| 00:02.01 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 00:17.12 | Kobaz | where can i download the NI2 spec? |
| 00:30.15 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 00:33.53 | *** join/#asterisk eliel (~eliels@186.18.108.106) |
| 00:34.39 | *** join/#asterisk Gareth (~gareth@www.wiked.org) |
| 00:37.13 | Gareth | has anyone seen an issue with 1.6 ignoring the specified context with incoming calls for a SIP account? type is set to friend, but its still looking in [default] |
| 00:37.49 | *** join/#asterisk homiziado (~ernestofr@88.210.104.187.rev.optimus.pt) |
| 00:38.17 | TJNII | I've seen that in 1.4 after a power outage caused a ungraceful shutdown and suspect reboot. Restarting * fixed it. |
| 00:39.02 | Gareth | could try a restart and see... |
| 00:39.23 | Gareth | nope. no change. |
| 00:46.57 | p3nguin | Where's the sip debug? |
| 00:51.23 | Gareth | I'll generate one. |
| 00:51.55 | p3nguin | Include your sip peer definition that you think the call should be matching. |
| 00:53.58 | *** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.pa.comcast.net) |
| 00:54.07 | TehRabbitt | Hello Hello :-D |
| 00:55.11 | p3nguin | tehrabbitt: I didn't make it 24 hours before changing back to SIP. :/ |
| 00:55.21 | TehRabbitt | p3nguin: seriously? :( lol |
| 00:55.29 | TehRabbitt | so far it's working great for my phone lol |
| 00:55.41 | p3nguin | Transfers don't work for me. How about for you? |
| 00:55.47 | TehRabbitt | oh and btw... if anyone wants to have their own OC3 lines in their own house lol :http://cgi.ebay.com/Marconi-Fore-Forerunner-LE-155-ATM-Workgroup-Switch-/180494711323?cmd=ViewItem&pt=COMP_EN_Hubs&hash=item2a0652ba1b#ht_733wt_1165 |
| 00:55.57 | TehRabbitt | p3nguin: transfers work great actually lol |
| 00:56.15 | TehRabbitt | well I transfered from the cisco phone to a SIP extention... havent tried the other way around yet though |
| 00:56.52 | p3nguin | tehrabbitt: I don't understand. When I'm on a call, if I hit the transfer button, it gives me a dial tone, I dial, that line answers, then I press the transfer key again and it says it cannot transfer. |
| 00:57.17 | TehRabbitt | hm weird 0_o |
| 00:57.25 | TehRabbitt | hold on lemme try on my phone again |
| 00:57.59 | Gareth | p3nguin: http://pastebin.com/DhS0hdbZ |
| 00:58.23 | TehRabbitt | p3nguin: same thing just happened to me... hm must be a setting |
| 00:58.35 | TehRabbitt | MoH works nice though... lol |
| 00:58.47 | Naikrovek | what are you guys talking about |
| 00:58.50 | Naikrovek | iax? |
| 00:58.50 | p3nguin | tehrabbitt: So you can't transfer, after all? |
| 00:59.01 | p3nguin | Cisco and chan_sccp. |
| 00:59.06 | Naikrovek | aah |
| 00:59.21 | TehRabbitt | no no it transfers, but then as soon as the transfer goes through my handheld started ringing again / it was put on hold 0_o |
| 00:59.23 | Naikrovek | sccp isn't supported well in asterisk, is it? |
| 00:59.32 | TehRabbitt | Naikrovek: nope lol |
| 00:59.42 | TehRabbitt | well I honestly think it depends on the phone too |
| 00:59.56 | p3nguin | chan_sccp (third party) is slightly better than chan_skinny (comes with asterisk). |
| 01:00.22 | TehRabbitt | and btw that link i sent to the ebay auction... it's something I came across / thought i'd share it... I have one... given the right ATM cards in routers, you can have OC3 lines running between floors of your house lol |
| 01:00.40 | TehRabbitt | well up to 155mbit/sec |
| 01:01.03 | Naikrovek | power line ethernet goes to 200mbit/s |
| 01:01.15 | Naikrovek | and gig-e .. well |
| 01:01.22 | TehRabbitt | Naikrovek: i know i know lol but still... it's more the "I have an OC3 in my house" factor |
| 01:01.23 | Naikrovek | oc3 is impressive, but only over distance |
| 01:01.28 | Naikrovek | trye |
| 01:01.31 | Naikrovek | true |
| 01:01.32 | TehRabbitt | lol |
| 01:01.34 | Naikrovek | that would be wicked |
| 01:01.40 | p3nguin | gareth: no matching peer... this is why it goes to default. |
| 01:01.46 | Naikrovek | fiber getting installed to my area this year :D |
| 01:02.31 | TehRabbitt | p3nguin: I got 411, 800, 866, 877, 888 all working :-D Still need to get that damn magicjack BS working though lmao other than that ummmm I'm still running into a slight NAT issue... which may be why MJ wont work either... |
| 01:03.26 | TehRabbitt | I tried connecting to my * server from my school.... it registers fine, but then it grabs the hostname "192.168.1.70" which is the internal LAN ip... so it's not performing the correct hostname lookup :( |
| 01:04.07 | *** join/#asterisk coppice (~chatzilla@m121-202-57-209.smartone-vodafone.com) |
| 01:04.27 | p3nguin | gareth: Your peer wants to match voip.lax.teliax.com (8.3.252.22), but the call seems to come from 8.3.252.23 |
| 01:04.54 | p3nguin | gareth: I would change host=voip.lax.teliax.com to host=8.3.252.23 then save and sip reload. |
| 01:05.59 | p3nguin | And if they also use the .22 address, I would create another peer definition for it. |
| 01:06.13 | TehRabbitt | p3nguin: any ideas on the NAT issue? I've specified the external IP but it seems like * doesn't wanna present the correct IP to the remote SIP host... Also i've heard IAX can get around the issues associated with this but idk if that will work the way I want it to |
| 01:07.02 | p3nguin | tehrabbitt: I would read the sipnat guide a couple more times and try to figure out why it isn't working. |
| 01:07.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:07.58 | TehRabbitt | p3nguin: yea can you send the link again? sorry lol |
| 01:08.06 | p3nguin | ~sipnat |
| 01:08.07 | 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 |
| 01:10.04 | TehRabbitt | p3nguin: if I had a truck I would TOTALLY buy this: http://cgi.ebay.com/network-tower-netgear-24-ports-linksys-phones-/160428095925?cmd=ViewItem&pt=COMP_EN_Hubs&hash=item255a4279b5#ht_500wt_1182 |
| 01:10.31 | TehRabbitt | though I think they are digital phones not IP phones |
| 01:10.44 | p3nguin | IP phones aren't digital? |
| 01:11.46 | p3nguin | tehrabbitt: For that price, you can find a way to transport the stuff. |
| 01:11.47 | TehRabbitt | p3nguin: Digital phones require a "special" digital circuit on lets say an old AT&T PBX |
| 01:11.51 | TehRabbitt | or Lucent PBX |
| 01:12.08 | p3nguin | It says there are seven Avaya Ethernet phones. |
| 01:12.13 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:12.28 | TehRabbitt | aka Transfer, Hold, Etc relies on the PBX's Digital circuit card |
| 01:12.39 | TehRabbitt | p3nguin: yea google the model # of the phones |
| 01:12.47 | TehRabbitt | i'm like tempted to buy it just for the phones to be honest heh |
| 01:13.27 | TehRabbitt | http://verticall.com/definity/phones/8403.htm |
| 01:14.06 | TehRabbitt | http://www.telecombiz.com/8403-refurb.html |
| 01:14.37 | p3nguin | The 8403 telephone is compatible with the Definity system. |
| 01:14.40 | TehRabbitt | http://en.wikipedia.org/wiki/Avaya_Definity |
| 01:14.48 | TehRabbitt | and Definity is? lol |
| 01:15.06 | TehRabbitt | or should I ask... would the 8403 work with * |
| 01:15.07 | TehRabbitt | lol |
| 01:15.17 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 01:15.30 | TehRabbitt | The long time protocol that the Definity telephones connect the switch is called the Digital Communication Protocol or DCP. In later releases of Communication Manager, the system was reworked as SIP being the primary systemwide signaling protocol, while all previous legacy protocols would run under SIP. |
| 01:22.28 | TehRabbitt | p3nguin: this is what I have between my server and my switch: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=280500353839&ssPageName=STRK:MEWNX:IT#ht_710wt_1165 |
| 01:22.29 | TehRabbitt | lol |
| 01:23.03 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 01:25.30 | Gareth | p3nguin: will give that a try. thanks |
| 01:25.55 | p3nguin | tehrabbitt: You're trying to say that you run fiber channel from your switch to your asterisk box? |
| 01:26.58 | p3nguin | gareth: Or call up teliax and ask them why they are sending from an IP address that does not match their hostname. |
| 01:28.01 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 01:29.46 | TehRabbitt | not fiberchannel, fiber ethernet |
| 01:30.03 | TehRabbitt | oh shit i did it again :( |
| 01:30.07 | TehRabbitt | grrr lol |
| 01:31.19 | p3nguin | So you're running FCoE? |
| 01:31.29 | TJNII | I thought "fiber ethernet" was ethernet encapsulated in fibre-channel. |
| 01:31.40 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:31.47 | TehRabbitt | lol didn't realize thsoe cards were Fiberchannel gah |
| 01:32.02 | TehRabbitt | no no i have a couple Intel Fiber Gigabit cards for ethernet |
| 01:32.37 | p3nguin | 1000BASE-SX? |
| 01:32.43 | TehRabbitt | yes |
| 01:33.04 | TehRabbitt | server is in the basement desktop is upstairs... fiber runs thru the air vent |
| 01:33.11 | TehRabbitt | (return air, not heated) |
| 01:33.38 | p3nguin | Don't worry, I'm not the code inspector. |
| 01:33.44 | TehRabbitt | lol |
| 01:33.46 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 01:34.11 | TehRabbitt | naw it's more just the last time I told someone I have fiber running through an air vent, they were like "you know it's gonna melt in the winter right?" and i was like "no it wont!" etc |
| 01:35.38 | p3nguin | I wonder if a 7912G would have any luck on chan_sccp. What do ya think? |
| 01:36.52 | Gareth | p3nguin: found the solution...host needed to be .net, not .com. thanks for the help :) |
| 01:37.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:37.25 | p3nguin | voip.lax.teliax.net has address 8.3.252.23 |
| 01:37.35 | Gareth | Yup. |
| 01:37.38 | p3nguin | voip.lax.teliax.com has address 8.3.252.22 |
| 01:37.42 | p3nguin | weird people! |
| 01:37.49 | TehRabbitt | heh |
| 01:38.01 | p3nguin | They shouldn't have made it like that. Too easy to get mixed up. |
| 01:39.08 | TehRabbitt | p3nguin: the NAT issue i'm having... could it have something to do with that whole "peer" or "Friend" thing? |
| 01:39.28 | p3nguin | It's possible, but I don't find it likely. |
| 01:41.01 | p3nguin | I'm surprised no one here is willing to help trouble shoot the "Bad Gateway" problem. |
| 01:42.24 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 01:42.40 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 01:43.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:43.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:45.02 | p3nguin | switching back to SCCP now. |
| 01:45.13 | p3nguin | Are you SURE that transfers are working for you? |
| 01:45.17 | TehRabbitt | yep |
| 01:45.38 | p3nguin | I'm going to try a little harder this time to see what the problem is. |
| 01:47.07 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 01:48.15 | Naikrovek | are there any differences in functionality on a cisco phone when using the sip firmware versus the sccp firmware |
| 01:48.23 | Naikrovek | like, does the phone act different? |
| 01:48.28 | p3nguin | yes |
| 01:48.30 | p3nguin | totally |
| 01:48.44 | Naikrovek | really |
| 01:48.47 | Naikrovek | example? |
| 01:49.23 | *** join/#asterisk cdose1 (~chris@ip72-219-50-148.br.br.cox.net) |
| 01:49.59 | cdose1 | when i launch asterisk using "-G asterisk -U asterisk" i get the following message: Unable to access the running directory (Permission denied). Changing to '/' for compatibility. |
| 01:50.16 | cdose1 | it starts up anyway; is this something to worry about and try and fix? |
| 01:50.58 | cdose1 | i don't know what it means by the running directory... |
| 01:52.14 | TehRabbitt | Naikrovek: Ok, it handles additional lines differently, allows more "native" cisco functionality |
| 01:52.30 | Naikrovek | sccp firmware does? |
| 01:52.55 | TehRabbitt | yes |
| 01:53.23 | Naikrovek | hmm. sounds like yet another reason to avoid cisco phones |
| 01:54.14 | TehRabbitt | Cisco's Call Manger is basically a propiatary PBX system similar to * however it requires the use of ALL CISCO equipment... aka you need to use Cisco IP phones, Cisco AP's, Cisco Switches, etc |
| 01:54.20 | TehRabbitt | it' "Unified Telephony" |
| 01:54.23 | TehRabbitt | its* |
| 01:54.25 | p3nguin | SCCP has softkeys that SIP doesn't have, such as Private, iDivert, Park, DirTrfr (might be BlindXfr on SIP), MeetMe ... |
| 01:54.46 | p3nguin | it's |
| 01:54.47 | TehRabbitt | p3nguin: too bad it still doesn't support PTT lol |
| 01:55.04 | *** join/#asterisk JoshF (~Josh@wsip-98-174-176-6.ok.ok.cox.net) |
| 01:55.31 | p3nguin | I received a call and was able to transfer it out. Now I need to make a call and see about transferring it. |
| 01:56.02 | JoshF | Any one ever mess around with SIP Expirey Timer? |
| 01:56.18 | joako | Is there a reason why asterisk would be more prone to crash when running under OpenVZ? |
| 01:56.52 | p3nguin | SCCP also has call forward busy and configurable DND. SIP has only call forward all and basic DND reject. |
| 01:58.38 | p3nguin | Okay... transfers are working now. I don't know what was wrong before. |
| 01:58.43 | p3nguin | The Park key doesn't do anything, though. |
| 01:59.32 | p3nguin | Wait, yes it does. |
| 01:59.38 | p3nguin | wtf |
| 01:59.41 | p3nguin | Now everything's working. |
| 02:00.09 | TehRabbitt | lol see :-D |
| 02:00.15 | p3nguin | So... |
| 02:00.19 | TehRabbitt | SCCP ftw? |
| 02:00.37 | TehRabbitt | haha wow... http://cgi.ebay.com/Motorola-3456-ModemSURFR-External-56k-dial-up-/350346520499?cmd=ViewItem&pt=PCC_Modems&hash=item519247a7b3#ht_1141wt_939&autorefresh=true |
| 02:00.46 | p3nguin | If everything is working this time, there's no reason to go back to SIP? |
| 02:00.58 | TehRabbitt | exactly haha |
| 02:01.09 | TehRabbitt | Come to the SCCP side, we have cookies ;) |
| 02:01.13 | *** join/#asterisk coppice (~chatzilla@m121-202-73-116.smartone-vodafone.com) |
| 02:02.16 | carrar | SIP has wiskey, sip a little |
| 02:02.17 | p3nguin | Should I switch my 7912G over to SCCP, too? |
| 02:02.37 | TehRabbitt | wanna get PTT working? |
| 02:02.38 | TehRabbitt | http://cgi.ebay.com/4-CISCO-AIRONET-350-AIR-AP350-WIRELESS-ACCESS-POINT-NR-/360256829891?cmd=ViewItem&pt=COMP_EN_Routers&hash=item53e0faf9c3#ht_2825wt_1165 |
| 02:02.39 | TehRabbitt | lmfao |
| 02:03.40 | carrar | thats cheap |
| 02:03.46 | carrar | B is better then nothing |
| 02:03.52 | carrar | and works fine in most cases |
| 02:04.36 | TehRabbitt | yep |
| 02:04.49 | TehRabbitt | and if you're using SCCP you can get PTT working haha |
| 02:04.50 | carrar | conver them to iso style which is easy, then you're good to go |
| 02:04.51 | TehRabbitt | possibly |
| 02:04.54 | TehRabbitt | lol |
| 02:04.55 | carrar | IS |
| 02:04.57 | carrar | err ios |
| 02:05.21 | carrar | I've converted 350's before ages ago |
| 02:05.42 | carrar | 3-4 years ago |
| 02:05.44 | TehRabbitt | p3nguin: i'd do the switch... |
| 02:05.45 | carrar | AGES |
| 02:07.40 | ChannelZ | Iggy Pop needs to start wearing clothes. |
| 02:07.46 | carrar | Better spending a hair more and getting the Cisco AIR-AP1231G |
| 02:07.59 | carrar | http://cgi.ebay.com/CISCO-AIRNET-WIRELESS-ACCESS-POINT-AIR-AP1231G-A-K9-/270570479857?cmd=ViewItem&pt=COMP_EN_Routers&hash=item3eff41e0f1 |
| 02:09.10 | p3nguin | tehrabbitt: If I keep going, I won't be using any SIP. :/ |
| 02:09.18 | TehRabbitt | Heh guy at the computer show i bought my 7921G at was selling those $100 each with the 5Ghz cards installed |
| 02:09.30 | TehRabbitt | p3nguin: that's ok :-p you didn't need SIP anyway haha |
| 02:09.42 | p3nguin | module unload chan_sip |
| 02:10.00 | TehRabbitt | lol |
| 02:10.15 | p3nguin | No, wait. I have DIDs that come in on SIP. |
| 02:10.23 | TehRabbitt | LOL |
| 02:10.38 | TehRabbitt | soo keep SIP for those only |
| 02:10.38 | p3nguin | backspaces |
| 02:11.13 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:11.43 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 02:12.02 | TehRabbitt | http://cgi.ebay.com/Cisco-7920-phone-two-cisco-7920-phones-/270570213562?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item3eff3dd0ba#ht_500wt_1182 |
| 02:13.05 | carrar | Crisco |
| 02:13.13 | carrar | for execs |
| 02:13.14 | TehRabbitt | Wait WTF is this for: http://cgi.ebay.com/Unlocked-Linksys-SPA-3000-VoIP-FXS-FXO-PSTN-spa-3000-/330429176698?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item4cef1d1b7a |
| 02:13.17 | TehRabbitt | what is this for |
| 02:13.18 | TehRabbitt | http://cgi.ebay.com/Unlocked-Linksys-SPA-3000-VoIP-FXS-FXO-PSTN-spa-3000-/330429176698?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item4cef1d1b7a |
| 02:13.28 | TehRabbitt | is that so you can use regular Analog lines as SIP lines? |
| 02:13.45 | carrar | ## Welcome to #eBay finder ## |
| 02:15.29 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 02:15.33 | TehRabbitt | now we're talking: http://cgi.ebay.com/Talkswitch-TS550i-TS-550i-PBX-IP-Phone-/140371792098?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item20aecf90e2 |
| 02:15.34 | TehRabbitt | heh |
| 02:17.44 | devmod | Any ideas on how to bridge an audio call into an existing video call between two endpoints? |
| 02:20.13 | TehRabbitt | Will a Nortel M5316 work on *? |
| 02:21.16 | TehRabbitt | or a NT9K16AC03 |
| 02:21.44 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 02:22.01 | boodu | hello |
| 02:22.06 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:22.10 | JoshF | Any one ever mess around with SIP Expirey Timer? |
| 02:23.31 | TehRabbitt | http://cgi.ebay.com/PACIFIC-BELL-PAYPHONE-PAY-PHONE-EMPTY-CASE-PROP-DISPLAY_W0QQitemZ220536720648QQcategoryZ11909QQcmdZViewItemQQ_trksidZp4340.m8QQ_trkparmsZalgo%3DMW%26its%3DC%26itu%3DUCC%26otn%3D20%26ps%3D63%26clkid%3D8785971726863614349#ht_518wt_1165 |
| 02:24.16 | TJNII | That is one blunt description |
| 02:25.05 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:26.02 | TehRabbitt | will a CP-7985 work on *? lol |
| 02:26.12 | carrar | yes |
| 02:26.19 | TehRabbitt | 0_o it'll support the video? |
| 02:27.32 | carrar | sorry 7975 does |
| 02:27.35 | carrar | haven't tried th e80 |
| 02:27.40 | carrar | err 85 |
| 02:28.02 | carrar | buy me one and I'll test it |
| 02:28.12 | TehRabbitt | lol |
| 02:28.21 | TJNII | Why you'd be stupid not to take him up on an offer like that! |
| 02:28.27 | *** join/#asterisk MetaMucil (~Omeras@99-2-200-244.lightspeed.milwwi.sbcglobal.net) |
| 02:28.54 | p3nguin | tehrabbitt: Are you using serviceURL in your sccp.conf? |
| 02:30.09 | TehRabbitt | p3nguin: lemme check |
| 02:30.45 | TehRabbitt | no i'm not... are you? |
| 02:30.49 | p3nguin | I want to. |
| 02:31.03 | p3nguin | I added a URL and it bitched about the wrong syntax. |
| 02:31.05 | TehRabbitt | I know you can specify what the soft keys do using that |
| 02:31.14 | TehRabbitt | can it be *any* url? |
| 02:31.27 | p3nguin | I use a regular web address in SIP. |
| 02:31.37 | TehRabbitt | I think that's how you can set up a "company directory" etc |
| 02:31.56 | TehRabbitt | apparently there's even a way to set the background image on the color phonews |
| 02:32.05 | p3nguin | So you think an SCCP services URL is different from a SIP services URL? |
| 02:32.29 | TehRabbitt | it probabbly is... most likely needs to be in XML |
| 02:32.38 | p3nguin | Yeah, I'm not too happy that I don't have an image on my screen. In SIP, I use my own image. |
| 02:32.45 | p3nguin | The one I use in SIP is XML. |
| 02:32.58 | TehRabbitt | I was reading something somewhere on how to setup SCCP with XML urls i forget where though |
| 02:35.08 | TehRabbitt | p3nguin: check this link out for info on the serviceURL |
| 02:35.08 | TehRabbitt | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg118867.html |
| 02:35.19 | *** join/#asterisk norrec (~Ghost@76-201-85-28.lightspeed.frokca.sbcglobal.net) |
| 02:35.53 | TehRabbitt | http://phone-xml.berbee.com/ |
| 02:35.57 | TehRabbitt | that might help heh |
| 02:36.33 | TehRabbitt | damn, you can have it pull RSS feeds and display them on the phone 0_o |
| 02:36.55 | TJNII | Oh, that's a pranking begging to happen. |
| 02:36.56 | p3nguin | Yeah, that's what I did on SIP, but serviceURL doesn't like my URL in SCCP. |
| 02:37.10 | TehRabbitt | check out hte link i sent you |
| 02:37.10 | TehRabbitt | http://phone-xml.berbee.com/ |
| 02:38.57 | p3nguin | I'm not sure what you're wanting me to see, but I don't notice anything addressing my issue with the serviceURL setting in sccp.conf. |
| 02:39.19 | norrec | hey guys, I've got an asterisk server with a sip link to my underlieing provider and an iax trunk to another asterisk server and I want the first server to just pass the call though the iax trunk to the 2nd server |
| 02:39.48 | norrec | but i keep getting this error chan_iax2.c:10523 socket_process: Rejected connect attempt from 67.203.87.70, request 'xxxxxxxxxx@from-outside' does not exist |
| 02:40.16 | TehRabbitt | nvm it's not working on mine either :( |
| 02:42.51 | TehRabbitt | anyone wanna help me with a bad gateway error? |
| 02:42.58 | TehRabbitt | regarding SIP |
| 02:44.19 | p3nguin | tehrabbitt: If you figure out how to set a custom screen image, let me know. I like my own image better than these straight lines. |
| 02:45.03 | TehRabbitt | p3nguin: i'll keep messing with it lol.. in the meantime i'm trying to figure out how to get SIP working on that proxy that i was talking to you about yesterday... If I can't get it working it makes a great intercom system between me myself and I |
| 02:45.04 | TehRabbitt | lmao |
| 02:45.31 | p3nguin | Personally, I would get an ITSP that doesn't suck. |
| 02:45.52 | TehRabbitt | mehhhh trying to use this one until I can save up some $$$ |
| 02:45.59 | p3nguin | Screw all that bad gateway crap. |
| 02:46.05 | TehRabbitt | this is kinda a first experiment into * and etc |
| 02:46.25 | TehRabbitt | the weird thing is the NAT issue still remains even with NAT on and the proper hostname.. |
| 02:46.25 | p3nguin | Minimum deposit in VoIP.ms is $25. |
| 02:46.53 | p3nguin | Termination to US numbers is 1.05 cents per minute. |
| 02:46.53 | TehRabbitt | it's like it just wont look up the hostname |
| 02:47.00 | TehRabbitt | Hm... not bad |
| 02:47.31 | p3nguin | That's who I use. |
| 02:47.48 | TehRabbitt | sooo you get roughly 2200 minutes for 25 bucks? |
| 02:48.14 | p3nguin | Flowroute is also good. I think the minimum deposit might be $35, but termination rates are slightly less. |
| 02:48.38 | p3nguin | If you do a lot of toll-free outbound calling, that $25 will last a LONG time. |
| 02:48.44 | TehRabbitt | hm... lol |
| 02:49.03 | TehRabbitt | well it'll be used primarally between 3 users :-\ lol 3 different locations |
| 02:49.06 | TehRabbitt | one in PA, 2 in NJ |
| 02:49.15 | TehRabbitt | myself being one of the NJ users |
| 02:49.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:49.38 | p3nguin | If you want a DID, you can spend 99 cents per month plus per minute rates on incoming calls, or get an unlimited incoming plan for around $7 per month. |
| 02:49.47 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-32-130.owb.bellsouth.net) |
| 02:49.48 | TehRabbitt | 99 cents a month? that's it? heh |
| 02:49.51 | p3nguin | yeah |
| 02:49.57 | TehRabbitt | hm |
| 02:50.30 | norrec | is anyone familiar with iax? |
| 02:50.37 | p3nguin | yes |
| 02:51.06 | TehRabbitt | p3nguin, so what do I do with the DID that is already assigned to that MJ proxy that keeps saying bad gateway |
| 02:51.06 | TehRabbitt | lol |
| 02:51.12 | TehRabbitt | htat one is already kinda public lol |
| 02:51.19 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:51.29 | norrec | p3nguin: i'm having some trouble setting up an iax trunk, do u know were i can find some documentation or can you give me a couple pointers |
| 02:51.29 | p3nguin | If you want to keep the number, port it to a better provider. |
| 02:51.30 | TehRabbitt | or should i just have that one foward calls to a different DID |
| 02:51.39 | p3nguin | or that |
| 02:51.41 | TehRabbitt | hm |
| 02:51.57 | p3nguin | You'll spend $25 to port it. |
| 02:51.58 | TehRabbitt | can I make outgoing calls using google voice? or did they shut that down? |
| 02:52.26 | p3nguin | norrec: Have you read The Book? |
| 02:52.30 | p3nguin | ~book |
| 02:52.31 | 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 |
| 02:52.50 | p3nguin | norrec: Show me what configuration you have already done and explain what the problem is. |
| 02:53.20 | norrec | k, give me a sec to get it all together, i'll post a link in a sec |
| 02:53.39 | p3nguin | Have you read the book? |
| 02:54.25 | norrec | p3nguin: no i havent, but i'm gonna dl the pdf |
| 02:55.24 | TehRabbitt | p3nguin: found a good tutorial on SIP and NAT gonna try this one out |
| 02:57.29 | TehRabbitt | *crosses fingers and hopes this works* |
| 03:00.29 | TehRabbitt | p3nguin: the call quality over this 1800 carrier is better than my cell phone 0_o |
| 03:00.37 | TehRabbitt | much crisper / clearer |
| 03:00.44 | p3nguin | heh |
| 03:00.50 | TehRabbitt | and the voice menus actually work lmfao |
| 03:00.54 | p3nguin | Which gateway are you using? |
| 03:01.03 | p3nguin | futurenine or ideasip? |
| 03:01.03 | TehRabbitt | "Say FEATURES to order FEATURES" |
| 03:01.08 | TehRabbitt | futurenine |
| 03:01.22 | p3nguin | Did you ever figure out how to get them to pay you for calls? |
