idk what you'd do with those lol`
`19:36.36`	`TehRabbitt`	`even a projector version`
`19:36.42`	`TehRabbitt`	`one had PTZ capability`
`19:36.44`	`p3nguin`	`set up video phones...`
`19:36.48`	`p3nguin`	`That's what you'd do with them.`
`19:36.49`	`TehRabbitt`	`hm`
`19:36.58`	`p3nguin`	`Asterisk should support video, I think.`
`19:36.59`	`TehRabbitt`	`one is like a polycom confrence one with a motorized camera`
`19:37.06`	`jpcansa`	`Hi, What is the maximum number of simultaneous calls Asterisk can handle??`
`19:37.11`	`TehRabbitt`	`you can apparently move the camera using IP`
`19:39.07`	`p3nguin`	`Now that I got chan_sccp going, I'm more likely to use sccp and I can use those extra line keys for speed dials.`
`19:39.30`	`*** join/#asterisk ruben23 (~ITadmin@125.212.40.2)`
`19:40.00`	`paparent`	`Whichi hardphone do you suggest ?`
`19:40.18`	`p3nguin`	`That depends on what you like and want.`
`19:40.33`	`TehRabbitt`	`:-\ sigh... http://ibot.rikers.org/%23asterisk/20050331.html.gz`
`19:40.40`	`TehRabbitt`	`read throught that... ipicasso`
`19:40.54`	`p3nguin`	`not good?`
`19:41.00`	`TehRabbitt`	`propiatary`
`19:41.05`	`TehRabbitt`	`simialr to sccp`
`19:41.06`	`TehRabbitt`	`i think`
`19:41.08`	`p3nguin`	`not SIP?`
`19:41.10`	`TehRabbitt`	`not sure yet`
`19:41.15`	`TehRabbitt`	`apparently they need firmware :-\`
`19:42.22`	`TehRabbitt`	`they had a TON of fxo / fxs cards and a ton of T1 cards 0_o`
`19:42.43`	`*** join/#asterisk diegomad (~mad@190.146.200.120)`
`19:42.53`	`TehRabbitt`	`i forget the brand though but i know it's a name brand too`
`19:43.10`	`drmessano`	`It takes CCCP to run SCCP`
`19:43.17`	`Naikrovek`	`jpcansa: depends entirely on the hardware you choose to run asterisk on`
`19:43.41`	`Naikrovek`	`p3nguin: chan_sccp working successfully? I've been non-lurking today`
`19:44.09`	`p3nguin`	`naikrovek: Yeah, tehrabbitt and I are both having success with chan-sccp-b and Cisco phones.`
`19:44.18`	`Naikrovek`	`nice`
`19:44.25`	`p3nguin`	`It seems like all the features are working.`
`19:44.39`	`TehRabbitt`	`yep`
`19:44.46`	`TehRabbitt`	`now if only i could get incoming SIP working haha`
`19:44.58`	`p3nguin`	`park, transfer, iDivert, direct transfer (blind transfer), group pickup`
`19:45.21`	`Naikrovek`	`nice!`
`19:45.23`	`Naikrovek`	`hrm.`
`19:45.31`	`p3nguin`	`I disabled meetme on the phone because it wasn't working like I expected. I probably just need to configure it.`
`19:45.36`	`Naikrovek`	`feels the desire for a Cisco phone grow a bit.`
`19:45.43`	`lordvadr`	`The ? character doesn't seem to work in *'s regular expression syntax. Is there another way to get a minimal match?`
`19:46.06`	`jpcansa`	`Naikrovek: so if i wanna use a pretty good HP server 4GB RAm 3.0Ghz with 2 Digium TE420 cards, can it handle 240 simultaneous incomming calls from the 8 E1´s ??`
`19:46.20`	`Naikrovek`	`jpcansa: easy`
`19:46.20`	`*** join/#asterisk MiserySoft (~lnd@109.180.131.48)`
`19:46.30`	`Naikrovek`	`jpcansa: probably a few thousand calls in total`
`19:46.41`	`jpcansa`	`really? thats great then`
`19:46.50`	`jpcansa`	`thanks`
`19:46.54`	`Naikrovek`	`not a few thousand in a single meetme room, but a fwe thousand simultaneous point to point calls easy`
`19:47.40`	`p3nguin`	`naikrovek: chan_skinny that ships with Asterisk does not even compare to chan_sccp. I was rather surprised.`
`19:47.49`	`jpcansa`	`Naikrovek: can it handle the 240 simultaneous calls into one single IVR?`
`19:48.12`	`TehRabbitt`	`Ah thats the company name... Digium... he had a ton of the T1/E1 digium cards lol`
`19:48.20`	`*** part/#asterisk paparent (~paparent@unaffiliated/paparent)`
`19:48.46`	`Naikrovek`	`jpcansa: yeah. people have tested hardware older than that, slower than that, and done 1000 simultaneous calls before they saw problems`
`19:48.53`	`Naikrovek`	`jpcansa: there's a voip-info wiki page on it`
`19:49.14`	`jpcansa`	`i´m looking at that right now ;)`
`19:49.20`	`*** join/#asterisk kartik (~koolkarti@117.199.127.147)`
`19:50.00`	`TehRabbitt`	`so wait, if that server can handle all that... my server which is dual 3.8 dual-core Xeons with 6GB of ram running x64 bit is way overkill? lmao`
`19:50.27`	`Naikrovek`	`TehRabbitt: well you could use it for other things too`
`19:50.31`	`Naikrovek`	`or just virtualize it like i did`
`19:50.33`	`TehRabbitt`	`well I am lol`
`19:50.34`	`Naikrovek`	`it works great, btw`
`19:50.39`	`TehRabbitt`	`using for other things i mean`
`19:51.39`	`jaytee`	`Intel has currently has 48-core chips based on the i7 series in their R&D labs for testing`
`19:51.40`	`Naikrovek`	`speaking of, i need to expand the virtual disk of my clearcase server, then i have to do something in the OS to make it use that extra space, i'm thinking`
`19:51.48`	`Naikrovek`	`jaytee: omfg`
`19:52.10`	`Naikrovek`	`what do you do in redhat to expand an existing partition`
`19:52.11`	`Naikrovek`	`hrm.`
`19:52.13`	`Naikrovek`	`google.s`
`19:52.17`	`Naikrovek`	`googles*`
`19:52.23`	`jaytee`	`LVM?`
`19:52.43`	`drmessano`	`jaytee: 48-core chips could scoop a lot of dip`
`19:52.43`	`Naikrovek`	`looks that way`
