| 00:00.29 | *** join/#asterisk asdx (n=diego@adsl-156-148.click.com.py) |
| 00:00.35 | asdx | can i use type=peer for inbound? |
| 00:00.41 | asdx | or it has to be type=user |
| 00:00.57 | asdx | inbound calls* |
| 00:01.47 | outtolunc | ac1djazz: because it is CallerID: Some Name <1234567890> |
| 00:02.14 | outtolunc | you did read teh sample file ... right? |
| 00:02.32 | ac1djazz | oooo |
| 00:02.36 | ac1djazz | negetaroy out thank you though |
| 00:03.28 | ac1djazz | CallerID: LOL DONGS <31337> |
| 00:03.29 | ac1djazz | didnt work |
| 00:04.39 | *** join/#asterisk rob0 (n=rob0@66.226.201.55) |
| 00:04.57 | outtolunc | #Callerid: Wakeup Call Service <(555) 555-5555> |
| 00:06.20 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 00:10.02 | ac1djazz | #? |
| 00:10.39 | outtolunc | it was a copy from 'sample.call' which that line was commented out |
| 00:10.47 | outtolunc | open your EYES |
| 00:11.01 | ac1djazz | not workin |
| 00:11.58 | ac1djazz | if i comment it out it just says "Call" |
| 00:12.06 | ac1djazz | if i let it stasy it doesnt give the one i specified, but teh real # |
| 00:15.00 | outtolunc | are your shoes tied? or you have velcro <G> |
| 00:16.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:16.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:16.16 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 00:16.20 | outtolunc | pastebin your call file so we can see what is hosed |
| 00:17.35 | exvito | hi all... i need to try a SVN zaptel version according to Digium suggestion... i know nothing about SVN and trying "svn co URL" fails... can anyone post a hint ? |
| 00:17.53 | exvito | (the URL i got is http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ |
| 00:18.21 | fakhir | svn co http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/ zaptel |
| 00:19.56 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 00:19.57 | exvito | ok, let me try that..., thanks fakhir |
| 00:20.57 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 00:21.05 | Horvath | Is there any real advantage to running asterisk on a 64bit kernel? |
| 00:21.18 | ac1djazz | it seems that my issue is that my voip carrier controls my callerid? |
| 00:21.33 | exvito | hmm, no, it still fails... "svn: PROPFIND request failed on [path]" + "svn: PROPFIND of [path] 301 Moved (http://svn.digium.com)" |
| 00:21.34 | *** join/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 00:21.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 00:21.50 | [T]ank | anyone here using asterisk in a call center environment? I want to compare notes. |
| 00:22.00 | [T]ank | 100 users or more |
| 00:22.07 | [T]ank | queuing and such |
| 00:23.05 | JunK-Y | tank: asking a specific questions will help. |
| 00:23.17 | exvito | sorry, i stand corrected, i was using the wrong URL ! checking it out now... thanks fakhir |
| 00:23.30 | fakhir | k |
| 00:24.11 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 00:24.23 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:24.53 | [T]ank | exvito: there are no specific questions... I want to compare notes. How do I compare in number of users, to others, how are others using queuing, are others doing a front end, if so, are they using agi etc... just wondering if there is anyone here using asterisk in a call center with queues. |
| 00:25.23 | exvito | [T]ank: ...i think you meant JunK-Y |
| 00:25.26 | exvito | :) |
| 00:25.33 | [T]ank | yep... sorry |
| 00:26.48 | [T]ank | ok, lets start with this... anyone here have more than 100 users on one asterisk system |
| 00:27.02 | MrTelphone | i have 50 users and it is flakey |
| 00:27.34 | *** join/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 00:27.45 | [T]ank | ok... I am at 200 and have not really had any real terrible issues. its not flakey |
| 00:28.12 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 00:28.21 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 00:28.24 | MrTelphone | i guess it depends what your doing |
| 00:28.29 | asdx | what more do i need to do apart from creating an extention/user for inbound calls? |
| 00:28.35 | asdx | i tried creating them but no luck :( |
| 00:28.43 | asdx | i try to call my teliax number and i get "service unavailable" |
| 00:28.52 | rob0 | Horvath: main advantage that I know of is that you can have more memory without the HIGHMEM kludge. |
| 00:29.36 | [T]ank | the only issue i have run into that i have not caused myself is that in 1.4.15+ with a ringall queue, once the users get a call and transfer it to another user, the call has to end at the second user before the first one can get another. digium provided me a patch, which came with a few issues :-(. anyone else come across that one? |
| 00:29.44 | Horvath | rob0, but I mean its asterisk it bearly uses any CPU/Mem |
| 00:30.16 | rob0 | if you had a busy enough PBX, you might appreciate the extra RAM :) |
| 00:30.37 | Horvath | rob0, I guess theres no reason not to run a 64bit kernel when you have the hardware |
| 00:30.58 | lirakis | ...Uses tons of cpu for transcoding |
| 00:31.21 | MrTelphone | im trying to switch to openser because my endpoints are very fussy |
| 00:31.34 | MrTelphone | the polycom phones work fine but its some other type of endpoint |
| 00:32.00 | MrTelphone | bad redundancy features |
| 00:32.18 | asdx | can someone help? i tried to use the book and their examples but it didn't work |
| 00:33.06 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 00:34.06 | MrTelphone | anyone use those sangoma a200 fxo cards on noisey lines? |
| 00:34.31 | comprookie2000 | asdx, https://www.teliax.com/pages/asteriskconfig |
| 00:35.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bffb14e2f70dae39) |
| 00:35.54 | Horvath | lirakis, I dont do transcoding :) |
| 00:36.08 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.187.78) |
| 00:36.17 | Horvath | lirakis, but you make a good point I wonder if the transcoding actually runs any faster on 64bit though |
| 00:37.02 | lirakis | ...If the transcoding app can make use of wider registers |
| 00:37.15 | Horvath | thats what im wondering |
| 00:37.30 | asdx | comprookie2000: that works but only for outbound calls, i'm trying to receive calls from normal telephones into my pbx |
| 00:37.36 | lirakis | ...But thats why they make transcoder cards ;) |
| 00:38.06 | jjshoe | MrTelphone 'noisey' ? |
| 00:38.34 | asdx | teliax gives me this number: (202) 470-5265 and i want to use that in order to receive calls |
| 00:39.14 | jjshoe | asdx do you have register line in sip.conf for them? |
| 00:39.14 | *** join/#asterisk OmiKrOn (n=fraktal@unaffiliated/omikron) |
| 00:39.22 | asdx | jjshoe: yes |
| 00:39.39 | jjshoe | asdx really? the url above says use iax. |
| 00:39.46 | asdx | register => user:pass@den.teliax.net |
| 00:39.49 | jjshoe | asdx why don't you read that article? |
| 00:40.18 | asdx | jjshoe: i cant use iax, my customers use sip devices |
| 00:40.29 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 00:40.39 | jjshoe | asdx it doesn't matter what end point they use. |
| 00:40.49 | jjshoe | hello clue bat |
| 00:41.35 | asdx | jjshoe: incoming calls are easier with iax? |
| 00:42.42 | comprookie2000 | asdx, For IAX2 inbound, you need port 4569 forwarding in to your Asterisk box |
| 00:43.32 | *** part/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 00:43.37 | asdx | ok |
| 00:44.12 | JT | udp port 4569 |
| 00:45.35 | asdx | what about with sip? |
| 00:45.38 | JT | you don't need to port forward to connect to an ITSP generally |
| 00:45.49 | JT | udp 5060 and udp 10000-20000 for rtp |
| 00:47.20 | JT | asdx: what are you trying to do? |
| 00:47.25 | [T]ank | is there a good script out there anywhere for reproducing call volume load on an asterisk server? The idea I have is that I have 2 trunk groups of pris. I could use one to dial the other and load test it. anything already built? |
| 00:47.42 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 00:47.53 | asdx | JT: i want to call from my normal telephone (PSTN) or any cellphone to my asterisk pbx |
| 00:48.03 | MrTelphone | jjshow, i tried digium fxo on analog lines with hum and they couldn't echo cancel at all without going into half duplex |
| 00:48.04 | JT | okay... how? |
| 00:48.13 | asdx | JT: and then press 123 and hear tt-monkeys |
| 00:48.39 | asdx | JT: i don't know, teliax gives me a number, i can use that number for that right? |
| 00:50.31 | JT | asdx: so the answer is VoIP |
| 00:50.38 | asdx | (202) 470-5265 Active den.teliax.net |
| 00:50.39 | JT | using some sort of protocol |
| 00:50.44 | JT | either iax2 or sip |
| 00:50.46 | JT | which one? |
| 00:50.48 | asdx | JT: yeah i know that, i want to use sip |
| 00:50.55 | JT | do they support sip? |
| 00:51.09 | asdx | yes |
| 00:51.14 | JT | asdx: well that's what the question "how?" means, there's a billion ways to connect asterisk to the PSTN |
| 00:51.16 | asdx | i make calls sometimes to normal telephones |
| 00:51.19 | asdx | trough sip |
| 00:51.28 | JT | asdx: you shouldn't need to forward any ports |
| 00:51.45 | asdx | i make calls to normal telephones, i want to do the other way around now |
| 00:51.51 | asdx | make calls from normal telephones to my voip phones |
| 00:52.17 | JT | make sure it's set to register with the provider |
| 00:52.26 | [T]ank | question regarding 1.4.19. Is there anyone that is on 1.4.19 that is using queues in a ringall? I have a bug in 1.4.15 that I want to see if exists in 1.4.19. I do not see anywhere that it is resolved. |
| 00:52.35 | [T]ank | i have a patch in place for it currently. |
| 00:52.43 | asdx | JT: it is, i see "registered" when i do sip show registry |
| 00:54.47 | ac1djazz | the sample in the asterisk book says to call or stream files like "degrees" or "with" or "fahrenheit" which dont seem to exist |
| 00:54.52 | ac1djazz | there some sound back imn missing? |
| 00:56.02 | asdx | JT: what about if i want to call from my softphone to that number (202) 470-5265 (my teliax number), can i do that? |
| 00:56.32 | asdx | JT: when i do that i get service unavailable |
| 00:56.36 | asdx | anyway |
| 00:56.56 | JT | asdx: where does the softphone call through? |
| 00:57.40 | asdx | i can call to any phone number with my softphone, or to any registered user |
| 00:57.44 | asdx | except to that number that teliax gives me |
| 00:58.36 | asdx | when i call to that number that teliax gives me, i expect to get connected to my * box |
| 00:59.01 | asdx | so i can dial any extention |
| 00:59.27 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:59.27 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:59.54 | asdx | that's how it should work right? |
| 01:00.06 | JT | i dunno, do you want to answer the question? |
| 01:00.50 | asdx | i don't know, i was hoping that someone knew more than me about this |
| 01:01.05 | JT | did you set up this softphone? |
| 01:01.12 | asdx | yes |
| 01:01.24 | asdx | it registers to my pbx |
| 01:01.32 | asdx | and i can dial extentions, etc |
| 01:01.34 | JT | and how does it make calls to the pstn? |
| 01:01.57 | asdx | _9.,1,Dial(SIP/teliax/${EXTEN:1}) |
| 01:02.00 | asdx | with that extention |
| 01:02.32 | JT | so it calls using the teliax ITSP |
| 01:02.34 | asdx | then i dial 901159521201964 in my softphone |
| 01:02.36 | asdx | and it just dials |
| 01:02.37 | JT | see, that wasn't that hard |
| 01:02.43 | asdx | yeah |
| 01:02.52 | JT | so you are calling yourself, through the same number |
| 01:02.58 | JT | your itsp might not allow that |
| 01:03.08 | JT | use a real phone to call the number. |
| 01:07.12 | asdx | calling myself to that number wont work? |
| 01:08.05 | jjshoe | asdx did you not read what JT said? |
| 01:08.10 | Horvath | Anyone here fimilar with sipp? Trying to bench asterisk |
| 01:09.16 | JT | jjshoe: reading people's answers back as questions is fun isn't it? |
| 01:09.19 | JT | :P |
| 01:09.31 | JT | asdx: seriously, already answered that. |
| 01:10.20 | asdx | ok |
| 01:10.31 | asdx | i'm going to ask my boss if he have skype or something like that so i could use that to test |
| 01:10.33 | asdx | s/test/call |
| 01:10.36 | asdx | i can't call from home |
| 01:10.45 | JT | don't you have a bloody landline or cell? |
| 01:10.54 | JT | how can you possibly test without one |
| 01:11.01 | JT | properly test anyway |
| 01:11.19 | asdx | i live in paraguay |
| 01:11.23 | asdx | i have a cell |
| 01:11.27 | asdx | but calling to that number is expensive |
| 01:12.36 | JT | maybe you need a voip account without a different provider on a softphone to test |
| 01:14.56 | asdx | yeah |
| 01:26.14 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 01:30.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 01:39.30 | asdx | JT: do you know if i can buy a phone number (i think it's called DID numbers?) from my country (Paraguay) in this case, and then call that phone number from my normal phone and then it gets connected to my * PBX? |
| 01:40.27 | JT | if there's someone who offers that, then sure |
| 01:41.21 | asdx | http://www.didww.com |
| 01:41.26 | asdx | damn, no paraguay |
| 01:42.33 | *** join/#asterisk iNick (n=noident@nmbug.org) |
| 01:42.37 | iNick | wow, lots of ppl here |
| 01:42.39 | iNick | :) |
| 01:42.49 | iNick | that's good. |
| 01:43.40 | *** join/#asterisk _daver_ (n=daver@entropy.tmok.com) |
| 01:47.22 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:48.52 | JT | asdx: goes paraquay have voip providers? |
| 01:50.59 | asdx | JT: yes |
| 01:52.27 | asdx | net2phone or something like that |
| 01:52.58 | JT | they should be able to offer a local DID |
| 01:53.17 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 01:54.51 | iNick | i've been asked to research Asterisk in possible replacement of our 3COM VoIP solution |
| 01:55.06 | asdx | JT: yeah i'm going to ask them |
| 01:55.07 | iNick | and I know the concepts, some of the terms, but have no CLUE on how to put it all together |
| 01:55.18 | JT | ~thebook |
| 01:55.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:56.23 | iNick | anyone willing to help me with a wizard on what components I need? (I'm d/ling the pdf book now) |
| 01:57.57 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:58.28 | JT | iNick: what do you need it to do? |
| 01:59.26 | asdx | JT: calling myself to that DID number that teliax gives me from my softphone is like calling my phone number from my normal phone right? |
| 01:59.37 | TJNII | I'm having a problem with my SIP provider where when I call them I get transferred from one server to another. Server A tells my server to hold, so my * box starts hold music. Server B never sends the message that I |
| 01:59.59 | TJNII | I'm having a problem with my SIP provider where when I call them I get transferred from one server to another. Server A tells my server to hold, so my * box starts hold music. Server B never sends the message that I'm no longer on hold, so I stay on hold forever. How can I get * to ignore this? |
| 02:00.34 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:01.03 | jameswf-home | anyone seen an osrisch running around here?? |
| 02:01.08 | jameswf-home | :) |
| 02:01.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:03.30 | asdx | how do you add the DID numbers into the pbx? |
| 02:04.18 | jameswf-home | ~ch5 |
| 02:04.19 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 02:04.44 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:05.31 | JT | asdx: no, it's not really like that at all |
| 02:05.41 | JT | it's not a good test |
| 02:05.47 | JT | too many variables |
| 02:06.47 | *** join/#asterisk SomethingISODD (n=noc@S010600a0d1757bfb.cg.shawcable.net) |
| 02:07.24 | TJNII | has to call his sip provider on a cell phone. That's embarassing. |
| 02:07.32 | asdx | JT: why doesn't work when i call myself to that DID number? |
| 02:07.38 | SomethingISODD | hello all i have a few questions about chan_h323 i am trying to register clients in strictly ip and route each ip to a different prefix |
| 02:07.40 | asdx | it should work... |
| 02:07.55 | SomethingISODD | but it only routes to the default context |
| 02:08.00 | JT | asdx: check with your itsp if they allow more than 1 simultaneous phone call |
| 02:08.14 | JT | asdx: you can also do a sip debug to see if the call comes in |
| 02:08.29 | asdx | yeah |
| 02:08.32 | asdx | let me try that |
| 02:10.25 | asdx | JT: it has 4 channels, and 1 number, it doesn't say more than that |
| 02:12.22 | jameswf-home | Doing asterisk without RTFM is like doing brain surgery with a swiss army knife and a box of bandaids..... sure you could but something will likely go terribly wrong |
| 02:12.30 | asdx | my plan is the business plan from teliax |
| 02:12.39 | asdx | https://www.teliax.com/plans/3 |
| 02:13.10 | asdx | jameswf-home: i RTFM from time to time |
| 02:13.24 | asdx | but there are times that i get stuck |
| 02:13.31 | TJNII | ~itsp-us |
| 02:13.33 | asdx | and i don't know wth goes wrong |
| 02:14.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:15.14 | TJNII | ~itsp |
| 02:15.14 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 02:15.22 | TJNII | ~itsplist |
| 02:15.27 | TJNII | ~itsp-list |
| 02:15.49 | TJNII | A little help? I'm looking for the list of US ITSPs |
| 02:16.00 | TJNII | ~itspus |
| 02:16.17 | ac1djazz | is there some kinda agi command reference doc url |
| 02:16.23 | ac1djazz | where can i check out commandses |
| 02:18.06 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582382.dsl.bell.ca) |
| 02:19.03 | TJNII | ac1djazz: tab completion works |
| 02:19.17 | TJNII | along with the help command |
| 02:19.41 | jameswf-home | ~itsplist-us |
| 02:19.42 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 02:20.00 | TJNII | ty jameswf-home |
| 02:20.31 | jameswf-home | ~wikis ac1djazz |
| 02:20.38 | jameswf-home | ~wikis |
| 02:20.39 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 02:21.00 | TJNII | Apparently Broadvoice doesn't support callerID. If that is their official position, I'm going to tell them to sit and spin. |
| 02:21.20 | JT | TJNII: inbound or outbound? |
| 02:21.27 | *** part/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 02:21.35 | *** join/#asterisk steliosk (n=Stelios@85.72.76.221) |
| 02:21.53 | TJNII | I've had service for months and outbound callerID just quit last week. Now they're telling me they never supported it. |
| 02:22.53 | TJNII | JT: outbound. Inbound still works OK. |
| 02:23.24 | ac1djazz | TJNII: im writing the commands in a diff language though, not in the CLI |
| 02:24.04 | TJNII | ac1djazz: What? |
| 02:24.17 | TJNII | Oh, AGI |
| 02:24.42 | TJNII | I misread that, sorry. voip-info has a good list of agi commands. |
| 02:24.55 | iNick | JT: sorry, was reading book. we have 1 T1 line inbound, and ~30 desktop phones that need to work. |
| 02:25.16 | iNick | The desktop phones don't have direct lines to the desk, they all go through the receptionist |
| 02:27.03 | *** join/#asterisk ming_zy1 (n=ming_zym@nat/yahoo/x-1cf661231e02748e) |
| 02:27.44 | asdx | JT: thats what i get when i try to call my DID number... http://pastie.caboo.se/181489 |
| 02:27.59 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 02:29.07 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 02:30.27 | asdx | err, no, thats the wrong one |
| 02:32.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:36.27 | asdx | http://pastie.caboo.se/181491 |
| 02:38.19 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 02:42.42 | ManxPower | TJNII: what callerid are you trying to send them? |
| 02:45.11 | SomethingISODD | hello all i have a few questions about chan_h323 i am trying to register clients in strictly ip and route each ip to a different prefix but it only routes to the default context |
| 02:46.19 | TJNII | ManxPower: Just the info I gave broadvoice, my billing name and number. Nothing fancy. |
| 02:46.48 | TJNII | ManxPower: Half the time it sends it and half the time it shows "restricted" or "private" |
| 02:52.42 | *** join/#asterisk mmurdock (n=tga@c-98-202-154-226.hsd1.ut.comcast.net) |
| 02:52.52 | jameswf-home | try a _. route |
| 02:54.35 | *** join/#asterisk korihor (n=humberto@190.74.120.245) |
| 02:54.57 | *** join/#asterisk suvir (n=suvir@ppp-124-120-141-89.revip2.asianet.co.th) |
| 02:56.28 | *** part/#asterisk suvir (n=suvir@ppp-124-120-141-89.revip2.asianet.co.th) |
| 03:03.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:04.41 | *** join/#asterisk andrewn (n=andrew@76-191-213-99.dsl.dynamic.sonic.net) |
| 03:04.46 | lmadsen | _. is not ideal |
| 03:05.08 | lmadsen | if you need to match on a letter instead of a number, do: _[a-zA-Z].,1,NoOp() |
| 03:05.27 | *** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 03:05.55 | lmadsen | otherwise do _X. |
| 03:06.28 | lmadsen | or _[a-zA-Z0-9]. will match everything except single letter extension (i.e. i for invalid, h for hangup, etc...) |
| 03:06.49 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:06.55 | Kyoshi | need help bad with asterisk real time. http://www.pastebin.ca/986980 - Thank you. |
| 03:09.21 | asdx | i tried the examples in the book for incoming calls and it didn't work, can someone help? |
| 03:10.02 | Corydon76-dig | Kyoshi: you don't specify the filename in odbc.ini, only in odbcinst.ini |
| 03:10.34 | Kyoshi | cor, lemme try then... just name the driver i take it |
| 03:10.34 | Corydon76-dig | Kyoshi: in odbc.ini, "Driver = MySQL ODBC 3.51 Driver" |
| 03:10.37 | Kyoshi | k |
| 03:12.13 | *** join/#asterisk erojasv (n=erojasv@190.43.10.68) |
| 03:12.22 | jameswf-home | muh bad :) |
| 03:13.21 | Kyoshi | cor, now i get a diff error... |
| 03:13.22 | Kyoshi | WARNING[10235]: res_odbc.c:559 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 03:14.05 | Kyoshi | before that it says 'NOTICE[10235]: res_odbc.c:288 load_odbc_config: registered database handle 'res_mysql' dsn->[dsn_mysql]' |
| 03:16.55 | Corydon76-dig | Kyoshi: what do you get when you type: 'isql dsn_mysql' ? |
| 03:17.45 | Kyoshi | errors |
| 03:17.59 | Kyoshi | isql dsn_mysql myusername mypassword |
| 03:18.01 | Kyoshi | errors |
| 03:18.03 | Corydon76-dig | Try: isql -vv dsn_mysql |
| 03:18.09 | Kyoshi | k |
| 03:18.53 | Kyoshi | [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 03:19.38 | Corydon76-dig | Kyoshi: you have a typo somewhere in odbc.ini |
| 03:19.38 | Kyoshi | but it worked before |
| 03:19.50 | Kyoshi | so strange |
| 03:20.13 | Corydon76-dig | Most likely the Driver= line does not match EXACTLY the category name in odbcinst.ini |
| 03:20.34 | Kyoshi | sorry for the paste.... |
| 03:20.37 | Kyoshi | [MySQL ODBC 3.51 Driver] |
| 03:20.37 | Kyoshi | Description = ODBC 3.51 for MySQL |
| 03:20.37 | Kyoshi | DRIVER = /usr/lib/libmyodbc3.so |
| 03:20.43 | Kyoshi | SETUP = /usr/lib/libmyodbc3S.so |
| 03:20.43 | Kyoshi | UsageCount = 3 |
| 03:20.53 | Kyoshi | ooo |
| 03:20.54 | Kyoshi | wait |
| 03:20.55 | Kyoshi | my bad |
| 03:20.57 | Kyoshi | one sec |
| 03:22.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f12939674162aca9) |
| 03:24.38 | Kyoshi | *CLI> Apr 14 19:59:59 NOTICE[10294]: chan_sip.c:11242 handle_request_register: Registration from '<sip:87216@192.168.27.177:5060>' failed for '192.168.27.203' - Username/auth name mismatch |
| 03:24.47 | Kyoshi | sorry the box is a day behind :p |
| 03:25.04 | Kyoshi | username 87216 / secret 87216 |
| 03:25.07 | Corydon76-dig | Kyoshi: the 'isql' command has to work first |
| 03:25.13 | Kyoshi | its working now |
| 03:25.16 | Kyoshi | sorry |
| 03:25.19 | Corydon76-dig | Okay, good |
| 03:25.21 | Kyoshi | im good with odbc and stuff |
| 03:25.28 | Kyoshi | just not ast realtime |
| 03:25.49 | Corydon76-dig | Okay, your second argument in extconfig.conf is incorrect |
| 03:25.55 | Kyoshi | figures |
| 03:26.07 | Kyoshi | im stupid as i said |
| 03:26.11 | Corydon76-dig | Are you sure you want sip.conf and not sippeers? |
| 03:26.21 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 03:26.21 | Kyoshi | i want umm |
| 03:26.36 | Corydon76-dig | sip.conf is for the old static realtime, not dynamic realtime |
| 03:26.42 | Kyoshi | sipusers |
| 03:26.49 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:26.56 | Corydon76-dig | In any case, the value should be "odbc,res_mysql,sipusers" |
| 03:27.15 | Corydon76-dig | since res_mysql is what you called it in res_odbc.conf |
| 03:27.24 | Kyoshi | currently my extconfig.conf says sipusers => odbc,asterisk,sipusers |
| 03:27.31 | asdx | should i need another line for trying incoming calls? |
| 03:27.35 | Corydon76-dig | right, change the second arg |
| 03:27.40 | Kyoshi | hmm |
| 03:27.43 | Kyoshi | tried that before |
| 03:27.45 | Kyoshi | lemme try again |
| 03:27.54 | Corydon76-dig | Kyoshi: yes, but isql wasn't working before |
| 03:28.08 | Corydon76-dig | isql tests JUST the odbc layer |
| 03:28.21 | Corydon76-dig | and that has to work before Asterisk has a chance of working |
| 03:29.21 | Kyoshi | i got too many dsn's spoiling my stuff :( |
| 03:30.01 | Kyoshi | ok now what? changed extconfig.conf |
| 03:30.05 | Corydon76-dig | btw, you probably want both sipusers and sippeers |
| 03:30.10 | asdx | i'm trying to make incoming calls work in my asterisk installation, but i only have my voip-provider for incoming/outgoing calls, will i need another voip provider for testing incoming calls? or i can just call from my softphone to my DID number and try that? |
| 03:30.13 | Kyoshi | k |
| 03:30.22 | Corydon76-dig | That registration error won't go away unless you have peers |
| 03:31.10 | asdx | i don't have a cellphone or landline |
| 03:31.14 | Corydon76-dig | asdx: have you confirmed that outgoing calls work? |
| 03:31.24 | asdx | Corydon76-dig: yes outgoing calls work |
| 03:31.34 | *** join/#asterisk blq (n=Bl@dslb-088-066-243-114.pools.arcor-ip.net) |
| 03:31.41 | Corydon76-dig | asdx: then the worst that you can do is to try calling your own number |
| 03:31.46 | *** join/#asterisk blq (n=Bl@dslb-088-066-243-114.pools.arcor-ip.net) |
| 03:31.51 | asdx | Corydon76-dig: why? |
| 03:32.16 | asdx | Corydon76-dig: yeah i believe you |
| 03:32.30 | asdx | Corydon76-dig: thats what i been trying to do sine 10 hours ago and it's been a nightmare |
| 03:32.37 | asdx | since |
| 03:32.53 | Corydon76-dig | asdx: then you need to talk to your provider |
| 03:33.02 | jameswf-home | asdx: do exten => _[a-zA-Z0-9].,n,NOOP(WHO YOU GOING TO CALL: ${EXTEN}) |
| 03:33.31 | jameswf-home | make sure your getting what you thin you are |
| 03:33.37 | jameswf-home | *think |
| 03:33.47 | asdx | jameswf-home: what would that do? |
| 03:34.22 | jameswf-home | it prints what the inbound call will be looking for in your context |
| 03:34.26 | asdx | Corydon76-dig: why do you say that it's the worst thing to do, because it wont work or?? |
| 03:34.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:35.30 | Corydon76-dig | asdx: See 'idiom' |
| 03:35.41 | jameswf-home | ~idiom |
| 03:35.46 | jameswf-home | doh |
| 03:36.09 | asdx | ??????? |
| 03:36.09 | Corydon76-dig | I never said it was the worst thing you can do... I said the worst you could do is to try it |
| 03:36.36 | Corydon76-dig | obviously lost on a non-native speaker |
| 03:36.37 | asdx | why? |
| 03:36.56 | asdx | why is the worst thing to try? |
| 03:36.57 | drmessano | Corydon76-dig: What if his Asterisk box sends U shaped packets? |
| 03:37.05 | Kyoshi | ok so i need the sipusers and sippeers tables and both configured in the extconfig.conf file right? |
| 03:37.15 | Corydon76-dig | asdx: Nevermind. |
| 03:37.28 | Corydon76-dig | asdx: it's an idiomatic expresssion |
| 03:37.58 | Kyoshi | ok so thats done |
| 03:38.05 | asdx | Corydon76-dig: ok |
| 03:38.08 | Corydon76-dig | Kyoshi: that's it |
| 03:38.26 | asdx | Corydon76-dig: it doesn't make much sense to call to my own number anyway |
| 03:38.31 | asdx | Corydon76-dig: i'm going to try to get another line |
| 03:38.45 | jameswf-home | WOW serious... |
| 03:38.53 | jameswf-home | ~freesupport |
| 03:38.54 | jbot | freesupport is, like, Remember people are in irc to help but their are limits to free support. you should try to google or call the manufacturers support for hardware issues. The fastest way to lose help here is to be impatient or combative. If you know so much more than us that you can say "no that's not it" maybe you shouldn't ask us questions |
| 03:39.21 | drmessano | Maybe you need a Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 03:39.41 | Corydon76-dig | I like my Hitachi 3000 |
| 03:39.54 | drmessano | ~happyclownpbx |
| 03:39.55 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 03:40.11 | drmessano | Things are looking up for the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 03:40.19 | Kyoshi | cory: how will i know if it was successful? do i need records in both the peers and users tables? |
| 03:40.50 | Corydon76-dig | Kyoshi: most people configure it to be the same table |
| 03:41.21 | Corydon76-dig | and call the table 'sipfriends' or some such |
| 03:42.49 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:42.49 | Kyoshi | ahh |
| 03:42.49 | Kyoshi | so im a moron |
| 03:42.49 | Kyoshi | thats ok |