| 03:01.27 | TehRabbitt | nope lol |
| 03:01.44 | p3nguin | Most of the stuff on their site does not jive. |
| 03:01.52 | TehRabbitt | heh |
| 03:02.02 | TehRabbitt | well it def keeps track of all the outgoing 800 numbers |
| 03:02.14 | p3nguin | They talk about service plans that you can elect to use, but if you look for it, they don't exist. |
| 03:02.16 | TehRabbitt | pretty quick connection too from dial to ringing |
| 03:02.22 | p3nguin | Which plan did you select? |
| 03:02.24 | TehRabbitt | oh they exist... you need to sign up first |
| 03:02.27 | TehRabbitt | the M3 business one |
| 03:02.31 | p3nguin | I have an account with them. |
| 03:02.39 | TehRabbitt | i think |
| 03:02.40 | p3nguin | Let me go look. |
| 03:03.24 | p3nguin | I'm currently on America Free. |
| 03:04.17 | TehRabbitt | hm should I bother setting up E911? |
| 03:04.18 | TehRabbitt | heh |
| 03:04.30 | p3nguin | there's no m3 business plan listed. |
| 03:04.33 | TehRabbitt | probabbly not since there will be multiple locations |
| 03:04.48 | TehRabbitt | PayG Premium I chose |
| 03:04.50 | TehRabbitt | i think |
| 03:04.51 | p3nguin | If you configure 911, you'll have to pay the 911 fees. |
| 03:05.05 | p3nguin | <PROTECTED> |
| 03:05.07 | TehRabbitt | What is "america Free" |
| 03:05.11 | TehRabbitt | lol |
| 03:05.20 | p3nguin | higher than PayG Premuim |
| 03:05.26 | TehRabbitt | what is PayG premium lol |
| 03:05.49 | norrec | p3nguin: http://pastebin.com/Rf0pw3LD |
| 03:07.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:10.50 | TehRabbitt | p3nguin: should I just go with flowroute and call it a day? lol |
| 03:11.16 | p3nguin | Oh, no wonder I didn't have any calls in my log. My futurenine peer didn't have any username in it. |
| 03:11.27 | p3nguin | flowroute or voip.ms |
| 03:12.01 | p3nguin | I changed my sip config to the way they said to configure it and now calls appear in their log. |
| 03:12.07 | TehRabbitt | lol |
| 03:12.13 | TehRabbitt | yupp |
| 03:12.29 | TehRabbitt | WHich has cheaper incoming? flowroute or voip.ms? |
| 03:12.37 | p3nguin | I don't have a DID with them, so I never configured a username. |
| 03:12.42 | TehRabbitt | oh |
| 03:12.45 | TehRabbitt | who has the cheapest DID? |
| 03:13.28 | p3nguin | per minute or montly unlimited? |
| 03:13.51 | TehRabbitt | per minute i suppose |
| 03:14.02 | TehRabbitt | i'd love to get the MagicJack working though :-\ |
| 03:15.01 | p3nguin | Flowroute incoming is $1.39/mo and 1.2 cents per minute... or $6.95/mo ulimited. |
| 03:15.27 | p3nguin | Which city would you want your phone number to be in? |
| 03:15.31 | TehRabbitt | p3nguin: http://www.magicjacksupport.com/magicjack-patch-for-asterisk-updated-t7243-90.html |
| 03:15.49 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:15.49 | TehRabbitt | Uhhhh area code 848 or 732 |
| 03:16.13 | TehRabbitt | apparently there's an asterisk 1.6 0_o |
| 03:16.13 | p3nguin | city, not numbers |
| 03:16.20 | p3nguin | ~versions |
| 03:16.37 | p3nguin | ~asterisk-versioning |
| 03:16.38 | infobot | i guess asterisk-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 03:16.39 | p3nguin | ~versioning |
| 03:18.09 | p3nguin | Okay, I'll let you figure out VoIP.ms's DID rates yourself. |
| 03:18.52 | TehRabbitt | sorry was reading this: http://www.magicjacksupport.com/asterisk-502-bad-gateway-t8327.html |
| 03:19.00 | p3nguin | norrec: What's the issue with your config? |
| 03:20.21 | TehRabbitt | p3nguin: do you think i'll be able to get this working? |
| 03:20.29 | ChannelZ | besides ridiculously long context names? |
| 03:20.31 | p3nguin | what, mj? |
| 03:20.36 | TehRabbitt | yes |
| 03:20.40 | p3nguin | maybe |
| 03:20.44 | p3nguin | Several people have. |
| 03:21.27 | norrec | p3nguin: chan_iax2.c:10523 socket_process: Rejected connect attempt from x.x.x.x, request 'xxxxxx5353@from-outside' does not exist |
| 03:21.37 | p3nguin | I guess you need to update your useragent and possibly apply the sip patch. |
| 03:22.52 | p3nguin | norrec: You don't seem to have the context matching the actual name of the context. |
| 03:23.06 | p3nguin | from-outside != from-outside-xxxxxx5353-tl-allhours |
| 03:23.59 | norrec | so how can i get it to just pass the call rather than give it a context? |
| 03:24.32 | p3nguin | All calls go into a context. |
| 03:25.02 | p3nguin | Change the context of the peer to coincide with the actual name of the context or vice versa. |
| 03:25.12 | p3nguin | Do you know what I mean? |
| 03:25.15 | norrec | i think so |
| 03:26.33 | norrec | so why does this setup work with sip and not iax? |
| 03:26.47 | p3nguin | You must not have broken contexts in sip. |
| 03:27.29 | norrec | are the contexts different for sip and iax? |
| 03:29.15 | p3nguin | Yours must be, since these are wrong and don't work, but you claim sip works. |
| 03:30.39 | norrec | well, i use the same scipts for routing to sip and i dont have any issues |
| 03:30.54 | norrec | i havent tried useing sip to this server though |
| 03:32.12 | p3nguin | Until you match the peer's context with the actual context, the call won't procede. |
| 03:33.21 | norrec | so i need to modify the config on the reciving server to accept the from-outside context? |
| 03:35.12 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:36.26 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 03:36.56 | norrec | p3nguin: i'm just kind of confused because the iax trunk is set to be in the from-outside context |
| 03:37.03 | TehRabbitt | p3nguin: how do I do this: http://www.magicjacksupport.com/magicjack-patch-for-asterisk-updated-t7243.html |
| 03:37.04 | p3nguin | norrec: line 57 and line 63 NEED TO MATCH. |
| 03:38.19 | norrec | p3nguin: so the "-tl-allhours" is whats messing up my routing then? |
| 03:38.39 | p3nguin | norrec: The context of the peer and the context in the dialplan MUST MATCH. |
| 03:39.24 | TehRabbitt | p3nguin: how do I patch chan_sip.so? |
| 03:39.25 | p3nguin | tehrabbitt: Are you asking me how to apply a patch? or what? |
| 03:39.54 | TehRabbitt | p3nguin: yes |
| 03:41.38 | p3nguin | tehrabbitt: Get asterisk source. Go to the directory where chan_sip.c is. Put the patch file there. Apply the patch with patch -p0 <magickjack.patch. Compile chan_sip.so. Copy it into /var/lib/asterisk/modules/. Run sip reload. |
| 03:42.12 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 03:42.22 | TehRabbitt | ah |
| 03:42.48 | p3nguin | Make sure you have the right patch and corresponding asterisk version. |
| 03:43.51 | p3nguin | What version are you using? |
| 03:44.05 | TehRabbitt | 1.4 |
| 03:44.23 | p3nguin | That's a branch, not a version. |
| 03:44.49 | TehRabbitt | i'm not sure how do i check |
| 03:44.54 | p3nguin | core show version |
| 03:45.01 | norrec | p3nguin: can you help me write something to just answer the call and like play a file or something just so i can see it work, i think i can fix it from there.... |
| 03:45.11 | TehRabbitt | Asterisk 1.4.21.2~dfsg-3+lenny1 built by buildd @ brahms on a x86_64 running Linux on 2009-12-14 19:40:23 UTC |
| 03:45.14 | norrec | p3nguin: when u get a second at least |
| 03:45.37 | p3nguin | norrec: Which system do you want to answer the call? Where is the call coming from? |
| 03:46.04 | p3nguin | tehrabbitt: That's an oldie. |
| 03:46.16 | norrec | oh, i want the 2nd one to answer it, i just want the first one to take the call from the sip provider and hand it off to the 2nd one via iax |
| 03:46.18 | [TK]D-Fender | TehRabbitt: I'm betting you don't even have the sources since you installed from a repo |
| 03:46.28 | TehRabbitt | [TK]D-Fender: nope :( |
| 03:46.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:46.32 | norrec | then the second one to process and terminate the call |
| 03:46.49 | TehRabbitt | [TK]D-Fender; how would I find the sources |
| 03:46.56 | p3nguin | norrec: So a call will come from the first one to the second one, and the second one needs to accept it. Right? |
| 03:47.20 | norrec | yeah, the first one seems to be handing it off alright but the 2nd one doesnt like the context |
| 03:47.28 | norrec | or so it seems |
| 03:47.41 | norrec | i can pastebin the output from the first server if u like |
| 03:48.51 | [TK]D-Fender | TehRabbitt: trash your packaged install and do it yourself |
| 03:48.53 | p3nguin | norrec: Seriously. Just change the name of the contexts to MATCH. Are you okay this? |
| 03:49.06 | TehRabbitt | soo basically do everything over again? :-\ |
| 03:49.16 | p3nguin | keep your confs! |
| 03:49.18 | [TK]D-Fender | TehRabbitt: I didn't say you had to erase your CONFIGS. |
| 03:49.21 | TehRabbitt | ah true |
| 03:49.39 | p3nguin | You should be able to compile 1.4.30 in five minutes. |
| 03:49.46 | TehRabbitt | so basically just backup all my configs, (and the sccp-b module)... and reinstall? |
| 03:50.37 | p3nguin | pretty much. Be sure to rm -f /var/lib/asterisk/modules/* |
| 03:51.11 | norrec | p3nguin: where do u want me to change it? i changed line 63 to from-outside-xxxxxx5353 and it still gave the same error |
| 03:51.18 | TehRabbitt | ah true... |
| 03:51.34 | TehRabbitt | should i bother saving the chan-b module or shoudl i just remake it once asterisk is made? |
| 03:52.02 | TJNII | Backup your current system and rebuild it all so it is matched. |
| 03:52.07 | p3nguin | tehrabbitt: It will probably be okay, but it only takes a minute to rebuild it against the current asterisk. |
| 03:52.13 | TehRabbitt | true |
| 03:52.21 | TJNII | If it explodes in a firey ball of failure restore the backup. |
| 03:52.26 | TehRabbitt | *crosses fingers* here we go... aptitude remove asterisk |
| 03:52.28 | p3nguin | i.e. I would rebuild chan-sccp. |
| 03:52.28 | TehRabbitt | lol |
| 03:53.01 | TehRabbitt | i backed up all the files in "etc/asterisk/*" into a sep directory |
| 03:53.03 | TehRabbitt | i should be good to go right? |
| 03:53.47 | p3nguin | norrec: You changed [from-outside-xxxxxx5353-tl-allhours] to [from-outside]? Then you saved the file, and ran iax2 reload? |
| 03:54.03 | TJNII | You may have wanted /var/lib/asterisk too, but nothing in there is irreplaceable. /etc/asterisk is the one you really want. |
| 03:54.06 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:54.37 | TJNII | covets his precious /var/lib/asterisk/moh directory |
| 03:55.00 | *** join/#asterisk coppice (~chatzilla@m121-202-80-24.smartone-vodafone.com) |
| 03:55.04 | TJNII | I had to turn off moh on incoming calls. |
| 03:55.05 | p3nguin | I rsync my /etc/asterisk/*.conf to an SD card once a day. |
| 03:55.10 | TJNII | people stopped calling. |
| 03:55.29 | norrec | p3nguin: well i had done [from-outside-xxxxxx5353] but i just tried it with out the number as well and same deal |
| 03:55.29 | TJNII | My lounge cover of Stairway to Heaven was very effective on telemarketers, though. |
| 03:55.42 | norrec | p3nguin: and i did reload after the change |
| 03:55.48 | p3nguin | norrec: Show me the proof. |
| 03:56.06 | TehRabbitt | *deleting* |
| 03:56.07 | TehRabbitt | 0_o |
| 03:56.29 | TehRabbitt | and it's gone 0_o lol |
| 03:56.47 | norrec | p3nguin: u just want me to copy my config file again? or can i output my dialplan though asterisk? |
| 03:56.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:56.59 | norrec | p3nguin: er in the cli |
| 03:57.04 | TehRabbitt | p3nguin: should I just stick with 1.4 or go with 1.6.2? |
| 03:57.04 | p3nguin | norrec: I want to see the proof of the failure. |
| 03:57.08 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 03:57.09 | p3nguin | tehrabbitt: 1.4.30 |
| 03:57.11 | TehRabbitt | k |
| 03:57.13 | [TK]D-Fender | norrec: from extensions.conf along with your lastest failed attempt |
| 03:57.29 | TehRabbitt | whats' the difference? |
| 03:57.34 | norrec | alright give me a sec to get that |
| 03:57.43 | p3nguin | tehrabbitt: stuff |
| 03:58.02 | p3nguin | tehrabbitt: and things. |
| 03:58.35 | [TK]D-Fender | TehRabbitt: Some stuf, and lots of things; Not necessarily in that order |
| 03:59.28 | p3nguin | tehrabbitt: Think of the 1.4 branch like Debian stable and the 1.6.x branches like Ubuntu. |
| 03:59.45 | TehRabbitt | p3nguin: ah... got it |
| 03:59.57 | TehRabbitt | so basically just "make clean && make"? |
| 04:00.24 | p3nguin | Don't forget to apply your patch. |
| 04:00.29 | TehRabbitt | ah true :) |
| 04:00.30 | TJNII | You may want to make menuconfig before you make, too. |
| 04:00.37 | TehRabbitt | hm true |
| 04:00.38 | p3nguin | yeah |
| 04:00.39 | TJNII | Kind of a handy step. |
| 04:01.12 | p3nguin | You'll want to make some other targets, as well. |
| 04:01.37 | p3nguin | I think make config and make samples wouldn't be a bad idea. |
| 04:03.02 | TJNII | There is also a make option to install the init script. |
| 04:03.08 | TJNII | Don't remember what it is called. |
| 04:03.17 | norrec | p3nguin [TK]D-Fender : http://pastebin.com/5jW7siYd |
| 04:03.27 | TJNII | It is a debian init script, but it sounds like that is what you want. |
| 04:03.38 | p3nguin | make config |
| 04:03.40 | p3nguin | :) |
| 04:03.45 | TJNII | There we go. |
| 04:04.46 | TehRabbitt | make command not found :( |
| 04:04.54 | p3nguin | hahaha |
| 04:05.03 | p3nguin | apt-get install build-essentials |
| 04:05.07 | TJNII | Denied. |
| 04:05.16 | TehRabbitt | already installed |
| 04:05.16 | TehRabbitt | 0_o |
| 04:05.17 | [TK]D-Fender | norrec: is that the dialplan on * #2? |
| 04:05.29 | TehRabbitt | thoth:~/asterisk-1.4.30# make menuconfig |
| 04:05.29 | TehRabbitt | -bash: make: command not found |
| 04:05.42 | p3nguin | make isn't part of build-essentials? |
| 04:05.55 | TehRabbitt | lmfao i just made the CHAN-SCCP-B driver last night |
| 04:06.00 | p3nguin | oh yeah! |
| 04:06.06 | TehRabbitt | w......t......f..... is wrong with debian lmao |
| 04:06.41 | TJNII | ponders |
| 04:06.48 | TJNII | Is make found as root? |
| 04:06.56 | TehRabbitt | ./configure would help heh |
| 04:07.14 | TehRabbitt | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 04:07.34 | *** join/#asterisk JJJones (~jerry@68-30-199-149.pools.spcsdns.net) |
| 04:07.40 | TJNII | gently pats his Gentoo box. |
| 04:07.47 | TehRabbitt | wtf? configure: error: *** Please install GNU make. It is required to build Asterisk! |
| 04:07.48 | p3nguin | rubs Arch |
| 04:07.49 | TJNII | No such nonsense from you. No. |
| 04:07.52 | norrec | [TK]D-Fender: huh? |
| 04:08.00 | TehRabbitt | TJNII: I am jealous lmao i miss my gentoo box :( |
| 04:08.05 | TehRabbitt | it died :( |
| 04:08.09 | TehRabbitt | and then I got lazy with debian |
| 04:08.12 | [TK]D-Fender | norrec: which * is that dialplan from? |
| 04:08.18 | JJJones | Anyone here know of any SIP providers who have support on staff? My main SIP just died and I need to get another setup ASAP tonite if possible |
| 04:08.20 | TJNII | Gentoo never dies with proper backups. |
| 04:08.29 | TJNII | And proper cflags. |
| 04:08.35 | TehRabbitt | umm more like the machine just wnet up in toast litterly |
| 04:08.42 | TehRabbitt | well flames |
| 04:08.52 | p3nguin | jjjones: Do you need your DID ported right now, or only termination service? |
| 04:08.58 | TehRabbitt | hahahahahahahaha MAKE needed to be installed... yet I used it last night 0_o explain that one |
| 04:09.00 | JJJones | termination only |
| 04:09.09 | norrec | [TK]D-Fender: i'm not really sure actually, I didnt write most of this =/ |
| 04:09.11 | p3nguin | jjjones: VoIP.ms is all customer configured. Deposit money, use the service. |
| 04:09.33 | [TK]D-Fender | morrYou just pastebinning some dialplan and you can't tell me WHICH SERVER you jsut got it from? |
| 04:09.39 | TehRabbitt | w00t I love the asterisk ACSII art :-D |
| 04:09.41 | TehRabbitt | configure: Package configured for: |
| 04:09.41 | TehRabbitt | configure: OS type : linux-gnu |
| 04:09.41 | TehRabbitt | configure: Host CPU : x86_64 |
| 04:09.46 | TehRabbitt | looks like i'm good to go |
| 04:09.49 | [TK]D-Fender | norrec: You just pastebinned some dialplan and you can't tell me WHICH SERVER you got it from? |
| 04:10.14 | TJNII | Once again [TK]D-Fender is defeated by the evil Dr. Tab Completion! |
| 04:10.20 | TehRabbitt | ok i'm in the build option selection, which ones do I want to select haha |
| 04:10.46 | norrec | [TK]D-Fender: oh, which server, thats the 2nd one |
| 04:11.12 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 04:11.25 | norrec | [TK]D-Fender: sip provider -> asterisk -> iax2 -> asterisk 2 |
| 04:11.46 | norrec | [TK]D-Fender: and that is the extensions.conf of the 2nd asterisk server |
| 04:12.01 | joobie | hrm.. trying to setup a boot server for my polycom 320.. got the phone pulling down the new bootrom (v4) and also the <mac>.conf and sip.ld.. it says its running the sip.ld, comes up with ip addr then says "Config file error, Error is 0x4020" .. anyone know wtf this is? I've looked at the log the phone spits out, which doesnt really say anything apart from "0503232359|app1 |4|00|Loaded application 2345-12200-002.sip.ld successfully, errors 0x20." |
| 04:12.08 | joobie | any help appreciated... |
| 04:12.14 | TehRabbitt | p3nguin: i'm guessing I can safely unselect Skinny from "modules to install" correct? lol |
| 04:12.17 | TehRabbitt | since i'll be using SCCP anyway |
| 04:12.20 | p3nguin | yes |
| 04:12.44 | TehRabbitt | boo gtalk is disabled lol |
| 04:12.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:13.26 | [TK]D-Fender | norrec: PB "dialplan show" on #2 |
| 04:14.00 | [TK]D-Fender | joobie: Syntax error in your configs |
| 04:14.03 | TehRabbitt | what is module embedding? |
| 04:14.13 | joobie | [TK]D-Fender, would that be in sip.cnf? |
| 04:14.14 | TehRabbitt | nothing is selected inside it |
| 04:14.50 | norrec | [TK]D-Fender: http://pastebin.com/yK9KZJDL |
| 04:15.33 | [TK]D-Fender | joobie: sip.cfg or any other files referenced |
| 04:15.41 | *** join/#asterisk TehRabbitt-2 (~rabbott@c-71-59-82-2.hsd1.pa.comcast.net) |
| 04:16.04 | joobie | thanks TK |
| 04:16.06 | TehRabbitt-2 | should I select any of those options? |
| 04:16.09 | TehRabbitt-2 | p3nguin? |
| 04:16.11 | joobie | is there a way I can quickly check for syntax errors? |
| 04:16.15 | p3nguin | tehrabbitt-2: no |
| 04:16.21 | [TK]D-Fender | norrec: extensions.conf is not even being READ on server #2 |
| 04:16.23 | joobie | .. short of manually going through the whole fiel |
| 04:16.34 | [TK]D-Fender | norrec: pb "ls -la /et/asterisk" |
| 04:16.46 | [TK]D-Fender | (etc) |
| 04:16.50 | norrec | [TK]D-Fender: oh fantastic |
| 04:18.00 | TehRabbitt-2 | *running make* |
| 04:18.02 | norrec | [TK]D-Fender: http://pastebin.com/VRKr1aWK |
| 04:20.45 | [TK]D-Fender | norrec: do a reload at CLI and see if it sees it |
| 04:20.52 | norrec | k |
| 04:21.59 | *** join/#asterisk jmcdowell (~airmadnes@174-156-119-55.pools.spcsdns.net) |
| 04:22.03 | jmcdowell | hello all |
| 04:22.10 | norrec | [TK]D-Fender: well it reloads with no errors |
| 04:22.22 | jmcdowell | anyone ever seen a phone "double" digits when entering a voice mail password? |
| 04:22.31 | [TK]D-Fender | norrec: and you see the dialplan being loaded? |
| 04:22.40 | joobie | TK, mind having a squizz at my config? It's very small - http://pastebin.com/HhTDNG0t .. the top part of that pastebin shows the files that the polycom is trying to grab. It only grabs that one config file |
| 04:22.50 | norrec | [TK]D-Fender: no i only see the ael dialplan being loaded |
| 04:23.08 | TehRabbitt-2 | running make install 0_o |
| 04:23.14 | TehRabbitt-2 | now it's downloading things heh |
| 04:23.37 | p3nguin | jjjones: How's that working out for you? |
| 04:23.39 | norrec | [TK]D-Fender: hmm, i wonder if this is because i'm using 1.6.2 and i fucked something up with the configs |
| 04:24.12 | joobie | norrec, i missed what the problem was.. |
| 04:24.48 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 04:24.58 | [TK]D-Fender | norrec: manually load pbx_config.so |
| 04:25.02 | norrec | joobie: well apperently extensions.conf isnt being loaded |
| 04:25.12 | TehRabbitt-2 | p3nguin: just realized what it's downloading, all the MoH that i selected lol |
| 04:25.43 | [TK]D-Fender | joobie: 0004f216faca.cfg <- is not a valid SIP APP config file |
| 04:25.57 | [TK]D-Fender | joobie: <APPLICATION APP_FILE_PATH="2345-12200-002.sip.ld" CONFIG_FILES="0004f216faca.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="logs/" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY="" LICENSE_DIRECTORY=""/> |
| 04:26.07 | *** join/#asterisk Polysics (~Luca@host207-51-dynamic.24-79-r.retail.telecomitalia.it) |
| 04:26.17 | [TK]D-Fender | joobie: that file is used byt he boot ROM to point to the application to load and the configs associated to it. |
| 04:26.32 | [TK]D-Fender | joobie: circular reference to a wrong file |
| 04:26.43 | norrec | [TK]D-Fender: it wasnt loaded |
| 04:26.46 | joobie | 2345-12200-002.sip.ld that file exists though - which is sip.ld |
| 04:26.59 | joobie | ahhhhhh |
| 04:27.04 | TehRabbitt-2 | I shouldn't install the sample .conf files since I have all the old ones, right? |
| 04:27.04 | joobie | i think i get you |
| 04:27.27 | norrec | [TK]D-Fender: alright, now it answers the call, yey |
| 04:27.57 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 04:27.57 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:27.57 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 04:27.58 | *** join/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 04:28.01 | norrec | [TK]D-Fender: well that fixed the problem |
| 04:28.02 | joobie | TK, thanks.. i was menat to have phone1_0004f216faca.cfg in there as opposed to 0004f216faca.cfg |
| 04:28.05 | joobie | updated, trying again... |
| 04:28.15 | norrec | [TK]D-Fender: any way to find out why the module wasnt being loaded? |
| 04:28.38 | [TK]D-Fender | norrec: check your modules.conf to see if some twit tried getting "smart" in hand-picking moduiles and forgot it |
| 04:29.25 | roe | what is the preferred method of ensuring that multiple digium cards come up in the same order at boot? |
| 04:30.14 | norrec | [TK]D-Fender: hmm, well autoload is set to yes |
| 04:30.28 | joobie | TK, http://pastebin.com/L9Qwcw67 that is what i have now.. same error. I don't see the phone even attempt to get those config files specified in CONFIG_FILES= btw |
| 04:31.03 | TehRabbitt | make: *** [.tmp/sccp_actions.o] Error 1 |
| 04:31.07 | TehRabbitt | any ideas p3nguin |
| 04:31.26 | TJNII | Pastebin the output. |
| 04:31.37 | TJNII | Specifically where the actual failure occours. |
| 04:31.49 | TehRabbitt | http://pastebin.com/fSvMrwaG |
| 04:31.50 | norrec | [TK]D-Fender: *shrug* well its working now i guess, if it doesnt load on a restart should i just add it to the modules.conf? |
| 04:31.52 | TehRabbitt | thats the output |
| 04:31.59 | jmcdowell | Ha ha |
| 04:32.08 | jmcdowell | I just went through that Polycum nightmare |
| 04:32.13 | TehRabbitt | ?? |
| 04:32.13 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 04:32.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:32.31 | jmcdowell | I missed the "problem", what is it again ? |
| 04:32.38 | [TK]D-Fender | joobie: Check your perms on the files, check your server settings on the phone |
| 04:32.43 | roe | the polycom mass provisioning is awesome |
| 04:32.59 | jmcdowell | Yeah, if you like to pull your hair out. |
| 04:33.06 | jmcdowell | Granted, once it's working, it's great |
| 04:33.10 | roe | it is robust and easily to centrally mange |
| 04:33.12 | TJNII | Did you patch that file? |
| 04:33.15 | jmcdowell | getting it there, takes a few years off your life. |
| 04:33.46 | TehRabbitt | No, the SCCP is the same one I was using before, never patched nor modified it |
| 04:33.47 | roe | the newer provisioning setup is easier than what it used to be |
| 04:33.52 | TehRabbitt | in any way shape or form |
| 04:34.03 | jmcdowell | I have only seen xml files. |
| 04:34.39 | roe | my biggest complaint is actually with *all* of the manufacturers web interfaces |
| 04:34.44 | TJNII | "the same one I was using before" <- What, exactly, do you mean by this? Did you copy code into your source tree or something? |
| 04:34.46 | ChannelZ | ARGH WTF |
| 04:34.54 | joobie | TK, it says "Running ..sip.ld" on the phone and the logs also indicate that the files are all being pulled down (have the right perms). After it says that, it shows me the ip address, then it goes to the Config file error notification.. once it hits that "Running" stage, I don't see any requests to the boot server at all |
| 04:35.09 | TehRabbitt | TJNII: I downloaded the tar.gz from the website... untarred it... and ran make and that is what it is giving me |
| 04:35.15 | ChannelZ | My Windoze box has suddenly decided to start opening menus (pulldown menus, cascading menus) to the LEFT of the menus instead of the right |
| 04:35.16 | [TK]D-Fender | joobie: check your logs (FTP as well) |
| 04:35.17 | TehRabbitt | unmodified, unchanged, same exact way I did it before |
| 04:35.45 | jmcdowell | Anyone know of a GOOD GSM pci card to add sim support to asterisk ? |
| 04:35.48 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 04:36.03 | joobie | ahh sec TK |
| 04:36.12 | joobie | im doing a tcpdump on my own gateway and seeing some https traffic |
| 04:36.22 | joobie | it's doing something, investigating |
