| 00:02.05 | comfrey | odd... so... this * box stopped allowing calls ... |
| 00:02.28 | comfrey | the only thing i see in the logs is file.c cant fing file... |
| 00:02.39 | comfrey | only warnings... |
| 00:02.53 | comfrey | what else might trip up an ast intance? |
| 00:03.01 | comfrey | instance |
| 00:14.17 | comfrey | codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 00:14.43 | comfrey | i am using sangoma hardware with dahdi... shoule this device exist? |
| 00:17.45 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:20.54 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:23.01 | *** join/#asterisk j0n23 (n=rkt@95.154.197.29) |
| 00:23.08 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:26.32 | xphree | quit |
| 00:45.19 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 00:45.55 | *** join/#asterisk [acer]lanman (n=lanman@173-102-7-136.pools.spcsdns.net) |
| 00:51.50 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:54.47 | comfrey | so i am having trouble understanding the dahdi transcode device. |
| 00:55.01 | comfrey | it is hw? or hardware dependant? |
| 01:00.14 | j0n23 | hey |
| 01:00.19 | j0n23 | any one around? |
| 01:00.31 | *** join/#asterisk jicksta (n=jicksta@c-67-188-112-50.hsd1.ca.comcast.net) |
| 01:00.32 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:01.09 | j0n23 | is it possible for me to autoselect calling cards,its pin number etc using asterisks? |
| 01:05.11 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:21.28 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:23.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4a5c28db312e16b2) |
| 01:37.42 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:37.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:38.42 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:40.59 | *** join/#asterisk ruben23 (n=RPL@203.177.91.193) |
| 01:41.42 | ruben23 | hi |
| 01:43.46 | carrar | HARRO |
| 01:44.13 | jaytee | rowr |
| 01:48.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:51.29 | *** join/#asterisk dshap (n=dshap@216-165-39-44.DYNAPOOL.NYU.EDU) |
| 01:52.34 | dshap | hey can someone here please help me figure out why Playback() isn't playing any audio on my channel? |
| 01:52.44 | dshap | i'm using 2 SIP providers and it works for 1 of them but not the other |
| 01:53.06 | dshap | which leads me to believe it is an issue with my configuration for that provider |
| 01:53.23 | dshap | here is my sip.conf: http://pastebin.com/d62ed2f17 |
| 01:53.33 | dshap | [voipms] works fine and can use Playback() |
| 01:53.55 | dshap | [flowroute] seems to be working but when i use Playback() i can't hear anything on the channel |
| 01:56.44 | Docteh | DIDs? its probably RTP problems? |
| 02:00.06 | dshap | huh |
| 02:00.10 | dshap | i forwarded all RTP ports |
| 02:05.07 | dshap | any other ideas? |
| 02:07.12 | Docteh | packet sniff the thing |
| 02:07.21 | dshap | you mean SIP DEBUG? |
| 02:08.18 | Docteh | ...no |
| 02:08.23 | Docteh | that'd be reading |
| 02:08.30 | dshap | you mean wireshark? |
| 02:08.34 | Docteh | with a packet sniff you can just look for packets going either way |
| 02:08.44 | Docteh | wireshark is a packet sniffer, yes |
| 02:08.51 | dshap | what would i be looking for |
| 02:08.53 | *** join/#asterisk chendy (n=chatzill@58.251.102.197) |
| 02:08.59 | Docteh | with a packet sniff you can just look for packets going either way <-- |
| 02:11.19 | dshap | im really not sure how to use a packet sniffer to solve this problem |
| 02:11.25 | dshap | i have minimal experience with packet sniffers |
| 02:11.30 | dshap | nothing looks wrong with my sip.conf? |
| 02:12.46 | carrar | no time like the present to learn! |
| 02:12.58 | dshap | hah hey carrar |
| 02:13.18 | carrar | This will increase your value |
| 02:14.03 | carrar | and then you can move into a UBER 31337 k-r4D status |
| 02:15.14 | dshap | can i do it remotely? |
| 02:15.19 | dshap | ive only ever used wireshark |
| 02:15.21 | dshap | w/the GUI |
| 02:15.22 | dshap | but i can't do that |
| 02:15.26 | dshap | because i'm now in new york |
| 02:15.30 | dshap | and the server is in my home in california |
| 02:15.43 | carrar | tcpdump captures import directly into wireshark |
| 02:16.03 | carrar | so yes |
| 02:16.26 | *** join/#asterisk macros73 (n=cs@c-71-61-74-104.hsd1.pa.comcast.net) |
| 02:17.39 | Docteh | dshap: either filter out ssh traffic or try to filter to only the range that flowroute uses |
| 02:18.38 | carrar | tcpdump -i eth0 -s 4000 -w MYCAPTUREFILENAMEHERE.cap host IPADDRESSHERE udp port 5060 |
| 02:19.03 | carrar | go forth and multiply |
| 02:19.38 | carrar | IPADDRESSHERE being the SIP peer that isn't working |
| 02:20.24 | carrar | then copy that file to your local machine and open in wireshark! |
| 02:20.40 | carrar | update wireshark too since they just released a newer version |
| 02:21.45 | carrar | (June 15, 2009; Wireshark 1.2.0) |
| 02:23.22 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 02:23.50 | carrar | tcpdump -i eth0 -s 4000 -w MYCAPTUREFILENAMEHERE.cap host IPADDRESSHERE and udp port 5060 |
| 02:23.52 | carrar | :) |
| 02:27.04 | carrar | This is handy too to just look at the SIP packets |
| 02:27.10 | carrar | ngrep -q -W byline -I MYCAPTUREFILENAMEHERE.cap |
| 02:27.18 | carrar | (may need to install ngrep) |
| 02:27.42 | Docteh | actually i was suggesting capture everything, sip and rtp |
| 02:28.42 | *** join/#asterisk tobias (n=tobias@cpe-174-099-111-162.nc.res.rr.com) |
| 02:31.10 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 02:31.35 | dshap | thanks guys |
| 02:31.36 | dshap | i will try this |
| 02:31.59 | dshap | what am i lookin for |
| 02:32.04 | dshap | in this SIP packets |
| 02:38.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c32b1fbd30612f86) |
| 02:40.03 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.23) |
| 02:42.00 | *** join/#asterisk mchou_ (n=quassel@unaffiliated/mchou) |
| 02:50.59 | *** join/#asterisk ingenius (n=alektro@188-213-17-190.fibertel.com.ar) |
| 02:56.18 | *** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net) |
| 03:02.22 | *** join/#asterisk propellerhead (n=yogurt2u@host115.190-224-222.telecom.net.ar) |
| 03:06.26 | *** join/#asterisk ruben23 (n=RPL@203.177.91.194) |
| 03:10.51 | *** part/#asterisk [pnp]tomas (n=xanith@96bus5.tampabay.res.rr.com) |
| 03:15.03 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 03:26.50 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:29.31 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 03:32.52 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 03:33.59 | rue_mohr | so then whats the latest stable asterisk version, I see 1.6.1.1 ? |
| 03:35.18 | rue_mohr | anyone want to place Bets on if the 1mw is fixed? |
| 03:38.15 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 03:42.31 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 03:46.24 | Docteh | whats a 1mw? |
| 03:50.18 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:50.33 | hesco | a megawatt? |
| 03:50.48 | hesco | I doubt it if it came up in asterisk |
| 04:00.22 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 04:03.52 | Docteh | oh |
| 04:03.58 | Docteh | 20:33 < rue_mohr> so then whats the latest stable asterisk version, I see 1.6.1.1 ? |
| 04:04.01 | Docteh | 20:35 < rue_mohr> anyone want to place Bets on if the 1mw is fixed? |
| 04:04.07 | Docteh | thats what i was asking about |
| 04:04.42 | Docteh | rue_mohr: wouldn't something like that be a hardware problem? so like drivers would fix it? |
| 04:29.43 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 04:40.15 | rue_mohr | Docteh, I dont know, all the codecs are in the astersik source |
| 04:40.27 | rue_mohr | and I think its a problem that revolves around slinear |
| 04:41.09 | rue_mohr | the core table didn't change, but how its used did |
| 04:41.29 | rue_mohr | I think someone redefined the 0db mark on the slinear |
| 04:41.45 | rue_mohr | I bleive peak shoudl be about +20db, not 0db |
| 04:41.52 | rue_mohr | but I cant prove any of it |
| 04:42.14 | rue_mohr | but I also didn't get any reply to "how are these codecs tested" |
| 04:45.06 | rue_mohr | I dont think there is a changelog for the zaptel->dahdi conversion |
| 04:45.52 | rue_mohr | but from what I can tell, alot of stuff got rolled in with the rewrite |
| 04:45.53 | rue_mohr | I also think someone has been going through the source and changeing the syntax to the way they like it |
| 04:46.31 | rue_mohr | I was told that the leap from 1.4 to 1.6 didn't require rewriting my configs, so here goes |
| 04:48.14 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 04:48.23 | buttons840 | Is anyone here familliar with neospeech? |
| 04:49.11 | buttons840 | I'm considering using it as a TTS in asterisk, it has the best tts quality i have found, and was wondering if anyone knew of something even better? |
| 04:49.29 | rue_mohr | <PROTECTED> |
| 04:52.26 | rue_mohr | I dotn get it, its the format for extensions.conf different in 1.6? |
| 04:52.35 | Docteh | whats the syntax? |
| 04:53.15 | Docteh | I do know you have to use , as a delimeter between arguments |
| 04:53.42 | rue_mohr | different than 1.4? |
| 04:53.53 | rue_mohr | just a min, I want to get my system back up |
| 04:55.24 | Docteh | well i didn't have trouble with dial, but i did with Voicemail and Voicemailmain |
| 04:55.52 | Docteh | voicemailmain(1000@home|s) -> voicemailmain(1000@home,2) |
| 04:56.31 | rue_mohr | exten => 9,1,Dial,DAHDI/7 |
| 04:56.39 | rue_mohr | thats the line that didn't work |
| 04:56.45 | Docteh | oh |
| 04:56.54 | rue_mohr | are () required now? |
| 04:57.02 | Docteh | exten => 9,1,Dial(DAHDI/7) |
| 04:57.20 | Docteh | possibly, i've only seen ()'s in use in examples |
| 04:57.20 | rue_mohr | I knew something changed about 1.6 that required me to do a bunch of config rewrite |
| 04:57.54 | *** part/#asterisk McL0VIN (n=chatzill@unaffiliated/dacs) |
| 04:58.43 | rue_mohr | I'll rewrite tonight and try 1.6 again tommorow |
| 04:59.58 | Docteh | so you press 9 and it just opens line 7? |
| 05:00.20 | rue_mohr | yep |
| 05:00.34 | rue_mohr | line 7 is the pots line |
| 05:01.38 | rue_mohr | tommorow I'll try 1.6 again, if it works with those changes, I'll test the 1mw again and see if its not still 10db out |
| 05:02.03 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 05:02.08 | Docteh | at some point i need to make my dialplan failover properly, right now i have 604xxxxxxx goes through pots and if i want via voip i do 1604xxxxxxx |
| 05:02.16 | Docteh | so 1mw is a test file? |
| 05:03.14 | rue_mohr | no, miliwatt() is a strength test signal, its 0db in amplitude, but somewhen it broke and its -10dbm instead of 0dbm |
| 05:03.53 | rue_mohr | and a lot of people have filed various complaints about things being about 10db quiet |
| 05:04.47 | rue_mohr | alot of things have been fixed, and they added a app_volume() , what I see as having been an overpatch for one of the complaints |
| 05:05.25 | rue_mohr | I positivly know its 10db short, cause I can measure it on my really rare system |
| 05:06.05 | Docteh | ah |
| 05:06.08 | rue_mohr | I seem to be the only one with an all analog system, a dbm meter and the knowledge to use it all :/ |
| 05:07.01 | rue_mohr | then again, polls on how many people use digium boards have come back pretty thin |
| 05:07.22 | rue_mohr | sounds like a lot of people use asterisk with sangnoma cards |
| 05:07.51 | rue_mohr | I'm trying to be loyal to digium on this and I think its biting my ass off :) |
| 05:09.14 | Docteh | I wouldn't know, I'm using linksys hardware which some people thumb thier nose down at |
| 05:09.27 | rue_mohr | hows it working for you? |
| 05:09.39 | rue_mohr | any volume/echo problems? |
| 05:10.34 | Docteh | nope |
| 05:10.39 | rue_mohr | cool |
| 05:10.48 | rue_mohr | I want to use some for a remote office |
| 05:10.51 | Docteh | I did have problems using a phone designed for the hard of hearing |
| 05:11.03 | *** join/#asterisk jicksta_ (n=jicksta@c-67-188-112-50.hsd1.ca.comcast.net) |
| 05:11.05 | rue_mohr | do tell |
| 05:11.44 | Docteh | the other side heard a lot of stuttering when i spoke, but it sounded fine on my end |
| 05:11.52 | [TK]D-Fender | rue_mohr: Linksys ATA's are OK, but their FXO's are very hit-or-miss |
| 05:12.10 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 05:12.36 | Docteh | yea |
| 05:12.46 | rue_mohr | which ones hit and which ones miss? |
| 05:12.50 | Docteh | strong "WorksForMe(TM)" notice from me |
| 05:13.09 | [TK]D-Fender | rue_mohr: Within the same model |
| 05:13.10 | Docteh | well there was someone in here having tons of trouble with an spa3102 which is what i use |
| 05:13.24 | rue_mohr | hmm |
| 05:13.29 | rue_mohr | thats what I wanted to use |
| 05:13.32 | Docteh | so its probably quality control or line calibration problems |
| 05:13.35 | [TK]D-Fender | rue_mohr: Some have real echo issues, others don't. It does gain drops and muting for suppression, etc |
| 05:13.54 | rue_mohr | tk, so your saying bad echo can? |
| 05:14.18 | [TK]D-Fender | rue_mohr: Only FXO from them I recommend is the SPA-3102 and only for home sue or MINIMAL office. For anything "real", FFS get a decent card |
| 05:14.34 | rue_mohr | tk, what version(s) would it be acceptable for me to file a bug on milliwatt() for? |
| 05:14.54 | [TK]D-Fender | rue_mohr: Yeah, reports have been random enough that someone in your neck of the woods had best avoid for business use |
| 05:15.12 | [TK]D-Fender | rue_mohr: I wouldn't touch these for QC measurement |
| 05:15.15 | hesco | Calls initiated on my * server with .call files are not being recorded in the CDR. I have read and heeded the advice about using Context/Extension/Priority in the call file |
| 05:15.26 | hesco | how would I proceed to debug this issue, please? |
| 05:15.51 | rue_mohr | I go ask in dev |
| 05:16.06 | Docteh | also, look at how cdr is triggered |
| 05:16.43 | hesco | Docteh: is that directed at me? |
| 05:17.23 | Docteh | yea, but its a suggestion not backed by any serious knowledge of asterisk |
| 05:17.43 | [TK]D-Fender | hesco: unanswered call-files do not generate CDR by default |
| 05:17.56 | hesco | what about answered ones? |
| 05:18.31 | hesco | this last .call file was answered, even if hung up on a few seconds later |
| 05:26.59 | [TK]D-Fender | checkout time, later all |
| 05:28.52 | hesco | g'night |
| 05:29.15 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 05:31.06 | ruben23 | good morning |
| 05:37.16 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 05:39.28 | *** part/#asterisk ruben23 (n=RPL@203.177.91.194) |
| 05:39.33 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 05:43.39 | zeeesh | yahoo SIP to Asterisk is it possible? |
| 05:51.15 | Docteh | is it standard sip or with weird stuff? |
| 05:54.43 | zeeesh | Yahoo is using TCP SIP and RTP codec SPEEX. |
| 06:00.16 | kaldemar | zeeesh: there's experimental support for TCP SIP in 1.6 and speex codec. go try it. |
| 06:03.35 | Nugget | tcp sip works great in 1.6 |
| 06:07.54 | hesco | I'm seeing in the CLI: |
| 06:07.57 | hesco | [2009-06-23 02:06:57] NOTICE[28051]: loader.c:592 ast_module_reload: The module 'cdr_manager.so' was not properly initialized. Before reloading the module, you must run "module load cdr_manager.so" and fix whatever is preventing the module from being initialized. |
| 06:08.53 | hesco | which yields: Unable to load module cdr_manager.so ; Command 'module load cdr_manager.so' failed. ; [2009-06-23 02:07:25] WARNING[28051]: cdr_manager.c:72 load_config: Failed to load configuration file. Module not activated. |
| 06:09.05 | hesco | any clues as to why? |
| 06:10.14 | kaldemar | failed to load configuration file may be a good hint |
| 06:10.54 | kaldemar | do you have cdr_manager.conf? |
| 06:12.07 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 06:12.12 | florz | erm, any hints as to how I make chan_dahdi compile such that it uses zaptel in 1.4? |
| 06:13.32 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:13.57 | hesco | I just redid the menu select thing and it indicates that I do have cdr_manager.so. |