| 03:42.49 | jameswf-home | I wasnt going to say it |
| 03:42.50 | jameswf-home | :) |
| 03:43.20 | drmessano | Kyoshi: There are no such things as morons here.. The fact you're trying gives you a +10 against the n00bkfletcher |
| 03:43.28 | drmessano | ~n00bkfletcher |
| 03:43.29 | jbot | Everyone was a n00b once, but if you're a non-listening n00b, expect a kvelching from the n00bkfletcher! |
| 03:45.57 | jameswf-home | is kvelching a KDE app |
| 03:46.29 | Corydon76-dig | Or is it kfelching ? |
| 03:47.09 | Kyoshi | trying for 3 days |
| 03:47.13 | drmessano | It's a V.. f is close, but you need more phlegm |
| 03:47.32 | Kyoshi | i just dont know how to tell if its working other than trying to get a softphone to register with the ast svr |
| 03:47.41 | Kyoshi | nothing in the logs or cli will say if it worked? |
| 03:48.26 | JunK-Y | pokes Corydon76-dig. |
| 03:48.35 | Corydon76-dig | Yes, love? |
| 03:48.58 | jameswf-home | for fun originate zap/1/victimA application dial zap/2/victimB |
| 03:49.12 | drmessano | yum install kvelching failed |
| 03:49.22 | JunK-Y | nothing, just want to say hi honey. |
| 03:49.28 | drmessano | Guess they ran afoul of the GPL |
| 03:49.45 | jameswf-home | ~wiki kvelching |
| 03:49.56 | *** join/#asterisk RoyK (n=roy@91.149.31.152) |
| 03:51.54 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 03:52.30 | MrTelephone | if you had two adit 600's one with fxo cards and the other with fxs cards.. if configured properly you can run cat5 between the t1 ports and it should extend the fxo lines? |
| 03:52.57 | JT | if one can generate internal clocksource |
| 03:53.18 | MrTelephone | i can't remember if it can |
| 03:53.23 | MrTelephone | im pretty sure it can |
| 03:54.34 | jameswf-home | You can with rhinos I dont see why adit would be different but could be |
| 03:54.46 | MrTelephone | im trying to help a business owner out |
| 03:55.24 | MrTelephone | he owns 2 hotels and im trying to feed a pbx with analog lines |
| 03:55.32 | jameswf-home | make sure you use a rj48c |
| 03:56.23 | MrTelephone | im almost sure it will need an asterisk box to be able to oversubscribe the fxo lines |
| 03:56.37 | drmessano | Im confused about the ilbc thing |
| 03:56.41 | MrTelephone | 10 fxo lines coming in for example.. but want to feed two pbxs with 8 fxs each |
| 03:57.11 | drmessano | If I get the tarball, I need to run the script.. if I use SVN, I can still check it out.. |
| 03:57.16 | *** join/#asterisk znoG (n=gs@host52.190-31-242.telecom.net.ar) |
| 03:57.19 | drmessano | Maybe not so confusing |
| 03:57.46 | MrTelephone | lines 1-5 goto building 1, lines 6-10 incoming goto building 2. outgoing called use reverse order on 1-5 and 6-10 until they are all in use |
| 03:57.53 | MrTelephone | i guess it can be down but it will require a machine |
| 03:57.56 | MrTelephone | yikes |
| 03:58.33 | MrTelephone | analog lines suck |
| 03:58.52 | Kyoshi | whoa |
| 03:58.58 | jameswf-home | I am not sure a channelbank is what you want.... alot of money to extend 3 extra pairs |
| 03:58.59 | Kyoshi | cory |
| 03:59.03 | Kyoshi | i love you |
| 03:59.05 | Kyoshi | :) |
| 03:59.15 | Kyoshi | i deeply and truly appreciate the help |
| 03:59.23 | Corydon76-dig | yw |
| 03:59.24 | MrTelephone | the buildings phone bill is 800 a month |
| 04:00.24 | jameswf-home | So you can get a t1 in and push physical analog lines out to the building... T1 4 pair (cat 5 had 4 pair) string 1 extra pair your their |
| 04:00.51 | jameswf-home | there has and misc other typos |
| 04:00.59 | MrTelephone | its 10 km out of town so t1 is too expensive |
| 04:01.26 | jameswf-home | I am lost as to where your trying to save 10 lines is 10 lines |
| 04:02.29 | MrTelephone | we are cutting down from 18 lines down to 10 or something |
| 04:02.43 | jameswf-home | How many rooms total |
| 04:02.53 | MrTelephone | 16 rooms on one side and 32 on the other |
| 04:03.13 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.32 | MrTelephone | the guy needs a whole new system really |
| 04:03.46 | MrTelephone | he's running some NEC pbx |
| 04:04.27 | JT | why can't you run ethernet between the sites? |
| 04:04.36 | MrTelephone | its across a major highway |
| 04:04.47 | JT | then you will do what? |
| 04:05.03 | MrTelephone | i installed a 10mbit wireless bridge last summer for internet |
| 04:05.19 | JT | so are you going to use it for VoIP? |
| 04:05.38 | MrTelephone | i was thinking about it but the WAN connection is a cable modem.. it will only handle a couple calls i think |
| 04:05.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 04:05.51 | MrTelephone | voip between the sites would be nice |
| 04:05.55 | *** part/#asterisk RoyK (n=roy@91.149.31.152) |
| 04:06.03 | MrTelephone | with one of those t1 over ip RAD units |
| 04:06.45 | JT | err |
| 04:06.51 | JT | why can't you use the wireless bridge? |
| 04:07.05 | MrTelephone | yeah I was planning on it |
| 04:07.16 | JT | running voip |
| 04:07.26 | jameswf-home | MrTelephone: check priv |
| 04:07.30 | JT | dunno what the story is with this back to back channel bank idea |
| 04:09.07 | MrTelephone | i didn't want to touch the original pbx |
| 04:09.13 | MrTelephone | voip -> channel bank -> pbx |
| 04:09.46 | drmessano | JT: It's channel bank night |
| 04:09.49 | drmessano | Err |
| 04:09.54 | drmessano | Week, actually |
| 04:10.00 | JT | heh |
| 04:10.04 | JT | seems so |
| 04:11.06 | drmessano | Asterisk w/FXO cards > Channel Bank > PRI <> PRI < channel bank < Asterisk w/FXO + Asterisk with T1 card < Channel Bank < Legacy FXO PBX |
| 04:11.17 | drmessano | So umm.. care to help me troubleshoot? |
| 04:11.35 | drmessano | I have some audio issues |
| 04:11.48 | jameswf-home | drmessano: welcome to my world |
| 04:11.51 | jameswf-home | daily |
| 04:12.04 | drmessano | I have a question |
| 04:12.13 | drmessano | Can I fix my audio issues by adding another channel bank? |
| 04:12.45 | jameswf-home | yes |
| 04:12.48 | jameswf-home | no |
| 04:12.50 | jameswf-home | maybe |
| 04:12.55 | drmessano | Great.. Sorta |
| 04:13.16 | drmessano | I think I may have a signalling issue.. or, not |
| 04:13.52 | drmessano | I turned on debug and got a PC_LOAD_LETTER error.. what that mean? |
| 04:14.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.218.194) |
| 04:14.21 | jameswf-home | THERE IS NO PAPER JAM |
| 04:14.22 | [TK]D-Fender | \o/ |
| 04:14.43 | [TK]D-Fender | jameswf-home, WTF does PC LOAD LETTER mean? |
| 04:14.54 | drmessano | lol |
| 04:14.57 | drmessano | You missed it.. |
| 04:15.01 | JT | yeah |
| 04:15.05 | [TK]D-Fender | I'm sure I did :) |
| 04:15.11 | drmessano | I was just commenting to JT how this is Channel bank week |
| 04:15.21 | drmessano | [00:10] <drmessano> Asterisk w/FXO cards > Channel Bank > PRI <> PRI < channel bank < Asterisk w/FXO + Asterisk with T1 card < Channel Bank < Legacy FXO PBX |
| 04:15.25 | JT | if you could go right ahead an ensure that you always attach the new cover sheets to your TPS Reports, that's be great |
| 04:15.27 | [TK]D-Fender | For me this is "public subnet week". |
| 04:15.35 | drmessano | [00:10] <drmessano> So umm.. care to help me troubleshoot? [00:11] <drmessano> I have some audio issues |
| 04:15.56 | drmessano | I figured JT would appreciate that |
| 04:15.57 | [TK]D-Fender | Learned a lot about netfilter NAT, subnet routing, etc today... just went live at home with my new /29 |
| 04:16.03 | drmessano | Cool |
| 04:16.15 | [TK]D-Fender | TONS more to learn of course... |
| 04:16.37 | MrTelephone | PSTN <> FXO ADIT600 T1 <> T1 ASTERISK-T1 <> RADVMUX1 <> WIRELESS BRIDGE <> RADVMUX1 <> ADIT600FXS <> LEGACY PBX |
| 04:16.51 | MrTelephone | ? |
| 04:16.52 | MrTelephone | hah |
| 04:17.03 | JT | remember to use a dialup connection too |
| 04:17.04 | drmessano | Sad thing is, MrTelephone, I took you serious |
| 04:17.05 | MrTelephone | those legacy pbx's are using regular analog phones right? |
| 04:17.22 | [TK]D-Fender | 2 NICs (1 x class C local NAT to dynamic ppp0), (1 x 1st IP of subnet/29) |
| 04:17.35 | [TK]D-Fender | JT, I am on ppp0 :) |
| 04:17.37 | MrTelephone | drmessano, im being semi-serious |
| 04:17.48 | drmessano | MrTelephone: Sadly, I know |
| 04:17.56 | [TK]D-Fender | JT, PPPoE of course.... |
| 04:17.57 | MrTelephone | jameswf-home had some good ideas to bring to this guy |
| 04:18.01 | JT | [TK]D-Fender: heh |
| 04:18.04 | drmessano | MrTelephone: Friends don't let friends use that many channel banks |
| 04:18.45 | JT | i'm glad the only channel banks i deal with is the one i have at home |
| 04:18.55 | MrTelephone | i think ip phones are too valueable to put in hotel rooms.. keep the same phones and upgrade the pbx |
| 04:19.19 | MrTelephone | the adit600 running a CICS 3.0 merdian i installed works the best on my whole asterisk system |
| 04:19.20 | [TK]D-Fender | I vaguely thought about running DHCP on my WAN subnet, but then NAH... I want those fixed anyways.... now my support options grow. |
| 04:19.23 | MrTelephone | faxing works 100% |
| 04:19.26 | drmessano | How much is a NEW hotel handset? |
| 04:19.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:19.38 | MrTelephone | he installed these cheap little analog handsets |
| 04:19.47 | MrTelephone | they look like they are 30 bucks or something |
| 04:20.26 | drmessano | you do hit an inverse curve where there's diminishing returns from buying a GOOD phone for a hotel room, but a cheap phone is going to be replaced more often |
| 04:20.51 | MrTelephone | I think what I meant is that its featureless |
| 04:20.58 | MrTelephone | 0-9 buttons * and # |
| 04:21.00 | MrTelephone | haha |
| 04:21.16 | JT | what, doesn't even have A, B, C and D?! |
| 04:21.22 | MrTelephone | I don't think so |
| 04:21.23 | drmessano | Anytime you put someting CHEAP in a high loss, high risk area, you risk overcheaping it for fear of loss |
| 04:21.27 | drmessano | and that is not good |
| 04:21.32 | MrTelephone | hahaha |
| 04:21.55 | JT | i want an analogue handset with a 16 character dtmf keypad |
| 04:22.50 | drmessano | I learned that with buying $4 mice for PCs in control rooms.. I found a good $15 optical fixed the issue with the mechanical wearing out, and I was replacing far less for coke and mountain dew spills than I was monthly for mechanical |
| 04:23.09 | MrTelephone | i agree |
| 04:23.43 | MrTelephone | i remember the owner of this place left his investment website open by accident and he had 1.2 million sittin in one of his accounts |
| 04:23.52 | MrTelephone | and I can tell you why, he's a cheap bastard |
| 04:24.12 | drmessano | So nothing wrong with being sensible, but you can actually spend MORE money by being too cheap |
| 04:24.19 | MrTelephone | i agree |
| 04:24.27 | MrTelephone | a good example is if you buy a KIA automobile |
| 04:24.43 | drmessano | heh |
| 04:24.59 | drmessano | Except I made out just fine with my two |
| 04:25.24 | MrTelephone | its sad because business is slow up here so its not like these phones are getting overused |
| 04:28.30 | MrTelephone | his current pbx has to plug into this little pad and everytime there is a phone call over 60 seconds it prints to a dot matrix printer |
| 04:29.01 | JT | old school |
| 04:29.16 | drmessano | JT: That's one way to put it |
| 04:29.18 | JT | i was thinking about setting up a dot matrix in line printer mode in the datacentre |
| 04:29.21 | JT | just to be a tool |
| 04:29.26 | MrTelephone | hahha |
| 04:29.26 | JT | and make everyone go "wtf" |
| 04:29.33 | MrTelephone | its good for what it does.. |
| 04:29.35 | JT | make it tail auth logs |
| 04:29.38 | droops | whats that noise? |
| 04:29.42 | MrTelephone | but why not just have a webpage pull cdr from asterisk and print? |
| 04:29.51 | MrTelephone | hahha |
| 04:30.12 | MrTelephone | make it print tcpdump on a 1gbs nic box |
| 04:30.17 | JT | lol |
| 04:30.18 | MrTelephone | eh |
| 04:30.24 | JT | it will run out of box paper very fast |
| 04:30.35 | JT | or ink ribbon |
| 04:30.39 | JT | or buffers, or something |
| 04:30.52 | MrTelephone | hahaa |
| 04:31.00 | drmessano | JT I have one I use for printing weather bulletins |
| 04:31.02 | MrTelephone | they still use dot matrix oki datas alot at stores though that need duplicate |
| 04:31.10 | drmessano | Its great when I get a warning and it cranks up |
| 04:31.52 | JT | yeah |
| 04:31.55 | JT | wakes you up |
| 04:32.09 | jameswf-home | hmmmm dotmatrix porn |
| 04:32.24 | MrTelephone | how do you do in room wake up calls and shit |
| 04:32.35 | MrTelephone | i guess you can just program that in there and charge 1000$ for the feature |
| 04:32.36 | MrTelephone | hahah |
| 04:32.37 | JT | schedule a .call file |
| 04:33.08 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:33.12 | joelsolanki | Hi room |
| 04:33.19 | [TK]D-Fender | jameswf-home, I remember those days... |
| 04:33.20 | MrTelephone | what do you use for a switchboard then |
| 04:33.29 | MrTelephone | right now the place has a huge button board with every number |
| 04:33.34 | [TK]D-Fender | jameswf-home, 9-pin shiznit y0! |
| 04:33.52 | joelsolanki | my setup was already working before but from 2 days i m having issues. |
| 04:34.02 | joelsolanki | i m using xlite to connect to asterisk server |
| 04:34.09 | joelsolanki | it has 6 g729 license installed. |
| 04:34.24 | joelsolanki | when in make call with g711 on xlite. and do show g729 on asterisk |
| 04:34.26 | drmessano | I have the wakeup.php running on my box |
| 04:34.28 | [TK]D-Fender | joelsolanki, And X-Lite supports all none of them. |
| 04:34.32 | drmessano | Actually, on several |
| 04:34.37 | drmessano | LOL |
| 04:34.41 | joelsolanki | it shows me this asterisk1*CLI> show g729 |
| 04:34.42 | joelsolanki | 0/0 encoders/decoders of 6 licensed channels are currently in use |
| 04:34.53 | drmessano | Fail X 6 |
| 04:34.55 | MrTelephone | shitty |
| 04:34.55 | joelsolanki | i remember it should show me 1/1 |
| 04:35.00 | joelsolanki | any suggestions ? |
| 04:35.02 | [TK]D-Fender | drmessano, I did a tt-monkeys .call file that auto-answer dialed my polycoms :) |
| 04:35.03 | drmessano | In a dream |
| 04:35.08 | drmessano | LOL |
| 04:35.18 | joelsolanki | due to this reason my voice is not going to remote person |
| 04:35.19 | drmessano | joelsolanki: X-lite doesn't do G729 |
| 04:35.20 | MrTelephone | whats with this .call file, new feature? |
| 04:35.39 | [TK]D-Fender | joelsolanki, Why 1/1? Where do you see something that would give you the impression that an encoder/decoder was in use? |
| 04:35.44 | MrTelephone | polycom 501's can't autopickup right? |
| 04:35.54 | drmessano | Ehh |
| 04:35.57 | [TK]D-Fender | MrTelephone, Sure they can. All Polycom IP's |
| 04:36.00 | drmessano | .call = new? Uh no |
| 04:36.04 | JT | autopickup |
| 04:36.06 | joelsolanki | 1/0 or something. because xlite is sending g711 and asterisk will convert to g729 |
| 04:36.08 | joelsolanki | correct ? |
| 04:36.11 | JT | with a klaxton horn recording |
| 04:36.38 | [TK]D-Fender | joelsolanki, You seem to assume that * was TOLD to convert. Who says you told ANY of your peers they were allows to USE G.729 in the first places? |
| 04:36.50 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 04:37.02 | [TK]D-Fender | joelsolanki, Go look at your actual peers and actual calls. |
| 04:37.10 | joelsolanki | ok let me check plz |
| 04:38.59 | joelsolanki | in my test peer it is set to accept all codecs |
| 04:39.08 | joelsolanki | and provider peer i have setup to go only g729 |
| 04:39.22 | joelsolanki | so that should translate the call to g729 |
| 04:39.32 | *** join/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 04:40.03 | [TK]D-Fender | joelsolanki, pastebin your configs and the SIP debug of a call from beginning to end. |
| 04:40.14 | joelsolanki | ok 1 sec |
| 04:40.56 | [T]ank | i upgraded to 1.4.19 and a bug still persists. I dont see anything in a bug status, wanted to see if anyone else knows about this first. |
| 04:40.56 | [T]ank | when i get a call from queue and transfer it to another agent, i cannot receive another call until that first transferred call is hung up by the agent I transferred to. |
| 04:41.01 | joelsolanki | this box is on private IP. behind DMZ |
| 04:42.08 | drmessano | UGh |
| 04:42.13 | drmessano | thats your problem |
| 04:42.50 | [TK]D-Fender | joelsolanki, read up : |
| 04:42.50 | drmessano | Take it out of the DMZ, open the proper ports, and set the NAT settings correctly |
| 04:42.51 | [TK]D-Fender | ~sipnat |
| 04:42.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:43.06 | [TK]D-Fender | [T]ank, ANCIENT news. |
| 04:43.33 | joelsolanki | ok i will checkout |
| 04:43.38 | luke-jr | notes DMZ is basically 'all ports forwarded' |
| 04:43.45 | [TK]D-Fender | [T]ank, transfers by any other means that * controlled DTMF will leave the agent as "on-call" until the transferred channel terminates |
| 04:43.59 | drmessano | DMZ is a pile of poo for setting NAT |
| 04:44.16 | [TK]D-Fender | notes the DMZ is only a SMALL part of the job and even then, overkill |
| 04:45.00 | luke-jr | notes "DMZ" is natural with NAT except when people assume NAT is supposed to be a firewall. |
| 04:45.15 | drmessano | notes that no one asked luke-jr for a networking class |
| 04:45.32 | MrTelephone | nat is a good firewire? why not? |
| 04:46.17 | drmessano | A NAT is not a firewall |
| 04:46.21 | MrTelephone | it saved me from the blaster worm quite a few times |
| 04:46.38 | luke-jr | MrTelephone: coincidence |
| 04:46.41 | [TK]D-Fender | luke-jr, pretty valid way of looking at it... I like many kinda leave it as "super basic" firewall in some sense... |
| 04:47.08 | MrTelephone | yeah its a good firewire for your basic home user |
| 04:47.19 | drmessano | "firewall" |
| 04:47.22 | luke-jr | learn the difference between firewire, firewall, and NAT |
| 04:47.30 | MrTelephone | hahahaha |
| 04:47.33 | MrTelephone | did i spell that |
| 04:47.35 | drmessano | NAT != firewall |
| 04:47.40 | MrTelephone | firewire... |
| 04:47.49 | MrTelephone | don't argue with me I'll win |
| 04:47.50 | MrTelephone | :P |
| 04:47.53 | luke-jr | ⦠|
| 04:48.03 | luke-jr | I don't think you can win that one |
| 04:48.07 | drmessano | MrTelephone: Well aware of what it's like to argue with you |
| 04:48.32 | MrTelephone | network address translation has a side effect of being a firewall |
| 04:48.41 | drmessano | LOL |
| 04:48.57 | [TK]D-Fender | MrTelephone, For only very little fires ;) |
| 04:49.07 | MrTelephone | hahaaa |
| 04:49.09 | MrTelephone | yeah |
| 04:49.39 | MrTelephone | i mean it keeps a lot of those little open ports that people don't know about from being attacked all the time |
| 04:49.41 | [TK]D-Fender | I think we can leave it at that.... "side effect" sums it up as far as intent is concerned. |
| 04:49.48 | MrTelephone | sure |
| 04:50.06 | luke-jr | MrTelephone: NAT does not inherently block anything |
| 04:50.13 | MrTelephone | if you don't agree with me switch your television to Lonestar channel.. commando is on |
| 04:50.35 | JT | nat makes it pretty hard to setup arbitary inbound connections to a pc on a private address |
| 04:50.36 | drmessano | NAT is just one piece of the puzzle.. it does not block anything |
| 04:50.38 | luke-jr | a NAT would be 100% correct in automatically detecting where to forward ports |
| 04:51.07 | luke-jr | eg, forward the initial SYN packet to every leased private IP and connect with the first to respond |
| 04:53.48 | MrTelephone | ? |
| 04:54.02 | [TK]D-Fender | luke-jr, that would be ip_conntrack_russianroulette right? ;) |
| 04:54.11 | [T]ank | [TK]D-Fender: can you explain that a little more dumb down? I did not understand. i am transfering using the transfer button on the polycom 301 |
| 04:54.21 | luke-jr | [TK]D-Fender: if there were multiple machines with that port open, yeah ⺠|
| 04:54.40 | MrTelephone | good luck with that then |
| 04:54.42 | luke-jr | [T]ank: basically, the transfer button on your phone will not work with that feature ⺠|
| 04:54.55 | [T]ank | ok... so how do others do it? |
| 04:55.01 | [TK]D-Fender | [T]ank, exactly. The queue is too dumb to know that you transferred the call so it thinks you're busy. If you want to avoid this you MUST use an * based DTMF transfer. |
| 04:55.10 | MrTelephone | im just referring to nat routers.. they have extra functionality to drop packets that are not meant for the computers behind the nat |
| 04:56.12 | [TK]D-Fender | MrTelephone, Or more like "drop packets not in response to a NAT'd outbound request and not forwarded explicitly which is never assumed." |
| 04:56.27 | [T]ank | [TK]D-Fender: how do i do a DTMF transfer? |
| 04:56.32 | luke-jr | MrTelephone: that's not extra functionality, its mere coincidence or (arguably) a bug ⺠|
| 04:56.39 | MrTelephone | hahaa |
| 04:57.05 | [TK]D-Fender | [T]ank, read up on features.conf and "core show application queue" |
| 04:57.09 | MrTelephone | anyways when ipv6 makes it big say goodbye to nat and everyone is going to get the blaster worm |
| 04:57.32 | [TK]D-Fender | MrTelephone, I don't think blaster is IPv6 compatible ;) |
| 04:57.33 | [T]ank | [TK]D-Fender: will do, thank you |
| 04:57.33 | MrTelephone | your computer will now restart in 59 seconds |
| 04:57.39 | MrTelephone | hahah |
| 04:57.55 | [TK]D-Fender | MrTelephone, pwned by its own mediocrity. |
| 04:58.52 | luke-jr | MrTelephone: you mean every Windoze user |
| 04:59.03 | MrTelephone | yeahf |
| 04:59.08 | MrTelephone | service pack 1 |
| 04:59.12 | MrTelephone | yay |
| 04:59.16 | luke-jr | and I totally support letting Windoze users get it :þ |
| 04:59.23 | luke-jr | especially if they're still vulnerable |
| 04:59.45 | MrTelephone | its just another example why war will never end |
| 04:59.48 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 04:59.51 | MrTelephone | people are brutal |
| 05:00.10 | khronos | Hi guys. |
| 05:00.23 | drmessano | Yeah.. because that argument makes sense.. Nothing to do at all with end users.. it's all about the OS |
| 05:00.36 | khronos | Trying to hook up a Linksys spa3102 to my Asterisk server. Outgoing calls work fine. |
| 05:00.40 | khronos | Incoming calls don't though. |
| 05:00.42 | drmessano | yawns at another "Windows is so insecure, Linux FTW" discussion |
| 05:00.51 | MrTelephone | i mean it only takes one person to make a big bad rpc worm.. |
| 05:00.58 | khronos | The device picks up and issues a dial tone instead of handing the user off to the pbx for processing. |
| 05:01.01 | MrTelephone | like a dictator |
| 05:01.13 | drmessano | Except the RPC holes have been closed for 3 years.. |
| 05:01.18 | MrTelephone | hahaha |
| 05:01.23 | [TK]D-Fender | drmessano, the line between "criminally stupid" and "insecure" isn't always so fine.... |
| 05:01.32 | MrTelephone | i mention it beacuse someone got blasted the other day |
| 05:01.52 | drmessano | [TK]D-Fender: Agreed.. |
| 05:01.53 | MrTelephone | they installed windows xp to wipe their machine out before selling it |
| 05:02.05 | MrTelephone | the new owners plugged it in and they were fucked within a minute |
| 05:02.05 | drmessano | SP0, no doubt |
| 05:02.05 | luke-jr | I know someone who runs Windows 98 with no antivirus |
| 05:02.11 | luke-jr | and has never had a virus infection |
| 05:02.38 | MrTelephone | i got a virus looking for a keygen for avg antivirus, how ironic is that? |
| 05:02.39 | MrTelephone | hahah |
| 05:02.53 | drmessano | Lame is more like it |
| 05:03.08 | MrTelephone | no kidding |
| 05:03.22 | MrTelephone | pretty sad day of events |
| 05:03.29 | [TK]D-Fender | And frankly that vast majority of people on the internet are indeed "sheep". I have clients who have serious trouble comprehending all the bad things that can happen to the,. Thy think "hey my band has all sorts of security so I'm safe, right?" and don't think "Maybe this keylogger I've caught with @#%^ me in the ass as soon as I type my user & pass" |
| 05:03.40 | [TK]D-Fender | bank* |
| 05:03.43 | MrTelephone | then i loaded a virus on purpose to see what happened and i tracked it down to a bunch of kids on an irc server |
| 05:04.24 | drmessano | [TK]D-Fender: You didn't even include the 80% of IT professionals that are complete morons that cause just as many problems as end users |
| 05:04.38 | MrTelephone | i sent an email out today.. your more likely to be victim of fraud from using your card at the store than using webbanking |
| 05:05.03 | [TK]D-Fender | drmessano, Not really.... they are too far outnumbered, but a little more guilty for having a clue and not being able to DO anything with it :) |
| 05:05.06 | luke-jr | only because the script kiddies have a lot more people to pick from |
| 05:05.13 | drmessano | MrTelephone: I got that e-mail from you, but I deleted it, knowing that you're prone to infect your own PC more than my grandmother |
| 05:05.20 | MrTelephone | hahaha |
| 05:06.06 | MrTelephone | i always enjoyed trying to get rid of them afterwards though |
| 05:06.07 | [T]ank | [TK]D-Fender: so if i add the "T", and enable the #1 in features.conf, I can make this work? did I read this right? |
| 05:06.23 | [TK]D-Fender | [T]ank, largely |
| 05:06.27 | MrTelephone | especially the ones that auto-generate .dll's everytime you try and delete the runtime service |
| 05:06.56 | [T]ank | ok, gonna try it |
| 05:06.59 | MrTelephone | if you make windows/system32 not writable by SYSTEM it will crash the trojan and stop making dll's |
| 05:07.35 | MrTelephone | file replication trojans = dead harddrive in a week |
| 05:07.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:07.58 | MrTelephone | oh well |
| 05:08.08 | MrTelephone | log into my bank account.. you need a code to add a payee |
| 05:08.22 | MrTelephone | you can only add credit cards to payees online |
| 05:08.28 | luke-jr | what kind of code? |
| 05:08.35 | MrTelephone | like a 4 digit number |
| 05:08.48 | luke-jr | ok, please add a payee with my keylogger installed |
| 05:09.07 | MrTelephone | then what are you going to add |
| 05:09.12 | MrTelephone | yoru visa number so you can be tracked? |
| 05:09.25 | drmessano | luke-jr: naah.. just send a message to the bank's support asking to have the access code reset, from within the logged in account |
| 05:09.38 | MrTelephone | nice |
| 05:09.44 | MrTelephone | whats my moms maiden name |
| 05:09.47 | MrTelephone | birthdate |
| 05:09.49 | MrTelephone | address |
| 05:09.52 | drmessano | Don't need it |
| 05:09.56 | drmessano | I am logged in as you |
| 05:10.00 | drmessano | Already authed |
| 05:10.12 | MrTelephone | don't matter they have policies to prevent that kind of attack |
| 05:10.16 | drmessano | LOL |
| 05:10.18 | drmessano | sure they do |
| 05:10.20 | MrTelephone | hahaha |
| 05:10.34 | drmessano | Banks are SO secure from social engineering attacks |
| 05:10.54 | drmessano | Really |
| 05:11.00 | drmessano | You go to the support box |
| 05:11.02 | MrTelephone | debit card fraud is big right now |
| 05:11.09 | MrTelephone | fake swipe machines at retail outlets |
| 05:11.23 | drmessano | Because most have an Email type app inside the account pages |
| 05:11.31 | luke-jr | ROFL |
| 05:11.35 | MrTelephone | one guy in toronto got busted with a caseload of blake magnetic stripe cards |
| 05:11.40 | luke-jr | email is 100% insecure |
| 05:11.52 | drmessano | HI MY NAME IS BLAH AND I LOST MY ACOUNT CODE AND I NEED IT RESET TO 3165 THANK YOU |
| 05:11.54 | MrTelephone | then you have to bust into the guys email |
| 05:11.57 | drmessano | Just like a newb |
| 05:12.11 | luke-jr | MrTelephone: nope |
| 05:12.16 | luke-jr | MrTelephone: anyone can read your emails |
| 05:12.21 | MrTelephone | THATS WHY MY EMAIL DIDN"T WORK the other day |
| 05:12.21 | drmessano | Support for the bank gets it |
| 05:12.22 | luke-jr | they are not secure in the slightest |
| 05:12.31 | drmessano | They process it |
| 05:12.33 | MrTelephone | hahah |
| 05:12.40 | drmessano | ALL CAPS looks like a 80 yr old man |
| 05:12.44 | drmessano | or a soccer mom |
| 05:12.46 | drmessano | or whoever |
| 05:12.52 | drmessano | So they push it right thru |
| 05:13.00 | MrTelephone | ALL CAPS lol |
| 05:13.08 | drmessano | and since it came from YOUR logged in account.. no reason to question it |
| 05:13.10 | MrTelephone | like a 13 year old with hyperactive disorder |
| 05:13.28 | MrTelephone | i've been through it before |
| 05:13.30 | drmessano | A lot of computer illiterate folks type in all caps.. or those that dont see well |
| 05:13.34 | MrTelephone | had to phone the bastards to reset my webpass |
| 05:13.42 | drmessano | Trust me.. you know nothing about this |
| 05:14.07 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 05:14.07 | MrTelephone | that reminds me of this kid who said he hacked a bank with notepad.exe |
| 05:14.07 | C4colo | aolers use all-caps |
| 05:14.22 | MrTelephone | computer people know everything though |
| 05:14.38 | C4colo | notepad? seriously? he was using windows? |
| 05:14.48 | MrTelephone | probably |
| 05:14.51 | C4colo | what kind of l33t haxx0r uses windows? |
| 05:14.51 | drmessano | "blah is so secure because it took me two hours to reset my password" |
