| 00:03.13 | *** join/#asterisk daven (n=dave@145.175.adsl.brightview.com) |
| 00:03.51 | dhahn | Nvr Mind - got it |
| 00:04.20 | *** part/#asterisk daven (n=dave@145.175.adsl.brightview.com) |
| 00:04.36 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 00:06.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:11.52 | linagee | wtf? is connect/voicepulse down?? |
| 00:12.00 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 00:14.43 | linagee | ack. that totally sucks |
| 00:14.50 | rob0 | robin_sz: /sbin/modinfo zaptel |
| 00:14.59 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 00:15.02 | `Tingles` | can someone tell me where this is from.. is it my asterisk box trying to connect and timing out or is it the other connection connecting and timing out? |
| 00:15.02 | `Tingles` | Sep 27 07:37:44 NOTICE[16508]: chan_sip.c:5372 sip_reg_timeout: -- Registration for 'matt@XXX.XXX.XXX.XXX' timed out, trying again (Attempt #1) |
| 00:15.05 | linagee | my faith in voip is going down the drain |
| 00:15.13 | *** join/#asterisk Rahail (i=rahail1@209.19.88.243) |
| 00:15.15 | rene1 | hello all |
| 00:15.39 | linagee | even if i had more than one outbound provider, what about my inbound number? |
| 00:15.46 | `Tingles` | nm i was a retard... |
| 00:16.09 | linagee | wtf? |
| 00:16.17 | linagee | that's wierd... i can still call going outbound.... |
| 00:16.38 | linagee | my inbound isn't working then |
| 00:16.42 | *** join/#asterisk X-Rob (n=rob-x@vg28.vodafone.com.au) |
| 00:16.58 | rene1 | small question: which is a more reliable way of doing screen popup for agents based on dialed/called number (IAX SendURL, AMI event parsing, or DialPlan i.e. Local Channel Agents and mysql |
| 00:18.34 | *** join/#asterisk Roguecmdr (n=roguecmd@145.175.adsl.brightview.com) |
| 00:19.09 | *** join/#asterisk daven (n=dave@145.175.adsl.brightview.com) |
| 00:19.17 | Roguecmdr | hey daven |
| 00:19.21 | daven | hi |
| 00:19.49 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 00:20.42 | Roguecmdr | Could some help me configure freepbx with the line .. i got a server finished and configured |
| 00:21.14 | Roguecmdr | i got a sip registeed.. now im trying to get out on the line and could use some help please |
| 00:21.42 | tuck3r | Roguecmdr: wrong channel. go to #freepbx |
| 00:21.58 | rene1 | Qwell: hey |
| 00:22.06 | Roguecmdr | thanks tuck3r |
| 00:22.22 | *** part/#asterisk Roguecmdr (n=roguecmd@145.175.adsl.brightview.com) |
| 00:23.07 | *** part/#asterisk daven (n=dave@145.175.adsl.brightview.com) |
| 00:23.08 | hmmhesays | nice 45 minute workout on the guitar |
| 00:23.15 | hmmhesays | i should really play more though |
| 00:24.20 | *** join/#asterisk haker0 (n=haker0@unaffiliated/fakhir) |
| 00:24.49 | *** part/#asterisk haker0 (n=haker0@unaffiliated/fakhir) |
| 00:26.00 | rene1 | hmmhesays: are you into screen pops using *? |
| 00:26.17 | MikeJ | hmmhesays I hear is into some really weird things ;) |
| 00:26.27 | MikeJ | wassup hmmhesays.. long time no talk |
| 00:27.19 | rene1 | this place is a bit dead. it used to be more active |
| 00:28.13 | joako | Yep, voicepulse inbound is down.... |
| 00:28.18 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:29.04 | SwK | screen pops with asterisk? |
| 00:29.58 | rene1 | SwK |
| 00:29.59 | tzanger | SwK: snapanumber.com? |
| 00:30.00 | SwK | yes |
| 00:30.02 | rene1 | well with a web browser |
| 00:30.25 | rene1 | triggered by asterisk incoming calls to a queue |
| 00:30.57 | rene1 | right now i am using agentconnect event in AMI, it is not in sync with the call connection actual event |
| 00:31.41 | rene1 | and agentcomplete event should happen on all call terminations but it happens only in ~95% i guess |
| 00:32.57 | rene1 | i have been told to use and iax phone and iax sendurl, but that dont work for me at the time |
| 00:33.10 | rene1 | and the third option would be to use local channels |
| 00:33.12 | adamowitz | does anyone here know anything about PsipTN or TeLTeL? |
| 00:34.24 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.104.247.2) |
| 00:34.52 | rene1 | and have something like this in the local channel: mysql(insert into agents (activecall) values (${callerid})) and just poll with javascript from the browser to see if the agent has a new call or not |
| 00:38.06 | rene1 | SwK: any comment? |
| 00:39.53 | joako | rene1: Have you looked at the Flash Panel, I think it could do what you want |
| 00:40.28 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:40.37 | teknoprep | anyone on voicepulse not have incomming calls right now? |
| 00:40.55 | DrukenHME | got a number? i'll call if you want... |
| 00:41.01 | teknoprep | its not working |
| 00:41.03 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 00:41.10 | joako | Well that makes three of us |
| 00:41.30 | teknoprep | i see the voicepulse IRC guy here |
| 00:41.33 | teknoprep | fix it fool |
| 00:41.44 | teknoprep | i was in the middle of explaining the new system |
| 00:41.50 | teknoprep | and all of a sudden everything stops working |
| 00:41.52 | teknoprep | thanx ass |
| 00:42.15 | rene1 | joako: well it sorts of but not really |
| 00:42.16 | carrar | hahahah |
| 00:42.30 | carrar | no pulse? |
| 00:42.37 | teknoprep | no voicepulse again |
| 00:42.42 | teknoprep | do they suck like really bad or something? |
| 00:45.05 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 00:45.16 | joako | Yes. |
| 00:45.35 | joako | For some reson the phone #'s I ported to them, the DTMF doesnt work from Sprint phones.... |
| 00:45.51 | teknoprep | joako really |
| 00:46.00 | teknoprep | one of my numbers doesn work with comcast voip phones |
| 00:46.09 | teknoprep | i am leaving voice pulse if this happens again |
| 00:46.10 | teknoprep | EVER |
| 00:46.29 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:47.23 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 00:47.38 | linagee | what happened to voicepulse? |
| 00:48.06 | teknoprep | it sucks |
| 00:48.10 | teknoprep | or they went out of business |
| 00:48.22 | teknoprep | i think they went out of business... took our money and ran |
| 00:48.31 | linagee | lol. i was just about to say |
| 00:49.06 | linagee | teknoprep: do you get a doo doo doo. "No Answer" when calling? |
| 00:49.13 | teknoprep | no |
| 00:49.16 | teknoprep | i get silence |
| 00:49.18 | teknoprep | nothing |
| 00:49.43 | teknoprep | i get a busy signal about 30 seconds into it |
| 00:50.05 | teknoprep | how many ppl in here on VoicePulse? |
| 00:50.46 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:51.15 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:52.19 | DrukenHME | anyone know how to reset a pap2-na without the admin password? |
| 00:52.47 | predder | did you need it to continue working? My method involved a hammer.. |
| 00:53.27 | DrukenHME | uhmm... yeah i'd like it to work... |
| 00:53.51 | DrukenHME | i already got one that's pooched.... |
| 00:55.01 | selsky | anyone get the imap storage backend for voicemail working? |
| 00:55.48 | *** join/#asterisk ComputerGuySteve (n=steve@wsip-70-183-114-254.no.no.cox.net) |
| 00:55.50 | ComputerGuySteve | hey guys |
| 00:55.58 | ComputerGuySteve | whats up with the UserEvent app |
| 00:56.10 | ComputerGuySteve | the wiki says to use ^ for new lines |
| 00:56.16 | ComputerGuySteve | but it takes it litterly |
| 00:58.00 | joako | I think voicepulse doesn't have too many customers, its always the same few people answering calls and emails. and besides even if you just email them they usally end up calling you to say the same thing the email does |
| 00:58.23 | linagee | can someone please help me with my voip? it used to work, but then my provider went down. nothing on the debug output when i dial in. :( |
| 00:58.30 | ComputerGuySteve | joako: whats wrong with your voicepulse |
| 00:58.44 | joako | Well right now the voicepulse inbound is down |
| 00:58.50 | ComputerGuySteve | voicepuslse rocks |
| 00:58.54 | linagee | joako: yes. it is for me too |
| 00:58.59 | linagee | ComputerGuySteve: are you using voicepulse? |
| 00:59.10 | ComputerGuySteve | absoultley |
| 00:59.19 | linagee | ComputerGuySteve: try to call your inbound number (with a cell or something) |
| 00:59.50 | DrukenHME | you guys toronto ? |
| 00:59.59 | teknoprep | ComputerGuySteve, you on voicepulse? |
| 00:59.59 | linagee | ComputerGuySteve: anything? |
| 01:00.08 | linagee | teknoprep: it seems lots of us are |
| 01:00.12 | ComputerGuySteve | oh |
| 01:00.19 | teknoprep | lol |
| 01:00.23 | teknoprep | good it isn't me |
| 01:00.24 | joako | however, voicepulse outbound does work |
| 01:00.26 | teknoprep | i am just happy to hear this |
| 01:01.06 | ComputerGuySteve | oh yeah... |
| 01:01.10 | ComputerGuySteve | we do got a problem |
| 01:01.23 | linagee | joako: yes. here too. |
| 01:01.25 | teknoprep | bullshit man.. i am pissed |
| 01:01.35 | ComputerGuySteve | linagee: did you get a hold of them ? |
| 01:01.37 | teknoprep | probably won't be fixed till tommorow |
| 01:01.44 | linagee | teknoprep: the guy should at least have his system page him when someone calls in pissed |
| 01:01.52 | teknoprep | probably some rogue idiot IT tech who thought a system change of sorts would do good |
| 01:01.54 | ComputerGuySteve | oh yeah, there tech support it gone for the night |
| 01:02.03 | linagee | ComputerGuySteve: i sent an email saying my server is timing out on the registration |
| 01:02.13 | linagee | ComputerGuySteve: gone for the night? hahaha |
| 01:02.22 | linagee | ComputerGuySteve: you don't "gone for the night" with phones. :p |
| 01:02.29 | linagee | i would never do such a thing |
| 01:02.44 | ComputerGuySteve | linagee: yeah, i guess you're right |
| 01:02.48 | teknoprep | lets all leave voicepulse |
| 01:02.51 | ComputerGuySteve | well did you try calling |
| 01:02.52 | teknoprep | skrew this crap |
| 01:03.08 | linagee | teknoprep: tell me someone else who charges $11/mo though. and iax support. seriously. :-/ |
| 01:03.19 | teknoprep | agreed |
| 01:03.25 | teknoprep | but you have to deal with this |
| 01:03.25 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 01:03.35 | droops | hey if i set up a con ference in marked mode, how do i set one of the conf attendies as the marked user |
| 01:03.39 | linagee | did anyone bother to try the main voicepulse phone number? i wonder if their non connect.voicepulse customers are affected. LOL |
| 01:04.30 | linagee | nope. same number on the regular customer page |
| 01:04.35 | teknoprep | linagee, its the same number |
| 01:04.36 | teknoprep | lol |
| 01:04.41 | linagee | maybe new jersey blew up |
| 01:04.46 | teknoprep | lol |
| 01:04.51 | teknoprep | good |
| 01:04.53 | teknoprep | place is a dump |
| 01:05.43 | linagee | i don't really see anything like that... hrm |
| 01:07.06 | ComputerGuySteve | the inbound service must of just stopped working cause I was getting calls about it hour ago |
| 01:07.17 | ComputerGuySteve | does anyone know about the UserEvent app ? |
| 01:07.51 | linagee | ComputerGuySteve: people are calling you now? |
| 01:07.54 | linagee | yikes |
| 01:08.09 | linagee | "ComputerGuySteve! why isn't my phone working!" heh |
| 01:08.27 | *** part/#asterisk Trazz (i=Trazz@c-71-57-36-106.hsd1.il.comcast.net) |
| 01:08.41 | ComputerGuySteve | linagee: no.. mine is broke too |
| 01:08.50 | linagee | ah |
| 01:09.38 | *** join/#asterisk Trazz (n=Trazz@c-71-57-36-106.hsd1.il.comcast.net) |
| 01:09.44 | ComputerGuySteve | does anybody use UserEvent |
| 01:09.45 | ComputerGuySteve | <PROTECTED> |
| 01:10.01 | linagee | never heard of it. agi? |
| 01:10.07 | ComputerGuySteve | no |
| 01:10.16 | ComputerGuySteve | its used to specify custom Event |
| 01:10.18 | ComputerGuySteve | its used to specify custom Events |
| 01:10.22 | ComputerGuySteve | to the event manager |
| 01:10.27 | ComputerGuySteve | I use YaaCID |
| 01:10.37 | ComputerGuySteve | and now I need to do somethign more custom with it |
| 01:10.57 | ComputerGuySteve | and I need to sepcify more then one line for the UserEvent body |
| 01:11.04 | ComputerGuySteve | the wiki says how to |
| 01:11.12 | ComputerGuySteve | but it doesn't work right |
| 01:14.50 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 01:15.23 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:15.33 | linagee | teknoprep: you would think a voicepulse technician could just call into their system, dial some secret numbers, and forward all calls to something |
| 01:15.40 | linagee | instead of just "drop all calls" |
| 01:16.39 | QbY_ | Does anyone here use Voicepulse? And currently receiving calls? |
| 01:16.55 | ComputerGuySteve | QbY_: yeah, its down for us too |
| 01:17.03 | Nivex | QbY_: stby lemme try a known VP node |
| 01:17.04 | QbY_ | damn, this sucks.. no way to call them |
| 01:17.14 | ComputerGuySteve | QbY_: linagee hacked thier servers and fucked it up |
| 01:17.29 | linagee | QbY_: yes. everyone here uses voicepulse. and yes, they're down. |
| 01:17.37 | QbY_ | great. |
| 01:17.40 | linagee | someone needs to change the topic. "Yes, voicepulse is down" |
| 01:17.42 | linagee | QbY_: i know |
| 01:17.52 | *** join/#asterisk truespace (i=truespac@S010600301b1eb05e.gv.shawcable.net) |
| 01:17.54 | linagee | QbY_: someone near the area go break into the building. anyone live near NJ? |
| 01:18.09 | truespace | i can launch a cruise missile if you want |
| 01:18.16 | linagee | truespace: you too?? |
| 01:18.19 | linagee | geez |
| 01:18.25 | truespace | ya i bought it off ebay |
| 01:18.40 | linagee | no, i mean you use voicepulse too? |
| 01:19.07 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 01:19.13 | truespace | voicepulse/ |
| 01:19.16 | truespace | no |
| 01:19.18 | truespace | i cant even read |
| 01:19.34 | linagee | ??? |
| 01:19.41 | linagee | truespace: how are you even getting by? |
| 01:19.53 | QbY_ | now this is odd.. * tells me it is regsitered with one of their servers, but the ip is different from what dns is reporting |
| 01:20.16 | truespace | hey can someone help me? i need to set up asterix NOW |
| 01:20.24 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 01:20.25 | QbY_ | i want to get laid NOW |
| 01:20.34 | Nivex | QbY_: you too? |
| 01:20.35 | Nivex | :) |
| 01:20.36 | QbY_ | scratch that.. i need to get laid NOW |
| 01:20.52 | jbroome | voicepulse01 is now unreachable. AWESOME |
| 01:20.57 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 01:20.57 | truespace | just put some ice on your hand till it goes numb |
| 01:21.02 | jbroome | stranger! |
| 01:21.08 | linagee | truespace: go use a trixbox |
| 01:21.12 | QbY_ | PING csw004.voicepulse.com (64.61.93.87) 56(84) bytes of data. |
| 01:21.12 | QbY_ | 64 bytes from static-64-61-93-87.isp.broadviewnet.net (64.61.93.87): icmp_seq=0 ttl=56 time=2.69 ms |
| 01:21.15 | Twister | is there any way to change the volume of a sip trunk, my calls seem awful quiet sounding on 2 diffrent phones |
| 01:21.20 | linagee | QbY_: csw004? |
| 01:21.22 | truespace | trixbox? |
| 01:21.33 | linagee | QbY_: maybe voicepulse was hacked and the hackers are providing outbound dialing |
| 01:21.42 | QbY_ | linagee.. that's what resolved when i pinged connect01 |
| 01:21.55 | linagee | QbY_: do you know the IP before? |
| 01:22.03 | QbY_ | nope |
| 01:22.12 | truespace | I got to the point where it says "asterisk ready" now I am stuck |
| 01:22.24 | linagee | truespace: "please enter command" |
| 01:22.38 | linagee | truespace: type in, "ok, now do the PBX thing" |
| 01:22.42 | linagee | lol |
| 01:22.43 | QbY_ | HAH |
| 01:22.50 | truespace | k |
| 01:23.04 | linagee | please tell me you didn't actually just do that |
| 01:23.16 | truespace | hold on i am a slow typist |
| 01:23.24 | linagee | lol |
| 01:24.25 | truespace | i have a D.I.D number how do i make that work |
| 01:24.40 | linagee | truespace: a DID over...? T1? |
| 01:24.52 | QbY_ | you'd think for a company who caters to the asterisk userbase would have a rep hanging out in here.. |
| 01:25.04 | linagee | QbY_: or at least someone on a pager |
| 01:25.10 | *** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net) |
| 01:25.15 | QbY_ | yeah.. answering service or somethin |
| 01:25.15 | linagee | or some extension that rings their cell |
| 01:25.55 | QbY_ | i hate it, it never fails, the moment i drop $500 on something (my voicepulse accounts) it dies.. |
| 01:26.00 | QbY_ | reminds me of that stripper in vegas |
| 01:26.03 | truespace | i'm gonna go nuts |
| 01:26.10 | QbY_ | our helpdesk is dead |
| 01:27.00 | linagee | QbY_: wow. hooked up to voicepulse???? |
| 01:27.05 | linagee | yikes |
| 01:27.18 | QbY_ | yeah, they wanted to go the cheap route |
| 01:27.27 | QbY_ | only part of it is down--the other part is through kall8.. |
| 01:27.31 | linagee | QbY_: you had $500 in the account?? they probably took off to vegas! |
| 01:27.33 | QbY_ | i have cell phone numbers there |
| 01:27.39 | QbY_ | hahaha |
| 01:27.45 | linagee | QbY_: you have their cell phone numbers? |
| 01:27.47 | linagee | good. call! |
| 01:27.48 | *** join/#asterisk SomethingISODD (n=Dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 01:27.55 | QbY_ | linagee.. for Kall8 |
| 01:28.00 | SomethingISODD | hello can you send sms messages over voip? |
| 01:28.16 | linagee | SomethingISODD: i don't think so |
| 01:28.22 | linagee | SomethingISODD: SMS is out of band. |
| 01:28.33 | linagee | (not in the analog/voice channel) |
| 01:28.38 | linagee | afaik |
| 01:29.03 | SomethingISODD | linagee, true but i had a company a few years ago buying usa/canada termination off of me and they where doing it |
| 01:29.12 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 01:29.40 | linagee | SomethingISODD: not sure then. you can probably call a gateway and enter the SMS in that way |
| 01:30.11 | linagee | somehow signal the gateway your SMS (modem? DTMF??) and have it rela |
| 01:30.13 | linagee | y |
| 01:30.14 | SomethingISODD | hrm. i guess this is time for some searching |
| 01:31.59 | QbY_ | damn.. i was gonna go out on a limb and look for voice pulse's corporate information on the secretary of state site.. but you gotta pay for it |
| 01:32.33 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 01:32.44 | teknoprep | anyone else down still on VoicePulse inbound ? |
| 01:33.10 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:33.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:33.39 | jbroome | i'm on vp02 ok, but vp01 is still unreachable |
| 01:33.51 | QbY_ | jbroome.. are you getting calls through 02? |
| 01:33.56 | teknoprep | you were able to recieve inboud on 02 ? |
| 01:33.58 | linagee | jbalcomb: vp02?? |
| 01:34.11 | teknoprep | connect02.voicepulse.com |
| 01:34.15 | teknoprep | yup |
| 01:34.17 | QbY_ | hehe |
| 01:34.23 | QbY_ | i've been connected to it--but not calls |
| 01:34.27 | jbroome | QbY_: testing |
| 01:35.01 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-41-14.dsl.tul2ok.sbcglobal.net) |
| 01:35.35 | linagee | hrm. 02 seems to not work either |
| 01:35.36 | jbroome | got nothing on incoming |
| 01:35.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:35.53 | linagee | jbroome: you too?? |
| 01:36.00 | jbroome | Nothing on the VP page under service/or upgrade notices. |
| 01:36.12 | jbroome | linagee: correct. |
| 01:36.25 | linagee | jbroome: like i said, mabe NJ blew up? |
| 01:36.38 | teknoprep | no i just downloaded mIRC from NJ mirror |
| 01:36.43 | teknoprep | lol |
| 01:37.27 | jbroome | linagee: i noticed recently the service notices mentioned they were adding a 03 and 04, but I haven't heard anything about it |
| 01:37.49 | linagee | QbY_: probably one 1U poweredge 1550. LOL |
| 01:38.08 | linagee | nope. 03 doesn't work either for me |
| 01:38.20 | linagee | jbroome: 04 seems to point to 01 or something |
| 01:38.22 | QbY_ | anyone ever ported away from vp? |
| 01:38.24 | linagee | lol |
| 01:38.50 | jbroome | connect 04 is an alias for connect2 according to host |
| 01:38.50 | linagee | QbY_: you could port to zap? but then you are stuck at one trunk unless you have a hunt group |
| 01:38.55 | linagee | es |
| 01:38.57 | linagee | y |
| 01:39.41 | linagee | 04 goes to 07, 02 goes to 10, and 03 goes to 08. lol |
| 01:40.01 | linagee | (yes, i realize there's nothing above 04) |
| 01:40.39 | jbroome | So, any alternatives to VP out there? this has been a problem recently |
| 01:42.12 | jbroome | Ok, according to one of my servers, vp01 went unreachable at 20:02:31 Eastern |
| 01:44.06 | truespace | oops i did it again |
| 01:45.19 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 01:45.47 | linagee | jbroome: how did you have it testing? it still pings for me... |
| 01:46.09 | linagee | voicepulse should give all of it's users a $20 credit |
| 01:46.40 | *** join/#asterisk Crescendo (n=GWing@cpe-024-211-177-002.nc.res.rr.com) |
| 01:46.55 | Crescendo | What ports do I need to forward to access SIP on asterisk remotely? |
| 01:48.48 | truespace | 6 and 4 i would guess if i had to |
| 01:49.41 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 01:50.13 | linagee | port 6 and 4? lol |
| 01:50.48 | truespace | oh ya, your write its 5 and 7 right? |
| 01:51.10 | truespace | never mind im' thinking of skype |
| 01:52.03 | truespace | hey can someone try calling me? |
| 01:52.10 | truespace | jmorken@213.91.9.210 |
| 01:52.11 | truespace | SIP |
| 01:52.16 | truespace | to see if it works |
| 01:53.16 | truespace | linagee call me |
| 01:53.31 | linagee | only IAX. SIP sucks |
| 01:53.38 | linagee | my NAT doesn't like it |
| 01:53.46 | jbroome | no ones nat likes it |
| 01:54.11 | linagee | exactly why you should just stick with IAX |
| 01:54.17 | linagee | (or iax2 or whatever) |
| 01:54.31 | truespace | ok call me here then with IAX |
| 01:54.41 | truespace | 24.68.151.87 |
| 01:55.05 | linagee | so we all paid $11 this month for nothing. :-/ |
| 01:55.06 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 01:55.14 | QbY_ | HA! |
| 01:55.18 | QbY_ | I've paid $500 |
| 01:55.19 | Crescendo | ...? |
| 01:55.25 | linagee | QbY_: for the month!?!?!? |
| 01:55.26 | linagee | yikes |
| 01:55.27 | linagee | ! |
| 01:55.32 | QbY_ | yeah |
| 01:55.34 | linagee | whoa |
| 01:55.41 | QbY_ | its cheaper than doing market activations all over the country and having to buy dids |
| 01:55.48 | linagee | QbY_: indeed |
| 01:56.05 | linagee | QbY_: i think we should all get their pager numbers as part of the settlement. :) |
| 01:56.05 | truespace | you could have bought a lot of cruise missles off ebay for that |
| 01:56.08 | QbY_ | however, i am porting all of my VP numbers away |
| 01:56.13 | linagee | QbY_: wow! |
| 01:56.17 | linagee | QbY_: just because of tonight? |
| 01:56.24 | linagee | lol! |
| 01:56.33 | ComputerGuySteve | can one port a number from voicepulse ? |
| 01:56.37 | linagee | truespace: cruise missles for $500? not even |
| 01:56.43 | QbY_ | linagee.. yeah.. its the fact that there is no way to get ahold of them.. |
| 01:56.45 | QbY_ | i like service |
| 01:57.09 | QbY_ | no information on their website, as to what's up |
| 01:57.17 | QbY_ | and their redundant servers, are obviously--not redundat |
| 01:57.48 | linagee | QbY_: i don't like when you google search for "2227 US Highway One new jersey", you get "pyramid schemes" in the results. heh. (could be a mailboxes etc or something?) |
| 01:57.52 | ComputerGuySteve | does anyone here use UserEvent ? |
| 01:57.59 | ComputerGuySteve | in thier dialplan ? |
| 01:58.00 | linagee | QbY_: lol. yes. they are all on the same subnet too! i noticed that. |
| 01:58.08 | linagee | (or so it seems) |
| 01:58.20 | QbY_ | linagee.. are you shitting me, the pyramid scheme? |
| 01:58.25 | jbroome | fcking odd: I can dial out from one of my customers' * machine, but not mine |
| 01:58.32 | linagee | no. lol. like i said, it's probably a mail boxes etc. |
| 01:58.38 | linagee | http://www.google.com/search?hl=en&q=2227+US+Highway+One+new+jersey&btnG=Google+Search |
| 01:58.54 | truespace | what do i type in to call another Asterisk server using IAX? |
| 01:58.56 | jbroome | and incoming on the one i can dial out from is no go |
| 01:58.58 | linagee | i was trying to find out if it was a mail boxes etc. but apparently, that's hard to do. |
| 02:00.13 | QbY_ | maybe they've packed up and left |
| 02:00.21 | QbY_ | or, worse, didn't pay their bill to GX |
| 02:00.31 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 02:02.42 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:03.39 | linagee | GX? |
| 02:03.39 | linagee | is that their provider? |
| 02:04.07 | *** join/#asterisk teknoprep (i=teknopre@unaffiliated/teknoprep) |
| 02:04.13 | teknoprep | so voicepulse is still down aye ? |
| 02:04.16 | QbY_ | Global Crossings.. |
| 02:04.26 | QbY_ | that's who all of our DIDs from VP actually belong to |
| 02:04.38 | truespace | i type "help" at the asterix command line interface (CLI) in the asteriskwin32 pbx window, and it does nothing |
| 02:04.40 | linagee | hrm... it seems 732-339-xxxx has several providers. Verizon NJ, Nextel, Snip Link, Paetec Communications |
| 02:05.04 | truespace | try les.net for DID's... |
| 02:05.09 | linagee | 858-605-xxxx (my prefix) has Paetec Communications listed. |
| 02:05.17 | linagee | i think Paetec Communications is their provider. |
| 02:05.53 | linagee | QbY_: did you say GX? isn't it Paetec Communications? |
| 02:06.15 | teknoprep | how is LES.net ? |
| 02:06.28 | linagee | never heard of them |
| 02:06.31 | QbY_ | me neither |
| 02:06.47 | QbY_ | linagee.. they must have multiples.. all of ours come from global crossing |
| 02:06.55 | QbY_ | according to http://fonefinder.net/ |
| 02:07.11 | linagee | QbY_: what is your area code and prefix? |
| 02:07.14 | tengulre11 | Hi,all |
| 02:07.38 | QbY_ | 360-219-1 |
| 02:07.51 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 02:07.51 | linagee | QbY_: hrm. mine says Paetec. heh |
| 02:07.58 | QbY_ | eh? |
| 02:08.03 | QbY_ | 253-444-1 |
| 02:08.07 | QbY_ | where are you searching? |
| 02:08.13 | linagee | same thing |
| 02:08.15 | linagee | same page |
| 02:08.17 | QbY_ | FoneFinder reports Global |
| 02:08.28 | linagee | try 858-605-0 |
| 02:08.29 | tengulre11 | I dial out with X100P, but it must cross a telelphony PBX, and the PBX need dial 0 first ,how to write the rule in /etc/asterisk/extensions.conf? |
| 02:08.40 | truespace | les.net seems fine so far, I have a DID number there, but i dont know how to set it up |
| 02:08.47 | linagee | QbY_: i mean for my number it says paetec. yes, your says GX |
| 02:08.52 | QbY_ | oh |
| 02:09.09 | QbY_ | the beer has me fucked up |
| 02:09.11 | linagee | QbY_: maybe GX has some sort of monopoly in washington. heh |
| 02:09.17 | truespace | i type in "help iax" and it says "no such command 'iax'" |
| 02:09.20 | QbY_ | "seems fine so far, i have a DID number there, but i don't know how to set it up" |
| 02:09.26 | QbY_ | truespace.. try iax2 |
| 02:09.31 | truespace | ok |
| 02:09.42 | truespace | thx |
| 02:10.04 | *** join/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 02:10.30 | teknoprep | i am wondering |
| 02:10.32 | truespace | what is the iaxfwd users? |
| 02:10.35 | teknoprep | does voicepulse even know its down? |
| 02:10.50 | QbY_ | teknoprep.. my guess.. nope.. |
| 02:10.55 | linagee | QbY_: interesting... paetec has their own voip service. lol |
| 02:10.57 | jbroome | There's no one at the office |
| 02:11.00 | linagee | i think |
| 02:11.05 | linagee | jbroome: which? |
| 02:11.07 | truespace | i think i will go shoot myself |
| 02:11.16 | truespace | cya |
| 02:11.17 | fjean5 | hi guys, anyone knows how to properly configure sendEmail in voicemail.conf without going into the code source ? |
| 02:11.18 | linagee | truespace: are you really losing that much money per hour?? |
| 02:11.19 | linagee | yikes |
| 02:11.24 | truespace | be back in a few |
| 02:11.33 | linagee | truespace: you will be back, right? :) |
| 02:11.40 | teknoprep | sounds fun truespace ... if you doit right... you'll only be able to doit once... try and write about it when you doit too.. so other ppl know what its like right before you doit... since you can't tell us later |
| 02:11.42 | QbY_ | *BANG* |
| 02:11.55 | QbY_ | leave a HOWTO |
| 02:12.05 | teknoprep | lol |
| 02:12.11 | linagee | QbY_: did you sell voicepulse to other customers, or do you really use $500 of phone? |
| 02:12.30 | linagee | QbY_: yikes. that has to suck if they start calling you up |
| 02:12.48 | QbY_ | linagee.. one of our customers runs an outsourced helpdesk.. so they need numbers all over the country.. voicepulse gives me unlimited incoming minutes with DIDs all over |
| 02:12.56 | linagee | QbY_: ouch |
| 02:13.11 | QbY_ | linagee.. and she is in pure panic.. but i gave them a toll free number to use tonight |
| 02:13.20 | linagee | wow |
| 02:13.49 | linagee | QbY_: business works fast. lol |
| 02:14.12 | teknoprep | QbY_, whats your ETA on this... have you seen them be down for this long before? |
| 02:14.35 | QbY_ | teknoprep.. i've never seen them down this long |
| 02:14.37 | linagee | maybe someone accidentally turned off the server when they left for the night. LOL |
| 02:14.39 | QbY_ | in fact, i've never seen them down |
| 02:14.47 | linagee | QbY_: never seen is key. hrm |
| 02:14.48 | QbY_ | linagee.. heheeh.. they hit the lightswitch |
| 02:14.50 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 02:14.52 | teknoprep | this is the 2nd time i have seen this |
| 02:14.57 | linagee | QbY_: i'm thinking about setting up something to monitor it too now. lol |
| 02:15.05 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 02:15.11 | linagee | QbY_: at least i will know when it's down! heh |
| 02:15.17 | QbY_ | yeah |
| 02:16.02 | linagee | QbY_: maybe someone should code something into asterisk. "when you can't register with any inbound contexts, and production = yes, then email and execute these commands" heh |
| 02:16.10 | *** join/#asterisk tuck3r_ (n=tuck3r@unaffiliated/tuck3r) |
| 02:16.12 | jbroome | vp01 just came back |
| 02:17.11 | linagee | woohoo. i'm back up |
| 02:17.27 | teknoprep | same |
| 02:17.54 | linagee | ah |
| 02:18.42 | jbroome | it's lagged to hell, but at least it's up |
| 02:18.43 | linagee | how many inbound channels do you think they have? LOL. 24? 48? |
| 02:18.48 | truespace | did you hear about the guy who shot himself twice in the head? |
| 02:18.54 | ComputerGuySteve | hey my DID is working !! |
| 02:18.54 | QbY_ | i'm hitting it a 3ms |
| 02:18.55 | linagee | jbroome: are you using PCM? lol |
| 02:19.05 | linagee | i bet it sucks when more people use PCM |
| 02:19.33 | linagee | QbY_: 3 ms? what kind of connection do you have?? |
| 02:19.43 | QbY_ | i'm co-lo'd at Voxel |
| 02:19.48 | SplasPood | Yea? |
| 02:20.07 | linagee | (cable modem) |
| 02:20.34 | SplasPood | QbY_: I'm a voxelian |
| 02:21.33 | *** part/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 02:21.45 | linagee | i have to go rr.com -> level3.net -> transbeam.com -> ??? -> cosmoweb.net -> ??? -> vp01 |
| 02:21.47 | QbY_ | SplasPood.. You work there or hosted there? |
| 02:21.59 | SplasPood | QbY_: both, actually |
| 02:22.14 | linagee | jbroome: stop using up all the vp bandwidth! hehe |
| 02:22.16 | QbY_ | cools.. you guys got it together |
| 02:22.18 | QbY_ | no complaints from me |
| 02:22.25 | SplasPood | QbY_: I like to think so :) Thank you |
| 02:22.29 | QbY_ | we are actually considering two more boxes |
| 02:22.31 | jbroome | linagee: ok, i'll share a little |
| 02:22.31 | SplasPood | QbY_: Mind if I ask what your company name is? |
| 02:22.45 | QbY_ | PM |
| 02:24.27 | truespace | what do i type into the console to make a sip call/ |
| 02:24.28 | truespace | ? |
| 02:24.40 | linagee | i just spoke with the voicepulse guy |
| 02:24.48 | linagee | he apologizes. :) |
| 02:24.52 | directory | profusely? |
| 02:24.53 | jbroome | now that your phone is back up? :) |
| 02:24.57 | linagee | directory: *shrug* |
| 02:25.06 | ComputerGuySteve | linagee: what did you say was the prbolem ? |
| 02:25.08 | linagee | i didn't get any caller id |
| 02:25.30 | linagee | ComputerGuySteve: i told him inbound calls weren't going through and it's been working before |
| 02:25.53 | ComputerGuySteve | linagee: no.. im sorry... i meant what did he say was the problem ? |
| 02:25.57 | linagee | (i think i may have said other #asterisk users were experiencing the problem too. ehh) |
| 02:26.01 | linagee | ComputerGuySteve: he didn't say. heh |
| 02:26.10 | tengulre11 | anybody know how to setting a SS7 gateway if I using E1 card? |
| 02:26.20 | linagee | ComputerGuySteve: ask _VoicePulse |
| 02:26.32 | ComputerGuySteve | can I version check here |
| 02:26.46 | ComputerGuySteve | Im running 2.12 |
| 02:26.54 | ComputerGuySteve | what about others ? |
| 02:26.55 | truespace | les.net has DID numbers for way cheaper than voicepulse btw |
| 02:27.09 | linagee | truespace: interesting... |
| 02:27.14 | linagee | truespace: 800 numbers too? |
| 02:27.17 | truespace | ya |
| 02:27.19 | ComputerGuySteve | versions please ? |
| 02:27.59 | jbroome | 1.2.10 & 1.2.12.1 |
| 02:28.30 | tengulre11 | how to dial another user with web call? |
| 02:28.45 | tengulre11 | is it need a SIP/IAX2 ocx? |
| 02:28.54 | ComputerGuySteve | jbroome: can you did a simple test on your servers please ? |
| 02:29.08 | jbroome | ComputerGuySteve: maaaaaybe |
| 02:29.33 | ComputerGuySteve | jbroome: add a line to an extension |
| 02:29.39 | pfn | so who's the preferred intl iax2 provider these days? |
| 02:29.54 | pfn | since nufone and voipjet both aren't terminating my intl calls and aren't answering my requests for support |