| 06:14.12 | kaldemar | florz: now why would you want that? |
| 06:14.16 | hesco | sorry, I see now. no I don't have a cdr_manager.conf file in /etc/asterisk |
| 06:14.27 | hesco | where might I find a sample to adapt from? |
| 06:14.44 | florz | kaldemar: because I don't wanna port all our zaptel patches for now |
| 06:15.00 | kaldemar | hesco: <asterisk_source_dir>/configs/cdr_manager.conf.sample |
| 06:16.04 | hesco | thanks. on it now |
| 06:18.10 | kaldemar | florz: sounds like some work anyway. |
| 06:20.52 | Docteh | heh |
| 06:21.24 | hesco | kaldemar: THANKS !!! That resolved a long term issue where I was not seeing CDR's for callfile initiated calls. |
| 06:21.27 | Docteh | i thought 1.4 was supposed to support zaptel or dahdi |
| 06:21.37 | hesco | On the the next issue now. |
| 06:21.45 | hesco | Thanks for the second set of eyes |
| 06:22.35 | kaldemar | Docteh: it does, but not chan_dahdi with zaptel or chan_zap with dahdi. at least not just like that. |
| 06:23.37 | Docteh | ah |
| 06:31.06 | *** join/#asterisk xrmx__ (n=rm@host176-254-dynamic.14-87-r.retail.telecomitalia.it) |
| 06:38.57 | *** part/#asterisk norgs (n=ken@WoWUIDev/Norganna/Administrator/Norgs) |
| 06:39.23 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:42.30 | florz | kaldemar: what then? |
| 06:42.50 | florz | chan_sip with zaptel support? |
| 06:43.00 | Docteh | chan_zaptel |
| 06:43.28 | florz | well, how would I make asterisk build that, then? |
| 06:44.52 | *** join/#asterisk ectodozing (n=ectospas@i.hate.your.vhosts.shellium.org) |
| 06:45.41 | kaldemar | just use chan_zap & zaptel with 1.4 until you port your patches to dahdi. |
| 06:46.07 | florz | kaldemar: well, how do I get chan_zap with current 1.4, then? |
| 06:46.52 | tzafrir_laptop | Porting a patch to chan_zap over to chan_dahdi takes some work but shouldn't be difficult. Mostly sed |
| 06:47.05 | kaldemar | now i see, there's no chan_zap in current 1.4. my bad. |
| 06:48.06 | tzafrir_laptop | (as was the case with the asterisk part of bristuff) |
| 06:48.09 | florz | tzafrir_laptop: the chan_zap patches I already have ported - what's missing yet are the driver patches |
| 06:48.25 | tzafrir_laptop | zaphfc ? |
| 06:48.31 | florz | nah |
| 06:48.35 | florz | well, that too, is you will ;-) |
| 06:48.45 | florz | s/is/if/ |
| 06:48.54 | tzafrir_laptop | frankly it shouldn't be a patch. Nobody uses the original |
| 06:49.08 | kaldemar | 1.4.21.2 seems to be the last release shipped with chan_zap |
| 06:49.56 | florz | tzafrir_laptop: well, yeah, try to get kapejod to actually include it with the driver, not just as an optional patch ;-) |
| 06:50.34 | florz | tzafrir_laptop: but actually, we have a bunch more patches for this and that which I'd rather like to avoid porting for now |
| 06:50.53 | florz | tzafrir_laptop: which is why I'd like to use chan_dahdi with zaptel ... |
| 06:53.49 | tzafrir_laptop | does he still care? http://www.junghanns.net/en/VOIPMG_produkt.html |
| 06:54.16 | tzafrir_laptop | asterisk 1.4 works with zaptel |
| 06:54.59 | florz | tzafrir_laptop: no clue? At least bristuff still contains the driver, after all ;-) |
| 06:55.06 | *** join/#asterisk linuxmaniac (n=maniac@239.Red-81-32-73.dynamicIP.rima-tde.net) |
| 06:56.00 | tzafrir_laptop | In DAHDI I get reports of it oopsing . I had not had time to debug them |
| 06:56.00 | florz | tzafrir_laptop: @zap with * 1.4: yeah, that's what I thought, but it doesn't seem to work ... even though it may be that the problem is somewhere else entirely, just downgraded to 1.4.21, and chan_zap still doesn't work ... |
| 06:56.24 | tzafrir_laptop | (I don't have the hardware, and remote debugging of oopses is no fun) |
| 06:56.51 | florz | you mean, zaphfc with my patch oopsing? |
| 06:58.18 | tzafrir_laptop | yes |
| 06:58.23 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 06:59.41 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
| 07:08.17 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:10.42 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 07:14.03 | florz | well, that problem seems to have been ... whatever, I have no clue, but after re-loading the kernel modules, chan_dahdi has no problem using zaptel, it seems |
| 07:14.04 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:15.37 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:19.23 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:19.56 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 07:24.07 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 07:24.39 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:49.47 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 07:53.06 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 07:53.25 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 07:56.53 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:06.57 | *** join/#asterisk [acer]lanman (n=lanman@173-133-130-32.pools.spcsdns.net) |
| 08:09.07 | *** join/#asterisk NaPs (n=Unknown@tecknet.org) |
| 08:10.11 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:24.10 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:35.05 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 09:00.07 | *** join/#asterisk [acer]lanman (n=lanman@173-133-130-32.pools.spcsdns.net) |
| 09:06.01 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:11.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:15.38 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 09:20.46 | *** join/#asterisk bionoid (i=terje@mesyah.net) |
| 09:33.31 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:06.08 | *** join/#asterisk salzh (n=Administ@122.144.138.5) |
| 10:17.28 | *** join/#asterisk mattbUK (n=mattbUK@82-46-93-20.cable.ubr16.stav.blueyonder.co.uk) |
| 10:23.07 | McL0VIN | question: what is the different between [context] in sip.conf and [context] in extension.conf? |
| 10:23.35 | kaldemar | everything. |
| 10:24.27 | McL0VIN | kaldemar: oh yeah that helps me alot,,,jeez thank man! |
| 10:24.36 | kaldemar | they have nothing in common. |
| 10:24.51 | kaldemar | besides syntax. be more specific. |
| 10:26.03 | ISO9001 | lol |
| 10:26.47 | McL0VIN | and thats whats confusing me |
| 10:28.45 | kaldemar | in extensions.conf you make the dialplan. in sip.conf you can define devices that may use the dialplan. the only things linking those two are the context parameter under some [context] in sip.conf and a Dial(SIP/<device>) in extensions.conf, where <device> is a [context] in sip.conf. |
| 10:29.13 | kaldemar | still confused? |
| 10:31.56 | McL0VIN | lol yes |
| 10:34.28 | kaldemar | ~book |
| 10:34.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 10:34.48 | kaldemar | McL0VIN: ^^ that will be helpful |
| 10:36.11 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 10:38.17 | McL0VIN | kaldemar: i have the hard copy |
| 10:42.59 | *** join/#asterisk [acer]lanman (n=lanman@173-133-130-32.pools.spcsdns.net) |
| 10:49.19 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 10:51.54 | *** join/#asterisk j_kroon (n=jkroon@196.35.70.126) |
| 10:54.24 | j_kroon | hi all, not sure whether this is a known issue, or whether my internal timing just isn't working, but I've got an IAX/2 trunk running out to my provider. tcpdump shows that the incoming packets are coming in at 20ms intervals, however, asterisk is sending the SIP side packets to the phone out at very irregular (10ms mostly, then suddenly at 170ms) intervals. |
| 10:54.55 | j_kroon | is there any way I can inspect the trunked timestamps to make sure they are sane? |
| 10:55.15 | j_kroon | iax/2 timestamps is imho the first thing to check, but I have no idea how. |
| 10:55.35 | j_kroon | ok, actually, that's the second thing, first I need to confirm that internal timing is working properly. |
| 11:00.40 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 11:01.29 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:01.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:07.37 | j_kroon | ok, update, it seems it's _fine_ when actually trunking. The issue is when it's the ONLY IAX/2 stream on the wire at the time. |
| 11:13.39 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 11:14.21 | McL0VIN | anyone using zoiper here |
| 11:18.15 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 11:25.12 | *** join/#asterisk Muisje (i=3e15ae19@gateway/web/freenode/x-d4c23760381a1a8b) |
| 11:25.51 | Muisje | Hello, i'm looking for a solution to connect my landline to skype accounts, is asterisk any good for me? + What would i need for hardware? and pricing? :) |
| 11:39.17 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 11:42.27 | bionoid | Muisje: Asterisk can probably do what you want, but you need to state more clearly your intentions (should a ring to skype account transfer to pots telephone, outgoing call routed from pots -> skype -> world, or what) |
| 11:42.45 | bionoid | Muisje: In general I would reccomend you open an account with a "real" telco and not use skype. |
| 11:47.05 | j_kroon | if i'm not mistaken there is a chan_skype. |
| 11:47.37 | j_kroon | or alternatively there is a sip provider that can do the "routing" for you. |
| 11:49.41 | madduck | why in the world would my asterisk send a double contact in the REGISTER packet? |
| 11:49.45 | madduck | Contact: <sip:s@77.109.139.86>;expires=120, <sip:madduck@77.109.139.86>;expires=83 |
| 11:49.53 | madduck | I do *not* have any register-> lines in sip.conf |
| 11:49.59 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:50.00 | madduck | I only have callbackextension in the peer sections |
| 11:50.17 | *** join/#asterisk Aiatek (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:51.26 | madduck | i have *no* idea where the madduck@ comes from |
| 11:52.09 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
| 11:54.03 | leifmadsen | McL0VIN: I am |
| 11:55.13 | zeeesh | i hv installed "yum -y install speex speex-devel" but when compiling asterisk-1.6.1, it still showing "XXX codec_speex"? |
| 11:55.30 | McL0VIN | leifmadsen: i am not able to get sound at all when i dial i don't hear the prompt |
| 11:56.28 | leifmadsen | McL0VIN: you need to provide more information about your topology, console output (in a pastebin), perhaps relevant sections of your dialplan, 'rtp debug' output to determine if you're getting audio flowing (and to the right IP), etc... |
| 11:56.48 | leifmadsen | needs to write an article on how to write good questions, and how to debug |
| 11:56.55 | leifmadsen | adds it to the long long list |
| 11:59.03 | *** join/#asterisk jeffspeff (n=jeffspef@c-68-53-54-238.hsd1.ky.comcast.net) |
| 11:59.23 | tzafrir_laptop | does codec_speex really need all the fancy features of speex? "somebody" needs to fix the configure script so codec_speex could be built even with the basic speex support of centos5 |
| 12:00.40 | jeffspeff | i'm needing to get some wireless phones deployed in a warehouse. not going to be using hardly an features, but phones need to small and have blue-tooth headset capability. and looking for something in the $100USD each area... any suggestions? |
| 12:00.51 | madduck | i am amazed by how much asterisk remembers across a reboot |
| 12:00.54 | madduck | seriously |
| 12:01.15 | madduck | the weirdest things are happening |
| 12:02.14 | tzafrir_laptop | you happen to use software suspend? |
| 12:02.22 | j_kroon | leifmadsen, people who can't ask good questions also generally can't read. |
| 12:02.23 | *** join/#asterisk michael-i (n=michael-@141.41.40.153) |
| 12:02.45 | madduck | tzafrir_laptop: i meant restart of asterisk; and no. |
| 12:03.12 | madduck | at someon point in time I had callbackextension=incoming for a peer |
| 12:03.18 | leifmadsen | j_kroon: ya, don't I know it -- but at least I can beat them over the head with an article |
| 12:03.24 | madduck | and that caused asterisk to register s@ and incoming@ with the peer |
| 12:03.34 | madduck | and then I used callbackextension=s |
| 12:03.43 | madduck | but it's still s@ and incoming@ |
| 12:03.46 | madduck | even after a reboot |
| 12:03.50 | madduck | how does it remember incoming? |
| 12:04.07 | tzafrir_laptop | leifmadsen, hmm.. that takes a book. And a thick one |
| 12:04.29 | leifmadsen | tzafrir_laptop: hmmm..... where is my copy of War and Peace.... |
| 12:05.04 | leifmadsen | Warring with my copy brings me Peace |
| 12:05.31 | leifmadsen | McL0VIN: sorry, you'll have to clean things up and post it to the room -- I don't do non-public support |
| 12:06.04 | tzafrir_laptop | madduck, asterisk regissters to some remote host? |
| 12:06.27 | McL0VIN | leifmadsen: it all good man , look like you need public to support you as well |
| 12:06.38 | leifmadsen | burn |
| 12:06.41 | madduck | tzafrir_laptop: due to callbackextension, which I think replaces register=> |
| 12:07.36 | madduck | and another peer, netvoip.ch -- i register with contact incoming@ and it just talks to me with my username |
| 12:10.54 | tzafrir_laptop | leifmadsen, well, there's the generic one, of course: http://www.catb.org/~esr/faqs/smart-questions.html |
| 12:11.16 | tzafrir_laptop | ~smart-questions |
| 12:11.16 | infobot | rumour has it, smart-questions is http://tinyurl.com/anel |
| 12:11.25 | j_kroon | <random>I've got two servers in the same rack, the one downloads from ftp.kernel.org at 2Mbps, the other at 50Kbps ... </random> |
| 12:11.27 | leifmadsen | tzafrir_laptop: right -- I meant a more general Asterisk debugging article |
| 12:11.44 | leifmadsen | j_kroon: I'd respond with, "Yes you do." |
| 12:11.46 | j_kroon | leifmadsen, would be useful. i'd like that too. |
| 12:12.02 | leifmadsen | primarily focused on debugging SIP issues |
| 12:12.05 | tzafrir_laptop | infobot, no, smart-questions is http://www.catb.org/~esr/faqs/smart-questions.html |
| 12:12.05 | infobot | tzafrir_laptop: okay |
| 12:12.16 | leifmadsen | yay! |
| 12:12.28 | j_kroon | leifmadsen, i've got a sip/iax gateway issue at the moment. if you're up for a challenge. |
| 12:12.34 | leifmadsen | I'm not |
| 12:12.38 | j_kroon | ok. |
| 12:12.39 | leifmadsen | my work day has started |
| 12:12.45 | j_kroon | ah, good luck. |
| 12:12.54 | leifmadsen | I don't need luck; I need time :) |
| 12:13.05 | j_kroon | get off irc. |
| 12:13.30 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:14.28 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:15.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.30 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:18.10 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 12:19.31 | leifmadsen | hands the torch off to [TK]D-Fender |
| 12:19.40 | leifmadsen | [TK]D-Fender: all yours! |
| 12:19.59 | [TK]D-Fender | grins and wonders who he'll get to ignite first :D |
| 12:20.12 | [TK]D-Fender | ~fire |
| 12:20.12 | infobot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 12:20.59 | guax | lol |
| 12:23.12 | madduck | argh! |
| 12:23.32 | madduck | now an outgoing call tries to treat callbackextension as a hostname! |
| 12:23.35 | madduck | chan_sip.c:4526 create_addr: No such host: incoming |
| 12:23.55 | WeazelON | hey guys, i've found this error in the /etc/asterisk/full right after asterisk crashed "segfault at 6ac ip 08080380 sp b65beba0 error 4 in asterisk[8048000+103000]" i'm running an asteirsk 1.4.24.1 on a gentoo 2.6.25-gentoo-r8 |
| 12:24.07 | WeazelON | can someone tell me what happend exactly ? |
| 12:25.02 | leifmadsen | WeazelON: probably not -- you'd have to generate a backtrace (per backtrace.txt in doc/) |
| 12:25.55 | leifmadsen | WeazelON: additionally, you'd have to open an issue on http://issues.asterisk.org with what happens to cause the crash (as best you can), and to provide the backtrace while you have DONT_OPTIMIZE compiled into Asterisk (from menuselect, under Compiler Flags) |
| 12:26.37 | madduck | <PROTECTED> |
| 12:26.37 | madduck | [Jun 23 14:25:07] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: incoming |
| 12:26.41 | madduck | wtf? |
| 12:26.49 | WeazelON | k thx |