`19:52.48`	`Naikrovek`	`drmessano: lol`
`19:52.57`	`Naikrovek`	`these ain't your daddy's chips!`
`19:53.08`	`*** join/#asterisk bsaxon (~bsaxon@174.47.44.19)`
`19:53.14`	`drmessano`	`NEW YORK CITY!`
`19:53.18`	`ecrane`	`get a rope`
`19:53.21`	`Naikrovek`	`lol`
`19:54.53`	`drmessano`	`Apparently all my trips to NYC never included travel through the Picante/Salsa manufacturing districts of the 5 boroughs`
`19:55.49`	`ruben23`	`hi is it fine to used intel Atom based PC client , using softphones and web access on an asterisk server..can it handle the load...`
`19:56.25`	`*** join/#asterisk moos3 (~rgenthner@216.52.121.66)`
`19:57.53`	`Naikrovek`	`ruben23: yeah probably`
`19:57.53`	`Naikrovek`	`not for a server, but for a softphone, sure`
`19:58.03`	`garymc`	`Anyone help me here I know its the wrong room but the only room Ive ever got any help. Ive setup this PPTP VPN on my asterisk box I can connect fine with my Iphone. But If i close the VPN on iphone then try to reconnect to the VPN it wont let me`
`19:58.09`	`drmessano`	`Easily.. Find something that rhymes with "You Bun Too?" and you're good to go`
`19:58.15`	`garymc`	`Anyone know where I can get some solid info for this?`
`19:58.22`	`TehRabbitt`	`anyone here heard of Avanti?`
`19:58.33`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`19:58.47`	`Naikrovek`	`it's an italian place on main street`
`19:58.49`	`Naikrovek`	`avantis`
`19:58.53`	`Naikrovek`	`oh Avanti`
`19:58.54`	`Naikrovek`	`no`
`19:59.00`	`TehRabbitt`	`:(`
`19:59.37`	`drmessano`	`Isn't the Avanti a car?`
`19:59.49`	`Naikrovek`	`garymc: what's the log on the asterisk box say when you try to connect again? also, how long after you connect do you try again?`
`20:00.06`	`garymc`	`im trying every 10 mins.`
`20:00.13`	`garymc`	`Maybe ill try again in an hour.`
`20:00.22`	`Naikrovek`	`garymc: could it be a NAT hold that's closed preventing you?`
`20:00.40`	`garymc`	`Well it works if I restart PPTP via ssh`
`20:00.47`	`jaytee`	`Naikrovek, here's an old link I used once for FC7 but it should work in RHEL http://allaboutfedora.blogspot.com/2007/01/how-to-resize-or-expand-lvm-partitions.html`
`20:01.07`	`garymc`	`I need to put this to bed`
`20:01.09`	`garymc`	`:S`
`20:02.09`	`garymc`	`Ill show you a log file, Naikrovek of being connected disconnecting then trying to reconnect.`
`20:02.25`	`TehRabbitt`	`avanti softphone SP15`
`20:02.28`	`TehRabbitt`	`google it...`
`20:02.28`	`pabelanger`	`Are you able to change the playback order of old message withing voicemail? oldest to newest or newest to oldest?`
`20:02.31`	`TehRabbitt`	`donno what protocol it uses`
`20:02.32`	`*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)`
`20:02.45`	`TehRabbitt`	`it asks for a DN and an ECC`
`20:02.52`	`TehRabbitt`	`then a TVSe`
`20:03.11`	`saisoma`	`I have an issue with grandstream 2000 phones and asterisk. I'm not in a position to show config info, but the phones don't ring like they should. They register and ring for a bit, but then seem to "timeout" even though I can pick up the phone and call out, incoming calls do not ring the phone.`
`20:03.37`	`jaytee`	`~gs`
`20:03.38`	`infobot`	`GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.`
`20:03.39`	`saisoma`	`yet asterisk shows the phone being called (SIP/230 is one extension)`
`20:04.04`	`saisoma`	`jaytee, i agree, but boss said he could save $10 per phone (idiot) and I got stuck with 12 of them. Wanted polycoms, which I know well`
`20:04.21`	`jaytee`	`really? I had an idiot boss just like him`
`20:04.32`	`jaytee`	`~cheap`
`20:04.33`	`infobot`	`methinks cheap is 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.`
`20:04.44`	`saisoma`	`hah. yup`
`20:04.45`	`jaytee`	`Polycom FTW!!!`
`20:04.52`	`leifmadsen`	`~polycom`
`20:04.52`	`infobot`	`polycom is probably The Polycom Song by dialing sip:polycom@leifmadsen.comISN 7659*460`
`20:04.57`	`Naikrovek`	`Polycom FTW!!!!!!`
`20:05.03`	`leifmadsen`	`haha I thought it said other stuff too`
`20:05.43`	`saisoma`	`i have a couple hundred polycoms at my main position that I manage and they work awesomely. He saved $120 and my time is $100/hour when I bill him. SHeesh. It's not even saving him any money.`
`20:06.12`	`leifmadsen`	`infobot: no, polycom is The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460. Polycom phone are devices that are favoured by much of the community and range in price from under $100 and upwards.`
`20:06.13`	`infobot`	`leifmadsen: okay`
`20:06.15`	`Naikrovek`	`polycoms are awesome to administer in large numbers`
`20:06.41`	`jaytee`	`I got stuck with two GSX-2000s and I had all kinds of problems. Jitter for one and the MOH would drop out unless you kept making some noise into the handset when you were on hold because of a firmware bug that showed silence suppression turned off when it was still enabled.`
`20:07.31`	`TehRabbitt`	`anyone hear of what i posted?`
`20:08.02`	`jaytee`	`when I finally got my first Polycom it felt like breaking up with a real ugly chick and then scoring a total hottie.`
