| 00:02.39 | pdtpatrick1 | might be the postgresql_server-dev-9.1 |
| 00:04.53 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 00:08.29 | SeRi | scubes13: I am now |
| 00:13.10 | FLeiXiuS | Is there a pseudo channel thats always registered in asterisk? |
| 00:13.22 | FLeiXiuS | Like a DAHDI/pseudo.. or something? |
| 00:14.09 | voipeng | if your using a virtual timing source |
| 00:14.14 | voipeng | for sure |
| 00:14.25 | voipeng | i dunno if you actually have hardware if its there, i dont have any hardware |
| 00:14.44 | voipeng | tdm cards heh |
| 00:16.51 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 00:17.20 | ChannelZ | if you have DAHDI there will be a /dev/dahdi with lots of junk in it |
| 00:17.46 | FLeiXiuS | THere is a DAHDI/pseudo, worked perfectly |
| 00:17.50 | ChannelZ | If not, you'll have res_timing_fdtimer or res_timing_pthreads loaded |
| 00:18.01 | ChannelZ | (probably) |
| 00:18.18 | p3nguin | seri: No 0 after verbose unless you just feel like typing it or if your string contains a comma. No level means 0. |
| 00:18.38 | SeRi | got it |
| 00:18.59 | SeRi | Thanks p3nguin I sort of figured... but wanted to make sure... |
| 00:22.05 | p3nguin | My friggin' desktop OS is hosed, and I don't have the time to fix it for at least a couple more days. How irritating. |
| 00:23.49 | SeRi | :( |
| 00:23.50 | SeRi | one sec |
| 00:27.52 | SeRi | does any body know how the fuck you change the hwclock in solaris? |
| 00:27.56 | SeRi | fucking piece of shit |
| 00:28.37 | pabelanger | really? you feel the need to swear? |
| 00:29.51 | SeRi | when I have been at it for a whole day it becomes very furstrating. we all have been there.... dont tell me you dont get frustrated? |
| 00:29.53 | jrrose | SeRi: I assume you've already seen this and that it's not that easy right? |
| 00:29.53 | jrrose | http://www.howtogeek.com/howto/solaris/set-the-date-and-time-on-solaris/ |
| 00:30.21 | SeRi | jrrose: hwclock does not sync to date! solaris suck. :( |
| 00:30.30 | SeRi | and yess I seen it |
| 00:30.33 | SeRi | Thanks though |
| 00:30.34 | jrrose | Alrighty |
| 00:30.40 | jrrose | I don't know Solaris. |
| 00:31.17 | SeRi | linux = hwclock --systohc very simple |
| 00:31.24 | SeRi | but not in solaris |
| 00:31.31 | SeRi | they have to make it difficult |
| 00:31.33 | SeRi | lol |
| 00:31.34 | SeRi | brb |
| 00:32.53 | jrrose | If it's the hardware clock... can't you set it from the BIOS setup screen of your motherboard? |
| 00:33.02 | jrrose | Oh wait |
| 00:33.06 | jrrose | I bet you are doing it remotely |
| 00:33.22 | jrrose | That would have been way too simple. |
| 00:38.35 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 00:52.11 | SeRi | jrrose: BIOS? on solaris? This is sparc nox x86 :) |
| 00:52.26 | SeRi | There is an EPROM but no BIOS :) |
| 00:55.14 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 00:57.30 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 01:04.21 | *** join/#asterisk coppice (~coppice@m121-202-20-54.smartone.com) |
| 01:07.22 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 01:20.48 | SeRi | p3nguin: you avail? |
| 01:24.22 | *** join/#asterisk jamesmills (~jamesmill@2402:c00:2:0:ca2a:14ff:fe3f:6227) |
| 01:24.33 | p3nguin | You're really going to have to learn how this IRC thing works. |
| 01:24.52 | SeRi | hu? |
| 01:25.43 | SeRi | p3nguin: what do you mean? |
| 01:25.45 | jamesmills | Hey all. Just trying to go through this advisory http://downloads.asterisk.org/pub/security/AST-2011-013.pdf ... Is there a way to test if our systems are vulnerable to this? |
| 01:26.47 | WIMPy | jamesmills: Just keep an eye on your bill :-) |
| 01:27.09 | jamesmills | Haha |
| 01:27.57 | jamesmills | AFAIK we don't have a general nat setting in our asterisk setups |
| 01:28.19 | jamesmills | But we do have some PABXs where peers/users register over the internet and nat is set to "yes" per extension/user |
| 01:31.31 | SeRi | p3nguin: I am not sure what you mean but I was wondering if I could do some test.... all days I have been getting choppy audio in both directions.... it only seems to be from cells.... |
| 01:33.23 | p3nguin | seri: What do you need me to do? |
| 01:33.47 | SeRi | p3nguin: one sec. can I call your conference once I am done with this damn solariz box? |
| 01:33.58 | p3nguin | Sure. |
| 01:33.59 | SeRi | I just want to test audio |
| 01:34.08 | SeRi | Thanks |
| 01:34.42 | p3nguin | If I don't respond right away, it's because I stepped out for a few minutes. I'm trying to get my stuff ready for work tomorrow, but with a broken desktop OS, things are an inconvenience. |
| 01:36.11 | SeRi | o wow that sucks |
| 01:36.32 | *** join/#asterisk adolfomaltez (~taro@190.62.215.67) |
| 01:39.18 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:39.25 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 01:39.51 | p3nguin | Let's just hope I can print. |
| 01:40.49 | p3nguin | I can hear you typing. |
| 01:41.01 | p3nguin | Yes. |
| 01:41.04 | p3nguin | Sounds normal. |
| 01:41.34 | p3nguin | You can't hear me at all? |
| 01:41.41 | p3nguin | Weird. |
| 01:41.52 | p3nguin | Yes. |
| 01:41.56 | p3nguin | I'm talking. |
| 01:42.19 | p3nguin | Do you hear moh? |
| 01:42.28 | SeRi | no sr |
| 01:42.49 | p3nguin | Did you disconnect? |
| 01:43.02 | SeRi | yes |
| 01:43.15 | SeRi | I just called dijib's and I hear the moh |
| 01:43.40 | p3nguin | Something went wrong with my system? |
| 01:43.49 | SeRi | Yes |
| 01:43.55 | p3nguin | wtf |
| 01:44.15 | p3nguin | That's unusual. |
| 01:44.42 | SeRi | can yo hear me? |
| 01:44.55 | SeRi | s/yo/you/ |
| 01:45.03 | p3nguin | I'm not on it right now. I have to fix something. |
| 01:45.21 | SeRi | ah |
| 01:45.23 | SeRi | pl |
| 01:45.24 | SeRi | ok |
| 01:45.26 | SeRi | * |
| 01:47.14 | p3nguin | Call again. |
| 01:47.39 | SeRi | ok one sec |
| 01:47.55 | p3nguin | If it's still messed up, I'll restart the Vyatta. |
| 01:50.23 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:53.19 | p3nguin | /bin/su is owned by coreutils 8.14-1 |
| 01:55.48 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 02:05.26 | p3nguin | eth0_rena |
| 02:07.13 | *** part/#asterisk jamesmills (~jamesmill@2402:c00:2:0:ca2a:14ff:fe3f:6227) |
| 02:09.37 | *** join/#asterisk Androo (~andy@li110-229.members.linode.com) |
| 02:09.44 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 02:10.50 | SeRi | face palms |
| 02:12.38 | WIMPy | Can anyone conform that you must not set ConfBridges mixing_interval to less than the slowest participant? |
| 02:13.17 | WIMPy | s/form/firm/ |
| 02:17.52 | Androo | I'm an Asterisk newbie and as such have a newbie question: If I didn't want to have an FXO card to connect an Asterisk server to PSTN, but wanted to use a remote provider of PSTN services, what exactly am I looking for (SIP provider? PSTN gateway? IAX? Too many acronyms for me to know exactly what I should be searching for) and in the US who are some reputable providers? |
| 02:21.50 | lanning | SIP provider, AKA ITSP |
| 02:21.52 | lanning | http://en.wikipedia.org/wiki/Internet_telephony_service_provider |
| 02:22.03 | WIMPy | ~itsp |
| 02:22.03 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:22.21 | lanning | that too :) |
| 02:22.36 | WIMPy | That doesn't neccessarily mean SIP. |
| 02:22.53 | lanning | true |
| 02:23.19 | *** join/#asterisk master_of_master (~master_of@p57B5403C.dip.t-dialin.net) |
| 02:23.46 | lanning | h323, sip, iax2... |
| 02:25.27 | Androo | les.net good enough for experimentation or do they lack something? I see them mentioned often but they seem to be incredibly cheap in comparison to other services. |
| 02:35.25 | SeRi | p3nguin: I am in. |
| 02:37.23 | *** join/#asterisk god-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 02:38.24 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 02:54.04 | *** join/#asterisk nafg (~quassel@ool-4355e4a2.dyn.optonline.net) |
| 02:54.26 | nafg | Hi, what's the ideal format to use for playback files? |
| 02:54.45 | nafg | Running asterisk without NAT, sip connection via IPKall. |
| 02:57.25 | p3nguin | If you are using ulaw for phone calls, use ulaw for the file format. |
| 03:00.28 | nafg | Thanks p3nguin. From www.ipkall.com: "IPKall supports G.711 u-Law, G.711 A-Law, GSM, G.729 A, and RFC2833." |
| 03:07.07 | nafg | p3nguin: So which is the best? |
| 03:07.27 | nafg | Specifically, I'm using ispeech.org to generate prompts. |
| 03:08.24 | WIMPy | If you want to be prepared for all of them use 8000s/s 1ch 16bit signed linear in a RIFF container (.wav). |
| 03:09.50 | nafg | Hmm, ispeech seemingly only does wavs as PCM. So that means I have to run it through sox? |
| 03:10.21 | WIMPy | PCM doesn'n mean more than digital. |
| 03:13.05 | nafg | So what does RIFF mean? |
| 03:13.54 | WIMPy | That's the correct name for .wav files. |
| 03:15.07 | nafg | How are ulaw, alaw, gsm etc. different than wav --- are they compressed? |
| 03:15.56 | WIMPy | reduced, yes. |
| 03:17.32 | nafg | But, since ispeech has a minimum sampling rate of 16kHz, so for wav I would have to downsample, is there anything to gain by using something other than wav? |
| 03:17.48 | nafg | Or should I upgrade to Asterisk 10? |
| 03:19.21 | WIMPy | Or just leave it that way and be prepared for wideband codecs. |
| 03:19.25 | nafg | WIMPy: (it's helpful if you write my id, that way I get a popup notification when you reply even when I'm on a different screen.) |
| 03:19.36 | nafg | WIMPy: What does that mean? |
| 03:19.56 | nafg | (only if you don't respond immediately) |
| 03:20.11 | nafg | What does it mean, be prepared for wideband codecs? |
| 03:20.25 | nafg | I thought Asterisk 1.8 can't play 16khz wavs? |
| 03:22.29 | WIMPy | nafg: AFAIK it will downsample. But that will cost CPU every time the file is played. |
| 03:23.32 | nafg | But in any case, since when it talks to the remote it has to encode it, can I lose less quality by converting it myself? |
| 03:25.25 | WIMPy | nafg: I have no idea how the internal resampler compares to others. |
| 03:25.43 | WIMPy | But it may be using the same lib. |
| 03:29.18 | nafg | Just converted to ulaw with javax.sound: |
| 03:29.26 | nafg | file newtest2.wav |
| 03:29.26 | nafg | newtest2.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 16000 Hz |
| 03:29.50 | nafg | Should that be .wav or .ulaw, or does extension matter? |
| 03:29.52 | WIMPy | G.711 with 10ks/s? No. |
| 03:30.03 | nafg | What? |
| 03:30.05 | WIMPy | Extension does matter. |
| 03:30.15 | WIMPy | But that format doesn't make sense. |
| 03:30.20 | nafg | What did you say about 10k/s? |
| 03:30.29 | WIMPy | Err 16 |
| 03:30.36 | nafg | What? |
| 03:30.42 | nafg | Don't understand. |
| 03:30.52 | WIMPy | It has to be 8ks/s not 16. |
| 03:31.48 | nafg | Before you said regarding wav that it could downsample. |
| 03:32.06 | nafg | Is that a limitation of asterisk 1.8 or of telephony in general? |
| 03:32.13 | WIMPy | Yes, but you sai G.711 ยต-law. |
| 03:32.23 | WIMPy | said |
| 03:33.55 | nafg | Is there a way to test it from the asterisk console? |
| 03:34.23 | WIMPy | If you have a console configured, yes. |
| 03:34.32 | nafg | asterisk -r |
| 03:34.38 | WIMPy | But I wouldn;t trust that to behave the same as another channel. |
| 03:34.54 | WIMPy | Better us a phone. |
| 03:35.15 | nafg | Oh well |
| 03:37.16 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:43.34 | *** join/#asterisk radic (~radic@dslb-178-002-217-143.pools.arcor-ip.net) |
| 03:44.57 | nafg | Okay, so with wav I get: [Dec 12 22:43:43] WARNING[13863]: format_wav.c:142 check_header: Unexpected frequency mismatch 16000 (expecting 8000) |
| 03:47.38 | *** join/#asterisk bmg505 (~leon@196-209-171-50.dynamic.isadsl.co.za) |
| 03:48.26 | nafg | With that ulaw file it doesn't error but the sound is wrong. |
| 03:49.06 | WIMPy | G.711 has a fixed samplerate and no header. |
| 03:49.11 | p3nguin | It's a 16k file. It should be 8k. |
| 03:49.42 | WIMPy | That's why I said the combination doesn't make sense. |
| 03:50.13 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 03:53.18 | p3nguin | seri: In the client interface, there is a paypal payment option. If I pay $5 there, where does the $5 go? |
| 03:55.25 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:01.59 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 04:11.54 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 04:12.17 | SwK | anyone around know a good wholesaler of mexican DIDs? (and i dont mean DIDWW or DIDX) |
| 04:13.15 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 04:19.41 | SeRi | p3nguin: you in? |
| 04:19.42 | *** part/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:20.09 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:20.12 | SeRi | p3nguin: you in? |
| 04:20.13 | *** part/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:20.17 | SeRi | :/ |
| 04:20.23 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:20.25 | p3nguin | What did you do to me?! |
| 04:20.29 | p3nguin | twice |
| 04:20.30 | SeRi | lol |
| 04:20.32 | SeRi | nothing |
| 04:20.41 | SeRi | I just mention your nick |
| 04:20.46 | SeRi | like so |
| 04:20.49 | SeRi | p3nguin: you in? |
| 04:20.51 | *** part/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:21.00 | SeRi | auto away |
| 04:21.02 | SeRi | lol |
| 04:21.07 | SeRi | nice |
| 04:21.41 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:21.57 | SeRi | is that an auto msg? |
| 04:22.29 | p3nguin | Can you just tell me whatever it is you're trying to tell me? |
| 04:22.51 | SeRi | :/ |
| 04:22.51 | p3nguin | I also asked you about the voipms paypal thing. |
| 04:23.00 | SeRi | Yes |
| 04:23.19 | SeRi | ok so in the client inteface do you see a pay pal link? |
| 04:23.36 | *** join/#asterisk davlefou (~david@41.225.223.69) |
| 04:23.37 | p3nguin | I did last time I looked. |
| 04:23.46 | davlefou | hi, |
| 04:23.52 | p3nguin | There was a paypal option. I could pay $5 up to like $50. |
| 04:23.58 | p3nguin | maybe more than that, I don't remember. |
| 04:24.05 | SeRi | ok do you have a pay pal link setup in the reseller interface? |
| 04:24.11 | p3nguin | Yes. |
| 04:24.51 | SeRi | ok so when the client send the money it will go to what ever pay pal account you have setup in the reseller interface |
| 04:24.51 | p3nguin | Oh, I have 5-100 |
| 04:24.57 | davlefou | i would like use an variable in callerid for the num and name, how i do? |
| 04:25.32 | SeRi | yes thats right. |
| 04:25.40 | SeRi | I have it the same way. |
| 04:25.42 | SeRi | so |
| 04:25.51 | SeRi | if the client sends you lets say 25.00 |
| 04:25.58 | WIMPy | davlefou: You need to be more specific. |
| 04:26.33 | SeRi | and they go to your pay pal account. BUT you need to take the profit out before you send it in. lets say the setup fee |
| 04:27.03 | SeRi | your now down to 20.00 dollars from the customer and up 5 dollars in profit |
| 04:27.10 | SeRi | now. |
| 04:27.21 | davlefou | i ll have sererval user and i want the have the same callerid but, i would like put the information in one point of the file! |
| 04:27.39 | SeRi | You have to send the meony back in to account for the 20.00 dollars. |
| 04:27.49 | davlefou | it was for team of call center! |
| 04:27.50 | WIMPy | davlefou: What kind of users and what file? |
| 04:28.14 | SeRi | BUT you cant. Why? because voip.ms does not allow you to pay pal them no less than 25.00 dollars |
| 04:28.18 | davlefou | i ll use the iax file! |
| 04:29.36 | *** join/#asterisk sjobeck (~sjobeck@67.136.135.134) |
| 04:29.54 | davlefou | WIMPy: do you undestand my problem? |
| 04:30.00 | SeRi | so what happens? not only you where deducted 2.5% for teh transaction but you have to send back in the 25.00 dollars so out of your pocket you have to add 2.5% back to complete the shortage and send them 25.00 |
| 04:30.25 | WIMPy | davlefou: No. Maybe you try to describe what you want. |
| 04:30.27 | SeRi | so your profit goes to them regardles |
| 04:31.23 | p3nguin | I guess I need to try it once. |
| 04:31.54 | p3nguin | So what did you propose they do to resolve the problem? |
| 04:31.54 | SeRi | I am pulling my site up as well |
| 04:32.34 | SeRi | to allow flexibility when pay paling back the money |
| 04:32.42 | SeRi | so not to have a limit of 25.00 |