| 12:28.28 | madduck | if I change callbackextension to 'dialin', asterisk remembers and gives me two warnings |
| 12:28.31 | madduck | [Jun 23 14:26:51] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: incoming |
| 12:28.34 | madduck | [Jun 23 14:26:51] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: dialin |
| 12:28.35 | leifmadsen | madduck: not much info for us to debug on -- I'd pastebin the 'sip debug' after removing anything you don't want us to see. Additionally -- why are you using callbackextension? |
| 12:28.54 | leifmadsen | I've never even used that configuration option |
| 12:29.09 | madduck | leifmadsen: why not? i understand it's the new way forward to prevent having to keep credentials in two separate places in sip.conf |
| 12:30.01 | leifmadsen | Guess that makes sense. |
| 12:30.11 | leifmadsen | I do all my stuff in realtime, so I don't have that problem |
| 12:30.31 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:30.38 | leifmadsen | anyways, debug output would be useful to see why it might be failing. Could even be that you've found a bug if it is a brand new option. |
| 12:30.49 | madduck | it just keeps growing |
| 12:30.53 | madduck | even after i restart asterisk |
| 12:30.54 | madduck | [Jun 23 14:29:29] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: dialin |
| 12:30.57 | madduck | [Jun 23 14:29:29] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: incoming |
| 12:31.00 | madduck | [Jun 23 14:29:29] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: in-de-sipgate |
| 12:31.03 | madduck | [Jun 23 14:29:29] WARNING[5788]: chan_sip.c:4526 create_addr: No such host: s |
| 12:31.07 | [TK]D-Fender | madduck: PAStebiN |
| 12:31.08 | madduck | anyway, this is *outgoing* |
| 12:31.15 | [TK]D-Fender | madduck: And you aren't showing anything |
| 12:31.21 | madduck | right, will prepare pastebin |
| 12:31.52 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:31.56 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:32.27 | ariel_ | Morning folks |
| 12:35.28 | AlmightyOatmeal | waves |
| 12:36.33 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 12:41.29 | tzafrir_laptop | madduck, what's called "incomming" in sip.conf? |
| 12:41.42 | madduck | tzafrir_laptop: callbackextension for the peer |
| 12:42.09 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 12:42.40 | madduck | well, it seems it only happens when I call my sipgate landline number using sipgate for outgoing |
| 12:42.44 | madduck | everything else seems to work |
| 12:42.45 | madduck | weird. |
| 12:42.59 | [TK]D-Fender | madduck: PASTEBIN |
| 12:43.07 | AlmightyOatmeal | sounds like a syntax issue, pastebin that stuff |
| 12:43.17 | AlmightyOatmeal | we aren't all psychics ya know |
| 12:43.32 | madduck | i am working on it |
| 12:43.40 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 12:43.41 | [TK]D-Fender | AlmightyOatmeal: I knew you were gonna say that... |
| 12:43.49 | AlmightyOatmeal | [TK]D-Fender: i did too... |
| 12:44.03 | AlmightyOatmeal | ;P |
| 12:44.17 | [TK]D-Fender | AlmightyOatmeal: That jsut makes me psychic and you premeditated ;) |
| 12:45.16 | AlmightyOatmeal | mmm |
| 12:45.30 | AlmightyOatmeal | it sounds sexy when you say it like that |
| 12:45.53 | madduck | [TK]D-Fender, AlmightyOatmeal: here you go: http://stikked.com/view/34805867 |
| 12:45.57 | madduck | leifmadsen: ^ |
| 12:46.33 | superbeef | Anybody have any preferred Windoze SIP client? One of my users has been using X-Lite, and suddenly now it keeps on having registration errors until the machine reboots. |
| 12:47.40 | AlmightyOatmeal | superbeef: i particularly like x-lite and even use that emulated under wine in *nix |
| 12:48.18 | superbeef | AlmightyOatmeal: it had been my favorite, but the this registration error thing is driving me crazy because its driving her crazy |
| 12:48.21 | madduck | i don't know why my machine would send something like INVITE sip:089xxxxxx852@incoming SIP/2.0 to the SIP peer |
| 12:48.40 | AlmightyOatmeal | superbeef: then fix it. |
| 12:48.43 | madduck | or well, try to |
| 12:48.55 | madduck | Reliably Transmitting (no NAT) to 217.10.79.9:5060: |
| 12:49.19 | [TK]D-Fender | madduck: 4989xxxxxx852 <- "sip show peer for this |
| 12:49.37 | superbeef | AlmightyOatmeal: I'll get right on that |
| 12:49.50 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 12:49.51 | AlmightyOatmeal | superbeef: specifically what error goes on? |
| 12:49.52 | [TK]D-Fender | madduck: And what are this incoming calls for? |
| 12:50.28 | AlmightyOatmeal | superbeef: is the error the softphone itself or your settings for your asterisk box or the settings for the phone for your asterisk box? |
| 12:50.28 | superbeef | AlmightyOatmeal: 401 registration error.... I spent an hour last week trying different accounts etc, then i just rebooted the stupid box and it connected without a problem |
| 12:50.29 | madduck | [TK]D-Fender: i am trying to call my own number via sipgate |
| 12:50.33 | madduck | http://stikked.com/view/37637285 |
| 12:50.39 | superbeef | AlmightyOatmeal: totally the softphone |
| 12:51.44 | AlmightyOatmeal | superbeef: sounds like a similar problem i was having with my asterisk box connecting to my sip proxy.. turns out it was the host operating systems caching of data that caused problems... so i would check the host operating system before writing off x-lite |
| 12:51.45 | [TK]D-Fender | madduck: 1st change your dial format to "SIP/peername/numbertodial" |
| 12:52.00 | madduck | ok |
| 12:52.01 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:52.11 | AlmightyOatmeal | superbeef: as in you may want to try the network "repair" before restarting the whole machine to see if it fixes the issue |
| 12:52.40 | [TK]D-Fender | madduck: And whats the point of"callbackextension=incoming"? |
| 12:52.43 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:53.06 | madduck | [TK]D-Fender: from what I understand in the comments of sip.conf, that replaces the register=> line for that peer |
| 12:53.14 | madduck | (or for all peers since I use it in a template) |
| 12:53.27 | superbeef | AlmightyOatmeal: you're probably right...... Vista creates broken homes |
| 12:53.31 | [TK]D-Fender | madduck: DON't |
| 12:53.37 | madduck | why not? |
| 12:54.02 | AlmightyOatmeal | superbeef: vista, vista_sucks++ |
| 12:54.03 | [TK]D-Fender | madduck: Because every time I see people trying to be "mart" they do stupid things that break their systems. E |
| 12:54.07 | [TK]D-Fender | smart* |
| 12:55.02 | superbeef | AlmightyOatmeal: for sure.. and guess what, we (mostly me) can quit worrying about that problem. Our stupid ticket system was sending me a notication that the ticket was closed, but the heading was wrong... |
| 12:55.46 | madduck | [TK]D-Fender: okay, but if I use "register => 72xxx27:xxxxxxxxxx@4989xxxxxx852/incoming" instead, I am told |
| 12:55.54 | madduck | WARNING[5788]: acl.c:376 ast_get_ip_or_srv: Unable to lookup '4989xxxxxx852' |
| 12:56.21 | madduck | that problem goes away if i duplicate the hostname from the peer to the register line too |
| 12:56.58 | [TK]D-Fender | madduck: Also undo your templating |
| 12:56.58 | madduck | indeed, with register=>, the call works |
| 12:57.01 | madduck | what a shame. |
| 12:57.15 | madduck | [TK]D-Fender: for the same reason that I am trying to do something smart? |
| 12:57.25 | madduck | I have already unrolled it earlier. |
| 12:57.39 | [TK]D-Fender | madduck: Bad things happen. |
| 12:57.53 | madduck | why does the feature exist then? ;) |
| 12:59.40 | [TK]D-Fender | madduck: More "nice ideas" that are buggy and people use to try to say typing 3 lines and put in wrong places, etc. Shit happens, and when you insist on working as far outside the box as possible you'll get whats coming, especially with * |
| 13:01.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:01.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:03.39 | madduck | well, to be honest, now it's really weird. register => 72xxx27:xxxxxxxxxx@4989xxxxxx852/incoming causes incoming calls to go to extension 'incoming' but in the context of the peer defined later |
| 13:03.39 | madduck | which i suppose is matched using the hostname |
| 13:03.39 | madduck | wait, register => 72xxx27:xxxxxxxxxx@sipgate.de/incoming |
| 13:03.39 | [TK]D-Fender | madduck: Regsiter does not auth your calls |
| 13:03.39 | madduck | because using the peer name does not work |
| 13:03.49 | madduck | [TK]D-Fender: not outgoing ones |
| 13:03.53 | madduck | but it routes incoming ones |
| 13:03.59 | madduck | and it uses information from the peer stanza for that |
| 13:04.17 | [TK]D-Fender | madduck: register has nothing to do with authing the incoming call |
| 13:04.30 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 13:04.32 | madduck | i said routing, no? |
| 13:04.38 | [TK]D-Fender | madduck: NO |
| 13:04.46 | madduck | 23 15:03 < madduck> but it routes incoming ones |
| 13:05.11 | [TK]D-Fender | madduck: 2 completely separate funcstion. Any association betweent he 2 is purely logical, not functional. |
| 13:05.49 | madduck | register causes asterisk to auth with the sip provider and inform it about a contact, e.g. incoming@myip |
| 13:05.57 | madduck | when a call comes in, it is matched with a peer |
| 13:06.08 | madduck | and thus placed in that context |
| 13:06.29 | madduck | but using the extension that the register=> instruction conveyed to the peer |
| 13:06.33 | [TK]D-Fender | madduck: It is matched to the peer because that what the perr matched, but it isn't matched against it because of the register |
| 13:06.43 | madduck | yeah i understand that (now) |
| 13:07.05 | madduck | it's really not very pretty though. :/ |
| 13:07.17 | [TK]D-Fender | madduck: 09:06]<madduck>but using the extension that the register=> instruction conveyed to the peer <--- nothing is converyed. Register has no relationship to your peer |
| 13:07.47 | madduck | [TK]D-Fender: why then do I append /incoming to the register command, to prevent the s default? |
| 13:07.56 | madduck | this causes a Contact: header to be appended to the INVITE |
| 13:08.07 | [TK]D-Fender | [09:07]<madduck>[TK]D-Fender: why then do I append /incoming to the register command, to prevent the s default? <- Yes |
| 13:08.13 | madduck | and when the peer receives a call for me and routes it on, it uses 'incoming' in the INVITE command to my asterisk |
| 13:08.13 | [TK]D-Fender | madduck: And nothing more than that. |
| 13:08.44 | madduck | register …/incoming means "hey sip registrar, if you get a call for me, route it here to incoming@myip" |
| 13:08.44 | [TK]D-Fender | madduck: Because some providers will NOT send authed calls and all of their junk will end up in the same context and you need some way to figure out whos ent the call |
| 13:08.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.02 | madduck | the registrar can ignore that |
| 13:09.08 | madduck | like netvoip.ch does |
| 13:09.37 | [TK]D-Fender | madduck: They can, but usually those who do will send the DID (if thats what they use) regardless. Or others will include an extra header |
| 13:10.11 | madduck | alright, i think i get it now |
| 13:11.06 | madduck | thank you |
| 13:13.19 | SuPrSluG | coppice: you around? having a fax issue |
| 13:14.07 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 13:14.13 | coppice | I'm not here right now, but if you leave your name and number I'll get back eventually |
| 13:15.57 | ariel_ | wow, nice message, But should it not be: Sorry I am not here right now, But if you leave your name and number, and I fell like I just might get back to you or not! |
| 13:17.26 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 13:17.59 | coppice | "eventually" gives hope without commitment |
| 13:18.13 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:18.14 | Katty | :> |
| 13:20.26 | eppigy | hi |
| 13:20.34 | SuPrSluG | coppice: i'm getting the dreaded PHASESTRING is Received a DCN while waiting for a DIS |
| 13:21.02 | SuPrSluG | coppice: what settings should I look at? |
| 13:21.42 | SuPrSluG | coppice:trying to do T38 |
| 13:21.46 | Katty | HAI |
| 13:22.03 | [TK]D-Fender | Katty: O HAI |
| 13:22.17 | coppice | you probably have one way communication for some reason |
| 13:24.22 | Katty | coppice: that's what she said. |
| 13:24.49 | SuPrSluG | coppice:firewall? maybe? |
| 13:24.59 | SuPrSluG | or nat? |
| 13:25.46 | coppice | poor old Nat gets blamed for everything |
| 13:26.02 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 13:26.27 | *** join/#asterisk ruchir (n=ruchir@122.169.82.43) |
| 13:26.55 | ruchir | hi all |
| 13:27.15 | eppigy | D: |
| 13:27.17 | ariel_ | that is due to "Nat" is almost always in the mix of issues. |
| 13:27.29 | eppigy | /NAT/ |
| 13:27.39 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-213-174.lns10.mel4.internode.on.net) |
| 13:29.11 | ruchir | any idea about non-standard rtp encryption? |
| 13:29.57 | SuPrSluG | coppice:nat should be set to no in sip.conf , correct? |
| 13:30.00 | *** join/#asterisk odenkos (n=feeds@85-135-156-24.adsl.slovanet.sk) |
| 13:30.03 | coppice | what's wrong with standard RTP encryption? |
| 13:30.19 | ruchir | coppice: it might be blocked by isp :) |
| 13:30.25 | ariel_ | ok it's time to go to the lab. I have a few dial plans to start testing and new configurations to do.... |
| 13:30.58 | ruchir | basically i would like to encrypt in a way that it doesnt look like rtp at all |
| 13:31.02 | ruchir | even i did it |
| 13:31.09 | ruchir | added padding |
| 13:31.13 | ruchir | encrypted pkt |
| 13:31.20 | ruchir | but then i get disturbance |
| 13:31.32 | ruchir | if i remove padding and keep encryption, audio is proper |
| 13:31.33 | [TK]D-Fender | ruchir: VPN <- |
| 13:31.44 | ruchir | but when padding added, audio broken |
| 13:32.00 | ruchir | [TK]D-Fender : that might be slow and also blocked :) |
| 13:32.16 | [TK]D-Fender | ruchir: Get a new ISP |
| 13:32.21 | ruchir | :D |
| 13:32.25 | ruchir | wish i could :( |
| 13:32.33 | ruchir | i have to live with that only |
| 13:32.35 | coppice | bomb them |
| 13:32.38 | ruchir | :D |
| 13:32.47 | ruchir | its out of my hands |
| 13:33.00 | ruchir | so no choice except making non-standard protocols ;) |
| 13:33.07 | coppice | well, when it goes off it had better be |
| 13:33.20 | [TK]D-Fender | ruchir: Good, because holding onto the bomb when it ggoes off could be hazardous to your health... |
| 13:33.29 | ruchir | :) |
| 13:33.44 | [TK]D-Fender | shakes his fist at coppice ! |
| 13:33.48 | ruchir | so anyone have idea how to remove disturbance? |
| 13:33.54 | rob0 | Hey, I need to make a silk purse, and all I have is this sow's ear. Can you help? |
| 13:34.00 | coppice | AK-47 |
| 13:34.02 | [TK]D-Fender | coppice: Get off my telepathic wavelength, this is registered band-space! |
| 13:34.39 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 13:38.49 | *** join/#asterisk VoipForces (n=quassel@firewall.prival.ca) |
| 13:42.55 | *** join/#asterisk xphree (n=xphree@unaffiliated/xpider) |
| 13:43.36 | j_kroon | [Jun 23 15:43:27] WARNING[27489]: chan_sip.c:2629 __sip_xmit: sip_xmit of 0x849ef48 (len 861) to 192.168.41.127:5060 returned -1: Operation not permitted |
| 13:43.38 | j_kroon | any ideas? |
| 13:43.40 | xphree | Hello all, Yesterday i was asking for the L() Param in the Dial() command... it's working for DAHDI calls, but for SIP calls the system doesn't hangup the call |
| 13:44.40 | [TK]D-Fender | j_kroon: SIP DEBUG <- |
| 13:44.51 | j_kroon | [TK]D-Fender, it resolved itself now ... |
| 13:44.57 | [TK]D-Fender | xphree: I'm not seeing anything to debug from you either |
| 13:45.02 | j_kroon | wish i could get the sip debug ::) |
| 13:46.33 | *** join/#asterisk xphree (n=xphree@unaffiliated/xpider) |
| 13:47.09 | xphree | it seems that the L() param or the S() param in the Dial() Command doesn't hangup the SIP calls, only DAHDI |
| 13:47.20 | VoipForces | Hi all, I'm about to write some code (dialplan centric) to implement the Nortel realtime SMDR serial port ouput. Before I do so, anyone has written something for that already? |