`20:08.16`	`garymc`	`Naikrovec : http://pastebin.com/aF0EiPxs message logs of connecting using internet via VPN. Closing the VPN connection then trying to establish a fresh VPN connection`
`20:08.48`	`saisoma`	`jaytee, lol. i started with polycoms and i'm feeling all nasty and dirty when working with the grandstreams:(`
`20:08.54`	`garymc`	`putty on the iphone`
`20:08.59`	`garymc`	`whoops wrong room`
`20:09.27`	`MiserySoft`	`jaytee: In my world a grandstream is a hottie !`
`20:10.17`	`beek`	`Polycom -- the desk set of the gods.`
`20:10.28`	`*** join/#asterisk delrocas (~delrocas@71-34-223-180.spkn.qwest.net)`
`20:12.18`	`delrocas`	`Does a TDM400p work as well in a 100 MHZ PCI-X slot as it would in a 133 MHZ PCI-X slot?`
`20:19.06`	`Naikrovek`	`when I hit my first home run, my Polycom was there. When I got my first car, Polycom was there. When I scored with the prom Queen, my GrandStream was there, and it turned out she had chlamidia. When I got married, my Polycom was there.`
`20:19.46`	`Naikrovek`	`delrocas: probably`
`20:20.04`	`Naikrovek`	`delrocas: there's not a lot of data going over voice calls, i don't think the speed will matter much.`
`20:20.28`	`Baylink-work\|afk`	`Grandstream; $59... and 24 hours setup. Aastra, $300... and 1 hour setup.`
`20:20.34`	`Naikrovek`	`hah`
`20:20.43`	`Baylink-work\|afk`	`Polycom... $150. Someone else doing the setup... priceless.`
`20:20.45`	`Naikrovek`	`polycom, $80, 3 minutes setup`
`20:21.04`	`Baylink-work\|afk`	`It was the best I could do on short notice; it's mediocre.`
`20:21.18`	`Naikrovek`	`it's good`
`20:21.43`	`[TK]D-Fender`	`Naikrovek: 3 ..... minutes? You'd better be including assemly in that...`
`20:21.49`	`Naikrovek`	`[TK]D-Fender: i am`
`20:21.59`	`Naikrovek`	`i don't even have to do it.`
`20:22.13`	`Naikrovek`	`i just hand the box to the user, user opens it`
`20:22.18`	`Naikrovek`	`sets it up, plugs it in, works.`
`20:23.03`	`jaytee`	`be back later`
`20:23.07`	`*** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`20:23.24`	`*** join/#asterisk moos3 (~rgenthner@216.52.121.66)`
`20:23.45`	`garymc`	`No worries I just bought a small Putty type app for the Iphone so if I lose VPN connection I just reboot the PPTP server using putty`
`20:23.50`	`garymc`	`small fix for now`
`20:23.53`	`garymc`	`:S`
`20:24.04`	`Baylink-work\|afk`	`When in doubt, use brute force.`
`20:24.27`	`garymc`	`yeah cost Â£2.99 for a putty APP too :eek`
`20:25.05`	`garymc`	`infact the money ive spent and time trying to do this has prob cost more than the calls Ill be making if I was to use the uk phone network to make them,`
`20:25.14`	`garymc`	`any how good night and god bless!`
`20:25.19`	`Naikrovek`	`night dude`
`20:25.22`	`Naikrovek`	`feh i'm always too late`
`20:25.42`	`[TK]D-Fender`	`checkout time, later all`
`20:30.39`	`*** join/#asterisk rare1980_ (~rare1980@203.175.76.218)`
`20:31.02`	`rare1980_`	`hi all in asterisk 1.4 we load g729 codec with this command "module load codec_g729.so"`
`20:31.17`	`rare1980_`	`how we can load g729 codec in asterisk 1.2`
`20:33.56`	`ChannelZ`	`modules.conf`
`20:34.04`	`Naikrovek`	`put it in the directory with the other modules and restart asterisk`
`20:34.07`	`Naikrovek`	`i think...`
`20:34.11`	`Naikrovek`	`or, what ChannelZ said`
`20:34.15`	`leifmadsen`	`ya it should load automatically`
`20:34.20`	`leifmadsen`	`assuming it can even load in 1.2 :)`
`20:34.22`	`ChannelZ`	`although I'm not sure that's what he's actually asking`
`20:34.32`	`leifmadsen`	`rare1980_: use the "load" command`
`20:34.41`	`*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)`
`20:35.25`	`*** join/#asterisk jnicola (~jnicola@186.124.123.2)`
`20:35.33`	`jnicola`	`hello all`
`20:35.55`	`jnicola`	`anybody know something about dynamic hints?`
`20:36.49`	`jnicola`	`i add a hint by dialplan whit exten => _XXX.,hint,SIP/${EXTEN}`
`20:37.38`	`jnicola`	`and then.. with extensionstate. i get always empty hint: and status = -1`
`20:38.29`	`rare1980_`	`leifmadsen: i wrote this in asterisk cli > load codec_g729.so but i got a msg`
`20:38.30`	`rare1980_`	`May 5 16:37:36 WARNING[15424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/codec_g729.so: undefined symbol: ast_trans_frameout`
`20:38.58`	`leifmadsen`	`rare1980_: probably means the version of codec_g729.so you're loading is incompatible with Asterisk 1.2`
`20:39.07`	`rare1980_`	`humm`
`20:39.08`	`rare1980_`	`may be`
`20:39.13`	`leifmadsen`	`likely :)`
`20:39.21`	`rare1980_`	`from where is can get for asterisk 1.2`
`20:39.22`	`rare1980_`	`:)`
`20:39.39`	`leifmadsen`	`not sure -- I don't know for sure if there is a codec_g729 for asterisk 1.2`
`20:39.42`	`delrocas`	`Naikrovek, thanks`
`20:40.13`	`rare1980_`	`humm`
`20:42.16`	`*** join/#asterisk lost_sou1 (~none@cpe-67-241-68-202.twcny.res.rr.com)`
`20:42.39`	`*** join/#asterisk halkun (~chatzilla@mke-67-208-60-183.milwpc.com)`
`20:44.06`	`halkun`	`Ok, I just bought a SPA3102. If setting this thing up causes me to go into a blood-thirsty rage, I swear I'll take a hostage.`
`20:44.51`	`Kobaz`	`anyone have a blackberry`
`20:45.05`	`drmessano`	`Millions do`
