| 00:00.56 | Nivex | *drumroll* |
| 00:01.39 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 00:02.06 | MRH2 | is current svn on version 1.2 compiling for everyone? |
| 00:02.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:08.32 | blitzrage | huh? |
| 00:08.42 | blitzrage | oh -- I see what you mean, I read it wrong |
| 00:20.57 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177817337.dsl.bell.ca) |
| 00:22.01 | MRH2 | itsok i am getting the same error at make codec_zap.o with a historical version as well - so something else I guess |
| 00:22.11 | fab5freddy | I am running Asterisk on a VMWare guest, and my want to connect to it with my host system via a soft phone software called Ekiga, how do I determine the IP address to use for the Registrar? |
| 00:22.17 | *** join/#asterisk _deg_ (n=deg@189.11.53.118) |
| 00:22.50 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 00:23.03 | drako | fab5freddy, ekiga sucks, period. |
| 00:23.26 | drako | fab5freddy, the ip address is always on ifconfig. |
| 00:23.57 | fab5freddy | i use the inet addr: correct? |
| 00:24.22 | drako | guess so |
| 00:24.29 | _deg_ | Anybody got the new Asterisk SPE? |
| 00:25.04 | _deg_ | With the new T39 fax support!!!! |
| 00:25.24 | jql | I'm holding out for t40! |
| 00:25.29 | _deg_ | <PROTECTED> |
| 00:25.33 | drako | jql, what? |
| 00:25.41 | _deg_ | jql, Yeah me too. |
| 00:25.45 | fab5freddy | drako: i am using that and it doesn't work, keeps saying Registration Failed |
| 00:26.10 | JT | T.38 maybe? |
| 00:26.36 | _deg_ | I heard that T40 will print color faxes on BW printers |
| 00:27.24 | BBHoss_Work | cool |
| 00:27.27 | JT | T.50 is photo quality |
| 00:27.30 | JT | 300dpi |
| 00:27.45 | BBHoss_Work | you mean 1200dpi |
| 00:27.54 | robin_sz | mmm tasty |
| 00:28.12 | JT | eh, peeople rarely print at 1200dpi in reality |
| 00:28.16 | robin_sz | ? |
| 00:28.48 | Nivex | hmm... can't get chan_cellphone to see this motorola v365 |
| 00:29.04 | JT | robin_sz: ? |
| 00:29.23 | Qwell | Nivex: in pairing mode? |
| 00:29.27 | robin_sz | JT, well, why would I buy a colour 1200dpi printer and then use anything less than 1200DPI? |
| 00:29.31 | _deg_ | JT, I usualy make some big porttraits with the successed fax receiving with Asterisk... |
| 00:29.43 | _deg_ | JT, So, I will use 1200dpi... |
| 00:29.51 | robin_sz | anyway, faxes suck |
| 00:29.57 | Nivex | Qwell: no, I'm just doing a cell search and a cell show devices |
| 00:30.01 | robin_sz | bigtime |
| 00:30.08 | jql | naw, T.50 transmits in PNG, w/alpha channel |
| 00:30.12 | Qwell | put it in pairing mode and do the search again |
| 00:30.16 | JT | robin_sz: because most professional printers don't even print that high |
| 00:30.22 | JT | human eye can hardly tell a difference |
| 00:30.40 | JT | consumer printer manufacturers just put crazy specs on their gear to try and bamboozle consumers |
| 00:30.45 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:30.57 | robin_sz | JT, yeah, well, whatever, it claims to do 1200dpi ... there no advantage in using less is there? |
| 00:31.21 | Nivex | Qwell: ok will do... as soon as I read up on how to do it :) |
| 00:31.29 | JT | you are limited by your source material, robin_sz |
| 00:31.34 | drako | lol |
| 00:31.57 | JT | the advantage maybe that you can actually print it at the size you want, if you use a lower res |
| 00:32.13 | robin_sz | JT, true, true ... unlike inkjets thoug, that claim sillyt resolutions and then the paper just soaks it all up to make a mess, this one does at least look sharp |
| 00:32.52 | JT | laser? |
| 00:33.04 | robin_sz | pahser hot wax |
| 00:33.09 | robin_sz | phaser |
| 00:33.21 | JT | oh, those |
| 00:33.30 | robin_sz | well, i like em |
| 00:34.37 | robin_sz | so ... |
| 00:35.16 | robin_sz | what alternative drivers can i use to chan_mISDN? the echo is baaaaad |
| 00:36.08 | JT | bristuff |
| 00:36.26 | robin_sz | yeah ... maybe try that again |
| 00:37.08 | robin_sz | that or an eicon Diva card |
| 00:38.54 | JT | yeah great idea, using a card that almost no-one uses ;) |
| 00:39.04 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 00:41.29 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 00:41.39 | Nivex | ahh nuts... hcitool cc seems to connect but doesn't do anything. Trying to connect the handsfree from the phone keeps saying "Service not supported" |
| 00:45.24 | hillct | isn't there a way to specify registration-specific timeouts like register => user:passwors@provider.com;timeout=120 |
| 00:45.32 | hillct | or something like that? |
| 00:46.13 | hillct | I have one provider that wants a timeot of under 1 minute and one that freaks out if I have a timeout of less than 5 minutes |
| 00:49.35 | blitzrage | You've got bankers! |
| 00:49.49 | Qwell | bankers? |
| 00:51.44 | robin_sz | oh well, chan_miSDN 0.3.1-rc29 here I come ... |
| 00:53.36 | JT | always suffering with misdn :P |
| 00:53.46 | robin_sz | JT, more a qustion of using a card I can afford really |
| 00:54.09 | JT | umm bristuff supports at least what misdn supports |
| 00:54.44 | robin_sz | yeah, but my first attempts with bri didnt work at all ... |
| 00:55.05 | robin_sz | so I tried chan_miSDN and it worked right out of the box |
| 00:55.17 | robin_sz | but .. then weve noticed echo is a problem |
| 00:55.39 | Nivex | Qwell: progress! I can get it to show up on a "cell search" but it doesn't stay in "cell show devices" |
| 00:56.19 | robin_sz | the eicon divas come up on Ebay from time to time at sensible money |
| 00:57.16 | joemon | bankers are rats some times |
| 00:58.08 | Nivex | ahh nuts "Usable: no" |
| 00:58.10 | robin_sz | not seen a digium isdn card I can afford yet :( |
| 00:58.17 | Nivex | *grumble* |
| 00:58.53 | joemon | btw off topic somone is used some kind of blue tooth scanner to listen to mcdonals atm transactions and made a whole back of duplicate atm cards with that info and made off with thousands in cash. It was in the local paper today. |
| 00:59.50 | Nivex | it thinks my cell phone is a headset :( |
| 00:59.52 | JT | robin_sz: and the dtmf issues? also no-one uses eicon cards, so the support would probably be even wrose |
| 01:00.16 | *** part/#asterisk hillct (n=H@cpe-024-211-129-051.nc.res.rr.com) |
| 01:00.28 | JT | joemon: dude, you should read your setences before sending them ;) |
| 01:00.40 | robin_sz | JT, the eicon cards I have deployed at $otherOffice work fine |
| 01:00.54 | JT | running what software? |
| 01:01.02 | robin_sz | dtmf on chan-mISDN is indeed flakey though |
| 01:01.06 | robin_sz | sw, for the eicon? |
| 01:01.19 | robin_sz | their software and chan_capi |
| 01:01.42 | JT | oh proprietary drivers |
| 01:01.49 | robin_sz | working driveers |
| 01:02.51 | robin_sz | right bedtime ... |
| 01:03.16 | JT | i'll steer clear of proprietary stuff where i can |
| 01:03.30 | JT | still, hardly anyone uses eicon cards with asterisk |
| 01:09.08 | Nivex | [Apr 1 21:08:37] DEBUG[27787]: chan_cellphone.c:952 sdp_search: sdp_connect() failed on device 00:19:2C:BB:39:A3. |
| 01:10.01 | joemon | nope |
| 01:21.07 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 01:27.33 | *** join/#asterisk MrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 01:28.02 | mroli | can asterisk 1.4 receive fax natively, or does it need third-party software ? |
| 01:28.45 | jql | it's not included |
| 01:29.30 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 01:35.08 | mroli | what's the best way that you know of, to handle incoming faxes ? |
| 01:36.17 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:51.19 | *** join/#asterisk marc7 (n=marc@S0106000f66461bdb.gv.shawcable.net) |
| 01:53.48 | [TK]D-Fender | mroli, a completely seperate analog line that does not come within 10' of * |
| 01:54.23 | marc7 | from what I understand, priority jumping (eg, when a System() call fails, it jumps to n+101) has been deprecated in Asterisk... and where setting "priorityjumping=yes" in 1.2 would still re-enable it... it's been completely turfed from the 1.4 codebase, correct? |
| 01:54.32 | Nivex | "cell scan" finds an unusable headset on port 0 |
| 01:54.56 | Nivex | when I manually iterate through the ports using rfcomm connect I find handsfree on port 7 |
| 01:55.17 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 01:56.30 | [TK]D-Fender | marc7, safe bet |
| 01:58.42 | kiwoneka | good evening |
| 01:59.14 | kiwoneka | i am hoping i can get instruction on how to make my polycoms hang up imediately after a call |
| 01:59.22 | kiwoneka | kinda like a cell phone does |
| 01:59.29 | JT | what do you mean? |
| 01:59.30 | jql | mine have always done that |
| 01:59.50 | jql | you have to change the default to make them play the off-hook tone |
| 01:59.51 | kiwoneka | well, lets i am talking with you, and you hang up |
| 01:59.58 | kiwoneka | i am using the speaker phone |
| 02:00.03 | jql | yeah |
| 02:00.05 | kiwoneka | i kust get a busy signal |
| 02:00.18 | JT | that's a dialplan issue |
| 02:00.18 | kiwoneka | i want the phone to hang up |
| 02:00.48 | kiwoneka | i know |
| 02:00.56 | kiwoneka | i just dont know what part |
| 02:01.11 | kiwoneka | very very green at this |
| 02:01.17 | JT | kiwoneka: actually, what technology is the other end of the call using? |
| 02:01.33 | kiwoneka | does it matter? |
| 02:01.38 | kiwoneka | well |
| 02:01.43 | JT | of course |
| 02:01.49 | JT | how are you getting your calls? |
| 02:02.02 | kiwoneka | its all voip here |
| 02:02.09 | JT | itsp? |
| 02:02.11 | Nivex | is there a flag I can pass in the config file to force a connection attempt even if a scan doesn't see it? |
| 02:02.24 | kiwoneka | unlimitel |
| 02:02.57 | JT | kiwoneka: IF they correctly signal hangup through SIP or IAX2, you can make your phone hung up immediately |
| 02:03.04 | JT | if they don't, you need to change provider |
| 02:03.13 | kiwoneka | they do |
| 02:03.20 | JT | ok |
| 02:03.31 | JT | so it's just something in your dialplan then |
| 02:03.32 | kiwoneka | it used to work when i was using trix box |
| 02:03.51 | kiwoneka | but i had to abandon trixbox |
| 02:03.58 | JT | :) |
| 02:04.27 | kiwoneka | so i had to start from scratch |
| 02:04.37 | kiwoneka | learning |
| 02:04.47 | kiwoneka | this place has helped alot |
| 02:04.50 | marc7 | Fender et al: okay, so I'm trying to use a System() command to check to see if a file (a .ulaw recording) exists on the asterisk server. Currently, if the script finds the file, it returns 0... it it doesn't, it returns n+101. Since that functionality is being deprecated, is there any way of assigning a variable based on what the System() value executes? Is it as simple as nesting System within a Set command? -- eg: Set(Recording=System(/ |
| 02:05.15 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:07.25 | Juggie | marc7, STAT() |
| 02:07.47 | Juggie | func stat. |
| 02:07.58 | marc7 | hah, awesome. exactly what I was looking for. |
| 02:08.18 | [TK]D-Fender | kiwoneka, paste the dialstring you are using to use unlimitel |
| 02:08.19 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-179-77.ny325.east.verizon.net) |
| 02:08.23 | marc7 | thanks Juggie |
| 02:08.41 | kiwoneka | sip.conf? |
| 02:08.45 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-179-77.ny325.east.verizon.net) |
| 02:08.55 | JT | extensions.conf |
| 02:08.56 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 02:09.08 | JT | just the relevant Dial line only |
| 02:09.12 | JT | otherwise pastebin |
| 02:09.13 | JT | ~pb |
| 02:09.15 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:12.14 | kiwoneka | i hate to sound really dumb, but wha is it ami looking for |
| 02:26.02 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:26.27 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:30.53 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 02:32.10 | kiwoneka | here it is http://pastebin.ca/420370 |
| 02:34.17 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-179-77.ny325.east.verizon.net) |
| 02:34.52 | Nivex | ah ha! progress! Apparently I had to completely unload the module |
| 02:35.05 | Nivex | now I'm getting a bunch of errors on the rfcomm line |
| 02:37.36 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-188-8.nycmny.east.verizon.net) |
| 02:38.08 | Nivex | ok, it just doesn't handle inadvertently sending it a null phone number |
| 02:38.10 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 02:38.38 | k-man | compleetly off topic, but does anyone know of some sort of billing system that can do recurring invoices, for renting things out |
| 02:38.42 | [TK]D-Fender | kiwoneka, You are getting caught when you call "Congestion". |
| 02:39.03 | kiwoneka | please help |
| 02:39.09 | kiwoneka | what do you mean |
| 02:39.12 | [TK]D-Fender | kiwoneka, Just get rid of that |
| 02:39.24 | kiwoneka | that is it |
| 02:39.31 | [TK]D-Fender | kiwoneka, Congestion will keep your phone off hook with an annoying tone indefinately. |
| 02:39.32 | kiwoneka | then it will just hang up |
| 02:39.49 | [TK]D-Fender | kiwoneka, HANGUP will tend to make phones hang up immediately ;) |
| 02:39.52 | Nivex | aaaand no audio |
| 02:39.56 | kiwoneka | it achieved its intended purpose |
| 02:39.58 | BosHaus | I've got my SPA3102 registering with the asterisk server, but it still doesn't pick up calls coming in from the outside. know what woudl cause that? |
| 02:40.05 | Nivex | but valuable progress has been made :) |
| 02:40.21 | [TK]D-Fender | BosHaus, Because your dialplan on the SPA doesn't tell it to pass the call to * |
| 02:40.34 | [TK]D-Fender | BosHaus, Go read the guides on www.voxilla.com |
| 02:41.03 | JunK-Y | [TK]D-Fender: on the polycom, do ya know a way to avoid the chord, but only on dnd? |
| 02:41.20 | BosHaus | ok, thanks [TK]D-Fender |
| 02:41.26 | [TK]D-Fender | JunK-Y, When do you gt the chord that you want to avoid? |
| 02:41.51 | JunK-Y | <PROTECTED> |
| 02:42.13 | [TK]D-Fender | JunK-Y, Oh, you mean you want to silence MWI while on DND? |
| 02:42.23 | JunK-Y | i can turn the chord off, on normal mode, but id like to turn it off, only on dnd. |
| 02:42.28 | JunK-Y | yep. |
| 02:42.49 | [TK]D-Fender | JunK-Y, Sorry, they don't do compound state-checking for things like that.... time for a feature request ;) |
| 02:43.30 | [TK]D-Fender | JunK-Y, Oh, working with my Aastra 57i CT at the office I have come to a rather solid conclusion that Polycom's call-handling capabilities DESTROY evrything else out there... hands down |
| 02:43.33 | JunK-Y | feel free to post that feature request :) |
| 02:43.53 | [TK]D-Fender | JunK-Y, And the 5i series switching to rubbery buttons = mistake. |
| 02:43.57 | JunK-Y | so flexible huh? |
| 02:44.06 | [TK]D-Fender | JunK-Y, I would... if I cared about it personally ;) |
| 02:44.31 | [TK]D-Fender | JunK-Y, Yeah, Polycom's attended vs blind transfer, JOIN***, etc > ALL |
| 02:44.32 | JT | [TK]D-Fender: the aastras are still usable, though? |
| 02:44.41 | [TK]D-Fender | JT, Yeah, largely. |
| 02:44.58 | [TK]D-Fender | JT : I really do like the backlight, and wish they made better use of the LCD. |
| 02:45.08 | MACscr | can i limit the number of RTP ports open in my firewall and also sync that number with rtp.conf? If so, how many ports do i need open for each simultaneus call? |
| 02:45.15 | kiwoneka | thanks that did it |
| 02:45.17 | [TK]D-Fender | JT : But real-life call handling is not so great. |
| 02:45.21 | kiwoneka | works perfectly |
| 02:45.28 | JT | [TK]D-Fender: in what way? |
| 02:45.36 | [TK]D-Fender | JT : I calso got it for the wireless handset which I can't seem to make completely seperate fromt he base... |
| 02:45.52 | kiwoneka | now |
| 02:45.54 | BosHaus | I really wish voicepulse connect had callerid name lookup :\ |
| 02:45.58 | kiwoneka | the next task |
| 02:46.02 | kiwoneka | hints |
| 02:46.21 | [TK]D-Fender | JT : Haven't figured how to do attended transfer, it adds annoying "#"'s into speed-dial tones, etc |
| 02:46.24 | JT | MACscr: yes it needs to sync, 1 port each way for each reinvite of each call |
| 02:46.28 | kiwoneka | i have 4 extensions using polycom ip601 |
| 02:46.44 | JT | hmm |
| 02:46.52 | kiwoneka | i would like tho have all the extensions are of each other when in use |
| 02:46.57 | [TK]D-Fender | JT : No JOIN feature like Polycom, and so on. |
| 02:47.06 | JT | i see |
| 02:47.22 | MACscr | JT: thanks, i limited the ports to 19978 - 20000 and things appear to be working fine, just wanted to make sure there wasnt some other issues i didnt know about |
| 02:47.44 | JT | MACscr: that's a pretty limited port range |
| 02:48.05 | [TK]D-Fender | JT : Also each line represents a call wheras Polycom's keys can support multiple simultaneous. |
| 02:48.05 | JT | MACscr: generally there is no advantage to limiting the ports unless you need them for something else |
| 02:48.52 | JT | i see |
| 02:49.21 | JT | this true of the 480i also? |
| 02:50.17 | [TK]D-Fender | JT : Seems identical |
| 02:50.33 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:50.48 | JT | [TK]D-Fender: so if an aastra has say 2 line keys, it can only handle 2 calls? |
| 02:51.08 | [TK]D-Fender | JT : Aastra's presence support is much better and reflects ringing as well though, and the way the soft-keys are state-based jsut rocks... |
| 02:51.43 | JT | damn, all these little differences really need to be committed to a document like a wiki entry,... |
| 02:51.52 | [TK]D-Fender | JT : almost. the 57i supports 9 lines (calls), yet only hs 4 keys... you can sort scroll through the lines to pick an outgoing past your 4 base # of line-keys. |
| 02:52.05 | [TK]D-Fender | JT : Is hould right a full review of them all. |
| 02:52.09 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 02:52.12 | JT | hmm |
| 02:58.00 | *** join/#asterisk f1r3w0rm (n=warirc@adsl-070-145-060-050.sip.bix.bellsouth.net) |
| 02:59.46 | kiwoneka | any ideas on setting up hinting |
| 02:59.53 | *** part/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
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| 03:14.25 | kiwoneka | do i have to configure my phones ip601 to work with hints |
| 03:17.12 | [TK]D-Fender | kiwoneka, Yes, you have to enable presence support in your provisioning files. |
| 03:17.36 | kiwoneka | how |
| 03:19.19 | kiwoneka | in sip.cfg |
| 03:19.33 | [TK]D-Fender | kiwoneka, Go download the admin guide from polycom's site, and check out the WIKI guide on provisiniong them. |
| 03:20.07 | kiwoneka | ok |
| 03:20.18 | kiwoneka | i have them |
| 03:34.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:34.54 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
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| 03:42.48 | ber_ | offtopic, does anyone know of free web software which allows users to manage their own lists of RSS feeds |
| 03:42.59 | ber_ | preferably in php or something easy to run on unix systems |
| 03:43.06 | ber_ | bloglines.com does this, but I woudl like to host it myself if possible |
| 03:45.44 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 03:50.02 | bkruse_home | the voip-info.org april fools jokes page was a total bust, no one posted, it sucked. |
| 03:50.45 | JunK-Y | bkruse_home: the only think i can say is that ive voicechanger for 1.4! |
| 03:51.59 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 03:51.59 | *** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
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| 03:53.37 | Ambrose | Anyone able to help me out? I'm getting "Got SIP response 400 "Bad Request" back from 192.168.1.70" when I try to call ext 100 from ext 101, which are two cisco 7960 phones setup next to each other, it goes right to voicemail. It works when calling in/out though... ? (thx) |
| 03:54.32 | Ambrose | The phones can call the iaxy phone too |
| 03:54.44 | Ambrose | but the iaxy phone can't call 100/101 |
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| 04:00.10 | voltagex | can asterisk run an arbitary command that isn't an AGI? |
| 04:00.18 | voltagex | like an exec() or system() call? |
| 04:00.28 | [TK]D-Fender | voltagex, You mean like SYSTEM. |
| 04:00.34 | voltagex | oh |
| 04:00.37 | [TK]D-Fender | voltagex, "show application system" |
| 04:00.49 | [TK]D-Fender | voltagex, Sometimes you just need to read the BIG PRINT |
| 04:00.56 | voltagex | SYSTEM('python status-test.py') |
| 04:01.05 | [TK]D-Fender | voltagex, no quotes |
| 04:01.10 | voltagex | oh |
| 04:01.11 | voltagex | ok |
| 04:02.23 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:02.29 | [TK]D-Fender | voltagex, for all of these questions the quic answer is in "show applications" |
| 04:02.50 | voltagex | ah |
| 04:02.53 | voltagex | sorry |
| 04:02.57 | voltagex | I should RTFM |
| 04:03.03 | JT | [TK]D-Fender: do you know how long the line key labels on the aastra consoles can be? |
| 04:03.30 | [TK]D-Fender | Ummm.. no :) |
| 04:03.39 | [TK]D-Fender | voltagex, indeed you should :) |
| 04:04.09 | [TK]D-Fender | JT : I think they were a fixed # of chars, not pixel based... |
| 04:04.15 | [TK]D-Fender | JT : I'll lte you know tomorrow |
| 04:05.46 | JT | yeah i was after a number in chars not pixels :) |
| 04:05.57 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:10.34 | [TK]D-Fender | JT : Well Polycom works on a pixel basis as the font is not monospaced. |
| 04:10.44 | [TK]D-Fender | JT : Aasta's IS monospaced |
| 04:10.48 | JT | heh |
| 04:10.54 | [TK]D-Fender | Another downside. |
| 04:12.25 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 04:24.15 | *** part/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
| 04:30.04 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 04:30.59 | mroli | bye all |
| 04:31.18 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@adsl-75-31-197-229.dsl.pltn13.sbcglobal.net) |
| 04:31.59 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:34.30 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 04:34.50 | *** join/#asterisk TEOX (n=TEX@203-184-35-128.jetbuster.co.nz) |
| 04:34.59 | TEOX | i can't get to my voicemails |
| 04:35.04 | TEOX | asterisknow |
| 04:35.07 | TEOX | dial 850 |
| 04:35.16 | TEOX | and it just keeps loopin the menu |
| 04:39.17 | tzafrir_laptop | TEOX, what do you see on the CLI? |
| 04:39.21 | tzafrir_laptop | asterisk -rvvv |
| 04:39.40 | tzafrir_laptop | probably needs to be run by root |
| 04:40.39 | *** join/#asterisk BOLIVIAN (n=ter@201.222.65.113) |
| 04:40.58 | BOLIVIAN | which one is better or faster? x windows, gnome or kde? |
| 04:43.05 | TEOX | humm |
| 04:43.10 | TEOX | I just rebooting it |
| 04:43.19 | *** join/#asterisk diclophis-work (n=jbardin@comet.cisdata.net) |
| 04:43.19 | TEOX | and then have another look |
| 04:43.48 | TEOX | cos there is a few funny things going on |
| 04:44.11 | TEOX | like when i connect to pstn through my clone card I hear myself very loud |
| 04:44.23 | TEOX | which is odd. But thought it might be my phone at fault |
| 04:45.46 | TEOX | Ok. Had a look at the cli |
| 04:45.48 | *** join/#asterisk mihinomenest (i=oLQR@cerebus.clandestineresearch.com) |
| 04:45.57 | TEOX | it tell me it is playing the audio files |
| 04:46.00 | TEOX | which it does |
| 04:46.08 | TEOX | but when i press 1 of inbox |
| 04:46.14 | TEOX | it doesn't do anything |
| 04:49.46 | TEOX | any ideas? |
| 04:55.09 | flenders | guys, quick questoin about CDR/billing... I have a bunch of analogue lines here, on 2 TDM400Ps, whenever I dial a number, through my Zap channels, I get a "ANSWERED" on Master.csv, even though the call wasn't answered. if I dial the same number through an ITSP (sip), I get the expected "NO ANSWER" if the call is not answered. |
| 04:55.11 | flenders | any ideas? |
| 04:57.54 | JT | analogue :( |
| 04:58.58 | carrar | yeah, don't trust your CDR's |
| 04:59.00 | ber_ | flenders, what is the map in extensions.conf which handles your analog lines |
| 04:59.11 | ber_ | my cdr has been pretty reliably |
| 04:59.13 | ber_ | reliable |
| 04:59.20 | flenders | ber_: map? |
| 04:59.40 | ber_ | map is wrong word, what is teh extension plan which handles your analog lines |
| 04:59.44 | carrar | what about billsec? |
| 04:59.54 | carrar | is it > 0 ? |
| 05:00.02 | flenders | carrar: yeah |
| 05:00.14 | carrar | so the call was completed? |
| 05:00.35 | flenders | no, it wasn't answered |
| 05:00.40 | flenders | that's what the problem is |
| 05:00.57 | flenders | ber_: you mean the Dial command? |
| 05:01.30 | carrar | billsec's not duration right? |
| 05:01.50 | flenders | carrar: let me double check |
| 05:01.52 | ber_ | whatever context you have assigned to the incoming analog lines, paste the behavior in your extensions.conf |
| 05:03.25 | flenders | carrar: billsec was 16 seconds and duration was 20 |
| 05:04.12 | flenders | ber_: sorry mate, what behaviour? can you rephrase that? |
| 05:04.54 | flenders | carrar: and the call was not anserwer |
| 05:04.57 | flenders | answered |
| 05:05.07 | carrar | Asterisk 1.4.1? |
| 05:05.12 | flenders | 1.4.0 |
| 05:05.28 | ber_ | flenders |
| 05:05.35 | ber_ | inside of extensions.conf |
| 05:05.44 | ber_ | you have a 'context' which you have chosen for your analog lines |
| 05:05.55 | ber_ | i think you choose that context in zapata.conf or something |
| 05:06.10 | flenders | ber_: yeah, it's called 'incoming' |
| 05:06.21 | ber_ | ok, what is the contents of that incoming context |
| 05:06.30 | ber_ | and is 'incoming' the same context you use for your sip phones |
| 05:06.50 | flenders | ber_: not sure why it matters, I'm talking about outgoing calls |
| 05:07.03 | ber_ | oh |
| 05:07.14 | JT | i think ber_ is totally on the wrong track |
| 05:07.16 | ber_ | DAR |
| 05:07.22 | ber_ | brain no workie |
| 05:07.42 | flenders | ber_: I have a bunch of contexts... incoming calls go to incoming contexts, all my sip phones are on a context called 'intern', and the outgoing dialplan is on a context called 'outgoing' |
| 05:08.18 | flenders | outgoing context is included in intern, so people can dial out |
| 05:08.36 | ber_ | for someone connected via analog line what context are they in |
| 05:08.59 | flenders | and my dial line is as simple as "exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN},60,rtT)" |
| 05:09.13 | flenders | ber_: I don't have FXS channels |
| 05:09.21 | flenders | ber_: all my handsets are SIP |
| 05:09.47 | ber_ | ah still on the wrong track ok im getting closer :) |
| 05:10.31 | ber_ | is there debugging you can turn on your zap channels to show it receiving an answer back from the PTT |
| 05:10.45 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-58-152.socal.res.rr.com) |
| 05:11.08 | flenders | dunno... |
| 05:11.42 | ber_ | the transition from seizure to answer is pretty basic telephony so I doubt the problem is with the PTT |
| 05:12.09 | flenders | what's PTT? |
| 05:12.29 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:12.36 | ber_ | phone company |
| 05:13.02 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 05:13.27 | ber_ | have you tried turning your debugging up to max through the CLI? |
| 05:13.46 | ber_ | i would try that and then making some test calls thru the Zap and SIP providers |
| 05:14.07 | JT | i have never ever heard of a phone company as the PTT |
| 05:14.12 | JT | PTT is Push To Talk :) |
| 05:14.24 | ber_ | ptt is pretty much what they call all the european telephone comapnies |
| 05:14.40 | JT | would you mind expanding upon it? |
| 05:14.41 | ber_ | deutsche telekom, spanish one, france telecom, danish one |
| 05:14.51 | ber_ | it is public telephone something |
| 05:15.01 | ber_ | they are basically all state owned monopolies |
| 05:15.02 | JT | "telco" or "carrier" is much more universal |
| 05:15.15 | ber_ | carrier isnt wireline |
| 05:15.24 | JT | yes it can be |
| 05:15.29 | ber_ | it can but it doesnt need to be |
| 05:15.37 | flenders | JT: seen this? http://www.google.com/tisp/ |
| 05:15.39 | ber_ | i thoguht he was in euro thats why i said PTT |
| 05:15.43 | JT | yep :) |
| 05:16.55 | cr4z3d | haha wow |
| 05:16.59 | JT | so telco or carrier are much better terms |
| 05:17.06 | JT | or telecommunications provider |
| 05:17.11 | JT | or something |
| 05:17.40 | ber_ | yup |
| 05:19.12 | Qwell | Are "HD cages" inside a PC pretty generic? |
| 05:19.14 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 05:19.30 | ber_ | agreed |
| 05:19.46 | JT | Qwell: generic, as in..? |
| 05:19.49 | [TK]D-Fender | qwell : only in as much as they typically take 3.5" drives. as for how they moun to the case, no, entirely proprietary |
| 05:20.03 | Qwell | well, I need to get another one |
| 05:20.17 | JT | yeah they're usually non standard |
| 05:20.22 | ber_ | order from your manufacturer |
| 05:20.30 | ber_ | or might find one for your model on ebay |
| 05:20.33 | JT | SCA has the most chance of being standard, but then that's just the connector :) |
| 05:24.07 | *** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at) |
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| 05:29.47 | flenders | JT: what was your hosting again? |
| 05:29.59 | flenders | the company I mean |
| 05:31.55 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 05:34.09 | flenders | thanks |
| 05:35.49 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 05:39.29 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:41.34 | TEOX | work out why the stupid voicemail prompts didn't work |
| 05:41.40 | TEOX | dtmf tones.. |
| 05:41.47 | TEOX | had to change setup in phone |
| 05:42.38 | TEOX | now have to work out why i getting a local echo |
| 05:43.34 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 05:43.40 | zeeesh | hi |
| 05:43.43 | *** join/#asterisk p4r14h (n=j0sh@69.92.145.178) |
| 05:48.38 | [TK]D-Fender | TEOX, What phones are you using? |
| 05:51.23 | TEOX | grandstream |
| 05:51.30 | TEOX | budgetone 100 |
| 05:51.57 | TEOX | when i connect to a pstn it i get a local echo of my own voice |
| 05:52.04 | [TK]D-Fender | .... yup.. they do that on severl firmware revisions... |
| 05:52.05 | [TK]D-Fender | ~gs |
| 05:52.07 | jbot | it has been said that gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 05:52.07 | TEOX | like it's amped 10x |
| 05:52.18 | TEOX | Yeah i gopt the leastest firmware flashed up |
| 05:52.23 | TEOX | lastest |
| 05:52.36 | [TK]D-Fender | No saying even that will help you.... so |
| 05:52.37 | TEOX | LOL |
| 05:52.39 | [TK]D-Fender | ~wglwat |
| 05:52.40 | jbot | wglwat is, like, well, good luck with all that |
| 05:52.46 | [TK]D-Fender | :| |
| 05:52.49 | TEOX | LOL |
| 05:52.59 | TEOX | yeah so wat do you recommend??? |
| 05:53.09 | TEOX | cos i got given this so can't complain |
| 05:53.14 | [TK]D-Fender | ~phones |
| 05:53.15 | jbot | it has been said that phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 05:53.24 | TEOX | wasn't sure if it was the phone or the clone card |
| 05:53.39 | TEOX | coll cheers |
| 05:53.43 | TEOX | cool cheers |
| 05:53.45 | [TK]D-Fender | TEOX, If you get it on internal SIP calls between phones / * then its the phones. |
| 05:54.28 | [TK]D-Fender | TEOX, Kind of the "undeniable evidence" side of things |
| 05:54.42 | TEOX | nah i tested that and between that phone and xlite then it sounds fine. |
| 05:54.52 | TEOX | but then i thought it might have been xlite |
| 05:54.54 | [TK]D-Fender | TEOX, Though any hardware card w/o hardware EC can be hit/miss |
| 05:54.57 | TEOX | will do a biut more testings |
| 05:55.13 | TEOX | http://forums.digium.com/viewtopic.php?t=13465&highlight=voice+menu |
| 05:55.22 | TEOX | that post had the same issues |
| 05:55.39 | TEOX | same sort of phone too |
| 05:56.09 | TEOX | yeah only to pstn |
| 05:56.15 | TEOX | sip to sip internally is fine |
| 05:57.37 | flenders | TEOX: what card is it? |
| 05:57.53 | [TK]D-Fender | TEOX, Ok, well tuning to reduce echo can work in many cases... |
| 06:01.07 | zeeesh | is it possible to receive a copy of ur voicemail at ur any web based e-mail address |
| 06:01.38 | [TK]D-Fender | zeeesh, Yes |
| 06:02.09 | zeeesh | i made some peers at my server .. |
| 06:03.10 | zeeesh | i have been configured voicemail .. any trying to receive a copy of voicemail at any web address but could not done . |
| 06:04.19 | zeeesh | could not understand how to use this command " mailcmd=/usr/sbin/sendmail -v -t -f asterisk-pbx@yourdomain.com "???? |
| 06:04.39 | [TK]D-Fender | what is "any web address" and have you checked your message queue in sendmail or even proved that its independantly functional? |
| 06:05.09 | *** join/#asterisk kiko69 (n=keith@adsl-75-16-91-106.dsl.irvnca.sbcglobal.net) |
| 06:07.08 | zeeesh | ok let me check .. again .. |
| 06:08.36 | *** part/#asterisk diclophis-work (n=jbardin@comet.cisdata.net) |
| 06:11.12 | [TK]D-Fender | ok, WAY late.. I'm off, back later |
| 06:11.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:12.00 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
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| 06:14.00 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 06:14.21 | *** join/#asterisk IPconfig (n=dd@mail.trueafrican.com) |
| 06:16.19 | *** join/#asterisk Caplain (n=matt@adsl-75-45-249-77.dsl.sfldmi.sbcglobal.net) |
| 06:16.21 | Caplain | hey |
| 06:16.33 | Caplain | i cant seem to get incomming calls or making outgoing calls |
| 06:18.17 | IPconfig | ? |
| 06:18.17 | IPconfig | checked your line ? |
| 06:21.06 | Caplain | how do i do that? |
| 06:21.13 | Caplain | i registered with asterlink |
| 06:21.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:22.07 | *** join/#asterisk TEOX (n=TEX@203-184-35-128.jetbuster.co.nz) |
| 06:22.17 | TEOX | ok changed to anothe fxo |
| 06:22.20 | TEOX | and it is better |
| 06:22.34 | TEOX | but now i get cracking when on pstn |
| 06:22.40 | TEOX | fricken cheap cards for ya! |
| 06:22.49 | flenders | TEOX: what card is it? |
| 06:22.51 | IPconfig | what are you using an analog line or digital line .. , |
| 06:22.51 | IPconfig | and plus does that line have a dial tone e.t.c. can it call out. |
| 06:23.12 | Caplain | i have an analog line but i dont know how to set that up |
| 06:23.17 | Caplain | can i use a regular modem? |
| 06:23.29 | Caplain | i got a winmodem and an isa modem |
| 06:24.05 | TEOX | Flenders: it is just coms cheapy ass thing ambinet chipset wildcard x101p |
| 06:24.28 | TEOX | this one works better than the other card but crackly |
| 06:24.36 | TEOX | but alteast the local echo not so bad |
| 06:24.44 | TEOX | but really crackly |
| 06:24.57 | TEOX | Still odd that it gets a local echo |
| 06:25.02 | TEOX | like hearing yourself |
| 06:26.10 | TEOX | like the mic too sensitive when through to pstn |
| 06:26.19 | IPconfig | ummh IIRC one of the warnings was to always stay away from winmodems |
| 06:26.20 | TEOX | have not playied with teh tx and rxgains |
| 06:26.32 | TEOX | yeah i know |
| 06:26.33 | IPconfig | if you have read the asterisk manual .. |
| 06:26.40 | IPconfig | think i read it some where . |
| 06:27.06 | TEOX | Just wanted to see if they are really shit as most ppl say they are :) |
| 06:27.48 | TEOX | so these "winmodems" are they just like normal modems which ppl have hacked teh firmware for? |
| 06:28.20 | TEOX | in otherword why coudl you not just use a decent modem with dsp onboard etc, |
| 06:30.17 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:32.05 | fbffff | are you talking about the x100p's? |
| 06:32.53 | canapa | can someone help me with this warning please, i get it like every second, which is a little annoying |
| 06:32.59 | canapa | <PROTECTED> |
| 06:33.09 | TEOX | yeah xp100ps |
| 06:33.10 | fbffff | what warning? |
| 06:33.22 | fbffff | humm |
| 06:34.11 | fbffff | canapa, one second i need to glance at the source really fast |
| 06:34.18 | zeeesh | i want to send a copy of voicemail at my any web e-mail address .. do i need to install "Exim" or asterisk can send a copy of voicemail????? |
| 06:35.55 | fbffff | canapa, do you have a sip dump of the registration you can paste? |
| 06:36.58 | canapa | ? |
| 06:38.33 | canapa | fbffff: a sip dump? |
| 06:38.42 | TEOX | so xp100 have a Z-Complex?? problem??? |
| 06:41.13 | canapa | fbffff: i dont even know which registration this should be about.. |
| 06:42.23 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:51.24 | *** join/#asterisk shay|work (i=shay@unaffiliated/shay) |
| 06:51.28 | shay|work | hello |
| 06:52.22 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 06:58.24 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 06:58.36 | zdrulio | hello :) |
| 07:00.36 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
| 07:03.08 | shay|work | zdrulio: hi |
| 07:03.27 | zdrulio | i have question :) i wish to simulate calls (20-30) to check how work my weak PC. do you know any appz or script ? |
| 07:05.55 | jql | like sipp? |
| 07:16.43 | zdrulio | what ? |
| 07:16.45 | Caplain | is it possible to use a modem with the pbx? |
| 07:17.42 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 07:24.11 | IPconfig | Caplain what are you trying to achieve ? |
| 07:24.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
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| 07:48.23 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 07:53.50 | zdrulio | i have question :) i wish to simulate calls (20-30) to check how work my weak PC. do you know any appz or script ? |
| 07:54.57 | flenders | can I generate calls on the manager api using 2 extensions? |
| 08:00.11 | *** join/#asterisk Dovid (n=Dovid@85.159.160.199) |
| 08:00.11 | Dovid | can anyone tell me what this means ? |
| 08:00.11 | Dovid | SIP/2.0 603 Declined (no dialog) |
| 08:03.58 | *** join/#asterisk Strom_M (n=strom@65.124.49.251) |
| 08:05.02 | Dovid | Strom_C: have u ever seen this ? |
| 08:05.03 | Dovid | SIP/2.0 603 Declined (no dialog) |
| 08:05.17 | Dovid | been happening with one of our providers ever since we upgraded to 1.2.17 |
| 08:05.21 | JT | zdrulio: there is an "application" called sipp |
| 08:05.29 | Strom_M | nope |
| 08:06.02 | Dovid | JT: isnt sipp the open source app that is used to create SIP test calls |
| 08:06.52 | JT | yes. |
| 08:07.00 | Dovid | what is a 603 error ? |
| 08:07.07 | zdrulio | Jt thx |
| 08:07.29 | JT | Dovid: i've seen that question 3 times in the last 6mins |
| 08:07.36 | JT | if i knew the answer, i'd say so. |
| 08:07.50 | JT | sounds like you have a missing SIP signalling dialogue though :) |
| 08:07.56 | Dovid | thx |
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| 08:20.48 | zdrulio | Jt |
| 08:22.37 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 08:24.32 | *** join/#asterisk saftsack (n=oliver@p54a7e5c9.dip.t-dialin.net) |
| 08:24.37 | saftsack | hi |
| 08:24.49 | flenders | JT: still around? |
| 08:25.02 | Dovid | when was SIP 2.0 released? |
| 08:25.05 | saftsack | is * on a compact flash disc a good thing? |
| 08:25.13 | Dovid | flenders: what is your qustion ? |
| 08:25.25 | Dovid | saftsack: why not? |
| 08:25.28 | *** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at) |
| 08:25.36 | flenders | Dovid: not asterisk related |
| 08:25.49 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:26.01 | saftsack | Dovid, i was just asking, because i had problems with my openwrt system. |
| 08:26.09 | flenders | I was bidding on a bunch of cisco phones on ebay, and ended up winning more than one |
| 08:26.25 | flenders | as JT thought it would happen |
| 08:26.34 | *** join/#asterisk joobie (i=joobie@c220-239-107-179.eburwd3.vic.optusnet.com.au) |
| 08:27.05 | joobie | Hey Guys.. I want to use Asterisk at a commerical level - is there a branch that is good for this sort of use? I've only really looked into the asteriskathome branch, but it doesnt sound suitable:P |
| 08:28.20 | flenders | joobie: a branch? |
| 08:28.24 | Dovid | saftsack: never used it but i cant understand why. if u had issues with open WRT then i would be cautios. email the users list and see what others have to say |
| 08:28.42 | flenders | jobbie: asterisk@home was based on asterisk, as trixbox and freepbx are |
| 08:28.54 | Dovid | joobie: I prefer good old vanila asterisk - where u write the configs urself |
| 08:28.54 | joobie | ahh |
| 08:28.58 | flenders | jobbie: I think you'd be better off installing asterisk from source |
| 08:28.58 | Dovid | www.asterisk.org |
| 08:29.03 | saftsack | Dovid, i think that my problems had the cause of high cpu usage |
| 08:29.18 | joobie | so asterisk @ home is basically asterisk but with preconfigs? |
| 08:29.28 | Dovid | saftsack: idk, see what others say |
| 08:29.32 | flenders | joobie: it was a linux with asterisk pre-installed |
| 08:29.34 | saftsack | ok |
| 08:29.37 | Dovid | joobie: and lots of messy and uneeded code |
| 08:29.48 | joobie | ahh |
| 08:29.59 | flenders | joobie: you'd be better off installing it yourself |
| 08:30.01 | Dovid | it's best to write it urself if u can |
| 08:30.14 | joobie | when you say write itmyself, this is just standard config files? |
| 08:30.18 | joobie | like no coding involved? |
| 08:30.19 | flenders | joobie: you can do a lot more if you install it from scratch |
| 08:31.02 | flenders | no coding to get a PBX up and running |
| 08:31.15 | flenders | just messing around with conf files |
| 08:33.21 | joobie | ahh |
| 08:33.24 | joobie | thanks flenders |
| 08:33.35 | joobie | hey btw, is there a good resource for hardware? |
| 08:33.48 | joobie | i have 2 PST lines coming into the building and 7 users in the office |
| 08:33.53 | Dovid | joobie: to purchase ? |
| 08:34.01 | Dovid | www.telephonydepot.com |
| 08:34.05 | Dovid | www.voipsupply.com |
| 08:34.05 | joobie | i want to give them each a phone hooked up to the asterisk box, and then handle the PSTN line coming in |
| 08:34.12 | joobie | what's some good hardware though |
| 08:34.18 | joobie | like "known to work well" in asterix? |
| 08:34.21 | joobie | -x+sk |
| 08:34.31 | joobie | is there like some trustworthy hardawre that is known to work well? |
| 08:35.54 | flenders | ~phones |
| 08:36.04 | jbot | [phones] http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 08:36.26 | Dovid | joobie: u can use soft phones like x-lite |
| 08:36.34 | Dovid | i like polycoms and snom's |
| 08:36.40 | joobie | softphones as in the software ones on the pc? |
| 08:36.47 | Dovid | joobie: yes |
| 08:36.59 | flenders | I use linksys spa-921s and they work pretty well for what they cost |
| 08:37.30 | joobie | i know the linksys brand, but only for computers |
| 08:37.40 | flenders | and for the phone lines, I guess you're best option is a TDM400P with 2 fxo modules |
| 08:37.43 | joobie | do you think the linksys spa would be better than the Polycom? |
| 08:37.45 | Dovid | flenders: i believe they are a replacement for the 841's which were real good |
| 08:37.58 | flenders | Dovid: yeah... I have an 841 too |
| 08:38.13 | Dovid | polycom is the king in moy book but costs more. also i had lots of issues with them with NAT |
| 08:38.39 | joobie | ahh |
| 08:38.43 | flenders | I just wish polycoms were a bit cheaper |
| 08:38.45 | joobie | issues with the polycom with nat? |
| 08:39.14 | flenders | joobie: to tell you the truth, you'll find NAT issues with pretty much any SIP phone |
| 08:39.17 | flenders | :D |
| 08:39.46 | Dovid | joobie: if u r using phones locally u shouldnt have NAT issues |
| 08:39.51 | flenders | joobie: linksys are pretty down here |
| 08:39.52 | joobie | yea |
| 08:39.57 | joobie | it's local, but just interesting about the NAT |
| 08:40.03 | joobie | to see what the problems could potentially be |
| 08:40.09 | joobie | how is the performance too? |
| 08:40.14 | flenders | joobie: you in sydney? |
| 08:40.18 | joobie | like compared to standard PABX, is it noticable? |
| 08:40.22 | joobie | flenders, melb.. you? |
| 08:40.27 | flenders | joobie: syd |
| 08:40.44 | joobie | ahh cool |
| 08:41.01 | joobie | well my problem is my client has a liking for PABX systems (commander) |
| 08:41.10 | joobie | im trying to convince them to go VOIP, but not sure myself if it's as good |
| 08:41.12 | flenders | joobie: for us here, it is only noticeable, because we're not using hardware echo cancellers, so it was a pain in the ass to get rid of echo |
| 08:41.15 | joobie | what are your thoughts? |
| 08:41.22 | joobie | ahh |
| 08:41.38 | Dovid | joobie: if it's internal on its own network u should have no isses what so ever. |
| 08:41.47 | Dovid | although they do tend to come up |
| 08:41.55 | joobie | what are teh common issues? |
| 08:42.04 | flenders | joobie: I have a separate network for the phone here, and I highly recommend that, so it works as well as our old NEC phone system |
| 08:42.14 | joobie | i mean if i suggest VOIP and the client goes for it, it needs to be as robust as a PABX..... the cost is not a huge issue |
| 08:42.24 | flenders | joobie: hangon |
| 08:42.31 | flenders | you want a full voip solution? |
| 08:42.42 | joobie | yea |
| 08:42.45 | flenders | they must have a VERY reliable internet connection too |
| 08:42.59 | joobie | why though? I was thinking they would use PSTN into the office |
| 08:43.01 | joobie | then in the LAN use voip |
| 08:43.08 | flenders | ahh, ok |
| 08:43.08 | joobie | so they still route out over PSTN |
| 08:43.22 | flenders | I thought you wanted voip to outside as well |
| 08:43.42 | joobie | i'm confident in bringing a stable / high performance LAN.. when we start talking internet links, it's a bit iffy |
| 08:43.46 | joobie | naa |
| 08:44.06 | flenders | inside the network, performance is great |
| 08:44.13 | joobie | just like to run numous lines into the office (PSTN) and route out over that.. and use voip within the office.. I Don't trust the internet uplink to route over the net as voip |
| 08:44.19 | flenders | the phones don't use that much bandwidth anyway |
| 08:44.25 | joobie | ahh |
| 08:44.40 | joobie | flenders, what do you recommend in terms of hardware |
| 08:44.44 | joobie | like say object was no cost |
| 08:44.48 | joobie | and we wanted to support say.. 30 users |
| 08:44.56 | flenders | you want to save money? |
| 08:45.47 | joobie | well .. it would be good, but just not at a tradeoff for performance |
| 08:45.47 | joobie | like if these guys go with VOIP, it would be in the understanding that the system will function as well as a standard PABX |
| 08:45.47 | joobie | that is my concern in mind... |
| 08:45.52 | joobie | so basically numerous PSTN lines coming into the asterisk box, then VOIP for the LAN office.. |
| 08:46.03 | flenders | it all depends on how much money they want to spend |
| 08:46.18 | flenders | I would definetely get a card with echo cancel |
| 08:46.31 | flenders | and that costs a bit more money |
| 08:46.46 | flenders | how many lines? |
| 08:46.48 | joobie | how does it do the echo cancel? |
| 08:47.01 | joobie | i dont know exactly.. i have to send a proposal.. but say 4 lines max? |
| 08:47.18 | flenders | it does it on the card, hardware, not on software. |
| 08:47.22 | joobie | i marketed the product as cheaper than a standard PABX solution to them, and as reliable. |
| 08:47.26 | joobie | ahhh |
| 08:47.46 | flenders | joobie: so you told them before you had any idea how it worked? |
| 08:47.47 | flenders | :D |
| 08:48.11 | flenders | what's the reason for the upgrade? |
| 08:48.23 | flenders | just to stop paying commander maintenance fees? |
| 08:48.32 | joobie | yea |
| 08:48.45 | joobie | i mean, the thing is if it's a PABX solution they go with, i won't do it for them because I can't |
| 08:48.52 | joobie | voip i can learn.. so i made some presumtions:P |
| 08:49.06 | joobie | well they haven't bought a commander yet, they don't have a PABX |
| 08:49.16 | flenders | do they have a commander system installed at the moment? |
| 08:49.21 | flenders | ahh ok |
| 08:49.24 | joobie | so rather than buy that commander and lose flexibility., i suggested VOIP.. so we can do stuff like voicemail and things wtihout extra costs |
| 08:49.36 | joobie | how muhc is a commander system btw on avergae? |
| 08:49.39 | joobie | say a small 30 user one..? |
| 08:49.46 | joobie | i estimated 10k+ to the client? |
| 08:49.55 | flenders | I have a quote for a commander system here, and I know they are pretty pricey |
| 08:50.21 | joobie | how many users on the commander system you have? |
| 08:50.23 | flenders | let me check |
| 08:50.25 | joobie | yea i heard they were.. |
| 08:50.31 | joobie | just guessed 10k+ |
| 08:50.44 | flenders | it's about 5K for 6 users |
| 08:51.10 | joobie | ahh |
| 08:52.28 | joobie | can you make any suggestions for a complete solutoin flenders? |
| 08:52.50 | joobie | i might just invest in it and give it a whirl, either way if the client takes it or not.. |
| 08:52.51 | flenders | if you buy a dell server for 600/700 bucks, plus UPS, plus all handsets, plus phone cards, I would say you would spend about 3K for 6 handsets |
| 08:53.15 | flenders | I'm trying to find the quote on my e-mail |
| 08:53.44 | joobie | ahh cool |
| 08:53.47 | joobie | thanks flenders |
| 08:54.00 | joobie | 3K aint bad at all |
| 08:54.57 | flenders | joobie: $4,987.40 |
| 08:55.03 | flenders | 6 handsets |
| 08:55.05 | joobie | ahh |
| 08:55.10 | flenders | and installation is not included |
| 08:55.12 | joobie | the linksys ones? |
| 08:55.19 | flenders | no, commander |
| 08:55.27 | joobie | ahh |
| 08:55.40 | joobie | is it limited to 6 users tho? how well does it scale out? |
| 08:55.41 | flenders | the 3K I said earlier was with linksys SPA922s, with PoE |
| 08:55.45 | joobie | ahhh |
| 08:55.47 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 08:55.53 | flenders | joobie: no, you can have up to 32, I think |
| 08:56.14 | flenders | with asterisk you can have as many as you want |
| 08:56.14 | joobie | flenders, is there any IP in that quote? do you mind forwarding it to me for my refernece? |
| 08:56.28 | flenders | joobie: sorry mate, I can't |
| 08:56.30 | joobie | yea.. these guys are limited in their capacity at the moment though |
| 08:56.37 | joobie | who did you get it through flenders? |
| 08:56.40 | joobie | maybe ill give htem a buzz |
| 08:56.51 | flenders | joobie: commander |
| 08:56.55 | joobie | ahh k |
| 08:56.55 | joobie | ta |
| 08:56.59 | flenders | 132777 |
| 08:57.05 | joobie | yea ill give them a ring |
| 08:57.22 | joobie | okok.. down to the dirty stuff |
| 08:57.24 | flenders | I can't send you cause it was a quote to a friend of mine, and we decided to go with asterisk |
| 08:57.32 | flenders | so he only sent it to me as a reference |