| 04:36.34 | jmcdowell | joobie : that's just some chineese hacker bot netting your system. |
| 04:37.22 | joobie | jmcdowell, ahh, i guess that explains the "prawncracker.net" src hostname |
| 04:37.43 | jmcdowell | lol |
| 04:37.56 | jmcdowell | Anyone know of a GOOD GSM pci card to add sim support to asterisk ? |
| 04:38.09 | TehRabbitt | anyone? |
| 04:38.21 | jmcdowell | everything I find related to GSM is it's own gateway |
| 04:38.25 | [TK]D-Fender | jmcdowell: not since you asked... THREE MINUTES AGO |
| 04:38.30 | jmcdowell | and that's not what I want.. |
| 04:38.54 | TehRabbitt | jmcdowell: http://cgi.ebay.com/4-PORT-GSM-Asterisk-Card-OpenVox-G400P-/180501173668?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item2a06b555a4#ht_1329wt_1165 |
| 04:39.10 | joobie | jmcdowell, why do you want sim support on asterisk? |
| 04:39.28 | jmcdowell | So my cell phone can be used on my PBX when I am @ home. |
| 04:39.40 | TehRabbitt | joobie: it could be useful for calling people who have lets say "free boost to boost mobile" calls |
| 04:39.47 | TehRabbitt | or "T-mobile to t-mobile" calls |
| 04:39.49 | joobie | nice |
| 04:39.58 | jmcdowell | I found something on ebay that claims it can interface with FPBX |
| 04:40.00 | [TK]D-Fender | jmcdowell: So you were planning on what... pulling the card out of your phone every time you come home? |
| 04:40.01 | jmcdowell | I mean asterisk |
| 04:40.15 | jmcdowell | No, just turning my phone off and using my clone sim. |
| 04:40.17 | TehRabbitt | p3nguin: you still here? |
| 04:40.51 | jmcdowell | I have callcentric set for termination at the end of this month, due to their non-standard compliance. |
| 04:41.07 | jmcdowell | So I am thinking about using only my GSM card @ home and on the road. |
| 04:41.50 | [TK]D-Fender | jmcdowell: chan_mobile <- |
| 04:42.10 | TehRabbitt | [TK]D-Fender: any ideas why my module won't compile? |
| 04:42.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:42.25 | jmcdowell | that's interesting.. |
| 04:42.32 | jmcdowell | I wonder if I could just us bluetooth.. :> |
| 04:42.53 | jmcdowell | and I can.. |
| 04:43.07 | jmcdowell | [TK]D-Fender : thanks |
| 04:43.44 | jmcdowell | Wow.. that's off the chain |
| 04:43.45 | [TK]D-Fender | jmcdowell: Your new wheel is not a unique and beautiful snowflake |
| 04:43.54 | jmcdowell | lol |
| 04:44.17 | [TK]D-Fender | A mixed-metaphor a day is woth two in a bush |
| 04:44.21 | [TK]D-Fender | orth* |
| 04:44.39 | [TK]D-Fender | TehRabbitt: No. |
| 04:45.13 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 04:53.53 | TehRabbitt | can anyone here help me? |
| 04:54.21 | TehRabbitt | make: *** [.tmp/sccp_actions.o] Error 1 |
| 04:56.08 | TehRabbitt | p3nguin: I figured out how to use serviceURL btw |
| 04:56.20 | TehRabbitt | serviceURL = Phonebook,http://webserver/phonebook.php |
| 04:56.41 | TehRabbitt | Anyway I can't get SCCP installed so i'm back to step 1 of having no operating * server |
| 05:00.05 | TehRabbitt | is there anyone here still? |
| 05:00.14 | p3nguin | yes |
| 05:00.53 | [TK]D-Fender | TehRabbitt: Any reason not to try chan_skinny on 1.6.2? |
| 05:02.28 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 05:02.51 | TehRabbitt | [TK]D-Fender: all my .conf are for 1.4 |
| 05:03.07 | TehRabbitt | p3nguin: looks like the update to g++ broke make |
| 05:04.56 | p3nguin | downgrade |
| 05:05.07 | p3nguin | Why did you upgrade, anyway? |
| 05:05.28 | p3nguin | Also, why are you not using chan-sccp from svn? |
| 05:05.42 | TehRabbitt | no SVN support :( |
| 05:06.05 | TehRabbitt | only GUI versions for gnome in debian no CLI version afaik |
| 05:06.08 | p3nguin | What does that mean? |
| 05:06.30 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:06.47 | TehRabbitt | p3nguin: http://pastebin.com/Y0j4zVpz |
| 05:07.02 | TehRabbitt | those are the only SVN packages available for debian |
| 05:07.30 | p3nguin | apt-get install subversion |
| 05:07.56 | joobie | TK, man this was weird |
| 05:08.08 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 05:08.23 | joobie | TK, I temporarily made the boot server http (just by changing dhcp option 66).. the phone boots |
| 05:08.28 | joobie | then i change ti back to https, and the phone now boots |
| 05:08.40 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 05:09.33 | p3nguin | tehrabbitt: Where did you find the info on serviceURL? |
| 05:09.58 | TehRabbitt | nice little file called "serviceURL" in the /doc/ dir of chan_sccp-b |
| 05:09.59 | TehRabbitt | lol |
| 05:10.12 | p3nguin | I knew I saw it somewhere! |
| 05:10.21 | TehRabbitt | yupp :-D same here i was going nuts trying to find it too |
| 05:10.24 | *** join/#asterisk diegomad (mad@190.158.77.205) |
| 05:10.58 | p3nguin | Google doesn't even have a copy of it. |
| 05:11.03 | TehRabbitt | yea :( |
| 05:11.09 | TehRabbitt | should... post that on my website for hits haha |
| 05:11.29 | TehRabbitt | jk |
| 05:11.32 | TehRabbitt | sighhh |
| 05:11.36 | TehRabbitt | sccp sitll wont compile |
| 05:11.48 | p3nguin | using svn this time? |
| 05:12.25 | TehRabbitt | got it 0_o |
| 05:12.27 | TehRabbitt | phew |
| 05:12.35 | TehRabbitt | ============================ |
| 05:12.35 | TehRabbitt | | | |
| 05:12.35 | TehRabbitt | | | | | |
| 05:12.35 | TehRabbitt | | :|: :|: | |
| 05:12.35 | TehRabbitt | | :|||: :|||: | |
| 05:12.36 | TehRabbitt | | .:|||||||:..:|||||||:. | |
| 05:12.36 | TehRabbitt | | CHAN_SCCP_v2 | |
| 05:12.37 | TehRabbitt | ============================ |
| 05:13.00 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 05:13.17 | frk2 | So are grandstreams still crap or have they gotten better in the past 2 years? |
| 05:13.26 | [TK]D-Fender | ... |
| 05:13.32 | [TK]D-Fender | thedo not flood like that again |
| 05:13.47 | TehRabbitt | p3nguin: and it's alive... again haha |
| 05:14.58 | frk2 | Grandstreams are the only semi affordable phones for the third world. too bad they always suck :) |
| 05:15.03 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
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| 05:16.07 | TehRabbitt | p3nguin: the serviceURL works :-D |
| 05:16.46 | p3nguin | tehrabbitt: What did you configure for it? |
| 05:17.19 | TehRabbitt | that XML file on that site from b4... once i select the softkey for it, it opens up the menu full screen and lets me choose weather, stocks, etc |
| 05:17.20 | *** join/#asterisk gospch (~gospch@p5088F4FB.dip.t-dialin.net) |
| 05:17.20 | TehRabbitt | lol |
| 05:17.40 | *** join/#asterisk GameGamer43 (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 05:17.53 | TehRabbitt | if Only I could specify what the different softkeys did 0_o |
| 05:18.04 | p3nguin | So what does your "serviceURL =" look like? |
| 05:18.07 | TehRabbitt | i've got one that looks like a "help" logo |
| 05:18.17 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 05:18.25 | TehRabbitt | serviceURL = menu,http://phone-xml.berbee.com/menu.xml |
| 05:18.38 | p3nguin | In SIP, you just specify the service URL and pressing the Services button on the phone brings up that service URL. |
| 05:18.56 | p3nguin | no softkey involved. |
| 05:19.08 | TehRabbitt | what u mean? |
| 05:19.39 | p3nguin | services_url: "http://phone-xml.berbee.com/menu.xml" ; |
| 05:20.14 | p3nguin | Then pressing the services/globe button runs the service. |
| 05:20.21 | TehRabbitt | MJ is still not registring :( |
| 05:20.26 | frk2 | so nobody there to tell me that grandstreams now rock? :) (thats what i wanna hear) hahah |
| 05:20.30 | p3nguin | Did you remember to patch? |
| 05:20.35 | TehRabbitt | Yep |
| 05:20.36 | TehRabbitt | :( |
| 05:20.44 | p3nguin | ~gs |
| 05:20.45 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 05:20.54 | TehRabbitt | lol |
| 05:20.57 | TehRabbitt | Grand Suck? |
| 05:21.03 | frk2 | nooooooooo |
| 05:21.08 | frk2 | :( |
| 05:21.15 | frk2 | do they still suck? |
| 05:21.22 | frk2 | how can they suck for 5+ years is my question |
| 05:21.28 | frk2 | and still be in business |
| 05:21.29 | TehRabbitt | how can what suck? |
| 05:21.36 | frk2 | grandstream |
| 05:21.45 | TehRabbitt | not using grandstream afaik |
| 05:21.46 | frk2 | I know they sucked bigtime 3 years ago |
| 05:21.50 | TehRabbitt | using magicjack lmao |
| 05:22.05 | Jumpie | S |
| 05:22.10 | Jumpie | Savitha....is that a female or male name? |
| 05:22.11 | Jumpie | its indian |
| 05:22.12 | Jumpie | hehe |
| 05:22.19 | Nugget | frk2: http://spreadsheets.google.com/ccc?key=0At-N6lnvzmbRdDQtek9qMS1uWXowOHVCYU03dmlhUUE&hl=en |
| 05:22.37 | frk2 | Nugget, whats that? |
| 05:22.51 | frk2 | hahahah |
| 05:22.54 | Nugget | :D |
| 05:23.08 | frk2 | Nugget, dont know what to make of that :D |
| 05:23.22 | frk2 | but i guess crappiness is going down |
| 05:23.34 | frk2 | isn't their sole livelihood based on selling IP phones? |
| 05:23.35 | Nugget | not sure the data is enlightening, but more data are always better. |
| 05:25.30 | [TK]D-Fender | checkout time. Later all |
| 05:25.50 | p3nguin | tehrabbitt: That's so very different from service_url in SIP. |
| 05:26.05 | *** join/#asterisk grind (~somebody@203.185.208.156) |
| 05:26.16 | grind | hey guys |
| 05:26.27 | p3nguin | I want to configure the services button as opposed to create a new button under my line key. |
| 05:26.47 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uicgqydejzgcierp) |
| 05:26.58 | grind | i have an external (out of the office) phone which rings but neither end can hear any audio. the asterisk server shows "bad event" when doing a TCPDUMP |
| 05:31.01 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 05:32.20 | frk2 | is there a simpler way to let the user add SPEED dial to the Cisco 7911 phones? |
| 05:37.54 | TehRabbitt | p3nguin: any way to get that google voice thing working? |
| 05:37.59 | TehRabbitt | instead of using MJ? |
| 05:38.04 | p3nguin | google it |
| 05:38.19 | TehRabbitt | all I find are articles about prior to google buying GC |
| 05:38.26 | *** join/#asterisk Faithful (~Faithful@121.91.127.126) |
| 05:38.38 | p3nguin | pygooglevoice |
| 05:38.46 | p3nguin | See if that turns up anything. |
| 05:39.14 | p3nguin | also orgasmatron |
| 05:39.34 | TehRabbitt | wtf is that lol |
| 05:40.45 | Jumpie | sup guys |
| 05:44.40 | TehRabbitt | which is better SIPgate or IPKall |
| 05:45.04 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:45.21 | TehRabbitt | SIP outgoing or IAX outgoing? which is better? |
| 05:45.59 | p3nguin | Depends on what you want to do. |
| 05:46.10 | p3nguin | ipkall doesn't do termination. |
| 05:46.18 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 05:46.21 | TehRabbitt | dont' need termination apparently |
| 05:46.25 | xheliox | SIP is a robust standard with lots of support.. |
| 05:46.29 | xheliox | IAX.. well.. |
| 05:46.44 | p3nguin | You decided not to make phone calls? |
| 05:46.48 | TehRabbitt | LOL |
| 05:47.01 | TehRabbitt | p3nguin: i'll just keep talking to the voices in my head ;-) jk |
| 05:47.30 | coppice | SIP has lots of support, but robust is stretching reality |
| 05:47.30 | xheliox | I find the voices in my head provide the best conversation. |
| 05:47.32 | p3nguin | IPKall will give you a free WA phone number, but that's about all. |
| 05:47.46 | xheliox | coppice: In comparison. :) |
| 05:48.02 | TehRabbitt | hmph |
| 05:48.11 | TehRabbitt | so basically IPKall is just incoming? |
| 05:48.21 | p3nguin | Not That's all it is. |
| 05:48.24 | p3nguin | err |
| 05:48.38 | p3nguin | No, that's all it is. Not basically, but entirely. |
| 05:48.54 | TehRabbitt | lol so basically it's an incoming trunk but it can do more? |
| 05:48.59 | p3nguin | no |
| 05:49.03 | p3nguin | It's not a trunk at all. |
| 05:49.06 | p3nguin | It's just a DID. |
| 05:49.06 | TehRabbitt | Unfortunately, we do not have a number in New Jersey for you at this time. Please, bear with us as we are continously expanding our footprint. In the meantime we can offer you a free number in California. |
| 05:49.08 | TehRabbitt | lmfao |
| 05:50.08 | p3nguin | You can also get a free DID from IPcomms. |
| 05:50.44 | TehRabbitt | DID == incoming calls? |
| 05:50.52 | p3nguin | ~did |
| 05:50.53 | infobot | did is, like, Direct Inward Dialing, or just a phone number |
| 05:51.02 | TehRabbitt | can you receive calls on it though lol |
| 05:51.05 | p3nguin | Yes, INCOMING CALLS. |
| 05:51.08 | Jumpie | haha |
| 05:51.14 | Jumpie | the next option for a NJ number is california? |
| 05:51.15 | Jumpie | mega fail |
| 05:51.19 | TehRabbitt | lmao yep |
| 05:51.23 | TehRabbitt | and there was only 1 available in cali |
| 05:51.24 | p3nguin | YOU CAN'T CALL OUT of a DID. |
| 05:51.44 | Jumpie | p3nguin i think it ust helps people visualize a DID as a path :) |
| 05:51.48 | Jumpie | but yea..thats true |
| 05:51.57 | TehRabbitt | so basically a DID is a number I can give to ppl and say "call me" and it'll allow the incoming call.. but i need to terminate outgoing calls to call outbound? |
| 05:51.59 | p3nguin | sipgate always offers a California phone number. |
| 05:52.00 | Jumpie | what are you tryin tod oe xactly TehRabbitt? |
| 05:52.10 | Jumpie | i can get numbers from anywhere, any time ;) |
| 05:52.12 | p3nguin | ~itsp |
| 05:52.13 | 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. |
| 05:52.18 | Jumpie | although sometimes, requests take a bit longer |
| 05:52.20 | p3nguin | tehrabbitt: read this ^^^ |
| 05:52.28 | Jumpie | i had to get 10 sequential washington dc numbers and verizon didnt have any readilly available |
| 05:52.52 | TehRabbitt | lmfao damn |
| 05:52.58 | TehRabbitt | why did you need 10 sequential DC numbers lol |
| 05:52.59 | p3nguin | SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination) |
| 05:53.05 | Jumpie | TehRabbitt lawfirm |
| 05:53.09 | Jumpie | they had OEN did |
| 05:53.19 | TehRabbitt | so IPKall == Free Origination? |
| 05:53.25 | p3nguin | yes |
| 05:53.26 | Jumpie | and were paying for 15 pots lines, long distance, and a 900/mo lease on a shitty pbx |
| 05:53.31 | p3nguin | one DID |
| 05:53.36 | Jumpie | their totalc osts were like $3800 a month |
| 05:53.41 | TehRabbitt | ah lol so they pay long distance to call me... but I get to receive their calls for free? lmao |
| 05:53.48 | p3nguin | right |
| 05:53.56 | Jumpie | i got them a bundle t1 w/ qos, 10 DIDS with unlimited incoming |
| 05:54.01 | TehRabbitt | haha cool :-p unless I have google voice foward calls TO that #? |
| 05:54.01 | Jumpie | for under 1600/mo |
| 05:54.02 | TehRabbitt | ;) |
| 05:54.11 | p3nguin | tehrabbitt: exactly |
| 05:54.12 | TehRabbitt | and my MJ # foward to that # as well |
| 05:54.14 | TehRabbitt | ;) |
| 05:54.32 | *** join/#asterisk Tim_Toady (~moi@193.92.246.150.dsl.dyn.forthnet.gr) |
| 05:54.34 | TehRabbitt | and then just pay outgoing via ERmmmmmm one of those 2 you sent me earlier at 1 cent a minute |
| 05:54.35 | TehRabbitt | lol |
| 05:54.40 | p3nguin | tehrabbitt: That's exactly what I do. My local GV number forwards to my IPkall number, which comes to me by SIP URI. |
| 05:54.52 | TehRabbitt | Hm... lol |
| 05:55.08 | Jumpie | what's gv? |
| 05:55.13 | TehRabbitt | Google Voice |
| 05:55.13 | p3nguin | Google Voice |
| 05:55.14 | Jumpie | oh google voice |
| 05:55.22 | Jumpie | p3nguin..isnt that a lot of hands in the pot? |
| 05:55.30 | Jumpie | i mean it works but...sounds like a lot of middlemen |
| 05:55.51 | p3nguin | It's absolutely no different than having GV calls go to Gizmo5, which then sends to me via SIP URI. |
| 05:56.13 | Jumpie | i guess im ust used to commercial itsp |
| 05:56.16 | TehRabbitt | hm |
| 05:56.18 | p3nguin | There aren't a lot of options with GV's call forwarding. |
| 05:56.19 | Jumpie | that just send it directly to me |
| 05:56.39 | Jumpie | i think googlevoice is ok for a one person thing but for business i tend to offer more cost effective/relaible solutions |
| 05:56.53 | p3nguin | directly to you... via SIP or IAX2 |
| 05:56.59 | TehRabbitt | Ummm google voice isn't working :( grrr lol |
| 05:57.11 | Jumpie | yeah |
| 05:57.19 | p3nguin | It's no different. |
| 05:57.48 | Jumpie | so gv is freeinbound |
| 05:57.52 | Jumpie | and you have to find somebody else for outobund? |
| 05:58.37 | p3nguin | If GV forwards a call to another phone number on the PSTN, then that routes to me via SIP... that is exactly the same as a call from one person to your ITSP and then to you via SIP. |
| 05:59.17 | p3nguin | You can originate your calls from their web interface for free. |
| 05:59.38 | p3nguin | You just don't pick up your phone and call outbound through google. |
| 05:59.45 | Jumpie | i guess i like having one stop provider |
| 05:59.46 | Jumpie | heh |
| 06:00.36 | Jumpie | what it sounds like is essentially then, you have 2 isps? |
| 06:00.55 | Jumpie | gv is basically one |
| 06:01.05 | TehRabbitt | sigh GV wont take the DMTF tones :( |
| 06:01.12 | TehRabbitt | through sipgate |
| 06:01.16 | p3nguin | tehrabbitt: been there! |
| 06:01.22 | TehRabbitt | how do i fix it haha |
| 06:01.46 | *** join/#asterisk sourcode (~code@ppp-58-8-238-176.revip2.asianet.co.th) |
| 06:02.23 | p3nguin | You just need a little bit of savvy and creativity. |
| 06:03.46 | p3nguin | You could try changing your dtmfmode, or even letting the dialplan send the digits. |
| 06:04.43 | TehRabbitt | it's calling my CELL :( and it wont accept lmfao |
| 06:06.26 | *** join/#asterisk gelo (~gelo@mx01.quobis.com) |
| 06:06.39 | p3nguin | I wish google knew something about the services key on my blasted phone. |
| 06:08.57 | grind | i have an external (out of the office) phone which rings but neither end can hear any audio. the asterisk server shows "bad event" when doing a TCPDUMP |
| 06:09.10 | p3nguin | ~sipnat |
| 06:09.11 | 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 |
| 06:09.15 | p3nguin | grind: ^^^^^^^ |
| 06:09.18 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 06:09.23 | grind | gracias |
| 06:10.11 | Jumpie | does gv work natively with android phones? |
| 06:10.16 | Jumpie | im getting the moto cliq |
| 06:10.22 | Jumpie | or at least any sip client? |
| 06:10.34 | frk2 | Has anybody here ever used LB/Magneto lines with Asterisk? |
| 06:10.42 | frk2 | shouldn't be work with the FXO interface? |
| 06:11.05 | TehRabbitt | p3nguin: SIP wont register with Gizmo |
| 06:11.05 | TehRabbitt | :( |
| 06:12.37 | p3nguin | Yeah? Does it need to? |
| 06:12.57 | p3nguin | I don't think they accept registrations from you unless you are paying for callOUT services. |
| 06:13.08 | TehRabbitt | oh lol |
| 06:13.13 | TehRabbitt | it doesnt need to? |
| 06:13.17 | p3nguin | http://pastebin.com/wLS9wmfC |
| 06:13.47 | p3nguin | You need to configure the service to send calls to you via SIP URI, exactly the same way IPKall does. |
| 06:13.59 | coppice | ah, magneto lines takes me back to my youth :-\ |
| 06:14.42 | p3nguin | tehrabbitt: How long have you had your Gizmo5 account? |
| 06:15.38 | TehRabbitt | A while lol why? |
| 06:15.46 | TehRabbitt | i wanna say umm september |
| 06:15.51 | TehRabbitt | never really used it though |
| 06:15.53 | TehRabbitt | why? |
| 06:15.55 | p3nguin | I was going to ask you how you weaseled an account out of them. |
| 06:16.14 | TehRabbitt | hahahaha ah |
| 06:16.19 | p3nguin | Registrations have been closed for a few months. |
| 06:16.31 | TehRabbitt | yea :( lol i forgot I had an account actually haha |
| 06:16.48 | TehRabbitt | I got the account in sept when i was messing around with PBXes |
| 06:17.47 | TehRabbitt | sooo, now that that's in the SIP... how do I specify which extension to DIAL() when a call comes in on that DID? |
| 06:19.59 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 06:20.13 | TehRabbitt | ??? |
| 06:20.24 | p3nguin | You don't dial extensions. |
| 06:20.29 | p3nguin | extensions dial phones |
| 06:20.43 | TehRabbitt | ok, lemme rephrase, how do I dial phones |
| 06:20.52 | TehRabbitt | a call comes in on my DID... then what? |
| 06:20.59 | p3nguin | exten => yourDIDnumber,1,Dial(SCCP/yourphone) |
| 06:21.19 | TehRabbitt | oohhhh lol |
| 06:21.33 | TehRabbitt | can I have it ring a group of phones? |
| 06:21.37 | p3nguin | sure |
| 06:21.46 | p3nguin | exten => yourDIDnumber,1,Dial(SCCP/yourphone&SIP/200) |
| 06:21.53 | TehRabbitt | ah |
| 06:22.07 | p3nguin | or sequential... |
| 06:22.12 | p3nguin | exten => yourDIDnumber,1,Dial(SCCP/yourphone) |
| 06:22.21 | p3nguin | exten => yourDIDnumber,n,Dial(SIP/200) |
| 06:22.23 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 06:22.26 | TehRabbitt | is there a way to define something like [users] and have it dial everyone in the [users] group? |
| 06:22.37 | p3nguin | Oh, need timeouts. |
| 06:22.38 | p3nguin | exten => yourDIDnumber,1,Dial(SCCP/yourphone,30) |
| 06:22.42 | p3nguin | exten => yourDIDnumber,n,Dial(SIP/200,30) |
| 06:22.42 | TehRabbitt | yea i remember the 1,Dial thning... did that with setting up voicemail |
| 06:22.50 | TehRabbitt | if it rings for 30 sec with no answer, go to voicemail etc |
| 06:23.22 | p3nguin | You can create groups, but I can't remember the specifics. |
| 06:23.38 | TehRabbitt | ah |
| 06:24.38 | p3nguin | It's not that hard to write the devices joined with a couple & symbols, so that's how I do it. |
| 06:26.45 | voxter | any of you have experience using asterisk on xen? |
| 06:28.50 | p3nguin | Hmm, I've developed a new problem. Every time I have made a change to sccp.conf, I've unloaded and loaded the chan_sccp module to load the changes. |
| 06:29.07 | p3nguin | Suddenly, unloading the chan_sccp module crashes asterisk. Every time. |
| 06:29.22 | grind | I tried alot of stuff from those pages p3nguin with no luck, i just dont get it. The phone rings, asterisk -r shows it pick up -- but no voice is transmissted -- asterisk then see's the call end |
| 06:29.45 | TehRabbitt | gizmo still isn't working :( |
| 06:29.56 | p3nguin | grind: Did you set up all the NAT stuff on the Asterisk server? |
| 06:29.58 | TehRabbitt | i call my GV number, and it rings, rings rings, GV voicemail |
| 06:30.16 | grind | yea |
| 06:30.43 | grind | this phone used to work in building "a" but now this person has moved to another building in other town and its a no go |
| 06:33.37 | frk2 | man. this customer is up my a** to integrate their LB/Magneto lines with asterisk |
| 06:33.43 | *** join/#asterisk Corydon76-dig (gray@c-69-137-80-31.hsd1.tn.comcast.net) |
| 06:33.44 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 06:33.51 | frk2 | Is that theoretically possible with a FXO interface? |
| 06:34.02 | grind | man i hate customers |
| 06:34.10 | TehRabbitt | wtf is a Magneto line? |
| 06:34.16 | frk2 | TehRabbitt, exactly |
| 06:34.20 | TehRabbitt | lol |
| 06:34.25 | frk2 | these are old school crank phones |
| 06:34.31 | TehRabbitt | 0_o |
| 06:34.31 | frk2 | like the ones they used in WW-2 |
| 06:34.40 | TehRabbitt | why would they want to use those? |
| 06:34.58 | p3nguin | tehrabbitt: What did you set gv to send calls to? |
| 06:35.35 | bn-7bc | grind: well,of corse you cheked tihis alredy, but did the phone register to an internal sip egistrat/gw with an internalip? |
| 06:36.13 | *** join/#asterisk [OpenSys] (~vasco@fw.vslinux.net) |
| 06:36.14 | TehRabbitt | Gizmo # |
| 06:36.21 | TehRabbitt | gv calls the gizmo 747 number |
| 06:36.39 | TehRabbitt | * isnt' getting the call though I dont think |
| 06:36.45 | p3nguin | Did you configure gizmo to send calls to you via SIP URI? |
| 06:36.54 | TehRabbitt | dont know how to do that |
| 06:36.55 | TehRabbitt | <PROTECTED> |
| 06:36.55 | TehRabbitt | Configure your IP phone or SIP device Learn More |
| 06:37.27 | Jumpie | frk2 hah!! i have seen those when i was in the military |
| 06:37.33 | Jumpie | they are pretty much relics |
| 06:37.41 | TehRabbitt | wait, foward calls to SIP what? |
| 06:37.41 | p3nguin | tehrabbitt: Click on the Call Forwarding tab. |
| 06:37.44 | TehRabbitt | yea then what 0_o |
| 06:37.58 | p3nguin | Mark forward all calls. |
| 06:37.58 | Jumpie | what on earth is the customer smoking? that's like insisting on using a typewriter |
| 06:38.00 | frk2 | Jumpie, i wonder if I can just shove them into a FXO interface |
| 06:38.09 | frk2 | Jumpie, it IS the army |
| 06:38.10 | p3nguin | Mark forward to SIP. |
| 06:38.10 | Jumpie | i dont remember their interface... |
| 06:38.10 | TehRabbitt | selected that.... |
| 06:38.16 | Jumpie | frk2 oh...shiz |
| 06:38.17 | Jumpie | heh |
| 06:38.21 | coppice | jumpie: you'd be surprised. in the early 90s financial districts were still installing lots of them |
| 06:38.30 | p3nguin | Put in your SIP URI for it to call. |
| 06:38.39 | frk2 | so shoving into FXO makes sense? |
| 06:38.42 | TehRabbitt | what's the SIP URI? |
| 06:38.48 | TehRabbitt | thats what I dont know 0_o |
| 06:38.51 | Jumpie | coppice yea..and that industry is also on the 20 year old mainframes thats 'too big a project to overhaul' and needed all those cobol programmers on y2k scare |
| 06:38.59 | Jumpie | congrats our money is ran by antiquated morosn |
| 06:39.07 | frk2 | Jumpie, indeed |
| 06:39.15 | frk2 | old equipment is better for the vendor's pockets |
| 06:39.16 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:39.23 | TehRabbitt | Jumpie: ADP is still using Token-ring and servers running very early OS/2 |
| 06:39.26 | p3nguin | tehrabbitt: mine is my_gizmo#@myhost.com |
| 06:39.32 | Jumpie | well they also want to suck out every last penny theyc an of investment before they are forced to adapt |
| 06:39.35 | Jumpie | same with telcos :D |
| 06:39.40 | frk2 | okay okay |
| 06:39.40 | TehRabbitt | I toured their datacenter once.... |
| 06:39.40 | frk2 | SO |
| 06:39.42 | frk2 | CAN |
| 06:39.44 | Jumpie | damn skippy they wanna suck all the copper ivnestment |
| 06:39.48 | TehRabbitt | they said if those machiens turned off, there is a chance they woudln't turn back on |
| 06:39.50 | TehRabbitt | lmfao |
| 06:39.51 | frk2 | I shove those two wires into a FXO interface? |
| 06:39.51 | coppice | there was nothing old about the kit we made, though I have no idea why they wanted to use such an old interface |
| 06:39.51 | Jumpie | TehRabbitt...so is NOAA |
| 06:39.54 | frk2 | or will things blow up |
| 06:39.56 | grind | bn-7bc - sip show peers shows it has an external ip address |
| 06:40.01 | Jumpie | frk2 i wouldnt do that yet... |
| 06:40.06 | Jumpie | is this the dark green/red wires? |
| 06:40.13 | TehRabbitt | p3nguin: what is myhost.com |
| 06:40.19 | frk2 | Jumpie, I have no idea |
| 06:40.20 | Jumpie | i need refreshment to look at the interface...do they even interface with PSTN? |
| 06:40.27 | p3nguin | tehrabbitt: whatever your asterisk box answers to, I guess. |
| 06:40.30 | frk2 | no they cannot |
| 06:40.31 | grind | wait wat, it has a random port |
| 06:40.37 | grind | 10243 |
| 06:40.38 | grind | wtf |
| 06:40.46 | p3nguin | tehrabbitt: IP address or hostname, as long as gizmo can reach it via internet. |
| 06:40.46 | TehRabbitt | ok... |
| 06:40.54 | Jumpie | frk..what is the official namenclature |
| 06:40.56 | Jumpie | model numbe, etc |
| 06:40.56 | frk2 | they send 12v over the two wires |
| 06:41.07 | frk2 | that rings the other magneto phones |
| 06:41.09 | Jumpie | i think you'd need some kinda custom interface first |
| 06:41.12 | frk2 | basically charges up the coil |
| 06:41.13 | Jumpie | yea.... |
| 06:41.17 | bn-7bc | grind: well then it is probably a firewall somwhere thet blocks sip |
| 06:41.20 | Jumpie | remember the dudes that would go on the front lines with a spool of copper |
| 06:41.28 | Jumpie | to connect that shit |
| 06:41.34 | Jumpie | its basically a 2 cups and string technology |
| 06:41.38 | frk2 | yeah |
| 06:41.40 | frk2 | so gay |
| 06:41.53 | Jumpie | whoever wants you to try to get those into asterisk iss moking crack |
| 06:41.55 | grind | bn-7bc - yea now that i see random port that sounds about right, any idea why its using 10243? ive set everything to 5060 |
| 06:42.06 | Jumpie | frk2 can you get me a make/model exactly? |
| 06:42.09 | Jumpie | i dont remember the official term |
| 06:42.33 | Jumpie | http://www.myinsulators.com/commokid/telephones/ww2_phones.htm ? |
| 06:42.41 | Jumpie | stuff liek that? |
| 06:42.42 | Jumpie | heeh |
| 06:42.45 | frk2 | Jumpie, let me ask |
| 06:42.46 | grind | bn-7bc - Status says OK and it still rings which stumps me :\ |
| 06:42.53 | frk2 | this is a field wireless set |
| 06:43.07 | frk2 | not very old |
| 06:43.12 | p3nguin | tehrabbitt: Are you still scratching your head? |
| 06:43.15 | frk2 | made in the USA :) |
| 06:43.17 | TehRabbitt | ... those are going to work with asterisk?!? |
| 06:43.18 | TehRabbitt | lmafao |
| 06:43.43 | Jumpie | well..i was in the military 1998-2006 |
| 06:43.46 | Jumpie | i helped field THSDN |
| 06:43.50 | bn-7bc | grind: so ring goes trough but bo sound,hmm |
| 06:43.56 | Jumpie | which was basically overhaulign the old x.25 crap |
| 06:43.58 | grind | bn-7bc - correct |
| 06:44.10 | Jumpie | the phones were integrated into our tactical equipment, normally isyscon stuff |
| 06:44.10 | grind | same codecs as the other identical phones too |
| 06:44.11 | Jumpie | we didtn use that |
| 06:44.19 | Jumpie | so when you say 'not very old' can be relative |
| 06:44.24 | frk2 | well |
| 06:44.26 | frk2 | 1970's |
| 06:44.28 | Jumpie | ok |
| 06:44.29 | frk2 | not 40's :) |
| 06:44.31 | Jumpie | rofl |
| 06:44.37 | Jumpie | still gettina model # will help |
| 06:44.42 | TehRabbitt | p3nguin: still nothing :( |
| 06:44.47 | frk2 | Jumpie, im trying |
| 06:44.50 | p3nguin | tehrabbitt: What have you done, now? |
| 06:45.01 | bn-7bc | grind: then rtp is blocked, dou you have access to a stun server |
| 06:45.15 | TehRabbitt | p3nguin: 17474945623@thoth.tenehawk.com |
| 06:45.21 | TehRabbitt | and it doesn't work |
| 06:45.28 | grind | bn-7bc - i dont believe so |
| 06:45.31 | Jumpie | frk2 http://www.csl.army.mil/usacsl/publications/NCWCS%20Volume%202/20%20NCWCS%20Volume%202%20%28Appendix%20D%29.pdf |
| 06:45.35 | Jumpie | this is pretty much what i was used to |
| 06:45.43 | Jumpie | now imagine physically altering those shelters |
| 06:45.53 | Jumpie | with cisco and hp gear, and welding FE/fiber interfaces |
| 06:46.09 | p3nguin | tehrabbitt: Does the call reach you at all? |
| 06:46.57 | Jumpie | btw that pdf is interesting readin :D |
| 06:47.17 | TehRabbitt | it is interesting reading |
| 06:47.25 | TehRabbitt | Call doesn't go through at all :( |
| 06:47.26 | Jumpie | i had to know every damn switch, fuse, wire, port, protocol, cable |
| 06:47.35 | Jumpie | in training they would put bugs like elbow deeps into the circuitry |
| 06:47.37 | Jumpie | turn out the lights |
| 06:47.40 | Jumpie | and give us 4 hours to figure it out |
| 06:48.02 | Jumpie | on my final exam i had a 1 inch 50 cent fuse about 3 hours worth of work deep into the base |
| 06:48.03 | p3nguin | tehrabbitt: Are you sure that your IP address is updated? |
| 06:48.05 | Jumpie | lol |
| 06:48.27 | TehRabbitt | lemme check |
| 06:49.27 | *** join/#asterisk Corydon76-dig (twelve@c-69-137-80-31.hsd1.tn.comcast.net) |
| 06:49.27 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 06:49.33 | TehRabbitt | p3nguin: yes it's valid |
| 06:49.38 | TehRabbitt | IP address is up to date |
| 06:49.44 | p3nguin | Now I have to test. |
| 06:49.53 | bn-7bc | grind: just to chek is there a nat between youre astreisk and the phone |
| 06:51.17 | Jumpie | wow.... |
| 06:51.31 | Jumpie | friedn just msg me the 2010 pirelli calendar topless photoshoot pic |
| 06:51.36 | Jumpie | VERY nice |
| 06:51.54 | grind | bn-7bc yea |
| 06:52.30 | p3nguin | tehrabbitt: Did you see that call? |
| 06:52.38 | TehRabbitt | nope |
| 06:52.55 | TehRabbitt | i'm hearing "that call cannot be completed as dialed" when I call the 747 number |
| 06:52.56 | p3nguin | tehrabbitt: I guess you don't have sip ports forwarded properly. |
| 06:53.08 | TehRabbitt | female voice... is that asterisk? |
| 06:53.11 | p3nguin | I just called you via SIP URI. |
| 06:53.13 | p3nguin | probably |
| 06:53.21 | TehRabbitt | Hm... |
| 06:53.31 | TehRabbitt | I have debugging off lemme turn it on |
| 06:53.35 | bn-7bc | grind: well there is yore problem no stun andthe nat does bot forward rtp to the phone |
| 06:53.35 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 06:53.55 | p3nguin | Have you ever received any SIP calls inbound? |
| 06:54.12 | TehRabbitt | Yes I have but no audio |
| 06:54.26 | grind | thanks for your help bn-7bc |
| 06:54.27 | bn-7bc | grind: does this costumer have one or multiplephones at the location in guestion |
| 06:54.32 | TehRabbitt | the thing is i'm not even seeing a ring on a phone here :( |
| 06:54.48 | grind | just 1 bn-7bc |
| 06:55.16 | p3nguin | I think you've failed in configuring the networking portion. |
| 06:56.23 | TehRabbitt | :( |
| 06:56.25 | TehRabbitt | howso |
| 06:56.37 | bn-7bc | grind: well then they nedd to setup portforwarding and a static localip on the phhone,, no problem gkad to help,but now I@m off to work |
| 06:56.38 | Jumpie | TehRabbitt he said you fail |
| 06:57.11 | p3nguin | no audio == fail |
| 06:57.11 | TehRabbitt | sighh... |
| 06:57.15 | TehRabbitt | yea :( |
| 06:57.27 | TehRabbitt | "The Call cannot be completed as dialed please check the number etc" |
| 06:57.34 | TehRabbitt | when I call the asterisk box through SIP |
| 06:57.37 | Jumpie | TehRabbitt there are about 126872068707826 reasons you can get that |
| 06:57.37 | p3nguin | What kind of router do you have? And if you say Belkin, I'm leaving. |
| 06:57.43 | grind | tips hat to bn-7bc |
| 06:58.03 | TehRabbitt | Linksys WRT160N running DD-WRT v24 |
| 06:58.29 | p3nguin | Did you bother forwarding the ports to the * box? |
| 06:58.33 | TehRabbitt | Yes |
| 06:58.37 | TehRabbitt | 10000-20000 |
| 06:58.41 | p3nguin | port 5060 and 10000-20000 all udp? |
| 06:58.51 | TehRabbitt | and 5060 to 5082 |
| 06:58.55 | TehRabbitt | all udp |
| 06:59.17 | bn-7bc | grind: : hold on,does that phone sypport IAX/iax2 |
| 06:59.20 | p3nguin | but not 10000-20000? |
| 06:59.37 | TehRabbitt | no 10,000-20,000 were enabled as well |
| 07:00.00 | grind | not sure bn-7bc, its a linksys SPA942, i'll check |
| 07:00.05 | p3nguin | Did you ever fix the whole DMZ problem you mentioned yesterday? |
| 07:00.15 | TehRabbitt | nope I think this is the same issue |
| 07:00.28 | p3nguin | You can't un-DMZ the IP address? |
| 07:00.42 | TehRabbitt | what do you mean? |
| 07:00.55 | grind | looks like it does bn-7bc |
| 07:00.56 | p3nguin | You said something about DMZ being enabled for the * box. |
| 07:01.08 | p3nguin | But now you're telling me that you're forwarding ports. |
| 07:01.14 | p3nguin | does not compute! |
| 07:01.17 | p3nguin | You can't do both. |
| 07:01.32 | p3nguin | Turn off the stupid DMZ setting and forward the damn ports. |
| 07:01.47 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 07:01.57 | bn-7bc | grind: great use that instead of sip+rtp everythong on one port, that makes it easier trou nat |
| 07:02.12 | TehRabbitt | hm.... should I bother updating firmware to support Milkfish SIP or is it pointless? |
| 07:02.13 | p3nguin | I wish they wouldn't even have a DMZ setting on it, because everyone wants to use it without even knowing why they want to use it. |
| 07:02.17 | grind | i'll lookinto it, thanks again bn-7bc |
| 07:02.46 | p3nguin | Fix your router. Then we'll continue testing. |
| 07:03.07 | TehRabbitt | ok ports are fowarded, DMZ is off |
| 07:03.20 | bn-7bc | :grind do that and drop me a line in årivate char so it does not get drowned in the channel |
| 07:03.27 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 07:03.34 | TehRabbitt | NICE "you have reached a non working number" now that DMZ is of |
| 07:03.35 | TehRabbitt | off* |
| 07:03.46 | grind | rgr bn-7bc |
| 07:04.00 | p3nguin | Good. Now watch sip debug while I call you. |
| 07:04.47 | TehRabbitt | Reliably Transmitting (NAT) to 198.65.166.131:5060: |
| 07:04.49 | TehRabbitt | is that you? |
| 07:05.09 | p3nguin | That's gizmo. |
| 07:05.22 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 07:05.26 | p3nguin | Did you enable NAT for gizmo? |
| 07:05.36 | TehRabbitt | Yes. |
| 07:05.39 | p3nguin | TURN IT OFF; they are not behind NAT! |
| 07:05.41 | TehRabbitt | ok... |
| 07:05.55 | p3nguin | Stop turning on NAT for things that are not behind NAT. |
| 07:06.15 | Jumpie | TehRabbitt p3nguin is hax0ring you |
| 07:06.20 | TehRabbitt | ok its off now |
| 07:06.50 | p3nguin | Make a call to your GV number while watching sip debug. |
| 07:07.06 | p3nguin | I try to call you via SIP URI, but I get nothing. |
| 07:07.24 | TehRabbitt | nothing |
| 07:08.07 | TehRabbitt | "your call cannot be completed as dialed" |
| 07:08.09 | TehRabbitt | nothing in SIP debug |
| 07:08.22 | p3nguin | Where do you hear that? |
| 07:08.43 | TehRabbitt | on my cell phone calling the 747 number |
| 07:08.44 | TehRabbitt | Reliably Transmitting (no NAT) to 198.65.166.131:5060: |
| 07:08.49 | p3nguin | lol |
| 07:08.49 | TehRabbitt | is all that came up |
| 07:09.14 | p3nguin | You know that the gizmo number is not a DID, right? |
| 07:09.30 | TehRabbitt | lmao (sorry tired again) haha |
| 07:09.32 | p3nguin | And that means you cannot call it from the PSTN. |
| 07:09.36 | TehRabbitt | *calls gv number* |
| 07:09.57 | TehRabbitt | wtf? |
| 07:10.08 | TehRabbitt | "your call is being answered by an automated voice mail system" |
| 07:10.22 | p3nguin | Check your gv settings. |
| 07:10.38 | p3nguin | Make sure you're sending calls where you think you're sending calls. |
| 07:10.52 | Jumpie | haha |
| 07:10.55 | Jumpie | i had that happen before |
| 07:10.59 | Jumpie | i accidentally called korea |
| 07:10.59 | TehRabbitt | that was definatally not my GV voicemail |
| 07:11.10 | TehRabbitt | that might have been gizmos though |
| 07:12.42 | TehRabbitt | it works 0_o |
| 07:12.48 | p3nguin | finally! |
| 07:12.48 | TehRabbitt | 2 way voice too.... hm |
| 07:12.50 | TehRabbitt | :-D |
| 07:12.57 | TehRabbitt | so inbound calls work now *phew* |
| 07:12.58 | TehRabbitt | lmao |
| 07:13.02 | TehRabbitt | now for outbound! jkjk |
| 07:13.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.204.199) |
| 07:13.53 | p3nguin | You called you gv number and the call went to gizmo, which went to your SIP URI? |
| 07:14.03 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 07:14.06 | TehRabbitt | Yep |
| 07:14.16 | p3nguin | And how does your SIP URI accept a call from gizmo but not from me? |
| 07:14.18 | TehRabbitt | my cisco phone started ringing really really loud lmao |
| 07:14.23 | TehRabbitt | try now 0_o |
| 07:14.32 | TehRabbitt | i changed a few things in the config |
| 07:14.49 | TehRabbitt | 17474945623@thoth.tenehawk.com |
| 07:14.49 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 07:15.31 | p3nguin | nothing |
| 07:15.46 | TehRabbitt | :( |
| 07:15.56 | *** join/#asterisk smooth_penguin (~smoove@59.96.228.68) |
| 07:16.08 | TehRabbitt | try calling 848-207-HAWK (4295) |
| 07:16.16 | TehRabbitt | i just wanna see if you can hear me lol |
| 07:16.22 | UQlev | can anyone recommend good tested provider for SIP/IAX termination? |
| 07:16.31 | p3nguin | What context does gizmo call into? |
| 07:16.39 | TehRabbitt | [users[ |
| 07:16.40 | TehRabbitt | err |
| 07:16.43 | TehRabbitt | [users] |
| 07:16.59 | TehRabbitt | for right nw |
| 07:17.00 | TehRabbitt | now |
| 07:17.02 | p3nguin | and there you have exten => 17474945623,1,blah ? |
| 07:18.14 | TehRabbitt | Yes |
| 07:18.31 | TehRabbitt | it's definatally working |
| 07:18.58 | p3nguin | That's really strange that I can't call you via SIP. |
| 07:19.27 | p3nguin | uqlev: VoIP.ms |
| 07:19.41 | TehRabbitt | try connecting an SIP phone to the asterisk box.... i made a test account |
| 07:19.52 | TehRabbitt | Hostname: thoth.tenehawk.com:5060 |
| 07:19.59 | TehRabbitt | Username: 300 |
| 07:20.04 | TehRabbitt | Secret: 300 |
| 07:20.09 | TehRabbitt | see if you can connect :-\ |
| 07:20.26 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 07:20.27 | TehRabbitt | if you can, dial extension 500 (my SCCP phone) |
| 07:20.54 | TehRabbitt | my friend from PA says he tries to connect, he can register, but when he calls me, he can hear me but I can't hear him |
| 07:20.59 | TehRabbitt | so idk if it's NAT or what |
| 07:21.25 | p3nguin | <PROTECTED> |
| 07:21.39 | TehRabbitt | :-\ |
| 07:21.40 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 07:21.56 | c0rnoTa | TehRabbitt: try to look for tshark's RTP caputre report |
| 07:22.17 | ChannelZ | seems firewalled or something |
| 07:22.21 | TehRabbitt | c0rnoTa: what do you mean? |
| 07:22.38 | TehRabbitt | p3nguin: http://pastebin.com/Z1fKqQz6 |
| 07:22.41 | ChannelZ | "Registration for '300@thoth.tenehawk.com' timed out" |
| 07:23.05 | TehRabbitt | shouldn't be timing out :-\ |
| 07:23.16 | ChannelZ | it's just dropping the packets |
| 07:23.21 | TehRabbitt | try now... |
| 07:23.57 | ChannelZ | nothing |
| 07:24.37 | TehRabbitt | :( |
| 07:24.59 | c0rnoTa | TehRabbitt: when call connection established, and you can hear your friend, start wireshark (tshark - console app) on server and look is there RTP flow exist to needed destination (external IP of your friend). If the flow exists that's mean that you have NAT problem. |
| 07:25.02 | p3nguin | 503 |
| 07:25.16 | TehRabbitt | try now... |
| 07:25.36 | c0rnoTa | TehRabbitt: otherwise it could be wrong destination ip or other things - idk |
| 07:25.37 | p3nguin | 503 |
| 07:25.44 | TehRabbitt | that's imposible 0_o |
| 07:25.56 | TehRabbitt | 71.59.82.2 |
| 07:25.58 | TehRabbitt | try using that IP |
| 07:26.02 | ChannelZ | still no response here |
| 07:26.03 | p3nguin | thoth.tenehawk.com has address 71.59.82.2 |
| 07:27.01 | ChannelZ | is this box directly on the net, or behind a NAT router or anything |
| 07:27.13 | ChannelZ | running iptables etc |
| 07:27.55 | TehRabbitt | ChannelZ: it is behind a Linksys 160N Router running DDWRT v24 |
| 07:27.59 | TehRabbitt | that's ALL there is |
| 07:28.09 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 07:28.18 | p3nguin | Your box denies me. |
| 07:29.11 | ChannelZ | Does your * have that actual IP or is it NAT behind the Linksys? |
| 07:29.23 | TehRabbitt | it has the actual IP |
| 07:29.39 | *** join/#asterisk AtLeT (~atletek@spletoknovsvet.spin.si) |
| 07:29.40 | p3nguin | uh, what? |
| 07:30.01 | TehRabbitt | http://pastebin.com/a3McHVN9 |
| 07:30.05 | TehRabbitt | look |
| 07:30.08 | ChannelZ | so this linksys is doing transparent bridging? |
| 07:30.24 | p3nguin | ifconfig |
| 07:31.18 | TehRabbitt | ChannelZ: no |
| 07:31.37 | TehRabbitt | the machine gets an internal LAN ip but i've specifeid the WAN Ip in the sip.conf file |
| 07:31.41 | ChannelZ | so then your * box must have a fake LAN IP. Your router has the real IP. |
| 07:32.12 | ChannelZ | You need to port-forward on the Linksys. It's bouncing all incoming SIP traffic because it has no idea what it's supposed to do with it. |
| 07:32.24 | TehRabbitt | it IS portfowarded |
| 07:32.37 | p3nguin | Did you turn on DMZ again? |
| 07:32.39 | TehRabbitt | ports 5060-5082 and ports 10000-20000 UDP and TCP |
| 07:32.39 | ChannelZ | Well it seems to be going nowhere. |
| 07:33.01 | p3nguin | You can dump the TCP, since this is all UDP. |
| 07:33.02 | TehRabbitt | DMZ is disabled p3nguin |
| 07:33.10 | TehRabbitt | If I dump TCP it stops workign idk why |
| 07:33.16 | TehRabbitt | (gizmo stops) |
| 07:33.31 | p3nguin | What do you do with ports 5061-5082? |
| 07:33.34 | Jumpie | gizmo uses some kinda tcp auth? |
| 07:33.41 | p3nguin | no |
| 07:33.50 | p3nguin | They send calls via SIP URI. |
| 07:33.56 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:34.05 | p3nguin | I don't have any TCP forwarded to my * box, and I can get gizmo calls just fine. |
| 07:34.13 | TehRabbitt | nvm it works with TCP disbaled |
| 07:35.03 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 07:35.17 | TehRabbitt | ports 5061-5082 was what a guide on NAT and SIP told me to open |
| 07:35.23 | TehRabbitt | along with ports 10000-20000 |
| 07:35.27 | p3nguin | That's silly. |
| 07:35.57 | p3nguin | Not that it's going to hurt anything, but silly nevertheless. |
| 07:36.05 | Dovid | 5060 is sip default |
| 07:36.13 | TehRabbitt | hm aight |
| 07:36.14 | p3nguin | You don't listen on those ports, so there wasn't any need to forward them. |
| 07:36.34 | TehRabbitt | I do listen for softphones to connect external to my LAN |
| 07:36.35 | TehRabbitt | no? |
| 07:36.50 | ChannelZ | You sure your LAN IP didn't change? Or that you have the right IP setup in your forwards? |
| 07:36.52 | p3nguin | If they are SIP phones, they should be connecting to you on 5060. |
| 07:37.06 | ChannelZ | and that you're not running iptables on the * box with a default of DROP or something? |
| 07:37.16 | TehRabbitt | No IPtables configured or running |
| 07:37.38 | p3nguin | Are you sure? iptables -L -nv |
| 07:37.40 | TehRabbitt | again using x-lite I can register from outside my LAN (over the net) but there is no audio |
| 07:38.17 | TehRabbitt | Positve |
| 07:38.21 | TehRabbitt | hain OUTPUT (policy ACCEPT 283K packets, 58M bytes) |
| 07:38.21 | TehRabbitt | <PROTECTED> |
| 07:38.27 | p3nguin | INPUT |
| 07:38.38 | TehRabbitt | Chain INPUT (policy ACCEPT 384K packets, 256M bytes) |
| 07:38.38 | TehRabbitt | <PROTECTED> |
| 07:39.00 | p3nguin | I don't understand why I can't register to you, but you claim another phone can. |
| 07:39.13 | TehRabbitt | p3nguin: are you on NAT? |
| 07:39.16 | p3nguin | And I can't send SIP calls to you, but you claim gizmo can. |
| 07:39.19 | p3nguin | yes |
| 07:39.25 | TehRabbitt | perhaps that's why? |
| 07:39.27 | ChannelZ | I tried an 'anonymous' call (not registering at all) and get zilch.. no rejection, just nothing. |
| 07:39.36 | TehRabbitt | call my 848-207-4295 |
| 07:39.39 | p3nguin | exactly, dead air. |
| 07:39.46 | TehRabbitt | call it as a regular # |
| 07:39.51 | TehRabbitt | should ring my cisco phone |
| 07:40.12 | TehRabbitt | then i'll put you on hold to confirm MoH works |
| 07:41.09 | p3nguin | dead air |
| 07:41.16 | TehRabbitt | aight |
| 07:41.17 | TehRabbitt | hm |
| 07:41.23 | TehRabbitt | apparently MoH isn't working again haha |
| 07:41.45 | p3nguin | That's your GV number? |
| 07:42.11 | p3nguin | Did you configure any ACLs when you did port forwarding? |
| 07:42.34 | TehRabbitt | thats the GV number yes |
| 07:42.43 | TehRabbitt | the 848 is the gv number |
| 07:43.14 | p3nguin | It is so very inconsistent. |
| 07:44.19 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 07:45.13 | TehRabbitt | :( |
| 07:45.38 | TehRabbitt | [May 4 03:45:31] WARNING[31862]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/drwho': No such file or directory |
| 07:45.43 | TehRabbitt | that's the on hold music... |
| 07:45.46 | TehRabbitt | drwho.mp3 |
| 07:46.00 | TehRabbitt | it's not the dr who you're thinking it's some techno song lmao |
| 07:46.02 | p3nguin | I guess that's why I cuoldn't here it. No such file or directory |
| 07:46.08 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:46.09 | p3nguin | hear |
| 07:46.14 | TehRabbitt | but it exists |
| 07:46.24 | TehRabbitt | -rwxr--r-- 1 root root 6857687 2010-05-04 03:45 drwho.mp3 |
| 07:46.53 | ChannelZ | did you install add-ons for mp3 support? |
| 07:47.10 | p3nguin | and is musiconhold.conf configured for mp3 playback? |
| 07:47.15 | TehRabbitt | [May 4 03:47:42] WARNING[31881]: file.c:664 ast_openstream_full: File /var/lib/asterisk/moh/drwho does not exist in any format |
| 07:47.18 | TehRabbitt | Ah that could be it |
| 07:47.53 | TehRabbitt | how do I do that haha |
| 07:48.10 | p3nguin | Go back to asterisk source and run make menuselect again. |
| 07:48.19 | TehRabbitt | I enabled mp3 though... |
| 07:48.20 | TehRabbitt | hmph |
| 07:48.24 | p3nguin | oh |
| 07:48.32 | p3nguin | check the .conf |
| 07:49.29 | TehRabbitt | [May 4 03:50:06] NOTICE[31901]: res_musiconhold.c:556 monmp3thread: Request to schedule in the past?!?! |
| 07:49.51 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:50.11 | TehRabbitt | and it just keeps saying it |
| 07:50.30 | ChannelZ | are you actually specifying the file in musiconhold? depending on what you've got going you generally tell it the directory to find files in, and it plays things in that dir |
| 07:50.36 | ChannelZ | you don't tell it a specific audio file |
| 07:51.48 | TehRabbitt | I didn't lol |
| 07:56.02 | TehRabbitt | hmph still not working |
| 07:56.45 | *** join/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 07:56.48 | hurdman | hi |
| 07:57.09 | hurdman | i'm trying to have a personnal ton during the ringing time |
| 07:57.13 | TehRabbitt | btw MP3 isn't one of the MoH packages |
| 07:57.23 | hurdman | i have use Dial(**** ,, m(blabla) ) |
| 07:57.40 | hurdman | en configure musiconhold.conf |
| 07:57.46 | hurdman | restart asterisk |
| 07:57.59 | hurdman | i can see : Started music on hold, class 'mickaelandyg', on DAHDI/27-1 |
| 07:58.13 | hurdman | but i here nothing |
| 07:58.15 | hurdman | any idea |
| 07:58.31 | hurdman | i have tried ulaw, alaw and gsm file format |
| 07:59.14 | p3nguin | tehrabbitt: /usr/lib/asterisk/modules/format_mp3.so |
| 08:00.27 | TehRabbitt | [May 4 04:00:40] WARNING[32141]: format_wav_gsm.c:142 check_header: Unexpected header size 40 |
| 08:00.27 | TehRabbitt | [May 4 04:00:40] WARNING[32141]: file.c:385 fn_wrapper: Unable to open format wav49 |
| 08:00.27 | TehRabbitt | [May 4 04:00:40] WARNING[32141]: format_wav_gsm.c:142 check_header: Unexpected header size 40 |
| 08:00.27 | TehRabbitt | [May 4 04:00:40] WARNING[32141]: file.c:385 fn_wrapper: Unable to open format wav49 |
| 08:00.27 | TehRabbitt | [May 4 04:00:40] WARNING[32141]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/Drwho': No such file or directory |
| 08:01.20 | p3nguin | Looks to me like you haven't specified that you'll be playing an mp3 file. |
| 08:02.15 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:03.01 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 08:03.11 | p3nguin | Time to go. |
| 08:03.15 | TehRabbitt | noooo lol |
| 08:03.19 | TehRabbitt | it is late :( |
| 08:03.23 | p3nguin | 03.00 |
| 08:03.28 | TehRabbitt | that's trying to play a WAV file btw |
| 08:03.39 | TehRabbitt | i have it both in WAV and MP3 |
| 08:07.40 | ChannelZ | wav should be 8 or 16-bit, 8kHz |
| 08:07.58 | hurdman | ( and mono ) |
| 08:08.35 | ChannelZ | that too |