`20:45.07`	`Kobaz`	`when i delete an email from my berry, it's supposed to delete it from the server too, but it doesn't`
`20:45.57`	`Kobaz`	`and speaking of blackberry... i recently found the bolt browser boldbrowser.com... it's free. and like 100x better/faster than the built in one`
`20:46.04`	`Kobaz`	`and beats the pants off of opera mini`
`20:46.10`	`Kobaz`	`er boltbrowser.com`
`20:49.09`	`Naikrovek`	`http://www.iggdawg.com/pics/misc/chew_it.jpg`
`20:49.14`	`*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)`
`20:50.18`	`lordvadr`	`Is there a way to get accesses to the sip.conf "subscribecontext" from within the dialplan?`
`20:51.01`	`*** part/#asterisk moos3 (~rgenthner@216.52.121.66)`
`20:51.06`	`nachoguy`	`Does anyone know the reason that calls waiting in queue have a number of "Stopped music on hold" "Started music on hold" statements? This is not the initial joing the queue or the final "stopping" before being answered.`
`20:51.25`	`nachoguy`	`Is it just the individual songs stopping and starting?`
`20:51.33`	`Qwell`	`nachoguy: prompts being played?`
`20:51.46`	`nachoguy`	`nope, no periodic announce on those queues`
`20:51.58`	`*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)`
`20:52.13`	`[TK]D-Fender`	`lordvadr: external script <-`
`20:53.16`	`*** join/#asterisk Spacer (~Space@65-120-33-210.dia.static.qwest.net)`
`20:53.30`	`lordvadr`	`I just want to lookup the subscribecontext and do something like VoicemailMain(${CALLERID(num)}@${SUBSCRIBECONTEXT})`
`20:53.51`	`[TK]D-Fender`	`lordvadr: external script <-`
`20:54.10`	`lordvadr`	`I was hoping there was a more elegant way`
`20:54.15`	`lordvadr`	`thanks though`
`21:01.25`	`*** join/#asterisk abradley (~abradley@72.12.1.38)`
`21:01.57`	`abradley`	`are there any distro's of asterisk that, by default, have a built-in fax-to-email gateway? i see that trixbox doesn't, and I can't find if elastix does or does not`
`21:03.21`	`Nugget`	`dunno, I use asterisk.`
`21:03.42`	`Nugget`	`there aren't many people in this channel who don't use asterisk`
`21:04.21`	`Qwell`	`I don't`
`21:05.18`	*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
`21:05.21`	`Nugget`	`yeah, but you're weird.`
`21:07.25`	`[TK]D-Fender`	`abradley: what is a "fax to e-mail gateway"?`
`21:08.38`	`Nugget`	`presumably some rube goldberg contraption that involves bailing wire, hylafax, and perl.`
`21:09.26`	`Qwell`	`Nugget: you said perl twice`
`21:09.31`	`Nugget`	`heh`
`21:10.17`	`abradley`	`well hylafax is pretty complicated. Is there a simple fax gateway integration built-into some variant of asterisk?`
`21:10.19`	`[TK]D-Fender`	`Nugget: don't forget the chewing gum, duct tape, hinges, WD40, and Swiss Army Knife`
`21:10.24`	`Qwell`	`[TK]D-Fender: he said perl`
`21:10.42`	`[TK]D-Fender`	`Qwell: Ok, that handles the SWA :)`
`21:10.51`	`[TK]D-Fender`	`abradley: What does" fax gateway integration " imply?`
`21:10.53`	`Qwell`	`that's like MacGyver-as-file.pl`
`21:11.07`	`[TK]D-Fender`	`abradley: You throw around a lot of the non-fixed definitions`
`21:11.13`	`Nugget`	`or MacGuiver-as-file.pm if you want the enterprise version.`
`21:11.46`	`abradley`	`I'm not sure how to word my initial question more simply`
`21:12.10`	`[TK]D-Fender`	`abradley: Receiving a fax = 1 line of dialplan. Sending it as an e-mail = 1 more.`
`21:12.30`	`[TK]D-Fender`	`abradley: What kind of "miracle integration" definition do you have that should overcomplicate this?`
`21:12.42`	`abradley`	`is there some version of asterisk (for example pbiaf, elastix, asterisk @ home, ...) that has a fax gateway built-in`
`21:13.01`	`abradley`	`and by fax gateway I mean a system that can receive faxes and send them to an email list`
`21:13.01`	`Nugget`	`most people in this channel are completely unaware of those products.`
`21:13.03`	`[TK]D-Fender`	`abradley: All of those use FreePBX which can already e-mail the fax.`
`21:13.13`	`[TK]D-Fender`	`Ad-Hoc: So... seriously.. whats the issue here?`
`21:13.24`	`[TK]D-Fender`	`abradley:*`
`21:15.20`	`abradley`	`I didn't realize freepbx had fax functionality built-in. I've never noticed it before`
`21:15.27`	`abradley`	`I'll look into that then. Thanks for the tip`
`21:24.56`	`p3nguin`	`No "version" of asterisk is going to provide you with an operation dialplan that emails faxes to you.`
`21:25.06`	`p3nguin`	`operationa;`
`21:25.10`	`p3nguin`	`operational`
`21:28.10`	`Qwell`	`ChrisHansen: crap.`
`21:28.33`	`xheliox`	`has a seat over there..`
`21:28.57`	`ChrisHansen`	`Qwell, problems elsewhere (I'm actually the registered owner of this nick)`
`21:29.09`	`Qwell`	`oh, PHEW ;)`
`21:31.27`	`*** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com)`
`21:38.40`	`bent_screwdriver`	`p3nguin: remember when we were talking about the choppy calls while doing dialplan reload?`
`21:41.52`	`bent_screwdriver`	`anyone have ever seen an issue where calls were choppy during a reload (dialplan reload)? The strange thing is that they are not choppy if i do it with asterisk -rx "dialplan reload"`
`21:42.25`	`bent_screwdriver`	`even doing it on my standby system that is different make/model. both ast 1.6.2`