| 05:14.53 | MrTelephone | this was 15 years ok |
| 05:15.13 | MrTelephone | ok = ago |
| 05:15.17 | C4colo | heh, well banks weren't that secure back then, so mabye |
| 05:15.23 | sbingner | drmessano, too bad they're not like us and just delete anything in all caps |
| 05:15.34 | drmessano | lol |
| 05:16.07 | MrTelephone | some little hackers brut forced their way into one of my linux boxes due to a weak password |
| 05:16.24 | MrTelephone | and i followed the little bitches to their irc channel and cussed them out |
| 05:16.26 | sbingner | wow that musta been an awfully weak pw |
| 05:16.34 | MrTelephone | upload/upload123 |
| 05:16.49 | drmessano | ALL CAPS is not the only way.. but a well crafted message for support is going to bring more red flags than looking like some n00b who tried their best to ask for a password reset and can barely sign on AOL |
| 05:16.52 | sbingner | MrTelephone, somebody got on my box once and installed an eggdrop. I took over their botnet and held their channel for a month |
| 05:16.58 | MrTelephone | not you can't login without a rsa key |
| 05:17.06 | MrTelephone | hahah sbingner, |
| 05:17.26 | MrTelephone | i phoned all the providers they used to brute force the servers and half of them responded to my calls |
| 05:17.39 | MrTelephone | i didn't think I'd get any responses |
| 05:17.57 | sbingner | drmessano, true enough |
| 05:17.59 | MrTelephone | sbingner, were they on efnet.org? |
| 05:18.11 | sbingner | MrTelephone, #playstation on efnet |
| 05:18.30 | MrTelephone | yeah same here.. guys from romania |
| 05:18.49 | sbingner | not the same channel I hope? lol |
| 05:19.10 | MrTelephone | no, #uptime and a bunch of other ones |
| 05:19.24 | drmessano | While I do think Kevin Mitnick got MUCH more credit than he deserved, since he was more of a social engineer than a hacker, a lot of what he's written still holds true today in this so-called secured world |
| 05:19.29 | C4colo | I had an old windows server 2000 box on my network once that got hit, I sniffed the traffic and found the irc channel username password etc and logged in, there were hundreds of these bots in there |
| 05:19.45 | sbingner | drmessano, social engineering is the easiest way to break into anything |
| 05:19.48 | MrTelephone | i still have a config from their energymech with a blowfish encrypted password.. don't know if i want to invest the time to try and crack it or not |
| 05:20.08 | sbingner | MrTelephone, blowfish? lol that takes like 8 hours |
| 05:20.12 | drmessano | sbingner: and it will NEVER get any harder unless humans evolve.. got 2 million years? |
| 05:20.23 | C4colo | sbingner: true, almost all security breaches are via people... employees, contractors, etc |
| 05:20.24 | drmessano | blowfish, lol |
| 05:20.33 | C4colo | and idiot IT companies |
| 05:20.41 | sbingner | nods |
| 05:20.44 | drmessano | People are easy to manipulate |
| 05:20.44 | MrTelephone | c4colo, did you take over? |
| 05:20.48 | C4colo | I worked in a doctor's office that was using WEP on their wireless |
| 05:21.02 | MrTelephone | sbingner, i have to read and download the tools to do it i guess.. i dunno |
| 05:21.17 | C4colo | MrTelephone, no I just reinstalled the OS and thought about ways to do horrible things |
| 05:21.31 | drmessano | I bet one could crack blowfish with notepad... or maybe DOS edit |
| 05:21.34 | [TK]D-Fender | drmessano, people evolve? No there is another solution. |
| 05:21.43 | sbingner | MrTelephone, omg I forgot the name of the program I used for testing pws wth |
| 05:21.46 | drmessano | [TK]D-Fender: Get rid of them completely? |
| 05:21.55 | C4colo | when I asked them about it, they told me the company that set it up ... get this, the wiring closet was 10 feet away from the workstation, over a suspended ceiling ... could have run a patch cable |
| 05:21.56 | MrTelephone | i dunno but that rainbow crack program sounds awesome |
| 05:22.10 | [TK]D-Fender | drmessano, time to cull the heard. People become proportionately smarter as the herd is culled ;) |
| 05:22.10 | sbingner | MrTelephone, hahah thats overkill for blowfish |
| 05:22.11 | C4colo | but the installed a crappy wifi router and a crappy (very crappy) wifi card |
| 05:22.31 | sbingner | MrTelephone, try john the ripper |
| 05:22.32 | drmessano | [TK]D-Fender: HAH... Must.. implement... Operation Darwin |
| 05:22.34 | MrTelephone | and charged like 1000 for doing the job probably |
| 05:22.39 | [TK]D-Fender | drmessano, :D |
| 05:22.47 | C4colo | probably more than that |
| 05:22.57 | C4colo | my god, these people are idiots |
| 05:23.00 | C4colo | and these are the pros |
| 05:23.02 | [TK]D-Fender | My company is DUMB with several of its most important resources. |
| 05:23.06 | C4colo | the trusted sources for IT |
| 05:23.15 | [TK]D-Fender | I want to kick several peoples ass at head-office... |
| 05:23.38 | MrTelephone | i find IT people always dis each others work though |
| 05:23.52 | MrTelephone | its always easy to improve on the base of someone elses work |
| 05:24.09 | drmessano | The company I work for now has done some amazing things for customers who were utterly screwed over by their previous IT company/person.. Couldn't even begin to tell you the crap some people have done. |
| 05:24.20 | sbingner | MrTelephone, there we go "Cain and Abel" is the one I liked |
| 05:24.33 | C4colo | oh, the other thing was, for that company I was doing some database translation and I called their accounting company (hipa certified) and asked them to send me the database of all of the clients of the local doctors office, they asked me what email and I gave them my gmail account... without any verification they emailed it off to me |
| 05:24.45 | MrTelephone | sbingner, yeah same.. it does the blowfish thing? |
| 05:24.53 | drmessano | MrTelephone: 80% of IT workers are INCOMPETENT.. That comes from a recent study. I firmly believe that. |
| 05:25.03 | [TK]D-Fender | MrTelephone, 64bit WEP to our LAN, and open telnet to our AS/400 which runs our corporate logistics SW. bad enough yet? |
| 05:25.04 | Nugget | telnet is eeeeeeevil! |
| 05:25.08 | drmessano | That's why I am listening to only 20% of you |
| 05:25.14 | sbingner | MrTelephone, not sure I think so.. I know john the ripper has blowfish support |
| 05:25.19 | C4colo | [TK]D-Fender: so the number has gotten better recently then? |
| 05:25.20 | MrTelephone | JD EDWARDS? |
| 05:25.25 | sbingner | isn't blowfish the original crypt()? |
| 05:25.27 | [TK]D-Fender | MrTelephone, yup |
| 05:25.29 | *** part/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 05:25.30 | MrTelephone | hahaa |
| 05:25.37 | MrTelephone | yeah thats not very good |
| 05:25.53 | drmessano | ZOMG |
| 05:26.02 | drmessano | 64-bit WEP.. didn't DOS support that |
| 05:26.03 | MrTelephone | 80% of it people are incompetent is pretty vague |
| 05:26.04 | C4colo | I only use ROT13 for my wifi |
| 05:26.17 | MrTelephone | i think people get conned into doing jobs they can't handle |
| 05:26.25 | MrTelephone | or they lie on their resume |
| 05:26.35 | drmessano | MrTelephone: Thats incompetence |
| 05:26.37 | C4colo | you don't have rot13 on your backtrack2 disc do you? |
| 05:26.39 | MrTelephone | or they get hired because they have computer science in college? |
| 05:26.41 | C4colo | lol |
| 05:26.48 | drmessano | If you get into a job you can't do, you're an effing idiot |
| 05:26.56 | [TK]D-Fender | MrTelephone, Yeah.. they piss me off. I have a Linux server for which I asked if they could configure LOTUS DOMINO (GAH!!!!) on our side to poll a box via Dovecot POP3 to pick up my server logs @ distribute for "IDS & general upkeep notices". Dumb-fucks don't seem to understand the word "prevention". |
| 05:27.05 | MrTelephone | if you work for a company who has a strict procedure in place you just follow directions. how smart do you have to be? |
| 05:27.37 | [TK]D-Fender | drmessano, Correction, the HR person choosing YOU is an idiot, and you're just in for a ride :) |
| 05:27.42 | drmessano | LOL |
| 05:27.55 | [TK]D-Fender | drmessano, Don't misdirect blame, there's plenty to go around! |
| 05:28.14 | drmessano | Well, I think part of being intelligent is knowing what your strengths and weaknesses are.. |
| 05:28.34 | drmessano | and to lack that, shows a lower level of intelligence |
| 05:28.56 | drmessano | "Oh, a Citrix metaframe? Yeah, I used SSH once, this should be a snap |
| 05:29.07 | [TK]D-Fender | drmessano, and to continue in blatant disregard of your weaknesses is a sign of flagrant stupidity. |
| 05:29.25 | MrTelephone | if you think your the smartest guy around you should be teaching your knowledge |
| 05:29.36 | MrTelephone | not gloating about it and being condescending to others |
| 05:30.06 | drmessano | Agreed.. Which is where that 80% incompetence comes from. Those same people who think you're a whiz because you can reboot windows are the ones who think THEY are a whiz because they can reboot windows, and pretty soon, they're looking for an IT job. |
| 05:30.08 | MrTelephone | when i feel overworked I take a step back and find out what I can teach my apprentice so I can do less |
| 05:30.22 | MrTelephone | ohhh |
| 05:30.24 | [TK]D-Fender | MrTelephone, Can't teach those that don't want to learn and if you pull the "I just haven't found the way to reach them" card, I've considered a few means of doing that, but all seem "malicious" |
| 05:30.40 | MrTelephone | yeah i understand you guys |
| 05:30.52 | [TK]D-Fender | MrTelephone, And would put my ass under a very unfriendly electron microscope at head-office. |
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| 05:31.13 | MrTelephone | i know a principal of a school who fails to sit and learn things computerwise... yet she is teaching our children and also thinks she is underpaid |
| 05:31.31 | [TK]D-Fender | MrTelephone, I'm a newb to Linux & I guess I could say even networking to some degree. But I'm constantly learning,a nd evolving fast.... |
| 05:31.58 | MrTelephone | at least your learning |
| 05:31.59 | sbingner | drmessano, lol |
| 05:32.38 | [TK]D-Fender | MrTelephone, these guys are dead-end burnouts who live with the "if it ain't broke" mentality and and ceased to innovate, or renovate, only repair when it breaks and aren't equiping themselves to find & stop pyromaniacs. |
| 05:33.17 | [TK]D-Fender | Today was a great step for me at home, more to come shortly... |
| 05:33.34 | MrTelephone | what are you talking about a subnet.. you rented a subnet from your isp? |
| 05:33.39 | [TK]D-Fender | As far as * is concerned I have to really start working with AGI, then real-time and other similar bits. |
| 05:34.24 | MrTelephone | it never ends |
| 05:34.27 | [TK]D-Fender | MrTelephone, yup, I had a fixed IP, now I route a subnet. Added a NIC for a local LAN (DHCPD'd), 2nd NIC for 1st IP of range, and iptables from scratch to route it all |
| 05:35.06 | [TK]D-Fender | MrTelephone, my LAN NAT's as my ppp0 and the next is to see if I can mangle it as another IP in my subnet. |
| 05:35.06 | MrTelephone | networking is fun stuff and opens a lot of doors |
| 05:35.18 | [TK]D-Fender | MrTelephone, All about traceability. |
| 05:35.31 | [TK]D-Fender | MrTelephone, Closing open doors is what networking is about right now :p |
| 05:35.48 | [TK]D-Fender | </irony> |
| 05:36.35 | [TK]D-Fender | ok, far too late again. Bed time. back in the morning.... |
| 05:36.37 | MrTelephone | you would have to mangle it with an ip of an interface on your systme no? |
| 05:36.38 | [TK]D-Fender | *yawn* |
| 05:36.55 | [TK]D-Fender | MrTelephone, Somthing like that I think, will be doing more reading shortly. |
| 05:36.59 | [TK]D-Fender | ok, I'm off. |
| 05:37.21 | MrTelephone | good night drmessano |
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| 05:51.15 | sione | Anyone know why Asterisk 1.6 does not work with CallCentric provider? |
| 05:51.47 | sione | seems to be a problem with the invite packet |
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| 06:55.36 | jeev | i'm ready to get a few voip phones.. |
| 06:55.49 | jeev | looking at polycom 330.. but i need something with 2 port ethernet + power, i can't do POE |
| 06:56.40 | jeev | any recommendations ? |
| 06:56.56 | JT | you could use the IP330, but you'd need to get a power supply for each one too |
| 06:57.00 | jeev | ya |
| 06:57.04 | jeev | got anything else in mind ? |
| 06:57.13 | jeev | i'm looking at amazon.com.. return policy + amazon prime |
| 06:57.17 | jeev | or newegg |
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| 06:57.44 | JT | telephonydepot.com is meant to be pretty good |
| 06:57.53 | jeev | yea |
| 06:57.56 | jeev | i will get from there eventually |
| 06:58.00 | jeev | i want to get from somewhere else and test |
| 07:00.00 | jeev | so i CANT connect my nortels to an adapter > ethernet.. right ? |
| 07:00.59 | JT | not cheaply |
| 07:01.08 | jeev | shit, a little spider dangled in front of me |
| 07:01.11 | jeev | i flicked it like 9 times |
| 07:01.16 | jeev | it's like almost dead, it fell on a cdr |
| 07:01.17 | jeev | lol |
| 07:01.27 | jeev | poor thing |
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| 07:05.55 | jeev | JT, i think i can find the adapter somewhere local.. it's a normal adapter.. |
| 07:08.52 | JT | normal adapter? is it just a plain analogue telephone? |
| 07:09.03 | jeev | no |
| 07:09.05 | jeev | no |
| 07:09.06 | jeev | sorry |
| 07:09.08 | jeev | i was talking about the power. |
| 07:09.16 | jeev | you ever heard of this ? http://www.newegg.com/Product/Product.aspx?Item=N82E16833365001 |
| 07:09.35 | JT | yeah it's just 24V |
| 07:09.49 | JT | if you have the right dc jack, should work |
| 07:10.14 | jeev | ever heard of them ? |
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| 07:12.01 | JT | the site is not loading for me |
| 07:13.13 | jeev | http://www.snom.com/en/snom300_voip_phone0.html |
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| 07:15.15 | JT | lol a Snom 300 is waay downmarket of an IP330 |
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| 07:15.55 | jeev | really |
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| 07:18.10 | jeev | http://www.telephonydepot.com/ProductDetails.asp?ProductCode=105-059-300 |
| 07:18.11 | jeev | without POE |
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| 07:18.16 | jeev | so does it come with the fuckin power cable or not |
| 07:18.20 | jeev | i fuckin hate stupid descriptions |
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| 07:33.56 | Telemac | Hello |
| 07:37.23 | Telemac | Could anyone give me any clue about an misdn error http://openpaste.org/en/6124/ ? |
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| 08:14.59 | mort_gib | Any procedure on * updating, like moving from 1.4.17 -> 1.4.19 |
| 08:15.10 | mort_gib | There seems to be issues with the modules |
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| 08:19.30 | rolek | mort_gib: I haven't seen any trouble so far, after an upgrade from 1.4.15 to 1.4.19 |
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| 08:20.27 | mort_gib | I just spotted some complaints about modules "not compiled for this version...bla bla" |
| 08:21.06 | ibob63 | I'm trying to get statistics on the calls (lenght, number of incoming and outgoing calls ) does anyone know some statistics software for asterisk? |
| 08:21.14 | rolek | mort_gib: I've seen that, when compiling asterisk-addons on a machine which had an older vserion of asterisk installed. |
| 08:22.14 | rolek | mort_gib: Do you install from source? |
| 08:22.24 | mort_gib | Yes.... |
| 08:22.56 | mort_gib | I recon I can just copy the old modules somewhere else when I compile |
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| 08:23.37 | rolek | mort_gib: Trouble is with old header files and/or libraries. |
| 08:24.04 | mort_gib | Well, I have downloaded * 1.4.19, and latest version of addons |
| 08:24.12 | mort_gib | Like latest version of Zaptel |
| 08:24.17 | rolek | mort_gib: I had to compile asterisk-addons with include dirs explicitly pointing to my freshly compiled asterisk source |
| 08:24.56 | rolek | mort_gib: But easiest was is probably compile *, the upgrade, then compile *-addons. |
| 08:24.59 | mort_gib | Yeah, how?? |
| 08:26.11 | mort_gib | That's what I wanted to do |
| 08:27.17 | mort_gib | 1. compile zaptel |
| 08:27.25 | mort_gib | 2. compile * |
| 08:27.32 | mort_gib | 3. compile*-addons |
| 08:27.46 | mort_gib | 4. (in my case) compile Sangoma's Wanpipe |
| 08:28.27 | rolek | mort_gib: Maybe you have old header files or libraries around, after step 2? |
| 08:28.58 | rolek | mort_gib: I compile on separate host, without asterisk installed, so my procedure might not work for you, but I use |
| 08:29.20 | mort_gib | Well, I DO make clean first |
| 08:29.29 | mort_gib | if that makes any difference |
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| 08:32.45 | rolek | make ASTERISK_INCLUDE="-Iinclude -I../include -I/usr/src/asterisk-1.4.19/include" |
| 08:33.26 | rolek | mort_gib: With this flag I make sure asterisk-addons uses include files from my freshly compiled asterisk source. |
| 08:33.48 | mort_gib | in /usr/src/asterisk-addons-1.4.6 of course |
| 08:34.04 | rolek | mort_gib: The trouble is probably not in your source, but in the fact that you still have old header files in /usr/include/asterisk. |
| 08:34.12 | rolek | mort_gib: yes, indeed |
| 08:34.18 | mort_gib | Thanks :-) |
| 08:34.27 | rolek | mort_gib: good luck |
| 08:34.39 | mort_gib | I'll let you know |
| 08:36.04 | Telemac | Could anyone give me any clue about an misdn error http://openpaste.org/en/6124/ ? |
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| 08:39.42 | saftsack | hi, is it possible to replace the unavailable message in the voicemail system via console? |
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| 08:50.22 | Dr-Linux | adeel :) |
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| 09:24.50 | masus | hi all, does anyone know how to adjust the voice volume from asterisk ? |
| 09:27.20 | mort_gib | rolek: Seemed to work fine |
| 09:27.23 | mort_gib | Thanks |
| 09:27.31 | rolek | mort_gib: Welcome. |
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| 09:31.52 | BrokenNoze | Hi, I am having some probelms compiling my sangoma wanpipe drivers, getting a n error linux/config.h not found. anyone any ideas? |
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| 09:43.39 | rolek | BrokenNoze: Do you have kernel headers installed on the machine on which you are compiling? |
| 09:44.01 | BrokenNoze | rolek: yes, and kernel-devel packages |
| 09:44.24 | BrokenNoze | asterisk, zaptel and libpri seem to have built fine |
| 09:45.06 | rolek | BrokenNoze: Maybe you need to define additional include dirs? |
| 09:45.27 | BrokenNoze | how do i do that? |
| 09:45.47 | BrokenNoze | in the Makefile you mean? |
| 09:46.35 | rolek | BrokenNoze: Usually, yes. I'm not familiar with compilation of sangoma drivers specifically. |
| 09:47.03 | rolek | BrokenNoze: which kernel version do you have? |
| 09:47.30 | rolek | BrokenNoze: I noticed my newer 2.6 kernels no longer have the config.h header file. |
| 09:47.31 | BrokenNoze | rolek: well i can't find the linux/config.h file on my machine to be honest. I assuemd as you did it was part of the kernel-devel package |
| 09:47.40 | mort_gib | rolek: Sangoma's ./Setup install process will ask for the kernel header dir |
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| 09:49.54 | rolek | BrokenNoze: IIRC the config.h file was deprecated some kernel versions ago. You might want to create it yourself. |
| 09:50.07 | rolek | BrokenNoze: See http://openpaste.org/en/6148/ for contents. |
| 09:51.32 | BrokenNoze | rolek: so this needs to go in the Makefile, uses autoconf.h instead of config.h? |
| 09:51.38 | rolek | BrokenNoze: Create it in the include/linux directory of wherever your kernel headers are. Might work, might not. |
| 09:51.55 | BrokenNoze | Oh, got you this IS config.h |
| 09:53.16 | rolek | BrokenNoze: Yep, it is. If you create it, you might fool sangoma into using your newer kernel version. |
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| 09:55.51 | BrokenNoze | rolek: Looks like that might have worked! awesome. thanks. seems odd thouigh that kernel-devel doesn't just include a stub config.h like this for backward compatibility anyhow though |
| 09:57.57 | rolek | BrokenNoze: I think they did that for quite a while, but lots of things changed, so there's no guarantee it works. Dlad it did in your case, though.. |
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| 09:59.03 | rolek | Can anyone tell me how I can get a specific * bug reopened? |
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| 10:03.06 | saftsack | where are the custom recorded voicemailfiles saved? |
| 10:05.57 | rolek | saftsack: In my installation they are in /var/lib/asterisk/sounds/custom/voicemail.. |
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| 10:33.20 | l0verb0y | hey |
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| 10:50.41 | exvito | tzafrir: ...are you there ? |
| 10:51.00 | tzafrir | exvito, yes |
| 10:51.30 | exvito | just checking email and need to clarify your email regarding the zaptel 1.4.10 possible regression... |
| 10:51.47 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 10:51.56 | exvito | when you ask if the crash is easy to reproduce... well, it's not exactly a crash in itself |
| 10:52.19 | l0verb0y | does anyone know of any DID management systems? |
| 10:52.28 | exvito | ...its a "soft hangup detected", the system keeps working... however, with zt1.4.10 it always gives me that error / stack trace. |
| 10:53.51 | exvito | i'll now try the svn zaptel version that shaun suggested... do you want me to test anything in particular or where you just suggesting the usage of live_zap to make things easier ? |
| 10:54.01 | exvito | where = were ! |
| 10:55.02 | tzafrir | exvito, sruffel has suggested a specific branch for you to test. |
| 10:55.26 | exvito | tzafrir: yep. that's what i'm about to do in the next half-hour... :) |
| 10:55.30 | tzafrir | The live_zap script is simply intended to save you the bother of actually installing it on your system |
| 10:55.43 | tzafrir | it is in the base directory of the zaptel tarball |
| 10:55.58 | exvito | ok, so it should be a matter of: ./configure && make + run live_zap ? |
| 10:56.46 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:59.49 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:00.06 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 11:08.48 | *** join/#asterisk RoyK (n=roy@91.149.31.152) |
| 11:09.29 | *** join/#asterisk plantseeker (n=chatzill@77.240.56.22) |
| 11:22.34 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 11:23.51 | masus | exit |
| 11:24.05 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-221-161.bflony.east.verizon.net) |
| 11:26.27 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 11:27.02 | *** join/#asterisk Siya (n=djerk@194.60.207.239) |
| 11:27.05 | Siya | re |
| 11:31.06 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:33.04 | rolek | oej: are you there? |
| 11:35.07 | Faustov | hi, could asterisk have a problem if 2 nats have the same addresation, like voipphoneA -> asterisk -> router -> net -> router -> asterisk -> voipphoneB |
| 11:35.17 | Faustov | when both routers got their local network the same? |
| 11:36.52 | oej | rolek: Yes, it feels that way |
| 11:37.53 | rolek | oej: Doesn't sound like you're really convinced.. :) |
| 11:38.03 | rolek | oej: Can I humbly ask you to reopen bug 6240 for me? |
| 11:38.24 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 11:38.50 | rolek | oej: .. or is there a more approriate way to accomplish that? |
| 11:40.25 | SuPrSluG | Faustov:no, the routers see each others public IP. And the routers do the rest |
| 11:41.24 | *** part/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 11:45.04 | Faustov | SuPrSluG: yeah in theory... i just made this work, however i dont understand one thing |
| 11:45.47 | Faustov | on one end i had to enable nat=yes,localnet=xxx,externip=yyy, but on the other one i had to DISABLE that |
| 11:45.55 | Faustov | while both asterisks are behind nat |
| 11:45.56 | Faustov | why? |
| 11:46.36 | *** join/#asterisk Dr-Linux (n=somethin@117.20.21.66) |
| 11:46.41 | Siya | SuPrSluG: that is if you have good routers |
| 11:47.31 | oej | M6240 |
| 11:47.37 | Siya | Faustov: NAt devices with support for 'stateful SIP' will translate the IP addresses in the SIP headers as well as the IP ones |
| 11:48.03 | *** join/#asterisk xenonex (n=xenonex@89.218.235.91) |
| 11:48.05 | Siya | for those NAT devices Asterisk doesn't need to know about NAT |
| 11:48.10 | oej | rolek: I could never test that stuff so I gave up on that bug report. |
| 11:48.20 | oej | rolek: Do you have a network where you can test? |
| 11:48.28 | oej | rolek: And see issues? |
| 11:48.52 | rolek | oej: yes I do (have network, see issues) |
| 11:50.06 | Siya | Faustov: what NAT devices (routers in your case) are on either side? |
| 11:50.48 | rolek | oej: I filed bug # 12322 before I found the one you had been working on. |
| 11:51.10 | rolek | oej: Probably it's a duplicate. |
| 11:51.59 | *** join/#asterisk jkirby (n=jkirby@196.25.36.129) |
| 11:55.54 | oej | rolek: I'm just adding a relationship between them |
| 11:56.00 | oej | I wonder if I still have that code around |
| 11:56.38 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 11:56.45 | oej | I guess I cleaned it away. We should be able to find it a few versions back in subversion |
| 11:57.57 | tzafrir | exvito, here? |
| 11:58.24 | rolek | oej: I've been trying to find it myself, but so far I've had no luck. |
| 12:01.25 | oej | rolek: http://svn.digium.com/view/asterisk/team/oej/strictrouting/?pathrev=50000 |
| 12:01.59 | oej | http://svn.digium.com/view/asterisk/team/oej/strictrouting/?pathrev=64190 |
| 12:02.03 | oej | Was the latest revision |
| 12:02.36 | rolek | oej: Thanks. I'll see if I can get those code changes to apply to 1.4.19. |
| 12:03.40 | rolek | oej: I'll post the results in bug 12322, after I've tested it. Probably tomorrow. |
| 12:04.12 | oej | rolek: Ok |
| 12:06.46 | hwt | hi. this isn't strictly asterisk talk, but has anyone gotten firmware upgrades through autoprovisioning to work on thomson st-2030? |
| 12:06.53 | hwt | ..to work. :) |
| 12:07.24 | hwt | i always get Error: Wrong firmware configuration! in the APS log. |
| 12:08.23 | glaz | hwt: http://www.voip-info.org/wiki/view/Thomson+ST2030 |
| 12:09.21 | hwt | glaz: i have done everything according to docs. |
| 12:09.41 | hwt | everything works, except firmware upgrades |
| 12:09.51 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.51 | glaz | Then your phone must be broken if you have done everything. |
| 12:10.26 | hwt | glaz: you completely rule out bugs in the thomson firmware. interesting. :) |
| 12:10.46 | hwt | glaz: and it's not only this phone, but all of them. |
| 12:10.58 | glaz | So you're doing something wrong |
| 12:12.22 | hwt | glaz: or there is a bug in how ST2030 handles HTTP preset provisioning and firmware upgrades, or there are some undocumented limitations. |
| 12:12.37 | hwt | glaz: what i went here for was to hear if anyone else have experienced the same problem. |
| 12:13.35 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:17.32 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 12:18.24 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:20.03 | rolek | hwt: We're using it without problems. Used the docs that hwt mentioned for configuration.. |
| 12:22.04 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:23.55 | *** join/#asterisk nighty^ (n=nighty@p5025-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 12:28.45 | hwt | rolek: what do you specify for [application] fw_filename? i have fw_filename=http:/fqdn/firware/v2030SG.080227.1.59.3.zz |
| 12:29.02 | hwt | rolek: in the INF. |
| 12:29.16 | hwt | rolek: also tried /firmware/foo.zz, foo.zz, etc. |
| 12:30.46 | rolek | hwt: fw_filename=v2030EG.071214.1.58.6.zz |
| 12:31.56 | hwt | rolek: so where do you specify the full url? |
| 12:32.27 | hwt | rolek: and i would appreciate it if you sent me the the INF on a private msg. |
| 12:33.00 | rolek | hwt: ST2030S_00147FE1F4CF.inf is here: http://openpaste.org/en/6149/ |
| 12:34.01 | rolek | hwt: We use pxe and the firmware file is stored in the pxe root. |
| 12:34.26 | rolek | hwt: So we never specifiy a full url at all. |
| 12:34.51 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 12:35.09 | Telemac | Hello |
| 12:35.32 | hwt | rolek: then it's not really the same. i am using HTTP for both config and firmware. |
| 12:36.35 | Telemac | Could anyone give me a hand about misdn ... http://openpaste.org/en/6124/ ? |
| 12:36.41 | hwt | rolek: you know where i can find more docs on the inf files than what is available from the zip-file from thomosn. |
| 12:36.53 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:38.09 | rolek | hwt: Sorry, I don't. |