| 02:30.05 | linagee | pfn: yikes |
| 02:30.21 | ComputerGuySteve | jbroome: UserEvent(TestEvent|Arg1:Value1^Arg2:Value2) |
| 02:30.36 | ComputerGuySteve | and see what your manager output says |
| 02:30.51 | pfn | meh, I've sent them both email like 3 or 4 weeks back and no response |
| 02:30.57 | linagee | ouch |
| 02:31.08 | ComputerGuySteve | jbroome: do you know how to the manager interface ? |
| 02:31.15 | linagee | pfn: phones are a here and now kind of business. heh |
| 02:31.33 | jbroome | ComputerGuySteve: i do, and i don't really want to futz with dialpans that are in production |
| 02:31.45 | ComputerGuySteve | ok |
| 02:31.51 | ComputerGuySteve | do you have some test server ? |
| 02:32.01 | jbroome | not currently, no |
| 02:32.06 | ComputerGuySteve | ok thaks |
| 02:32.16 | ComputerGuySteve | can someone run this simple test ? |
| 02:33.58 | tengulre11 | >Running Asterisk Server.... |
| 02:34.12 | *** join/#asterisk Toadyus (i=Toadyus@S010600121746f9fe.mh.shawcable.net) |
| 02:34.57 | ComputerGuySteve | linagee: can you help me out with something by testing a line on your server ? |
| 02:35.35 | linagee | ComputerGuySteve: how? |
| 02:36.02 | ComputerGuySteve | add a line to some extension in your dialplan or just create a test extension that just does or two things |
| 02:36.02 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 02:36.14 | ComputerGuySteve | and find the output on your manager output |
| 02:36.44 | ComputerGuySteve | linagee: do you know how to use the manager service ? |
| 02:37.03 | linagee | ComputerGuySteve: not really. i wouldn't consider myself an asterisk expert by a long shot |
| 02:37.15 | ComputerGuySteve | oh ok. |
| 02:37.16 | ComputerGuySteve | well |
| 02:37.19 | ComputerGuySteve | its not hard |
| 02:37.35 | ComputerGuySteve | ok nevermind |
| 02:37.37 | linagee | lol |
| 02:37.51 | ComputerGuySteve | fuck ! |
| 02:37.58 | ComputerGuySteve | userEvent is broken !! |
| 02:38.46 | linagee | FSCK! les.net is no setup fee and $3.99/mo! wow |
| 02:38.55 | linagee | wowowowow. :-D :-D |
| 02:38.58 | jbroome | woah |
| 02:39.01 | linagee | nice |
| 02:39.08 | jbroome | might have to get a number for home to play with for that cost |
| 02:39.21 | linagee | maybe that's a no DID pricve |
| 02:39.39 | *** part/#asterisk QbY_ (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 02:40.32 | jbroome | linagee: http://les.net/products/product_ipdidusa.php |
| 02:40.48 | jbroome | eeenteristing |
| 02:40.51 | linagee | yep |
| 02:41.09 | linagee | it says 1 DID, doesn't really give details until you sign up i guess |
| 02:42.32 | linagee | hrm... the international destination of my choice is $0.01/min higher |
| 02:43.01 | *** join/#asterisk Star568 (n=steve@pool-71-118-163-45.lsanca.dsl-w.verizon.net) |
| 02:43.04 | linagee | in states dialing is 1.5 cents per minute. hrm |
| 02:44.12 | Star568 | i need a sample config which can tranfer PSTN calls from SER /OPENSER to Asterisk, Asterisk will terminate the calls to pstn |
| 02:44.31 | linagee | VoIP termination via SIP/H323/IAX for Canada/USA |
| 02:44.35 | linagee | so they support IAX. nice. |
| 02:46.54 | linagee | toll free number = $15/setup, $2.50/mo, $0.04/mo. not bad. or is there better? |
| 02:47.00 | linagee | er, $0.04/min |
| 02:47.06 | tengulre11 | which client using IAX?? and have source code for windows platform. |
| 02:48.21 | benjk | iaxclient |
| 02:50.09 | *** join/#asterisk LabRat021 (n=booker56@70.146.196.22) |
| 02:50.17 | LabRat021 | hi all |
| 02:50.52 | LabRat021 | does anyone here know Timo? |
| 02:52.05 | LabRat021 | hello? |
| 02:52.13 | LabRat021 | why is there so much silence in the room? |
| 02:52.37 | mog | yes |
| 02:52.50 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:52.50 | mog | a guy that it |
| 02:52.51 | mog | er is |
| 02:53.10 | LabRat021 | mog => you know who that guy is? |
| 02:53.20 | mog | timo at digium |
| 02:53.23 | LabRat021 | yep |
| 02:53.28 | mog | yeah |
| 02:53.30 | mog | i know him |
| 02:53.33 | directory | hehe... Timo... |
| 02:53.49 | LabRat021 | is he online right now? |
| 02:54.01 | mog | i dont see him online |
| 02:54.06 | mog | it is 10 pm in hsv |
| 02:54.12 | mog | i dont think anyone is still at digium |
| 02:54.14 | LabRat021 | same here. |
| 02:54.28 | brookshire | why do you ask about timo? |
| 02:54.32 | brookshire | mog!!! |
| 02:54.39 | directory | brookshire: book. a. flight. |
| 02:54.43 | directory | brookshire: write. a. bio. |
| 02:54.51 | directory | brookshire: take. a. headshot. |
| 02:54.55 | brookshire | lol |
| 02:54.57 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 02:54.58 | Qwell | brookshire: (re)sell. your. soul. |
| 02:55.01 | LabRat021 | well, let me know when mark comes back. |
| 02:55.26 | Twister | ./join #blackmarket WTS SOUL for vanilla ice cream |
| 02:55.33 | Twister | :) |
| 02:55.41 | LabRat021 | lol |
| 02:56.00 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 02:56.06 | mog | okies |
| 02:56.23 | LabRat021 | does anyone know where mark is? |
| 02:56.30 | linagee | Qwell: ebay? |
| 02:56.32 | directory | on Earth. |
| 02:57.01 | LabRat021 | alls i can say is that it's a long ways away from h'ville. |
| 02:58.17 | LabRat021 | Oh well, I'll give Timo a call tomorrow. |
| 02:58.34 | *** join/#asterisk Katty (n=Administ@dialup-4.244.159.63.Dial1.StLouis1.Level3.net) |
| 02:59.09 | Qwell | That was..random |
| 02:59.38 | Katty | directory: would you stop changing /nicks. |
| 02:59.43 | Katty | directory: i'm gettin all confuzzled. |
| 02:59.51 | directory | I have to keep the authorities off my tail! |
| 02:59.51 | directory | :D |
| 03:00.08 | Katty | maybe you should just remove your tail. |
| 03:00.10 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 03:00.15 | Katty | then the authorities can't be on it. |
| 03:00.19 | Katty | regardless of /nick |
| 03:01.08 | mog | i hate all your nicks |
| 03:01.20 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 03:02.01 | brookshire | all your nicks are belong to me |
| 03:02.08 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:02.23 | Qwell | /msg brookshire ident supersecretpassword |
| 03:02.29 | Crescendo | Myself and my boss are banging our heads over this error: Sep 26 23:02:28 NOTICE[4863]: chan_iax2.c:5138 register_verify: No registration for peer 'phone2' (from 66.26.63.1) |
| 03:02.37 | Crescendo | What is it and how do we fix it? |
| 03:03.24 | Katty | Crescendo: it's aliens. |
| 03:03.35 | Katty | Crescendo: and you must appease them by sending pillows. |
| 03:03.44 | Crescendo | What of the comfy chair! |
| 03:03.51 | Crescendo | Gah, that doesn't help. |
| 03:04.04 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:04.07 | Katty | have you tried thinking about it while placing a waterbottle on your head? |
| 03:04.08 | linagee | Crescendo: voicepulse? heh |
| 03:04.28 | Katty | that helps me, on occasion |
| 03:04.28 | linagee | j/l |
| 03:04.28 | linagee | j/k |
| 03:04.29 | Crescendo | I don't want voicepulse. |
| 03:04.32 | linagee | no? |
| 03:04.50 | linagee | Crescendo: who do you use then? |
| 03:04.53 | Crescendo | We're just doing a simple SIP setup. |
| 03:04.57 | Katty | Crescendo: so my best guess is that there's a device on that ip... |
| 03:05.07 | Katty | Crescendo: trying to authenticate with the ID phone2 |
| 03:05.12 | Crescendo | He's trying to connect from that IP, yes. |
| 03:05.20 | Crescendo | That's what the ID is set up as for that ID. |
| 03:05.36 | Katty | Crescendo: but you probably don't have iax.conf configured correctly. |
| 03:05.42 | Katty | Crescendo: or at all. |
| 03:06.24 | directory | Crescendo: do you have a peer entry for phone2? |
| 03:06.36 | Crescendo | I haven't touched iax.conf, only sip.conf and some other thing... extensions.conf |
| 03:06.40 | Crescendo | I think so. |
| 03:07.00 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:07.32 | Katty | well, to authenticate a client device (softphone or hardphone) with an iax protocol |
| 03:07.37 | Katty | you have to put the registration info into iax.conf (= |
| 03:07.49 | Katty | sip.conf is for the sip protocol. |
| 03:07.56 | Crescendo | What is the format? |
| 03:08.09 | Katty | look at your conf file |
| 03:08.13 | Katty | there will be examples |
| 03:08.22 | Katty | otherwise, there are some on voip-info i'm sure. |
| 03:08.37 | *** part/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 03:10.01 | droops | im trying to use the w option for a meetme conf, but if the marked user doesnt enter in time, it just hangs up the conf user, shouldnt this send you to another priority? |
| 03:10.01 | *** join/#asterisk kimo_sabe (n=nlopez@70-57-228-216.tcsn.qwest.net) |
| 03:10.21 | Katty | directory: directory directory |
| 03:10.40 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:10.43 | Katty | directory: dir /? |
| 03:11.54 | Katty | directory: you know what i think? |
| 03:12.03 | Katty | directory: i think newbies should recieve hugs. |
| 03:12.17 | Katty | directory: because it's not everyday that the average person will feel like tackling something new. |
| 03:12.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:12.27 | directory | then hug them |
| 03:12.31 | directory | or hug [TK]D-Fender |
| 03:12.35 | Katty | okay! |
| 03:13.57 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:14.09 | shmaltz | hi everyone |
| 03:16.02 | [TK]D-Fender | z0mg |
| 03:16.24 | directory | [TK]D-Fender: do you just want want... |
| 03:17.09 | [TK]D-Fender | directory : ! ! ! |
| 03:17.24 | directory | :D |
| 03:21.00 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:25.19 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 03:27.21 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 03:28.28 | wulfy814 | hello folks, I'm working on a fairly new asterisk system, latest stable, ubuntu dapper drake |
| 03:28.34 | wulfy814 | polycom Ip430's |
| 03:28.47 | wulfy814 | I can check voicemail and make calls |
| 03:29.19 | wulfy814 | but when I call from one station to another [200] --> [201] and then hangup [201] then [200] will lockup and reboot |
| 03:29.34 | wulfy814 | any ideas? I'm running the 2.01 firmware |
| 03:29.55 | james_ | probably a bug |
| 03:29.56 | CunningPike | wulfy814: Hmmm - no ideas, sorry - sounds like a problem with that set rather than asterisk |
| 03:30.08 | *** part/#asterisk Prelius (n=pzotov@70.88.213.25) |
| 03:33.07 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 03:33.18 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.139.173) |
| 03:34.58 | smackus | when starting asterisk i get the following error. could someone please help me understand why? it started fine before, than I rand the ./Setup install for the wanpipe stuff and rebooted... now this http://pastebin.ca/183697 |
| 03:36.15 | tuck3r | for caller ID, is "num" and ani the same thing? |
| 03:36.18 | shmaltz | smackus, looks like you got something lin /etc/asterisk/modules.conf |
| 03:36.28 | shmaltz | tuck3r, no |
| 03:37.01 | shmaltz | tuck3r: |
| 03:37.02 | shmaltz | http://www.voip-info.org/wiki/index.php?page=PBX+Calling+ID |
| 03:37.59 | smackus | where are the modules actually found... |
| 03:38.15 | shmaltz | it's at the bottom of your mesage in that pb |
| 03:38.32 | smackus | yeah, ok... im blind thanks |
| 03:38.42 | shmaltz | smakcus, or try chmod on that file |
| 03:38.44 | shmaltz | /usr/lib/asterisk/modules/res_odbc.so |
| 03:39.05 | shmaltz | smackus, do the following: |
| 03:39.06 | shmaltz | ls -l /usr/lib/asterisk/modules/res_odbc.so |
| 03:39.07 | shmaltz | what do you get? |
| 03:39.24 | smackus | well, i just deleted them to reinstall so it would return nothing now... |
| 03:39.37 | shmaltz | :P |
| 03:40.39 | tuck3r | shmaltz: it seems that "num" and "ani" are doing the same thing tho |
| 03:42.05 | smackus | i copied /etc/asterisk from another server on to this one... may be just a permissions thing... reinstalling to make sure its all correct. |
| 03:42.43 | shmaltz | tuck3r, it could be for you, but how did you come to that conclusion? |
| 03:42.50 | shmaltz | ani should be just a 2 digit code |
| 03:44.23 | tuck3r | but I have seen examples with the ani being 10 digits |
| 03:45.27 | shmaltz | tuck3r, could be I'm wrong, and the 2 digit thingy is only for ani2 |
| 03:45.35 | truespace | HOLY CRAPPERS |
| 03:45.43 | tuck3r | shmaltz: it seems your right |
| 03:46.42 | shmaltz | tuck3r: |
| 03:46.44 | shmaltz | http://en.wikipedia.org/wiki/Automatic_number_identification |
| 03:46.51 | tuck3r | i was just reading that |
| 03:48.37 | shmaltz | tuc3r, from that article it appears that ani is the service and ani2 is the specs on the 2 digit code, now to asterisk, ani then means the 10 digit CNID, and ani2 is the code |
| 03:48.58 | wulfy814 | CunningPike: it turned out to be the Polycom 2.01 firmware |
| 03:49.06 | wulfy814 | reverted to 1.67 and the world is well |
| 03:49.50 | tuck3r | shmaltz: i'm just wondering the proper way to set outbound caller id number in 1.4 |
| 03:49.53 | CunningPike | wulfy814: OK good - the downgrade worked OK? I thought they wouldn't downgrade |
| 03:50.12 | wulfy814 | CunningPike: yeah downgrade worked fine from 2.01 to 1.67 |
| 03:50.14 | shmaltz | tuck3r, just use set (CALLERID(num)) |
| 03:50.33 | CunningPike | wulfy814: Good to know - thanks |
| 03:50.54 | *** join/#asterisk h0 (n=haker0@unaffiliated/fakhir) |
| 03:52.49 | h0 | does anyone have a rough estimate when asterisk 1.4 will be out of beta. Just want to know if I should wait or just install 1.2 for my new install. |
| 03:53.13 | brookshire | h0: october sometime |
| 03:53.44 | h0 | k thanks |
| 03:55.02 | *** part/#asterisk Toadyus (i=Toadyus@S010600121746f9fe.mh.shawcable.net) |
| 03:57.28 | *** part/#asterisk h0 (n=haker0@unaffiliated/fakhir) |
| 04:08.57 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:10.42 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 04:12.07 | truespace | can 1 DID number be used for concurrent multiple inbound calls? |
| 04:12.16 | denon | yes |
| 04:12.24 | denon | depending on the restrictions your provider may have |
| 04:12.45 | denon | you could have 1 DID tied to hundreds or thousands of pstn trunks, or SIP sessions |
| 04:12.55 | *** join/#asterisk evilmnky (n=evilmnky@216-106-185-169.ds1-static.mia1.net.ststelecom.com) |
| 04:14.23 | *** join/#asterisk Borgon (n=l3orgon@12.173.143.25) |
| 04:15.14 | Borgon | IF my iax works find behind a double router nat... how can i get sip to work and make sound audible? Can a ssh tunnel to a open vps make my incoming connections play? |
| 04:15.32 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 04:15.34 | linagee | denon: thousands? lol |
| 04:15.40 | linagee | denon: if you operated a call center? |
| 04:15.41 | denon | linagee: sure, or more |
| 04:15.44 | denon | right |
| 04:15.50 | linagee | denon: that is scary |
| 04:16.01 | denon | think of 1-800-Pick-UPS |
| 04:16.02 | kimo_sabe | linagee: 1800-dell-sux, or any other call-center sort of environment |
| 04:16.11 | linagee | denon: 10,000? lol |
| 04:16.12 | denon | or 1-800-Go-FEDEX |
| 04:16.21 | kimo_sabe | denon has better examples :) |
| 04:16.30 | linagee | kimo_sabe: i would think dell went with a commercial thing. nortel. |
| 04:16.35 | *** join/#asterisk Math` (n=privmath@modemcable238.43-82-70.mc.videotron.ca) |
| 04:16.44 | linuxbangalore | I am doing some testing.. |
| 04:16.46 | denon | linagee: the platform you use doesnt matter, it's the same technology |
| 04:16.48 | truespace | what if someone uses multiple free outbound PSTN accounts to do an attack on your DID#? and then you pay the charges for the answering machine time |
| 04:16.48 | linuxbangalore | here is the problem |
| 04:16.49 | linuxbangalore | chan_iax2.c:5138 register_verify: No registration for peer 'Phone4' |
| 04:16.55 | linagee | denon: fedex uses asterisk? |
| 04:17.01 | kimo_sabe | linagee: I'm sure they did, I was just trying to think of an example of one DID an bajillions of channels |
| 04:17.03 | Math` | uhm why is there a password for documentation on digium's site? (redirected to staging.digium.com) |
| 04:17.10 | denon | linagee: doesnt matter, it's the same technology for what he's talking about |
| 04:20.29 | truespace | what came first? the chicken, the egg or KFC? |
| 04:20.39 | Math` | lol |
| 04:22.18 | truespace | i am going to start up a call center with my DID |
| 04:24.27 | truespace | erectile dysfunction support call center |
| 04:24.49 | truespace | I will call it: DID you come? |
| 04:25.17 | kimo_sabe | truespace: and what will it say when it answers? |
| 04:25.17 | linagee | does anyone know why my lag to voicepulse is going up? |
| 04:25.28 | linagee | used to be 68ms, now it's 94 ms... |
| 04:25.28 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 04:25.34 | linagee | :( |
| 04:25.41 | Math` | linagee: traceroute will tell you |
| 04:25.55 | linagee | there's even noticable jitter in the audio now |
| 04:26.06 | linagee | any way to fix that in a conf somewhere? :-/ |
| 04:26.37 | Math` | you can increase the jitterbuffer |
| 04:26.46 | Math` | 94ms shouldnt be noticable |
| 04:26.49 | Math` | unless you drop packets |
| 04:26.56 | linagee | maybe the packets are dropped |
| 04:26.58 | linagee | i'm using IAX2 |
| 04:27.03 | denon | or unless it varies a lot |
| 04:27.10 | denon | if you fire off a continuous ping, what's the deviation look like? |
| 04:27.33 | linagee | checking now... |
| 04:28.12 | linagee | ok, just did 30 packets, normal ping rate. 31 packets transmitted, 31 received, 0% packet loss, time 30035ms |
| 04:28.12 | linagee | rtt min/avg/max/mdev = 55.994/83.450/112.947/12.716 ms, pipe 2 |
| 04:28.26 | linagee | 56ms to 112 ms. hrm |
| 04:28.58 | *** part/#asterisk linuxbangalore (n=karsansu@59.92.139.173) |
| 04:29.02 | Math` | dont forget rtp packets are bigger and more frequent than pings |
| 04:29.10 | Math` | or iax2 even |
| 04:29.11 | linagee | Math`: what size? |
| 04:29.17 | linagee | Math`: i could set the ping size |
| 04:29.20 | denon | that's quite a spread |
| 04:29.34 | truespace | you saw October playboy too? |
| 04:29.43 | linagee | denon: so if i could get my firewall to make the latency more normalized, that's better? |
| 04:29.45 | Math` | Im sure asterisk has a command to display dropped packets |
| 04:29.46 | denon | linagee: hard to get a real accurate reading with icmp, as your voice packets are different sized and udp |
| 04:29.56 | linagee | oic |
| 04:30.02 | linagee | denon: iptraf? |
| 04:30.19 | Math` | denon: does IAX2 have seqnos? |
| 04:30.26 | denon | linagee: quite honestly, make your carrier do the work - open a ticket with them and complain |
| 04:30.37 | denon | they may have dropped some providers hoping to save a few bucks |
| 04:30.37 | Math` | lol |
| 04:30.58 | *** part/#asterisk Star568 (n=steve@pool-71-118-163-45.lsanca.dsl-w.verizon.net) |
| 04:31.04 | denon | now's the time to say, "hey, dont drop xyz backbone, that's close to me!" |
| 04:33.35 | linagee | interesting. all of the voip packets were between 75 and 150 bytes |
| 04:33.39 | linagee | but it's iax2... |
| 04:35.13 | linagee | denon: ok. i'm now trying using 148 byte pings. (ping -s 140) |
| 04:35.27 | truespace | anyone have a link to a good tutorial on setting up asterisk for simple peer to peer VOIP? |
| 04:36.27 | linagee | denon: look at the results now. (previous packets were about half the size) |
| 04:36.28 | linagee | 61 packets transmitted, 61 received, 0% packet loss, time 60069ms |
| 04:36.29 | linagee | rtt min/avg/max/mdev = 54.785/81.801/123.074/14.001 ms, pipe 2 |
| 04:36.45 | linagee | does mdev mean something? |
| 04:36.48 | Math` | 60 packets in 60 seconds |
| 04:36.56 | Math` | voip goes much faster than that |
| 04:37.18 | denon | linagee: you really do want to put the burdain of diagnostic on your provider |
| 04:37.37 | linagee | Math`: i could push them all down. heh. -l 60 -c 60 |
| 04:37.41 | truespace | fuck i'm a nerd... |
| 04:37.51 | Math` | linagee: but then your provider might rate-limit icmp packets so your result will not be accurate |
| 04:37.56 | linagee | --- csw004.voicepulse.com ping statistics --- |
| 04:37.56 | linagee | 60 packets transmitted, 60 received, 0% packet loss, time 2ms |
| 04:37.56 | linagee | rtt min/avg/max/mdev = 85.392/190.138/275.580/48.508 ms, pipe 61 |
| 04:37.58 | linagee | Math`: true... |
| 04:38.24 | linagee | that was (ping -s 140 -l 60 -c 60) |
| 04:39.01 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 04:41.04 | truespace | anyone here used trixbox? |
| 04:41.19 | linagee | wow. wtf. hahaha. look what happens when i don't use traffic shaper. |
| 04:41.24 | linagee | --- csw004.voicepulse.com ping statistics --- |
| 04:41.24 | linagee | 60 packets transmitted, 39 received, 35% packet loss, time 13ms |
| 04:41.24 | linagee | rtt min/avg/max/mdev = 101.112/161.502/230.881/26.157 ms, pipe 61 |
| 04:41.30 | linagee | that sucks |
| 04:41.45 | truespace | is there any way to disable the backdoors in my VOIP hardware that were put in there by mossad? |
| 04:41.54 | linagee | truespace: who is mossad?? |
| 04:42.08 | truespace | israeli spy agency |
| 04:42.22 | linagee | truespace: there's an easy way. never say anything classified over the phone. :p |
| 04:42.41 | truespace | ya but how can i buy and sell cruise missiles on ebay then |
| 04:43.17 | linagee | truespace: wtf. i know you're just trolling, but what does ebay have to do with phones? |
| 04:43.41 | truespace | ebay owns skype |
| 04:43.59 | truespace | nevermind i will just read the tutorials |
| 04:45.03 | linagee | truespace: there's a cruise missle tutorial? |
| 04:46.20 | truespace | see this guy put a backdoor in my VOIP adapter: "http://www.ecomtrade.co.il/emall/content.asp?cc=76221" |
| 04:46.57 | truespace | excerpt: "Prior to Orckit, Mr. Avrahami was associated with the Israeli Ministry of Defense (Intelligence Corp)." |
| 04:50.24 | truespace | what can you expect i guess.. remember Bill Clinton telling monica lewinsky that he couldnt talk about sex on the phone cause a foreign government was tapping his phone :) |
| 04:50.37 | *** join/#asterisk drcode (n=user1@87.69.59.186.cable.012.net.il) |
| 04:50.40 | drcode | hi all |
| 04:54.13 | truespace | does trixbox work on windows or only linux? |
| 04:54.25 | Math` | linux |
| 04:56.28 | truespace | i wish there was a drug that could decrease procrastination |
| 04:57.24 | rob0 | caffeine? :) |
| 04:57.30 | brookshire | redbull? |
| 04:57.34 | brookshire | meth? |
| 04:58.25 | truespace | i was thinking more like ritalin |
| 04:58.28 | truespace | or something |
| 04:58.57 | truespace | if i had less procrastination i could actually install this software i think |
| 04:59.21 | directory | brookshire: BROOKSHIRE |
| 04:59.24 | linagee | ah nice. here we go. UDP packets, size = 54. (IAX2) |
| 04:59.32 | brookshire | file! |
| 04:59.34 | linagee | wait, isn't IAX2 TCP based? |
| 04:59.40 | brookshire | no |
| 04:59.40 | Math` | no |
| 04:59.41 | directory | linagee: ha no |
| 04:59.44 | linagee | interesting... |
| 04:59.47 | brookshire | iax2 is UDP based |
| 04:59.51 | linagee | so SIP and IAX2 are both UDP |
| 04:59.54 | Math` | hell no would be the right answer |
| 05:00.04 | linagee | IAX2 just has as better chance with NATs? |
| 05:00.06 | brookshire | well.. you can do sip over tcp |
| 05:00.11 | truespace | do the linksys VoIP Broadband Routers work well with asterisk? |
| 05:00.14 | brookshire | i think there is a standard somewhere |
| 05:00.59 | Math` | truespace: if they are sip (which most of them are) they will work |
| 05:01.02 | directory | brookshire: I need to get my hair cut :( |
| 05:01.23 | truespace | cool |
| 05:02.29 | brookshire | directory: flowbee? |
| 05:03.03 | directory | brookshire: are you INSANE?!? |
| 05:03.08 | brookshire | http://www.flowbee.com/images/webmombot2flat%20copy.jpg |
| 05:03.35 | Math` | to configure a 2400, what kind of framing/coding should I specify in span=? those should only matter for isdn links.... |
| 05:03.54 | brookshire | umm.. |
| 05:04.09 | brookshire | tdm2400p is analog |
| 05:04.16 | brookshire | you do not have framing or coding with that |
| 05:04.22 | Qwell | brookshire: Don't diss the flowbee |
| 05:04.28 | Math` | yeah thats what I was thinking |
| 05:04.33 | Math` | but I need to create a span to use the card no^ |
| 05:04.34 | Math` | ? |
| 05:04.39 | brookshire | qwell: was that your mom? |
| 05:04.40 | Qwell | Math`: no |
| 05:05.15 | directory | Qwell: you should grow your hair out ... |
| 05:05.20 | Qwell | meh |
| 05:05.31 | Math` | I have a TE110P with a TDM2401P in that box, the span is supposed to be for the TE110P but when I cat /proc/zaptel/1 it tells me "Span 1: WCTDM/0 "Wildcard TDM2400P Board 1" ... |
| 05:05.34 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 05:05.51 | brookshire | math: change the order the modules load |
| 05:06.00 | Math` | ah cuz it matters |
| 05:06.21 | brookshire | i think that will fix it |
| 05:06.22 | brookshire | heh |
| 05:06.26 | brookshire | never tried.. just guessing |
| 05:07.54 | Math` | uh my config started working automagically, was giving me some "Invalid argument" error on SPANCONFIG before |
| 05:08.36 | [shodan] | eh, funny thing just happened , I picked up a phone connected on the same line as my x100p , and I loudly whistled into the phone , and on the asterisk console I saw "Got event 18 (Ring Begin)..." and it started playing my welcome message ! |
| 05:08.51 | Math` | lol |
| 05:08.57 | Math` | callprogress=yes |
| 05:09.02 | Qwell | You whistled at two frequencies? |
| 05:09.51 | [shodan] | callprogress is set to no |
| 05:10.43 | Math` | is it normal I dont have any files in /proc/zaptel when asterisk isnt started yet? (but I ztcfg'd everything properly) |
| 05:10.45 | linagee | wow. when i dial into asterisk with a cell phone, there is a lot more latency than when i dial with a land line! (put voip phone on speaker phone, press DTMF keys while connected to extension, listen to relative time before DTMF heard) |
| 05:10.58 | Math` | yeah cells are laggy |
| 05:11.05 | linagee | i'd almost guestimate that the latency for cell phones is twice! lol |
| 05:11.43 | linagee | Math`: interesting too... i think the latency on cell phones is one sided. i think it takes longer for the cell signals to reach the tower than the tower to reach the cell... |
| 05:12.20 | Math` | linagee: RX speed is always faster because the antennas can have a stronger transmit power while meeting the FCC regulations |
| 05:12.28 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 05:12.28 | linagee | aha |
| 05:12.32 | Math` | and by being plugged in to an electrical network lol |
| 05:12.45 | linagee | Math`: also, cell towers have a proper antenna instead of a stub (if even that. lol) |
| 05:13.02 | Math` | your cell is limited to a certain transmit power, which lowers the bandwidth available to your phone, which adds latency |
| 05:13.10 | linagee | Math`: i wonder if the cell phone is "slower" too because of it's voice processor? hrm. |
| 05:13.23 | Math` | thats a bandwidth issue |
| 05:13.27 | linagee | Math`: i'm sure the GSM encoding in a cellphone adds some latency |
| 05:13.38 | Math` | it does, but its done hardware |
| 05:13.43 | Math` | so it doesnt add too much |
| 05:14.10 | [shodan] | turns out, softly blowing on the phone's mic with the volume turned up is causing all kind of signals to be received by the x100p |
| 05:14.25 | linagee | [shodan]: all kinds of signals? lol. aliens? |
| 05:14.56 | [shodan] | no. ring signal , answered signal etc.. |
| 05:15.04 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 05:15.08 | brookshire | linagee: actually the encoder chips are pretty quick on cell phones.. |
| 05:15.16 | truespace | to save battery life, the cellphone will turn its transmit power down as low as possible while still maintaining adequate audio quality, while the cell tower has unlimited power |
| 05:15.23 | linagee | Math`: i guess people just don't notice cell phone latency? i mean, will it ever improve? lol |
| 05:15.31 | brookshire | i think it has more to do with the cell towers being connected with microwaves |
| 05:15.33 | linagee | brookshire: hrm. 1ms? 2ms? |
| 05:15.54 | linagee | i don't think that's something you'll percetibly notice |
| 05:16.01 | denon | not all cell towers are connected via microwave |
| 05:16.04 | brookshire | like you call this tower, it bounces the call off of three more towers before it finds a hard line |
| 05:16.16 | linagee | hrm... i see.. |
| 05:16.18 | denon | many carriers actually do T1s right to the tower |
| 05:16.40 | brookshire | denon: a lot still use microwaves though |
| 05:16.54 | linagee | denon: and i'm guessing the cell carriers have a certain latency SLA expected by the T1 provider or the call will sound like shiznit? lol |
| 05:17.24 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:17.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:17.34 | linagee | so i wonder if latency changes with distance to the cell tower, or what tower it's using. hrm |
| 05:17.48 | Math` | changes with your signal strength |
| 05:17.48 | truespace | not noticeabley |
| 05:17.53 | Math` | but slightly |
| 05:18.11 | Math` | anyways as long as you dont have echo you're not going to notice latency |
| 05:18.23 | truespace | cellphone range is so short compared to the distance required to notice latency from lights finite propagation speed |
| 05:18.41 | Math` | I had a call with a VSAT connection in jamaica, over 500ms delay (one way) |
| 05:18.52 | Math` | was seamless |
| 05:19.11 | [shodan] | hmm I have a problem , when I call people with my x100p (which is going to be replaced by a spa-3*02 real soon now) sometime asterisk hangs up on them , especially if they are on cellphones , what is causing the x100p to think they have hung up ? on another x100p I have , it won't detect hang up at all until I get the busy signal 15 seconds after they hungup , what setting can I adjust for "far side hangup detection |
| 05:19.11 | [shodan] | sensitivity" ? |
| 05:19.41 | brookshire | it's a x100p.. nuff said |
| 05:19.51 | Qwell | [shodan]: Got a soldering iron? |
| 05:20.00 | Math` | had the same problem with a TDM400 |
| 05:20.00 | Qwell | Take that, and stab it repeatedly through the pcb |
| 05:20.18 | [shodan] | I like the first part |
| 05:20.20 | [shodan] | ohhh |
| 05:21.25 | brookshire | i should really trade my x100p for a fxo module |
| 05:21.31 | *** part/#asterisk truespace (i=truespac@S010600301b1eb05e.gv.shawcable.net) |
| 05:21.53 | [shodan] | dunno it just feels wrong to shell out 90$ per fxo :( |
| 05:22.06 | Qwell | [shodan]: Try pricing a nortel :) |
| 05:22.34 | linagee | hahahha. great. i just found a firmware bug in my phone. lol. wow. |
| 05:22.44 | linagee | (motorola) |
| 05:22.50 | brookshire | razr? |
| 05:22.51 | Math` | and it impresses you? |
| 05:22.54 | Math` | firmware are full of bugs |
| 05:23.08 | linagee | Math`: it's in field test mode. you hit the wrong key and screen goes all crazy with bars |
| 05:23.09 | Math` | try spraying some Off on it |
| 05:23.28 | [shodan] | hmm |
| 05:23.29 | Math` | it might just be a screen test |
| 05:23.35 | Math` | press the same wrong key over and over |
| 05:23.38 | brookshire | you are running a 1.0 beta.. touch this it's 1.5 |
| 05:23.40 | linagee | ut ohh. i hope it's a screen test. LOL |
| 05:23.42 | linagee | oh whoa!!! |
| 05:23.45 | linagee | cool |
| 05:23.45 | brookshire | farm fresh! |
| 05:23.46 | linagee | text |
| 05:23.49 | Math` | lol |
| 05:24.02 | linagee | it has all this mad text looks like linux booting. lol |
| 05:24.18 | brookshire | linagee: i believe motorola phones do run linux |
| 05:24.24 | Qwell | some do |
| 05:24.30 | linagee | oic. there is a signal strength number |
| 05:24.32 | Math` | off the shelf? |
| 05:24.34 | linagee | V557. probably not |
| 05:24.38 | Qwell | Math`: yeah |
| 05:24.43 | Math` | :o |
| 05:24.45 | linagee | PoC flip open phone |
| 05:24.56 | Math` | can you run * on the phone and make it a gsm gateway? lol |
| 05:25.05 | linagee | oh fsck. lol |
| 05:25.14 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:25.21 | linagee | i can't get rid of it. lol. it's like an "always on top" sort of thing |
| 05:25.25 | Math` | AST_TYPE_TEXT is there for SMS already :P |
| 05:26.15 | linagee | haha! got out. menu/left soft key. internet told me. :-D |
| 05:27.36 | Qwell | I like that my cell does fax detection.. |
| 05:27.45 | Qwell | and can call forward based on that |
| 05:28.45 | linagee | interesting... you can have it play a tone when you switch networks. geez that might get annoying. lol |
| 05:31.11 | [shodan] | is there some x100p hardware mod page ? since they all use the same design it should be easy to improve , and the most common problem is echo that could easily be fixed by matching the impedance by changing the capacitors on it ! |
| 05:32.24 | Math` | Qwell: oh, it doesnt redirect it to your carrier then Multimedia Message you the fax? |
| 05:32.39 | Qwell | Math`: never tried it... |
| 05:33.04 | Qwell | I highly doubt it'll mms it though |
| 05:33.21 | Qwell | never know.. |
| 05:33.26 | Math` | maybe email but I still doubt it |
| 05:33.32 | Math` | and for free I doubt it even more |
| 05:36.31 | *** join/#asterisk NDT (i=NDT@cpe-24-195-72-2.nycap.res.rr.com) |
| 05:36.39 | drcode | is there open source project to connect sip-to-astrikes |
| 05:37.09 | brookshire | drcode: i have no idea what you are asking |
| 05:37.10 | Qwell | drcode: huh? |
| 05:40.08 | Qwell | brookshire: see msg :D |