| 08:57.40 | joobie | what specific hardware would you recommend flenders.. like the dell server, handsets, phone cards,etc? |
| 08:57.45 | joobie | yea it's cool man |
| 08:57.49 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:57.55 | joobie | ill give them a buzz and find out.. I wasn't sure commander did the quotes direct |
| 08:58.06 | flenders | how many users? |
| 08:58.18 | joobie | they have 7 now.. but to scale out to say 30 ? |
| 08:58.26 | flenders | ok. |
| 08:58.26 | joobie | so buy for 7 users now |
| 08:58.31 | joobie | but capability to scale to 30 |
| 08:58.33 | flenders | how many lines? |
| 08:58.38 | joobie | say 4 incoming? |
| 08:58.49 | joobie | i don't know, taking a stab in the dark .. but most likley around that figure |
| 08:59.10 | joobie | so still want incoming and outoging to be over swtandard PSTN.. just to use the voip around the LAN |
| 08:59.25 | joobie | and if possible, support stuff like "on hold music".. "voicemail" |
| 08:59.34 | joobie | hunt groups.. those sort of features |
| 08:59.35 | flenders | joobie: can't they get ISDN? |
| 09:00.03 | joobie | possibly.. but isn't it expensive in comparison to numerous PSTN's? |
| 09:00.25 | Ahrimanes | depends on the definition of "numerous" |
| 09:00.36 | joobie | hey i have to jet |
| 09:00.39 | joobie | got a dinner to go to |
| 09:00.42 | flenders | if you get ISDN2s (BRI), they cost the same as 2 POTS lines |
| 09:00.42 | joobie | ill be on later tonight |
| 09:00.48 | flenders | around 30 for line rental |
| 09:00.57 | joobie | flenders, have to run man |
| 09:01.04 | joobie | thanks for the help, ill jump on tonight |
| 09:01.07 | flenders | I think an ISDN2 is 70 for line rental (2 lines) |
| 09:01.08 | joobie | speka to you then if you're on |
| 09:01.11 | joobie | got a dinner to go to now |
| 09:01.12 | joobie | sorry |
| 09:01.19 | joobie | ahh k |
| 09:01.20 | joobie | not bad |
| 09:01.22 | flenders | joobie: come back tomorrow |
| 09:01.31 | joobie | ok |
| 09:01.33 | joobie | speka to you then |
| 09:01.45 | joobie | thanks again |
| 09:01.56 | flenders | no worries |
| 09:02.03 | joobie | later |
| 09:05.17 | *** join/#asterisk TEOX (n=TEX@203-184-35-128.jetbuster.co.nz) |
| 09:06.07 | TEOX | can you measure/change the zcomplex??? |
| 09:06.17 | TEOX | cos i can't seem to find much info about ir |
| 09:07.49 | Dovid | zcomplex = ? |
| 09:12.58 | *** join/#asterisk kremiq (n=kremo@213.215.118.28) |
| 09:15.27 | TEOX | a setting to do with the impedance of the pstn line or something |
| 09:15.49 | TEOX | x100 cards have to havs specific settings for zcomplex else you get noise |
| 09:15.53 | TEOX | something likethat |
| 09:21.20 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-205417.home.otenet.gr) |
| 09:23.53 | zeeesh | is "sendmail" option is a part of asterisk or linux ???? |
| 09:29.21 | JT | zcomplex == complex impedence |
| 09:36.24 | TEOX | yeah so is it a hardware "setting" or a software one |
| 09:36.46 | JT | what makes you sure there is such a setting |
| 09:37.38 | TEOX | I didn't say there was a setting |
| 09:38.00 | TEOX | I got told that some cards can be changed |
| 09:38.10 | TEOX | for various places where they are used |
| 09:38.25 | TEOX | so that they have the same impedance as the country they are used in |
| 09:38.26 | JT | hrm there might be some localisation setting |
| 09:38.48 | TEOX | I understand the x100p cards don't have this setting??? |
| 09:38.59 | TEOX | but the flasher 4 port fxo do? |
| 09:39.02 | JT | i don't know |
| 09:39.02 | TEOX | something like that? |
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| 09:41.47 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:48.21 | nasls_lsa | when I put the BeroNet BN4S0 card on the pc I loose network conectivity , any ideas ? Suse 10.1 operating system .. |
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| 10:14.54 | Ahrimanes | oej: morning |
| 10:14.54 | oej | morning |
| 10:15.51 | *** join/#asterisk Brokenheart` (n=ganesh@202.143.4.253) |
| 10:16.12 | Brokenheart` | hi |
| 10:17.46 | Brokenheart` | I has a query regarding asterisk system. I will be installing asterisk for the first time for my office with max 50 users. But before installation i wanted to make sure how much hardware configuration will be needed to run the system for 50 users |
| 10:17.52 | Brokenheart` | i'll only be using voip |
| 10:18.31 | Brokenheart` | Anyone ? |
| 10:19.14 | nasls_lsa | there is a PDF |
| 10:19.26 | Ahrimanes | Brokenheart`: hm, 1 ghz+, 512 mb ram should do |
| 10:19.34 | TEOX | decent card |
| 10:19.37 | nasls_lsa | that you can take an idea of what hardware you need |
| 10:20.02 | Brokenheart` | nasls_lsa: yeh i'm going through the pdf .. but there were plenty of example so was not sure enuf |
| 10:20.07 | Brokenheart` | thanks guyz for the help |
| 10:20.20 | Brokenheart` | i'll try the installation on dual xeon with 4gb |
| 10:20.31 | Brokenheart` | i hope that will take care of the hard ware requirement |
| 10:20.43 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 10:21.13 | TEOX | more than enough |
| 10:21.21 | Brokenheart` | ;) TEOX |
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| 10:23.05 | TEOX | i guessing you got a few old rack mounts by the sounds of it :) |
| 10:24.23 | Brokenheart` | nah its not rack mountable |
| 10:24.47 | Brokenheart` | yeh i have couple of spare servers basically used for testing purpose |
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| 10:57.23 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 10:57.56 | voltagex | hi, is there a way I can call a function, like system() after every call connects and disconnects? |
| 10:58.42 | voltagex | so a call coming in would trigger the function, a call being made would trigger the function and a hangup would trigger the function |
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| 11:08.28 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 11:10.34 | voltagex | anyone awake? |
| 11:16.41 | tzafrir | no |
| 11:17.27 | tzafrir | voltagex, the answer is generally "the dialplan". That's what it is for |
| 11:17.41 | voltagex | yeah |
| 11:17.55 | tzafrir | Dial(whatever) |
| 11:17.59 | voltagex | but I don't want to have to add a function to everything |
| 11:18.18 | voltagex | or can I use a pattern match? |
| 11:18.22 | tzafrir | System(whatever); note that this is the application, not the function. function called system()? |
| 11:18.54 | voltagex | yeah |
| 11:19.24 | tzafrir | maybe you can use the h (hangup action) extension |
| 11:20.16 | voltagex | well honestly, I'll be running script to stop and start my torrents |
| 11:20.54 | *** join/#asterisk dj-fu (i=aj@unaffiliated/dj-fu) |
| 11:20.56 | voltagex | so I need it to also work when calls first connect |
| 11:20.56 | voltagex | either incoming or outgoing |
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| 11:34.53 | voltagex | . |
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| 12:14.57 | voltagex | anyone alive? |
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| 12:48.29 | jeremy_g | hi |
| 12:48.52 | jeremy_g | what is calling presentation? is it the Display Name in SIP |
| 12:49.13 | jeremy_g | isn't it a part of callerid? |
| 12:50.28 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 12:51.34 | JT | no |
| 12:52.20 | JT | calling presentation is what the status of calling presentation permissiability is :) |
| 12:52.23 | anonymouz666 | is it possible to change the From field in CDR? the telco sends 1248888-8888 as callerID I wanna record only 248888-8888 |
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| 13:03.50 | saftsack | hi |
| 13:03.55 | saftsack | does somebody use the asterisk fop? |
| 13:04.35 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 13:04.46 | zdrulio | hello |
| 13:04.59 | zdrulio | i dl stersstest SIPp |
| 13:05.22 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 13:05.31 | zdrulio | but i cant start IT |
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| 13:09.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:13.23 | zdrulio | can anybody help me ? |
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| 13:26.47 | *** join/#asterisk Jubei (n=jubei@adsl46-40static.access.altecnet.gr) |
| 13:27.45 | Jubei | guys, i have 3 Zap groups. Zap/g1 etc... Is there anyway , in my dialplan, to make a call with whichever is available? |
| 13:28.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:28.19 | dj-fu | yea |
| 13:28.37 | dj-fu | you already got it half done |
| 13:28.40 | dj-fu | /g1 |
| 13:28.57 | dj-fu | OUTBOUNDTRUNK=Zap/g1 |
| 13:29.29 | dj-fu | Dial(Zap/g1/number) |
| 13:29.42 | dj-fu | I'm not sure if you can group groups, though. |
| 13:29.59 | blitzrage | that'll call the first available channel in g1 |
| 13:31.09 | Jubei | so.. how do I make sure that if all channels in g1 are busy to move on to g2 ?:) |
| 13:31.33 | blitzrage | ${DIALSTATUS} |
| 13:31.33 | anonymouz666 | put everything on g1? is it right? |
| 13:32.02 | blitzrage | GotoIf($[${DIALSTATUS} = CONGESTION | ${DIALSTATUS} = CHANUNAVAIL]?dial_group_2,1) |
| 13:32.25 | anonymouz666 | blitzrage, is it possible to change the From field in CDR? the telco sends 1248888-8888 as callerID I wanna record only 248888-8888 |
| 13:32.33 | Jubei | anonymouz666: can I do that? in zapata.conf? |
| 13:33.00 | anonymouz666 | Jubei: the best approach is what blitzrage said, but yes zapata.con |
| 13:33.01 | anonymouz666 | f |
| 13:33.18 | fbcit | Qwell[]: How do I compile just the chan_skinny driver? Right now I am recompiling all of * for every hack to skinny. This takes too long. |
| 13:33.41 | blitzrage | fbcit: when you make a change to the code, just run 'make' and it'll only compile the module that changed |
| 13:34.07 | Jubei | blitzrage: where exactly does that line go? :) |
| 13:34.10 | fbcit | So no 'make clean'... that was probably my mistake. |
| 13:34.20 | blitzrage | Jubei: in the dialplan |
| 13:34.32 | fbcit | blitzrage: Thanks. |
| 13:34.36 | Jubei | blitzrage: ye, but.. where? after Dial/ ? :D |
| 13:34.38 | blitzrage | fbcit: make clean will remove all compiled modules, causing Asterisk to rebuilt the whole thing -- so yes, that was your problem. |
| 13:34.46 | blitzrage | Jubei: yes |
| 13:35.19 | Jubei | blitzrage: i see. So in my case, having 3 zap groups. I'd change that to dial_group_3,2,1 ? |
| 13:35.32 | blitzrage | no |
| 13:35.43 | blitzrage | dial_group_1,1 is an extension and priority |
| 13:35.49 | blitzrage | you need to read more documentation on how dialplans work |
| 13:35.59 | nasls_lsa | I installed a BeroNet BN4S0 card on my linux , and installed mISDN . How do I check a) if all are fine - line works b) configure it with asterisk. can't see any midsn command in CLI ... |
| 13:36.04 | Jubei | blitzrage: ok thanks :) |
| 13:36.19 | blitzrage | i.e.: exten => dial_group_1,1,Dial(Zap/g2/${MY_EXTENSION}) |
| 13:36.39 | Jubei | blitzrage: i see. thank you I'll read up on that |
| 13:37.29 | blitzrage | or even: exten => _dial_group_X,1,Dial(Zap/g${EXTEN:-1}/${DID}) |
| 13:37.57 | blitzrage | which would work for "dial_group_[0-9]" |
| 13:38.09 | blitzrage | ~docs |
| 13:38.11 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
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| 13:45.12 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 13:45.56 | Jubei | which is a good place to read up on setting up a dialplan (but for asterisk 1.2) |
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| 13:48.16 | saftsack | is it possible to originate new outbound calls with FOP? |
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| 13:53.48 | Jubei | here's a strange thing that's occuring. I have say 2 BRI lines going into my MultiBRI card. 1st line's number is 505050, 2nd line's number is 515151 for instance. I've setup zapata.conf so that it sends everything to [default]. In [default] I've got exten => s,1,..do something .. When people dial the 1st bri line (505050) asterisk picks it up with the exten => s extension in default. But when ppl dial the other line asterisk rejects the |
| 13:54.26 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
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| 14:10.47 | fbcit | Any idea what causes this error: WARNING[27556]: chan_skinny.c:1267 find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on 'ext1'? |
| 14:11.56 | Qwell | fbcit: it's a warning - ignore it |
| 14:12.07 | aydiosmio | anyone know how to pronounce IDEFISK? |
| 14:12.25 | LeddyHM | Do you guys know of/recommend any user interfaces for the "follow me" feature instead of editing it in the config file manually? |
| 14:12.37 | aydiosmio | heheh, freepbx? |
| 14:13.15 | LeddyHM | We're not starting over :| |
| 14:13.32 | aydiosmio | can't handle the truth eh? |
| 14:13.55 | Mercestes | <PROTECTED> |
| 14:14.06 | LeddyHM | depends on if that "truth" is the only option |
| 14:14.30 | aydiosmio | Mercestes: ah -- any idea what it means? |
| 14:14.37 | aydiosmio | I'm so confused |
| 14:14.45 | Mercestes | LeddyHM, I wasn't aware that asterisk had a "follow me" feature. |
| 14:15.00 | aydiosmio | he can't handle the truth |
| 14:15.20 | Mercestes | aydiosmio, The truth is in Telepathy |
| 14:15.24 | LeddyHM | Ours does |
| 14:15.34 | LeddyHM | We didn't set it up though, vendor di |
| 14:15.37 | Mercestes | LeddyHM, That's some magic asterisk then. |
| 14:15.37 | LeddyHM | then they went awol |
| 14:15.48 | aydiosmio | probably copied it out of freepbx |
| 14:16.15 | Mercestes | I've always had to manually configure my dialplan to try a series of numbers off of varying available technolgoies. Man, it would be nice if Asterisk was intuitive enough to implement that as a feature. |
| 14:16.37 | LeddyHM | I guess that answers my question then |
| 14:16.42 | LeddyHM | we got lucky ;) |
| 14:16.43 | aydiosmio | ringall, ringall-prim, hunt, hunt-prim, memoryhunt, memoryhunt-prim |
| 14:16.51 | LeddyHM | and that it's a manual process to configure |
| 14:16.57 | aydiosmio | look familiar? |
| 14:17.09 | Cybertoy | good morning |
| 14:17.17 | Mercestes | LeddyHM, That is correct. |
| 14:17.33 | Cybertoy | woops.. wrong chat window ... :) |
| 14:17.36 | Mercestes | LeddyHM, It's just like dialing a sip phone, except you try a series of numbers instead of just one. |
| 14:17.39 | Cybertoy | but wish you guys a good morning anyway... |
| 14:17.44 | Mercestes | Good morning Cybertoy |
| 14:17.53 | LeddyHM | it looks like they created a macro |
| 14:18.14 | Mercestes | LeddyHM, That'd be a good way to do it. |
| 14:18.32 | LeddyHM | awesome |
| 14:18.48 | LeddyHM | I'll make sure not to lose the config then :) |
| 14:18.49 | Mercestes | before you change a whole lot..... |
| 14:18.50 | Mercestes | it wouldn't reference a DB would it? |
| 14:19.15 | aydiosmio | LeddyHM: what's the macro called? |
| 14:19.30 | LeddyHM | macro-findme, macro-followme |
| 14:19.40 | LeddyHM | merc: doesn't look that way |
| 14:19.52 | LeddyHM | ayd: looking to see if it's a freepbx rip? |
| 14:20.36 | Mercestes | LeddyHM, Does it call an AGI at anypoint? |
| 14:20.43 | aydiosmio | LeddyHM: yeah |
| 14:21.00 | aydiosmio | freepbx both have meetme and followme configurations built in |
| 14:21.06 | aydiosmio | but meh |
| 14:21.15 | LeddyHM | I haven't seen an AGI call |
| 14:21.20 | LeddyHM | what would I look for? |
| 14:21.31 | Mercestes | LeddyHM, AGI() |
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| 14:22.41 | LeddyHM | no |
| 14:22.59 | LeddyHM | but it has "s-NOANSWER" |
| 14:23.11 | LeddyHM | is that "standard"? |
| 14:23.56 | Mercestes | Yea. |
| 14:24.07 | Mercestes | There should be a Goto(s-${DIALSTATUS} somewhere. |
| 14:24.16 | Mercestes | could hire a consultant |
| 14:25.10 | nasls_lsa | I have asterisk 1.4.0 , and at CLI> I can't see any support of misdn .. any ideas ? |
| 14:25.35 | Jubei | nasls_lsa: have you setup chann_mISDN? |
| 14:26.05 | nasls_lsa | yes , as the instructions said , but even that I don't know how to check if it works fine .. |
| 14:26.16 | nasls_lsa | I didn't get any error .. |
| 14:26.21 | LeddyHM | Merc: naw I'll just hardcode the changes |
| 14:26.23 | LeddyHM | whee |
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| 14:26.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:26.52 | LeddyHM | as I'd like to stay away from freepbx |
| 14:27.25 | Mercestes | LeddyHM, tar -zcvf save_my_job.tgz /etc/asterisk /etc/zaptel.conf |
| 14:27.34 | nasls_lsa | I followed the instructions from beronet |
| 14:28.24 | LeddyHM | Merc: *grin* |
| 14:29.59 | aydiosmio | ah here it is |
| 14:30.00 | aydiosmio | [ext-findmefollow] |
| 14:30.23 | fbcit | Qwell[]: Maybe I'm chasing a rabbit, but the subchannel warning seems to be related to the failure I am seeing in handling the speeddial stimulus for the 30VIP. It appears that the template is applied, but I receive this debug output with no number dialed: unknown1 in handle_stimulus_message is '0' Received Stimulus: SpeedDial(1) |
| 14:30.54 | aydiosmio | fpbx uses the sql server to store the settings and dumps that context to the conf file on each reload |
| 14:31.30 | aydiosmio | ${DB(AMPUSER/500/followme/grpconf)} |
| 14:33.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:34.00 | LeddyHM | interesting |
| 14:34.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:34.25 | LeddyHM | Our #'s are hardcoded in the conf file |
| 14:36.08 | aydiosmio | I dunno, though. If you're using asterisk as a general pbx and without a lot of custom applications freepbx is pretty great for easy user and feature management |
| 14:36.25 | aydiosmio | otherwise it does get in the way |
| 14:39.12 | fbcit | Qwell[]: Am I correct in thinking that there is no code to handle STIMULUS_SPEEDDIAL? Is this some of what is 'not implimented?' |
| 14:40.50 | Mercestes | aydiosmio, With the small side-effect of loosing all respect/support in this channel the moment you install it. |
| 14:40.50 | Mercestes | aydiosmio, It also causes testicular shrinkage. |
| 14:41.20 | aydiosmio | Mercestes: you don't respect me?:( |
| 14:41.57 | Mercestes | aydiosmio, Not if you use fpbx. |
| 14:42.44 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 14:44.24 | aydiosmio | cpm: it doesn't "suck" it just has a mind of it's own and can clobber your asterisk customizations |
| 14:44.43 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 14:44.57 | cpm | aydiosmio, no thanks |
| 14:44.58 | aydiosmio | 2.2.0 is a really good release, which totally fixed all the stupid shit I hated about the 1.x branch |
| 14:45.15 | aydiosmio | fpbx sucked back then |
| 14:46.32 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 14:48.23 | Mercestes | fpbx still sucks. |
| 14:48.42 | Mercestes | it doesn't *do* anything that an intelilgent being can't accomplish with nominal reading |
| 14:48.51 | aydiosmio | it does it faster |
| 14:49.02 | Mercestes | no it doesn't. |
| 14:49.07 | aydiosmio | I don't want to worry about managing IVRs or peers |
| 14:49.21 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 14:49.38 | anonymouz666 | 2.2.0 integrate SugarCRM? |
| 14:49.50 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 14:49.51 | aydiosmio | I write AGIs, but them in the custom.conf and they appear in the IVR, it's great |
| 14:49.59 | aydiosmio | s/but/put |
| 14:50.06 | aydiosmio | anonymouz666: yeah it's in there |
| 14:50.11 | Mercestes | Trolls |
| 14:50.17 | Mercestes | talk about #freepbx in #freepbx. |
| 14:50.42 | blitzrage | Mercestes: I'd have to say your attitude is less than ideal |
| 14:50.44 | aydiosmio | Mercestes: I think you're the troll |
| 14:51.18 | aydiosmio | but yeah little too much fpbx talk |
| 14:51.20 | aydiosmio | I'll stfu |
| 14:51.26 | Mercestes | aydiosmio, Oh? I talk about Asterisk in the #asterisk channel and stay out of #freepbx in my "it sucks" attitude. |
| 14:51.37 | aydiosmio | yeah go ahead |
| 14:51.38 | Mercestes | Not vice verse. |
| 14:52.03 | aydiosmio | or you could get all defensive and snobby |
| 14:52.07 | Mercestes | blitzrage, noted. Thanks. Ditto, btw. |
| 14:52.08 | aydiosmio | that's fine too |
| 14:52.34 | nasls_lsa | exit |
| 14:52.40 | nasls_lsa | ?ex |
| 14:53.01 | Mercestes | aydiosmio, It' snot defensive and snobby. We don't support freepbx in here and it's a convoluted mess and theres no one in #freepbx to offer support for it. |
| 14:53.13 | Mercestes | I personally think that it's poor advice to direct anyone to it. |
| 14:53.27 | Mercestes | and I don't think yo ushould be doing so as "support advice" in #asterisk. |
| 14:53.40 | Mercestes | Need help with Asterisk? Just install this thing that will cause #asterisk to not support you anymore. |
| 14:53.44 | aydiosmio | well, correct me if I am wrong |
| 14:53.51 | Mercestes | You are wrong. |
| 14:53.57 | aydiosmio | but he asked for a configuration tool for a feature that freepbx supports natively |
| 14:54.04 | Mercestes | You are no D-fender, Qwell or file and last time I checked there aren't any in #freepbx. |
| 14:54.23 | aydiosmio | there are, I've talked to their devs and had issues resolved |
| 14:54.27 | *** join/#asterisk Fieldy (i=3Jrz7nUe@gentoo/contributor/Fieldy) |
| 14:54.39 | Mercestes | freepbx doesn't support anything "natively" i thas a template to plug in values. |
| 14:54.56 | Mercestes | It's like saying that Microsoft word supports fax cover pages "natively." |
| 14:55.16 | aydiosmio | so I was wrong to suggest that to him? |
| 14:55.32 | aydiosmio | noting cavets |
| 14:55.32 | Mercestes | Yes. Unless your going to offer him ongoing support in #freepbx. |
| 14:56.02 | aydiosmio | caveats that is |
| 14:56.05 | Mercestes | In my experience, about 90% of the people we point to #freepbx are here becuase nobody answered their question. |
| 14:56.19 | Mercestes | I would actually appreciate it if you offered support there so they would quit coming here. |
| 14:56.26 | aydiosmio | I do |
| 14:56.35 | Mercestes | Great! |
| 14:57.56 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 14:58.30 | Qwell[] | what the hell |
| 14:58.36 | Qwell[] | are there 988 numbers now? |
| 14:59.01 | aydiosmio | cool |
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| 14:59.13 | Mercestes | Qwell[] are they toll free? |
| 14:59.19 | aydiosmio | are they direct hotlines to sexy, lonely babes? |
| 14:59.20 | Qwell[] | Mercestes: no, they wouldn't be |
| 14:59.25 | Corydon-w | Qwell[]: there always were. Malawi |
| 14:59.30 | Qwell[] | Malawi? |
| 14:59.40 | Corydon-w | Caribbean island |
| 14:59.40 | Mercestes | is that like an evil wii? |
| 14:59.45 | Mercestes | oh. |
| 15:00.07 | anonymouz666 | Corydon-w: there are some CDR vars that are only read-only right? |
| 15:00.09 | Qwell[] | dunno, the way the number is formatted (some message to the -dev list), it looks...weird |
| 15:00.10 | Corydon-w | That's one of those "don't call it unless you feel like getting charged $2/minute for international |
| 15:00.19 | Corydon-w | anonymouz666: correct |
| 15:00.43 | Corydon-w | anonymouz666: all of the standard variables, with the possible exception of accountcode and userfield are readonly |
| 15:00.53 | anonymouz666 | the damn telco send the calleridnum to me 1248888-8888 i wanna strip off the 1 and save in cdr like 248888-8888 |
| 15:01.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:01.15 | Corydon-w | anonymouz666: then you need to change CALLERID(num) |
| 15:01.30 | Corydon-w | Or CALLERID(ani) |
| 15:01.47 | Corydon-w | That's the reason the CDR is readonly -- because it reflects another setting |
| 15:02.20 | anonymouz666 | yeap I need to change the ANI |
| 15:05.27 | anonymouz666 | I think callerid(ani) is only in 1.4? |
| 15:06.22 | Corydon-w | It's in 1.2, too |
| 15:06.28 | Corydon-w | but it's CALLERID(ani) |
| 15:06.34 | Corydon-w | case sensitive |
| 15:06.55 | *** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at) |
| 15:07.38 | canapa | can someone gimme a hint for a good recording tool and converting tool to gsm format ? |
| 15:07.56 | putzz | sox |
| 15:08.09 | putzz | not recording tho |
| 15:08.28 | aydiosmio | /dev/linein > soundfile.wav |
| 15:08.54 | aydiosmio | tricky, but it'll work |
| 15:09.07 | Corydon-w | You can use Asterisk with the Record app to record |
| 15:09.22 | Corydon-w | best of all, you can save it directly in gsm format |
| 15:10.56 | fbcit | Qwell[]: Am I correct in thinking that there is no code to handle STIMULUS_SPEEDDIAL? Is this some of what is 'not implimented?' |
| 15:11.18 | Qwell[] | fbcit: see response to -users list. But yes, you are correct |
| 15:11.55 | anonymouz666 | exten => _1XXXXXXXXXX/s,1,Set(CALLERID(ani)=${ANI:1}) |
| 15:12.36 | anonymouz666 | look Corydon-w, what a beautiful exten |
| 15:12.38 | anonymouz666 | hehe |
| 15:12.40 | fbcit | Thanks. Sorry for the double post. I'm going to work on fixing it. No guarantees..... :P |
| 15:12.54 | Corydon-w | ANI is not a standard variable, you know |
| 15:13.01 | Qwell[] | fbcit: the fix is probably pretty trivial, really |
| 15:13.08 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 15:13.16 | pigpen | If I had: [Apr 2 09:08:50] VERBOSE[21843] logger.c: == Spawn extension (from-sip, 1031, 24) exited non-zero on 'SIP/1031-51923810<ZOMBIE>' |
| 15:13.16 | pigpen | [Apr 2 09:08:50] WARNING[21843] channel.c: Channel 'SIP/1031-51923810<ZOMBIE>' may not have been hung up properly |
| 15:13.23 | Corydon-w | But you could do Set(CALLERID(ani)=${CALLERID(ani):1}) |
| 15:13.40 | pigpen | ....then had asterisk 1.4.2 die....would this point to the issue? (doesn't me) |
| 15:13.42 | Qwell[] | Corydon-w: that works? |
| 15:13.45 | fbcit | It works in sccp, but so much else is broke there. |
| 15:13.51 | Corydon-w | Qwell[]: yep |
| 15:13.54 | Qwell[] | neat |
| 15:14.32 | pigpen | MOH on another channel stopped at the same time. |
| 15:14.44 | Corydon-w | Qwell[]: when I pointed out how many people do ${CALLERIDNUM:1}, we had to make it do that |
| 15:14.46 | pigpen | I didn't know if the "ZOMBIE" thing meant much. |
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| 15:24.21 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:25.49 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:26.02 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:31.53 | *** join/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 15:31.54 | kieran491 | hi |
| 15:32.07 | kieran491 | what is a good software phone for asterisk? |
| 15:34.56 | pigpen | eyebeam (sip) and idefisk (iax) |
| 15:35.12 | blitzrage | idefisk does both SIP and IAX |
| 15:35.18 | blitzrage | and is my favourite softphone |
| 15:35.40 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:35.42 | kieran491 | hmm k |
| 15:35.46 | kieran491 | sweet thanks |
| 15:35.52 | pigpen | it does? shit. how did I miss that. |
| 15:35.56 | *** join/#asterisk toombaloomba (n=phph@do.you.like.my.frippers.com) |
| 15:36.00 | pigpen | I have been using it .... a long time. |
| 15:36.02 | blitzrage | pigpen: new version |
| 15:36.07 | blitzrage | idefisk2 |
| 15:36.11 | pigpen | Windozzz? |
| 15:36.14 | blitzrage | of course |
| 15:36.17 | pigpen | ugh. |
| 15:36.19 | blitzrage | windows/linux/mac |
| 15:36.29 | blitzrage | I'm using the linux version |
| 15:36.32 | pigpen | I use mac..so I will have to check. |
| 15:36.49 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:36.50 | pigpen | I feel like a hippie when I use my mac. |
| 15:36.54 | blitzrage | heh |
| 15:37.01 | pigpen | but a hacker when I use my linux box. |
| 15:37.09 | pigpen | and a failure when I use windows. |
| 15:37.11 | pigpen | :) |
| 15:37.44 | blitzrage | I'm not a Mac fan. I use Linux as my desktop and servers, and Windows in VMware to run Quickbooks |
| 15:37.45 | [TK]D-Fender | pigpen: Get a haircut, and get a real job.... |
| 15:37.52 | pigpen | ha! |
| 15:38.00 | blitzrage | Just like your big brother Bob |
| 15:38.27 | [TK]D-Fender | pigpen: Clean your act up, and don't be a slob |
| 15:38.31 | kieran491 | Hi there is a list for all of these is'nt there? for the software phones that is.. |
| 15:38.35 | pigpen | I actually cut my hair yesterday: 1/16th" |
| 15:38.48 | pigpen | [TK]D-Fender, you can see my office? |
| 15:38.50 | pigpen | shit.. |
| 15:40.04 | [TK]D-Fender | pigpen: I've let mine grow out. Nobody I've seen a year ago would recognize me for that and having done Lasik. |
| 15:40.10 | pigpen | On the serious side....if I get a channel.c "bla bla" ZOMBIE, right before Asterisk dies...would this point to the issue? |
| 15:40.24 | pigpen | [TK]D-Fender, getting sexy? |
| 15:40.28 | [TK]D-Fender | pigpen: No on to phase 3 - body-building again.... |
| 15:40.37 | [TK]D-Fender | s/no/now/ |
| 15:40.50 | pigpen | [TK]D-Fender, yeah...same here....but I need to get -back- to bodybuilding. |
| 15:40.54 | [TK]D-Fender | pigpen: Sexy-ER ;) |
| 15:41.20 | [TK]D-Fender | pigpen: If such a state is even possible.... |
| 15:41.48 | [TK]D-Fender | pigpen: I'm a no-drugs kinda guy (with a NATURAL chemical imbalance like mine... who NEEDS them?! :D ) |
| 15:41.53 | pigpen | yeah...for me it is mostly an issue of living forever... |
| 15:42.09 | pigpen | Well, 15 years ago...steroids wern't bad.... |
| 15:42.35 | bulle | pigpen: steroids surely was just as bad 15 years ago as they are today |
| 15:42.46 | pigpen | yeah..just better educated. |
| 15:43.02 | pigpen | The worst I "do" now is starbucks. |
| 15:43.11 | blitzrage | "The Man Whose Arms Exploded". Nuff said. |
| 15:43.30 | pigpen | Man, there are some amazing videos of Ronnie Coleman on youtube. |
| 15:43.44 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 15:44.28 | bulle | scary, so several people here actualy more or less train bodybuilding more or less seriously ?' |
| 15:45.17 | pigpen | Well, I haven't for some time...(ie: 10 years). |
| 15:45.37 | pigpen | the first thing I did was hit the weights as I used to do... |
| 15:45.47 | pigpen | heh....not a good idea... |
| 15:46.02 | bulle | he, i can imagine |
| 15:46.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:46.17 | bulle | i just do some basic weightlifting to keep somewhat in shape |
| 15:46.17 | pigpen | yeah...I could do pretty good...but I paid for it for 2 weeks. |
| 15:46.20 | *** join/#asterisk thapa (n=thapa@d86-33-73-99.cust.tele2.at) |
| 15:46.43 | pigpen | ...but until I have my hardwood floor down...no workouts... |
| 15:46.47 | bulle | i still look like a miniature hammer-thrower though |
| 15:47.06 | pigpen | I look like homer simpson. |
| 15:47.08 | pigpen | ah. |
| 15:47.13 | bulle | hehe |
| 15:47.14 | pigpen | JK. |
| 15:47.26 | bulle | 110kb/169cm |
| 15:47.27 | red9012 | is there a way to accept/refuse a queue call, once a call is received from a queue? |
| 15:47.28 | bulle | ehm, kg |
| 15:47.36 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 15:47.37 | pigpen | anybody reporting that asterisk 1.4.2 is dieing? |
| 15:47.50 | JacksLivr | morning all, if I want to get input from a dialpad and save it as a variable and then i want to get another input from dialpad and save it as a different variable, how would i do that? |
| 15:47.58 | bulle | pigpen: nope, only had some sip problems, eg sip channels lingering when phoning non existing extensions |
| 15:48.08 | JunK-Y | pigpen: yep, see bugs.digium.com |
| 15:48.21 | FreezeS | I've got a problem with ChanIsAvail. Sometimes I get this: SIP Seeding peer from astdb .... |
| 15:48.46 | FreezeS | and that seeding is false because the user is actually not registered to that computer |
| 15:49.08 | FreezeS | does anyone has an idea about this ? |
| 15:49.24 | *** part/#asterisk thapa (n=thapa@d86-33-73-99.cust.tele2.at) |
| 15:50.35 | pigpen | JunK-Y, referring to the meetme segfault? |
| 15:51.46 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 15:52.14 | Mercestes | pigpen: Heard about a few segfaults here too. Dunno what caused them. |
| 15:53.18 | pigpen | yeah..mine (customers) died after running for 3 days. |
| 15:53.23 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:53.29 | pigpen | but today is the first heavy day.... |
| 15:54.02 | pigpen | after updating to 1.4.1, every box I installed it on died....but ran fine afterwards |
| 15:54.04 | *** join/#asterisk eald (n=eald@189.157.162.200) |
| 15:56.15 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:57.59 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:00.07 | JunK-Y | pigpen: generate a text file, like requested in doc/backtrace.txt and i will see what i can do. |
| 16:00.29 | pigpen | k..I will see if I can get it to do it again. |
| 16:00.32 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:00.43 | *** join/#asterisk RTman (n=rtorrey@208.236.9.10) |
| 16:01.54 | RTman | Anybody else getting this error with 1.4 (chan_zap.c: !! Got reject for frame 52, retransmitting frame 55 now, updating n_r!) |
| 16:02.04 | [TK]D-Fender | JacksLivr: "show application read" |
| 16:02.21 | RTman | Used to be just a warning under 1.2.17 |
| 16:02.41 | *** join/#asterisk ManxPower (n=manxpowe@155.sub-75-203-175.myvzw.com) |
| 16:03.20 | RTman | what about "show application read"? |
| 16:04.08 | *** join/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 16:05.22 | JunK-Y | %ast core show application read |
| 16:06.44 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 16:08.01 | RTman | Yeah - and what do you want to know from ast core show application read |
| 16:08.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:08.51 | wunderkin | RTman, that was not for you |
| 16:09.02 | RTman | oh sorry |
| 16:09.19 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 16:09.23 | pigpen | me. |
| 16:09.25 | RTman | <PROTECTED> |
| 16:10.03 | RTman | pigpen - any clue what;'s doing it. message is from libri module |
| 16:10.04 | pigpen | RTman, you may have some slip errors or possibly a weak db off of your smartjack |
| 16:10.20 | FreezeS | what is astdb ? |
| 16:10.34 | pigpen | asterisk database. |
| 16:10.38 | pigpen | based on db 1.3 ? (ver? |
| 16:10.43 | giasai68 | hello, can you help me to fix my problem? in my extensions i write exten => 39XXXXXXXX.,1,dial...... this exten work if i dial 8 digits after 39..asterisk processing dial if i digit less of 8 time digits tha call not start if i don't' have the number of time digit how can i do to start call from asterisk? |
| 16:10.49 | RTman | maybe slips...but we are in carrier's collo. |
| 16:11.16 | wunderkin | that isnt a slip, i dont think it is anything to worry about |
| 16:11.23 | FreezeS | pigpen: so why is ChanIsAvail doing this: SIP Seeding peer from astdb ? |
| 16:12.03 | RTman | well I took 410000 calls and got close to 67000 of these errors. |
| 16:12.09 | DarKnesS_WolF | anyone paged giptel200 sip phone before ? |
| 16:12.17 | pigpen | FreezeS, running RTA? |
| 16:12.35 | FreezeS | pigpen: what's RTA ? |
| 16:12.44 | pigpen | Real Time Asterisk |
| 16:12.55 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:12.58 | FreezeS | pigpen: yes |
| 16:13.10 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:13.17 | FreezeS | and on one workstation it works perfectly. On it's clone it doesn't |
| 16:13.18 | pigpen | I would google your error...I see some posts about it. |
| 16:13.21 | pigpen | what DB are you using? |
| 16:13.26 | FreezeS | mysql |
| 16:13.33 | FreezeS | I did google, but nothing relevant |
| 16:13.35 | pigpen | Sorry..I am running Postgres. |
| 16:13.54 | FreezeS | what pisses me off is that this error is kinda random |
| 16:14.06 | FreezeS | happends on some computers, doesn't on others |
| 16:14.19 | *** join/#asterisk Fieldy (i=ueymRuZi@gentoo/contributor/Fieldy) |
| 16:14.59 | *** join/#asterisk [shodan] (n=shodan@ip094.99-113-216.pppoe4.joliette.intermonde.net) |
| 16:15.02 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 16:15.16 | RTman | pigpen - any other thoughts? |
| 16:19.19 | *** join/#asterisk hypnox (n=dan@lleuad.ocq.omnicea.net) |
| 16:19.40 | hypnox | anyone know of a sipchaninfo equivalent to IAX? I want to find out the source IP of an incoming IAX call |
| 16:19.45 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:20.34 | FreezeS | doesn't "IAX show peers" work ? |
| 16:20.56 | pigpen | RTman, in your zapata.conf put: resetinterval=never |
| 16:21.00 | hypnox | well i want to do it from an agi script.. |
| 16:21.10 | RTman | pigpen yup |
| 16:21.24 | pigpen | yup = already there? |
| 16:21.27 | RTman | pigpen - already had that in there. |
| 16:21.41 | pigpen | hmm...check for slips and low db. |
| 16:21.53 | pigpen | even though you are in a colo, you could still be getting a weak signal. |
| 16:22.19 | pigpen | slips are normally a timing issue. make sure you don't have interrupt confilcts. |
| 16:22.22 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:22.31 | RTman | pigpen...why do the errors disappear under 1.2.17? |
| 16:22.52 | pigpen | I got up from my chair and sat back down: update! |
| 16:23.08 | pigpen | I moved from ver 1.2.9 to 1.2.16 and all was fine. |
| 16:23.29 | red9012 | is there a way to customize the directory cmd dir-intro? |
| 16:23.49 | RTman | pigpen - 1.4 has better SendDTMF module and I have backend systems that need that. |
| 16:25.14 | *** part/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 16:25.17 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 16:25.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:26.10 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 16:26.18 | pigpen | yeah...we are running 1.4.2 and zap "to be" 1.4.2 |
| 16:28.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-1-13.red.bezeqint.net) |
| 16:28.14 | *** join/#asterisk feistel (n=feistel@host79.201-253-140.telecom.net.ar) |
| 16:28.34 | feistel | hi |
| 16:28.50 | feistel | I need help configuring my SPA3000 con asterisk |
| 16:28.54 | feistel | anyone can help me ? |
| 16:29.24 | pigpen | there are very good example online. |
| 16:29.37 | feistel | pigpen, please get me one |
| 16:33.43 | quidpro | Hmm... is there a way to pick-up a call from a queue, if the queue isn't trying extensions (ie. just missed the phone)? |
| 16:33.50 | RTman | pigpen - how's that working? |
| 16:33.59 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:34.36 | Hmmhesays | ok how do I make mod_dingaling spit out more debug info |
| 16:34.53 | Hmmhesays | <param name="debug" value="1"/> |
| 16:35.00 | Hmmhesays | in dingaling.conf.xml |
| 16:35.12 | giasai68 | <PROTECTED> |
| 16:35.13 | pigpen | RTman, working pretty good....had a hickup this am...but may be nothing. |
| 16:35.27 | hedge77 | quidpro: i dunno the answer to your question but you're not using dynamic members by any chance are you? |
| 16:35.30 | Hmmhesays | crap wrong channel |
| 16:35.47 | quidpro | hedge: No static |
| 16:35.58 | quidpro | hedge: No,,, they are static. |
| 16:36.47 | hedge77 | quidpro: ugh i'm trying to find someone else with this problem -> http://bugs.digium.com/view.php?id=9421 |
| 16:37.31 | quidpro | hedge: nope, doesn't apply here |
| 16:37.31 | GreyFoxx | Is astman proxy generally considered the "best" manager proxy out there? |
| 16:38.36 | GreyFoxx | I've been asked to install it for a couple of clients to use with "SNAP" but I want to make sure I can configure it so that there is no way they can see events from other users who at least users who are not members of a predefined group |
| 16:38.37 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:41.02 | RTman | pigpen..what libpri version are you using? |
| 16:42.13 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:43.32 | *** join/#asterisk lorinc (n=ang@pool-2681.adsl.interware.hu) |
| 16:45.46 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 16:50.03 | JunK-Y | http://tools.ietf.org/html/rfc4824 |
| 16:53.44 | hedge77 | which codecs will be best at handling high-latency connections? |
| 16:56.43 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 16:58.31 | *** join/#asterisk rrocha (n=ruy@201.22.26.180.adsl.gvt.net.br) |
| 16:59.02 | *** part/#asterisk rrocha (n=ruy@201.22.26.180.adsl.gvt.net.br) |
| 16:59.16 | *** join/#asterisk friedrich| (n=friedric@e177254204.adsl.alicedsl.de) |
| 16:59.32 | Mahmoud | hello |
| 16:59.45 | Mahmoud | what's a hot line phone? is it the number that's dailed by simply going off-hook? |
| 16:59.47 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:00.28 | hedge77 | i've never heard the term used outside of the "OMG RED PHONE NUKE 'EM" but it makes sense |
| 17:00.41 | hedge77 | your description i mean |
| 17:01.04 | Mahmoud | the book i read is given examples in US |
| 17:01.14 | Mahmoud | giving* |
| 17:03.06 | Mahmoud | to me it sounds like security phones that are only used to dial 911 or whatever similar by simple going off hook |
| 17:03.21 | hedge77 | i agree |
| 17:03.23 | Mahmoud | so, just pickup the phone and speak, no need to dial |
| 17:03.55 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 17:04.01 | Mahmoud | i'll make my house phone this way, so whenever any one picks the phone, he calls me, i'll be the centre of the whole universe! |
| 17:04.08 | Mahmoud | j/k |
| 17:05.27 | *** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
| 17:05.31 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:05.45 | Ambrose | Anyone able to help me out? I'm getting "Got SIP response 400 "Bad Request" back from 192.168.1.70" when I try to call ext 100 from ext 101, which are two cisco 7960 phones setup next to each other, it goes right to voicemail. It works when calling in/out though... ? (thx) |
| 17:05.46 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 17:06.49 | kannan | hello, i am not able to connect to the asterisk server to use the Asterisk .Net classes , any poiters or howtos , i am not able to find any except gotdotnet.com, and it doesnt seem to have all the answers |
| 17:07.14 | kannan | for AGIs in .NET |
| 17:08.38 | ManxPower | Ambrose: does it give anything ELSE on the console? |
| 17:10.54 | Ambrose | ManxPower : 2 lines that look "out of the ordinary" to me: -- SIP/100-3d58 is circuit-busy |
| 17:10.55 | Ambrose | <PROTECTED> |
| 17:11.22 | ManxPower | Ambrose: does "sip show peers" show the correct IP address for both devices? |
| 17:11.42 | Ambrose | Looks like it, just verifying |
| 17:11.58 | *** join/#asterisk fetcher (n=jnh@ip-209-172-35-240.static.privatedns.com) |
| 17:12.21 | Ambrose | ManxPower : Yes, 101/101 192.168.1.83 D 5060 Unmonitored |
| 17:12.26 | ManxPower | Ambrose: For testing use disallow=all and allow=ulaw in sip.conf [general] and no other allow or disallow lines |
| 17:12.27 | Ambrose | same for ext 100 |
| 17:12.37 | Ambrose | ManxPower : Okay thanks |
| 17:12.54 | ManxPower | do a "reload" then try again. If it works, then you have a codec issue. |
| 17:13.27 | Ambrose | Ok, changing now. Strange (to me) though is that it gets call from outside the pstn and can call out too |
| 17:14.19 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 17:14.38 | Ambrose | ManxPower : Ok, I changed the disallow and allow and did a reload but it still wont call the other ext |
| 17:15.05 | ManxPower | Ambrose: looks like you will have to do the very unpleasant task of looking at sip debug |
| 17:15.20 | Ambrose | sounds fun :p |
| 17:15.24 | Ambrose | ManxPower : Ok, thanks |
| 17:17.52 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:18.10 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:18.21 | RTman | Manxpower -- is Sip/100 registered? if not then it can't get any calls. |
| 17:18.51 | Ambrose | How do I find that out? |
| 17:18.58 | RTman | sip show peers |
| 17:19.13 | Ambrose | the ext/ip shows up in there |
| 17:19.32 | RTman | Does it show registered? |
| 17:19.42 | Ambrose | Status: Unmonitored |
| 17:19.42 | Ambrose | ? |
| 17:20.07 | JunK-Y | add a qualify=yes for that peer. |
| 17:20.20 | Ambrose | in the sip.conf ? |
| 17:20.26 | JunK-Y | ya |
| 17:20.28 | Ambrose | ok |
| 17:20.49 | RTman | does host field show ip address or unspecified? |
| 17:20.54 | JunK-Y | and do a sip reload, then sip show peers, ya will see. |
| 17:21.06 | Ambrose | RTman : It shows the IP |
| 17:21.18 | Mercestes | Qwell[]: I vote we change the subject. |
| 17:21.25 | Qwell[] | To? |
| 17:21.40 | RTman | Check NAT configuration on client |
| 17:22.32 | Mercestes | Asterisk: Today's schedule: 6a-10a: trollchat, what freepbx can do for you. 10a-7p, Asterisk: the open source PBX. 7p - 6a. The soothign sounds of crickets to pass the time as the channel idles. |
| 17:22.34 | RTman | Anybody else getting this error with 1.4 (chan_zap.c: !! Got reject for frame 52, retransmitting frame 55 now, updating n_r!)? |
| 17:22.34 | RTman | * darkskiez has quit IRC (Client Quit) |
| 17:22.57 | RTman | I am still slooking of help on this one. |
| 17:25.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:27.17 | Mercestes | RTman: Sorry, dunno. What is this on? PRI? BRI? FXO? FXS? Telepathy? |
| 17:27.40 | Qwell[] | Mercestes: shh, chan_telepathy is secret |
| 17:27.51 | Mercestes | oh, yea, sorry. |
| 17:28.04 | Qwell[] | jbot: remove Mercestes' tongue |
| 17:28.27 | Mercestes | :( |
| 17:29.38 | red9012 | I like to have the option of accepting/refusing calls received from queue. How can I do this? |
| 17:29.44 | ManxPower | RTman: perhaps you missed me telling him to do a "sip show peers" and then him telling me that both phones have an IP address listed. |
| 17:31.20 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@63.145.228.3) |
| 17:32.22 | Ambrose | :-) |
| 17:32.39 | Ambrose | Now instead of Unmonitored it shows OK (X ms) |
| 17:32.50 | Ambrose | But still shows the IP's |
| 17:33.22 | JunK-Y | so its fine. |
| 17:34.21 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-49fb92c3b8ff3f6d) |
| 17:34.59 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:35.43 | RTman | Mercestes - On PRI - Digium TE412P |
| 17:36.23 | JunK-Y | anyway to send a register and set an outbound-proxy at the same time? |
| 17:38.03 | Mercestes | RTman, Is it broken in any way? |
| 17:38.30 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 17:39.56 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@63.145.228.3) |
| 17:44.39 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 17:45.39 | kratzers | Is there a trick to getting ChanIsAvail to set ${AVAILSTATUS} to something other than 0 for a SIP channel? |
| 17:47.11 | Mahmoud | why? |
| 17:48.05 | kratzers | it never does, for me |
| 17:50.58 | *** join/#asterisk Jubei (n=Stormtro@147.27.46.26) |
| 17:51.11 | Jubei | guys what's the best way to make a dialplan? .ael or .conf? |
| 17:51.19 | file | kratzers: are you using call limits? |
| 17:51.19 | Jubei | (for * 1.2) |
| 17:51.31 | kratzers | yeah, just found http://bugs.digium.com/view.php?id=8301 |
| 17:54.33 | *** join/#asterisk rrocha (n=ruy@201.22.26.180.adsl.gvt.net.br) |
| 17:54.38 | RTman | Seems to answer calls ok. I just don't like seeing that mayn errors in the log file. |
| 17:55.28 | codefreeze | Jubei: AEL, of course (but, then, I'm a little biased! |
| 17:55.59 | codefreeze | Jubei: oh-oh! 1.2. I take it back. AEL2 then. |
| 17:57.40 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 18:03.13 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 18:10.56 | DarKnesS_WolF | i'm trying to connect a GSM device " as extnsion to asterisk " i connected the device port called phone " which provides tone " to FXO analog port. i can make calls and get calls .. when if someone calling from mobile ---> to the device asterisk pickup the call .. when when the mobile hangs up the device dosn't hang up . any idea ? |
| 18:11.20 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:11.38 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:11.38 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 18:12.06 | chefrs | So this is more of a phone question but, say a line is busy with Asterisk and someone tries to call it. I get just a "Line is busy" message and nothing. These are normal PSTN wired lines. |
| 18:12.21 | chefrs | Would a T1 fix this issue? Is there some digital solution that would allow multiple incoming connections on the same line? |
| 18:12.33 | Qwell[] | chefrs: define "line" |
| 18:12.48 | chefrs | A single phone line. A run of the mill twisted pair. |
| 18:13.27 | Mercestes | chefrs, A T1 would allow 23 simultaneous channels, yes. |
| 18:13.28 | Qwell[] | You may need to redefine what you consider to be a "line" |
| 18:13.28 | Qwell[] | You're probably thinking "telephone number", which shouldn't be related to "lines" at all. A T1 can have 1 number, or it can have 500 |
| 18:13.29 | hedge77 | chefrs: yes a T1 would give you multiple "lines" |
| 18:13.30 | Jubei | codefreeze: any good guides to learn making dialplans? I'm also interested in learning good practices |
| 18:13.31 | bkruse | Qwell[]: good point |
| 18:13.32 | Qwell[] | in fact, multiple T1s can share a number |
| 18:13.39 | bkruse | DID != zap line |
| 18:13.44 | chefrs | We have 14 lines into the Asterisk box. Three are reserved for extensions for our executive folk. Thing is if they're on the line they get a busy signal, I want it to still come in and goto their voicemail. |
| 18:13.49 | DarKnesS_WolF | Jubei: the book |
| 18:13.53 | DarKnesS_WolF | ~book |
| 18:13.57 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:14.03 | hedge77 | does "the book" cover ael though? |
| 18:14.09 | bkruse | codefreeze: teach me to be awesome! |
| 18:14.10 | Jubei | does it? |
| 18:14.17 | Qwell[] | chefrs: sure, a T1 would definitely help with that. Just tell the provider that all numbers should come in on all lines of the T1, and then route the call based on incoming DID |
| 18:14.21 | bkruse | it does, ael and ael2 i believe |
| 18:14.46 | Mercestes | chefrs, Actually...you can have the telco huntgroup those existing lines as well instead of assigning them on a 1-1 ratio |
| 18:14.53 | chefrs | Qwell[]: So I can have 3/23 T1 lines saying "I'm from 303-555-1234" when 555-1234 is a certain extension and even if it is busy, it'll come in |
| 18:14.53 | codefreeze | bkruse: I cannot teach you, that which you already know/are! |
| 18:14.56 | hedge77 | I did my dialplan in AEL with just the ael.txt from the source so I had no idea |
| 18:15.00 | bkruse | ! |
| 18:15.05 | chefrs | Mercestes: We have one. But DID doesn't get set for some reason. |
| 18:15.06 | Mercestes | so you could have 12 numbers hunting across 12 lines independently |
| 18:15.07 | Qwell[] | chefrs: yep |
| 18:15.19 | chefrs | Qwell[]: Perfect. I appreciate the help. |
| 18:15.20 | bkruse | codefreeze: hows all your ael2 stuff coming? good? |
| 18:15.49 | codefreeze | I seem to be keeping up with up. Any probs, bkruse? |
| 18:16.25 | bkruse | no, not at all, i just see all those commits and im like, "yay!" |
| 18:16.26 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 18:16.39 | codefreeze | hedge77: any probs with AEL? Q's? Now's the time. |
| 18:17.13 | hedge77 | codefreeze: no I think i've got all that done now. The only problem I have left is with dynamic queue members since I'm trying to avoid agentcallbacklogin |
| 18:17.30 | vooduhal | I have a PRI that I shows this "Status: Provisioned, Down, Active" for the status and I keep getting this if I try to make a call on it: Short write: 0/5 (Broken pipe) and zap show status shows OK. |
| 18:17.31 | bkruse | hedge77: good idea. |
| 18:17.33 | vooduhal | Any ideas? |
| 18:17.46 | bkruse | vooduhal: google? |
| 18:17.49 | vooduhal | Tried. |
| 18:17.51 | codefreeze | hedge77: have you seen the examples in the doc dir, I think, on queues? |
| 18:17.54 | bkruse | gotcha |
| 18:17.58 | vooduhal | Didn't couldn't find anything of use. |
| 18:18.03 | bkruse | i wasnt trying to be mean, i was just wondering, im not sure :[ |
| 18:18.20 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 18:18.23 | hedge77 | codefreeze: yeah the actual queues are set up but i've hit a bug: http://bugs.digium.com/view.php?id=9421 |
| 18:18.38 | hedge77 | logging in/out call distribution etc works fine though |
| 18:18.59 | codefreeze | OK, good to hear; I'm not the expert on queues, tho! |
| 18:19.04 | kuku5 | I have a question regarding the manager. Action: ParkedCalls has a paramaneter called ActionID, do I just leave ActionID in there or do I make one up or whats the deal..? |
| 18:19.12 | hedge77 | the docs weren't terribly clear so I might write something better once I'm done with this |
| 18:19.44 | codefreeze | hedge77: we can use all the help we can get! |
| 18:19.47 | vooduhal | Is there an alternative to wanpipemon with the sangoma cards to monitor a digium card? |
| 18:21.06 | bkruse | vooduhal: you have zttool, zttest |
| 18:22.53 | hedge77 | hmm why wasn't zttest installed when I built zaptel? |
| 18:23.07 | hedge77 | do you have to do it seperately? |
| 18:23.59 | JerJer | you need the dev package of ..... |
| 18:24.11 | JerJer | newt |
| 18:24.49 | hedge77 | no i mean it built it (left in /usr/src/zaptel/) but didn't install it to somewhere in the path |
| 18:26.47 | tuan_modulis | anyone here know when will asterisknow become stable (or estimation thereof)? |
| 18:28.43 | *** join/#asterisk Winkie (n=urmom@87-194-8-125.bethere.co.uk) |
| 18:29.02 | Qwell[] | tuan_modulis: soon |
| 18:29.43 | hedge77 | does the gui for that jack it up so your hand edits are screwed up like freepbx? |
| 18:29.46 | Qwell[] | nope |
| 18:30.09 | bkruse | :| |
| 18:31.20 | Jubei | how can I check to see if my asterisk supports ael2 ? |
| 18:33.59 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 18:34.09 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 18:34.32 | [[blah]asfd | I am trying to dial from one server to another using sip and am getting a username error... can anyone take a look at my paste and see where I am not understanding this process? |
| 18:34.34 | [[blah]asfd | http://pastebin.ca/421224 |
| 18:34.51 | hedge77 | are you going between asterisk servers? |
| 18:34.54 | [[blah]asfd | yes |
| 18:35.35 | bkruse | ya |
| 18:35.44 | bkruse | your not authenticating correctly |
| 18:35.53 | bkruse | [usernamehere] in sip.conf |
| 18:36.04 | bkruse | try changing [local] to [asterisk] and then registering |
| 18:36.35 | codefreeze | Jubei: no checking necc. 1.2 == AEL v.1; 1.4 == AEL v.2, trunk == AEL v.2 |
| 18:36.37 | [[blah]asfd | I want it to be the other way around... local is what I wanted the user name to be... I dont know where it is getting asterisk from. |
| 18:36.51 | [[blah]asfd | I am going to have local and ld and internal users |
| 18:37.01 | Jubei | codefreeze: thanks |
| 18:37.10 | [[blah]asfd | connecting from multiple server to this single box. |
| 18:37.35 | hedge77 | codefreeze: acutally i did think of something. This is how I'm handling holidays http://pastebin.ca/421226 . is there a better way? |
| 18:38.27 | *** part/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
| 18:38.40 | Jubei | codefreeze: voip-info describes of a way to get your 1.2 * patched for AEL2 |
| 18:38.46 | Jubei | codefreeze: i might try it |
| 18:40.43 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 18:40.45 | codefreeze | hedge77: looks OK to me. I do something similar in my home dialplan, where I say Merry Christmas! instead of just plain "hello" when the system answers the phone on a holiday. |
| 18:43.29 | codefreeze | I also say "good morning" "good afternoon", etc. based on the time of day. Your structure looks OK, tho. Better than OK, actually, I see very few gotos. |
| 18:44.19 | hedge77 | codefreeze: yeah it works it just seems kind of over-complicated to me now (it was the first thing I did when I started this). I didn't now if there was a more elegant way to do it. I tried different contexts with gotoIfTime but I couldn't get it working right so bleah. |
| 18:44.51 | hedge77 | codefreeze: yeah i'm trying to avoid jumping around as much as possible. The dialplan I'm trying to upgrade has tons of them and nobody can figure out exactly what it does. We're trying to avoid that this time. |
| 18:44.54 | codefreeze | Jubei: OK; let me know if you have probs! |
| 18:45.19 | codefreeze | hedge77: good strategy! |
| 18:46.29 | hedge77 | totally random unrelated question: If I have a T1 in one city with Bellsouth, and analog lines/#'s in another city with Sprint, can I take those Sprint numbers and have them point to the BS T1? |
| 18:47.18 | Corydon-w | You cannot without making them BS numbers |
| 18:47.26 | Corydon-w | It's called "porting" the numbers |
| 18:48.05 | bkruse | which i think the company has to due, by law, but are not obeying it. At least in the case of going from bell -> voip |
| 18:48.08 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 18:48.30 | blebleble | what is the best way to pass lets say 48 channels to another asterisk server dundi? |
| 18:48.33 | Corydon-w | bkruse: they will, but sometimes it takes longer than you're willing to wait |
| 18:48.52 | Qwell[] | blebleble: depends on what you mean by pass |
| 18:49.07 | bkruse | Corydon-w: oh, thats how they get around it....interesting |
| 18:49.15 | blebleble | Qwell: basically need the lines registered on pbx1 but then passed / able to be used on pbx2 |
| 18:49.29 | Qwell[] | blebleble: an iax2 trunk would work well for that |
| 18:49.54 | Corydon-w | bkruse: yeah, it's a case of "we'll get around to scheduling it sometime in the next 90 days" |
| 18:50.08 | bkruse | Corydon-w: eww......thats just mean :[ |
| 18:50.19 | *** join/#asterisk sysreq (n=sysreq@193.203-ppp.3menatwork.com) |
| 18:50.24 | Corydon-w | bkruse: welcome to BellSouth |
| 18:50.26 | hedge77 | Corydon-w: right porting is what I meant, I just wasn't sure if it applied for going telco->telco or not. |
| 18:50.28 | blebleble | Qwell[]: thanks i'll do some reading |
| 18:50.33 | bkruse | haha |
| 18:50.55 | hedge77 | oh that sucks :( |
| 18:51.10 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 18:51.24 | Corydon-w | "Welcome to BellSouth. If we do it, what makes you think it would be convenient for you?" |
| 18:51.40 | hedge77 | would they be faster/more agreeable if it were a *new* T1? |
| 18:52.01 | Corydon-w | Nope, the issue is the numbers themselves |
| 18:52.25 | hedge77 | suck suck suck |
| 18:52.26 | Corydon-w | If you understand a bit about how calls are routed, you'll understand why it created a problem |
| 18:52.53 | hedge77 | I know a little but I'm no expert, but I can kind of see why it is a problem |
| 18:53.01 | Corydon-w | It's really all about billing, not about technical issues |
| 18:53.14 | bkruse | omg, i remember when matt brooks got a t1 from bellsouth way back in the day and how much of a pain that was |
| 18:53.46 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:53.53 | Corydon-w | Technically, porting the number takes 5 minutes |
| 18:54.06 | Corydon-w | if that |
| 18:54.15 | hedge77 | right but you have to keep track of who owes who money |
| 18:54.47 | Corydon-w | Well, it's more about routing the call through the cloud to the right provider |
| 18:55.13 | bkruse | Corydon-w: i would agree. |
| 18:55.14 | Corydon-w | In the old days, numbers were allocated in blocks of 10,000 to each provider |
| 18:55.37 | Corydon-w | so you could use the prefix (NPA/NXX) to discover how to route the call |
| 18:56.00 | Corydon-w | Now you need to use the whole number to route the call correctly |
| 18:56.34 | hedge77 | ooooohh |
| 18:57.09 | Corydon-w | So to port a number, two things need to happen: 1) the old provider needs to stop claiming it knows how to route that number from the cloud, and 2) the new provider needs to start claiming it |
| 18:58.22 | hedge77 | and the old provider is in no hurry to release it? |
| 18:58.54 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:59.03 | jm|laptop | hullo |
| 18:59.43 | bkruse | hedge77: exactly |
| 18:59.47 | jm|laptop | I have problems with my Asterisk 1.2.16 |
| 18:59.47 | bkruse | thats how they get "around" the law |
| 19:00.14 | jm|laptop | Description Alarms IRQ bpviol CRC4 |
| 19:00.14 | jm|laptop | Wildcard X100P Board 1 OK 0 0 0 |
| 19:00.27 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 19:00.27 | jm|laptop | should that IRQ really be 0 ?! |
| 19:00.46 | jm|laptop | the problem I have is very noisy line from the card; people can hear me but I can't hear them :| |
| 19:01.13 | jm|laptop | <PROTECTED> |
| 19:03.01 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 19:04.26 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 19:05.15 | rene- | hello, does Zap/107-1 means the first call placed or recieved on zapata channel 107 and should that number increase should i get a new call on that channel? |
| 19:05.34 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:10.24 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 19:10.31 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 19:11.33 | hedge77 | voice T1's should be waaay cheaper than internet ones, right? |
| 19:12.09 | hedge77 | since you're not paying for anything but the actual line? |
| 19:12.31 | Corydon-w | Other way around |
| 19:12.32 | [[blah]asfd | bkruse: hey, changing user name to asterisk did not resolve that issue you were helping me with. |
| 19:12.51 | [[blah]asfd | I am still getting the same error. |
| 19:12.57 | [[blah]asfd | it has to be something else. |
| 19:13.04 | Corydon-w | Data T1s are subject to less tariffs, so they tend to be cheaper than voice T1s |
| 19:13.04 | hedge77 | Corydon-w: really? more than $500/mo? |
| 19:13.53 | Corydon-w | hedge77: yes, much more |
| 19:13.53 | hedge77 | we;lp |
| 19:17.10 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 19:19.06 | *** join/#asterisk darylvoip (n=darylvoi@c-71-225-231-66.hsd1.pa.comcast.net) |
| 19:19.30 | darylvoip | Hey all....anyone know how to make Asterisk (1.2.x or 1.4.x) write CDRs to TWO tables rather than just one? |
| 19:19.57 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:19.58 | mvanbaak | why you want that ? |
| 19:19.59 | Corydon-w | darylvoip: run two CDR backends |
| 19:20.41 | darylvoip | Hmmm....I've got 2 Postgres databases.....I guess maybe I could do one with ODBC. |
| 19:20.41 | mvanbaak | ah well |
| 19:20.54 | mvanbaak | too drunk to think correctly |
| 19:20.56 | mvanbaak | going to bed |
| 19:20.57 | Corydon-w | Otherwise, not possible in the current release |
| 19:21.13 | darylvoip | Well, goo to know (at least I can stop searching for how to do it ;) ) |
| 19:21.18 | mvanbaak | darylvoip: you can use replication for it |
| 19:21.37 | darylvoip | Not really. We're talking about a heavily loaded database. |
| 19:22.10 | darylvoip | I was thinking of using triggers on the the primary database, but it's busting at the seams. This is a stop-gap measure until the DB architecture is properly repaired. |
| 19:22.28 | darylvoip | Right now, users are killing the DB running reports on the CDRs. |
| 19:22.55 | darylvoip | And some brain surgeon decided to mix all administartive functions with the production database. |
| 19:23.06 | mvanbaak | ah |
| 19:23.27 | mvanbaak | I have never seen a database workload where replication was impossible because of the load |
| 19:24.05 | darylvoip | When you have users pulling inefficient reports that sends your load average into the 15's on a regular basis, you'll know you've found one. |
| 19:24.10 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 19:24.22 | darylvoip | I've got to works the administrative reporting queries off piece by piece. |
| 19:24.44 | darylvoip | In the future, replilcation will work well I'm sure (once only production functionality is on that database) |
| 19:24.51 | mvanbaak | indeed |
| 19:24.58 | mvanbaak | bad design there ;) |
| 19:25.03 | darylvoip | But....got to do the messy hack for now ;) |
| 19:25.15 | darylvoip | Yeah...but that's why I have a job. To clean up messy hacks. |
| 19:25.24 | mvanbaak | hhmm |
| 19:25.44 | mvanbaak | I remember there are some sql relay programs that can connect to multiple backends |
| 19:25.58 | darylvoip | Hmmm....that might be interesting. |
| 19:26.00 | ctaloi | Hey guys - Question - does a way exist to have use the MixMonitor recording function of a dialplan in a queue? i.e. - I want to record queue calls, but not in seperate audio streams |
| 19:26.11 | mvanbaak | sqlbalancer or something |
| 19:26.13 | darylvoip | Like a load balancing front-end type of thing? I'll have to search around. That might be easier. |
| 19:26.21 | mvanbaak | yeah |
| 19:26.28 | mvanbaak | install it on the asterisk box |
| 19:26.34 | darylvoip | Nice. Thanks. Gotta go talk to my friend google. |
| 19:26.35 | mvanbaak | and let it connect to both sql boxen |
| 19:26.40 | darylvoip | I like the sound of that. |
| 19:26.50 | mvanbaak | good luck |
| 19:26.57 | mvanbaak | hope you get stuff sorted out |
| 19:27.25 | mvanbaak | ok, I'm off as well |
| 19:28.08 | *** join/#asterisk RA25 (n=RA25@76.24.24.118) |
| 19:28.12 | darylvoip | Thanks. I appreciate the pointer. |
| 19:35.58 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:36.45 | *** join/#asterisk doolph (n=doolph@200.105.35.219) |
| 19:36.48 | doolph | hi |
| 19:37.00 | doolph | what's the best qos for asterisk SIP? |
| 19:39.03 | *** join/#asterisk VaRMiT (n=varmit@CPE-76-177-75-124.natcky.res.rr.com) |
| 19:39.17 | VaRMiT | hey all |
| 19:39.35 | VaRMiT | i have been fooling with asteriskNow all day... |
| 19:39.52 | l0rdr0ck | http://i167.photobucket.com/albums/u121/l0rdr0ck/asterisk.png |
| 19:39.59 | VaRMiT | and im trying to figure out how to setup where the IVR answears the calls and routes them correct |
| 19:40.00 | l0rdr0ck | whoops sorry |
| 19:40.11 | l0rdr0ck | asterisk on aTV |
| 19:40.26 | tuan_modulis | is there a way to specifically load a particular config file and not the whole thing? |
| 19:40.36 | tuan_modulis | like sip reload <somefile> |
| 19:40.49 | hedge77 | what are you trying to reload? |
| 19:40.55 | tuan_modulis | voicemail.conf |
| 19:41.34 | VaRMiT | i have just about tore my hair out...ive tried trixbox...but dont like it....more bugs than a flea circus |
| 19:41.47 | dlynes_laptop | tuan_modulis: unload app_voicemail.so, and then load app_voicemail.so |
| 19:41.58 | tuan_modulis | thx |
| 19:41.59 | VaRMiT | anyone help me.... |
| 19:42.00 | dlynes_laptop | tuan_modulis: or alternatively, reload app_voicemail.so |
| 19:42.03 | hedge77 | VaRMiT: I tried and gave up on the guis and did it by hand, it actually wasn't too bad. |
| 19:42.29 | tuan_modulis | dlynes_laptop: excellent |
| 19:42.32 | VaRMiT | i have the GXP 2000 phones talking to one another..no problem.... |
| 19:42.47 | VaRMiT | i just cant get the IVR to do the answearing |
| 19:42.52 | dlynes_laptop | tuan_modulis: just bear in mind that not all modules like to be reloaded; some require an unload and a load |
| 19:43.03 | dlynes_laptop | tuan_modulis: a prime example is chan_zap.so |
| 19:43.56 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 19:44.37 | dlynes_laptop | tuan_modulis: just bear in mind that not all modules like to be reloaded; some require an unload and a load |
| 19:44.39 | dlynes_laptop | tuan_modulis: a prime example is chan_zap.so |
| 19:44.48 | dlynes_laptop | Just in case you didn't get that before your disconnection |
| 19:44.54 | VaRMiT | is there a howto for the gui somewhere? |
| 19:44.57 | tuan_modulis | thx |
| 19:45.13 | dlynes_laptop | VaRMiT: try /join #asterisk-gui and /join #asterisk-now |
| 19:45.21 | dlynes_laptop | VaRMiT: the peeps in those channels should be able to help you out |
| 19:45.29 | VaRMiT | thanks dlynes_laptop |
| 19:45.35 | *** part/#asterisk VaRMiT (n=varmit@CPE-76-177-75-124.natcky.res.rr.com) |
| 19:47.23 | doolph | anyone know any working qos firewall for asterisk with SIP (sipphones) |
| 19:47.50 | hedge77 | dlynes_laptop: that actually helped with a bug i'm hitting. Do you know if app_queue is one of the ones that hates being reloaded? |
| 19:47.52 | jm|laptop | AX-100P ... any good? |
| 19:50.38 | darylvoip | doolph: Yout aren't getting answers because your questions don't make a whol lot of sense. What are you trying to QoS? Asterisk isn't any different than any other SIP device, if that's what you're tlaking about. |
| 19:50.39 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 19:50.40 | dlynes_laptop | hedge77: no idea..never used that app |
| 19:50.42 | doolph | hello |
| 19:51.17 | darylvoip | And QoS is rarely an issue on the client side. Since your QoS won't be honored upstream, and you're more likely to have bottlenecks coming downstream, QoS won't likely help too much. |
| 19:51.18 | dlynes_laptop | doolph: all qos firewalls work with asterisk |
| 19:51.45 | doolph | like what |
| 19:51.51 | dlynes_laptop | like everything |
| 19:52.02 | dlynes_laptop | pick one |
| 19:52.18 | hedge77 | dlynes_laptop: well it didn't barf when I tried it and it didn't kick the caller out of the queue so it works for me :) |
| 19:52.35 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 19:52.51 | dlynes_laptop | or try /join #tell_me_what_firewalls_are_available_for_my_os(insert_os_name_here) |
| 19:53.08 | darylvoip | doolph.....what are you trying to accomplish and where? I can tell you what firewalls over $15,000 will work for you, but if you're trying to do this at home I'm guessing there is more appropriate equipment. We have noidea what you're trying to do and on what scale. |
| 19:53.09 | *** join/#asterisk Gambold (n=root@80-46-38-124.static.dsl.as9105.com) |
| 19:54.07 | jm|laptop | hmm |
| 19:54.15 | jm|laptop | maybe I should get a Sipuar |
| 19:54.17 | jm|laptop | maybe I should get a Sipura* |
| 20:00.32 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:02.48 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 20:10.56 | *** join/#asterisk Xteven (n=deepstar@void.singularity.be) |
| 20:11.14 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:11.54 | hedge77 | is there a way to specify the module loading order during startup or a reload? |
| 20:12.06 | JerJer | modules.conf |
| 20:12.56 | Xteven | hi there, I've been told we use "digital phones" at work, but I have no idea what that could be. can asterisk work with those ? also, how can I check whether those phones are not just regular analog phones ? (they have a display that shows who's calling you, fwiw) |
| 20:13.45 | hedge77 | JerJer: it will load exactly in the order specified in modules.conf every time? |
| 20:14.09 | JerJer | if you do autoload=no then specify load=[foo].so |
| 20:14.14 | JerJer | it would get loaded top down |
| 20:16.19 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:24.26 | jm|laptop | ztmonitor 1 -v shows the RX audio level at 100% all the time |
| 20:26.12 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:28.01 | *** join/#asterisk Voice2 (n=Voice2@145-27.mc.cite.net) |
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| 20:32.25 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:33.24 | *** join/#asterisk ManxPower (n=manxpowe@217.sub-75-203-213.myvzw.com) |
| 20:34.17 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 20:35.41 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 20:35.42 | b11d | allo |
| 20:36.25 | bkruse | ello b11d :D |
| 20:36.30 | b11d | :) |
| 20:36.46 | b11d | how are ya mate? |
| 20:41.23 | hedge77 | what is the best codec for a high latency connection? |
| 20:41.27 | bkruse | good man, and yourself? |
| 20:41.48 | bkruse | high latency? i would guess something like g723.1 or g729 since its low bandwith |
| 20:41.59 | bkruse | but a codec cannot makeup for latency(delay) unfortunatly :[ |
| 20:42.11 | Mercestes | hedge77, Anything you want as long as that latency doesn't fluctuate |
| 20:42.40 | hedge77 | Mercestes: that's the problem is that it is a wonky sattelite connection and it goes up and down |
| 20:42.50 | hedge77 | bandwidth is fine though |
| 20:42.53 | Mercestes | hedge77, Try telepathy |
| 20:42.57 | ManxPower | the codec has nothing to do with latency, it has to do with bandwidth |
| 20:43.18 | bkruse | ManxPower: exactly |
| 20:43.25 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 20:43.41 | bkruse | my guess is if you have a long latency, your probably worried about bandwith consumption also. |
| 20:44.08 | ManxPower | bkruse: not so much on satellite |
| 20:44.21 | ManxPower | I get 150Kbps download speeds, but 900ms+ latency |
| 20:44.24 | bkruse | ManxPower: ya, i was just thinking that also..... |
| 20:44.31 | ManxPower | and 200ms - 3000ms jitter |
| 20:44.34 | bkruse | well, i hope your not running a call center from it |
| 20:44.35 | bkruse | :[ |
| 20:44.40 | bkruse | :| |
| 20:44.55 | hedge77 | hahaha no just one guy |
| 20:45.01 | bkruse | :P |
| 20:45.33 | bulle | hedge77: is packet traffic cheaper then talk, over satellite ? |
| 20:45.40 | hedge77 | sometimes it's fine, but sometimes it's just terrible. Oh wait this might be microwave now not sattelite i'll have to ask. |
| 20:45.57 | hedge77 | uhh it's cheaper when the caller is on the other side of the planet |
| 20:51.43 | ber_ | anyone here have experience querying lidb |
| 20:52.24 | ber_ | wondering if there is a way thru asterisk to do it or some external app which can be invoked on linux to hit it via ss7 or sip |
| 20:52.39 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:52.50 | ManxPower | satellite .vs. microwave is pretty important |
| 20:53.11 | Mercestes | hedge77: Your best bet...go ahead and run up your latency to about oh, 1000ms or so, maybe even 1500ms, and set an insane jitterbuffer. |
| 20:54.06 | ManxPower | hedge77: and teach the users to say "Over". |
| 20:54.23 | ManxPower | microwave would be fairly low latency |
| 20:55.45 | Mercestes | ManxPower, LOL. "over" I love it. |
| 20:56.09 | hedge77 | ManxPower: I don't think the vendors would quite get it |
| 20:56.18 | hedge77 | ManxPower: but it's an amusing idea. |
| 20:57.06 | ManxPower | hedge77: you may have no option if you are really on comsumer grade satellite (i.e. not VSAT) |
| 20:57.35 | ManxPower | with VSAT you have dedicated 64K channels and can do QoS at least to eliminate jitter. |
| 20:59.08 | hedge77 | no it's commercial if anything, I asked the guy to clarify which connection he had. I'm starting to think it's microwave and the latency is in all the hops to get to the US |
| 21:07.47 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 21:11.44 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 21:13.13 | hedge77 | hey i remembered how I really wanted to phrase my question: are any codecs better about handling *packet loss*? |
| 21:13.24 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 21:15.55 | Mercestes | hedge77, with packet loss, the less compression, the better. But....packet loss isn't giong to pass good voice. |
| 21:16.40 | Mercestes | hedge77, Your best bet is to use a VPN with a high qos priority and use it to give you packet error correction and then try to pass voice through the VPN in ulaw. |
| 21:16.56 | Mercestes | hedge77, My disclaimer is....that won't work. :D But it is a best bet. |
| 21:17.17 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:17.44 | hedge77 | Mercestes: ah well i think that is exactly what we are already doing (maybe using ilbc instead of ulaw). |
| 21:18.02 | Mercestes | hedge77, if ilbc has less compression than ulaw then it'd be better. |
| 21:18.08 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 21:18.16 | Mercestes | hedge77, compression saves your bandwidth, no compression is less susceptible to errors. |
| 21:18.24 | Mercestes | hedge77, but errors are very very bad regardless. |
| 21:18.40 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:19.22 | *** join/#asterisk Laureano (n=Laureano@OL155-33.fibertel.com.ar) |
| 21:25.04 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 21:37.29 | red9012 | how can i customize the directory command playbacks? |
| 21:38.18 | Mercestes | red9012, Edit the source. |
| 21:47.24 | *** join/#asterisk Crudy (n=crudi@ppp-70-247-203-146.dsl.snantx.swbell.net) |
| 21:49.08 | Juggie | ilbc is the worst codec ever. |
| 21:49.09 | Juggie | avoid. |
| 21:49.11 | Juggie | use gsm. |
| 21:51.24 | *** join/#asterisk Vec (n=Vec@dsl-244-208-173.telkomadsl.co.za) |
| 21:52.11 | [TK]D-Fender | Juggie, LIES |
| 21:52.18 | [TK]D-Fender | Juggie, ....LPC10 ;) |
| 21:53.24 | bkruse | [TK]D-Fender: watch out there..... |
| 21:53.25 | bkruse | lol |
| 21:54.16 | [TK]D-Fender | bkruse, hey... I'm going for the 31's ;) |
| 21:54.22 | [TK]D-Fender | #1's* |
| 21:55.14 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 21:56.08 | brodiem | I have a question on CNAM... is there a "central" CNAM DB out there, or do LECs/ISPs just use "whoever" where CNAM may not actually be available but is provisioned at a different DB not being used by that particular LEC/ISP |
| 22:00.11 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:02.15 | Qwell[] | bkruse: ha, nice hack |
| 22:02.45 | Qwell[] | I see exactly what you mean now |
| 22:05.33 | cr4z3d | huh? |
| 22:05.49 | cr4z3d | oh nevermind |
| 22:06.03 | doolph | anyone know a good hardware or anyway to have quality of service in a mixed network with data & Asterisk using SIP, and with softphone. pc/softphone <--> (need qos here) <--> internet <--> asterisk |
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| 22:16.09 | cr4z3d | woah now they're back |
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| 22:17.52 | hedge77 | derp derp der |
| 22:17.52 | Qwell[] | ...and thank you for flying freenode air |
| 22:17.52 | tzanger | heh |
| 22:17.52 | doolph | anyone know a good hardware or anyway to have quality of service in a mixed network with data & Asterisk using SIP, and with softphone. pc/softphone <--> (need qos here) <--> internet <--> asterisk |
| 22:17.52 | brodiem | I have a question on CNAM... is there a "central" CNAM DB out there, or do LECs/ISPs just use "whoever" where CNAM may not actually be available but is provisioned at a different DB not being used by that particular LEC/ISP |
| 22:17.53 | tzanger | brodiem: I think it's done very similarly to TDM clocking |
| 22:17.53 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 22:17.53 | tzanger | doolph: as soon as you leave your own network QoS guarantees go right out the door |
| 22:17.54 | cr4z3d | how do i enable ztdummy? i got the zaptel software installed but can't figure out where exactly ztdummy is |
| 22:17.54 | tzanger | brodiem: as in if your LEC isn't authoratative, it contacts the authoratative database owner |
| 22:17.54 | Qwell[] | cr4z3d: modprobe it |
| 22:17.54 | tzanger | russellb: what's your email addy? I just found you on freenode |
| 22:17.55 | tzanger | er linkedin |
| 22:17.55 | mmartinn | Hey folks -- is "app_dial.c: Unable to forward voice" not okay? Google results indicate that unallocated numbers may trigger this warning -- my outbound calls may frequently call those, and I'm wondering if I can ignore the warning? |
| 22:17.55 | brodiem | tzanger ahh makes sense, like DNS then basically.. |
| 22:17.55 | doolph | tzanger what u mean |
| 22:17.55 | cr4z3d | Qwell[], it says it doesn't exist |
| 22:17.56 | [hC] | Anyone have any idea why sometimes calls get dropped when calling somewhere and getting an answering machine, and when it plays the 'beep' to begin recording, asterisk decides to end the call?? |
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| 22:18.07 | bkruse | someone say the name? |
| 22:18.07 | cr4z3d | ? |
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| 22:23.48 | [TK]D-Fender | . |
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| 22:28.04 | Vec | Does anyone know of any asterisk redundent solutions similar to Ranch Networks solution, maybe one implemented in software, eg, I want to be able to run to Asterisk servers for redundancy ? |
| 22:28.25 | Vec | Each asterisk box would have its own E1 line. |
| 22:30.53 | mmartinn | Vec: see http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
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| 22:34.33 | Vec | mmartinn : ta |
| 22:39.48 | cr4z3d | when i try to make install zaptel modeul i get "install: cannot stat *.ima: no such file or directory" what exactly does that mean |
| 22:47.16 | file | cr4z3d: what version? |
| 22:48.52 | cr4z3d | no idea whatever ubuntu has in apt-get install zapte-source |
| 22:48.58 | cr4z3d | but i think i figured it out |
| 22:49.16 | cr4z3d | i used m-a build zaptel and now i'm doing a dpkg on the .deb file and so far no errors |
| 22:50.10 | cr4z3d | woohoo ztdummy is finally running |
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| 22:56.24 | [hC] | Why might my PRI be hanging up the call when hearing a 'beep' from some answering machines? |
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| 23:07.50 | ManxPower | [hC]: busydetect or callprogress being enabled. DO NOT EVER DO THAT ON A PRI |
| 23:10.47 | [hC] | ManxPower: it was actually faxdetect of all things. |
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| 23:13.12 | Mercestes | [hC] Don't ever do that on a PRI either. |
| 23:13.20 | Mercestes | faxdetect==evil |
| 23:13.33 | ManxPower | Just don't do faxdetect on OUTGOING calls. |
| 23:13.44 | ManxPower | we use NVFaxDetect for incoming calls. |
| 23:14.47 | ManxPower | faxdetect also supports a direction rather than yes or no |
| 23:16.46 | [hC] | I use NVFaxDetect now too, I just forgot to disable faxdetect... |
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| 23:17.35 | hrmphh | can someone recommend either cheap sip phones or cheap ata adapters? |
| 23:17.48 | Qwell[] | ManxPower: That README file is severely out of date.. |
| 23:18.12 | Qwell[] | like...it doesn't even include the dual-span card or the tdm2400p |
| 23:18.51 | hrmphh | and am i correct that theres no way to use a skype based phone with say the x-lite softphone+asterisk? |
| 23:18.58 | Qwell[] | hrmphh: yes |
| 23:19.13 | Qwell[] | ~phones |
| 23:19.15 | jbot | [phones] http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
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| 23:20.32 | hrmphh | hmm |
| 23:20.57 | hrmphh | is the linksys pap2t-na a good ata? |
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| 23:22.08 | cr4z3d | what's the difference between a pap2t-na and the spa=2102 |
| 23:22.25 | doolph | nothing |
| 23:22.58 | cr4z3d | really? |
| 23:23.04 | doolph | i think so |
| 23:25.41 | hrmphh | ok in that case |
| 23:25.47 | hrmphh | is the spa2102 good? want to run 2 lines through it |
| 23:25.50 | hrmphh | 2 diff sip connections |
| 23:25.56 | hrmphh | to 2 diff analog phones |
| 23:28.41 | doolph | it is its job |
| 23:31.03 | hrmphh | heard some issues w/it could only handle like one simulataneous high compression codec |
| 23:32.58 | ManxPower | hrmphh: I am not aware of any ATA that does two G729/G723.1 calls at the same time. |
| 23:34.22 | Mercestes | I've heard of one but I can't recall which one it was |
| 23:35.21 | *** join/#asterisk anthm (n=anthm@m810f36d0.tmodns.net) |
| 23:35.21 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 23:45.58 | *** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz) |
| 23:47.23 | flenders | hey anyone tested jaxtr? |
| 23:50.41 | Jubei | guys what is a good way to camp on an extension until it hangs up then ring it? |
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| 23:57.25 | mmartinn | Hey folks -- is "app_dial.c: Unable to forward voice" not okay? Google results indicate that unallocated numbers may trigger this warning -- my outbound calls may frequently call those, and I'm wondering if I can ignore the warning? |
| 23:58.13 | ManxPower | mmartinn: the value of HANGUPCAUSE will tell you what the telco is telling you the reason for the disconnect is. |
| 23:58.34 | ManxPower | mmartinn: Unable to forward voice CAN be a symptom of a problem, but not always. |
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| 23:58.59 | mmartinn | ManxPower: Mmmkay :) |
| 23:59.16 | ManxPower | Dial() sets HANGUPCAUSE |
| 23:59.44 | ManxPower | If you call into Voicemail Main and then hangup while it is playing the messge, you will get that message, for example. |
| 23:59.46 | mmartinn | Yes, all is fine with HANGUPCAUSE and Dial() |
| 23:59.56 | *** part/#asterisk jmacz (n=jmacz@190.24.101.197) |