| 08:12.55 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 08:14.18 | hurdman | ( i have found, i have forgotten the answer ) |
| 08:14.22 | hurdman | <PROTECTED> |
| 08:14.50 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-236.canterbury.ac.uk) |
| 08:14.57 | AtLeT | hi |
| 08:14.57 | AtLeT | I have problems with one way audio. When I'm in call, sometimes happen (call length vary), |
| 08:14.57 | AtLeT | that I can't hear the client in the middle of the call, but the client can hear me. |
| 08:14.57 | AtLeT | One rtp stream get "disconected". I can't find anything on google. I also used wireshark |
| 08:14.57 | AtLeT | and sip debug, rtp debug, but there is nothing strange. Any idea, what can be wrong or |
| 08:14.58 | AtLeT | where can I look for the problem? |
| 08:16.13 | TehRabbitt | i'm goin to bed... night |
| 08:17.33 | *** join/#asterisk smooth_penguin (~smoove@59.95.32.157) |
| 08:19.32 | ChannelZ | AtLeT: is your * box behind a firewall of any kind? |
| 08:19.43 | AtLeT | I have 2 * |
| 08:19.50 | AtLeT | one is 1.6 and is behind nat |
| 08:19.58 | AtLeT | another is 1.4 and isn't behind nat |
| 08:20.07 | AtLeT | I have on the both the same isue |
| 08:20.27 | AtLeT | on both I have iptables |
| 08:20.38 | AtLeT | and allow only sip and rtp |
| 08:21.14 | AtLeT | but this happen sometime in middle of the call |
| 08:21.23 | AtLeT | http://www.pingvincek.com/img/Graf.PNG |
| 08:21.32 | AtLeT | http://www.pingvincek.com/img/Call.PNG |
| 08:22.02 | AtLeT | as you can see on the second picture, tne one rtp stream finished 10 seconds |
| 08:22.13 | AtLeT | before I hang up |
| 08:22.37 | ChannelZ | Is * in the media stream of the call (canreinvite=no) or are the two end points talking directly to each other? Are you sure it's not your phone? |
| 08:22.55 | AtLeT | I have canreinvite=no |
| 08:23.09 | AtLeT | this happen on phone (grandstream gxp2000) and also on x-lite |
| 08:23.43 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 08:28.24 | ChannelZ | hmm not sure. I've not had that/heard of that before, that'd be a hard one to debug |
| 08:28.41 | ChannelZ | Does the audio get choppy or strange before it stops? |
| 08:30.03 | AtLeT | no |
| 08:30.07 | AtLeT | the audio is fine |
| 08:30.33 | AtLeT | at the some point it get "cut" |
| 08:30.41 | AtLeT | I can't hear nothing |
| 08:30.47 | AtLeT | but client can ear me |
| 08:30.50 | AtLeT | hear |
| 08:33.05 | smooth_penguin | AtLeT, what are you using for those PNGs? |
| 08:33.44 | AtLeT | wireshark |
| 08:34.07 | smooth_penguin | oh thanks |
| 08:34.15 | AtLeT | np |
| 08:35.47 | smooth_penguin | <PROTECTED> |
| 08:36.24 | AtLeT | ok, I will |
| 08:36.36 | smooth_penguin | any QoS routers etc? |
| 08:36.44 | AtLeT | no |
| 08:42.52 | AtLeT | it is possible because of ping 400ms |
| 08:43.00 | AtLeT | sometime happen, tat ping is realy big |
| 08:43.21 | AtLeT | but most of time ping is 20ms |
| 08:43.38 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
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| 08:54.20 | xheliox | why so angry? :( |
| 08:55.58 | Preatorian | O.o |
| 08:59.09 | xheliox | o.O |
| 08:59.11 | xheliox | what? |
| 08:59.30 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
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| 09:33.53 | Dovid | anyone here have IP services from TW Telecom ? |
| 09:35.28 | pasqu | hi, I need help with t38 fax in Asterisk (always T1_TIMEOUT). Someone have a working setup to compare? |
| 09:36.10 | Gido-E | pasqu i am using agx_fax, works phine |
| 09:37.15 | pasqu | never hear about it |
| 09:38.02 | imcdona | morning all. This isn't an asterisk issue but I figure someone may have an answer: I've got a strange problem. Whenever I try and configure an inbound route with certain numbers, freePBX generates the dword equivelent. For example, an inbound route for the number "2535551212" is written as "-1759416084" in extensions_additional.conf. Any Idea's? |
| 09:38.48 | *** join/#asterisk Z_God (~julius@wlan226046.mobiel.utwente.nl) |
| 09:39.06 | pasqu | Gido-E, have t38 support agx_fax? |
| 09:41.28 | Gido-E | pasqu i can recieve faxes and send. And i thought so... :-) |
| 09:44.14 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:49.27 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 09:49.35 | elliot98 | greetings! |
| 09:49.53 | elliot98 | I am testing out some T38 capabilities |
| 09:50.25 | elliot98 | how/where do I check the logs/cli to ensure that the channel is in fact sending the call with T38 fax? |
| 09:56.13 | pasqu | elliot98, I see: [May 4 10:52:55] NOTICE[1345]: res_fax.c:1083 receivefax_t38_init: Negotiating T.38 for receive on SIP/192.168.99.129-b91772e8 |
| 09:56.13 | pasqu | [May 4 10:52:55] NOTICE[1345]: res_fax.c:1125 receivefax_t38_init: Negotiated T.38 for receive on SIP/192.168.99.129-b91772e8 |
| 10:03.38 | Jumpie | my faxing seems to workk but for some reason res_fax_digium.conf takes like..8 minutes to load on bootup |
| 10:03.53 | Jumpie | i cant really find any documentation as to why its doing that and my network si fine all other services satart fine |
| 10:05.46 | *** join/#asterisk Akiraaa (~Akira@92.81.197.101) |
| 10:08.39 | elliot98 | what is res_fax_digium.conf? |
| 10:10.40 | elliot98 | I don't recall coming across any res_fax when installing the digium fax driver |
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| 10:43.53 | elliot98 | hmm....there is no such NOTICE in my logs |
| 10:44.09 | elliot98 | res_fax.c:1083 receivefax_t38_init: Negotiating T.38 for receive on SIP/192.168.99.129-b91772e8 |
| 10:46.22 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 10:56.55 | plundra | So, I'm trying to get those channelvariables set when bridging the call from the queue, with a queue member. |
| 10:57.13 | plundra | Yesterday I figured it wasn't sufficient to just queue reload all, but I did a full restart this morning. |
| 10:57.30 | plundra | And still can't see any variables, what might I be doing wrong? :) |
| 10:59.06 | Gido-E | you sould have stayed in bed. |
| 10:59.13 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 10:59.17 | plundra | I've checked all channels involved, but can't find any of the MEMBER*, QE* and QUEUE* variables. Are they just set when doing a gosub or macro? |
| 10:59.36 | plundra | Gido-E: Tell me about it, snoozed seven times this morning... |
| 11:00.20 | *** join/#asterisk kartik (~koolkarti@117.199.121.144) |
| 11:01.14 | Gido-E | plundra ok, than you are maxed out! |
| 11:09.25 | elliot98 | when I run "fax show stats" in the CLI for the digium fax device, it shows: Capabilities : SEND RECEIVE G.711 |
| 11:09.31 | elliot98 | why isn't T38 there? |
| 11:11.51 | Gido-E | elliot98 that is logic |
| 11:12.09 | *** join/#asterisk sp4rc (~sp4rc@178-83-239-81.dclient.hispeed.ch) |
| 11:12.32 | elliot98 | isn't the Digium driver also for T38? |
| 11:13.13 | sp4rc | guys, i am new to asterisk and would like to know if i can e.g. set an asterisk server which runs at my home and offers two accounts, one for sipgate and one for switzernet |
| 11:13.44 | sp4rc | then i would like to use ekiga to make calls over a openvpn tunnel to this server which should forward them to sipgate or switzernet |
| 11:14.04 | Gido-E | sp4rc you know the first second timing problem with ekiga? |
| 11:14.42 | sp4rc | Gido-E: what do you mean by "timing problem"? are you talking about the latency which occurs over the vpn link= |
| 11:15.19 | Gido-E | no, ekiga, does not let hear the first second of a conversation. |
| 11:15.23 | Gido-E | about one second. |
| 11:16.05 | elliot98 | Gido-E: why doesn |
| 11:16.07 | elliot98 | woops |
| 11:16.27 | elliot98 | Gido-E: why doesn't the Digium driver show T38 capabitilies? |
| 11:17.09 | sp4rc | Gido-E: okay so this is a specific ekiga problem |
| 11:18.29 | sp4rc | Gido-E: but let's say i would like to use any other sip softphone |
| 11:18.58 | sp4rc | would it be possible to route to two sip providers? |
| 11:19.01 | kaldemar | sp4rc: yes, it is possible. |
| 11:19.14 | sp4rc | i am not aware of all the specific pbx words |
| 11:19.24 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:20.22 | sp4rc | kaldemar: at the moment i have two different sip accounts on two different providers which i would like to bring together and make accessible via vpn |
| 11:21.40 | Gido-E | sp4rc that is possible. But at the end is everything possible :-) |
| 11:23.03 | sp4rc | hm, what about freeswitch |
| 11:24.34 | sp4rc | the thing is, i am running pfsense on my firewall, and there is a package for freeswitch |
| 11:24.54 | sp4rc | that would make things easier, and there would be no need to start up my computer to place calls |
| 11:26.42 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-sjsqiwonyqqhgvou) |
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| 11:27.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:28.49 | kaldemar | sp4rc: or you could just set up both accounts in your soft phone. |
| 11:32.04 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
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| 11:37.51 | plundra | Ok, so I did try calling a macro when connecting caller/queuemember, in which I did a DumpChan. But, no QE*, QUEUE*, MEMBER* variables there either. Surely I'm missing something very vital :-) |
| 11:37.56 | sp4rc | kaldemar: yes thats what i am practising right now, but if i e.g. want to place calls from my office, where the firewall restricts RTP traffic, then i would like to route this through my vpn tunnel |
| 11:44.39 | joobie | burp |
| 11:46.17 | *** join/#asterisk suprstar (~suprstar@216.56.88.4) |
| 11:46.34 | elliot98 | does the digium driver for asterisk 1.4 have t38 capabilities? |
| 11:55.44 | Gido-E | elliot98 you mean the closed source driver? |
| 11:56.01 | elliot98 | Gido-E: yes |
| 11:56.12 | elliot98 | the closed source one from Digium.com |
| 11:57.52 | Gido-E | elliot98 than it sould al be nice documented. watch there. |
| 11:57.57 | Gido-E | look there :-) |
| 11:58.02 | Gido-E | i am not native english. |
| 11:58.39 | *** join/#asterisk chasecrum (~chasecrum@adsl-065-082-196-004.sip.asm.bellsouth.net) |
| 11:59.11 | chasecrum | Ive just done a fresh install of asterisk now and I'm having a pretty simple problem . can anyone here help me out for a sec ? |
| 11:59.49 | beek | ~ask |
| 11:59.49 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:00.19 | Gido-E | chasecrum it is verry annoying if you ask your questions that way. |
| 12:00.26 | chasecrum | the asterisk manual I'm studying gives the first command as a dial hello world, console dial 1001, but when I put that in I get No such command console dial |
| 12:00.42 | chasecrum | (appologies for the anoyance) |
| 12:00.58 | Gido-E | chasecrum if you never do it again. It is forgiven. |
| 12:01.04 | chasecrum | :) |
| 12:01.37 | Gido-E | chasecrum |
| 12:01.39 | beek | chasecrum: Which asterisk manual are you looking at? The "book"? |
| 12:01.53 | Gido-E | core show application dial (works?) |
| 12:01.55 | chasecrum | Practical Asterisk 1.4 and 1.6 |
| 12:02.06 | chasecrum | dunno, lemme check |
| 12:02.37 | Gido-E | probably you are on the CLI, ant dial is part of the dial plan. You sould read the book more and more, til you understand asterisk principals. |
| 12:02.45 | chasecrum | I put that in and got what looks like a man page |
| 12:02.54 | beek | chasecrum: It's supposed to. |
| 12:03.16 | beek | chasecrum: which version of asterisk are you running? |
| 12:03.17 | Gido-E | chasecrum yep, than you are on the CLI, and you have no idea what you are doing :-) |
| 12:03.20 | chasecrum | ok. I'll be the first to state the obvious, I have no idea what I'm doing. |
| 12:03.43 | chasecrum | I just got the current iso for asterisk now |
| 12:03.46 | Gido-E | ok, you sould first play/test/read/learn more about asterisk. |
| 12:04.01 | chasecrum | (this is how I'm trying to learn, a book, and a image) |
| 12:04.10 | beek | chasecrum: The standard for Asterisk manuals is available online. |
| 12:04.13 | beek | ~thebook |
| 12:04.14 | infobot | it has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 12:04.38 | beek | chasecrum: Do yourself a BIG, BIG favor and get that book and read it from the beginning. |
| 12:04.48 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 12:04.49 | beek | It will make your adventure into Asterisk land so much smoother. |
| 12:04.51 | chasecrum | ok, pulling down the pdf now |
| 12:04.54 | carrar | all the way to the end |
| 12:05.22 | chasecrum | the plan WAS to install, and start at step one in this book, but i failed at step one... |
| 12:06.06 | beek | chasecrum: I'm not familiar with the book you have but I do know the quality of "the book." We don't refer to it as "the book" in this channel for nothing! ;-) |
| 12:06.47 | chasecrum | Well that's a great start ! and I appreciate the pointer. Would you install asterisk over a distro, or use the asterisknow iso built on cent ? |
| 12:07.20 | beek | chasecrum: Do yourself a marvelous favor and forget pre-configured versions. |
| 12:07.33 | beek | Just start with a distro (I use CentOS), download the source and compile it yourself. |
| 12:07.46 | beek | Then you'll know exactly what it has been configured for. |
| 12:08.13 | chasecrum | sure. sounds reasonable. I take it the instructions for that are in THE book ? |
| 12:08.17 | beek | When I started a few years ago I used Trixbox to "get me going." I spent more time figuring out what they did than I would have spent just learning from the ground up. |
| 12:08.26 | beek | Yes |
| 12:08.42 | chasecrum | (I got the asterisknow iso from digium, thought that would be the way to go...) |
| 12:09.06 | beek | Nothing beats having access to any number of versions, depending on what you want to do. |
| 12:09.47 | chasecrum | Sweet God that's a huge book..... |
| 12:09.55 | Gido-E | chasecrum yep :-) |
| 12:09.59 | beek | chasecrum: What you want to do is avoid a GUI at all costs. Learn to write 'em. |
| 12:10.19 | beek | chasecrum: it's not as big as it appears... the last 1/2-1/3 is reference material. |
| 12:10.30 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:10.35 | chasecrum | ah... |
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| 12:10.43 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:10.44 | chasecrum | (company printer.....) |
| 12:11.09 | beek | chasecrum: It's available in dead tree form from your favorite bookseller, too. |
| 12:11.11 | chasecrum | does it make any noticeable difference which distro I use ? |
| 12:11.14 | *** join/#asterisk TimeRider (steve@5ac7b3c5.bb.sky.com) |
| 12:11.18 | chasecrum | dead tree form.... ha! |
| 12:11.30 | beek | chasecrum: The general consensus is one of two: Debian or CentOS. |
| 12:11.40 | beek | I use the latter. |
| 12:12.04 | chasecrum | we use centos on our servers here, but ubuntu on our desktops... |
| 12:12.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:12.08 | suprstar | i use centos for EVERYTHING, not just * boxes |
| 12:12.18 | beek | Me too. |
| 12:12.27 | beek | chasecrum: Asterisk is a server... |
| 12:12.31 | chasecrum | yes |
| 12:12.37 | Gido-E | deamon |
| 12:12.42 | plundra | Uhm, so, shouldn't the stuff set in [general] in queues.conf apply to all queues? Unless it's overridden of course? |
| 12:13.05 | plundra | Or are you supposed to inherit [general] as a template? :) (This sounds weird...) |
| 12:13.05 | [TK]D-Fender | plundra: For things you ACN specify there, sure |
| 12:13.07 | [TK]D-Fender | CAN |
| 12:13.24 | beek | Mornin' [TK]D-Fender |
| 12:13.29 | plundra | [TK]D-Fender: I assume you are supposed to be able to set the stuff which are under there in the example config. |
| 12:13.49 | [TK]D-Fender | plundaGot something to show us? |
| 12:13.49 | plundra | [TK]D-Fender: The set...var=yes in this case. |
| 12:14.18 | plundra | Either way, I had to put it under the queue, not just general. |
| 12:15.02 | beek | chasecrum: One last thing... if you choose to load it manually you can ask questions in this channel. If you choose a GUI-based solution you'll be directed elsewhere. |
| 12:16.00 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 12:17.10 | chasecrum | yeah, that seems to be the consensus everywhere... |
| 12:17.33 | plundra | I do use templating, but can't see how this should affect anything. I'll try creating a queue without template and see if the stuff under [general] is applied. |
| 12:18.28 | [TK]D-Fender | plundra: You can't put everything in [general]. Just like other confs there are things you can, and cannot put there. |
| 12:19.21 | chasecrum | i get that learning the cli means knowing the software, we have that attitude here about our linux servers, but really, why does everyone seem to be anti-gui ? |
| 12:19.39 | leifmadsen | sip:polycom@leifmadsen.com |
| 12:19.41 | plundra | [TK]D-Fender: YEah I do realize that. |
| 12:19.46 | leifmadsen | If you love polycom :) |
| 12:19.51 | leifmadsen | ~polycom |
| 12:19.52 | infobot | [polycom] the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 12:19.55 | Gido-E | chasecrum not anti gui, but don't ask gui questions here. |
| 12:20.03 | beek | chasecrum: Because the GUI limits your options. Writing it by hand gives you the maximum flexibility. |
| 12:20.21 | beek | chasecrum: Here's an example: |
| 12:20.24 | beek | ~freepbx |
| 12:20.25 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:20.27 | [TK]D-Fender | chasecrum: * gives you control over how you get to process calls. You can do almost anything. Once you run a GUI you are stuck with a toaster-builder and a bloated pile of shit as one at that |
| 12:20.28 | leifmadsen | infobot: polycom is also The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460 |
| 12:20.29 | infobot | okay, leifmadsen |
| 12:20.37 | chasecrum | I'd just as soon learn it from the cli anyway, seems you force yourself into knowing the software that way... |
| 12:21.16 | plundra | [TK]D-Fender: Ah, ehm, so I didn't really notice it was in the [markq]-example block. And not [general] |
| 12:21.44 | [TK]D-Fender | plundra: You may now plunge that hot poker into your left eye-socket :p |
| 12:22.05 | plundra | [TK]D-Fender: Certainly! :-P |
| 12:22.24 | beek | chasecrum: Like I said... I had a working PBX with Trixbox (CentOS + Freepbx + SugarCRM) and it worked fine. Up until I wanted to do something that the GUI wouldn't allow. Then I had to figure out Trix's method to allow manual configuration. It would have been easier just to go straight to "Asterisk from Source." I changed and haven't looked back. |
| 12:22.25 | plundra | Thanks anyway :-) |
| 12:23.15 | chasecrum | thats good to know just starting out, and I appreciate that. |
| 12:25.00 | beek | chasecrum: Bottom line is that the learning curve is a bit steeper doing it this way but the long term payoff is well worth it. "The Book" is a great introduction. |
| 12:25.35 | chasecrum | well, since it's my job we're talking about, I'll call that good advice... |
| 12:26.33 | beek | Once you have specific questions this channel is a great resource. |
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| 12:37.48 | leifmadsen | ~isn |
| 12:37.57 | leifmadsen | ~itad |
| 12:38.07 | leifmadsen | wow really? huh well I better fill those in |
| 12:42.17 | *** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
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| 12:42.27 | leifmadsen | infobot: itad is an IP Telephony Administrative Domain. Similar in nature to an AS (Autonomous System) number, it is administered by IANA (Internet Assigned Numbers Authority). An ITAD number is part of the TRIP specification in RFC 3219. Although TRIP never took off, ITAD numbers are being used by Freenum.org as part of an ISN (ITAD Subscriber Number). For more information about ISN numbers, see 'isn'. |
| 12:42.28 | infobot | leifmadsen: okay |
| 12:44.57 | *** join/#asterisk coppice (~chatzilla@191.193.17.210.dyn.pacific.net.hk) |
| 12:45.48 | leifmadsen | infobot: isn is ITAD Subscriber Number. (For more information about what an ITAD number is, see 'itad'.) An ISN is a method of dialing SIP URI's via a standard keypad on a telephony. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers however simplifies this by utilizing DNS lookups to map the ISN number to a domain, and to dial that end point w |
| 12:45.49 | infobot | leifmadsen: okay |
| 12:45.49 | leifmadsen | ith the returned data. An ISN has the format of <resource>*<location> where the <resource> is some number (like an extension number) and the <location> is an ITAD number. See http://www.freenum.org for more information. |
| 12:45.57 | leifmadsen | hmmm... I have a feeling that got cut off |
| 12:45.59 | *** join/#asterisk smooth_penguin (~smoove@59.95.47.187) |
| 12:46.04 | fenrus | looks like that |
| 12:46.06 | leifmadsen | infobot: tell leifmadsen about isn |
| 12:46.16 | leifmadsen | stupid xchat |
| 12:46.18 | leifmadsen | will fix in another window |
| 12:51.30 | leifmadsen | there |
| 12:51.36 | leifmadsen | had to shorten it quite a bit :( |
| 12:51.38 | leifmadsen | ~itad |
| 12:51.39 | infobot | methinks itad is an IP Telephony Administrative Domain. Similar in nature to an AS (Autonomous System) number, it is administered by IANA (Internet Assigned Numbers Authority). An ITAD number is part of the TRIP specification in RFC 3219. Although TRIP never took off, ITAD numbers are being used by Freenum.org as part of an ISN (ITAD Subscriber Number). For more information about ISN numbers, see 'isn'. |
| 12:51.42 | leifmadsen | ~isn |
| 12:51.43 | infobot | i heard isn is ITAD Subscriber Number (see 'itad'). An ISN is a method of dialing SIP URI's via a standard keypad on a telephone. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers simplifies this by utilizing DNS lookups to map the ISN number to a domain. See http://www.freenum.org for more information. |
| 12:51.51 | leifmadsen | nice. |
| 12:51.59 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
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| 13:08.34 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 13:15.53 | elliot98 | I am getting a refused to negotiate T.38" error when faxing |
| 13:16.22 | elliot98 | the two endpoints are using the Digium driver, which has T38 capabilties |
| 13:16.41 | elliot98 | the middlepoint is an asterisk 1.4 with t38 passthrough enabled |
| 13:16.50 | elliot98 | what does "refused to negotiate T.38" mean? |
| 13:20.48 | Kobaz | that T.38 failed to work |
| 13:21.03 | Kobaz | misconfiguration, or the protocols dont match on both ends |
| 13:21.15 | Kobaz | or one side has a broken implementation of T.38 |
| 13:21.22 | smooth_penguin | Hey can the dcap exam be given without taking the training |
| 13:21.46 | smooth_penguin | like what sort of knowledge/experience level is needed |
| 13:25.09 | *** join/#asterisk bmg505 (~leon@196-209-71-68-rndf-esr-3.dynamic.isadsl.co.za) |
| 13:26.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:27.44 | chasecrum | just finished printing "the book." everyone here hates me now. |
| 13:28.02 | elliot98 | it's actually using the same digium driver (the same server looping) |
| 13:28.32 | elliot98 | Kobaz: except there is another server in the middle |
| 13:28.40 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:29.23 | beek | chasecrum: It's compelling reading. Great plot. Well-developed characters. A real page turner. |
| 13:29.57 | chasecrum | and apparently the end to the war and peace series... |
| 13:30.07 | coppice | ah, but you never really find who did it |
| 13:30.12 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:32.23 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:32.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:32.24 | elliot98 | however, if t38 is not working, why can't it fallback on g711?? |
| 13:33.17 | jaytee | spoiler alert!: great book but Dumbledore dies on page 573 |
| 13:33.53 | Baylink | Heh |
| 13:34.18 | chasecrum | wow. WHATS THE POINT NOW ? |
| 13:36.06 | *** join/#asterisk seelen (~seelen@190.145.97.138) |
| 13:36.46 | smooth_penguin | what book |
| 13:37.01 | pabelanger | ~book |
| 13:37.02 | 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 |
| 13:37.11 | smooth_penguin | eh |
| 13:37.15 | smooth_penguin | there is no Dumbledore |
| 13:37.19 | smooth_penguin | in that book |
| 13:37.37 | jaytee | ~humor |
| 13:37.38 | infobot | methinks humor is Q: Why are the streets of Paris lined with trees? A: Because Germans like to march in the shade. |
| 13:37.45 | jaytee | hahaha |
| 13:38.17 | beek | morning jaytee |
| 13:38.26 | jaytee | morning beek |
| 13:39.22 | beek | jaytee: Did you get what you needed from my emails? |
| 13:39.30 | jaytee | beek, yes thanks |
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| 14:07.00 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:10.23 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 14:14.49 | pabelanger | leifmadsen: ping |
| 14:16.34 | leifmadsen | pabelanger: pong |
| 14:17.22 | pabelanger | m17266 |
| 14:17.24 | MuffinMan | [ready for testing] [Asterisk] Channels/chan_sip/General 0017266: [patch] Failed to register peers from realtime config reported by Nick_Lewis https://issues.asterisk.org/view.php?id=17266 |
| 14:17.47 | pabelanger | looks like we can move to ready for review and possible merge. Seems like a straight forward patch. |
| 14:18.39 | russellb | yes, looks fine |
| 14:18.42 | russellb | needs some spaces after "if" |
| 14:19.02 | *** join/#asterisk wnunes (~under_lin@200-233-56-25.corp.ajato.com.br) |
| 14:19.17 | russellb | pabelanger: note added, feel free to merge |
| 14:19.43 | pabelanger | russellb: will do. Thanks. |
| 14:19.53 | russellb | wrong channel btw :-) |
| 14:20.59 | pabelanger | It was related to a bug after all ;). I'll be sure to post it into -dev next time. |
| 14:21.05 | russellb | we're not in bugs |
| 14:21.13 | pabelanger | hah |
| 14:21.16 | russellb | :-p |
| 14:21.19 | pabelanger | time for coffee then |
| 14:21.22 | russellb | yay coffee |
| 14:21.28 | russellb | i graduated to black coffee today |
| 14:22.04 | pabelanger | yar! I rolled back to it last week. Started using sugar, was giving me too many headacks. |
| 14:23.45 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:23.57 | coppice | why does coffee have to be so black and white. what about some shades of grey? |
| 14:24.14 | *** join/#asterisk snapple42 (~snapple42@h216-18-80-131.gtconnect.net) |
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| 14:30.27 | leifmadsen | I don't like the idea of gray coffee |
| 14:30.31 | *** join/#asterisk darkskiez_ (~dz@62-50-207-121.client.stsn.net) |
| 14:31.08 | coppice | I think you might find white coffee a little off putting too |
| 14:32.25 | Naikrovek | finds the idea of coffee offputting |
| 14:34.13 | smooth_penguin | hello Naikrovek |
| 14:34.21 | Naikrovek | greetings |
| 14:34.35 | smooth_penguin | hows stuff with the Indian folks |
| 14:38.51 | Naikrovek | they suck |
| 14:38.53 | Naikrovek | heh |
| 14:39.08 | Naikrovek | not indians in general, just my guys seem to not want to learn anything new |
| 14:44.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:44.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:51.22 | *** join/#asterisk ThoMe (tm@tm.muc.de) |
| 14:51.24 | ThoMe | hiho |
| 14:51.31 | ThoMe | is app_devstate not available for asterisk 1.6 ? |
| 14:51.48 | [TK]D-Fender | thontried looking for a FUNCTION like that? |
| 14:51.58 | [TK]D-Fender | ThoMe: tried looking for a FUNCTION like that? |
| 14:52.17 | ThoMe | [TK]D-Fender: i need the appilication |
| 14:52.27 | ThoMe | i need this for the asterisk console |
| 14:52.47 | ThoMe | is it available as application? |
| 14:52.56 | leifmadsen | no |
| 14:53.08 | ThoMe | leifmadsen: and a alternate? |
| 14:53.10 | leifmadsen | Exec(DEVSTATE(...) ) |
| 14:53.18 | leifmadsen | or somethign like that |
| 14:53.25 | ThoMe | leifmadsen: oh, realy? |
| 14:53.29 | leifmadsen | possibly |
| 14:53.39 | ThoMe | leifmadsen: i can run EXEC on the console and then a functino? |
| 14:53.41 | leifmadsen | you may have to be creative -- I'm not sure what you're trying to do |
| 14:53.57 | leifmadsen | app_devstate does not exist though -- it is func_devstate |
| 14:55.58 | ThoMe | hmm ok ok |
| 14:57.58 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.25 | *** join/#asterisk Z_God (~julius@wlan226046.mobiel.utwente.nl) |
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| 15:08.43 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 15:08.58 | kruemeltee | hello all together ... |
| 15:10.38 | 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? |
| 15:11.17 | *** join/#asterisk hfb (~hfb@pool-98-112-146-69.lsanca.dsl-w.verizon.net) |
| 15:11.18 | kruemeltee | better to ask, where is the Request-URI set? Within the definition of the peer within sip.conf? |
| 15:11.29 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 15:11.56 | kruemeltee | something like "fromuser" ?!? |
| 15:17.51 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 15:22.53 | kruemeltee | ha! I got it ... fromuser is the right thing! |
| 15:28.50 | ThoMe | hmmm |
| 15:28.57 | ThoMe | leifmadsen: my asterisk said |
| 15:28.58 | ThoMe | [May 4 17:27:39] ERROR[18974]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe |
| 15:29.04 | ThoMe | what is ast_carefulwrite ? |
| 15:29.20 | leifmadsen | it's a function in asterisk source code |
| 15:29.30 | ThoMe | and why error? |
| 15:29.36 | leifmadsen | that command generally means audio could not be sent to a channel because it didn't exist |
| 15:29.43 | leifmadsen | s/command/error/ |
| 15:29.56 | leifmadsen | I get it all the time when I hang up while a prompt is playing |
| 15:35.20 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 15:35.23 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:37.47 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 15:38.12 | idespinner | any idea what this message means: WARNING[5009] chan_dahdi.c: Got restart ack on channel 0/1 span 1 with owner |
| 15:38.47 | idespinner | 4 port pri card running a recent asterisk 1.4 |
| 15:43.36 | idespinner | or in other words, what is the meaning in chan_dahdi.c if our PRI gets the message PRI_EVENT_RESTART_ACK? |
| 15:44.22 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:44.36 | Corydon76-dig | idespinner: it means the channel was successfully restarted |
| 15:45.27 | idespinner | Corydon76-dig, TY, but I wasnt aware individual channels could be restarted on a PRI... |
| 15:45.38 | Corydon76-dig | Yes |
| 15:46.12 | idespinner | is there anything you know of that would cause asterisk to restart individual channels? |
| 15:46.35 | Corydon76-dig | Yes, it automatically restarts every idle channel once an hour, by default |
| 15:47.20 | Corydon76-dig | The purpose is to ensure that both ends agree on each channel's state, to avoid channel inactivation |
| 15:47.56 | Corydon76-dig | There's a parameter you can tweak, if you want to turn that off or at least reduce its occurrence |
| 15:48.04 | idespinner | ive never seen the message before... odd |
| 15:49.04 | idespinner | what is the parameter that we tweak? |
| 15:49.55 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
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| 15:59.49 | *** join/#asterisk brandonf (~bran@vaoffice.inmotionhosting.com) |
| 16:02.02 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:05.00 | brandonf | Currently using freepbx/asterisk, trying to find a way to set the outbound cid (to one of our DIDs) based on the number I dial (from a lookup in our database). Does anyone know of an existing module, or do I have a bit of customizing to do? |
| 16:05.39 | Qwell | brandonf: #freepbx |
| 16:05.53 | brandonf | kk.. thx |
| 16:06.39 | p3nguin | Calls to a DID is not outbound CID. |
| 16:07.30 | brandonf | i want to set the outbound caller id to a specific DID (one of ours) |
| 16:07.49 | p3nguin | still wrong |
| 16:08.02 | p3nguin | DID is for INBOUND CALLING. |
| 16:08.16 | p3nguin | It has nothing to do with CallerID nor outbound calls. |
| 16:08.34 | brandonf | gotcha |
| 16:10.12 | *** join/#asterisk iscario (~div@laureades.davout.pck.nerim.net) |
| 16:11.51 | *** join/#asterisk imox1234 (~imox1234@p4FC5C167.dip0.t-ipconnect.de) |
| 16:12.11 | imox1234 | hello, i have problems to connect asterisk with mysql |
| 16:12.51 | imox1234 | here are my pastbin http://pastebin.com/suhLcpD0 |
| 16:16.03 | imox1234 | can somebody help me to connect to mysql server ? |
| 16:16.15 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 16:16.21 | [TK]D-Fender | imox1234: You oconnected via CLI but didn't prove database rights |
| 16:17.06 | imox1234 | the database have all rihts |
| 16:17.08 | imox1234 | rights |
| 16:17.13 | [TK]D-Fender | imox1234: prove it |
| 16:17.16 | imox1234 | in the pastbin you can sea it |
| 16:17.18 | imox1234 | see |
| 16:17.31 | imox1234 | you look in the pastbin |
| 16:17.35 | [TK]D-Fender | imox1234: No, I can't You didn't use the db. You didn't list tables. You didn't query one |
| 16:18.01 | imox1234 | mysql -u asterisk -p asteriskcdrdb |
| 16:18.08 | imox1234 | and then all ok ? |
| 16:18.27 | [TK]D-Fender | imox1234: You CONNECTED with a user. You did NOT touch a single F-ING OBJET while you were connected. |
| 16:18.30 | imox1234 | the user asterisk has ALL rights für this DB |
| 16:18.31 | [TK]D-Fender | imox1234: MEANINGLESS |
| 16:20.03 | imox1234 | http://img413.imageshack.us/img413/1179/bildschirmfoto20100504u.png |
| 16:20.07 | imox1234 | here please look |
| 16:20.08 | imox1234 | all rights |
| 16:20.50 | [TK]D-Fender | imox1234: Don't give me a story. Get off your damn ass and show me via CLI. I also don't see the configs to match |
| 16:21.12 | imox1234 | i dont know how ?? |
| 16:21.25 | imox1234 | what you want ? |
| 16:21.33 | imox1234 | the user asteirsk has alll rights |
| 16:21.45 | imox1234 | why i have to go on the cli ? and i dont know how |
| 16:21.49 | p3nguin | Perhaps that is part of the reason things are not working. |
| 16:22.07 | imox1234 | i can connect with mysql administrator |
| 16:22.08 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 16:22.22 | imox1234 | with ecplise |
| 16:22.25 | imox1234 | too |
| 16:22.36 | imox1234 | its work |
| 16:22.37 | spiceycurry | hey guys, would this card Te420 ("http://www.asteriskexchange.com/listings/120") work with Fax For Asterisk? |
| 16:22.48 | spiceycurry | (in Canada/US?) |
| 16:22.58 | Qwell | spiceycurry: sure |
| 16:23.14 | spiceycurry | ok, its 2400US, so I wanted more than my guess |
| 16:23.35 | spiceycurry | is there a better card, or is it about the same as the rest in its clasS? |
| 16:23.42 | spiceycurry | (or a suggested card) |
| 16:23.56 | Qwell | spiceycurry: that would be the suggested card |
| 16:24.12 | Qwell | and other (acceptable..) vendors are about the same price |
| 16:24.16 | spiceycurry | Ok cool |
| 16:24.21 | Qwell | You could buy a cheaper card, but we would all laugh at you. |
| 16:24.26 | spiceycurry | rofl |
| 16:24.32 | imox1234 | [TK]D-Fender: ??? |
| 16:24.43 | spiceycurry | infobot, what is spiceycurry |
| 16:24.44 | infobot | you are probably a bot molester. He touches my no-no area! |
| 16:24.58 | spiceycurry | :O |
| 16:25.03 | spiceycurry | On that note, I am out! :D |
| 16:25.15 | spiceycurry | Thanks again |
| 16:25.22 | coppice | $2400 sounds a lot for a card with no EC |
| 16:25.56 | p3nguin | Fax needs EC? |
| 16:25.58 | spiceycurry | well, with software only echo cancellation - its 1500 us |
| 16:26.09 | spiceycurry | # |
| 16:26.09 | spiceycurry | # With DSP echo cancellation - $2326.50 USD |
| 16:26.09 | spiceycurry | # Software only echo cancellation - $1446.50 USD |
| 16:26.20 | coppice | if you only want to do FAX, EC is not needed |
| 16:26.25 | imox1234 | ok now it works |
| 16:26.36 | imox1234 | [TK]D-Fender: the rights was not the problem :-P |
| 16:26.39 | spiceycurry | ok, whats the echo cancellation for? |
| 16:26.50 | Qwell | canceling echo |
| 16:26.57 | p3nguin | lol |
| 16:27.00 | spiceycurry | rofl |
| 16:27.27 | spiceycurry | Ok, ok. I will get it without echo cancelling |
| 16:27.49 | spiceycurry | though it's always great to get extra expensive stuff I don't need! :D |
| 16:28.12 | coppice | then why use fax for asterisk? |
| 16:28.43 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 16:28.43 | spiceycurry | it would be a lot easier I think to deploy on large scale installs |
| 16:29.40 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 16:29.59 | coppice | dunno. i've never heard from anyone with a big install of it. people usually use it for one channel, because that's free :-) |
| 16:30.04 | *** join/#asterisk stmaher (~stephen@80.68.89.200) |
| 16:30.13 | stmaher | hi guys.. |
| 16:30.27 | stmaher | I have an asterix box with a digium pstn 8 port card.. |
| 16:30.39 | stmaher | calls seems to drop after 30 seconds everytime.. |
| 16:30.43 | stmaher | any idea what could be causing it? |
| 16:31.06 | spiceycurry | we have 10 channels plus the free one, its pretty good so far. We're going to be packing 110 more licenses on it |
| 16:31.15 | spiceycurry | then replicating the same setup for another 10 machines |
| 16:31.38 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 16:31.48 | *** join/#asterisk maurox (~maurox@190.209.184.32) |
| 16:32.33 | coppice | well, that will cost a lot more than the card, so I guess the price of the card is less important |
| 16:32.58 | spiceycurry | yea |
| 16:33.15 | spiceycurry | I'd prefer to take that extra 1000 and put it back into the channel cost |
| 16:33.31 | spiceycurry | so i'll get that card, without hardware echo cancellation if possible |
| 16:33.33 | coppice | then use a free fax option |
| 16:34.07 | spiceycurry | Does Hyla/IAXModem up with T.38 yet? |
| 16:35.31 | coppice | that's an either or. iaxmodem+hylafax or t38modem+hylafax. the genuinely free option within asterisk supports T.38, though. It looks like they might even have it doing T.38 gateway soon |
| 16:36.18 | spiceycurry | hmm, interesting, I will do a bit of reading today |
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| 16:47.48 | garymc | hi peeps, im going to dubai in 2 weeks (hopefully unless ash cloud stops me) and Im told it eithe illegal to voip all from there and possibly the hotel will blcok port 5060. How would I get aourn the blocked port? |
| 16:48.06 | Qwell | garymc: we aren't going to help you break the law.. |
| 16:48.08 | garymc | *either *call |
| 16:48.31 | garymc | yes I heard (like rumour) |
| 16:48.47 | garymc | cant confirm its true, but if it is illegal I wont be doing it |
| 16:49.03 | Qwell | then you don't need help getting around it |
| 16:49.12 | Qwell | problem solved |
| 16:49.19 | KavanS | get a vpn? |
| 16:49.44 | garymc | So ok im going to a hotel where they block port 5060 but wirless is in each room. It is legal to make calls in this particular country how do I get around port 5060 |
| 16:49.55 | Nugget | a vpn doesn't make it legal, it just makes it harder to get caught. |
| 16:49.56 | KavanS | garymc, consult an attorney |
| 16:50.07 | KavanS | legality is not a subject that would be discussed here |
| 16:50.14 | p3nguin | If it's legal, they probably won't block the ports. |
| 16:50.18 | garymc | ok so a VPN? |
| 16:50.40 | garymc | no its legal they just block ports 5060 so you have to use there phones at high cost |
| 16:50.58 | garymc | or your own mobile network again at high cost |
| 16:51.07 | Nugget | sounds like it's illegal if the googles are to be beleived. |
| 16:51.37 | Nugget | in UAE/Dubai |
| 16:52.26 | garymc | yeah so i seen on google. |
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| 16:52.54 | p3nguin | If it is legal but they are trying to railroad you into using their services, it's easy to get around a few blocked ports. |
| 16:53.48 | beefpastry | I could see a hotel blocking the port on their own wireless network, but illegal for a country? Thought that stuff only happened in America after the Telcos make a few campaign "contributions." |
| 16:54.06 | imox1234 | when i want to use the accouncode, I have to set this in the dialplan or it is possible to set the accoundcode for a extension in the sip.conf ? |
| 16:54.14 | p3nguin | beefpastry: In Kuwait, for example, it is not legal to call internationally using VoIP. |
| 16:54.30 | p3nguin | It's okay within the country, but that is all. |
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| 16:55.00 | p3nguin | imox1234: extensions go in extensions.conf, not sip.conf. |
| 16:55.03 | beefpastry | Well, proximity in area likely translates to proximity in law. |
| 16:55.13 | beefpastry | I suppose. |
| 16:55.42 | imox1234 | when i have to set for every dialplan the exten with the accountcode ? |
| 16:55.56 | [TK]D-Fender | imox1234: You can set it in your sip peer |
| 16:56.06 | p3nguin | imox1234: You can set accountcode for each sip peer in sip.conf. |
| 16:56.07 | imox1234 | ahh ok thanks |
| 16:56.40 | p3nguin | that's not relative to extensions, though. You can override account codes in the extensions if you wanted. |
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| 17:01.46 | Qwell | beefpastry: don't forget - in many countries the telcos are owned and operated by the government |
| 17:02.02 | Qwell | they don't *need* campaign contributions |
| 17:02.05 | beefpastry | Yup |
| 17:02.19 | garymc | yeah so why would these fookers make it illegal for me to call home through my Asterisk box? |
| 17:02.26 | garymc | UAE in particular |
| 17:02.35 | Qwell | it doesn't matter why. it just matters that it is. |
| 17:02.54 | garymc | yeah so what sort of punishments if caught and how could you be caught~? |
| 17:02.59 | Qwell | you are not a citizen, so you have absolutely no say in the matter |
| 17:03.04 | garymc | How can you get caught? |
| 17:03.09 | Qwell | consult an attorney. |
| 17:03.12 | beefpastry | A little surprising for Dubai, though...oil and tourism rich (although I guess it's a good way to keep as much of the tourism cash as possible). |
| 17:03.41 | garymc | yeah so how would they catch you if port 5060 is open? Can they tell your making a Voip call? |
| 17:04.02 | beefpastry | Wouldn't be that difficult to monitor. |
| 17:04.41 | mr_ian | have you tried using a different port for SIP? |
| 17:05.03 | *** join/#asterisk voipnoob (shiva@detroit.slack.net) |
| 17:05.03 | Ad-Hoc | hi ppl |
| 17:05.31 | [TK]D-Fender | garymc: Fucking with the law in a country like that will land you in prison. |
| 17:05.38 | mr_ian | garymc: best answer it know is actually "Witopia" |
| 17:05.47 | mr_ian | a VPN service for about $30/yr |
| 17:06.35 | garymc | Yeah probably [TK]D-Fender but it looks like the whole of the internet over there has port 5060 blocked even if the hotel doesnt block it |
| 17:07.05 | p3nguin | My VPN service is $free/yr. |
| 17:07.13 | mr_ian | no offence, but in the UAE people ignore the legislation *way* more than here, that is WHY they are so much more severe with punishment. |
| 17:07.15 | [TK]D-Fender | garymc: because the gov't regulates the internet there. |
| 17:07.30 | garymc | yeah fuk sake, im gonna have to leave the iphone at home |
| 17:07.37 | garymc | bastards |
| 17:07.44 | [TK]D-Fender | garymc: Or *gasp* use it to place PHONE CALLS |
| 17:08.10 | garymc | no i cant afford my Networks charges they are way too expensive |
| 17:08.15 | mr_ian | take it anyway, you can always use it to play "Plants vs. Zombies" ;) |
| 17:08.23 | garymc | thats why i was gonna use sip |
| 17:08.41 | garymc | yeah suppose so on the beach in 40c temps |
| 17:08.58 | mr_ian | just don't try to use "fring", they are an Israeli company ;P |
| 17:09.18 | voipnoob | hello - i am newbie to VOIP & Asterisk etc - is this a good forum to ask some basic questions - i am trying to learn stuff |
| 17:09.22 | garymc | Couldnt I just deny using it? |
| 17:09.31 | [TK]D-Fender | voipnoob: So far... yes |
| 17:09.41 | voipnoob | tx |
| 17:10.02 | [TK]D-Fender | garymc: PAYG local SIM card... |
| 17:10.09 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 17:10.26 | voipnoob | I am trying to figure out when exactly something like OpenSER becomes neccessary - i.e. can VOIP telephony be done just with Asterisk |
| 17:10.30 | McBoingbo | calls in asterisk use the SIP protocol correct? |
| 17:10.39 | leifmadsen | yes |
| 17:10.48 | voipnoob | or is something like OpenSer always neccessary |
| 17:10.54 | leifmadsen | voipnoob: might want to check out http://www.asteriskdocs.org |
| 17:11.02 | leifmadsen | voipnoob: no, OpenSER is not necessary |
| 17:11.15 | mr_ian | garymc: try to use SIP on alternate port, or over VPN, you may or may not FAIL... but no one will give you trouble. |
| 17:11.18 | voipnoob | when exactly does OpenSER become neccesary? |
| 17:11.30 | voipnoob | i.e what does it provide? scalability or something more than that? |
| 17:11.51 | p3nguin | Why are proxies ever useful for anything? |
| 17:11.52 | leifmadsen | voipnoob: when you need to modify headers, handle LOTS of registrations, or want to load balance amongst many asterisk boxes and want a central registration authority |
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| 17:12.10 | voipnoob | got it |
| 17:12.17 | leifmadsen | unless you're doing a LOT (and I meant a LOT) of minutes, you probably don't need it |
| 17:12.43 | McBoingbo | With Asterisk do you choose the protocol or is it a static one? (SIP, H.323, SCCP) |
| 17:12.49 | voipnoob | Assuming 1000 users spread across different cities & 150 maximum concurrent calls |
| 17:12.53 | mr_ian | garymc: fring is only a problem because it routes calls through HQ... which is in Israel. you probably don't even want to *mention* fring, as Arabs can be very touchy about Israelis monitoring their calls... |
| 17:12.56 | voipnoob | it's not neccesary? |
| 17:13.07 | leifmadsen | McBoingbo: answer should be available to you in the documentation |
| 17:13.27 | leifmadsen | voipnoob: probably not -- depends what you're doing (lots of call recording or transcoding can add significant load) |
| 17:13.35 | voipnoob | no recording at all |
| 17:13.37 | voipnoob | no video |
| 17:13.41 | voipnoob | just voice calls |
| 17:13.42 | leifmadsen | in general though, Asterisk can handle 150 simultaneous calls |
| 17:13.50 | leifmadsen | depending on CPU power, etc... |
| 17:13.57 | voipnoob | okie |
| 17:14.03 | leifmadsen | (which is pretty much any modern quad-core system) |
| 17:14.17 | voipnoob | what kind of transcoding are you talking about |
| 17:14.28 | leifmadsen | I'm talking about converting between one codec and another |
| 17:14.29 | *** join/#asterisk CatLynx (~booga@173-11-77-182-SFBA.hfc.comcastbusiness.net) |
| 17:14.39 | voipnoob | when would something like that be neccessary? |
| 17:14.42 | leifmadsen | I'm going to direct you again to the documentation :) |
| 17:14.51 | voipnoob | same doc - asteriskdocs? |
| 17:14.54 | leifmadsen | these are answers readily available |
| 17:14.57 | voipnoob | i have started download |
| 17:14.58 | leifmadsen | yes |
| 17:15.08 | voipnoob | i am on a slow connection from home |
| 17:15.15 | leifmadsen | http://astbook.asteriskdocs.org is the HTML version of the same book |
| 17:15.16 | CatLynx | that is a good doc, best to keep handy everywhere :) |
| 17:15.22 | voipnoob | ok, tx |
| 17:15.38 | p3nguin | ~answers |
| 17:15.39 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 17:15.53 | voipnoob | one question about SBCs unless that's covered in the book? |
| 17:16.10 | [TK]D-Fender | voipnoob: It isn't |
| 17:16.17 | CatLynx | anyeone else having issues with callcentric keeps timeing out, and after a while asterisk gives up trying anymore? have to do "sip reload" to get it back up? |
| 17:16.24 | [TK]D-Fender | voipnoob: * is not a "SIP server. it is a B2BUA |
| 17:16.31 | p3nguin | CallCentric is pretty sucky. |
| 17:16.47 | CatLynx | ya my logs are starting to proove that now :) |
| 17:16.54 | leifmadsen | ya, I see bugs about them on https://issues.asterisk.org a bunch |
| 17:17.14 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:17.20 | leifmadsen | they tend to change the way they handle calls which is incompatible with Asterisk -- which is funny since Asterisk is probably one of the most robust platforms for SIP :) |
| 17:17.21 | CatLynx | is there a setting to keep asterisk not from giving up? :) |
| 17:17.58 | CatLynx | it had single passwor failer and thats when my asterisk gave up |
| 17:18.00 | voipnoob | [TK]D-Fender - I was reading this article -> http://www.voipuser.org/forum_topic_8289.html |
| 17:18.16 | voipnoob | and it said that SBC is neccessary at the edge of your VOIP clouse |
| 17:18.19 | voipnoob | cloud |
| 17:18.37 | *** join/#asterisk Systemt` (~lol@109.67.23.122) |
| 17:18.38 | p3nguin | I think you have to compile it with --work-with-callcentric=YES |
| 17:18.44 | [TK]D-Fender | voipnoob: What "cloud"? |
| 17:18.48 | CatLynx | they was timing out from 1am to 4am and they at 4am I got password reject from them, and then asterisk gave up after that. |
| 17:18.56 | voipnoob | assuming i don't need roaming or call routing or presence |
| 17:18.59 | CatLynx | p3nguin: heheh thats a good one :) |
| 17:19.01 | voipnoob | i guess I don't need SBC |
| 17:19.16 | [TK]D-Fender | voipnoob: I guess you are are asking about means without declaring needs |
| 17:19.23 | Systemt` | p3nguin: how can i transfer anonymous call with the real number ? |
| 17:19.36 | Systemt` | but just on loges |
| 17:19.46 | p3nguin | systemt`: How do you know the real number if the call was anonymous? |
| 17:19.56 | voipnoob | [tk]d-fender - i am trying to understand the basics of VOIP telephony |
| 17:20.07 | voipnoob | what are the bare minimum stuff |
| 17:20.16 | [TK]D-Fender | voipnoob: What is your functional goal? |
| 17:20.25 | CatLynx | System: I do something like but I transfer it to my voice mail insted :) |
| 17:20.33 | [TK]D-Fender | voipnoob: Bare minimum is getting * set up with a simple phone and a simple ITSP |
| 17:20.35 | voipnoob | this is more of an academic exercise - but let me explain the scenario i am studying |
| 17:20.48 | Systemt` | p3nguin: From My sip |
| 17:20.53 | voipnoob | assume 1000 different offices |
| 17:20.55 | Systemt` | p3nguin: is show me |
| 17:21.06 | Systemt` | p3nguin: but just on the logs |
| 17:21.15 | CatLynx | exten => _NXXXXXX,1,GotoIf($["${CALLERID(num)}" = "" ]?20:2) |
| 17:21.56 | CatLynx | oh wait that my wrong one :) |
| 17:22.05 | voipnoob | 1000 different offices - no relation to each other - there is one Asterisk Server on the internet - Asterisk box is trunked to PSTN |
| 17:22.32 | voipnoob | people in each of these offices use a hardphone or softphone to connect to the asterisk box & make calls to either each other or to landline phones |
| 17:22.42 | CatLynx | exten => _1234,2,GotoIf($[${CALLERID(number)} = Restricted]?20:3) |
| 17:22.49 | voipnoob | it's 1000 different users |
| 17:22.56 | voipnoob | and 150 concurrent calls at maximum |
| 17:23.06 | voipnoob | [td-k]fender - did you get the scenario |
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| 17:23.33 | [TK]D-Fender | voipnoob: 1000 users starts getting pretty heavy... |
| 17:23.56 | [TK]D-Fender | voipnoob: Need a big server for it, You basically want only loose phones at sites? |
| 17:24.07 | [TK]D-Fender | voipnoob: No "local" PBX per site? |
| 17:24.22 | voipnoob | but out of the 1000 users, max of 150 concurrent calls |
| 17:24.27 | voipnoob | and only voice, not video |
| 17:24.29 | devmod | inside asterisk dialplan, can I bring EXTEN to all uppercase? |
| 17:24.38 | Qwell | what? |
| 17:24.54 | voipnoob | as of i am thinking if it can be done without a PBX per site |
| 17:24.59 | paulc | you want to write EXTEN => instead of exten => ? |
| 17:25.06 | voipnoob | because this isnt going to be used for internal calling |
| 17:25.32 | devmod | paulc, i meant like a toUpper function I could apply to the extension number if it was to be composed of characters as well |
| 17:25.39 | [TK]D-Fender | voipnoob: Sure |
| 17:26.08 | voipnoob | will the load be 2 heavy for the one external Asterisk server on the internet? |
| 17:26.36 | CatLynx | anyone know off hand, how to keep asterisk to retry password if it got a password fail repsonse from callcentric? |
| 17:26.39 | voipnoob | also OpenSer like stuff, you said won't be neccessary |
| 17:26.53 | voipnoob | so i am trying figure out how SBCs relate to this |
| 17:26.53 | devmod | paulc, nevermind it was actually called TOUPPER :) |
| 17:26.54 | p3nguin | The extension number is a number... there is no upper or lower case. |
| 17:27.05 | voipnoob | is it neccessary to be running an SBC |
| 17:27.10 | [TK]D-Fender | voipnoob: No, 1 big box should do fine |
| 17:27.12 | devmod | p3nguin, TOUPPER was introduced in 1.6 |
| 17:27.19 | p3nguin | fine |
| 17:27.33 | voipnoob | if i don't want hard or soft roaming, or presence or call routing |
| 17:27.40 | voipnoob | is an SBC still neccessary? |
| 17:28.02 | [TK]D-Fender | voipnoob: So far roaming doesn't matter, presence works, and "call routing" is completely vague |
| 17:28.07 | [TK]D-Fender | voipnoob: No SBC |
| 17:28.18 | voipnoob | ok |
| 17:28.31 | Mark22 | voipnoob: I don't see the problem with what you want |
| 17:28.35 | leifmadsen | Set(RESULT=${TOUPPER(${EXTEN})}) |
| 17:28.50 | voipnoob | when you say 1 Big Box - what kind of specs are you talking about? |
| 17:29.23 | devmod | leifmadsen, yup found it thx |
| 17:29.55 | voipnoob | Mark22 - i am not sure i understand your question - i don't have any problem - just trying to understand some VOIP telephony related concepts - i have read a lot of documents, but i am still not able to see the big picture of how the different components come together |
| 17:30.05 | [TK]D-Fender | voipnoob: Server. Mostly carbon, iron, copper, silicon, etc |
| 17:30.37 | voipnoob | :-) |
| 17:30.48 | Mark22 | voipnoob: probably something I would put in a VPS ;) |
| 17:31.12 | voipnoob | what's VPS? |
| 17:31.26 | Mark22 | voipnoob: what do you want to know exactly? |
| 17:31.33 | Mark22 | A VPS is a virtual private server |
| 17:32.03 | voipnoob | i am trying to figure out when SBCs & OpenSER become neccessary - TKDFender has cleared my doubts |
| 17:32.06 | voipnoob | tx, tkdfender |
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| 17:35.09 | voipnoob | one last question - is a SIP proxy same as SIP server? |
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| 18:11.38 | pabelanger | Anybody have a decent URL for somebody to learn Linux commands / concepts? |
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| 18:12.19 | pabelanger | ~linux |
| 18:12.20 | infobot | i guess linux is an operating system that beats the socks off Windows and Mac! Go Linux! |
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| 18:45.07 | JasonL | Hello, wondering if someone can help me.. I'm using the Dial() cmd with the TK arguments to allow parking and transfer, however this prevent DTMF from being transmitted to the remote end which prevents the ability to dial options on remote IVR's... Is there a way around this? I've been searching and can't find any details on this... |
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| 18:49.11 | paulc | Why use TK? Are you using analog phones? |
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| 18:55.26 | drfreeze | Anyone have a voip account that they have connected a Polycom phone to directly? |
| 18:55.58 | [TK]D-Fender | drfreeze: ...What is an "indirect" voip account? |
| 18:58.23 | drfreeze | [TK]D-Fender: no * involved. Just a polycom phone, a router and an internet connection |
| 19:00.11 | idespinner | sounds like a silly question but how does one find out which version of dahdi theyre running? |
| 19:00.48 | [TK]D-Fender | idespinner: dahdi_cfg -vvvv |
| 19:00.57 | [TK]D-Fender | drfreeze: Same as anything else |
| 19:02.04 | drfreeze | I see some VoIP accounts that will try to sell you a phone and claim only their phone works with the service |
| 19:02.53 | [TK]D-Fender | drfreeze: Yes, because their servers are made to recognize the UA that is contacting them and if they don't like what they see they tell you to GTFO |
| 19:03.27 | Slugs_ | GTFOB |
| 19:03.28 | Slugs_ | ;)_ |
| 19:04.27 | drfreeze | So, anyone recommend a voip account that works with polycom |
| 19:05.00 | leifmadsen | unlimitel.ca ? |
| 19:07.45 | Slugs_ | drfreeze, voicenetwork.ca |
| 19:07.45 | drmessano | You could always go with the free hosted pbx from Aretta and just use whatever provider with it |
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| 19:08.13 | [TK]D-Fender | drfreeze: jsut about any that will work with * |
| 19:09.16 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 19:09.51 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:10.02 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:10.49 | DHE | I'm probably not going to find what I'm looking for, but does anybody know of software like iaxmodem but with data modem support? Like for real dial-up? (56k a bonus) |
| 19:11.31 | raden_work | is there a way to like page all phones with a message ? |
| 19:11.35 | raden_work | or just an extension ? |
| 19:11.58 | [TK]D-Fender | raden_work: "core show application page" |
| 19:12.01 | *** join/#asterisk norrec (~Ghost@76-201-85-28.lightspeed.frokca.sbcglobal.net) |
| 19:12.06 | raden_work | [TK]D-Fender, APPRECIATED :) |
| 19:13.14 | norrec | hey, I need to debug a sip call and i know how to start the debug but is there way to to get just the debug info into a log? |
| 19:16.02 | Slugs_ | [TK]D-Fender, y arnt you an op? they afraid you'll ban everybody? |
| 19:17.52 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 19:23.02 | drmessano | or he is, and he gets some satisfaction from going +o and banning newbs that call him names and going off on him when they realize they're clueless |
| 19:23.33 | Slugs_ | lol |
| 19:23.39 | drmessano | Two sides to every coin.. Unless you're handy with tools |
| 19:23.42 | Qwell | because it's much funner to do this... |
| 19:23.43 | *** mode/#asterisk [-o Qwell] by Qwell |
| 19:23.53 | Slugs_ | runs |
| 19:23.55 | Qwell | GODMODE ACTIVATE |
| 19:23.56 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:23.59 | Slugs_ | lol |
| 19:23.59 | Qwell | see? |
| 19:24.03 | drmessano | HA |
| 19:24.39 | Slugs_ | Qwell, definitly more effective |
| 19:24.48 | norrec | lol |
| 19:24.59 | drmessano | ~STAB ME |
| 19:24.59 | infobot | ACTION runs at drmessano with an origami Swiss Army knife, and inflicts a nasty paper cut. |
| 19:25.11 | Qwell | looks at leifmadsen |
| 19:25.16 | leifmadsen | looks at Qwell |
| 19:25.20 | Qwell | looks at /topic |
| 19:25.24 | russellb | ~thwack Qwell |
| 19:25.25 | infobot | ACTION hits Qwell on the knee with a sink |
| 19:25.29 | leifmadsen | looks at Qwell looking at the topic |
| 19:25.42 | Qwell | deflects the look using a mirror pointing towards the topic |
| 19:25.49 | russellb | staring is impolite |
| 19:25.50 | Slugs_ | this is /me abuse |
| 19:25.53 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7, 1.6.1.19, 1.6.0.27 (2010/05/04), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 19:26.13 | leifmadsen | breaks the mirror then breaks Qwell |
| 19:26.23 | Qwell | |
| 19:26.32 | leifmadsen | is not Qwell |
| 19:26.42 | leifmadsen | in whole or in part! |
| 19:26.44 | norrec | hey, does any1 know how to get the sip debug output into a log file |
| 19:26.50 | drmessano | Guess I need to run an svn update to see what you people have chosen to inflict on me now |
| 19:27.02 | drmessano | Ahem, I mean.. Update, sweet |
| 19:27.16 | Slugs_ | hehe |
| 19:27.32 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:27.48 | leifmadsen | norrec: use 'tee' or log it via the 'verbose' method in logger.conf |
| 19:28.08 | norrec | leifmadsen: tee? |
| 19:28.13 | leifmadsen | norrec: man tee |
| 19:28.15 | Slugs_ | tee hee |
| 19:28.46 | leifmadsen | asterisk -rvvv | tee /tmp/some-file-for-output.txt |
| 19:29.23 | norrec | i wanted to avoid logging all the other output |
| 19:29.27 | norrec | i just wanted the debug output |
| 19:29.40 | drmessano | I don't know how to run a tee, therefore should I man a tee? |
| 19:29.57 | Qwell | drmessano: I'll tee you, man. |
| 19:30.04 | Slugs_ | lol |
| 19:30.10 | drmessano | Don't tee me off |
| 19:30.14 | norrec | drmessano: thanks for the sarcasm, didnt realise it was a command =( |
| 19:30.30 | norrec | lol |
| 19:30.59 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:31.08 | drmessano | man welcome |
| 19:31.29 | Slugs_ | norrec on some distros you can man on man |
| 19:32.08 | norrec | haha, that seems a bit redundant |
| 19:32.19 | Qwell | norrec: why? |
| 19:32.43 | Micc | Do PRI cards work kind of like an ethernet card, can I use a special cable to go from one pri card on one machine to another pri card on another asterisk machine and provide analog service to that other asterisk machine? |
| 19:32.47 | drmessano | Don't man woman ... People claim Asterisk is so "poorly documented', try finding documentation on woman |
| 19:33.24 | norrec | Qwell: well i didnt realise man had some many options, so i guess it isnt redundant |
| 19:33.36 | norrec | so many* |
| 19:33.57 | Slugs_ | man can do many things |
| 19:34.12 | Micc | you guys are so punny |
| 19:34.15 | norrec | anyways, so is there a way to just output the sip debug info to a txt file rather than all the cli messages? |
| 19:34.21 | drmessano | Micc: Why would you want to do that if you can do so via IAX2, SIP, or ____? Is one of these machines non-asterisk? |
| 19:34.24 | Slugs_ | norrec, also less is more |
| 19:34.28 | Slugs_ | just remeber that |
| 19:34.43 | Micc | drmessano, because I need to send faxes over these lines. |
| 19:34.47 | Micc | and receive faxes. |
| 19:34.50 | Micc | both are asterisk. |
| 19:34.52 | Qwell | Micc: T.38 |
| 19:34.58 | drmessano | Yep T.38 |
| 19:35.03 | Qwell | but, to answer your question - yes, of course. That's all a channelbank is |
| 19:35.05 | Micc | One does not support T.38 |
| 19:36.22 | Micc | which version of asterisk do T.38 gateway? Would it have to receive and forward or could it T.38 to my fax server which has the fax for asterisk licenses? |
| 19:36.43 | Qwell | none do gatewaying |
| 19:36.49 | *** join/#asterisk kartik (~koolkarti@117.199.113.128) |
| 19:36.56 | Qwell | you would have to store/forward, yes |
| 19:37.01 | drmessano | What about TDMoE? |
| 19:37.08 | Micc | Or would that be a passthrough? Right now its goes from PRI I think to SIP to my server, then I SIP it to my fax server from there where it does receivefax |
| 19:37.12 | fenrus | TDMoE is neat. |
| 19:37.51 | Micc | does TDMoE work on asterisk 1.4? |
| 19:38.00 | fenrus | ATOM <3 |
| 19:38.01 | Micc | does it require a stand alone NIC? |
| 19:38.50 | drmessano | I found a TDMoE howto dated 2002, so I would say so |
| 19:40.31 | drmessano | http://www.projectmf.com/cgi-bin/ikonboard.cgi?act=ST;f=1;t=16;&#top |
| 19:41.42 | drmessano | I would probably spring for the extra NIC cards and run NIC <> NIC |
| 19:42.07 | drmessano | Since you're saving like $1000+ already, a couple $20 NICs aren't hard to swallow |
| 19:49.20 | *** join/#asterisk Mango (~iMango@d154-20-110-91.bchsia.telus.net) |
| 19:49.22 | jdoe | does the time range in gotoiftime wrap? |
| 19:49.41 | jdoe | ie can I specify 17:00-9:00? |
| 19:52.08 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 19:54.04 | Mango | I want to keep a log of the date and time a phone registers to the Asterisk server. Is there a better way to do this than parsing sip messages? |
| 19:55.41 | bn-7bc | jdoe: hmm tru splitting it in to parts 17:00-23:59 an 00:00-09:00 |
| 20:02.01 | *** join/#asterisk c0dyhi11 (~c0dyhi11@ip70-190-105-213.ph.ph.cox.net) |
| 20:05.36 | sbrath | So if i use a Digium Wildcard TE110P, the T1's from the telco are encoded in ulaw, correct? Can I default Voice mail to ulaw? |
| 20:08.05 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 20:09.03 | *** join/#asterisk suprstar (~suprstar@216.54.131.253) |
| 20:09.34 | sbrath | I figured it out.... I guess it sounds "Tinny" are there any adjustments? |
| 20:10.17 | Qwell | voicemail.conf, check what you're storing them as |
| 20:18.31 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 20:19.36 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:19.54 | *** join/#asterisk hatpanda (~pete-joh@triton.dsv.su.se) |
| 20:19.55 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 20:23.59 | JasonL | paulc: sorry for the late response.. I'm not using analog phones, using SIP phones... Arent the T and K args required to allow park and transfer? If I use the Dial() without, then I can't use the park and transfer |
| 20:25.36 | DHE | is there something like iaxmodem but with support for traditional dialup? (56k a bonus) |
| 20:26.45 | *** join/#asterisk MiserySoft (~LND@89.193.117.121) |
| 20:28.20 | *** join/#asterisk homiziado (~ernestofr@62.169.96.192.rev.optimus.pt) |
| 20:28.27 | *** join/#asterisk defslap (~andy@defsdoor.gotadsl.co.uk) |
| 20:31.36 | [TK]D-Fender | checkout time, BBIAB |
| 20:33.45 | *** join/#asterisk acxty (~acxty@201.220.136.118) |
| 20:34.36 | acxty | Hi guys, I am create a .call file but when I copy it to the outgoing directory it says Permision denied. I did chmod 777 to the file before mv it |
| 20:37.32 | Defraz | it has to be owned by asterisk |
| 20:37.40 | Defraz | or the user that asterisk is runnign as |
| 20:37.44 | leifmadsen | do you have permission to write to the /var/spool/asterisk/outgoing/ directory? |
| 20:37.47 | Defraz | and moving the file is better than copyy |
| 20:37.55 | acxty | I also did chown asterisk file.call |
| 20:38.16 | Defraz | a copy will copy it as the user you are coping as |
| 20:38.24 | Defraz | so if you chmod it to asterisk |
| 20:38.39 | acxty | I did chmod 777 on outgoing also |
| 20:38.39 | Defraz | then do a copy it will copy it as the user you are. |
| 20:38.49 | Defraz | so chmod it then move it. |
| 20:39.35 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 20:39.54 | chazzam | you mean chown it to asterisk? |
| 20:40.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:41.14 | TheDavidFactor | I'm trying to move some ael code from 1.4.x to 1.6.2 I'm getting a bunch of these: ERROR[27190]: ael.flex:654 ael_yylex: Unhandled char(s): |
| 20:41.31 | TheDavidFactor | it's kind of hard to track down an empty error messsage :-S |
| 20:41.55 | TheDavidFactor | any suggestions on what to look for? |
| 20:43.16 | staffmember | am i suppose to configure extensions.conf or .ael? also, can someone point me in the right direction, I am using asterisk solely as a fax server, I want my faxes to go to PDF, is there a step by step guide on getting this done from scratch |
| 20:45.04 | *** join/#asterisk stix_ (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:45.34 | leifmadsen | staffmember: there is no step by step guide that I know of (actually, maybe on google now that I think about it) -- you want to search for iaxmodem + hylafax + asterisk |
| 20:45.51 | leifmadsen | staffmember: you can use either extensions.conf or extensions.ael -- they are two different methods to achieve the same goal |
| 20:46.58 | staffmember | leifmadsen: if i am using extensions.conf, do i need to delete extensions.ael ? how will asterisk know which of the 2 to use |
| 20:47.10 | leifmadsen | staffmember: actually it's smart enough to merge them |
| 20:47.18 | leifmadsen | staffmember: you can technically use both -- just don't overlap contexts |
| 20:47.33 | staffmember | if i delete .ael, will that cause problems? |
| 20:47.35 | leifmadsen | staffmember: otherwise, just don't compile AEL, or disable the AEL modules in modules.conf |
| 20:47.43 | staffmember | ok |
| 20:47.51 | leifmadsen | it might cause a WARNING or something on the console that says extensions.ael is not available |
| 20:48.28 | c0dyhi11 | is there an issue with having mulitple analog cards installed in a system? |
| 20:48.42 | keith4 | no |
| 20:48.50 | c0dyhi11 | Sweet. thx. |
| 20:48.55 | keith4 | depending on what you mean by "an issue" |
| 20:49.10 | c0dyhi11 | Umm... not be able to make calls |
| 20:49.19 | c0dyhi11 | or recieve calls |
| 20:49.32 | keith4 | then no |
| 20:49.39 | leifmadsen | no, that should allow you to receive more calls :) |
| 20:49.57 | keith4 | but you might need to fight with them to make them come up in the same order all the time |
| 20:50.09 | paulc | JasonL: Delayed reply - No, just use the native transfer feature on the SIP phone.. ditto for call parking (xfer to 700 by default) |
| 20:51.49 | spenguin[work] | AEL is so nice, why doesnt everyone just use AEL? |
| 20:52.52 | leifmadsen | I prefer extensions.conf :) |
| 20:56.32 | chazzam | I still have yet to even look at AEL |
| 20:56.39 | TheDavidFactor | any one able to help me with this ael error: |
| 20:56.41 | TheDavidFactor | ERROR[27190]: ael.flex:654 ael_yylex: Unhandled char(s): |
| 20:58.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:59.39 | acxty | My peer is 110. I want to make a phone call between 109 and a cellphone. I receive the call on 109 but when it dial the cellphone it says that the status is "CONGESTION" http://dpaste.com/190893/ |
| 21:00.12 | *** join/#asterisk MiserySoft (~lnd@89.193.35.14) |
| 21:00.52 | ChannelZ | that's not right |
| 21:01.00 | leifmadsen | TheDavidFactor: what version? I thought I saw something like that in the bug tracker recently... |
| 21:01.09 | leifmadsen | TheDavidFactor: do you have a space or something where it shouldn't be? |
| 21:01.44 | ChannelZ | acxty: SIP/110 is a device? (A phone?) |
| 21:02.08 | acxty | is the asterisk account |
| 21:02.21 | ChannelZ | to where? |
| 21:02.35 | acxty | to a panasonic pbx |
| 21:02.46 | TheDavidFactor | 1.6.2.6 and I'm getting the error on almost every line (if not every line, I didn't count it out) |
| 21:03.05 | leifmadsen | TheDavidFactor: 1.6.2.7 is out now :) |
| 21:03.17 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:03.22 | TheDavidFactor | :-P |
| 21:03.32 | JasonL | paulc: oh, i see why you're asking that.. actually we're intergrating the server with a Nortel BCM, and for those phones to park calls we need to use DTMF |
| 21:03.41 | leifmadsen | TheDavidFactor: https://issues.asterisk.org/view.php?id=17215 |
| 21:04.13 | paulc | acxty: So your panasonic is a SIP peer? (in which case I'd expect something like SIP/123@panasonic) |
| 21:04.44 | leifmadsen | TheDavidFactor: forget to close with a bracket or something? |
| 21:05.03 | paulc | JasonL: Ah.. we have a BCM here and had a few DTMF issues generally.. so you're feeding calls from Nortel handsets through Asterisk (direct via SIP) and want them to be able to transfer etc? |
| 21:05.10 | acxty | I am connected to a panasonic pbx. They gave me a sip account which is 110. I can receive calls |
| 21:05.19 | ChannelZ | acxty: if you dial that number 60848678 from your device (that is extension 109) does it work? |
| 21:05.31 | TheDavidFactor | no it's code moved from 1.4 no errors on 1.4; and it works, the errors don't prevent anything from working |
| 21:05.47 | acxty | my extension is 110 |
| 21:05.53 | leifmadsen | TheDavidFactor: huh, well that's odd then |
| 21:06.14 | raden_work | [TK]D-Fender, i have no application page ! |
| 21:06.15 | paulc | acxty: So what about Dial(SIP/somenumber@110) (assuming you have [110] in sip.conf) |
| 21:06.15 | ChannelZ | ok but you're dialing SIP/110/109 |
| 21:06.21 | TheDavidFactor | yea, I can ignore it, but errors on the CLI bug me :-S |
| 21:06.26 | ChannelZ | which means extension 109 at whatever SIP/110 is |
| 21:06.42 | acxty | I get the call on 109 |
| 21:07.10 | ChannelZ | either way can you pick up 109 or 110 and dial that number? |
| 21:07.50 | chazzam | doesn't the call file start a channel on extension 109 right away, and then the dial-plan execution tries to create another one? |
| 21:08.03 | raden_work | [TK]D-Fender, server100*CLI> core show version |
| 21:08.04 | raden_work | Asterisk 1.6.0.10 built by root @ linux-zm7c on a i686 running Linux on 2009-07-27 20:11:39 UTC |
| 21:08.06 | ChannelZ | IE you are sending a call to SIP/110 as 60848678 - taking your call file out of the equation, is that even valid for how your system is setup? |
| 21:08.42 | *** join/#asterisk Alagar (~Administr@122.164.43.211) |
| 21:09.35 | acxty | that is the idea, to call first 109 |
| 21:09.46 | acxty | and then the extension which is the cellphone |
| 21:09.59 | ChannelZ | 109 is connected to the OTHER PBX |
| 21:10.01 | [TK]D-Fender | raden_work: Guess you didn't have DAHDI installed prior |
| 21:10.03 | acxty | yes |
| 21:10.13 | ChannelZ | And your ability to dial out is via the OTHER PBX |
| 21:10.26 | raden_work | [TK]D-Fender, the only system i dont crap |
| 21:12.19 | acxty | yes |
| 21:12.59 | ChannelZ | So first of all why is Asterisk involved in this? |
| 21:14.32 | acxty | It is connected to a vehicle tracking server. When the clients press a bottom on the vehicle it receive the signal on the server and the asterisk make the call to both sides |
| 21:15.12 | JasonL | paulc: thats right, we're actually talking to the BCM over a PRI... and the BCM can park and transfer calls... but when they call outside IVR the DTMF isnt getting through, so I'm trying to find a way around that |
| 21:15.55 | ChannelZ | As in, you're telling asterisk to call SIP to another PBX (109). You are then telling asterisk to connect it to a local extension, which in turn calls SIP to the same PBX. It's probably confused. |
| 21:16.16 | chazzam | It sounds like you are basically trying to do an originate to two external numbers from Asterisk' perspective |
| 21:16.23 | ChannelZ | yeah |
| 21:16.24 | raden_work | [TK]D-Fender, I presume the only way to install dahdi is reinstall asterisk ? |
| 21:16.25 | chazzam | but both go through the same "trunk" |
| 21:16.39 | paulc | JasonL: Is your call to external IVR over the PRI? |
| 21:17.11 | paulc | JasonL: I had some iffy results with DTMF to our own IVR systems.. it was kind of weird.. out via PRI to PSTN, then in via SIP to our IVRs on the other side. |
| 21:17.36 | [TK]D-Fender | raden_work: Correct |
| 21:17.51 | raden_work | May as well update while im at it too then |
| 21:18.12 | JasonL | BCM -> (pri) -> Asterisk -> (pri) -> PSTN (DTMF works fine when I don't have the TK options in the Dial cmd |
| 21:21.17 | *** join/#asterisk MiserySoft (~LND@89.193.233.193) |
| 21:22.03 | acxty | Well guys, what I would do is to use 2 extensions. One will call 109 and the other will call the cellphone |
| 21:22.37 | *** join/#asterisk MiserySoft (~LND@89.193.233.193) |
| 21:22.46 | acxty | One question. When I need to dial to a cellphone on the panasonic pbx I need to press 9 so it can open la line. How can I tell that on Dial() |
| 21:23.27 | chazzam | you put a 9 in the Dial command... ? |
| 21:24.13 | chazzam | ~book |
| 21:24.13 | 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 |
| 21:24.20 | chazzam | acxty: check that |
| 21:27.05 | acxty | I had been login on voip-info but hadn't find anything |
| 21:27.40 | [TK]D-Fender | acxty: add the 9. |
| 21:28.10 | [TK]D-Fender | acxty: Extension: 60848678 <---- this didn't start with a 9. By your own statement the number is not valid |
| 21:29.51 | brandonf | woooot.. wrote a custom agi script, checks a local phone number database to see which 'brandname' the customer belongs to, and sets the appropriate outgoing caller id depending on the phone number we dial, strangely very easy :) |
| 21:31.34 | ChannelZ | That's sort of why I asked " if you dial that number 60848678 from your device (that is extension 109) does it work?" but, what do I know.. |