| 05:40.36 | drcode | I want to connect skype into astriks |
| 05:40.42 | Qwell | skype isn't sip |
| 05:40.50 | drcode | it has sip no? |
| 05:40.55 | Qwell | no |
| 05:43.30 | linagee | would it be better to have a voip provider on the same coast as you? |
| 05:43.38 | linagee | does that make a difference in lag? |
| 05:43.49 | Qwell | linagee: sure |
| 05:44.18 | drcode | k |
| 05:44.40 | linagee | my first level3 hop out onto the intarweb gets 13ms. that's not that bad. |
| 05:44.57 | Math` | skype is using sip for skypeOut with level3 but they dont let you use it |
| 05:45.17 | linagee | the level3 path from san diego to chicago makes it jump from 13ms to 80ms |
| 05:45.24 | linagee | yikes |
| 05:45.58 | linagee | Math`: "but they don't let you use it"? |
| 05:46.19 | Math` | skype is using sip to interconnect with level3 for the skypeout service |
| 05:46.25 | Math` | but they dont offer any sip connection to their customers |
| 05:46.30 | linagee | hmmmm |
| 05:46.32 | linagee | that sucks |
| 05:46.39 | Math` | indeed |
| 05:47.09 | linagee | Math`: "i don't care if you have hardware phones that work better. you MUST use our client!!" lol |
| 05:47.36 | Math` | or a skype hardware adapter that paid their royalty fees to skype |
| 05:48.20 | linagee | why am i always connecting through level3? with cox and rr. does level3 have some sort of huge deal with cable internet? heh |
| 05:50.04 | Math` | with those 2 they do lol |
| 05:53.37 | linagee | who is broadviewnet.net? |
| 06:00.12 | brookshire | a company with broken links on their homepage |
| 06:00.37 | Qwell | must be a subsidiary of Cisco |
| 06:00.53 | Qwell | page is just as unusable |
| 06:01.03 | linagee | ew. even les.net is about just as latent as voicepulse... |
| 06:01.56 | linagee | rtt min/avg/max/mdev = 70.409/85.731/119.955/11.749 ms, pipe 2 |
| 06:02.15 | linagee | what is with people hosting on the east coast. :( |
| 06:03.15 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 06:03.20 | aadilismail | hi guys |
| 06:05.09 | linagee | huh?? |
| 06:05.25 | linagee | teknoprep: are you still here? |
| 06:05.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:11.02 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 06:13.17 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.139.173) |
| 06:13.49 | linuxbangalore | hi.. when I am placing call from one softphone to the other.. both on same machine.. and asterisk on same network.. I am getting this error |
| 06:13.49 | linuxbangalore | Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 06:14.10 | linuxbangalore | softphones I am using.. Xlite and DIAX |
| 06:14.44 | linuxbangalore | I am able to make from XLITE to XLITE.. I mean, I am calling the same number and able to receive it on line2 |
| 06:15.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:15.57 | DarKnesS_WolF | linuxbangalore: seems that ur other phone not registered to astersik |
| 06:16.04 | kimo_sabe | linuxbangalore: channel type SIP for calling DIAX? |
| 06:16.05 | DarKnesS_WolF | sip show peers and see if it's registered |
| 06:16.30 | linuxbangalore | ha.. you are rite.. this is what I am getting |
| 06:16.31 | linuxbangalore | amar/amar (Unspecified) D 0 Unmonitored |
| 06:16.31 | linuxbangalore | santosh/santosh 192.168.1.100 D 62168 Unmonitored |
| 06:16.33 | DarKnesS_WolF | DIAX = IAX client ? or both sip ? |
| 06:16.40 | Nugget | finally, someone in india has to call (upon) the US for tech support. At last we see the tables have turned! ;) |
| 06:16.43 | kimo_sabe | IAX only |
| 06:16.44 | linuxbangalore | oh.. both sip |
| 06:17.31 | linuxbangalore | kimo_sabe, is DIAX .. IAX only? I did not get you |
| 06:18.13 | DarKnesS_WolF | linuxbangalore: if DIAX is a IAX only then u need a IAX account not a SIP one. |
| 06:18.20 | kimo_sabe | linuxbangalore: according to it's webpage, yes, it's IAX2 only |
| 06:20.09 | linuxbangalore | thank you.. Darkness and kimo |
| 06:20.28 | *** join/#asterisk Star568 (n=steve@pool-71-118-163-45.lsanca.dsl-w.verizon.net) |
| 06:21.11 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 06:21.50 | Star568 | if i have have 50 sip phones try to connected to asterisk, should i need to add 50 contexts to sip.conf? |
| 06:22.24 | kimo_sabe | Star568: 50 contexts? Not unless you want them all to have unique dialplans. Extensions, sure, but not contexts |
| 06:23.30 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 06:23.57 | zouzou | what the differencre between callgroup and pickupgroup? |
| 06:25.28 | Star568 | <kimo_sabe> chan_sip.c:11084 handle_request_register: Registration from '"200"<sip:200@test.test.com:5060>' failed for '71.118.100.100' - Username/auth name mismatch |
| 06:26.35 | Star568 | <kimo_sabe> if i set a context [200] in sip.conf, there 200 will be ok for register |
| 06:27.27 | *** join/#asterisk daysmen3 (n=primus@host81-153-174-207.range81-153.btcentralplus.com) |
| 06:27.30 | Star568 | <kimo_sabe> does it means that i need to create a lot of contexts inside sip.conf? [201],[202],[203]... |
| 06:28.58 | linagee | oh wow. there's this really neat voip provider, real cheap. only 25 hops away from me! LOL. yikes. |
| 06:30.50 | *** join/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 06:31.21 | Narkov- | how long should I use NVBackgroundDetect for? 1 second? 20 seconds? |
| 06:31.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:31.57 | DarKnesS_WolF | Star568: what ur talking about here is not a context it called extention name |
| 06:32.08 | linuxbangalore | cool.. now I am using the other sip phone.. "Express Talk" and I am able to make call from "Express Talk to Xlite" But not able to make a call from XLITE to Express Talk |
| 06:32.17 | DarKnesS_WolF | make sure that the username 200 and the password are mached in ur sip.conf and in ur phone |
| 06:32.43 | DarKnesS_WolF | linuxbangalore: u didn't add the extention number of the express talk in ur extention |
| 06:32.58 | linuxbangalore | and here is the sip show peers |
| 06:32.59 | DarKnesS_WolF | exten => balba,1,Dial(SIP/balba) |
| 06:33.07 | linuxbangalore | i did.. |
| 06:33.08 | linuxbangalore | amar/amar (Unspecified) D 0 Unmonitored |
| 06:33.08 | linuxbangalore | santosh/santosh 192.168.1.100 D 62168 Unmonitored |
| 06:33.10 | DarKnesS_WolF | where is balba is ur express talk phone number |
| 06:33.22 | DarKnesS_WolF | linuxbangalore: this is something related to the dialplan as i think |
| 06:33.50 | DarKnesS_WolF | Narkov-: i have no idea i failed to get faxing with asterisk even i have a detecated phone line. |
| 06:34.12 | linuxbangalore | here is the extension.conf.. I cannot see any error in this |
| 06:34.12 | linuxbangalore | [testsip] |
| 06:34.12 | linuxbangalore | exten => 1234,1,Dial(SIP/santosh) |
| 06:34.12 | linuxbangalore | exten => 1235,1,Dial(SIP/amar) |
| 06:34.30 | DarKnesS_WolF | i didn't use NVBackgroundDetect too just tx_fax and rx_fax |
| 06:34.37 | Narkov- | DarKnesS_WolF: i'm more keen to NOT send a fax with asterisk....so I want to detect it and then hang up |
| 06:35.04 | DarKnesS_WolF | Narkov-: did u read the asterisk faxing page in voip-info ? |
| 06:35.21 | Narkov- | yeah the NV pages don't specify how long is desireable |
| 06:35.29 | DarKnesS_WolF | linuxbangalore: sip show peers again please ? |
| 06:35.44 | DarKnesS_WolF | Narkov-: i think u can give it a try or so |
| 06:36.13 | Narkov- | i can try it yes but I want to know what the optimal values are |
| 06:36.21 | linuxbangalore | Name/username Host Dyn Nat ACL Port Status |
| 06:36.22 | linuxbangalore | amar/amar (Unspecified) D 0 Unmonitored |
| 06:36.22 | linuxbangalore | santosh/santosh 192.168.1.100 D 62168 Unmonitored |
| 06:36.22 | linuxbangalore | 2 sip peers [2 online , 0 offline] |
| 06:36.52 | linuxbangalore | now.. I just got this error.. when I placing call from xlite to express talk.. |
| 06:37.05 | linuxbangalore | <PROTECTED> |
| 06:37.11 | linagee | does anyone have the voipjet hostnames? i want to check latency |
| 06:37.45 | DarKnesS_WolF | linuxbangalore: please add qualify=yes in ur sip.conf for both phones and reaload * |
| 06:38.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:38.45 | linagee | interesting... voipjet has servers on both coasts? |
| 06:39.09 | linuxbangalore | same error even afetr reload |
| 06:39.10 | linuxbangalore | Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 06:39.29 | linuxbangalore | i added qualify=yes for both the phones |
| 06:39.33 | linuxbangalore | in sip.conf |
| 06:39.35 | DarKnesS_WolF | linuxbangalore: yes show me sip show peers |
| 06:39.51 | linuxbangalore | Name/username Host Dyn Nat ACL Port Status |
| 06:39.51 | linuxbangalore | amar/amar (Unspecified) D 0 UNKNOWN |
| 06:39.51 | linuxbangalore | santosh/santosh 192.168.1.100 D 62168 OK (102 ms) |
| 06:39.51 | linuxbangalore | 2 sip peers [1 online , 1 offline] |
| 06:40.03 | DarKnesS_WolF | seen ur amar phone is not registered with asterisk |
| 06:40.05 | DarKnesS_WolF | go fix this |
| 06:40.14 | linuxbangalore | ha |
| 06:40.36 | *** join/#asterisk lorinc (n=ang@caracas-3200.adsl.interware.hu) |
| 06:40.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:42.59 | *** join/#asterisk oej (n=oej@231.Red-88-14-197.dynamicIP.rima-tde.net) |
| 06:43.43 | *** part/#asterisk oej (n=oej@231.Red-88-14-197.dynamicIP.rima-tde.net) |
| 06:45.11 | *** join/#asterisk angom_h (n=Angel@red-corp-200.38.30.241.telnor.net) |
| 06:48.44 | *** join/#asterisk tropobot (n=tropobot@202.149.38.32) |
| 06:52.38 | linuxbangalore | DarKness, I did make any more changes in the extension.conf and sip.conf file.. but just switched the registrations of the phones.. |
| 06:53.02 | linuxbangalore | now.. amar is on xlite and santosh is on express.. and the happens.. with xlite everything is good |
| 06:53.31 | linuxbangalore | but with express it is not.. do you think it the problem with some port which the express talk listens |
| 06:53.47 | DarKnesS_WolF | linuxbangalore: sip show peers |
| 06:54.04 | linuxbangalore | Name/username Host Dyn Nat ACL Port Status |
| 06:54.04 | linuxbangalore | amar/amar 192.168.1.100 D 37102 OK (103 ms) |
| 06:54.04 | linuxbangalore | santosh/santosh (Unspecified) D 0 UNKNOWN |
| 06:54.04 | linuxbangalore | 2 sip peers [1 online , 1 offline] |
| 06:54.14 | DarKnesS_WolF | linuxbangalore: and seems that ur express talk is not working correctly .. so u can try anohter SIP client like ekiga |
| 06:54.36 | linuxbangalore | ok thank you.. I will change the softphone.. to ekiga |
| 06:58.23 | DarKnesS_WolF | linuxbangalore: ur welcome but seems to me that the problem in ur softphone configurations not * |
| 06:59.05 | linuxbangalore | rite.. as you said I will use some other softphone and test it.. |
| 07:02.34 | *** join/#asterisk Sir_Diddymus (n=doe@pd95b08e3.dip0.t-ipconnect.de) |
| 07:03.45 | stephane_ | amis du jour bonjour |
| 07:04.49 | *** part/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 07:10.46 | *** join/#asterisk |nix (n=inix@bb58-185-129-130.singnet.com.sg) |
| 07:11.09 | |nix | hi, i would like to check if asterisk supports extension numbers that begin with 0? say 0902 |
| 07:12.04 | Math` | why wouldnt it |
| 07:12.21 | zouzou | salut |
| 07:13.30 | zouzou | when i pickup a ringing phone (*8) a 3 way call is established between the 3 phones ? |
| 07:13.35 | zouzou | anyone know why? |
| 07:14.04 | |nix | Math`: i've been trying to set it up, bu faces problem with extensions that begin with 0. any other prefixes, like 1xxx, 2xxx all work perfect |
| 07:14.33 | |nix | Sep 27 15:14:24 NOTICE[28014]: chan_iax2.c:7323 socket_read: Rejected connect attempt from 10.10.50.50, request '0902@management' does not exist |
| 07:15.01 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:15.19 | Math` | does 0902@management exists? |
| 07:15.23 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 07:15.26 | |nix | Math`: yes it does. |
| 07:15.40 | Math` | paste the line |
| 07:15.52 | |nix | its in realtime |
| 07:16.22 | |nix | -- Executing Macro("SIP/0902-0815edd8", "stdexten|1902|Staff|20|0|0|0") |
| 07:16.28 | |nix | i just did a SIP call to 1902 |
| 07:21.09 | linuxbangalore | how to solve this notice |
| 07:21.09 | linuxbangalore | res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 07:22.46 | |nix | here is the realtime line - stdexten|0902|management|10|0|0|0 |
| 07:24.38 | Sir_Diddymus | linuxbangalore: did you change your system time after starting asterisk? should be something like that i think... |
| 07:25.49 | linuxbangalore | no.. i did not but after gooogling for some time I found that it is a problem with mpg123.. on amd64bit...as it does nto complie well on 64bit |
| 07:27.41 | scanna | hi all, can someone help me with call transfer and cdr? |
| 07:30.35 | |nix | Math`: any help? |
| 07:34.23 | DarKnesS_WolF | scanna: ask ur question directly |
| 07:35.49 | *** join/#asterisk hatamen (i=hatamen@222.183.27.247) |
| 07:36.02 | hatamen | hi,all |
| 07:36.17 | hatamen | i have a meetme problem |
| 07:36.48 | hatamen | i have a conf =>1234 in meetme.conf |
| 07:37.38 | hatamen | but, -- Playing 'conf-invalid' (language 'en') , why!!! |
| 07:38.04 | hatamen | who can help me? :) |
| 07:38.24 | scanna | A dial B and B transfer the call to C |
| 07:39.14 | scanna | B and C are sip phones, A is an external phone that use a zap line |
| 07:41.03 | scanna | in cdr the callerId of A when dial B is empty |
| 07:41.15 | scanna | why? |
| 07:46.11 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:46.44 | *** join/#asterisk T42X (n=tadas@omega.vic.lt) |
| 07:49.10 | inspired | hmm, if I use 15 as timeout in Dial, will it continue at next priority after 15 seconds and no answer, or do I need to mess with DIALSTATUS etc? |
| 07:50.18 | hatamen | <PROTECTED> |
| 07:53.13 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 07:53.16 | hatamen | localhost*CLI> |
| 07:53.16 | hatamen | <PROTECTED> |
| 07:53.16 | hatamen | <PROTECTED> |
| 07:53.17 | hatamen | [Sep 27 16:23:45] WARNING[3192]: chan_zap.c:921 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
| 07:53.17 | hatamen | [Sep 27 16:23:45] ERROR[3192]: chan_zap.c:7612 chandup: Unable to dup channel: No such file or directory |
| 07:53.17 | hatamen | [Sep 27 16:23:45] WARNING[3192]: app_meetme.c:611 build_conf: Unable to open pseudo channel - trying device |
| 07:53.19 | hatamen | [Sep 27 16:23:45] WARNING[3192]: app_meetme.c:614 build_conf: Unable to open pseudo device |
| 07:53.21 | hatamen | [Sep 27 16:23:45] WARNING[3192]: chan_zap.c:921 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
| 07:53.24 | hatamen | [Sep 27 16:23:45] ERROR[3192]: chan_zap.c:7612 chandup: Unable to dup channel: No such file or directory |
| 07:53.26 | hatamen | [Sep 27 16:23:45] WARNING[3192]: app_meetme.c:611 build_conf: Unable to open pseudo channel - trying device |
| 07:53.28 | hatamen | [Sep 27 16:23:45] WARNING[3192]: app_meetme.c:614 build_conf: Unable to open pseudo device |
| 07:53.30 | hatamen | <PROTECTED> |
| 07:53.32 | hatamen | <PROTECTED> |
| 07:53.34 | hatamen | localhost*CLI> |
| 07:54.16 | inspired | you don't have zaptel |
| 07:54.21 | hatamen | what's the wrong? |
| 07:54.23 | inspired | you need it for timing |
| 07:54.50 | hatamen | meetme must need Zaptel support ? |
| 07:54.53 | inspired | and you also need a device to get timing correctly |
| 07:54.54 | inspired | yes |
| 07:55.23 | inspired | or you could use zaprtc or ztdummy to emulate a timing device |
| 07:55.49 | hatamen | thanks ! I will try! |
| 07:55.57 | inspired | download zaptel. make ztdummy or something like that |
| 07:56.01 | inspired | and make install ztdummy I guess |
| 07:56.03 | inspired | never used it though |
| 08:02.10 | CtRiX | or use app_conference which does not need timers. |
| 08:02.16 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 08:02.42 | hatamen | make: *** [ztdummy] Error 1 |
| 08:02.43 | hatamen | :( |
| 08:03.56 | linagee | inspired: what is the diff? ztdummy and zaprtc? |
| 08:04.09 | Stephnie | I want to dial 2 number at the same time using net2phone ...is this correct ? Dial(SIP/19496666921@nettophone&SIP/19496611321@nettophone) |
| 08:04.20 | linagee | hatamen: are you using trixbox, or straight asterisk? |
| 08:04.21 | inspired | ztdummy is made by digium, zaprtc by someone else |
| 08:04.28 | linagee | hm |
| 08:04.39 | inspired | can't help you with that hatamen. never used it |
| 08:05.25 | linagee | Stephnie: why dial two numbers at the same time? |
| 08:06.24 | Stephnie | want to ring at two numbers...and connected to the one who picked up first... |
| 08:06.53 | *** join/#asterisk RoyK (n=roy@ti211210a080-1773.bb.online.no) |
| 08:07.27 | Stephnie | yes...I think that Dial syntax is correct |
| 08:08.45 | drcode | is there open source project to use skype and astrikes that work? |
| 08:08.48 | drcode | under linux env |
| 08:09.54 | tzafrir | there basically can't be, unless you manage to reverse-engeneere the skype protocol |
| 08:10.42 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 08:10.51 | drcode | k |
| 08:11.09 | tzafrir | The license of skype basically requires that you have an interactive skype client running |
| 08:11.13 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 08:12.51 | tzafrir | Go for sip or for Jingle |
| 08:13.18 | RoyK | is the skype protocol really that hard to understand? |
| 08:13.38 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 08:13.40 | RoyK | since there've been so much talk about it, i mean |
| 08:13.54 | RoyK | but still I haven't seen a single project doing it |
| 08:13.55 | james_ | there's more problems than just reverse engineering the protocol |
| 08:14.01 | jeremy_g | quick question: does default context really mean a default context always processed by *? |
| 08:14.13 | RoyK | james_: what are those? |
| 08:14.29 | jeremy_g | please tell quickly, i am in a middle of an important task |
| 08:14.37 | james_ | are wideband codecs still a problem in asterisk? |
| 08:14.55 | james_ | dunno, haven't really kept up with dev for a while |
| 08:15.24 | jeremy_g | is [default] just another user defined context, or is default a keyword and having things inside it makes things different? |
| 08:15.29 | RoyK | james_: yes.... asterisk can't do 16kHz sample rate |
| 08:16.12 | RoyK | jeremy_g: if the (whateverthenameofthatfunctionwas()) is called to jump to a context with the context given as NULL, asterisk jumps to [default] |
| 08:16.17 | james_ | so there's 3 issues... the licensing of skype and it's protocol, the reverse engineering of the protocol, and asterisk's ability to support the protocol |
| 08:16.19 | RoyK | which of course is bloody ugly |
| 08:17.37 | linagee | james_: skype will find a way to recognize your client, then blackhole it to oblivion. |
| 08:17.39 | RoyK | (that last one was for jeremy_g ) |
| 08:18.09 | james_ | yeah, that's covered in the first issue :P |
| 08:18.24 | *** join/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 08:18.29 | linagee | james_: they could have an auto updater where they can tell all the clients, "ok, today we are going to append 'official skype client' to the end of all transmissions". then they tell their skype servers to look for it and bam. borked protocol. |
| 08:19.23 | james_ | indeed |
| 08:19.30 | james_ | so everything's against asterisk supporting skype |
| 08:20.05 | linagee | maybe you could somehow hack skype so you're using their binaries, running within wine, and somehow get asterisk to present a sound card interface and click the buttons on demand, but it would be very hard and annoying to use. |
| 08:20.25 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 08:21.12 | james_ | skypes api supports linux |
| 08:21.25 | linagee | skype api? |
| 08:21.27 | *** join/#asterisk xnon (i=xnon@200.82.222.159) |
| 08:21.34 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 08:21.44 | linagee | james_: well there you go then. make a skype api -> asterisk bridge |
| 08:21.54 | james_ | https://developer.skype.com/Docs/ApiDoc/Using_the_Skype_API_on_Linux |
| 08:22.03 | james_ | haha, i'm not the one interested in doing it |
| 08:22.26 | xnon | hello anybody speak spanish here? |
| 08:22.26 | jeremy_g | Thanks RoyK. From the web i learnt "The context "[default]" should also always be there, this is where Asterisk starts to look" |
| 08:22.33 | xnon | i need some help |
| 08:22.42 | james_ | i wouldn't bother with anything else but SIP |
| 08:23.04 | jeremy_g | As you say, its not the case rather asterisk would only go to default in case a func() has a context missing |
| 08:23.33 | *** join/#asterisk klem_ (n=klem@klem.estpak.ee) |
| 08:23.44 | RoyK | jeremy_g: not exactly right. [default] is a context that should indeed always be there, but should hold one line exten => _X.,1,Hangup |
| 08:24.06 | RoyK | [default] is just silly |
| 08:24.08 | xnon | my phone HandyTone later of some time is away status |
| 08:24.33 | xnon | this is for the register expiration? |
| 08:24.39 | xnon | 3600 |
| 08:24.43 | xnon | ??? |
| 08:25.06 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:25.17 | jeremy_g | RoyK:thanks again!it was more confusing than i expected. now i know. |
| 08:25.23 | linagee | intersting.... nufone.net is in chicago, but i get a relatively low ping... nice |
| 08:25.46 | RoyK | jeremy_g: just define a context name other than default, and keep default as 'don't go here' |
| 08:26.04 | RoyK | nuphone is evil |
| 08:27.32 | jeremy_g | RoyK:i am actually trying to understand someone else's configuration, who has included other contexts in default. those context are not included by any other context in the program and i was wondering about its security |
| 08:27.53 | StyleWarz | Anyone can recommend any specific hfc-s card? |
| 08:28.20 | linagee | RoyK: nuphone is evil? how so? :( |
| 08:28.21 | jeremy_g | RoyK:there are [outgoing-sip]... was wondering if someone can really land into it and make calls? |
| 08:29.03 | linagee | are all voip termination services evil? :( |
| 08:31.23 | RoyK | StyleWarz: they're all the same |
| 08:32.24 | StyleWarz | RoyK: Hm i just need some name to look for :) |
| 08:32.41 | RoyK | billion, for instance |
| 08:32.45 | RoyK | there are several |
| 08:33.28 | *** join/#asterisk oQPa (n=Spain@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 08:34.58 | StyleWarz | i think the Longshine LCS-8051 should be enough right? |
| 08:35.34 | RoyK | omfg. trying to report a bug to digium as like trying to beg from a beggar |
| 08:36.34 | RoyK | getting answers like "the modifcations you are making to one makefile don't have the same effect as make don-optimize" |
| 08:36.51 | RoyK | when I've changed -O6 to -O0 |
| 08:37.05 | RoyK | strangely the makefile does the exact same thing |
| 08:37.10 | RoyK | bastards |
| 08:39.58 | benjk | RoyK, the default context should have two lines |
| 08:40.25 | benjk | one with a NoOp() to print some info on how tried to make the call |
| 08:40.27 | *** part/#asterisk cornet (n=nathan@cpc3-heck3-0-0-cust802.hudd.cable.ntl.com) |
| 08:40.30 | benjk | then Hangup |
| 08:40.43 | RoyK | verbose() perhaps |
| 08:40.54 | benjk | or that |
| 08:41.28 | benjk | if you do have something go in to default, you probably want to know a little about the circumstances, so print or log some info before hanging up |
| 08:41.36 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:43.37 | ichilton | any cisco 79xx users around? |
| 08:44.10 | linagee | ack. can anyone recommend an iax provider on the west coast with a low ping? |
| 08:44.29 | linagee | that sounds like a tall order i guess |
| 08:45.07 | benjk | West Coast? Kenia? |
| 08:45.18 | benjk | or Sydney? |
| 08:49.02 | RoyK | Bergen |
| 08:49.11 | RoyK | Dublin, perhaps |
| 08:49.20 | RoyK | hm. no that's not west coast |
| 08:49.22 | RoyK | Limerick |
| 08:49.45 | benjk | West Coast of Antarctica |
| 08:49.49 | RoyK | or Bayonne |
| 08:49.52 | RoyK | lol |
| 08:50.06 | benjk | that's a tough one |
| 08:50.17 | benjk | where is the West Coast of Antarctica? |
| 08:51.21 | hatamen | inspired ~~~ Thank you very much! I did it ~ |
| 08:51.34 | inspired | :) |
| 08:52.20 | james_ | haha |
| 08:52.27 | james_ | benjk: sydney is far from the west coast :P |
| 08:52.32 | rob0 | Perth |
| 08:53.04 | benjk | I know |
| 08:53.42 | rob0 | Perhaps the western hemisphere part of Antartica could be called "west coast" |
| 08:53.54 | benjk | perhaps |
| 08:55.02 | benjk | heh |
| 08:55.15 | RoyK | never trust inspired |
| 08:55.25 | inspired | :O |
| 08:55.45 | RoyK | :) |
| 08:56.02 | jeremy_g | RoyK:i am so surprised. chan_local.c on line 498 (asterisk 1.2.10) uses [default] context as temporary context |
| 08:56.29 | RoyK | welll. you have a @asdf style in chan_local |
| 08:57.00 | jeremy_g | yup |
| 08:57.23 | jeremy_g | wtf is default doing in chan_local.c ? |
| 08:58.08 | jeremy_g | does that break ur definition of RoyK |
| 08:58.15 | jeremy_g | of default |
| 09:00.07 | stoffell | hm, is there a way to change the separators used in "show channels concise" ? |
| 09:00.43 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 09:01.44 | benjk | yes there is |
| 09:01.57 | benjk | edit, make, make install |
| 09:03.06 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.57 | stoffell | benjk: hm, according to bugs.digium.com it should be changed already? -> http://bugs.digium.com/view.php?id=6086 |
| 09:04.17 | linagee | bah. there is only one west coast. :-D |
| 09:04.22 | linagee | US of A. :) |
| 09:04.46 | linagee | maybe i should have said "wess side" that might have made more sense. :) |
| 09:04.54 | RoyK | jeremy_g: no. it only underlines the definition of the asterisk code, "INCONSISTENT" |
| 09:05.40 | RoyK | linagee: the underlined shite of america? |
| 09:06.25 | linagee | ha ha ha |
| 09:06.41 | linagee | united oil stealers of america |
| 09:06.47 | linagee | :-D |
| 09:08.18 | linagee | i don't get the saying, "free as in beer"... beer isn't free.... |
| 09:09.13 | stoffell | benjk: okay, it's changed in 1.4, so i'll manually adjust in my 1.2 setup, tnx |
| 09:10.41 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:16.26 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:17.33 | ichilton | any cisco 79xx users around? |
| 09:19.08 | stoffell | ichilton: probably there are.. |
| 09:19.14 | darkskiez | ichilton: 'sup |
| 09:20.50 | Nugget | I've got a few 7960s |
| 09:23.33 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 09:24.24 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 09:25.02 | ichilton | ah, cool |
| 09:25.05 | darkskiez | anybody have any ketchup ? |
| 09:25.07 | ichilton | I just got one yesterday |
| 09:25.14 | ichilton | (a 7940) |
| 09:25.24 | ichilton | I've got it working with asterisk for outgoing calls but it wont work for incoming |
| 09:25.59 | ichilton | and there is a little cross next to the line button |
| 09:26.05 | darkskiez | ah, that'd explain it |
| 09:26.11 | Nugget | sounds like it's failing to register correctly |
| 09:26.14 | darkskiez | its not registering |
| 09:26.23 | ichilton | ian/ian (Unspecified) D 255.255.255.255 0 UNKNOWN |
| 09:26.39 | ichilton | the same configuration works with my budgettone |
| 09:26.47 | darkskiez | yeh u must have it wrong |
| 09:26.50 | darkskiez | on the phone |
| 09:26.52 | ichilton | (although the cisco wont connect at all with a numeric username, only a text one) |
| 09:26.52 | darkskiez | then |
| 09:27.02 | ichilton | any ideas how I find out what's wrong? |
| 09:27.19 | Nugget | watch the asterisk console and see if you are seeing the failed registration |
| 09:27.21 | ichilton | the voip-info.org page said you could telnet to the cisco's but I can't - there are no ports open on it according to nmap |
| 09:27.21 | Nugget | telnet is eeeeeeevil! |
| 09:27.36 | ichilton | not seeing anything on the asterisk console |
| 09:27.38 | Nugget | you can only telnet to the cisco phones if you have that enabled in the config. |
| 09:27.38 | darkskiez | ichilton: you have to enabled it in the config. |
| 09:28.06 | ichilton | until I try to call it that is, when I get: |
| 09:28.07 | ichilton | <PROTECTED> |
| 09:28.07 | ichilton | Sep 27 09:54:31 NOTICE[14990]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 09:28.14 | ichilton | darkskiez: where is that option? |
| 09:28.23 | Nugget | telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged |
| 09:28.34 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:28.34 | Nugget | and you'll also want to set phone_password |
| 09:28.36 | ichilton | humm |
| 09:28.36 | darkskiez | ^-- that in SIPDefault.cnf |
| 09:28.43 | ichilton | what is SIPDefault.cnf? :) |
| 09:28.50 | Nugget | heh |
| 09:28.57 | ichilton | I just set everything on the UI on the phone |
| 09:28.57 | Nugget | how did you configure the phone? |
| 09:29.01 | Nugget | ah, dunno then. |
| 09:29.13 | ichilton | preseed unlock, then went to sip settings and entered the details |
| 09:29.30 | ichilton | wonder if that's my problem :) |
| 09:29.35 | ichilton | how do you guys do it? |
| 09:31.57 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-104978.otenet.gr) |
| 09:32.21 | jgoo | Hey guys, the TDM400P, I have one here with 4 red FXO modules |
| 09:32.47 | jgoo | does that mean I cannot plug this into the wall unit (no fxs?) I am worried it does... |
| 09:33.07 | *** join/#asterisk drbeaker (n=dave@150.2.169.217.in-addr.arpa) |
| 09:33.09 | *** join/#asterisk Zouzou (n=test@mail.splendor.net) |
| 09:33.24 | drbeaker | hi folks |
| 09:33.35 | jgoo | hi drbeaker |
| 09:33.49 | drbeaker | can anyone help me set up asterisk to work with SIP on (for now) a local network |
| 09:33.56 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 09:34.08 | drbeaker | ive read what docs I can find and im getting nowhere |
| 09:34.23 | tengulre11 | hi,all |
| 09:34.39 | tengulre11 | is it need SS7 gateway if I select E1 card? |
| 09:34.46 | drbeaker | is there an idiot's guide anywhere? |
| 09:34.54 | jgoo | either of you two familiar with the fxo/fxs modules? I have a tdm400p card, 4 red modules, fxo I believe |
| 09:35.17 | jgoo | drbeaker: lots of idiots guides, that is one reason I have gotten as far as I have, but with isdn card, now I am on analogue testing :s |
| 09:35.27 | hads | Yes, red = FXO |
| 09:35.38 | tzafrir | jgoo, FXO units are the ones you connect to the telco. no worry |
| 09:35.40 | drbeaker | lol can u suggest one tha works? |
| 09:35.57 | jgoo | haos: damn, so this card is completely useless |
| 09:36.09 | ichilton | Nugget / darkskiez: still around? |
| 09:36.10 | jgoo | haos/hads |
| 09:36.24 | hads | Useless why? |
| 09:36.41 | jgoo | it cannot connect to the PSTN socket |
| 09:36.50 | jgoo | only connect office phones to it right? |
| 09:36.51 | darkskiez | ichilton: with a tftp-server and dhcp server, the phones dont work amazingly well if configured from the phone itself. |
| 09:36.59 | hads | ~FXOFXS |
| 09:37.03 | jbot | [fxofxs] An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 09:37.30 | hads | Like tzafrir said, other way around :) |
| 09:37.32 | jgoo | aaah, somewhere along the line I used S for signal and O for office to distinguish them |
| 09:37.38 | linagee | the bot should tell you what happens if you plug an FXS into a landline port and get a ring. :-D |
| 09:37.44 | jgoo | so I can plug this into 4 wall sockets, but no phones into it. |
| 09:37.47 | linagee | RELEASE THE MAGIK SMOKE! :-D |
| 09:37.51 | linagee | lol |
| 09:38.00 | hads | linagee: :) |
| 09:38.09 | jgoo | so red is FXO, which means you connect to the wall |
| 09:38.37 | hads | jgoo: Correct |
| 09:38.47 | jgoo | thank buggery for that, so it ain't useless |
| 09:39.00 | linagee | jgoo: i like to think of it like S = supply and O = something else. :-D |
| 09:39.19 | linagee | jgoo: (supply as in voltage supply. sometimes denoted as VSS) |
| 09:39.38 | jgoo | erm acronymfinder says foreign exchange offce or French Xtrem Overclockers |
| 09:39.41 | jgoo | :p |
| 09:40.10 | linagee | foreign exchange orifice? |
| 09:40.23 | ichilton | darkskiez: I can do tftpd |
| 09:40.33 | ichilton | darkskiez: do you have an examples of the files I need please? |
| 09:40.34 | jgoo | okilydokily folks, I am happier! linagee: that is what happens when you travel across europe :) |
| 09:40.35 | linagee | jgoo: french. lol |
| 09:40.45 | linagee | jgoo: ??? |
| 09:40.55 | jgoo | <linagee> foreign exchange orifice? |
| 09:41.07 | darkskiez | ichilton: voip-info and google has everything. |
| 09:41.11 | linagee | jgoo: what's in europe that makes that funnny? lol. or do i want to know |
| 09:41.14 | darkskiez | ichilton: make an attempt first. |
| 09:41.52 | drbeaker | can anyone suggest a guide (that works) to setting up asterisk using sip? or give me some clue as to how to start debugging a setup that doesnt? |
| 09:41.56 | ichilton | darkskiez: ok, thanks |
| 09:42.15 | ichilton | darkskiez: once you have tftp'd a config file up there, does it remember it or do it every time it boots? |
| 09:42.22 | linagee | drbeaker: try getting the server out of the NAT. hehehe |
| 09:42.25 | ichilton | darkskiez: i.e could you set it up and then move it to a network without tftp? |
| 09:42.39 | drbeaker | its on a local network so no NAT |
| 09:42.58 | linagee | drbeaker: connect to sip what then? phones? |
| 09:43.30 | drbeaker | im testing it with x-lite as a softphone but it doesnt seem to connect |
| 09:43.37 | linagee | *shrug* |
| 09:43.39 | jgoo | linagee:europe: easy access to lots of different cultures and females :-D, excuses to try new beers |
| 09:43.43 | linagee | try the tutorial. :-D |
| 09:43.46 | jgoo | ok back to work |
| 09:43.52 | linagee | jgoo: females? |
| 09:44.00 | drbeaker | ive done a telnet to port 5060 and that doesnt seem to be open (but it may use udp?) |
| 09:44.22 | drbeaker | and yeah, i have started astersk ;) |
| 09:44.26 | festr__ | hello |
| 09:45.05 | hads | drbeaker: Asterisk uses UDP |
| 09:45.16 | hads | (for SIP) |
| 09:45.19 | drbeaker | yeah i thought it might |
| 09:45.53 | jgoo | does jbot have a quote file? :p |
| 09:45.57 | jgoo | ~quote |
| 09:45.58 | tzafrir | use netstat -lnup |
| 09:46.05 | hads | netstat -planu |
| 09:46.08 | drbeaker | at the moment i'd be happy to get to x-lites to talk to each other over the local network |
| 09:46.20 | festr__ | i've problem with asterisk 1.2 sip.conf register =>. I've two asterisk servers and A register with B this is ok. But if B calls to A (Dial/SIP/festr) on A i see Failed to authenticate user "242413010" altough i Dial/festr (242413010 is CIDNUM) |
| 09:46.23 | tzafrir | hads, what's the point of using both -a and -l ? |
| 09:46.24 | hads | Oh, tzafrir bet me to it :) |
| 09:46.52 | hads | tzafrir: Habit I always remember it as -plant |
| 09:47.04 | festr__ | s/Dial\/festr/Dial(SIP/festr) |
| 09:47.23 | hads | So no point at all, just a bad habit of mine |
| 09:47.25 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 09:49.06 | drbeaker | ok - it's listening so that's a start |
| 09:53.20 | *** join/#asterisk ahewitt (n=ahewitt@203-166-225-64.dyn.iinet.net.au) |
| 09:54.38 | ahewitt | hi guys...I am trying to get a sip trunk connecting to my VoiP provider and I can see the traffic leaving the asterisk box on port 4569 going to the sip server on port 4569 but not returning....can someone tell me what the port 4569 is supposed to be for and if I am expected to see return traffic on that port? |
| 09:55.42 | Zouzou | what does register a call mean? |
| 09:56.34 | jeremy_g | is this supported in asterisk1.2-10 (exten=>701,2,Goto(s-${DIALSTATUS}) |
| 09:56.35 | jeremy_g | ) |
| 09:56.53 | drbeaker | when i try to place a call I get "handle_request_invite: Unable to create/find SIP channel for this INVITE" |
| 09:57.03 | jeremy_g | this goto dialstatus stuff |
| 09:57.31 | hads | jeremy_g: Yes |
| 09:58.00 | drbeaker | what's it mean? what do i need to do to fix it |
| 10:06.57 | jeremy_g | had:http://pastebin.ca/183866 what am i doing wrong? it shud work |
| 10:07.33 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 10:07.42 | RoyK | I hate reporting asterisk bugs. WHATEVER you do to collect facts and do it the right way, someone is bound to ask you to do it otherwise. it's like "nowedonthavenaybugsinasteriskandweknowitandyourewrongandallwereallywanttodoisbitchyouarounduntilyougiveup" |
| 10:08.34 | darkskiez | I'm scared off reporting potential bugs as i dont want negative karma |
| 10:08.59 | darkskiez | "technical support issue" |
| 10:12.16 | darkskiez | I cant see how thats a good environment to promote |
| 10:13.11 | darkskiez | i wouldnt mind if digium had a scoring thing, but either only a) positive only, b) for internal review only |
| 10:13.47 | RoyK | only you don't get positive karma unless you suck digium dick |
| 10:14.13 | darkskiez | yay for bitterness |
| 10:14.17 | RoyK | that is, a few points perhaps, but not like +200 like some people have |
| 10:14.25 | RoyK | nonono |
| 10:14.40 | RoyK | I just think the whole karma thing is bullshit |
| 10:14.40 | mut | butt secks |
| 10:15.06 | mut | what site are you talkin about? |
| 10:15.34 | RoyK | bugs.digium.com |
| 10:15.34 | RoyK | ? |
| 10:15.34 | darkskiez | RoyK: I'm not going to argue. |
| 10:15.44 | mut | o |
| 10:16.01 | mut | it has karma? |
| 10:16.07 | darkskiez | very little |
| 10:16.33 | mut | where do ya even see someons karma |
| 10:16.53 | darkskiez | mut, change t,1 to 702,n |
| 10:17.36 | darkskiez | jeremy_g: change t,1 to 702,n |
| 10:17.42 | ahewitt | do any of you guys know if it is possible to use a voip enabled adsl modem in bridged mode with the asterisk server making the pppoe connection for the ADSL, then setup a local network between the ethernet interface on the modem and the asterisk server and have the VoIP ports register with the asterisk server? |
| 10:18.20 | ahewitt | o_0 |
| 10:18.32 | ahewitt | thats not as out there as it sounds |
| 10:19.05 | RoyK | chan_htcpcp, perhaps |
| 10:19.07 | darkskiez | ahewitt: I see what you're trying to do, I guess if the modem has a private address thats ok |
| 10:19.36 | ahewitt | darkshiez: hrmm...well I guess thats not the problem then....cheers |
| 10:20.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:20.37 | jeremy_g | i have a confusion, the extension t gets executed when the no digit is entered. how can this time out be changed by using SomeSetDigitTimeout() sort of function... |
| 10:21.02 | jeremy_g | is it true that t gets executed on failing to type in a digit |
| 10:21.43 | *** join/#asterisk Red15 (n=RedXV@d54C323D1.access.telenet.be) |
| 10:22.59 | RoyK | jeremy_g: show function TIMEOUT |
| 10:30.38 | *** part/#asterisk Red15 (n=RedXV@d54C323D1.access.telenet.be) |
| 10:39.34 | *** join/#asterisk hermuli (n=Eladamri@a88-112-252-73.elisa-laajakaista.fi) |
| 10:42.10 | ichilton | darkskiez: yay, got it working - thanks! |
| 10:43.34 | *** join/#asterisk RoyK (n=roy@ti211210a080-1773.bb.online.no) |
| 10:45.58 | X-Rob | anyone in here speak spanish? |
| 10:46.20 | DarKnesS_WolF | any recomendation about any IAX / SIP java applet ? |
| 10:46.36 | X-Rob | DarKnesS_WolF, java doesn't do UDP |
| 10:46.39 | X-Rob | so you can't 8) |
| 10:47.47 | darkskiez | java does do UDP! |
| 10:49.23 | DarKnesS_WolF | X-Rob: hmmm strange i'm not java expert |
| 10:49.31 | DarKnesS_WolF | X-Rob: i have failed compiling jiaxclient |
| 10:49.36 | X-Rob | Oh, it does? |
| 10:49.38 | X-Rob | I thought it didn't. |
| 10:49.43 | X-Rob | (er than usual) |
| 10:50.18 | backblue | jaaaaaavvvvvaaaaaaaaaaaaaaaaaaaaa it's slow even just writing the name. |
| 10:51.05 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 10:51.51 | DarKnesS_WolF | what other way i can use to add voice support in any website? |
| 11:03.27 | tzafrir | X-Rob, there's a rather empty #asterisk-es |
| 11:04.17 | backblue | DarKnesS_WolF: what do you mean by voice support in your web-site? |
| 11:04.57 | X-Rob | tzafrir, yeah, I was in there, tried to ping the guy and he didn't respond. |
| 11:05.03 | X-Rob | ended up getting by with him using google translate 8) |
| 11:05.19 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 11:07.21 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 11:07.27 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-104978.otenet.gr) |
| 11:08.10 | jgoo | meh, the box I just bought to test this install, I boot it clean without anything and it cannot even find the SATA disk, freaking hell, every time someone else sticks a box toegther for you the chances of it working are 1:278279873 |
| 11:08.40 | jgoo | anyone know about SATA? maybe I will fix this myself instead of waiting for some hardware guy to come and start poking it |
| 11:08.53 | X-Rob | plug two cables in |
| 11:09.00 | X-Rob | one goes to the power supply, other goes to the motherboard |
| 11:09.04 | X-Rob | that's about it. |
| 11:09.22 | X-Rob | Make sure it's enabled in the bios and that's about it. |
| 11:09.50 | jgoo | check, and check :( |
| 11:10.01 | X-Rob | different sata socket on the motherboard? |
| 11:10.06 | jgoo | I was worried there was a trixbox SATA issue, but that would be madness |
| 11:10.11 | jgoo | I could try that, thanks |
| 11:10.14 | X-Rob | can the bios see the drive? |
| 11:11.42 | jgoo | well, 'see' it, it lists is as a boot device and as a drive, yes |
| 11:12.10 | jgoo | I moved to SATA2 socket, but the problem is this is going to be a trixbox so I ain't got a screen on it, gotta move mine, and now I am deep in email :s |
| 11:12.15 | jgoo | so ill know in a mo |
| 11:12.18 | X-Rob | then it's a kernel issue |
| 11:12.20 | X-Rob | use ubuntu |
| 11:12.28 | jgoo | X-Rob, but POST complains about no drives |
| 11:12.29 | X-Rob | www.aussievoip.com/freepbx-ubuntu |
| 11:12.42 | X-Rob | how can it complain about no drives when it sees it? |
| 11:12.43 | fourcheeze | hmm |
| 11:12.47 | jgoo | I googled trixbox and SATA, didn't find any notice |
| 11:12.51 | jgoo | trixbox 1.2 |
| 11:12.53 | fourcheeze | my e61 registers fine with asterisk but hangs when I do an internet call :/ |
| 11:12.54 | X-Rob | does it say NO BOOT DEVICE OR DRIVE INACCESSABLE or something like that? |
| 11:13.09 | fourcheeze | any ideas? |
| 11:13.16 | *** join/#asterisk dacleric (n=dacleric@p54823CA1.dip0.t-ipconnect.de) |
| 11:13.18 | jgoo | first, it says name of driver company.... SATA driver... |
| 11:13.23 | jgoo | detecting drives... no drive detected |
| 11:13.31 | jgoo | (that is before BIOS can show up) |
| 11:13.42 | fourcheeze | e61 is a nokia smartphone |
| 11:13.47 | X-Rob | you don' thave sata on the motherboard? |
| 11:14.15 | X-Rob | secondly, don't use trixbox 1.2, it's broken in lots of ways |
| 11:14.17 | jgoo | X-Rob, yes I do, I just moved cable from SATA1 to SATA2, it is anew box picked up 20 minutes ago from a fairly reliable PC store |
| 11:14.18 | fourcheeze | e61: http://www.europe.nokia.com/nokia/0,,81718,00.html |
| 11:14.30 | X-Rob | thirdly, the topic says 'trixbox, A@h support in #freepbx' |
| 11:14.59 | X-Rob | <X-Rob> does it say NO BOOT DEVICE OR DRIVE INACCESSABLE or something like that? |
| 11:15.05 | X-Rob | that's an important question I'm waiting for you to answer |
| 11:15.19 | X-Rob | because you said "<jgoo> well, 'see' it, it lists is as a boot device and as a drive, yes" |
| 11:15.34 | jgoo | X-Rob, ah, 1.1 is better? I will go back there |
| 11:15.46 | X-Rob | jgoo, if you're having driver issues, ubuntu us the way to go |
| 11:15.50 | jgoo | my answer was to that question, it does say no boot device when I boot the machine emtpy yes |
| 11:15.57 | X-Rob | well that's normal |
| 11:16.00 | fourcheeze | anyone else got an e61 working with asterisk? |
| 11:16.11 | X-Rob | that's what happens when you try to boot from a drive without any partitions |
| 11:16.37 | jgoo | X-Rob, it doesn't say no system boot disk, it says no boot device... ok, ill try the ubuntu based thing you recommended, is it like trixbox? |
| 11:16.42 | jgoo | (link was 040 btw) |
| 11:16.45 | jgoo | 040/404 |
| 11:17.17 | X-Rob | http://aussievoip.com/wiki/freepbx-Ubuntu |
| 11:17.20 | jgoo | v |
| 11:17.30 | jgoo | yeah I googled it :) |
| 11:17.38 | X-Rob | ubuntu is a distro, as is centos |
| 11:17.44 | X-Rob | trixbox is centos+freepbx+misc other bits |
| 11:18.05 | RoyK | trixbox == asstrix? |
| 11:18.12 | X-Rob | RoyK, heh |
| 11:18.35 | tzafrir | BTW: freepbx is making its way slowly into Debian . Hopefully will be in Etch... |
| 11:18.49 | X-Rob | tzafrir, who's driving it? |
| 11:19.00 | tzafrir | Diego, obviously |
| 11:19.04 | X-Rob | Good point 8) |
| 11:21.11 | fourcheeze | is trixbox some gui? |
| 11:21.21 | X-Rob | fourcheeze, freepbx is a gui for asterisk |
| 11:21.29 | X-Rob | trixbox is centos+asterisk+freepbx |
| 11:21.41 | X-Rob | www.freepbx.org |
| 11:21.44 | fourcheeze | ahh ok |
| 11:22.43 | in-pt | Hello Can anybody please tell me how many simultaneous calls asterisk can handle on a high end server machine??? |
| 11:22.54 | X-Rob | in-pt, 9. |
| 11:23.01 | X-Rob | maybe 10. |
| 11:23.34 | X-Rob | (for a more accurate answer, try googling 'asterisk sizing') |
| 11:23.46 | in-pt | X-Rob: and how many sip clients max i can register with it ? |
| 11:23.55 | X-Rob | 2 or 3. |
| 11:25.06 | DarKnesS_WolF | backblue: i want to make phone calls to me * from website |
| 11:26.29 | in-pt | X-Rob: your knowledge is wrong mr australian..i had already registered 30 sip users.. |
| 11:26.45 | X-Rob | <X-Rob> (for a more accurate answer, try googling 'asterisk sizing') |
| 11:26.48 | jeremy_g | :d |
| 11:30.02 | stoffell | hm, i'm using mISDN and iaxmodem for faxes.. but outgoing faxes always fail, the number doesn't get dialed completely... is there no overlapdial available? |
| 11:32.05 | merlinn | is X-Rob being deliberately obstructive? |
| 11:32.16 | qdk | X-Rob: hehe, good answers. :-) |
| 11:32.39 | X-Rob | merlinn, he's asking a stupid question. |
| 11:32.46 | X-Rob | and there's no such thing as stupid questions, only stupid people. |
| 11:32.52 | X-Rob | so therefore, he's stupid. |
| 11:33.10 | X-Rob | I'm trying to _increase_ his knowledge by encouraging him to google his question before exposing himself. |
| 11:33.15 | X-Rob | So therefore, I'm actually being nice. |
| 11:34.24 | in-pt | X-Rob: no you are not nice infact have mis understanding about your self ...you you were nice then you have to tell me go to google infact giving answers |
| 11:34.48 | merlinn | it was a slightly retarded way to go about saying 'google asshole' |
| 11:35.16 | in-pt | X-Rob: and what do you mean by exposing yourself ...is it a porn show happening here ..of exposing |
| 11:35.18 | X-Rob | merlinn, I always accept constructive criticism. However, his question is still stupid. |
| 11:35.27 | merlinn | i don't deny that |
| 11:35.46 | X-Rob | in-pt, I have a new car. How fast will it go? |
| 11:35.48 | qdk | in-pt: whats highend? what kind of calls? what codecs? some many missing factors. |
| 11:36.19 | X-Rob | and the answer to his question is, in fact, googling 'asterisk sizing' |
| 11:36.20 | qdk | X-Rob: no a new highend car. ;-) |
| 11:38.14 | X-Rob | heh |
| 11:38.20 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 11:40.08 | jeremy_g | what is a label of an extension? |
| 11:40.21 | jeremy_g | <PROTECTED> |
| 11:40.28 | jeremy_g | either i have gone made or * |
| 11:40.35 | jeremy_g | s/made/mad |
| 11:41.15 | qdk | jeremy_g: instead of a priority number you can have labels, and such is missing. |
| 11:41.18 | jeremy_g | when i clearly have , exten => s-CHANUNAVAIL,1,Dial(SIP/02323@somewhere,30,r) in the right context |
| 11:41.26 | tzafrir | is there n(s-${CHANSTATUS}) ? |
| 11:41.56 | RoyK | ~docs |
| 11:41.57 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 11:41.59 | RoyK | ~book |
| 11:42.00 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:42.06 | RoyK | ~lart jeremy_g for not rtfm |
| 11:42.27 | X-Rob | jeremy_g, you want to goto incoming-sip,s-CHANUNAVAIL,1 |
| 11:42.53 | X-Rob | or s-CHANUNAVAL,1 |
| 11:43.08 | jeremy_g | what a simple mistake me making, i had noted that Goto(haha) would go to prioritiy haha but some how i created an extension for that |
| 11:43.15 | tzafrir | rather than "priority" |
| 11:43.20 | jeremy_g | qdk: love u for reminding me, thanks man |
| 11:43.33 | jeremy_g | tzafrir:it says in the log file * |
| 11:44.44 | *** join/#asterisk xxxAGENTExxx (n=x@201.65.48.130) |
| 11:45.20 | qdk | jeremy_g: and if you didnt ask as you did, i could have saved you the time... and best of all, my own time. |
| 11:46.42 | jeremy_g | qdk:why is it that sometime a person forgets even the most basic things. is there a configuration editor for .conf files of * |
| 11:46.53 | jeremy_g | that could color code and drop down :p |
| 11:48.25 | jeremy_g | one line typo can fuck your whole day |
| 11:49.15 | jeremy_g | i was so fucked up because of this Goto(context,extension,priority) thingie. so obvious knew it, but mistyped Goto(extension) shud have Goto(extension,priority) |
| 11:49.44 | jeremy_g | damn! i need an editor for .confs |
| 11:49.48 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 11:49.54 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:49.54 | RoyK | jeremy_g: vi |
| 11:50.05 | RoyK | ~editor |
| 11:50.06 | jbot | ACTION 's favorite editor is vim, what else? |
| 11:50.21 | QbY_ | the tones you hear when you dial a wrong number ("do do do")--are those available in Asterisk? |
| 11:50.37 | RoyK | playtones(info)? |
| 11:50.46 | QbY_ | k. |
| 11:50.59 | RoyK | i don't know if it says 'do' but it does indeed indicate that something is wrong |
| 11:51.37 | QbY_ | hrmm.. playtones(info) does nothing for me |
| 11:51.45 | coppice | do do do do, da da da da, that's all I want to say to you. |
| 11:51.51 | QbY_ | ;) |
| 11:53.02 | qdk | QbY_: try putting it into * and see if * does something with it. ;-) |
| 11:54.37 | tzafrir | RoyK, so it is your idea to set joe=vim ? |
| 11:56.03 | tzafrir | QbY_, take a look at indications.conf for the names |
| 11:56.05 | QbY_ | qdk.. funny. |
| 11:56.25 | QbY_ | tzafrir.. I did just make sure it was there.. it is.. On the cli, I'm getting: Executing PlayTones("SIP/69.39.22.206-08a44d38", "info") in new stack |
| 11:56.27 | QbY_ | but.. nothing |
| 11:56.52 | tzafrir | Maybe you have no indications set? |
| 11:56.55 | QbY_ | indications.conf: info = !950/330,!1400/330,!1800/330,0 |
| 11:57.36 | QbY_ | exten => ruri_as1,n,Playtones(info) |
| 11:59.36 | *** join/#asterisk mr_canny (n=mr_canny@200.138.113.82) |
| 12:01.12 | mr_canny | I have one modem us. robotics and I cant to use with asterisk how fxo port ? |
| 12:02.01 | *** join/#asterisk riddlebox (n=blah@64.241.37.140) |
| 12:02.16 | QbY_ | Is there any thing else that must be turned on, flags set, prior to calling Playtones? |
| 12:03.17 | mr_canny | I cant use modem us robotics with asterisk ? working ? |
| 12:03.24 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 12:03.43 | mr_canny | I like one fxo port.... |
| 12:03.51 | coppice | I can't use a house brick with asterisk. isn't it annoying :-) |
| 12:04.02 | creativx | i can build a brick house with asterisk |
| 12:05.17 | coppice | ah, that's easy. just phone the builder and tell him to get started |
| 12:07.51 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
| 12:08.23 | QbY_ | w00t.. |
| 12:13.09 | RoyK | coppice: unless he can't hear you for the jitter |
| 12:16.39 | jeremy_g | QbY_:is playtones working for you |
| 12:16.44 | QbY_ | jeremy_g.. yes sir.. |
| 12:16.56 | *** join/#asterisk Niklas- (i=niklas@moo.dk) |
| 12:16.59 | QbY_ | A Wait(1) after Playtones is necessary |
| 12:17.13 | jeremy_g | QbY_:what module to load for Playtones |
| 12:18.08 | Niklas- | Hi, i have a little problem getting the voicemail to work with personally greetings. The greeting gets saved and i can play the file in a media player without any problems, but when the voicemail is activated for a user it plays the default greeting - anything i need to configurere? |
| 12:18.17 | QbY_ | uhm.. i do not know |
| 12:18.48 | *** join/#asterisk Zouzou (n=test@mail.splendor.net) |
| 12:18.50 | jeremy_g | QbY_:at your cli, do show modules |
| 12:19.17 | Zouzou | how can i tell asterisk which mp3 file tp play when music on hold? |
| 12:19.45 | X-Rob | put the different mp3's into different moh contexts |
| 12:20.22 | jeremy_g | is playtones available in asterisk1.2-10 |
| 12:20.40 | Zouzou | in musiconhold.conf????? |
| 12:20.43 | QbY_ | i'm most definitely sure it is |
| 12:20.52 | jeremy_g | yes it is |
| 12:21.00 | jeremy_g | QbY_:its available in modules res_indications |
| 12:21.57 | backblue | jeremy_g: yes |
| 12:22.16 | backblue | there or something like readdtmf() or seomthing? |
| 12:22.39 | X-Rob | Zouzou, 'show application setmusiconhold' |
| 12:23.39 | DrukenHME | Niklas-: are you putting a u infront of their mailbox? |
| 12:24.23 | Zouzou | expplain nothing |
| 12:24.55 | hwt | how can i find out which context an ip is being dropped into? |
| 12:25.11 | hwt | anyone got a good tutorial on how to use sipp and asterisk? |
| 12:27.49 | Niklas- | DrukenHME: yes |
| 12:28.09 | Niklas- | It's only needed in extensions.conf right? |
| 12:30.11 | DrukenHME | yes |
| 12:30.45 | DrukenHME | so if their mailbox was 101@default then you put u101@default for the unavailable and b101@default for busy |
| 12:30.55 | *** part/#asterisk QbY_ (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 12:31.17 | *** join/#asterisk lorinc (n=ang@caracas-3143.adsl.interware.hu) |
| 12:32.11 | Seba_soy | new hard work day! |
| 12:32.21 | Seba_soy | I will try my question again :) |
| 12:32.48 | DrukenHME | hardwork ? |
| 12:33.01 | DrukenHME | since when is computer work hardwork? :) |
| 12:33.30 | Seba_soy | I cant make a call from an ip-pbx to my asterisk if call comes with a FROM = to some user from my asterisk... |
| 12:33.37 | Seba_soy | I need some help |
| 12:33.46 | Seba_soy | jejej, I dont do only computer things |
| 12:34.05 | Seba_soy | for example |
| 12:34.27 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:34.52 | Seba_soy | I have a user [2000] into my asterisk, and a call from an pbx ip comes with "fromuser: 2000@ip_pbx" (it means both my asterisk box and ip pbx have same users definitions) |
| 12:34.57 | Seba_soy | then call is rejected |
| 12:35.38 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:35.55 | X-Rob | Seba_soy, no it's not. Check /var/log/asterisk/full for why it's being rejected really. |
| 12:38.06 | coppice | DrukenHME: computing work gets hard when you run out of fingers. even harder when you run out of toes as well |
| 12:38.59 | DrukenHME | coppice: very true, but thank christ we all have a tounge :) |
| 12:39.38 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:39.40 | coppice | hum. base 21 numbers :-\ |
| 12:40.18 | X-Rob | geez. |
| 12:41.11 | Seba_soy | X-Rob: I know why it is rejected |
| 12:41.22 | Seba_soy | I can see it in sip debug |
| 12:41.38 | Splat | you'd do better using your fingers in base 2.. you'll get upto higher numbers that way.. :P |
| 12:41.45 | Seba_soy | asterisk uses fromuser field to check if exist some user on his definitions |
| 12:41.53 | X-Rob | 21, 42, 63, 84, so (base10)100 would be 3f |
| 12:41.59 | X-Rob | it's sad that I worked that out. |
| 12:42.07 | Seba_soy | so, call is coming from 2000, and it is a user on my asterisk box TOO!! |
| 12:42.11 | X-Rob | Seba_soy, sip debug doesnt' tell you why it's rejected |
| 12:42.17 | X-Rob | /var/log/asterisk/full tells you why it's rejected. |
| 12:42.18 | Seba_soy | then, call is rejected because authorization |
| 12:43.02 | Seba_soy | how I actevate full on log asterisk |
| 12:43.37 | X-Rob | edit /etc/asterisk/logger.conf |
| 12:44.03 | Seba_soy | I have to activate debug => debug? |
| 12:44.25 | Seba_soy | messages => debug |
| 12:44.30 | Seba_soy | that is? |
| 12:44.57 | X-Rob | Seba_soy, I strongly suggest you read all of the files in /usr/src/asterisk/configs |
| 12:45.17 | X-Rob | then, after that, |
| 12:45.18 | X-Rob | ~docs |
| 12:45.27 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:45.27 | X-Rob | read ATFOT |
| 12:45.28 | X-Rob | ~atfot |
| 12:45.34 | X-Rob | ~tfot |
| 12:45.36 | jbot | rumour has it, tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 12:46.27 | Seba_soy | i know all the files X-Rob |
| 12:46.45 | X-Rob | No, you don't. |
| 12:47.03 | X-Rob | you've already proved that |
| 12:47.06 | DrukenHME | even the developers don't know ALL the files :) |
| 12:47.14 | X-Rob | I know all the files _exist_ |
| 12:47.16 | X-Rob | somewhere. |
| 12:47.18 | X-Rob | in the ether. |
| 12:47.29 | Seba_soy | I know files I have to know, of course |
| 12:47.34 | Seba_soy | I never activate debug |
| 12:48.05 | Seba_soy | config files sure |
| 12:48.06 | Seba_soy | not all |
| 12:48.36 | DrukenHME | i've been using asterisk for 2 years, and even i don't know all the config files :) |
| 12:48.37 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 12:48.38 | Seba_soy | I will read doc |
| 12:48.56 | DrukenHME | i know the ones i dip into everyday.... |
| 12:49.02 | Seba_soy | exactly |
| 12:49.07 | Seba_soy | that's what I want to say |
| 12:49.17 | Seba_soy | i know files I constantly work in |
| 12:49.52 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 12:50.08 | DrukenHME | hey X-Rob, do you know if i need todo anything special to make vmwi work with realtime? |
| 12:50.09 | *** join/#asterisk s0lid (n=jlq@124.6.191.22) |
| 12:50.20 | X-Rob | DrukenHME, we haven't touched realtime yet. |
| 12:50.35 | X-Rob | but I can't see why there would be anything 'special' |
| 12:50.43 | X-Rob | as long as you have the correct device in the voicemail table? |
| 12:50.56 | DrukenHME | device..... |
| 12:51.55 | DrukenHME | isn't it usually the otherway around? the device tells asterisk what it's mailbox is? |
| 12:51.59 | DrukenHME | or in sip.conf ? |
| 12:52.09 | DrukenHME | or the sipfriends table.... |
| 12:52.13 | X-Rob | uh yeah |
| 12:52.14 | X-Rob | sorry |
| 12:52.15 | X-Rob | don't mind me |
| 12:52.26 | X-Rob | correct voicemail in the device table |
| 12:52.28 | X-Rob | *sigh* |
| 12:52.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:52.37 | DrukenHME | it's still early :) |
| 12:52.42 | X-Rob | 11pm here |
| 12:52.50 | X-Rob | done bugger all today apart from fiddle with the website |
| 12:52.52 | DrukenHME | exactly, it's still early :) |
| 12:52.53 | DrukenHME | hehehe |
| 12:52.55 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 12:56.25 | Niklas- | DrukenHME: eya |
| 12:56.27 | Niklas- | done that |
| 12:57.16 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 12:57.22 | DrukenHME | are you sure the directory exsists? i've had problems with asterisk creating the voicemail directories lately |
| 13:01.37 | DrukenHME | oh, oops, one tooo many ss'sss in there :) |
| 13:04.18 | hmmhesays | heh |
| 13:04.30 | hmmhesays | people don't notice types like that |
| 13:04.56 | DrukenHME | :) i realize |
| 13:05.05 | DrukenHME | we notice, we just ignore |
| 13:05.43 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 13:05.49 | dezent | is it possible to make queues stop ringing an agent once he allready in a call ? |
| 13:05.54 | hmmhesays | the latest trixbox is freaking out on one of my machines |
| 13:06.45 | DrukenHME | is it turning trix? hehehe it'll have to call it's pimp if the john refuses to pay.... |
| 13:06.55 | dezent | i recived spam from a trixbox yesterday =) |
| 13:06.59 | hmmhesays | heh, yeah it is |
| 13:07.27 | hmmhesays | toying around with it, I like freepbx as a gui, and it has sugarcrm integrated, but ... man what a pain in the ass |
| 13:07.40 | hmmhesays | I can't reload sip |
| 13:08.11 | DrukenHME | dezent: depends on how you do it, if your using member in queues.conf then set the device to incominglimit=1, if your agent is logging in, then the queue should do it already |
| 13:09.09 | FlatFoot | afternoon all |
| 13:09.17 | FlatFoot | anybody used the snom320 ? |
| 13:09.24 | FlatFoot | got a weird problem |
| 13:09.42 | DrukenHME | sounds too much like snow, i wouldn't touch one... :) |
| 13:09.49 | FlatFoot | can get an identity to register np |
| 13:10.15 | FlatFoot | but as soon is i tell the identity it has a mailbox number thats when the prob appears |
| 13:10.26 | *** part/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 13:10.40 | FlatFoot | i get failed to authenticate and a stale nonce message |
| 13:10.43 | *** join/#asterisk onurx (n=ddsdd@85.98.79.130) |
| 13:10.59 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 13:11.05 | FlatFoot | any ideas ? |
| 13:11.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.59 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E274.versanet.de) |
| 13:12.44 | fourcheeze | FlatFoot: which firmware version? |
| 13:13.24 | FlatFoot | 6.2.3 |
| 13:13.47 | FlatFoot | same prob on some with 6.2.2 |
| 13:14.06 | fourcheeze | FlatFoot: do you mean the mailbox number on asterisk, or the one on the snom? |
| 13:14.45 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:14.45 | FlatFoot | on the snom , the * config looks quite normal |
| 13:14.45 | fourcheeze | FlatFoot: how many IP numbers on your * box? |
| 13:14.53 | FlatFoot | but a soon as i enter the mailbox number in the snom identity it goes all wrong |
| 13:14.56 | dezent | DrukenHME: i used members in queues.conf |
| 13:15.14 | FlatFoot | IP Numbers in what respect ? |
| 13:15.41 | fourcheeze | FlatFoot: ifconfig |
| 13:15.48 | fourcheeze | how many ip numbers listed |
| 13:15.58 | FlatFoot | fourcheeze: hang on |
| 13:16.01 | DrukenHME | dezent: so limit the devices to 1 in sip.conf |
| 13:16.09 | dezent | DrukenHME: hm ok |
| 13:16.45 | dezent | DrukenHME: thats no good solution.. then i need to change that everytime someone needs more lines |
| 13:16.49 | FlatFoot | fourcheeze: eth0 listed as normal , eth1 empty , lo as normal |
| 13:16.54 | dezent | DrukenHME: i think ill try with agi |
| 13:17.00 | DrukenHME | why would someone require more lines? |
| 13:17.09 | dezent | DrukenHME: i use more channels often |
| 13:17.12 | fourcheeze | FlatFoot: ok sometimes the stale nonce thing happens if * sends out on one IP and listens on another |
| 13:17.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.102) |
| 13:17.30 | FlatFoot | fourcheeze: but as far as i am aware there is only one network adapter |
| 13:17.37 | DrukenHME | how many people can ya talk to at once? |
| 13:17.56 | coppice | at a conference? :-) |
| 13:17.57 | fourcheeze | FlatFoot: each adaptor can have many IP numbers |
| 13:18.02 | fourcheeze | FlatFoot: however it sounds like it's not it |
| 13:18.13 | dezent | DrukenHME: its for my customers.. they want it |
| 13:18.14 | DrukenHME | coppice: i'm sure you know what i'm gettin at... |
| 13:18.16 | FlatFoot | fourcheeze: just seems strange that it only happens with the mailbox ID |
| 13:18.16 | [TK]D-Fender | DrukenHME: Don't interrupt the Voices when they are speaking!!!!! |
| 13:18.29 | fourcheeze | FlatFoot: you can define the key if you need to |
| 13:18.43 | DrukenHME | :P |
| 13:19.13 | fourcheeze | FlatFoot: are you talking about the "mailbox" field in the Login section of the line? |
| 13:19.39 | FlatFoot | fourcheeze: yes |
| 13:19.45 | [TK]D-Fender | coppice: I think it might almost be better worded as speaking "to" or "at", rather than "with" :) |
| 13:19.58 | DrukenHME | hey [TK]D-Fender do you know if anyone around here uses realtime with their voicemail and stuff that has vmwi ? |
| 13:20.02 | DarKnesS_WolF | [TK]D-Fender: the greatest of all time ;-) how are u |
| 13:20.06 | fourcheeze | FlatFoot: that's just the number that gets dialled |
| 13:20.24 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 13:20.31 | [TK]D-Fender | DrukenHME: You keep asking, and I keep answering : I don't use realtime or know anything about those who do. |
| 13:20.39 | coppice | [TK]D-Fender: oh, I never speak at my audience. its always personal for every single one. :-) |
| 13:21.00 | [TK]D-Fender | coppice: "in their minds".... the grand deception! |
| 13:21.02 | DrukenHME | bah, sorry man, great memory, only about half a second lo.... what was i typing? |
| 13:21.03 | FlatFoot | fourcheeze: yeah understand that but why would that fail to auth and give a stale nonce |
| 13:21.12 | fourcheeze | FlatFoot: you should be able to leave that blank and use |
| 13:21.12 | fourcheeze | vmexten=1234 in your sip.conf (where 1234 is the number they dial for mailbox) |
| 13:21.25 | FlatFoot | fourcheeze: just seems really weird |
| 13:21.30 | fourcheeze | yeah it is |
| 13:21.32 | fourcheeze | I've never had that |
| 13:21.33 | *** join/#asterisk onurx (n=ddsdd@85.98.79.130) |
| 13:22.06 | FlatFoot | fourcheeze: i shall try vmexten , ta for that |
| 13:23.44 | DarKnesS_WolF | do i need libiax while i'm compiling * ? |
| 13:25.45 | FlatFoot | fourcheeze: nope that don't work , the retrieve button seems not to be functioning correctly now . i think i shall have to get the hammer out |
| 13:26.52 | fourcheeze | FlatFoot: try leaving a vm |
| 13:28.29 | FlatFoot | fourcheeze: trying now |
| 13:30.01 | *** join/#asterisk blop (n=blop@204.53-201-80.adsl-dyn.isp.belgacom.be) |
| 13:30.07 | blop | hi there |
| 13:30.54 | blop | i'm trying to get colors in the CLI (asterisk -r) on a debian, i tried exporting TERM=linux and such things but cant get it working, any idea? |
| 13:32.13 | *** part/#asterisk onurx (n=ddsdd@85.98.79.130) |
| 13:32.48 | *** join/#asterisk Unistim_junky (n=rover@216-230-88-10.client.cypresscom.net) |
| 13:35.00 | FlatFoot | fourcheeze: i think this is gonna be a brain burner , still no good |
| 13:35.42 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:37.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:39.31 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 13:39.39 | jeremy_g | hello again my dear friends |
| 13:39.45 | jeremy_g | another issue is biting!!! |
| 13:40.08 | scanna | hi all, can someone help me with call transfer? |
| 13:40.10 | jeremy_g | a sip client registers with * with registration expirey period of 60seconds |
| 13:40.32 | jeremy_g | right after registration, the sip client looses network connectivity |
| 13:41.02 | scanna | i have 3 sip phone A call B and B transfer to C, i don't understand why in cdr the caller id of A is missing... |
| 13:41.23 | jeremy_g | now at that point if that client is dialled by another * user, what DIALSTATUS message would you do expect? |
| 13:41.39 | jeremy_g | scanna:explain more, i think i can help |
| 13:41.51 | Seba_soy | jeremy_g: CHANUNAVAIL I think |
| 13:42.41 | jeremy_g | * shows as if sip client is not answering the phone, while the fact is that it that client actually lost network connectivity |
| 13:42.51 | jeremy_g | Seba_soy:no |
| 13:42.58 | jeremy_g | Seba_soy:it gives NOANSWER |
| 13:42.59 | ichilton | 4 channels configured. |
| 13:43.00 | ichilton | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 13:43.08 | Seba_soy | what SIP SHOW PEERS showing? |
| 13:43.16 | ichilton | anyone know what that means? |
| 13:43.33 | Seba_soy | if it shows UNRACHEABLE then it will be dialstatus CHANUNAVAIL i think |
| 13:43.48 | Zouzou | Executing Answer("SIP/1001-b7e073c8", "") in new stack |
| 13:43.48 | Zouzou | <PROTECTED> |
| 13:43.48 | Zouzou | <PROTECTED> |
| 13:43.48 | Zouzou | <PROTECTED> |
| 13:43.48 | Zouzou | <PROTECTED> |
| 13:43.50 | Zouzou | Executing Answer("SIP/1001-b7e073c8", "") in new stack |
| 13:43.50 | Zouzou | <PROTECTED> |
| 13:43.52 | Zouzou | <PROTECTED> |
| 13:43.54 | Zouzou | <PROTECTED> |
| 13:43.56 | Zouzou | <PROTECTED> |
| 13:44.01 | Zouzou | Music on Hold Immediately Stopping |
| 13:44.17 | Seba_soy | Zouzou: IT IS A PROBLEM WITH WAV FILE!! |
| 13:44.26 | Seba_soy | for some reason astrisk cant play it |
| 13:44.42 | Zouzou | musiconhold stop immediately(afetr i changed the defaut class0 |
| 13:45.00 | Zouzou | sorry |
| 13:45.12 | hmmhesays | ~pastebine |
| 13:45.18 | jbalcomb | jbroome: could you change your nick? i'm getting a lot of false alarms cause people are hitting jb<tab> and getting my nick instead of yours. |
| 13:45.18 | jbalcomb | ;) |
| 13:45.20 | hmmhesays | ~pastebin |
| 13:45.24 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 13:45.30 | jbalcomb | !paste |
| 13:45.36 | hmmhesays | ~hmmhesays |
| 13:45.37 | jbot | somebody said hmmhesays was not really here... |
| 13:45.44 | Zouzou | i copeid the same mp3 file to anothe folder than mohmp3 |
| 13:45.49 | Zouzou | and its not working |
| 13:45.53 | DrukenHME | if realtime doesn't find a match shouldn't it goto +101? |
| 13:46.04 | Seba_soy | so is a mp3 player problem? |
| 13:46.05 | rob0 | jbalcomb: complain to jbot too. ;) |
| 13:46.08 | Seba_soy | paste your config files |
| 13:46.10 | Zouzou | after changing the directory to the new file |
| 13:46.20 | jbalcomb | DrukenHME: i though priority jumping was deprecated? |
| 13:46.35 | jbalcomb | jbot: could you change your nick? i'm getting a lot of false alarms cause people are hitting jb<tab> and getting my nick instead of yours. |
| 13:46.36 | jbot | that's too long, jbalcomb |
| 13:46.46 | jeremy_g | Seba_soy:sip show peers even after disconnectivity showed the sip client IP it had when it was on the network |
| 13:47.04 | hmmhesays | it still exists in some app's with options |
| 13:47.04 | jbalcomb | jbot: You, sir, are Futt Bucker. |
| 13:47.05 | jbot | jbalcomb: what are you talking about? |
| 13:47.07 | Seba_soy | but it has to show UNRACHEABLE |
| 13:47.26 | [TK]D-Fender | jbalcomb: *'s idea of "deprecation" is to "say" that it should no longer be used, and in a few case tell you to use a switch to enable it by application, but it remains never theless, and there is a global var in 1.2 to set it more generally. |
| 13:47.28 | scanna | jeremy: do you have the same problem? |
| 13:47.55 | [TK]D-Fender | ~lart jbot |
| 13:47.59 | scanna | with attended transfer the caller id of the original caller is not posted in cdr |
| 13:48.02 | jeremy_g | Seba_soy:once i dialled the dead sip client, * returned with DIALSTATUS noanswer and then on doing a show peer, * showed 'Unspecified' |
| 13:48.04 | [TK]D-Fender | FAILURE! |
| 13:48.56 | rob0 | Just a minute here ... okay ... can someone help put out the fire? |
| 13:48.59 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 13:49.27 | *** join/#asterisk ppyy (n=lala@222.185.19.218) |
| 13:49.27 | ichilton | Any zaptel card experts here? |
| 13:49.48 | jbalcomb | [TK]D-Fender: yeah, there are a few common linux commands that have that. like using : for chown instead of . |
| 13:50.00 | FlatFoot | exit |
| 13:50.08 | FlatFoot | oops sorry wrong screen |
| 13:50.39 | jbalcomb | [TK]D-Fender: ten years of seeing 'don't use blah switch, someday it won't be supported' |
| 13:50.43 | [TK]D-Fender | jbalcomb: However much of what was "deprecated in 1.2 seems to be "removed" from 1.4 thus completing its demise |
| 13:50.54 | [TK]D-Fender | jbalcomb: How very M$ ;) |
| 13:51.04 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 13:51.48 | jbalcomb | [TK]D-Fender: hrm.. I've always thought of M$ as making change without notice, deprecation, or reverse compatibility. |
| 13:52.14 | jbalcomb | [TK]D-Fender: that's why you have to get recertified in Windows every two years. |
| 13:52.23 | [TK]D-Fender | jbalcomb: that is.... when they "get around to it" |
| 13:52.30 | *** join/#asterisk sxpert-work (n=sxpert@raph.imag.fr) |
| 13:52.38 | jbalcomb | [TK]D-Fender: oh yeah, that part for sure |
| 13:52.44 | jeremy_g | Seba_soy:no it doesnt |
| 13:52.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:52.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:52.51 | jeremy_g | Seba_soy:its Unspecified |
| 13:54.24 | *** join/#asterisk Kernel_Core (n=I@83.217.236.42) |
| 13:54.29 | Kernel_Core | hi all |
| 13:54.37 | jeremy_g | In sip show peers, there is a status field, it gives 'Unmonitored', what do i change to change things under that field??? |
| 13:54.43 | jeremy_g | do i have to enable monitoring |
| 13:55.09 | Kernel_Core | I have 1000 number , how do I generate call with asterisk to test this 1000 number ?! |
| 13:55.27 | Kernel_Core | for every number , 2 minutes |
| 13:56.05 | scanna | jeremy: do you have the same problem with attended transfer? |
| 13:56.33 | jeremy_g | scanna:havent tried that |
| 13:56.48 | fourcheeze | FlatFoot: in the function keys sectino you can just define the mailbox key as dialling a particular number |
| 13:57.33 | jeremy_g | scanna:it returns NOANSWER until the time when dead sip client's registration expiry period finishes, * then flushes it entry and then * starts returning CHANUNAV.. |
| 14:00.57 | jeremy_g | scanna:it works now if i set qualify=yes in that dead sip client's sip account |
| 14:01.16 | jeremy_g | man, this is such an issue |
| 14:01.51 | jeremy_g | why cant * verify the registered sip clients, some might be dead |
| 14:02.00 | jeremy_g | why it requries qualify=yes option |
| 14:04.31 | Seba_soy | it requires qualify=xxx to maintaing nat sessions |
| 14:04.40 | jeremy_g | Seba_soy:I know that |
| 14:04.52 | jeremy_g | Seba_soy:nat=yes, qualify=yes then |
| 14:05.05 | jeremy_g | only then it is used for maintaining nat sessions |
| 14:05.05 | Seba_soy | nat=yes qualify=500 |
| 14:05.11 | Seba_soy | to use 500ms delay |
| 14:05.24 | jeremy_g | yup i have read the manuals, i know that |
| 14:05.27 | jeremy_g | problem is * |
| 14:05.52 | directory | jeremy_g: ...that's what qualify is for, to see if the SIP client is there, how else do you expect it to know? |
| 14:05.58 | directory | well... SIP peer |
| 14:06.16 | jeremy_g | when * dials out to a sip client that just registered and lost network connecitivty, it some how thinks that it is still there and not answering my call |
| 14:06.50 | Seba_soy | i think asterisk send sip invite to port where your sip have registered |
| 14:06.56 | Seba_soy | if not reponse |
| 14:07.08 | jeremy_g | directory:u know by dialing to it and seeing if it even on the network or not, if not then please return CHANUNAVAIL. right now its returning NOANSWER and i am so pissed off |
| 14:07.10 | hmmhesays | because your registration timeout has not expired |
| 14:07.11 | Seba_soy | then I think it will be CHANUNAVAIL or maybe NOANSWER as you say |
| 14:07.31 | *** join/#asterisk oej (n=oej@231.Red-88-14-197.dynamicIP.rima-tde.net) |
| 14:07.32 | hmmhesays | jeremy_g: qualify=yes in sip.conf |
| 14:07.37 | directory | jeremy_g: you didn't say that though |
| 14:07.38 | jeremy_g | * behaves this way for 60 seconds, which is the expiry period of the reigstration |
| 14:07.57 | *** part/#asterisk jmacz (n=jmacz@190.24.96.103) |
| 14:07.58 | Seba_soy | qualify and expiry period are not same thing |
| 14:08.03 | *** join/#asterisk bmg505 (n=leon@c1-69-3.rndf.isadsl.co.za) |
| 14:08.18 | ichilton | Any zaptel card experts here? |
| 14:08.22 | jeremy_g | Seba_soy: :D |
| 14:08.44 | jeremy_g | Seba_soy:u almost had me. why do you love to read from the book |
| 14:09.14 | jeremy_g | why NOANSWER |
| 14:09.29 | Seba_soy | ask sip developers! |
| 14:09.30 | Seba_soy | :) |
| 14:09.31 | jeremy_g | it shud be CHANUNAVAIL cuz then i want to make it to dial to another number |
| 14:09.52 | jeremy_g | ok |
| 14:09.59 | Seba_soy | put it in bug |
| 14:10.05 | Seba_soy | maybe is a bug |
| 14:12.29 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 14:13.24 | *** join/#asterisk eMZed (n=marko@rikom.sk-branik.si) |
| 14:14.18 | DoDaT69 | so with echo cancel, if a card offers more taps, is it better? |
| 14:14.28 | DoDaT69 | ie-- comparing digium and sangoma analog cards |
| 14:14.44 | DoDaT69 | digium offers 256 while sangoma offers 1024 |
| 14:15.30 | MikeJ | I was under the impression that they were both using the same octasic chip with the same number of taps.. |
| 14:15.50 | Seba_soy | what are taps? |
| 14:15.52 | MikeJ | you may want to double check your numbers.. but typically, more taps is better.. |
| 14:16.14 | MikeJ | it's just a matter of how bad (long) your echo is... |
| 14:16.17 | *** part/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 14:16.29 | DoDaT69 | I am just shopping around for cards atm |
| 14:16.31 | MikeJ | and how many taps you need to see the echo |
| 14:16.32 | hmmhesays | taps are those things BEER comes out of |
| 14:16.42 | DoDaT69 | yea |
| 14:16.49 | DoDaT69 | so more is better |
| 14:16.53 | DoDaT69 | thats what I needed to know ;) |
| 14:17.01 | DoDaT69 | thnx :-D |
| 14:17.18 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 14:18.13 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:18.37 | hmmhesays | haha Walker Texas Ranger is on... ROCK |
| 14:19.06 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 14:19.36 | MrChimpy | is there anywhere selling FXO/FXS modules for TDM400P at non-extortionate prices? |
| 14:19.44 | MrChimpy | digium's prices are ridiculous! |
| 14:20.14 | MikeJ | what are they selling the modules for? |
| 14:20.24 | MrChimpy | $80 each |
| 14:20.34 | MrChimpy | there's very little to them |
| 14:20.46 | MikeJ | hehe.. |
| 14:20.48 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-cfeaabf69e97c02a) |
| 14:21.17 | coppice | make your own, then. :-) |
| 14:21.29 | coppice | I wonder if the OpenVox ones fit |
| 14:22.05 | MrChimpy | quite possibly |
| 14:23.29 | *** join/#asterisk kuto (n=kuto@125.60.241.24) |
| 14:23.44 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:25.14 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-168-240.dsl.chcgil.sbcglobal.net) |
| 14:25.18 | Flauto | hi all |
| 14:26.14 | Flauto | anyone here is using iksemel |
| 14:26.39 | *** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 14:26.43 | MikeJ | I do |
| 14:26.55 | Flauto | mike, when i tried to comlile it |
| 14:27.18 | Flauto | -bash: ./configure: No such file or directory |
| 14:27.31 | Flauto | would you help with that? |
| 14:27.37 | Flauto | is there anything else i need to do |
| 14:27.49 | *** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net) |
| 14:27.50 | Flauto | i got it through svn |
| 14:27.53 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 14:28.14 | MikeJ | if you get it through svn, it probably doesn't have configure built yet.. |
| 14:28.35 | MikeJ | either just run autoreconf, or there may be a shell script in the root dir to build all the autodrool |
| 14:29.00 | Cubber | what is the best way to connect an asterisk box to a nortel norstar pbx system? We have 5 lines comming into the norstar system and about 20 digital extensions off of that? Would my best be to get a anolog converter for one of the extensions or go PRI off of the norstar sytem into the asterisk server? |
| 14:30.42 | Flauto | [root@Encore iksemel]# autoreconf |
| 14:30.42 | Flauto | aclocal: configure.ac: 48: macro `AM_PATH_LIBGNUTLS' not found in library |
| 14:30.42 | Flauto | autoreconf-2.5x: aclocal failed with exit status: 1 |
| 14:33.31 | Flauto | mikej, would you mind talking to me in private |
| 14:33.48 | Flauto | i opened a window |
| 14:41.43 | *** part/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 14:47.06 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 14:47.10 | FlatFoot | fourcheeze: sorry for not replying had to sort out SQL probs for one of the useless planks i work with ( what port do i need to open for EM ) |
| 14:47.43 | FlatFoot | fourcheeze: i have got round it , sent them to a please enter your vm number menu |
| 14:48.29 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
| 14:49.27 | AsteriskMonkey | Im having some ivr dtmf issue can anyone help? |
| 14:50.19 | AsteriskMonkey | I know how to calibrate tx/tx and all that |
| 14:50.41 | AsteriskMonkey | this one is stumping me, im having occasional issue with people entering extension of an ivr |
| 14:50.54 | AsteriskMonkey | it seems to not catch the first digit alot |
| 14:51.54 | *** part/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 14:53.33 | dgilmore | how can i tell a TE110P to setup for T1 not E1? |
| 14:54.25 | AsteriskMonkey | there is a jumper on it |
| 14:54.38 | dgilmore | gahh thanks i missed that on this one |
| 14:54.43 | dgilmore | AsteriskMonkey: cheers |
| 14:55.42 | *** join/#asterisk Aces1Up (n=Aces1Up@ip68-96-234-176.lv.lv.cox.net) |
| 14:57.45 | Aces1Up | has anyone here used asterisk with a GSM Gateway as a call back Service? |
| 14:59.49 | AsteriskMonkey | Kram/ digium guys you need to fix your phone system |
| 15:00.13 | AsteriskMonkey | im talking to some one digium there all crakly and choppy.ca |
| 15:00.37 | trelane_ | might be the voip provider |
| 15:01.20 | *** join/#asterisk scud (n=scud@gate.hhsys.org) |
| 15:01.32 | AsteriskMonkey | i was callign on a pstn |
| 15:01.40 | AsteriskMonkey | pstn to there 256 :P |
| 15:01.53 | brookshire | AsteriskMonkey: they are probably on a cordless phone |
| 15:03.53 | AsteriskMonkey | brookshire: I wish |
| 15:04.53 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:05.40 | *** join/#asterisk Katty (n=Administ@64.82.232.54) |
| 15:05.55 | Katty | allo (= |
| 15:07.14 | Zouzou | what is the best sound format to use with IVR????? |
| 15:08.01 | brookshire | AsteriskMonkey: then who did you talk to? |
| 15:09.47 | Katty | so. i was thinking. |
| 15:09.57 | Katty | it'd sure be nice to put a sort of polycom address book on ftp. |
| 15:10.01 | scud | someone port asterisk to run on the qtopia greenphone |
| 15:10.13 | Katty | so i can update it centerally, and everyone doesn't have to pester me about who's extension is who. |
| 15:10.15 | scud | i will bounty a bag of skittles and a snickers |
| 15:10.29 | Katty | is that possible? or am i just dreaming? |
| 15:10.34 | scud | taste the rainbow... |
| 15:11.06 | *** join/#asterisk ajl119 (n=a_lemin@62-105-121-host-201.as15758.net) |
| 15:15.10 | angler_ | scuddles |
| 15:15.21 | Katty | [TK]D-Fender: you around? |
| 15:16.24 | scud | angler: que pasa |
| 15:16.32 | scud | angler: mark? |
| 15:16.38 | ichilton | Any zaptel card experts here? |
| 15:17.03 | coppice | just ask your question. it usually works better that way |
| 15:17.14 | ichilton | :) |
| 15:17.14 | ichilton | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 15:17.18 | ichilton | any ideas? |
| 15:17.59 | angler_ | scud, when u said skittles it reminded me of mark's nickname for u |
| 15:18.11 | scud | yeah |
| 15:18.30 | [TK]D-Fender | Katty: Somewhat, brb |
| 15:18.39 | scud | mark an his fruityness, hes a crazy and whacky kind of guy!! ::toots big red horn:: |
| 15:18.40 | Corydon-w | ichilton: sounds like the kernel drivers aren't loaded |
| 15:18.45 | scud | an/and |
| 15:19.05 | Corydon-w | ichilton: is this a TDM400 card, by chance? |
| 15:19.13 | scud | angler: how is the new family life? |
| 15:19.24 | ichilton | Corydon-w: yes - TDM04P |
| 15:19.27 | Katty | [TK]D-Fender: kay. |
| 15:19.30 | angler_ | good.... go to msg |
| 15:19.32 | AsteriskMonkey | ok this is driving me batty... dtmf issues.. ztmonitor show 10800 on dtmf presses on the rx, is this considered too hot? |
| 15:19.38 | ichilton | Corydon-w: the modules are loaded but I get this in the logs: |
| 15:19.43 | Corydon-w | ichilton: did you plug in the molex connector? |
| 15:19.45 | ichilton | Sep 27 15:25:39 asterisk kernel: ACPI: PCI interrupt 0000:01:01.0[A] -> GSI 22 (level, low) -> IRQ 217 |
| 15:19.48 | ichilton | Sep 27 15:25:39 asterisk kernel: Freshmaker version: 73 |
| 15:19.51 | ichilton | Sep 27 15:25:39 asterisk kernel: Freshmaker passed register test |
| 15:19.53 | ichilton | Sep 27 15:25:40 asterisk kernel: Module 0: Not installed |
| 15:19.56 | ichilton | Sep 27 15:25:40 asterisk kernel: Module 1: Not installed |
| 15:19.58 | ichilton | Sep 27 15:25:40 asterisk kernel: Module 2: Not installed |
| 15:20.01 | ichilton | Sep 27 15:25:40 asterisk kernel: Module 3: Not installed |
| 15:20.03 | ichilton | Sep 27 15:25:40 asterisk kernel: wctdm: probe of 0000:01:01.0 failed with error -5 |
| 15:20.12 | ichilton | Corydon-w: what's molex? - the power connector? |
| 15:20.22 | Corydon-w | DONT do that |
| 15:20.35 | ichilton | sorry, it didn't look that many lines on the screen :) |
| 15:20.36 | Corydon-w | Use pastebin if you want to show more than 2 or 3 lines |
| 15:21.16 | ichilton | ok |
| 15:22.19 | ichilton | Corydon-w: any ideas? |
| 15:22.42 | Corydon-w | 1) plug in the molex connector, 2) make sure the modules are plugged in |
| 15:22.53 | ichilton | is molex the power connector? |
| 15:22.57 | Corydon-w | yes |
| 15:23.01 | ichilton | that's in |
| 15:23.14 | ichilton | and I just unplugged all modules andplugged them back in + changed the pci slot |
| 15:23.19 | ichilton | it's all definatly in |
| 15:23.25 | Corydon-w | ichilton: Please call Digium support |
| 15:23.31 | ichilton | ok |
| 15:23.43 | ichilton | does it matter I bought it from ebay? |
| 15:23.53 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 15:24.14 | Corydon-w | Did the seller sell it as-is or do they have a money back guarantee? |
| 15:24.33 | Corydon-w | And is it truly a Digium card or is it a clone card? |
| 15:25.19 | ichilton | doesn't actually say |
| 15:25.26 | ichilton | it has Digium on it so I think it's a real one |
| 15:25.42 | ichilton | http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&ih=004&item=140027304421 |
| 15:26.20 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.206) |
| 15:27.01 | Corydon-w | Call the seller and ask for support |
| 15:27.14 | ichilton | ok |
| 15:28.03 | ichilton | thanks |
| 15:30.19 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-104978.otenet.gr) |
| 15:30.46 | jgoo | well despite the weirdness of trixbox 1.2 (yes yes #freepbx!) I got it all installed, even though I had to swap out SATA for IDE, aaaah well |
| 15:30.53 | *** join/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 15:31.43 | jgoo | I asked this question a while back but I was still too noob to get very far with it - basically a 'best practice' for calling external scripts / services from a dialplan, is there an in depth dialplan hacking guide? |
| 15:32.36 | *** join/#asterisk GiabboO (n=GiabboO@host-84-220-151-170.cust-adsl.tiscali.it) |
| 15:32.39 | GiabboO | hi all |
| 15:32.43 | Corydon-w | jgoo: just the one you're going to go write |
| 15:32.43 | GiabboO | i have stupid question |
| 15:32.44 | GiabboO | :) |
| 15:33.01 | GiabboO | I am going to buy some voip phones so i have to choice wich protocol use |
| 15:33.08 | GiabboO | is better IAX2 or SIP? |
| 15:33.19 | GiabboO | in ur own experiences |
| 15:33.20 | *** join/#asterisk s0lid (n=jlq@61.28.130.145) |
| 15:33.56 | Corydon-w | Like everything else, the answer is "it depends" |
| 15:33.58 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 15:34.26 | GiabboO | :) |
| 15:34.33 | GiabboO | i shouldnt have nat problem |
| 15:34.58 | razu | hi, is there any possibility to change CALLERIDNAME and CALLERIDNUM separately with AGI ? SET VARIABLE CALLERIDNAME <value> doesn't work and SET CALLERID can only change name or number as far as I can understand ... :( |
| 15:34.59 | jgoo | =] |
| 15:35.05 | razu | can anyone help me please ? |
| 15:35.46 | *** join/#asterisk batphone (n=bugz@cpe-24-162-13-42.houston.res.rr.com) |
| 15:35.52 | Corydon-w | ~ask |
| 15:35.53 | jbot | somebody said ask was 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. See also ... |
| 15:36.51 | *** part/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 15:37.44 | GiabboO | oh finallythey removed the k line in my office brb |
| 15:38.10 | [TK]D-Fender | Corydon-w: He did just ask a rather specific question.... |
| 15:38.25 | Tall-guy | yeah, he did....(jbot Nazi) |
| 15:38.42 | *** join/#asterisk lirakis (n=tbright@h-68-165-94-219.nycmny83.covad.net) |
| 15:38.44 | Corydon-w | Sorry, didn't see it |
| 15:38.46 | lirakis | helllo all |
| 15:38.56 | *** join/#asterisk GiabboO (n=GiabboO@host101-246-static.73-81-b.business.telecomitalia.it) |
| 15:38.58 | GiabboO | woww:) |
| 15:39.01 | GiabboO | thanks freenode stuff:D |
| 15:39.03 | GiabboO | brb |
| 15:39.05 | *** part/#asterisk GiabboO (n=GiabboO@host101-246-static.73-81-b.business.telecomitalia.it) |
| 15:39.17 | backblue | i'm trying to setup a gsm gateway in one card i have, but it looks like, everytime i answer the call, the gsm gateway does not know, i have answer it. |
| 15:39.25 | Corydon-w | razu: use the function call: SET VARIABLE CALLERID(name) ... |
| 15:39.46 | lirakis | i need to set P-Asserted identity headers ... i dont really know much about it .. does anyone know how to do this? |
| 15:40.15 | razu | Corydon-w : damn ... of course ... thx a lot :) |
| 15:40.54 | backblue | Corydon-w: do you know, why asterisk does not follow sip rfc, and cuts the "." (dots)?? |
| 15:41.42 | Corydon-w | backblue: what specific section of what RFC (there's more than one for SIP) do you think Asterisk is not compliant with? |
| 15:42.13 | CtRiX | Corydon-w, UTF support, fot example ! |
| 15:42.13 | backblue | Corydon-w: why cant i have sip users with dots? |
| 15:42.57 | Corydon-w | CtRiX: AFAIK, RFC's are numeric, not alpha |
| 15:43.21 | backblue | no, they are alpha-numeric |
| 15:43.22 | *** join/#asterisk gandhijee (n=gandhije@static-66-16-235-31.dsl.cavtel.net) |
| 15:43.23 | Corydon-w | backblue: I don't know offhand. |
| 15:43.36 | backblue | well, ok. |
| 15:43.43 | CtRiX | AFAIK: From sip header is not numeric |
| 15:44.01 | Corydon-w | CtRiX: again, that's not an RFC |
| 15:44.01 | backblue | i will keep adding my patch's |
| 15:44.24 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:44.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:44.29 | CtRiX | backblue, can you send me your patch, PLZ? |
| 15:44.34 | CtRiX | i'll check it and add to openpbx |
| 15:45.01 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:45.18 | backblue | what does asterisk has to do with openpbx? |
| 15:45.39 | CtRiX | hpefully nothing. Openpbx is a fork of asterisk |
| 15:45.49 | Corydon-w | The openpbx developers hang out in here because there aren't that many openpbx users |
| 15:45.53 | *** join/#asterisk Gourou69 (n=test@par69-3-82-224-162-203.fbx.proxad.net) |
| 15:46.02 | backblue | so my patch does nothing in openpbx, it's only one line or two i think |
| 15:46.03 | jbalcomb | what config options for the Polycom match up with the sip.conf username and secret? |
| 15:46.16 | CtRiX | Corydon-w, yeah... of course. |
| 15:46.31 | jgoo | jbalcomb: just about ot ask about polycom myself, asteriskguru and voip info don't have this |
| 15:46.35 | backblue | and we have another patch for the incoming sip domain |
| 15:46.40 | backblue | that asterisk does not handle it |
| 15:46.54 | backblue | but i think it breaks something, so it's not finish |
| 15:47.00 | Corydon-w | backblue: have you submitted any patches to the bugtracker? |
| 15:47.20 | backblue | Corydon-w: i think we have submited one patch do rawplayer |
| 15:47.24 | backblue | s/do/to |
| 15:47.30 | backblue | but not this ones, i have thought |
| 15:47.41 | Corydon-w | backblue: cool, just add a new issue to the bugtracker |
| 15:47.45 | Tall-guy | jbalcomb/jgoo: what phones are you config'ing? ( I have a working ip501) |
| 15:47.47 | backblue | i tryed to discuss this in ML but no one answer me. |
| 15:47.48 | Aces1Up | has anyone here used asterisk with a GSM Gateway as a call back Service? |
| 15:47.58 | bkw_ | Aces1Up, SMS? |
| 15:48.05 | backblue | i dont really have many time to do that now, sorry |
| 15:48.15 | jbalcomb | Tall-guy: IP 501 |
| 15:48.16 | Aces1Up | bkw sms, ani, |
| 15:48.16 | backblue | Aces1Up: i have |
| 15:48.21 | backblue | and i'm doing that right now |
| 15:48.24 | bkw_ | Aces1Up, try smstools |
| 15:48.29 | backblue | but this gsm gateway it's freaking out |
| 15:48.48 | backblue | i have never used sms service |
| 15:48.49 | Tall-guy | jbalcomb: not too tricky, only pain in the ass is the slow reboots |
| 15:49.45 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 15:49.50 | jbalcomb | Tall-guy: yeah, i just need those two config options. i'm looking through the manual but its like a developers scratch pad. |
| 15:50.00 | Tall-guy | jbalcomb: msg me if you want, I'll walk thru it with you.... |
| 15:50.12 | Tall-guy | jbalcomb: manual blows :) |
| 15:50.33 | [TK]D-Fender | jbalcomb: "address" is the [peerentry" and "auth.password" is the "secret=" |
| 15:50.46 | [TK]D-Fender | Tall-guy: I beg to differ..... |
| 15:51.00 | Tall-guy | tkd: you don't think the manual blows? |
| 15:51.13 | [TK]D-Fender | Tall-guy: No, I am completely successful with it. |
| 15:51.39 | [TK]D-Fender | Tall-guy: From-scratch to 100% functional in 5 minutes flat. |
| 15:51.44 | Tall-guy | tkd: I'm pretty sure there are different "versions" of the manual...as the one I got with the phone was horrid. |
| 15:52.17 | jbalcomb | [TK]D-Fender: I'm afraid I agree that the manual is poor but I don't dispute that you do well. |
| 15:52.24 | [TK]D-Fender | Tall-guy: You mean the USERS GUIDE? That has nothing to do with the ADMIN guide whose layou has been extremely consistant for years. |
| 15:52.40 | jbalcomb | [TK]D-Fender: I suspect it's 'in spite of' rather than 'because of' the manuals helpfullness. |
| 15:52.52 | Tall-guy | tkd: I was speaking of the manual that CAME with the phone.....not the one that a person eventually finds online. |
| 15:53.00 | [TK]D-Fender | jbalcomb, Tall-guy : You just need a controlled frame of mind when going over it and it becomes clear. |
| 15:53.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:53.48 | jbalcomb | i have not idea what a controlled frame of mind is there. Is that like suspension of disbelief when I see moves like hacks and the net? |
| 15:53.56 | jbalcomb | s/hacks/hackers |
| 15:54.03 | [TK]D-Fender | And notice to all : I have recently begun working with Aastra 480i's and find them to be very decent phones and highly configurable. A very worthy option to add to the list. So I am no longer just a "Polycom Zealot" :D |
| 15:54.44 | [TK]D-Fender | jbalcomb: No, you don't "suspend" reality so much as direct it through a slightly different filter. |
| 15:54.57 | *** join/#asterisk adamowitz (n=adamowit@ip68-9-201-27.ri.ri.cox.net) |
| 15:55.09 | Tall-guy | tkd: I've got a mitfull of the non-ip Aastras....I like their ADSI capability and their voice quality..... |
| 15:55.11 | jbalcomb | [TK]D-Fender: how much for one of those filters? Can you ship international? ;) |
| 15:55.23 | adamowitz | anyone here using the UTstarcom F3000? |
| 15:55.31 | [TK]D-Fender | jbalcomb: Sure, just ship your brain down and I'll get to work ;) |
| 15:59.01 | fourcheeze | wahey got my e61 working with * |
| 15:59.06 | fourcheeze | life is sweeeeeeeeeeeeeeeeet |
| 15:59.37 | *** join/#asterisk Ebola (i=1000@user-3e8876ab.telcl21.dsl.pol.co.uk) |
| 15:59.44 | fourcheeze | you guys have to get an e61 |
| 15:59.49 | fourcheeze | just so cool |
| 16:00.15 | Tall-guy | humor me, whats an e61 |
| 16:00.17 | CtRiX | Corydon-w, |
| 16:00.18 | CtRiX | <CtRiX> AFAIK: From sip header is not numeric |
| 16:00.18 | CtRiX | <Corydon-w> CtRiX: again, that's not an RFC |
| 16:00.24 | CtRiX | SIP is a text-based protocol that uses UTF-8 encoding |
| 16:00.28 | CtRiX | http://www.voip-info.org/wiki-SIP |
| 16:00.42 | CtRiX | now, tell me from where your convinction comes from ... |
| 16:01.43 | fourcheeze | Tall-guy: e61 is a smart phone from nokia |
| 16:01.54 | fourcheeze | Tall-guy: http://www.europe.nokia.com/nokia/0,,81718,00.html |
| 16:02.20 | Corydon-w | CtRiX: I asked for an RFC number and section from someone else, and you gave me something that was neither an RFC number nor a section |
| 16:02.42 | CtRiX | ah ok. |
| 16:02.44 | jbalcomb | [TK]D-Fender: After I have secrets enabled for all my [peer] entries in sip.conf, how do I make asterisk not allow someone without a password or registered to make calls? |
| 16:02.55 | Tall-guy | fourcheeze: define "working with *" for me.......hooked direcT???? |
| 16:02.57 | CtRiX | so you're telling me to show you what's the content of an RFC. |
| 16:02.58 | CtRiX | ok |
| 16:03.08 | [TK]D-Fender | jbalcomb: Thats implicit. |
| 16:03.09 | fourcheeze | Tall-guy: registered with * via wlan and making calls |
| 16:03.15 | tecnico | fourcheeze: I just got one too :) I'm waiting for it to ship. Did you set it up using SIP ? is your contacts book tied to that software? |
| 16:03.30 | Tall-guy | fourcheeze: ah...it's wifi too..... |
| 16:03.34 | [TK]D-Fender | jbalcomb: So unless you added "allowguest=yes" and a context under [general] you will have successfully secured your system. |
| 16:03.34 | fourcheeze | tecnico: you can choose whether to dial each number via sip or cell |
| 16:03.48 | jbalcomb | [TK]D-Fender: why the hell does my Polycom IP 501 at extension 4444 get to make calls then? |
| 16:04.02 | fourcheeze | Tall-guy: it does everything |
| 16:04.03 | Corydon-w | Considering there's multiple RFCs that specify SIP, that's not usually a problem for someone who wants to challenge a particular behavior of the SIP stack |
| 16:04.08 | fourcheeze | it's easily the best phone around ATM |
| 16:04.10 | [TK]D-Fender | jbalcomb: I'd have to look at it. |
| 16:04.12 | fourcheeze | (cell phone that is) |
| 16:04.26 | Tall-guy | fourcheeze: snazzy |
| 16:04.26 | CtRiX | Corydon-w, ok |
| 16:04.30 | CtRiX | 7 SIP Messages |
| 16:04.30 | CtRiX | <PROTECTED> |
| 16:04.30 | CtRiX | <PROTECTED> |
| 16:04.36 | [TK]D-Fender | jbalcomb: You reloaded SIP w/ the secrets and don't have the global stuff I mentioned? |
| 16:04.42 | CtRiX | that's rfc3261 .... |
| 16:04.48 | fourcheeze | quite cheap too |
| 16:04.59 | tecnico | fourcheeze: nice :) Is it STUN enabled yet ? I know an update for the E61 came out recently, I wonder if they fixed that |
| 16:05.02 | CtRiX | that should not be an unknown or new RFC |
| 16:05.29 | Tall-guy | fourcheez: how much (I assume you're in europe?) |
| 16:05.38 | fourcheeze | tecnico: I haven't seen any stun things in there. My update is very recent - my company services mobile phones so I get rapid access to those |
| 16:05.42 | tecnico | Tall-guy: I got it for 329 in ebay |
| 16:05.52 | Corydon-w | CtRiX: have you filed a bug on the bugtracker? |
| 16:05.53 | fourcheeze | Tall-guy: price to us is about £200 |
| 16:06.05 | CtRiX | Corydon-w, i'm not interested in that. |
| 16:06.14 | Corydon-w | CtRiX: then what's the point? |
| 16:06.28 | CtRiX | the point is: RFC are written to be read. |
| 16:06.53 | Corydon-w | Great, and they'll be read. Thank you for your concern. |
| 16:07.10 | CtRiX | and another point is: you cannot say * is RFC compliant cause one of the main requirements is not there. |
| 16:07.26 | CtRiX | you may say "worksforme(tm)" but not that ot's compliant. |
| 16:07.33 | jbalcomb | [TK]D-Fender: I have allowguest = no |
| 16:07.40 | *** join/#asterisk Borgon (n=l3orgon@12.173.143.25) |
| 16:08.28 | ManxPower | CtRiX, Asterisk is "compliant enough" to work with most SIP devices. |
| 16:08.42 | Borgon | hello, If am using a 3rdparty sip app behind a double nat router... How can i get the sound to play? I can manage to get my asterisk working on iax but sip doesnt rtp sound, is it possible to make a ssh tunnel to a outside vps to bypass the nat? |
| 16:08.54 | CtRiX | that's a synonim of "worksforme(tm)" |
| 16:08.55 | Corydon-w | CtRiX: please work with an Asterisk developer to get your concerns addressed |
| 16:09.09 | Corydon-w | CtRiX: first step would be to file a bug |
| 16:09.18 | ManxPower | Borgon, It works just find with a double nat. Is the Asterisk SERVER behind NAT, or just the phones? |
| 16:10.16 | Borgon | ManxPower:i have 2 asterisk servers.. 1 on a vps working fine, but my laptop is behind a router and nat... and when i have the other asterisk behind router/nat i cant hear incoming sound. |
| 16:10.24 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 16:10.59 | Borgon | ManxPower: am on a wifi and i dont think the router has nat configured correctly but i dont have access to it. |
| 16:11.03 | ManxPower | Borgon, you need to forward the SIP signalling port and the RTP ports on the router that handles NAT for the Asterisk server. |
| 16:11.18 | CtRiX | Corydon-w, i know. But as i already explained, i'm not interested in filing a bug. * developers are reading so i expect at least this thing to be added to a TODO list without filing a bug. |
| 16:11.25 | ManxPower | You do not have to do anything for the NAT router that handles NAT for the SIP client. |
| 16:11.44 | ManxPower | CtRiX, don't expect anything if you don't file a bug |
| 16:11.53 | CtRiX | ManxPower, i know !! |
| 16:12.02 | Corydon-w | CtRiX: so you're not here to help, you're just here to shout out problems and hope someone fixes them? |
| 16:12.10 | Borgon | ManxPower: would the forwarding be done on winxp shared connection port configs.. on via a ssh tunnel from laptop to vps or vice versa to forward and compress the sound better bypassing the firewalls? |
| 16:12.39 | ManxPower | Borgon, an ssh tunnel will screw up packet latency. |
| 16:12.56 | CtRiX | Corydon-w, pointing out that there is a non compliance, to me, is enough for a person involved in * developement to take a note. |
| 16:13.09 | Corydon-w | CtRiX: or are you hoping someone doesn't fix them and users just notice that you're shouting? |
| 16:13.16 | ManxPower | Borgon, I already told you where the port forwarding is needed. I don't know WHAT is doing the NAT for the Asterisk server, but whatever it is you need to do portforwarding on thet bpx. |
| 16:13.24 | Qwell | CtRiX: This is totally the *WRONG* place for you to find somebody to work on something like that |
| 16:13.46 | CtRiX | Corydon-w, i couldn't care less if someone fixes it. |
| 16:14.09 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 16:14.10 | CtRiX | Qwell, the discussion arose when corydon said that * is sip compliant and i just noted that it is not. |
| 16:14.13 | TripleFFFF | yo got a deadlock |
| 16:14.27 | TripleFFFF | show channels says dealock |
| 16:14.28 | CtRiX | i'm not trying to open a bug or have it solved. |
| 16:14.31 | TripleFFFF | shit |
| 16:14.34 | Corydon-w | CtRiX: I never said it was RFC compliant |
| 16:15.07 | CtRiX | anyway, it's compliant enough to work. |
| 16:17.20 | TripleFFFF | oh shit |
| 16:17.46 | CtRiX | Triple(FFFF)-shit ! |
| 16:18.32 | Katty | where's a good article about how to get a company directory into a polcom phone? |
| 16:18.45 | TripleFFFF | ok asterisk is officaly dumped for good.. |
| 16:18.50 | TripleFFFF | deadlock and now box fucked |
| 16:19.00 | *** join/#asterisk Crescendo (n=GWing@cpe-024-211-177-002.nc.res.rr.com) |
| 16:19.04 | ManxPower | TripleFFFF, what version of Asterisk? |
| 16:19.24 | Katty | TripleFFFF: you clearly didn't treat it gentleman style. |
| 16:19.25 | Qwell | TripleFFFF: Didn't you say that yesterday? |
| 16:19.26 | *** join/#asterisk Crescendo (n=martinda@adsl-144-167-184.rmo.bellsouth.net) |
| 16:19.43 | Tall-guy | tripleffff: I had that happen to me when I had horrible interrupt conflicts on the zaptel cards.... |
| 16:20.07 | CtRiX | or when you send a call to an undefined extension in an undefined context from spool |
| 16:20.33 | ManxPower | TripleFFFF, I've basically stopped doing new Asterisk installs until I can get a stable version. |
| 16:20.42 | Crescendo | I'm connecting two SIP clients over NAT - and we can't hear any audio. |
| 16:20.53 | ManxPower | Crescendo, Is Asterisk behind NAT? |
| 16:21.00 | Crescendo | Yes, from his POV. |
| 16:21.14 | Crescendo | I'm right next to it, and can hear anything not beyond the NAT. |
| 16:21.15 | ManxPower | No. Does Asterisk has a public IP address? |
| 16:21.19 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:21.32 | ManxPower | is the router that connects Asterisk to the internet doing NAT? |
| 16:21.35 | Crescendo | Yes. |
| 16:21.53 | TripleFFFF | well im fucked |
| 16:21.57 | TripleFFFF | then i wanted to reboot |
| 16:21.58 | Crescendo | He can connect and dial, answer... everything. |
| 16:22.00 | Crescendo | Just no audio. |
| 16:22.06 | Katty | hmmhesays: you around? |
| 16:22.07 | TripleFFFF | but did a stupid shutdown -h now |
| 16:22.32 | ManxPower | Crescendo, put context=INVALID in [general] in sip.conf, then put the correct context= line for each of the SIP devices. |
| 16:23.03 | rob0 | Hey, just curious, if I add modules to a wctdm card (right now only one FXS), does the card need more interrupts? |
| 16:23.15 | ManxPower | Crescendo, this will make SURE the incoming calls to Asterisk will match the correct sip.conf entry. |
| 16:23.22 | ManxPower | rob0, no |
| 16:23.24 | Crescendo | I'm not sure exactly what a context is. |
| 16:23.32 | Katty | rob0: it just can't share an interrupt with /another/ computery thing |
| 16:23.36 | ManxPower | Crescendo, you are using an Asterisk GUI? |
| 16:23.40 | Katty | rob0: the card itself all uses the same one, i believe. |
| 16:25.07 | TripleFFFF | so my box is down |
| 16:25.07 | TripleFFFF | lol |
| 16:25.11 | rob0 | great, thanks |
| 16:25.13 | TripleFFFF | you know why ? |
| 16:25.23 | Tall-guy | tripleffff: short of Zoloft? |
| 16:25.24 | Katty | because it lacks hugs? |
| 16:25.25 | Crescendo | No, no GUI. |
| 16:25.27 | Tall-guy | :) |
| 16:26.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:26.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:27.14 | Crescendo | ManxPower, what WOULD be the correct context for each SIP device? |
| 16:27.30 | ManxPower | Crescendo, that would depend on your dialplan. What is your current context= lines? |
| 16:28.07 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 16:28.10 | Crescendo | No device even has a context= line. |
| 16:28.15 | Crescendo | >_> |
| 16:28.23 | ManxPower | how about in [general]> |
| 16:28.24 | ManxPower | ? |
| 16:28.31 | TripleFFFF | i had dial (LOCAL/something&Local/SOMETHINGELSE) |
| 16:28.32 | TripleFFFF | folowed by a voiemail |
| 16:28.34 | TripleFFFF | and it deadlocked |
| 16:28.43 | Crescendo | context=default |
| 16:28.44 | TripleFFFF | if it does this again its 100 % reproducable |
| 16:28.44 | Seba_soy | someone has a basic dialer for asterisk? |
| 16:29.04 | TripleFFFF | it deadlocked on the actualy voicemail part |
| 16:29.04 | CtRiX | TripleFFFF, i think it's reproducible 100% |
| 16:29.11 | ManxPower | Crescendo, then put context=default in each of the device sections in sip.conf and put context=INVALID in [general] then do a reload |
| 16:29.20 | *** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 16:29.29 | TripleFFFF | and since asterisk and failover sucks.. well.. we doomed |
| 16:29.31 | TripleFFFF | ;) |
| 16:29.53 | TripleFFFF | dadlock froze screen.. then i killed asterisk process.. BUT then mysql was not connecting |
| 16:30.23 | backblue | how can i do a svn checkout of the last svn asterisk version? i need something like a changelog or something. |
| 16:30.35 | TripleFFFF | -r |
| 16:30.44 | Qwell | backblue: define "last svn asterisk version" |
| 16:30.45 | CtRiX | svn log |
| 16:30.51 | Qwell | There are 3 different branches in svn |
| 16:31.16 | backblue | svn co http://svn.digium.com/svn/asterisk/trunk/ asterisk-svn this? |
| 16:31.41 | backblue | Qwell: well maybe someone has just commited a patch right about now, and i want to check out the last version. |
| 16:31.49 | backblue | i think this will done it |
| 16:31.50 | Qwell | backblue: That would get svn trunk.. |
| 16:31.59 | Qwell | If that's truly what you want, then yes |
| 16:32.13 | Crescendo | Nope, still no audio. Perhaps I haven't routed the correct ports through my firewall? |
| 16:32.28 | ManxPower | Crescendo, you have a firewall? |
| 16:32.56 | Crescendo | Just a router. |
| 16:32.59 | ManxPower | Thank you for wasting 15 mins of my life that I will never get back by not telling me that. |
| 16:33.04 | Crescendo | >_< |
| 16:33.07 | Tall-guy | haahah |
| 16:33.18 | ManxPower | Crescendo, A ROUTER != FIREWALL |
| 16:33.29 | TripleFFFF | router can be firewall |
| 16:33.30 | ManxPower | Now, is your router doing any nat or packet filtering or firewall functions?? |
| 16:33.35 | TripleFFFF | if a csico router.. |
| 16:33.36 | Crescendo | Yes, it is. |
| 16:33.41 | TripleFFFF | but not linksys ceap 100$ crap |
| 16:33.48 | ManxPower | Crescendo, What, specifically, is your router doing? |
| 16:33.51 | Qwell | TripleFFFF: no, a router is always a router... but the same box can also act as a firewall |
| 16:34.01 | Qwell | TripleFFFF: It's a distinction of semantics, but it matters |
| 16:34.07 | TripleFFFF | anything can block ports hence FIREwall |
| 16:34.07 | TripleFFFF | ;) |
| 16:34.48 | ManxPower | Crescendo, SIP Signaling goes over port 5060 UDP. Audio uses the RTP protocol and can go over any ports |
| 16:34.48 | Crescendo | ManxPower, it's one of those DSL gateways. Modem/Router/Firewall - I don't know if it IS a firewall - but to connect any other service externally, I need to put a pinhole through it. |
| 16:35.00 | ManxPower | by default, I think Asterisk defaults to ports 10000 - 20000 |
| 16:35.10 | Qwell | ManxPower: You are correct |
| 16:35.11 | TripleFFFF | ok so.. can one try this ? LOCAL/soething&local/somehingelse that pushed to SIP /outbound crap.. all followed by vmail if no answer..see if deadlock reprodui.. |
| 16:35.32 | Crescendo | 5060 is routed to the appropriate IP on both TCP AND UDP. |
| 16:35.41 | TripleFFFF | also i can garatee that linksys wrt54g with sveasoft QOS.. will reset the tables every 30 secs for traffic shapping dropping the calls |
| 16:35.45 | Crescendo | So... should I forward 10k through 20k as well? |
| 16:35.45 | Qwell | TripleFFFF: pastebin a dialplan sample that causes the issue for you. try to get the smallest/simplest sample you can |
| 16:35.49 | ManxPower | Crescendo, which is why call setup works, but audio does not. |
| 16:35.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:35.57 | hmmhesays | YES the band has a place to practice again |
| 16:35.58 | TripleFFFF | qwell ill try |
| 16:36.03 | ManxPower | Crescendo, you don't FORWARD if you are not doing NAT. You just permit those ports |
| 16:36.11 | Qwell | Crescendo: You don't need tcp 5060, just udp |
| 16:36.49 | backblue | Qwell: that will give me trunk, and it's 1.2.x or 1.4.x? |
| 16:37.01 | Qwell | backblue: neither. like I said, there are 3 branches in svn |
| 16:37.03 | ManxPower | backblue, trunk is trunk |
| 16:37.05 | Qwell | 1.2, 1.4, and trunk |
| 16:37.34 | Qwell | 3 "main line" branches, anyhow |
| 16:37.41 | ManxPower | I'm starting to wonder if people should be required to understand basic IP networking before being allowed to use Asterisk |
| 16:38.08 | Nivex | ManxPower: couldn't hurt |
| 16:38.27 | hmmhesays | ManxPower: yes, 2nd that .. also for apache |
| 16:38.27 | ManxPower | backblue, "trunk" is currently what will become 1.6 |
| 16:38.35 | backblue | ManxPower: ok tks |
| 16:38.54 | backblue | so i have to checkout branches/1.2/ |
| 16:38.57 | rene1 | elitism doesnt help really.. you can always point people to the fiendly manual |
| 16:39.14 | *** part/#asterisk eMZed (n=marko@rikom.sk-branik.si) |
| 16:39.20 | rene1 | s/fiendly/friendly/ |
| 16:39.47 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:41.03 | hmmhesays | ok, how do I keep asterisk from writing any cdrs at all |
| 16:41.27 | Qwell | hmmhesays: disable them in the configs, or remove the modules (or noload them) |
| 16:41.42 | [TK]D-Fender | hmmhesays: Symlink "cdr_csv" to /dev/null :) |
| 16:43.11 | bkw_ | or no load it |
| 16:43.23 | Crescendo | Alright, I've removed the unnecessary TCP port, and the option to simply "allow" a port isn't part of this particular device. I forwarded them instead, to the IP of the server. |
| 16:43.49 | Crescendo | Would it make better sense to place the server in the DMZ to avoid all of this? |
| 16:43.52 | Qwell | bkw_: I said that, nub :p |
| 16:44.22 | bkw_ | what RFC states that you must support both TCP and UDP for signalling? |
| 16:44.23 | [TK]D-Fender | Crescendo: Only if you want to open everything on it. You only need 5060-5070 (fatety range), and 10000-20000 anll UDP |
| 16:44.55 | [TK]D-Fender | bkw_: That'd be "No Comment" ;) |
| 16:45.07 | bkw_ | I was reading it the other night and it clearly said you MUST do it |
| 16:45.25 | Crescendo | Currently, we only have 5060 pointed to the server, and 10000-20000 |
| 16:45.36 | Crescendo | I'll adjust the first hole to 5060-5070 |
| 16:45.47 | Tall-guy | crescendo: what router is it? I'm about to give you some dmz advice :) |
| 16:45.54 | [TK]D-Fender | Crescendo: Ok, well then your router's side is taken care of and you just need to ensure your * SIP setup is configured accordingly |
| 16:46.03 | lirakis | does any one know how to add P-Asserted ID's ???? |
| 16:46.04 | TripleFFFF | http://pastebin.ca/184179 |
| 16:46.16 | TripleFFFF | Qwell: http://pastebin.ca/184179 |
| 16:46.25 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:46.36 | Crescendo | This is a belkin cable/dsl gateway, Version Info |
| 16:46.36 | Crescendo | <PROTECTED> |
| 16:46.36 | Crescendo | 4.05.03 |
| 16:46.37 | Crescendo | <PROTECTED> |
| 16:46.37 | Crescendo | 2.01.09 |
| 16:46.37 | Crescendo | <PROTECTED> |
| 16:46.41 | Crescendo | F5D7230-4 |
| 16:46.43 | Crescendo | >_< |
| 16:46.49 | TripleFFFF | Qwell cant be more simple that that |
| 16:47.13 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:47.32 | Qwell | Why are you using ${MACRO_EXTEN} in a context? |
| 16:47.57 | Tall-guy | crescendo: to echo tkd-fender, the DMZ is a "pants around your knees, wide open, let the world into my box setting"....so don't do it. |
| 16:48.23 | TripleFFFF | no idea |
| 16:48.29 | TripleFFFF | i copied part of code from a macro |
| 16:48.50 | Crescendo | Ah, excellent. |
| 16:48.56 | TripleFFFF | bu that shouldnt break it.. would be an empty var |
| 16:49.01 | Crescendo | Lol. |
| 16:49.06 | Crescendo | So no DMZ for me. |
| 16:49.22 | Crescendo | Looks like I fixed it with the routing of ports, though. |
| 16:49.37 | Crescendo | So my pants are only unzipped a little bit. |
| 16:49.41 | Crescendo | Thanks for everyone's help. |
| 16:49.42 | Crescendo | :) |
| 16:49.48 | Tall-guy | crescendo: just enough to get caught for flashing |
| 16:49.49 | bkw_ | um |
| 16:49.57 | bkw_ | the cow still got out of the barn |
| 16:50.04 | TripleFFFF | dialplan is alot bigger then that but.. thats the simplified.. actualy get number does crap then pushed to rebounds if a failover follow me.. then finds if usa/canada in a macr then finds the best providers for that destiantion then routes to outbound-usa or canada.. that does this |
| 16:50.05 | *** join/#asterisk sageang (n=ang40@machine77.Level3.com) |
| 16:50.23 | TripleFFFF | actualy its 2k lines |
| 16:51.32 | Borgon | hello, my sip incoming calls isnt playing. does port 8000 need to show on netstat -pan ? i see port 5060 showing and open |
| 16:51.53 | TripleFFFF | no for 8000 |
| 16:51.56 | *** join/#asterisk tRSS (n=tRSS@202.125.133.130) |
| 16:51.57 | TripleFFFF | DMZ the box then try |
| 16:52.56 | Borgon | TripleFFFF: the asterisk server is a vps on dmz already..... my home laptop is behind a wifi/router nat firwall whatever |
| 16:53.34 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:53.53 | sageang | can anyone recommend how to ... when someone leaves a message on a certain extension, i need to have asterisk call a numeric pager and then send a predefined 10 digit number to that numeric pager |
| 16:54.24 | Tall-guy | sageang: is the predeifned 10 digit number unique to eqch voicemailbox/extensions? |
| 16:54.41 | sageang | yes |
| 16:55.26 | Tall-guy | sageang: thinking.... |
| 16:55.44 | sageang | no worries, thanks for your help |
| 16:57.41 | *** join/#asterisk GiabboO (n=GiabboO@host101-246-static.73-81-b.business.telecomitalia.it) |
| 16:57.43 | GiabboO | hey ppl |
| 16:57.48 | GiabboO | when a phone have dual RJ45 |
| 16:57.55 | Qwell | GiabboO: it's a switch |
| 16:57.59 | GiabboO | :D |
| 16:58.04 | De_Mon | Tall-guy isn't that in voicemail.conf already? |
| 16:58.05 | GiabboO | Qwell you nostradamus |
| 16:58.05 | Qwell | simple layer 2, 2 port switch |
| 16:58.16 | GiabboO | thanks, I need it :) |
| 16:58.22 | GiabboO | i am about to buy this |
| 16:58.27 | GiabboO | http://store.pbxeq.com/store/product.php?productid=16162&cat=249&bestseller |
| 16:58.32 | GiabboO | what do u think Qwell |
| 16:58.41 | De_Mon | I bet it's a hub not a switch |
| 16:58.51 | GiabboO | Its ok |
| 16:58.54 | Qwell | De_Mon: no, it's gonna be a switch |
| 16:59.15 | GiabboO | Support for Layer 2 |
| 16:59.16 | GiabboO | :) |
| 16:59.17 | [TK]D-Fender | NostraDumbAss ;) |
| 16:59.21 | Tall-guy | de_mon: dunno, I don't page at all |
| 16:59.23 | GiabboO | LOL TK |
| 16:59.32 | Tall-guy | haahahah |
| 16:59.42 | [TK]D-Fender | (idiot who thinks they can predict whats going on) |
| 16:59.52 | GiabboO | hihi |
| 16:59.55 | [TK]D-Fender | :D |
| 17:00.00 | [TK]D-Fender | <- for teh winy0! |
| 17:00.02 | GiabboO | Fender Mark III ? |
| 17:00.02 | GiabboO | lol |
| 17:00.42 | GiabboO | ok |
| 17:00.47 | De_Mon | Tall-guy go look at what voicemail CAN do and then come back |
| 17:00.47 | GiabboO | Its cheap |
| 17:00.49 | GiabboO | i will buy that |
| 17:01.00 | Qwell | GiabboO: I wouldn't recommend a grandstream |
| 17:01.07 | GiabboO | ups |
| 17:01.16 | GiabboO | sucker components ? |
| 17:01.22 | Tall-guy | de_mon: I didn't ask....YOU go look :) |
| 17:01.24 | Qwell | I guess I need to be vendor neutral now :p |
| 17:01.31 | GiabboO | :P |
| 17:01.38 | GiabboO | I have to buy around 80 ip phones |
| 17:01.43 | Qwell | GiabboO: for a business? |
| 17:01.44 | GiabboO | to my company |
| 17:01.48 | GiabboO | yes |
| 17:01.49 | Qwell | I would recommend polycoms |
| 17:01.53 | De_Mon | Tall-guy oops tell sageang to go look |
| 17:01.57 | GiabboO | let me check |
| 17:02.01 | Tall-guy | de_mon: I'll be sure and let him know :) |
| 17:02.01 | Qwell | They're a bit more pricey, but they are worth it |
| 17:02.19 | GiabboO | omg |
| 17:02.20 | GiabboO | 195$ |
| 17:02.27 | [TK]D-Fender | Polycom / Aastra get my seal of approval. |
| 17:02.47 | [TK]D-Fender | GiabboO: Where are you located? |
| 17:02.47 | GiabboO | Italy |
| 17:02.49 | justinu|laptop | qwell recommending polycom? what alternate reality did I wake up in? |
| 17:02.55 | GiabboO | im checking prices on pbexq |
| 17:02.57 | TripleFFFF | ok |
| 17:03.03 | Strom_C | yeah, thats what im wondering too, justinu :D |
| 17:03.04 | GiabboO | Italy is more expensive |
| 17:03.14 | [TK]D-Fender | GiabboO: Look at Aastra if Polycom is pricy. Please provide links to the vendors you are searching at as well. |
| 17:03.25 | sageang | i have looked at voicemail and it seems to only be able to send email |
| 17:03.32 | Tall-guy | giaboo: is that $USD? |
| 17:03.34 | [TK]D-Fender | justinu|laptop: Yeah, like WTS eh? :) |
| 17:03.53 | GiabboO | we got € |
| 17:03.53 | GiabboO | :)) |
| 17:04.03 | GiabboO | where is pbexq located ? |
| 17:04.04 | [TK]D-Fender | GiabboO: What models are you looking at? And plesae link directly if possible |
| 17:04.18 | GiabboO | I am looking for a SIP phone with dual lan |
| 17:04.22 | GiabboO | dual eth |
| 17:04.32 | justinu|laptop | most of the good ones have that |
| 17:04.37 | GiabboO | i dont want spend tons of money |
| 17:04.49 | [TK]D-Fender | GiabboO: Many phones have that. Aastra 9133, 4801, Polycom IP series ALL have dual RJ45's |
| 17:04.58 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 17:05.17 | GiabboO | Aastra is much expensive |
| 17:05.20 | Tall-guy | giaboo: cheap IP phones are....well, .....cheap.... |
| 17:05.31 | GiabboO | :) |
| 17:05.38 | Tall-guy | I can buy the ip501 and the aastra480i for essentially the same price..... (200$ CDN) |
| 17:05.41 | GiabboO | well I dont need a supercool phone |
| 17:05.46 | [TK]D-Fender | GiabboO: Please provide a link to the place you are looking at and not just say "expensive". |
| 17:05.49 | GiabboO | I just need something over ip that ring |
| 17:06.11 | GiabboO | im looking at www.pbexq.com |
| 17:06.32 | Strom_C | GiabboO: you want to consider durability and usability as well |
| 17:06.36 | justinu|laptop | do you already have POTS phones? get a bunch of ATAs |
| 17:06.39 | Strom_C | especially if this is for a business |
| 17:06.44 | GiabboO | yes.. |
| 17:06.47 | [TK]D-Fender | GiabboO: That site is not working for me... |
| 17:06.58 | Qwell | [TK]D-Fender: it's REALLY slow |
| 17:07.02 | GiabboO | opsssssss |
| 17:07.08 | GiabboO | www.pbxeq.com |
| 17:07.10 | GiabboO | sorry |
| 17:07.11 | GiabboO | :) |
| 17:07.30 | GiabboO | I got PBX (PSTN) |
| 17:07.38 | GiabboO | and pots phones |
| 17:07.50 | justinu|laptop | and you want to do what exactly... |
| 17:07.55 | GiabboO | but I want to reach those phones without pass trought the pbx |
| 17:08.05 | GiabboO | its boring in my business |
| 17:08.11 | GiabboO | call must be direct to the phones |
| 17:08.22 | GiabboO | i am going to structure 15 stores |
| 17:08.29 | justinu|laptop | so you want DID capability? |
| 17:08.30 | *** join/#asterisk c_lees (n=c_lees@machine77.Level3.com) |
| 17:08.31 | [TK]D-Fender | GiabboO: And these guys are in Italy? |
| 17:08.32 | GiabboO | every stores needs 4/5 phones |
| 17:08.43 | GiabboO | its my stores point of sales |
| 17:08.49 | GiabboO | supermarkets :P |
| 17:08.58 | GiabboO | yes justinu |
| 17:09.06 | GiabboO | ill get a PRI E1 15channel line |
| 17:09.11 | justinu|laptop | and your existing pbx can't do DID? |
| 17:09.12 | GiabboO | with DID support |
| 17:09.14 | [TK]D-Fender | GiabboO: I mean that vendor is local to you and would sell at the price stated in USD$ to your door? |
| 17:09.20 | GiabboO | Im using Asterisk PBX |
| 17:09.24 | c_lees | sageang you ever get the answer? |
| 17:09.33 | justinu|laptop | oh, you're using asterisk already? |
| 17:09.35 | GiabboO | I could buy in $ |
| 17:09.42 | sageang | no but tallguy was thinking about it |
| 17:09.47 | GiabboO | justinu, actually, each stores got a own pbx |
| 17:09.50 | Tall-guy | (and still is:) ) |
| 17:09.51 | justinu|laptop | so how are your pots phones connected to asterisk? |
| 17:09.54 | c_lees | lol |
| 17:10.02 | GiabboO | now I installed dsl in each point |
| 17:10.09 | GiabboO | and ill move the actual pbx |
| 17:10.14 | GiabboO | to ip phones |
| 17:10.25 | justinu|laptop | oh, good luck... hope that DSL is solid |
| 17:10.30 | GiabboO | it is. |
| 17:10.38 | Tall-guy | giaboo: what kinda pbx? |
| 17:10.49 | GiabboO | all calls will pass trough the master dsl |
| 17:10.58 | GiabboO | which have the pri channels |
| 17:11.10 | GiabboO | old pbx |
| 17:11.16 | GiabboO | isdn pbx |
| 17:12.21 | Tall-guy | giaboo: what are you handsets now? |
| 17:12.26 | Tall-guy | s/your/you |
| 17:12.50 | GiabboO | uh ? |
| 17:12.56 | GiabboO | are you talking about phones ? |
| 17:12.59 | Tall-guy | yup |
| 17:13.03 | Tall-guy | aka handset |
| 17:13.05 | GiabboO | normal pstn |
| 17:13.17 | GiabboO | very cheap :P |
| 17:13.17 | Tall-guy | um....your PBX...what kinda pbx is it? |
| 17:13.19 | *** part/#asterisk oej (n=oej@231.Red-88-14-197.dynamicIP.rima-tde.net) |
| 17:13.24 | Tall-guy | ie: brand |
| 17:13.30 | GiabboO | Uh |
| 17:13.51 | GiabboO | something like Bticino PABX |
| 17:14.07 | GiabboO | Italian stuff Tguy |
| 17:14.13 | *** join/#asterisk gissi (n=silvio@32.104.18.240) |
| 17:14.22 | justinu|laptop | does scuderia ferrari make a PBX yet? |
| 17:14.28 | GiabboO | rofl |
| 17:14.28 | Tall-guy | giaboo: I was only asking because you could potentialy use your existing handsets (phones), if they were analog, with Zaptel cards... (not saying it was best...just a thought) |
| 17:14.28 | GiabboO | :D |
| 17:14.31 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 17:14.50 | GiabboO | those pbx are not in the same place eh... |
| 17:14.59 | [TK]D-Fender | Zaptel cards for FXS deployment = a MISTAKE and total waste...... |
| 17:15.03 | GiabboO | and they are old, i dont think they support external cards |
| 17:15.06 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 17:15.47 | GiabboO | look... |
| 17:15.47 | GiabboO | I show u |
| 17:15.47 | GiabboO | ) |
| 17:15.47 | gissi | Hello all... I have just tried to compile * 1.4 beta2 on Ubuntu (Kubuntu actually) and had problems with PWLib and OpenH323... Is this available on a wiki anywhere? |
| 17:15.47 | Tall-guy | tkd: each situation is unique, I would hesitate to label all projects with fxs a total waste. |
| 17:15.47 | GiabboO | http://www.bticino.es/pfw_files/cma/news/pbx.jpg |
| 17:15.47 | GiabboO | old school! |
| 17:16.01 | GiabboO | rs232 support ^^ |
| 17:16.20 | GiabboO | have u ever setted up a pabx like that |
| 17:16.25 | GiabboO | is boring stuff |
| 17:16.25 | GiabboO | :P |
| 17:16.32 | [TK]D-Fender | Tall-guy: I said "ZAPTEL" FXS.... |
| 17:16.47 | [TK]D-Fender | Tall-guy: Important adjective! |
| 17:16.51 | GiabboO | I will use FXS |
| 17:16.59 | GiabboO | only to connect asterisk to fax machines |
| 17:17.05 | GiabboO | or ATA |
| 17:17.13 | [TK]D-Fender | Tall-guy: FXS is just fine, but get it running through a SIP gateway of some sort and out of *'s DTMF grabbing hands! |
| 17:17.14 | GiabboO | i have to choise is not really important |
| 17:17.29 | GiabboO | how does work FAX over SIP ATA ? |
| 17:17.32 | TripleFFFF | Qwell any idea ? |
| 17:17.42 | justinu|laptop | not to well, if your qos isn't perfect |
| 17:17.55 | GiabboO | i have cisco 1800 series |
| 17:17.58 | GiabboO | i can control the qos |
| 17:18.17 | GiabboO | anyhow, voip system is expensive |
| 17:18.33 | GiabboO | but in years i will reduce telecomunication costs |
| 17:18.49 | GiabboO | expecially "company inside" traffic |
| 17:18.51 | justinu|laptop | fax over ip is really tricky... read steve underwood's page about it |
| 17:18.53 | justinu|laptop | http://soft-switch.org/foip.html |
| 17:18.58 | GiabboO | that is 80% of actual costs |
| 17:19.19 | Seba_soy | somebody knows why a zap-generic fxo card gives me answered instead ringing when I place calls? |
| 17:19.55 | Seba_soy | destination phone start ringing, but zaptel report ANSWERED!! |
| 17:19.55 | Seba_soy | so, bad billing info |
| 17:19.55 | GiabboO | actually |
| 17:19.55 | GiabboO | my asterisk pbx |
| 17:19.55 | GiabboO | is connected to isdn line |
| 17:19.59 | GiabboO | with a bri card |
| 17:20.03 | GiabboO | and fax works |
| 17:20.08 | GiabboO | fax to mail i mean |
| 17:20.39 | Seba_soy | can it be some miss configuration on zapata.conf? |
| 17:20.39 | justinu|laptop | fax to mail is different... read the site |
| 17:20.58 | *** part/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 17:21.00 | justinu|laptop | store and forward fax is fairly easy to do... realtime fax otoh |
| 17:21.16 | directory | Seba_soy: getting accurate call progress on an analog line is very very difficult, so if you need that then you shouldn't use analog |
| 17:21.41 | Seba_soy | directory: :), sorry, I am newbie, what? |
| 17:22.27 | directory | Seba_soy: knowing whether something is ringing or busy is known as call progress... on analog you don't get that really |
| 17:22.34 | Borgon | Crescendo: did u sovle ur audio problem? |
| 17:23.00 | Crescendo | Somewhat. |
| 17:23.06 | directory | there is an option called callprogress which tries to determine it.... BUT it may cause your calls to hang up in the middle of them |
| 17:23.24 | Crescendo | As far as I can tell, he can hear audio now. I suspect that the reason I can't hear him is that his mic may not be working. |
| 17:23.36 | Seba_soy | i read it, it says it is for FXS |
| 17:23.43 | Seba_soy | i am using FXO |
| 17:23.52 | Seba_soy | aaaa ok... |
| 17:23.56 | Seba_soy | it is inverted |
| 17:23.59 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:24.14 | directory | don't say I didn't warn you if it hangs up on you |
| 17:24.24 | directory | or doesn't work, because it only works for some countries |
| 17:24.32 | Seba_soy | yes, US only |
| 17:24.38 | Seba_soy | I live in argentina |
| 17:24.44 | hmmhesays | yes you do |
| 17:24.53 | hmmhesays | there are lots of attractive women their |
| 17:24.56 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 17:25.25 | Seba_soy | :) |
| 17:25.53 | Assid | hey how come asterisk hasnt posted a changelog for 1.4 |
| 17:26.06 | directory | Assid: because it's being written |
| 17:26.15 | stubert | Is there a way to use the Dial command and then drop the calling connection, but have the Dial command continue until the M(macroname) command completes? |
| 17:26.15 | Assid | hrmm |
| 17:26.17 | brif8 | If Asterisk Box is on Public IP with NIC a and the Phones are on the Private LAN with NIC b do you still need to NAT / forward packets from NIC a to NIC b with iptables or not ? |
| 17:26.21 | Assid | more waitin.. |
| 17:26.22 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 17:26.22 | kink0 | hi |
| 17:26.23 | directory | it's not something that can be done overnight, there are a LOT of changes... even minor ones |
| 17:26.29 | directory | plus it's in beta, not released yet |
| 17:26.33 | Assid | right. |
| 17:26.44 | Assid | but i would like to know what the update before i do update |
| 17:27.05 | Assid | oh well. no issues.. i aint going anywhere |
| 17:27.13 | [TK]D-Fender | brif8: Don't need to, and I'd set "canreinvite=no" globally if I were you. |
| 17:27.17 | justinu|laptop | stubert app dial is one of asterisk's Achilles heels IMO |
| 17:27.38 | Assid | stubert app dial ? |
| 17:27.44 | [TK]D-Fender | brif8: Taht will have all calls pass through the * server and should wok just fine |
| 17:29.01 | stubert | Assid: yes, I'm using a call file to dial a numeric pager, but it's a bit ugly... |
| 17:29.32 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:29.33 | Assid | i should ty playing wih that someday |
| 17:29.42 | brif8 | [TK]D-Fender: do you mean me ? |
| 17:29.48 | shmaltz | anybody here know the Avaya Partner ACS system? |
| 17:29.52 | stubert | I'd love to be able to just fire the page process off as a Dial command |
| 17:30.22 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:30.30 | CunningPike | If I have an SPA-3000 and it shares a POTS line with other analog phones, is there a way to way to make it ring through to a SIP phone on Asterisk while leaving the other analog phones ringing? |
| 17:30.30 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:30.53 | shmaltz | CunningPike, yes, don't answer the line |
| 17:30.55 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:31.21 | [TK]D-Fender | brif8: I did at least type your name right in there :) |
| 17:31.23 | shmaltz | CunningPike, what firmware? |
| 17:31.47 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:31.53 | [TK]D-Fender | CunningPike: Yes, check out the forums on voxilla.com for info on the exact how-to's |
| 17:31.58 | CunningPike | shmaltz: Not sure yet - it's still in the box :) It's a Linksys rather than a Sipura, though |
| 17:32.03 | CunningPike | [TK]D-Fender: WIll do - thanks |
| 17:32.07 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:32.30 | shmaltz | CunningPike, then it will work out of the box |
| 17:32.35 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 17:32.44 | CunningPike | shmaltz: Perfect |
| 17:33.14 | shmaltz | anybody here know the Avaya Partner ACS system? |
| 17:33.26 | CunningPike | I'm replacing the POTS phone in my home office with a line appearance on a Polycom 501, but I want the POTS phones in the rest of the house to ring as well |
| 17:33.41 | Seba_soy | I will try adding zonedata for argentina! |
| 17:34.50 | [TK]D-Fender | CunningPike: Keep in mind the CID delay factor or be prepared to give it up. |
| 17:35.03 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-5d8c1a2377240c16) |
| 17:35.03 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:35.06 | CunningPike | [TK]D-Fender: Aye |
| 17:35.17 | directory | WAIT! NO! |
| 17:35.22 | directory | better. |
| 17:35.47 | Cresl1n | hey directory |
| 17:35.57 | [TK]D-Fender | Cresl1n:tackled? |
| 17:36.03 | directory | Cresl1n: I know where you are... |
| 17:36.10 | Cresl1n | lol |
| 17:36.11 | Qwell | [TK]D-Fender: Cresl1n[tackled] |
| 17:36.32 | Cresl1n | ummmm |
| 17:36.35 | [TK]D-Fender | :O |
| 17:36.51 | *** join/#asterisk huslage (n=huslage@hact.net) |
| 17:36.53 | [TK]D-Fender | Where RTP stream meed Field & Stream! |
| 17:36.57 | huslage | howdy |
| 17:36.59 | [TK]D-Fender | meet* |
| 17:37.52 | huslage | does anyone know what this stuff means: http://rafb.net/paste/results/1EPAcr38.html |
| 17:38.59 | huslage | i'm getting it a TON in my messages log |
| 17:42.23 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:43.37 | *** join/#asterisk argentinaasteris (n=mbianchi@200.68.94.91) |
| 17:43.55 | argentinaasteris | hello |
| 17:44.46 | argentinaasteris | DTMF tones does not pass trough a zap channel to a sip phone |
| 17:48.16 | fafnir | and then something maginal happens |
| 17:48.21 | fafnir | and a zerbra is formed |
| 17:49.45 | argentinaasteris | oh, how funny! |
| 17:49.50 | argentinaasteris | hilarious |
| 17:50.05 | huslage | big problem. using AEL with a lot of SQL queries, asterisk and mysql need to be restarted every few hours |
| 17:50.15 | huslage | there is no obvious error. |
| 17:50.34 | argentinaasteris | what do you mean "is no obvius error" |
| 17:50.43 | huslage | i mean it just stops passing calls |
| 17:50.56 | argentinaasteris | and how can you solve this isue |
| 17:51.10 | argentinaasteris | i wrote relaxdtmf=yes in zapata.conf |
| 17:51.16 | argentinaasteris | and relaxdtmf=no also |
| 17:51.33 | huslage | the sequence: call comes in, mysql connection is started, queries issued,call end, mysql connection is torn down. |
| 17:52.02 | huslage | yes, argentinaasteris, i'm searching for ideas on solving this mystery |
| 17:52.12 | argentinaasteris | thank you very much |
| 17:56.18 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:56.42 | ghenry | hi all, anyone done a simple Perl program to connect and send a few command via the Manager? |
| 17:56.57 | ghenry | I'm pasting mine, which doesn't seem to be working |
| 17:57.13 | CunningPike | 1, 2, 3...... |
| 17:57.15 | CunningPike | ~pb |
| 17:57.21 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:57.23 | Aces1Up | hey kink :) |
| 17:57.50 | benjk | huslage, prolly deadlocking |
| 17:59.07 | ghenry | CunningPike: http://scsys.co.uk:8001/4217 |
| 17:59.20 | ghenry | maybe return/newlines? |
| 17:59.36 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 17:59.38 | ghenry | with debug/verbose 10 I can see the admin login/logoff |
| 17:59.50 | ghenry | but no info about the command I am trying to do |
| 18:00.01 | Juggie | are you not using enough returns |
| 18:00.08 | Juggie | its two to issue your command |
| 18:00.23 | Juggie | so you have to do two after your login, then two after your command |
| 18:00.45 | *** join/#asterisk hassler (n=hassler@r-corp.hcst.com) |
| 18:00.46 | ghenry | If I have 2 after login, the login doesn't work |
| 18:00.50 | ghenry | will try Ju |
| 18:00.53 | ghenry | will try Juggie |
| 18:01.02 | Juggie | it should. |
| 18:01.09 | ghenry | tesing, thanks |
| 18:01.21 | Juggie | by 2 i mean the one at the end of the line |
| 18:01.24 | Juggie | and then one black line return |
| 18:01.27 | jtexter3 | ghenry: Another option is to modify your code to dump what you are sending to a file. Then, copy that text, and paste it into a telnet session to see if it's working |
| 18:01.40 | ghenry | true, tahnks |
| 18:01.52 | huslage | benjk, how would i debug the deadlock? |
| 18:01.54 | Juggie | or jsut output it to the shell :) |
| 18:01.56 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:01.57 | Juggie | *just |
| 18:02.03 | hassler | Hi Folks, anyone familiar with BLF - Busy LIne Function / Indication (sip "hints")? Having problems getting it to work on one *, works fine on another, but can't figure out why! |
| 18:02.29 | jtexter3 | ghenry: I also found I have to use carriage return and line feeds (\r\n in C), not just new line |
| 18:02.34 | benjk | huslage, debugging deadlocks is a bit of a challenge |
| 18:02.50 | ghenry | I have that jtexter3 |
| 18:02.55 | huslage | benjk, okay...for instance. if i do this: |
| 18:02.56 | huslage | mysql(Connect connid2 ${MASTER-MYSQL-SERVER} ${MYSQLUSER} ${MYSQLPASSWORD} asteriskdb); |
| 18:03.15 | huslage | multiple times within a context, does that actually create a new connection with the same handle each time |
| 18:03.29 | benjk | one method is to log every single lock with a unique ID into a log file and thereby try to see which one caused the deadlock when it happens |
| 18:03.29 | CunningPike | hassler: What phones? |
| 18:04.26 | hassler | Grandstream GXP2000 -- one extension tied to my "home" PBX (freepbx) works fine, extensions to work PBX do not. |
| 18:04.40 | ghenry | I just get admin logon/logoff Will try dumping to a file and testing first |
| 18:04.42 | ghenry | thanks all |
| 18:04.48 | hassler | I don't see the work PBX even sending out the SIP traffic, although the subscriptions appear to be registered |
| 18:04.59 | De_Mon | how can I monitor the pbx and verify it is receiving calls on a relgular basis? |
| 18:05.13 | Juggie | log files, console, snmp on 1.4+ |
| 18:05.30 | huslage | explain how to log the locks, benjk |
| 18:05.49 | benjk | you have to go into the source code for that |
| 18:05.52 | CunningPike | hassler: Hmmm - we'on Polycoms here, but from an Asterisk pov, all we did was create entries like exten => foo,hint,SIP/foo in our extensions.conf |
| 18:05.55 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 18:06.06 | CunningPike | hassler: The rest is up to the phones |
| 18:06.10 | De_Mon | Juggie yesterday asterisks stopped responding to all sip peers, nothing showed up in the logs or console to indicate there was a problem |
| 18:06.19 | CunningPike | hassler: What asterisk version is your work install? |
| 18:06.44 | De_Mon | someone had to call me and say my phone doesnt work help! |
| 18:06.47 | benjk | De_Mon, that's a likely deadlock too |
| 18:07.10 | De_Mon | I thought deadlocks generaly locked the whole system, I was able to restart it fine... |
| 18:07.18 | Juggie | hmmmm |
| 18:07.18 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:07.23 | De_Mon | it meaning asterisk |
| 18:07.28 | Juggie | what did like show channels or sip show peers or etc show you |
| 18:07.28 | justinu|laptop | asterisk sip stack will crash/deadlock if it gets fragmented packets |
| 18:07.46 | De_Mon | show channels was empty |
| 18:07.50 | justinu|laptop | i had a misconfigured switch sending it bad packets, and it didn't like it one bit |
| 18:08.11 | De_Mon | show peers said the status of all peers was UNKNOWN |
| 18:08.17 | Juggie | justinu, i would have rathered you fixed * then fixed the switch |
| 18:08.23 | Juggie | if you cuold recreate that problem it would be helpful |
| 18:08.33 | Juggie | because if sending malformed packets can crash sip then thats a DDOS |
| 18:09.28 | De_Mon | any other ideas? |
| 18:09.28 | huslage | alright. another question. is a MYSQL(Connect) statement context specific or is it call specific? |
| 18:09.32 | justinu|laptop | yeah, i think what happened is sip stack read partial packet, locked the channel the packet was destined for, then barfed, never unlocking the channel |
| 18:09.57 | justinu|laptop | unfortunately, i'm not interesting in disclaiming my code, so no fixes |
| 18:10.03 | justinu|laptop | s/interesting/interested |
| 18:11.28 | De_Mon | huslage neither |
| 18:11.36 | aptura | Cunning did you get your polys from ccp? |
| 18:11.47 | huslage | De_Mon, then what is it? |
| 18:11.55 | benjk | justinu, you are just trolling here, go away |
| 18:12.09 | justinu|laptop | that's mr. troll to you, sir |
| 18:12.11 | De_Mon | hmmm, huslage I take that back, it'd be call specific |
| 18:12.12 | justinu|laptop | :) |
| 18:12.23 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:12.29 | TripleFFFF | ep 27 14:11:47 WARNING[6809]: app_cut.c:320 cut_exec: The application Cut is deprecated. Please use the CUT() function instead. |
| 18:12.30 | TripleFFFF | ? |
| 18:12.43 | [TK]D-Fender | justinu|laptop != troll |
| 18:12.52 | TripleFFFF | ok so we talking about casxe sensitive or ? 1.2.12.1 |
| 18:12.56 | [TK]D-Fender | TripleFFFF: That message is pretty blantant.... |
| 18:12.59 | CunningPike | aptura: No, from Microserve |
| 18:13.01 | De_Mon | benjk is just trolling ;P |
| 18:13.03 | huslage | ok De_Mon ... that's what i thought, but i wanted some external confirmation |
| 18:13.04 | TripleFFFF | well cut=CUT |
| 18:13.31 | [TK]D-Fender | aptura: I do. |
| 18:13.34 | benjk | justinu, how about contributing your fix to opbx then, no disclaimer required |
| 18:13.41 | TripleFFFF | Cut(name=CHANNEL,,1) |
| 18:13.44 | TripleFFFF | im using this |
| 18:13.45 | justinu|laptop | certainly |
| 18:13.48 | aptura | microserv the isp? |
| 18:13.49 | De_Mon | could it even be a channel deadlock if there were no channels? |
| 18:13.58 | [TK]D-Fender | TripleFFFF: : Show the ENTIRE line please. |
| 18:14.04 | De_Mon | sip show channels doesn't show any |
| 18:14.12 | TripleFFFF | exten => s,1,CUT(name=CHANNEL,,1) |
| 18:14.27 | [TK]D-Fender | TripleFFFF: You are calling the old APPLICATION, not the new FUNCTION. |
| 18:14.27 | *** join/#asterisk Unistim_junky (n=rover@216-230-88-10.client.cypresscom.net) |
| 18:14.27 | TripleFFFF | Demon |
| 18:14.41 | TripleFFFF | DEADLOCKS will stop the thread of channel list from being accessed |
| 18:14.48 | De_Mon | ah |
| 18:14.50 | benjk | De_Mon, in certain cases * locks the entire list, in that event, you couldnt do show channels |
| 18:14.52 | TripleFFFF | meaning you cant list them or create new ones.. meaning no new calls etc |
| 18:14.58 | TripleFFFF | i just got a deadlock 1 hour ago |
| 18:15.06 | TripleFFFF | had to reboot.. since it will also kill mysql |
| 18:15.07 | De_Mon | that sounds like what happened alrighty |
| 18:15.23 | TripleFFFF | screen freeze + killall -9 asterisk wont help.. |
| 18:15.24 | benjk | Rocky Mountain locking |
| 18:15.35 | [TK]D-Fender | TripleFFFF: exten => s,1,Set(name=${CUT(CHANNEL,,1)}) |
| 18:15.45 | TripleFFFF | ok whats the diff tk ? |
| 18:15.46 | justinu|laptop | when my particular deadlock was happening, I got a repeating message in full log |
| 18:15.50 | TripleFFFF | oh |
| 18:15.50 | TripleFFFF | lol |
| 18:15.58 | De_Mon | I was able to do a restart now and everything went back to normal |
| 18:16.14 | benjk | latest RML technology |
| 18:16.36 | justinu|laptop | anyways, if you look at the chan_sip code, i'm sure you could find TONS of ways to DDOS it |
| 18:16.39 | TripleFFFF | heyu ? how you do that |
| 18:16.52 | CunningPike | aptura: Microserve the reseller |
| 18:16.59 | benjk | isnt TON a PRI thing? |
| 18:17.00 | [TK]D-Fender | TripleFFFF: On is an application that is going to be removed entirely in 1.4 and the other is a FUNCTIOn that you can call in-line with other things |
| 18:17.08 | TripleFFFF | oh ok |
| 18:17.11 | TripleFFFF | thanks.. |
| 18:17.23 | TripleFFFF | but |
| 18:17.34 | CunningPike | aptura: http://microserve.ca/ |
| 18:17.40 | [TK]D-Fender | TripleFFFF: String-based apps are being removed in place of functions all-around |
| 18:17.52 | TripleFFFF | cuT(name=CHANNEL,,1) = cisco1 |
| 18:17.52 | TripleFFFF | Set(name=${CUT(CHANNEL,,1)}) = SIP/cisco1 |
| 18:18.28 | benjk | couldn't you have some more parentheses in there? |
| 18:18.34 | [TK]D-Fender | TripleFFFF: "show function CUT" |
| 18:18.35 | benjk | I mean it looks kinda empty |
| 18:18.52 | [TK]D-Fender | benjk: You're right... we CAN. Let me att hem now! |
| 18:18.53 | TripleFFFF | Sep 27 12:13:21 WARNING[32131]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0xb7db2bc8', 10 retries! |
| 18:18.55 | TripleFFFF | again |
| 18:18.56 | TripleFFFF | hmm |
| 18:19.23 | benjk | :) |
| 18:20.02 | benjk | defintely the SIP/cisco1 part looks very lonely without parens |
| 18:20.16 | TripleFFFF | ? |
| 18:20.27 | TripleFFFF | i mean that the result of noop |
| 18:20.48 | TripleFFFF | s,2,NoOp(${name:4:20}) |
| 18:20.51 | TripleFFFF | as in this |
| 18:22.17 | justinu|laptop | tripleFFFF you can cut out the trolling now :P |
| 18:22.18 | *** join/#asterisk mr_canny (n=mr_canny@200.138.113.82) |
| 18:22.26 | mr_canny | alguem do Brasil ai ? |
| 18:22.49 | TripleFFFF | kek |
| 18:22.56 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:23.42 | justinu|laptop | lol |
| 18:24.25 | *** part/#asterisk hassler (n=hassler@r-corp.hcst.com) |
| 18:26.45 | *** join/#asterisk tomlobato (n=tomlobat@201-43-93-88.dsl.telesp.net.br) |
| 18:29.08 | hmmhesays | wee haw |
| 18:29.27 | Aces1Up | what are your opinions on the best OS to put asterisk on? |
| 18:29.28 | *** join/#asterisk Avalone (n=Avalone_@dial-474.vl-cen-as3.avtlg.ru) |
| 18:29.46 | justinu|laptop | any recent linux distro |
| 18:29.54 | jbroome | jbalcomb: i'm sorry, that isn't an option |
| 18:29.59 | *** join/#asterisk ambriento (n=melcon@200-158-14-51.dsl.telesp.net.br) |
| 18:30.53 | benjk | MS-DOS |
| 18:31.42 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:32.14 | [TK]D-Fender | CP/M |
| 18:32.29 | benjk | that would be somewhat messy though |
| 18:32.41 | justinu|laptop | MPE/iX |
| 18:32.50 | benjk | cause it hasn't go a hierarchical filesystem |
| 18:33.02 | justinu|laptop | Pr1meOS |
| 18:33.46 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 18:36.15 | CunningPike | Aces1Up: Sensible answer alert - your favorite distro, if you have one - otherwise CentOS |
| 18:36.40 | Aces1Up | why centos? |
| 18:36.52 | CunningPike | Aces1Up: Because I like it :) |
| 18:37.09 | CunningPike | Aces1Up: And will be able to help you if you have problems |
| 18:37.18 | justinu|laptop | OS/400 |
| 18:37.33 | Aces1Up | cunning i have not much exp with linux can you suggest a good linux distro suited to newbie?? |
| 18:37.34 | CunningPike | Now, _that's_ trolling ;) |
| 18:37.43 | hmmhesays | I like centos, nice and stable for me |
| 18:37.55 | CunningPike | Aces1Up: CentOS - am I not saying it right? |
| 18:38.06 | Aces1Up | ok okok |
| 18:38.09 | mr_canny | Alguem do Brasil ai ?? |
| 18:38.09 | CunningPike | ;) |
| 18:38.36 | mr_canny | algum brazuca ... ai rssss |
| 18:38.51 | CunningPike | mr_canny: Exactly |
| 18:39.20 | justinu|laptop | RiscOS! |
| 18:39.34 | benjk | OpenBSD |
| 18:39.40 | rob0 | TorsOS |
| 18:39.41 | benjk | definitely some BSD |
| 18:39.54 | rob0 | HellOS |
| 18:39.59 | *** join/#asterisk ambriento (n=melcon@200-158-14-51.dsl.telesp.net.br) |
| 18:40.14 | benjk | isn't that called Windows now? |
| 18:40.42 | justinu|laptop | QNX would make a good softswitch platform.. too bad it's not FOSS yet |
| 18:40.57 | benjk | there's always freeRTOS |
| 18:41.38 | mr_canny | how to configure the asterisk for conect in one host sip on internet ? |
| 18:42.03 | hmmhesays | what a fantastically vague question |
| 18:42.20 | rob0 | Fritos? |
| 18:42.23 | benjk | give him a break, he's having difficulties with asking in English |
| 18:42.51 | benjk | maybe a link to the PDF of that * book might help him |
| 18:43.00 | hmmhesays | maybe |
| 18:43.09 | benjk | or some Brazilian asterisk forum |
| 18:43.23 | justinu|laptop | ~book |
| 18:43.24 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:43.32 | benjk | I think there is one, cant recall now what it was called |
| 18:44.51 | benjk | mr_canny are you ok reading a book in English? |
| 18:45.37 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 18:45.39 | mr_canny | benjk, ok |
| 18:45.58 | benjk | then download that book http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:46.18 | benjk | there will be examples how to configure sip |
| 18:47.38 | hmmhesays | will banks normally take a check that has been taped back together? |
| 18:47.53 | Juggie | umm no |
| 18:47.55 | benjk | cant hurt to ask |
| 18:48.05 | De_Mon | depends on how badly it was torn up |
| 18:48.19 | hmmhesays | it doesn't say "void" on it, i accidentaly ripped it into 3 pieces |
| 18:48.30 | hmmhesays | it got in my junk pile |
| 18:48.31 | De_Mon | I usually write VOID across checks I don't want cashed |
| 18:48.38 | benjk | I think up to 42 pieces is ok |
| 18:48.48 | benjk | if it is more than 42, then not |
| 18:49.00 | De_Mon | 10 peices is the most I would bother trying to put back togeather :P |
| 18:49.17 | hmmhesays | yeah just 3 here |
| 18:49.38 | hmmhesays | and they can call the issuer |
| 18:49.50 | benjk | indeed |
| 18:50.04 | benjk | might take an extra day or two to clear |
| 18:50.29 | justinu|laptop | ime, banks here will use any excuse they can do hold checks, or not cash them |
| 18:50.32 | benjk | you can always blame it on the dog |
| 18:50.48 | hmmhesays | it was just a stupid mistake |
| 18:50.50 | benjk | the dog ate the check |
| 18:51.07 | benjk | or cheque actually |
| 18:51.59 | Juggie | deposit it at a bank machine |
| 18:52.07 | hmmhesays | i'd get freaky with major kira |
| 18:52.14 | justinu|laptop | i bet kira is freaky |
| 18:52.19 | justinu|laptop | often thought the same thing |
| 18:52.26 | hmmhesays | lol |
| 18:52.29 | hmmhesays | ds9 is on right now |
| 18:52.40 | justinu|laptop | i think dax is a stoner |
| 18:52.47 | hmmhesays | which one? |
| 18:52.47 | justinu|laptop | and i'd hit that before kira |
| 18:52.52 | justinu|laptop | dax, the trill |
| 18:52.57 | justinu|laptop | the one with spots on her face |
| 18:53.02 | hmmhesays | jadzia? |
| 18:53.05 | justinu|laptop | yeah |
| 18:53.08 | hmmhesays | or the last one |
| 18:53.10 | hmmhesays | she's a hottie too |
| 18:53.13 | justinu|laptop | der, jedzia |
| 18:53.21 | justinu|laptop | that other girl sucked |
| 18:53.26 | benjk | that's nice |
| 18:53.28 | hmmhesays | yes i'd let her |
| 18:53.31 | justinu|laptop | :) |
| 18:53.45 | hmmhesays | t'pal from enterprise too, oooohh myyyy |
| 18:53.47 | benjk | alhtough I have absolutely no clue what you guys are talking about |
| 18:53.54 | justinu|laptop | my wife watches all this shit |
| 18:54.10 | justinu|laptop | so I just sit around and tell her which girls I'd nail :P |
| 18:54.38 | hmmhesays | t'pal was THE hottest woman on star trek |
| 18:54.44 | De_Mon | /sighs/ startrek geeks |
| 18:54.47 | justinu|laptop | i remember the name, but not the face |
| 18:55.21 | hmmhesays | http://www.vidiot.com/st-ent/images/Enterprise_Blalock_2003-01-800.jpg |
| 18:55.50 | jbroome | sweet jesus, anti-gravity tits |
| 18:56.03 | justinu|laptop | bah, i'd go for 7of9 over that |
| 18:57.00 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 18:57.03 | justinu|laptop | enterprise was when ST started to go more softporn than scifi |
| 18:57.16 | jbroome | i believe the real question is, if you're doing porn on the holodek, is it possible to knock one of them up? |
| 18:58.08 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 18:58.18 | Nivex | I prefer Hoshi. |
| 18:59.20 | Nivex | jbroome: no, and the janitors probably get really cranky about having to clean up the mess when the hologram gets shut off. |
| 19:01.12 | justinu|laptop | generally the hottest chicks are on xena |
| 19:01.28 | jbroome | didn't worf get shot on the holodek, and it caused a real wound? |
| 19:02.00 | justinu|laptop | yeah, that's because the safety system got overridden |
| 19:02.52 | *** join/#asterisk knarfly_wk (n=bwatson@12.42.132.26) |
| 19:03.20 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 19:04.10 | jbroome | ahhh |
| 19:04.38 | justinu|laptop | it's just whatever is convenient for the plot |
| 19:05.07 | justinu|laptop | i stopped watching because it was too retarted |
| 19:05.12 | hmmhesays | speaking of tng |
| 19:05.16 | hmmhesays | its on now |
| 19:09.59 | AsteriskMonkey | anyone know if there is dtmf issues with latest branches? |
| 19:10.12 | AsteriskMonkey | or the new digium hardware cards? |
| 19:11.43 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 19:14.22 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:15.18 | tzanger | I dunno, but I'm kind of disappointed with Sangoma's A104d |
| 19:15.41 | tzanger | screeching echo from hell, which they fixed but 4 weeks later it's just "fuzzy audio occassionally several times during the call" now |
| 19:17.17 | [TK]D-Fender | tzanger: First I've heard of that.... wow |
| 19:17.24 | tzanger | [TK]D-Fender: yeah |
| 19:17.47 | tzanger | I emailed them (had a ticket open from the screeching audio from hell) and they gave me a new beta to try |
| 19:17.51 | tzanger | just waiting for the system |
| 19:18.39 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:18.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:18.46 | [TK]D-Fender | tzanger: Only bug I ran into was a PID going bad in the EC subsystem that caused AUDIO to never leave the pipe. |
| 19:18.50 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 19:18.56 | [TK]D-Fender | tzanger: Took 5 mins to fix and tht was it |
| 19:19.25 | [hC] | i JUST put in an a104d |
| 19:19.28 | [hC] | so far i had to add relaxdtmf in because it was dropping my first digit on ivr's |
| 19:19.41 | Cresl1n | tzanger: did ever get a TE407 to see if the problem went away? |
| 19:19.48 | aptura | is astdb installed in all versions of asterisk |
| 19:20.09 | tzanger | Cresl1n: I'm waiting on the TE407 :-) |
| 19:20.24 | aptura | just was wondering if its included in the install as default. |
| 19:20.38 | tzanger | I had a TE405 that I swapped for a TE406 which was worse than sw echo can, and when the 407 was announced I got my 406 RMA'd for the 407 |
| 19:20.38 | [hC] | tzanger: what hurdles have you come across with the a104d? I swapped from the a102, to try to solve some echo issues. |
| 19:20.41 | tzanger | so I am hopeful |
| 19:21.47 | tzanger | [hC]: it could be my mobo but it's a goodstuff asus server platform... alien voice was the orinal problem (it's like the echo can would go sideways), now the problem is periodic "fuzzy audio" during calls |
| 19:21.47 | AsteriskMonkey | i have a 407 works good |
| 19:21.47 | AsteriskMonkey | just having dtmf issue :P |
| 19:21.53 | AsteriskMonkey | intermittant ones :P |
| 19:22.46 | [hC] | tzafrir: hmm havent run into that yet. fingers crossed. |
| 19:24.00 | *** part/#asterisk Avalone (n=Avalone_@dial-474.vl-cen-as3.avtlg.ru) |
| 19:27.08 | sageang | i posted this earlier but didnt get back a response. when someone leaves a message for an extension, i need asterisk to dial a numeric pager and then send the extension number to the numeric pager |
| 19:28.22 | denon | sageang: show application senddtmf |
| 19:29.13 | sageang | sorry not sure what you meant |
| 19:29.24 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:29.36 | denon | Im saying SendDTMF is what you're looking for |
| 19:29.46 | denon | and if you go to you asterisk console, and type what I said, you'll see some info on it |
| 19:30.24 | sageang | got it. how do you intercept the message signal? |
| 19:30.36 | denon | intercept the message signal/. |
| 19:30.37 | denon | ? |
| 19:31.09 | sageang | sorry. when a message is left on an extension, isnt there messaging from the asterisk server to the phones |
| 19:31.20 | denon | just do it in the dialplan .. |
| 19:31.26 | denon | after the voicemail for that user |
| 19:31.44 | sageang | good one. |
| 19:41.23 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 19:44.29 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:44.43 | *** join/#asterisk sid (n=unstable@tor/regular/sid) |
| 19:44.47 | sid | I'm having a convention at my University(theme is free/libre software + entertainment), and I need to take calls from vendors/speakers/sponsors etc for when they call. Anyone know a good service for a voice mail box, so the vendors can leave a message and I can check the messages and call them back? |
| 19:47.16 | aptura | Why ask for a company to supply vm when you can setup asterisk to do that |
| 19:47.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-30.red.bezeqint.net) |
| 19:47.59 | sid | I was thinking to have the "phone guy" at my school give us a number, and we could put a box in this teachers office, and hook up a phone adapter to it, using asterisk. Then I could have phone calls routed to me on my laptop when I'm on campus... |
| 19:48.16 | sid | But i'm not sure how much work this entails, and if it's easier just to use a vm box |
| 19:49.34 | DarKnesS_WolF | sid: i'm planning to do somehting like this but i don't have the money to get the hardware yet .. i'll have a wifi SIP phone and asterisk at my home with nomral phone line so i have my line with me where ever i want .. " i already hocking my self in my work * box " while i'm at home it's cool |
| 19:50.01 | *** join/#asterisk BooDy (n=boody@213.212.240.16) |
| 19:50.30 | sid | Well that sounds expensive with the wifi sip phone |
| 19:50.37 | *** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 19:50.38 | sid | I didn't want to spend more then $20 USD |
| 19:50.50 | *** join/#asterisk eivindtr (n=wingnut-@062016176152.customer.alfanett.no) |
| 19:50.52 | DarKnesS_WolF | sid: yep teh wifi phone is a fancy stuff |
| 19:51.16 | DarKnesS_WolF | sid: i don't know how much the phone addaptor cost .. it's a analog line or SIP provider ? |
| 19:51.23 | sid | I already have a bunch of junk pIII machines, and I have a laptop. SO I wanted to setup a junk machine in a teacher's office. with GNU/Linux + astersisk on it. THen make it so I could use some software on my laptop and connect and receive calls. |
| 19:51.37 | sid | Like wengophone on my laptop with debian or something, and get calls from the asterisk box(the PIII) |
| 19:51.45 | sid | DarKnesS_WolF: analog |
| 19:51.50 | sid | It's my university phone system |
| 19:52.22 | DarKnesS_WolF | sid: yes u can do that but i'm not sure how much the hardware will cost but more than 20$ as i think |
| 19:52.59 | sid | DarKnesS_WolF: Don't I only need the phone adapter to hook up to the computer? |
| 19:53.02 | sid | Those seem pretty cheap. |
| 19:53.15 | DarKnesS_WolF | sid: for softphones u can use ekiga is Free/Open source software or xlite " free of charge for stander edtion " but not open source or u can use IAX protocol too and IAX softphone |
| 19:53.22 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 19:53.50 | DarKnesS_WolF | sid: i'm not sure about this kind of hardware i use digium TDM400P ... but i can check and see let me see something |
| 19:54.19 | sid | DarKnesS_WolF: How would I have a number hooked up to ekiga, so when vendors/sponsors call me it rings ekiga? |
| 19:55.51 | Beighto | I am having an issue where the dtmf is sometimes duplicating numbers pressed. Currently I have dtmf set as inband, which works significantly better than rfc2833 for the SIP connection I am using. |
| 19:56.03 | DarKnesS_WolF | sid: many ways i can think of now .. ekiga is a SIP softphone .. so in * u setup a SIP extension then when the analog phone got the call u can use * to fwd this call to u .. but i didn't use adaptors at all but i think it can be done .. |
| 19:58.47 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:01.37 | sid | Well the box right now has debian on it, with a ethernet nic for cat5 wire(rj45), and the box also has an rj11 nic in it(phone line wire) for dialup services. Can I use that? Do I even need the adapter? |
| 20:01.42 | *** join/#asterisk toonz (n=dexter@dsl-145-250-19.telkomadsl.co.za) |
| 20:02.38 | toonz | Hi Guys, having a weird problem with bristuffed production version. just installed it on a box. If I got show modules, I see app_dial is loaded. application dial, seems to be missing. |
| 20:02.44 | toonz | sorry to barge in and ask. but i'm stumped. |
| 20:02.47 | toonz | any ideas ? |
| 20:02.52 | DarKnesS_WolF | sid: i think the RJ11 is a dialup modem ? |
| 20:03.50 | sid | DarKnesS_WolF: yes |
| 20:04.32 | sid | DarKnesS_WolF: sorry, I don't know anything about phones or the phone system. Just a noob who wants a single phone line / voice mail / way to receive calls on my laptop while in the LAN |
| 20:04.35 | DarKnesS_WolF | sid: i have no idea if u can use it but i think no |
| 20:06.08 | toonz | :/ |
| 20:06.17 | toonz | No such command 'dial' (type 'help' for help) |
| 20:06.21 | toonz | as in lol ? |
| 20:06.28 | *** part/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 20:08.50 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 20:12.14 | *** join/#asterisk Ebola (n=Ebola@user-5445f8fb.lns4-c13.telh.dsl.pol.co.uk) |
| 20:13.58 | sid | Do I need a tdm400p with one fx/fxs? Or can I just get a voice modem? |
| 20:14.10 | sid | I only want one phone line, I won't be getting more then 20 calls a week. |
| 20:16.12 | toonz | I scheme u gonna need a tdm400 there hey |
| 20:18.47 | toonz | could the dial command screw out because the soundcard in the box is not working or disabled or something ? |
| 20:18.50 | sid | toonz: I am at my university where I want to setup my one single phone line. My University uses analog / POTS |
| 20:19.03 | toonz | or am i pullating at straws here ? |
| 20:19.22 | sid | I just want to put a box in an office, and have one phone number router to that box, so when people call it, asterisk will have voice mail setup, and I can answer it. |
| 20:19.35 | *** join/#asterisk saftsack (n=saftsack@p54A7C48D.dip.t-dialin.net) |
| 20:19.37 | toonz | sid: I'm not shure of using a modem with *, I have only use digium and junghanns cards. even cirrix. but never a modem. I'm shure u can use one tho |
| 20:19.59 | toonz | have u checked google about that ? |
| 20:20.10 | sid | Is there a good hardware list of supported modems for GNU/Linux? |
| 20:20.14 | *** join/#asterisk YaBa_PT (n=Ya_Ba@bl7-183-159.dsl.telepac.pt) |
| 20:20.19 | YaBa_PT | hi everyone |
| 20:21.06 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 20:21.38 | toonz | sid: u can use anything hey. |
| 20:21.44 | toonz | just try not to get a winmodem |
| 20:22.00 | toonz | try go for a Duxbury or 3com / US. Robotics. or something |
| 20:22.16 | toonz | Hi YaBa_PT |
| 20:25.50 | DarKnesS_WolF | sid: http://voip-info.org/wiki/view/FXS-FXO+Converters <-- check out this but i never done it before and i don't know may be it will help u understand |
| 20:27.14 | DarKnesS_WolF | toonz: i don't understand u have exten => balba,1,Dial(SIP/balba) in ur dial plan and when u try to dial balba u got dial command not found ? where do u get dial coammnd not found ? |
| 20:27.45 | Strom_C | sid: best would be to use a TDM01B (single FXO module), but if you're brave, you can use an x100p, which is crappier quality but will probably do the trick inexpensively |
| 20:27.55 | toonz | on asterisk console |
| 20:28.17 | toonz | if I go, "dial 200@main" on console, it tunes me, No such command 'dial' (type 'help' for help) |
| 20:30.20 | DarKnesS_WolF | toonz: ? i don't think this should work ! |
| 20:30.22 | DarKnesS_WolF | let me try |
| 20:30.46 | sid | Strom_C: That tdm01b is $150 USD, I need a phone line for 2 months, one single phone line and that's it. |
| 20:30.58 | sid | Why is it so expensive to hook up a phone line to a computer. |
| 20:31.16 | Strom_C | sid: because voice is difficult to do well |
| 20:31.19 | DarKnesS_WolF | toonz: it should work yes but ur sayng that ur soundcard not counfgured ? |
| 20:31.22 | toonz | sid: u'll be surprised how much more expensive traditional pabx's are... |
| 20:31.42 | toonz | DarKnesS_WolF: yeah, the boxes sound card is disabled for some reason.... |
| 20:32.05 | DarKnesS_WolF | toonz: hmmmm let me try to make mine busy and see what will happen |
| 20:32.13 | toonz | can't pick it up on lspci. i'm just bummed cause the box is sitting 80km's from my office. don't wanna drive there. |
| 20:32.26 | toonz | nah its kewl DarKnesS_WolF, don't stress. I think its this sound card hey |
| 20:32.58 | toonz | I have another identical box, same instalation and thats works. cept its sound card is enabled. |
| 20:33.01 | toonz | oh well. |
| 20:33.17 | DarKnesS_WolF | toonz: yes seems to be u need voice |
| 20:33.18 | DarKnesS_WolF | see here |
| 20:33.29 | DarKnesS_WolF | i just installed fresh copy of 1.4 and check what happen when i did dial |
| 20:33.38 | sid | Is there no easy way to have one single phone line(one wire) hook up to a box, so I can get phone calls on it? |
| 20:33.50 | DarKnesS_WolF | <PROTECTED> |
| 20:33.51 | DarKnesS_WolF | <PROTECTED> |
| 20:33.51 | DarKnesS_WolF | <PROTECTED> |
| 20:33.51 | DarKnesS_WolF | <PROTECTED> |
| 20:34.01 | DarKnesS_WolF | toonz: so i think it needs the sound ? what do u think ? |
| 20:34.12 | DarKnesS_WolF | it's addressing the alsa hardware here |
| 20:34.13 | toonz | yeah, i scheme so hey. def needs sound card. |
| 20:34.14 | sid | I have to buy these expensive cards and adapters? I can't get a voice modem for $5 dollars on newegg? |
| 20:34.32 | toonz | DarKnesS_WolF: the weird thing is, the experimental version works fine without it. |
| 20:34.46 | Strom_C | sid: like I said, you can try an x100p for $20, but it's going to be more of a challenge to get sounding as good as a tdm400 |
| 20:34.47 | DarKnesS_WolF | toonz: ur using 1.4 too ? |
| 20:35.02 | toonz | nah, I'm using BRIstuffed. from junghanns.net |
| 20:35.09 | sid | Strom_C: Will it sound like shit? Can I hold up a conversation with someone? |
| 20:35.23 | Strom_C | sid: it depends on your telephone line |
| 20:36.12 | *** join/#asterisk celophane (n=e@ip68-104-251-230.ph.ph.cox.net) |
| 20:36.30 | sid | Strom_C: It's a regular analog / POTS system. |
| 20:36.42 | Strom_C | sid: thats not what I meant |
| 20:36.46 | Strom_C | not all POTS lines are the same |
| 20:37.00 | Strom_C | some are harder to compensate for than others |
| 20:39.17 | sid | Strom_C: How can I tell if it's hard to compensate? |
| 20:39.31 | Strom_C | sid: do you know the impedance? loop length? |
| 20:40.04 | sid | Strom_C: no |
| 20:40.37 | Strom_C | well, those would be good things to find out |
| 20:40.53 | *** join/#asterisk arguile (i=user224@74.14.224.125) |
| 20:41.22 | sid | Strom_C: Someone here is saying that none of that is going to be a problem. impedance/look length etc |
| 20:41.35 | saftsack | which softswitch has the most advanced GUI (webinterface) for doing some settings? |
| 20:41.44 | Strom_C | sid: who is that? the local telco person? |
| 20:43.40 | sid | http://www.newegg.com/Product/Product.asp?Item=N82E16825180008 |
| 20:43.43 | sid | Will that work? |
| 20:43.48 | *** join/#asterisk alexis101 (n=as@MTRLPQ02-1177996380.sdsl.bell.ca) |
| 20:43.59 | sid | Strom_C: no, some technie for the school that does system administration. |
| 20:44.41 | Strom_C | sid: i can almost guarantee you he doesn't have enough experience with POTS to know what he's talking about |
| 20:45.03 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 20:46.02 | celophane | Would like a recommendation on a good, solid motherboard... Have: P4 Core 2 Duo CPU, Digium TE110P, and two 240 Pin PC2-5300 Unbuffered x64 Non-ECC memory modules (512MB ea). The Intel board and the Asus board we tried aren't working properly...and the only advice we've been given is just "try another board". This will be our 3rd board...I'd really like to get it right this time. Any ideas? Anyone using a close config that can recommend |
| 20:46.04 | sid | Strom_C: It seems easy enough to do. I have a phone wire, I can plug a phone into it and it works. If someone calls it it rings. I just want to hook up a GNU/Linux box to it, and have voice mail setup, and be able to ssh -X so I can use ekiga/gnome for when someone calls. |
| 20:46.05 | alexis101 | I got some problem with my music on hold ... the volume is a way to loud ... |
| 20:46.16 | alexis101 | is there a way to set it |
| 20:47.10 | Strom_C | sid: regular analog phones do equalization in the set. Because asterisk introduces a time-division switching matrix into the call path, equalization is much harder to do. |
| 20:47.34 | Strom_C | sid: like I said, get an x100p for $20 |
| 20:47.55 | Strom_C | if that's too much for you, go to the thrift store and get a used answering machine for $3 |
| 20:48.03 | *** join/#asterisk vhatz (n=Vlasis@ppp146-5.adsl.forthnet.gr) |
| 20:48.22 | sid | Strom_C: The cheapest I can find is for $30 USD, the ENCORE ENF656-ESW-AGPR 56Kbps Modem is only $5 |
| 20:48.31 | Strom_C | sid: that wont work |
| 20:48.34 | sid | Strom_C: But I want to be able to be anywhere on campus, and receive calls. |