| 04:34.30 | davlefou | i have some team in an call center, so i need to create count for each user but i want to put the same callerid number. I depend of the customer. |
| 04:37.02 | davlefou | WIMPy: is it more clear? |
| 04:37.28 | p3nguin | seri: I found the reason there was no moh. The stream apparently died. No more Christmas moh stream. :( |
| 04:37.38 | p3nguin | Back to default. |
| 04:38.15 | WIMPy | davlefou: So yo want to identify them by callerID and then pass the calls on with always the same callerID? |
| 04:38.28 | p3nguin | And my desktop doesn't print, but my laptop did! |
| 04:38.43 | SeRi | awesome |
| 04:38.44 | SeRi | ! |
| 04:39.07 | p3nguin | I really want to get this issue resolved ASAP. |
| 04:39.18 | SeRi | go for the update :/ |
| 04:40.55 | davlefou | WIMPy: yes for each team! |
| 04:41.56 | WIMPy | davlefou: Do you need it in your CDRs or where? |
| 04:43.26 | davlefou | CDR? |
| 04:43.39 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 04:43.51 | raden | is it possible to provision polycoms right from tftp server ? |
| 04:44.16 | WIMPy | davlefou: Where do you want to identify your teams? |
| 04:45.02 | SeRi | raden: I think polycom treat tftp as read only mode. ftp is used for read and write. |
| 04:45.10 | SeRi | s/treat/treats/ |
| 04:46.18 | raden | SeRi, how on earth do i make a config for polycom |
| 04:46.24 | SeRi | I could not get my polycom to use tftp for upgrades and cfg updates because I wanted to write to a log file. |
| 04:46.26 | davlefou | WIMPy: i think create an user count for each in the iax.conf with the same context! |
| 04:46.36 | p3nguin | It seems that ftp works better with polycom phones. |
| 04:46.44 | SeRi | p3nguin: +1 |
| 04:46.45 | raden | ok ftp is fine |
| 04:46.56 | raden | again how do i create a config per mac address |
| 04:47.00 | raden | google driving me nuts |
| 04:47.05 | WIMPy | davlefou: You can't count in iax.conf. Where do you want the information to appear? |
| 04:47.09 | SeRi | raden: did you download it the corresponded files to your model from polycom? |
| 04:47.38 | SeRi | just rename 00000000000.cfg to macofmyphone.cfg |
| 04:47.52 | SeRi | and so forth and so on for each config file |
| 04:48.00 | p3nguin | Won't you want to keep 00000000000.cfg as the template? |
| 04:48.09 | p3nguin | copy it, don't rename it. |
| 04:48.10 | SeRi | noope |
| 04:48.20 | p3nguin | You'll want the file for each phone. |
| 04:48.21 | davlefou | WIMPy: why i can't count in iax.conf? i want the target of the call see the number i choise! |
| 04:48.31 | SeRi | 0000000000.cfg is the file that tells polycom which files are where |
| 04:48.54 | p3nguin | You'll want to make copies of the file with each MAC, though. |
| 04:49.18 | WIMPy | davlefou: You said you want to use the same number externally. If that's not your question, I don't get it. |
| 04:49.26 | p3nguin | You'll need to keep the original as the template to copy another off of it. |
| 04:49.44 | raden | SeRi, where do i get a configuration file for my phone ? I have been searching everywhere :( |
| 04:49.58 | p3nguin | If you "mv 00000000000.cfg macofmyphone.cfg" for the first phone, what are you going to do about the next 50 phones? |
| 04:50.05 | SeRi | Thats one way to do it. I do keep copys of the config files. not the 000000000.cfg because that file tells the system where is everything so having to of those files is a bit redundant.... |
| 04:50.13 | SeRi | p3nguin: each phone is different |
| 04:50.16 | SeRi | they are not the same |
| 04:50.31 | SeRi | my 301 uses a different 0000000000.cfg than my 501 |
| 04:50.39 | SeRi | s/301/321/ |
| 04:50.51 | p3nguin | What if you have 20 of the 321 phones? |
| 04:50.56 | SeRi | ah! |
| 04:51.00 | SeRi | than I agree |
| 04:51.11 | p3nguin | You'll change the name of your "template" file. |
| 04:51.17 | [TK]D-Fender | CRAZY PEOPLE |
| 04:51.17 | p3nguin | Then you have one file for one phone. |
| 04:51.29 | p3nguin | And all the other phones can suck it, I guess. |
| 04:51.35 | SeRi | sup [TK]D-Fender to the rescue |
| 04:51.35 | p3nguin | NO MAC.cfg FOR YOU! |
| 04:51.54 | SeRi | s/sup/super/ |
| 04:52.10 | [TK]D-Fender | 00000000 is a 1-time only file and you should never attempt to do anything productive with it. |
| 04:52.12 | SeRi | p3nguin: you just change the whole scenario |
| 04:52.14 | SwK | sup [TK]D-Fender |
| 04:52.39 | [TK]D-Fender | Only good as a means to take the most basic config possible to wipe a phone and bring to "base spec" |
| 04:52.39 | davlefou | WIMPy: yes, i want the same number externally for an team an time! |
| 04:52.48 | davlefou | WIMPy: yes, i want the same number externally for an team at an time! |
| 04:53.02 | SeRi | raden: go to the polycom site and go to support |
| 04:53.09 | SeRi | there you can download your files |
| 04:53.12 | raden | Im there been there for ever |
| 04:53.17 | raden | Not finding it |
| 04:53.21 | SeRi | what model? |
| 04:53.36 | SeRi | [TK]D-Fender: I see. |
| 04:53.45 | SeRi | raden: What model? |
| 04:54.20 | WIMPy | davlefou: Ok, now I read that you want to change it based on time. Maybe you should start again and try to give the whole picture. |
| 04:54.54 | p3nguin | [tk]d-fender: Are you saying the 0000000000.cfg is not supposed to be the starter file for the first use of a new phone? |
| 04:54.59 | raden | SeRi, 321 |
| 04:55.08 | raden | SeRi, I'm totally blind or something tonight |
| 04:55.13 | [TK]D-Fender | p3nguin: Only to reset a phone, never as a final actual config |
| 04:55.23 | p3nguin | I'm asking about first use. |
| 04:55.24 | [TK]D-Fender | p3gits the "catch-all" for when you don't have a MAC specific |
| 04:55.30 | [TK]D-Fender | p3nguin: its the "catch-all" for when you don't have a MAC specific |
| 04:55.42 | SeRi | raden: http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip321_331.html?product_type=2&category=%2FPolycomService%2Fsupport%2Fus%2Fsupport%2Fvoice%2Fsoundpoint_ip%2Fsoundpoint_ip321_331.html%23document#document |
| 04:55.43 | [TK]D-Fender | p3nguin: Good for when you are upgrading firmware |
| 04:56.09 | SeRi | [TK]D-Fender: most polycom know which file to download correctly? |
| 04:56.22 | raden | SeRi, thats where I was what file do I want ? |
| 04:56.23 | p3nguin | How are you supposed to get you first time file for new phone that you just got? Is there a template file that you are supposed to copy to your MAC.cfg? |
| 04:56.24 | SeRi | *correct?* |
| 04:56.26 | raden | I'm missing something |
| 04:56.45 | SeRi | raden: 3.3.3 split |
| 04:56.58 | SeRi | dont get 4.0.1 it wont upgrade :/ |
| 04:57.28 | [TK]D-Fender | SeRi: it downloads what you TELL it to |
| 04:57.44 | raden | everything will be in there ? |
| 04:57.45 | [TK]D-Fender | SeRi: If you don't have a <mac>.cfg then it will look for 00000000 |
| 04:57.54 | raden | SeRi, ^^ |
| 04:58.03 | SeRi | raden: Yes |
| 04:58.21 | SeRi | raden: make sure you update your boot rom as well |
| 04:58.24 | SeRi | [TK]D-Fender: got it |
| 04:58.46 | [TK]D-Fender | ONLY upgrade your BootROM is you absolutely have to |
| 04:58.57 | [TK]D-Fender | If your SIP/UA will work with what you have, don't mess with it |
| 04:59.05 | [TK]D-Fender | UC* |
| 04:59.16 | SeRi | raden: I have a 321 in my kids room and is an awesome little phone. I am now waiting on a 560 *fingers cross* |
| 04:59.18 | [TK]D-Fender | This is where you stand a chance of bricking a phone |
| 04:59.29 | p3nguin | <[TK]D-Fender> SeRi: If you don't have a <mac>.cfg then it will look for 00000000 <----- How do you GET <mac>.cfg in the first place? |
| 04:59.47 | [TK]D-Fender | p3nguin: You copy it and personalize the copy of course :) |
| 04:59.53 | p3nguin | Copy it from what? |
| 04:59.54 | [TK]D-Fender | cp <- |
| 05:00.22 | [TK]D-Fender | cp 00000000.cfg 12345678.cfg |
| 05:00.30 | [TK]D-Fender | p3nguin: Exactly as you suggested |
| 05:00.33 | p3nguin | So that is _exactly_ what I said 10 minutes ago. |
| 05:00.38 | [TK]D-Fender | Correct |
| 05:00.43 | [TK]D-Fender | I'm reinforcing that |
| 05:01.01 | [TK]D-Fender | But you leave the 00000000 around as a means of clean-resetting a new phone / upgrading |
| 05:01.02 | SeRi | how is renaming that file any different? |
| 05:01.05 | p3nguin | Just making sure we're on the same page. I sometimes misunderstand affirmation for indifference. |
| 05:01.19 | [TK]D-Fender | SeRi: because if you rename... it won't be there for NEW phones for a basic reset |
| 05:01.31 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 05:01.34 | p3nguin | seri: It's different because it is the MAC.cfg and you still have your 0000000000.cfg for the next phone so you can copy it again. |
| 05:02.02 | [TK]D-Fender | SeRi: You use 10th generation photocopy forms don't you? |
| 05:02.07 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-gmrzyardokgquqpe) |
| 05:02.09 | p3nguin | haha |
| 05:02.18 | [TK]D-Fender | "Don't fuck the master copy" |
| 05:02.22 | p3nguin | I love the photocopies that say ma... |
| 05:02.28 | p3nguin | I love the photocopies that say master on them. |
| 05:02.46 | p3nguin | photocopied, of course. |
| 05:03.12 | SeRi | [TK]D-Fender: no I just dont do things like you guys do is all. I have 00000 on my default system if for some reason I need it I just move it over in place again. never thought your way. I am not doing large deployments. everything is not in larg scales. |
| 05:03.13 | p3nguin | It's like they couldn't be bothered to stick a post-it over the word Master when they copied it. |
| 05:03.47 | [TK]D-Fender | SeRi: That's fine when its just you at risk... please reconsider when giving this as "advice" for others. |
| 05:04.04 | p3nguin | seri: It's a matter of "If I rename the default file to mac.cfg, then I make changes to it, what the hell am I going to do when I get another phone that needs that default file?" |
| 05:04.23 | SeRi | p3nguin: Thats not what [TK]D-Fender said. |
| 05:04.47 | SeRi | [TK]D-Fender: how is it a risk? I just want to learn so i dont make teh same mistake again |
| 05:05.42 | p3nguin | If you copy the default file to your mac1.cfg and mac2.cfg now, you edit mac1.cfg and mac2.cfg for those phones. Now when you get a third phone, you just copy the default file to mac3.cfg and continue like the previous two phones. |
| 05:05.54 | SeRi | p3nguin: I understand that it makes sense. |
| 05:06.05 | SeRi | but how is that my advice is putting people at risk? |
| 05:06.25 | p3nguin | But you'd rather mv 000000000000.cfg MAC.cfg and to hell with the default original file. |
| 05:06.27 | [TK]D-Fender | p3nguinseri: It's a matter of "If I rename the default file to mac.cfg, then I make changes to it, what the hell am I going to do when I get another phone that needs that default file?" |
| 05:06.31 | [TK]D-Fender | seri that IS what I said |
| 05:07.12 | [TK]D-Fender | [00:02][TK]D-Fender"Don't fuck the master copy" |
| 05:07.22 | [TK]D-Fender | [00:01][TK]D-FenderBut you leave the 00000000 around as a means of clean-resetting a new phone / upgrading |
| 05:07.38 | [TK]D-Fender | Rather precisely. |
| 05:08.18 | p3nguin | I don't understand the aversion to keeping the default file lying around for a rainy day. |
| 05:08.36 | SeRi | [TK]D-FenderI"Don't fuck the master copy" <---- That should make it to the boot. |
| 05:10.24 | SeRi | p3nguin: I understand what you guys are saying and it makes total sense. what you guys are doing is jumping the gun though.... I never siad I got rid of my 000000 file. I did fail to convey that information and for safe keeping to others cp is best. |
| 05:12.06 | SeRi | Like I said I just do it different. I do understand now what [TK]D-Fender said about re considering my advice. |
| 05:13.43 | SeRi | :/ |
| 05:14.07 | p3nguin | Customer service at that Altus carrier whatever SUCKS. |
| 05:14.20 | SeRi | lol |
| 05:14.35 | SeRi | I can say that about comcast :) |
| 05:14.36 | p3nguin | You remember that guy on here a little while ago? |
| 05:14.40 | SeRi | Yes |
| 05:14.53 | p3nguin | I _STILL_ don't have my account with them up. |
| 05:15.05 | SeRi | :/ :\ |
| 05:15.19 | SeRi | And than he wants you to resommend them. |
| 05:15.22 | SeRi | bah.... |
| 05:15.52 | p3nguin | He was supposed to set it up. Never did. I finally sent an email because they still don't have the self sign-up online. |
| 05:16.58 | p3nguin | Days go by without any answer. Then someone finally says "I'll take your account," as if it's a commissioned sort of thing. I reply back with the info he asked for, and now it's still another waiting game for who knows how many days. |
| 05:17.04 | SeRi | have they reply? |
| 05:17.19 | SeRi | oh |
| 05:17.21 | p3nguin | It should be as easy as it is with voip.ms. |
| 05:17.28 | SeRi | +1 |
| 05:17.38 | p3nguin | Open a ticket, it is answered within hours usually. |
| 05:17.52 | p3nguin | Or next business day at the most. |
| 05:18.48 | SeRi | Thats true. I have one open with them right now :) |
| 05:18.55 | p3nguin | Some things take more time than others to complete, like your hard limit issue, but they worked on it and got it done. And the initial response right away was that they would look into it. |
| 05:19.20 | p3nguin | I got nothing for days with this place. |
| 05:19.37 | p3nguin | And I'm not even asking for anything hard... just enable my account that I've been waiting on. |
| 05:19.47 | SeRi | :/ |
| 05:19.55 | SeRi | no good review for them |
| 05:20.03 | SeRi | you should have your own blog spot |
| 05:20.11 | SeRi | I bet it will get sponsor in no time |
| 05:21.15 | p3nguin | They say they have "operated at the carrier level" for some time, and have recently gone into retail sales. |
| 05:21.52 | p3nguin | What does operating at the carrier level even entail? |
| 05:22.09 | SeRi | that they are full of it? |
| 05:22.10 | p3nguin | They certainly can't wholesale to carriers such as callcentric, can they? |
| 05:22.37 | SeRi | I am sure the guy was trying to attract people in... |
| 05:23.03 | p3nguin | The only reason I even care about their services is so I can do cost-based routing instead of a flat rate for all calls. |
| 05:24.17 | *** join/#asterisk Casper_v2 (~Casper@fedora/fantom) |
| 05:24.39 | Casper_v2 | hello |
| 05:24.42 | SeRi | I heard you saying something about that yesterday |
| 05:25.37 | Casper_v2 | I'm packaging asterisk-gui and I've a little problem |
| 05:25.42 | Casper_v2 | https://bugzilla.redhat.com/show_bug.cgi?id=765651 |
| 05:26.08 | Casper_v2 | could you update your COPYING file please ? |
| 05:26.34 | p3nguin | If I can pay less for a call to one destination than I pay with voipms, I should use that route. If it costs more, use voipms. Makes sense to me. |
| 05:27.09 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 05:27.48 | Sean-Der | Hey everyone I made a CEL PHP frontend, it is really basic and it may be only good for inspiration. Anyone interested? |
| 05:28.26 | Sean-Der | I made it to run reports and I have been working with it ever since I was shown how to install cel actually only a day or two so it is still young |
| 05:29.40 | Casper_v2 | because the old COPYING file make a warning by rpmlint (incorrect free software foncdation adress in the header) |
| 05:29.52 | Casper_v2 | *fondation* |
| 05:32.53 | SeRi | p3nguin: any luck with arch? |
| 05:33.42 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 05:33.42 | Casper_v2 | is there a bugzilla for asterisk ? |
| 05:34.58 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 05:35.00 | *** join/#asterisk ponyofdeath (~vladi@cpe-75-80-175-217.san.res.rr.com) |
| 05:35.03 | WIMPy | Casper_v2: http://issues.asterisk.org/jira/ we here in #asterisk are not in to guis, Try #asterisk-gui |
| 05:35.06 | *** part/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 05:35.25 | Casper_v2 | okay thanks WIMPy :) |
| 05:36.19 | p3nguin | seri: It will remain the same until I have a chance to work on it. |
| 05:36.35 | SeRi | ah I see. |
| 05:36.36 | ponyofdeath | him i am getting ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? any ideas? |
| 05:37.10 | p3nguin | Maybe tomorrow. If I get time. |
| 05:37.41 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001725.dsl.bell.ca) |
| 05:38.13 | SeRi | cool |
| 05:38.22 | SeRi | let me know how it goes. |