| 12:43.33 | *** join/#asterisk Modde (n=root@c-9370e255.020-2012-73746f32.cust.bredbandsbolaget.se) |
| 12:46.16 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:47.54 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:47.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:49.51 | Faustov | Siya: sorry for the delay, i got simple linksys router on one side (wan + 4x f/eth switch) and a coresponding model of dlink |
| 12:50.20 | Faustov | Siya: of course with the right ports open, any other firewall tools disabled |
| 12:56.18 | *** join/#asterisk CCFL_Man2 (i=1d218ae2@pool-70-105-212-150.scr.east.verizon.net) |
| 12:56.19 | Siya | Faustov: no I mean that a firewall can translate the IP addresses contained in the SIP header. But not all can do this or are configured to do so by default |
| 12:56.54 | CCFL_Man2 | i got a cingular go phone for a cheap sms account |
| 12:57.02 | Siya | of course you can configure port forwards, either statically or dynamically but that does not mean the NAt process on the router or firewall fixes the SIP header too. |
| 12:57.24 | CCFL_Man2 | err, gsm account, i mean |
| 12:57.57 | Faustov | Siya: i thought routing takes place on IP level, so it doesn't matter if it is sip, tcp or whatever? |
| 12:59.19 | [TK]D-Fender | Faustov: Several D-Link Routers royally screw up NAT. |
| 12:59.43 | [TK]D-Fender | Faustov: That aside, read up : |
| 12:59.45 | [TK]D-Fender | ~sipnat |
| 12:59.46 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:59.47 | [TK]D-Fender | ^^^^^^ |
| 13:00.16 | Siya | :) |
| 13:00.54 | Siya | Faustov: I thought you were talking about a dual NAT setup, if so then you have to consider more than just IP routing |
| 13:01.13 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:01.35 | rob0 | FWIW SIP works great without NAT. :) That's what I do, no cheapo routers in the way. |
| 13:01.46 | [TK]D-Fender | * > NAT > internet < NAT < * isn't a problem. |
| 13:02.16 | Siya | rob0: indeedy. I used to have my * in a datacenter but now I don't have that privilege |
| 13:02.30 | rob0 | I know, it just seems to be a mindset that these routers are always there. |
| 13:02.32 | Siya | and additional public addresses are not cheap on my ADSL2+ provider |
| 13:03.01 | [TK]D-Fender | Siya: I just got a /29 on mine set up last night... |
| 13:03.04 | [TK]D-Fender | 5$ |
| 13:03.08 | Siya | rob0: nor do I use cheap routers :) |
| 13:03.14 | Siya | [TK]D-Fender: nice |
| 13:03.36 | Siya | they want to charge me 10 GBP per month for a /29 |
| 13:03.56 | *** join/#asterisk igascream (n=igascrea@bzq-84-109-83-104.red.bezeqint.net) |
| 13:03.59 | Siya | which afaik is not allowed as a 10,- per month admin cost is a little too high |
| 13:04.06 | [TK]D-Fender | $45 CAD = 5000/800 Kbps ADSL, /29 subnet, unlimited transfer & dry-loop |
| 13:04.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:04.21 | igascream | Need help with busyderect!!!! |
| 13:04.54 | rob0 | nice! |
| 13:04.56 | Siya | 22 GBP (not sure what that is in CAD) gives me ADSL2+ 17400+/1241-ish |
| 13:05.05 | exvito | tzafrir: ...back from lunch. Tell me |
| 13:05.08 | Siya | unlimited and a static IP |
| 13:05.40 | Siya | igascream: you scream indeed, wanna give some more info in a more calm manner? |
| 13:06.02 | lmadsen | GBP? british pounds? |
| 13:06.11 | Siya | sadly my Cisco 877 doesn't do annex-M otherwise I'd have 2.4Mbps uplink |
| 13:06.17 | Siya | lmadsen: yup |
| 13:06.26 | lmadsen | Siya: ah, then you can basically double that number for CAD |
| 13:06.51 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 13:06.57 | Siya | lmadsen: so not that diff then |
| 13:07.05 | Siya | speed versus IP's :) |
| 13:07.21 | Siya | I still need to test my * with NAT config removed from sip.conf |
| 13:07.35 | Siya | my router should be able to do statefull NAT for SIP |
| 13:07.36 | *** join/#asterisk bikr (n=bikr@63.174.231.5) |
| 13:07.54 | bikr | Good morning!! Can anyone tell me what ports to forward on my router to allow external sip? |
| 13:08.13 | Siya | bikr: welcome to the NAT conversation... :) |
| 13:08.18 | Siya | depends on your router |
| 13:08.34 | Siya | start with UDP 5060 and add RTP ports as needed |
| 13:08.52 | tzafrir | exvito, basically: ./live_ast install; and once: ./live_ast config |
| 13:08.56 | bikr | <---watchguard firebox 7050 |
| 13:09.01 | MaliutaLap | the godd think about * is that you can configure the RTP ports |
| 13:09.05 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 13:09.14 | tzafrir | kind of like 'make install' and 'make config' (that you only need to run once) |
| 13:09.55 | tzafrir | then use: './live_ast reload' which stops asterisk, unloads all of the zaptel modules, attempts to load all the needed modules and starts asterisk again |
| 13:10.10 | exvito | tz |
| 13:10.20 | exvito | tzafrir: ...thanks, I had figured it. |
| 13:10.21 | tzafrir | ("attmpts" - it's tricky, as I could not use modprobe there) |
| 13:10.34 | lmadsen | 1 simultaneous call = 2 RTP, and 2 RTCP ports, so setup as appropriate |
| 13:10.46 | Siya | ~sipnat |
| 13:10.47 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:10.57 | Siya | bikr: ^^^^^^ |
| 13:11.05 | Siya | copies [TK]D-Fender :) |
| 13:11.28 | igascream | Asterisk doesn't recognize the busy tones. The config for IL in indicasions.conf are for other operator I have already checked the tone by myself but for some reason it doesn't work I have Kavey Zahav operator in IL if somebody knows. |
| 13:11.47 | bikr | woot tks!! |
| 13:12.53 | Siya | goes back to doing some 'real' work |
| 13:13.00 | tzanger | morning all |
| 13:13.37 | MaliutaLap | tzanger: evening |
| 13:13.44 | MaliutaLap | well actually it's night |
| 13:15.07 | lmadsen | tzanger: morn! |
| 13:15.11 | tzanger | werd |
| 13:15.16 | tzanger | listening to some good vocal trance |
| 13:15.17 | *** part/#asterisk fish_ (n=fish@freigeist.org) |
| 13:15.18 | tzanger | I love ah.fm |
| 13:15.22 | tzanger | perfect work music |
| 13:15.25 | CCFL_Man2 | are there any gsm to pots adapters? like one that accepts your sim card and gives you a pots interface? |
| 13:15.33 | tzanger | 181.fm is pretty good too |
| 13:15.39 | tzafrir | igascream, indications.conf / tonezones are not related to busydetect |
| 13:16.02 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:16.34 | tzafrir | Kavey Zahav (cables) give basically the same tones as Bezeq |
| 13:17.49 | igascream | So it has to work with basic IL configs. |
| 13:18.22 | jackson__ | If an asterisk box is on a dual homed host, (eth0 Internet, eth1 local LAN) and asterisk is bound to 0.0.0.0, do we need to specify the externip and the localnet? |
| 13:19.23 | bikr | does the external ip have to be specified in a conf, if the router is port forwarding? |
| 13:19.53 | jsmith | bikr: It doesn't have to, but you might get strange results |
| 13:20.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:20.44 | bikr | k |
| 13:21.02 | Siya | bikr: depends if the router translates the IP address contained in the SIP header when it does NAT on the IP header... |
| 13:21.22 | Siya | so to be on the safe side (if you don't know) |
| 13:24.06 | [TK]D-Fender | jackson__: if its the border, then no, though I'd specify it anyways.... |
| 13:24.40 | [TK]D-Fender | bikr: If you're behind NAT, YES. |
| 13:24.44 | jackson__ | [TK]D-Fender, Thanks for the response. There will be sip clients on the LAN who will be talking to the asterisk box via the LAN. So as you say, I spose it's a good idea. |
| 13:25.00 | bikr | hmm.. calls are working |
| 13:25.10 | [TK]D-Fender | jackson__: jsut "healthy" I'd say. Don't forget to prevent re-invites. |
| 13:25.17 | bikr | audio from internal to external is working |
| 13:25.21 | bikr | otherway around doesn't seem to be |
| 13:25.34 | bikr | but could be my cell connection |
| 13:25.37 | [TK]D-Fender | bikr: Exactly. So go follow the guide : |
| 13:25.38 | bikr | in this buidling so i'll test later |
| 13:25.39 | [TK]D-Fender | ~sipnat |
| 13:25.39 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:25.43 | jackson__ | [TK]D-Fender, yes, I was beating my head against the wall yesterday due to the reinvites :) |
| 13:25.48 | bikr | looking at guide now |
| 13:25.59 | [TK]D-Fender | jsmith: howdy neighbour. |
| 13:26.14 | [TK]D-Fender | jsmith: Any idea who's coming up my way for next months boot-camp? |
| 13:26.56 | *** join/#asterisk ManxPower (n=manxpowe@220.sub-75-203-100.myvzw.com) |
| 13:27.18 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:27.43 | Katty | if i were wanting to set the outgoing callerid of my PRI, where would i look? |
| 13:27.44 | lmadsen | I think there's someone in MTL teaching now isn't there? |
| 13:27.49 | Katty | jbot: callerid? |
| 13:27.59 | Katty | s/were/was |
| 13:28.08 | [TK]D-Fender | Katty: Set(CALLERID(num)=2128675309) |
| 13:28.27 | Katty | [TK]D-Fender: right before the Dial() command? |
| 13:28.39 | lmadsen | anytime before you dial :) |
| 13:28.40 | [TK]D-Fender | lmadsen: No clue. Also its been common to see some out of town faces show up |
| 13:28.46 | [TK]D-Fender | Katty: yup. |
| 13:28.49 | Katty | [TK]D-Fender: neat. |
| 13:28.49 | lmadsen | [TK]D-Fender: ya... I'm not sure either |
| 13:28.51 | Katty | [TK]D-Fender: thank you deary. |
| 13:28.56 | [TK]D-Fender | Katty: Mew. |
| 13:29.27 | Katty | [TK]D-Fender: would Set(callerid(name)=foobar) also be acceptable? |
| 13:30.55 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-233.usadatanet.com) |
| 13:31.08 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 13:31.23 | Katty | hmmhesays: hai |
| 13:31.27 | Katty | hugs on hmmhesays |
| 13:31.37 | hmmhesays | Good morning |
| 13:32.03 | bikr | yay 4 sip.conf |
| 13:32.10 | bikr | internal and external working now |
| 13:32.23 | bikr | have you guys tried fring? it's pretty good |
| 13:32.43 | bikr | iphone / windows mobile multi messenger client , has sip also |
| 13:32.54 | *** join/#asterisk erwinpogz (n=niwre@67.159.178.21) |
| 13:32.54 | bikr | works with google talk / skype / sip / aim / msn / yahoo / etc.. |
| 13:35.24 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 13:35.56 | Telemac | Could anyone give me a hand about misdn ... http://openpaste.org/en/6124/ ? |
| 13:36.40 | *** join/#asterisk rushowr (n=rushowr@adsl-67-66-238-42.dsl.spfdmo.swbell.net) |
| 13:36.43 | rushowr | Morning all |
| 13:38.52 | *** join/#asterisk af_ (n=getsmart@88-149-230-168.dynamic.ngi.it) |
| 13:39.21 | Katty | jbot: set callerid |
| 13:39.25 | Katty | jbot: callerid |
| 13:39.33 | Katty | jbot: you are just one uninformative bot, mister! |
| 13:39.33 | jbot | I think you lost me on that one, Katty |
| 13:39.40 | Katty | yeah i can tell. |
| 13:39.40 | ManxPower | Katty: CLID name may be transported across the PSTN, but the telco on the other side will ignore the name and look up the name based on number. |
| 13:39.50 | Katty | ManxPower: ahhh. |
| 13:39.53 | Katty | ManxPower: righto then. |
| 13:39.55 | Katty | ManxPower: thank you (= |
| 13:40.31 | ManxPower | Katty: if the call stays on your carrier's network, CLID name you set might be seen on the far end. |
| 13:41.01 | rushowr | ManxPower: Some CLECS _do_ accept CLID name from other networks, depends on what network |
| 13:41.24 | lirakis_work | bikr: iphone and windows mobile client huh? ... i guess i will NEVER EVER in my life see it |
| 13:41.25 | lirakis_work | ;) |
| 13:41.38 | rushowr | Level3's Voip<->PSTN network is one example of a network I've worked with that other CLECS accepted all CLID info without prejudice |
| 13:42.13 | lirakis_work | since.. you know.. just looking at either of those things burns my retinas |
| 13:42.54 | *** join/#asterisk DirtyDD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 13:43.16 | DirtyDD | Does anyone use SLAtrunk command? |
| 13:43.22 | CCFL_Man2 | lirakis_work: i hear level 3 is the shit when it comes to voip to pstn termination |
| 13:43.36 | CCFL_Man2 | err |
| 13:43.39 | CCFL_Man2 | rushowr |
| 13:43.41 | rushowr | most definitely |
| 13:43.44 | rushowr | Man2 |
| 13:43.46 | rushowr | ;-) |
| 13:43.49 | rushowr | but they're EXPENSIVE |
| 13:43.50 | ctooley | s/the shit/shit/ |
| 13:43.58 | rushowr | but yeah, they're a Tier1 provider of bandwidth |
| 13:44.05 | rushowr | so yeah |
| 13:44.18 | lirakis_work | rushowr: level3 is a joke... |
| 13:44.24 | rushowr | if you say so |
| 13:44.25 | CCFL_Man2 | rushowr: i bet a voice t1 from them will cost lots |
| 13:44.34 | lirakis_work | from a network perspective |
| 13:44.42 | ctooley | L3 has severe issues with SIP compliance |
| 13:44.49 | rushowr | never had problems with them |
| 13:45.00 | rushowr | anyway, I'm not here to argue about providers I don't even use anymore |
| 13:45.01 | lirakis_work | for the past month ... EVERY weekend .. a DS3 goes down in a switch site in miami |
| 13:45.05 | CCFL_Man2 | what? wow |
| 13:45.19 | lirakis_work | and every time they are like "uhhh.. huh? what? .. " |
| 13:45.27 | ctooley | 12 different switch platforms and they don't integrate them well. |
| 13:45.29 | rushowr | dunno, I dealt with NY |
| 13:45.35 | lirakis_work | they have a bad fibre mux and dont give a crap |
| 13:45.35 | rushowr | and all my shit was coloc'd |
| 13:45.51 | CCFL_Man2 | for years i hear they were the best |
| 13:46.06 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-114-66.w90-30.abo.wanadoo.fr) |
| 13:46.17 | bougie | hello :) |
| 13:46.17 | ctooley | We have to deal with L3 issues all day long, every day. |
| 13:46.19 | lirakis_work | CCFL_Man2: .. uunet is the best quality badnwitdth ive used |
| 13:46.56 | CCFL_Man2 | their dialup accessisn't as fast as splitrock's was though |
| 13:46.58 | CCFL_Man2 | ahh |
| 13:48.15 | lirakis_work | ctooley: ++ |
| 13:48.54 | bikr | lirakis: yah, my job sorta requires me to carry some sort of pda, and i hate blackberry edge b/s even though their mail client is good, so i carry a blackjack. |
| 13:49.09 | lirakis_work | bikr: i got a n810 recently |
| 13:49.11 | lirakis_work | nice |
| 13:49.14 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:49.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:49.21 | CCFL_Man2 | ewww, blackberry |
| 13:49.23 | erwinpogz | hello, i have problem dialling out, i think i missed some of the extensions.conf, here is my current setup. http://www.nomorepasting.com/getpaste.php?pasteid=14613 can anyone please help. Thanks |
| 13:49.43 | lirakis_work | bikr: thought about waiting for the wimax version .. but the U.S. probably wont get Xhom or any other wimax for like .. 20 years |
| 13:50.08 | CCFL_Man2 | lirakis_work: such is usually typical |
| 13:50.52 | CCFL_Man2 | lirakis_work: why introduce a better technology when the current crappy ones make the carriers lots of money |
| 13:50.53 | bikr | lirakis: probably right |
| 13:51.05 | bikr | lirakis: and 2nd comment, 100% right |
| 13:51.06 | bikr | lol |
| 13:51.24 | lirakis_work | once hsdpa or some other 3g becomes available .. ill just get a tiny little 3g phone with data only and pair it via bluetooth |
| 13:52.07 | CCFL_Man2 | lirakis_work: carriers won't you you do that, it'll prevent them from making lots of money |
| 13:52.09 | DirtyDD | I'm reading through some documentation on SLAtrunk. I'd like to create a "virtual trunk" but here is the problem. My PBX is fed by a PRI line. I'd like to setup the trunk for a specific DID that is routed through the PRI. How can I make the virtual mapping out of a DID instead of a physical channel? |
| 13:52.25 | lirakis_work | .. but .. really i didnt get the n810 as a phone.. if it can replace one thats cool .. but.. i got it as an ultra portable computer... |
| 13:52.42 | lirakis_work | CCFL_Man2: right.. which is why several charge you more for tethering |
| 13:52.45 | lirakis_work | .. scumbags |
| 13:52.57 | bikr | lirakis: grrr@ bluetooth, that is a cursed word to me , i had chan_mobile working flawless on my asterisk , but i just couldn't get a grasp on the configs etc, so i switched to asterisknow and dammit chan_mobile doesn't want to work at all! |
| 13:53.09 | *** join/#asterisk DarthclueWork (n=Darthclu@fw149.northside.isd.tenet.edu) |
| 13:53.20 | CCFL_Man2 | lirakis_work: unless you can enable tethering yourself without them having to charge you for it |
| 13:53.27 | bikr | <--got lucky on my blackjack.. i had a blackberry unlimited data w/e from cingular which included tethering and 750 voice for like 59.99/month |
| 13:53.43 | bikr | when i got my blackjack i convinced the sales agent to leave my bbunltd price alone |
| 13:53.52 | lirakis_work | yeah .. t-mobile doesnt charge extra .. i think att does.. and verizon def. does |
| 13:53.54 | bikr | and not chrage me for that pda plan krap |
| 13:54.00 | bikr | and now bam , unlimited 3g tethering |
| 13:54.01 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 13:54.11 | CCFL_Man2 | bikr: why didn't you just swap sim cards? |
| 13:54.27 | lirakis_work | bikr: what network is your phone on? is it hsdpa? or umts? |
| 13:54.34 | bikr | ccfl_man2: they bill you blindly for tethering with unusually high traffic |
| 13:54.48 | bikr | ccfl_man2: i couldn't just swap the sims because it was a upgrade phone at upgrade price |
| 13:55.02 | CCFL_Man2 | bikr: ahh, ewww |
| 13:55.05 | bikr | lirakis: no idea, it's at&t formally cingular |
| 13:55.21 | lirakis_work | bikr: so.. how do you know its 3g and not gprs or edge? |
| 13:55.22 | CCFL_Man2 | i do hate att |
| 13:55.40 | bikr | lirakis: little 3g icon at top of phone when i'm on 3g lol.. also the speeds are insane when tethered. |
| 13:55.50 | lirakis_work | bikr: hmm thats pretty sick |
| 13:55.50 | bikr | lirakis: when i'm on edge i get a "E" |
| 13:56.06 | bikr | and a "G" when i'm in ocala lol |
| 13:56.16 | bikr | fuggin podunk towns!!!! |
| 13:56.18 | lirakis_work | bikr: .. do you usualyl get 3g ? |
| 13:56.29 | *** join/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 13:56.33 | bikr | lirakis: outside 99% of time inside 5% maybe |
| 13:56.39 | CCFL_Man2 | i got a $20 prepaid go phone for the intention on using the sim in another phone |
| 13:57.04 | bikr | ccfl: was it sim locked? |
| 13:57.35 | RockHound | good day ... is there a way to program an automatic skip of a entry in the dialplan if the extension is not reachable (SIP phone not connected)? |
| 13:57.42 | CCFL_Man2 | bikr: i'm not sure, when i registered it asked for the imei |
| 13:57.47 | lirakis_work | bikr: .. hmm im on the att site.. it shows like .. $60 for unlimited data +tethering |
| 13:57.57 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:58.48 | bikr | lirakis: no clue how the rate plan works , but the rate code on their system is BBUTL |
| 13:59.04 | bikr | and when you pair that with voice it comes out to like 59 |
| 13:59.17 | bikr | it is a blackberry rate plan |
| 13:59.19 | CCFL_Man2 | bikr: if i try the sim in another unlocked phone and it works it means the sim isn't locked? |
| 13:59.28 | bikr | yup |
| 13:59.49 | bikr | what they usually do on those prepays in my experience is lock the sim to the phone, kinda like they did with iphone |
| 13:59.49 | CCFL_Man2 | bikr: it worked when the sim was unregistered though |
| 14:00.02 | lirakis_work | bikr: ill have to talk to them about it b/c if i can get unlimited 3g tether + a phone (just incase) .. i will def do that b/c right now my phone is like $60mo |
| 14:00.17 | bikr | lirakis: word |
| 14:00.42 | bikr | lirakis: we switched to a business account now from a personal account and nothing changed on that line either , so it should still be doable |
| 14:00.50 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:00.56 | CCFL_Man2 | i never got tracphone because i wasn't sure if it was gsm |
| 14:01.11 | bikr | ccfl: I've been impressed with boost |
| 14:01.19 | Telemac | I really don't understand misdn error message (NO FREE CHAN) -> http://openpaste.org/en/6150/ ? Chan list dump indeed show an in use channel but I've only one incoming call so why ? |
| 14:01.28 | bikr | ccfl: too bad they'll be going out of business soon cuz they targeted all their ad money to kids |
| 14:01.38 | erwinpogz | is this right for extension to call outside dial 9? exten => _9.,99,Dial(SIP/cbiout/${EXTEN:1},20,tr) |
| 14:01.46 | CCFL_Man2 | bikr: are they gsm? |
| 14:01.57 | bikr | ccfl: yes |
| 14:02.11 | CCFL_Man2 | bikr: ahh, cdma is evil |
| 14:02.30 | bikr | ccfl: agree :) |
| 14:02.40 | bikr | ccfl: stupid verizon!$%@!#% <--my personal phone is verizon |
| 14:02.41 | CCFL_Man2 | hehe |
| 14:02.41 | RockHound | Telemac: have you configured port groups? |
| 14:02.55 | Telemac | RockHound: Yes, 1 |
| 14:03.06 | CCFL_Man2 | bikr: thats unfortunate, i'd never want shitty verizon |
| 14:03.15 | CCFL_Man2 | unless they went gsm |
| 14:03.20 | bikr | my wife has been a customer since they were "primeco" she won't leave lol |
| 14:03.33 | CCFL_Man2 | because using 3rd party cdma equipment is hard to do |
| 14:03.38 | bikr | ccfl: supposidly they are doing "open" platform this year, which will allow gsm phones on their network somehow |
| 14:03.39 | erwinpogz | anyone please help |
| 14:03.40 | CCFL_Man2 | lol |
| 14:03.41 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 14:03.46 | RockHound | Telemac: how many ports does your card have? what kind of card is it? |
| 14:03.52 | Telemac | RockHound: You can se my misdn.conf at http://openpaste.org/en/6151/ |
| 14:04.07 | Telemac | RockHound: 4 port with a b410p digium card |
| 14:04.12 | bikr | erwinpogz: looks to be |
| 14:04.14 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:04.24 | CCFL_Man2 | bikr: i heard that, i'm sure i can use a cdma modem on their network too |
| 14:04.27 | bikr | erwinpogz, but i'm a noob |
| 14:04.36 | erwinpogz | bikr, it doesnt worj here |
| 14:04.50 | CCFL_Man2 | bikr: for years i'd never get a cell phone because of how carriers only allow their phones |
| 14:05.13 | bikr | ccfl: they gotta rope you somehow lol.. as soon as they open their platform i'm getting an iphone and JB'ing it |
| 14:05.29 | bikr | ccfl: and hopefully the 3g versoin will be out lol |
| 14:05.43 | CCFL_Man2 | i wanted to use whatever i wanted when i wanted |
| 14:05.45 | CCFL_Man2 | yeah |
| 14:06.05 | CCFL_Man2 | i wanted the go phone account with the intention of using it with an iphone |
| 14:06.07 | bikr | question: why don't home phones have txt msging yet lol |
| 14:06.18 | RockHound | Telemac: why use misdn with a digium card? why not use bristuff? but that does not solve your problem ... are you sure you are dialing to the groups? via g:GROUPNAME |
| 14:06.23 | CCFL_Man2 | bikr: thats what smsd is for :P |
| 14:06.25 | RockHound | and g:CONTEXT |
| 14:06.26 | bikr | ccfl: you can do that, but gotta sim break it |
| 14:06.35 | CCFL_Man2 | bikr: yeah |
| 14:06.36 | bikr | ccfl: lolz!! |
| 14:06.44 | igascream | Asterisk doesn't detect busy tones some suggestions about it? Realy need your help.... |
| 14:06.54 | bikr | ccfl: i haven't done one yet, but it doesn't look easy to do (iphone simunlock) |
| 14:07.13 | bikr | igascream: * doesn't? or your sip client doesn't? |
| 14:07.16 | rupa | bikr, one click, real easy. got my ipohone unlocked w/out any trouble |
| 14:07.16 | CCFL_Man2 | i hear it is |
| 14:07.28 | rupa | running my iphone on t-mobile |
| 14:07.28 | bikr | rupa: the iphone or the sim unlock ? |
| 14:07.33 | bikr | rupa: nice!!! |
| 14:07.35 | CCFL_Man2 | nowadays unlocking an iphone is super easy |
| 14:07.36 | rupa | the iphone |
| 14:07.38 | Telemac | RockHound: chan_misdn is recommanded way for that card. It's not an outgoing call be an incoming one (but I also get a NO FREE CHAN or for outgoing call on port) |
| 14:07.46 | bikr | <--can't wait for verizonopen then :) |
| 14:08.07 | igascream | bikr, No Asterisk from analog line |
| 14:08.14 | RockHound | Telemac: can you post your relevant pieces of the dialplan? |
| 14:08.30 | bikr | igascream: dialing out of your pbx? or internal? |
| 14:08.36 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 14:08.40 | *** join/#asterisk fmueller (n=user@p548F6C21.dip.t-dialin.net) |
| 14:08.40 | CCFL_Man2 | bikr: i don't know why they didn't sell the iphone unlocked anyway without any carrier's infection in it, one with a cdma modem, one with a gsm modem |
| 14:09.02 | bikr | ccfl: because they didn't make any money on the retail , they made money when the phone was activated |
| 14:09.03 | igascream | bikr, out of it |
| 14:09.06 | rupa | CCFL_Man2, apple gets a cut of your monthly |
| 14:09.14 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.235.113) |
| 14:09.32 | CCFL_Man2 | ewww, i have no respect for apple |
| 14:09.41 | bikr | igascream: what effect are you expecting on an outbound? does it not even relay the signal? |
| 14:09.50 | bikr | ccfl: me either, but damn their shit is cool! lol |
| 14:09.56 | igascream | bikr, it doesn't hang up if caller hung up. |
| 14:10.02 | bikr | ccfl: cept the air , that thing sucks |
| 14:10.09 | CCFL_Man2 | i purchased an iphone for a person in canada, he gave me the money for it, i feel proud i screwed them |
| 14:10.11 | Telemac | RockHound: http://openpaste.org/en/, When I manually use "dial" in asterisk console (eg : dial 1234@from-pstn) I properly get the call |
| 14:10.14 | ManxPower | bikr: You must be new in the industry. Verizon will not open up anything. They will find a way around the rules. |
| 14:10.17 | bikr | igascream: the trunk stays open???? that's like, outside of default behaivior for sure |
| 14:10.18 | rupa | I do miss the home screen I had on my winmobile phone, but overall the iphone is real nice |
| 14:10.32 | bikr | manx: they already press released it before the open rules were announced, it's happening this year |
| 14:10.33 | CCFL_Man2 | ManxPower: so true |
| 14:10.39 | igascream | yes |
| 14:10.46 | bikr | manx: and i'm not in the industry per say.. I'm just Systems Engineer for a toy company |
| 14:10.55 | CCFL_Man2 | iphone is cool, a pocket version of darwin |
| 14:10.56 | bikr | manx: who happens to want a PBX in his house |
| 14:10.59 | bikr | manx: lol |
| 14:11.16 | RockHound | Telemac: the link is not complete |
| 14:11.20 | igascream | it still recording the message in voicemail if person hung up |
| 14:11.27 | ManxPower | igascream: detecting a far end hangup using analog is very difficult to make work |
| 14:11.31 | Telemac | RockHound: ? |
| 14:11.38 | ManxPower | igascream: you must not be in the USA or Canada. |
| 14:11.41 | CCFL_Man2 | i want a pbx in my house, as well as a private digital cable system, i'm in the process of building both |
| 14:11.47 | RockHound | Telemac: you did not post the entire link to openpast.org |
| 14:11.56 | bikr | ccfl: with tuning!!!??? |
| 14:11.56 | igascream | I'm in IL. |
| 14:12.00 | ManxPower | CCFL_Man2: I built a private analog cable system |
| 14:12.03 | Telemac | RockHound: oh sorry, http://openpaste.org/en/6152/ |
| 14:12.09 | ManxPower | CCFL_Man2: want to buy some equipment? |
| 14:12.15 | CCFL_Man2 | ManxPower: what were the sources? |
| 14:12.17 | bikr | manx: with tuning????? how the hell do they do that?? |
| 14:12.35 | Faustov | Siya: sorry again for the delay, what do you mean more than ip routing? i thought it's all about port forwarding and some settings in sip.conf, that i mentioned before. Is there more to it? |
| 14:12.35 | RockHound | Telemac: yes ... that is what I thought ... you need to name the groups |
| 14:12.37 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.121) |
| 14:13.26 | CCFL_Man2 | bikr: using an off the shelf atsc/qam tuner to tune the qam channels |
| 14:13.27 | RockHound | Telemac: ehhmm ... where are the DIAL() statements? |
| 14:13.28 | Telemac | RockHound: ? |
| 14:13.31 | ManxPower | CCFL_Man2: Master Antenna -> Blonder Tongue boxes to shift the channel and DVD player -> Blonder Tongue modulators |
| 14:13.57 | bikr | ccfl: that is awesome! how many channels can you get and how much does that cost? |
| 14:13.59 | CCFL_Man2 | ManxPower: master antenna us a uhf/vhf antenna? |
| 14:14.00 | ManxPower | igascream: is your analog line connected directly to a major telco? |
| 14:14.02 | bikr | wants |
| 14:14.09 | ManxPower | CCFL_Man2: correct. |
| 14:14.28 | bikr | Manx: I think he was looking for busy detection, not trunk release no? |
| 14:14.30 | Telemac | RockHound: Dial is elsewhere in extensions.conf (exten => mysip-sip,1,Dial(SIP/account-sip)) |
| 14:14.36 | ManxPower | CCFL_Man2: almost all of my equipment will be put on eBay within the next 6 months |
| 14:14.54 | igascream | ManxPower, yes it's simple telephone line |
| 14:15.00 | ManxPower | bikr: if he has "hangup problems" then it's not a busy issue. |
| 14:15.01 | CCFL_Man2 | bikr: it costs a crapload, the sources are c band dishes, the receivers are super expensive, as are the qam modulators, but thats all thats required |
| 14:15.02 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:15.04 | RockHound | Telemac: are you using the cards to connect phones internally? or to dial out? |
| 14:15.12 | CCFL_Man2 | ManxPower: you have no digital stuff? |
| 14:15.14 | [TK]D-Fender | Telemac: pastebin your entire dialplan and CLI output of a failed attempt at verbose 10 & SIP debug enabled |
| 14:15.19 | RockHound | Telemac: nevermind |
| 14:15.28 | ManxPower | igascream: I've never heard of your problem on an analog standard telco line in the USA |
| 14:15.49 | ManxPower | CCFL_Man2: Digital CATV stuff is 4x as expensive as the analog stuff on eBay |
| 14:15.51 | RockHound | Telemac: afaik, you need to configure one group per phone in misdn ... |
| 14:15.53 | tzafrir | ManxPower, that line is not in the US |
| 14:15.57 | igascream | But I'm not in US |
| 14:16.12 | Telemac | RockHound: for now to get call from isdn to sip phone |
| 14:16.18 | ManxPower | tzafrir: "igascream: I'm in IL." |
| 14:16.19 | tzafrir | igascream, try removing callprogress=yes |
| 14:16.22 | RockHound | Telemac: okay |
| 14:16.22 | CCFL_Man2 | i got a broked digicipher II ird... i feel so stupid thinking i could fix it |
| 14:16.28 | tzafrir | ManxPower, Israel |
| 14:16.31 | RockHound | Telemac: so forget about the one port per phone thing ... |
| 14:16.41 | tzafrir | .il , not .il.us |
| 14:16.53 | ManxPower | tzafrir: Mayne the jerk could have said that. |
| 14:17.05 | CCFL_Man2 | the jerk |
| 14:17.06 | ManxPower | he's screwed. |
| 14:17.09 | RockHound | Telemac: to dial out, you need to do something like DIAL(misdn/g:TEports/${EXTEN}) |
| 14:17.39 | Telemac | RockHound: Yes but for incoming ? |
| 14:17.57 | CCFL_Man2 | ManxPower: i know it is, but i want the discovery networks c band feed reaching all my tvs, the only trouble is the activation |
| 14:18.01 | igascream | tzafrir, I have already tried it doesn't work |
| 14:18.18 | tzafrir | have you tried using fxs_ks ? |
| 14:18.24 | RockHound | just do a DIAL(SIP/2000) |
| 14:18.30 | ManxPower | CCFL_Man2: C Band, you mean using a BUD? |
| 14:18.38 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 14:18.42 | ruied | I'm having a billing proble... If the operator places the call to the outside (A), I can trace the call and bill that call, but, if the operator places an inside call (B) and then transfers the call from (A) to (B) I don't know how to identify call (A) and (B) to sum the total call time and make the total billing... |
| 14:18.44 | tzafrir | or maybe hanguponpolarityswitch? |
| 14:18.45 | RockHound | Telemac: where the 2000 is the SIP phone |
| 14:18.54 | CCFL_Man2 | ManxPower: i need to use a couple to get all the stations i want |
| 14:18.57 | ManxPower | CCFL_Man2: The fact you get the signal as digital does NOT mean you have to send it across your network as digital. |
| 14:19.00 | [TK]D-Fender | ManxPower: Outside of North America that makes a good match for PAL TV signalling ;) |
| 14:19.09 | igascream | i have only fxo_ks ports if I'm not wrong |
| 14:19.10 | *** join/#asterisk anthm (n=anthm@adsl-69-210-62-125.dsl.milwwi.ameritech.net) |
| 14:19.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:19.13 | ruied | I don't see any fild in cdr where I can make the match for what I've said... any ideas? |
| 14:19.25 | ManxPower | Half of my channels came in as digital to a converter box, then turned into analog using commercial modulators |
| 14:19.30 | CCFL_Man2 | ManxPower: i don't have to, but i want to |
| 14:20.02 | CCFL_Man2 | oh, you used off the air atsc? |
| 14:20.06 | ManxPower | CCFL_Man2: There is no good way without spending MASSIVE amounts of money. |
| 14:20.11 | *** join/#asterisk af_ (n=getsmart@88-149-240-161.dynamic.ngi.it) |
| 14:20.26 | ManxPower | CCFL_Man2: Yes, I used OTA ATSC for as many channels as I could |
| 14:20.28 | Telemac | RockHound: That's what I do, but I'd like to trigger that from incoming isdn call |
| 14:20.31 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:20.45 | tzafrir | ManxPower, and if someone tells you he's from CA, what would you assume? |
| 14:20.46 | ManxPower | Until recently OTA ATSC boxes were over $100 each. For 8 local stations that adds up. |
| 14:21.03 | ManxPower | tzafrir: California unless I had reason to think otherwise. |
| 14:21.22 | tzafrir | Right. Nobody is from Canada |
| 14:21.23 | ManxPower | Now if they say .ca I would think of that wonderful country north of the USA with the horrible weather. |
| 14:22.00 | ManxPower | tzafrir: But it really is the job of the other person to SPEAK CLEARLY |
| 14:22.08 | *** join/#asterisk xxnickxx (n=chatzill@bas3-montreal19-1177825199.dsl.bell.ca) |
| 14:22.09 | RockHound | Telemac: just put this: exten => _X.,1,DIAL(SIP/103) in your [from-pstn] ... thats all |
| 14:22.11 | RockHound | for starts |
| 14:22.19 | CCFL_Man2 | ManxPower: i purchased everything used off ebay except the qam modulators |
| 14:22.29 | ManxPower | CCFL_Man2: *nod* |
| 14:22.36 | *** part/#asterisk erwinpogz (n=niwre@67.159.178.21) |
| 14:22.40 | Telemac | RockHound: that's what I have, but I get a NO CHAN ERROR (I don't think it's an extension issue in fact) |
| 14:22.52 | ManxPower | My CATV network was supposed to be connected to analog TVs and there was NO reason to use digital |
| 14:23.05 | RockHound | what does misdn show stacks say? |
| 14:23.32 | RockHound | Telemac: but I thought you have all this goto stuff in there ... |
| 14:23.33 | Telemac | RockHound: http://openpaste.org/en/6154/ |
| 14:23.37 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-c6f1304cd685feb8) |
| 14:23.39 | CCFL_Man2 | ManxPower: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=250235787718 <--this puppy i snagged for $50 i will use it's asi output to feed into my qam modulator, i just want pbs |
| 14:23.51 | ManxPower | CCFL_Man2: The project was 10 times as complicated as I thought it would be. |
| 14:24.03 | CCFL_Man2 | ManxPower: it usually us |
| 14:24.28 | igascream | So I can't understand the process of hanging up by asterisk is configurable or it is made by the phonecard If yes where can I change it in zapata.conf??? |
| 14:25.05 | *** join/#asterisk jfg (n=jfg@dyn-91-165-220-45.ppp.tiscali.fr) |
| 14:25.18 | RockHound | Telemac: PMP mode is correct? |
| 14:25.26 | ManxPower | CCFL_Man2: That is 1/4 the cost of the stuff I looked at. Granted I was doing this 12 - 24 months ago |
| 14:25.27 | Telemac | RockHound: I've "exten => _X.,1,Goto(internalc|103|1)" (http://openpaste.org/en/6153/) in [from-pstn] and I dial sip phone from internalc |
| 14:25.29 | CCFL_Man2 | ManxPower: though the qam modulators and rf upconverters are small modules that slide into a chassis, you can have 4 of them in one chassis, thats 4 multiplexes |
| 14:26.01 | Telemac | RockHound: mISDN found PMP (I'm in France), and when I try PTP (I could retry) isdn doesn't get packets |
| 14:26.03 | CCFL_Man2 | ManxPower: i'm wondering myself why it was so cheap |
| 14:26.15 | ManxPower | igascream: Asterisk expects power to be removed from the line for .5 seconds to indicated a hangup. Your telco is not providing that signal and so the line is not being detected as being hung up. During a normal phone call, YOUR side hangs up, so asterisk hangs up all associated channels. |
| 14:26.30 | Telemac | RockHound: mISDN scan I mean |
| 14:27.08 | CCFL_Man2 | ManxPower: i'm usually just looking for some receiver with asi output, i really want to get abc, cbs, nbc, and fox via their national hd c band feeds |
| 14:27.09 | tzafrir | Actually that is one way to signal hangup. I know Bezeq does that, but only on some of the lines |
| 14:27.21 | ManxPower | igascream: for the most part the only telcos that remove battery fro the line for .5 second to indicate a hangup are located only in the USA. |
| 14:27.35 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:27.43 | ManxPower | you need to find out how your telco signals a hangup and hack asterisk to detect that. This is not a problem on ISDN or T-1/E-1 |
| 14:27.46 | tzafrir | Another common method is by polarity reversal. Bezeq does not use it. But HOT? who knows |
| 14:28.01 | RockHound | Telemac: what was connected to the line before the digium card? |
| 14:28.51 | Telemac | RockHound: I get the cable directly from ISDN jack. Previously that cable go to a Nortel PBX port. |
| 14:29.11 | igascream | ManxPower, so that mean that I have nothing to do with it Just once a day to delete msgs with beeps from voice mail? |
| 14:29.12 | RockHound | Hmm ... Sounds to me that you need to run the port in PTP ... |
| 14:29.13 | CCFL_Man2 | ManxPower: i wanted to try to get a c band provider to authorize my dcII ird for discovery networks, but that will be a challenge |
| 14:29.27 | RockHound | Telemac: but I could be wrong |
| 14:29.43 | ManxPower | igascream: I suggest you enable the silence detection and min message length in voicemail.conf That will HELP, but not REMOVE the problem |
| 14:29.54 | Telemac | RockHound: I could try it again, I'm trying various combination from last week :( |
| 14:30.11 | ManxPower | CCFL_Man2: *nod* There are ala carte packages from several providers. |
| 14:30.20 | ManxPower | But they are expensive for an individual |
| 14:30.46 | RockHound | A PBX is usually connected to a PTP port ... |
| 14:31.00 | tzafrir | RockHound, running the port in PtP is generally recommended, I guess |
| 14:31.19 | CCFL_Man2 | i'm just sick of the cable and pizza dish providers that compress the crap out of the signal to fit more crappy channels to make you think you are getting a good deal |
| 14:31.23 | Telemac | RockHound: I've change pmp to ptp in /etc/mISDN.conf and added "ptp=yes" in /etc/asterisk/misdn.conf but "misdn show config" tell me "ptp=no" |
| 14:31.33 | tzafrir | Unless you actually need the extra features of PtMP (connecting several phones on the same line) |
| 14:31.44 | RockHound | Telemac: that config is take from /etc/misdn.conf |
| 14:32.02 | RockHound | tzafrir: I have my setup working, it is Telemac struggeling |
| 14:32.27 | Telemac | RockHound: I've /etc/mISDN.conf and /etc/misdn-init.conf so in misdn-init.conf I suppose |
| 14:32.28 | RockHound | Telemac: I type to fast ...sorry |
| 14:32.44 | RockHound | Telemac: or better, I type before I read ... anyhow .. .try unloading an reloading drivers |
| 14:32.51 | boblutz | RockHound: yea |
| 14:33.06 | RockHound | boblutz: ? |
| 14:33.10 | CCFL_Man2 | ManxPower: though music choice is unencrypted, so i can use a dcII ird with asi out to get the 53 channels of music |
| 14:33.22 | Telemac | RockHound: I've unloaded them and do a "mISDN start" |
| 14:33.40 | igascream | ManxPower, Something else : After some time not hanging up teleco send me some loud signals so I can hear if I forgot to hang up my phone and asterisk hang up after it why ? |
| 14:34.13 | boblutz | RockHound: I was agreeing with you |
| 14:34.27 | RockHound | boblutz: about typing before reading? :-P |
| 14:34.58 | boblutz | heh |
| 14:35.03 | RockHound | Telemac: stop asterisk and tell me what misdnportinfo says? |
| 14:35.27 | RockHound | Telemac: that should say something about Interface is Poin-To-Point. |
| 14:35.40 | Telemac | RockHound: Yes, http://openpaste.org/en/6156/ |
| 14:35.57 | Telemac | RockHound: But chan_misdn in Asterisk doesn't seem to know that |
| 14:37.28 | ManxPower | igascream: does asterisk hangup when it receives the tones? |
| 14:38.00 | RockHound | Telemac: I had this problem ... the chan_misdn looks for a misdn-init.conf file and not for mISDN.conf ... this was the old module config file |
| 14:38.18 | igascream | ManxPower, yes after some this loud ones . |
| 14:38.26 | Telemac | RockHound: I've "te_ptp=1,2,3,4" in /etc/misdn-init.conf |
| 14:38.33 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:38.35 | lirakis_work | .. right.. after the loud ones |
| 14:38.40 | RockHound | Telemac: ok |
| 14:38.40 | ManxPower | igascream: that is as good as you can hope for. |
| 14:39.24 | Telemac | RockHound: i've tried adding "ptp=yes" in [general], [default] sections and in [isdn1] group |
| 14:39.32 | Telemac | RockHound: I still have ptp=no |
| 14:39.39 | CCFL_Man2 | i see igascream and i get hunfry for ice cream |
| 14:39.46 | CCFL_Man2 | hungry |
| 14:39.52 | Telemac | RockHound: "ptp: no" in misdn show config |
| 14:40.00 | RockHound | Telemac: then asterisk is not reading your misdn-init.conf correctly ... |
| 14:40.23 | Telemac | RockHound: Maybe... I don't see how to fix that |
| 14:40.28 | lirakis_work | CCFL_Man2: im hungry too.. heh heh |
| 14:40.30 | igascream | CCFL_Man2, do you want me to change my nick? |
| 14:40.42 | CCFL_Man2 | hehe |
| 14:40.42 | *** join/#asterisk s0lid (n=s0lid@210.213.199.98) |
| 14:40.51 | CCFL_Man2 | igascream: not at all, i love ice cream |
| 14:41.01 | CCFL_Man2 | one of my guilty pleasures |
| 14:41.20 | CCFL_Man2 | if i wasn't lactose intolerant i'd eat a half a tub |
| 14:41.34 | lirakis_work | <PROTECTED> |
| 14:41.51 | CCFL_Man2 | lirakis_work: just not the same :P |
| 14:42.07 | CCFL_Man2 | but it's good too |
| 14:42.39 | CCFL_Man2 | i should be having fiber rich things, not ice cream |
| 14:42.47 | CCFL_Man2 | but i just love it |
| 14:43.21 | igascream | I realy happy that I remind you about ice cream ,but it doesn't solve my problem |
| 14:43.55 | CCFL_Man2 | hehe, i wish i could help |
| 14:45.12 | igascream | CCFL_Man2, Did you saw what I have been writing about? |
| 14:45.52 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
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| 14:50.31 | CCFL_Man2 | igascream: yeah, i'm not sure how to fix the problem |
| 14:51.08 | igascream | CCFL_Man2, did you hear about it at all? |
| 14:51.11 | *** join/#asterisk Levan75 (n=levan@210.1.193.217) |
| 14:52.00 | Levan75 | hi all I was wondering if I can get some help compiling format_mp3 for a Ubuntu asterisk install - specifically with some compiler errors |
| 14:52.05 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:53.02 | CCFL_Man2 | igascream: only thing i can think of is to use a channel bank for your fxo ports |
| 14:53.12 | ZPertee | has anyone been able to get asterisk compiled with IMAP support on ubuntu? |
| 14:54.06 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:54.48 | CCFL_Man2 | hangup detection can be a screwy thing on pots lines |
| 14:54.50 | Levan75 | I ran make install in the format_mp3 directory and got the following output... |
| 14:54.59 | rupa | ~pb |
| 14:54.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:54.59 | Levan75 | In file included from /usr/include/asterisk/logger.h:28, |
| 14:54.59 | Levan75 | <PROTECTED> |
| 14:56.02 | CCFL_Man2 | in the US they cut battery for a second? |
| 14:56.34 | jsmith | CCFL_Man2: Correct |
| 14:56.36 | *** join/#asterisk frk2 (n=fkhan@117.20.28.50) |
| 14:57.07 | frk2 | Is there anybody who can provide me with the SIP firmware for a cisco 7911G? |
| 14:57.31 | Levan75 | can someone take a look at http://paste.debian.net/plain/906 and tell me what they think regarding the errors? |
| 14:57.36 | rob0 | Cisco? |
| 14:57.36 | CCFL_Man2 | jsmith: why do zaptel fxo interfaces have such a difficult problem with hangup detection? |
| 14:57.36 | frk2 | my cisco distributor is rather retarded |
| 14:57.40 | rob0 | oh |
| 14:58.28 | CCFL_Man2 | frk2: i got a 7912, how is the 7911? |
| 14:58.43 | frk2 | same thing, newer model |
| 14:58.58 | CCFL_Man2 | is it any better? |
| 14:58.59 | frk2 | I would know if I can get it to work :) |
| 14:59.22 | Levan75 | no compiler experts floating around? |
| 14:59.35 | frk2 | but it looks the same, menus look the same too. datasheets also look the same :) |
| 14:59.36 | CCFL_Man2 | just purchase an $8 phone contract |
| 14:59.39 | ManxPower | Levan75: this is not a dev channel |
| 14:59.43 | *** join/#asterisk af_ (n=getsmart@88-149-241-171.dynamic.ngi.it) |
| 14:59.49 | jsmith | CCFL_Man2: I'm not sure what you mean... The TDM410P does great with both detecting the open switching interval and/or polarity reversal |
| 15:00.00 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:00.10 | frk2 | CCFL_Man2 can I do this online? |
| 15:00.11 | ManxPower | CCFL_Man2: because anyone with the skills to fix the hangup detection (it requires great DSP foo) is using digital |
| 15:00.21 | Levan75 | I get that - but then I am not tryng to develop...just compile some source for use on Gutsy |
| 15:00.24 | CCFL_Man2 | jsmith: i hear so many issues with it |
| 15:00.26 | jsmith | CCFL_Man2: The only thing it doesn't do spectacularly is tone detection... |
| 15:00.32 | CCFL_Man2 | ahh |
| 15:00.38 | CCFL_Man2 | i see |
| 15:00.38 | Levan75 | Mankpower: I get that - but then I am not tryng to develop...just compile some source for use on Gutsy |
| 15:00.49 | jsmith | CCFL_Man2: And if you're doing tone detection for hangup supervision, you're just gonna have to learn to deal with heartache |
| 15:00.50 | tzanger | jsmith: the TDM4xx doesn't do *any* tone detection at all to my knowledge |
| 15:01.04 | CCFL_Man2 | frk2: yeah, i got mine from cdw but it wasn't instant |
| 15:01.08 | jsmith | tzanger: Well, the card doesn't... it's done in the DSP routines in Asterisk |
| 15:01.15 | tzanger | right |
| 15:01.15 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) [NETSPLIT VICTIM] |
| 15:01.16 | *** join/#asterisk plantseeker (n=chatzill@77.240.56.22) [NETSPLIT VICTIM] |
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| 15:01.21 | tzanger | but that's card-agnostic |
| 15:01.27 | ManxPower | Levan75: looks like you didn't do a ./configure |
| 15:01.42 | CCFL_Man2 | jsmith: i'll just stick with a channel bank |
| 15:01.46 | ManxPower | or you installed packages and did not remove and unpack the asterisk source again |
| 15:02.07 | jsmith | CCFL_Man2: That's fine... nothing wrong with a good channel bank |
| 15:02.36 | CCFL_Man2 | jsmith: i'll reduce problems, i think |
| 15:02.51 | CCFL_Man2 | use it for fxs/fxo interfaces |
| 15:02.54 | tzanger | <-- <3 channel banks |
| 15:02.55 | codefreeze | zippety |
| 15:03.00 | Levan75 | Manx:ok the wierd thing is I untarred a fresh source. ran the ./configure from the addons src . ran a make on addons and it reported that it ran fine...BUT it didnt run the make on format_mp3 .... |
| 15:03.15 | CCFL_Man2 | tzanger: what kind? |
| 15:03.26 | jsmith | Levan75: Did you do "make menuselect" and select format_mp3? |
| 15:03.35 | tzanger | CCFL_Man2: personally I like the Carrier Access stuff and the Adit 600 in particular |
| 15:03.58 | ManxPower | Levan75: Oh! You are using Asterisk addons! I can't help you with that. |
| 15:04.06 | jsmith | CCFL_Man2: I too will vouch for the Adit 600, as well as the Adtran TA series |
| 15:04.09 | CCFL_Man2 | tzanger: yeah, theykick ass |
| 15:04.11 | ManxPower | Levan75: maybe you don't have any MP3 libraries installed? |
| 15:04.12 | Levan75 | manx: yup but it reports XXX on Format_mp3 indicating its missing the asterisk dependency -- which is bizarre considering my asterisk box is running nicelyt |
| 15:04.28 | CCFL_Man2 | jsmith: ahh, i wanted to try the adtran a750 |
| 15:04.34 | ManxPower | Levan75: it's not bizarre at all. You don't have the required libs installed to support MP3 |
| 15:04.43 | Levan75 | manx: I actually have the mpg123 package installed... |
| 15:04.43 | frk2 | I guess nobody here has it |
| 15:04.49 | jsmith | CCFL_Man2: The Adtran TA750 works great |
| 15:04.54 | frk2 | CCFL_Man2 do you remember the cisco part number? |
| 15:05.07 | jsmith | Levan75: format_mp3 doesn't use mpg123 |
| 15:05.07 | [TK]D-Fender | Levan75: mpg123 has nothing to do with format_mp3.so |
| 15:05.32 | Levan75 | manx: ok so which lib do I need ? |
| 15:05.42 | CCFL_Man2 | jsmith: ahh, nice |
| 15:06.11 | CCFL_Man2 | frk2: not for the 7911, it's the smartnet contract |
| 15:06.17 | Levan75 | manx: I know it doesnt....use mpg123 that is... |
| 15:06.30 | *** part/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 15:06.45 | ManxPower | Levan75: mpg123 and mpg321 have their own built in mp3 codec |
| 15:06.53 | *** join/#asterisk bts3685 (n=bts@c-69-249-136-23.hsd1.pa.comcast.net) |
| 15:07.03 | ManxPower | Levan75: there are NO docs for format_mp3 in asterisk-addons? |
| 15:07.21 | Levan75 | manx: no unfortunately |
| 15:07.39 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:07.41 | frk2 | CCFL_Man2 does CON-SNT-CP7911 ring a bell? |
| 15:07.43 | ManxPower | Levan75: why do you want to go thru all this pain for something that's not recommended anyway? |
| 15:08.06 | Levan75 | manx: using mp3's not recommended? |
| 15:08.23 | Levan75 | manx: I think i missed something.... |
| 15:08.32 | CCFL_Man2 | frk2: yeah, it's $8 |
| 15:08.38 | ManxPower | Levan75: They suck up CPU and there is no reason to use MP3 unless you are trying to play a live mp3 stream. |
| 15:08.44 | *** join/#asterisk xenonex (n=xenonex@89.218.235.91) |
| 15:08.56 | ManxPower | and the one thing Asterisk wants lots of is CPU |
| 15:09.00 | jsmith | Levan75: For what it's worth, the only dependency I see in format_mp3 is on Asterisk itself |
| 15:09.05 | Levan75 | manx: no I wasnt I had ripped my onhold music to mp3 ... |
| 15:09.35 | ManxPower | Levan75: convert it to the format most of your calls use and you'll significantly reduce CPU usage of MoH |
| 15:09.50 | Levan75 | manx: fair enough... |
| 15:10.12 | bts3685 | does anyone know why my polycom 501 (firmware 2.2.2.0084) isn't pulling the correct time/date from the pbx box? |
| 15:10.26 | Levan75 | manx: shrug its a test box anyway... |
| 15:10.35 | ManxPower | bts3685: perhaps the NTP server on "your box" is wrong? |
| 15:11.04 | Levan75 | manx: a trial compiling asterisk/freepbx/op_panel on Ubuntu server.... |
| 15:11.30 | ManxPower | Levan75: Well you should not be here if you are using FreePBX |
| 15:12.03 | vlt | Hello. All our phones are connected to Asterisk via SIP, Asterisk is connected to the Telcos via IAX. We are facing problems with echos on incoming calls from Telco. Having read the wiki and using VoIP only on this side I assume the echo is caused by the remote peer. Any idea how we can avoid it? |
| 15:12.26 | ManxPower | bts3685: the polycoms get their time from whatever IP address you configure as an NTP server in the polycom config |
| 15:12.58 | ManxPower | vlt: you can't do anything about echo. Echo must be removed AT THE POINT where the call is converted to/from VoIP |
| 15:13.06 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 15:13.32 | ManxPower | vlt: since you don't have control over that point (PSTN/VoIP gateway) you can't fix the echo. |
| 15:13.40 | ManxPower | You can change ITSPs or live with the echo |
| 15:13.48 | Levan75 | manx: right....the fact is I am using the digium asterisk source's compiled atm with no extras running... |
| 15:14.43 | ManxPower | Levan75: the major problem we have with GUI Asterisks is that their config files were designed by the Marquis de Sade himself. |
| 15:14.48 | vlt | So, any reccomendation for a good ITSP provider for inbound and outbound calls in Germany? |
| 15:15.07 | Levan75 | manx: manx: completely understand... |
| 15:16.10 | Levan75 | manx: 1 issue I did note not sure that it was relevant was that the deb package for ubuntu doesnt put any of the "include" files found in the asterisk src in the includes path meaning any other src depending on them will fail |
| 15:17.15 | ManxPower | Levan75: we don't support packages here either. If you have a problem with a package talk to the author of the package. |
| 15:17.33 | Levan75 | manX: fair enough... |
| 15:17.36 | ManxPower | BTW, I do NOT speak for Digium |
| 15:19.36 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 15:19.54 | Levan75 | manx: can I ask one more question - where does the format_au.so come from...I get an error on the asterisk load sayings its looking for it....but havent seen it referenced anywhere? |
| 15:20.54 | ManxPower | Levan75: never heard of it. |
| 15:22.32 | [TK]D-Fender | Levan75: Looks like its in OpenPBX |
| 15:23.25 | ManxPower | Levan75: looks like you are now starting to experience the What The Hell Was The Packager Thinking |
| 15:23.43 | ManxPower | Levan75: if you want help here then remove all asterisk related stuff and COMPILE FROM SOURCE |
| 15:23.45 | Levan75 | manx: ....shiver... |
| 15:24.28 | ManxPower | be sure to remove /etc/asterisk and /usr/lib/asterisk and /var/spool/asterisk (if this is a test box, if not then it's more complicated) |
| 15:24.49 | ManxPower | I'd save a copy of the configs somewhere as well |
| 15:24.54 | *** part/#asterisk rolek (n=rolek@87.215.195.98) |
| 15:25.07 | Levan75 | manx: thanks..... |
| 15:25.22 | Levan75 | manx: the configs are backed up... |
| 15:25.24 | ManxPower | Levan75: then you will be starting out the way most of us started out. |
| 15:25.39 | ManxPower | Rather than some franken-asterisk that we know nothing about. |
| 15:26.13 | Levan75 | manx: but what if I like my monster..... |
| 15:26.27 | Levan75 | manx: jking,,, |
| 15:26.58 | Levan75 | manx: thanks for the advice mate if I am going to scrap this install then I had better get to t |
| 15:29.04 | [TK]D-Fender | Levan75: If you like your monster, do us a favour and enjoy it silently :) |
| 15:33.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:33.36 | puzzled | hi |
| 15:35.25 | SomethingISODD | hello all i have a few questions about chan_h323 i am trying to register clients in strictly ip and route each ip to a different prefix but it only routes to the default context |
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| 15:54.23 | Faustov | [TK]D-Fender: is there a model or brand of simple routers (like those for home usage) that are known to route/translate nat well with sip? |
| 15:57.07 | *** join/#asterisk erwinpogz (n=niwre@67.159.178.21) |
| 15:57.26 | erwinpogz | hi, how can i solve this problem? NOTICE[31906]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
| 15:57.35 | [TK]D-Fender | Faustov: Linksys is usually fine. Netgear is iffy for some in general, but no news of SIP specifically. I'm partial to netfilter personally ;) |
| 15:57.50 | [TK]D-Fender | erwinpogz: Turn off CNG in your client :) |
| 15:58.13 | [TK]D-Fender | erwinpogz: And thats a "warning", not a "problem". |
| 15:58.48 | Faustov | [TK]D-Fender: ok thanks, i got to get rid of all this dlink shit then :> |
| 15:59.09 | [TK]D-Fender | Faustov: before you blame your equipement, get another trustworthy test piece. |
| 15:59.24 | [TK]D-Fender | Faustov: And let process of elimination determine the rest |
| 16:00.47 | Faustov | [TK]D-Fender: i think i'm at it, because i got (asterisk server -> router -> internet -> router -> asterisk server) and on one end i configured asterisk as the nat+sip howtos explained, but when i did the same on the other end it didn't work. I had to unset nat=yes on the other end or it wouldn't work |
| 16:01.08 | Faustov | [TK]D-Fender: with both routers having the same port forwarding rules, corresponding to asterisk, sip, rtp needs |
| 16:01.50 | Faustov | [TK]D-Fender: and eventually, the place where i had to unset nat=yes is the place with a dlink router |
| 16:02.12 | Faustov | any ideas what else might give me a better answer? |
| 16:02.36 | [TK]D-Fender | Faustov: IS IT * ON BOTH SIDES? |
| 16:04.28 | Faustov | [TK]D-Fender: yes |
| 16:05.40 | *** join/#asterisk af_ (n=getsmart@88-149-230-169.dynamic.ngi.it) |
| 16:06.51 | Faustov | [TK]D-Fender: such setup is defined in that second sipnat howto |
| 16:07.16 | Faustov | [TK]D-Fender: well it works for me, just not as i'd expect :) |
| 16:07.26 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:07.35 | erwinpogz | [TK]D-Fender thanks |
| 16:07.38 | flujan | hello all. |
| 16:08.00 | Faustov | bbl |
| 16:14.08 | flujan | guys, I have two box running asterisk . I need to place calls originated in box 2 need to be placed to box 1. It is working. But, when I receive a call in box 1 I need to deliver it to box 2. (this is not working) |
| 16:14.14 | flujan | here goes the files http://pastie.caboo.se/181755 |
| 16:14.19 | flujan | could you please take a look? |
| 16:15.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:15.44 | ManxPower | flujan: register => belongs in [general] not the config for the device |
| 16:16.04 | ManxPower | since registration has nothing to do with the device, it just notifies the remote server what IP is associated with which user/pass |
| 16:16.30 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 16:16.36 | ManxPower | Also it's Dial(SIP/destination@sipconfentry) |
| 16:16.43 | Telemac | Hello again |
| 16:16.50 | ZPertee | I am trying to get asterisk working with IMAP Support. I have libc-client2007 installed. how do I get asterisk working with it? |
| 16:16.57 | ManxPower | flujan: I would have thought you would have learned at least somethig in the last 6 months |
| 16:17.12 | ManxPower | ZPertee: I suggest imap-client-devel or similar. |
| 16:17.49 | flujan | ManxPower: register is in the [general] |
| 16:17.54 | ManxPower | ZPertee: I've never heard of anyone getting IMAP working in a production envirment in Asterisk. This has been discussed on the mailing list -- perhaps something |
| 16:18.34 | ManxPower | flujan: yur formatting is so screwed up it looked wrong |
| 16:18.37 | ZPertee | ManxPower, so you wouldn't try, or I just shouldn't expect too much? |
| 16:18.51 | ManxPower | ZPertee: Do you NEED VM IMAP? |
| 16:19.03 | ManxPower | flujan: don't enable both ulaw and alaw |
| 16:19.04 | *** join/#asterisk sione (i=sione@ocs.net) |
| 16:19.20 | flujan | ManxPower: why ? |
| 16:19.22 | SomethingISODD | hello all i have a few questions about chan_h323 i am trying to register clients in strictly ip and route each ip to a different prefix but it only routes to the default context |
| 16:19.44 | ManxPower | flujan: 1) there is no need 2) I've seen transcoding issues |
| 16:20.03 | ManxPower | SomethingISODD: almost nobody on this channel uses H323. Good luck and you might want to check the mailinglist archives |
| 16:20.08 | flujan | ManxPower: ow, thanks for the tip. |
| 16:20.13 | ZPertee | ManxPower, I guess it depends on the definition of NEED? It would most definitely be used and since I only have voicemail access through through phone port it would be nice to lighten the load |
| 16:20.28 | ManxPower | ZPertee: give it a try, but don't expect too much |
| 16:20.37 | sione | Anyone else have issues with CallCentric on Asterisk 1.6Beta7.1 ? |
| 16:20.49 | ZPertee | ManxPower, what is this imap-client-devel that you speak of? how do I get it? |
| 16:20.57 | flujan | ManxPower: so what is wrong with my Dial app at line 37 |
| 16:21.12 | file | mighty active in here. |
| 16:21.32 | ManxPower | exten => 35149399,1,Dial(SIP/${EXTEN}@1135149399) |
| 16:22.00 | ZPertee | ManxPower, would this http://packages.ubuntu.com/search?keywords=IMAP+CLIENT&searchon=names&suite=all§ion=all be it? |
| 16:22.00 | ManxPower | assuming you are trying to dial the phone number 35149399 via the gateway at [1135149399] |
| 16:22.28 | ManxPower | if not then a simple Dial(SIP/1135149399) will be just fine of the destination device only has 1 number |
| 16:22.50 | flujan | ManxPower: yes, it will have just one number. But it is not working... I dunno why. |
| 16:22.56 | ManxPower | ZPertee: I would be shocked if that was the correct one |
| 16:23.06 | ManxPower | flujan: its not working because you screwed up your dial line |
| 16:23.12 | sione | I think there is something wrong with SIP INVITE on asterisk 1.6 while using CallCentric :( |
| 16:23.13 | flujan | The second box, do not receive the call. Nor the inititation. |
| 16:23.18 | ManxPower | but you did not provide any CLI output so I really don't know for sure. |
| 16:24.18 | ManxPower | ZPertee: Those packages are for PERL, not C |
| 16:24.42 | ManxPower | flujan: I have you 2 possible Dial lines. Do neither of them work? |
| 16:25.11 | flujan | the second that places calls from box 2 to outside is working. |
| 16:25.14 | ZPertee | ManxPower, ok yeah I see. do you know how to compile asterisk with IMAP support? what would I enter in /usr/src/asterisk-1.4.18? |
| 16:25.31 | [TK]D-Fender | ZPertee: What do you want to do with IMAP? |
| 16:25.35 | flujan | The first, that receives a call in number 35149399 and dial to SIP/1135149399 is not working. |
| 16:25.58 | flujan | ManxPower: 35149399,1,Dial(SIP/1135149399/${EXTEN},60,rtT) |
| 16:26.05 | ManxPower | ZPertee: I can only tell you what I've seen on the mailing lists and here. I don't use IMAP support as I believe it it FAR too recent of a feature for my anal retentive manic depressive hateful users. |
| 16:26.17 | *** join/#asterisk vicram (n=vr@cm136.delta11.maxonline.com.sg) |
| 16:26.22 | ManxPower | flujan: where is the CLI output of the Dialing server? Where? Where? |
| 16:26.37 | [TK]D-Fender | ManxPower: So no dating clients, right? :) |
| 16:27.19 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-f927d12b6ec1a075) |
| 16:27.25 | ManxPower | [TK]D-Fender: My medical clinic customer users were always so nice and helpful. The real estate ones are prima donnas that don't like it when you say no and they have power. |
| 16:27.48 | ManxPower | [TK]D-Fender: you want to help flujan? I need to do errands. |
| 16:27.48 | ZPertee | [TK]D-Fender, my idea is to sync asterisk voicemail with outlook. Then when checking email can check voicemail at the same time. I know that I can just send out email attachment but that isn't going to turn off the mwi light |
| 16:28.10 | ManxPower | [TK]D-Fender: I think he's disabled. He takes SO long to do anything and doesn't listen. |
| 16:28.11 | flujan | ManxPower: http://pastie.caboo.se/181767 |
| 16:28.24 | [TK]D-Fender | ZPertee: "delete=yes" <- |
| 16:28.33 | [TK]D-Fender | ZPertee: Compilcating things for nothing. |
| 16:28.53 | lmadsen | ZPertee: you could give it a shot and see where you get... testing is useful, and then you could report bugs! :) |
| 16:29.26 | ZPertee | [TK]D-Fender, what if I don't want to delete just want to mark read |
| 16:29.26 | ManxPower | flujan: put a Noop(DIALSTATUS is ${DIALSTATUS} and HANGUPCAUSE is ${HANGUPCAUSE}) between the Dial and the Hangup |
| 16:30.02 | flujan | ManxPower: ok... doing it... It takes sometime, since It is a production server. with more than 300 users on it. |
| 16:30.07 | sione | thinking of moving away from Asterisk 1.6 and go to Asterisk 1.4 |
| 16:30.09 | ZPertee | lmadsen, thanks for your vote |
| 16:30.12 | [TK]D-Fender | ZPertee: * isn't that smart, and IMAP storage isn't a means of sharing signalling, its jsut a way to store it elsewhere (remote) |
| 16:30.12 | flujan | its is difficult to parse the output. |
| 16:31.03 | [TK]D-Fender | sione: Smart move. |
| 16:31.41 | sione | ya i should of done it long time ago |
| 16:31.55 | ZPertee | [TK]D-Fender, according to this article it is that smart http://blogs.reucon.com/srt/2007/02/20/asterisk_imap_store_for_voicemail.html |
| 16:32.42 | flujan | ManxPower: http://pastie.caboo.se/181769 |
| 16:32.44 | [TK]D-Fender | ZPertee: The go follow his guide |
| 16:32.48 | sione | I moved CallCentric to one of my 1.4 box and it works fine so I figure that 1.6 is too buggy now that i need to move off of it |
| 16:33.45 | ZPertee | [TK]D-Fender, ok point taken |
| 16:34.09 | jsmith | sione: Asterisk 1.6.0 hasn't even been released yet -- it's still in beta |
| 16:34.25 | sione | ya i know |
| 16:34.37 | sione | i just thought they would have the sip down right on it |
| 16:35.00 | jsmith | If you think there's a problem with the way 1.6.0 betas handle SIP messaging, then fill out a bug report |
| 16:35.04 | [TK]D-Fender | sharpens his "bleeding edge" and prepares to decapitate another shmuck |
| 16:35.34 | [TK]D-Fender | sione: put SER in front of it then |
| 16:35.39 | sione | I would love too, just I dont have access to the CallCentric side to see what their box is doing |
| 16:36.42 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 16:36.45 | sione | I see I get repsonse with "Trying, Ringing" but then "Invite" its dead |
| 16:36.58 | sione | the ACK on the INVITE packet |
| 16:37.04 | sione | no ack |
| 16:37.50 | jsmith | sione: Like I said... if you think Asterisk is doing something wrong (and can back it up with SIP traces, etc.) then open a bug report. |
| 16:37.50 | sione | go down load Asterisk 1.4.19 now |
| 16:38.04 | *** part/#asterisk Mongos (n=alex@mongos.devnull.ru) |
| 16:38.04 | sione | jsmith: will do |
| 16:38.22 | sione | at the time being I move the 1.6 box to Asterisk 1.4 |
| 16:40.44 | flujan | ManxPower: and the box 2 is not busy or have congestion... that is the point. |
| 16:43.57 | ManxPower | flujan: do a sip show peers and tell paste JUST the line for 1135149399 |
| 16:44.03 | *** join/#asterisk troubled_ (n=troubled@pdpc/supporter/sustaining/troubled) |
| 16:44.18 | ManxPower | I must leave soon. The faster you work the better chance of a solution before I leave |
| 16:46.16 | flujan | 1135149399/1135149399 192.168.1.9 D 5060 OK (1 ms) |
| 16:47.01 | flujan | ManxPower: this is from box 1 |
| 16:47.07 | [TK]D-Fender | jblack: You about? |
| 16:47.11 | flujan | need something from box 2? |
| 16:47.22 | ManxPower | flujan: YOU have a NAT PROBLEM! |
| 16:47.46 | jeev | [TK]D-Fender, told the guy to f himself and i ordered the 330's. |
| 16:47.51 | *** join/#asterisk shinao1 (n=shinao1@41.219.221.101) |
| 16:47.54 | flujan | the machines are in the same network. 192.168.1.6 box1 |
| 16:49.02 | flujan | ManxPower: and 192.168.1.9 box 2 |
| 16:50.32 | ZPertee | [TK]D-Fender, so how do I setup Voicemail to email functionality, what all do I need to do this |
| 16:51.25 | [TK]D-Fender | ZPertee: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+IMAP+storage |
| 16:52.46 | ZPertee | [TK]D-Fender, I mean to do it your way |
| 16:53.12 | ZPertee | [TK]D-Fender, without IMAP storage just as attachment delete=yes |
| 16:53.15 | [TK]D-Fender | ZPertee: "my way" does not remove MWI on read, just permanently. |
| 16:53.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:53.51 | ZPertee | [TK]D-Fender, I know but "your way" is about a million times easier |
| 16:54.56 | [TK]D-Fender | ZPertee: and I already told you, you just aren't looking. |
| 16:55.16 | ManxPower | flujan: no they are not, you have a 200.x.x.x address for one and a 192.x.x.x. address for the other |
| 16:55.49 | ManxPower | flujan: someone else can help you, I must leave |
| 16:55.50 | *** part/#asterisk ManxPower (n=manxpowe@220.sub-75-203-100.myvzw.com) |
| 16:55.51 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 16:55.58 | flujan | ManxPower: I see, i am receiving the call in box 1 from a sip trunk. |
| 16:56.02 | watchy | i need some hot luvin |
| 16:56.12 | ZPertee | [TK]D-Fender, so I need to install Sendmail or something like that to get it work right |
| 16:56.26 | tzafrir | Or something like that |
| 16:56.30 | [TK]D-Fender | ZPertee: yup |
| 16:56.38 | watchy | i'm about to implement 20 501s and 10 601s, should i go with POE? |
| 16:56.54 | [TK]D-Fender | watchy: 501's = waste in most cases |
| 16:57.06 | watchy | what should i replace the 501s wityh |
| 16:57.18 | watchy | 301s? |
| 16:57.22 | [TK]D-Fender | watchy: I'd suggest IP 320/330 unless you intend on doing quite a bit with the microbrowser |
| 16:57.35 | watchy | nope, not really |
| 16:57.50 | [TK]D-Fender | watchy: IP 320/330 then. www.telephonydepot.com |
| 16:57.50 | watchy | that'd probably save quite a bit |
| 16:57.57 | [TK]D-Fender | watchy: Quite a lot. |
| 16:58.02 | [TK]D-Fender | watchy: And definietly PoE |
| 16:58.16 | watchy | Whats POE gonna give me advantage wise? |
| 16:58.23 | [TK]D-Fender | watchy: And I MIGHT streth the 601's to 650's just for the longer term outlook |
| 16:58.48 | watchy | will the sidecars work with the 650s |
| 16:59.00 | [TK]D-Fender | watchy: It'll cost you the same for the external power bricks, and PoE is more elegant and allows you the ability to power backup them in 1 spot |
| 16:59.11 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 16:59.12 | [TK]D-Fender | watchy: Yes, 601 & 650 sidecars |
| 16:59.36 | watchy | 650 sidecars is backlit? |
| 16:59.56 | sione | is it common when do "make -j8" that it fails then you do "make" it works just fine? |
| 17:00.02 | [TK]D-Fender | watchy: yup |
| 17:00.40 | watchy | ah ok i think i'll go with 650s and 330s instead of 501s and 601s |
| 17:01.01 | watchy | with the savings from 501s to 330s i can get the 650s |
| 17:01.34 | watchy | tk: what POE switch should i use? |
| 17:02.18 | *** join/#asterisk dlynes (n=dlynes@216.251.149.69) |
| 17:02.33 | [TK]D-Fender | watchy: I've been happy with D-Link's personally... |
| 17:02.58 | [TK]D-Fender | watchy: http://www.antonline.com/p_D-Link-Systems--Inc.--DES-1228P--Web-Smart-24-PORT-10-100-Poe-switch--4-Gigabit--2-Combo-Sfp-_275788.htm |
| 17:03.00 | watchy | ok next question, should the ip phones be in their own VLAN? |
| 17:03.24 | [TK]D-Fender | watchy: 340$ / 24 = 14.16$ / port. |
| 17:03.31 | watchy | nice |
| 17:03.52 | [TK]D-Fender | watchy: I tend to prefer to run completely separate wiring for phones. |
| 17:04.44 | watchy | so 1 drop for a pc and 1 for a phone? |
| 17:04.50 | watchy | so %100 seperate networks? |
| 17:04.51 | [TK]D-Fender | watchy: invest in the core wiring, and how & what you want to do can permutate on demand. Separate wiring means you could cross-connect a digital or TDM phone system to replace the setup you're about to install further down the line. |
| 17:04.56 | [TK]D-Fender | watchy: Yes. |
| 17:05.30 | watchy | ok, check this. this is a hard one on my end |
| 17:05.35 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 17:05.46 | [TK]D-Fender | watchy: And by doing this on a completely separate network, you could get IP320's instead of 330's,a dn save $25/phone for which to PAY for the wiring job. Smart deployments can pay for themselves in many cases |
| 17:05.57 | brodiem | anyone recommend a soft phone other than Xlite or the other counterpath products? |
| 17:05.58 | watchy | I'll have 2 phone systems but I want to connect them together |
| 17:06.32 | watchy | if I connect them via IAX2 will I beable to transfer calls across etc? |
| 17:06.37 | watchy | like they were in the same building |
| 17:06.51 | jsmith | watchy: Yes |
| 17:06.51 | dlynes | watchy: yes, it's all dependent on your dialplan |
| 17:06.52 | [TK]D-Fender | watchy: thats just *, |
| 17:07.04 | watchy | ah, well i'm gonna need help on that one |
| 17:07.18 | watchy | i guess I need to get the implementation done firsty |
| 17:07.18 | watchy | first |
| 17:07.21 | watchy | and wired etc |
| 17:07.44 | watchy | i'll have 30 phones on 1 building and 50 phones in another building |
| 17:08.01 | watchy | so i'll need 2 seperate servers correct? i don't want that much traffic going eithr way on 1 server |
| 17:08.21 | flujan | [TK]D-Fender: I create a summary of my problem which Manx was helping me with. Could you check it? |
| 17:08.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:09.26 | watchy | i'm trying to do this implementation correct the last one was so rushed its not exactly what i want |
| 17:10.05 | [TK]D-Fender | flujan: Sorry, can't dedicate that kind of attention span right now... |
| 17:10.46 | flujan | [TK]D-Fender: ok thanks. :) |
| 17:10.54 | *** part/#asterisk mfedy2 (n=Administ@adsl-71-134-153-203.dsl.irvnca.pacbell.net) |
| 17:10.57 | boblutz | o ok |
| 17:12.48 | *** join/#asterisk CVirus (n=GoD@82.201.178.187) |
| 17:14.03 | dlynes | I'm having an issue whereby dtmf is detected in voicemail(), I've got two-way audio on phone conversations, but voicemail() is still telling me there's no incoming audio from the caller, just after it receives a video frame (which is impossible, because i'm not using any devices that do video) |
| 17:14.04 | watchy | so tk i should use 2 seperate servers right? |
| 17:14.28 | dlynes | Does anyone have any ideas what might be causing something strange like that on asterisk 1.4.13/1.4.16.2? |
| 17:14.32 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 17:15.10 | dlynes | I've even tried setting silencethreshold as high as 1024, with no difference |
| 17:16.45 | rupa | watchy, doesn't hat depend on how much available bandwidth there is between buildings, the latency, and also your call patterns (do you expect to have a lot of cross building conversation or mostly within the building) |
| 17:16.49 | jeev | [TK]D-Fender you there |
| 17:17.09 | [TK]D-Fender | jeev: fire away |
| 17:17.23 | rupa | eg: if the buildings are on the same campus and you are running gigabit fiber between the buildings... why bother with two servers? |
| 17:17.23 | [TK]D-Fender | watchy: 2 servers for what? |
| 17:17.23 | jeev | i ordered the 330.. waiting for it to come, i can't do PoE.. i just can't. |
| 17:17.36 | [TK]D-Fender | jeev: Oh well, get a brick as well. |
| 17:17.47 | jeev | adapter? yea, it's not worth $15.. is it ? |
| 17:18.05 | [TK]D-Fender | jeev: Not if you can get it somewhere/how else for less :p |
| 17:18.06 | jeev | i dont like the sneekiness that site does, for an addition, it's $17.95.. but if you look carefully, on the right, it's $14.95 separate. |
| 17:18.18 | rupa | jeev, haha, I just did the same |
| 17:18.24 | jeev | that shits not cool man |
| 17:18.27 | jeev | that's immoral |
| 17:18.29 | jeev | stupid bastards |
| 17:18.35 | jeev | rupa, where you located? |
| 17:18.45 | rupa | jeev, texas |
| 17:18.46 | [TK]D-Fender | jeev: Or a product pricing/linking error. |
| 17:18.48 | jeev | how soon did they ship ? |
| 17:18.58 | rupa | jeev, dunno, ordered today |
| 17:19.13 | rupa | got tired of trying to load SIP firmware no a cheap siements optipoint 300 phone |
| 17:19.32 | jeev | ahh |
| 17:19.33 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 17:19.42 | jeev | ohwell |
| 17:19.48 | jeev | fender, how does it work? you put the config on the phone ? |
| 17:19.50 | jeev | and connect to ethernet ? |
| 17:19.56 | jeev | or you need someo ther type of config, tftp.. ? |
| 17:21.26 | [TK]D-Fender | jeev: on phone, via HTTP, provisioning. Many ways. Although if you do anything but provision you are a shmuck. |
| 17:21.49 | [TK]D-Fender | jeev: Go download the admin guide and look up "polycom provisioning" on the WIKi, and get reading. |
| 17:22.54 | rupa | anyone play witht he linksys ip300 or ip330 phones? |
| 17:23.57 | watchy | rupa: its about 6miles down the road |
| 17:24.07 | watchy | there is no direct link. except a wireless link |
| 17:24.21 | rupa | watchy, then two servers would make more sense |
| 17:24.27 | watchy | yea thats what i though |
| 17:24.45 | watchy | with a min of 30 phones at each location i thought it would be wise for 2 servers |
| 17:25.22 | rupa | or let me rephrase a question. if I wante dto replace our current wireless phones with sip phones -- what would you recommend? |
| 17:26.12 | jeev | ok cool fender |
| 17:26.21 | watchy | Sangoma T1 PCI or PCIE? |
| 17:28.54 | boblutz | rupa: Always leave room to add an explanation if it doesnt work out |
| 17:31.31 | rupa | oh, jeez, battery life for these suck... maybe a DECT phone that supports SIP at the base station |
| 17:32.35 | [TK]D-Fender | watchy: Either |
| 17:32.54 | CCFL_Man2 | $0.15 sms mobile originate message |
| 17:33.37 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 17:33.54 | [TK]D-Fender | rupa: WIP300 = **GARBAGE**. I wouldn't pay for the 330 either. |
| 17:34.06 | Qwell | 330 runs...Windows |
| 17:34.11 | CCFL_Man2 | [TK]D-Fender: i hear they are royal shit |
| 17:34.18 | rupa | [TK]D-Fender, battery life kills it for all these 802.11 handset thingies |
| 17:34.30 | Qwell | the joystick on the WIP300 is pretty slick though |
| 17:34.47 | alrs | I'm spectralink-curious |
| 17:35.03 | [TK]D-Fender | rupa: I've got one here that gets < 4 hours between small-talk & standby |
| 17:35.11 | CCFL_Man2 | there still are no good wireless phones? |
| 17:35.26 | [TK]D-Fender | ~wifivoip |
| 17:35.26 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 17:35.27 | Qwell | ATA + cordless handset |
| 17:35.39 | jjshoe | CCFL_Man2 aastra dect phone. |
| 17:35.42 | rupa | aastra has a SIP-DECT phone |
| 17:35.49 | jjshoe | CCFL_Man2 Haven't and won't try the snom dect phone. |
| 17:35.50 | Qwell | isn't that a desk phone too though? |
| 17:35.54 | jjshoe | Qwell no. |
| 17:36.04 | jjshoe | Qwell you're thinking 480ict |
| 17:36.08 | Qwell | yes |
| 17:36.08 | rupa | Qwell, they have a deskphone model too, but the SIP-DECT isn't |
| 17:36.12 | CCFL_Man2 | hmm.. |
| 17:36.15 | [TK]D-Fender | Qwell: I believe they have a more standalone solution now. |
| 17:36.18 | Qwell | neat |
| 17:36.24 | jjshoe | Qwell the range is incredible. |
| 17:36.27 | CCFL_Man2 | vinage uses the uniden dect phone, no? |
| 17:36.47 | jjshoe | Qwell pain in the ass to configure though |
| 17:37.05 | jeev | i want a 330 SO BAD |
| 17:37.11 | Qwell | jeev: no you don't |
| 17:37.14 | rupa | oh, ouch. ok, I have to get a RFP32 + the handsets, that looks expensive |
| 17:37.16 | Qwell | WIP330, you mean? |
| 17:37.17 | jeev | :< |
| 17:37.17 | CCFL_Man2 | jeev: don't give in! |
| 17:37.19 | jeev | yeas |
| 17:37.22 | jeev | dood it's so addictive |
| 17:37.22 | Qwell | no, you don't |
| 17:37.27 | jeev | everywhere i go, i have wifi.. |
| 17:37.31 | jeev | i'm tired of using my 2 cell phones |
| 17:37.39 | watchy | Sangoma T1 PCI or PCIE? |
| 17:37.40 | jeev | i CANT STOP wanting! |
| 17:37.42 | jjshoe | http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-13A15A11/03/hs.xsl/21540.htm |
| 17:38.00 | CCFL_Man2 | watchy: it doesn't matter |
| 17:38.03 | jjshoe | watchy what on earth are you trying to ask? |
| 17:38.15 | watchy | should i go with a PCI or PCIE card? |
| 17:38.28 | rupa | watchy, whatever you computer has |
| 17:38.36 | CCFL_Man2 | how about the ut starcom voip phone, is it shit or not? |
| 17:38.42 | seanbright | is there a way to have the whispering (ChanSpy with whisper mode turned on) mixed in with the recording produced on the channel via MixMonitor? |
| 17:38.50 | seanbright | an automagic way, that is. |
| 17:39.13 | [TK]D-Fender | watchy: Already answered, they make BOTH. |
| 17:39.33 | CCFL_Man2 | watchy: buy me a single port pci 5v model plz |
| 17:39.35 | watchy | yes i know |
| 17:39.38 | watchy | but which should i go with? |
| 17:39.55 | [TK]D-Fender | jjshoe: Who retails them currently? |
| 17:39.56 | watchy | PCIE is new so i was just curious if there were issues with it |
| 17:39.58 | jjshoe | watchy WHICH EVER YOUR COMPUTER SUPPORTS. |
| 17:40.06 | jjshoe | [TK]D-Fender don't know, we got them straight from aastra. |
| 17:40.07 | [TK]D-Fender | watchy: If possible, PCI. More transportable |
| 17:40.14 | watchy | ah |
| 17:40.24 | watchy | ok cool |
| 17:40.28 | [TK]D-Fender | jjshoe: Let us know when/where the general population can get them. |
| 17:40.39 | watchy | i think i got everything worked out for this install then. Thanks TK and rupa |
| 17:40.42 | watchy | and everyone else |
| 17:41.14 | jjshoe | [TK]D-Fender I'm not an aastra sales person :) |
| 17:41.23 | CCFL_Man2 | i am tempted to get a cisco 7920 and use chan_sccp ... |
| 17:41.32 | CCFL_Man2 | should i resist? |
| 17:41.38 | Qwell | yes |
| 17:41.44 | Qwell | chan_sccp is terrible |
| 17:41.49 | Qwell | and unmaintained |
| 17:41.53 | jjshoe | Qwell I thought you were working on sccp? |
| 17:41.57 | Qwell | skinny |
| 17:42.03 | jjshoe | ahh that's right |
| 17:42.10 | [TK]D-Fender | CCFL_Man2: and why on earth do you want to buy a dead-end phone like that? |
| 17:42.23 | rupa | the aastra SIP-DECT solution is pricey. 650+ for a base station and $250 per handset |
| 17:42.27 | Qwell | and yeah, getting a 7920 is a horrible idea, even with CCM |
| 17:42.45 | CCFL_Man2 | [TK]D-Fender: i hear it's the best wifi voip phone other than the sccp only firmware |
| 17:43.12 | CCFL_Man2 | ahh |
| 17:43.17 | [TK]D-Fender | Standard Li-ion battery life: up to 3.5-hour talk time, up to 30-hour standby - Extended Li-ion battery life: up to 4.25-hour talk time, up to 40-hour standby |
| 17:43.19 | Qwell | CCFL_Man2: the 7920 isn't even good |
| 17:43.25 | [TK]D-Fender | CCFL_Man2: Big bloody deal. |
| 17:43.29 | jjshoe | rupa pricey? if you say so. |
| 17:43.29 | Qwell | it's...monochrome |
| 17:43.36 | [TK]D-Fender | CCFL_Man2: And thats the spec sheet, not real-world |
| 17:43.39 | rupa | jjshoe, for a home solution, pricey |
| 17:43.45 | CCFL_Man2 | ahh |
| 17:43.52 | [TK]D-Fender | Home = ATA + cordless |
| 17:43.55 | jjshoe | rupa the quality is unmatched wirelessly in my experience |
| 17:43.55 | CCFL_Man2 | then i'll stay away from it |
| 17:44.18 | rupa | [TK]D-Fender, yeah, thats what I'm doing now... no biggie |
| 17:44.49 | CCFL_Man2 | though most cordless phones are crap too |
| 17:44.58 | CCFL_Man2 | except the panasonic ones |
| 17:45.57 | CCFL_Man2 | i miss Ma'Bell technology |
| 17:46.18 | jjshoe | it sucks when money is an object ;) |
| 17:46.23 | boblutz | pours some out for ma'bell |
| 17:46.36 | jjshoe | Qwell market has tanked out for your old neighborhood, houses are hella cheap over there now |
| 17:46.42 | CCFL_Man2 | boblutz: lol |
| 17:46.45 | Qwell | doesn't surprisie me |
| 17:46.56 | Qwell | it's...Covina FFS ;p |
| 17:47.04 | jjshoe | :P |
| 17:47.10 | jjshoe | I'm still waiting for the west side to tank :\ |
| 17:47.11 | Qwell | practically san bernardino |
| 17:47.57 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 17:48.08 | jjshoe | oh I thought you were in the norwalk area |
| 17:48.12 | Qwell | nah |
| 17:48.17 | southtel | Other than zttool, is there a good way to troubleshoot/debug T1 lines? |
| 17:48.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:48.51 | southtel | I inherrited an asterisk PBX at a remote office, and after months of working fine, it's suddenly showing "Red". |
| 17:48.57 | Qwell | jjshoe: 91791 |
| 17:49.09 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:49.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:49.46 | jjshoe | southtel what's your telco say? |
| 17:50.01 | southtel | They swear up and down that the circuit is fine all the way to the PBX. |
| 17:50.27 | jjshoe | southtel if they loop the t1 do you show it looped? |
| 17:50.56 | jjshoe | Qwell ah west covina area, I'm still in Hermosa Beach. |
| 17:51.12 | jjshoe | southtel do you have a loop back plug? |
| 17:51.34 | southtel | jjshoe: Well, I'm not on site, so even if I did, it wouldnt' do any good. |
| 17:52.11 | jjshoe | any monkey can plug a loop back plug in for you, but *shrug* |
| 17:52.25 | jjshoe | you could try restarting asterisk, maybe the t1 will restart, but I doubt it. |
| 17:52.41 | CCFL_Man2 | i forget if asterisk has any way to tell if there is an alarm, etc |
| 17:52.42 | southtel | Yeah, a full restart of the box didn't fix anything. |
| 17:53.03 | southtel | I'll call the telco and ask them to loop the t1. |
| 17:53.10 | CCFL_Man2 | it's a PRI? |
| 17:53.12 | southtel | (I don't know enough to ask the right questions) |
| 17:53.35 | jjshoe | I don't know how the telco can tell that it's fine all the way to the pbx, they can only tell to the csu. |
| 17:54.15 | jjshoe | you have a smart jack right? |
| 17:54.33 | CCFL_Man2 | jjshoe: the T1 card has a csu, no? |
| 17:55.15 | jjshoe | CCFL_Man2 yeah, sorry. |
| 17:55.38 | jjshoe | i mixed up csu and smart jack |
| 17:55.40 | southtel | By smart jack, you mean the CSU? |
| 17:55.59 | CCFL_Man2 | i think he just means smart jack |
| 17:56.20 | southtel | Now we're all confused. |
| 17:56.20 | CCFL_Man2 | can zaptel tell you if you have loss of frame, etc? |
| 17:56.27 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 17:56.34 | jjshoe | southtel I meant smart jack, not csu. |
| 17:56.45 | jjshoe | southtel smart jack takes a signal from the carrier and performs the loop. |
| 17:56.54 | southtel | Gotcha. |
| 17:57.08 | southtel | Let me find out. |
| 17:57.39 | CCFL_Man2 | jjshoe: the cpe can tell if the T1 is in loopback? |
| 17:58.12 | jjshoe | CCFL_Man2 yes. you'll get a fairly obvious error repeated continuously on the console. |
| 17:58.25 | CCFL_Man2 | ahh |
| 17:58.52 | CCFL_Man2 | jjshoe: if it's loss of frameing, will that say on the console? |
| 17:59.41 | jjshoe | CCFL_Man2 isn't that a yellow alarm? |
| 17:59.48 | CCFL_Man2 | it is |
| 18:00.05 | CCFL_Man2 | but what is a red alarm? |
| 18:00.25 | jjshoe | no timing. |
| 18:00.30 | Corydon76-dig | Incorrect |
| 18:00.42 | Corydon76-dig | Yellow alarm simply means the other side sees red |
| 18:00.43 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:00.52 | CCFL_Man2 | oh |
| 18:01.04 | Qwell | how can we get an alarm if the other side can't see us? O.o |
| 18:01.18 | Qwell | I never understood that |
| 18:01.30 | Corydon76-dig | Qwell: typically, yellow occurs because of local issues or when the span is recovering |
| 18:01.56 | Corydon76-dig | Qwell: you can receive just fine on one side, even if the other side isn't |
| 18:02.01 | Corydon76-dig | Remember, 2 pairs |
| 18:02.30 | jjshoe | Corydon76-dig is there a document up anywhere discussing the various alarm states of a pri? |
| 18:02.48 | Corydon76-dig | presumably, yes |
| 18:02.57 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:02.59 | Corydon76-dig | but other than red, yellow, and green, there's only 1 more |
| 18:03.10 | Qwell | blue? |
| 18:03.15 | CCFL_Man2 | green is normal, no? |
| 18:03.26 | Corydon76-dig | blue alarm, which means that the circuitry is sending all 1s |
| 18:03.37 | Corydon76-dig | correct, green is normal |
| 18:04.31 | CCFL_Man2 | they put blue leds in equipment? |
| 18:05.13 | Corydon76-dig | I believe the red/yellow/green alarms are already done with a multicolor led |
| 18:05.28 | jjshoe | yes, on sangoma and digium cards they are done that way |
| 18:05.37 | jjshoe | I've personally never experienced a blue alarm, so I have no comment there. |
| 18:05.52 | Corydon76-dig | I haven't, either. I don't know what would cause that |
| 18:06.23 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:06.43 | CCFL_Man2 | i thought there was an amber alarm |
| 18:06.49 | Corydon76-dig | http://www.techfest.com/networking/wan/t1.htm |
| 18:09.09 | jjshoe | Corydon76-dig thanks |
| 18:09.25 | jjshoe | blue alarm situation makes far more sense reading that :P |
| 18:13.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:14.41 | *** part/#asterisk korihor (n=humberto@190.74.120.245) |
| 18:15.36 | CCFL_Man2 | T1s are so cool |
| 18:15.52 | CCFL_Man2 | Ma'Bell did a great job on it |
| 18:16.03 | Corydon76-dig | I wish they were more affordable |
