| 00:02.10 | peanut- | no |
| 00:02.38 | MrTelephone | are u using cisco ata186? |
| 00:02.52 | Laureano | I answer my self; yes, it support. |
| 00:03.07 | *** join/#asterisk KingDavid_ (i=CHRIS@ool-43555029.dyn.optonline.net) |
| 00:03.12 | MrTelephone | it won't udpate the analog handsets to the right time |
| 00:04.39 | KingDavid_ | hello guys, can anybody please recommend me a public domain library for a sip client? |
| 00:05.11 | KingDavid_ | I am trying to write a web embeded sip client |
| 00:05.49 | MrTelephone | good luck |
| 00:06.12 | MrTelephone | search for sip developer kit |
| 00:08.35 | KingDavid_ | not bad... I couldn't find much under 'sip clients' |
| 00:08.43 | KingDavid_ | thanks MrTelephone |
| 00:11.54 | peanut- | ludicrous speed! |
| 00:17.19 | MrTelephone | yeah i remember seeing something what you were looking for |
| 00:17.25 | MrTelephone | not sure what language it was in though |
| 00:17.43 | MrTelephone | i had to put the timezone in the cisco ata to 21 for eastern and its supposed to be 20 |
| 00:17.49 | MrTelephone | i guess it doesn't do dst on its own |
| 00:19.21 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:23.45 | sigmounte | salut mrtelephone |
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| 00:36.21 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.144.94) |
| 00:36.58 | MrTelephone | hi sigmounte |
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| 00:55.57 | Mw3 | hi. hm i just watching the logs and there are some of these lines: Nov 5 01:50:19 localhost asterisk[15442]: rc_avpair_new: unknown attribute 1490026597. what it can be? i just found 1 hit on google, but no explanation |
| 01:05.37 | Corydon76-dig | Mw3: it's from cdr_radius |
| 01:06.19 | Corydon76-dig | Most people aren't using Radius for CDR logging |
| 01:07.04 | phix | Nov 4 17:31:06 WARNING[23931]: dsp.c:1424 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 01:07.28 | phix | this is setting dtmfmode to auto, if I set it to anything else the dial pad does not work |
| 01:07.52 | phix | but I still get these warnings flood my asterisk console, any way to disable that warning message? |
| 01:07.55 | Corydon76-dig | phix: Then set your codec to ulaw |
| 01:08.17 | Corydon76-dig | inband will NEVER work with g729 |
| 01:08.29 | phix | Corydon76-dig: no, ulaw runs bad |
| 01:08.45 | Corydon76-dig | phix: then buy more bandwidth from your carrier |
| 01:08.51 | phix | Corydon76-dig: yes but the dialpad works then set to auto, and I get these inband message flooding |
| 01:08.56 | Corydon76-dig | phix: or use a different phone |
| 01:09.14 | phix | Corydon76-dig: It is the area, can only get ADSL and the phone lines are bad |
| 01:09.25 | Corydon76-dig | phix: Asterisk is absolutely correct on the message |
| 01:09.39 | Corydon76-dig | phix: then use a different phone |
| 01:09.42 | phix | Corydon76-dig: ok but it is stupid since the keys do work |
| 01:09.52 | Corydon76-dig | phix: I hear that Polycom phones are excellent |
| 01:10.08 | phix | It has nothing to do with phones |
| 01:10.37 | phix | it has to do with disabling messages |
| 01:10.40 | MrTelephone | does anyone here use ata186's and linux ntp protocol? |
| 01:10.43 | Corydon76-dig | It has everything to do with the phone, if it won't send rfc2833 messages |
| 01:11.07 | phix | The phone is a standard analog phone connected to a TDM2400 |
| 01:11.24 | ManxPower | phix: It is the OTHER SIDE of the connection. |
| 01:11.33 | Corydon76-dig | Oh, then it's your provider that is at fault |
| 01:11.44 | Corydon76-dig | Get a different provider |
| 01:11.50 | robl^ | MrTelephone: prolly need to update the ata186 firmware -- DST start / end dates changed in many countries recently, and you will need updated firmware |
| 01:12.06 | phix | ManxPower: Yes, they use a payload type of 96, which I have patched asterisk and dtmfmode=auto now works, but I am getting flooded with inband messages |
| 01:12.14 | ManxPower | The only way you are going to disable that message is to edit the source code and remove it. The reason it's so hard to do is because it is a very bad idea, don't expect DTMF to work well. Maybe 60% to 80% success rate. |
| 01:12.24 | phix | Corydon76-dig: Already been through 2 providers |
| 01:12.36 | ManxPower | phix: then don't use audo. use dtmfmode=rfc2833 or dtmfmode=info |
| 01:13.17 | phix | dmtfmode=rfc2833 does not work, I have not tried info but I doubt that they support that either |
| 01:13.17 | ManxPower | inband dtmf does not work with compressed codecs because those codecs distort continuous tones like DTMF. |
| 01:13.31 | ManxPower | phix: did it ever occur to you that maybe if you figured out why rfc2833 does not work, you will have solved your problem. |
| 01:13.45 | robl^ | dtmf inband only works with ulaw/alaw\ |
| 01:13.52 | phix | ManxPower: yes well the thing is the dialpad DOES WORK but I am still getting inband warnings, so obviously it is not using inband because that won't work over compressed codec but I am still getting warnings about it |
| 01:14.00 | ManxPower | robl^: don't bother. he doesn't want to listen to the truth. |
| 01:14.05 | ManxPower | phix: define "work" |
| 01:14.32 | phix | ManxPower: ...... BECAUSE THE VOIP PROVIDER USES A PAYLOAD TYPE OF 96!!!! I have already said this |
| 01:14.35 | ManxPower | you called into your bank's IVR and looked up your blalance? You called into an airline's IVR. |
| 01:14.44 | ManxPower | phix: payload 96 is not inband |
| 01:14.53 | phix | ManxPower: I KNWO! |
| 01:15.00 | phix | but asterisk is reporting it as inband! |
| 01:15.09 | phix | that is my problem, I am getting warnings flooded through console! |
| 01:15.13 | ManxPower | phix: it does that when you set auto. |
| 01:15.14 | phix | I want to diable them |
| 01:15.48 | ManxPower | if you set audo you are saying "guess at the dtmf mode" |
| 01:15.48 | Corydon76-dig | You can't disable them |
| 01:15.48 | ManxPower | phix: then go into the source code and disable it. |
| 01:15.48 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 01:15.48 | phix | ManxPower: :\ sure, which line? :) |
| 01:15.48 | phix | and file |
| 01:16.01 | phix | rtp.c? |
| 01:16.14 | Corydon76-dig | If you're going to be messing with your source, you should know what you're doing |
| 01:16.38 | phix | I know very little about how asterisk is coded, but I do know how to code in C |
| 01:16.58 | ManxPower | main/dsp.c: ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(af->subclass)); |
| 01:17.03 | Corydon76-dig | I don't want to be on the receiving end of "but Corydon told me to comment out that line..." |
| 01:17.07 | ManxPower | now that I've grep'd it for you... |
| 01:17.21 | phix | ManxPower: haha I should of done that :P |
| 01:17.25 | Qwell | can an RJ11 cable be trimmed down to RJ9? heh |
| 01:17.27 | ManxPower | yes, you should havve. |
| 01:17.42 | phix | ManxPower: but I don't really want to recompile again :/ |
| 01:17.48 | Corydon76-dig | Qwell: What's the pinout for RJ9? |
| 01:18.00 | phix | RJ9. wow that would be tiny |
| 01:18.18 | Qwell | RJ10? |
| 01:18.26 | Qwell | 4 wire, 4 connector |
| 01:18.33 | Qwell | handset cable |
| 01:18.39 | ManxPower | Corydon76-dig: when he has forgotten he did that and has problems with DTMF (that he does not currently realize he has), revenge will be sweet. |
| 01:19.13 | Corydon76-dig | Qwell: you'll bloody your fingertips trying to trim it down |
| 01:19.31 | Qwell | stupid radio shack |
| 01:19.48 | Corydon76-dig | It's very hard plastic |
| 01:19.53 | Qwell | yeah... |
| 01:20.23 | Corydon76-dig | You need an extra handset cord? |
| 01:20.32 | Qwell | to butcher |
| 01:20.32 | ManxPower | I'll bet he things "dialpad works" because he can call PSTN numbers via his provider, and we all know that the number is sent to his provider, not as DTMF, but as part of the SIP call setup. |
| 01:20.48 | Corydon76-dig | You can get one at Wally Mart |
| 01:21.00 | Qwell | yeah, probably cheaper than radio shack too |
| 01:21.04 | Corydon76-dig | Heck, I think Kroger has them, too |
| 01:21.06 | Qwell | they wanted like $9.99 |
| 01:21.15 | Qwell | I only need one like...6" long |
| 01:21.25 | Corydon76-dig | The actual plugs without wires are nearly impossible to find |
| 01:21.28 | Qwell | and I don't want curly |
| 01:21.34 | Qwell | heh, if we had Fry's here, this would be trivial |
| 01:21.40 | Corydon76-dig | I had to find them at Graybar |
| 01:21.41 | ManxPower | cheaper than getting your finger re-attached. |
| 01:21.42 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 01:21.50 | Qwell | is graybar a local shop? |
| 01:21.59 | Corydon76-dig | It's a national chain |
| 01:22.07 | Qwell | never been to one |
| 01:22.22 | Corydon76-dig | There's one in Nashville. Too late to get you a box for tomorrow, but maybe for next Monday |
| 01:22.35 | Corydon76-dig | I think they come in bags of 25 |
| 01:22.56 | Corydon76-dig | www.graybar.com |
| 01:23.02 | Qwell | I'm probably just going to order online |
| 01:23.54 | Corydon76-dig | Hot dog, there's one in HSV |
| 01:24.00 | Qwell | neat |
| 01:24.02 | Corydon76-dig | 2811 University Dr |
| 01:24.06 | ManxPower | Cool! |
| 01:24.20 | ManxPower | now I know where to buy cable (if I ever need cable again) |
| 01:24.35 | Qwell | hmm |
| 01:24.39 | Strom_M | i cant believe you do not know the ultimate awesomeness that is Graybar |
| 01:25.16 | ManxPower | Strom_M: I know it is virtually impossible to get the 66-block I wanted based on the info on their web site. |
| 01:25.27 | Strom_M | which 66 block is that? |
| 01:25.33 | Corydon76-dig | Qwell: it's a couple blocks east of Jordan Ln |
| 01:25.39 | ManxPower | Strom_M: It was at least a year ago. |
| 01:25.53 | ManxPower | IIRC, it was one with the Amp connector already wired into it. |
| 01:26.19 | WilliamK | ManxPower, could also try www.altex.com |
| 01:26.23 | Corydon76-dig | ManxPower: Heh, you can get two Amphenol connectors, if you like |
| 01:26.24 | WilliamK | they do alot of qty 1 items |
| 01:26.37 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 01:26.50 | Strom_M | ManxPower: i see those at graybar all the time |
| 01:26.51 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:26.51 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:27.44 | WilliamK | Strom_M, yep about every major telco supplier has em - anixter, graybar, grainger, etc.. |
| 01:29.32 | WilliamK | One of the things I like the most in regards to surge protectors is utilizing the Tripplite rackmount ones around my house |
| 01:30.12 | *** join/#asterisk macr0 (n=macr0@adsl-074-229-244-187.sip.bhm.bellsouth.net) |
| 01:30.18 | WilliamK | mention plugging a FiOS connection into a surge protector and the techs freak out.... then you tell them to hold on a min and you'll be back, and then show off the nice rackmount one |
| 01:30.24 | WilliamK | like a whole new world to them |
| 01:33.09 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:33.09 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:38.56 | *** join/#asterisk mcreedjr (n=mikejr@h241.126.16.98.ip.alltel.net) |
| 01:40.20 | mcreedjr | quick question... i've got a TDM2422 that won't pick up a ringing phone line in the US. the CLI shows a ring event, and shows the Answer app running, but the phone on the other end continues to ring. Is this a signalling problem? |
| 01:41.13 | mcreedjr | oh and I can place phone calls out this phone line from Asterisk, but it won |
| 01:41.18 | mcreedjr | won't answer. |
| 01:42.18 | *** part/#asterisk agentzh (i=agentz@nat/yahoo/x-a1763b79de2d9f16) |
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| 02:07.46 | Sci_05 | anyone got any ideas was to why when I upgrade from 1.4.5 asterisk to 1.4.13 I no longer have zaptel equipment showing up? I have the latest zaptel installed |
| 02:08.58 | Sci_05 | everything stays the same, if I go back to 1.4.5 it shows up with no problem, but I install 1.4.13 and I get notta |
| 02:10.12 | Strom_M | are you unloading the old drivers from memory and loading the new drivers into memory? |
| 02:12.22 | Sci_05 | I am running zaptel 1.4.6 and ahve rebooted the server sence installing it. I stop asterisk do the "make install" start asterisk (asterisk -p) and it will load up showing the correct version I just get no zaptel stuff If I go beyond 1.4.5 |
| 02:12.43 | *** join/#asterisk techie (n=techie@adsl-76-214-11-114.dsl.lsan03.sbcglobal.net) |
| 02:13.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:14.46 | Sci_05 | its the damndest thing, I just don't get why the zaptel would stop working in 1.4.13 but no in 1.4.5 |
| 02:21.28 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584436.dsl.bell.ca) |
| 02:29.08 | *** join/#asterisk salviadud (n=noyb@189.156.177.212) |
| 02:29.31 | salviadud | i'm getting this error, all the time |
| 02:29.34 | salviadud | Username/auth name mismatch |
| 02:29.41 | salviadud | and as far as i know |
| 02:29.46 | salviadud | my username and password |
| 02:29.47 | salviadud | are correct |
| 02:30.00 | salviadud | i'm so pissed right now... |
| 02:33.29 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-38-156.lns10.syd7.internode.on.net) |
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| 02:34.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:39.29 | MrTelephone | salviadud, what does your sip.conf look like |
| 02:39.41 | MrTelephone | [username] |
| 02:39.45 | MrTelephone | secret=abc123 |
| 02:39.54 | MrTelephone | password=assword? |
| 02:41.00 | Mw3 | i get a lot of messages like this in the iaxmodem log: [2007-11-05 03:31:10] IAX2 jitter - last_ts: 42024, ts: 42032. its running on localhost. what the hell can cause jitter on localhost? |
| 02:42.16 | MrTelephone | mw3, are you actually gettin jitter? |
| 02:42.29 | MrTelephone | it looks like an informational line |
| 02:42.31 | JT | Mw3: interrupt sharing |
| 02:42.54 | JT | but maybe there isn't really jitter |
| 02:44.58 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 02:49.52 | Mw3 | well, the iaxmodem line is linked to hylafax, and faxing is not working. in the asterisk console i get this: [Nov 5 03:31:06] WARNING[9794] chan_iax2.c: Max retries exceeded to host 127.0.0.1 on IAX2/iaxmodem0-2 (type = 6, subclass = 11, ts=10021, seqno=9) |
| 02:50.15 | Mw3 | i have no zaptel hardware, i use ztdummy as timing source |
| 02:52.32 | MrTelephone | max retries means its sending a message to your modem thing and its not responding |
| 02:53.18 | Mw3 | MrTelephone: in the source the message looks like this: printlog(LOG_ERROR, "IAX2 jitter - last_ts: %d, ts: %d\n", last_ts, iaxevent->ts); because of LOG_ERROR i thought it isnt just informational |
| 02:53.18 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:53.42 | MrTelephone | what is ts? |
| 02:53.55 | MrTelephone | time sequence? |
| 02:54.01 | Mw3 | i think so |
| 02:54.38 | Mw3 | it generated 55088 lines of this message just because of some test calls in the past few hours |
| 02:55.03 | MrTelephone | its dropping frames or something |
| 02:55.16 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 02:55.40 | MrTelephone | check your irq misses? |
| 02:56.20 | Mw3 | in zap show status there is no irq miss |
| 02:56.26 | Mw3 | but it is unrelated as i think |
| 02:56.52 | MrTelephone | what is the test condition for the pringlog() function? |
| 02:56.54 | Mw3 | you only need timing for iax trunking |
| 02:56.57 | salviadud | mr telephone |
| 02:57.01 | salviadud | my sip.conf |
| 02:57.07 | salviadud | could i pastebin to ya? |
| 02:57.17 | MrTelephone | i can try and help salviadud |
| 02:57.38 | Mw3 | MrTelephone: if (!nojitterbuffer && last_ts && iaxevent->ts <= last_ts) { |
| 02:58.42 | salviadud | http://pastebin.com/d220a2dc1 |
| 02:58.45 | MrTelephone | im still stumped on what ts is |
| 02:58.46 | salviadud | please, check it out |
| 02:58.56 | salviadud | i'm trying to register |
| 02:58.59 | salviadud | a nokia e-70 |
| 02:59.07 | salviadud | i got my username and pass correct |
| 02:59.11 | salviadud | maybe, i |
| 02:59.16 | salviadud | i'm missing a parameter |
| 02:59.35 | *** join/#asterisk BeeBuu (n=chatzill@61.145.77.5) |
| 02:59.40 | BeeBuu | hello,all |
| 02:59.44 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 03:01.26 | MrTelephone | hey salviadud, your going to have to sip debug <peer> and try and register |
| 03:02.00 | salviadud | damn... |
| 03:02.03 | salviadud | this is gonna take a while |
| 03:02.04 | salviadud | brb |
| 03:02.35 | Mw3 | MrTelephone: ts is timestamp in the libiax sources |
| 03:05.56 | MrTelephone | salviadud, what network is your phone on? behind a nat? put nat=yes |
| 03:06.00 | MrTelephone | mw3, do you need hardware for iaxmodem? |
| 03:07.36 | Mw3 | MrTelephone: what? i dont understand your quesstion |
| 03:07.57 | MrTelephone | I have to read about that iaxmodem.. I don't know anything about it |
| 03:08.05 | MrTelephone | is it all software? |
| 03:08.26 | ManxPower | it is all software |
| 03:08.30 | Mw3 | MrTelephone: its a software which creates a /dev/ttyIAX device |
| 03:08.56 | Mw3 | MrTelephone: and you can use that device from hylafax. its a modem emulator or something |