| 05:40.07 | SeRi | p3nguin: 3 days left on that phone :/ .... I am starting to get anxious. [TK]D-Fender mention it was a good deal for 125.00 |
| 05:40.30 | SeRi | and up to 160.00 |
| 05:41.02 | SeRi | dijib: got your psql going? |
| 05:41.18 | [TK]D-Fender | Casper_v2: You can also bring it up in #asterisk-gui |
| 05:41.35 | [TK]D-Fender | Casper_v2: You can practically double their non-employee occupancy ;) |
| 05:41.52 | Casper_v2 | ^^ |
| 05:41.53 | dijib | nope SeRi |
| 05:41.58 | dijib | havnt been working on it |
| 05:42.01 | dijib | wish to though |
| 05:42.29 | SeRi | :/ |
| 05:42.46 | dijib | you up for a bit? |
| 05:42.53 | dijib | im thinking about pulling an allnighter |
| 05:42.59 | SeRi | lol |
| 05:43.06 | SeRi | I am around |
| 05:43.46 | dijib | well i might go to the smokers room. i should give you access maybe with teamviewer.. dont know why im getting this authentication error |
| 05:44.04 | *** join/#asterisk gerhard7 (~gerhard7@82.171.103.215) |
| 05:44.12 | dijib | let me go to the back, and ill be back |
| 05:44.15 | dijib | 5min |
| 05:44.17 | SeRi | to what? |
| 05:44.18 | SeRi | ok |
| 05:50.56 | davlefou | WIMPy: i am not clear, i have team for mission, i need to change for each mission the number phone display for custumer target. There are serveral workers in each team. |
| 05:51.04 | davlefou | Is it more clear? |
| 05:51.49 | WIMPy | davlefou: Ok, so it is only about changing the number the callee will see? |
| 05:52.05 | davlefou | yes! |
| 05:53.06 | WIMPy | Ok, and how do you want to change it? From their phone? Is is just a few fixed numbers or any number? |
| 05:53.38 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:54.12 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 05:54.39 | dijib | yo |
| 05:54.40 | dijib | im back |
| 05:57.05 | dijib | k working on some teamviewer access |
| 05:58.22 | SeRi | dijib: what errors you are talking about? |
| 05:59.33 | davlefou | i want to change it in from the file of asterisk, by the administrator! |
| 06:01.06 | Casper_v2 | have a nice day everyone ;) |
| 06:06.14 | WIMPy | davlefou: Then either change the callerid setting of the peer(s) or set it in the dialplan. |
| 06:06.54 | davlefou | WIMPy: ok, did you have an exempl? |
| 06:07.34 | dijib | cant find the guide i was working off of |
| 06:07.35 | dijib | gggrrr |
| 06:07.48 | dijib | teamviewer up on a seperate computer |
| 06:08.18 | WIMPy | davlefou: In the peer just add callerid=... or in the dialplan use Set(CALLERID(num)=...). |
| 06:09.56 | davlefou | ok |
| 06:10.03 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 06:12.14 | dijib | psql: FATAL: Ident authentication failed for user |
| 06:12.17 | dijib | is the error |
| 06:12.26 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 06:13.00 | SeRi | I see |
| 06:13.13 | WIMPy | Oh, cool. I plugged in a telephone and Asterisk crashed. |
| 06:13.34 | pdtpatrick | :( |
| 06:13.52 | SeRi | WIMPy: :/ not funny but funny. |
| 06:14.38 | dijib | so i guess i cant connect to the postgres server |
| 06:15.09 | WIMPy | The worrying thing is that you get used to it. |
| 06:16.02 | SeRi | dijib: check your msg |
| 06:17.33 | dijib | ive done the /var/lib/pgsql/data/pg_hba.conf modification to no avail |
| 06:20.02 | *** join/#asterisk EugeneKay (eugene@itvends.com) |
| 06:20.46 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 06:22.02 | gajini | Hi, How to check my Echo chancellor module is removing echo or not? Please help me |
| 06:22.15 | davlefou | i have un g729 codec and i juste by an liscence! |
| 06:23.07 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 06:23.11 | davlefou | I have generate the liscence, my codec is ok but it don't see my liscence! |
| 06:23.15 | dijib | buy the license. :) |
| 06:23.29 | davlefou | dijib: have buy it! |
| 06:23.38 | dijib | you have bought it? |
| 06:23.43 | davlefou | Yes! |
| 06:23.48 | dijib | i have not. |
| 06:24.02 | AliRezaTaleghani | dijib: lisences of what? |
| 06:24.14 | WIMPy | What else than transfer=no do I need to keep dahdi from doing internal transfers? |
| 06:24.16 | dijib | u711 is plenty clear, sans license |
| 06:24.28 | dijib | AliRezaTaleghani: i have no clue, ask davlefou |
| 06:24.32 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 06:24.56 | davlefou | g729 show version : Digium G.729A Module Version 1.8.0_3.1.5 (optimized for generic_64) |
| 06:25.16 | nafg | WIMPy and p3nguin: Thanks for your help, just letting you know it's working great. I feed the http response into sox's stdin from java (scala) so only one file is output. |
| 06:25.39 | davlefou | but the "g729 show licenses" give me back nothing! |
| 06:26.00 | nafg | Also I think it's a lot faster than running text2wave locally. |
| 06:26.22 | SeRi | dijib: he is talking about g729. digiums codec and is licensed |
| 06:26.56 | p3nguin | If you've bought the license, you'll get a key. Use the register utility, and insert the key. Then your license will show up. |
| 06:27.19 | dijib | i realise SeRi |
| 06:27.29 | dijib | but im just saying u711 is better |
| 06:27.31 | dijib | for free |
| 06:27.49 | WIMPy | G.722 is much better and also free. |
| 06:27.57 | SeRi | +1 ^^ |
| 06:28.05 | dijib | (y) |
| 06:28.25 | dijib | 1834688 |
| 06:28.31 | dijib | remember those? |
| 06:28.48 | SeRi | wideband (HD Audio) |
| 06:28.57 | dijib | 722? |
| 06:29.11 | SeRi | g722 |
| 06:29.23 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 06:29.27 | dijib | seri did you ever get that tV? |
| 06:29.44 | SeRi | ? |
| 06:30.12 | SeRi | hey |
| 06:30.22 | dijib | teamviewer |
| 06:30.22 | SeRi | ? |
| 06:30.28 | SeRi | lol |
| 06:30.38 | dijib | oh what version |
| 06:30.40 | SeRi | look at your msg damit |
| 06:30.56 | dijib | 7 |
| 06:31.09 | WIMPy | We don't have anyon here on IRC into the wct4xxb module, do we? |
| 06:31.15 | davlefou | p3nguin: i have already use the Register and he have create an liscence! |
| 06:33.29 | SeRi | go back to 6 |
| 06:36.56 | SeRi | I guess not. |
| 06:40.37 | WIMPy | would like to find out what's wrong with the Digium cards or their driver. But I have no clue, where to go. |
| 06:40.37 | dijib | go back to 6 he says |
| 06:42.01 | SeRi | dijib: dont stress it. I am sure you can manage. |
| 06:42.28 | SeRi | WIMPy: ask Qwell or pabelanger ? |
| 06:43.45 | SeRi | awaits for the magical *fail* |
| 06:44.23 | WIMPy | I have some FAIL to spare. How much do you want? |
| 06:44.25 | ChannelZ | which is the wct4xxb? |
| 06:45.04 | SeRi | WIMPy: lol. No thank you I have some of my own :P |
| 06:45.09 | WIMPy | ChannelZ: The module for Digiums PRI cards. |
| 06:45.39 | SeRi | *fail* is near... I can feal it... |
| 06:46.53 | SeRi | hahaha! fail! |
| 06:47.52 | WIMPy | Oh, shit. |
| 06:48.09 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:48.18 | WIMPy | Looks like the echocanceller is killing the interrupt handling. |
| 06:48.32 | dijib | my battery pack has been running a motor for 1.5days now. |
| 06:49.16 | WIMPy | I've got hardware for that! :-( |
| 06:49.31 | dijib | think i do to |
| 06:49.41 | dijib | o |
| 06:50.45 | WIMPy | For EC that is. |
| 06:51.47 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 06:52.41 | dijib | SeRi: i cant get 6 |
| 06:53.08 | SeRi | Thats fine. I am busy right now. *compiling* |
| 06:54.14 | dijib | library hell |
| 06:54.33 | SeRi | nope |
| 06:54.43 | dijib | i always have that |
| 06:54.50 | dijib | when compiling |
| 06:55.01 | dijib | but packages are just as much of a nightmare |
| 06:55.26 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:57.57 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 06:59.36 | dijib | [unixODBC]FATAL: IDENT |
| 06:59.42 | dijib | oops |
| 06:59.51 | tzanger | hm, is there any way to convince a sip phone (polycom) to update its state in realtime? the hints only seem to update when an actual call is in progress, and seting DND on the phone doesn't change anything, for example |
| 07:00.13 | tzanger | I know that technically even though it's offhook it's not in use, but that's the kind of thing I'm trying to produce |
| 07:01.32 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:01.47 | *** part/#asterisk EugeneKay (eugene@itvends.com) |
| 07:03.53 | SeRi | dijib: I rarely run in to dependency hell... I do my research before compiling anything. Thoug ocasionally they do pop up.... |
| 07:04.51 | SeRi | my issue right now is that the devel env and tool chain on this specific build is some what broken and is not following standards |
| 07:05.33 | SeRi | so trial an error is the only path here |
| 07:06.12 | *** join/#asterisk radic (~radic@dslb-178-002-217-143.pools.arcor-ip.net) |
| 07:07.50 | SeRi | right now I disable asterisk-gui from been compiled to see if I get arround res_http issue with gmime |
| 07:08.12 | SeRi | on this specific build |
| 07:14.21 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:14.22 | schmidts | good morning |
| 07:21.28 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:24.18 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
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| 07:28.37 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 07:37.19 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001642.dsl.bell.ca) |
| 07:38.17 | dijib | d |
| 07:39.15 | SeRi | p3nguin: how can I exclude res_http_post from trying to build? |
| 07:39.22 | SeRi | in asterisk |
| 07:39.32 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 07:41.19 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:43.38 | jacc0 | curl |
| 07:44.00 | jacc0 | sorry, you should d; : make menuselect |
| 07:45.21 | WIMPy | MFBS |
| 07:45.54 | WIMPy | Looks like the current dahdi doesnt support the old HWEC any more. |
| 07:46.23 | WIMPy | Not an excuse for the SWEC to make IRQs fail, however. |
| 07:47.05 | dijib | http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 07:47.38 | dijib | im down to the conf files.. what do i need to do for the pgsql odbc ... |
| 07:47.43 | SeRi | menuconfig --disable res_http_post |
| 07:47.47 | SeRi | :) |
| 07:47.59 | SeRi | I shouldnt need it since I am not using asterisk-gui |
| 07:48.41 | SeRi | is now happy and compiling again :) |
| 07:49.33 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 07:50.36 | dijib | what do you guys think>? |
| 07:50.57 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 07:53.04 | dijib | which configuration files do i need to have in /etc/asterisk/ to have the proper modules loaded? res_odbc is it? |
| 07:54.32 | SeRi | 3.43 1.78 1.17 |
| 07:54.43 | SeRi | brun! baby burn! |
| 07:54.54 | SeRi | s/brun/burn/ |
| 07:54.58 | SeRi | :P |
| 07:55.51 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 07:56.19 | dijib | so do you know seri? |
| 07:58.10 | SeRi | dijib: not sure. I am not very concentrated right now. I am wired up with caffein and compiling... |
| 07:58.32 | dijib | im running on ciggies and i should crack a beer |
| 07:58.52 | SeRi | beer sounds good |
| 07:58.57 | SeRi | too late for me though |
| 07:59.08 | SeRi | must compile! |
| 07:59.26 | *** join/#asterisk timahvo1 (~rogue@197.177.77.110) |
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| 08:08.17 | dijib | need to find realtime_pgsql.sql |
| 08:10.37 | SeRi | :/ |
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| 08:25.30 | dijib | having an issue with echo "local asteriskDB asterisk md5" >> /etc/postgresql/8.1/pg_hba.conf |
| 08:25.31 | SeRi | find all my issues. ha! |
| 08:25.33 | SeRi | all done |
| 08:25.40 | SeRi | ok time to nap for an 1hr |
| 08:25.42 | SeRi | bbl |
| 08:25.48 | dijib | just sleep |
| 08:26.21 | *** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net) |
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| 08:44.42 | dijib | anybody running pgsql? |
| 08:44.45 | dijib | and asterisk? |
| 08:45.04 | dijib | im looking for settings.py |
| 08:46.17 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
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| 08:49.05 | *** part/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 08:50.54 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 08:52.41 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:52.49 | *** join/#asterisk brdude (~brdude@c-24-130-20-142.hsd1.ca.comcast.net) |
| 08:53.09 | IsUp | hi |
| 08:54.11 | *** join/#asterisk kodomo (~gregor@g225036191.adsl.alicedsl.de) |
| 08:54.15 | dijib | sup up? |
| 08:56.53 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 08:57.19 | nny | any late night geeks on, how can I use a password with a "=" in it in a register statement? Don't ask why the vendor facerolled the password |
| 08:58.50 | WIMPy | \= ? |
| 08:59.28 | *** join/#asterisk oej (~olle@fw1.gml.g.icnet.infracom.se) |
| 08:59.28 | nny | still working on setting this up, their instructions are convoluted as all hell |
| 08:59.56 | nny | let me get it to fail with bad pass before I put the right password in, I wish I could share this email, I am either a complete idiot or their system was setup by 100 monkeys |
| 09:01.10 | nny | example is this, their suggested register string: 6782185796@arson.com: 6782185796:6782185796@ pbx.ga1.providerdomain.us/6782185796 |
| 09:01.27 | nny | i have changed it to not leak the actual info, but that's the register string they provided |
| 09:01.36 | nny | mind you the password they have given is no where in there |
| 09:01.42 | nny | and it's not in the example either |
| 09:01.57 | nny | arson.com is the client's domani |
| 09:01.59 | nny | domain |
| 09:02.12 | nny | and they want me to change host.conf to have that domain resolve to another ip |
| 09:02.15 | nny | i have no idea what this is about |
| 09:02.37 | WIMPy | Sounds evil. |
| 09:02.43 | nny | this is how I normally do register strings: register => username:secret@inbound6.vitelity.net:5060 |
| 09:02.50 | nny | example in my configs |
| 09:03.07 | nny | sigh, another vendor, another cryptic process |
| 09:03.53 | nny | WIMPy: what do you make of this? Trunk will register to the resolved IP of pbx.ga1.cproviderdomain.us and use arson.com in the domain portion of the SIP URI: |
| 09:05.06 | nny | Got 404 Not found on SIP register to service 6782185796@pbx.ga1.providerdomain.us, giving up |
| 09:05.27 | nny | trying the simplest setup first to see what they meant to tell me |
| 09:06.11 | WIMPy | Unknown account, probably. |
| 09:07.06 | nny | i put the ip that it pbx.ga1.providerdomain.us resolves to in place of the url too. Jesus |
| 09:07.48 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 09:08.03 | nny | and this http://pastebin.com/7yG7dfMQ |
| 09:08.07 | nny | no idea what it does |
| 09:08.14 | ocx | i have an asterisk box that i need to connect to a PBX |
| 09:08.40 | ocx | i connect the asterisk via the FXO port which will go to the pbx system right? |
| 09:08.57 | WIMPy | That sets a callerID. |
| 09:08.59 | ocx | PBX acts as an FXS ? |
| 09:09.49 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:09.51 | nny | hmm ok one thing a time, need to get it to register first ha |
| 09:09.59 | WIMPy | ocx: You can do it either way. Depends on how you want it. |
| 09:10.30 | ocx | i've read if i connect an FXS port to an FXS i might corrupt the card |
| 09:10.40 | ocx | since 2 signals would be sending at the same time |
| 09:10.44 | nny | dammit |
| 09:10.52 | WIMPy | You always connect two different kinds. |
| 09:10.59 | nny | when I put the password in I get Got 404 Not found on SIP register to service hjkasd8dj@domain, |
| 09:11.18 | WIMPy | But which side uses FXO and which FXS is your choice. |
| 09:11.26 | nny | the password is (like) jksjs=hjkasd8dj, so it's breaking the string |
| 09:11.38 | ocx | ok you mean i can connect my asterisk via FXS port (which acts as a source for signals) to the FXO port on the PBX (which will receive the signal) |
| 09:11.40 | ocx | ? |
| 09:11.43 | nny | so the = sign is being interpreted as a break. ffs |
| 09:12.04 | WIMPy | nny: Have you tried \= instead of = ? |
| 09:12.06 | nny | i tried jksjs\=hjkasd8dj |
| 09:12.12 | nny | yeah, says the same thing |
| 09:12.50 | dijib | anybody done this? |
| 09:12.53 | dijib | http://www.cdr-stats.org/documentation/installation/ |
| 09:12.57 | WIMPy | ocx: Signals should go both directions. Otherwise it's pretty pointless. |
| 09:13.00 | dijib | im lost |
| 09:13.25 | ocx | i mean ringing signals, battery etc |
| 09:13.30 | WIMPy | moves dijib to /lost+found |
| 09:13.45 | dijib | lol |
| 09:13.56 | nny | WIMPy: if I replace the password with a random string i get the 404 still though, gah |
| 09:14.15 | nny | but my thought was their end is denying it based on a bad pass, which I can't send due to the = |
| 09:15.45 | ocx | a telephone is an FXS device? |
| 09:15.54 | ocx | if connected to a pbx |
| 09:15.57 | ocx | analoue phone |
| 09:17.49 | WIMPy | I still haven't fully figured that analog terminology, but you connect a telephone to an FXS port. |
| 09:19.45 | IsUp | hostname = strrchr(username, '@'); /* allow @ in the first part */ |
| 09:20.17 | ocx | WIMPy: you connect a telephone to an FXS port ??? |
| 09:20.25 | IsUp | ocx: as i understand in chan_sip.c source, probably you should ask to your provider to change your username/password |
| 09:20.35 | ocx | you mean FXS Port on the PBX side |
| 09:20.46 | WIMPy | ocx: Yes |
| 09:21.14 | ocx | yea so the wall mount on the telephone side acts as an FXO |
| 09:21.54 | ocx | IsUp: what user/pss? |
| 09:22.37 | IsUp | ocx: sorry, it was wrong tab i was talking to nny :) |
| 09:25.40 | IsUp | ocx: this explains fxo/fxs :P http://www.digium.com/images/graphics/pitch_catch.gif |
| 09:26.35 | WIMPy | So are devices calles the same or the opposite of their interfaces? |
| 09:26.54 | WIMPy | The signalling is the opposite, IIRC? |
| 09:27.04 | IsUp | WIMPy: signalling is opposite |
| 09:27.42 | WIMPy | Luckily I never had to care about that. |
| 09:28.31 | schmidts | IsUp thanks for the link, not i understand it :D :D |
| 09:28.38 | schmidts | s/not/now |
| 09:29.23 | WIMPy | schmidts: Isn't the question of channelised E3 more relevant to you? |
| 09:29.45 | IsUp | WIMPy: yeah its confusing me all the time |
| 09:37.55 | *** join/#asterisk ven0m (~ven@87.69.145.169) |
| 09:38.26 | IsUp | WIMPy: my fxo/fxs configuration, you can have an idea or save it for future :p http://pastebin.com/raw.php?i=1C7dPhz1 |
| 09:38.46 | ven0m | Error: Cannot retrieve repository metadata (repomd.xml) for repository: asterisk-current. Please verify its path and try again |
| 09:39.00 | ven0m | I am unable to install asterisk package on red hat |
| 09:39.08 | ven0m | i am getting 404 on the repomd.xml |
| 09:40.17 | WIMPy | IsUp: "analog gsm gateways"? You connect two digital networks via an analog link? |
| 09:41.17 | IsUp | WIMPy: umm actually these devices are FCTs, like this one: http://www.telsyco.cz/eng/analogova-pruchozi-fxo-fxs-gsm-brana |
| 09:41.35 | IsUp | WIMPy: so im using them for outgoing calls |
| 09:42.23 | WIMPy | Seems like a bad idea. |
| 09:42.47 | IsUp | WIMPy: yes, but we have a live broadcasting studio, it's their own equipments |
| 09:43.12 | IsUp | WIMPy: we are using PRI GSM gateways normally |
| 09:43.47 | WIMPy | Well, that's a little bigger, but they also exist in BRI and SIP. |
| 09:43.54 | IsUp | WIMPy: also we have another FCT, it's using Globalstar satellite to make calls. really buggy equipment |
| 09:56.27 | *** join/#asterisk ven0m (~ven@bzq-218-39-34.cablep.bezeqint.net) |
| 09:57.20 | ven0m | Hey, i am trying to isntall asterisk package on red hat and i am getting 404 on the repomd.xml link |
| 09:57.26 | ven0m | anybody knows whats wrong? |
| 09:57.57 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:03.52 | *** join/#asterisk ven0m (~ven@87.69.145.169) |
| 10:04.17 | ven0m | Hey, i am getting 404 while trying to install asterisk package on Redhat. anybody has any idea what could be wrong? |
| 10:04.40 | WIMPy | Apart from using Red Hat? |
| 10:04.53 | ven0m | Thats what i got from Amazon, sorry :) |
| 10:06.16 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 10:06.26 | ven0m | seriously though |
| 10:06.47 | ven0m | what am i doing wrong? I did exactly what asterisk's guide says |
| 10:14.45 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001642.dsl.bell.ca) |
| 10:16.12 | wdoekes2 | ven0m: most of us don't use rpm's.. I don't know what the repomd.xml file is you're referring to |
| 10:24.42 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 10:25.14 | ocx | is there any standard FXO configuration that works on all PBX? |
| 10:27.52 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 10:28.51 | *** join/#asterisk benj4min (~Benjamin@ep161bs.univ-nancy2.fr) |
| 10:29.01 | benj4min | hi, |
| 10:29.11 | benj4min | I'm using ael language for asterisk configuration |
| 10:29.43 | benj4min | I'm using if("${CALLERID(num)}" < "3667"){..... |
| 10:29.56 | benj4min | is it possible to add an AND option in my if ? |
| 10:30.06 | benj4min | what would be the syntax ? |
| 10:30.10 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 10:33.18 | schmidts | benj4min sure but it depends on your asterisk version |
| 10:34.27 | schmidts | normaly you can use something like this ExecIf($[ $[ a < 5] & $[ a > 3] ]?Noop(a is 4):Noop(a is not 4)) |
| 10:34.41 | IsUp | ocx: what you mean by "FXO configuration"? |
| 10:34.45 | schmidts | ah sorry have overread you use ael but i guess it should be the same |
| 10:35.04 | ocx | configure my asterisk to connect to CO |
| 10:36.22 | IsUp | ocx: whats your hardware on Asterisk and on endpoint? |
| 10:36.32 | benj4min | schmidts, I'm using Asterisk 1.8.7.1 |
| 10:36.38 | benj4min | I'll try :) |
| 10:42.25 | benj4min | schmidts, it works |
| 10:42.35 | benj4min | if("${CALLERID(num)}" > "3654" & "${CALLERID(num)}" < "3668"){ |
| 10:42.40 | benj4min | thanks :) |
| 10:42.43 | schmidts | your welcome ;) |
| 10:42.43 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 10:42.53 | schmidts | you can also use | for an OR |
| 10:44.00 | jacc0 | you might want to make it like this: |
| 10:44.27 | jacc0 | if("0${CALLERID(num)}" > "3654" & "0${CALLERID(num)}" < "3668") |
| 10:44.49 | benj4min | what's the difference jacc0 ? |
| 10:44.54 | jacc0 | I'm not sure how ael handles NULL vars |
| 10:44.59 | benj4min | what does 0$ mean ? |
| 10:45.14 | jacc0 | in normal dialplan it will break execution if ${CALLERID(num)} is NULL |
| 10:45.24 | jacc0 | placing the 0 in front overcomes this |
| 10:45.49 | benj4min | ok, but what would be the case where ${CALLERID(num)} is NULL ? |
| 10:45.51 | schmidts | jacc0 it will not be a problem with the " " around |
| 10:46.45 | jacc0 | benj4min: when there is no callerID number set |
| 10:46.48 | jacc0 | :P |
| 10:47.17 | jacc0 | schmidts: are you sure? |
| 10:47.21 | benj4min | lol |
| 10:47.23 | schmidts | no :D but i think so |
| 10:47.27 | benj4min | I never met this case |
| 10:47.40 | benj4min | we just use asterisk for voicemail |
| 10:47.57 | benj4min | we have cisco solution for phone :) |
| 10:48.05 | jacc0 | SPA? |
| 10:48.26 | schmidts | i guess you mean a Cisco Call Manager or CCM |
| 10:48.48 | jacc0 | okay cisco solution for PBX |
| 10:49.15 | schmidts | jacc0 i am sure now :D |
| 10:49.28 | benj4min | CCM yes |
| 10:49.28 | schmidts | tested it with: same => n,ExecIf($[ "" > "123"]?Noop(a):Noop(b)) |
| 10:49.29 | schmidts | and there is no warning |
| 10:49.39 | jacc0 | okay |
| 10:49.48 | benj4min | and we just have a sip trunk for voicemail |
| 10:50.01 | benj4min | and IVR^^ |
| 10:56.47 | benj4min | tks schmidts and jacc0 , bye |
| 11:20.03 | tompaw | What's the best way to call CLI commands from python? Manager + py-telnet? |
| 11:20.21 | tompaw | I tried installing py-asterisk 0.2, but it fails. |
| 11:21.31 | IsUp | umm, asterisk -rx 'command' maybe? (local env.) |
| 11:22.05 | tompaw | It's not local, they're on separate servers. |
| 11:26.18 | *** join/#asterisk fromol (~n1x@188.129.255.238) |
| 11:27.00 | fromol | hi guys , i need to make direct or grouppickup ( i dont know exactly name ) when ring is comming from siptrunk to my local phone , i need to pickup this ring from another local phone |
| 11:27.48 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 11:27.59 | IsUp | fromol: 'core show application Pickup' |
| 11:29.13 | schmidts | tompaw there is also a manager command to exec cli commands |
| 11:33.04 | tompaw | schmidts: it looks like Manager's QueueStatus action is what I need :-) |
| 11:33.59 | schmidts | ;) |
| 11:36.11 | fromol | IsUp: thanks :) |
| 11:36.24 | IsUp | fromol: good luck |
| 11:37.07 | *** part/#asterisk fromol (~n1x@188.129.255.238) |
| 11:41.50 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:46.00 | *** join/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 11:46.50 | catphish | can anyone explain what asterisk uses to determine where to send the 200 response to a NOTIFY request from a SIP peer |
| 11:47.29 | catphish | i am seeing NOTIFYs arrive from client:1083 then 200 OK send back to client:5060 |
| 11:48.31 | catphish | the peer is in nat: yes and the NOTIFY request carries VIA: 192.168.0.5:5060 |
| 11:51.50 | catphish | at the same time, the OPTIONS requests asterisk sends out are sent to the correct port and get a response from the client as expected |
| 11:55.22 | schmidts | catphish which version do you use? and what kind of notify is this? after a refer or just for nat keep alives? |
| 11:56.00 | *** join/#asterisk flyaster (~flyaster@94.45.77.151) |
| 11:56.19 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 11:57.25 | flyaster | hi, somebody know how to use asterisk 10 rc2 as t.30 (711) to t.38 fax gateway? |
| 11:57.32 | flyaster | i receive error [Dec 13 13:47:36] ERROR[12788]: res_fax.c:1009 fax_session_new: Could not locate a FAX technology module with capabilities (GATEWAY) |
| 11:57.46 | flyaster | and no found nothing at google( |
| 11:58.52 | schmidts | flyaster ask irroot about it but i think you have to use something like faxopt(gateway) for this |
| 11:59.26 | flyaster | yes, i set setvar=FAXOPT(gateway)=yes to peer |
| 11:59.31 | flyaster | but still have this error |
| 12:00.07 | flyaster | and i have buyed fax channels from digium |
| 12:00.30 | schmidts | sorry thats everything i know so far about the fax gateway stuff ;) |
| 12:00.34 | flyaster | )) |
| 12:00.56 | schmidts | flyaster i will ask irroot to come in here, maybe he can help you ;) |
| 12:01.07 | coppice | does the digium FAX module do T,38 gateway? |
| 12:01.10 | flyaster | thanks, also https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway here - is same trouble at last post) |
| 12:01.26 | flyaster | i have res_fax.so and res_fax_digium.so + 3 buyed licenses |
| 12:01.57 | coppice | I thought T.38 gateway only worked with spandsp |
| 12:02.14 | flyaster | hmm |
| 12:02.15 | flyaster | not sure |
| 12:02.21 | flyaster | but will check it now |
| 12:02.22 | schmidts | no idea |
| 12:02.30 | schmidts | but could be a possible problem |
| 12:02.55 | coppice | why are you buying FAX channels anyway? free ones are better value :-) |
| 12:03.17 | flyaster | i have clients who want to use G729 + T.38 razer then G711 |
| 12:03.23 | flyaster | https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway - i check it again - no worlds 'span' here.. |
| 12:03.45 | flyaster | and have clients who have endpoint devices without T.38 support |
| 12:03.52 | flyaster | so need channels... |
| 12:04.26 | coppice | it doesn't mention the digium FAX module. all the development of asterisk's T.38 gateway mode was done with spandsp |
| 12:04.28 | flyaster | [Dec 13 13:47:36] ERROR[12788]: res_fax.c:1009 fax_session_new: Could not locate a FAX technology module with capabilities (GATEWAY) |
| 12:04.29 | flyaster | [Dec 13 13:47:36] ERROR[12788]: res_fax.c:2503 fax_gateway_start: Can't create a FAX session, gateway attempt failed. |
| 12:04.29 | flyaster | [Dec 13 13:47:36] ERROR[12788]: res_fax.c:2666 fax_gateway_detect_t38: error starting T.38 gateway for T.38 channel SIP/2329422-0000067a and G.711 channel SIP/021908-00000679 |
| 12:04.41 | flyaster | full error |
| 12:04.50 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 12:05.02 | flyaster | i really think that it is broken res_fax.. i check it for 10.0.0 beta1 and rc2 |
| 12:05.08 | flyaster | same issue |
| 12:05.21 | coppice | sounds about right. use spandsp, but kick out the digium module |
| 12:06.36 | flyaster | will try spandsp today.. thanks |
| 12:08.25 | catphish | schmidts: 1.8.7.1 and just keepalives |
| 12:10.33 | tompaw | Is Manager returning new lines as \r\n or \n ? |
| 12:13.38 | tompaw | It's \r\n. Thanks mate. You're welcome. |
| 12:14.07 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 12:14.18 | Faustov | tompaw: ;) |
| 12:16.40 | tompaw | <: |
| 12:16.52 | davlefou | hi, |
| 12:17.04 | davlefou | in the cli when i write "reload" |
| 12:17.32 | davlefou | it said me "No such command 'reload' (type 'core show help reload' for other possible commands)" |
| 12:17.50 | schmidts | davlefou core reload is what you are searching |
| 12:17.52 | catphish | try core reload |
| 12:18.34 | catphish | schmidts: did you have any idea where i could start looking with regard to the NOTIFY replies? |
| 12:18.34 | davlefou | ok |
| 12:19.24 | schmidts | catphish i am trying to remember if there was a known problem with incoming notifys and wrong nat handling but nothing comes to my mind. Could you pastebin one of these notifys and the response |
| 12:19.40 | catphish | sure |
| 12:23.46 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 12:29.27 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 12:30.17 | davlefou | before, i use "reload", why i need to use "core reload" now? |
| 12:31.37 | catphish | because they changed the command :) |
| 12:32.12 | catphish | schmidts: i'll gather some sip debug in a while |
| 12:33.28 | davlefou | euh, but i use since two week asterisk in 1.8 lts |
| 12:33.42 | davlefou | i do that since this morning! |
| 12:35.20 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 12:36.39 | fprior | hi, little doubt: I'm apply patch on https://issues.asterisk.org/view.php?id=17635; I must execute configure+make+make install or can I compile only chan_mobile.c ? |
| 12:39.34 | schmidts | fprior after you have applied the patch, just do make and make install |
| 12:39.39 | schmidts | no need to run configure again |
| 12:41.06 | fprior | schmidts: thank you |
| 12:55.05 | *** join/#asterisk vetal (~chatzilla@198-179-113-92.pool.ukrtel.net) |
| 12:55.16 | vetal | Hi all |
| 12:55.50 | vetal | Please help on issue https://issues.asterisk.org/jira/browse/ASTERISK-18940 |
| 12:57.17 | schmidts | vetal can you reproduce this? |
| 13:00.43 | jacc0 | can I set autofallthrough in diaplan? or only as a global? |
| 13:00.55 | schmidts | jacc0 its a global option |
| 13:01.52 | vetal | schmidts: yes, this happend 80% of calls to this peer, 20% go well, no issue |
| 13:02.35 | tompaw | With 1.8, is it possible to trigger an action if a queue call is not picked up by any agent and the customer hangs up? |
| 13:02.36 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:02.39 | schmidts | vetal could you pastebin the sip config of this peer, cause in the issue pabelanger said its a config problem |
| 13:04.52 | vetal | schmidts: http://pastebin.com/pKsSdpys |
| 13:05.56 | vetal | schmidts: [general] section I take from asterisk 1.8 samples, with no changes |
| 13:06.07 | schmidts | vetal ok i will take a look |
| 13:07.03 | vetal | schmidts: when i install 1.6, no problem with default configs |
| 13:07.56 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 13:08.59 | schmidts | vetal very funny thing this is, with the normal default sip config peer ez doesnt even have qualify enabled so it should never become lagged |
| 13:09.21 | schmidts | but there was someone in here yesterday with a similar problem |
| 13:09.33 | schmidts | have you tried earlier 1.8 versions if this happens there too? |
| 13:10.56 | asteriskATmarmuD | hi guys. I need fax detection on outgoing calls. If a fax is detected the call should be hung up and I need some kind of special hangupcause/dialstatus for that... all with asterisk 1.4 btw |
| 13:11.37 | jacc0 | ${CHANNEL} holds the channel ID of the call-leg it is executed in; how can I get the channel id of the call-leg it is connected to? |
| 13:12.16 | schmidts | jacc0 depends on the version but imho there is something in the cdr values |
| 13:12.54 | jacc0 | 1.8.8.0-rc4 |
| 13:16.56 | leifmadsen | tomaw: I believe you can have the call continue on in the dialplan, or you might have to perform the action in the 'h' extension |
| 13:17.03 | leifmadsen | (likely the latter) |
| 13:17.07 | vetal | schmidts: this happend only with 1.8 and Mera Softswitch, and not all calls, only someone |
| 13:17.43 | tompaw | leifmadsen: thanks, I will look into it. |
| 13:19.11 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:19.31 | schmidts | vetal i have attached a patch, please try it and atleast try to reproduce it |
| 13:20.41 | tompaw | leifmadsen: something like that: http://pastecode.com/iz ? |
| 13:22.11 | leifmadsen | tomaw: if it works, sure :) |
| 13:22.16 | tompaw | (: |
| 13:22.36 | leifmadsen | if the caller hangs up, it might not continue in the dialplan, which is why I mentioned the 'h' extensions |
| 13:22.47 | leifmadsen | it's early for me, so my brain probably isn't thinking about it too clearly |
| 13:22.59 | leifmadsen | and I'm looking at puppet code |
| 13:23.20 | schmidts | leifmadsen good morning btw ;) |
| 13:23.28 | leifmadsen | morning :) |
| 13:23.32 | leifmadsen | 8:23am! |
| 13:23.38 | leifmadsen | been at my desk nearly 45 mins now |
| 13:24.00 | schmidts | its allready 2:23 pm here, sitting on my desk allready too long for today :D |
| 13:25.20 | leifmadsen | indeed |
| 13:25.24 | leifmadsen | I'm going christmas shopping today... |
| 13:25.49 | tompaw | what's a puppet code? |
| 13:25.55 | schmidts | hehe, i allready finished all christmas terror shopping |
| 13:25.58 | tompaw | do you code your dogs? |
| 13:26.43 | vetal | schmidts: after patch, 10+ calls no more errors! |
| 13:26.51 | tompaw | schmidts: that's a proper time zone sir! |
| 13:27.52 | schmidts | tompaw ;) |
| 13:28.33 | leifmadsen | tomaw: no, http://www.puppetlabs.com |
| 13:28.45 | schmidts | vetal if possible let this run for another hour just to make sure but i think this solves this problems |
| 13:28.53 | schmidts | just the same problem as another guy had yesterday ;) |
| 13:29.22 | vetal | schmidts: yesterday I was here, but do not get solved |
| 13:29.25 | leifmadsen | schmidts: am I reopening ASTERISK-18940 then? |
| 13:29.43 | schmidts | leifmadsen i will do ;) |
| 13:29.51 | leifmadsen | schmidts: ok, please assign to yourself then :) |
| 13:30.32 | schmidts | damn i should have thought about this :( |
| 13:31.06 | tompaw | cool, seems like a complicated enterprise-class project that does something! |
| 13:31.10 | vetal | schmidts: this issue can cause wrong tarification of calls? The issue come, when operator give their CDR, and I see that some calls were with 2400 sec, but I have 0 in my cdr. is it assosiated? |
| 13:32.23 | schmidts | vetal yes cause if its really the problem i have in my mind, for example the operator sends you an 200 ok but asterisk thinks this is just an options message and so from your side the call is still not active but its active from the other side |
| 13:33.25 | *** join/#asterisk oej_ (~olle@87.96.134.129) |
| 13:33.29 | vetal | schmidts: now I apply patch to asterisk with live traffic to test |
| 13:34.28 | *** join/#asterisk fofware (~fabian@host83.190-225-15.telecom.net.ar) |
| 13:36.26 | p3nguin | seri: make menuselect ; turn it off. If not there, I don't know. |
| 13:36.41 | SeRi | o I did :) |
| 13:36.44 | SeRi | Thanks! |
| 13:37.59 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 13:41.06 | *** join/#asterisk oej (~olle@fw1.gml.g.icnet.infracom.se) |
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| 14:01.35 | schmidts | vetal have you allready tried it in production? |
| 14:02.33 | schmidts | UnixDev are you there? |
| 14:04.20 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
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| 14:11.01 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:11.58 | vetal | no, I will apply it in 1 hour |
| 14:12.26 | vetal | schmidts: I have problem with applying one more patch, that I need |
| 14:12.44 | schmidts | vetal ok ;) |
| 14:13.01 | schmidts | i have allready put the patch on reviewboard so i guess this will be fixed in the next release version |
| 14:14.02 | vetal | schmidts: thank you, now I try to apply patch for textmessages outbound of call, but patch gives error, because I get last version of asterisk 1.8.7.2 |
| 14:14.51 | schmidts | vetal if you mean my patch, do it by hand its just one row |
| 14:15.05 | vetal | schmidts: not yours patch, other one |
| 14:15.14 | vetal | your patch I apply successed |
| 14:15.37 | schmidts | ok ;) |
| 14:16.17 | vetal | schmidts: can you help me with other patch to apply? |
| 14:16.33 | schmidts | i can try ;) |
| 14:16.35 | vetal | schmidts: I think I need to modify patch, but I failed to do |
| 14:19.43 | vetal | schmidts: http://pastebin.com/TdPN09ug |
| 14:19.51 | vetal | I've got Hunk #6 failed at 4421. |
| 14:20.03 | vetal | schmidts: I past to you only problem part |
| 14:22.14 | schmidts | vetal you are using 1.8.7.2 right? cause even in 1.8.7.1 the row which should be changed is 4437 and not 4420 |
| 14:22.27 | schmidts | vetal maybe just patch this row by hand, shouldnt be a problem |
| 14:22.47 | *** part/#asterisk roham (~ali@31.184.187.2) |
| 14:23.58 | vetal | schmidts: I can modify by hand, but I want to correct patch |
| 14:24.29 | schmidts | hmm ok thats not so easy |
| 14:24.51 | vetal | schmidts: in your patch, line in patch do not match line in chan_sip.c but it success |
| 14:25.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:28.26 | schmidts | vetal yes cause the lines doesnt differ too much i guess |
| 14:28.49 | vetal | schmidts: how to make patch, I want to make by hand, and than make a patch |
| 14:29.04 | vetal | schmidts: my own patch |
| 14:29.29 | schmidts | you have to use two files, the original one and the changed one and then just use diff |
| 14:29.52 | schmidts | diff -u channels/chan_sip.orig channels/chan_sip.c > patch |
| 14:31.03 | *** join/#asterisk inaciano (~diego@189.24.123.105) |
| 14:32.11 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 14:35.14 | vetal | thank you |
| 14:39.43 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:39.51 | *** join/#asterisk ipconfeng (~ipconfeng@pool-98-114-57-252.phlapa.fios.verizon.net) |
| 14:41.21 | vetal | schmidts: wow, one more mistake, http://pastebin.com/c2w3rMsz look two chan_sip.c 1.8.5 and 1.8.7 original files |
| 14:41.50 | vetal | schmidts: pastbin line 4 and 14 |
| 14:50.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:51.02 | UnixDev | schmidts: hi there |
| 14:51.12 | UnixDev | schmidts: I am here now, just saw your msg |
| 14:56.36 | jacc0 | why does this sip/555 exit non-zero when bridgin": http://pastebin.com/dCUnPLQY |
| 14:59.02 | akrohn | what could cause a SIP phone to receive a Page() when DND is turned on? |
| 15:00.19 | vetal | schmidts: patch applied to production |
| 15:01.31 | [TK]D-Fender | akrohn, Show us the call |
| 15:02.36 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 15:02.49 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:03.17 | akrohn | I can't right now. I'll do some more testing and come back later with useful info. just wondering if you knew a place I could start looking. |
| 15:03.55 | [TK]D-Fender | akrohn, No details were provided, thus no means to suggest. |
| 15:04.24 | akrohn | kk bbl =) |
| 15:04.45 | [TK]D-Fender | Page() is a call like any other Dial() |
| 15:05.49 | *** join/#asterisk lftsy (~lftsy@194.38.160.113) |
| 15:06.13 | akrohn | what if I were to ask you something like, I have a phone on DND. By what mechanism can I override this and force the call to come through? |
| 15:06.40 | akrohn | autoanswer sip headers, perhaps? |
| 15:06.56 | [TK]D-Fender | Might happen. Depends on the phone |
| 15:07.19 | [TK]D-Fender | Or it might not be really on "DND |
| 15:08.21 | [TK]D-Fender | <akrohn> what could cause a SIP phone to receive a Page() when DND is turned on? <- is this a request? Or is this a question about a time that it actually happened? |
| 15:08.53 | schmidts | UnixDev and also vetal i will submit my patch you both got to code soon |
| 15:09.29 | UnixDev | schmidts: you found something else related to the options snafu ? |
| 15:10.47 | akrohn | [TK]D-Fender, i have a customer telling me how their DND is interrupted by their Page feature. Just wanted to get an idea of how that could happen |
| 15:10.47 | schmidts | UnixDev it was the change i have sent you yesterday. it was possible that an incoming message was handled as an incoming option response even if it was something else |
| 15:11.07 | schmidts | UnixDev have you tried it on your system? were you able to still reproduce this problem? |
| 15:11.09 | akrohn | [TK]D-Fender, I will try reproducing it on my test system to find out more |
| 15:11.24 | UnixDev | schmidts: I'm testing now, about to start heavy fax hour, so we will know soon |
| 15:11.25 | [TK]D-Fender | akrohn, What "DND" would they have set exactly? |
| 15:11.29 | UnixDev | seems better already though |
| 15:12.31 | schmidts | ok |
| 15:12.32 | akrohn | [TK]D-Fender, DND on a cisco 7940. on a per-endpoint basis |
| 15:13.13 | voipeng | what is the general consenus... build asterisk from source or rpm? is one more preferred than the other? |
| 15:13.35 | [TK]D-Fender | akrohn, interesting... they support a header for it now as opposed to treating a 2nd registration to a auto-answer account? |
| 15:13.44 | leifmadsen | I don't think you can override the DND on the phone |
| 15:14.05 | [TK]D-Fender | voipeng, either works. |
| 15:14.14 | [TK]D-Fender | voipeng, As long its built right for your system |
| 15:15.00 | *** join/#asterisk mhaddog (~gabriel@z65-50-116-231.ips.direcpath.com) |
| 15:15.50 | akrohn | [TK]D-Fender, That's what I was wondering. Our paging macro does some sip header insertion so I was thinking that might cause it. leifmadsen, thanks for the tip. I will try reproducing the scenario locally |
| 15:16.30 | leifmadsen | voipeng: you could build your own RPMs from source too :) |
| 15:16.37 | mhaddog | is there a way to log every step that asterisk does? |
| 15:16.41 | leifmadsen | nothing wrong with packages |
| 15:16.43 | mhaddog | outside asterisk's logs? |
| 15:16.54 | leifmadsen | mhaddog: log everything into syslog |
| 15:17.13 | mhaddog | like creating an asterisk loggin outside asterisk? to be abel to log all that it does? |
| 15:17.22 | leifmadsen | mhaddog: see above |
| 15:17.30 | leifmadsen | put it into syslog |
| 15:17.40 | voipeng | gotcha, i was under the assumption since voiceaxis in our scenario has a repo we pull from and it didnt build with zaptel working, if trying to compile from source would yeild better results |
| 15:17.42 | leifmadsen | otherwise, you're going to have to be more specific on your question |
| 15:18.11 | mhaddog | asterisk is getting stuck, and I don't see anything in syslog |
| 15:18.19 | mhaddog | that's like the issue |
| 15:18.39 | leifmadsen | voipeng: asterisk would need to be compiled against zaptel/dahdi in order to build the timing modules, so likely you'd need to build separate RPMs that depended on zaptel existing (and was aware of it) |
| 15:18.50 | leifmadsen | mhaddog: you have to configure asterisk to write to syslog |
| 15:18.55 | leifmadsen | it doesn't do it automatically |
| 15:19.04 | leifmadsen | it writes to logfiles by default |
| 15:19.19 | leifmadsen | and you need to have the appropriate verbosity and debug levels set at the console |
| 15:19.21 | mhaddog | ok, that means recompile it? |
| 15:19.23 | leifmadsen | core set verbose 5 |
| 15:19.26 | leifmadsen | core set debug 5 |
| 15:19.29 | leifmadsen | for example |
| 15:19.34 | leifmadsen | mhaddog: no, that means enabling it |
| 15:19.42 | leifmadsen | mhaddog: in logger.conf |
| 15:19.44 | mhaddog | ok so just enable it... |
| 15:19.49 | mhaddog | verbose 5 or 6? |
| 15:19.56 | leifmadsen | 800km |
| 15:20.38 | mhaddog | I saw something on dmesg, seems like the T1 card had an issue, but I kill/start asterisk and all goes back to normal... the only problem is that this freezing in happening too many times during the day.... |
| 15:21.44 | voipeng | thanks leif |
| 15:22.43 | leifmadsen | runs away |
| 15:23.07 | mhaddog | ok so I need to see logger.conf then? |
| 15:23.31 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 15:24.07 | schmidts | UnixDev and vetal patch is submitted into the source code |
| 15:24.09 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:24.25 | schmidts | leifmadsen is it too late for this one to come into 1.8.8 ? |
| 15:24.27 | UnixDev | ahh perfect, will I get that if I update my branch to the latest SVN ? |
| 15:24.34 | UnixDev | using /trunk |
| 15:24.38 | mhaddog | or could verbosity be increase to the files where it ussually writes to? |
| 15:24.42 | leifmadsen | schmidts: yes |
| 15:25.15 | leifmadsen | pending an issue that simply causes anarchy in the world, 1.8.8.0-rc5 will go out as 1.8.8.0 |
| 15:25.15 | schmidts | is 1.8.8 allready released? thought there were only rcs available ;) |
| 15:25.21 | leifmadsen | only RCs |
| 15:25.33 | leifmadsen | you can check with mjordan, but this is not a blocker in my opinion |
| 15:25.40 | leifmadsen | 1.8.8.0 has been delayed far too long |
| 15:25.43 | schmidts | not really ;) |
| 15:25.46 | schmidts | yes thats true |
| 15:25.53 | leifmadsen | 1.8.9.0-rc1 would come out very shortly after 1.8.8.0 |
| 15:25.56 | jacc0 | true |
| 15:25.59 | jeroen_h | hello, I have a problem with my newly installed 1.8.8.0~rc4 |
| 15:26.07 | leifmadsen | so people won't have to wait long for it, and they can easily backport it if it's that big of a deal |
| 15:26.09 | jeroen_h | I am not able to use the following in my dialplan: |
| 15:26.14 | leifmadsen | jeroen_h: pastebin |
| 15:26.15 | jeroen_h | exten => 999,3,Set(DB(test/count=1)) |
| 15:26.17 | schmidts | leifmadsen what anarchy issue do you mean? :D |
| 15:26.22 | mjordan | if it goes into the 1.8 branch, it would be slated for 1.8.9 |
| 15:26.24 | leifmadsen | jeroen_h: because you're using it wrong |
| 15:26.31 | [TK]D-Fender | jeroen_h, You aren't setting it to anything. |
| 15:26.35 | jeroen_h | that would explain it :) |
| 15:26.35 | leifmadsen | jeroen_h: exten => 999,n,Set(DB(test/count)=1) |
| 15:26.43 | schmidts | mjordan thanks ;) |
| 15:26.46 | jeroen_h | hm |
| 15:26.50 | jeroen_h | wrong bracket |
| 15:26.54 | leifmadsen | wrong spot |
| 15:26.55 | jeroen_h | thanks :) |
| 15:27.00 | jacc0 | hehehe |
| 15:27.09 | leifmadsen | bracket was right |
| 15:27.14 | leifmadsen | if you used a } it would be the wrong bracket |
| 15:27.17 | jeroen_h | fresh eyes help for stupid problems :p |
| 15:27.22 | jeroen_h | true |
| 15:27.25 | [TK]D-Fender | } = brace ;) |
| 15:27.30 | leifmadsen | [TK]D-Fender: I know... shhhhh |
| 15:27.38 | leifmadsen | don't be pedantic! |
| 15:27.45 | leifmadsen | I WILL FIGHT YOU TO THE DEATH |
| 15:27.54 | [TK]D-Fender | leifmadsen, pedantic=yes <- always in my sip.conf :) |
| 15:28.32 | jacc0 | pedantic=yes only works with notsofunny=yes |
| 15:28.37 | jacc0 | :P |
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| 15:30.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:31.38 | IsUp | hello |
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| 15:39.09 | phix | :D |
| 15:40.14 | vetal | schmidts: thank you, patch work good for me, no more errors |
| 15:42.34 | schmidts | vetal your welcome! thank you for noticing this so we could solve it |
| 15:42.43 | *** part/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 15:43.33 | vetal | schmidts: I will spend in irc chat more time :) |
| 15:44.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:48.01 | schmidts | vetal ;) |
| 15:48.53 | *** join/#asterisk psharmor (~tekoholic@97-118-209-77.hlrn.qwest.net) |
| 15:51.52 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377107.home.otenet.gr) |
| 15:56.08 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 15:57.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:02.29 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:02.29 | *** join/#asterisk flexnsniff (~flexnsnif@173-18-192-234.client.mchsi.com) |
| 16:04.14 | flexnsniff | So, I guess I'm new to telephony...? |
| 16:04.54 | flexnsniff | I know a little bit, but obviously not enough yet.... |