| 13:47.43 | ruchir | any idea about removing disturbance? |
| 13:48.13 | VoipForces | ruchir: disturbance??? |
| 13:48.30 | ruchir | well i encrypted rtp pkts |
| 13:48.36 | ruchir | for special purpose |
| 13:48.45 | ruchir | which worked fine |
| 13:48.49 | ruchir | then i added padding to pkt |
| 13:48.54 | ruchir | then get disturbance |
| 13:48.58 | ruchir | 7 bits added |
| 13:49.02 | VoipForces | ruchir: oh, never worked with that sorry. |
| 13:49.04 | *** join/#asterisk mattbUK (n=mattbUK@82-46-93-20.cable.ubr16.stav.blueyonder.co.uk) |
| 13:49.09 | ruchir | what coujld be causing disturbance |
| 13:49.49 | [TK]D-Fender | xphree: No. |
| 13:51.11 | xphree | [TK]D-Fender: do you have any idea? |
| 13:51.54 | [TK]D-Fender | xphree: Yes, show us the problem |
| 13:52.12 | xphree | we have a system with Trunks in DAHDI and SIP |
| 13:52.12 | beek | morning [TK]D-Fender |
| 13:52.37 | [TK]D-Fender | xphree: I read that already, I said SHOW us the problem. Actual CLI output showing the FAILURE |
| 13:52.53 | [TK]D-Fender | beek: Morning it is, condition fading fast... |
| 13:52.56 | BlargMaN00 | When I use a call file to dial a number, I answer my handset, and then the outside phone I dialed starts ringing, but there is no ringing tone on my end... Just silence, then audio ~1-2 secs after both legs are established... is it supposed to work like that?? |
| 13:53.14 | xphree | hmm, the CLI doesn't show errors.. |
| 13:53.17 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:54.00 | xphree | simply the call isn't hangup by the timeout |
| 13:54.02 | rue_mohr | core set debug 10 |
| 13:54.07 | rue_mohr | core set verbose 10 |
| 13:54.16 | rue_mohr | then you see errors |
| 13:54.18 | Katty | peeks in |
| 13:54.35 | xphree | rue_mohr: thanks, i know that, i have the debug and verbose |
| 13:54.56 | rue_mohr | and it dosn't say anything giving clues about your problem? |
| 13:55.54 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 13:56.17 | xphree | i have this error always [Jun 23 08:55:13] ERROR[2103]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 13:56.28 | rue_mohr | ah |
| 13:56.37 | rue_mohr | broken pipes leak data |
| 13:56.58 | rue_mohr | your cpu could be swimming in gigabytes of bits |
| 13:57.18 | xphree | and i'm using Asterisk 1.4.25 |
| 13:57.46 | rue_mohr | xphree, maybe you shoudl pastebin the log in the area of that error so someone who isn't me can help you diagnose it |
| 13:59.15 | xphree | hhmm that is not my error, my error is that in SIP calls the L() param in the DialCommand doesnt' work... and i testet with the S() param and nothing.. it only hangup calls via DAHDI trunk |
| 13:59.48 | Katty | hum. this Alexander Rybak is quite talented. |
| 14:00.09 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:00.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.23 | *** join/#asterisk ingenius (n=alektro@186.13.19.185) |
| 14:00.41 | kaldemar | Katty: please don't curse. |
| 14:01.45 | Katty | kaldemar: baroo? |
| 14:02.48 | kaldemar | Katty: yes. rybak + talented. :) |
| 14:04.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:05.34 | *** join/#asterisk chendy (n=chatzill@219.134.230.50) |
| 14:06.18 | Katty | kaldemar: if you go through life acting like that, you're going to be missing out (= |
| 14:06.35 | Katty | kaldemar: so stop being so negative! |
| 14:07.40 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:07.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:08.43 | *** join/#asterisk SparFux (n=raoul@e182022183.adsl.alicedsl.de) |
| 14:09.23 | SparFux | Hello all. Has anybody sucessfully installed dahdi with zaphfc on a debian box yet? |
| 14:10.38 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:10.56 | *** join/#asterisk uluatu (n=uluatu@189.32.44.21) |
| 14:14.10 | *** join/#asterisk mazpe (n=mazpe@c-98-203-52-170.hsd1.fl.comcast.net) |
| 14:16.11 | kaldemar | Katty: based on what i've heard, i will be completely happy missing the rest of him out. |
| 14:17.08 | xphree | thank guys, is working right now... it was a misconfiguration of the SIP trunks.. |
| 14:20.02 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 14:22.21 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:22.53 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:23.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:23.36 | mazpe | are polycoms still recommended over ciscos? I have a choice to make between a 7960 and Polycom 330.. the main features to support is intercom |
| 14:23.58 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
| 14:24.40 | mazpe | based on voip-info.org it seems that only the Polycom 600 with auto-answer will support a paging/intercom |
| 14:25.14 | mazpe | 500/600 that is |
| 14:26.05 | Joel | mazpe polycom. |
| 14:26.26 | mazpe | the customer really needs 2 lines per phone. |
| 14:26.28 | j_kroon | ok, who here is familiar with the SIP and IAX/2 timing code that would be able to assist me a little later? |
| 14:26.38 | j_kroon | probably about two hours. |
| 14:26.58 | mazpe | Joel: the 330 seems like a good choice... I just want to make sure that the intercom/paging is supported and can be done with asterisk |
| 14:30.42 | SuPrSluG | coppice: i checked for one way audio on a softphone and I got 2 way audio. any other reasons I'd that DCN message? |
| 14:30.58 | SuPrSluG | I'd get.... |
| 14:31.40 | Katty | eppigy: poke. |
| 14:31.45 | coppice | you said the FAX is T.38. Do you still have 2 way communication after the switchover? |
| 14:32.17 | eppigy | hi |
| 14:32.18 | eppigy | HI |
| 14:32.22 | Katty | sleepy :< |
| 14:32.31 | eppigy | I am hungry |
| 14:32.35 | eppigy | :[ |
| 14:32.38 | Katty | waffles? |
| 14:32.47 | eppigy | oh man that sounds good |
| 14:32.49 | Katty | with peanut butter. |
| 14:33.03 | Katty | or honey. or coolwhip |
| 14:33.05 | eppigy | D: |
| 14:33.13 | eppigy | i just like butter and sirup |
| 14:33.18 | *** join/#asterisk odenkos (n=feeds@85-135-156-24.adsl.slovanet.sk) |
| 14:33.18 | eppigy | I am old fashioned |
| 14:33.21 | rob0 | kewlwhip or loopwhip? |
| 14:33.23 | Katty | blubber. |
| 14:33.36 | [TK]D-Fender | mazpe: All of Polycom's SPIP series support paging |
| 14:33.38 | Katty | i likes teh mapley syrups tho |
| 14:33.46 | eppigy | yesh |
| 14:34.00 | Katty | nutrigrain makes waffles you can stick in the toaster |
| 14:34.05 | Katty | they are whole grain waffles. |
| 14:34.26 | eppigy | I like eggos with aunt jemima |
| 14:34.27 | mazpe | [TK]D-Fender: so probably a good choice to go with the Polycom 330 eh? that price is not bad at all |
| 14:34.28 | Katty | with lots of fiber, so you stay full longer |
| 14:34.37 | eppigy | BUTTERMILK |
| 14:34.40 | eppigy | WAFFLES |
| 14:34.42 | [TK]D-Fender | mazpe: Do you require the passthrough port? |
| 14:34.44 | eppigy | 8[] |
| 14:34.55 | Katty | never had buttermilk waffles. |
| 14:35.10 | mazpe | [TK]D-Fender: for the codec? |
| 14:35.13 | eppigy | OH JEEZE |
| 14:35.15 | eppigy | whoops |
| 14:35.24 | eppigy | I have never had home made |
| 14:35.30 | eppigy | but the toaster ones are pretty good |
| 14:35.31 | Katty | WHAT?!?!!?!?! |
| 14:35.40 | eppigy | i mean buttermilk |
| 14:35.41 | Katty | that is unacceptabuhls. |
| 14:35.47 | eppigy | I have had wafflehouse waffles |
| 14:35.47 | [TK]D-Fender | mazpe: What codec? I'm talking about your choice of the 330 over the 320 |
| 14:35.51 | eppigy | :D |
| 14:35.53 | Katty | those are not homemade. |
| 14:35.56 | eppigy | they are kind of home made |
| 14:35.58 | eppigy | o |
| 14:36.00 | eppigy | :< |
| 14:36.07 | Katty | the best type of homemade waffles uses muffin mix. |
| 14:36.13 | mazpe | [TK]D-Fender: oh, i thought you meant pass through for g729 |
| 14:36.23 | Katty | and pancakes too. it is my secret. |
| 14:36.27 | Katty | that you may have. |
| 14:36.30 | [TK]D-Fender | mazpe: No, the IP 320 = IP 330 w/o the switched port |
| 14:36.31 | mazpe | [TK]D-Fender: yes i do |
| 14:36.37 | mazpe | thats why the 330 |
| 14:36.41 | [TK]D-Fender | mazpe: And saves a good bit of extra $ |
| 14:36.45 | Katty | muffin mix + a bit of water = perfect pancake or waffle mix. |
| 14:36.50 | eppigy | I like jiffy mix pancakes |
| 14:36.54 | eppigy | apple cinnamon |
| 14:37.00 | Katty | fluffy blueberry muffin pancakes. |
| 14:37.08 | Katty | banana nut pancakes. |
| 14:37.09 | [TK]D-Fender | mazpe: the difference in cost might validate creating a more diverse infrastructure. |
| 14:37.21 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-3f913c3f93c6fffa) |
| 14:37.21 | Katty | drizzled with honey, and powdered sugar |
| 14:37.23 | eppigy | blueberry sounds good |
| 14:37.41 | mazpe | [TK]D-Fender: Yeah, we actually have one drop per desk, so we will need the passthrough port of the 330 |
| 14:38.30 | Katty | cinnamon streusel pancakes. |
| 14:38.42 | Katty | lemon poppy seed pancakes. |
| 14:39.15 | [TK]D-Fender | mazpe: What I was saying is that depending on the number of stations it might be worth it to make separate drops for your telephony network with the money saved in choosing the IP 320 instead. Worth considering |
| 14:39.47 | mazpe | 20 stations |
| 14:39.58 | eppigy | OH MAN |
| 14:40.33 | mazpe | but will have to get drops and switches... which using the catalyst 2950 (very inexpensive now a days) with QoS might be something to consider |
| 14:40.49 | mazpe | I'm with you on that one. |
| 14:43.08 | [TK]D-Fender | mazpe: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=4648934&CatId=2800 |
| 14:43.15 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 14:43.23 | [TK]D-Fender | mazpe: Power your phones (actualy covers the cost of power bricks |
| 14:44.16 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 14:44.23 | mazpe | aside from the lack of PoE, it seems that no one likes the 2950 |
| 14:44.41 | [TK]D-Fender | mazpe: So cheap PoE switch = cost of bricks, 320 vs 330 price drop helps pay for wiring infrastructure, and your overall deployment looks cleaner and allows you not to tie down your stations to 10/100 |
| 14:45.18 | mazpe | Yeah, this is a good choice. |
| 14:46.05 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:46.52 | [TK]D-Fender | mazpe: well, this is jsut a suggestion, take a close look at your physical building and consider the particulars. Might well be a good idea. Everything "depends". |
| 14:48.00 | merkurie | whats everyone using for a call/queue statistics/metrics package? |
| 14:48.16 | merkurie | something with pretty numbers and graphs |
| 14:48.28 | [TK]D-Fender | merkurie: MS Excel |
| 14:48.40 | mazpe | :) |
| 14:48.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:48.47 | merkurie | just suck in the csv data and make some charts and what not? |
| 14:49.00 | merkurie | anything more automated? I was thinking something with some php in it |
| 14:49.23 | merkurie | i see there are some that people want money for |
| 14:49.29 | merkurie | i also found queue-tip |
| 14:49.36 | merkurie | doesn't seem too popular |
| 14:49.53 | [TK]D-Fender | merkurie: Depends where they stuck it ;) |
| 14:50.09 | merkurie | [TK]D-Fender, =) |
| 14:50.22 | [TK]D-Fender | merkurie: Go look at QueueMetrics, and check out whichever one Trixbox bundles |
| 14:50.42 | merkurie | [TK]D-Fender, k |
| 14:52.31 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:54.12 | jplank | queuemetrics is pretty good, I have a few installations running it, all the end users are happy with it. its not cheap though |
| 14:54.25 | jplank | any comments on lumenvox? |
| 14:56.27 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:57.13 | j0n23 | hey,is it possible for me to conenct a 4 port calling card with asterisks and can assign 4 independed accounts to those? |
| 14:58.53 | [TK]D-Fender | j0n23: What is a "4 port calling card"? And what "accounts" are trying to "tie" to it? |
| 14:59.02 | merkurie | anyone have comments on Asterisk Guru Queuestats? |
| 15:02.31 | j0n23 | [TK]D-Fender: i want to conenct 4 phone lines to pci device or something like that,and bind each lines with dif sip account? |
| 15:03.09 | Joel | jplank what about lumenvox? |
| 15:03.18 | j0n23 | and managing those using asterisks |
| 15:03.20 | [TK]D-Fender | j0n23: Digium & Sangoma both made suitable cards for this, for which I HIGHLY recommend you get one with ahrdware echo cancellation. |
| 15:03.26 | [TK]D-Fender | j0n23: the rest is all dialplan |
| 15:03.39 | j0n23 | aah nice |
| 15:04.30 | j0n23 | [TK]D-Fender: so is it possible for me to configure stuff inside asterisks such that the call wil get disconencted after a time interval randomly in 4 ports? |
| 15:04.31 | superbeef | j0n23: so you want to connect 4 POTS to your asterisk box, then route them to sip extensions |
| 15:04.42 | j0n23 | superbeef: exactly |
| 15:05.00 | [TK]D-Fender | j0n23: Sure |
| 15:05.19 | j0n23 | [TK]D-Fender: can you recommend a device to me? |
| 15:05.32 | [TK]D-Fender | j0n23: Sangoma A200d |
| 15:05.34 | *** join/#asterisk s0lid (n=s0lid@61.28.187.210) |
| 15:05.36 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:05.38 | superbeef | j0n23: A200d |
| 15:05.45 | [TK]D-Fender | j0n23: Sorry, actually, get the Sangoma B600d instgead. |
| 15:05.52 | [TK]D-Fender | j0n23: Unless you're looking to expand |
| 15:05.59 | Katty | eppigy: can't wake up :< |
| 15:06.16 | eppigy | D: |
| 15:06.20 | eppigy | JOG IN PLACE |
| 15:06.25 | eppigy | choke someone |
| 15:06.37 | jplank | Joel: I just wanted to hear if anyone has used it and if they had any comments on it. I have a customer requesting it, and I never worked before with it |
| 15:06.54 | Katty | eppigy: i'll do some stretching |
| 15:06.56 | Joel | jplank we use it, with our magic button :) |
| 15:07.01 | eppigy | good idea |
| 15:07.26 | jplank | merkurie: Asterisk Guru's QueueStats is good. Not as good as QueueMetrics, but good. |
| 15:07.29 | j0n23 | [TK]D-Fender: thx,and if want to expand in teh future which card you will recommend |
| 15:07.42 | jplank | Joel: it works well? |
| 15:09.15 | Joel | jplank it works |
| 15:09.39 | Katty | i feel like going to some little coffee shop with an oversized couch and really good hot chocolate. |
| 15:09.45 | Joel | jplank its a bit spendy per channel |
| 15:10.20 | Katty | and maybe some accoustic guitar soloist. |
| 15:10.53 | talirk81 | Is it possible to set the CallerID name from an AGI directly, SET CALLERID sets the name to the number specified not like callerid(name)="NAME" or callerid(full)="NAME"<NUMBER> |
| 15:11.29 | [TK]D-Fender | j0n23: I already told you |
| 15:11.47 | j0n23 | oh ok :) |
| 15:11.51 | [TK]D-Fender | j0n23: A200d if you're thinking of expanding, B600d if you aren't |
| 15:12.35 | j0n23 | so this 2 cards perfectly work with asterisks,and linux kernel got support for this one,do i need to compile the kernel manually? |
| 15:14.06 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:15.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:15.28 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 15:15.41 | [TK]D-Fender | j0n23: If your kernel changes you've halve to recompile DAHDI & in the case of Sangoma, their Wanpipe driver as well |
| 15:17.26 | j0n23 | [TK]D-Fender: any idea if the default debian/centos/ubuntu kernels got teh basic support compiled in for these devices,and if so i can manually compile sangoma driver |
| 15:17.44 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:18.21 | [TK]D-Fender | j0n23: Any will do |
| 15:19.34 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:19.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:20.42 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:22.43 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:27.15 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:28.00 | Katty | eppigy: i do believes i might possibly be awake :> |
| 15:28.08 | eppigy | oh grood! |
| 15:28.16 | Katty | just in time for lunch :> |
| 15:28.47 | eppigy | yes indeed |
| 15:28.50 | eppigy | I am so readu |
| 15:28.53 | eppigy | ready |
| 15:28.55 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 15:29.00 | Katty | eppigy: brunch? |