| 03:09.01 | BeeBuu | i want record at zap/3 when answer,how to do that? |
| 03:09.51 | MrTelephone | i was using spandsp for faxes and it worked 100% |
| 03:10.07 | ManxPower | MrTelephone: then keep using spandsp. It's what I use. |
| 03:10.14 | BeeBuu | MrTelephone: can you help me?please? |
| 03:10.41 | MrTelephone | mw3, that if statement is testing to see if the current timestamp is less then the last timestamp i think.. if its the localhost then there is a buffer issue or irq issue |
| 03:11.03 | MrTelephone | manxpower, i was just telling mw3 maybe he should try it |
| 03:11.55 | Mw3 | iaxmodem is actually using spandsp. by spandsp you mean app_rxfax and app_txfax? |
| 03:12.20 | MrTelephone | bebuu, not sure.. google dialplan commands asterisk and look for record |
| 03:12.51 | BeeBuu | i don't know how to detect the line is answer |
| 03:13.52 | MrTelephone | you want to answer the ringing line and record what is said? |
| 03:14.12 | BeeBuu | yes,no "du,du..." |
| 03:14.16 | MrTelephone | mw3, thats what I'm saying |
| 03:14.36 | BeeBuu | record when someone answer |
| 03:14.44 | MrTelephone | mw3, I find it very odd that the iax transmissions are arriving in different order on local host?? |
| 03:14.58 | MrTelephone | beebuu, did you get it to answer yet? |
| 03:15.57 | BeeBuu | zap/1 dial to zap/2 |
| 03:16.06 | MrTelephone | you got that working? |
| 03:17.31 | BeeBuu | but zap/2 not answer,still record ... |
| 03:17.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:17.41 | BeeBuu | i don't want this |
| 03:18.00 | Mw3 | MrTelephone: i just figured out that my kernel is complied with HZ = 100. i found some docs which recommends that you should use HZ = 1000 for better timing when you are using asterisk. i will recompile my kernel with HZ=1000 tomorrow. but now its sleeping time. its 4:17 am here and at 8 a.m. i got to work :) thank you |
| 03:18.21 | MrTelephone | ok take it easy mw3 |
| 03:18.23 | MrTelephone | what is HZ? |
| 03:19.18 | Mw3 | some kernel thing related to timing. i dont know exactly |
| 03:19.39 | MrTelephone | if it fixes it what a piss off eh |
| 03:19.52 | MrTelephone | all that hard ache for a decimal place :( |
| 03:19.59 | Mw3 | i will let you know if i found you around tomorrow |
| 03:20.07 | MrTelephone | alright cya |
| 03:21.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:23.19 | Mw3 | by the way i've already sucked with this HZ thing today. the BRI poll defaults to 128 but it should be multiple of 8000 / HZ. in my case 8000 / HZ = 80, so i set it to 80 and it didnt work. after 2 hours i set it to 160 and it started to work :). with HZ=1000 8000/1000 would be 8 and the default 128 is multiple of 8 and it would work in the first place ... |
| 03:23.47 | Mw3 | but the help in the kernel said that 100 for servers, 250 and 300 for general usage, and 1000 for desktops |
| 03:23.56 | alpha232 | Ahhh back at work |
| 03:24.29 | MrTelephone | odd |
| 03:25.30 | JT | Mw3: asterisk is not a normal server |
| 03:26.01 | alpha232 | BRI? bri how dare he have a BRI that is starting to work |
| 03:28.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:28.16 | fujin | Hey, anyone know what type of headset the IP330's take? |
| 03:28.55 | fujin | standard 2.5m?? |
| 03:29.26 | *** join/#asterisk TSCHAK (n=thomas@c-65-96-160-105.hsd1.ma.comcast.net) |
| 03:29.41 | TSCHAK | hello, is there anyone in here with experience with Cisco 7970 phones? |
| 03:30.14 | BeeBuu | anyone tell me what's this for? "Record(asterisk-recording%d:ulaw) " |
| 03:30.44 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:30.56 | BeeBuu | %d:ulaw? |
| 03:31.34 | flenders | BeeBuu: *CLI> show application record |
| 03:31.53 | flenders | thats the format you'll be recording your files |
| 03:32.18 | citats | the %d will be replaced with a number that increments, so you dont have duplicates |
| 03:32.30 | BeeBuu | o,i got it |
| 03:32.47 | citats | but flenders has the right idea. 'show application record' should tell you all of that |
| 03:33.46 | MrTelephone | im getting a lot of these stale nonce sip messages now |
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| 03:47.46 | MrTelephone | all those us army commercials make me want to join the army |
| 03:47.49 | MrTelephone | not |
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| 03:54.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:54.57 | alpha232 | why join the army when you can sleep with a soldier |
| 03:55.53 | MrTelephone | thats what happens in the motel? |
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| 06:00.31 | alpha232 | pfft |
| 06:03.08 | Snake-eyes | How can make asterisk keep going through a macro if the caller hangs up (like deadAGI) ? |
| 06:03.26 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 06:05.15 | alpha232 | why? |
| 06:05.27 | alpha232 | if there is no one on the other side... whats the use |
| 06:06.42 | ai-a[zZ] | Snake-eyes: you mean the called party,, or the calling party ? |
| 06:06.56 | litage|w | alpha232: why ask why Snake-eyes wants to do this? if he wants to, he wants to =) |
| 06:07.36 | ai-a | alpha232: there is reasons to continue, ie, do some db clean ups. |
| 06:08.07 | Snake-eyes | ai-a, the caller/calling party phones into asterisk. In this case to do a recording |
| 06:08.24 | CunningPike | Doesn't the 'g' option to Dial() do that? |
| 06:08.43 | alpha232 | Snake-eyes: ok the caller is making a recording.. |
| 06:09.40 | alpha232 | then what |
| 06:10.01 | Snake-eyes | CunningPike, thanks |
| 06:10.54 | CunningPike | Snake-eyes: ytw. However, giyf |
| 06:11.40 | alpha232 | CunningPike: sometimes in coding people ask simple questions but truely it's a more complex issue with a better solution |
| 06:12.12 | litage|w | CunningPike: what does "ytw" mean? |
| 06:12.19 | CunningPike | ~ytw |
| 06:12.44 | CunningPike | Hmm - [TK]D-Fender must have cleaned that one out - You're Totally Welcome |
| 06:12.46 | litage|w | ytw == yellow turgid waterbuffalo |
| 06:13.04 | litage|w | i like mine better :) |
| 06:13.17 | flenders | You Tell Wayne! |
| 06:13.20 | alpha232 | the idea of anything on a waterbuffalo being turgid scares me |
| 06:21.25 | [TK]D-Fender | CunningPike, nope :) |
| 06:21.35 | CunningPike | OK, then |
| 06:21.37 | [TK]D-Fender | CunningPike, not my doing... |
| 06:21.57 | CunningPike | jbot: ytw is You're totally welcome! |
| 06:21.58 | jbot | okay, CunningPike |
| 06:22.15 | CunningPike | ~giyf |
| 06:22.16 | jbot | [giyf] Google Is Your Friend, or see also: STFW |
| 06:22.25 | CunningPike | Cool |
| 06:22.29 | flenders | ~stfw |
| 06:22.30 | jbot | from memory, stfw is Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
| 06:22.38 | CunningPike | How's it going, [TK]D-Fender? |
| 06:22.50 | CunningPike | And, do you ever sleep? |
| 06:22.53 | CunningPike | :) |
| 06:23.32 | [TK]D-Fender | CunningPike, getting by. Just in from late coffee. Spent a lot of money this weekend. new shoes and fall jacket, and that new custom blade (that last one is kinda an early Christmas gift to myself. |
| 06:23.44 | [TK]D-Fender | CunningPike, And sure.... plenty of time when I'm dead :) |
| 06:23.48 | CunningPike | ;) |
| 06:25.16 | [TK]D-Fender | CunningPike, I thing you're looking for THIS : |
| 06:25.19 | [TK]D-Fender | ~jfgi |
| 06:25.20 | jbot | http://www.google.com/search?q=jfgi |
| 06:25.24 | [TK]D-Fender | ?! |
| 06:25.35 | [TK]D-Fender | Oh.. same net result! :P |
| 06:25.38 | CunningPike | That'll work |
| 06:25.43 | CunningPike | ~giym |
| 06:26.01 | [TK]D-Fender | CunningPike, It used to "just say it", but the redirect surprise is still worth it :) |
| 06:26.49 | CunningPike | Heh - I see that http://www.googleityoumoron.com/ is no more |
| 06:26.53 | CunningPike | Pity |
| 06:28.49 | [TK]D-Fender | and this month Bell is coming out with the HTC touch (probably DUO) and others.... another final gift to myself. |
| 06:29.06 | Maliuta | [TK]D-Fender: new toy? |
| 06:29.29 | [TK]D-Fender | Maliuta, Custom katana built to spec |
| 06:30.01 | CunningPike | And he doesn't mean a bike |
| 06:30.04 | CunningPike | :) |
| 06:30.10 | CunningPike | Which I would much prefer |
| 06:30.14 | Maliuta | well duh! :P |
| 06:30.58 | Maliuta | 4 weeks until I can move back to Brisneyland! sooner if work finishes me before the end of the notice period |
| 06:31.07 | [TK]D-Fender | ? |
| 06:31.25 | [TK]D-Fender | Maliuta, "Brisneyland"? |
| 06:31.26 | alpha232 | congrats! |
| 06:32.05 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 06:32.05 | Maliuta | [TK]D-Fender: AKA, Brisvegas or Brisbane |
| 06:32.44 | alpha232 | brisvegas? brisneyland? sounds like a jewish boys nightmare |
| 06:32.48 | alpha232 | snip snip |
| 06:33.20 | flenders | Maliuta: not sure why you would call Brisbane that |
| 06:33.33 | flenders | unless you spend a lot of time in the casino there |
| 06:33.46 | Mw3 | alpha232: why were suprised about i have a working BRI? :) |
| 06:34.19 | Maliuta | Brisvegas is a term that popped up when the Treasury opened. I prefer Brisneyland, everyones always smiling and happy :) |
| 06:34.44 | flenders | you were obviously born there |
| 06:35.30 | Maliuta | I refer to the suburb of Springwood (where my parents place is) as Spingvegas, because as you come off the freeway there is all this gaudy lighting like the vegas strip |
| 06:35.34 | alpha232 | Mw3: yeah, here in the US getting a working BRI on * has so far been unheardof |
| 06:35.51 | Maliuta | flenders: yeah, and lived just about everywhere else in .au |
| 06:37.55 | Mw3 | alpha232: oh, i see. im in europe :) |
| 06:38.06 | [TK]D-Fender | CunningPike, mine is based on this one : http://www.roninswords.com/custom%20horse%20and%20plum.htm |
| 06:38.16 | alpha232 | Mw3: lucky |
| 06:39.38 | Mw3 | i wish i would be in the US. i wouldn't care about not working BRIs :) |
| 06:39.53 | alpha232 | Mw3: grass is greener |
| 06:41.52 | alpha232 | [TK]D-Fender: rolf |
| 06:42.07 | alpha232 | i'm so disappointed |
| 06:42.15 | alpha232 | i want my bri to WORK! |
| 06:42.35 | alpha232 | nothing like a system that can answer after 0 rings and get caller id :D |
| 06:42.52 | Mw3 | but its a little bit funny that the digium bri card (which is made by an US based company) is not working in north america |
| 06:43.20 | CunningPike | Hey, dlynes_laptop |
| 06:43.22 | alpha232 | Mw3: the market for ISDN is EuroAsia |
| 06:43.41 | dlynes_laptop | CunningPike: hey |
| 06:43.42 | alpha232 | Mw3: plus they rape you on the price |
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| 06:43.56 | CunningPike | dlynes_laptop: More at http://www.members.optusnet.com.au/fischermk/otherkats.htm |
| 06:44.01 | CunningPike | :) |
| 06:44.27 | dlynes_laptop | CunningPike: well..not saying i wanted one, but if i did want a katana, i'd much rather have the motorbike variety, than the sword |
| 06:44.43 | CunningPike | Indeed |
| 06:45.03 | dlynes_laptop | at least they go vroom vroom, and they attract chicks |
| 06:45.11 | [TK]D-Fender | And strangely enough, your choice is far more likely to get you killed :p |
| 06:45.12 | dlynes_laptop | the katana kinda turns the chicks off |
| 06:45.16 | alpha232 | Mw3: same reason why a software voice modem isn't used as a voice board even though all they need is the software |
| 06:45.22 | dlynes_laptop | erm the sword kind i mean |
| 06:45.47 | CunningPike | Well, I know which one I'd rather have between my legs........ |
| 06:45.51 | CunningPike | :O |
| 06:46.05 | [TK]D-Fender | All this talk about cats.... hmmm |
| 06:46.07 | [TK]D-Fender | :O |
| 06:46.19 | [TK]D-Fender | ok, bed time! |
| 06:46.23 | [TK]D-Fender | later all! |
| 06:46.24 | [TK]D-Fender | <PROTECTED> |
| 06:47.25 | dlynes_laptop | anyways...g'night, CunningPike |
| 06:47.30 | dlynes_laptop | beddy bye time |
| 06:47.31 | CunningPike | Night |
| 07:08.19 | alpha232 | it would be nice to setup a streaming on hold music source that buffers say, 15 seconds, when it reaches 5 seconds it fades out and plays local static music until it buffers X amount of audio |
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| 07:11.19 | Mw3 | it would be nice if early audio would work with pri in my country :) |
| 07:14.40 | Strom_M | Mw3: you've got to send a proceeding message first :) |
| 07:15.05 | Mw3 | yes, with Progress |
| 07:15.08 | Mw3 | i did |
| 07:15.20 | Mw3 | ztmonitor shows that there is tx |
| 07:15.32 | Mw3 | but i didnt hear anything on the other end |
| 07:15.46 | Mw3 | i think its a pay service here or i dunno |
| 07:16.03 | Mw3 | but if i call the support they wont even know what i am talking about |
| 07:16.47 | Mw3 | i took them half a day to figure out that caller id presentation is not enabled by default on their pri |
| 07:17.00 | Mw3 | hopeless |
| 07:17.03 | alpha232 | lol |
| 07:17.50 | alpha232 | what is early audio? |
| 07:18.39 | alpha232 | you mean audio prior to supervised answer? |
| 07:19.32 | Mw3 | yes |
| 07:19.39 | Mw3 | so i can have custom ringtone:) |
| 07:20.07 | alpha232 | not just custom ring but also for system status messages |
| 07:21.13 | alpha232 | though i knew one company who had their IVR prior to supervision and got in naughty naughty for it |
| 07:22.43 | Mw3 | all i get is big silence |
| 07:22.52 | alpha232 | ouch |
| 07:22.55 | Mw3 | no ringing, no custom tone |
| 07:23.05 | Mw3 | ztmonitor shows tx |
| 07:23.11 | Mw3 | but its lost somewhere |
| 07:23.30 | Mw3 | and that somewhere got to be my provider :) |
| 07:24.18 | Mw3 | we are paying shitload of money per month for that pri. but there is no support. if its not working -> your fault, our equipment never fail |
| 07:25.53 | alpha232 | lol but it could be provisioning |
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| 09:27.32 | OzyWebMaster | Hi all, does anyone know what the option STATIC_BUILD is in the makefile? |
| 09:32.14 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
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| 09:52.29 | Uatec | ach |
| 09:52.32 | Uatec | why is sip show piss |
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| 09:59.58 | billybongo | hmm~ |
| 10:06.06 | tzafrir | ~hmm |
| 10:06.07 | jbot | hmm is, like, hidden markov model |
| 10:06.19 | tzafrir | oops |
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| 10:16.06 | linxroute | dtmf |
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| 10:22.47 | Zefk | Hi all, after upgrade from 1.4.11 to 1.4.13 the cdr does not work any more. The message is: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/voiceblue-084c8a10' not posted. Any advice ? thx |
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| 10:38.29 | blueneon | hi, im trying to setup call monitoring (recording), but asterisk seems to be splitting the conversation into two files, 1 for in and 1 for out, is there no option to setup to make asterisk merge this into 1 file rather? |
| 10:39.29 | IPetrov | blueneon: use MixMonitor |
| 10:41.02 | kaldemar | or option m on Monitor with sox installed. |
| 10:41.14 | blueneon | ye i just saw the optional m param |
| 10:41.16 | blueneon | doh |
| 10:41.17 | blueneon | hehe ta |
| 10:41.54 | blueneon | how can i get asterisk to use the basename as the current date/time |
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| 10:42.24 | blueneon | in the Monitor() function the option for basename, i'd like it to be the current date and time instead of a set file name |
| 10:42.28 | blueneon | if possible |
| 10:43.35 | kaldemar | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 10:43.48 | blueneon | cool, thanks |
| 10:49.00 | GigaWork | Hi, is there a way to merge attended and blind transfer? (asterisk 1.4.12.1) |
| 10:49.07 | GigaWork | so when person A transfers to person B, it should be an attended transfer, but if person A hangs up before person B picks up, it becomes a blind transfer |
| 10:49.13 | GigaWork | and is it possible to call back person A if person B doesn't pick up at all? |
| 10:49.19 | GigaWork | i tried this: http://bugs.digium.com/view.php?id=8413 |
| 10:49.23 | GigaWork | but that doesn't work |
| 10:49.31 | Poincare | With the 'sip debug' messages, on received messages, the date field is that the time the sip message was received or when it was sent? |
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| 10:56.18 | *** join/#asterisk case_ (n=case@pdpc/supporter/student/case-) |
| 10:56.23 | case_ | hello |
| 10:56.55 | phix | hi |
| 10:58.00 | case_ | i try to recieve fax with asterisk and to do so i want to name the incoming fax with the current date and time. i can't find a working example... something like FAX_`date "+%Y%m%d%H%M%S"`.tif |
| 10:58.28 | phix | hmmmmm |
| 10:58.43 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:58.48 | phix | does asterisk handle faxes by itself or does it pass it off to hylafax? |
| 10:59.00 | case_ | it handle faxes by itself |
| 10:59.05 | case_ | with rxfax() |
| 10:59.30 | case_ | anyway, my problem is just to make asterisk generate the filename |
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| 11:00.11 | phix | h nice |