`21:47.29`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`21:59.41`	`*** join/#asterisk sgtarr (~sgt@grim.rasterburn.org)`
`22:00.17`	`sgtarr`	`Any chance I can run some AGI commands from the CLI? For example, I first dial an extension (this works), then I want to do "say digits 1234". Later I'd put it in an AGI script but for now I'd like to try that on the CLI.`
`22:03.16`	`[TK]D-Fender`	`sgtarr: No`
`22:03.30`	`[TK]D-Fender`	`sgtarr: CLI is not a place to direct a call by hand through the dialplan`
`22:03.51`	`[TK]D-Fender`	`sgtarr: When you dial an extension that IS dialplan.`
`22:04.31`	`sgtarr`	`[TK]D-Fender: And to execute an AGI script that just dials extension 100 and executes "say digits 1234", how is that done?`
`22:06.07`	`[TK]D-Fender`	`staffmember: AGI is not self initiating`
`22:06.17`	`[TK]D-Fender`	`sgtarr: AGI is not self initiating`
`22:06.43`	`[TK]D-Fender`	`sgtarr: It is a dialplan app in and of itself. an existing channel must execute it`
`22:07.21`	`sgtarr`	`[TK]D-Fender: what is the best way for me to initiate an AGI script from let's say a shell script?`
`22:07.44`	`[TK]D-Fender`	`sgtarr: What is this AGI going to do all by itself?`
`22:07.48`	`sgtarr`	`I assumed that there were some asterisk command line tool that could have some data piped into it`
`22:08.20`	`[TK]D-Fender`	`sgtarr: CLI doesn't hand-hold calls`
`22:08.28`	`sgtarr`	`[TK]D-Fender: I understand that the AGI will then just be the commands that I need to run. But how to initiate a call?`
`22:08.46`	`[TK]D-Fender`	`sgtarr: What kind of things is your AGI going to do?`
`22:08.47`	`sgtarr`	`[TK]D-Fender: I can always use the Asterisk API`
`22:09.16`	`sgtarr`	`[TK]D-Fender: I just need the following: Initiate a call to a given extension, play a wav file or do a "say digits 1234", then hangup`
`22:09.30`	`[TK]D-Fender`	`sgtarr: So far this has NOTHING to do with AGI.`
`22:09.36`	`[TK]D-Fender`	`sgtarr: You are looking to ORIGINATE a call`
`22:09.43`	`sgtarr`	`[TK]D-Fender: I thought the "SAY DIGITS" was an AGI command :)`
`22:09.51`	`[TK]D-Fender`	`sgtarr: For which there is already the Originate CLI command, and AMI command`
`22:10.07`	`*** join/#asterisk boodu (~boodu@175.158.129.128)`
`22:10.09`	`[TK]D-Fender`	`sgtarr: And call-files`
`22:10.59`	`sgtarr`	`[TK]D-Fender: Originate does not do anything within the Asterisk CLI`
`22:11.19`	`[TK]D-Fender`	`sgtarr: What version are you using?`
`22:11.20`	`sgtarr`	`you mean as in a separate command line app?`
`22:11.23`	`sgtarr`	`[TK]D-Fender: let me check`
`22:11.29`	`sgtarr`	`1.2.30`
`22:11.30`	`[TK]D-Fender`	`sgtarr: Thats the problem. 1.6+`
`22:11.41`	`[TK]D-Fender`	`sgtarr: Welcome to this decade...`
`22:11.45`	`sgtarr`	`yes, I can't use a newer version. Out of my control`
`22:11.57`	`[TK]D-Fender`	`sgtarr: Yeah I hear stuff like that a lot.`
`22:12.00`	`sgtarr`	`[TK]D-Fender: There is no way of doing this with pre-1.6 ?`
`22:12.04`	`[TK]D-Fender`	`sgtarr: call-files / AMI originate it is`
`22:12.19`	`boodu`	`ola!`
`22:13.20`	`[TK]D-Fender`	`is off for a bit`
`22:14.28`	`*** join/#asterisk devdvd (Jason@173-31-175-3.client.mchsi.com)`
`22:16.05`	`*** join/#asterisk brad_mssw (~brad@shop.monetra.com)`
`22:23.06`	`sgtarr`	`[TK]D-Fender: call-files it is.`
`22:29.35`	`devdvd`	`im trying to implement an ivr and i want the caller to have 5 seconds to enter the extension but i also want it to pass through to the next priority once the caller is finished imputting digits. Right now what is happening is the call is waiting 5 seconds before dropping to the next priorty, here is the part of my dialplan that has the menu. Ive tried taking out everyting except answer and background`
`22:29.50`	`devdvd`	`exten=>s,1,Answer(900)`
`22:29.50`	`devdvd`	`exten=>s,n,Background(ivr)`
`22:29.58`	`devdvd`	`any thoughts? or is it even possible`
`22:30.12`	`devdvd`	`the user will press 1 or 2 or dial a 3 digit extension`
`22:30.37`	`devdvd`	`maybe press another single digit down the line but i would think the logic would be the same`
`22:30.40`	`devdvd`	`any thoughts on this.`
`22:30.51`	`devdvd`	`or maybe someone can offer up a better way`
`22:31.33`	`leifmadsen`	`devdvd: that is not possible because asterisk is going to wait for digits if there is a possibility for it to possibly match on something else`
`22:32.01`	`devdvd`	`hwo would you recommend i go about it?`
`22:32.14`	`devdvd`	`right now the delay seems way to long`
`22:32.18`	`leifmadsen`	`devdvd: ideally your extensions will not overlap with either the options, then asterisk can say, "oh there is nothing else that could possibly match here, so we'll use this extension"`
`22:32.29`	`leifmadsen`	`you can modify the response timeout and the digit timeout too`
`22:33.05`	`leifmadsen`	`So use options like 1, 2, 3, 6, 7, 8, 0 and have your extensions in the range of either 4XX or 5XX`
`22:33.30`	`devdvd`	`yea, i think yoru right, i think i did something stupid early on and never fixed it`
`22:33.34`	`devdvd`	`let me change that and see`