| 15:29.03 | Katty | eppigy: bacon and eggs. |
| 15:29.20 | eppigy | dude I love breakfast food |
| 15:29.20 | degrade | Does anybody using Asterisk with IBM Power6 plataform? |
| 15:29.28 | eppigy | I will eat it any time anywhere |
| 15:29.35 | Katty | in a tortilla? |
| 15:29.58 | eppigy | I like breakfast burritoes too |
| 15:30.02 | eppigy | :D |
| 15:30.44 | Katty | bob evans would be super for lunch. |
| 15:30.53 | Katty | or crackerbarrel. |
| 15:31.13 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 15:32.03 | VoipForces | Anyone using freepbx paging? I get phones that get stuck in meetme rooms following a page. |
| 15:32.27 | eppigy | agreed |
| 15:32.37 | eppigy | I love crackerbarrel so much |
| 15:32.42 | rue_mohr | no, but anyone know how to make it so that current calls dont get put on hold while the phone plays a page? |
| 15:33.14 | *** join/#asterisk ming_zym (n=ming_zym@220.181.34.198) |
| 15:33.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:33.18 | [TK]D-Fender | rue_mohr: Shouldn't happen on Polycom phones |
| 15:33.32 | rue_mohr | hmm |
| 15:33.40 | rue_mohr | I did only try it on an aastra |
| 15:34.34 | VoipForces | rue_mohr: Check the "BargeIn" setting under preferences in the phone gui. |
| 15:34.42 | rue_mohr | hmm |
| 15:35.49 | VoipForces | rue_mohr: make sure you have the latest firmware, if bargein is enabled, it will put the current call on hold. |
| 15:36.37 | rue_mohr | k, checking now |
| 15:36.54 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 15:37.14 | rue_mohr | :) there it is |
| 15:42.46 | Katty | eppigy: hashbrown casserole :> |
| 15:43.18 | eppigy | oh snap |
| 15:43.19 | eppigy | yes |
| 15:43.22 | eppigy | I love it |
| 15:44.19 | VoipForces | nlinux: works now? |
| 15:45.33 | Katty | eppigy: is it acceptable to go to cracker barrel alone? |
| 15:45.57 | Katty | eppigy: my lunch person has other things she needs to do :< |
| 15:46.32 | eppigy | I have before |
| 15:47.39 | Katty | k |
| 15:47.45 | SuPrSluG | coppice: thanks for the help. I can pass faxes through to an audio codes fine. But when I try to use callweaver as a fax server it gets the DCN error. |
| 15:50.09 | coppice | it sounds like a one way signal. a few things mess up when the change from RTP to UDPTL occurs and there doesn't seem to be universal cure. a wireshark trace mih show something interesting |
| 15:52.03 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 15:54.35 | hackt0r | is there any way to configure the senddtmf(which is in misdn.conf) from the Asterisk GUI? |
| 15:55.08 | [TK]D-Fender | hackt0r: Go ask in the GUI's support channel |
| 15:55.16 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:55.51 | odenkos | hackt0r: /join #asterisk-gui |
| 15:56.00 | hackt0r | k thnx |
| 16:01.08 | rue_mohr | hmm paging requires alert-info to be different to the different phones |
| 16:01.13 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 16:07.58 | *** join/#asterisk s0lid (n=s0lid@61.28.187.210) |
| 16:10.28 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:11.18 | *** part/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 16:11.48 | VoipForces | rue_mohr: yup... that's why I avoid mixing makes at all cost |
| 16:14.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:14.41 | *** join/#asterisk BadHAL (n=nn@70.99.106.38) |
| 16:16.25 | e0n` | What's a pretty decent setup to run a 1-30 employee asterisk server, I've got it externally at this point and issues with multiple carriers inbetween us and the phone system kind of degrades phone system quality. |
| 16:17.07 | KavanS | e0n`, a pentium 3 might take care of you then |
| 16:17.24 | e0n` | Figured I didn't need anything beefy. |
| 16:17.49 | VoipForces | e0n`: I never use anything below a Xeon with server grade motherboard. |
| 16:18.50 | *** join/#asterisk thansen (n=thansen@c-24-10-236-153.hsd1.ut.comcast.net) |
| 16:19.07 | KavanS | lol |
| 16:19.26 | e0n` | heh |
| 16:19.45 | KavanS | e0n unless you are planning on setting up 30 pr0n video streams, I seriously doubt you will need anything beefy |
| 16:19.48 | e0n` | None of my equipment is below a tyan based or supermirco based rig, just trying to get a good estimate |
| 16:20.17 | e0n` | KavanS: We're growing though so i figured in contingency plan for the future as well as the current |
| 16:20.33 | KavanS | then definitely grab something new |
| 16:21.12 | e0n` | Little xeon 1000 series wouldn't be too bad I don't think |
| 16:21.40 | e0n` | s/1000/3000/ |
| 16:21.50 | e0n` | :) |
| 16:26.52 | *** join/#asterisk talirk81 (n=talirk@rrcs-67-78-39-22.sw.biz.rr.com) |
| 16:28.30 | talirk81 | If I am bridging 2 dialed in calls in asterisk when one leg hangs up the other leg is stay up and active , is there a way inside of AGI or a dial plan, I can send a tone that tells the other leg to also hang up? |
| 16:33.02 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:36.58 | *** join/#asterisk wpbrown (n=wpbrown@wh-gtw-0001.woolfharris.com) |
| 16:37.21 | wpbrown | Hi peeps |
| 16:37.49 | wpbrown | Anyone want to translate this for me |
| 16:37.54 | wpbrown | [930234.670349] asterisk[18281]: segfault at c ip b7e709f0 sp b397e414 error 4 in libpthread-2.8.90.so[b7e69000+15000] |
| 16:37.56 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:37.59 | wpbrown | in english |
| 16:39.08 | russellb | "it crashed" |
| 16:39.26 | wpbrown | lol |
| 16:39.35 | wpbrown | well yeah |
| 16:39.41 | wpbrown | but what does it mean |
| 16:39.44 | russellb | "it crashed because of a seg fault" |
| 16:39.55 | russellb | that's honestly the best i can do from that |
| 16:40.17 | deadpigeon | google it. segfault in libpthread |
| 16:40.29 | deadpigeon | run debug. |
| 16:40.40 | russellb | googling is unlikely to be productive in this case ... |
| 16:40.51 | russellb | a backtrace from a core dump from the crash is ideal |
| 16:41.00 | russellb | (and having asterisk compiled with DONT_OPTIMIZE) |
| 16:41.28 | deadpigeon | ive never encountered much for segfaults myself. |
| 16:41.39 | deadpigeon | i get kernel panics on reboot sometimes, wont shut down properly. |
| 16:42.47 | deadpigeon | dealing with a strange issue, where one of my extensions hangs up his call and immediately his extension is ringing again, incoming from itself. |
| 16:43.08 | deadpigeon | cant quite figure out why. must be the ata. |
| 16:43.48 | e0n` | hmm, now for bandwidth, just running voip what's the average rate per phone call on different codec's |
| 16:44.03 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:44.56 | deadpigeon | average for me is 40.0kb per call. |
| 16:45.06 | e0n` | deadpigeon: What codec are you using? |
| 16:45.13 | deadpigeon | ulaw |
| 16:45.15 | e0n` | ah |
| 16:45.27 | e0n` | same here, haven't been able to average it out yet, so 40kb/sec on average? |
| 16:45.39 | deadpigeon | yea i average it based on what my mikrotik router says. |
| 16:45.54 | deadpigeon | considering on that connection is only 1 sip extension |
| 16:46.00 | deadpigeon | its definitely 40kb/sec |
| 16:46.31 | Qwell | ulaw is about 80k with overhead |
| 16:47.16 | deadpigeon | i havnt seen it up to 80k, but that could be right. |
| 16:47.26 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:47.37 | coppice | Qwell: not when you use 1ms packets, to avoid that nasty latency |
| 16:48.06 | Qwell | coppice: and violate the specs? never! |
| 16:48.17 | coppice | what does that violate? |
| 16:48.22 | Qwell | no clue |
| 16:48.42 | coppice | some people do actually use 1 or 2ms RTP packets |
| 16:48.55 | Qwell | yeah, that's a bit absurd |
| 16:49.10 | Qwell | I don't even want to think about what the overhead is there |
| 16:49.24 | coppice | ulaw at 40kb/s would be interesting. I think the laws against perpetual motion would have something to say about that |
| 16:49.46 | coppice | Qwell: people do us that kind of thing in conference setups |
| 16:49.54 | Qwell | why? |
| 16:50.00 | deadpigeon | i could be very well mistaken, my estimate is based on a quick glance at my connection tracking |
| 16:50.07 | Qwell | I mean, sure, the latency, but...for what purpose? |
| 16:50.55 | deadpigeon | of course i was probably only looking at what was being received |
| 16:50.55 | coppice | if you want the audio from multiple mic shunted around and reaching speakers without echo, you might take latency pretty seriously |
| 16:51.27 | Qwell | coppice: yeah, makes sense |
| 16:51.50 | coppice | CELT has been designed for 2ms algorithmic latency for this type of use |
| 16:52.11 | e0n` | so for 20 phones constant i'd need about 2 megabit upstream? |
| 16:52.28 | e0n` | with ulaw @ average of 80kbps |
| 16:54.39 | deadpigeon | that sounds about right. if its being shared with other traffic you might want to prioritize voip packets |
| 16:55.56 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:56.02 | [TK]D-Fender | Or not use SIP... or not use G.711... or not use 20ms packets... |
| 16:58.12 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:01.25 | j0n23 | hey,is it possible for me to upload the pin numbers of voip card using asterisks? |
| 17:01.37 | *** join/#asterisk jicksta (n=jicksta@c-67-188-112-50.hsd1.ca.comcast.net) |
| 17:01.42 | j0n23 | i mean in a more user firendly way |
| 17:01.44 | *** join/#asterisk JackStorm (n=no@ip24-252-118-155.no.no.cox.net) |
| 17:03.05 | [TK]D-Fender | j0n23: HUH?! |
| 17:03.18 | [TK]D-Fender | j0n23: What "voip card"? Upload into what? to do what? |
| 17:04.09 | j0n23 | <--noob here |
| 17:04.12 | j0n23 | hehe |
| 17:04.31 | rob0 | Count all the pins & sockets on a DAHDI card, then blog it? |
| 17:04.56 | j0n23 | [TK]D-Fender: in asterisks,for voip,do we need to enter the voip cards pin number? |
| 17:05.04 | j0n23 | for access |
| 17:05.10 | j0n23 | liek we enter in a voip modem? |
| 17:05.15 | Qwell | let's just go with yes |
| 17:06.13 | kaldemar | j0n23: what do you mean by a voip card? what do you mean by PIN? what do you mean by a voip modem? |
| 17:06.43 | [TK]D-Fender | j0n23: there is no such thing as a "voip card". |
| 17:06.50 | *** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net) |
| 17:06.56 | [TK]D-Fender | j0n23: this is not a long distance calling card. |
| 17:07.16 | j0n23 | [TK]D-Fender: if i buy a voip service,they will give me an account with thier usr/pas? |
| 17:07.20 | [TK]D-Fender | j0n23: You typically prepay a SERVICE PROVIDer and the give you an account you setup with * |
| 17:07.26 | [TK]D-Fender | jonYES |
| 17:07.29 | j0n23 | thats what :) |
| 17:07.32 | j0n23 | i mean |
| 17:07.33 | j0n23 | sorry |
| 17:07.58 | j0n23 | and for voip to work in asterisks do i need to enter those details in it? |
| 17:08.09 | j0n23 | access info |
| 17:08.32 | talirk81 | j0n23, most providers give you details on what settings ot put in sip.conf |
| 17:08.45 | j0n23 | yes |
| 17:08.48 | rob0 | For SIP, there are statements like "register" in sip.conf |
| 17:09.04 | Jumpie | anybody know why my phones wont accept input but the 9 key? |
| 17:09.38 | rob0 | Jumpie: they're from side 4 of the Beatles white album. |
| 17:09.50 | Jumpie | and trying on xlite i cant call off the sip "trunk" i always get channel congestion/unavail |
| 17:09.53 | Jumpie | lol rob0 |
| 17:09.54 | rob0 | number 9 ... number 9 ... |
| 17:09.59 | j0n23 | talirk81: in those settings do they give us a pin number? |
| 17:10.00 | talirk81 | Anyone know if i had a longrunning DEADAGI script, how i can get * to actually hang up that sip channel while its running |
| 17:10.25 | talirk81 | j0n23, you dont ussaly have a pin number , just a user/pass in the register command |
| 17:10.48 | rob0 | j0n, ideally your service provider will tell you how to set up your asterisk. |
| 17:11.45 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 17:12.03 | KavanS | any suggestions for good prices on ATA adapters? |
| 17:12.09 | KavanS | i.e. not ebay |
| 17:12.35 | carrar | criagslist! |
| 17:12.57 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:13.03 | [TK]D-Fender | rob0: JOHNIE FIVE ALIVE! |
| 17:13.08 | Jumpie | this is driving me bananas |
| 17:13.24 | talirk81 | http://www.google.com/products?q=pbx+ATA+adapters+-ebay |
| 17:13.29 | [TK]D-Fender | Jumpie: then get some ice-cream & chocolate syrup |
| 17:13.37 | KavanS | lol thanks talirk |
| 17:13.45 | KavanS | I was looking for first hand recommendations |
| 17:13.53 | rob0 | mmmm, nana split! |
| 17:13.54 | [TK]D-Fender | KavanS: www.telephonydepot.com |
| 17:13.55 | KavanS | I got a counterfeit pap2, and don't want to repeat |
| 17:14.46 | KavanS | [TK]D-Fender, thanks, that's what I was looking for |
| 17:14.53 | ariel_ | I did not know there were counterfeit pap2 |
| 17:15.03 | KavanS | ariel_, http://text.broadbandreports.com/forum/r21331513-Equipment-Counterfeit-Linksys-PAP2 |
| 17:15.31 | j0n23 | rob0: talirk81 i agree,but is there an easy interface available in asterisks for updating that info,i mean for common people? |
| 17:15.48 | talirk81 | j0n23, FREEPBX is good. |
| 17:16.00 | [TK]D-Fender | j0n23: man vi <- |
| 17:16.01 | talirk81 | even installs the os for you and give you a web gui for configs |
| 17:16.12 | Qwell | FreePBX is not an OS. |
| 17:16.27 | talirk81 | [TK]D-Fender, is they any way i can force a SIP BYE before the h extension in a dial plan is done ? I have a long running DEADAGI command i want to run, but the chan stays active until all the DEADAGI's finish |
| 17:16.36 | talirk81 | FreePBX installs an OS |
| 17:16.41 | [TK]D-Fender | Qwell: the new Sony car steroe even buys your car for you and isntalls itself! |
| 17:16.53 | talirk81 | it uses CentOS5 currently i believe if you get their iso |
| 17:17.02 | [TK]D-Fender | Talkradio: FreePBX is 1 tiny piece of bolt-on software. If does not install a friggen OS |
| 17:17.02 | Qwell | FreePBX doesn't have an ISO. |
| 17:17.23 | [TK]D-Fender | TalFreePBX is ONLY the dumb GUI, and not even the DATABSE is uses to stare the data |
| 17:17.27 | talirk81 | oh sorry, i was think Pbx in a box |
| 17:17.43 | [TK]D-Fender | talirk81: PBX In a FLASH. |
| 17:17.58 | [TK]D-Fender | talirk81: Might want to hold off on your use of nouns for a while... |
| 17:18.10 | [TK]D-Fender | talirk81: till you can get the straight at least :p |
| 17:18.14 | [TK]D-Fender | them* |
| 17:18.49 | talirk81 | lol true enough, i hate that product but it was good as a starter when i first started dealing with Asterisk and didnt know much on the conf files |
| 17:19.13 | [TK]D-Fender | Talkradio: trying to learn for that GUI is such a waste. |
| 17:19.18 | [TK]D-Fender | talirk81: ^^ |
| 17:20.10 | [TK]D-Fender | talirk81: So much convoluted crap that 99% of those deluded enough to think it'll be their launching pad to learning * end up lUSERs |
| 17:21.09 | talirk81 | [TK]D-Fender: regarding my DEADAGI question , the only thing i can think of is using a fork in php to split off the long running process of the php script so the the channel can close, but that seems kinda ugly :( |
| 17:21.32 | rob0 | As with so many other things in life, there really is no substitute for learning, and no learning without effort. |
| 17:22.05 | [TK]D-Fender | talirk81: Or fork off a countdown process targeting the PID of the spawned AGI |
| 17:22.25 | eppigy | TRANAJO |
| 17:22.28 | eppigy | TRABAJO |
| 17:22.51 | [TK]D-Fender | eppigy: PHAIL |
| 17:22.58 | eppigy | yes |
| 17:23.04 | eppigy | el missfire |
| 17:23.10 | eppigy | DONDE ESTA |
| 17:23.29 | rob0 | Mooey bean, gracious. |
| 17:23.40 | ariel_ | wow |
| 17:23.44 | ariel_ | spanish 101 |
| 17:27.03 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 17:27.24 | jplank | anyone have a ip330 handy running 3.0? |
| 17:27.48 | jplank | does it have the extension number on the screen? |
| 17:28.01 | ariel_ | humm no |
| 17:28.07 | ariel_ | just a little phone icon |
| 17:28.52 | [TK]D-Fender | jplank: It has a tag nedt to whatever line keys are allocated towards a registration... |
| 17:32.35 | jplank | in mac-cfg? |
| 17:32.52 | comfrey | Your search is over! |