| 11:00.20 | phix | ok |
| 11:00.25 | phix | good luck with that |
| 11:00.28 | Mw3 | case_: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 11:00.34 | case_ | thanks |
| 11:02.58 | blueneon | any idea where i could find a nice voice saying something like "all calls are recorded for quality control purposes" |
| 11:05.23 | *** join/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 11:07.34 | Mw3 | ask your girlfriend to say those words :) |
| 11:09.25 | case_ | blueneon, i haven't listened them, but there are a lot of spy-*.gsm in /usr/share/asterisk/sounds , none of them fit your needs? |
| 11:10.20 | IPetrov | GigaWork: it merged in trunk version now |
| 11:12.02 | blueneon | case_ i only see a very small amount of spy-* files in /var/lib/asterisk/sounds/ eg. /var/lib/asterisk/sounds/spy-zap.gsm, which just says "zap" heh |
| 11:12.23 | case_ | sorry then. |
| 11:12.35 | GigaWork | but asterisk doens't do this IPetrov |
| 11:13.16 | blueneon | is there not more of these i could find somewhere? |
| 11:13.26 | blueneon | perhaps my version doesnt incl. the whole lot |
| 11:13.39 | GigaWork | when person A transfers to person B, and person A hangs up before person B picks up, the call is lost |
| 11:19.32 | penguinFunk | blueneon: have you installed asterisk-sounds ? |
| 11:25.48 | IPetrov | GigaWork: it works in SVN TRUNK |
| 11:25.56 | IPetrov | GigaWork: 814 patch is applied there |
| 11:26.03 | IPetrov | GigaWork: 8413 i mean |
| 11:26.37 | IPetrov | http://svn.digium.com/svn/asterisk/trunk/ |
| 11:29.21 | GigaWork | ok thx, i'll check that out |
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| 11:31.01 | IPetrov | GigaWork: but u need to know - it unstable version, it 1.5 beta in fact :) |
| 11:31.43 | GigaWork | ah |
| 11:31.55 | GigaWork | is there another way to get this functionality? |
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| 11:39.15 | puzzled | hi |
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| 12:19.35 | ThoMe | hiho |
| 12:19.50 | ThoMe | is it posible a fritzcard of avm use with mISDN ? |
| 12:20.22 | *** join/#asterisk cy3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 12:20.27 | cy3o3 | re |
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| 12:21.44 | ThoMe | cy3o3: hi |
| 12:21.49 | ThoMe | masus: hi, kennst du dich mit asterisk aus? |
| 12:21.51 | ThoMe | is it posible a fritzcard of avm use with mISDN ? |
| 12:22.31 | masus | ThoMe: ask |
| 12:22.46 | ThoMe | masus: kann man ne fritzcard aus nutzen mit misdn? |
| 12:22.49 | ThoMe | aus = auch |
| 12:23.02 | masus | ne soweit bin ich noch nicht |
| 12:23.04 | masus | :) |
| 12:23.07 | ThoMe | :-) |
| 12:23.35 | ThoMe | masus: ich hab scho ne 4s0-port karte mit asterisk am laufen.. aber noch ned asterisk und ne isdnkarte |
| 12:23.39 | ThoMe | aaeh isdn karte von avm |
| 12:24.55 | masus | ThoMe: ich hab mir eben gerade ne te205p karte gekauft |
| 12:25.05 | masus | und versuch es gerade zu konfigurieren |
| 12:25.27 | masus | könntest du mir villeicht weiter helfen |
| 12:25.49 | waKKu | o.0 |
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| 12:27.29 | ThoMe | masus: hmm. was ist das fuer ne karte? |
| 12:29.05 | masus | ThoMe: Digium Wildcard TE205P (3rd Gen) |
| 12:29.18 | ThoMe | Hat die Multiplexanschluesse? |
| 12:29.28 | masus | jep |
| 12:29.29 | ThoMe | Ich hab bis jetzt nur ne ganz normale 4xs0-port karte. |
| 12:29.40 | ThoMe | sollte aber wohl ned viel anders sein? |
| 12:29.44 | ThoMe | ich nutze dafuer misdn |
| 12:29.47 | ThoMe | und bin sehr zufrieden damit. |
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| 12:56.32 | masus | ThoMe: http://pbx-manager.de/installation-fritzcard-asterisk.php |
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| 13:00.09 | dror99 | hi |
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| 13:00.16 | ThoMe | masus: Ja.. seh schon. Die nutzen halt chan_capi-cm-0.6.4. und ich dachte, mISDN koennt ja auch gehen? |
| 13:00.32 | lirakis | morning |
| 13:00.51 | dror99 | I've made a little patch to asterisk code and would like to discuss it |
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| 13:05.36 | shido6 | discuss |
| 13:06.17 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 13:06.21 | ZaVoid | morning guys |
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| 13:11.18 | dror99 | dror99> In the agi interface, I've added another vaiable that is passed from asterisk agi_processid. |
| 13:11.19 | dror99 | <dror99> I use this variable in the script I call from the AGI command. |
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| 14:04.14 | thewiizle | hey can anyone explain what the purpose of user=phone is |
| 14:04.21 | thewiizle | and if its required specifically |
| 14:04.47 | Katty | mew. |
| 14:05.15 | Wonka | *stroke* |
| 14:05.34 | thewiizle | is it the context for the invite |
| 14:05.56 | *** join/#asterisk ManxPower (n=manxpowe@143.sub-70-220-220.myvzw.com) |
| 14:06.25 | [TK]D-Fender | Katty: Mew. |
| 14:06.36 | Katty | herro Wonka |
| 14:06.39 | Katty | how's your chocolate factory? |
| 14:07.19 | Wonka | bad. down. |
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| 14:11.08 | case_ | if anyone know a way to set up an incremental counter (for name generation) without involving mysql or other dependancies, it'll help me a lot |
| 14:11.27 | Uatec | Hey, has anybody had issues with SIP traversing netgear routers? Particularly a DG834PN? |
| 14:11.44 | ManxPower | case_: there is no such feature than I can think of. If you are using 1.4 I have a couple of ideas. |
| 14:11.58 | Uatec | i configured my sip device to use port 5060 and 10,000-20,000, and forwarded all the ports from the router to the phone |
| 14:12.00 | thewiizle | fo $1+1 |
| 14:12.02 | thewiizle | *for |
| 14:12.15 | Uatec | and although calls can be created and destroyed, no audio works eithe rway |
| 14:12.17 | thewiizle | *for $i+1 |
| 14:12.20 | case_ | ManxPower, 1.2 :/ |
| 14:12.23 | ManxPower | Uatec: does the box have specific support for SIP+NAT? |
| 14:12.31 | Uatec | in the end i had to put the phone in the DMZ, and it works fine. but it's not exactly secure |
| 14:12.37 | Uatec | ManxPower, no, it does not |
| 14:12.45 | ManxPower | case_: let me check a few things. Of course $i+1 would have a RACE CONDITION! |
| 14:13.06 | Uatec | it DOES however have specific DOS protection, and when that was turned on, it flagged up every routed UDP RTP packet as a potential DOS attack |
| 14:13.46 | case_ | ManxPower, i try to reproduce the beaviour of hylafax (where faxes are faxXXXXXXX.tif ), where XXXXX is incremented... |
| 14:14.23 | thewiizle | can someone confirm this |
| 14:14.35 | ManxPower | case_: I don't have any suggestions for 1.2 that does not have a significant chance of a race condition. MY idea was use something like MacroExclusive (1.4 specific) to lock the code for incementing the variable. |
| 14:14.42 | thewiizle | user=phone is a PBX sided feature to indicate to termination gateways/systems the type of Number that is being passed |
| 14:14.49 | thewiizle | useful for routing tables etc |
| 14:14.56 | ManxPower | case_: are you just looking for a unique string that will never be repeated? |
| 14:15.31 | case_ | ManxPower, that the 2nd choice (uniqueid) , but i'm was looking for that incremental counter feature |
| 14:15.35 | ManxPower | thewiizle: I can't confirm THAT, but I can say that user=phone is a normal thing in sip and really has no significant meaning to Asterisk. |
| 14:15.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.58 | case_ | would it be that ugly if i use a file to store the counter? can i prevent race condition with file locks ? |
| 14:16.02 | ManxPower | case_: You could write an AGI that uses locking to prevent race conditions. |
| 14:16.18 | case_ | sorry, AGI ??? |
| 14:16.25 | ManxPower | case_: Asterisk supported NO dialplan based locking before 1.4 |
| 14:16.28 | ai-a | case_: why not use a db / |
| 14:16.31 | ManxPower | case_: external application. |
| 14:16.39 | ManxPower | ai-a: race conditions. |
| 14:16.51 | thewiizle | Im just trying to understand the importance, if any |
| 14:16.52 | ai-a | race conditions of what ? |
| 14:17.05 | ai-a | a db doesnt have race conditions. |
| 14:17.07 | thewiizle | or how it should be used and when it shouldnt |
| 14:17.11 | thewiizle | this seemed the logical place to asl |
| 14:17.14 | thewiizle | *ask |
| 14:17.16 | case_ | ai-a, what kind of db ? sgdb like mysql? |
| 14:17.28 | ai-a | all db's have row / table locking. |
| 14:17.40 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.249) |
| 14:17.49 | ManxPower | ai-a: two threads read the current value of the counter from the database, one thread increments and writes, then the other thread increments and write. BINGO! race condition. |
| 14:17.54 | case_ | ai-a, i don't need a sgdb right now. using mysql to implement a counter is like using a nuclear device to open a locked door... |
| 14:18.16 | ai-a | BEGIN update tab set v=v+1 where line = 4; COMIT; |
| 14:18.21 | ManxPower | the problem is that you can't prevent the value your thread read from being modified by another thread. |
| 14:18.39 | ManxPower | ai-a: that might work. |
| 14:18.43 | ai-a | it WILL work. |
| 14:18.48 | ai-a | been around ince the 60s. |
| 14:18.50 | ai-a | *since |
| 14:18.54 | ManxPower | that is SQL, not AsteriskDB, however. |
| 14:19.04 | J4zen | Does anyone happen to have a SNOM320? |
| 14:19.21 | ManxPower | so you would have to build and install an SQL server, which to me seems much more complicated than writing a simple AGI that does locking. |
| 14:19.30 | ai-a | query = "exec inc(4)" - create stored procedure inc(_line) BEGIN ... |
| 14:19.44 | ai-a | mysql supports procedures and functions |
| 14:19.58 | ManxPower | ai-a: but that locking happens in side the database/application, still does not do it using locking from the dialplan. |
| 14:20.15 | ai-a | why lock it on the dialplan ? |
| 14:21.47 | ManxPower | ai-a: because then you would not have to install a database and figure out how to get asterisk-apps to compile and link against the database. |
| 14:22.15 | ai-a | ManxPower: does asterisk support file locking / semaphores / race conditions ? |
| 14:22.30 | ManxPower | It just seems to me to be sort of silly to run a multimegabyte application just to update a counter. |
| 14:22.43 | ManxPower | ai-a: no it does not (at least in the dialplan). |
| 14:22.44 | case_ | ManxPower, +1 |
| 14:23.04 | ai-a | ManxPower, later you'll want sql anyway Heh. |
| 14:23.14 | ManxPower | ai-a: at least in 1.2. In 1.4 there is a way to basically have a semaphore. |
| 14:23.19 | ai-a | for your recordings,, call costs per phone, ddi lookup. |
| 14:24.02 | ManxPower | ai-a: all he wants to do is have a counter be incremented. |
| 14:24.36 | ai-a | okay, so we conclude it is possible, but you want to do it the hard way by not installing a free db. |
| 14:24.57 | case_ | ai-a, i have an asterisk working for 2 years and don't need any sgdb, i won't install one just for that |
| 14:25.27 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:26.07 | ManxPower | case_: an AGI is an external application for Asterisk, similar to CGI is used for Web servers. |
| 14:26.22 | case_ | i have to have a look on that |
| 14:26.23 | ManxPower | case_: do you know any programming languages? |
| 14:26.35 | case_ | some :) |
| 14:26.41 | ManxPower | case_: which one(s)? |
| 14:26.58 | ManxPower | case_: AGI is still pretty heavy weight, but still much less so than the overhead of a database. |
| 14:27.11 | case_ | c c++ c# python php... bash, perl a little bit |
| 14:27.51 | ManxPower | case_: Cool. Look up AGI info on the wiki. Just remember that the Wiki has a lot of wrong information on it. |
| 14:28.05 | case_ | ok, thanks a lot |
| 14:28.05 | ai-a | write a c app that locks / increment the contents and writes it back out then. |
| 14:28.11 | [TK]D-Fender | the BOOK ahs a lot of AGI info |
| 14:28.15 | [TK]D-Fender | ~book |
| 14:28.15 | 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 |
| 14:28.20 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 14:28.20 | ai-a | basicly doing same as the db would. |
| 14:28.32 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 14:28.42 | robl^ | the wiki is always 100% accurate, but Asterisk is often wrong and not matching up to the wiki ;-) |
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| 14:29.09 | twisted | lol |
| 14:29.30 | ManxPower | <-- humor impared -- still work on first cup of coffee. |
| 14:29.52 | Zefk | Hi all, after upgrading from 1.4.11 to 1.4.13 cdr is not working any more: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/voiceblue-087d1110' not posted. Any solutions ? |
| 14:32.06 | twisted | time for a little bit of morning rock-n-roll: http://www.youtube.com/watch?v=eBGIQ7ZuuiU |
| 14:32.52 | ManxPower | Zefk: nothing was listed in the changelog for 1.4.13 that might be helpful? |
| 14:33.09 | tzanger | coppice: hahaha I just read your copyright notice in the dsp routines |
| 14:33.23 | *** part/#asterisk mocker (n=user@198.247.173.227) |
| 14:33.37 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 14:36.24 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:37.16 | pif | hi, when chanspy'ing on asterisk I often hear only the local leg of the conversation |
| 14:39.59 | thewiizle | Tk |
| 14:40.07 | thewiizle | dude |
| 14:40.10 | thewiizle | show me some love :) |
| 14:40.42 | thewiizle | tell me what the point in user=phone is |
| 14:40.46 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 14:40.55 | thewiizle | im reading googles million page search results but nothing clearly states what it is |
| 14:42.53 | [TK]D-Fender | thewiizle: It is mysterious and can be best defined as what it isn't :) |
| 14:43.12 | ManxPower | thewiizle: nobody cares about it, as it does not affect people using Asterisk. |
| 14:43.30 | ManxPower | thewiizle: try reading the SIP RFC instead. |
| 14:43.33 | coppice | tzanger: I think that copyright notice was given an award somewhere :-) |
| 14:43.43 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:44.55 | thewiizle | yeh im reading it now |
| 14:45.05 | thewiizle | it seems to have something to do with RFC number compliance |
| 14:45.30 | robl^ | its pretty much a "deprecated" requirement |
| 14:45.33 | thewiizle | when user=phone is specified it indicates the To: field contains a number which should be formatted in a particular way |
| 14:45.40 | thewiizle | thats my summary so far |
| 14:45.52 | thewiizle | *indicates THAT the |
| 14:46.12 | *** join/#asterisk blueneon (n=blueneon@dsl-146-77-86.telkomadsl.co.za) |
| 14:47.13 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:48.29 | thewiizle | seems it was introduced due to compuserce |
| 14:48.33 | thewiizle | *Compuserve |
| 14:48.41 | *** join/#asterisk ajohnstone (n=ajohnsto@85.211.235.68) |
| 14:48.42 | thewiizle | and their stupid account naming policies |
| 14:49.04 | De_Mon | Zefk I get that warning too, thought it was because I adding debugging to the CLI, as the CDR is still running normally. |
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| 14:50.42 | Zefk | De_Mon: This wasn't in 1.4.11. |
| 14:52.00 | blueneon | im trying to get asterisk to use the Monitor() function only once the call has been answered, how would i do this, I tried just placing it after the Dial() function in the dial plan, but it seems that doesnt work, if I place it before the Dial() function then it records the onhold music etc until eventually the call is answered and continues to rec. |
| 14:52.28 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
| 14:52.51 | De_Mon | Zefk I didn't have dubugging sent to CLI till upgrading (for a different reason) so you could be right. I'm just saying my CDRs are working normally after upgrading and I get that WARNING too |
| 14:54.26 | De_Mon | blueneon priorities after dial are execute after the call is hung up. When you say music on hold, are you talking * MOH or music from the number you called? |
| 14:54.31 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:54.37 | Zefk | De_Mon: yes. it seems that it is working for me also. |
| 14:54.46 | ManxPower | blueneon: Dial blocks the dialplan until the Dial exits. |
| 14:54.57 | ManxPower | you can look at the M() option of Dial to execute a macro on answer. |
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| 15:06.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:08.03 | blueneon | thanks ManxPower |
| 15:10.31 | tzanger | any other Rogers cell users here? Did your phone autoupdate the time? |
| 15:10.37 | tzanger | I had to cycle power on my L6 to get it to come up right |
| 15:12.39 | thewiizle | hmmm |
| 15:12.42 | thewiizle | Monitor() |
| 15:13.07 | robl^ | tzanger: more than likely its a cell phone firmware issue. I know at least with the blackberry, you have to apply a DST patch |
| 15:13.25 | tzanger | robl^: fun times |
| 15:13.28 | tzanger | my CDMA phones just worked |
| 15:13.32 | tzanger | but my rogers one did not |
| 15:13.45 | tzanger | not sure if it's the phone or the network not sending a "change your time you fuckwit" |
| 15:16.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:16.15 | `Sauron | Hum, anyone know why I'm getting /tmp/zaptel-1.4.6/zaptel-base.c:788: error: `fcstab' undeclared (first use in this function) when trying to compile zaptel on Linux 2.6.5-7.201-smp x86_64 ? |
| 15:17.31 | tzafrir | `Sauron, because you also get the same warning in i386 |
| 15:18.26 | `Sauron | tzafrir: And then the compile bombs. |
| 15:18.58 | tzafrir | `Sauron, hmm... undeclared? on what version of Zaptel? |
| 15:19.02 | tzafrir | 1.4.6? |
| 15:19.05 | `Sauron | ya |
| 15:19.46 | `Sauron | So maybe somebody removed the table and didn't clean up the rest.. :p :) |
| 15:19.49 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 15:21.21 | `Sauron | hum |
| 15:21.31 | tzafrir | No. It should never be used directly, IIRC |
| 15:22.17 | `Sauron | tzafrir: http://www.pastebin.ca/762268 |
| 15:23.47 | `Sauron | hehn |
| 15:23.58 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:23.59 | `Sauron | I love how there's no 1.4.5 in the download directory |
| 15:24.09 | *** join/#asterisk kensukeido (i=be0210a1@gateway/web/cgi-irc/ircatwork.com/x-be703b177a85f349) |
| 15:24.09 | `Sauron | 1.2 latest and 1.4 latest |
| 15:24.11 | `Sauron | grumble |
| 15:24.13 | kensukeido | hi people |
| 15:24.25 | tzafrir | `Sauron, older versions are in the releases/ subdirectory |
| 15:24.31 | `Sauron | yeah, just figured that out |
| 15:24.32 | tzafrir | and use 1.4.5.1, not 1.4.5 |
| 15:24.55 | `Sauron | roger that |
| 15:25.05 | tzafrir | Or better, svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4 |
| 15:25.20 | tzafrir | This is much less download for each backtrack |
| 15:25.27 | `Sauron | can't svn through firewall |
| 15:26.02 | robl^ | umm.. why not?? its going through http, not svn: ;-) |
| 15:26.13 | `Sauron | svn does PROPFIND and stuff |
| 15:26.22 | `Sauron | and our proxy is retarded and can't handle them |
| 15:26.25 | `Sauron | not like squid |
| 15:26.39 | file | try port 8080 |
| 15:26.53 | file | I *think* the public mirror has it running on there too for this reason |
| 15:27.09 | `Sauron | Hehn. |
| 15:27.10 | `Sauron | wtf |
| 15:27.15 | `Sauron | 1.4.5.1 has same problem |
| 15:27.24 | `Sauron | <PROTECTED> |
| 15:27.24 | `Sauron | /tmp/zaptel-1.4.5.1/zaptel-base.c: In function `calc_fcs': |
| 15:27.25 | `Sauron | /tmp/zaptel-1.4.5.1/zaptel-base.c:786: error: `fcstab' undeclared (first use in this function) |
| 15:27.53 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:28.09 | `Sauron | huh |
| 15:28.15 | `Sauron | #if !defined(LINUX26) |
| 15:28.15 | `Sauron | static |
| 15:28.15 | `Sauron | __u16 fcstab[256] = |
| 15:28.15 | `Sauron | { |
| 15:28.25 | `Sauron | I'm on 2.6 |
| 15:28.28 | robl^ | you must be missing something. I just built zaptel about 45 mins ago |
| 15:28.36 | `Sauron | So it's not getting compiled |
| 15:28.40 | robl^ | kernel-devel? |
| 15:28.51 | `Sauron | kernel-source-2.6.5-7.201 |
| 15:29.08 | iCEBrkr | I knew a Sauron back in the oldschool BBS days. |
| 15:29.26 | `Sauron | hum |
| 15:29.30 | `Sauron | depends on the BBS's |
| 15:29.33 | case_ | i kewn a sauron back in the oldschool lord of the ring days... |
| 15:29.43 | robl^ | I knew a Sauron back in the LOTR days |
| 15:29.55 | `Sauron | I've used this nick (with our w/o the `) for > 10 years |
| 15:29.58 | `Sauron | possibly > 15 years |
| 15:30.02 | *** join/#asterisk d3wayne (n=d3wayne@pool-71-187-1-180.nwrknj.fios.verizon.net) |
| 15:30.08 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 15:30.55 | iCEBrkr | `Sauron: You know anything about THG? ACiD? iCE? etc.. I'm trying to recall what the affiliation was.. hrrm |
| 15:31.07 | `Sauron | Hum |
| 15:31.23 | `Sauron | Vaguely ring a bell, but I either don't know enough, or have forgotten. |
| 15:32.24 | `Sauron | I took out the #if !defined ... |
| 15:32.26 | `Sauron | and #endif |
| 15:32.50 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:32.51 | iCEBrkr | Ahh, then you're not him :) |
| 15:33.15 | `Sauron | I was in the euro BBS scene long enough ago |
| 15:33.21 | `Sauron | So yeah, not surprised. :) |
| 15:33.32 | iCEBrkr | Ahh |
| 15:33.33 | iCEBrkr | ok |
| 15:33.45 | iCEBrkr | Sometimes IRC can be a small world.. |
| 15:33.55 | Nugget | CONNECT 300 |
| 15:33.59 | iCEBrkr | I still have people /msging me from the scene. |
| 15:34.14 | iCEBrkr | haha |
| 15:34.42 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 15:34.59 | `Sauron | Nugget: ++ATH0 |
| 15:35.43 | `Sauron | hum |
| 15:35.45 | `Sauron | wth |
| 15:35.54 | `Sauron | why does "make install" in zaptel try to download stuff? |
| 15:36.33 | Nugget | if you guys haven't seen http://www.bbsdocumentary.com/ yet you should track down a copy. |
| 15:36.36 | Nugget | it's fan-fucking-tastic |
| 15:36.54 | iCEBrkr | Nugget: Yea, my buddy was interviewd for the art scene part. |
| 15:37.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:37.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:37.37 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:43.50 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.249) |
| 15:44.27 | twisted | iCEBrkr: i remember all of those :) |
| 15:44.49 | iCEBrkr | twisted: nerd. :P |
| 15:44.54 | iCEBrkr | oh wait. |
| 15:45.08 | twisted | lol |
| 15:45.38 | iCEBrkr | TSRI, 1911, hrrm who else..:P |
| 15:45.43 | iCEBrkr | When loaders rocked your world. |
| 15:45.46 | twisted | yeah |
| 15:45.55 | twisted | it's been ages |
| 15:46.52 | coppice | youngster |
| 15:46.57 | iCEBrkr | twisted: Late bloomer |
| 15:47.02 | twisted | pfft |
| 15:47.06 | iCEBrkr | 89-93 here :) |
| 15:47.07 | twisted | i was 12 |
| 15:47.09 | iCEBrkr | hahah |
| 15:47.14 | iCEBrkr | coppice: ^5 |
| 15:47.30 | coppice | did people still run a BBS as late as 1992? |
| 15:47.46 | twisted | are you kidding? |
| 15:47.50 | [TK]D-Fender | coppice: I ran one through 1994... |
| 15:47.54 | iCEBrkr | coppice: in 216 we did. The internet wasn't popular until 94 or so |
| 15:48.04 | Katty | twisted: ah HA! |
| 15:48.05 | Nugget | I shut mine down around 1994. |
| 15:48.12 | Katty | twisted: i have found you my pretty!! bwahahahha |
| 15:48.16 | twisted | i finally shut down around 98 |
| 15:48.20 | iCEBrkr | Nugget: Me too, cuz I discovered IRC |
| 15:48.21 | Nugget | not coincidentally, the same year I registered slacker.com :) |
| 15:48.26 | iCEBrkr | haha |
| 15:48.31 | twisted | was averaging about 2 calls a day, down from over 100 |
| 15:48.47 | coppice | I gave up before moving to asia, which was in 1991. in the UK BBSes had already gone pretty quiet |
| 15:49.09 | iCEBrkr | I ported ShockWavE:Pro over to Linux using freePascal and setup a telnet BBS, but no one used it. I think I had 2-3 connections a week. |
| 15:49.10 | Nugget | telnet is eeeeeeevil! |
| 15:50.04 | iCEBrkr | I need to hunt down the C!A art packs I was in. |
| 15:50.15 | twisted | argh |
| 15:50.16 | iCEBrkr | I found them once, but I can't remember where I put'm |
| 15:50.17 | twisted | i did telnet bbsing |
| 15:50.22 | twisted | part of the reason i shut it down |
| 15:50.29 | twisted | PITA |
| 15:50.32 | iCEBrkr | haha |
| 15:50.51 | iCEBrkr | I was going to just run Tradewars as telnetable. |
| 15:50.59 | twisted | i wanted to keep LoRD |
| 15:51.02 | iCEBrkr | haha |
| 15:51.07 | twisted | man, i loved that game |
| 15:51.16 | iCEBrkr | Blacknova Traders == Web-based Tradewars :) |
| 15:51.21 | twisted | good thing there's www.lotgd.net |
| 15:51.27 | iCEBrkr | oh dear. |
| 15:51.37 | De_Mon | yeah thank goodness for that |
| 15:51.50 | De_Mon | I was partial to LoD myself |
| 15:52.17 | De_Mon | it had vga graphics ;P |
| 15:52.26 | twisted | ugh. |
| 15:52.26 | coppice | we used to run BBSes over strings and cans back in the good old days |
| 15:52.30 | iCEBrkr | oh the LoD client |
| 15:52.44 | iCEBrkr | coppice: 10cps? |
| 15:52.44 | twisted | might as well have used a RIP board |
| 15:52.56 | iCEBrkr | RIP rocked, but it was 5yrs too late. |
| 15:53.04 | twisted | yeah... i wasn't too impressed |
| 15:53.11 | coppice | yep. 10cps - 110 baud modems :-) |
| 15:53.17 | twisted | it's graphics compression was good though |
| 15:53.38 | iCEBrkr | coppice: I remeber when I got my 1200 baud and it wasn't about CPS anymore.. I was like WOW! Fly'n! |
| 15:54.06 | iCEBrkr | Ony of my friends wrote his own graphics protocol which just extended ANSI, so you could do animations without the need for more bandwidth. |
| 15:54.17 | iCEBrkr | Timebanks where the shit! |
| 15:54.20 | coppice | 110baud with an ASR33 teletype was more about chugging |
| 15:54.20 | iCEBrkr | lol |
| 15:55.00 | twisted | coppice: you think iaxmodem would work well as an interface for a bbs? :P |
| 15:56.09 | coppice | yeah. you could recreate that old modem (was it US Robotics) that ping ponged a V.29 FAX modem to kinda get sorta 9600bps :-) |
| 15:56.10 | *** join/#asterisk ManxPower (n=manxpowe@143.sub-70-220-220.myvzw.com) |
| 15:56.19 | twisted | hehe |
| 15:57.01 | coppice | actually, some POS terminals do just that |
| 15:57.03 | iCEBrkr | HST |
| 15:57.08 | twisted | well yeah |
| 15:57.11 | twisted | for cc validation |
| 15:57.12 | iCEBrkr | Dual-Standard 38400 yo |
| 15:57.54 | twisted | i just think it'd be funny to run a bbs off an asterisk box :P |
| 15:58.09 | iCEBrkr | ha |
| 15:58.10 | twisted | dosemu + renegade/wildcat |
| 15:58.25 | twisted | ( i still have those damn bbs packages too ) |
| 15:58.25 | nestAr | lol |
| 15:59.40 | nestAr | i was late to that, my first modem was on 2400 |
| 15:59.49 | nestAr | 110 sounds terrible. |
| 16:00.17 | *** join/#asterisk tulcod (n=auke@a62-251-21-22.adsl.xs4all.nl) |
| 16:00.33 | coppice | its not the 100 that sounds terrible, its the bloody ASR33 teletype that does |
| 16:01.22 | tulcod | tzafrir: can't you put the ebuild in sunrise? |
| 16:02.25 | tzafrir | tulcod, I don't know anything about Gentoo's systems, and don't have the time to learn |
| 16:03.03 | tzafrir | If this is of minimal importance to anybody, then he should take that, give it some sanity check and testing and pput it whereever necessary |
| 16:03.21 | tulcod | tzafrir: then how come you're able to make an ebuild :p |
| 16:03.39 | *** part/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 16:03.40 | tzafrir | tulcod, because I know shell scripts and I know zaptel |
| 16:04.01 | tzafrir | I know how to avoid errors. I don't know what I need to test |
| 16:04.07 | tulcod | tzafrir: that's all you need to know to be able to put stuff in sunrise ;) |
| 16:04.46 | tulcod | tzafrir: join #gentoo-sunrise and ask around |
| 16:04.46 | tzafrir | tulcod, then go ahead and put it there |
| 16:04.46 | tulcod | tzafrir: I don't even have intel hardware, would be no use |
| 16:04.46 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:04.50 | tulcod | zaptel |
| 16:04.51 | tulcod | whatever |
| 16:04.53 | tzafrir | tulcod, given that I don't even have a working build system |
| 16:05.01 | tzafrir | I really can't maintain it |
| 16:05.10 | tzafrir | And hence won't have my name on it |
| 16:05.17 | tulcod | ah |
| 16:05.26 | tulcod | well then, file a bug on bugzilla |
| 16:05.38 | `Sauron | Hum. |
| 16:05.40 | tzafrir | Which requires registering. Yuck |
| 16:05.41 | tulcod | maybe someone else with zaptel stuff can maintain it |
| 16:05.45 | tulcod | :-/ |
| 16:05.49 | iCEBrkr | I'll just write a BBS in AEL |
| 16:05.51 | iCEBrkr | :) |
| 16:05.53 | `Sauron | [Nov 5 10:05:45] WARNING[3799]: pbx.c:1797 pbx_extension_helper: No application 'MeetMe' for extension (from-trunk, 49901, 2) |
| 16:05.57 | `Sauron | How odd. |
| 16:06.07 | tzafrir | someone must be maintaining the package. He must be hanging out around he. Or someone who knows him |
| 16:06.11 | tulcod | tzafrir: yeah, might take up to 5 minutes of YOUR WHOLE LIFE :p |
| 16:06.36 | tulcod | tzafrir: just file a bug on bugs.gentoo.org, attach the ebuild and everything else needed (ie, patches), and run |
| 16:06.47 | tulcod | tzafrir: I mean, that's the easiest way to get us maintaining a package |
| 16:06.49 | [TK]D-Fender | `Sauron: Go setup a Zaptel timing source and recompile * and maybe, just MAYBE, you'll get MeetMe, and Page compiled ;) |
| 16:06.57 | tzafrir | And dump the temporary email address? |
| 16:07.44 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:07.56 | tulcod | tzafrir: you won't get a single email after registering if you don't want to |
| 16:08.10 | `Sauron | TKD: Hum. Blah. |
| 16:08.19 | `Sauron | I compiled * before zaptel |
| 16:08.20 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:10.02 | `Sauron | I wish I too could drink of the digium kool-aid, so their install practices would make sense. |
| 16:10.32 | ManxPower | `Sauron: uh, all the docs say to install zaptel before Asterisk |
| 16:10.49 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:10.56 | ManxPower | MeetMe uses Zaptel, so zaptel must be installed or it won't even find the header files for zaptel. |
| 16:11.15 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:11.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:11.50 | `Sauron | ManxPower: the * README makes no mention of installing Zaptel first. |
| 16:12.01 | `Sauron | * NEW INSTALLATIONS |
| 16:12.26 | `Sauron | compiler, clib, openssl, ncurses, zlib |
| 16:12.34 | `Sauron | are the only prerequisites listed |
| 16:12.44 | ManxPower | except of course, for the actual readme |
| 16:12.59 | [TK]D-Fender | *sigh* |
| 16:13.35 | iCEBrkr | [TK]D-Fender: oh stop |
| 16:13.58 | `Sauron | I'm just saying. The asterisk readme (there's no INSTALL) does not mention having to install zaptel first. |
| 16:14.28 | iCEBrkr | When geeks write how-to's and documentation there's a lot of assumed knowledge. |
| 16:14.29 | [TK]D-Fender | `Sauron: ... |
| 16:14.31 | [TK]D-Fender | ~book |
| 16:14.31 | 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 |
| 16:14.40 | `Sauron | eyeroll |
| 16:15.07 | iCEBrkr | But it's like compiling php + mysql. PHP uses MySQL so you might want to build it first. |
| 16:15.30 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:15.42 | `Sauron | icebrkr: You should've drank the php kool-aid |
| 16:15.46 | `Sauron | or mysql |
| 16:16.04 | iCEBrkr | Man, back in the day when you wanted to do PHP + Apache + MySQL yadda yadda |
| 16:16.09 | iCEBrkr | It was all a catch .22 |
| 16:16.22 | `Sauron | Heh. php is still a bitch to install |
| 16:16.30 | iCEBrkr | You had to ./configure parts, make ;; make install other parts.. |
| 16:16.31 | ManxPower | `Sauron: I whined about the issue on #asterisk-dev |
| 16:17.05 | `Sauron | Aww... |
| 16:17.33 | `Sauron | I mean. All it takes is to add a line to the README saying "Please also install zaptel-blahblah before compiling asterisk." |
| 16:17.44 | ManxPower | `Sauron: /join #asterisk-dev and complain. |
| 16:18.14 | ManxPower | `Sauron: ACTUALLY it should say if you need the following features, you have to install zaptel first. |
| 16:18.17 | *** part/#asterisk tulcod (n=auke@a62-251-21-22.adsl.xs4all.nl) |
| 16:18.21 | ManxPower | Asterisk does NOT require zaptel. |
| 16:18.41 | ManxPower | only a few specific features require zatel |
| 16:18.44 | ManxPower | zaptel too |
| 16:20.01 | [TK]D-Fender | `Sauron: Zaptel isn't a requirement for *, but it is "suggested" :) |
| 16:20.10 | tzafrir | the README does not list other optional components |
| 16:20.15 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:20.29 | ManxPower | it is required for MeetMe, IAX2 trunking, or Zaptel compat cards. |
| 16:20.33 | `Sauron | Okay, I take it back. |
| 16:20.44 | [T]ank | have 4 pris in a digium TE410P and periodically I get this error and drop a few calls: http://pastebin.ca/762323 any help would be appreciated. |
| 16:20.52 | `Sauron | I was wrong, and you all are right. How dare I criticize the almighty asterisk. |
| 16:21.31 | [TK]D-Fender | `Sauron: You gain wisdom child :D |
| 16:21.42 | ManxPower | [T]ank: no other errors like HDLC errors? |
| 16:21.56 | [T]ank | i saw one earlier today |
| 16:21.59 | [T]ank | but that was it |
| 16:22.08 | [T]ank | nope i was wrong... just got one |
| 16:22.15 | [T]ank | [Nov 5 23:37:56] NOTICE[18768]: chan_zap.c:8462 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 |
| 16:22.27 | [T]ank | so that makes 2 in the last hour |
| 16:22.51 | ManxPower | HDLC errors mean one of two things. 1) you are getting line errors from the PRI or 2) your system is locking interrupts for so long data is lost. (2) is the most common) |
| 16:23.11 | [T]ank | how do i check for (2) |
| 16:23.59 | ManxPower | Onboard GigEthernet, onboard SATA, running in graphics mode or framebuffer mode, and RAID cards can all cause this issue. Fixes were put into zaptel at one point to help with these issues. Make sure you are running the latest zaptel for your major version. |
| 16:24.45 | ManxPower | also, IRQ sharing could EASILY cause this issue. |
| 16:24.45 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 16:24.58 | ManxPower | cat /proc/interrrupts to see if anything else is on the same IRQ as the card. |
| 16:25.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.57 | [T]ank | here is my output of act /proc/interrupts: http://pastebin.ca/762334 |
| 16:26.45 | [TK]D-Fender | [T]ank: What ver of * & zaptel? |
| 16:27.05 | [T]ank | 1.4.10.1 |
| 16:27.26 | ManxPower | no conflicts, but you to have USB enabled as well as acpi |
| 16:27.44 | [T]ank | is that bad? |
| 16:27.44 | ManxPower | I sort of doubt you will be needing ACPI to go into and out of sleep mode. |
| 16:27.59 | ManxPower | [T]ank: the more stuff loaded, the slower IRQs are serviced. |
| 16:28.07 | [T]ank | ok |
| 16:28.27 | [T]ank | but do you think that is the cause of my issue? |
| 16:28.34 | [T]ank | never had an issue in 4 months until today |
| 16:28.38 | ManxPower | [T]ank: there is no way to know. |
| 16:28.48 | ManxPower | well if it just started today then maybe it is a line problem. |
| 16:28.59 | [T]ank | worth reporting to the phone company? |
| 16:29.07 | ManxPower | also, you realize that the difference between getting the data off the card and not getting the data off the card is like 5 microseconds, right? |
| 16:29.19 | [T]ank | yeah |
| 16:29.47 | ManxPower | You are SURE there is nothing on your system that could be adding a few microseconds to the time it takes to service the interrupts from the Zaptel card? |
| 16:30.00 | [T]ank | no. not sure |
| 16:30.03 | [T]ank | i never am |
| 16:30.06 | ManxPower | Exactly. |
| 16:30.22 | [T]ank | ;-) at least i admit it :-D |
| 16:30.26 | ManxPower | so the fact that it started having problems recently could just be because more people are leaving voicemail or stuff like that. |
| 16:30.47 | [T]ank | good point. we do have more traffic as of today |
| 16:30.53 | ManxPower | the solution to HDLC errors is to try to fix every possible cause and hope one of them works. |
| 16:33.09 | ManxPower | [T]ank: asterisk has a fair number of issues that only show up under load. |
| 16:33.19 | [T]ank | thanks |
| 16:33.35 | ManxPower | and those are the hardest for the programmers to fix, since they can't reproduce them. Anyway, make sure you have the LATEST zaptel installed. |
| 16:33.48 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:35.39 | destructure | they're also largely mitigated by how inexpensive asterisk is |
| 16:36.00 | destructure | it's easier to drop in a second box than to create crazily optimized configurations |
| 16:37.23 | ZaVoid | hey guys.. any ideas what would cause a sound recording to play "stuttery" and random points and different each time.. and its not network bandwidth related |
| 16:37.50 | penguinFunk | hardware resources completely consumed? |
| 16:37.58 | ZaVoid | nope |
| 16:38.00 | penguinFunk | CPU/MEM/Disk IO ? |
| 16:38.02 | ZaVoid | load average is 0.10 |
| 16:38.11 | ZaVoid | cpu is like 1% |
| 16:38.13 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:38.17 | ZaVoid | plenty of disk space |
| 16:38.23 | penguinFunk | not disk space |
| 16:38.25 | penguinFunk | disk usage |
| 16:38.32 | penguinFunk | IO != size |
| 16:38.40 | outtolunc | what else is consuming irqs |
| 16:38.43 | penguinFunk | decent sound card? |
| 16:38.45 | ZaVoid | lets say the file says "hi how are you today mr wilco and i love that dress your wearing".... depending on the random time i have the file played.. different parts will stutter out or not at all |
| 16:38.52 | ManxPower | ZaVoid: You didn't even tell us what Digium card or SIP phone you are using. |
| 16:39.29 | ZaVoid | ManxPower: no digium cards. its usally on DIDs inbound via sip the box but i can reproduce it using sj phone clients and sound point IP phones and grandstream phones |
| 16:40.05 | ManxPower | ZaVoid: do you have zaptel even installed on the system? |
| 16:40.06 | penguinFunk | ~gs |
| 16:40.07 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:40.15 | penguinFunk | ZaVoid ^ |
| 16:40.30 | ZaVoid | hmm good question on this one ManxPower |
| 16:40.31 | ZaVoid | it should be |
| 16:40.33 | penguinFunk | try a different sip phone |
| 16:40.41 | ZaVoid | yes i know gs phones suck penguinFunk but it happens elesewhere |
| 16:40.48 | penguinFunk | ok |
| 16:40.52 | ZaVoid | mostly happens on SIP inbound DID's so no hardware but asterisk involved |
| 16:41.03 | ManxPower | ZaVoid: does the problem go away if you stop asterisk, unload (rmmod) all the zaptel and ztdummy stuff) and start Asterisk again? |
| 16:41.16 | ZaVoid | that i haven't tried yet ManxPower |
| 16:41.29 | ZaVoid | thats my best bet to troubleshoot it ya think? |
| 16:41.38 | ManxPower | ZaVoid: it's the best place to start. |
| 16:41.46 | ZaVoid | ok thanks man i'lls tart ther |
| 16:42.39 | ManxPower | I seem to be seeing more reports of issues with ztdummy then there should be. |
| 16:43.54 | ZaVoid | trying zttest |
| 16:43.59 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:44.09 | ZaVoid | i should mention this is my only trixbox too.. my str8 asterisk installs work fine |
| 16:44.39 | ManxPower | ZaVoid: then it will be YOUR job to translate our instrucitons for plain asterisk into the dark, evil world of trixbox. |
| 16:44.44 | ManxPower | don't expect and help with that. |
| 16:44.51 | ZaVoid | yesp i know |
| 16:44.54 | ZaVoid | i'm fine with that |
| 16:45.01 | ZaVoid | ]didn't think it was a trix issue specfically.. |
| 16:45.04 | ZaVoid | trying the zt junk |
| 16:45.14 | ZaVoid | its my only trix box on this network.. the rest are good old reliable 1.4.9 |
| 16:45.23 | ZaVoid | the only good build of asterisk ever :) |
| 16:48.15 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:48.22 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 16:50.41 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 16:51.02 | ZaVoid | zttest worst 80.1 |
| 16:51.03 | ZaVoid | lol |
| 16:52.16 | ZaVoid | oh yeah its also running as a VM on vmware |
| 16:52.24 | ZaVoid | but the only vm server on the box |
| 16:52.30 | [TK]D-Fender | ZaVoid: Go get some KY... you're gonna need it |
| 16:52.35 | ZaVoid | lol sup fender |
| 16:53.02 | ManxPower | ZaVoid: VMWare would be expected to create these issues. |
| 16:53.05 | [T]ank | ManxPower: would you think that these errors reflect the same issue as we were discussing earlier? http://pastebin.ca/762363 |
| 16:53.09 | ManxPower | don't use ztdummy. |
| 16:53.30 | ManxPower | [T]ank: YES! |
| 16:56.05 | *** join/#asterisk shtoom (n=godson@59.93.116.46) |
| 16:56.44 | linagee | wow. hah. so many police men and such depend on caller ID. drawing guns based upon that information. http://youtube.com/watch?v=BQD_NOngwWE |
| 16:57.02 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 16:57.33 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 16:57.58 | shtoom | Hi I am using asterisk 1.2.17 music on hold is not working here is the cli output http://www.pastebin.ca/762372 |
| 16:58.42 | shtoom | I've also installed asterisk addons but still not working |
| 16:59.18 | twisted | shtoom: do you have zaptel installed and ztdummy loaded? |
| 16:59.42 | shtoom | twised: yes I've it installed |
| 16:59.43 | ManxPower | shtoom: that pastebin does NOT indicate an error. |
| 16:59.55 | ManxPower | shtoom: try removing ztdummy and see if that helps. |
| 17:00.08 | twisted | shtoom: is the ztdummy module loaded though? (lsmod) |
| 17:00.26 | *** join/#asterisk Silicium (n=marco@core.forkbomb.ch) |
| 17:00.28 | Silicium | hi there |
| 17:00.39 | Silicium | How are the Dial Menues named? |
| 17:00.40 | twisted | ManxPower: i've only seen the moh act that way when no timing source exists (ie, ztdummy not loaded) |
| 17:00.43 | shtoom | ManxPower: give me a minute |
| 17:01.09 | ManxPower | twisted: I've only seen that on slower systems, regardless of if a timing source was available or not. |
| 17:01.14 | Silicium | for navigate with the phone |
| 17:01.26 | ManxPower | Silicium: we do not understand you |
| 17:01.27 | alrs | ~book |
| 17:01.28 | 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 |
| 17:01.35 | Silicium | ManxPower: i think :D |
| 17:01.56 | twisted | therefore you are? |
| 17:02.25 | Silicium | If i call to a number, then i can choose my destination with the Phonekeys |
| 17:02.54 | ManxPower | Silicium: That is called an IVR. Example: Calling your bank and making menu selections. |
| 17:02.55 | [T]ank | ManxPower: just came across the date being off. Was set to a day in advance. Could this cause timing issues with the PRI? I wonder if that is the cause of all of this. |
| 17:02.59 | Silicium | yep |
| 17:03.00 | Silicium | thanks |
| 17:03.06 | ManxPower | [T]ank: no it could not. |
| 17:03.08 | shtoom | twisted: now I've zaptel and ztdummy loaded now I am not getting that NOTICE on cli but moh is still not working |
| 17:03.14 | [T]ank | hmm. |
| 17:03.28 | twisted | shtoom: are you using file based moh or mpg123 based? |
| 17:03.29 | ManxPower | shtoom: did you have it loaded BEFORE? |
| 17:03.36 | twisted | and did you restart asterisk? |
| 17:04.01 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 17:04.18 | shtoom | twisted:I've not installed mpg123 i've installed addons i think that should use format_mp3.so |
| 17:04.34 | shtoom | i've explicitly specified to load it |
| 17:04.38 | twisted | if you're trying to use format_mp3.so you need to edit your musiconhold.conf and use the settings for filesystem based moh |
| 17:04.57 | ManxPower | if ztdummy is having issues with the hardware timer (USB or RTC) that would cause audio issues. |
| 17:04.58 | shtoom | ManxPower:Previously they were not loaded |
| 17:05.25 | ManxPower | shtoom: are you running asterisk in a virtual machine? |
| 17:05.44 | twisted | k ManxPower, you have fun with this. i think i'm on the right track, though. |
| 17:07.32 | shtoom | twisted:i've default setting for moh here is the config http://www.pastebin.ca/762382 |
| 17:07.45 | ManxPower | shtoom: for one thing that is NOT a valid config file. |
| 17:07.54 | ManxPower | quietmp3 expects to use mpg123, IIRC. |
| 17:08.00 | twisted | that's correct |
| 17:08.04 | ManxPower | you would, of course, want mode files, right? |
| 17:08.11 | shtoom | ManxPower:cli shows that started moh and immediately stopped moh |
| 17:08.30 | shtoom | ManxPower:I am running it on dedicated server |
| 17:08.36 | ManxPower | shtoom: I can see that happening if the config was wrong. |
| 17:08.39 | twisted | shtoom set mode=files |
| 17:08.54 | ManxPower | twisted: isn't that the SECOND time you told him to do that? |
| 17:08.56 | twisted | reload res_musiconhold.sh |
| 17:09.13 | twisted | ManxPower: not explicitly, now |
| 17:09.16 | twisted | s/now/no |
| 17:10.23 | twisted | oh, and res_musiconhold.so, not .sh |
| 17:10.32 | twisted | fingers typing faster than mind moving + dxm == fun |
| 17:11.43 | putnopvut | dxm? Reallly? |
| 17:12.07 | shtoom | twisted:i've reloaded it reloaded but still the same |
| 17:12.17 | shtoom | hmm, i'll restart once |
| 17:12.19 | twisted | putnopvut: yes, i've been sick |
| 17:12.34 | twisted | shtoom: did you change mode=files in musiconhold.conf like i said before? |
| 17:12.39 | putnopvut | right. sick ;) |
| 17:12.56 | twisted | yes, sick. my desk looks like a pharmacy right now |
| 17:13.05 | putnopvut | I'm just messing with you. |
| 17:13.12 | shtoom | twisted,ManxPower: Thanks for your help its working now after restart ! |
| 17:13.27 | twisted | shtoom :) |
| 17:13.51 | coppice | as Michael Palin would say |
| 17:14.00 | *** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 17:14.29 | shtoom | twisted :) BTW is your nick in any way inspired #twisted ? |
| 17:14.44 | twisted | shtoom: i doubt it, but i bet my nick came before that channel |
| 17:14.50 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 17:16.02 | shtoom | twisted: Ok , I was just asking to know if you are a pythonista. |
| 17:16.10 | twisted | no, not really |
| 17:16.14 | twisted | i try to steer away from python if i can |
| 17:16.42 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 17:17.18 | shtoom | twisted: Anyway thanks once again for you help and patience. I gtg! |
| 17:17.40 | twisted | shtoom: np, have fun |
| 17:21.35 | *** join/#asterisk bantu (n=Miranda@p54A32D06.dip0.t-ipconnect.de) |
| 17:26.07 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:26.24 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:26.24 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 17:26.59 | *** join/#asterisk dalbaech (n=dalbaech@darkvoip.net) |
| 17:30.56 | agx | just uploaded version 1.4.1 that allow rx/tx fax at 14000 (instead of fixed speed of 9600), have a look http://sourceforge.net/projects/agx-ast-addons/ |
| 17:31.28 | *** part/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 17:31.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:31.43 | *** join/#asterisk Aeudian (n=somewher@75.148.21.113) |
| 17:32.30 | Aeudian | Are there any options/configuration files which allow for modifcation to Directory() I do not like how the directory speaks each letter of the persons full name and rather it just speak the last name. |
| 17:32.37 | *** join/#asterisk asdx (n=diego@adsl-151-247.click.com.py) |
| 17:33.05 | *** part/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 17:34.56 | *** join/#asterisk guillote_GNU (n=guillote@host101.200-82-48.telecom.net.ar) |
| 17:38.12 | [TK]D-Fender | Aeudian: have your users record their name for directory in VOICEMAILMAIL like they're supposed to then... |
| 17:39.18 | Aeudian | TK: so the spelling is because they have been lazy and not recorded the name as i told them too 3 weeks ago? lol |
| 17:39.54 | [TK]D-Fender | Aeudian: That would be a resounding "yes" |
| 17:40.09 | Aeudian | TK: lazy people i sware, lol. |
| 17:41.58 | [TK]D-Fender | Aeudian: Start wiring each of their chairs up to your taser so you can switch in on demand :) |
| 17:42.49 | [TK]D-Fender | Who else thinks the one on the left looks a lot like the Paramount logo? http://gizmodo.com/gadgets/world.s-tallest/china-begins-construction-on-worlds-tallest-ferris-wheel-318855.php |
| 17:44.52 | Aeudian | TK: lol, it would definitly light a fire under their ass lol |
| 17:45.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584888.dsl.bell.ca) |
| 17:46.37 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 17:47.13 | mrtelephone | it seems I have to turn off DST on most of my equipment such as polycom 501's and cisco ata186's |
| 17:47.17 | [TK]D-Fender | ~fire |
| 17:47.18 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 17:47.30 | mrtelephone | ntp sends UTC time and the device should figure out the current time based on timezone? right? |
| 17:47.34 | [TK]D-Fender | mrtelephone: Seems you never set it up PROPERLY to account for this years DST change |
| 17:48.02 | coppice | ~fish |
| 17:48.03 | jbot | i guess fish is FISHFISHFISH! DO THE FISH DANCE! "Give a man a fish and you'll feed him a day. Teach him how to fish and he'll feed himself for the rest of his life." This is so appropriate, instead of asking us to tell you exactly what to do, why not read some docs, then come back and ask specific questions which aren't covered?, or ... |
| 17:48.07 | mrtelephone | when I check the phones log files it shows the right time but its not right on the display |
| 17:48.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:49.03 | [TK]D-Fender | mrtelephone: Where in the log.. the file timestamp, or the ENTRIES in it? |
| 17:49.30 | coppice | jbot fish is also "Give a man a fish and you'll feed him for a day. Teach him how to fish and he'll undercut you, and drive you out of business." |
| 17:49.30 | jbot | ACTION slaps is also "Give a man a fish and you'll feed him for a day. Teach him how to fish and he'll undercut you, and drive you out of business." around with a large trout |
| 17:49.36 | mrtelephone | it says got time from ntp server and current time is blah blah.. |
| 17:49.44 | mrtelephone | maybe fixed day has to be enabled? |
| 17:50.23 | thewiizle | lol |
| 17:50.24 | thewiizle | fish |
| 17:50.28 | [TK]D-Fender | mrtelephone: Polycom had a seperate doc explaining this specifically and repeated in the newr admin guides, etc. |
| 17:50.31 | thewiizle | ? |
| 17:50.33 | thewiizle | fish? |
| 17:50.39 | thewiizle | what is fice |
| 17:50.43 | thewiizle | what is fish |
| 17:50.44 | mrtelephone | im starting to wonder if ntptime should change during dst or just flag dst with an extra parameter? |
| 17:50.50 | thewiizle | grr |
| 17:51.43 | mrtelephone | the cisco ata186s don't seem to add an hour during dst :( |
| 17:52.29 | [TK]D-Fender | mrtelephone: each device should know its TZ rule, not as implemented in NTP. |
| 17:52.34 | destructure | what would be a good way to turn off zap without losing timing for meetmes |
| 17:52.45 | destructure | without much configuration change |
| 17:52.49 | [TK]D-Fender | destructure: Clarify "turn off zap" |
| 17:53.07 | destructure | prevent outbound calls via dial(ZAP...) |
| 17:53.36 | _x86_ | destructure: change the signalling method on all the ports |
| 17:53.52 | _x86_ | destructure: if it's FXO, switch it to E&M, etc |
| 17:54.10 | [TK]D-Fender | destructure: Oh, just change your DIALPLAN. |
| 17:54.15 | destructure | it pri |
| 17:54.21 | [TK]D-Fender | DIALPLAN <------- |
| 17:54.42 | [TK]D-Fender | Changing zaptel/zapata means potentially killing CALLS in progress |
| 17:55.16 | Poincare | With the 'sip debug' messages, on received messages, the date field is that the time the sip message was received or when it was sent? |
| 17:55.27 | destructure | the dial is in agi. I guess I can change it there. |
| 17:56.01 | destructure | no existing calls, so no problem. zap is just used to gateway to our legacy stuff, I just don't want to gateway right now |
| 17:56.05 | destructure | it's irritating the cs reps |
| 17:56.06 | destructure | heh |
| 17:56.21 | destructure | but I don't want to make code changes, since it's really a temporary config |
| 17:57.18 | *** join/#asterisk rnovotny22 (n=root@h460dfd16.area2.spcsdns.net) |
| 17:58.14 | [TK]D-Fender | destructure: You should change it in your code as I'm sure you want calls routed through the new appropriate resource. |
| 17:59.47 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 17:59.55 | rnovotny22 | Has anyone been able to get Hudlite working with Asterisk? |
| 17:59.56 | destructure | nah, appropriate resource=/dev/null. I guess the Right Thing would be to make the gateway variable |
| 18:01.34 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 18:02.15 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 18:03.30 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:06.27 | *** part/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 18:09.07 | mrtelephone | yeah I had to enable fixedday dst on the phone |
| 18:09.44 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:09.55 | hmmhesays | I wonder how you're feeling, theres ringing in my ears |
| 18:11.04 | *** join/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 18:13.23 | [TK]D-Fender | hmmhesays: Better influences.... |
| 18:13.58 | mrtelephone | fender do you use 7960s? |
| 18:14.13 | hmmhesays | i've been feeling mellow lately |
| 18:16.38 | [TK]D-Fender | mrtelephone: Nope |
| 18:21.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:24.03 | *** join/#asterisk PepOSX (n=pepOSX@190.72.156.74) |
| 18:29.22 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 18:30.01 | *** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34) |
| 18:30.09 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 18:30.15 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:30.49 | DataCompBoy | Hi All! i have small problem... Upgraded asterisk to 1.4.13, and it won't start now -- no core dump, no errors, just stop load after "WARNING[13282]: pbx.c:2948 ast_register_application: Already have an application 'Directory'" |
| 18:31.04 | DataCompBoy | tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvdddddddddddddddddddddddddddTgfc -- still no info get |
| 18:32.11 | hmmhesays | you didn't clean out your modules directory |
| 18:33.39 | DataCompBoy | hmmhesays: hmm... remove everything and reinstall? |
| 18:34.16 | hmmhesays | from the modules directory yes |
| 18:34.40 | DataCompBoy | hmmhesays: ok, will try. |
| 18:36.40 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 18:36.55 | Shaun2222 | with the polycom phones can the ACD login/logout buttons work? |
| 18:38.06 | DataCompBoy | hmmhesays: still won't work :( and no info... no custom modules tried |
| 18:39.17 | robl^ | Shaun2222: set them as speed dials to login / logout extensions defined in your asterisk |
| 18:40.35 | *** join/#asterisk bigwilson (n=tim@ppp-70-128-184-241.dsl.rcsntx.swbell.net) |
| 18:40.50 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 18:41.03 | rantsh | heloo all |
| 18:41.24 | Shaun2222 | bah.. whats the deal.. why cant the features that exist on these phones work with asterisk... they must send some type of event when you use them, why cant asterisk pick that up and use it. |
| 18:41.32 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 18:42.15 | rantsh | I need help from a queue's guru |
| 18:42.23 | [TK]D-Fender | Shaun2222: You have the source jsut like everyone else. Get to work :p |
| 18:42.35 | bigwilson | Hello everyone |
| 18:43.20 | Shaun2222 | [TK]D-Fender: haha, if i had the skill/time i would :) |
| 18:43.35 | [TK]D-Fender | Shaun2222: Till then, stop whining :) |
| 18:43.41 | DataCompBoy | hmmhesays: zttest works fine (99.96% accuracy [ztdummy]). was absolutely empty modules directory before "aptitude reinstall asterisk". and still same -- slently dead. |
| 18:43.52 | DataCompBoy | How can I obtain core point of problem? |
| 18:43.56 | bigwilson | Anyone know how to import exsiting voicemail messages into mysql databases |
| 18:43.56 | [TK]D-Fender | Shaun2222: And there was a patch on Mantis for this. |
| 18:44.08 | Shaun2222 | [TK]D-Fender: some of these features just seem like they should exist already, polycom is a popular phone in the asterisk world, asterisk been around a while, these are commenly wanted features... |
| 18:44.10 | fetcher | Does Asterisk have a way of logging SIP re-register timeouts, short of turning on the full 'sip debug' packet-by-packet detail? |
| 18:44.24 | Shaun2222 | [TK]D-Fender: i did find somthing but it looked to be abandoned. |
| 18:44.36 | [TK]D-Fender | Shaun2222: Guess you have no idea how hard it is to get anything added to a channel driver around here... |
| 18:44.37 | fetcher | (trying to debug intermittent connectivity trouble at one site, to particular IP phones) |
| 18:46.04 | Shaun2222 | [TK]D-Fender: i dont, i'm not even saying i understand what needs to be done, if it's a polycom problem or a asterisk problem.. just thinking that features like this that are used in normal phone systems would be a priority to get working. |
| 18:46.06 | bigwilson | Anyone know how to import exsiting voicemail messages into mysql databases |
| 18:46.28 | [TK]D-Fender | Shaun2222: its not a problem, its a FEATURE. this is NOT a standard SIP offering. |
| 18:46.39 | [TK]D-Fender | Shaun2222: Thats your first mistake. |
| 18:47.24 | Shaun2222 | [TK]D-Fender: i see, kind of like each phone vendor has there own proprietary method of doing this... nothing set in stone :) |
| 18:47.56 | [TK]D-Fender | Shaun2222: In fact most vendors DON'T have a way of doing this. It was created to accomodate OTHER solutions. |
| 18:50.03 | file | BJ has worked on reverse engineering the ACD login/logout buttons for the Polycom actually |
| 18:50.11 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 18:50.50 | dijungal | hello, why would i have BAD audio quality on asterisk when I get to about 20 simultaneous g.711 sip cals |
| 18:50.52 | dijungal | calls |
| 18:51.06 | dijungal | cpu usage is about 25% |
| 18:51.32 | dijungal | and this is a dual quad core xeon system |
| 18:52.22 | robl^ | bandwidth?? running a GUI on the server? |
| 18:52.42 | destructure | running an smp kernel? |
| 18:52.47 | gvasterisk | any good softphone? eyebeam falls down with dualcore proccesor |
| 18:52.51 | dijungal | no GUI |
| 18:53.17 | dijungal | i am using eyebeam |
| 18:53.35 | dijungal | smp kernel... hmmm.. i'm on fedora core 6 |
| 18:53.46 | destructure | uname -a |
| 18:54.03 | DataCompBoy | hmmhesays: uff! added noload => for app_directory_odbc.so, app_voicemail_imap.so, app_voicemail_odbc.so -- now it started fine :) |
| 18:54.16 | dijungal | 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20 EDT 2006 i686 i686 i386 GNU/Linux |
| 18:54.55 | destructure | looks good. try running top and hit "1". that'll show an individual cpu summary for each at the top |
| 18:55.41 | dijungal | u normally use htop |
| 18:55.45 | dijungal | nicer looking :) |
| 18:55.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:55.59 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 18:56.09 | dijungal | what does this mean: load average: 2.91, 3.15, 2.80 |
| 18:56.11 | dijungal | ? |
| 18:56.40 | dijungal | how do i know if the CPU is being over worked? |
| 18:57.28 | dijungal | any ideas? |
| 18:57.38 | dijungal | i have a 10mb connection to that box |
| 18:57.52 | russellb | what CPU(s) do you have? |
| 18:58.03 | dijungal | xeon |
| 18:58.28 | dijungal | anyone willing to call in and listen to how bad the audio sounds i can give u a number to try |
| 18:59.27 | destructure | as for cpu load, you're already using top so you should see reporting there. is any one cpu getting pegged? |
| 18:59.56 | destructure | load average is the number of processes waiting for cpu, which can be hard to mentally model on smp |
| 19:00.00 | russellb | %CPU usage is a better indicator, really |
| 19:01.19 | destructure | I would try lowering the number of calls, and then artificially load the box (run a benchmark) and see if you can replicate the behavior |
| 19:01.34 | destructure | if you can, it's load, if not, look elsewhere (bad network connection?) |
| 19:02.02 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:03.15 | dijungal | hos do i run a benchmark ? |
| 19:04.06 | tzafrir | 'vmstat 1' may layout stats nicer than top, BTW |
| 19:04.17 | destructure | I'm sure fedora has benchmark packages. find one and run it on the cmd line. you could also write and run a loop in C to load the cpu |
| 19:04.38 | tzafrir | while :; do:; done |
| 19:04.47 | *** join/#asterisk agx (n=badpengu@81-174-8-37.dynamic.ngi.it) |
| 19:04.48 | destructure | or the shell |
| 19:04.48 | destructure | heh |
| 19:05.00 | destructure | an infinite loop runs faster in C though |
| 19:05.00 | destructure | heh |
| 19:05.06 | De_Mon | heh |
| 19:05.16 | [TK]D-Fender | dijungal: You have NOT answered the bandwidth question... |
| 19:05.30 | dijungal | there was one? |
| 19:05.41 | destructure | I thought he said 10 Mbits |
| 19:05.51 | dijungal | yep 10 Mbits |
| 19:05.59 | destructure | is that to the public internet? |
| 19:06.24 | [TK]D-Fender | destructure: "And at warp 10, we're going nowhere mighty fast!" - Scotty |
| 19:06.46 | [TK]D-Fender | destructure: And the clients? What are they on? And conferencing? |
| 19:06.59 | [TK]D-Fender | dijungal: rather |
| 19:07.18 | dijungal | 10 Mbits to service provider |
| 19:07.24 | dijungal | local service provider |
| 19:07.54 | dijungal | but 25 calls at g.711 is about 2 MB of bandwidth |
| 19:08.59 | *** join/#asterisk steven_elvisda (n=Steven_E@202.47.107.60) |
| 19:09.01 | dijungal | i'm not doing any fancy conferencing... etc... it's just incoming calls from a sip provider (which is the same internet provider), going into a queue and distributed amongst agents |
| 19:09.25 | dijungal | i wish there was someway to measure the bandwidth consumption on that box |
| 19:10.02 | dijungal | this really does not sound like the box being overloaded |
| 19:10.17 | *** join/#asterisk Op3r (n=Op3r@121.97.246.229) |
| 19:10.29 | destructure | dijungal: based on tzafrir's suggestion, try running this in a bash shell a few times: ( while true; do echo -n "" ; done ) & |
| 19:10.34 | destructure | and see how many calls you can get to |
| 19:10.53 | destructure | that should raise you cpu load nicely |
| 19:11.00 | destructure | try running one per cpu |
| 19:15.16 | dijungal | lol |
| 19:15.36 | dijungal | that's an endless look |
| 19:15.41 | dijungal | loop |
| 19:15.41 | destructure | yes |
| 19:15.48 | destructure | but it fits your purpose nicely |
| 19:16.11 | dijungal | what about the benchmark test? |
| 19:16.31 | destructure | if you want to go find one, go ahead |
| 19:17.13 | destructure | but the only purpose was to exercise the cpu |
| 19:18.21 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:18.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:18.26 | dijungal | k |
| 19:18.36 | dijungal | how do i end it... kill ? |
| 19:19.30 | dijungal | hah.. ran it.. it's only affecting 1 cpu |
| 19:19.30 | destructure | try running jobs in the shell you spawned them in |
| 19:19.39 | destructure | 1 instance should effect 1 cpu |
| 19:19.46 | *** join/#asterisk fastfeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 19:19.51 | destructure | jobs will show you the instances running |
| 19:19.56 | destructure | kill %1 would kill the first instanec |
| 19:20.36 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:20.56 | *** part/#asterisk fastfeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 19:21.47 | dijungal | hmmm. |
| 19:22.02 | *** join/#asterisk pepo-- (n=pepOSX@190.72.156.74) |
| 19:22.07 | dijungal | what do u know i learn a new command today 'jobs' |
| 19:22.08 | tzafrir | destructure, that also highly depends if asterisk is run with priority -p |
| 19:22.31 | dijungal | priority -p |
| 19:22.33 | dijungal | ok |
| 19:22.40 | Shaun2222 | [TK]D-Fender: what hardware would you recommend for brining a PRI T1 into asterisk? |
| 19:22.41 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:22.51 | dijungal | so instead of running just "asterisk" i should use "asterisk -p" |
| 19:22.52 | tzafrir | with -p, those cpu hogs will have very little effect |
| 19:22.53 | [TK]D-Fender | Shaun2222: Sangoma A101d |
| 19:23.16 | dijungal | ok |
| 19:23.19 | dijungal | i'll remember that |
| 19:23.32 | dijungal | but i really don't think it's the CPU |
| 19:23.43 | Shaun2222 | $1k card... sweet :) got any hookups haha |
| 19:23.46 | destructure | regardless, it's a good way to narrow down the reasons |
| 19:25.21 | _x86_ | anyone know of an asterisk LiveCD? |
| 19:25.38 | [TK]D-Fender | _x86_: No, but I know of a lot that will kill your system :p |
| 19:25.58 | Shaun2222 | _x86_: http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM |
| 19:26.37 | _x86_ | i've got a digium TDM2400P with an AMP connector, and I was going to replace that whole server with an HP DL380 G5 with a Sangoma A102D-x.... turns out i'm missing a channel bank, and the TDM2400P wont fit in the server (slots are PCIe in the server, Digium card is PCI) |
| 19:27.15 | _x86_ | i'm miles from my warehouse, which has an FXS channel bank as it turns out... |
| 19:28.11 | rantsh | anyone knows why may asterisk not be able to use MixMonitor |
| 19:28.29 | Shaun2222 | rantsh: whats the error your seeing. |
| 19:28.51 | rantsh | it's producing 2 separate files (in and out) |
| 19:29.50 | rantsh | there's no error in my CLI |
| 19:32.48 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 19:33.48 | gvasterisk | is there any good solution with nat? cause I can register from outside but I can't receive sip calls from a trunk |
| 19:33.50 | gvasterisk | ? |
| 19:34.00 | [TK]D-Fender | gvasterisk: Read up : |
| 19:34.02 | [TK]D-Fender | ~sipnat |
| 19:34.12 | 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 |
| 19:34.13 | _x86_ | gvasterisk: use IAX |
| 19:34.50 | gvasterisk | IAX? but I'm calling from outside, my SIP provider doesn't use IAX |
| 19:34.54 | gvasterisk | or yes? |
| 19:35.06 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 19:35.07 | [TK]D-Fender | gvasterisk: Go. Read. NOW. |
| 19:38.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:40.03 | *** join/#asterisk Aeudian (n=somewher@75.148.21.113) |
| 19:40.22 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 19:41.12 | *** join/#asterisk |Rain| (i=rain@blackhole.themuffin.net) |
| 19:41.45 | |Rain| | how can I delete a global variable from the dialplan? Set(GLOBAL(foo)=); doesn't seem to work |
| 19:43.18 | bigwilson | Anyone know how to import exsiting voicemail messages into mysql databases |
| 19:43.53 | Aeudian | [TK]D-Fender: maybe my syntex is wrong for this applciation but the direction dialing is still speaking the letters when there is a greet.*** My syntex under extensions is exten = 9,2,Directory(iveia-voicemail,iveia-dial-by-name) where iveia-dial-by-name is located in extensions whichs tats exten = 301,1,Dial(SIP/301) and where iveia-voicemail under voicemail.conf has 301 => 301,John Smith. Asterisk then speaks J,O,H,N,Space,S,M,I,T,H |
| 19:44.08 | Aeudian | rather then reading the mailbox folder 301 and look for greet.*** |
| 19:44.54 | [TK]D-Fender | Aeudian: please provide a comprehensive pastebin of your configs and folders, as well as CLI output of the failed attempt./ |
| 19:45.53 | ThoMe | what is |
| 19:45.53 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 19:45.53 | ThoMe | P[ 1] jb_fill overflow:-1 |
| 19:45.54 | ThoMe | ? |
| 19:45.56 | Aeudian | tk, will do 1 second |
| 19:49.28 | Aeudian | [TK]D-Fender: http://pastebin.com/d29d1ea86 |
| 19:50.16 | [TK]D-Fender | .... |
| 19:50.20 | [TK]D-Fender | Aeudian: And the rest? |
| 19:51.49 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 19:52.12 | gvasterisk | where does asterisk saves announcements for IVR? |
| 19:53.09 | Aeudian | gvasterisk: have you looked in /var/lib/asterisk/sounds ? |
| 19:53.54 | [TK]D-Fender | gvasterisk: And doesnt' save anything ANYWHERE. When you use Record, you choose where you want to recording things to. |
| 19:54.10 | [TK]D-Fender | Aeudian: Careful o fthe questions you're answering... |
| 19:55.13 | *** join/#asterisk bucketfan99 (i=bucket@S010600010301ffb9.vc.shawcable.net) |
| 19:56.47 | gvasterisk | ok :D |
| 19:59.38 | dijungal | how do i enable echo canceling on sip channels ? |
| 20:00.41 | file | dijungal: it is done on the device, Asterisk does not do it |
| 20:00.48 | dijungal | k |
| 20:00.50 | rantsh | I just noticed my problem is that my queues.conf file is (for some reason) not accepting this line "monitor-type = MixMonitor" |
| 20:01.27 | file | rantsh: what version of Asterisk? |
| 20:01.44 | rantsh | 1.2.24 |
| 20:02.02 | dijungal | no built in echo canceling |
| 20:02.05 | dijungal | in asterisk.. |
| 20:02.06 | dijungal | hmmm |
| 20:02.07 | dijungal | k |
| 20:02.13 | file | monitor-type is not valid for 1.2 |
| 20:02.56 | file | dijungal: not for SIP devices, echo cancellation exists in Zaptel for Zap hardware though |
| 20:03.37 | *** join/#asterisk bucketfan99 (i=bucket@S010600010301ffb9.vc.shawcable.net) |
| 20:03.48 | bucketfan99 | hey anyone here ever have problems with faxes coming in on * ? |
| 20:04.15 | file | bucketfan99: without more information the answer is "yes, no, sometimes" |
| 20:04.32 | bucketfan99 | yeah, i was kind of looking for a quick poll. |
| 20:04.44 | bucketfan99 | i was going to set one up, buddy said, he always had problems with faxes on *.. so i thought id ask you guys |
| 20:04.45 | dijungal | i have a TE410P how do i enable echo canceling on it? |
| 20:05.15 | file | bucketfan99: well it depends on how faxes are coming in and what they are going over... |
| 20:05.27 | [TK]D-Fender | dijungal: "echocancel=yes" before your channel declaration in zapata.conf |
| 20:07.03 | dijungal | thnks |
| 20:09.17 | dijungal | ahhh... i already have echocancel=yes and echotraining=yes |
| 20:09.26 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:09.47 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 20:09.58 | l2trace99 | anyone know if there is a way of triggering an event when a sip device registers ? |
| 20:09.58 | *** part/#asterisk dalbaech (n=dalbaech@darkvoip.net) |
| 20:15.06 | [TK]D-Fender | l2trace99: Go read up on AMI to see if "regexten" can be seen. |
| 20:18.10 | *** join/#asterisk D_Asterisk (n=mail@82-136-226-200.ip.tiscali.nl) |
| 20:18.19 | D_Asterisk | Hello all! |
| 20:18.34 | l2trace99 | anyway besides the manager interface |
| 20:18.37 | D_Asterisk | may i ask a question please?? |
| 20:18.41 | l2trace99 | ? |
| 20:18.42 | *** join/#asterisk ManxPower (n=manxpowe@143.sub-70-220-220.myvzw.com) |
| 20:19.23 | D_Asterisk | I have a strange problem: Sometimes (Approximetely 2 times a day) Calls are dropped unexpectedly |
| 20:19.34 | D_Asterisk | error message in my asterisk box: |
| 20:19.45 | D_Asterisk | Sep 17 09:16:37 WARNING[30001] chan_sip.c: Maximum retries exceeded on transmission B8728175-642411DC-BDF2CB3C-616AECB7@83.98.222.254 for seqno 200 (Critical Response) |
| 20:19.45 | D_Asterisk | Sep 17 09:16:37 WARNING[30001] chan_sip.c: Hanging up call |
| 20:20.03 | D_Asterisk | Ports 10000-20000 and port 5060 are open. |
| 20:20.39 | D_Asterisk | oh, i'm using asterisk 1.2.22 |
| 20:22.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:28.55 | rantsh | file: I got it from the sample it created :s But I was kind of guessing that |
| 20:29.11 | asdx | ~book |
| 20:29.12 | 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 |
| 20:29.36 | file | rantsh: I checked the sample queues file and it is not there |
| 20:30.03 | dijungal | i wonder what edition i have |
| 20:30.06 | *** join/#asterisk yoshiznit123 (n=sciyoshi@142.157.233.50) |
| 20:30.07 | rantsh | btw, how can one know what to find through jbot? |
| 20:30.56 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@68.90.41.25) |
| 20:32.33 | _x86_ | oh my god... WHY does sangoma have to use non-standard "RJ9" plugs for their FXO/FXS cards?! |
| 20:32.53 | tzanger | RJ9? |
| 20:33.02 | _x86_ | whatever it is... looks like RJ9 |
| 20:33.09 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 20:33.10 | _x86_ | the plug that a handset uses to connect to a phone |
| 20:33.11 | tzanger | 4-wire "mini" modulear plug? |
| 20:33.15 | tzanger | you're kidding me |
| 20:33.21 | _x86_ | tzanger: more mini than an RJ11 heh |
| 20:33.29 | tzanger | I'm gonna phone david mandelstam and ask him wtf |
| 20:33.47 | _x86_ | yeah man... this sucks because i dont have any of those plugs (although i do have a crimper that is capable of crimping them) |
| 20:33.57 | _x86_ | tzanger: please do! |
| 20:34.28 | _x86_ | tell him to use standard RJ11 like everyone else |
| 20:35.03 | _x86_ | Digium figured out how to put 8 RJ11's on a single card.... why can't sangoma figure out how to put 4 RJ11's on theirs? |
| 20:35.31 | Qwell | _x86_: we wondered the same thing |
| 20:35.36 | fetcher | oh, there's only 4 jacks? I was thinking perhaps the RJ9 was to help squeeze more connectors onto the slot backplate |
| 20:35.40 | D_Asterisk | does anybody knows how to solve the call drop problem in asterisk |
| 20:35.53 | tzanger | I have an old octal serial board that uses 8 RJ11s on the backplane |
| 20:35.56 | tzanger | it's not rocket science |
| 20:36.08 | Qwell | _x86_: I told you about the card that used an svga plug, right? |
| 20:36.14 | D_Asterisk | i've found a lot of solutions on google but they don't work for me ... |
| 20:36.16 | Qwell | or was going to, or whatever |
| 20:36.23 | fetcher | D_Asterisk: possibly network issues. What's between you and the other SIP peer? |
| 20:36.31 | _x86_ | Qwell: haha yeah i think you did.... i think this whole conversation is de ja vu ;) |
| 20:36.33 | yoshiznit123 | where can i find more info about the new Bridge app in svn? (sorry new to asterisk :-)) |
| 20:36.38 | Qwell | indeed |
| 20:36.54 | D_Asterisk | just a normal router |
| 20:37.02 | _x86_ | gah... gonna have to stop at a rat shack and get some RJ11 couplers then... since I doubt they have RJ9 ends |
| 20:37.07 | D_Asterisk | alcatel Speedtouch 780WL |
| 20:37.12 | Qwell | heh |
| 20:37.17 | Qwell | _x86_: you know what's funny as hell about that? |
| 20:37.20 | D_Asterisk | i forwarded all the ports to my asterisk server |
| 20:37.24 | D_Asterisk | TCP and UDP |
| 20:37.35 | Qwell | I was *JUST* at Radio Shack - LAST NIGHT, looking for RJ10 ends/jacks |
| 20:37.57 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 20:38.04 | _x86_ | Qwell: am i talking about RJ9 or RJ10? i can't remember what the hell they are :) |
| 20:38.18 | Qwell | AFAIK, it's the same :D |
| 20:38.23 | Qwell | like RJ11/RJ12 |
| 20:38.32 | _x86_ | 4/6 wire |
| 20:38.35 | fetcher | D_Asterisk: I mean, is the other end a device in your building? Or an ITSP out on the Internet somewhere? |
| 20:38.43 | _x86_ | same physical plug/jack tho |
| 20:38.43 | rantsh | file: I just figured at some point I downgraded my asterisk from 1.4.x to 1.2.24, that's how I got the wrong file |
| 20:38.43 | Qwell | but the jacks are the same |
| 20:38.47 | _x86_ | yep |
| 20:38.55 | Qwell | RJ9 might just use 2 wires or something |
| 20:39.00 | D_Asterisk | the other end is my voip provider |
| 20:39.10 | Qwell | RJ10 is safe to say, because you probably do need 4 wires. |
| 20:39.10 | _x86_ | Qwell: that's all I need anyway |
| 20:39.15 | _x86_ | nah |
| 20:39.17 | D_Asterisk | may i pm you fetcher |
| 20:39.20 | _x86_ | these are CO lines |
| 20:39.21 | Qwell | oh, right, breakout crap |
| 20:39.24 | mrtelephone | I got a problem with asterisk ignoring register requests |
| 20:39.32 | rantsh | file: but Set(MONITOR_FILENAME=foo) which is in the right sample ain't working either |
| 20:39.43 | mrtelephone | not putting in the IP address properly or something |
| 20:40.07 | fetcher | D_Asterisk: the network you're on might just have periods of poor connectivity to that site. VoIP over the public Internet can be hit & miss. Try another VoIP upstream, if possible |
| 20:41.05 | D_Asterisk | okay i will do that |
| 20:41.05 | fetcher | D_Asterisk: preferably one on the same major backbone as your ISP... traceroutes can be helpful |
| 20:42.32 | D_Asterisk | Fetcher: i think you're right |
| 20:42.54 | D_Asterisk | my upstream: 768kb |
| 20:43.23 | D_Asterisk | i have 6 Grandstream GXP-2000 phones and 2 fax machines |
| 20:43.32 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:43.48 | D_Asterisk | problem occurs only when someone is calling us |
| 20:44.11 | D_Asterisk | so only incoming calls from outsite... |
| 20:44.45 | rantsh | ~ |
| 20:44.51 | rantsh | ~help |
| 20:45.23 | D_Asterisk | sometimes, everything is working great for several days... |
| 20:45.34 | D_Asterisk | Thanks for your help Fetcher ! |
| 20:48.04 | rantsh | are there instructions on how to use jbot? |
| 20:48.13 | Qwell | ~instructions |
| 20:48.18 | Qwell | nope |
| 20:48.29 | Qwell | rantsh: msg him the word help |
| 20:49.10 | mrtelephone | wow |
| 20:49.14 | mrtelephone | my asterisk box is crazy busy |
| 20:50.04 | rantsh | thanks Qwell, I did but really not understand much, sorry for the n00bnes |
| 20:50.38 | dijungal | ~instructions |
| 20:51.11 | GreggB | ~jbot |
| 20:51.11 | jbot | [jbot] a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 20:51.32 | [TK]D-Fender | :D |
| 20:52.02 | *** join/#asterisk hexorx (n=joshr@63-211-239-34.teliax.com) |
| 20:52.20 | mrtelephone | fembots |
| 20:52.25 | rantsh | haha |
| 20:52.39 | D_Asterisk | asterisk is a great VOIP telephony engine |
| 20:52.48 | *** join/#asterisk toomba (n=hola@do.you.like.my.frippers.com) |
| 20:53.17 | D_Asterisk | my only problem is that some calls a dropped several times a week |
| 20:53.50 | mrtelephone | too bad you couldn't do sip debug on a peername without an ip address |
| 20:53.51 | mrtelephone | :( |
| 20:54.13 | D_Asterisk | how do you mean mytelephone ? |
| 20:54.33 | D_Asterisk | Sep 17 09:16:37 WARNING[30001] chan_sip.c: Maximum retries exceeded on transmission B8728175-642411DC-BDF2CB3C-616AECB7@83.98.222.254 for seqno 200 (Critical Response) |
| 20:54.42 | [TK]D-Fender | mrtelephone: "ship debug peer [peername}" <------ |
| 20:54.53 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:54.57 | D_Asterisk | oh i didn't know that ... |
| 20:55.10 | mrtelephone | fender, if the peer isn't registered it won't work |
| 20:55.23 | *** join/#asterisk techie (n=techie@adsl-76-214-12-127.dsl.lsan03.sbcglobal.net) |
| 20:55.27 | *** part/#asterisk techie (n=techie@adsl-76-214-12-127.dsl.lsan03.sbcglobal.net) |
| 20:55.42 | mrtelephone | if you do sip debug I want it to show messages with certain words I guess.. |
| 20:55.56 | mrtelephone | say you see a peer trying to register but it isn't.. |
| 20:56.07 | [TK]D-Fender | mrtelephone: and if the peer isn't registered you should be doing a general debug to find out why |
| 20:56.17 | mrtelephone | what is your suggestion then |
| 20:56.25 | mrtelephone | one peer is saying nonce is stale.. another one I'm not sure of |
| 20:56.39 | mrtelephone | I had a local polycom 501 not register to my asterisk box even after a reset |
| 20:56.49 | mrtelephone | there is something buggy going on |
| 20:58.28 | *** join/#asterisk blq (n=Bl@dslb-088-064-128-247.pools.arcor-ip.net) |
| 20:59.32 | [TK]D-Fender | mrtelephone: perhaps if you just enabled global sip debug and a PASTEBIN we could actually have something to comment on... |
| 20:59.52 | mrtelephone | I'm getting 401 Unauthorized |
| 21:00.03 | [TK]D-Fender | mrtelephone: Good. Now go fix your auth |
| 21:00.25 | ManxPower | one would think you would get an actual error message. |
| 21:00.43 | mrtelephone | Nov 5 15:57:28 NOTICE[12894]: chan_sip.c:6532 check_auth: stale nonce received from |
| 21:00.50 | mrtelephone | the client is an ata186 |
| 21:01.43 | [TK]D-Fender | mrtelephone: What are we talking about here, an IP 501 or an ATA 186? |
| 21:02.49 | mrtelephone | right now Im trying to figure out the ata186, why its not registering.. but the same was happening with one of my ip 501 phones.. the wierd thing is that I could dial out with the 501 but yet it wasn't registering properly.. |
| 21:03.02 | ManxPower | mrtelephone: that is not weird AT ALL. |
| 21:03.30 | ManxPower | ALL registration does is tell the server what ip address is associated with which SIP account/password. It does nothing else. |
| 21:03.54 | mrtelephone | I know thats why I think there is a bug but I can't restart because there is a bunch of calls in progress |
| 21:03.58 | ManxPower | since the server doesn't have to know your IP address to ACCEPT a call from that device, calling out would work just fine. |
| 21:04.27 | ManxPower | mrtelephone: what version of Asterisk are you using? |
| 21:04.54 | mrtelephone | 1.2.21.1 |
| 21:05.12 | ManxPower | If you are going to use 1.2.x, you could at least use the latest one. |
| 21:06.02 | mrtelephone | i was going to but I had some trouble with the ncs patch |
| 21:06.11 | mrtelephone | im migrating everything to sip |
| 21:06.20 | ManxPower | to sip from what? |
| 21:06.25 | mrtelephone | MGCP/NCS |
| 21:06.38 | ManxPower | Ah. Good luck with that. |
| 21:06.42 | mrtelephone | hah |
| 21:06.45 | mrtelephone | yeah tell me about it |
| 21:06.53 | mrtelephone | arris cable modems with sip firmware now |
| 21:06.57 | mrtelephone | very easy to setup |
| 21:07.36 | mrtelephone | the company has really good support |
| 21:08.00 | Kobaz | hmm, is there anything special you need to do to get a polycom 320 going... i have 501's going fine |
| 21:08.32 | ManxPower | Kobaz: 2.x sip.cfg and phone1.cfg should be ahout it. |
| 21:08.38 | ManxPower | and the firmware, of course. |
| 21:08.44 | Kobaz | mmm, the firmware |
| 21:09.29 | ManxPower | the 320 would have shipped with the correct firmware, but not with the correct sip.cfg and phone1.cfg. If you use older versions of those files the volume on the phone will be low and the icons next to the buttons will be wrong. |
| 21:10.12 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 21:10.26 | kraptv | Has anyone ever experienced Digium T1 hardware failing? |
| 21:10.42 | *** join/#asterisk X-Scale (n=none@89.181.21.212) |
| 21:10.53 | mrtelephone | kraptv, never heard of it |
| 21:11.01 | mrtelephone | but you should have a spare card around |
| 21:11.03 | ManxPower | http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_2_1_2_release_sig.zip |
| 21:11.08 | ManxPower | no registration required. |
| 21:11.28 | ManxPower | found on http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
| 21:12.02 | Kobaz | kraptv: yeah, we had three fry |
| 21:12.06 | ManxPower | kraptv: Yes, but not very often |
| 21:12.09 | X-Scale | hi...i'm trying to make/install asterisk on archlinux...it all goes well until it tries to make editline...it crashes saying: "makelist: line 147: /usr/bin/awk: No such file or directory" Any hints how to solve this issue ? |
| 21:12.18 | kraptv | I do have a spare... garf. |
| 21:12.19 | Kobaz | kraptv: we switched to rhino and havent had problems since |
| 21:12.23 | mrtelephone | hwo the hell did you fry 3 t1 cards |
| 21:12.24 | ManxPower | X-Scale: install awk |
| 21:12.33 | [TK]D-Fender | X-Scale: I dunno... install AWK maybe? |
| 21:12.34 | X-Scale | it is installed by default |
| 21:12.35 | Kobaz | mrtelephone: not sure, they decided to not work one day |
| 21:12.45 | mrtelephone | induction maybe |
| 21:12.49 | mrtelephone | long runs? |
| 21:12.58 | Kobaz | not really |
| 21:13.14 | mrtelephone | Im planning on frying one cuz I got a line nextdoor that hooks up to a channel bank on a different hydro service |
| 21:13.25 | mrtelephone | i guess in network runs thats a bad thing |
| 21:13.39 | ManxPower | X-Scale: what is version listed in the output of "/usr/bin/awk --version" |
| 21:14.18 | X-Scale | ManxPower: awk is located on /bin/awk |
| 21:14.19 | X-Scale | GNU Awk 3.1.5 |
| 21:14.21 | ManxPower | you connect ethernet to a channelbank? |
| 21:14.30 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583145.dsl.bell.ca) |
| 21:14.41 | ManxPower | X-Scale: then it's not located where Asterisk expects it to be. symlink it and yell at your distro |
| 21:15.02 | X-Scale | i did it...then a whole bunch of errors appear |
| 21:15.08 | mrtelephone | nah i run cat 5 to a channel bank |
| 21:15.27 | *** part/#asterisk yoshiznit123 (n=sciyoshi@142.157.233.50) |
| 21:15.27 | ManxPower | mrtelephone: so you are running a T-1 to the channel bank, not a network to a channel bank. |
| 21:15.41 | mrtelephone | no but who says there can't be a surge on the line that fries the t1 card? |
| 21:16.10 | ManxPower | mrtelephone: a surge and a groundloop are different. one would hope a groundloop was handled by the telecom devices. |
| 21:16.23 | mrtelephone | i hope so |
| 21:16.36 | mrtelephone | adit 600? |
| 21:16.37 | ManxPower | The smartjack is connected to different mains power. |
| 21:16.58 | ManxPower | so having different mains circuits at the two ends of the circuit should not be a problem. |
| 21:17.37 | ManxPower | I *HAVE* had induction issues with long runs of ethernet underground. |
| 21:17.52 | mrtelephone | i've never had problems with any of the t1 circuits yet.. if everything ran off of t1 asterisk would be flawless |
| 21:17.53 | ManxPower | but never POTS lines in the same conduit. |
| 21:18.19 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583145.dsl.bell.ca) |
| 21:18.41 | mrtelephone | don't mix pots and ethernet? |
| 21:19.13 | *** join/#asterisk tehfox (n=tehfox@158.193.95.101) |
| 21:19.59 | ManxPower | not mixing pots and ethernet in the same conduit is -- difficult. |
| 21:20.33 | ManxPower | The point is not that pots blows ethernet ports, the point is that telecom is much more resistant to electrical issues. |
| 21:22.31 | X-Scale | ManxPower: i'm getting this oddity |
| 21:22.31 | X-Scale | [root@myhost asterisk-1.4.13]# ls -l /usr/bin/awk |
| 21:22.31 | X-Scale | lrwxrwxrwx 2 root root 4 Sep 17 16:03 /usr/bin/awk -> gawk |
| 21:22.31 | X-Scale | [root@myhost asterisk-1.4.13]# /usr/bin/awk |
| 21:22.31 | X-Scale | -bash: /usr/bin/awk: No such file or directory |
| 21:23.10 | ManxPower | X-Scale: rm /usr/bin/awk && ln -s /bin/awk /usr/bin/awk |
| 21:23.34 | ManxPower | X-Scale: and if you flood the channel again instead of using pastebin, Bad Things Will Happen |
| 21:24.22 | |Rain| | sigh. |
| 21:24.22 | *** part/#asterisk |Rain| (i=rain@blackhole.themuffin.net) |
| 21:25.08 | X-Scale | sorry ManxPower |
| 21:25.28 | ManxPower | Oh, and how do you want to pay? I do Asterisk help for free, but I charge for Linux help. |
| 21:25.38 | ManxPower | perhaps someone on #linux will help you for free. |
| 21:25.42 | X-Scale | http://pastebin.com/d5837f339 |
| 21:26.26 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 21:26.46 | ManxPower | X-Scale: looks to me like a kernel header issue. |
| 21:27.21 | ManxPower | What version of Asterisk are you using? |
| 21:27.34 | X-Scale | It compiled so many files without any problem |
| 21:27.46 | X-Scale | the lastest...1.4.13 |
| 21:28.00 | ManxPower | Ah, I see it now. |
| 21:28.16 | X-Scale | this editline lib ported from netbsd is not working |
| 21:28.29 | ManxPower | I suspect some of the header files were installed in places Asterisk does not see them. If awk was installed in the wrong place, I'm sure other stuff is too. |
| 21:29.13 | [TK]D-Fender | ok, time to head home. Later all |
| 21:29.32 | *** join/#asterisk macros73 (n=cs@dsl093-063-236.pit1.dsl.speakeasy.net) |
| 21:29.46 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 21:32.29 | *** join/#asterisk Skarmeth (n=Skarmeth@201009082245.user.veloxzone.com.br) |
| 21:35.04 | *** part/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 21:36.07 | wwalker | if, from and AGI script, I have a call join a meetme conference room, how do I get the DTMF seen by meetme instead of the AGI until they hang up (I need to write specialized CDRs at call end using state data that's in the running AGI) |
| 21:36.38 | wwalker | a/and/an/ |
| 21:40.51 | mrtelephone | asterisk should tell the client to use ulaw if the call is directed through voicemail/zap |
| 21:41.08 | mrtelephone | but it seems if you set g729 to priority 1 it will use it even though ulaw is available |
| 21:41.34 | mrtelephone | is there another setting to use g729 when feasible to do so? |
| 21:43.26 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 21:44.51 | Katty | anyone around that works for digium? |
| 21:46.03 | mrtelephone | russellb |
| 21:47.24 | _x86_ | i remember in asterisk 1.2.x, you could do 'zap show channels' to see the available zap channels |
| 21:47.31 | _x86_ | what's the equivelant in 1.4.x? |
| 21:48.58 | wwalker | zap show channels shows the channels. core show channels shows which channels are in use |
| 21:49.16 | _x86_ | what does it mean if zap show channels does not work? |
| 21:49.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583145.dsl.bell.ca) |
| 21:49.35 | mrtelephone | asterisk wasn't compiled with zaptel support |
| 21:49.36 | _x86_ | *CLI> zap show channels |
| 21:49.36 | _x86_ | No such command 'zap show' (type 'help' for help) |
| 21:49.45 | _x86_ | sure it was |
| 21:49.55 | Maliuta | so you don't have something installed |
| 21:49.56 | mrtelephone | check if the module is loaded |
| 21:50.05 | _x86_ | *CLI> show modules like chan_zap.so |
| 21:50.05 | _x86_ | Module Description Use Count |
| 21:50.09 | _x86_ | chan_zap.so Zapata Telephony 0 |
| 21:50.10 | Maliuta | rivne*CLI> zap show ch |
| 21:50.11 | Maliuta | channels channel |
| 21:50.11 | Maliuta | rivne*CLI> zap show channels |
| 21:50.11 | Maliuta | <PROTECTED> |
| 21:50.11 | Maliuta | <PROTECTED> |
| 21:50.12 | _x86_ | 1 modules loaded |
| 21:50.13 | rantsh | anyone knows if asterisk 1.2.24 has problems setting ${MONITOR_FILENAME} |
| 21:50.13 | Maliuta | <PROTECTED> |
| 21:50.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:50.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:50.15 | Maliuta | <PROTECTED> |
| 21:50.27 | Maliuta | works for me |
| 21:50.32 | _x86_ | hmm |
| 21:50.46 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:50.53 | Maliuta | _x86_: you have something broken |
| 21:51.06 | _x86_ | that's what i'm trying to figure out ;) |
| 21:51.15 | _x86_ | WHATs broken :P |
| 21:51.19 | Maliuta | _x86_: sure your zaptel drivers are installed properly? |
| 21:51.30 | _x86_ | they are... ztcfg doesn't complain |
| 21:51.35 | *** join/#asterisk agx (n=badpengu@81-174-8-37.dynamic.ngi.it) |
| 21:51.37 | Maliuta | and that the versions are compatible with your * versions |
| 21:51.38 | _x86_ | zttool shows both my T1's |
| 21:52.45 | mrtelephone | are they configured in zapata.conf? |
| 21:52.50 | _x86_ | yes |
| 21:53.14 | mrtelephone | i dunno have problems with console commands if asterisk can't resolve your own hostname |
| 21:53.24 | mrtelephone | does your machines host name resolve? |
| 21:53.39 | _x86_ | yep |
| 21:53.45 | Maliuta | forward or reverse? |
| 21:53.48 | _x86_ | both |
| 21:54.05 | Maliuta | what version of zaptel? |
| 21:54.29 | _x86_ | # host `uname -n` |
| 21:54.30 | _x86_ | rpc-pbx-urb-01.royalpublishing.com has address 10.46.27.252 |
| 21:54.36 | _x86_ | # host 10.46.27.252 |
| 21:54.36 | _x86_ | 252.27.46.10.in-addr.arpa domain name pointer rpc-pbx-urb-01.royalpublishing.com. |
| 21:55.38 | _x86_ | ah |
| 21:55.40 | _x86_ | i figured it out |
| 21:55.45 | mrtelephone | what |
| 21:55.52 | mrtelephone | waht was it |
| 21:55.58 | *** part/#asterisk agx (n=badpengu@81-174-8-37.dynamic.ngi.it) |
| 21:56.02 | _x86_ | i had removed a span that i was previously using, but not removed it from zaptel.conf/zapata.conf |
| 21:56.07 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:56.19 | _x86_ | i thought zaptel would just skip that span.... not completely bitch out |
| 21:56.20 | mrtelephone | oh |
| 21:56.23 | mrtelephone | hahah |
| 21:56.24 | mrtelephone | yeah |
| 21:56.27 | Maliuta | ahh! the old genius gene issue ;) |
| 21:56.32 | mrtelephone | must crash the module loading process |
| 21:56.38 | _x86_ | pfft |
| 21:56.39 | _x86_ | lame |
| 21:56.45 | mrtelephone | if you don't like it program it yourself |
| 21:56.47 | _x86_ | but why did CLI show me chan_zap.so was loaded? |
| 21:56.57 | Maliuta | I get the "genius gene" from time to time |
| 21:57.02 | mrtelephone | if (using span that doesn't exist) { skip and load anyways casuing the system to catch on fire } |
| 21:57.17 | mrtelephone | not sure |
| 21:57.21 | mrtelephone | it was probably loaded |
| 21:57.30 | mrtelephone | but it didn't get to load the commands into console |
| 21:58.12 | _x86_ | ugh, wish i could goto sleep... and it's only 3:57pm here ;) |
| 21:58.18 | mrtelephone | :P |
| 22:00.07 | robl^ | what does time have to do with sleep? power napping for the win! |
| 22:00.54 | mrtelephone | whats this in 0x hex form ? 00000000 00000000 00000001 000000101 |
| 22:00.57 | wwalker | I'm running an AGI from the dial plan. The AGI can accept DTMF and is receiving it well. the AGI exits and we drop into meetme() but meetme doesn't receive any of my DTMF....??? any ideas? |
| 22:01.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:01.43 | peanut- | granstream the lowest order of ip phones? |
| 22:02.20 | [TK]D-Fender | peanut-, No, but plenty low enough |
| 22:02.53 | peanut- | still pretty pricy |
| 22:03.16 | JT | granDstream is pretty low, but there are lower |
| 22:03.20 | JT | peanut-: stingey much? |
| 22:03.34 | mrtelephone | does asterisk still decode g729 if the rtp goes through the asterisk box but both clients are g729 enabled? |
| 22:04.12 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:04.26 | peanut- | JT: yes. |
| 22:04.51 | wwalker | peanut-: they are definitely the lowest order, no matter how much they cost. |
| 22:05.39 | rantsh | mrtelephone: I believe it does bridge the calls, don't thnk it decodes/encodes anything though |
| 22:06.14 | [TK]D-Fender | mrtelephone, it only decodes if it has a reason to decode |
| 22:07.12 | JT | wwalker: you can get worse crap from lesser known manufacturers |
| 22:07.33 | mrtelephone | can you tell asterisk to use ulaw only for internal functions such as voicemail? |
| 22:07.36 | rantsh | is there any place I can download the original sample files for a specific asterisk version? |
| 22:08.04 | [TK]D-Fender | mrtelephone, no. |
| 22:08.12 | hmmhesays | func_odbc is driving me insane at the moment |
| 22:08.20 | mrtelephone | i'll biab |
| 22:08.21 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 22:09.46 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-235-245.pppoe.mtu-net.ru) |
| 22:10.07 | hmmhesays | is there a verbose or debug level that will show you the actual query func_odbc is trying to execute? |
| 22:12.00 | ManxPower | The more I look at the 1.4 branch SVN commits, the less I'm interested in 1.4. |
| 22:13.00 | ManxPower | assigning values to variables in the dialplan was not thread-safe |
| 22:16.13 | *** join/#asterisk Anthro (n=kzjdfhfv@pdpc/supporter/active/Anthro) |
| 22:16.27 | *** part/#asterisk Anthro (n=kzjdfhfv@pdpc/supporter/active/Anthro) |
| 22:19.56 | peanut- | in that case, anyone have somen crappy grandstreams they wanna offload? |
| 22:20.15 | asdx | can you recommend me a good dedicated server? |
| 22:20.22 | hmmhesays | I can't freaking figure this out |
| 22:20.34 | ManxPower | peanut-: they don't usually work after the person throws them against the wall in frustration |
| 22:20.56 | ManxPower | asdx: I suggest one of the Intel reference server boards. |
| 22:21.31 | [TK]D-Fender | asdx, www.ibm.com |
| 22:22.00 | asdx | yeah, thanks |
| 22:23.18 | wwalker | Anyone use meetme? does every DTMF command in the meetme menu always require pressing * first? (* 4 to decrease volume then * 6 to increase volume, etc...)??? |
| 22:24.45 | ManxPower | wwalker: no. * brings you into the menu, then the number picks the option. |
| 22:25.07 | ManxPower | Is that a problem for you? |
| 22:31.11 | X-Scale | ManxPower: that symbolic link solved it...thanks :) |
| 22:33.21 | wwalker | ManxPower: but I can only enter 1 command after each *. so If I want to lower the volume 3 clicks it is * 4 wait * 4 wait * 4 |
| 22:35.11 | wwalker | or can I enter commands after * until I hit 8? |
| 22:38.56 | *** join/#asterisk jmacz (n=jmacz@201.244.175.134) |
| 22:40.26 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 22:40.26 | _x86_ | [TK]D-Fender: wait... there are phones lower on the totem pole than grandstream?! |
| 22:41.11 | [hC] | anyone else notice that their polycom did not seem to change after daylight savings? Ive checked my config and they're up to date with the correct 'stop' date/time, but people claim the time is still wrong. |
| 22:42.13 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:42.17 | file | creepy. |
| 22:42.21 | [hC] | hrmm. |
| 22:42.43 | file | according to TK Polycom put out a document detailing the changes needed for DST... perhaps you should find it |
| 22:43.06 | [hC] | Ive already followed it. Im looking at the config options and they are up to date |
| 22:43.28 | [hC] | Its told to stop on Nov 4, at 2 oclock. yet, the time is wrong. |
| 22:43.38 | [hC] | I'll have to load up this clients config and see if i can make it happen to me. |
| 22:46.36 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:46.56 | *** part/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 22:54.48 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 23:01.32 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 23:02.34 | *** part/#asterisk pepo-- (n=pepOSX@190.72.156.74) |
| 23:10.24 | syzygyBSD | does anyone have an example dhcpd.conf file that works with both polycom 501 and 601's for ftp provisioning |
| 23:16.15 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 23:16.15 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 23:19.20 | [TK]D-Fender | syzygyBSD, http://pastebin.com/m25447c4b |
| 23:20.42 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 23:20.51 | syzygyBSD | thanks [TK]D-Fender you rock, like always |
| 23:21.16 | syzygyBSD | trying to migrate from tftp for security and cuz 601 doesn't like what worked before... |
| 23:23.16 | [hC] | that looks like config for tftp, not ftp? |
| 23:23.47 | [TK]D-Fender | [hC], Polycom's pick up the server address via Opt 66 and the TYPE depends on what you set into the BootROM. |
| 23:24.13 | [hC] | [TK]D-Fender: so, with what you're saying you have to manually select FTP yourself, in the phone menu? |
| 23:24.22 | [TK]D-Fender | [hC], yes |
| 23:25.34 | [hC] | I have found that the polycom will default to trying FTP, out of the box, but if you want to specify (forcefully) FTP or HTTP, if you send a fully qualified url, ie ftp://user:pass@host or http://url as the tftp-server, that will force it to take ftp or http without any manual intervention. |
| 23:27.33 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 23:28.12 | remmo | nice |
| 23:30.51 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 23:31.30 | syzygyBSD | [hC]: no 601's? |
| 23:31.54 | syzygyBSD | I found the 601 is looking up the hostname of "tftp://192.168.0.1" |
| 23:32.04 | syzygyBSD | could just be a bad bootrom |
| 23:32.30 | asdx | one guy is asking me "what happens if the internet connection goes down, will i lose my call?" is there any workaround on this? |
| 23:33.33 | [hC] | syzygyBSD: if you hand out ftp://user:pass@192.168.0.1 as the tfp server, it should try ftp instead |
| 23:33.38 | `Sauron | asdx: No. |
| 23:33.48 | `Sauron | Er |
| 23:33.52 | `Sauron | Yes, and no. |
| 23:33.53 | asdx | `Sauron: hm? |
| 23:34.12 | `Sauron | "will i lose my call?" is there any workaround on this?" |
| 23:34.19 | `Sauron | Yes, and no. |
| 23:34.31 | asdx | `Sauron: ah |
| 23:34.34 | asdx | `Sauron: ok |
| 23:34.47 | De_Mon | `Sauron how is that not 'yes and yes' |
| 23:35.00 | De_Mon | oh, the workaround would be to use a different media right ? |
| 23:35.04 | `Sauron | De_Mon: Because I'm a bitch. |
| 23:35.05 | `Sauron | :) |
| 23:35.30 | asdx | maybe two simulataneous connections? |
| 23:36.29 | `Sauron | asdx: Sure, it can be worked around. It's a matter of $$. |
| 23:37.13 | asdx | `Sauron: ok |
| 23:37.54 | destructure | how many seconds is "down"? |
| 23:38.57 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 23:39.28 | destructure | short enough and the call won't drop, you'll just experience some missing audio |
| 23:39.50 | `Sauron | asdx: The point is, that as with most technology, when the underlying technology (in this case, the physical media) breaks, all things relying on that technology, cease to function (properly). |
| 23:39.52 | [TK]D-Fender | `Sauron, Your call is TOAST. |
| 23:40.12 | `Sauron | [TK]D-Fender: I'm not making the call. :p |
| 23:40.18 | [TK]D-Fender | `Sauron, WHOEVER'S |
| 23:40.39 | `Sauron | [TK]D-Fender, That's the point I'm trying to make. Duh. :) |
| 23:40.45 | asdx | destructure: i see |
| 23:40.59 | `Sauron | You might want to read the context. It was asdx who was asking. |
| 23:41.08 | asdx | `Sauron: yeh |
| 23:47.55 | *** join/#asterisk CyberSyx^ (n=cybersyx@ppp-221-162.32-151.iol.it) |
| 23:48.08 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 23:51.48 | CyberSyx^ | hi all, i've a big problem, i've configured asterisk with 3 internal, now the internal works great from context [internal] but can i using my sip provider to out frominternal ? |
| 23:52.29 | [TK]D-Fender | CyberSyx^, You can dial anything your device can reach by its context |
| 23:52.38 | [TK]D-Fender | CyberSyx^, names are irrelevant |
| 23:54.51 | CyberSyx^ | [TK]D-Fender, you can see my sip.conf and extesions ? are little lines |
| 23:55.04 | [TK]D-Fender | ~pb |
| 23:55.07 | jbot | A Pastebin is a place to paste your stuff 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 |
| 23:55.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 23:55.13 | [TK]D-Fender | CyberSyx^, and what matters is your DIALPLAN. |
| 23:55.13 | CyberSyx^ | yep |
| 23:55.19 | CyberSyx^ | i using pastebin |
| 23:55.25 | CyberSyx^ | w8 |
| 23:55.53 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:55.58 | CyberSyx^ | excuseme for my bad english :) |
| 23:58.27 | CyberSyx^ | http://pastebin.com/m512260cd |
| 23:58.38 | CyberSyx^ | my problem is this |
| 23:58.41 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:59.08 | CyberSyx^ | if i can using context interno ringing the internal |
| 23:59.33 | hmmhesays | exten => s,n,Set(qty=${FIELDQTY(7015412201&7012122140,\&)}) should that not set ${qty} to 2? |
| 23:59.58 | *** join/#asterisk Op3r (n=Op3r@121.97.246.229) |