| 20:48.44 | Strom_C | there is no driver for that to use with asterisk |
| 20:48.55 | Strom_C | sid: have you checked ebay? |
| 20:49.03 | sid | Rosewill RC-403 56Kbps PCI Bus (Plug and Play) Conexant Modem ? |
| 20:49.05 | sid | Will that work? |
| 20:49.30 | Strom_C | NO |
| 20:49.35 | Strom_C | NO modem will work |
| 20:49.43 | Strom_C | there are no drivers. |
| 20:49.57 | *** join/#asterisk Roach4 (n=Roach4@65.93.166.211) |
| 20:49.59 | alexis101 | anyone know how i could resolve my litlle problem i really need to fix it ... |
| 20:53.35 | Strom_C | sid: and I'm still trying to figure out how you're going to tunnel calls over SSH |
| 20:56.03 | celophane | Any recommendations on a motherboard? |
| 20:56.12 | Seba_soy | ASUS? |
| 20:56.37 | celophane | Seba: which one? or which chipset? |
| 20:56.46 | Seba_soy | wich CPU? |
| 20:56.59 | celophane | P4 Core 2 Duo |
| 20:57.11 | Seba_soy | P4, so... INTEL MOTHER |
| 20:57.26 | Seba_soy | I thought AMD |
| 20:59.13 | Assid | damn |
| 20:59.18 | Assid | i need to learn about securid |
| 20:59.20 | celophane | I did the Intel board and an MSI board and we get nmi errors, it was not sharing an irq, they made changes and then it changed from nmi errors to loc errors... patlooptest was failing, the card is fine but a te110p or t100p will not work in there. |
| 20:59.49 | Seba_soy | wich LINUX? |
| 21:00.25 | celophane | Fedora Core 4 |
| 21:00.36 | Seba_soy | try a INTEL BOARD wich no issues |
| 21:00.41 | Seba_soy | call DIGIUM for that |
| 21:00.53 | Seba_soy | they know chipsets with problems |
| 21:01.16 | celophane | I used the Intel DG965WHMKR board... |
| 21:01.25 | Seba_soy | Recently I had same problem and it was solved buying another INTEL board with another chipset |
| 21:01.31 | Seba_soy | just we ask to digium |
| 21:01.32 | celophane | I just e-mailed them.....waiting for response....thought I'd check here too. |
| 21:01.57 | Seba_soy | wait for response, I am sure it is a chipset problem |
| 21:02.08 | wunderkin | they would not make any suggestions other than trying another motherboard, i think the guy was new because he did not know what 2,3,4 for timing would do |
| 21:02.22 | wunderkin | we just need to try something other than 965 or 975 chipset |
| 21:02.38 | celophane | My local supplier only has the 975 chipset and the 946 chipset....not that I'm not willing to order something else....just saying, that's all they have. |
| 21:02.46 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 21:02.54 | Seba_soy | instead look on asus mainsite |
| 21:03.03 | Seba_soy | maybe they have some mothers with other chipsets |
| 21:03.22 | Seba_soy | you can found chipset problems on digium pages or googling I think |
| 21:03.49 | celophane | Checked that....didn't come up with any significant results. |
| 21:04.40 | celophane | Found people saying "having problem with this MB....having problems with that MB", but nowhere did I see "this MB works great...you should use it" (yes, I live behind rose-coloured glasses). |
| 21:04.48 | kink0 | <kinko> a question about OOH323, normally I got a channel named OOH323/10.10.10.10 where 10.10.10.10 is the ip address of the peer, but sometimes I got something like peer h323 id instead IP , i.e. OOH323/Gate1 |
| 21:04.48 | kink0 | <kinko> anyway to avoid this ? |
| 21:05.14 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-7d5e401432ec0f72) |
| 21:05.14 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 21:07.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:07.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:07.41 | syzygyBSD | kink0: are those users setup in your h323.conf file |
| 21:08.01 | kink0 | syzygyBSD, no. |
| 21:08.18 | syzygyBSD | hmm.... |
| 21:08.27 | *** join/#asterisk adamowitz (n=adamowit@ip68-9-201-27.ri.ri.cox.net) |
| 21:08.29 | syzygyBSD | reverse DNS maybe then... |
| 21:08.30 | kink0 | syzygyBSD, just ussing all the same trunk, as before users was ignored in my h323.conf |
| 21:08.56 | kink0 | syzygyBSD, hmmmm I don't think reverse DNS; because I get something like Gate1 and so |
| 21:08.57 | adamowitz | Does anyone here use FreeWorldDialup? |
| 21:09.24 | kink0 | no domain names, and I got the IP and did a host, and that was not the reverse, I think I am getting some h323 id |
| 21:10.02 | syzygyBSD | I dont' know exactly how that is setup, but if a connection is registered it will use the username instead of the IP |
| 21:10.47 | syzygyBSD | I have never used h323, but from the console something like 'h323 show peers' should tell you what will show up as what |
| 21:10.50 | kink0 | syzygyBSD, yes, but with h323 that comes included in Asterisk that doesn't work |
| 21:11.17 | kink0 | none peers :) none users !! just all are forwareded to the same context |
| 21:14.01 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-ef8a5560556cf03e) |
| 21:14.08 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 21:14.24 | vhatz | Now that we're talking about h323 was anyone able to compile chan_h323 frm snv or 1.4beta2??? |
| 21:16.51 | kink0 | vhatz, hmmm no, I am ussing ooh323 now, to avoid the terrible deadlock crashes |
| 21:16.52 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 21:16.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:18.33 | *** join/#asterisk daysmen3 (n=primus@host81-153-174-207.range81-153.btcentralplus.com) |
| 21:20.23 | *** join/#asterisk Orangeville (n=Bear@GLPHON22-1168101592.sdsl.bell.ca) |
| 21:20.41 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 21:20.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:22.10 | vhatz | kink0: I thought that chan_h323 was stalled or something, but I saw that new code was added in 1.4 )this is when I started noticing it at least) |
| 21:22.23 | vhatz | kink0: and as I see in the code it works with realtime database so I thought I should give it a try... |
| 21:22.44 | kink0 | vhatz, is , I saw from 1.4, anywise I am very cautelously, and still ussing 1.2.x , then I choose to use ooh323 |
| 21:23.02 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 21:23.12 | kink0 | vhatz, about databases... may be, we use our own interfaces, because we are ussing also for GSM |
| 21:23.27 | kink0 | where we need to get accounting based on SIM cards id. |
| 21:23.54 | *** join/#asterisk NormanASD (n=franke@206.135.58.98) |
| 21:24.28 | *** part/#asterisk huslage (n=huslage@hact.net) |
| 21:24.46 | *** join/#asterisk norm- (n=qn@81.5.175.218) |
| 21:24.53 | NormanASD | Hi, I have an odd question if anyone can help... |
| 21:24.59 | norm- | does Asterisk have any support for modem over ip yet? |
| 21:25.19 | NormanASD | I have a Digium T100P and am running Asterisk 1.2.12.1 |
| 21:25.22 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 21:25.56 | NormanASD | I get random hangups when connecting from the POTS world via the T100 (connected to a T1 to our PBX) and a viop phone. The voip phones work fine between each other. |
| 21:27.36 | NormanASD | I searched Digium's web page, voip-info.org and google to no avail. |
| 21:29.15 | kink0 | NormanASD, I am not familiarized and not experience with T1 signalling, just E1 .. but try this |
| 21:29.37 | kink0 | set your PCI bus clocking to 66Mhz , no Auto for the digium TE10x card slot. |
| 21:29.56 | NormanASD | BTW, it often happens if I make a loud noise, e.g. bang the phone or switch from speaker to handset on a phone. |
| 21:30.34 | kink0 | the problem with some mb cards, if you set to auto, is the bus is auto-setting to 133Mhz |
| 21:30.54 | kink0 | that will *NOT* cause red or yellow alarm on the span... but a lot of inestabilities. |
| 21:31.22 | justinu|laptop | that worked for you, kink0/ |
| 21:31.26 | NormanASD | can i tell what speed it selected via some /proc file? |
| 21:31.32 | justinu|laptop | first time I've ever heard anyone with a solution to that |
| 21:33.33 | *** join/#asterisk TwentyOneEuro (n=pirch@ipd120.51.tellas.gr) |
| 21:33.33 | justinu|laptop | NormanASD: are you using busydetect or callprogress in your zapata.conf? |
| 21:33.50 | TwentyOneEuro | Hi to evryone |
| 21:34.02 | NormanASD | busydetect=no and callprogress=no and transfer=no |
| 21:34.24 | justinu|laptop | interesting... the fact that you can repeat it by banging hte phone indicates a gain issue, perhaps |
| 21:34.42 | justinu|laptop | since you're not running any inband detection |
| 21:34.43 | TwentyOneEuro | I hcould someone give me a little help? |
| 21:35.24 | NormanASD | I have gain set to 0.0 |
| 21:36.05 | NormanASD | and it doesn't always happen, sometimes banging it doesn't work. however, switching from handset to speaker almost always causes it to hangup, but not the other way around. |
| 21:36.13 | justinu|laptop | that is funky |
| 21:36.29 | justinu|laptop | pots phone? |
| 21:36.32 | justinu|laptop | or voip phone? |
| 21:36.48 | NormanASD | PBX phone to VoIP phone. (I can't easily test PBX to PBX yet.) |
| 21:37.15 | NormanASD | voip phone to voip phone works just fine, as does analog phone via a Digium FXS |
| 21:37.23 | TwentyOneEuro | I have a network with 20-100 users and want my users to use voip to communicate with each other. At the moment I do not want to connect to another network such as PSTN. The question is do I need asterisk for something like this? |
| 21:37.52 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-c0551a9a5ecd9c66) |
| 21:39.40 | justinu|laptop | NormanASD: can you tell if the PSTN is releasing the call, or if asterisk is? (on PRI side) |
| 21:39.55 | TwentyOneEuro | actually it is a WiFi network and I am thinking of WiFi VoIP phones for all our users |
| 21:39.56 | NormanASD | kink0, lspci -vv tells me the card is running at 66 Mhz. |
| 21:40.25 | NormanASD | I don't have a PRI, just T1, sadly. I'm using E&M signaling. |
| 21:40.37 | justinu|laptop | ok, still you should be able to tell which side releases |
| 21:40.50 | NormanASD | Based on timing, it seems Asterisk gets the hangup before either my PBX phone or the voip phone detects a hangup. |
| 21:41.00 | kink0 | NormanASD, I agree with justinu|laptop , that was my experience with Digium cards |
| 21:41.06 | NormanASD | Actually, the PBX seems to delay hangup detection, so I think it's coming from the T100 |
| 21:41.20 | kink0 | anywise if you has your digium running at 66Mhz there not much problem. |
| 21:41.33 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 21:42.01 | justinu|laptop | in my experience E&M support on asterisk isn't all that great |
| 21:42.02 | kink0 | for me, digium TE405 I am ussing are a bit problematics, because sometimes without any alarm, if I restart, doesnt syncing well |
| 21:42.09 | justinu|laptop | ppl say it works tho |
| 21:43.02 | justinu|laptop | you can use zttool to look at the signalling bits on your T1 |
| 21:43.11 | justinu|laptop | maybe that might give you some insight into whats happening |
| 21:43.57 | TwentyOneEuro | Could someone help me plz |
| 21:44.16 | NormanASD | Sadly, we can only do E&M, as that's all our PBX can generate. I may be able to get a PRI card, but we'll be going to an all T1 solution anyway once things are working (and we remove the PBX.) |
| 21:44.40 | *** join/#asterisk VitoMG (n=Wael@85.112.64.168) |
| 21:44.57 | NormanASD | Asterisk tells me "Hangup 'Zap/1-1'" but the PBX phone seems to take a little while to determine the hangup and it plays the fast busy. |
| 21:45.53 | NormanASD | kink0, for now, each restart seems to sync up just fine. however, I only have like 12 feet of cable between the PBX T1 card and my T100 |
| 21:46.05 | justinu|laptop | yeah - i'm just wondering if we can determine which side hung up first |
| 21:46.09 | justinu|laptop | pbx or zaptel |
| 21:46.13 | justinu|laptop | that message is a bit vague |
| 21:46.41 | NormanASD | I can run zttool, but I'm not really sure what it all means... |
| 21:46.58 | kink0 | NormanASD have you tried to invert TE/TM in both extrems of the cable interfaces ? |
| 21:47.00 | justinu|laptop | on e&m you have signalling bits that indicate the hook state of each side |
| 21:47.20 | justinu|laptop | 0 means on hook, 1 means off hook |
| 21:47.39 | justinu|laptop | typically, all bits change at the same time. AB for D4/SF, and ABCD for ESF |
| 21:47.56 | NormanASD | right now, it's 1111 1100 |
| 21:48.25 | justinu|laptop | hmm, that's odd |
| 21:48.28 | NormanASD | that was weird, it just hung up while connected and very low ambient noise. (Then all the bits went to zero.) |
| 21:48.36 | NormanASD | That is for channel 1, reading down. |
| 21:48.37 | justinu|laptop | 1100 sounds like a different setup than E&M |
| 21:48.53 | NormanASD | RxC and RxD are 0, is that normal? |
| 21:49.03 | justinu|laptop | you are running ESF right? |
| 21:49.27 | NormanASD | esf? |
| 21:49.42 | NormanASD | ah, no ami |
| 21:49.46 | justinu|laptop | t1 or e1 |
| 21:49.50 | NormanASD | t1 |
| 21:49.53 | *** join/#asterisk ajohnson_laptop (n=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 21:49.54 | justinu|laptop | ami/d4 then? |
| 21:49.56 | NormanASD | d4 |
| 21:49.57 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:50.03 | justinu|laptop | ok, then there are no C and D bits |
| 21:50.06 | justinu|laptop | so it's ok that they're 0 |
| 21:50.13 | NormanASD | good to know... |
| 21:50.42 | NormanASD | maybe a clocking issue? it says "internally clocked" |
| 21:50.51 | justinu|laptop | it always says that :( |
| 21:50.57 | justinu|laptop | another bug that apparently won't get fixed |
| 21:51.13 | NormanASD | I'm using: span=1,1,0,d4,ami |
| 21:51.13 | wunderkin | aha |
| 21:52.13 | ajohnson_laptop | I am getting ready to post a bug report for what appears to be a random crash in 1.2.12 and I just want to make sure I do it right |
| 21:52.13 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:52.21 | ajohnson_laptop | I've read the guidelines as well |
| 21:52.25 | NormanASD | now it went from hangup to "unknown extension 's'" |
| 21:52.52 | ajohnson_laptop | I have done a bt full with a version of Asterisk that was compiled using "make dont-optimize" |
| 21:52.53 | NormanASD | it then plays the ss-noservice to the PBX |
| 21:53.03 | gooagle | 0 |
| 21:53.08 | NormanASD | I also get lots of "chan_zap.c:5395 ss_thread: getdtmf on channel 1: Operation now in progress" is that normal? |
| 21:53.39 | justinu|laptop | no |
| 21:53.40 | TwentyOneEuro | DO I NEED Asterisk for a local VoIP network to work? |
| 21:53.51 | justinu|laptop | norman: are you trying to run wink-start? |
| 21:54.18 | ajohnson_laptop | Can anyone take a look at a backtrace and tell me if anything stands out? |
| 21:54.22 | gooagle | has anyone gotten blf working with the thomson st2030? |
| 21:54.31 | justinu|laptop | ajohnson_laptop: perhaps try #asterisk-dev? |
| 21:54.32 | NormanASD | i am using wink start, yes. |
| 21:54.46 | justinu|laptop | NormanASD: is it possible to try immediate start on both sides, just for fun? |
| 21:55.02 | justinu|laptop | i have had bad experiences with wink start e&m on asterisk |
| 21:55.09 | NormanASD | so change signalling=em in zapata.conf? |
| 21:55.11 | ajohnson_laptop | was told to try here first |
| 21:55.16 | justinu|laptop | lol |
| 21:55.27 | justinu|laptop | they expect non devs to look at a backtrace? |
| 21:55.52 | TwentyOneEuro | Thank you for your time |
| 21:55.54 | *** part/#asterisk TwentyOneEuro (n=pirch@ipd120.51.tellas.gr) |
| 21:55.59 | ajohnson_laptop | dunno, just relaying what Kevin told me :) |
| 21:56.02 | justinu|laptop | NormanASD: yeah, don't forgot to change it on your pbx too |
| 21:56.04 | ajohnson_laptop | I'll head over to -dev then |
| 21:56.08 | wunderkin | heh yes thats what i was sayin |
| 21:56.09 | justinu|laptop | forget |
| 21:56.20 | NormanASD | i'm not sure our old pbx can do non-wink... |
| 21:56.35 | *** part/#asterisk ajohnson_laptop (n=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 21:56.36 | *** join/#asterisk PeeVee (n=nomail@sd511723c.adsl.wanadoo.nl) |
| 21:57.09 | NormanASD | I can ask our tech guy tomorrow. |
| 21:57.42 | NormanASD | while online here, i've tried connecting and just leaving them go. it always hangs up after a few min, sometimes less. |
| 21:58.00 | justinu|laptop | yeah... some timer expiring |
| 21:58.09 | justinu|laptop | pbx is waiting for wink, and never gets it? |
| 21:58.14 | justinu|laptop | so it abandons the call |
| 21:58.28 | justinu|laptop | or asterisk is expecting wink from pbx, never detects it |
| 21:58.30 | justinu|laptop | so abandons |
| 21:58.50 | AsteriskMonkey | when you do a restart of asterisk it will take new rx/tx correct? or does zaptel need restarting also? |
| 21:59.03 | AsteriskMonkey | just asterisk right? |
| 21:59.08 | justinu|laptop | i think a restart of asterisk should cover it |
| 21:59.19 | AsteriskMonkey | thanks |
| 21:59.22 | VitoMG | hi im trying to use SPA3102 PSTN line as a trunk. can anyone help in configuration plz? |
| 21:59.49 | justinu|laptop | NormanASD: i've had similar problems with e&m winkstart, so I am not surprised to hear you are having trouble |
| 21:59.58 | NormanASD | it gets the wink, otherwise it would never accept the call. setting the wink to low makes this happen. |
| 22:00.09 | NormanASD | I tried running em (not em_w) and it couldn't receive calls. |
| 22:00.13 | *** join/#asterisk ajohnson_laptop (n=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 22:00.38 | justinu|laptop | that's because the pbx is expecting a wink from asterisk |
| 22:01.55 | NormanASD | I know the wink to make an outgoing call works, because if I set rxwink too low, it will never get dialtone. setting it to 300 worked, and I can make outgoing calls. However, those will randomly hangup as well, e.g. calling my cell phone from a voip phone. |
| 22:02.04 | *** join/#asterisk K-eSaR (n=ircap@unaffiliated/k-esar) |
| 22:02.14 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:02.30 | justinu|laptop | hmm |
| 22:02.46 | NormanASD | i'll ask the tech guy about e&m no wink. i'd be disappointed to hear that this digium card won't work for us, though. i wonder if a sangoma one will work better? |
| 22:02.59 | justinu|laptop | well, it certainly seems like a signalling issue... not sure I can help much more |
| 22:03.06 | NormanASD | maybe I need to play with the wink times? |
| 22:03.12 | justinu|laptop | no, sangoma won't improve anything, as signalling is handling by asterisk |
| 22:03.26 | NormanASD | ah, good to know. |
| 22:03.53 | NormanASD | perhaps our pbx tech can monitor the system to see what's going on, i guess... |
| 22:04.15 | justinu|laptop | yeah, if you have a T1 test set, or a good debug display in the PBX you should get some clues |
| 22:05.39 | norm- | does Asterisk have any support for modem over ip? |
| 22:06.13 | justinu|laptop | i don't think so |
| 22:06.35 | NormanASD | FYI, I did notice that if I enable "usecallerid=yes" it always transfers to s,1 and kills the call. That's odd. |
| 22:06.52 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 22:07.07 | justinu|laptop | what kind of outpulsing is your PBX doing when it calls * |
| 22:07.20 | norm- | can anybody think of any similar ways i could bridge a modems signal across the net to where the POTS originates? |
| 22:07.23 | NormanASD | what do you mean? |
| 22:07.42 | justinu|laptop | meaning: how does it signal asterisk the destination number/calling number? |
| 22:07.45 | justinu|laptop | dtmf? |
| 22:08.13 | justinu|laptop | *CLID*DNIS* or something |
| 22:08.13 | NormanASD | ah, I believe dtmf |
| 22:08.20 | justinu|laptop | because you have messages in your log indicating some trouble with dtmf decoding |
| 22:08.27 | justinu|laptop | and you dont always reach the correct exten, right? |
| 22:08.34 | NormanASD | indeed, after the disconnect. |
| 22:08.43 | NormanASD | no, i always get the correct exten |
| 22:08.51 | justinu|laptop | oh, you said sometimes it drops to s,1 |
| 22:09.07 | NormanASD | here is a bit more, just before the hangup, I get this: |
| 22:09.08 | NormanASD | Spawn extension (internal, 1272, 1) exited non-zero on 'Zap/1-1' |
| 22:09.41 | NormanASD | then the hangup zap/1-1, then "starting simple switch" then the ss_thread dtmf error |
| 22:09.55 | NormanASD | C&P: == Spawn extension (internal, 1272, 1) exited non-zero on 'Zap/1-1' |
| 22:09.55 | NormanASD | <PROTECTED> |
| 22:09.56 | NormanASD | <PROTECTED> |
| 22:09.56 | NormanASD | Sep 27 18:07:45 WARNING[5997]: chan_zap.c:5395 ss_thread: getdtmf on channel 1: Operation now in progress |
| 22:09.56 | NormanASD | <PROTECTED> |
| 22:10.04 | justinu|laptop | ok, that tells me that PBX went on hook for a very brief moment (like a wink) |
| 22:10.14 | justinu|laptop | asterisk thought it was receiving a new call |
| 22:10.23 | NormanASD | ah, that's interesting. thanks. |
| 22:10.40 | linagee | whoa! i found a voip provider with ~30ms! nice. |
| 22:10.43 | justinu|laptop | actually, my bad... wink would be going off hook for a very brief moment |
| 22:10.51 | linagee | stupid east coast voip providers. :( (too many of them) |
| 22:10.56 | justinu|laptop | linagee: we get 5ms to broadvoice from our miami site |
| 22:10.58 | linagee | (USA east coast! :P) |
| 22:11.12 | linagee | justinu|laptop: so you are east coast to east coast? |
| 22:11.24 | linagee | justinu|laptop: what sort of connection do you have to the net? |
| 22:11.37 | justinu|laptop | that box is colo'ed at valueweb |
| 22:11.42 | NormanASD | so, the pbx is dropping the link for a short period of time, which signals to asterisk something? |
| 22:11.43 | justinu|laptop | at home I have shitty dsl like everyone else |
| 22:11.56 | linagee | justinu|laptop: i'm talking about from my cable modem |
| 22:12.03 | NormanASD | I just got FIOS at home, which I want to test out VoIP to the office if I can get this working... |
| 22:12.04 | justinu|laptop | NormanASD: sounds like pbx is droping it's AB bits for a brief second, so asterisk sees a new call |
| 22:12.05 | linagee | justinu|laptop: lag under 50ms is good |
| 22:12.12 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 22:12.13 | justinu|laptop | linagee: agreed, nice |
| 22:12.21 | linagee | justinu|laptop: for ping packets that is. yes, i realize continual packets are different... |
| 22:12.24 | *** part/#asterisk vhatz (n=Vlasis@ppp146-5.adsl.forthnet.gr) |
| 22:12.51 | NormanASD | can i get asterisk to ignore that dropping of the AB? |
| 22:13.03 | justinu|laptop | NormanASD: no, because that is how hook state is communicated to each side |
| 22:13.17 | NormanASD | maybe increase wink times? |
| 22:13.27 | linagee | justinu|laptop: wow. and it's even $4/mo. nice. |
| 22:13.36 | NormanASD | it should never signal another call is coming in, since I'm the only user. |
| 22:13.47 | NormanASD | if a new call comes in, it should use another T1 channel. |
| 22:13.52 | justinu|laptop | i'm not sure this is a wink problem |
| 22:14.16 | linagee | justinu|laptop: and they have paypal payments. this seems almsot too good. lol |
| 22:14.16 | NormanASD | sadly, my telecom knowledge isn't all that great for T1s... |
| 22:14.44 | justinu|laptop | E&M is very simple |
| 22:14.52 | justinu|laptop | maybe I can find a link for you |
| 22:15.13 | NormanASD | I want to move to PRI, but our phone company charges like 2x for a PRI vs a T1 |
| 22:15.34 | justinu|laptop | ok, if I ever badmouthed cisco, I apologize... their website rocks |
| 22:15.35 | justinu|laptop | http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml#ems |
| 22:15.49 | NormanASD | cool, thanks. |
| 22:19.58 | NormanASD | maybe I'll play around with ztmonitor a bit... |
| 22:20.15 | linagee | justinu|laptop: nm. they are using sipphone i guess. (that's what i was pinging.) i think sipphone.com doesn't support iax |
| 22:20.21 | linagee | sucks! :( |
| 22:20.29 | justinu|laptop | use sip :) it works |
| 22:20.35 | linagee | justinu|laptop: no, it doesn't. |
| 22:20.41 | justinu|laptop | why not? |
| 22:20.42 | linagee | justinu|laptop: not behind a nat |
| 22:20.48 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 22:20.49 | linagee | not with voodoo or something |
| 22:20.55 | justinu|laptop | sip works behind nat for me |
| 22:21.02 | justinu|laptop | doublenat can sometimes fuck stuff up |
| 22:21.15 | linagee | justinu|laptop: are you using dyndns? |
| 22:21.31 | linagee | justinu|laptop: or do you have a static ip |
| 22:21.37 | justinu|laptop | i've used both |
| 22:21.46 | adamowitz | what sort of problems? |
| 22:21.53 | justinu|laptop | you have asterisk behind nat, and you want to register to a SIP ITSP? |
| 22:22.02 | linagee | justinu|laptop: sip behind nat with a dynamic IP sort of sucks. (unless i can get it using my dyndns hostname) |
| 22:22.10 | justinu|laptop | it works, make sure you have externip= and localnet= set correctly in your sip.conf |
| 22:22.23 | linagee | justinu|laptop: externip? what do i set that to? :P |
| 22:22.34 | linagee | justinu|laptop: my dynamic IP? :P |
| 22:22.39 | linagee | :( |
| 22:22.39 | justinu|laptop | oh, ic. that's because asterisk can't properly resolve the DNS name you give it |
| 22:22.44 | justinu|laptop | or STUN for it's extern ip |
| 22:22.46 | justinu|laptop | :( |
| 22:22.50 | justinu|laptop | you're right. |
| 22:22.51 | linagee | justinu|laptop: to tell the truth i haven't used it much |
| 22:23.04 | NormanASD | justinu|laptop, thanks for the help, off to dinner. |
| 22:23.10 | linagee | (i heard you can use externhost now or something, haven't tried it yet) |
| 22:23.20 | justinu|laptop | anyways, there was some talk of being able to use a DNS name for externip, ... |
| 22:23.23 | justinu|laptop | yeah, that was it |
| 22:23.31 | justinu|laptop | NormanASD: good luck |
| 22:23.44 | NormanASD | thanks! |
| 22:23.48 | *** part/#asterisk NormanASD (n=franke@206.135.58.98) |
| 22:23.57 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:24.35 | adamowitz | anyone here use FWD? |
| 22:28.29 | *** join/#asterisk shodan- (n=shodan@ip185.99-113-216.pppoe4.joliette.intermonde.net) |
| 22:33.34 | *** join/#asterisk oQPa (i=name@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 22:38.06 | adamowitz | Is there some other service like FWD, but more popular? |
| 22:46.07 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 22:46.28 | *** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net) |
| 22:47.41 | *** join/#asterisk Magicianx (n=patateki@116-22.dr.cgocable.ca) |
| 22:55.49 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 22:58.30 | *** join/#asterisk kieranmullen (n=kieranmu@static-71-245-97-83.ptldor.fios.verizon.net) |
| 22:58.34 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:58.46 | TripleFFFF | is this still a valid option i n sip.conf 1.2.12.1 rtautoreg=yes |
| 22:59.18 | kieranmullen | has anyone heard of vitelity.com ? I am looking at getting some dids |
| 23:02.45 | linagee | why is sip such a PITA? :( |
| 23:06.30 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:15.27 | sid | 16:53 < Strom_C> sid: and I'm still trying to figure out how you're going to tunnel calls over SSH |
| 23:15.53 | sid | Strom_C: I'm going to use ssh X fowarding, ssh -X, and use gnome/ekiga on the box that asterisk is on. |
| 23:16.59 | *** join/#asterisk kph100 (n=kph100@206-248-159-225.dsl.teksavvy.com) |
| 23:18.27 | JT | that's a bad idea |
| 23:18.28 | JT | sid |
| 23:19.16 | jbroome | try openvpn |
| 23:21.46 | TripleFFFF | hey |
| 23:22.05 | TripleFFFF | when u adjut music on hold its not actualy changing the -a tenuation even on a reload re_s_musiconhold |
| 23:22.14 | TripleFFFF | anyway to force it without a restart ? |
| 23:25.01 | JT | sid: why ask for assistance if you are always going to go AFK? |
| 23:25.09 | JT | makes it hard for people to assist |
| 23:25.17 | TripleFFFF | ? |
| 23:25.20 | TripleFFFF | ay -Q -o raw:- --mono -R 8000 -a -5 fpm-suns |
| 23:25.27 | TripleFFFF | this shows -5 |
| 23:25.50 | TripleFFFF | 2 mintues ago i restarted * .. i saw 3 process on madplay one -5 one -10 one -15 lol |
| 23:26.10 | JT | TripleFFFF: why the ? |
| 23:27.27 | TripleFFFF | welll.. its shing |
| 23:27.34 | JT | shing? |
| 23:27.38 | JT | wtf are you talking about? |
| 23:27.59 | TripleFFFF | shiting |
| 23:28.05 | TripleFFFF | its using soem random values |
| 23:28.11 | TripleFFFF | now idea where from |
| 23:29.14 | TripleFFFF | guess i need a moh server for my farm |
| 23:29.35 | TripleFFFF | nothign in * seem to be reliable enough ..hmmm ok thanks nayhow |
| 23:29.40 | JT | OK |
| 23:29.49 | JT | well i have no idea what was with the question mark |
| 23:29.56 | JT | as i was not talking to you |
| 23:31.47 | TripleFFFF | dont get me wrong asterisk as gone a long way |
| 23:33.21 | TripleFFFF | but if i was to chose between 10 stable business ed of Asterisk @ 10,000$ (1 k each)+ a router (2000$) + SER +mysql (another $10,000 ) .. for 25k .. i prefer going with a cisco as5400 and get rid of all the crap |
| 23:33.25 | JT | yesit could definately be more reliable |
| 23:33.47 | TripleFFFF | right now my left nut and right nut are both in cisco ball camp |
| 23:33.48 | TripleFFFF | ;) |
| 23:33.56 | JT | you need to spend $10k on mysql? |
| 23:34.00 | TripleFFFF | more |
| 23:34.04 | JT | why |
| 23:34.09 | TripleFFFF | to get nay suport at all from them you need a license |
| 23:34.12 | JT | i wouldn't spend a cent on it |
| 23:34.19 | TripleFFFF | my point exaclt |
| 23:34.25 | JT | for such a mediocre rdbms |
| 23:34.29 | TripleFFFF | call them you will see |
| 23:34.41 | JT | nah i have no interest in it |
| 23:34.49 | JT | it only seems to interest web php noobs |
| 23:34.51 | TripleFFFF | pg ? |
| 23:35.01 | JT | pg is one option |
| 23:35.05 | JT | mysql is just junk |
| 23:35.15 | JT | otherwise there's commercial options |
| 23:35.15 | TripleFFFF | what u suing |
| 23:35.46 | JT | depends, we have a number of different databases here for different things |
| 23:41.06 | kieranmullen | any suggestions for who to use form incoming dids? |
| 23:41.53 | TripleFFFF | where from |
| 23:41.56 | TripleFFFF | use dringme ;) |
| 23:42.20 | kieranmullen | vitelity.com seems to have good prices? I am in oregon |
| 23:42.32 | kieranmullen | what is dringme? |
| 23:43.00 | TripleFFFF | ;) |
| 23:43.01 | TripleFFFF | oups |
| 23:43.22 | kieranmullen | ringme.com? |
| 23:43.33 | TripleFFFF | dring dring |
| 23:44.00 | kieranmullen | 503 |
| 23:44.09 | kieranmullen | looking to transfer 503-343 |
| 23:44.20 | TripleFFFF | oh you need a port ? |
| 23:44.25 | kieranmullen | y |
| 23:44.50 | TripleFFFF | ok |
| 23:45.45 | *** join/#asterisk jimbo- (n=jhio8838@cpe-74-65-239-78.nyc.res.rr.com) |
| 23:45.46 | jimbo- | hello, got a question re AGI |
| 23:45.46 | jimbo- | how can I retrieve the calling party's IP address? |
| 23:46.04 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 23:46.05 | TripleFFFF | unless you pass it along i think yuo cant |
| 23:46.12 | jimbo- | how do i pass it along? |
| 23:46.12 | Givemelove | hey there |
| 23:46.15 | TripleFFFF | no you cant |
| 23:46.17 | TripleFFFF | no idea |
| 23:46.21 | TripleFFFF | maybe parse the header |
| 23:46.25 | TripleFFFF | for contact info |
| 23:46.33 | jimbo- | contact info can be faked |
| 23:46.35 | TripleFFFF | and to do that you rtm |
| 23:46.40 | Givemelove | is there a way to check onto extensions.conf if a call is coming from a transfer or not? |
| 23:46.41 | TripleFFFF | then you cant |
| 23:46.56 | TripleFFFF | OR.. you make a peer based on ip.. in sip.conf and use the name |
| 23:46.57 | TripleFFFF | ;) |
| 23:47.12 | jimbo- | how do I get the peername from AGI? |
| 23:47.22 | TripleFFFF | rtm |
| 23:47.30 | jimbo- | i did, can't find anythin |
| 23:47.37 | jimbo- | no variable name i can query |
| 23:47.48 | TripleFFFF | wahts last page you checked ? |
| 23:47.53 | TripleFFFF | ill go from there |
| 23:47.54 | jimbo- | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 23:48.36 | Givemelove | is there a way to check onto extensions.conf if a call is coming from a transfer or not? |
| 23:48.57 | TripleFFFF | check in AGI section would help |
| 23:49.06 | TripleFFFF | and dump the env |
| 23:49.23 | TripleFFFF | Gimelove no diea |
| 23:49.31 | jimbo- | you mean this |
| 23:49.31 | jimbo- | http://voip-info.org/wiki/view/Asterisk+AGI |
| 23:50.13 | TripleFFFF | yeah |
| 23:50.15 | TripleFFFF | i gues |
| 23:50.36 | jimbo- | nada |
| 23:50.39 | jimbo- | nothing there |
| 23:50.42 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:51.40 | TripleFFFF | peername ? |
| 23:51.52 | TripleFFFF | hmm u can find the peer in the channel name. do a cut |
| 23:51.54 | TripleFFFF | on it |
| 23:52.55 | TripleFFFF | hmm is releasing 1.4 so fast any way related to g729 coming out in the open ? |
| 23:53.47 | brookshire | umm.. no.. it's illegal to use g729 without a license |
| 23:56.02 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:56.33 | jimbo- | g729 copyright is about to expire |
| 23:56.42 | jimbo- | 2007 if i recall |
| 23:56.55 | jimbo- | sorry, the patent |
| 23:57.13 | kieranmullen | who do music and books copyrights last so long? |
| 23:57.20 | kieranmullen | 100 years? |
| 23:57.27 | kieranmullen | technology patents only last what 10 years? |
| 23:57.31 | orlock | Cos of Disney |
| 23:57.35 | orlock | and Mickey Mouse |
| 23:58.03 | kieranmullen | 50 years for each would be good |
| 23:58.15 | brookshire | which one.. there are like 20 or so patents |
| 23:58.28 | kieranmullen | more books in music in public domain |
| 23:58.39 | kieranmullen | I thoguht all technology patents only last 10 years |
| 23:59.09 | kieranmullen | http://law.freeadvice.com/intellectual_property/patent_law/patent_duration.htm |