| 17:32.52 | comfrey | Club Bisexual has all the bi porn action you\u2019ve been looking for! Hot guys loving to suck cock, fuck and get fucked! The horny babes suck cocks while the guys take one in the ass! Incredible fuck combos in wild orgies! And if you like it, let\u2019s not forget the girls with strapons! We are the best and biggest bisexual movies collection online! |
| 17:32.57 | comfrey | The best collection online |
| 17:32.59 | comfrey | Over 450 movies: MMF threesomes, bisexual orgies, girl on guy strapon action! Download and keep everythin |
| 17:33.03 | comfrey | Over 450 movies: MMF threesomes, bisexual orgies, girl on guy strapon action! Download and keep everythinYour search is over! |
| 17:33.06 | comfrey | Club Bisexual has all the bi porn action you\u2019ve been looking for! Hot guys loving to suck cock, fuck and get fucked! The horny babes suck cocks while the guys take one in the ass! Incredible fuck combos in wild orgies! And if you like it, let\u2019s not forget the girls with strapons! We are the best and biggest bisexual movies collection online! |
| 17:33.08 | Aiatek | kill it |
| 17:33.12 | comfrey | The best collection online |
| 17:33.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.29 | jplank | damn pidgin doesn't have /ignore |
| 17:33.58 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 17:34.21 | jplank | fender: are you talking about the reg.1.label field? |
| 17:34.30 | [TK]D-Fender | jpYup |
| 17:34.41 | jplank | that doesn't seem to work on a ip330 running 3.0 |
| 17:34.47 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:34.53 | *** mode/#asterisk [+b *!*@66.166.226.6] by [TK]D-Fender |
| 17:34.54 | *** kick/#asterisk [comfrey!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 17:37.58 | KavanS | wow |
| 17:40.41 | russellb | thanks [TK]D-Fender |
| 17:41.18 | ariel_ | does the digium dsp board for g729 show up when you do show translation |
| 17:41.45 | russellb | it should, yes |
| 17:41.58 | russellb | it should be accounted for in the translation paths involving g729 |
| 17:42.29 | ariel_ | how do we load it up or start it up from the prompt |
| 17:42.44 | russellb | you'll have to contact tech support for detailed assistance |
| 17:42.54 | russellb | i know you need the dahdi driver loaded, and codec_dahdi.so loaded in asterisk |
| 17:42.57 | jplank | ariel_: guess there is an option sip.cfg (not reg.1.label in mac-cfg) that controls that. I just grabbed the newest sip.cfg, edited it, and the extension now shows up |
| 17:42.57 | russellb | that's about as far as i can go |
| 17:43.07 | jplank | funny I never seen that problem in 3.1 |
| 17:43.44 | ariel_ | russellb: no problem, I am beginning to be a regular call as of late to digium, I have to call them about an rma anyway.. |
| 17:44.09 | jplank | maybe you could check on my HPEC while your on with them :) |
| 17:44.14 | ariel_ | it's just faster getting a quick response here |
| 17:44.19 | russellb | nods |
| 17:44.23 | russellb | wish i had the answer |
| 17:45.32 | [TK]D-Fender | russellb: Still haven't broken double-digits on the KB counter ;) |
| 17:45.52 | russellb | heh |
| 17:46.05 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 17:47.30 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:47.58 | *** mode/#asterisk [+b *!comfrey@*] by [TK]D-Fender |
| 17:48.00 | [TK]D-Fender | better |
| 17:48.14 | [TK]D-Fender | He's on dynamic IP's from the same ISP I see |
| 17:48.28 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 17:49.37 | Qwell | [TK]D-Fender: #freenode is usually a pretty good place to report that too. typically they spam a bunch of channels, and an oper can just kline it |
| 17:49.58 | [TK]D-Fender | Qwell: Thing is he was here legimately late laste year |
| 17:50.04 | [TK]D-Fender | (checked IRC logs) |
| 17:50.09 | Qwell | O.o |
| 17:50.14 | jplank | zombie? |
| 17:50.21 | [TK]D-Fender | Qwell: Yeah, I know... |
| 17:50.21 | Qwell | unlikely |
| 17:50.23 | [TK]D-Fender | jplank: Possible... |
| 17:50.35 | Qwell | zombies don't typically ident to nickserv |
| 17:50.43 | Qwell | (using a valid nick) |
| 17:56.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:57.08 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 17:57.43 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:58.33 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 18:01.01 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 18:03.02 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 18:04.50 | *** join/#asterisk rdeuce (n=rdeuce@66.54.249.50) |
| 18:05.18 | rdeuce | Hey gents, does anyone know of any problem with using Asterisk on a system with bonded NICs (mode=4, LACP)? |
| 18:05.46 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 18:06.11 | grandpapadot | Hey guys, I need a reliable/trustworth source for some Polycom 330's. Telephony Depot is out and I need 21 of them quick. |
| 18:06.12 | rdeuce | I am really freaking out here. We have a 60 terminal deployment with a private T1 used for SIP trunks, but every now and then we get dropped calls/one-way audio. |
| 18:06.14 | grandpapadot | Any suggestions? |
| 18:06.48 | *** join/#asterisk Deepa|Away (n=lisa@wrongplanet/deepa) |
| 18:06.50 | rdeuce | It happens on any terminal we try. We have some Polycom units and some LinkSys. |
| 18:06.54 | rdeuce | SPA-942's |
| 18:07.12 | Deepa|Away | Is it possible to call directly to an IP Phone? the adapter itself, the box. |
| 18:07.31 | rdeuce | Networking components are a trio of Cisco 2960-PoE switches connected to an HP ProCurve 5412zl. |
| 18:07.45 | grandpapadot | TK: You recommend Telephony Depot, good choice and we've used them a lot, any suggestions for an alternate for some 330's? They are backordered from them for two weeks. |
| 18:07.58 | [TK]D-Fender | Deepa|Away: How do you call an IP phone INDIRECTLY? |
| 18:08.16 | [TK]D-Fender | grandpapadot: http://www.ipphone-warehouse.com/ |
| 18:08.32 | rdeuce | Each Cisco switch has their dual gige ports bonded back to the ProCurve, the output of 'show lacp' shows the trunks are up and active. |
| 18:08.36 | [TK]D-Fender | grandpapadot: No personal experience, but have heard recommendations of others and their price is agressive |
| 18:09.51 | rdeuce | Asterisk servers are CentOS on HP DL360g5's with dual gige NICs configured in a LACP trunk to the ProCurve. |
| 18:10.03 | Deepa|Away | [TK]D-Fender, good point |
| 18:10.24 | rdeuce | There aren't any port issues/disconnects that I can see. |
| 18:10.27 | Deepa|Away | Through a PBX :-)? |
| 18:10.35 | [TK]D-Fender | Deepa|Away: Grab a softphone, dial by IP, presto |
| 18:10.36 | rdeuce | Voice traffic is on an isolated VLAN. |
| 18:10.50 | Deepa|Away | Got any windows softphone to recommend? |
| 18:11.00 | Deepa|Away | I got ekiga |
| 18:11.08 | rdeuce | The calls just flat out seem to disappear. |
| 18:11.23 | rdeuce | End users also report static. |
| 18:12.15 | rdeuce | Echoing as well. |
| 18:12.31 | rdeuce | I can't find any latency in the network... obviously, this is overprovisioned like a son of a bitch. |
| 18:12.35 | [TK]D-Fender | rdeuce: That does not sound like a network issue |
| 18:12.48 | [TK]D-Fender | rdeuce: That sounds like an analog interface failure |
| 18:12.55 | [TK]D-Fender | Deepa|Away: Ekiga |
| 18:12.58 | rdeuce | Well, static yes -- but we have nothing analog here. |
| 18:13.09 | Deepa|Away | Sweet, I'll hunt through the manual even more. Thank you :-) |
| 18:13.09 | rdeuce | It seems random, on any handset. |
| 18:13.15 | [TK]D-Fender | rdeuce: Same goes for echo |
| 18:13.37 | rdeuce | Are you suggesting that there's a problem with the handsets? |
| 18:14.42 | [TK]D-Fender | rdeuce: No, with your termination provider |
| 18:17.07 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 18:18.52 | Deepa|Away | Can I call an IP on a non-softphone? |
| 18:19.00 | Deepa|Away | normal wired phone connect to PAP2T |
| 18:20.30 | Deepa|Away | connected* |
| 18:26.41 | rdeuce | [TK]D-Fender: Thanks... I just can't believe it's our equipment... christ |
| 18:27.00 | rdeuce | Irritating because we've lost a sale over shitty phones and so now, it's a freak out. |
| 18:27.29 | rdeuce | The guy called us to tell us he was leaving because he couldn't reach our support department. In the middle of THAT call, it dropped again. |
| 18:27.33 | rdeuce | Damned irritating. |
| 18:27.58 | Qwell | rdeuce: who is your provider? |
| 18:28.51 | [TK]D-Fender | Deepa|Away: Its all jsut IP, and jsut about every IP phone can dial by IP. |
| 18:29.14 | Katty | stretches |
| 18:29.26 | Katty | that was a lovely nap |
| 18:29.44 | Deepa|Away | I added <11S0<:@80.244.xxx.x:5060> at the end of my dial plan |
| 18:29.47 | Deepa|Away | That ought to work right? |
| 18:30.34 | Qwell | Deepa|Away: is your username 11S0< ? |
| 18:30.55 | Deepa|Away | Nope |
| 18:31.12 | Deepa|Away | To be honest, I don't know the username of the other device. it's pretty tightly locked down and I haven't been able to reach the webinterface yet |
| 18:32.27 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:35.25 | grandpapadot | TK: Thanks! |
| 18:36.12 | *** join/#asterisk ingenius (n=alektro@186.13.19.185) |
| 18:38.22 | [TK]D-Fender | Deepa|Away: Vendor-loceked device? |
| 18:38.44 | [TK]D-Fender | Qwell: Thats an SPA dialplan |
| 18:39.34 | Deepa|Away | Not sure really, haven't looked close at it |
| 18:40.16 | [TK]D-Fender | Deepa|Away: This is your device isn't it? You don't know what you have? |
| 18:40.38 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 18:41.02 | Deepa|Away | Well it's at my Mother's house |
| 18:43.34 | [TK]D-Fender | Deepa|Away: And you don't know where it came from? |
| 18:45.17 | Deepa|Away | It's a custom tailored one :-) |
| 18:45.43 | Deepa|Away | sorta |
| 18:45.51 | Deepa|Away | The website had three links and the manual was a small flyer |
| 18:47.36 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 18:48.35 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:48.49 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-128-42.telkomadsl.co.za) |
| 18:51.29 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:51.38 | *** join/#asterisk jicksta (n=jicksta@c-67-188-112-50.hsd1.ca.comcast.net) |
| 18:51.39 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:51.54 | [TK]D-Fender | Deepa|Away: Ah... "Generic Crap". Gotcha |
| 18:52.30 | Deepa|Away | Indeed |
| 18:52.36 | Deepa|Away | It also overheats and is plugged in before the ruter :P |
| 18:52.37 | Deepa|Away | router* |
| 18:53.08 | Deepa|Away | I dial 11 and it just goes beep beep beep beep beep beep |
| 19:01.17 | nkohh | anyone here use metrostat for CNAM? |
| 19:08.21 | *** join/#asterisk s0lid (n=s0lid@122.53.110.183) |
| 19:09.22 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 19:12.36 | *** join/#asterisk lowtek (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 19:13.24 | lowtek | Anybody even seen a Polycom 550 give this error when trying to download sip.ld? http://pastebin.com/m9d6149d (It's firmware spip_ssip_3_1_1RevB_release_sig.zip from polycom.com) |
| 19:14.48 | pa | is astman/gastman any good/useful or is it old? |
| 19:15.53 | [TK]D-Fender | Pauseless and ancient |
| 19:15.58 | [TK]D-Fender | Pa: useless and ancient |
| 19:16.13 | pa | oh |
| 19:16.14 | pa | i see.. |
| 19:19.07 | *** join/#asterisk ingenius (n=alektro@186.13.153.16) |
| 19:19.25 | ariel_ | argh why is ABE so different.... |
| 19:20.37 | [TK]D-Fender | ariel_: From what? |
| 19:21.26 | ariel_ | abe gets configured from the iso with rpath, but my issue is it does not setup the zaptel or dahdi in any normal place so you can get the version number from it. |
| 19:22.04 | [TK]D-Fender | ariel_: ABE *is* the version number |
| 19:22.16 | [TK]D-Fender | ariel_: You should know the release if you bought it |
| 19:22.41 | ariel_ | yes I know it C . 2.3.3 |
| 19:23.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:24.55 | *** join/#asterisk yerwish (n=geekangl@86.51.53.125) |
| 19:25.26 | yerwish | Hi everyone.. |
| 19:27.31 | yerwish | I am in a middle of building an embedded Aserisk but really confused about the hardware. Soekris is currently a good option for me (especially the price), but still I am looking for one box that contain an analog card (FXO/FXS). Any suggestoins? |
| 19:30.24 | ariel_ | digium has a box already made with that on it. |
| 19:31.21 | Nugget | http://www.digium.com/en/products/appliance/ |
| 19:31.45 | ariel_ | is looking at a AA50 that he one day will have time to play with.... |
| 19:32.09 | yerwish | I believe that building a box from scratch will be more cost-effective, isn't it? |
| 19:32.10 | ariel_ | now puts it back on the todo shelf. |
| 19:32.26 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:32.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:33.03 | Nugget | yerwish: doubt it. you'll spend a little bit less and then pay forever in headaches and hemorrhoids. |
| 19:33.21 | seanbright | ~sipnat |
| 19:33.21 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:34.05 | ariel_ | hehehehe I like that statement of the hemorrhoids.... it's fitting. |
| 19:34.51 | ariel_ | wonders if the new ones are coming with the asterisk-gui or the AsteriskNow 1.5 gui? |
| 19:35.34 | Docteh | wow those are expensive |
| 19:35.58 | yerwish | Nugget: LoL, however, still the economic downturn is really making it the thing you desrscribed. |
| 19:36.57 | Nugget | well if you truly need to save every last penny, you shouldn't try to get an embedded solution and just build it on some mini-tower desktop. |
| 19:37.19 | Nugget | it's the "embedded" that causes a lot of the hassles that make the digium box the better buy |
| 19:37.52 | ariel_ | I have to say the digium box is nice, just funny the tag they have in the top/front don't cover holes... |
| 19:38.25 | Nugget | but, really, its the wrong approach to look at asterisk as a cost-saving solution. asterisk is a flexible and powerful solution that gives you a lot more control over things than a more traditional pbx. |
| 19:38.43 | Nugget | but cost savings aren't really the main point and you'll be frustrated if you try to make it that way |
| 19:39.02 | Nugget | it's not cheap to do it right and if you do it cheaply you'll be unhappy with the results. |
| 19:39.39 | Nugget | If you're so broke you can't afford phones just get everyone skype and a headset. :) |
| 19:39.56 | ariel_ | we use the appliance for our remote offices, it's easy to deploy, and most of our really good routing is from our corp office anyway.... |
| 19:42.12 | Nugget | I'd love an asterisk appliance if it came with a PRI connection instead of those FXO/FXS ports. I'm chuffed to bits at my TE212s, but the host machines are ghetto clones. |
| 19:42.42 | yerwish | One of the main areas that should be clarified is the processor specs in which you can size the solution appropriately (Desktop/Embedded/whatever!). For instance, 30MHz/Active Call will certainly make the solution expensive. |
| 19:43.16 | Docteh | Nugget: well i have an idea that calls for asterisk + 1 FXO and embedded, networked answering machine |
| 19:43.55 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 19:43.55 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:44.27 | Nugget | When's the last time you saw or called an answering machine, networked or otherwise? :) |
| 19:45.07 | Nugget | the only people I know who still use answering machines are old people and their parents, who aren't exactly going to be enthused about replacing their microcasettes with ethernet. |
| 19:45.57 | Nugget | if anything, most people I know suffer from having too many "answering machines" -- voicemail on their mobile, voicemail at the office, voicemail on their land line (if they even still have a land line) |
| 19:48.15 | yerwish | Nugget: comparing to the current IP-PBX devices (Panasonic/Nortel), they can provide "almost" the same features with a competitive price. What Asterisk can provide/add in this case? |
| 19:48.48 | VoipForces | yerwish: an open platform that you can build on. |
| 19:49.12 | Nugget | exactly. |