`22:33.43`	`leifmadsen`	`you're* :)`
`22:34.08`	`devdvd`	`grumbles freaking grammar nazi :)`
`22:34.38`	`BarthezZ`	`hmm, are there other sources, apart from the asterisk book etc, of documentation? Im currently looking to change some behavior for "dumb" sip clients for (for example) call forwarding... and have no clue where to start`
`22:34.55`	`leifmadsen`	`devdvd: just trying to help :)`
`22:35.03`	`devdvd`	`thank you :)`
`22:35.10`	`leifmadsen`	`BarthezZ: define what you're trying to do...`
`22:35.35`	`leifmadsen`	`BarthezZ: documentation is pretty much the voip-info wiki, the asterisk book, the doc/ directory of your asterisk source, and some various websites found via google`
`22:35.40`	`BarthezZ`	`well, I to be honest just done a small brief search, so I haven't done all the research I should do yet`
`22:36.38`	`BarthezZ`	`but what i actually want: with an attended transfer, when the phone transfering hangs-up while being attended (so speaking to the guy who needs to get the call) the external call is transfered to him`
`22:37.24`	`BarthezZ`	`my experience so far is just with some more advanced sip-phones where you can just slame the transfer button to call the other guy and slam transfer again to actually foward the call`
`22:39.20`	`leifmadsen`	`BarthezZ: that sounds like it should already work. That sounds like a "blonde transfer"`
`22:39.27`	`leifmadsen`	`~blondetransfer`
`22:39.32`	`leifmadsen`	`~blonde-transfer`
`22:39.35`	`leifmadsen`	`huh...`
`22:39.58`	`leifmadsen`	`anyways, blonde transfer is when you use an attended transfer like a blind transfer, and that sounds similar to what you're doing unless I misunderstand what you're meaning.`
`22:40.38`	`leifmadsen`	`infobot: a blondetransfer is when you use an attended transfer as a blind transfer. You should really try and do a blind transfer if that is really what you're meaning to do.`
`22:40.38`	`infobot`	`leifmadsen: okay`
`22:41.37`	`BarthezZ`	`well, the steps on the dumb phone should be: pickup external line (say hi :p), press *<internal number>#, external line goes on hold, you speak to the other internal guy, and as soon as you hangup the other interal guy gets the external channel`
`22:42.38`	`leifmadsen`	`that just sounds like an attended transfer :)`
`22:42.50`	`leifmadsen`	`just enable it in features.conf and use the 't' or 'T' option in Dial()`
`22:43.02`	`BarthezZ`	`but shouldn't you normally press an extra key on the phone to actually do the transfer?`
`22:43.07`	`leifmadsen`	`you don't have to`
`22:43.12`	`leifmadsen`	`it should "just work"`
`22:43.19`	`BarthezZ`	`hmm oke great :)`
`22:43.21`	`leifmadsen`	`pretty sure I even tried that the other day`
`22:43.40`	`BarthezZ`	`getting the hardware for a new system tommorow, so i definitely goanna test it :p`
`22:43.43`	`kyosh`	`what would be the difference in a dialplan of having the _ such as _1X. versus 1X. ? why is the _ necessary?`
`22:44.04`	`leifmadsen`	`kyosh: the _ means "pattern match"`
`22:44.16`	`leifmadsen`	`without it, 1X. means to LITERALLY match 1X.`
`22:44.32`	`kyosh`	`ooo so _ means that the X is a wildcard?`
`22:44.39`	`leifmadsen`	`with the underscore, it means, "match any extension starting with 1 followed by any number 0-9, followed by 1 or more characters"`
`22:44.50`	`leifmadsen`	`~docs`
`22:44.51`	`infobot`	`from memory, docs is for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book)`
`22:44.58`	`leifmadsen`	`~book`
`22:44.59`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`22:45.01`	`leifmadsen`	`please read chapter 5`
`22:45.04`	`kyosh`	`im looking on voip-info but its not explained well`
`22:45.11`	`*** join/#asterisk megalomano (~klonstein@38.124.169.126)`
`22:45.37`	`BarthezZ`	`leifmadsen do you happen to have experience with the dCap trainings?`
`22:45.44`	`leifmadsen`	`BarthezZ: yes`
`22:45.56`	`Qwell`	`BarthezZ: leifmadsen practically wrote the book on Asterisk.`
`22:45.57`	`leifmadsen`	`BarthezZ: I have marked several dozen dCAP tests :)`
`22:45.59`	`Qwell`	`oh wait`
`22:46.13`	`leifmadsen`	`BarthezZ: and I helped write the initial dCAP training and testing`
`22:46.20`	`BarthezZ`	`oh lol, now I sound stupid for asking :p`
`22:46.33`	`leifmadsen`	`why? I just answered your question :)`
`22:46.37`	`leifmadsen`	`Qwell: HI!`
`22:46.40`	`*** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca)`
`22:46.41`	`leifmadsen`	`waves furiously`
`22:46.42`	`Qwell`	`leifmadsen: BAI`
`22:46.47`	`Qwell`	`dodges`
`22:46.57`	`BarthezZ`	`well, I'm currently orientating if it would be usefull for me to do the asterisk advanced course and the dCap examn`
`22:47.45`	`BarthezZ`	`but it's a little unclear to me if there actually is stuff explained which goes beyond the Asterisk book`
`22:47.55`	`leifmadsen`	`BarthezZ: I'd ask jsmith`
`22:48.00`	`leifmadsen`	`I presume there would be`
`22:48.10`	`leifmadsen`	`but I don't know for sure`
`22:48.17`	`leifmadsen`	`woh... what does this mean? [May 5 18:46:31] WARNING[15548]: pbx.c:2496 pbx_find_extension: Maximum PBX stack exceeded`
`22:48.36`	`BarthezZ`	`yeah I would presume too, but it's alot of money (as i'm self employed the boss paying for it is... well me :p )`
`22:48.36`	`Qwell`	`leifmadsen: nesting`
`22:48.47`	`leifmadsen`	`Qwell: include going to an include back to the original include?`
`22:49.00`	`Qwell`	`maybe. probably macroish stuff though`
`22:49.11`	`leifmadsen`	`hmmm... this is just on loading the dialplan`
`22:49.14`	`leifmadsen`	`not on executing anything`
`22:49.24`	`kyosh`	`hey lief, the site says i should contact you :-p`
`22:49.24`	`BarthezZ`	`but ok, i'll keep an eye out for jsmith on irc :p`
`22:49.26`	`Qwell`	`oh. yeah, maybe multi-level nested includes then`
`22:49.49`	`Qwell`	`I think a>b>a is detected, but a>b>c>a wouldn't be`
`22:50.19`	`leifmadsen`	`Qwell: WOH!`
`22:50.38`	`Qwell`	`oh!`
`22:50.39`	`leifmadsen`	`Qwell: all it ended up being was a typo -- I have a priority 'b' instead of 'n'`
`22:50.39`	`Qwell`	`you.`
`22:50.43`	`Qwell`	`weird`
`22:50.46`	`leifmadsen`	`yes, very`
`22:50.52`	`Qwell`	`anyways, I have something for you!`
`22:50.58`	`leifmadsen`	`I think it actually caused me to get disconnected from the trunk console too...`
`22:51.03`	`leifmadsen`	`how very odd :)`
`22:51.11`	`Qwell`	`funky`
`22:51.18`	`Corydon76-dig`	`Qwell: yes, a->b->c->a is detected fine`
`22:51.20`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`22:51.27`	`Qwell`	`codefreeze-lap: oh`
`22:51.28`	`Qwell`	`err`
`22:51.30`	`Qwell`	`Corydon76-dig:`
`22:51.58`	`Corydon76-dig`	`Now if you have 20 includes, that's where it breaks down`
`22:52.30`	`Corydon76-dig`	`That will exceed the maximum stack`
`22:55.50`	`*** join/#asterisk MiserySoft (~LND@nat67.mia.three.co.uk)`
`22:56.44`	`devdvd`	`leifmadsen, I think I love you....`
`22:56.54`	`leifmadsen`	`WILD THING!`
`22:56.58`	`devdvd`	`haha`
`22:56.58`	`leifmadsen`	`YOU MAKE EVERYTHING!`
`22:57.00`	`leifmadsen`	`Grooooooooovy`
`22:57.10`	`BarthezZ`	`lol wth :+`
`22:57.41`	`leifmadsen`	`Wild thing. I think I love you. But I wanna know for sure!`
`22:57.45`	`Qwell`	`note to self: must fresh prince of belair leifmadsen now.`
`22:58.03`	`spenguin[work]`	`lol`
`22:58.42`	`BarthezZ`	`note to self... must kill supplier :\`
`22:58.48`	`leifmadsen`	`Qwell: http://www.youtube.com/watch?v=AVbQo3IOC_A&fmt=18&annotation_id=annotation_322797&feature=iv`
`22:58.49`	`devdvd`	`there was 2 things that was stopping me`
`22:58.53`	`devdvd`	`from doing what i want`
`22:58.58`	`Qwell`	`leifmadsen: I'm not clicking that.`
`22:59.00`	`devdvd`	`had an extension 101 (DOH)`
`22:59.10`	`leifmadsen`	`Qwell: F7ESH!`
`22:59.13`	`devdvd`	`exten=>_X.`
`22:59.26`	`devdvd`	`which i wouldnt have thought of till you said something`
`22:59.34`	`BarthezZ`	`I just remebered it's been a week since they promised me to call me back "in a few days" about a demonstration for the polycom wirless server`
`23:01.39`	`*** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net)`
`23:03.12`	`*** join/#asterisk jksM (jks@193.189.93.254)`
`23:03.18`	`*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)`
`23:03.52`	`*** join/#asterisk orioni (~chatzilla@95.107.226.183)`
`23:04.07`	`orioni`	`is there an ooption on asterisk to enable the callerID ?`
`23:04.22`	`orioni`	`i dont get any callerid on my hardphone`
`23:04.27`	`*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)`
`23:08.03`	`leifmadsen`	`orioni: you don't really have to enable it... it should just come in. What does ${CALLERID(all)} give you on the incoming channel?`
`23:08.13`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`23:08.22`	`leifmadsen`	`orioni: you may or may not have to enable the 'o' option on Dial()... athough that should really be unnecessary`
`23:09.20`	`orioni`	`where does the callerid come from asterisk ? the extension or the extension name ?`
`23:09.35`	`leifmadsen`	`depends what you have setup`
`23:09.41`	`leifmadsen`	`is this from an incoming line, or from a phone?`
`23:09.45`	`leifmadsen`	`what is the technology?`
`23:11.41`	`orioni`	`2 voip hardphones registered on asterisk`
`23:11.54`	`orioni`	`phone - asterisk - phone`
`23:12.11`	`orioni`	`voip phones`
`23:12.36`	`leifmadsen`	`ok, then either the phone passes along the callerID (you may need to enable 'sendrpid=yes' and 'trustrpid=yes' in sip.conf), or you can set it with the callerid= option in sip.conf within the peer definition`
`23:13.20`	`sgtarr`	`So I got a dial plan defined in extensions.conf now`
`23:13.29`	`leifmadsen`	`you sure do :)`
`23:13.43`	`sgtarr`	`And it does SayDigits(1234), but I need the script that puts the call file in asterisk-outgoing to decide what number it is going to be`
`23:13.53`	`sgtarr`	`Any way to pass that along with the .call file?`
`23:14.21`	`leifmadsen`	`sgtarr: please define your requirements and what you expect to happen more carefully`
`23:15.24`	`sgtarr`	`leifmadsen: No problem. Currently I have a .call file being generated with Channel/Context/Extension. It looks up [test] in extensions.conf which defines Answer(), SayDigits(1234), Hangup() and so on. That works fine.`
`23:15.34`	`leifmadsen`	`ok`