| 21:33.39 | paulc | JasonL: I'm wondering (bit rusty here) - could you use features.conf (featuremap.conf?) (that "other" config file) to trigger transfer and parking, and not use TK on the Dial string? |
| 21:34.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:34.18 | paulc | brandonf: congrats :-) Using channel vars then? I did something similarish using CURL and was happy too :-) |
| 21:34.44 | *** part/#asterisk mnick86 (~Matthias@whhem00016.cip.uni-regensburg.de) |
| 21:51.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:52.41 | Linuturk | so, I've got an extension that's getting a busy signal when you try to access the mailbox |
| 21:53.10 | Linuturk | the configs look right, (just like the other phones), and no errors show up in the console when the attempt is made |
| 21:53.18 | Linuturk | yet, the phone can make calls and such |
| 21:53.24 | Linuturk | and shows registered in sip peers |
| 21:53.44 | Chainsaw | Linuturk: Make sure you have verbosity & debug dialled up to 10. |
| 21:56.10 | Linuturk | blah, figured it out. lol. I forgot to put the voicemail extension in the list for this new mailbox |
| 21:56.13 | Linuturk | dur |
| 21:56.15 | Linuturk | sorry :( |
| 21:58.13 | *** join/#asterisk ManxPower (~manxpower@190.sub-75-216-186.myvzw.com) |
| 22:00.39 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 22:04.13 | *** join/#asterisk Yudaisrael1984 (~Yuda@77.127.144.138) |
| 22:06.16 | *** join/#asterisk garymc (~chatzilla@host86-135-203-71.range86-135.btcentralplus.com) |
| 22:06.34 | garymc | Hi Guys can I setup a VPN on my asterisk server? |
| 22:07.26 | Yudaisrael1984 | hi guys quick question: is it possible to have the following setup running freepbx and a2billing same server and have a extension of freepbx setup as a trunk in a2billing (for billing purposes) |
| 22:07.27 | [TK]D-Fender | garymc: No. |
| 22:07.32 | [TK]D-Fender | garymc: We forbid you |
| 22:07.39 | garymc | lol |
| 22:07.51 | garymc | come on some help here... its possible though? |
| 22:08.02 | Yudaisrael1984 | of course its possible |
| 22:08.16 | Qwell | Yudaisrael1984: #freepbx, and #a2billing |
| 22:08.41 | Yudaisrael1984 | qwell question isnt on the actual scripts of freepbx and a2billing |
| 22:08.47 | Yudaisrael1984 | its the theory behind it |
| 22:08.50 | [TK]D-Fender | garymc: Its #&$^ING LINUX. Of course you can can do VPN on it |
| 22:08.57 | paulc | garymc: There was a series of articles in Linux Journal recently on how to set up OpenVPN |
| 22:09.05 | [TK]D-Fender | garymc: So go pick your method and go to their support channel |
| 22:09.08 | Qwell | Yudaisrael1984: we can't know the answer. we don't know how they will work together. they, however, will. |
| 22:09.09 | garymc | is it easy? |
| 22:09.18 | paulc | "easy" is subjective ;-) |
| 22:10.11 | paulc | garymc: When I'm away, I SSH to my linux box, and proxy all local traffic across the tunnel (web, email, MSN, etc).. not quite the same as VPN but works good enough |
| 22:10.37 | Yudaisrael1984 | ok so i will ask in a asterisk method this way u guys wont give me crap of other channels because its useless to ask there because they probably will not know |
| 22:12.00 | Yudaisrael1984 | so in the extension.conf to have a line that says dial(SIP/local/number) and SIP/local is in sip.conf as a friend and has a context of outgoing-dialing |
| 22:12.35 | Yudaisrael1984 | that in the outgoing dialing theres another dial command Dial(SIP/TRUNK/Number) |
| 22:13.04 | garymc | Well no help in the centos channel |
| 22:13.28 | garymc | well i dont understand how VPN integrates or works |
| 22:13.36 | garymc | so ill go see if i can find that out |
| 22:13.47 | Yudaisrael1984 | [TK]D-Fender now i asked the question in asterisk terms only |
| 22:13.57 | Yudaisrael1984 | so please dont tell me to go to the channel #sip |
| 22:14.00 | Qwell | Yudaisrael1984: I didn't see a question there |
| 22:14.13 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 22:14.21 | Yudaisrael1984 | can i do something like that |
| 22:14.31 | [TK]D-Fender | Yudaisrael1984: I fail to see where you are going with this |
| 22:14.33 | Qwell | sure, it's your config files. do whatever you like |
| 22:14.38 | Yudaisrael1984 | <PROTECTED> |
| 22:15.00 | [TK]D-Fender | Yudaisrael1984: Where is the PEER pointing to? |
| 22:15.20 | Yudaisrael1984 | Sip/local is pointing nowhere |
| 22:15.27 | [TK]D-Fender | Yudaisrael1984: So is your idea |
| 22:15.36 | Yudaisrael1984 | thats mean |
| 22:15.37 | [TK]D-Fender | Yudaisrael1984: You can't "fake" this shit |
| 22:16.11 | [TK]D-Fender | Yudaisrael1984: There is no such thing as "pointing nowhere". It will CALL that peer. if it has nowhere to go it will FAIL. What is the point then? |
| 22:16.43 | Yudaisrael1984 | because that SIP/Local has a account ID this way it can be billed |
| 22:16.58 | Yudaisrael1984 | and i want anything coming from that other script to be billed as that account |
| 22:17.04 | Yudaisrael1984 | and not a different one |
| 22:17.07 | garymc | So since im running Asterisk NOW and it uses CENTOS.... I take it I need to google Centos VPN? |
| 22:17.15 | Qwell | garymc: correct |
| 22:17.29 | garymc | Can anyone help me as to understand how VPN with Asterisk would work? |
| 22:17.37 | [TK]D-Fender | Yudaisrael1984: how does this impact the channel that calls that dial in ANY way? |
| 22:17.53 | [TK]D-Fender | Yudaisrael1984: the peer isn't used. |
| 22:18.01 | Yudaisrael1984 | so that was my question |
| 22:18.07 | Yudaisrael1984 | if the peer isnt used it wont work |
| 22:18.31 | Yudaisrael1984 | there is no such thing as a local /virtual peer |
| 22:18.35 | CunningPike | Can anyone familiar with Polycom phones explain the effect of setting the voip.outboundProxy settings has? |
| 22:18.51 | Yudaisrael1984 | in other words i cant put as an example a voip phone on the same asterisk server |
| 22:18.57 | Yudaisrael1984 | because it cannot connect to itself |
| 22:19.10 | Yudaisrael1984 | (voip softphone) |
| 22:19.19 | Qwell | of course it can connect to itself. that doesn't make it pointing to "nothing" |
| 22:19.27 | [TK]D-Fender | Yudaisrael1984: Why would a softphone connect to itself?Says who? |
| 22:19.42 | [TK]D-Fender | garymc: * doesn't know, or need to know about VPN |
| 22:19.45 | Yudaisrael1984 | theoreticly acording to what your saying it cant |
| 22:20.06 | Qwell | Yudaisrael1984: no, YOU said it was pointing nowhere |
| 22:20.11 | [TK]D-Fender | Yudaisrael1984: You are the one who jsut brought up a softphone. Where does it fit in this mess? |
| 22:20.12 | Yudaisrael1984 | i cannot put a softphone on a asterisk server since what would be in host??? 172.0.0.1 |
| 22:20.29 | [TK]D-Fender | Yudaisrael1984: Yes, you can |
| 22:20.39 | *** join/#asterisk Ta^3 (~tacvbo@189.146.183.88) |
| 22:20.39 | [TK]D-Fender | Yudaisrael1984: Now I also don't see why you WOULD do this anyway |
| 22:20.48 | Yudaisrael1984 | because the softphone is part of the context that is creating that dial/local (local is a softphone) |
| 22:20.59 | Qwell | Yudaisrael1984: chan_local |
| 22:21.05 | garymc | cool so VPN wont affect my office phones at all? |
| 22:21.29 | paulc | garymc: shouldn't do |
| 22:21.30 | [TK]D-Fender | garymc: it can if your attempt to set one up screws with your servers other routings. |
| 22:21.49 | garymc | fuk |
| 22:21.52 | paulc | (proviso: "so long as you set it up right" *doffs hat to TKD*) |
| 22:22.02 | garymc | I need a tech guy, by next week |
| 22:22.04 | garymc | :( |
| 22:22.42 | paulc | garymc: what about getting a UAE phone number? any providers of that out there? (didx?) or does that go against the "we forbid VoIP" thing too I guess? |
| 22:23.03 | paulc | garymc: prepaid SIM once you're there, then build yourself a call back service using Asterisk? |
| 22:23.25 | paulc | (might be great as in a lot of non North American countries, inbound calls are free on prepaid) |
| 22:24.10 | garymc | hmmm |
| 22:24.35 | [TK]D-Fender | garymc: And what are you going to use as a client? |
| 22:24.41 | garymc | my Iphone |
| 22:25.01 | garymc | apparently it will connect to a VPN |
| 22:25.05 | CunningPike | We are in the process of inserting SER between our phones and Asterisk, and if we specify the hostname of the SER server, the phone registers and places calls. If we use DNS SRV records, SER returns a 480 response |
| 22:25.17 | [TK]D-Fender | garymc: really? What protocols does it support? Wil that let you run your softphone on it as well? |
| 22:25.18 | garymc | unless im way off on the Iphone as a client idea |
| 22:27.06 | garymc | it says L2TP , PPTP IPSec |
| 22:27.28 | garymc | also says Send all traffic |
| 22:28.04 | garymc | so im guessing the softphone would run through it |
| 22:28.20 | garymc | once VPN was set. They are a good peice of kit |
| 22:28.36 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:28.42 | *** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.nj.comcast.net) |
| 22:29.26 | TehRabbitt | I'm... back!!! LOL |
| 22:29.34 | russellb | l...o ...... l? |
| 22:30.14 | TehRabbitt | any suggestions on why dtmf tones were working yesterday when calling 1800 #'s using future-nine but now they dont work at all? |
| 22:30.26 | Qwell | TehRabbitt: because future-nine changed something. |
| 22:30.31 | Qwell | (or you did) |
| 22:30.44 | TehRabbitt | I haven't changed anything, so it's gotta be on their end then? |
| 22:30.49 | russellb | or something further upstream changed |
| 22:30.54 | TehRabbitt | hm |
| 22:30.56 | russellb | the key word being 'change' |
| 22:31.02 | TehRabbitt | dmtfmode= inband correct? |
| 22:31.03 | Qwell | blackbox. their fault regardless :D |
| 22:36.37 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:38.54 | TehRabbitt | how do you set outgoing caller-id? |
| 22:39.09 | TehRabbitt | Is it something I add to the sip.conf file? |
| 22:40.11 | [TK]D-Fender | TehRabbitt: Your call out uses whatever callerid the channel started with |
| 22:40.23 | garymc | Wow im having a difficult time finding out about this VPN shit |
| 22:40.23 | TehRabbitt | [TK]D-Fender: ah got it... |
| 22:40.39 | TehRabbitt | [TK]D-Fender: any ideas why DMTF tones won't work at all? :-\ |
| 22:40.53 | [TK]D-Fender | TehRabbitt: Wrong mode |
| 22:43.27 | TehRabbitt | got it working... |
| 22:43.55 | TehRabbitt | Set "disallow=all" then "allow=ulaw, alaw" (took out GSM and g729) then it worked |
| 22:44.05 | TehRabbitt | apparently inband doesn't like going over GSM or g729 |
| 22:44.41 | Baylink | Yeah, inband is probably reasonably limited to g.711 |
| 22:44.55 | [TK]D-Fender | TehRabbitt: Clearly not. Expecting audio decoing over a heavily compressed codec is like trying to read a 15th gen fax |
| 22:45.16 | TehRabbitt | well now tones work better than my cell phone lol |
| 22:45.31 | TehRabbitt | [TK]D-Fender: reminds me of the old GOES WEFAX service lmao |
| 22:45.56 | TehRabbitt | now THAT would be interesting to get working lmao |
| 22:47.47 | TehRabbitt | <PROTECTED> |
| 22:48.20 | Qwell | TehRabbitt: if your provider lets you, sure |
| 22:48.38 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:48.39 | TehRabbitt | p3nguin mentioned two yesterday |
| 22:53.23 | TehRabbitt | Wow 2 thumbs up to future-nine for 1800 termination btw... |
| 22:53.36 | TehRabbitt | i'm kinda suprised based on the quailty of their website lol |
| 22:54.35 | TehRabbitt | @Qwell: Do you know if flowroute supports the CID to display my CID? |
| 22:55.12 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 22:56.10 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 22:56.52 | [TK]D-Fender | TehRabbitt: I all but certain they let you rig the CID |
| 22:56.52 | *** join/#asterisk Doc (~scott@2001:470:1:8::2) |
| 22:57.10 | Doc | dont suppose anyone is a polycom expert? |
| 22:57.27 | Doc | having trouble with a phone that's reporting voicemail on the wrong line |
| 22:57.27 | Qwell | [TK]D-Fender plays one on TV |
| 22:57.30 | Baylink | "expert"? No, but whatcha need? |
| 22:57.43 | Baylink | "reporting" voicemail. |
| 22:57.52 | Doc | MWI |
| 22:57.59 | Baylink | You mean it has more than one line appearance, one line has voicemail, and the other line indicates with an envelope? |
| 22:58.10 | Doc | baylink: exactly! |
| 22:58.21 | Baylink | Hmmm. |
| 22:58.23 | Doc | 2nd line has voicemail, but it shows as being on the first line |
| 22:58.36 | Doc | even shows the right count of messages, just on the wrong line |
| 22:58.49 | Doc | line 1 doesn't have VM setup at all. line 2 does |
| 22:59.14 | Baylink | Hmm. That the envelope shows up on the first button might be significant. Pastebin the .cfg file for that phone? |
| 22:59.34 | Baylink | Did the phone ever only have one line appearing? |
| 23:00.00 | Doc | month or two ago maybe |
| 23:02.07 | TehRabbitt | [TK]D-Fender: so basically if I signed up for an account with flowroute, i could specify my DID as the CID so people can call me back over my DID and it will be routed properly? |
| 23:03.20 | Doc | http://pastebin.com/mcb4xp3b |
| 23:03.32 | Baylink | But it properly *works* on both lines now, Doc? Did that take more than one rewrite of the cfg file? I've seen phones that weren't running the cfg I thought they were cause I introduced an error, and wasn't logging from the phone. looking... |
| 23:03.42 | Doc | that the current version, although i've tried multiple variations |
| 23:04.08 | Doc | baylink: yah, it's defintiely reading the config. i can change the behavour - just cant get it to work |
| 23:04.10 | TehRabbitt | http://patfleet.com/funstuff.php I like the second one "for callers that don't listen" |
| 23:04.24 | Baylink | And you're saying that when there's VM on the SIP account, you get an envelope, but it's on the internal line. |
| 23:04.37 | Doc | like i said, it works perfectly - displays the number of messages waiting and all |
| 23:04.46 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 23:04.47 | Doc | just shows them as being on the wrong line |
| 23:04.59 | Baylink | Could it be that it wants "msg.2.mwi..."? |
| 23:05.19 | Doc | i'm thinking bug (because after all, it shouldn't be possible to do what it's doing even if i wanted to) |
| 23:05.20 | Baylink | I'm not completely familiar with the syntax, there, but I thought it always wanted the appearance as the second item. |
| 23:05.48 | Doc | doubt it, as it's correctly picking up the callback number and the like |
| 23:05.52 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:06.19 | Baylink | Yeah: it would interpret it as "msg.1.mwi" if my surmise is correct, leading to exactly the behaviour you see. |
| 23:06.55 | Baylink | Try it as I've changed, it, just for giggles |
| 23:06.59 | Doc | every example i can find, along with the default sample config files has it the way i have it |
| 23:07.03 | Baylink | Hmmm. |
| 23:07.04 | Doc | but i'll try anything at the moment |
| 23:07.14 | Doc | jsut restarting to try something else first |
| 23:07.16 | Baylink | Lemme go check one of mine, too. |
| 23:07.17 | Chainsaw | It wouldn't be the first time that all examples are wrong. |
| 23:07.40 | Chainsaw | (Or that no workable examples exist, looking at you Patton) |
| 23:08.20 | Baylink | No, damnit; mine is how yours was. Disregard. |
| 23:08.43 | Baylink | Could you, maybe, need msg.mwi.1.subscribe="0"? |
| 23:09.05 | Baylink | Maybe the bug is "expects all to be defined, uses slot numbers as relative" |
| 23:09.39 | Doc | that's what i'm testing at the moment |
| 23:09.48 | Doc | except the phone appears to have hung while booting... blah |
| 23:09.52 | Baylink | Yay. |
| 23:10.30 | TehRabbitt | hmph you have to pay USF fees? :( |
| 23:12.24 | TehRabbitt | how much are USF fees usually for outgoing termination in the US? |
| 23:12.41 | Doc | ok, progress! tell it to do MWI for both lines and now it shows no MWI light |
| 23:14.16 | Baylink | Envelope either? |
| 23:14.57 | Doc | no Envelopes, no messages on either line |
| 23:15.04 | Doc | (although line 2 does still have 1 VM waiting) |
| 23:15.59 | Baylink | Hmmm. Poly's are known to get pissed unless the XML syntax is *perfect*. Do you have both lines pointing to valid servers? |
| 23:16.11 | Doc | screw it. i'll just put in a linksys |
| 23:17.26 | Doc | syntax is fine, because everything else i change works fine |
| 23:17.35 | Baylink | Your call, but the Poly's are *really* nice phones; disproportionately nice if you pay $175 for them as I did. :-) |
| 23:18.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:18.07 | Doc | oh this isn't for me. my polycom isnt going anywhere :) |
| 23:18.22 | chazzam | polycoms can also just randomly mess up and require multiple levels of resetting the phone to get anything to work again |
| 23:18.33 | chazzam | doing the format reset doesn't actually do the two before it apparently.. |
| 23:18.36 | Baylink | That problem I haven't had... |
| 23:18.40 | TehRabbitt | Baylink: how are they nice phones? what model do you have |
| 23:18.59 | Doc | i've got a 501 at home, and a 430 at work |
| 23:19.13 | Baylink | I have entirely 601's, and they have better keyboards, and better audio, than any other phone I've ever touched. The Nortel 7215's are only close. |
| 23:19.22 | Doc | the 501 is visibly slow (plus it needs to download the full software on every boot... obviously something screwed with it) |
| 23:19.40 | Baylink | The *only* think I don't like about the physical phone is the convex side buttons, which are semi-incompatible with overhead lights. |
| 23:19.48 | Baylink | (thing) |
| 23:19.59 | TehRabbitt | hmph |
| 23:20.16 | Baylink | In short: they're *real* key telephones, as opposed to most of the shoddy crap most companies sell. |
| 23:20.32 | Baylink | This really does matter, given the audience of, y'know, office phones. |
| 23:20.50 | Baylink | They're a *cast iron* sonuvabitch to get programmed right. |
| 23:21.01 | Baylink | My present load took me almost a week to get working. |
| 23:21.07 | Baylink | *Zero* problems since. |
| 23:21.15 | Doc | once you get your head around how they work the programming isnt that hard |
| 23:21.28 | Doc | (baring problems like the one I'm seeing here obviously, but i'm sure that's a bug) |
| 23:22.14 | *** join/#asterisk homiziado (~ernestofr@88.210.101.145.rev.optimus.pt) |
| 23:22.29 | Baylink | That's the problem: lots of stuff's a bug. They didn't clean up *that* particular interface, as they didn't contemplate the aftermarket -- or contemplated it, and wanted to stiff it for money. Given how nice they are to me supporting my ViewStation MP, I choose the former... |
| 23:23.50 | TehRabbitt | voip.ms seems pretty good comapred to the flowroute service due to the fact they don't charge USF fees |
| 23:26.34 | TehRabbitt | anyone here have experiance with voip.ms? |
| 23:27.09 | Doc | rabbitt: yes |
| 23:27.46 | TehRabbitt | Doc: good? bad? how many stars would you give them? |
| 23:28.12 | Doc | somewhere between 1 and 4.5 out of 5, deending on the day |
| 23:28.16 | Doc | what do you want to use them for? |
| 23:28.47 | [TK]D-Fender | TehRabbitt: Generally the reports have been good |
| 23:29.09 | TehRabbitt | Doc: outgoing calls from my * box to PTSN |
| 23:29.12 | Doc | generally i'd say they are great, but with a few catches... |
| 23:29.23 | TehRabbitt | what catches? |
| 23:29.28 | Doc | first catch is that they dont do auto-topup in any shape or form |
| 23:29.29 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:29.39 | TehRabbitt | so basically when you're out of minutes, you're out? heh |
| 23:29.50 | TehRabbitt | (you have to log in and add more manually?) |
| 23:29.51 | Doc | so you need to manually go and recharge your acct when you get low. that may or may not be an issue depending on what you're using it for |
| 23:29.59 | TehRabbitt | yea prob won't be an issue lol |
| 23:30.33 | Doc | second is that their "value" routes to some countries are occasionally screwy (calls fail, bad quality, etc). once you report it they are fairly quick to fix it tho |
| 23:30.49 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 23:30.51 | TehRabbitt | oh :-\ how bout US domestic calling? |
| 23:30.57 | TehRabbitt | thats all i'll be using lol\ |
| 23:31.03 | Doc | no issues that i've heard of domestically |
| 23:31.09 | TehRabbitt | CID? |
| 23:31.15 | TehRabbitt | passes ok? |
| 23:31.19 | Doc | in or out? |
| 23:31.23 | TehRabbitt | out |
| 23:31.31 | Doc | as things stand today, you can set CID to whatever you want |
| 23:31.48 | Doc | although it'll be interesting to see if they change that with the recent law changes over faking caller-id |
| 23:32.23 | TehRabbitt | hm what i mean is i've heard some providers allow CID to pass some days showing the CID you specify and on other days, the CID of the trunk etc |
| 23:32.27 | Doc | anyway, this reminds me that i need to go give them some more money... |
| 23:32.32 | TehRabbitt | lol |
| 23:32.43 | TehRabbitt | i'm guessing minimum increment with them is 25? |
| 23:32.50 | Doc | got the "less than $20" email this morning |
| 23:33.07 | Doc | i forget... for work i do a few hundred at a time. for home i dont remember the last time i topped it up |
| 23:33.16 | TehRabbitt | hmph cool |
| 23:34.06 | TehRabbitt | Doc: they are reliable though? |
| 23:34.12 | Baylink | TehRabbitt: Source? I know some people pass what you send, and some don't, but I'd never heard of anyone being non-deterministic about it. |
| 23:34.18 | Doc | other than the US problems, i've been happy |
| 23:34.34 | TehRabbitt | Doc: what US problems lol |
| 23:34.35 | Doc | you can register to multiple locations too, with failover |
| 23:34.46 | Doc | err.. other than the (occasional) international problems i mean |
| 23:34.51 | TehRabbitt | ohh lol |
| 23:34.57 | TehRabbitt | so US shouldn't be an issue? |
| 23:35.20 | Doc | i've been using them for > 1 year, and for US it's been great |
| 23:35.27 | Doc | and as they are canadian they dont enforce E911 |
| 23:35.27 | TehRabbitt | aightt |
| 23:39.54 | *** join/#asterisk Systemt` (~lol@89-138-105-197.bb.netvision.net.il) |
| 23:40.03 | Systemt` | hey :) |
| 23:40.12 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 23:41.12 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 23:49.20 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 23:49.24 | TehRabbitt | anyone know of another termination provider that's good other than voip.ms since apparently my account needs manual activation and can take up to 3-4 days to be activated? lmao |
| 23:51.05 | [TK]D-Fender | TehRabbitt: You in the kind of rush thats worth it? |
| 23:51.24 | boodu | hello |
| 23:52.02 | TehRabbitt | [TK]D-Fender: idk finals start thursday, and I know once that happens I wont have free time until at least next thursday |
| 23:52.13 | *** join/#asterisk Yudaisrael1984 (~Yuda@77.127.144.138) |
| 23:52.42 | Yudaisrael1984 | anyone ever see this error before? [2010-05-05 02:52:35] NOTICE[3071]: chan_sip.c:13885 handle_request_invite: Call from '5001' to extension '18005558355' rejected because extension not found. |
| 23:52.56 | TehRabbitt | then again, I *still* can't get SIP clients from outside my network to be able to register / voice to work :-\ |
| 23:53.04 | TehRabbitt | so perhaps fix that first |
| 23:53.23 | leifmadsen | Yudaisrael1984: yes, it is a common error when your request is entering a context that does not contain extension 18005558355 |
| 23:53.39 | Yudaisrael1984 | its supposed to be dialing out |
| 23:53.40 | leifmadsen | Yudaisrael1984: using 'sip set debug on' to debug the problem |
| 23:53.47 | Yudaisrael1984 | i am |
| 23:53.58 | leifmadsen | Yudaisrael1984: your request is still going to enter the dialplan prior to calling Dial() |
| 23:54.12 | leifmadsen | phone request --> asterisk --> enter context --> Dial() --> outbound call |
| 23:54.32 | Yudaisrael1984 | has anyone heard of mor |
| 23:54.33 | Yudaisrael1984 | ? |
| 23:54.42 | TehRabbitt | [TK]D-Fender: http://pastebin.com/kLz3NUyg |
| 23:54.45 | TehRabbitt | any ideas? |
| 23:55.24 | TehRabbitt | they can connect, they can make calls, but I can't call them / I can't hear them but they can hear me |
| 23:56.27 | CatLynx | sounds like firewall issue or codec |
| 23:56.52 | TehRabbitt | CatLynx: wanna try to connect? heh use the credentials in the pastebin above and host: thoth.tenehawk.com |
| 23:57.11 | TehRabbitt | see if it'll let you register... yesterday nobody could :-\ |
| 23:57.20 | CatLynx | I can't at this moment, busy working on stuff |
| 23:57.25 | TehRabbitt | oh |
| 23:57.41 | Yudaisrael1984 | thanks i found it |
| 23:57.45 | Yudaisrael1984 | and found my mistake |
| 23:57.48 | CatLynx | trying to make it out the door by 5:30 :) |
| 23:57.48 | Yudaisrael1984 | arghhhhhhhhhhhhh |
| 23:58.01 | TehRabbitt | lol anyone wanna try to register and see if it works? |
| 23:58.31 | ChannelZ | same as last night? |
| 23:59.09 | TehRabbitt | ChannelZ: Yep |
| 23:59.15 | TehRabbitt | host: thoth.tenehawk.com |
| 23:59.18 | TehRabbitt | User: 300 |
| 23:59.20 | TehRabbitt | Secret: 300 |
| 23:59.46 | ChannelZ | hey looks like you finally got it |
| 23:59.52 | TehRabbitt | it's working? |
| 23:59.56 | TehRabbitt | try calling extension 500 |
| 23:59.57 | ChannelZ | well I registered |