| 18:16.30 | Corydon76-dig | If I could get a hybrid data/voice PRI for $200 a month, I'd get one at home |
| 18:17.16 | CCFL_Man2 | Corydon76-dig: i cheat, i run a DSX1 signal from asterisk to a channel bank |
| 18:17.37 | Corydon76-dig | Cheater! |
| 18:17.43 | CCFL_Man2 | hehe |
| 18:18.18 | [TK]D-Fender | Corydon76-dig: I've got a data T1 & PRI here. I wish I could do dynamic data spead over BOTH as B-Chans get freed up.... XO offers that in the US in a bunch of places. |
| 18:18.39 | CCFL_Man2 | for cheap? |
| 18:18.45 | Corydon76-dig | [TK]D-Fender: no, they don't |
| 18:18.59 | Corydon76-dig | [TK]D-Fender: what they offer is a data T1 with SIP service |
| 18:19.43 | [TK]D-Fender | Corydon76-dig: What I saw showed a Cisco router that would take in 2 X T1 pump out a PRI & ethernet |
| 18:19.44 | CCFL_Man2 | i'd love if a CAS T1 with one channel to my house was affordable |
| 18:19.48 | Corydon76-dig | [TK]D-Fender: I've looked into it before for a customer. They are NOT doing dynamic management of a channelized T1, but delivering just a data T |
| 18:19.51 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:20.09 | [TK]D-Fender | Corydon76-dig: What I've heard / read in the specs. |
| 18:20.20 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 18:20.34 | Corydon76-dig | and then they deliver a PRI that only goes as far as their Adtran on the wall |
| 18:20.44 | CCFL_Man2 | [TK]D-Fender: i bet with a multiflex T1 VIC? |
| 18:21.01 | [TK]D-Fender | CCFL_Man2: it was a Cisco IAD24XX IIRC |
| 18:21.06 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 18:21.08 | CCFL_Man2 | ahh |
| 18:21.12 | [TK]D-Fender | I *think* that was the series |
| 18:21.17 | grandpapadot | Hey guys, what does this mean: Got SIP response 482 "Loop Detected" back from <this server's ip address> |
| 18:21.30 | [TK]D-Fender | grandpapadot: Means you talk to yourself too much. |
| 18:21.47 | Corydon76-dig | The ZapRAS technology is really cool, but I've yet to encounter a telco to support it |
| 18:22.03 | CCFL_Man2 | [TK]D-Fender: you can map channels from the incomming T1 i think |
| 18:22.33 | [TK]D-Fender | CCFL_Man2: I don't know the down & dirty details, jsut to overlaying idea. |
| 18:22.40 | CCFL_Man2 | [TK]D-Fender: ahh |
| 18:23.12 | grandpapadot | [TK]D-Fender: yea, I already know that, lol |
| 18:23.21 | [TK]D-Fender | CCFL_Man2: Free B's bond with T1 #2 so taht you get up to 24-47 B worth of data bonded dynamically |
| 18:23.46 | southtel | Sorry for the delay guys...I did check and there is a smart jack... |
| 18:23.56 | [TK]D-Fender | CCFL_Man2: Makes perfect sense... complicated? Dunno... SOUNDS like a solid concept. |
| 18:24.29 | southtel | They say that the loop test works, and suggested that I try to look for framing/timing errors on our equipment...I guess my question is about how to do that. |
| 18:24.45 | [TK]D-Fender | Corydon76-dig: I jsut wish I could get better than ADSL1 at home.... ADSL2+ isn't in town just yet, not VDSL/etc |
| 18:25.21 | [TK]D-Fender | southtel: Have you restarted the server? |
| 18:25.28 | southtel | I have |
| 18:25.39 | file | that reminds me! |
| 18:25.50 | file | [TK]D-Fender: Bell is finishing up taking over Aliant |
| 18:25.56 | southtel | Still showing yellow alarm in dmesg and red alarm in zttool. |
| 18:26.10 | file | [TK]D-Fender: I expect badness. |
| 18:26.13 | [TK]D-Fender | southtel: then pastebin the following : "zaptel.conf", "zapata.conf", "ztcfg -vvvv", "cat /proc/interrupts", and "dmesg" |
| 18:26.15 | jjshoe | southtel what did you see in the logs while it was looped? |
| 18:26.51 | southtel | jjshoe which logs? I didn't see anything that seemed to relate to zap. |
| 18:27.01 | jjshoe | southtel /var/log/asterisk/messages |
| 18:27.19 | jjshoe | southtel is asterisk even running currently? |
| 18:27.26 | CCFL_Man2 | i don't think they did a loopback] |
| 18:27.36 | southtel | Nothing. |
| 18:27.42 | grandpapadot | [TK]D-Fender: Yep, exactly what I was doing. Thanks. |
| 18:27.47 | jjshoe | southtel /var/log/asterisk/messages is an empty file? |
| 18:28.00 | southtel | No, just nothing during the loopback. |
| 18:28.12 | jjshoe | southtel what kind of card is in this box? |
| 18:28.22 | [TK]D-Fender | ~pb |
| 18:28.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:28.32 | southtel | TE 110P |
| 18:28.41 | [TK]D-Fender | southtel: Please pastebin whats been requested of you. |
| 18:28.44 | CCFL_Man2 | suma: i got the same card |
| 18:28.47 | southtel | Yeah, working on it. |
| 18:28.54 | CCFL_Man2 | err |
| 18:29.04 | southtel | (the worst part about this is that it's been working for quite some time just fine) |
| 18:29.05 | CCFL_Man2 | southtel: i got the same card |
| 18:29.36 | CCFL_Man2 | southtel: your T1 provider a clec? |
| 18:30.40 | kamanashisroy | hi, can asterisk call any module while a call is being processed ? Like bill in each 2 minutes .. is asterisk manager interface the only solution for this ? |
| 18:30.44 | southtel | Cox communications. |
| 18:31.06 | CCFL_Man2 | hah! cocks communications |
| 18:31.46 | CCFL_Man2 | cable companies are known for shitty support and things going down for no reason |
| 18:31.51 | southtel | I'll save any opinions for after I get this fixed. |
| 18:33.00 | CCFL_Man2 | southtel: i don't think they did the loopback |
| 18:33.25 | jjshoe | I highly doubt they would lie about looping the line. |
| 18:33.34 | Qwell | jjshoe: a telco? |
| 18:33.44 | Qwell | you're kidding, right? |
| 18:33.48 | jjshoe | Qwell no. |
| 18:34.01 | CCFL_Man2 | jjshoe: then why didn't he see anything in his dmesg? |
| 18:34.03 | Qwell | I would be shocked if a telco didn't lie... |
| 18:34.19 | jsmith | jjshoe: Often times they *think* they've looped up your line, but accidentally put a loop on somebody else's |
| 18:34.24 | jjshoe | Qwell detecting if a telco looped back the t1 or not is too easy to decide if they have lied. |
| 18:34.32 | jsmith | jjshoe: Telcos are not infallible |
| 18:34.34 | jjshoe | jsmith that's not a lie, that's an accident :P |
| 18:34.46 | CCFL_Man2 | thats a failure |
| 18:34.55 | jsmith | jjshoe: I've seen both accidents and outright lies |
| 18:35.08 | jjshoe | jsmith I've never had a telco lie about looping, but *shrug* to each their own |
| 18:35.14 | jsmith | jjshoe: Of course, I've seen just about everything a telco can throw at you |
| 18:35.27 | southtel | Well, they've said several things that made me think they're troubleshooting the wrong line. |
| 18:35.35 | CCFL_Man2 | it's cocks communications |
| 18:35.46 | CCFL_Man2 | almost as bad as cumschat |
| 18:36.23 | southtel | http://pastebin.com/m5c223a6e |
| 18:37.43 | CCFL_Man2 | ear and mouth signalling is so crappy |
| 18:38.21 | *** join/#asterisk frieze (n=frieze@cpe-69-203-15-230.nyc.res.rr.com) |
| 18:38.43 | southtel | They're "re-looping" now. |
| 18:38.49 | CCFL_Man2 | it's gotta be your provider |
| 18:39.06 | frieze | are there any recommended linux distros for running asterisk? I'm looking at ubuntu, redhat and suse |
| 18:39.12 | southtel | I hope you're right...I just don't know what else could be wrong otherwise. |
| 18:40.15 | rob0 | The recommended distro is the one you know how to use best. |
| 18:40.45 | frieze | so there are no weird digium driver hangups or anything? |
| 18:41.28 | rob0 | Um, for that you might want to search your chosen distro's bug database first. |
| 18:43.04 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 18:43.10 | CCFL_Man2 | southtel: i'd complain to high heaven |
| 18:43.32 | zackz | how can I find what exact version of zaptel i have installed? |
| 18:45.39 | CCFL_Man2 | why is ear and mouth signalling used? |
| 18:45.57 | southtel | CCFL_Man2: I hope you're not asking me. |
| 18:49.15 | CCFL_Man2 | southtel: oh, you aquired it |
| 18:49.39 | southtel | Yeah, didn't get the option to set it up otherwise. |
| 18:50.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:50.17 | CCFL_Man2 | i'd bet the problem is on there end |
| 18:50.28 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 18:50.48 | jsmith | E&M, E&M Wink, Feature D.... boy that brings back memories |
| 18:50.58 | jsmith | (of course, they're all memories I'd much rather forget) |
| 18:51.19 | CCFL_Man2 | jsmith: what aweful signalling |
| 18:51.31 | outtolunc | E&M gets the job done <G> |
| 18:51.53 | CCFL_Man2 | lies! |
| 18:53.22 | x86 | we're doing E&M wink here to the Telco |
| 18:53.27 | southtel | Well, they're claiming that it's looped right now, but I'm not seeing anything in /var/log/messages |
| 18:53.30 | x86 | best possible signalling on a non-PRI |
| 18:53.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:53.50 | CCFL_Man2 | x86: not CAS? |
| 18:54.03 | x86 | CCFL_Man2: E&M Wink is a CAS signalling method, yes |
| 18:54.49 | CCFL_Man2 | err, not fxs_ls |
| 18:56.00 | x86 | E&M Wink works far better than fxs_ls for sure |
| 18:56.08 | *** join/#asterisk quigon (n=matias@201.216.205.13) |
| 18:57.02 | *** join/#asterisk quigon (n=matias@201.216.205.13) |
| 18:58.28 | southtel | Any thoughts then about where else I might see logging other than /var/log/messages? |
| 18:59.40 | *** join/#asterisk pakipenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:00.10 | pakipenguin | hi |
| 19:01.23 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 19:03.01 | tzafrir | southtel, /var/log/asterisk/messages ? |
| 19:03.06 | *** join/#asterisk JEns0r (n=jens@92.227.192.38) |
| 19:03.09 | tzafrir | southtel, see /etc/asterisk/logger.conf |
| 19:04.01 | *** part/#asterisk JEns0r (n=jens@92.227.192.38) |
| 19:04.18 | southtel | Oh, yeah, misread the earlier...the conf is sending everything to full. |
| 19:05.04 | jackson__ | woohoo, found a cheapo used Motorola SM56 modem to play with as an FXO |
| 19:05.18 | southtel | And there's nothing in there about the loop. |
| 19:06.12 | grandpapadot | I have like 10 4-port digium cards if anyone wants them, $100 each. TDM400P with FXO daughter boards. |
| 19:08.55 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 19:10.20 | rob0 | no FXS? |
| 19:10.42 | grandpapadot | No... |
| 19:11.02 | grandpapadot | On Polycom 330's, is there a way to boost the microphone gain? It seems they are quieter (by default?) than our 501's ... |
| 19:12.04 | CCFL_Man2 | southtel: it's gotta be your provider |
| 19:12.14 | UnixDog | look in the sip.cfg |
| 19:12.15 | CCFL_Man2 | infact i'm almost sure it is |
| 19:12.23 | UnixDog | for rx tx gain |
| 19:12.50 | jackson__ | grandpapadot, the tdm400P's are populated with 4 fxo boards? |
| 19:12.58 | grandpapadot | Yep. |
| 19:13.38 | jackson__ | grandpapadot, any pitfall's I should know about with those particular cards? I'm new to the hardware fxo equipment. |
| 19:14.11 | lmadsen | jackson__: don't use the TDM400, use the new TDM410Ps with the new chipset |
| 19:14.20 | grandpapadot | Use the HSEC, but other than that they work pretty good. |
| 19:14.28 | lmadsen | TDM400P has an old chipset, and no H/W echo can. |
| 19:14.34 | *** join/#asterisk korihor (n=humberto@190.78.209.202) |
| 19:14.42 | UnixDog | but these card are cheaper |
| 19:14.49 | UnixDog | and need a good home |
| 19:14.53 | lmadsen | cheaper == crappier :) |
| 19:14.56 | UnixDog | if they work they work |
| 19:15.01 | UnixDog | get real |
| 19:15.07 | lmadsen | depends what you need them to do |
| 19:15.10 | grandpapadot | Just use the HSEC and you'll be fine. |
| 19:15.37 | rob0 | is sorely tempted |
| 19:16.04 | rob0 | only personal (SOHO) needs |
| 19:16.15 | grandpapadot | @lmadsen: Not always, your mother is really good and cheapest I've found... (bada bing -- sorry, obviously just kidding and couldn't resist) |
| 19:16.40 | lmadsen | notes he could just kick/ban grandpapadot :) |
| 19:16.51 | UnixDog | why |
| 19:16.58 | jackson__ | hehe |
| 19:17.06 | grandpapadot | notes he was kidding and didn't realize @lmadsen was sensative ... |
| 19:17.22 | rob0 | smiley |
| 19:17.31 | *** mode/#asterisk [-o lmadsen] by twisted |
| 19:17.33 | twisted | no you can't :P |
| 19:17.38 | UnixDog | all digium workers are overly sensative |
| 19:17.48 | *** mode/#asterisk [+o lmadsen] by twisted |
| 19:17.49 | lmadsen | grandpapadot: if I was sensitive, I'd just have done it |
| 19:17.53 | rob0 | grandpapadot needs new lenses to see the emoticons |
| 19:18.41 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 19:19.07 | rob0 | grandpapadot: /msg ? |
| 19:20.13 | nny_2 | Hi, getting sched_settime: Request to schedule in the past?!?! in a system's console and message log. I have read that this can mean that the cpu is being worked too hard, although my system load is real low (.08 max) and the system itself is 1 sip client and a 1.5 GHZ based box. Any other reasons why this error would appear? |
| 19:20.53 | C4colo | what source are you using for your zap timer? |
| 19:20.59 | C4colo | and is music on hold playing when you get that message? |
| 19:22.08 | nny_2 | C4colo: no moh when it happens, i am not sure how to check what zap it using for the timer |
| 19:22.27 | C4colo | is it in a virtual environment? such as xen? |
| 19:22.31 | nny_2 | C4colo: no |
| 19:22.37 | C4colo | hmm |
| 19:22.47 | C4colo | what processor is in the box? |
| 19:23.49 | nny_2 | C4colo: Intel(R) Celeron(R) CPU 220 @ 1.20GHz |
| 19:23.57 | nny_2 | ha excuse me 1.2 GHZ* |
| 19:24.07 | nny_2 | (really basic system) |
| 19:24.09 | southtel | By the way guys, it looks like our problem was on the telco side. |
| 19:24.31 | southtel | I finially got the tech to back off his "we're seeing the loop just fine" statements. |
| 19:24.51 | southtel | They're "checking their network more thourughly" right now. |
| 19:25.09 | *** join/#asterisk ManxPower (n=manxpowe@220.sub-75-203-100.myvzw.com) |
| 19:25.18 | [TK]D-Fender | nny_2: You don't know what your server is using for zaptel? |
| 19:25.34 | C4colo | well, of course they deny any problem on their end... you are running "non-standard" hardware from their perspective |
| 19:26.02 | nny_2 | [TK]D-Fender: it's a "stock" install more or less, I have never reviewed that information before (i.e. download zaptel source, make make install etc etc) |
| 19:26.16 | southtel | I never said the word "asterisk"...I know I'm as likely as not to hear a dialtone if I do. |
| 19:26.26 | [TK]D-Fender | nny_2: Well go look NOW. |
| 19:26.44 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 19:26.47 | ManxPower | nny_2: there really is no such thing as a "stock" asterisk. |
| 19:26.56 | ManxPower | Since you have to build every config file yourself. |
| 19:27.01 | nny_2 | yeah i understand |
| 19:27.03 | nny_2 | one sec |
| 19:27.06 | nny_2 | learning/looking |
| 19:27.07 | C4colo | cough*trixbox*cough |
| 19:27.10 | nny_2 | i know what the hardware is |
| 19:27.15 | nny_2 | lol not trixbox :D |
| 19:27.46 | C4colo | do you have physical zap hardware? |
| 19:27.52 | C4colo | or are you using ztdummy ? |
| 19:28.01 | nny_2 | it is using a TDM402B |
| 19:28.12 | C4colo | then you have a hardware zap timing source |
| 19:28.25 | nny_2 | k this is the first time i have used the newer TDM402B |
| 19:28.37 | C4colo | that rules out kernel/usb/rtc timing errors |
| 19:29.18 | [TK]D-Fender | make sure ztdummy isn't loading. |
| 19:29.21 | nny_2 | sorry yeah I am still/always a student, as far as the configs go, I am using modified extensions.conf, zapata.conf and sip.conf |
| 19:29.29 | nny_2 | [TK]D-Fender: let me check one sec |
| 19:29.31 | [TK]D-Fender | and verify your zaptel version. |
| 19:29.40 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:29.41 | [TK]D-Fender | nny_2: And then just stop using MP123 for MoH |
| 19:29.44 | [TK]D-Fender | MPG123 |
| 19:29.49 | nny_2 | not using and mp3s |
| 19:30.00 | nny_2 | standard 8 bit wav |
| 19:30.02 | nny_2 | mono* |
| 19:30.10 | nny_2 | zaptel 1.4.9.2 |
| 19:30.41 | nny_2 | the ztdummy module *is names "ztdummy" right? |
| 19:30.44 | nny_2 | named* |
| 19:31.39 | nny_2 | gah i suck, the moh files are: Microsoft PCM, 16 bit, mono 8000 Hz |
| 19:31.47 | nny_2 | 16 bit, 8 khz |
| 19:31.59 | robeph | how many amps in the 90vac for the ring? |
| 19:35.21 | nny_2 | fwiw zaptel is loading the full range of wc* modules, as we generally dont do a make menuconfig on install, if this is an issue, I can change it, but not sure if this is the source of my problem |
| 19:35.24 | *** part/#asterisk iNick (n=noident@nmbug.org) |
| 19:35.33 | nny_2 | zaptel 218320 17 zttranscode,xpp,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2 |
| 19:39.24 | nny_2 | also fwiw there is no apparent "problem" with the system, but I would prefer to figure out what it is and fix it vs. waiting for something to appear |
| 19:39.43 | nny_2 | ahh |
| 19:39.49 | nny_2 | another piece of info i didn |
| 19:39.52 | nny_2 | 't see* |
| 19:40.15 | nny_2 | getting chan_zap.c: Detected alarm on channel 1: No Alarm [Apr 16 11:56:48] NOTICE[26848] chan_zap.c: Alarm cleared on channel 1 |
| 19:42.53 | *** join/#asterisk droops (n=droops@74.193.237.138) |
| 19:44.25 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 19:51.49 | nny_2 | seems that the Alarm Warning is right after the call is hung up |
| 19:52.01 | nny_2 | any thoughts appreciated, insults a plus! |
| 19:52.56 | tzafrir | What version of Zaptel? |
| 19:53.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:53.13 | nny_2 | tzafrir 1.4.9.2 |
| 19:54.22 | tzafrir | It is generally caused by the temporary disconnect your provider uses to signal the end of a call |
| 19:54.24 | nny_2 | tzafrir not sure if you saw the other half of the conversation up there but also getting sched_settime: Request to schedule in the past?!?! in the log at some points, using TDM402B hardware and the cpu load is minimal |
| 19:55.18 | *** join/#asterisk ta^3 (n=tacvbo@189.146.192.175) |
| 19:55.56 | nny_2 | tzafrir that 1st error is generally a nonissue, as there is no problems caused by it, the later issue (the schd_settime one) is not presenting any problems AFAIK, but don't want to find out it is the hard way, if you know what i mean |
| 19:56.07 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 19:56.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:56.07 | *** join/#asterisk axisys (n=axisys@155.70.141.45) [NETSPLIT VICTIM] |
| 19:56.07 | *** join/#asterisk DarthclueWork (n=Darthclu@fw149.northside.isd.tenet.edu) [NETSPLIT VICTIM] |
| 19:56.08 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 19:56.08 | *** join/#asterisk bmg505 (n=leon@196-209-79-122-tbnb-esr-2.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 19:56.08 | *** join/#asterisk liri (n=liran@bzq-179-128-126.static.bezeqint.net) [NETSPLIT VICTIM] |
| 19:56.08 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) [NETSPLIT VICTIM] |
| 19:56.08 | *** join/#asterisk jblack (n=jblack@pool-72-79-189-156.sctnpa.east.verizon.net) [NETSPLIT VICTIM] |
| 19:56.08 | *** mode/#asterisk [+o denon] by irc.freenode.net |
| 19:58.00 | ac1djazz | i got the mp3 extension goin w/ asterisk, any reason i cant stream an mp3 via the AGI? |
| 19:58.27 | jsmith-away | Nope... as long as you've got format_mp3 working, it *should* work |
| 19:58.27 | ac1djazz | using STREAM FILE ? |
| 19:58.39 | ac1djazz | jsmith-away: how can i test to see if i have format_mp3 working? |
| 19:58.55 | jsmith | "core show file formats" |
| 20:00.07 | ac1djazz | Format Name Extensions |
| 20:00.07 | ac1djazz | slin mp3 mp3 |
| 20:00.12 | ac1djazz | that the one? |
| 20:00.42 | rupa | in my silly quest to get this old siemens optipoint 300 to work with asterisk I got h323 working (since I can't get the SIP firmware to load). I can directly dial the IP addr of asterisk and the call is routed correctly, but I end up with the error chan_oh323.c:3324 setup_h323_connection: Call 'ip$192.168.1.22:1048/25910-b4b4c875' requests connection re-setup (1). <sigh> |
| 20:00.56 | rupa | (netmeeting works) |
| 20:01.02 | rupa | and the phone can call netmeeting too |
| 20:01.42 | *** join/#asterisk asdx (n=diego@adsl-152-180.click.com.py) |
| 20:02.29 | ac1djazz | <PROTECTED> |
| 20:02.38 | ac1djazz | i dont need to put .mp3 right |
| 20:02.42 | rupa | no |
| 20:02.48 | asdx | do i need to specify something in my dialplan for incoming calls? |
| 20:02.51 | ac1djazz | do i need to convert it to like 8kbps? |
| 20:02.54 | asdx | or i just have to register to the provider? |
| 20:03.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:03.53 | grandpapadot | Does asterisk 1.2.24 properly support SRV records in sip peer definitions? |
| 20:04.00 | [TK]D-Fender | asdx: *duh* |
| 20:04.23 | [TK]D-Fender | asdx: you aren't processing a call if you aren't in your dialplan. |
| 20:04.33 | asdx | [TK]D-Fender: right |
| 20:04.55 | asdx | [TK]D-Fender: sorry, i'm new with asterisk |
| 20:05.00 | *** join/#asterisk af_ (n=getsmart@88-149-241-246.dynamic.ngi.it) |
| 20:05.16 | nny_2 | asdx: have you gotten a hold of the asterisk book yet |
| 20:05.25 | nny_2 | ~thebook |
| 20:05.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:05.42 | asdx | i read that from time to time |
| 20:05.43 | nny_2 | may help with some basic tutoring on how to construct your dial plan etc |
| 20:05.50 | nny_2 | asdx: yeah it helps loads :D |
| 20:06.08 | ac1djazz | is there any reason my mp3 music file wont play vai agi doing STREAM FILE song ? |
| 20:06.54 | grandpapadot | More to the point, how can I load balance outbound calls to two different SIP proxies in my dialplan? (Asterisk 1.2.24) |
| 20:07.43 | jsmith | ac1djazz: And you're sure you're passing the proper parameters, etc. to the STREAM FILE command? |
| 20:07.50 | grandpapadot | Just use GROUP()? |
| 20:08.00 | ac1djazz | jsmith: if i change it to 'temperature' which is a gsm sound it plays just find |
| 20:08.34 | Qwell | ac1djazz: yes, 8khz |
| 20:08.53 | ac1djazz | qwell; thast what i thought, how do i do that? thorugh lame? you konw the command? |
| 20:10.12 | asdx | [TK]D-Fender: teliax gave me a DID number, i want to use that for my incoming calls, where should i look for this? |
| 20:10.15 | asdx | :S |
| 20:10.47 | asdx | the book i guess |
| 20:11.34 | ac1djazz | any1 in here konw how to converrt an mp3 to 8khz using something like lame? |
| 20:11.59 | asdx | i tried calling from skype to the DID number but i hear "tu tu tu tu" as if the line is busy or don't exist |
| 20:12.18 | nny_2 | asdx: i generally have my [default] (they may use [incoming] set to exten => s,1,Goto(your desired path) with a _x.,1 and a _x, below it |
| 20:14.03 | nny_2 | so s,1,Goto(mainmenu,s,1) _x.,1,Goto(mainmenu,s,1) and _x,1,Goto(mainmenu,s,1) you may want it to just go to your defined extension(s) like Goto(people,100,1) , and then in [people] define a way for it to reach the intended peer etc |
| 20:14.27 | ManxPower | asdx: There is NO difference between a regular IAX/SIP device and Teliax. |
| 20:14.38 | ManxPower | nny_2: The incoming call will NEVER EVER match "s" |
| 20:14.44 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 20:14.55 | nny_2 | ManxPower: eh yeah, why the hell is* that there? |
| 20:15.01 | ManxPower | _x will also NEVER EVER match the DID |
| 20:15.13 | ccvp | hello fellow internet addicts |
| 20:15.13 | ccvp | are we all looking forward to another long & glorious week of irc/internet addiction? :) |
| 20:15.17 | asdx | thanks guys |
| 20:15.26 | ManxPower | nny_2: "s" is for FXO signaled ports (they don't support sending the dialed number to Asterisk" |
| 20:15.26 | asdx | i changed it to type=peer and it stays in "connecting" |
| 20:15.27 | asdx | now |
| 20:15.37 | ManxPower | asdx: you want type=friend |
| 20:15.51 | ManxPower | asdx: What is your actual DID? |
| 20:16.02 | asdx | (202) 470-5265 |
| 20:16.17 | rupa | quick,s tart dialing :) |
| 20:16.25 | ManxPower | exten => 2024705265,1,Goto(destcontext,destexten,1) |
| 20:16.44 | ccvp | Manks Pow-Urrrr |
| 20:16.54 | ManxPower | that will send incoming calls to that DID to whatever extension you put in for destexten in whatever context you put in destcontext |
| 20:16.55 | asdx | ManxPower: nice, can i still use _X. ? |
| 20:17.04 | ManxPower | asdx: Yes, but don't. |
| 20:17.10 | ManxPower | It will screw you up |
| 20:17.13 | ac1djazz | i converted my mp3 to 8khz using lame w/ lame -q -p --mp3input -a --preset 8 -m mono heat.mp3 8kheat.mp3 , im trying to play it via AGI via STREAM FILE music/8kheat and it just hangs up |
| 20:17.17 | ManxPower | Especially if you ever get a 2nd did |
| 20:17.20 | ac1djazz | <PROTECTED> |
| 20:17.20 | ac1djazz | <PROTECTED> |
| 20:17.27 | asdx | ManxPower: ok |
| 20:17.35 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 20:17.44 | *** join/#asterisk asencioles (n=andre@200.216.205.42) |
| 20:17.51 | nny_2 | ManxPower: heh that is from someone who originally set up our first basic dialplan, i will review why it is bad and sdjust accordingly |
| 20:18.29 | ManxPower | nny_2: If you have FXO ports _X silly and if you have FXS, SIP, IAX it is overly broad match |
| 20:18.46 | asencioles | guys, is there a channel specific to asterisk w/ libss7? |
| 20:19.03 | nny_2 | ManxPower: I think the original intention was to send all to the same context regardless |
| 20:19.27 | ManxPower | ac1djazz: stop asterisk, start asterisk as "asterisk -cvvv" and rerun your scipt, you should get more AGI info when you run asterisk in the coreground |
| 20:19.43 | ManxPower | and foreground too |
| 20:20.04 | asdx | [Apr 16 15:19:42] NOTICE[5372]: chan_sip.c:13888 handle_request_invite: Failed to authenticate user "0000123456" <sip:0000123456@63.211.239.14>;tag=as79d84056 |
| 20:20.10 | asdx | i get that when i call |
| 20:20.28 | nny_2 | ManxPower: which it does, but your saying that s is useless unless i use FXO signaled ports, and _x. is overly broad? |
| 20:20.36 | ac1djazz | ManxPower: -cvvv as a service? or for the CLI? |
| 20:20.53 | nny_2 | and _x is useless entirely as it states a pattern that should never exist |
| 20:21.08 | nny_2 | I probably should have reviewed this earlier before trying to help someone lol |
| 20:22.29 | ac1djazz | ManxPower: -> |
| 20:22.30 | ac1djazz | <PROTECTED> |
| 20:22.30 | ac1djazz | <PROTECTED> |
| 20:22.30 | ac1djazz | <PROTECTED> |
| 20:22.34 | ac1djazz | no errors |
| 20:22.34 | *** join/#asterisk Meaw (n=dino@a151-171.adsl.paltel.net) |
| 20:22.35 | ac1djazz | just hangs up |
| 20:22.38 | ac1djazz | and this file is a whole song |
| 20:22.57 | ccvp | acidjazz |
| 20:22.59 | fmueller | ac1djazz: can you paste your agi-scipt somewhere in the internet? |
| 20:22.59 | ccvp | ircle 3.1 sucks |
| 20:23.05 | SomethingISODD | Hello question with an agi script is there anyway to get the sip callers ip address |
| 20:23.08 | ac1djazz | [root@whatdood(/var/lib/asterisk/sounds/music)]: file 8kheat.mp3 |
| 20:23.08 | ac1djazz | 8kheat.mp3: MPEG ADTS, layer III, v2.5, 32 kBits, 8 kHz, Monaural |
| 20:23.21 | nny_2 | ManxPower: ironically, it doesn't work wihthout s,1 |
| 20:23.23 | nny_2 | without* |
| 20:23.23 | jjshoe | asdx is still trying to setup his teliax account? :P |
| 20:23.30 | ac1djazz | fmueller: the script is fine cuz when i replace music/9kheat w/ demo or temperature it plays the sound fine |
| 20:23.39 | asdx | jjshoe: yeah lol |
| 20:23.43 | nny_2 | Executing [s@default:1] Goto("Zap/1-1", "mainmenu|s|1") in new stack |
| 20:23.50 | Kobaz | i have a polycom phone connect to my box, and i make a call, i route the call out a sip account and i get no audio |
| 20:23.51 | ccvp | acidjazz |
| 20:23.54 | ccvp | "cuz" ? |
| 20:23.55 | asdx | jjshoe: my boss gave me a skype account to test incoming calls |
| 20:23.56 | ac1djazz | yes ccvp |
| 20:23.57 | ccvp | whose your cuz? |
| 20:23.59 | Kobaz | i connect using a sip soft phone and i do get audio... anyone have any idea? |
| 20:24.01 | ac1djazz | lol |
| 20:24.14 | ccvp | irc slang shorthand for "because" |
| 20:24.17 | ccvp | or "brother" HEH |
| 20:24.47 | asdx | jjshoe: outgoing works fine |
| 20:24.53 | asdx | jjshoe: incoming is what i'm trying to do now |
| 20:25.28 | fmueller | ac1djazz: ahh, ok. can you do 'core set verbose 6' and retry it please? |
| 20:25.31 | ccvp | asdx, i was doing that 'cuz' my codec was to strong for my weak system |
| 20:25.43 | asdx | http://pastebin.com/m5936dcb0 |
| 20:25.45 | asdx | i'm getting that |
| 20:25.51 | asdx | my sip account is registered |
| 20:26.04 | asdx | when i dial from skype to * i get that in the asterisk cli |
| 20:26.24 | asdx | ccvp: ah |
| 20:26.26 | *** part/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 20:27.02 | ac1djazz | fmueller: same output |
| 20:27.03 | ac1djazz | <PROTECTED> |
| 20:27.03 | ac1djazz | <PROTECTED> |
| 20:27.03 | ac1djazz | <PROTECTED> |
| 20:28.04 | fmueller | ac1djazz: ok, can you do "agi debug" and retry please and paste the output to pastebin? |
| 20:28.13 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-d1d2054b964ce151) |
| 20:28.22 | ac1djazz | fmueller: yesir |
| 20:28.26 | mknerd | g'day everyone |
| 20:28.36 | jsmith | g'day, mkillebrew |
| 20:28.41 | jsmith | g'day, mknerd |
| 20:28.50 | jsmith | yells at his tab-completion |
| 20:28.57 | mknerd | been using asterisk for awhile now for voip, just got my first TDM4xx card today |
| 20:29.15 | ac1djazz | fmueller: http://pastebin.com/m14963283 |