`23:15.45`	`sgtarr`	`leifmadsen: However, the program that generates the call file is the one deciding what number to put in SayDigits()`
`23:15.47`	`leifmadsen`	`can you set a variable with the value?`
`23:15.51`	`leifmadsen`	`ok`
`23:16.00`	`sgtarr`	`leifmadsen: I can pass anything along, I just don't know anything about Asterisk :)`
`23:16.08`	`sgtarr`	`All I've learned is in the last 30 minutes`
`23:16.32`	`sgtarr`	`if in extensions.conf I could do something like SayDigits($SOMEVAR)`
`23:16.44`	`sgtarr`	`And SOMEVAR was defined in my .call file, it would be perfec`
`23:16.46`	`sgtarr`	`+t`
`23:16.49`	`leifmadsen`	`look in the sample.call file because you can pass the data you want to use, or you can set a variable`
`23:16.50`	`*** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net)`
`23:16.57`	`leifmadsen`	`sgtarr: you can do exactly that`
`23:17.22`	`leifmadsen`	`sgtarr: look for the section in sample.call about channel variable`
`23:17.24`	`leifmadsen`	`s`
`23:17.28`	`sgtarr`	`leifmadsen: Think I understand now. Looking for sample.call now`
`23:17.39`	`leifmadsen`	`it's in the asterisk source directory`
`23:18.02`	`sgtarr`	`I see now: "Set: blabla=1234"`
`23:18.22`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:19.50`	`sgtarr`	`Works like a charm, tak leif`
`23:20.23`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:24.31`	`*** join/#asterisk orioni (~chatzilla@95.107.226.183)`
`23:24.36`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`23:25.01`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:25.59`	`orioni`	`how can i achive an type of secretary function`
`23:26.19`	`orioni`	`s.o calls , the secretary get the call, than ask the employ if he wants to accept the call`
`23:26.37`	`BarthezZ`	`attended transfers`
`23:26.53`	`trelane`	`does anyone have an /etc/sudoers from a switchvox system?`
`23:27.10`	`orioni`	`if yes that it connect the caller with the employee , if the staff doesnt want to be disturbed thatn the secretary get again the call and says "sorry ihe is busy"`
`23:27.15`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:30.21`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:30.39`	`p3nguin`	`orioni: Are you having an issue with attended transfers?`
`23:32.18`	`*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)`
`23:33.15`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:35.22`	`*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)`
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`23:53.25`	`Naikrovek`	`polycom wireless server, eh`
`23:53.38`	`Kobaz`	`does it do power over wireless`
`23:53.50`	`Naikrovek`	`heh`
`23:54.05`	`TJNII`	`PoWireless rocks. Just don't stand between the AP and the device.`
`23:54.10`	`BarthezZ`	`what with it Naikrovek? :p`
`23:54.12`	`Kobaz`	`bzzzz`
`23:54.40`	`*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)`
`23:54.40`	`Naikrovek`	`what's with what`
`23:54.48`	`BarthezZ`	`polycom wireless server? :p`
`23:54.56`	`Naikrovek`	`i'm just curious about a polycom wireless server`
`23:55.12`	`ChannelZ`	`Huh. The Pat Fleet sound set seems to be missing digits/at`
`23:55.29`	`BarthezZ`	`what do you want to know? :p`
`23:55.44`	`Naikrovek`	`what is it, and what does it do?`
`23:56.08`	`BarthezZ`	`uhmm a VoIP DECT gateway :p`
`23:56.16`	`Naikrovek`	`i see`
`23:56.22`	`Naikrovek`	`wasn't DECT cracked a few days ago`
`23:56.39`	`*** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net)`
`23:56.39`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`23:56.55`	`BarthezZ`	`who cares, it's much better than VoWifi`
`23:57.25`	`Naikrovek`	`not if people can decrypt DECT (which is possible) but not WAP2 (which ins't)`
`23:57.31`	`Naikrovek`	`isn't*`
`23:57.36`	`Naikrovek`	`but still sounds cool`
`23:58.20`	`Naikrovek`	`so it's meant for Polycom wireless VoIP phones then?`
`23:58.20`	`Naikrovek`	`i guess i could just google this`
`23:58.20`	`Naikrovek`	`or look on polycom's site`
`23:58.20`	`BarthezZ`	`but the range of an dect station is much better than a wifi ap :p`
`23:58.21`	`Naikrovek`	`aah`
`23:58.22`	`Naikrovek`	`nice`
`23:58.28`	`Naikrovek`	`true advantage there`
`23:58.36`	`BarthezZ`	`it's gap compliant`
`23:58.41`	`Naikrovek`	`i dunno gap`
`23:58.45`	`BarthezZ`	`so every handset should work`
`23:58.51`	`Kobaz`	`wow`
`23:58.52`	`BarthezZ`	`standard for dect phones :p`
`23:58.54`	`Kobaz`	`the handsets are cheap`
`23:58.57`	`Naikrovek`	`aah i see`
`23:59.03`	`Naikrovek`	`this is what i was waiting for actually`
`23:59.05`	`Naikrovek`	`this is important`
`23:59.11`	`BarthezZ`	`lol`
`23:59.13`	`Naikrovek`	`a voip base to DECT handsets`
`23:59.13`	`Kobaz`	`lik 80 bucks for the cheapo phones`
`23:59.21`	`BarthezZ`	`i'm getting a demo soon`
`23:59.24`	`BarthezZ`	`hopefully`
`23:59.34`	`Naikrovek`	`how many different handsets can you have on one base, how many distinct extensions`
`23:59.42`	`*** join/#asterisk coppice (~chatzilla@191.193.17.210.dyn.pacific.net.hk)`
`23:59.54`	`BarthezZ`	`check the product sheet :p`
`23:59.58`	`Naikrovek`	`will do`

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