| 19:49.31 | Nugget | with asterisk you get a platform you can completely control, code against, integrate, and customize. |
| 19:50.11 | VoipForces | yerwish: think of asterisk as a swissknife, it has build-in features that you can use as is, but you can also use it to build something far bigger |
| 19:50.47 | jaytee | when I want a PBX that can brew a fine cup of coffee or sharpen my #2 pencils I always reach for Asterisk :-) |
| 19:51.18 | ariel_ | brew? Sharpen Pencils. argh |
| 19:52.13 | yerwish | VoipForces: Coudln't agree more. But, still, the core features are provided off-the-shelf which the end-user is usually looking for. PBXs/IP-PBXs are always stable specialy when it compared to desktop-based "software" telephony solutions. |
| 19:52.17 | [TK]D-Fender | yerwish: * makes me coffE, is my jukebox, routes my calls to me only if my hockey team won their last home game, its tuesday night and its raining. |
| 19:52.49 | Nugget | at ud.com we had our coffee maker hooked up to the intranet and you could see how fresh the coffee was and how full the pot was from a web page. |
| 19:52.50 | VoipForces | yerwish, install asterisk with freepbx and you have an off-the-shelf solution. |
| 19:52.54 | Nugget | it was AWESOME |
| 19:53.10 | [TK]D-Fender | yerwish: Yes, and when you pay per VM box like you do with closed platforms, and per extension... and per EVERYTHING, and you have to do everything its way well... you get what you pay for. |
| 19:53.14 | Nugget | you can do all that just monitoring the power draw via an inductive coil |
| 19:53.20 | [TK]D-Fender | yerwish: Proprietary solutions = dead end |
| 19:53.28 | rob0 | http://coffee/ |
| 19:53.37 | *** join/#asterisk czindy (n=czindy@gprs4f7a0458.pool.t-umts.hu) |
| 19:54.23 | czindy | Hello Good Evening, |
| 19:54.29 | VoipForces | yerwish: I have asterisk deployements handing 1000s of calls per day that have been running non stop for years. |
| 19:55.01 | Nugget | the only asterisk instability I've wrestled with has been my fault. |
| 19:55.13 | VoipForces | yerwish: I probably replaced at least 10 Nortel systems with asterisk so far |
| 19:55.34 | Nugget | that said, it's quite easy to build an instable asterisk server. :) |
| 19:55.43 | yerwish | VoipForce: Wow! |
| 19:56.03 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:56.14 | yerwish | This is really informative... |
| 19:56.30 | yerwish | Appreciate your time guys.. |
| 19:56.48 | VoipForces | yerwish: Actualy doing one right now that I need to reproduce the nortel smdr port in asterisk. Off course rs-232 smdr does not exist in asterisk, but after 2 days I'm almost done implementing it in asterisk. |
| 19:58.02 | yerwish | VoipForce: How do you usually size such solution (1000s of calls)? |
| 19:59.24 | VoipForces | yerwish: Bigger is better... for small install I go Dual or Quad core xeon, for big install I use multi-processor system. The cost of High quality servers (like an HP ML350G5) is not that high. |
| 20:00.02 | VoipForces | yerwish: and I usually do like 1Gb of RAM per PRI. I personally tested asterisk 1.2 doing broadcast on 8 PRI. |
| 20:00.19 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 20:00.23 | yerwish | VoipForces: Aha, what about High Avalability (HA) & Virtualization? |
| 20:01.30 | VoipForces | yerwish: No, I do not do virtualization, hardware is cheap. For HA, when the customer requires it I just install 2 servers in a active standby configuration |
| 20:01.53 | VoipForces | yerwish: where the active server copies it's configuration via a private lan link every X minutes. |
| 20:02.07 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 20:02.13 | VoipForces | yerwish: In the case of a failure you just switch the PRI link from 1 server to the other. |
| 20:02.19 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 20:02.30 | yerwish | VoipForces: Manually. |
| 20:02.57 | VoipForces | yerwish: Yes, but there are boxes that do it automatic, just never tried it. |
| 20:03.14 | czindy | I have an asterisk 1.2 server. And there is a problem with the behave of queues. When client is sended to queue, and asterisk send the client to an agent, it is calling a macro. I cannot find how / where queues are connected to that macro. Could you help please? |
| 20:03.23 | VoipForces | yerwish: there is a talk that is going to happen at TAUG regarding using sangoma hardware for that also. |
| 20:03.53 | yerwish | VoipForces: Is it the processor or memory that we should be concerned about during the sizing phase? |
| 20:05.07 | VoipForces | yerwish: depends on the application. If you plan to do a lot of transcoding, then it's cpu intensive. If you plan to do conversation recording then it's RAM and fast HDD. If it's a high queue application, CPU and RAM. |
| 20:05.14 | *** join/#asterisk Weezey (n=ohno@lan3.LO.iasl.com) |
| 20:05.31 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:05.45 | Weezey | I'm having a problem with an AudioCodes MP-118 8port FXO |
| 20:06.35 | Weezey | when the call ends up at voice mail from the PSTN the audiocodes sends a "BYE" to the asterisk box and hangs up 10 seconds into the message |
| 20:06.41 | yerwish | VoipForces: For sure, 64-bit processor architecture is preferred, isn't it? |
| 20:07.20 | VoipForces | yerwish: helps for cpu intensive tasks like transcoding, but voice will not travel faster because you have a faster cpu... |
| 20:10.21 | *** join/#asterisk uluatu (n=uluatu@189.32.44.21) |
| 20:10.27 | luckyaba | Anyone know what the deal with "HD Phones" are? is there some technology built into them that actually makes calls better or is it just a marketing gimmick? |
| 20:10.30 | yerwish | VoipForces: Have you used G.711 codec in Asterisk? |
| 20:11.05 | VoipForces | yerwish: that's about the only one I use expect soetimes g729 or gsm for asterisk-to-asterisk trunks |
| 20:11.06 | ariel_ | yerwish: most setups use g.711 = ulaw |
| 20:11.29 | ruben23 | hi is my g729 codec and g723 codec installed in this format http://pastebin.com/m1ce7f5a2 |
| 20:11.34 | Qwell | VoipForces: well...that isn't entirely true |
| 20:11.42 | Qwell | VoipForces: if you transcode faster, you can send the audio faster |
| 20:11.49 | ariel_ | luckyaba: HD Phones are really nice we have a few of them from Polycom and they do sound better. |
| 20:12.02 | Qwell | ruben23: If you have a Digium transcoder board, you should call them for support on it. |
| 20:12.05 | Qwell | If you don't, please leave. |
| 20:12.14 | VoipForces | Qwell: what I meant is that is that you will not talk faster LOL |
| 20:12.21 | Qwell | VoipForces: it might! :) |
| 20:12.50 | Qwell | ruben23: describe what you're doing |
| 20:12.55 | pfn | whoop, my voicemail patch is in :) |
| 20:12.57 | luckyaba | ariel_, Yeah I have heard them and they do seem to sound better. But I have also seen Monster Cables scandalous ploy to show how their cables are better |
| 20:13.14 | luckyaba | I am trying to find the technical differences for them |
| 20:13.19 | Nugget | Module 'chan_chipmunkvoice.so' could not be found. |
| 20:13.32 | ruben23 | <PROTECTED> |
| 20:13.40 | rob0 | Nugget: just playback() at double speed |
| 20:13.43 | Qwell | luckyaba: The HD phones do sound rather amazing - even in G.711, because of the increased hardware quality. |
| 20:13.44 | Nugget | heh |
| 20:13.58 | Qwell | ruben23: The ones in that list are what are installed/loaded. |
| 20:14.54 | ruben23 | Qwell: codec having with no value are not loaded seeng like speex - - - - |
| 20:15.00 | Qwell | correct |
| 20:15.14 | ruben23 | Qwell:ok thanks |
| 20:16.42 | luckyaba | Qwell, So what I have found is "HD" is really wideband and that only g.722 or AMR-WB are "HD" quality |
| 20:17.00 | luckyaba | so does that mean I need to find a SIP provider that is using that codec to benefit from it? |
| 20:17.01 | Qwell | luckyaba: yes or SIREN |
| 20:17.11 | Qwell | luckyaba: yes and no.. |
| 20:17.39 | Qwell | if you want to use it in wideband mode, yes. however, like I said earlier, even with G.711, it sounds noticeably better than other phones |
| 20:17.53 | Qwell | the audio hardware (speakers/mic) is vastly superior |
| 20:17.55 | luckyaba | but that is simply because of better hardware? |
| 20:17.58 | Qwell | right |
| 20:17.59 | luckyaba | ok |
| 20:18.13 | luckyaba | right on, appreciate the confirmations! |
| 20:20.04 | yerwish | Comparing G.729 to G.711, G.711 is not compressed but free. The former is the opposite (compressed but has to be licensed). |
| 20:21.42 | seanbright | how does the transcoder card work, anyway? |
| 20:21.54 | seanbright | does it come with codec_g729.so? |
| 20:22.01 | Qwell | seanbright: codec_dahdi |
| 20:22.05 | seanbright | ahh. |
| 20:22.09 | Qwell | it provides a translator to g.729 and g.723 |
| 20:22.17 | seanbright | cool |
| 20:22.22 | Qwell | or, rather, a translation path I guess |
| 20:22.24 | seanbright | we ordered one on friday |
| 20:22.27 | Qwell | nifty |
| 20:22.36 | seanbright | i early await its arrival |
| 20:22.40 | seanbright | eagerly, too. |
| 20:27.20 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 20:28.12 | Katty | eppigy: ! |
| 20:28.19 | Katty | eppigy: i just learned you can make a facial mask with an EGG |
| 20:28.38 | [TK]D-Fender | checkout time, later all |
| 20:29.40 | VoipForces | yerwish: you can use g729 in passthrough mode, that is if asterisk does not have to do transcoding because all your endpoints are g729 you can use g729 free |
| 20:29.47 | Weezey | Anyone use an Audiocodes FXO device with asterisk? |
| 20:30.13 | VoipForces | Weezey: I have used the FXS devices, and stay away from those. Use Mediatrix instead. |
| 20:30.21 | ariel_ | Weezey: audiocodec has a great support department |
| 20:30.47 | ariel_ | wow VoipForces I actually would say it the other way around... |
| 20:31.34 | VoipForces | ariel_: You got support from Audiocodes??? How much did you pay yearly for that? |
| 20:31.52 | VoipForces | ariel_: Also Audiocodes have a 1 year warranty, Mediatrix have 3 year. |
| 20:32.24 | ariel_ | did not have too, it comes with there product for setup. Mediatrix is hit or miss and calling them on the west cost in CA sometimes is a mess. |
| 20:32.41 | eppigy | Katty: NICE |
| 20:32.50 | ariel_ | VoipForces: but I have used both and once there configured both I will say they work well. |
| 20:32.56 | Katty | eppigy: do you know how cheap eggs are?! |
| 20:32.58 | VoipForces | ariel_: I'm in montreal so it's easy to call them :-) |
| 20:33.07 | Katty | eppigy: that's like a 10 cent facial mask! |
| 20:33.11 | ariel_ | We are now using Quintum Gateways instead due to there support is 10 times better |
| 20:33.21 | eppigy | CANT BEAT THAT |
| 20:33.25 | Katty | their. |
| 20:33.44 | VoipForces | ariel_: Audiocode sis not offer me any setup support or post install support for that matter. |
| 20:34.13 | Katty | eppigy: yeah, now if i could just find something for the tips of my hair. |
| 20:34.30 | Katty | eppigy: the last time i went to get my hairs did, they razored the ends to give the effect of layers, rather than doing real layers. |
| 20:34.38 | Katty | eppigy: so now i'm all frizzy and ...tangly |
| 20:34.53 | rdeuce | Anyone here from SF area? |
| 20:34.58 | Katty | eppigy: this is a very serious problem. |
| 20:35.09 | Katty | eppigy: frizz is not to be taken lightly. |
| 20:35.28 | Katty | almost as serious as echo problems. |
| 20:35.31 | Katty | unacceptable. |
| 20:36.04 | Katty | we need more females in here :< |
| 20:36.08 | rdeuce | Name the Asterisk problem and I've got it. |
| 20:36.31 | rdeuce | Echo, static, dropped calls... it just keeps going and going... |
| 20:36.45 | Katty | well stop giving it viagra. |
| 20:36.55 | rdeuce | Someone today said "I dialed a number and it took two minutes to connect!" |
| 20:37.19 | Katty | rdeuce: asterisk does not fit every senario. |
| 20:37.31 | rdeuce | It should be able to handle 50 users. |
| 20:37.41 | Katty | rdeuce: it may take you 5 years to figure that out. maybe longer. |
| 20:37.45 | rdeuce | LOL |
| 20:37.46 | Katty | rdeuce: but eventually, you will figure it out. |
| 20:38.01 | eppigy | Katty: lol |
| 20:38.02 | rdeuce | Should have went with Shoretel. |
| 20:38.10 | eppigy | I have a simple yet elegant solution for the hair problem |
| 20:38.19 | Katty | mayo?! |
| 20:38.28 | rdeuce | My integrator is telling me that they want to downgrade to 1.4 from 1.6 |
| 20:38.29 | Katty | don't you dare suggest cutting it all off |
| 20:38.35 | rdeuce | To solve the problem. |
| 20:38.43 | rdeuce | Saying "SIP support may be an issue in 1.6" |
| 20:38.50 | Katty | i have downgraded to 1.4 to fix 'problems' |
| 20:39.01 | alrs | rdeuce: If your integrator is comfortable with 1.4, do it |
| 20:39.02 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 20:39.08 | rdeuce | It's fucked because... yesterday we had 10 calls dropped. |
| 20:39.15 | alrs | I solved a lot of problems at my current job by scrapping 1.6 |
| 20:39.23 | rdeuce | So I put on wireshark and am monitoring the vlan. |
| 20:39.24 | Katty | rdeuce: okay first, you need to just RELAX |
| 20:39.26 | rdeuce | Today, no dropped calls. |
| 20:39.32 | Katty | ReDNeQ: breathe deeply |
| 20:39.44 | Katty | s/ReDNeq/rdeuce/ |
| 20:40.06 | Katty | rdeuce: maybe listen to some soft music. |
| 20:40.28 | Katty | rdeuce: ARE YOU BREATHING?! |
| 20:40.29 | rdeuce | Oh, I am relaxed. Just really irritated. |
| 20:40.44 | Katty | irritated and relaxed are not in the same sentance. |
| 20:40.54 | Katty | relax more. |
| 20:41.39 | Katty | eppigy: so what is your solution for the frizz |
| 20:42.05 | eppigy | shave head |
| 20:42.28 | Katty | eppigy: i hate you, peach fuzz. |
| 20:42.47 | eppigy | :D |
| 20:42.52 | Katty | rdeuce: the earth is nearly 5 billion years old. |
| 20:43.28 | rdeuce | I've heard similar figures. |
| 20:43.34 | Katty | rdeuce: do you know what 100 years of 5 billion years is? |
| 20:44.15 | Katty | rdeuce: you're ENTIRE life span is like one itty bitty little piece of sand, in flordia. |
| 20:44.25 | Katty | probably more like the sahara |
| 20:45.05 | Katty | rdeuce: we're scheduled for another ice age in about 15 thousand years. |
| 20:45.30 | Katty | rdeuce: so... |
| 20:45.33 | Katty | rdeuce: yeah. |
| 20:45.48 | Katty | rdeuce: talk about being irritated... |
| 20:45.59 | Katty | rdeuce: because a server is dropping calls. |
| 20:46.47 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:47.26 | eppigy | DONDE ESTA |
| 20:47.32 | Katty | el gato con queso |
| 20:47.52 | rdeuce | Well, I am not sure if it's dropping the calls or what is dropping the calls. Regardless, I am not worried about the next ice age. What I am worried about is a whole team of pissed off sales people during the last few days of the quarter. |
| 20:48.13 | Katty | and the pissed off sales people will eventually grow old and die |
| 20:48.21 | Katty | and in another 20 years no one will remember or care |
| 20:48.28 | rdeuce | Right, but in the interim, it needs to be fixed. |
| 20:48.28 | Katty | nor will it make any huge difference in the grand scheme of things |
| 20:48.53 | Katty | if it will make you happy to fix it, then fix it |
| 20:48.59 | Katty | otherwise, go do something else. |
| 20:49.03 | Katty | live your life to be happy. |
| 20:49.03 | alrs | rdeuce: Sorry if you've already answered. Is this a trixbox or a Switchvox system? You can probably get around your consultant and contact the vendor directly. |
| 20:49.19 | Katty | things have gotten too complicated for me. |
| 20:49.39 | Katty | i'd rather think about squirrels and how to make my hair defrizz |
| 20:49.43 | rdeuce | alrs: It's a general Asterisk 1.6 system, generic if you will. |
| 20:49.53 | Katty | cause i've only got a few years left. |
| 20:49.58 | alrs | rdeuce: And it's all SIP? |
| 20:50.15 | rdeuce | alrs: We have a variety of terminals -- yes, all SIP. |
| 20:50.22 | rdeuce | alrs: Polycom and Linksys. |
| 20:50.24 | alrs | and your trunk to the PSTN is SIP as well? |
| 20:50.40 | rdeuce | alrs: Yes, the trunk is a private T1 -- all SIP, not PRI. |