| 20:29.41 | mknerd | I believe I have it all working as far as the OS, but I am having trouble with the FXS and an analog phone, I want to be able to dial a SIP extention |
| 20:29.44 | jjshoe | asdx put your broken sip.conf stuff for teliax on pastebin. |
| 20:29.55 | asdx | jjshoe: k |
| 20:30.20 | mknerd | I get starting simple switch on 'Zap/2-1' and then Hungup 'Zap/2-1' |
| 20:30.21 | fmueller | ac1djazz: ok, thanks. i inspect it. |
| 20:30.25 | mknerd | not even a chance to dial |
| 20:31.22 | jjshoe | mknerd new or used card? |
| 20:31.22 | mknerd | If I dial the extension for the zap/2 from a sip extension, it says invalid extension |
| 20:31.25 | mknerd | new |
| 20:31.40 | mknerd | i do get a dialtone |
| 20:31.45 | fmueller | ac1djazz: well, crazy. sorry. i cannot help you. |
| 20:32.16 | ac1djazz | fmueller: itneresting right? |
| 20:32.21 | ac1djazz | fmueller: you think the file is too bigt? |
| 20:32.33 | ac1djazz | fmueller: should i not have converted it to 8khz? maybe i converted it wrong? |
| 20:32.45 | asdx | jjshoe: http://pastebin.com/m157d6070 |
| 20:34.32 | jjshoe | asdx try changing [teliax] to [adilwali] |
| 20:34.41 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 20:35.47 | asdx | jjshoe: [foo] is different for incoming calls? |
| 20:35.49 | asdx | i mean, the sections |
| 20:36.07 | asdx | [Apr 16 15:35:17] NOTICE[5372]: chan_sip.c:13952 handle_request_invite: Call from 'adilwali' to extension 's' rejected because extension not found. |
| 20:36.11 | asdx | i get that now |
| 20:36.15 | asdx | interesting |
| 20:36.28 | asencioles | anyone using libss7? |
| 20:36.35 | asdx | looks like i need extention 's' right? |
| 20:37.09 | jjshoe | what do you think? |
| 20:37.15 | mknerd | what else do I need to call a zap extension from a sip extension? |
| 20:37.15 | jsmith | mknerd: Can you pastebin your zapata.conf file? |
| 20:37.22 | mknerd | roger |
| 20:37.34 | fmueller | ac1djazz: interesting, indeed. what is the size of that file? but actually i do not think that this is the reason ... |
| 20:37.40 | mknerd | http://mibbit.com/pb/ad1V46 |
| 20:37.42 | jsmith | mknerd: It should be as simple as Dial(Zap/2), assuming you're dialing zap channel 2 |
| 20:38.01 | jsmith | mknerd: (Watch the capitalization ... it needs to be Zap, not zap or ZAP) |
| 20:38.04 | asdx | jjshoe: i think it needs a 's' extention, but why? |
| 20:38.19 | jjshoe | asdx when are you going to read the book? |
| 20:38.30 | *** join/#asterisk RoyK (n=roy@91.149.31.152) |
| 20:38.34 | mknerd | exten => 2224,1,Dial(Zap/2) |
| 20:38.44 | *** join/#asterisk RoyK (n=roy@91.149.31.152) |
| 20:39.00 | ac1djazz | fmueller: -rw-r--r-- 1 root root 381096 Apr 16 12:55 8kheat.mp3 |
| 20:39.01 | ac1djazz | -rw-rw-r-- 1 acidjazz acidjazz 14811136 Apr 16 11:03 heat.mp3 |
| 20:39.14 | mknerd | chan_sip.c:13952 handle_request_invite: Call from '204' to extension '2224' rejected because extension not found. |
| 20:39.22 | *** join/#asterisk dssman (n=no@209.8.41.75) |
| 20:39.44 | dssman | Anyone around to gimme a quick pointer on a dial plan? |
| 20:39.47 | mknerd | 14 meg mp3? |
| 20:39.54 | mknerd | dont ask to ask |
| 20:39.56 | hwt | has anyone gotten BLF + call-pickup using the function keys (not *8XX) on the Thomson ST2030? |
| 20:40.06 | ac1djazz | <PROTECTED> |
| 20:40.10 | hwt | thomson has some docs, but that's only using *XXX. |
| 20:40.14 | asdx | jjshoe: should i read all or just the incoming calls part? |
| 20:40.17 | hwt | for asterisk 1.4 |
| 20:40.32 | hwt | all i can find is an undocumented patch for 1.2.12.1 :) |
| 20:41.13 | fmueller | ac1djazz: what happens when you try to playback heat.mp3 with stream file? |
| 20:41.26 | ac1djazz | fmueller: same thing exactly |
| 20:41.32 | jsmith | mknerd: It sounds like you don't have an extension 2224 in the context that the SIP phone is sending calls to |
| 20:41.39 | ac1djazz | fmueller: im gonna try .wav |
| 20:41.48 | jsmith | mknerd: First do a "sip show users" and see what context the calls from that SIP phone are coming into |
| 20:42.24 | jsmith | mknerd: Then do a "dialplan show 2224@yourcontext", where "yourcontext" is the name of the context you got from the previous step |
| 20:42.35 | fmueller | ac1djazz: ok, let me know the result ... |
| 20:42.39 | dssman | okay... my paste bin is: http://pastebin.com/d98d7061. I have all calls going thru new-sip right now, but I am getting busy here... how can I make it roll over to diallink2voip? |
| 20:43.14 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 20:43.37 | dssman | and I would like voicenetwork to be the primary trunk still |
| 20:45.04 | mknerd | jsmith, my context for sip is local |
| 20:45.26 | *** part/#asterisk asencioles (n=andre@200.216.205.42) |
| 20:45.31 | jsmith | mknerd: And the output of "dialplan show 2224@local"? |
| 20:45.34 | ac1djazz | lame --decode --preset 8 -m mono heat.mp3 8kwavheat.wav |
| 20:45.41 | ac1djazz | fmueller: i dont ever have to specify the file extension right? |
| 20:45.47 | mknerd | There is no existence of 'local' context |
| 20:45.53 | mknerd | is that right? |
| 20:46.04 | mknerd | should I change it in sip.conf to default? |
| 20:46.29 | fmueller | ac1djazz: i think that is true, yes. |
| 20:46.58 | ac1djazz | fmueller: |
| 20:46.58 | ac1djazz | [Apr 16 13:27:28] WARNING[26727]: format_wav.c:148 check_header: Not in mono 2 |
| 20:46.58 | ac1djazz | [Apr 16 13:27:28] WARNING[26727]: file.c:322 fn_wrapper: Unable to open format wav |
| 20:48.03 | ac1djazz | sigh |
| 20:48.49 | fmueller | ac1djazz: argh ... |
| 20:48.54 | ac1djazz | fmueller: you konw what? |
| 20:48.57 | ac1djazz | when i do a core show codecs audio |
| 20:49.01 | ac1djazz | i dont see mp3 |
| 20:49.13 | ac1djazz | fmueller: how do i mkae sure format_mp3 is loaded? |
| 20:50.03 | ZPertee | why doesn't asterisk answer the call immediately? rings 2X |
| 20:50.28 | Kobaz | ZPertee: do you have a long delay in your answer() |
| 20:51.42 | fmueller | ac1djazz: i did not know the command you mentioned. but when i type it i also get: |
| 20:51.44 | fmueller | Disclaimer: this command is for informational purposes only. |
| 20:51.45 | fmueller | <PROTECTED> |
| 20:51.45 | fmueller | <PROTECTED> |
| 20:51.53 | ZPertee | Kobaz, no I just have pots coming in to the context and exten => s,1,Answer() exten => s,2,NoOP(do somethin else) |
| 20:52.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:53.17 | jsmith | mknerd: Yeah... if your user in sip.conf is pointed at a context that doesn't exist, then you aren't going to be able to make very many calls from your SIP phone |
| 20:53.29 | *** join/#asterisk sacitec (n=tobi@201.144.211.83) |
| 20:55.49 | grandpapadot | Hi all. Did the g726 issues get fixed in the 1.4 trunk? |
| 20:56.23 | Kobaz | ZPertee: your provider is delaying it |
| 20:56.34 | sacitec | hello, i'm working with asterisk 1.2.x over Centos 5, i have plans to link my asterisk box with Cisco UCM 6.1. What is your recommendation ? Compile H.323 support or use SIP ?. |
| 20:56.37 | jsmith | grandpapadot: That depends on which g726 issues you're referring to. Do you have bug numbers? |
| 20:56.45 | jsmith | sacitec: Use SIP |
| 20:57.07 | ZPertee | Kobaz, ok thanks |
| 20:57.14 | sacitec | thanks so much |
| 20:57.18 | grandpapadot | jsmith: I meant in general. I know there was an issue a while back to where you raised your voice and it would distort the sound. |
| 20:57.19 | fmueller | ac1djazz: what asterisk version do you use by the way? (show version) |
| 20:57.29 | ZPertee | Kobaz, I have asterisk connected as extension on Avaya PBX probably the problem |
| 20:57.57 | jsmith | grandpapadot: I'm not aware of any such issues... if they haven't been reported on the bug tracker, they don't exist ;-) |
| 20:58.19 | pakipenguin | hi |
| 20:58.19 | grandpapadot | Ok, cool. |
| 20:58.31 | asdx | jjshoe: added the 's' extention, works! :D |
| 20:58.33 | pakipenguin | a small question , which text to speech engine , opensource / free is the best? |
| 20:58.51 | asdx | now i just need to figure out why it wanted the 's' extention |
| 20:58.54 | grandpapadot | pakipenguin: cepstral is best, imho, but it's not free |
| 20:59.06 | *** join/#asterisk anthm (n=anthm@adsl-69-210-62-125.dsl.milwwi.ameritech.net) |
| 20:59.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:59.41 | jsmith | pakipenguin: Festival or flite |
| 20:59.57 | jsmith | pakipenguin: Cepstral is much better, but not free. (That being said, it's fairly inexpensive) |
| 21:00.03 | pakipenguin | hmmms i see |
| 21:00.31 | pakipenguin | thanks :D i'll look into them |
| 21:00.56 | ac1djazz | fmueller: Asterisk 1.4.19 built by acidjazz @ whatdood.com on a i686 running Linux on 2008-04-11 22:06:24 UTC |
| 21:01.36 | fmueller | ac1djazz: ok. you mentioned format_mp3 several times know. is there a format_mp3 in the sources? i cannot find it. |
| 21:01.54 | lmadsen | probably in addons |
| 21:01.59 | lmadsen | since mp3 would have licensing issues |
| 21:01.59 | ac1djazz | fmueller: its in addons |
| 21:02.15 | ac1djazz | if i didnt have mp3 support i think itd fail..? |
| 21:02.44 | ac1djazz | ok here's a question |
| 21:02.57 | ac1djazz | in AGI, what would be the recommended commadn to play an MP3 file? |
| 21:03.05 | dlynes | I'm having an issue whereby dtmf is detected in voicemail(), I've got two-way audio on phone conversations, but voicemail() is still telling me there's no incoming audio from the caller, just after it receives a video frame (which is impossible, because i'm not using any devices that do video) |
| 21:03.07 | dlynes | Does anyone have any ideas what might be causing something strange like that on asterisk 1.4.13/1.4.16.2? |
| 21:03.08 | dlynes | I've even tried setting silencethreshold as high as 1024, with no difference |
| 21:03.20 | Kobaz | dlynes: check your phone config |
| 21:03.21 | mknerd | jsmith, thanks for your help, i see that my problem lies in contexts now, i need to do some reading .. |
| 21:03.37 | dlynes | Kobaz: check my phone config? |
| 21:03.42 | dlynes | Kobaz: what specifically? |
| 21:03.47 | Kobaz | dlynes: dtmf handling |
| 21:04.04 | Kobaz | dlynes: make sure it's sending it in rtp as well as audio |
| 21:04.13 | dlynes | Kobaz: the dtmf is working; it's the caller's voice that asterisk voicemail is saying is non-existent |
| 21:04.43 | dlynes | Kobaz: and it's not sip; it's iax2 |
| 21:04.52 | jsmith | mknerd: No worries |
| 21:05.11 | dlynes | Kobaz: so there is no phone config |
| 21:05.24 | Kobaz | mmmm |
| 21:05.35 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:06.09 | dlynes | Kobaz: the only thing abnormal about the set up is that the phone system is being port mapped (ITSP -> port 4570 on nat -> port 4570 on asterisk) |
| 21:06.25 | dlynes | Kobaz: because the NAT already has an asterisk phone system running on port 4569 |
| 21:06.33 | Kobaz | oh i misread, hhe, i was also on the phone |
| 21:06.38 | fmueller | ac1djazz: sorry, but i have no idea how to playback mp3. i will try to find out how to do that in the next days though. |
| 21:07.02 | Kobaz | mmm port forwarding evil |
| 21:07.30 | Kobaz | callers voice? |
| 21:07.33 | Kobaz | what exactly is the error |
| 21:08.35 | dlynes | Kobaz: [Apr 16 13:26:46] VERBOSE[15652] logger.c: -- IAX2/sol21_experion-12 requested a video update, passing it to IAX2/experion-9 |
| 21:09.09 | dlynes | Kobaz: I get that logger output, and then it says the caller is silent (voicemail check, I'm guessing, from the silence code), and then the call is disconnected |
| 21:09.53 | dlynes | Kobaz: the particular silence error is: [Apr 16 13:26:50] WARNING[15653] app.c: No audio available on IAX2/sol21_experion-13?? |
| 21:10.35 | dlynes | but like i said...dtmf is working fine from voicemail(), as I can hit '*' to make it jump back out to the dialplan to run voicemailmain() |
| 21:10.58 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:11.42 | fmueller | ac1djazz: i have just found this note in format_mp3.c: MP3 format [Any rate but 8000hz mono is optimal] |
| 21:12.49 | fmueller | ac1djazz: if i don´t get it wrong that means that 8000hz mono is not the best decision. |
| 21:12.54 | ac1djazz | fmueller: i was able to play the file via exec playback |
| 21:13.08 | ac1djazz | lol sounds like crap |
| 21:13.29 | *** join/#asterisk jfg_ (n=jfg@dyn-88-121-43-115.ppp.tiscali.fr) |
| 21:14.30 | ZPertee | is the flash() application the equivalent of a switchook? |
| 21:14.51 | dlynes | ZPertee: flash() is the equivalent of the 'link' button on a Nortel system |
| 21:16.20 | *** join/#asterisk intralanman (n=lanman@207.44.172.12) |
| 21:16.25 | ZPertee | dlynes, I am trying to get asterisk to work with an Avaya system. So I would say that it should do the trick based on what you said |
| 21:17.31 | dlynes | ZPertee: yeah...i know nothing about avaya, but if the switchhook on the avaya does the same as a link button on a nortel system then, yeah |
| 21:19.47 | ZPertee | dlynes, I know nothing of Nortel so I guess all I can do is just experiment and see what happens :-) |
| 21:20.31 | jjshoe | asdx isn't it amazing what you can learn when you read? |
| 21:28.03 | ac1djazz | ok so is it at all possible to send text messages via asterisk - iax2 - VOIP ? |
| 21:28.20 | jjshoe | ac1djazz yes, there's an sms command. |
| 21:28.29 | ac1djazz | SendText ? |
| 21:28.35 | ac1djazz | in the agi or callplan? |
| 21:28.59 | jjshoe | [14:28] <jjshoe> ac1djazz yes, there's an sms command. |
| 21:29.11 | jjshoe | s/sms/SMS/ |
| 21:29.35 | ac1djazz | where can i learn how to use it |
| 21:29.45 | ac1djazz | and send a test text to myself via my setup :) |
| 21:29.46 | asdx | jjshoe: indeed |
| 21:30.13 | jjshoe | ac1djazz voip-info.org |
| 21:32.18 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 21:32.18 | *** join/#asterisk intralanman (n=lanman@207.44.172.12) |
| 21:35.34 | fmueller | ac1djazz: you wrote that you were able to playback the mp3 with exec playback. so that solved your problem, didn´t it? |
| 21:35.48 | ac1djazz | fmueller: yes in voip info ti says that FILE ASTREAM is buggy and should not be used |
| 21:35.52 | ac1djazz | fmueller: and to use exec playback |
| 21:40.24 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 21:42.03 | *** join/#asterisk mwalling (i=mwalling@2001:470:1f05:1ff:0:0:0:1) |
| 21:42.15 | fmueller | ac1djazz: there is also an application called MP3Player (core show application MP3Player). this probably removes the need to install format_mp3. |
| 21:42.36 | grandpapadot | Is the g726 protocol widely used by the asterisk community? |
| 21:43.10 | fmueller | ac1djazz: the MP3Player application just uses the commandline program mpg123 to playback the mp3 |
| 21:45.26 | [hC] | Can anyone think of a way to pair up CDR's from a customer site PBX's cdr records, with the same CDR from my asterisk PSTN gateway that their PSTN calls go through? I store a special accountcode client side that i'd like to reference from the PSTN gateway's CDRs and cant figure out how to pair those two CDR's up to match to the single call... |
| 21:49.17 | ac1djazz | tried rtunning smsq to send a text to my cell from my aterisk box i keep getting retrys [Apr 16 14:29:37] NOTICE[27185]: chan_local.c:597 local_alloc: No such extension/context 17094009@default creating local channel |
| 21:49.25 | ac1djazz | any way i can remove these from the queue? |
| 21:49.46 | ac1djazz | or whatever ud call it |
| 21:49.51 | ac1djazz | n/m guess it went away |
| 21:50.02 | ac1djazz | any1 here in the US successfully setup SMS via saterisk? |
| 21:53.10 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 21:53.14 | Katty | allo |
| 21:53.27 | lmadsen | WELL HI THERE |
| 21:53.31 | Katty | hai |
| 21:53.33 | lmadsen | howdy |
| 21:53.36 | Katty | i was having a thought about blind xfers. |
| 21:53.40 | lmadsen | don't do that |
| 21:53.41 | Katty | a confusing thought. |
| 21:53.44 | Katty | but |
| 21:53.45 | Katty | butbut |
| 21:53.48 | lmadsen | but! |
| 21:53.50 | lmadsen | butt? |
| 21:53.52 | Katty | buter. |
| 21:53.56 | lmadsen | bueler |
| 21:54.08 | Katty | when i do a blind xfer to a polycom, it shows the telco callerid. |
| 21:54.33 | Katty | when i do a blind xfer to a cell phone, it shows my company |
| 21:54.44 | Katty | there anyway to get around that so i can see who's /really/ calling my cell? |
| 21:55.00 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:55.12 | Katty | [TK]D-Fender: i bet you'd know |
| 21:56.04 | [TK]D-Fender | O>O |
| 21:56.18 | [TK]D-Fender | ->- |
| 21:56.19 | [TK]D-Fender | ->- |
| 21:56.21 | [TK]D-Fender | O>O |
| 21:56.27 | [TK]D-Fender | Mew? |
| 21:56.38 | Katty | [TK]D-Fender: i'm tired of my company doing blind xfers to my cellphone. and it looking like it's from my company... and then i have to deal with some stupid bullshit about exchange. |
| 21:57.02 | Katty | [TK]D-Fender: there a way to keep the callerid info when i do a blind xfer to external number? |
| 21:57.41 | [TK]D-Fender | Katty, It should keep the callerid. make sure your telco is letting you set it. |
| 21:57.55 | Strom_C | Katty: I'd guess that whatever set of dialplan code is being executed to call the outside number may be re-setting the caller ID |
| 21:58.07 | [TK]D-Fender | Katty, And there is no such thing as an "external number" when transferring a call. |
| 21:58.26 | [TK]D-Fender | Katty, Indeed that sounds like what I was helping you do earlier |
| 21:59.40 | Katty | [TK]D-Fender: |
| 21:59.47 | Katty | [TK]D-Fender: xfer blind to internal polycom keeps callerid |
| 22:00.03 | Katty | [TK]D-Fender: so i know asterisk is okay. i suppose. |
| 22:00.21 | Katty | [TK]D-Fender: i asked the telco if they reset the callerid, and they told me on a pri it's all my server. nothing else. |
| 22:00.27 | [TK]D-Fender | Katty, Go look what your exten you're forwarding to is doing. |
| 22:00.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:02.37 | Katty | very odd. when i call my cell from my desk phone, it shows my extension. |
| 22:02.46 | Katty | that callerid is working. |
| 22:03.25 | lesouvage | Fring released a SIP/IM/Skype client in beta stage but working for the iPhone (see http://www.fring.com/iphone/). |
| 22:05.19 | lesouvage | It would be nice if my iPhone register as an internal phone (like the Nokia E61) as soon as I'm home and I expect that will be possible in short time. |
| 22:06.17 | lesouvage | Katty: normally the phoneprovider blocks the cid set within Asterisk if it doens't fit into the hosted numberrange. |
| 22:06.21 | twisted | HAHAHAHAHAHAHHA |
| 22:06.24 | twisted | http://www.bash.org/?841435 |
| 22:06.25 | [TK]D-Fender | lesouvage, how will your sever know that you are "home"? |
| 22:06.29 | ManxPower | lesouvage: that is not correct |
| 22:06.29 | twisted | I *LOVE* it! |
| 22:06.47 | cpm | heh |
| 22:06.58 | cpm | that's pretty fun |
| 22:07.07 | *** join/#asterisk dagobart (n=dagobart@ip-62-143-215-156.1411O-CUD12K-01.ish.de) |
| 22:07.14 | dagobart | hi |
| 22:08.13 | lesouvage | [TK]D-Fender: As soon as I walk in my phone makes a connection with my access point and then register at my Asterisk server. |
| 22:08.25 | lesouvage | ManxPower: what is not correct? |
| 22:08.32 | Katty | lesouvage: hmmm, that's interesting |
| 22:08.37 | cpm | that's just wrong |
| 22:08.43 | ManxPower | lesouvage: Carriers allowing your own callerid. It TOTALLY depends on the carrier. |
| 22:09.16 | ManxPower | Some carriers only require 10 digit CID that could be a valid number, some just require 10 digits, some require it to be one of your DIDs |
| 22:09.18 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:10.08 | ManxPower | Katty: don't be surprised if some day calls stop working until you send 10 digits |
| 22:10.44 | [TK]D-Fender | lesouvage, Oh, so it does that over WiFi? |
| 22:11.07 | lesouvage | ManxPower: In Holland none of the big telco's allows you to set your own CID afaik. Some SIP providers do. But yes it totally depends on the carrier |
| 22:11.13 | Katty | ManxPower: i'm going to contact the telco. i can't figure out why the callerid is us instead of the blind xfered number. |
| 22:11.41 | dagobart | some of you helped me last Friday when my boss prohibited me from rtfm, there's some news. Interested to learn? [TK]D-Fender? |
| 22:12.12 | dagobart | boss came in to show me how easy it was to config asterisk by trial and error. Fail. As you folks predicted (and me too). |
| 22:12.22 | dagobart | Hope he learned a bit of it. |
| 22:12.29 | [hC] | Can anyone think of a way to match up CDR's between two * boxes, for two legs of the same call? (ie Client PBX: Internal Phone -> PSTN Gateway, Client SIP peer -> PSTN) I need to find some sort of unique identifier or something to match CDR's up between these two boxes. |
| 22:12.30 | lesouvage | [TK]D-fender: Yes, I think you can use the edge connection but that will not work properly I'm afraid. IM works great using edge |
| 22:13.24 | [TK]D-Fender | lesouvage, well you need something that proves location. if you can do it via WiFi, then you can check for LAN registration, etc. |
| 22:14.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:14.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:15.28 | lesouvage | [TK]D-Fender: The phone has a list of known access points and as soon the phone comes in range it will connect. If it is an uknown open access point it will ask. What do you mean by "proves location"? |
| 22:17.04 | [TK]D-Fender | lesouvage, can confirm that you were "home". I wasn't sure the iPhone had WiFi, but since you confirmed all of that, the rest is pretty easy |
| 22:17.24 | [TK]D-Fender | lesouvage, Was thinking you were referring to edge at first |
| 22:20.28 | lesouvage | [TK]D-Fender: If edge was capable of doing that you could rgister everywhere and aswer homecalls everywhere. That will not happen for the coming decade, it would be the end of the mobile telco's. |
| 22:24.02 | lesouvage | [TK]D-Fender: I tried calling over umts once with a Nokia E60 registered on my Asterisk box. It was actually working but there was a delay of 5 seconds so it was not usable. |
| 22:24.27 | pakipenguin | lesouvage, i use e60 sip over edge and it works fine |
| 22:26.48 | asdx | jjshoe: why the incoming call needed an 's' extention, the book didn't mention this |
| 22:27.00 | ManxPower | Cellular internet is too high latency and far too high jitter to be generally useful for VoIP |
| 22:27.19 | lesouvage | pakipenguin: You could make phonecalls using edge without delay, that is interesting. Maybe the telco's here sabotage/delay sip traffic to protect there moneymaker. |
| 22:27.25 | ManxPower | asdx: Asterisk will match the "s" extension when the called/dialed number is NOT KNOWN |
| 22:27.37 | ManxPower | If Asterisk knows what number was dialed it will NEVER match "s" |
| 22:27.54 | lmadsen | just got a cell phone with wifi in it for VoIP |
| 22:28.31 | [hC] | lmadsen: nokia? |
| 22:28.38 | pakipenguin | lesouvage, qos profile over you grps/edge apn |
| 22:29.02 | pakipenguin | lesouvage, i work for the mobile operator and we have a seperate profile for us |
| 22:30.00 | lmadsen | [hC]: of course :) |
| 22:30.17 | pakipenguin | lmadsen, which one :) |
| 22:30.46 | ManxPower | pakipenguin: Well if you are the CARRIER you can make it work 8- |
| 22:30.46 | Meaw | The nokia n95 comes with wifi |
| 22:30.47 | lesouvage | pakipenguin: so they do protect there per minute mobile phone service by simply not offering flat rate umts bundles. |
| 22:31.17 | *** join/#asterisk imcdona (n=t@216.141.118.212) |
| 22:31.19 | ManxPower | I'm on an EVDO Rev A connection at the moment |
| 22:31.31 | lesouvage | pakipenguin: fast enough to be used for sip connection. |
| 22:31.37 | CanWood | anyone have BLF indicating "Do not disturb" for GXP-2000 phones connected to an asterisk system? |
| 22:32.01 | imcdona | in ztool I am having an issue. I am seeing the bits flipping like mad |
| 22:32.03 | imcdona | any idea's |
| 22:32.07 | [TK]D-Fender | CanWood, plenty of people. |
| 22:32.26 | CanWood | great. then there's hope :) |
| 22:32.37 | ManxPower | imcdona: that could be normal or not normal. |
| 22:32.38 | pakipenguin | lesouvage, here in pk , we have a 8$/month and otherwise Rs.15/mb :) |
| 22:32.58 | imcdona | manxpower: how so? |
| 22:33.09 | pakipenguin | Meaw, i use E61 and E61i with wifi and will try fring on iphone tomorrow at work |
| 22:33.15 | CCFL_Man2 | ManxPower: wonder how voip works over that |
| 22:33.34 | lmadsen | pakipenguin: Nokia E61i |
| 22:33.49 | lmadsen | I like the E61i because the design is much better and doesn't feel so cheap |
| 22:33.49 | CanWood | if I have the key mode set to BLF, it works for on hook, ringing, available, all that, but DND it's still green. eventlist BLF and nothing works |
| 22:33.53 | pakipenguin | lmadsen, very very nice choice , i love the phone! |
| 22:34.00 | lmadsen | yes, it's very nice :) |
| 22:34.03 | CanWood | are there any common gotchas with this? |
| 22:34.05 | lmadsen | imported mine from Europe (I'm in Canada) |
| 22:34.11 | lmadsen | anyways, dinner time |
| 22:34.18 | Meaw | lmadsen, how much you got the E61i ? |
| 22:34.31 | Meaw | I just bought the 6500 S 2 days ago :/ |
| 22:34.34 | *** join/#asterisk RoyK (n=roy@91.149.31.152) |
| 22:34.36 | Meaw | it doesnt have wifi |
| 22:34.47 | pakipenguin | :) Meaw tough luck |
| 22:34.51 | ManxPower | imcdona: if you have a Channelized T-1, bit flipping is normal. On PRI, chances are it's not. WHAT problem are you having? |
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| 22:35.00 | ac1djazz | when i specify escape_digits that the # you can hit to stop the command right ? what do i put for 1? just '1' ? |
| 22:35.01 | pakipenguin | e61i is ~400$ here |
| 22:35.23 | Meaw | got the 6500 S for like 450$ |
| 22:35.35 | ac1djazz | <- e90 rules you all |
| 22:35.52 | pakipenguin | ac1djazz, e90 is too expensive! |
| 22:35.59 | ac1djazz | noway |
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| 22:36.06 | ac1djazz | when i specify escape_digits that the # you can hit to stop the command right ? what do i put for 1? just '1' ? |
| 22:36.11 | ManxPower | pakipenguin: coolness comes at a price |
| 22:36.11 | ac1djazz | like i wanan RECORD FILE |
| 22:36.18 | ac1djazz | so RECORD FILE filename gsm 1 -1 |
| 22:36.23 | ac1djazz | thatll record until i hit 1 right? |
| 22:36.29 | ManxPower | ac1djazz: asking over and over will guarntee you don't get your question asked |
| 22:37.10 | pakipenguin | ManxPower, i've used every communicator , just not the e90 ;) although planning to buy one :p |
| 22:37.13 | ManxPower | ac1djazz: Of course, you did not say if you are trying to do this using the C API, Perl AGI, PHP, AGI, Raw AGI, or extensions.conf |
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| 22:38.10 | ac1djazz | PHP |
| 22:38.12 | ac1djazz | and its working n/m :) |
| 22:38.13 | ac1djazz | agi |
| 22:38.37 | ManxPower | ac1djazz: I assumed C API |
| 22:38.40 | CanWood | [TK]D-Fender: is there an application that needs to be installed for the eventlist BLF to work?. Anything I can look for in "core show applications"? |
| 22:46.56 | ac1djazz | ManxPower: you have experiencing getting SMS working w/ asterisk <-> voip ? |
| 22:47.33 | ManxPower | ac1djazz: I'm in the USA, we don't use SMS |
| 22:47.46 | ac1djazz | ManxPower: as i cant? |
| 22:47.51 | ac1djazz | as in cant* ? |
| 22:47.52 | ManxPower | But since SMS is a MODEM protocol, don't expect it to work over VoIP |
| 22:48.20 | ManxPower | ac1djazz: As in the cell carriers do not provide a publically accessable SMSC |
| 22:49.05 | lesouvage | ac1djazz: I use fastsms of bayham system and it is working great and it is very costeffective to use, especially with international sms traffic. |
| 22:49.33 | ac1djazz | lesouvage: really cool, and you can text US as well etc? |
| 22:50.09 | ac1djazz | lesouvage: howd you get started? |
| 22:50.35 | ManxPower | ac1djazz: As lesouvage pointed out there are MANY Web<->SMS gateways |
| 22:50.51 | ManxPower | but that's not using the app_sms that comes with Asterisk |
| 22:51.45 | lesouvage | ac1djazz: It is a perl script that I run with a system call and the proper parameters. You can call the script from everywhere but I use it with Asterisk. |
| 22:52.20 | ac1djazz | lesouvage: is bay the cheapest? |
| 22:54.26 | lesouvage | ax1djazz: I haven't completely search the market but I think the are cheap. depending on the destionation it is between EUR 0,09 and EUR 0,15 per message if you add EUR 50,- at a time. You can also by 80.000 sms messages and that is much cheaper ;-) |
| 22:56.31 | lesouvage | ac1djazz: check http://www.bayhamsystems.com/asterisk-pricing.html Most destinations are 3 credits. |
| 22:57.12 | ac1djazz | thats expensive |
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| 23:01.31 | lesouvage | ac1djazz: You have to be carefull in your dialplan when using an sms script. I once kept on receiving sms messages because the dialplan looped and I could not stop it sending out an sms every 20 seconds |
| 23:02.47 | ac1djazz | brutal |
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