| 16:05.05 | WIMPy | flexnsniff: You don't even know if you are new? |
| 16:05.07 | flexnsniff | Well |
| 16:05.10 | carrar | Please fill in the blank what you know ____ |
| 16:05.17 | flexnsniff | I know quite a bit about computers |
| 16:05.38 | flexnsniff | I know a decent amout about ubuntu |
| 16:05.41 | flexnsniff | And |
| 16:05.49 | flexnsniff | I want to run asterisk/google voice |
| 16:06.08 | [TK]D-Fender | Uphill battle there. GV is always breaking... |
| 16:06.09 | carrar | Great, grab the source and run! |
| 16:06.13 | flexnsniff | I have got it compiled and everything, libssl, etc |
| 16:06.29 | flexnsniff | What it seems I dont know about is SIP |
| 16:06.32 | voipeng | anyone have a good distribution site that ships to us/philly rather quickly? Was hoping for a usb adapter |
| 16:06.46 | [TK]D-Fender | voipeng, Which one? |
| 16:06.55 | voipeng | ut50 ? |
| 16:07.15 | voipeng | i went to the manufacturer directly but they require quotes which i thought was odd |
| 16:07.16 | [TK]D-Fender | Ah.. was thinking about that for you.. |
| 16:07.19 | voipeng | hehe |
| 16:07.29 | voipeng | i did purchase the last suggestion you had |
| 16:07.32 | flexnsniff | hmmm. |
| 16:07.33 | voipeng | but i didnt get tracking info |
| 16:07.37 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 16:07.37 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 16:07.37 | flexnsniff | so GV breaks alot huh. |
| 16:07.41 | [TK]D-Fender | voipeng, You don't buy direct from mfg's normally.. distribution channels... |
| 16:07.42 | voipeng | i wanted a fall back option since i knew usb would work |
| 16:07.45 | voipeng | ah |
| 16:07.52 | voipeng | well i guess i asked the question right :) |
| 16:08.04 | voipeng | just executed backwards |
| 16:08.06 | [TK]D-Fender | voipeng, might do the job, and certainly a more portable solution |
| 16:08.33 | *** join/#asterisk screenn (~screenn@212-56-202-46.pool.ukrtel.net) |
| 16:08.38 | flexnsniff | so, my next question, you can use SIP to run the outbound side of asterisk, |
| 16:08.47 | flexnsniff | Any reccomended SIP providers> |
| 16:08.54 | carrar | ~itsp |
| 16:08.54 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 16:08.55 | flexnsniff | that include a US number |
| 16:09.04 | [TK]D-Fender | flexnsniff, voip.ms, vitelity, les.net |
| 16:09.18 | flexnsniff | oh snap. |
| 16:09.35 | flexnsniff | i love irc bots, lol. they can do so many things >.< |
| 16:09.51 | carrar | [TK]D-Fender is one of the better bots in the channel |
| 16:09.59 | pdtpatrick1 | Question .. besides timing issues.. what other issues are there with regards to having asterisk in a VM ? |
| 16:10.04 | flexnsniff | lol |
| 16:10.11 | [TK]D-Fender | jbot |
| 16:10.16 | [TK]D-Fender | ~jbot |
| 16:10.16 | infobot | [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, or suck, or a pain in the ass |
| 16:10.26 | [TK]D-Fender | MAH BITCH! |
| 16:10.33 | pdtpatrick1 | haha |
| 16:11.14 | [TK]D-Fender | For those who've been around to remember when infobot used to be jbot... |
| 16:11.22 | [TK]D-Fender | "long enough to" |
| 16:12.21 | *** join/#asterisk devaf (kannan@123.238.108.34) |
| 16:13.27 | WIMPy | Oh, BTW: |
| 16:13.32 | WIMPy | ~freepbx |
| 16:13.32 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 16:13.52 | WIMPy | [TK]D-Fender: Can you remove that ", or fpbx" part? |
| 16:14.16 | WIMPy | Unless someone can explain what that means. |
| 16:14.29 | [TK]D-Fender | ~freepbx |
| 16:14.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:15.02 | WIMPy | Coo, now we don't have to guess any more. |
| 16:15.03 | [TK]D-Fender | done |
| 16:15.36 | *** join/#asterisk Hanoon (~Hanoon@static-71-190-139-66.nycmny.fios.verizon.net) |
| 16:15.59 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 16:16.00 | devaf | i am not abl to install libsrtp for my asterisk 1.8.7.2 on Ubuntu Lucid 10.04 LTS 64 bit server , can anyone guideme ? |
| 16:16.54 | pabelanger | devaf: you have to manually compile it or backport it from maverick |
| 16:17.34 | Hanoon | Hello Everyone , This s my first connect and I have a question... Not sure If I just ask it here or wait until the question in progress complete.. |
| 16:17.57 | WIMPy | ~ask |
| 16:17.58 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:18.05 | WIMPy | Hanoon: ^ |
| 16:18.40 | devaf | @pabelanger , thanks i will try the back port ; it does build from source |
| 16:18.43 | devaf | does not |
| 16:20.10 | Hanoon | I have a fresh install on 1.8.7.2 on Centos 6.0 - Everything works except that the RTPTIMEOUT=X and RTPHOLDTIMEOUT=X will not work , I have looked and found the patch 18559 in the bug tracker, Applied patch redone make and make install - problem still exist - Asterisk will not hangup call when there is not RTP. |
| 16:21.24 | pdtpatrick1 | devaf is there an error when you run ./configure ? |
| 16:21.30 | pdtpatrick1 | here's what i usually do |
| 16:22.03 | pdtpatrick1 | sudo apt-get install build-essentials libxml2-dev libssl-dev curl |
| 16:22.04 | WIMPy | Hanoon: Where does redfone go in there? |
| 16:22.23 | Hanoon | No errors while running ./configure |
| 16:22.28 | pdtpatrick1 | and then compile.. if there's an error, it will tell you that something is missing and you'll then have to look for the library |
| 16:22.30 | WIMPy | I only know that rtptimeout if "off by one". |
| 16:22.52 | flexnsniff | ok, so far as SIP+Asterisk goes, rock solid for years to come right? |
| 16:23.35 | Hanoon | WIMPY - Sorry what is redfone means? |
| 16:23.36 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:23.38 | wcselby | o/ |
| 16:24.04 | WIMPy | Hanoon: Oh, sorry, misread that. |
| 16:24.47 | devaf | hello, to build libsrtp i ubuntu server 10.04 , http://srtp.sourceforge.net/download.html -> current version 1.4.2 works || but http://sourceforge.net/projects/srtp/ -> currently version 1.4.4 does NOT |
| 16:27.35 | Hanoon | this is a link for the patch applied https://issues.asterisk.org/jira/browse/ASTERISK-18559 |
| 16:27.36 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 16:28.59 | vader-- | If you had to build a 75-100 phone asterisk system, would you choose to build asterisk by hand with dial plans by hand or use a front end such as freepbx? I work for a school which is using an older version of asterisk 1.2.26... The system works great, but I am looking for a way to delegate some fo the simple tasks to other users in the IT department who aren't exactly smart enough to write andtest dialplans. I built our current system 6 |
| 16:29.59 | vader-- | THey are currently using a 23 channel pri, and a 24 channel TDM2400P board for analog channes, with 60-70 Cisco 7940G IP Phones |
| 16:31.36 | pabelanger | vader--: If they are not smart enough, train them. What makes you think using something like FreePBX will make it easier? |
| 16:31.46 | vader-- | point and click |
| 16:32.00 | pabelanger | and when you run into problems with asterisk? |
| 16:32.14 | Hanoon | WIMPy I meant Re-Done (Make and Make install) after applying patch |
| 16:32.41 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 16:32.47 | [TK]D-Fender | vader--, Depends what your needs are beyond letting questionable people near your PBX |
| 16:32.56 | vader-- | right now to provision a phone, I have to edit a cfg file, upload it to the tftp section of the er with WinSCP, then edit our dialplans upload them, update the sip.conf |
| 16:33.25 | WIMPy | Hanoon: yes, got it. Did you try to set the values in the global section? |
| 16:33.35 | wcselby | vader-- I find freepbx to be easier to use on smaller systems *(less than 25 end points / users). if you get higher than 25, it's much easier and faster to do everything by hand |
| 16:33.39 | vader-- | it's not me really pushing for this |
| 16:33.54 | vader-- | my boss is trying to take stuff off my plate and give some things to my coworker |
| 16:34.30 | Hanoon | yes and per peer too - None of them take effect. Call stays up after unplugging network cable from phone. |
| 16:34.35 | wcselby | get your coworker to do some of the easier shit |
| 16:36.31 | *** join/#asterisk cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 16:36.39 | vader-- | ya i was just tossing the idea out there |
| 16:36.49 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 16:36.49 | vader-- | it doesn't bother me to edit the confs and do everything manually |
| 16:36.58 | vader-- | just looking for a point/click method for them |
| 16:37.18 | _Corey_ | vader--: I used FreePBX as a front-end at a 600 endpoint school district if that makes you feel any better |
| 16:37.22 | devaf | Hanoon , thanks i just saw your message |
| 16:37.42 | vader-- | corey was it a pia to use it with that many end points? |
| 16:37.43 | cadey | Hi guys, when you make a call in asterisk and it goes out over a PRI does the system convert all DTMF tones to in-band? |
| 16:38.03 | Hanoon | Thank you - Little more details about my setup I have (phone -----> opensips "as proxy" ---->Asterisk---->opensips ----->carrier) |
| 16:38.24 | [TK]D-Fender | <vader--> just looking for a point/click method for them <- well you already have a solution for this... |
| 16:38.39 | [TK]D-Fender | cadey, Yes |
| 16:38.50 | _Corey_ | vader--: Well, it was a migration so it was all populated via spreadsheets and import scripts... The phone provisioning doesn't really change much so we found it effective enough for their staff |
| 16:40.22 | WIMPy | cadey: That depends on the channeltype and configuration. |
| 16:40.33 | _Corey_ | vader--: That said, I wouldn't go near FreePBX on my own pbx. ;) |
| 16:45.27 | Hanoon | devaf: Would you need any more details? |
| 16:46.09 | ponyofdeath | hi, guys I am not getting any ring tones or hear any of the .slin sounds played on my asterisk setup? any ideas? do i need to load a special module for those. i have the indications.conf set up. |
| 16:49.33 | [TK]D-Fender | ponyofdeath, show us the call and escribe everything about the chain betweent he device & * |
| 16:49.43 | [TK]D-Fender | ~pb |
| 16:49.43 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:49.45 | [TK]D-Fender | ^^^ |
| 16:50.27 | [TK]D-Fender | Also, indications are generated, not "played back". So there should be no relation to static sound files not working |
| 16:50.54 | [TK]D-Fender | Id sooner believe a general audio issue like NAT & reinvites is involved |
| 16:53.52 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 16:54.27 | Hanoon | @devaf Are you there? Do you need any more details? |
| 16:54.28 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:54.49 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ygdldrfmwsdfsyle) |
| 16:55.11 | ponyofdeath | [TK]D-Fender: ok http://pastebin.com/kRFMfVVj |
| 16:56.21 | ponyofdeath | [TK]D-Fender: i have the iptables modules loaded for sip |
| 16:56.30 | [TK]D-Fender | Don't |
| 16:56.42 | WIMPy | ponyofdeath: Which one? |
| 16:56.49 | [TK]D-Fender | And that output does nto contain a call nor do we have any details about what you're trying this with |
| 16:57.08 | WIMPy | pf_conntrack_sip is usefull, pf_nat_sip will most certainly cause trouble. |
| 16:57.47 | ponyofdeath | nf_nat_sip nf_conntrack_sip |
| 16:57.54 | WIMPy | s/fp/nf/g |
| 16:58.07 | ponyofdeath | [TK]D-Fender: let me do a call this is with my cell to my voip provided number |
| 16:58.23 | WIMPy | Make sure you never load nf_nat_sip again. |
| 16:59.16 | ponyofdeath | [TK]D-Fender: http://pastebin.com/sbuHS1B8 for an call |
| 17:01.06 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 17:01.08 | [TK]D-Fender | ponyofdeath, I don't see an answer... or SIP debug in there to see if your setup looks at all sane |
| 17:01.12 | ocx | hello |
| 17:01.42 | ocx | do i need to reload zaptel modules once the /etc/asterisk/zapata.conf is modified? |
| 17:01.59 | ocx | the loaded configuration seems not taking new effects |
| 17:01.59 | anonymouz666 | no |
| 17:02.13 | anonymouz666 | some configuration needs to reload the chan_zap in your case |
| 17:02.22 | anonymouz666 | using 'restart now' |
| 17:02.27 | ocx | you mean chan_zap reload? |
| 17:02.28 | WIMPy | ocx: You need to upgrade. But reloading te channel is enough. |
| 17:02.30 | ocx | from asterisk cli ? |
| 17:02.35 | ponyofdeath | [TK]D-Fender: how do i do an sip debug |
| 17:02.50 | [TK]D-Fender | "sip set debug on" |
| 17:02.56 | ponyofdeath | [TK]D-Fender: well no one to answer right now and i dont hear an ring tone during the ringing |
| 17:03.03 | [TK]D-Fender | ponyofdeath, And remove that module as suggested |
| 17:03.11 | ponyofdeath | [TK]D-Fender: already removed |
| 17:03.48 | *** join/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com) |
| 17:04.17 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 17:04.21 | [TK]D-Fender | ponyofdeath, In SIP indications are your phone's job. there is no forced audio indication there |
| 17:05.04 | ponyofdeath | [TK]D-Fender: dont know what ur asking |
| 17:05.09 | ponyofdeath | [TK]D-Fender: in the indications.conf? |
| 17:07.16 | [TK]D-Fender | ponyofdeath, No. |
| 17:07.47 | [TK]D-Fender | ponyofdeath, this is SIP end to end, * has not apparently answered the call therefor it is up to each end to generate their own ringing |
| 17:09.10 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:09.42 | ponyofdeath | weird i had this setup running with debian asterisk 1.6 and now i switched to gentoo 1.8 and im getting these issues |
| 17:09.54 | [TK]D-Fender | Show the call. |
| 17:10.04 | ponyofdeath | [TK]D-Fender: ok doing the sip debug now |
| 17:10.21 | ponyofdeath | will call in a few on a conf call sorry |
| 17:10.29 | wcselby | ponyofdeath you could try adding an Answer() or Progress() before your Dial() statement, sometimes that helps with issues like you describe |
| 17:10.42 | ponyofdeath | wcselby: ok thanks |
| 17:11.14 | ponyofdeath | wcselby: i do have an 1,Answer |
| 17:12.16 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 17:12.54 | Hanoon | I have a fresh install on 1.8.7.2 on Centos 6.0 - Everything works except that the RTPTIMEOUT=X and RTPHOLDTIMEOUT=X will not work , I have looked and found the patch 18559 in the bug tracker, Applied patch redone make and make install - problem still exist - Asterisk will not hangup call when there is not RTP. |
| 17:15.50 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:15.51 | ponyofdeath | [TK]D-Fender: http://pastebin.com/peVbcPQb |
| 17:17.03 | ocx | i have configured in zapata.conf my FXO to be associated with context=incoming |
| 17:17.10 | ocx | and FXS to be with context phones |
| 17:17.18 | ocx | did module reload zap_chan |
| 17:17.27 | ocx | but i am getting an error when dialing into my FXO |
| 17:17.30 | ocx | Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 17:17.31 | devaf | when i try to run the asterisk-src-dir/contrib/scripts/ast_tls_cert , i am gettng error "hostname: Name or service not known" |
| 17:17.35 | ocx | this error |
| 17:17.38 | wcselby | ocx have you restarted asterisk and zaptel? |
| 17:17.41 | ocx | yes |
| 17:17.47 | ocx | not asterisk no |
| 17:17.53 | ocx | i only restarted the module zap_chan |
| 17:17.55 | wcselby | i mean |
| 17:17.59 | wcselby | service asterisk stop |
| 17:17.59 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-rqnmecjmuqfzurow) |
| 17:17.59 | WIMPy | ocx: add an exten => s,1,WaitExten(20) or something. |
| 17:18.02 | wcselby | service zaptel stop |
| 17:18.07 | wcselby | then start zap, then asterisk |
| 17:18.26 | ocx | i am actually working on a mini-os |
| 17:18.28 | wcselby | with zaptel I remember a lot of changes required a full restart of asterisk / zaptel |
| 17:18.31 | ocx | linux based |
| 17:18.40 | ocx | i dont have the "service" installed |
| 17:18.43 | WIMPy | Or turn off immediate, if you don;t mind waiting for timeouts. |
| 17:19.05 | wcselby | ocx well do whatever is the appropriate command for your os to accomplish the same goal |
| 17:19.18 | wcselby | /etc/init.d/rc.d/asterisk stop maybe? |
| 17:19.26 | wcselby | or /etc/init.d/rc.d/zaptel stop ? |
| 17:19.29 | ocx | i will try ti figure it out |
| 17:19.41 | wcselby | or just reboot the box, that should restart everything :) |
| 17:19.41 | [TK]D-Fender | ponyofdeath, Ok, add the ",r" dial option and retest |
| 17:19.42 | ocx | you mean i restart asteris kand zaptel after reloading the modules from asterisk cli? |
| 17:20.03 | wcselby | ocx I've found with zaptel some changes required a full restart of both asterisk and zaptel |
| 17:20.03 | ocx | wcselby: tried to reboot the box but the config stayed the same :) |
| 17:20.10 | ocx | ok |
| 17:20.19 | ocx | i will give it a shot |
| 17:20.24 | [TK]D-Fender | ocx, You have a faily clear dialplan error according to what you pasted |
| 17:20.45 | ocx | if i set the context in zapatel to default it works fine |
| 17:20.48 | ocx | but i dont want to use default |
| 17:20.54 | wcselby | [TK]D-Fender the error says the call is dumping into the default context, but he says he's made the context incoming, not default |
| 17:20.55 | ocx | i want to use "incoming" context for FXO |
| 17:21.00 | wcselby | so the error probably lies more with his config |
| 17:21.10 | wcselby | ocx can you show us your zaptel config? |
| 17:21.15 | ocx | sure |
| 17:21.22 | wcselby | you may have the context line in the wrong place |
| 17:21.25 | [TK]D-Fender | That would be nice... |
| 17:21.27 | [TK]D-Fender | ~pb |
| 17:21.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:21.28 | [TK]D-Fender | ^^^ |
| 17:22.01 | ponyofdeath | [TK]D-Fender: same no ringtone |
| 17:22.19 | ponyofdeath | [TK]D-Fender: Executing [7342226082@default:2] Dial("SIP/provider-00000000", "SIP/ares-pap&SIP/milko-lt,35,r") in new stack |
| 17:22.42 | [TK]D-Fender | ponyofdeath, Ok, do a "playback(silence/1)" between your answer and your Dial |
| 17:22.50 | ocx | http://pastecode.com/iG |
| 17:23.19 | ocx | seems it pasted incorrectly :) |
| 17:23.22 | ocx | let me repaste |
| 17:23.34 | [TK]D-Fender | First one looks clearly wrong |
| 17:23.38 | ponyofdeath | [TK]D-Fender: exten => 7342226082,2,Playback(silence/1) |
| 17:23.56 | [TK]D-Fender | ponyofdeath, Yes, something to set up the audio. |
| 17:23.58 | ocx | http://pastecode.com/iH |
| 17:24.01 | ocx | here you go |
| 17:24.03 | ocx | this is zapatel.conf |
| 17:24.13 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:24.16 | ponyofdeath | [TK]D-Fender: error No application 'Playback' for extension (default, 7342226082, 2) |
| 17:24.50 | [TK]D-Fender | ocx, Why do we see the contents twice, and differently? |
| 17:24.55 | wcselby | ocx why so much duplication in the conf? |
| 17:25.14 | ocx | it seems it pasted incorrectly again |
| 17:25.15 | ocx | 1 sec please |
| 17:25.19 | WIMPy | wcselby: It's the same file twice. |
| 17:25.28 | [TK]D-Fender | No, they are not the same |
| 17:25.31 | ocx | something wrong with that pastebin |
| 17:25.32 | wcselby | WIMPy but it's nto the same |
| 17:25.33 | WIMPy | But it looks ok. |
| 17:25.35 | [TK]D-Fender | neither of those 2 PB's |
| 17:26.18 | [TK]D-Fender | ponyofdeath, If you don't even have the playback app you would appear to be running a pretty messed up install |
| 17:26.59 | WIMPy | No, right. They are different. |
| 17:27.28 | ocx | http://bin.cakephp.org/view/1539958827 |
| 17:27.32 | ocx | this is the correct one |
| 17:27.33 | WIMPy | But neither version explains any default context being used. |
| 17:27.35 | ponyofdeath | [TK]D-Fender: ok got that loaded and now its ringging |
| 17:27.50 | ponyofdeath | [TK]D-Fender: thanks for your help! |
| 17:31.31 | [TK]D-Fender | ponyofdeath, ok.... |
| 17:31.32 | ocx | if i change the context=incoming into context=default the dialplan hits correctly |
| 17:31.32 | [TK]D-Fender | You're welcome... |
| 17:31.32 | [TK]D-Fender | ocx, we haven't seen a clear pictute of your config yet. |
| 17:31.32 | [TK]D-Fender | picture* |
| 17:31.32 | ocx | http://bin.cakephp.org/view/1539958827 this is the zapata |
| 17:31.32 | ocx | i will paste the dialplan |
| 17:31.33 | ocx | extensions.conf |
| 17:31.33 | WIMPy | ocx: Ok, so it does go to "phones"? Did you see my message where I told you what to add to your dialplan? |
| 17:31.33 | [TK]D-Fender | ocx, Now show us your reload chan_zap |
| 17:31.33 | ocx | ok |
| 17:31.33 | [TK]D-Fender | ocx, and a call that follows |
| 17:31.37 | ocx | http://bin.cakephp.org/view/1386615559 |
| 17:31.37 | Hanoon | update, Works find when it's not locally bridging... |
| 17:31.37 | ocx | this is the reload and the call that follows |
| 17:32.47 | WIMPy | ocx: Try to unload and load chan_zap or restart Asterisk. |
| 17:33.04 | wcselby | ocx what version of asterisk / zaptel are you using? |
| 17:33.09 | wcselby | and why are you using such old versions? |
| 17:33.18 | ocx | 1.4.21.2 |
| 17:33.41 | ocx | it came with the box |
| 17:33.49 | Hanoon | Can Anyone help please... |
| 17:33.58 | wcselby | Hanoon what's the issue? |
| 17:34.18 | [TK]D-Fender | ocx, I see what looks like USER.CONF junk in there <- |
| 17:34.20 | Hanoon | I have a fresh install on 1.8.7.2 on Centos 6.0 - Everything works except that the RTPTIMEOUT=X and RTPHOLDTIMEOUT=X will not work , I have looked and found the patch 18559 in the bug tracker, Applied patch redone make and make install - problem still exist - Asterisk will not hangup call when there is not RTP traffic. |
| 17:34.27 | wcselby | [TK]D-Fender I was thinking the same thing |
| 17:34.34 | [TK]D-Fender | And I now recognize that embedded box |
| 17:34.52 | Hanoon | Works fine when I enable transcoding and not locally bridging. |
| 17:35.19 | ocx | user.conf ? |
| 17:35.25 | [TK]D-Fender | ocx, "ls -la /etc/asterisk" |
| 17:35.32 | [TK]D-Fender | ocx, yes, typically used by asterisk-gui |
| 17:35.38 | ocx | yea |
| 17:43.43 | pdtpatrick1 | what's the different between GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavailable) and GotoIf($[${DIALSTATUS} = BUSY]?busy:unavailable) |
| 17:43.58 | pdtpatrick1 | or are quotes just more efficient in case there's a space between words? |
| 17:44.03 | pdtpatrick1 | sorta like bash code ? |
| 17:46.52 | [TK]D-Fender | pdtpatrickNo, in case a variable is completely blank and * bitches because there is nothing on one side of the "+" |
| 17:46.57 | [TK]D-Fender | "=" |
| 17:47.10 | pdtpatrick1 | oh |
| 17:47.24 | pdtpatrick1 | so with "" asterisk is less bitchy? cool deal. |
| 17:47.33 | WIMPy | So just like in bash, yes. |
| 17:47.35 | [TK]D-Fender | GotoIf($[junk${DIALSTATUS} = junkBUSY]? |
| 17:47.38 | [TK]D-Fender | just as valid |
| 17:48.00 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 17:48.03 | ocx | sorry dc... |
| 17:48.11 | ocx | did i miss any comments? |
| 17:48.28 | WIMPy | no |
| 17:48.41 | ocx | what is the problem with this context ? :/ |
| 17:48.43 | wcselby | ocx except the one that completely solves your issues |
| 17:48.46 | wcselby | but you missed it |
| 17:48.48 | wcselby | so sorry |
| 17:48.50 | ocx | which is |
| 17:48.53 | ocx | :) |
| 17:49.00 | wcselby | :) |
| 17:49.44 | ocx | btw ls -al points to /persistent |
| 17:49.50 | ocx | it seems like a symbolic link |
| 17:50.19 | wcselby | Hanoon sorry I don't have any advice :( |
| 17:50.53 | Hanoon | Is there is any suggestions for how to get help with the issue I'm having? Maybe another channel.. |
| 17:50.58 | ocx | wcselby: really did you find anything syspecious in configration? |
| 17:52.32 | [TK]D-Fender | Hanoon, post it on the tracker |
| 17:53.08 | Qwell | Hanoon: Is RTP even going through Asterisk? |
| 17:54.26 | Hanoon | yes - works 2 way without a problem through Asterisk |
| 17:54.58 | Qwell | if RTP isn't going through Asterisk, how would Asterisk know that RTP stopped? |
| 17:55.41 | Hanoon | RTP going through asterisk |
| 17:55.58 | Qwell | You said it worked if RTP was going through Asterisk |
| 17:56.43 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 17:59.25 | Hanoon | both ways Asterisk is handling RTP, It only works when I make Asterisk transcode between legs. |
| 18:00.59 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 18:01.26 | Hanoon | I do this by forcing alaw on the phone and allow asterisk to ulaw then alaw |
| 18:01.58 | Qwell | How did you verify that RTP was going through Asterisk? |
| 18:02.01 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
| 18:04.08 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-zcxetlawsxjihlny) |
| 18:06.57 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-gdnrxexkfnbhuzam) |
| 18:08.32 | pdtpatrick1 | What is the command to lookup something like Verbose and how it should be used? I thought it would be core show application Verbose. But that does not seem right |
| 18:10.10 | wcselby | core show application verbose would be correct, at least in 1.6+ |
| 18:10.26 | wcselby | to see a list of all apps loaded, you use "core show applications" |
| 18:10.32 | wcselby | same with functions and specific functions |
| 18:11.35 | pdtpatrick1 | nevermind my eyes deceived me .. i typed ? .. and i thought the list was alphabetical so i was looking for V towards the end.. it was mixed in there somewhere else |
| 18:12.05 | wcselby | you can also type core show application v then hit the tab key to see all the apps that start with V |
| 18:12.12 | Hanoon | Sip trace RTP packets, |
| 18:13.01 | pdtpatrick1 | wcselby, even better.. thanks :) |
| 18:13.37 | Qwell | Hanoon: Do you see packets in Asterisk if you enable rtp debug? |
| 18:15.21 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:18.14 | afink | Hello everyone, I am having some trouble getting cdr mysql to work on my asterisk install. I used an AsteriskNow cd to install asterisk 1.6. It seems that asterisk is having trouble loading the mysql modules. This is the error I am getting: cdr_addon_mysql.c:418 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf |
| 18:18.32 | Qwell | afink: Don't use cdr_mysql |
| 18:18.38 | Qwell | cdr_odbc is far better |
| 18:20.24 | afink | Qwell: I don't think that will work with cdr-stats |
| 18:20.39 | Qwell | Why not? It doesn't care how the data gets there. |
| 18:21.15 | afink | I was just about to say I may be wrong. Its been known to happen |
| 18:21.20 | *** join/#asterisk cyford (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 18:24.12 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:24.40 | Hanoon | Qwell: Yes just tested ... |
| 18:26.04 | *** join/#asterisk flyaster (~flyaster@94.154.210.188) |
| 18:26.32 | wcselby | afink what does your /etc/asterisk/cdr_mysql.conf file look like? |
| 18:27.00 | wcselby | use pastebin to show us |
| 18:27.01 | wcselby | ~pb |
| 18:27.01 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 18:27.53 | afink | wcselby: non-existent...the config file should be created when asterisk is restarted with the module installed correct? |
| 18:28.07 | wcselby | afink no why would you think that? |
| 18:28.10 | Qwell | Asterisk doesn't create config files.. |
| 18:28.55 | wcselby | afink the most you can hope for is a sample config file being copies in place after you use the command "make samples" |
| 18:29.03 | wcselby | after installing the module from asteirsk-addons |
| 18:29.07 | Qwell | He's using FreePBX. |
| 18:29.10 | Qwell | just fyi |
| 18:29.33 | afink | Qwell: I am not using FreePBX and I am semi offended by that remark |
| 18:29.35 | wcselby | he still has to install the module from asterisk-addons |
| 18:30.07 | Qwell | cdr-stats isn't part of FreePBX? |
| 18:30.10 | [TK]D-Fender | <PROTECTED> |
| 18:30.36 | wcselby | cdr-stats is not the tool I prefer for reviewing cdrs |
| 18:30.45 | [TK]D-Fender | afink, And no, * never makes configs that don';t exist |
| 18:30.51 | wcselby | but then again, I just use manual sql queries, so I guess I'm not one to talk :) |
| 18:30.56 | [TK]D-Fender | afink, tarballs come with samples you have to make |
| 18:31.13 | wcselby | [TK]D-Fender you don't have to make the samples, you can go in and copy them out one by one if you like |
| 18:31.25 | [TK]D-Fender | wcselby, Well.. if you want to do it the hard way.. sure |
| 18:31.29 | wcselby | they're in the /configs/ folder of the tarballs :) |
| 18:34.14 | afink | ok thanks for the help, created cdr_mysql.conf pretty sure I'm all good now |
| 18:35.36 | afink | yep working great thanks everyone... always bitch slapped with knowledge in this channel |
| 18:35.49 | Qwell | I still say you did it wrong. |
| 18:35.51 | Qwell | but w/e |
| 18:36.10 | wcselby | how? because he used cdr_mysql instead of cdr_odbc? |
| 18:36.14 | Qwell | yes |
| 18:36.23 | Qwell | cdr_mysql should die in a fire. It's horrible. |
| 18:36.49 | wcselby | it's never caused me any issues across the many systems I've installed it on |
| 18:37.14 | Qwell | There's a reason it's deprecated. |
| 18:37.27 | wcselby | which version did it get deprecated in? |
| 18:37.32 | Qwell | 1.0? |
| 18:37.49 | Qwell | really, it never should've existed |
| 18:38.02 | wcselby | cdr_mysql has been a part of asterisk addons at least through 1.6.2 |
| 18:38.09 | wcselby | or is that what you meant, not part of the main? |
| 18:38.13 | Qwell | that doesn't mean it should be used. :) |
| 18:38.23 | wcselby | but it's also not what I would call deprecated |
| 18:38.32 | wcselby | since it's still maintained and updated along with each version release |
| 18:38.41 | Qwell | We maintain lots of deprecated stuff.. |
| 18:47.28 | *** join/#asterisk kotis_ (~kotis@72.253.138.39) |
| 18:48.45 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 18:53.15 | [TK]D-Fender | Qwell, You just can't quit them ;) |
| 18:58.01 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 18:58.20 | *** join/#asterisk vetal (~chatzilla@20-143-134-95.pool.ukrtel.net) |
| 19:01.55 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:03.11 | vetal | Hi, I use to get packets from channel by CHANNEL(rtpqos,audio,local_count) - but since several upgrade of asterisk, this function begin to give me only 0 counters |
| 19:04.41 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 19:11.01 | pdtpatrick1 | Question .. i have a dialplan like so |
| 19:11.02 | pdtpatrick1 | http://paste.pound-python.org/show/15483/ |
| 19:11.05 | pdtpatrick1 | works fine |
| 19:11.12 | pdtpatrick1 | im using a softphone |
| 19:11.27 | pdtpatrick1 | it would call and then after completing everything |
| 19:11.33 | pdtpatrick1 | it says call failed |
| 19:11.54 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-hzbvdczwpqyrkptu) |
| 19:11.55 | pdtpatrick1 | cli is fine |
| 19:11.56 | pdtpatrick1 | http://paste.pound-python.org/show/15484/ |
| 19:12.04 | WIMPy | Sure. |
| 19:12.08 | WIMPy | You never answered it. |
| 19:12.14 | pdtpatrick1 | aww damn |
| 19:12.49 | WIMPy | Hmm. NFI if SayNumber would do it for you. |
| 19:13.02 | pdtpatrick1 | NFI = ? |
| 19:13.10 | pdtpatrick1 | im not all caught up on my abbreviations |
| 19:13.16 | WIMPy | No F***ing Idea |
| 19:13.30 | pdtpatrick1 | haha nice. THat's a new one i haven't heard |
| 19:15.56 | vetal | any ideas about CHANNEL(rtpqos,audio,local_count) ? |
| 19:32.14 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 19:40.57 | vetal | One more question, how can i analazy who finish call caller or colling part? |
| 19:43.03 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wacngbgjjzolckpd) |
| 19:44.10 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 19:44.16 | wcselby | aren't the caller and the calling party the same person? |
| 19:44.31 | sereal-work | man is there not some decent documentation on a2billing ? It seems like it's all over the place. |
| 19:44.43 | sereal-work | http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Documentation is useless |
| 19:44.48 | wcselby | sereal-work yeah it is. i eventually got it to work, let me see if I can find the right link I used |
| 19:45.33 | sereal-work | thanks wcselby |
| 19:48.52 | vetal | I mean A and B part |
| 19:51.43 | wcselby | sereal-work sorry I couldn't find it, and it's time for me to jet. i'll look for it and if I can't find anything I'll write something up in the next day or two on how I did mine |
| 20:01.48 | Qwell | shido6: nat/yahoo/ ? |
| 20:02.06 | p3nguin | Good old San Jose. |
| 20:04.08 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 20:05.11 | *** join/#asterisk vetal (~chatzilla@70-33-134-95.pool.ukrtel.net) |
| 20:12.54 | fprior | Hi, how about recording all calls: I must use MixMonitor before ever Dial() or exists another option ? |
| 20:13.54 | afink | fprior: thats what I do |
| 20:15.36 | [TK]D-Fender | fprior, read features.conf (automon), and the other apps must be called before dial (monitor / mixmonitor) |
| 20:18.25 | pdtpatrick1 | when you use core show application <something> .. how can u make it so asterisk doesn't just dump everything all at once? Is there a pipe to something similar to less ? |
| 20:19.04 | r0m|u | when asterisk starts how many asterisk process should spawn up? |
| 20:21.27 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-cybmsgvyzieoofzh) |
| 20:22.03 | jrrose | As far as I can recall, it should just be one process. |
| 20:23.25 | fprior | [TK]D-Fender, I'm reading on Definitive Guide about automixmon but, what happens is the caller press *3 (for example), he can terminate the recording ? |
| 20:23.26 | jrrose | I'm pretty sure it should just be one process. |
| 20:25.09 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 20:32.02 | [TK]D-Fender | fprior, Doubt it |
| 20:33.36 | anonymouz666 | jrrose: it would be nice if we have lots of children processes |
| 20:35.05 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 20:35.09 | patrickod | I'm trying to load test my asterisk instance with sipp |
| 20:35.24 | patrickod | 30 calls with music on hold is causing very high CPU usage |
| 20:35.34 | patrickod | what's the best way to figure out what's causing the high load ? |
| 20:36.01 | anonymouz666 | you know better than anybody since we don't know what is happening in your dialplan. |
| 20:38.40 | patrickod | hmm ok. I'll go looking so |
| 20:45.16 | [TK]D-Fender | patrickod, If all they are doing is MoH, then that is it. |
| 20:45.33 | [TK]D-Fender | patrickod, Consider the codec of the call and what's being used for MoH |
| 20:51.10 | *** join/#asterisk macroevolve (macroevolv@c-98-234-125-202.hsd1.ca.comcast.net) |
| 20:51.40 | macroevolve | I currently have two 2U servers colocated. I am planning on bringing them home. I was wondering if anyone happens to know if I can set both these servers up using my Comcast home internet connection (one server will be redundancy for the other server)? This is just for development. After development, when we are going commercial, it'll be colocated again.. Was wondering whether anyone |
| 20:51.41 | macroevolve | knows if this can be done? |
| 21:09.57 | *** join/#asterisk ocx (b2874709@gateway/web/freenode/ip.178.135.71.9) |
| 21:14.54 | WIMPy | macroevolve: Are you sure you want to get involved in that if you have to ask that question? |
| 21:16.40 | ocx | something really weird is happening... when i set context in zapata.conf and restart the zaptel and asterisk , the lines of contexts added are reordered in a different , ambigious way |
| 21:17.16 | ocx | like context=x and below it directly context=y |
| 21:17.26 | ocx | which makes no sense |
| 21:17.56 | WIMPy | ocx: Configure whatever configures your Asterisk or remove it so you can. |
| 21:18.39 | ocx | you think its the 3rd party trhing playing some games? |
| 21:19.02 | WIMPy | If it isn't your other self... |
| 21:19.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:20.33 | Guest708 | I been having major issues with asterisk for a few days now. out of no where I beenhaving this choppines on my lines. its only when the person talks ot me that the choppiness starts. They can hear me just fine but I cant heare them. |
| 21:21.16 | Guest708 | s/I cant heare them./I cant heare them very well./ |
| 21:21.23 | ocx | :) |
| 21:21.42 | r0m|u | that was me by the way |
| 21:21.49 | r0m|u | I have no clue why I was as guest |
| 21:21.56 | WIMPy | r0m|u: What connection? |
| 21:22.05 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 21:22.13 | ocx | the challenge is to install asterisk on uClinux |
| 21:22.22 | r0m|u | WIMPy: voip.ms |
| 21:22.25 | WIMPy | Maybe you need to share your porn collection with your neighbours so they download less? |
| 21:22.30 | r0m|u | comcast |
| 21:22.59 | r0m|u | ocx: a challenge indeed. I just finish making my own build |
| 21:23.11 | ocx | on uCLinux? |
| 21:23.16 | r0m|u | ocx yes |
| 21:23.24 | ocx | can you share with me? |
| 21:23.34 | ocx | which version you got running? |
| 21:23.42 | WIMPy | Something I need to look in to, as well. |
| 21:25.13 | r0m|u | ocx I used a the source from astlinux and modified it. |
| 21:26.13 | ocx | on which hardware did you get it running? |
| 21:26.32 | r0m|u | alix |
| 21:26.53 | r0m|u | I strip pretty much everything of their src and mademy own |
| 21:26.59 | r0m|u | *off* |
| 21:27.22 | ocx | i am using atcom ip02 |
| 21:27.29 | ocx | they got their software installed |
| 21:27.45 | ocx | i want to have a clean uClinux sith a fresh asterisk software on top of it |
| 21:27.49 | ocx | so its quite a challenge |
| 21:29.04 | r0m|u | It shouldnt bee complicated. here is an easy way at it. |
| 21:29.06 | r0m|u | http://buildroot.uclibc.org/ |
| 21:29.38 | r0m|u | http://crosstool-ng.org/ <--- for cross compiling. |
| 21:30.35 | r0m|u | so far I have been getting good results. my only issues where ast itself which is some what well obscure and broken..... (No comments :P ) |
| 21:31.30 | WIMPy | It barfed on ncurses for me. Not sure if they are actually needed at all. |
| 21:32.16 | r0m|u | WIMPy: its ncurses base. |
| 21:32.31 | r0m|u | s/its/it's/ |
| 21:32.42 | WIMPy | The build. |
| 21:32.51 | r0m|u | astlinux? |
| 21:33.06 | WIMPy | The uclinux |
| 21:33.09 | *** join/#asterisk kovalyshyn (~kovalyshy@88.155.72.104) |
| 21:33.29 | r0m|u | ooo yea for the menuconfig |
| 21:33.59 | WIMPy | Yes, but I don't need that on the uclinux system. |
| 21:34.09 | r0m|u | Indeed. |
| 21:34.33 | WIMPy | So I guess, I could just disable it somewhere. |
| 21:34.36 | r0m|u | It should only be for the make. not for the build. |
| 21:34.40 | r0m|u | .config |
| 21:35.23 | *** part/#asterisk kovalyshyn (~kovalyshy@88.155.72.104) |
| 21:35.52 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:41.34 | r0m|u | WIMPy: found my issue |
| 21:41.54 | r0m|u | my upload is at .78Mbps.... what ever that means |
| 21:42.01 | r0m|u | is shot to shits |
| 21:42.27 | r0m|u | that would explain me hearing the choppines |
| 21:47.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:47.24 | *** join/#asterisk pdtpatrick_ (~pdtpatric@12.249.4.226) |
| 21:49.25 | pdtpatrick_ | Question .. if u are using parked calls (features.conf) .. after adding tT to your Dial app .. do u have to use kK in order for the calls to be parked? I think the default is #72 to park a call |
| 21:52.46 | leifmadsen | you don't need either |
| 21:53.23 | leifmadsen | actually I guess you're not using SIP if you're using tT for transfers |
| 21:53.26 | pdtpatrick_ | ? i thought u had to include => parkedcalls .. and then in Dial(Technology,timeout, option) .. u had to have tT as option |
| 21:53.39 | pdtpatrick_ | with sip it should automatically work ? |
| 21:53.45 | pdtpatrick_ | and with PRI u have to use that right? |
| 21:53.54 | pdtpatrick_ | at least that's what i understood |
| 21:54.03 | leifmadsen | pdtpatrick: with SIP, you just use a SIP transfer, not DTMF enabled transfers |
| 21:54.18 | pdtpatrick_ | okay |
| 21:54.19 | leifmadsen | in which case, you just do an attended transfer to extension 700 by default when you include parkedcalls |
| 21:54.27 | pdtpatrick_ | oh nice |
| 21:54.36 | pdtpatrick_ | but for the PRI u have to include tT |
| 21:54.37 | leifmadsen | the options you mention are only for DTMF based transfers |
| 21:54.38 | pdtpatrick_ | right? |
| 21:54.42 | pdtpatrick_ | okay cool |
| 21:55.12 | leifmadsen | you only need k and/or K to park calls based on the description in dial |
| 21:55.25 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:55.28 | leifmadsen | t and/or T is for transfers other than parking |
| 21:55.44 | pdtpatrick_ | oh okay |
| 21:55.45 | pdtpatrick_ | thanks again |
| 21:56.48 | pdtpatrick_ | leifmadsen: this is what i was looking at |
| 21:56.49 | pdtpatrick_ | http://cl.ly/0v2x1Q2s2k3m2g1g171m |
| 21:57.05 | [TK]D-Fender | <PROTECTED> |
| 21:57.14 | leifmadsen | pdtpatrick: ok... I just looked at the documentation that comes with Asterisk |
| 21:58.07 | pdtpatrick_ | [TK]D-Fender: correct. |
| 21:58.24 | [TK]D-Fender | Very odd. |
| 21:58.37 | pdtpatrick_ | ? |
| 21:58.38 | WIMPy | You could have a PBX on a PRI. |
| 21:58.47 | [TK]D-Fender | pdtpatrick: What are you doing with your system exactly? |
| 21:59.00 | pdtpatrick_ | quemetrics |
| 21:59.07 | pdtpatrick_ | call center to be exact |
| 21:59.22 | pdtpatrick_ | i just want the agents to be able to park a call and pick it up somewhere else |
| 21:59.29 | pdtpatrick_ | and since we are using a PRI for that particular server |
| 21:59.36 | [TK]D-Fender | Your agents are vai PRI? |
| 21:59.40 | pdtpatrick_ | i figured (based on what i've read) that you need the tT extension |
| 21:59.41 | [TK]D-Fender | via* |
| 22:00.03 | [TK]D-Fender | and "Tt" isn't an extension |
| 22:00.08 | leifmadsen | pdtpatrick: how are your agents connected to asterisk? |
| 22:00.25 | pdtpatrick_ | sorry didn't mean extension .. i meant i added that as an option |
| 22:00.34 | leifmadsen | why? |
| 22:00.47 | leifmadsen | how are your agents connected to asterisk? |
| 22:01.19 | pdtpatrick_ | i guess we are registering via sip |
| 22:01.24 | leifmadsen | per 7 minutes ago |
| 22:01.24 | [TK]D-Fender | With what? |
| 22:01.25 | leifmadsen | <leifmadsen> pdtpatrick: with SIP, you just use a SIP transfer, not DTMF enabled transfers |
| 22:01.28 | [TK]D-Fender | "guess"? |
| 22:01.35 | leifmadsen | <leifmadsen> in which case, you just do an attended transfer to extension 700 by default when you include parkedcalls |
| 22:01.44 | pdtpatrick_ | im just taking over this server and im still yet to understand what their admin did |
| 22:01.52 | leifmadsen | then I suggest not making changes |
| 22:02.19 | [TK]D-Fender | pdtpatrickIf you don't know what your agents are using for phones I've worried for your ability to administer your won system... |
| 22:02.21 | [TK]D-Fender | own* |
| 22:02.37 | pdtpatrick_ | they are registering via sip |
| 22:02.42 | [TK]D-Fender | this looks like 50% blind |
| 22:02.48 | [TK]D-Fender | pdtpatrickAnd what are they using? |
| 22:02.57 | pdtpatrick_ | Cisco IP phones |
| 22:03.34 | [TK]D-Fender | Those have transfer buttons. You don't need DTMF transfers for anything |
| 22:04.23 | leifmadsen | taps the microphone.... test |
| 22:05.11 | pdtpatrick_ | Thanks |
| 22:05.38 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 22:06.25 | [TK]D-Fender | leifmadsen: Yeah, I'll pass you back a portion of credit, but you want to make sure they're nt using X-Lite or other similarly crippled SIP app/phone without a real transfer option :) |
| 22:07.02 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:07.33 | pdtpatrick_ | ouch .. sheesh |
| 22:07.43 | pdtpatrick_ | thanks anyway :) |
| 22:09.21 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:14.44 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 22:15.00 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 22:16.58 | IsUp | hello |
| 22:20.07 | *** join/#asterisk Crowb4r (u1012@gateway/web/irccloud.com/x-oczbldsneaqwqezm) |
| 22:26.31 | IsUp | what causes a "OPTIONS -> 404 not found" between peer and me? its happening every minute i think. im sending OPTIONS, peer returns me back "404 not found". does it caused by qualify=yes ? |
| 22:29.43 | WIMPy | yes |
| 22:30.10 | IsUp | is there any way to get a proper "200 OK" from peer? i can talk to my telco :p |
| 22:31.38 | [TK]D-Fender | IsUp: * always responds 404. this is normal |
| 22:32.14 | WIMPy | As long as you get an answer, you're ok, It doesn't matter what it is. |
| 22:32.36 | IsUp | [TK]D-Fender: um it looks like im sending OPTIONS, right? i can pastebin if you want. |
| 22:32.49 | IsUp | WIMPy: okay |
| 22:33.04 | [TK]D-Fender | * doesn't care what the answer is, and doesn't care what it answers. |
| 22:33.14 | [TK]D-Fender | It's used as a dumb keep-alive, nothing more |
| 22:34.02 | IsUp | okay, understood and what causes a "481 Call leg/transaction does not exist" from a peer? lets say i have 1k calls, im just having this on 10 calls. |
| 22:34.50 | IsUp | when i trace calls, i mean im matching caller id with Call-ID, channels looks "not there", ive already found that problem caused by "Session Timer" |
| 22:35.14 | IsUp | session timer is set from my provider, it's 1800secs and it says 'refresher uac', so i have to refresh session timer as i understand |
| 22:35.35 | IsUp | but im running 1.4, does it supports session timer? should i ask to my telco for remove their session timer? |
| 22:36.17 | [TK]D-Fender | IsUp: No, 1.8 does though |
| 22:38.36 | IsUp | [TK]D-Fender: to be honest, i was unaware this problem. and it caused me big problems. my all CDRs are wrong, it's financially messing up my system |
| 22:39.20 | IsUp | [TK]D-Fender: ive already have 3600secs max call time on my numbers, caller was going to talk 45 minutes, but session timer just lets 30 minutes. after 30 mins, call is dropped but it's closing on my side after 1 hour (on my max call time) |
| 22:45.11 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:48.06 | IsUp | so ive set my max call time to 29.5 minutes for resolve this problem temporary |
| 22:48.19 | IsUp | now it looks good but caller is cant talk over 30 mins. |
| 22:51.30 | IsUp | i have call monitor from both call legs, caller & agent. i am working on this issue, listening call records, looking logs, SIP capture. even i m using soundforge to detect voices after 30 mins to see if theres any silence or not from caller side. so im all stuck. |
| 22:52.51 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:54.19 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yckuzfuvyghnossm) |
| 22:57.49 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 23:02.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:04.09 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 23:04.24 | F2Knight | Q: Is Dahdi 2.5 stable? |
| 23:05.54 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 23:08.53 | [TK]D-Fender | Yes |
| 23:13.09 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:14.34 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:15.19 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 23:18.22 | FLeiXiuS | In AMI is it possible to get the state of a channel, whether the remote end answered/hungup? |
| 23:18.52 | [TK]D-Fender | You can see if it is currently bridged... |
| 23:19.04 | [TK]D-Fender | And you can access channelvars |
| 23:19.10 | WIMPy | And you get events when th state changes. |
| 23:20.13 | FLeiXiuS | I see that there are Hangup events, but I was wondering if that sort of information is stored in a status object some where |
| 23:20.27 | [TK]D-Fender | DIALSTATUS <- |
| 23:22.18 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ritunvcaubvvhhwq) |
| 23:23.39 | FLeiXiuS | DialStatus might work...I'll have to see how to incorporate it. |
| 23:24.01 | FLeiXiuS | I'm integrating rails/ruby scripts to determine if a call is answered by remote and its not voicemail. |
| 23:24.44 | [TK]D-Fender | You don't do that through external processes, you do that on the channel itself |
| 23:24.49 | [TK]D-Fender | M() + AMD |
| 23:25.42 | FLeiXiuS | Let me go over a bit more of what im doing |
| 23:26.39 | FLeiXiuS | I have a web page that will have a 'call' function available. Used to test if a phone is answered or unavailable. The call button will establish a pseudo call to the extension. |
| 23:26.56 | FLeiXiuS | I don't care abut audio, I only care about the status of the call. |
| 23:27.06 | [TK]D-Fender | M() + AMD <- |
| 23:28.09 | FLeiXiuS | What's M()? |
| 23:28.17 | [TK]D-Fender | "core show application dial" |
| 23:29.12 | FLeiXiuS | Ah macro's, gotcha. |
| 23:32.45 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
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| 23:40.37 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:49.50 | FLeiXiuS | When executing Wait() before Answer, the DIALSTATUS returns ANSWERED, is that expected behavior? |
| 23:51.09 | [TK]D-Fender | FLeiXiuS: It shouldn't and I'm skeptical about what's actually happening... |
| 23:59.43 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-jhipavmgijihpsrc) |