| 20:51.36 | rdeuce | alrs: Latency to the upstream gateway is... 7-9ms... |
| 20:51.36 | alrs | rdeuce: If it was me and I had to get something working TONIGHT I'd pull down a reasonably powered desktop machine, go in to the bios and turn off all of the sound/usb/serial/parallel/hyperthreading/irq hogging shit |
| 20:51.43 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 20:51.46 | alrs | download an iso of trixbox/asterisknow/pbxinaflash |
| 20:51.50 | alrs | get the thing installed |
| 20:52.20 | alrs | make sure that some port on your firewall is forwarding to port 22 on the trixbox/switchvox/whatevs |
| 20:52.24 | rdeuce | alrs: I suppose that would be one option. |
| 20:52.36 | seanbright | in a SIP invite, what are the codecs? a=rtpmap? |
| 20:52.38 | alrs | and then call one of the ~100/hr support services like Fonality, pbxtechs, whatever |
| 20:52.39 | *** join/#asterisk voxter (n=voxter@74.221.128.174) |
| 20:53.20 | alrs | as long as they can SSH in to your machine they caan usually get a lot done |
| 20:53.28 | rdeuce | Hmm. |
| 20:54.49 | rdeuce | See, as I said earlier, I have dumpcap going on a laptop connected to a monitor port on my core switch, and it's watching the entire public VLAN... I am just waiting for a call to drop at this point, so I can go and look at the entire trace. |
| 20:55.19 | rdeuce | I should not call it dropped calls, I believe it's actually just that audio goes away on one channel. |
| 20:55.28 | *** join/#asterisk shazaum (n=oidjiowe@unaffiliated/shazaum) |
| 20:55.30 | rdeuce | The call never actually ends, just one party can't hear the other. |
| 20:55.39 | alrs | rdeuce: That sounds like NAT trouble |
| 20:55.47 | rdeuce | We don't have a NAT. |
| 20:55.56 | rdeuce | The pbx has a public interface. |
| 20:56.06 | rdeuce | Well, a public vlan interface. |
| 20:56.11 | rdeuce | With a routable IP. |
| 20:56.24 | alrs | rdeuce: then it sounds like a router that has some lame options like "SIP Fixup" that screws things up |
| 20:56.33 | rdeuce | Hmm. |
| 20:56.38 | alrs | or a crap T1 |
| 20:56.49 | rdeuce | Cisco switches wouldn't have anything like that, would they? |
| 20:57.51 | alrs | rdeuce: worth looking in to |
| 20:57.57 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 20:58.10 | rdeuce | I know the routers have fixup stuffs. |
| 20:58.56 | alrs | http://help.fonality.com/index.php?title=Troubleshooting/One-way_Audio&highlight=fixup |
| 20:59.56 | rdeuce | Hmm. |
| 21:00.05 | rdeuce | Would those ports need to be allowed inbound as well? |
| 21:00.42 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:6420:b769:966:bb78) |
| 21:00.55 | alrs | yes |
| 21:01.05 | rdeuce | Hmm. |
| 21:01.26 | rdeuce | Looking at the iptables setup they built for us doesn't show any rules set to allow their gateway to access our pbx. |
| 21:02.22 | alrs | rdeuce: you can verify that they are using ports 10000-20000 for media by looking in /etc/asterisk/rtp.conf |
| 21:02.31 | rdeuce | Assuming that to be true, how would that even word period? |
| 21:02.32 | rdeuce | work |
| 21:03.03 | alrs | rdeuce: Dunno. Definately turn off anything that says "SIP Fixup" |
| 21:03.23 | alrs | s/definately/definitely |
| 21:09.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:10.38 | rdeuce | Okay. |
| 21:10.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:10.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 21:11.06 | *** join/#asterisk intralanman (n=lanman@va-67-76-163-226.sta.embarqhsd.net) |
| 21:11.15 | rdeuce | I chatted with the integrator, and they said that they've never really worried about that... but it does seem logical. |
| 21:12.06 | rdeuce | Now, do I agree with them? It does make sense that the sip/rtp ports be open between the pbx and the gateway, bidirectional. |
| 21:16.36 | *** join/#asterisk czindy (n=czindy@gprs5e1b993b.pool.t-umts.hu) |
| 21:16.52 | czindy | Hello again... |
| 21:17.26 | czindy | Could you explain me please why it is happening: |
| 21:17.29 | czindy | [Jun 23 23:08:38] -- Executing Queue("Zap/1-1", "ENT|tT|||1800") in new stack |
| 21:17.30 | czindy | [Jun 23 23:08:38] -- Started music on hold, class 'default', on channel 'Zap/1-1' |
| 21:17.30 | czindy | ----------------- |
| 21:17.30 | czindy | [Jun 23 23:08:38] -- outgoing agentcall, to agent '325', on 'Local/325@default-b442,1' |
| 21:17.30 | czindy | [Jun 23 23:08:38] -- Called Agent/325 |
| 21:18.08 | czindy | what is it mean outgoing agent call to agent 325, and why exit from queue? |
| 21:18.20 | Qwell | Who said it exited the queue? |
| 21:19.07 | czindy | because it is jumping to the extension which handles _325 |
| 21:19.21 | Qwell | and? |
| 21:19.42 | czindy | and if the 32 agent is busy it is hanging up the line |
| 21:20.18 | czindy | ok my question is... what is it mean outgoing agentcall? |
| 21:20.38 | Qwell | Did you use a GUI to configure this? |
| 21:21.06 | czindy | no... everyting are in the conf files |
| 21:23.05 | [TK]D-Fender | czindy: You logged your agent is and it uses a local channel to reach them |
| 21:23.22 | [TK]D-Fender | czindy: How is it you have no clue how the login system works, yet have DONE it already? |
| 21:24.38 | bionoid | <PROTECTED> |
| 21:24.39 | [TK]D-Fender | czindy: And its an outgoing agent call, because thats what Queues does. It CALLS AGENTS. |
| 21:24.50 | eppigy | oh boy |
| 21:25.10 | [TK]D-Fender | bionoid: He didn't just know it, he also pre-BUILT it apparently |
| 21:25.31 | bionoid | It's the new thing, Precognfiguration |
| 21:26.31 | czindy | Ok thank you for the answer, but why queue going out to an extension in the extension.conf? |
| 21:26.31 | *** join/#asterisk scalex000 (n=chatzill@218puntacana02.codetel.net.do) |
| 21:26.46 | Docteh | ~callback |
| 21:26.54 | scalex000 | hello |
| 21:27.27 | czindy | Docteh the callback means thet the queue will jump out to an extension? |
| 21:27.44 | Docteh | well i just remembered i need to look into callbacks |
| 21:27.55 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:28.33 | scalex000 | I need help to buy a good switch for voip with PoE |
| 21:28.52 | [TK]D-Fender | czindy: Agents are contacted through the dialplan, thats what your login dos. It says what to dial to reach them |
| 21:29.55 | czindy | So because of AgentCallbackLogin the agents are called throught the dialplan. Am I right? |
| 21:30.02 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 21:33.15 | Docteh | how do call files work? I'd like to call my handset, and if I pick up within ten seconds it'll dial out, and maybe send me an email if i ignore it |
| 21:34.28 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:34.34 | [TK]D-Fender | czindy: Yes. |
| 21:34.49 | [TK]D-Fender | czindy: that is the ONLY way "Agents" are called. |
| 21:35.07 | [TK]D-Fender | czindy: Agent = chan_agent = Dialplan via chan_local |
| 21:36.01 | [TK]D-Fender | Docteh: there are docs in your taball for this |
| 21:36.16 | Docteh | k |
| 21:37.07 | czindy | [TK]D-Fender: And if I don't want to agents called via extension, I have to logging them in with AgentLogin function? |
| 21:37.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:38.08 | rdeuce | [TK]D-Fender: Earlier we were talking about firewall rules perhaps being the cause of the strange one-way audio issue we're having... Do you know if inbound 5060 and 10000:20000 is truly necessary between Asterisk and our upstream? |
| 21:38.47 | rdeuce | We previously had no inbound SIP/RTP open. |
| 21:39.38 | rdeuce | I'm reading information that says these ports are needed -- but even with them closed, our calls did seem to work with the odd and sudden loss of one channel of audio. |
| 21:39.44 | [TK]D-Fender | rdeuce: firewall / reinvites = trouble |
| 21:39.59 | rdeuce | Our vendor checked the reinvites and claims they are disabled. |
| 21:41.15 | *** join/#asterisk DarkRift (n=dark@65.92.165.166) |
| 21:42.52 | bionoid | rdeuce: Can you put a sniffer (Wireshark or such) in front of your router? - ie verify that the packets are actually received? |
| 21:43.01 | rdeuce | Yes. |
| 21:43.21 | rdeuce | I have dumpcap running on a monitoring station right now. Once we experience another bad call, I can go back and analyze. |
| 21:43.39 | rdeuce | Nearly a gigabyte of captured traffic so far... :) |
| 21:44.50 | rdeuce | The problem is that today everything is working. |
| 21:45.01 | rdeuce | Yesterday we had 10 reports of audio issues. |
| 21:45.03 | rdeuce | Today, zero. :O |
| 21:45.12 | rdeuce | Maybe it fears Wireshark? I dunno. |
| 21:45.23 | bionoid | Sounds like you're up for quite a few hours of fun ;) |
| 21:45.40 | czindy | [TK]D-Fender: so as I read, agents are always dialed via dialplan, and if I don't want to use extension, I have to simly add a queue member? Am I right? |
| 21:45.45 | rdeuce | I just need one call to look at... Once I get a bad call, that will be it. |
| 21:45.58 | rdeuce | It certainly does seem like a reinvite issue. |
| 21:48.21 | rdeuce | Because that's exactly what happens... calls could be going on just fine when suddenly audio gets lots. |
| 21:48.22 | rdeuce | lost. |
| 21:48.56 | suma | got a problem with asterisk manager originate, can someone please help ? |
| 21:49.06 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 21:49.13 | suma | when i make a call through originate, asterisk immediately sends a cancel |
| 21:49.51 | suma | when i make a call through originate, it sends an invite, immediately sends a cancel |
| 21:52.05 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 21:52.34 | [TK]D-Fender | czindy: All queues have member,s its a question of what TYPE |
| 21:53.15 | [TK]D-Fender | czindy: A fixed device will ring a fixed device. Agent will be based on where they have logged in to and use Local. Local... is local.. DUH. SIP will just right that SIP device |
| 21:53.40 | *** join/#asterisk [jmc] (n=John@93-45-199-191.ip103.fastwebnet.it) |
| 21:55.17 | [jmc] | hi guys |
| 21:55.36 | suma | jmc, yes |
| 21:55.46 | czindy | [TK]D-Fender: ok thank you for the help. :) I think I have to test it. |
| 21:56.08 | [jmc] | can someone shortly explain to me what is the "peer" type of user in sip.conf? |
| 21:56.25 | suma | peer means it is for outgoing calls |
| 21:56.32 | [jmc] | ok, "user" can only receive calls, "friend" can receive and make calls |
| 21:56.40 | [jmc] | only outgoing ones? |
| 21:56.43 | suma | yes |
| 21:56.47 | [jmc] | oh, ok |
| 21:56.54 | [jmc] | I wasn't sure of that |
| 21:57.07 | suma | user is only for incoming |
| 21:57.13 | suma | friend can do both |
| 21:57.46 | [jmc] | ok :) |
| 21:57.52 | [jmc] | thanks suma |
| 21:57.56 | suma | np |
| 21:59.44 | [TK]D-Fender | peer can do both |
| 22:00.10 | [TK]D-Fender | Off to martial arts, BBL |
| 22:01.29 | [jmc] | both? |
| 22:01.31 | [jmc] | hmm ok |
| 22:01.38 | [jmc] | so I don't really get the difference |
| 22:09.40 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:16.18 | *** join/#asterisk xpot-mobile (n=james@hsc-4-173.hotspot.utah.edu) |
| 22:21.36 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:22.20 | e0n` | does the CDR log instantly |
| 22:22.23 | e0n` | or is there a pause? |
| 22:23.11 | e0n` | so if I generate a call file in the outgoing spool, would it show up instantly? Or not till the call completes |
| 22:24.31 | ehsjoar | e0n: The CDR contains data that describes the call. You know like billable time and so on. You don't have the information until the call completes. On a real switch you can get partials though (for very long calls) |
| 22:24.49 | e0n` | hmm |
| 22:25.01 | e0n` | Damn, that really kills my idea i had lol |
| 22:25.09 | *** join/#asterisk bbryant1 (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 22:25.34 | e0n` | ehsjoar: So if a disconnect was there, (busy signal and so on) would it log the instant that is hit? |
| 22:25.51 | *** join/#asterisk awkfu (n=awkfu@66.162.90.57) |
| 22:26.13 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 22:27.42 | ehsjoar | e0n: That is my understanding yes. From what I can see in the log the CDR gets logged immediately once the call has ended. Are you using MySQL for the CDRs? |
| 22:28.16 | e0n` | ehsjoar: PostgreSQL CDR |
| 22:29.21 | ehsjoar | e0n: I have never done that. There is a NoCDR command in the dial plan that one could use if you don't want a CDR |
| 22:29.47 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:30.06 | e0n` | hmm, I was depending on the CDR to post immediately |
| 22:30.55 | ehsjoar | e0n: I logged on quite recently. Did you mention what you are trying to accomplish? |
| 22:31.42 | e0n` | I have a script that throws calls in the * outgoing spool. |
| 22:31.56 | e0n` | I wanted to see, from sql if the message was disconnected or not after Nth seconds |
| 22:35.20 | ehsjoar | e0n: You can definitely find the duration of the call by looking at the CDR. There are also ways to log data into a table through sql (ODBC_SQL). If you just want to see in realtime when the call gets disconnected there is the manager interface where you can subscribe to events and stuff. I guess I am still not 100% sure of what you are trying to do here |
| 22:37.52 | e0n` | ehsjoar: It's ok, thanks for the suggestions though |
| 22:38.14 | ehsjoar | e0n: No problems |
| 22:42.10 | *** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com) |
| 22:43.03 | citywok | I want to use asterisk to make a phone call to a handful of people, until any one of them hits 1 (acknowledge phone call) during the phone call. anybody have any suggestions on where to start? |
| 22:43.13 | *** join/#asterisk BadHAL (n=nn@70.99.106.38) |
| 22:44.03 | ariel_ | citywok: there are a few dialer applications for asterisk you need to google them |
| 22:44.05 | citywok | I know how to originate the phone call using the API, but i don't know how to find out if the person hit 1 during that call, or what happened. does anybody have any suggestions on how to get that result back in to a perl script when using the AMI? |
| 22:44.09 | ariel_ | but vicidial is one |
| 22:45.45 | citywok | oh, that's for call center application, i just want to have my system monitor call the entire support team, until somebody hits 1 (yes, i'll take care of it) |
| 22:46.12 | citywok | dial all 5 cell phones round robin until 1 of the IT people responds by saying i can take care of it |
| 22:46.29 | ariel_ | that is just a ring group and can be done with a macro |
| 22:48.14 | ariel_ | It's time to go home, g'nite folks |
| 22:48.35 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 22:50.45 | *** join/#asterisk khronos (i=tom@sip02.kineticvoip.com) |
| 22:56.51 | tompaw | Hi. |
| 22:57.36 | tompaw | Is it - in your opinion - possible to detect this moment: http://pastebin.com/d736aebca with just adjusting AMD's parameters or does it require changing its source code? |
| 23:07.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:23.48 | tompaw | I enabled console => debug in logger.conf, but I still cannot see the messages from ast_debug(...) function |
| 23:23.54 | tompaw | Did I miss anything? |
| 23:25.22 | seanbright | core set debug 100 |
| 23:26.14 | tompaw | thanks |
| 23:33.21 | *** join/#asterisk mazpe (n=mazpe@c-76-108-3-76.hsd1.fl.comcast.net) |
| 23:34.01 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 23:35.03 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 23:45.53 | citywok | what is the common text to speech method? |
| 23:48.35 | Docteh | I've been looking at cepstral |
| 23:48.36 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 23:49.04 | e0n` | citywok: there is always carnival |
| 23:49.26 | citywok | i dont care how bad it sounds, just needs to read out letters |
| 23:49.42 | citywok | read out a server name to me over the phone C S G R E D E X 2 |
| 23:53.20 | Docteh | oh so you want to just use the sayletters() or w/e |
| 23:55.29 | Docteh | SayAlpha or SayPhonetic try both? |
| 23:55.40 | citywok | SayALpha, hmmmm ok i'll look at it |
| 23:56.10 | citywok | that will probably work perfectly, awesome |
| 23:56.58 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |