00:00.24 | dovid | raidenz: try using the help function in CLI. i am not playing with 1.4 till all issues are fixed and wiki is updated. dont have time to play to run in to issues |
00:00.47 | *** part/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
00:01.38 | raidenz | I did and I saw the help on the function but still doesn't help. |
00:02.10 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
00:02.10 | *** mode/#asterisk [+o mog] by ChanServ |
00:02.23 | dovid | wur kind of phone r u using ? |
00:02.55 | dovid | wut* |
00:08.15 | teknoprep | everything i am readying say do this |
00:08.16 | teknoprep | parkedhints=yes |
00:08.20 | teknoprep | where do i put that ? |
00:08.22 | teknoprep | on the extension ? |
00:09.57 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.161.255) |
00:13.40 | dovid | again what phone r u uisng ? |
00:15.24 | C6Vette | guess its a secret. |
00:16.21 | *** join/#asterisk docelmo (n=vircuser@m015f36d0.tmodns.net) |
00:17.18 | C6Vette | when doing a "show queue whatever"... How can i tell hold long its been since the received a call. It shows a (last was 30 secs ago) but |
00:17.24 | C6Vette | <PROTECTED> |
00:17.37 | C6Vette | when they are not on a call |
00:18.01 | teknoprep | dovid for this extension its a gxp-2000 |
00:19.19 | dovid | ah. didnt play much with those. sorry. they arent worth the time of day. |
00:19.51 | dovid | c6Vette: dont know much with queues. try help command or wiki |
00:20.13 | diablopico | can anyone help me with an insmod command for the te212p ? |
00:20.55 | teknoprep | to patch asterisk.. do i have to fully recompile it ? |
00:21.41 | diablopico | when you make , it should only remake any files that change |
00:22.12 | diablopico | and of course those dependant on the changed files |
00:27.01 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
00:28.49 | *** part/#asterisk cbm11211 (n=Administ@66.28.182.170) |
00:34.26 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:34.36 | aptura | long day. |
00:34.52 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
00:35.13 | kronic | is there a setting required for the agent function to return values? |
00:35.42 | kronic | I have an agent logged in, attempting to return values from the ${AGENT} function yields nothing :S |
00:35.57 | aptura | Do most pbxs use the same numbering sceem be it accessing vm or call forwarding or is that a customer choice? Example one pbx may use *98 to access its voice mail vs another pbx uses *67 |
00:39.16 | justinu|laptop | http://www.nanpa.com/number_resource_info/vsc_definitions.html |
00:41.15 | justinu|laptop | are there any SIP phones that support DND override? |
00:41.43 | dovid | by dnd over ride u mean ? |
00:42.02 | justinu|laptop | i mean if someone sets their phone in DND, I should be able to make it ring anyways |
00:42.14 | aptura | Thanks |
00:42.16 | justinu|laptop | with a proper passcode or some other authentication |
00:42.37 | aptura | for example a familly emergency |
00:42.54 | justinu|laptop | for example: i'm the pbx admin, and i don't care whether you want to be disturbed or not |
00:42.59 | justinu|laptop | s/pbx admin/boss |
00:43.52 | aptura | Unless your in a very important meeting or involved in a presentation. |
00:43.54 | Hymie | well, >I can tell you the uniden UIP/200 won't be doing that any time soon :Þ |
00:44.24 | justinu|laptop | this is yet another reason why the phone should just be a dumb device, ala SCCP |
00:44.34 | justinu|laptop | not some pseudo intelligent network element, like SIP |
00:44.59 | Hymie | just implement asterisk do not disturb, via dialplan logic and database variables |
00:45.11 | Hymie | then that can be over ridden as you choose |
00:45.17 | justinu|laptop | that doesn't prevent someone from placing their SIP phone in DND |
00:45.20 | justinu|laptop | which you can't override |
00:45.33 | Hymie | it does, if you turn that option off on the phone.. what phone is it? |
00:45.44 | Hymie | I can disable that in the uniden UIP200, and that's a crappy phone ;) |
00:46.10 | justinu|laptop | well, basically I like the idea of DND... i just think there should be a way to override it if needed |
00:46.24 | Hymie | I just told you how |
00:46.30 | dovid | justinu|laptop: i believe not because asterisk is told to not let anything get to it |
00:47.00 | justinu|laptop | having to turn off the DND button on the phone isn't my idea of a solution |
00:47.02 | dovid | unless u create a seperate extension that they have to authenticate and from ur dial plan u change thier dnd status but then they will get all other calls. |
00:47.36 | *** part/#asterisk r_evolution (i=r_evolut@208.251.203.246) |
00:48.05 | Hymie | justinu: SIP phones won't ring when the DND button is on.. Asterisk can't do anything about that. You said you wanted a dumb phone, but you don't want to disable the DND button on the phone.... not quite getting you here |
00:48.08 | justinu|laptop | the problem is that the DND status is something the phone can set without having to check with the switch to see if its allowed to set DND status |
00:48.18 | dovid | or create two extensions that go to one sip account and have them dial an exten to put them at unavail and when the gen exten is called it checks a variable id they wana be botherd |
00:48.35 | justinu|laptop | what I meant was that I want a phone which works with a protocol like SCCP |
00:48.38 | Hymie | dovid: that won't work |
00:48.51 | *** join/#asterisk Malawar (n=Malawar@adsl-75-21-232-9.dsl.sgnwmi.sbcglobal.net) |
00:48.54 | justinu|laptop | where when ppl press buttons on the phone, it sends messages to the softswitch, and the softswitch decides how the phone behaves |
00:48.55 | Hymie | dovid: when the phone has DND on, it won't ring.. period |
00:48.58 | dovid | Hymie: y not ? |
00:49.17 | justinu|laptop | yeah, that's a defect in the design of the phone, IMO |
00:49.19 | dovid | Hyme: for his/her case i wouldnt use dnd |
00:49.29 | dovid | I would have this |
00:49.36 | Hymie | justinu: hey, I'm not saying other ideas aren't good, but I'm not interested in that right now, >I have to deal with what I have |
00:49.52 | dovid | i would use a gotoif to see if they want to be botherd |
00:49.54 | justinu|laptop | i understand where you're coming from |
00:49.56 | dovid | for thier reg exten. |
00:50.12 | justinu|laptop | it's not something a SIP phone couldn't do |
00:50.14 | dovid | and have the user dial en exten to set the variable |
00:50.15 | Malawar | I need.. a business friendly VOIP provider |
00:50.16 | justinu|laptop | allow DND override |
00:50.26 | justinu|laptop | if we had an opensource SIP phone, we could easily implement it |
00:50.31 | dovid | Malwar: teliac isnt the cheapest but good |
00:50.48 | dovid | teliax* |
00:51.06 | Malawar | It doesn't have to be the cheapest, just reliable, and not some company that's going to turn around and charge extra/cut me off for using too much call volume (coughbroadvoicecough) |
00:51.27 | justinu|laptop | dovid: i saw it, but you're kinda missing the point |
00:51.44 | dovid | ok. explain |
00:51.47 | justinu|laptop | the problem is that with most sip phones, the user can press DND, and the phone will never ring again |
00:51.51 | justinu|laptop | unless they disable DND |
00:52.09 | justinu|laptop | well, then these friggen morons who press DND by accident, and never notice the display flashing "DO NOT DISTURB" call you up |
00:52.13 | justinu|laptop | and bitch to you their phone never rings |
00:52.26 | justinu|laptop | and that /I/ obviously broke something |
00:53.04 | dovid | lol |
00:53.06 | dovid | i got that too |
00:53.11 | justinu|laptop | you see my point now? |
00:53.12 | dovid | so just disable dnd on the phone |
00:53.19 | dovid | and make em dial the exten |
00:53.52 | justinu|laptop | again, i don't see that as a solution |
00:54.08 | justinu|laptop | because then I have to train them what exten to dial, and the damn phone has a DND button... why should I disable it? |
00:55.14 | dovid | and u cant disable it on the phone its self ? |
00:55.29 | dovid | and if its an idiot issue then ur SOL people goto learn |
00:56.03 | justinu|laptop | i wonder if SIP 3PCC might allow it |
00:56.14 | Malawar | hmm |
00:56.22 | Malawar | what I really need, is an incoming-only line |
00:59.43 | aptura | dovid, reliability should be the most important feature. |
01:00.26 | aptura | a customer or client should not have to give there pbx a second thought when making that call. |
01:01.26 | dovid | aptura: what point are u talkin about ? it is reliable to an extent. i trust it |
01:02.25 | aptura | I am talking about voip carriers |
01:02.46 | dovid | ah ok |
01:02.56 | *** join/#asterisk cekc (n=cekc@adsl-71-131-128-183.dsl.sntc01.pacbell.net) |
01:03.11 | kronic | what's the best method for obtaining an agents name? |
01:03.12 | dovid | apturs: thats y i have failover routes. if route A is out then use Route B |
01:03.22 | kronic | since the agent func is non-existent |
01:03.30 | dovid | voip just isnt as relaible as pots and i dont think it ever will in the near future |
01:03.35 | aptura | I would rather charge my customers a little more because my reputation is counting on the carriers quality. |
01:03.38 | justinu|laptop | this is interesting: http://www.ietf.org/rfc/rfc3880.txt |
01:03.41 | dovid | same here |
01:03.41 | justinu|laptop | wonder if anything implements that |
01:04.28 | dovid | those are too long for me to read. can u sum it upo ? |
01:04.29 | dovid | up* |
01:04.30 | justinu|laptop | heh, someone posted on asterisk-biz about it |
01:04.31 | aptura | but |
01:04.36 | justinu|laptop | but no one replied |
01:04.56 | aptura | voip is a good substitute should a local telco go down...like in new orelans. |
01:05.07 | Malawar | yarr, all these other providers seem to want to stuff their hardware down my throat as well as a service plan :( |
01:05.43 | aptura | Which providers? |
01:05.43 | Malawar | i'm just looking through them atm |
01:05.45 | Malawar | several so far |
01:05.49 | dovid | i use myphonecompany for inbound and voipjet and teliax for out |
01:06.27 | Malawar | all I need is inbound, but I need to be able to handle multiple calls concurrently |
01:06.49 | aptura | dovid, how many Minutes |
01:06.50 | aptura | <PROTECTED> |
01:08.13 | aptura | looking at this call forward bussy query dont know what it means. ; query |
01:08.14 | aptura | exten => *45*,1,Playback(call-fwd-on-busy) |
01:08.14 | aptura | exten => *45*,2,Set(temp=${DB(CFB/${CALLERIDNUM})}) ; Get CFB key |
01:08.39 | aptura | Is it quering if its set or not set? |
01:09.28 | dovid | yes |
01:09.39 | dovid | apturs: it sets the status to the vairable temp |
01:10.13 | dovid | aptura: myphonecompany gives unlimited inbound. however if u hog multiple channels all the time they may get pissy |
01:10.29 | aptura | of course |
01:10.30 | aptura | :) |
01:10.43 | dovid | its $5.00 for inbound. I once was able to get 9 concurent channels. (it was the most cell phones we had in the office) |
01:11.04 | aptura | like a netmeeting? |
01:11.11 | aptura | what were you trying to do |
01:11.20 | dovid | i did one client that needed 4 concurent so they said to just sign up for 2 lines so it costs $10.00 a month and they can use 4 concurent with no sweat |
01:11.33 | dovid | i was just testing to see how many channels they would give me |
01:11.38 | *** join/#asterisk dir (n=dir@58.69.13.29) |
01:11.38 | aptura | 5 dollars unlimited right? |
01:11.46 | dovid | yes. just inbound |
01:11.55 | aptura | Thats pretty good. |
01:11.56 | Malawar | I don't see that plan on their site :/ |
01:12.03 | dovid | than again if u hog it they wont be happy like the avg. provider |
01:12.09 | dovid | for some reason its not on the sire |
01:12.10 | aptura | Whats the reliability of the voice like |
01:12.13 | dovid | site* |
01:12.21 | dovid | its called a mydeviceplan |
01:13.28 | dovid | u have to email cc@myphonecompany.com |
01:13.28 | justinu|laptop | ~seen [TK]D-Fender |
01:13.36 | jbot | [tk]d-fender is currently on #asterisk (4h 23m 45s). Has said a total of 50 messages. Is idling for 3h 2m 49s, last said: 'ok, BBIAB'. |
01:13.36 | aptura | okay what is the quality of the call like |
01:13.36 | dovid | i have been with them for a year and a half and they were down once for 20 min |
01:13.36 | dovid | otherwise no issues with them |
01:13.36 | dovid | they are a brnach of exchange telecom |
01:13.36 | aptura | and Quality of the call? as in clear or garbled? |
01:13.45 | aptura | or jitter? |
01:13.45 | dovid | real clear |
01:13.52 | aptura | Where is there noc at that does the termination? |
01:14.00 | dovid | i believe in NYC |
01:14.10 | aptura | voipjet is there also |
01:14.16 | aptura | And you are? |
01:14.26 | aptura | where are you that is |
01:14.40 | dovid | just do a who is on thier IP's and u can see where they are |
01:14.46 | dovid | where am i ? in NJ and Israel |
01:14.49 | aptura | I know |
01:15.01 | dovid | u can ask them |
01:15.09 | dovid | i deal with some one there by the name of sam litt |
01:15.12 | aptura | when you make the calls you get the incoming calls you are in nj right? |
01:15.33 | dovid | no. it hits my dedicated server on the west coast |
01:15.56 | dovid | which uses teliax to call my cell phone being that I am in Israel now |
01:16.15 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.191.179) |
01:16.18 | dovid | thier outbound rates arent the best for US48 |
01:16.27 | aptura | I am just trying to picture the distance from where you are to myphonecompany noc is. |
01:16.39 | aptura | how far the rtp traffic has to flow. |
01:16.40 | dovid | ill to a tracert |
01:16.41 | dovid | one sec |
01:17.33 | aptura | not really interested in ping statistics |
01:18.15 | aptura | People on the west coast..well some of them company about call quality comming from vonage which is east coast. I suspect with multiple router hops and traffic that jitter would be a issue. |
01:19.25 | dovid | here u go |
01:19.26 | dovid | http://pastebin.ca/192567 |
01:19.38 | dovid | has never been for me |
01:20.20 | dovid | i would suggest u try a few providers and do the "can u hear me now" test |
01:20.29 | dovid | goto for a smoke |
01:20.30 | dovid | brb |
01:21.13 | aptura | I have worked with a number over two years |
01:23.24 | dovid | back |
01:23.29 | dovid | a number of ? |
01:23.48 | Malawar | anyone know of any other inbound-only services? Just shot off an email to myphonecompany |
01:23.49 | Malawar | :P |
01:23.57 | aptura | voip providers |
01:24.24 | dovid | Malwar: teliax is good but they charge per minute |
01:24.25 | quid246 | not many inbound-only providers. |
01:24.30 | aptura | sixtel has improved and is working for me. |
01:24.53 | aptura | ohh inbound only well sixtel does both. |
01:25.08 | dovid | u dont have to use em for both ways |
01:25.11 | Malawar | the provider doesn't have to be inbound only, just looking for an inbound-only plan :P |
01:25.14 | dovid | <PROTECTED> |
01:25.33 | dovid | MPC is 2 way just set up asterisk to only use em for inbound |
01:25.38 | quid246 | Axvoice has unlimited in, AFAIK... $10 for 2 DIDs. |
01:26.00 | quid246 | They even have 212's last time I looked. |
01:26.21 | dovid | wow. i like 212 |
01:26.26 | dovid | axvoice.com ? |
01:26.29 | quid246 | yep |
01:27.03 | *** join/#asterisk Yarrick40k (n=jmoser@static-71-98-31-11.mdsnwi.dsl-w.verizon.net) |
01:27.14 | quid246 | My 212 actually is phonetic... took me an hour or two seraching through their list to find it htough |
01:27.27 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
01:27.32 | dovid | all i see on axvoice is 18.99 a month. none cheaper |
01:27.36 | quid246 | BYOD |
01:27.39 | quid246 | Bring Your Own Device |
01:28.08 | quid246 | Pay As You go... onle $9 now |
01:28.11 | dovid | didnt see byod. looking again |
01:28.17 | dovid | okies |
01:28.39 | dovid | how many channels do they give u ? |
01:28.40 | Malawar | I'm also trying to avoid places like BroadVoice that have scary Terms of Service |
01:28.45 | dovid | lol |
01:28.50 | dovid | i would never use em again |
01:28.56 | DonX | Anyone use ngt? |
01:28.58 | dovid | they can keep ur number if u port in |
01:29.09 | dovid | call quality sux with them |
01:29.10 | quid246 | haha nice |
01:29.11 | dovid | ngt ? |
01:29.32 | DonX | yah, www.ngt.com |
01:29.32 | Malawar | at first glance Broadvoice was very nice looking.. but their ToS states they can charge $100/day if you're using the service in a way they arbitrarily deem unfit :/ |
01:29.44 | DonX | they have onvoip.net |
01:30.09 | quid246 | Malawar: Adn don't forget the clause for "in house legal fees". |
01:30.14 | Malawar | quid246, exactly |
01:30.34 | Malawar | scary :/ |
01:30.38 | Jason99 | Malawar: What area are you look for numbers in? |
01:31.46 | Malawar | not really picky, but something within the state of NY would be nice :P |
01:31.56 | dovid | u can also try voxbone.com |
01:32.25 | DonX | Has anyone ever used Broadworks? |
01:32.55 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
01:33.05 | *** join/#asterisk b4ka (n=jh@71-226-114-200.fibertel.com.ar) |
01:33.10 | Jason99 | Malawar: just checking, we're in Canada only. |
01:33.28 | b4ka | hey, im a bit of a problem, trying to make an asterisk talk to a strata dk424 |
01:33.36 | b4ka | it works from time to time |
01:33.49 | b4ka | but mostly not, complaining about no dchan |
01:33.54 | b4ka | anyone has a clue? |
01:34.23 | b4ka | its a t1 crossover cable to a sangoma A102 |
01:36.17 | justinu|laptop | b4ka: one side needs to use internal clock, otherside needs to be clocked from the loop |
01:38.48 | b4ka | justinu|laptop it is |
01:39.01 | b4ka | the telco is in master on the sterisk |
01:39.09 | b4ka | and i server as master to the pbx |
01:39.17 | b4ka | serve* |
01:39.55 | b4ka | and its weird, now it is working flawlessly, if i disconnect the cable and plug it back, it all goes to hell |
01:42.47 | aptura | justinu|laptop ever seen compatibility between a nortel mcs 5200 media server and asterisk? I cannot use a company as my termination service because there may be timming issues. the mcs is a 2 million dollar class IV switch. |
01:43.41 | justinu|laptop | shouldn't be a problem |
01:43.56 | justinu|laptop | not T1 timing at least |
01:44.21 | *** join/#asterisk cekc (n=cekc@adsl-71-131-128-183.dsl.sntc01.pacbell.net) |
01:45.04 | aptura | well there engineer did try in the past and the voice exchange between two parties would drop off every 15 min. |
01:45.09 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
01:45.13 | justinu|laptop | * connects to the DMS all the time |
01:45.24 | justinu|laptop | big brother to their mcs :) |
01:45.56 | justinu|laptop | aptura: talking about pri link? |
01:46.24 | justinu|laptop | asterisk periodically restarts the B channels, that might piss off some switches |
01:46.28 | justinu|laptop | you can disable it |
01:46.28 | aptura | yes |
01:46.45 | aptura | Where would it be disabled? |
01:46.52 | justinu|laptop | zapata.conf i think |
01:47.06 | aptura | okay |
01:47.48 | aptura | Is the restarting not a option or the code needs to be changed? |
01:48.02 | justinu|laptop | its an option, iirc |
01:48.07 | aptura | I see |
01:59.48 | kronic | what's the best method for obtaining an agents name (${AGENT} does is non-existent)? |
02:13.19 | *** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it) |
02:13.34 | *** join/#asterisk nvrs (n=RUR@Quebec-HSE-ppp3613717.sympatico.ca) |
02:15.01 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
02:18.13 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
02:18.48 | dovid | seen ~me |
02:18.58 | dovid | seen ~jbot |
02:20.06 | b4ka | oh now, i found the way to make the t1 between the strata and asterisk to work. i have to plug the cable SLOW LOL |
02:20.35 | Qwell | b4ka: get a new cable |
02:20.45 | b4ka | tried 3 |
02:20.58 | Qwell | pebcac? |
02:21.15 | ShadowHntr | stupid question |
02:21.15 | Qwell | (I haven't decided on what the c's stand for yet) |
02:21.21 | b4ka | could it be that the signal is too strong? |
02:21.36 | ShadowHntr | if i set up asterisk in a strictly SIP setup with SIP phones, do i need a digium card? |
02:21.40 | b4ka | and it makes ground before stablishing the connection |
02:21.48 | Qwell | ShadowHntr: no |
02:22.06 | ShadowHntr | cool. cause i read about the digium cards, and i knew *what* they were, just didn't know why i'd need one. |
02:22.10 | ShadowHntr | Qwell: thanks. :) |
02:23.08 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
02:23.53 | dovid | ShadowHntr: see: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
02:24.09 | ShadowHntr | yeah i've been reading that wiki |
02:24.17 | ShadowHntr | just been trying to make heads or tails of some of it. |
02:24.53 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
02:25.12 | *** join/#asterisk angom_h (n=Angel@www.aysco.com.mx) |
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02:31.02 | dovid | ShadowHntr: the wiki is the king. and remember google is ur friend |
02:31.28 | ShadowHntr | yeah i've kinda discovered that it may be easier to avoid the cisco phones. too much hassle to configure for a first-time voip user. |
02:31.42 | ShadowHntr | although i am a highly proficient linux admin and a network guru |
02:31.50 | ShadowHntr | just learning as i go |
02:31.55 | ShadowHntr | appreciate the pointers. :) |
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02:44.23 | wulfy814 | if I have just changed the echo cancellor and recompiled zaptel |
02:44.32 | wulfy814 | do I need to reboot for the change to take effect? |
02:44.49 | dovid | no |
02:44.55 | dovid | just stop and start asterisk |
02:45.21 | dovid | well if u just recompiled then it isnt running so just start asterisk |
02:46.06 | wulfy814 | let me rephrase... |
02:46.21 | wulfy814 | I just edited zconfig.h and did a make, make install |
02:46.25 | wulfy814 | asterisk was running at the time |
02:46.46 | wulfy814 | I changed to Mark3 |
02:47.18 | *** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
02:48.24 | *** join/#asterisk [Outcast] (n=bill@222-154-75-119.jetstream.xtra.co.nz) |
02:48.54 | heison | ~seen coppice |
02:49.07 | jbot | coppice <n=chatzill@229.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 9h 25m 18s ago, saying: 'I get weird reports from anything after 1.2.9 about NULL frames being thrown at rxfax, which cause a crash when they are freed'. |
02:52.07 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
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02:55.56 | dovid | stop asterisk. make and restart |
02:57.17 | b4ka | anyone knows how to send a restart to all the B channels on a pri line from asterisk? |
02:57.36 | Yarrick40k | anyone here using a Cisco 7940 with SIP firmware? |
03:02.36 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
03:03.32 | aptura | b4ka whats the issues? |
03:09.34 | sudhir492 | Will a kind soul please help me configure Polycom 301 phones for paging |
03:09.44 | sudhir492 | I mean Polycom 501 |
03:11.11 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
03:12.25 | [TK]D-Fender | sudhir492 : PM me root and I'll get you set up. |
03:13.53 | aptura | [TK]D-Fender I would also like to know |
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03:17.23 | AJaymn | Anyone know of a good reference for using the sound files. as in how to have Allison speak #s like "Your Account Balance is" "1 hundred |
03:17.30 | AJaymn | " and 24 dollars" |
03:17.37 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
03:19.50 | b4ka | aptura: i cant friggin make asterisk to talk to the strata through the pri properly |
03:20.04 | aptura | strata? |
03:20.11 | b4ka | toshiba strat |
03:20.13 | b4ka | pbx |
03:20.19 | [TK]D-Fender | *shudder* FUGLY phones.... |
03:20.22 | b4ka | some dude documented it working on the wiki |
03:20.33 | b4ka | but i cant fucking make it work |
03:20.38 | aptura | never even heard of them |
03:20.51 | b4ka | they are very common |
03:20.55 | b4ka | small pbxs |
03:21.45 | b4ka | this is so retarded, why the fuck connecting the cable slowly makes it work! |
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03:26.22 | wmandra | Yarrick40k: I'm using 7940's and 7960's with sip firmware |
03:26.41 | xpato | has anyone worked with the sipura spa3000? |
03:27.12 | Yarrick40k | wmandra: I think I got it worked out. Have to do firmware updates in stages apparently |
03:28.04 | wmandra | yarrick: just follow the instructions on the wiki..... they'll get you through it |
03:28.49 | Yarrick40k | wmandra: But winging it is so much more fun! |
03:28.50 | Yarrick40k | ;) |
03:28.57 | wmandra | lol |
03:29.26 | Yarrick40k | there a way to reboot a 7940 without jackin the power cord? |
03:29.33 | Qwell | Yarrick40k: sip or sccp? |
03:29.38 | Yarrick40k | sip |
03:29.46 | Qwell | * + 6 + settings |
03:30.22 | Yarrick40k | thanks |
03:31.02 | [TK]D-Fender | Qwell++ |
03:31.09 | Qwell | ? |
03:31.25 | [TK]D-Fender | Qwell : Jbot rating thing.... |
03:31.30 | Qwell | right |
03:31.46 | [TK]D-Fender | ~Qwell++ |
03:32.02 | [TK]D-Fender | jbot : Qwell |
03:32.06 | jbot | qwell is probably a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
03:32.16 | [TK]D-Fender | lol |
03:32.26 | [TK]D-Fender | I forget how to do it now! |
03:32.35 | [TK]D-Fender | <- dismal failure |
03:33.39 | linuxmigration | jbot: Qwell++ |
03:40.26 | kronic | I'm receiving this error with Monitor(): ast_writefile: Unable to open file |
03:40.47 | kronic | permissions are correct, does asterisk create directories if their non-existent |
03:53.15 | *** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za) |
03:57.35 | X-Rob_ | kronic, that's correct. it does not create directories. |
04:01.08 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
04:01.28 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
04:01.28 | kronic | it does, its actually due to the fact it doesn't handle relative filepaths |
04:04.13 | aptura | Here we go, Poe overhead paging system. |
04:04.16 | aptura | http://www.cyberdata.net/products/voip/voip-loudspeakeramp.html |
04:04.43 | [TK]D-Fender | Cute, but "ick |
04:05.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
04:05.19 | *** mode/#asterisk [+o mog] by ChanServ |
04:05.40 | aptura | ick? |
04:06.45 | aptura | Tk, what about this product you dont like? |
04:07.34 | *** part/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
04:09.00 | [TK]D-Fender | aptura : Low power guaranteed, and the device is tied to one tech. Better off with an FXS based unit + ATA |
04:09.23 | [TK]D-Fender | aptura : much more flexible, not to mention cheaper |
04:09.34 | aptura | ohh because it is poe not enoug power to run the speakers |
04:10.13 | [TK]D-Fender | aptura : And again we've been talking about phones who we know are capable of paging as it is. |
04:10.27 | [TK]D-Fender | aptura : Not that is won't have enough power, jsut that it'll be weak. |
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04:11.48 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
04:13.59 | aptura | So what do you know works well |
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04:15.09 | [TK]D-Fender | aptura : the Viking paging units have a good rep |
04:15.23 | aptura | Yea just saw them. |
04:15.45 | aptura | And then there is the one way page across all phones such as a emergency. |
04:16.38 | aptura | Anyway I am going to call it a night wife is home and dinner is cooking :) |
04:16.50 | aptura | Thanks for the advice on the vikings. |
04:17.00 | linuxmigration | any distro that is particularly well suited for asterisk or most people just go by personal preference? |
04:17.51 | Yarrick40k | centos/trixbox is the preferred choice |
04:18.02 | Yarrick40k | though I've seen it work with redhat, suse, etc |
04:18.09 | Yarrick40k | depends on if you want to spend time rolling your own |
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04:18.52 | linuxmigration | say, for manual install in a production env |
04:19.12 | JT | Yarrick40k: ???? |
04:19.21 | JT | Yarrick40k: trixbox is NOT the preffered choice here |
04:19.46 | JT | there is no advantage to using centos for asterisk either |
04:19.46 | Yarrick40k | JT: sorry, didn't mean to step on toes |
04:19.54 | JT | whatever distro you are most comfortable with |
04:20.26 | Yarrick40k | JT: so what /is/ the preferred choice then? |
04:20.55 | JT | Yarrick40k: standard asterisk, on the distro you are most comfortable/confident in personally |
04:21.11 | JT | no point learning a new distro just to do asterisk |
04:21.39 | linuxmigration | how is http://powerontech.com/freepbx-on-debian.htm ? |
04:22.06 | linuxmigration | anyone use that tutorial? |
04:22.16 | JT | linuxmigration: i'd recommend against freepbx in a production environment |
04:22.53 | [TK]D-Fender | linuxmigration : Whichever distro you are most comfortable administering |
04:23.22 | Yarrick40k | JT: why the recommendation against freepbx in production? |
04:23.25 | linuxmigration | i'm not actually setting up a production server. that was just an example |
04:23.48 | [TK]D-Fender | linuxmigration : Though I'd pick from Slackware, Debian, CentOS, RHEL if I were you. Just more stable sources and regular devel stuff available |
04:23.49 | JT | Yarrick40k: it's a toy which puts a gui on top of asterisk |
04:24.29 | Yarrick40k | JT: Right, but the underlying pieces (ie asterisk) is no different. |
04:24.57 | JT | Yarrick40k: except that freepbx obsfucates access to it |
04:25.03 | JT | and you run versions of freepbx, not asterisk |
04:25.07 | JT | so it's a fork |
04:25.19 | [TK]D-Fender | Yarrick40k : When you have a GUI controlling everything for you, who cares? You are basically turning a massively flexible PBX into a TOASTER. |
04:25.30 | linuxmigration | heh |
04:25.33 | Yarrick40k | true, don't get me wrong. |
04:25.50 | [TK]D-Fender | Yarrick40k You'd be almost better to buy a proprietary PBX. |
04:25.50 | Yarrick40k | I'd rather do all the management from the command line anyways /shrug |
04:26.22 | [TK]D-Fender | Which is why some managers still go with far less functional proprietary PBX's. |
04:26.32 | Yarrick40k | true |
04:26.38 | JT | at least there's a company to back it up |
04:26.42 | Yarrick40k | right |
04:26.50 | file | [TK]D-Fender: ! ! ! |
04:26.57 | JT | the proprietary pbxes, that is |
04:28.23 | [TK]D-Fender | file : I don't want relationship! |
04:28.40 | [TK]D-Fender | file : I just want ... |
04:28.47 | file | ! ! ! |
04:28.54 | [TK]D-Fender | z0mg |
04:29.29 | file | [TK]D-Fender: wazzzzzup |
04:33.02 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
04:34.33 | [TK]D-Fender | file : Enjoying a little time off. Contract last night took me from 5pm to 1 AM. |
04:34.54 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
04:34.55 | [TK]D-Fender | So trying to relax. Not sure if I'll have another job lined up for tomorrow night or not. |
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04:35.34 | file | eep, evil |
04:38.05 | Juggie | file! |
04:38.09 | file | moo |
04:38.35 | Juggie | you hurt that poor irc virgin today :) (nic) |
04:38.51 | file | ha |
04:39.15 | Juggie | i just got home from the sens/leafs game |
04:39.18 | Juggie | was a blast, did you watch it? |
04:39.25 | file | no |
04:40.03 | Juggie | you missed out, leafs won 6-0 :) |
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04:50.06 | bungalow | anyone here familiar with building modules for asterisk 1.4? |
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05:30.16 | *** join/#asterisk Leoman (n=leo@70.241.166.244) |
05:31.09 | Leoman | Good evening. |
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05:35.30 | Leoman | I am looking for a way to allow a person at x place to be able to dial a number on y's local pstn. Could I do that with asterisk? I'm thinking that by placing to pbx's, one at x place and the other at y's place I can pick up a phone at x and call someone thru y's phone. x an y connect via the internet. Am I correct in my thinking? |
05:37.34 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
05:38.23 | Leoman | juanjoc estas? |
05:38.34 | juanjoc | Si |
05:38.51 | Leoman | que tal como estas? tengo una pregunta de principiante |
05:39.00 | Leoman | es mas bien teoria |
05:39.26 | Leoman | hablas ingles? ya lo pregunte en ingles pero nadie respondio.... |
05:39.33 | juanjoc | Si, hablo inglés |
05:39.42 | Leoman | mira esto |
05:39.45 | Leoman | I am looking for a way to allow a person at x place to be able to dial a number on y's local pstn. Could I do that with asterisk? I'm thinking that by placing to pbx's, one at x place and the other at y's place I can pick up a phone at x and call someone thru y's phone. x an y connect via the internet. Am I correct in my thinking? |
05:40.07 | juanjoc | What kind of connection to the PSTN do you have? |
05:40.13 | juanjoc | A Zaptel card? SIP? |
05:40.39 | Leoman | none yet...I am in the planning stages. I only have a regular phone with dsl connections |
05:41.27 | juanjoc | Then you need to get a Zaptel card to connect you computer to the PSTN |
05:41.37 | juanjoc | Or get a SIP connection to the PSTN |
05:42.14 | Leoman | say I get a zaptel card. With that in mind, could I do what I am thinking about doing? |
05:42.42 | JT | yes |
05:43.00 | Leoman | something like: phone -> pbx -> internet -> pbx w zaptel card -> pstn? |
05:43.31 | JT | sure |
05:43.48 | Leoman | how do I connect the 2 pbx's? |
05:44.14 | Leoman | over the internet that is . |
05:44.21 | JT | sip or iax protocol |
05:44.34 | juanjoc | You connect them via a VoIP protocol, either SIP, IAX or H.323 |
05:44.50 | juanjoc | IAX might be the best option if you're connecting two Asterisk's |
05:45.15 | Leoman | will I need to buy services from someone to do connect the two pbx's via IAX? |
05:46.40 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
05:46.45 | juanjoc | If you just want to connect your PBX's you just need an Internet connection |
05:46.55 | Leoman | awesome. |
05:47.40 | juanjoc | If you want to access the telephone network you'd have to buy the service from a telephony provider |
05:48.35 | juanjoc | That might come in handy, for example, if you want to have a phone number in another country assigned to your PBX |
05:49.26 | Leoman | exactly what I'm looking for. To be able to pick up the phone at x location and get a dial tone at y location , which by the way is in another country. |
05:49.48 | juanjoc | You'll normally get those services via SIP |
05:49.58 | juanjoc | There are lots of providers for this |
05:50.19 | Leoman | how much are they per month? set fee or variable fee? |
05:50.36 | juanjoc | It depends, but they are generally cheap |
05:51.01 | juanjoc | The price per minute is normally cheaper than the one you'd get for a normal PSTN connection |
05:51.16 | juanjoc | What country are you interested in? |
05:51.44 | Leoman | what if I just to a point to point connection between pbxs with IAX? Is the quality of the call lower? |
05:51.46 | Leoman | Panama |
05:52.17 | JT | iax and sip are just voip protocols |
05:52.22 | JT | codecs run over them |
05:52.27 | juanjoc | Not necessarily, the quality will be maily determined by your Internet connection and the codec you use |
05:52.30 | JT | and they are the main determinate of call quality |
05:52.47 | Leoman | I don't why the need for the SIP provider. |
05:53.02 | juanjoc | To connect to the telephone network |
05:53.05 | Leoman | sorry...I meant to say, I don't see why the need for the sip provider |
05:53.20 | JT | Leoman: you don't if you are connecting to the pstn directly |
05:53.30 | juanjoc | Unless you want to connect to the PSTN on Panama via a PBX there |
05:54.01 | Leoman | yes, that's what I'm looking to do. place another pbx there and connect it to the one here. |
05:54.35 | JT | yes it will do it |
05:54.41 | JT | asterisk will do a lot of things |
05:54.54 | juanjoc | Then you only have to connect both instances of Asterisk with an IAX link |
05:55.05 | juanjoc | Or a SIP link |
05:55.06 | orlock | hmm |
05:55.10 | juanjoc | Any of them will do |
05:55.15 | orlock | our provider is giving us a few trial sip trunks |
05:55.20 | orlock | multiple DID's on one account |
05:55.21 | orlock | :) |
05:55.42 | juanjoc | Leoman: I'm going to sleep, it's very late here, bye |
05:55.45 | Leoman | excelent |
05:55.56 | Leoman | thanks for your help juanjoc |
05:55.59 | juanjoc | np |
05:56.32 | orlock | JT: ever heard of "TDM encryption"? |
05:56.37 | Leoman | JT, the connection between instances of asterisk is documented in asterisk? |
05:57.35 | JT | Leoman: www.voip-info.org |
05:57.46 | JT | there's stuff on some approaches to take there |
05:58.06 | JT | orlock: no, but i've thought of it |
05:58.14 | Leoman | ok, thanks. I'll get on reading there. Thanks for the answers. |
05:58.42 | orlock | JT: http://www.accessproviders.com.au/index.cfm?content=104 |
06:00.48 | JT | dodgy, 500000 is hardly any encryption keys |
06:01.12 | orlock | JT: yeah, plus it doesnt actually have any real details |
06:02.38 | JT | of course |
06:02.50 | JT | it's proprietary! you wouldn't want hackers to know would you |
06:03.37 | orlock | JT: heh, "xor" i bet |
06:04.16 | JT | maybe they're talking about spread spectrum spreading codes, they're XORed with the data :PO |
06:04.19 | JT | -O |
06:04.39 | JT | although it says TDM, so i doubt there's spread spectrum also |
06:04.47 | orlock | yeah, its 5.4Ghz |
06:04.49 | JT | tdm is poo for range and SNR |
06:05.31 | orlock | Server not found |
06:05.31 | orlock | <PROTECTED> |
06:05.34 | orlock | <PROTECTED> |
06:05.36 | orlock | <PROTECTED> |
06:05.38 | orlock | <PROTECTED> |
06:05.40 | orlock | argh |
06:05.42 | orlock | fuck |
06:05.44 | orlock | goddman |
06:05.46 | orlock | sorry! |
06:05.58 | jmls | who's godman ? |
06:06.48 | JT | who said godman? |
06:07.36 | jmls | sorry, orlock asked for goddman. I know, he meant goddamn. it's early. give me a break |
06:07.45 | jmls | <PROTECTED> |
06:07.48 | JT | heh |
06:08.04 | jmls | bloody kids |
06:08.23 | jmls | got me up at 6am. |
06:08.30 | jmls | grumpy daddy. |
06:08.38 | jmls | back to * development ... |
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06:14.31 | Leoman | JT, is this the right card needed ? -> "WildCard X100P card", single FXO Port, PCI interface For Asterisk IP-PBX |
06:15.02 | JT | you cannot buy any real X100Ps anymore |
06:15.11 | JT | the ones around don't work very well |
06:15.34 | JT | you'd be better off getting a TDM400P or an external hardware FXO device like a Sipura |
06:15.41 | Leoman | that one says it's 100% compatible with the wildcard X100P |
06:15.59 | JT | bullshit |
06:16.03 | Leoman | that's at least, what they say in the website I found that one. |
06:16.03 | JT | don't believe it |
06:16.06 | JT | they're all fake |
06:16.13 | JT | or made with rejected chips |
06:16.21 | JT | they may work |
06:16.28 | JT | they may work for a little bit |
06:16.30 | Leoman | it's only 16 bucks.,... |
06:16.32 | JT | or may not work at all |
06:16.40 | JT | well if you want to try, go for it |
06:16.48 | JT | just expect to buy something better at some stage |
06:17.20 | Leoman | so what does one do to get a real X100P? |
06:19.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
06:19.13 | JT | find someone who has one, and buy it... they're no longer produced |
06:19.20 | JT | and it's hard to verify if it's real |
06:19.28 | Leoman | I see...ouch |
06:19.49 | wmandra | i bought mine from digium years ago |
06:20.22 | Leoman | do the fake ones produce echo problems? |
06:22.26 | wmandra | i'd actually consider selling mine.... i'd rather replace one of my unused FXS modules with an FXO module |
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06:42.32 | Joel1978 | so then what are they selling at http://www.x100p.com ? |
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06:44.50 | JT | i have no idea, Joel1978 |
06:45.05 | Joel1978 | they say they sell authentic original x100p cards |
06:45.36 | JT | probably also fakes |
06:45.45 | JT | digium were the ones who sold real X100Ps |
06:45.53 | Joel1978 | for fakes their site is quite impressive |
06:45.54 | Joel1978 | lol |
06:47.09 | JT | yeah |
06:47.33 | JT | it's possible they're not, but i have no idea how that would be.. |
06:48.23 | Joel1978 | probably farmed out production to another company as i don't think digium wanted to play on the low end of the hardware spectrum |
06:48.57 | JT | err |
06:49.07 | JT | the Intel chips are no longer in production |
06:49.18 | JT | so they'd have to be seconds or copies |
06:49.27 | Joel1978 | probably better than before =) |
06:49.40 | JT | ask about the X100P when the regulars are awake and active in here |
06:49.47 | JT | some of which work for digium |
06:49.59 | JT | if they farmed it off, it's a secret to everyone |
06:51.09 | wmandra | is there a way to have * match 1001 and +1001 to the same extension without creating 2 seperate entries and without using "." ?? |
06:59.03 | Bobcat991966 | My vote is stay away from the X100P. I had one from Digium last year and it sucked. |
06:59.30 | Joel1978 | i've heard that as well |
06:59.55 | JT | it was always just a rebadged winmodem with a heatsink on the chipset IC |
07:00.09 | Bobcat991966 | I have the TDM400 now and I am not all that impress with it, but at least it works most of the time. |
07:00.28 | Joel1978 | why not impressed? |
07:00.37 | Bobcat991966 | Echo |
07:00.49 | Bobcat991966 | not really bad but its still does echo |
07:01.09 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
07:01.15 | Joel1978 | and you've changed settings applicable to resolving that with no improvement? |
07:01.24 | ShadowHntr | question - how resource hungry is asterisk in general |
07:01.30 | ShadowHntr | pure sip |
07:01.38 | Bobcat991966 | Oh yes, I have spent hundreds of hours trying to tune it |
07:01.58 | Bobcat991966 | It depends on what codec you use |
07:02.22 | JT | depends if you transcode |
07:02.51 | Bobcat991966 | I have heard good things about Rino cards,,, they have hardware echo cancelation |
07:03.02 | Bobcat991966 | Rhino that is |
07:03.22 | JT | if it's that imortant an application, you'd be using digital connections, too |
07:04.29 | ShadowHntr | i'm looking at a <1000MHz server |
07:04.53 | ShadowHntr | well maybe |
07:04.54 | JT | ShadowHntr: how many connections, what codec? |
07:05.04 | ShadowHntr | one line out |
07:05.06 | Bobcat991966 | How man simultanious cards and what codec are you going to use. will you be using the conf bridge |
07:05.10 | ShadowHntr | not sure on the codec yet |
07:05.23 | aiksa[LV] | any idea why outgoing digital call on chan_misdn would fail? |
07:05.28 | JT | one line out, how many voip connections? |
07:05.39 | ShadowHntr | let's see, probably one or two at most. |
07:05.40 | aiksa[LV] | without even getting to remote equipment? |
07:06.13 | JT | ShadowHntr: that's hardly any load, you'll be fine |
07:06.18 | Bobcat991966 | you will be fine with a 1G processor and 512 megs ram |
07:06.19 | ShadowHntr | cool |
07:06.53 | ShadowHntr | i was looking for a VoIP provider that let me use my own equipment. then i found broadvoice. :) |
07:07.13 | Bobcat991966 | na go with Telasip |
07:07.16 | ShadowHntr | cause i could use asterisk and put an IP phone in each room. |
07:07.58 | aiksa[LV] | http://pastebin.ca/192776 |
07:08.12 | Bobcat991966 | Broadvoice has a bad rep with their so call unlimited plan |
07:08.20 | aiksa[LV] | anyone? |
07:08.39 | ShadowHntr | Bobcat991966: who? |
07:09.01 | ShadowHntr | who are they |
07:09.02 | ShadowHntr | :/ |
07:09.07 | Bobcat991966 | Telasip |
07:09.13 | wmandra | i have broadvoice now, and i'm shopping for a new provider |
07:09.17 | JT | aiksa[LV]: hmm, you haven't even said anything about what the setup is |
07:09.28 | wmandra | broadvoice has been a huge headache |
07:09.50 | Bobcat991966 | Good BYOD provider that lets you set you own outbound callerid and never gives me a problem with the anount of calls I make |
07:10.13 | Joel1978 | ha |
07:10.17 | Joel1978 | really? |
07:10.33 | Joel1978 | that's the single thing i've been looking for is setting cid |
07:10.38 | diablopico | hey ,, does anyone know how to make Zaptel compile using bigmem ? |
07:11.08 | Bobcat991966 | They wont let you set name, but they will let you set any CLID number you like |
07:11.08 | ShadowHntr | i'm not impressed with their website. |
07:11.31 | aiksa[LV] | JT: i am trying to do txfax through misdn channel |
07:11.34 | Joel1978 | number is good enough for me |
07:11.51 | aiksa[LV] | JT: with call spool file |
07:12.14 | Bobcat991966 | Telasip is not much on their website but thay are excelent at customer service |
07:12.16 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
07:12.39 | ShadowHntr | any others that are asterisk friendly? |
07:13.04 | aiksa[LV] | JT: chan_misdn documentation (in source folder) said that now options can supplied at the end of channel string |
07:13.16 | aiksa[LV] | where /h means outgoing digital call |
07:13.33 | aiksa[LV] | asterisk 1.2.9.1 |
07:13.40 | aiksa[LV] | lastest chan_misdn |
07:26.46 | JT | aiksa[LV]: you haven't even said what card you're using |
07:29.54 | *** join/#asterisk elduffy (n=elduffy@pd9569322.dip0.t-ipconnect.de) |
07:32.02 | ShadowHntr | Bobcat991966: telasip is BYOD? |
07:33.42 | aiksa[LV] | JT, oh - very very sorry, beronet 4 port isdn bri |
07:34.04 | aiksa[LV] | ports configured as terminal equipment |
07:34.41 | JT | oh hrm |
07:34.51 | JT | i have no experience with that card of chan_misdn myself |
07:34.59 | JT | is that card compatible with bristuff? |
07:37.46 | ShadowHntr | hmmm |
07:37.49 | ShadowHntr | anyone use voicepulse? |
07:39.03 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
07:43.28 | aiksa[LV] | JT: not sure, but i remember trying to get it work with bristuff with 2.4 kernel |
07:43.56 | aiksa[LV] | yet after migrating to 2.6, we chose mISDN |
07:45.08 | aiksa[LV] | asteriskguru.com says "- BN4S0 : HFC based 4 port BRI card, known to work with chan_misdn" |
07:53.29 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
07:54.12 | diablopico | can anyone help me with a zaptel compile problem ? |
07:55.19 | diablopico | hmmmm |
07:58.20 | *** join/#asterisk Cyt (n=danielcy@85.75.176.202) |
08:12.24 | aiksa[LV] | what is chan_fax? |
08:12.56 | *** part/#asterisk Joel1978 (n=Joel1978@12-226-85-195.client.mchsi.com) |
08:13.27 | *** join/#asterisk scottmcl (n=scott@host217-40-20-25.in-addr.btopenworld.com) |
08:15.01 | aiksa[LV] | and how does it work anyway? |
08:15.09 | scottmcl | Hi i am having a big problem getting a call to transfer from a remote interface, can anyone help? |
08:17.16 | scottmcl | Is anyone actually on the channel at the moment? |
08:17.41 | aiksa[LV] | yup |
08:18.01 | aiksa[LV] | what are you trying to achieve |
08:18.01 | aiksa[LV] | ? |
08:18.17 | scottmcl | is that is just aiksa and scottmcl.... :o( |
08:19.51 | scottmcl | aiksa - are you looking for help on asterisk or able to help with techy problems? |
08:19.56 | aiksa[LV] | no, some just dont look into channel untill they get notified |
08:20.13 | aiksa[LV] | scottmcl: i can solve some of them, and i am still looking for help on some of them |
08:20.30 | aiksa[LV] | like chan_misdn with outgoing data call |
08:20.38 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
08:21.37 | scottmcl | aiksa: ok cool, let me know when you are free to have a chat about my problem .....i have hunted high and low for a solution but can't find one |
08:21.39 | aiksa[LV] | scottmcl: now tell me what are you trying to achieve: call transfer from extenranl line? |
08:25.02 | scottmcl | aiksa: if someone external calls into a phone from a web interface the user has i am trying to transfer the call else where. |
08:25.30 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
08:25.45 | scottmcl | aiksa: i have looked at calling the transfer command on the phone via *2 by injecting DTMF tones but this does not seam to work. |
08:27.19 | aiksa[LV] | you have that enabled in features.conf? |
08:27.52 | aiksa[LV] | does the transfer work on other phones (internal)? |
08:28.17 | *** join/#asterisk oQPa (i=name@237.Red-83-44-33.dynamicIP.rima-tde.net) |
08:28.46 | scottmcl | yep |
08:29.35 | aiksa[LV] | so only softphones from web is affected? |
08:34.02 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
08:35.46 | scottmcl | i hae not tried softphones.......the web interface is an application i have written that lets you speeddial people |
08:36.35 | aiksa[LV] | oh, its using asterisk manager interface? |
08:36.41 | aiksa[LV] | or spool? |
08:36.56 | scottmcl | yep....asterisk manager interface... |
08:37.10 | aiksa[LV] | you are using orginate? |
08:37.34 | scottmcl | i am using ELMEG 290 and i have even tried calling the commands on the phone but there is no techy instructions for the phone |
08:37.55 | scottmcl | yep using orginate |
08:38.13 | aiksa[LV] | have not worked with originate command, but i would suspect that it has to do with the 'tT' command option for Dial application command |
08:38.58 | scottmcl | when it is a new call....but if you already have a call in place how do you get that call to go intro an attended transfer? |
08:39.15 | scottmcl | without hitting the key on the phone |
08:40.22 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
08:40.28 | aiksa[LV] | oh you want to perform redirect from the same interface on the web |
08:40.33 | aiksa[LV] | ? |
08:40.37 | scottmcl | yep |
08:40.38 | RoyK | morning |
08:40.45 | aiksa[LV] | RoyK: morning |
08:40.46 | scottmcl | morning |
08:41.31 | aiksa[LV] | scottmcl: and right now you are just trying to perfomr another caqll with *2 ? |
08:42.57 | scottmcl | aiksa: there is very little help on how to do it so i have guest that if i called via Asterisk Manager Interface playDTMF *2 on the active call line ....then the number this would start a transfer |
08:43.12 | aiksa[LV] | it appears that AMI has a function called Redirect |
08:43.13 | scottmcl | currently it just plays the tones to the caller |
08:43.38 | aiksa[LV] | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect |
08:43.55 | aiksa[LV] | I beleive you should be able to achieve functionality you desire with that |
08:44.36 | aiksa[LV] | i dont think playing dtmf on the line will work, AMI is not exactly replacement of keypad |
08:45.18 | scottmcl | aiska: yep...is redirect an attended transfer....the wiki is not clear? |
08:45.39 | aiksa[LV] | http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer |
08:47.17 | aiksa[LV] | scottmcl: i believe you should implement that behaviour through scriptting AMI |
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09:01.29 | scottmcl | aiksa: i will have a look at the redirect command in a bit, thanks for your help |
09:01.53 | aiksa[LV] | scottmcl: ur welcome |
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09:23.53 | ebi | Hi all. I'm looking for nice voip hardware and probably someone of you can help me :) I'm looking for a wireless (wlan or dect with station doesn't matter) phone that can easily switch between 2 profiles [I like the Siemens SL75 WLAN but didn't find out if it's possible to switch profiles] |
09:26.27 | *** join/#asterisk xnon (n=xnon@200.8.86.187) |
09:33.55 | Cyt | Hi!!! Please, can I adapt this application: 0000,1,dial(${TRUNK}c/9871234321,20,r) to dial out usign a sip line on my PBX? |
09:34.53 | *** join/#asterisk madafaka (i=hts@mobile.dusan.info) |
09:46.04 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
09:46.28 | S^P | Hi I have a sip provider account how I connect to it using asterisk? |
09:46.36 | S^P | register=> ? |
09:50.08 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
09:52.31 | qdk | S^P: sip peer or friend. |
09:53.27 | S^P | friend |
09:54.04 | S^P | my ATA was connected to it. |
09:56.11 | *** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) |
09:57.48 | L|NUX | S^P : use this register => user:pass@host:port/extension |
09:57.48 | L|NUX | :) |
09:57.59 | *** part/#asterisk ebi (n=ebi@ebi.bitflux.ch) |
09:58.45 | S^P | I did and reloaded conf , but no listing in sip show registry :( |
09:59.14 | L|NUX | humm |
09:59.25 | L|NUX | sip reload |
09:59.28 | S^P | ya |
09:59.34 | L|NUX | which version you are using ? |
09:59.41 | L|NUX | and which provider ? |
10:00.04 | S^P | Asterisk 1.2.12.1 |
10:00.08 | S^P | provider RNK |
10:00.12 | scottmcl | aiski: are you online still? |
10:00.13 | L|NUX | hummm |
10:00.35 | L|NUX | well pb your sip.conf |
10:01.50 | *** join/#asterisk dir (n=dir@124.106.223.26) |
10:07.02 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
10:09.56 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
10:14.40 | scottmcl | ....stund you all into silence :o) |
10:20.57 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
10:20.58 | jeremy_g | hi |
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10:35.17 | lintechnokrats | hi all |
10:35.21 | lilalinux | when I have 2 extensions where the first is a prefix of the second, will the seconds be executed ever? |
10:36.55 | x86 | give an example |
10:37.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
10:37.31 | lilalinux | exten => _X.,1,Dial(CAPI/ISDN1/...) |
10:37.45 | lilalinux | exten => 1000, 1, ... |
10:37.59 | S^P | lilalinux: no |
10:38.00 | x86 | those are two different extension patterns |
10:38.18 | lilalinux | which will be executed? |
10:38.46 | lilalinux | when 1000 is dialed |
10:38.56 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-aeec775f9d4813c4) |
10:39.08 | S^P | as far as I knw _X. |
10:40.02 | lilalinux | just tried it: the second is executed |
10:40.37 | S^P | :( |
10:42.28 | lilalinux | but the question is why. |
10:44.04 | S^P | I experienced the opposite of it while asterisk book says it mathes the best match that what you just expereince. |
10:45.29 | *** join/#asterisk tparcina (n=tparcina@36-25.dsl.iskon.hr) |
10:46.15 | lilalinux | can we exclude that both are executed at the same time? |
10:46.18 | tparcina | hi asterisk guru's |
10:46.35 | lilalinux | hi tparcina |
10:46.57 | tparcina | which one of you knows where unatuhenthicated incoming IAX2 call goes? |
10:47.22 | tparcina | if you don't have anything defined in general section? |
10:47.50 | tparcina | i mean, where it goes => in which context |
10:48.35 | tparcina | who answers this question in a minute gets a plus ;)) (ps I know the answer) |
10:49.37 | S^P | check ur guest setting |
10:50.07 | S^P | by default it goes in default context of extension.conf |
10:50.38 | S^P | number depend on what they dialed or s, (I guess) |
10:50.46 | tparcina | yes, and I don't have default context so unatuhenthicated call goes to context that is defined for last user in iax.conf :)) |
10:51.02 | tparcina | soeurity threat? it sure is! |
10:51.09 | S^P | haha |
10:52.22 | tparcina | for example, if i have no default context and i define context that gives posibility to make outgoing calls, someone could make outgoing calls on my account |
10:52.57 | S^P | hmmmmm |
10:55.20 | tparcina | question: in iax.conf i hav edefined users with Ip address. because of that i can't put username/password. and when that user is calling me, he is treated as unauthenticated user. why? |
10:55.36 | tparcina | how to make calls from that user as aunthenticated? |
10:56.10 | tparcina | problem is that in iax.conf i can't define context in general section as I can in sip.conf |
11:02.23 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
11:08.46 | *** join/#asterisk xnon (n=xnon@200.8.86.187) |
11:08.55 | *** join/#asterisk zorball (n=adf@dsl51B686F0.pool.t-online.hu) |
11:08.56 | zorball | hello |
11:10.12 | zorball | I have an passive isdn card /dev/ttyI0 i4l mode |
11:10.16 | zorball | It is work |
11:10.29 | zorball | If somebody call it I can handle |
11:10.49 | zorball | But If i want to forward the call to another isdn card /dev/ttyI3 |
11:10.53 | *** join/#asterisk xnon (n=xnon@200.8.86.187) |
11:10.58 | zorball | it is not ringing |
11:11.03 | zorball | it say called |
11:11.10 | zorball | but doesnt happen anything |
11:11.33 | zorball | The /dev/ttyI3 is connected to an isdn telephone |
11:11.43 | zorball | Can it work ? |
11:13.09 | florz | zorball: You mean an isdn telephone is connected to the card that can be "reached" through /dev/ttyI3 as if it were an NT1? |
11:14.44 | zorball | There is a pbx and there are some telehone on the isdn pbx and I want a gateway between the isdn pbx and the telehone line |
11:14.48 | zorball | with two isdn card |
11:15.58 | zorball | florz: ... yes |
11:16.37 | florz | zorball: Well, that won't work, you need a card that supports NT mode and a driver that does, too. |
11:17.20 | S^P | I register to a sip provider using register=> .... now how i send call on it? |
11:17.57 | zorball | florz: Thanks |
11:18.12 | zorball | florz: And if I put my /dev/ttyI0 as an extension? |
11:18.35 | zorball | I can only forward one line to another extension in same time? |
11:18.46 | zorball | florz: And if I put my /dev/ttyI03 as an extension? |
11:18.54 | florz | zorball: hmm? |
11:19.05 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
11:19.29 | zorball | so /dev/ttyI3 I put the isdn pbx as an extension |
11:19.48 | zorball | and /dev/ttyI0 put into NT |
11:20.23 | florz | zorball: I4L doesn't support NT mode |
11:23.04 | zorball | ok thx |
11:23.28 | zorball | but if i put to /dev/ttyI3 to the isdn pbx as a another telephone |
11:23.57 | zorball | and if the call comeing i forward it to another phone which is on the isdn pbx too |
11:24.21 | Lloydie-t | Can I get some help with a q931.c problem |
11:26.08 | *** join/#asterisk szundi (n=szundi@152.66.243.163) |
11:26.43 | szundi | hi everyone, can somebody help me with a sip nat problem? |
11:26.58 | razu | tzafrir , tzafrir_home : are you here ? |
11:27.20 | tzafrir_home | only one of us |
11:27.50 | *** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92) |
11:28.32 | KriS83 | Hi |
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11:29.08 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
11:29.29 | szundi | i have a snom 300 ip phone, it is on a nat 172.* ip, but it has a virtual public ip address over an encrypted channel. so the server receives the sip session init from 10.* ip address of the phone, but the phone puts it's ip 172* in the sip headers. asterisk replies well to 10.* via sip, but the voicemal voip sound packets go to 172.* |
11:30.26 | szundi | can somebody tell me the proper sip.conf nat=yes|no|... options or anything? it's 1.4.0 beta checked ot 2 weeks ago from trunk |
11:30.46 | scottmcl | define('CMS_VERSION', 'v2.26' ); |
11:30.54 | scottmcl | oops wrong window |
11:30.56 | scottmcl | :) |
11:31.05 | szundi | i'v tried google but nothing found about that, it seems my config is correct (but we see not) |
11:31.06 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
11:32.25 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
11:32.53 | szundi | hallo, can someone help me? |
11:33.01 | szundi | ring ring :) |
11:35.28 | zeppelin_ | hello all !!! :) |
11:35.34 | leopardus1 | is trixbox using asteriskrealtime? |
11:35.57 | RoyK | asteriskonceuponatime |
11:35.58 | zeppelin_ | leopardus1, no ! |
11:38.00 | *** part/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
11:43.05 | zorball | FRITZ! CARD PCI can run in NT mode? |
11:44.21 | zorball | What kind of card know NT mode which cheep? |
11:47.00 | florz | zorball: HFC PCI-A based cards |
11:51.41 | HarryR | ouch |
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11:51.49 | KriS83 | Short and simple question.. how would I be able to do this: When a call comes in on a certain line, I'd like to see this in my snom: "sales: Extension it's coming from" |
11:51.51 | x86 | weeeeeee |
11:51.51 | HarryR | maybe I shouldn'tve pressed that big red button :) |
11:51.51 | x86 | lol |
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11:57.13 | S^P | hi i register to a sip provide using register=> ... how i can dial thought it? |
11:58.46 | benjk | only your provider can tell you |
12:01.17 | S^P | when i connect ATA i can dial easily :( |
12:04.27 | madafaka | where does asterisk keep sound files it uses for SayNumber? |
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12:04.32 | madafaka | i want to record them in my language |
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12:06.41 | jrprado | hi, hi |
12:07.08 | jrprado | somebody already configured telextreme in asterisk ? |
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12:10.09 | creativx | madafaka: /var/lib/asterisk/sounds |
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12:10.25 | madafaka | creativx, tnx man...just found /var/lib/asterisk/sounds/digits :) |
12:10.46 | madafaka | just to find out how to record voice into gsm/ulaw format :) |
12:11.04 | benjk | you can record them in other formats, then use sox to convert |
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12:12.36 | creativx | madafaka: record() |
12:12.43 | creativx | although using that for digits might not be the best way |
12:12.49 | creativx | i'd rather use a prof sound editor |
12:12.56 | creativx | so that you dont get odd gaps and shit |
12:13.12 | madafaka | sure :) |
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12:19.12 | jrprado | somebody already configured telextreme in asterisk ? |
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12:21.02 | fourcheeze | does the Polycom IP501 have buddies? |
12:21.13 | fourcheeze | buddies as in those people whose presence you can see |
12:22.57 | fourcheeze | I've found a Contact Directory but it doesn't seem to show status |
12:23.01 | fourcheeze | am I missing something |
12:25.35 | [TK]D-Fender | fourcheeze: Yes it supports buddy watch |
12:25.54 | [TK]D-Fender | fourcheeze: You need to enable presence support in sip.cfg first |
12:26.17 | fourcheeze | [TK]D-Fender: I'm not provisioning them from the network - can I do this in the gui? |
12:26.23 | [TK]D-Fender | fourcheeze: If you don't have a "Buddies" soft-key on Idle then you haven't done this |
12:26.32 | [TK]D-Fender | fourcheeze: I doubt it. |
12:26.36 | fourcheeze | damn |
12:26.48 | fourcheeze | I hate all that tftp crap |
12:27.21 | [TK]D-Fender | fourcheeze: The web GUI should never be used and I hope they remove it outright ASAP to make room for needed features. |
12:27.34 | [TK]D-Fender | fourcheeze: It supports much more than just TFTP..... |
12:30.31 | tzanger | Mugatu? |
12:30.32 | tzanger | heh |
12:30.45 | tzanger | "My entire panty line is made in sweatshops along the Malaysian border!" |
12:31.32 | tzanger | [TK]D-Fender: did you see that commit yesterday? They fixed (worked around) presence for 2.0.1 polycom fw |
12:32.01 | jrprado | somebody already configured telextreme in asterisk ? |
12:32.02 | [TK]D-Fender | tzanger: Polycom release? |
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12:32.15 | [TK]D-Fender | jrprado: Who are they? |
12:32.23 | tzanger | [TK]D-Fender: no, * commit worked around it |
12:32.51 | [TK]D-Fender | tzanger: Whats the reason for the failure exactly, and how much is * to blame for the disparity? |
12:33.08 | leopardus1 | how much does a small asterisk system sell? ;| |
12:34.13 | jrprado | a company VoIP |
12:34.32 | [TK]D-Fender | leopardus1: * is free, and the "system" is just a computer. Buy whatever you want. Cards if needed vary based on those needs. |
12:34.41 | [TK]D-Fender | jrprado: Link please... |
12:34.51 | jrprado | he work with user, pass and authenticationID |
12:35.29 | [TK]D-Fender | jrprado: Sounds like jsut about every provider out there. I'm sure you can set them up nearly identical to some other popular provider. |
12:35.41 | tzanger | [TK]D-Fender: the patch says "polycom phones only handle xpidf+xml, even if they say they can handle pidf+xml as well" |
12:36.00 | tzanger | it's a VERY small patch |
12:36.11 | tzanger | basically if the useragent is Polycom, force XPIDF_XML |
12:36.13 | [TK]D-Fender | tzanger: 5 chars or less? ;) |
12:36.26 | tzanger | otherwise allow them to select PIDF_XML (RFC3863) |
12:36.34 | tzanger | 1 extra line |
12:36.45 | tzanger | remove 3 lines, add 4 |
12:37.04 | leopardus1 | [TK]D-Fender : what do you mean - cards if needed vary based on those needs? |
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12:37.52 | [TK]D-Fender | tzanger: Cool... unfortunately I'll have to wait for 1.4's release before taking advantage of it.... |
12:38.45 | tzanger | ? 1.4 isn't released? |
12:38.52 | tzanger | they fixed it in 1.2 as well |
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12:40.28 | [TK]D-Fender | tzanger: Well I wonder if we'll see another 1.2 release before 1.4 goes "gold" |
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12:41.21 | [TK]D-Fender | leopardus1: Well you can run * without any special hardware if you just want to use it for VoIP calls. If you want to plug other kind of lines & phones in you will need extra equipment. The cost of which varies depending on that need. |
12:43.00 | leopardus1 | [TK]D-Fender : What I really mean, is this system really selling? |
12:43.27 | mut | does anyone know anyone i can talk to about a lucent stinger config problem? getting desprate now =\ |
12:43.36 | leopardus1 | [TK]D-Fender : sorry I'm actually talking about trixbox! |
12:48.00 | [TK]D-Fender | leopardus1: Trixbox is just a crappy packaged up Linux ddistro CD. Costs NOTHING. Its all just free software. |
12:48.01 | sudhir492 | D-Fender: I am still struggling with paging on Polycom phones :-( |
12:48.18 | [TK]D-Fender | sudhir492: I msg'd you last night offering to log in to help you fix it..... |
12:48.32 | [TK]D-Fender | sudhir492: You must have waled away from your computer right after asking it. |
12:48.43 | sudhir492 | Thanks for that. I must have stepped out |
12:49.06 | sudhir492 | I remember, I got a call and I had to step out in emergency |
12:49.11 | [TK]D-Fender | sudhir492: PM me connect details and I'll get you fixed up. |
12:49.30 | sudhir492 | thanks. How do I PM |
12:50.25 | *** join/#asterisk Jan_Chu (n=jan@213.150.51.91) |
12:51.09 | Jan_Chu | hi, i have a "problem" with asterisk, i have a fully functional IVR, but when i use it, it has a LONG wait time before it forwards me to the chosen destination |
12:51.25 | Jan_Chu | anything i can do about this... we are talking 3 sec of silence |
12:53.25 | [TK]D-Fender | Jan_Chu: Pastebin your IVR dialplan and CLI output for it |
12:53.43 | [TK]D-Fender | Jan_Chu: www.pastebin.ca |
12:53.59 | fourcheeze | [TK]D-Fender: what's my quickest route getting the polycom to read a sip.cfg from somewhere? |
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12:55.13 | Jan_Chu | <- somewhat of a noob, [TK]D-Fender where do i find these config files ? |
12:55.20 | creativx | /etc/asterisk |
12:55.29 | creativx | :P |
12:55.37 | creativx | no wait.. polycom. my bad |
12:55.38 | creativx | :) |
12:55.51 | Jan_Chu | hehe ya, but in witch files :D |
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12:56.17 | benjk | witch files, haha, I like that |
12:56.29 | Jan_Chu | hehe =) |
12:56.35 | Jan_Chu | i think FreeBPX is more of my place. |
12:56.44 | Jan_Chu | this is more develobment i think..... |
12:56.44 | [TK]D-Fender | fourcheeze: FTP |
12:56.46 | joelsolanki | Hello all. I had visited call center client. i got the exact scenario now. |
12:57.12 | [TK]D-Fender | Jan_Chu: If you are using FreePBX, you indeed should not be asking that in here. |
12:57.25 | benjk | Jan_Chu, what config files are you looking for> |
12:57.27 | benjk | ? |
12:57.43 | Jan_Chu | sorry wront place, going to the noob channel now =) |
12:57.44 | joelsolanki | They are using Parsec call center software/hardware in that they have 2 E1 cards and both E1 cards are connected to cisco 3640 2 E1 cards. |
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12:58.00 | joelsolanki | now i want to replace cisco 3640 with asterisk + 2 E1 cards. |
12:58.34 | joelsolanki | what i see is that they Parsec software/hardware is using g711 and cisco 3640 is doing transcoding to g729 |
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12:59.02 | joelsolanki | now i m planning to keep asterisk with P4 / 1GB ram and 2 E1 card in that. |
12:59.17 | fourcheeze | [TK]D-Fender: does the 301 support buddys too? |
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12:59.25 | [TK]D-Fender | fourcheeze: All of them |
12:59.26 | joelsolanki | so will P4 with 1 GB ram will transcode 60 channels on that hardware ? |
12:59.49 | flackes | think that is to low spec tbw |
12:59.50 | flackes | tbh |
13:00.08 | joelsolanki | ? |
13:00.11 | joelsolanki | means ? |
13:00.13 | flackes | does anyone here use the GXP - 2000 and have problems with it not responding? |
13:00.18 | [TK]D-Fender | "To Be Honest" |
13:00.20 | flackes | dont think that spec is high enough |
13:00.39 | joelsolanki | any one can recommend me ? |
13:00.46 | joelsolanki | RoYK: u there ? |
13:00.52 | [TK]D-Fender | flackes: Everything Grandstream is flakey junk and should be avoided. |
13:01.01 | flackes | well i have 100 phones and im using 2 x 3.0gb xeons |
13:01.05 | flackes | lol |
13:01.16 | RoyK | joelsolanki: sure. whatup_ |
13:01.24 | flackes | what would you recomend.. the phone has a few bugs but they seem to work ok |
13:01.45 | joelsolanki | RoyK: i visited the callcenter customer |
13:01.46 | flackes | Asterisk is very processor intensive |
13:01.54 | tzanger | flackes: depends on what you're doing |
13:02.06 | RoyK | joelsolanki: so good for you. what has that got to do with me? |
13:02.50 | joelsolanki | Nothing I have question regarding transcoding from g711 to g729 |
13:03.04 | joelsolanki | let me give u complete scenario so u can understand ok ? |
13:03.23 | benjk | Grandstream Barbietones are alright for putting into your grandparents house or the kids room or as a kitchen phone, stuff like that |
13:03.41 | Holos | Can I fork a Playback or Background to start to play and immediatly go into the next step? I have am using a Background() followed by a Read() and If I enter digits before the Background() is up, it tires to locate that extension in the context. I'd like it to be at the Read() right away. |
13:03.49 | [TK]D-Fender | flackes: Polycom or Aasta are the first shoice for any kind of business, and Linksys only for places that its hard to get the other 2 at a reasonable price in. |
13:05.02 | joelsolanki | RoyK: Parsec software/hardware has 2 E1 cards in it and both are connected to cisco 3640 2 E1 cards. and parsec software/hardware is using g711 and i guess cisco is transcoding to g729. now i m planning to keep 1 asterisk server with config of P4/2.4 Ghz / 1 gb ram and 2 E1 cards in it. so question is that will the hardware will transcode 60 channels from g711 to g729 ? |
13:05.05 | [TK]D-Fender | Holos: : Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) |
13:05.11 | joelsolanki | Or i need higher end server ? |
13:05.25 | RoyK | joelsolanki: quite possibly. try it |
13:05.27 | [TK]D-Fender | Holos: I do believe if you pass Read the file to play it will play up to the point of the first digit. |
13:05.40 | Holos | [TK]D-Fender: Of course... thats why the Filename is in there.. Duh.. I just looked at that yesterday... |
13:06.08 | joelsolanki | do u recommend any hardware ? |
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13:07.52 | fourcheeze | [TK]D-Fender: ok, I've set me up an ftp site - do I need a firmware on there or can I just upload the config? |
13:09.06 | pigpen | Incoming call: Got SIP response 500 "Internal Server Error" back from 10.3.1.7 |
13:09.16 | pigpen | ^^^I am getting this from my polycom 601...ideas? |
13:09.30 | pigpen | SIP 2.0.1 |
13:09.34 | Holos | pigpen: All the time? or just intermittantly? |
13:09.45 | pigpen | intermittantly... |
13:10.10 | pigpen | I should also note that my buddy watch is no longer working on 2.0.1 |
13:10.17 | [TK]D-Fender | fourcheeze: You don't need firmware there, just the proper config files. |
13:10.20 | pigpen | It was working fine on 1.6.7 |
13:10.23 | Holos | pigpen: Ya, I get those as well, I haven't worried much about it. |
13:10.32 | pigpen | yeah...just annoying. |
13:10.52 | [TK]D-Fender | pigpen: There is a bug in the way Polycom claims to support presence which has JUST been fixed. |
13:11.07 | pigpen | ah....next ver of the sip software I am sure... |
13:11.10 | pigpen | thanks for the info... |
13:11.16 | [TK]D-Fender | tzanger>[TK]D-Fender: the patch says "polycom phones only handle xpidf+xml, even if they say they can handle pidf+xml as well" |
13:11.21 | Holos | pigpen: I have seen these a lot more when the call is forwarded. But I have been getting them on all firmwares. |
13:11.26 | [TK]D-Fender | <PROTECTED> |
13:12.21 | pigpen | I see this mostly when one of my buddy watch exten's recieves a call... |
13:12.30 | pigpen | thanks. |
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13:16.43 | [TK]D-Fender | ~pb |
13:16.54 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
13:16.55 | Hymie | hey guys.. does anyone know why pickup() won't work when an extension is being run via a dial(something&something&something) instead of a plain dial(something)? |
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13:17.59 | [TK]D-Fender | tzanger: I can't see reference to the patch on Mantis.... |
13:19.08 | pigpen | [TK]D-Fender, thanks...but I am trying to get things "cleaned up" with the way that the Siemens switch sees messages, etc... After this, then we can get down to the isssue... |
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13:27.48 | tzanger | [TK]D-Fender: it's not on mantis |
13:28.13 | tzanger | in branch 1.2 it's r44432 |
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13:28.29 | tzanger | in 1.4, r44433, and trunk r44434 |
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13:31.08 | [TK]D-Fender | tzanger: Sigh. Ok, guess I'll just have to sit back and wait for a release now.... |
13:31.26 | [TK]D-Fender | tzanger: Any word on the "500 internal error" flooding? |
13:32.01 | KriS83 | Hi, is there anyway I can prefix incoming calls to be shown like this on a SIP Phone: "sales: ${CALLERID}" |
13:32.37 | tzanger | [TK]D-Fender: I think that may fix that if the polycoms are saying they support PIDF |
13:32.40 | [TK]D-Fender | KriS83: Yes, change the callerid. |
13:32.43 | tzanger | not sure, I'll have to update and see |
13:33.11 | [TK]D-Fender | KriS83: Set(CALLERID(name)=sales: ${CALLERID(name)}) |
13:33.26 | KriS83 | Thanks |
13:33.27 | Hymie | hey guys.. does anyone know why pickup() won't work when an extension is being run via a dial(something&something&something) instead of a plain dial(something)? |
13:36.35 | [TK]D-Fender | Hymie: Pastebin your related dialplan and CLI output |
13:38.47 | Hymie | [TK]D-Fender: give me 5, I'm trying to pickup(extension&extension) to match the dial(extension&extension). FYI, pickup(extenstion) works if there is no & in the dial statement |
13:39.49 | [TK]D-Fender | Hymie: Pickup(extension[@context]): This application can pickup any ringing channel |
13:39.51 | [TK]D-Fender | that is calling the specified extension. If no context is specified, the current |
13:39.52 | [TK]D-Fender | context will be used. |
13:40.11 | [TK]D-Fender | Hymie: Nowhere in there doe is imply that you can do this to multiple channels. Why do you suddenly think it will? |
13:40.32 | Hymie | [TK]D-Fender: I didn't say I wanted it to pickup multiple channels |
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13:40.48 | Hymie | [TK]D-Fender: it won't pickup any extension in a dial(blah&blah) statement |
13:40.58 | Hymie | so I tried pickup with the same as the dial line, just as a test |
13:41.03 | Hymie | I dont' know what the code says |
13:41.13 | Hymie | hold on, it may be the context |
13:41.19 | Hymie | might be my bonehead |
13:44.23 | [TK]D-Fender | Hymie: I was suspecting that possibility but was waiting for the incriminating pastebin ;) |
13:44.43 | Hymie | hmm |
13:44.47 | Hymie | oddly, that didn't work |
13:44.51 | Hymie | although it seemed like the answer |
13:45.06 | Hymie | wups |
13:45.08 | Hymie | hold |
13:47.39 | Hymie | #$@$@#)($ |
13:48.07 | Hymie | pastebin.ca is slow, wtf |
13:48.24 | Hymie | !pb |
13:48.26 | Hymie | ~pb |
13:48.31 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
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13:50.40 | Hymie | [TK]D-Fender: log output -> http://channels.debian.net/paste/3959 |
13:52.59 | Hymie | [TK]D-Fender: http://channels.debian.net/paste/3960 <--- the dialplan parts |
13:54.10 | Hymie | <PROTECTED> |
13:54.10 | Hymie | <PROTECTED> |
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13:56.09 | skywriter | is asterisk supported on fedora core 5 |
13:56.11 | fourcheeze | [TK]D-Fender: ok, I'm officially a polycom convert |
13:59.48 | intralanman | wow, WTF just happened there |
13:59.49 | skywriter | what happened? |
13:59.49 | Hymie | it's just a netsplit, heh |
13:59.49 | skywriter | is asterisk supported on fedora core 5? |
13:59.49 | fourcheeze | what exactly is a netsplit? |
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13:59.49 | Hymie | skywriter: not sure what you mean there... it will compile on it, I'm sure |
13:59.49 | Hymie | skywriter: and there are likely rpms for it |
13:59.49 | flackes | Any one have a clue how to Reboot an asterisk phone from Asterisk |
13:59.49 | Hymie | fourcheeze: one server has lost connection with the node it was connected to |
13:59.49 | skywriter | i hade the following message when i tried to compile zaptel |
13:59.49 | Hymie | flackes: what type of phone? SIP? some phones support reboot with SIP, but I don't know how to issue the command |
13:59.49 | skywriter | You do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel instal led. |
13:59.50 | flackes | yea they are SIP |
13:59.50 | Hymie | skywriter: install the kernel headers, for the kernel you have installed |
13:59.50 | skywriter | any ideas? |
13:59.50 | skywriter | how? |
13:59.50 | Hymie | skywriter: there should be something yum can download |
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13:59.57 | TexasJay | morning :) |
13:59.57 | skywriter | any command should i type |
13:59.57 | Hymie | flackes: I'd search on voip-info for sip reset.. look at the uniden UIP200 page for a ghost of a hint, talking about it |
13:59.58 | skywriter | i m a beginner |
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14:01.15 | Hymie | skywriter: I don't know anything about fedora, someone here might |
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14:01.24 | Hymie | skywriter: I don't know anything about fedora, someone here might |
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14:01.27 | flackes | will do :D |
14:01.28 | flackes | so its possible then? |
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14:02.56 | skywriter | when i try to install zaptel on fedora core 5 i got the following message |
14:02.56 | skywriter | You do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel instal led. |
14:02.56 | skywriter | any ideas |
14:02.56 | TexasJay | quick queue question: what's the correct queue type for making asterisk dial the next agent, remembering who the last successful agent was? |
14:02.56 | Hymie | skywriter: I told you what you need to do, now you just need to find out how to install the kernel headers for your kernel. You need to use YUM, or whatever you use to install packages on fedora... |
14:03.51 | Hymie | skywriter: search for the kernel number above, and "headers" |
14:03.51 | Hymie | or some such |
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14:05.33 | quid246 | skywrite: yum install kernel-devel |
14:05.33 | Hymie | there you go! ;) |
14:05.33 | [TK]D-Fender | TexasJay: RRMEMORY |
14:05.33 | skywriter | how can i know? |
14:05.33 | Hymie | crazy fedora! ;Þ |
14:06.41 | quid246 | gee, thanks would be nice |
14:08.42 | Hymie | [TK]D-Fender: any ideas? |
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14:09.39 | fall0ut | so, what is #asterisks reccomendations for appliances/front-ends/all-in-one solutions for small officez? |
14:11.09 | [TK]D-Fender | Hymie: I didn't get your 2nd pastebin due to the netsplit |
14:12.14 | [TK]D-Fender | fall0ut: We don't have one. GUI's are not supported here. If you want something cheap and easy, go with Trixbox. For a bigger office, call www.williamsglobal.com and ask about Fireworx |
14:12.30 | fall0ut | just want reccomendations |
14:13.47 | pigpen | Hi all...Siemens tech support is saying that Asterisk is sending the "channel number" in the "connect message". Apparently, it doesn't affect functionality, but it make more warning messages during troubleshooting...any way to disable this? |
14:14.47 | [TK]D-Fender | fall0ut: I just gave you 2. |
14:17.02 | fall0ut | thnx |
14:20.05 | Hymie | [TK]D-Fender: http://channels.debian.net/paste/3960 <--- the dialplan parts |
14:20.10 | Hymie | [TK]D-Fender: there you go |
14:22.03 | [TK]D-Fender | Hymie: Pretty obvious problem... there is no Exten 100 in [reception] ..... |
14:22.38 | Hymie | [TK]D-Fender: hmm, I didn't try 9@reception... |
14:22.40 | Hymie | er |
14:22.42 | Hymie | 0@ |
14:22.47 | Hymie | hold on |
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14:23.02 | Hymie | er |
14:23.15 | [TK]D-Fender | Hymie: there is no NUMBER in [reception] period. Please adjust your vision and direction :) |
14:23.23 | Hymie | yeah |
14:23.53 | Hymie | I thought it was checking for the actual extension, I started with SIP/100 in there, then moved it to 100 without thinking about the paragram shift |
14:23.57 | Hymie | will try |
14:24.02 | [TK]D-Fender | Hymie: I can already see how I'd fix this.... use your imagination..... |
14:24.13 | Hymie | yeah, a simple goto |
14:24.20 | [TK]D-Fender | Hymie: SIP/100 is a DEVICE, not and EXTEN.... theres a clue for you.... |
14:24.27 | [TK]D-Fender | an* |
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14:24.40 | Hymie | paragram shift |
14:25.17 | pifiu-laptop | morning fender |
14:25.26 | [TK]D-Fender | Hymie: Excellent :) |
14:25.33 | [TK]D-Fender | pifiu-laptop: y0 |
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14:26.04 | pifiu-laptop | i got one of those linksys 1 line voip desktop phones, im going to give it a try sometime in the next couple of days |
14:26.15 | pifiu-laptop | they look pretty simple which is good |
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14:28.49 | [TK]D-Fender | pifiu-laptop: EW..... hope you didn't spend personal money on it.... |
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14:30.40 | Tili | i get this error in compiling zaptel on debian libtonezone.a(tonezone.lo)(.gnu.linkonce.t.__i686.get_pc_thunk.bx+0x0):/usr/src/zaptel-1.2.6/tonezone.c:41: first defined here |
14:33.00 | [TK]D-Fender | Hymie: I believe the word you were looking for is "paradigm". Indeed it was sort of grasping for it in place of my use of "direction". |
14:34.13 | Hymie | [TK]D-Fender: no.. no.. my spelling is correct if you are welsh |
14:34.56 | [TK]D-Fender | Hymie: Watergate 2.0! |
14:35.27 | mut | watergate was about bad spelling? |
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14:35.41 | Hymie | mut: Nixon will tell you so ;) |
14:35.46 | mut | heh |
14:35.52 | Hymie | I WAS FRAMED, IT WAS THE SPELLING! |
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14:43.31 | Hymie | [TK]D-Fender: thanks dude, mucho appreciated and such |
14:43.57 | [TK]D-Fender | Hymie: Quite welcome |
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14:50.36 | pingwin | is there a way to "expire" a variable after say 6 hours? |
14:50.40 | pingwin | or reset? |
14:50.48 | pingwin | a method perhaps |
14:50.51 | pingwin | something that would work in my dialplan |
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14:52.29 | pablus | morning |
14:52.31 | Hymie | pingwin: you could always place a file in /tmp that had the same name as the variable |
14:52.31 | intralanman | pingwin: for the same 6 hours every day? or just 6 hours from when you set it? |
14:52.56 | Hymie | pingwin: then put the time when you set that variable, in it, using a CLI command from asterisk, such as date using unix time |
14:53.05 | Hymie | pingwin: then, next time you reference the variable |
14:53.23 | Hymie | pingwin: you could run another cli command, one that subtracts the current date from the one in the file |
14:53.25 | pingwin | hrmmm. that may work |
14:53.33 | Hymie | pingwin: and resets the variable if it is > number seconds |
14:53.41 | pingwin | wouldn't that be the same as just setting two variables. one with the timestamp it was created? |
14:54.01 | Hymie | pingwin: sure, could do that too, I suppose ;) |
14:54.08 | pingwin | or can asterisk dialplan even deal with timestamps? |
14:54.14 | Hymie | pingwin: I don't know if it can |
14:54.20 | Hymie | pingwin: which is why I mentioned the above |
14:54.36 | Hymie | pingwin: but, even if it can't, you can likely set a variable to the output of a date command for timestamp |
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14:54.49 | Hymie | pingwin: then exclude the additional step I had, of writing it to a file |
14:54.53 | rene1 | ~seen oej |
14:55.24 | jbot | oej <n=oej@23.Red-88-7-53.staticIP.rima-tde.net> was last seen on IRC in channel #asterisk, 7d 23h 42m 52s ago, saying: '~seen kpfleming'. |
14:55.24 | Hymie | oej was never here, rene1 |
14:55.24 | rene1 | ahh i se |
14:55.24 | rene1 | thx |
14:55.24 | Hymie | rene1: you are in an alternate universe, rene1 |
14:55.28 | Hymie | rene1: please give me money, rene1 ;) |
14:56.01 | pingwin | Hymie: cool, thank you, I'm going to give that a try |
14:56.20 | rene1 | Hymie: sure, to your nigerian acct? |
14:56.24 | rene1 | ;) |
14:56.30 | Hymie | hehe |
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15:01.15 | pingwin | anyone know of a good tutorial for writing perl scripts for asterisk? or does asterisk only really work via the system call? |
15:02.17 | macTijn | pingwin: see AGI |
15:02.20 | HarryR | pingwin, you could start at http://www.voip-info.org/wiki-Asterisk+AGI or http://home.cogeco.ca/~camstuff/agi.html |
15:02.26 | macTijn | that :) |
15:02.37 | pingwin | sweet, thanks for all your help :) |
15:02.44 | pingwin | asterisk fucking rocks guys! |
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15:04.00 | brettnem | hey all |
15:04.11 | brettnem | anyone in here using gnudialer? |
15:05.01 | brettnem | wow.. quiet in here today |
15:05.21 | TexasJay | anyone here an knowledgable about the aastra 480i? |
15:06.19 | brettnem | I've had a lot of trouble getting that phone to work |
15:06.23 | brettnem | where in Texas are you? |
15:07.33 | pigpen | brettnem, careful ..he may be "packin" |
15:07.45 | pigpen | You know Texan's |
15:07.58 | brad_mssw | it's quiet b/c everyone is taking the day off for GatorGrowl ... obviously, that's a national holiday, right ? I mean most businesses around here are closed or closing early |
15:08.48 | brettnem | pigpen: where abouts? |
15:08.52 | pigpen | middle. |
15:09.03 | pigpen | San Antonio |
15:09.10 | brettnem | ahh, I'm in Dripping Springs |
15:09.23 | pigpen | ah yes...the spring with a leak. |
15:09.37 | brettnem | which is why all our wells are drying up |
15:09.41 | pigpen | I am working today out of Boerne... |
15:10.21 | brettnem | for those of you not from texas, that's pronounced "BERNIE" |
15:10.33 | brettnem | or I think it is.. heh |
15:10.51 | pigpen | yep...dam Germans...hehe |
15:11.00 | brettnem | german texans |
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15:11.12 | pigpen | yep... |
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15:12.18 | brettnem | ok, so no one in here has used gnudialer? |
15:12.18 | pigpen | and sorry, I have never touched an aastra.... |
15:12.18 | pigpen | or gnudialer... |
15:12.18 | pigpen | I use macdial... |
15:12.18 | pigpen | and no I am not a hippie. |
15:12.18 | brettnem | macdial |
15:12.52 | jmls | leave the poor gnu alone. Pick on something bigger. Like a Kongdialler .. |
15:13.16 | TexasJay | brettnem: what problems are you having with the 480i? I quite like it. Only issue is I wish I could figure out a way to disable the "X missed calls" thing that flashes on the screen. |
15:13.40 | pigpen | some people are really hung up on that.... |
15:13.47 | brettnem | pigpen: macdial isn't a call center dialer.. it's a TAPI |
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15:14.14 | brettnem | TexasJay: I've had a heck of a time configuring it to work behind nat with asterisk |
15:15.00 | TexasJay | brettnem: with Asterisk behind the NAT with you or not? |
15:15.13 | TexasJay | I presume not otherwise you wouldn't have mentioned it. :) |
15:15.34 | brettnem | asterisk public... phones nat |
15:17.20 | TexasJay | can't say i've been in that position... yet |
15:17.26 | pingwin | one other question please... when the p (call screening) flag is raised on a dial, it only records the name of the person if the callerid info is missing. is there anyway, other than something custom, to require the "say your name" option to occur for every phone call coming in? |
15:17.36 | pingwin | or for that dial. |
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15:36.48 | brad_mssw | what are the recommended pay-as-you-go voip providers these days ? |
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15:47.12 | quid241 | If you need outgoing only... check out rapidvox.com |
15:47.28 | fall0ut | Guys at carriers.icall.com are nice |
15:47.32 | fourcheeze | brad_mssw: depends a lot where in the world you are |
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15:47.45 | fall0ut | they are using XO for origination, nice front-end to setup stuff |
15:47.46 | brad_mssw | fourcheeze: hah, true, US ... Eastern |
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15:48.10 | fall0ut | I'm using them for testing out new softswitch |
15:48.20 | brad_mssw | i've used so many providers over the last couple of years ... most of them are aweful ... just wondering if there is some 'fresh talent' |
15:48.43 | fall0ut | Check out icall |
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15:49.09 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
15:49.34 | aptura | brad, it may be the nature of the backbones that are causing the latency and jitter. What do you think |
15:49.36 | fall0ut | either unlimited or $0.01/min origination, and different rates for termination |
15:50.09 | fall0ut | its all pre-paid, too |
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15:51.14 | Synoptic | hi everyone |
15:51.54 | [TK]D-Fender | scottmcl: How do you imagine doing that? How does AMI get the audio channel to complete? |
15:52.10 | Synoptic | i'm having a little problem with a cheap ATA. Might just be some configuration. After hanging up the phone on the ATA, asterisk thinks it is still connected. Could it be the Polarity setup in the ATA ? |
15:52.11 | aptura | If the voip carriers suck then I wonder if anyone who has used asterisk has had any issues with site to site direct conectivity issues. |
15:52.13 | brad_mssw | aptura: yeah, I always do some basic latency/jitter tests before trying out a carrier |
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15:52.47 | [TK]D-Fender | Synoptic: Doesn't make sense. Hung up is a 100% guarantee |
15:53.20 | quid241 | fall0ut: Some good rates there, but then you look at some of the prices for other Area codes.. 14c a min... ouch. |
15:53.55 | aptura | brad, where do you think the problem exist? the voip carrier? the backbone? |
15:54.10 | olivier__ | Synoptic> your ata is connected to * via SIP ? |
15:54.37 | brad_mssw | aptura: well, a lot of problems are the voip carriers themselves ... especially got fed up with teliax because they'd just reboot their systems in the middle of the day (or so it seemed), no one would be able to connect for minutes |
15:54.46 | Synoptic | olivier__: yes, and i'm in North America. |
15:55.06 | aptura | brad, thats a very bad idea. Telcos drop channels for maintence at 2 am. |
15:55.55 | brad_mssw | junction networks has been generally good ... though they do glitch from time to time |
15:56.00 | aptura | thats because no one is on. The telco technicans would often listen for the last calls made..unknown to the caller waiting for them to get off the phone before thay change trunks to remove a frame out. |
15:56.46 | Holos | Can Asterisk do math like "${EXTEN} minus 10"? |
15:56.59 | aptura | I bet the life of a CO tech or egnineer was probebly lonely |
15:57.12 | fall0ut | quid241: yea |
15:57.18 | fall0ut | but who uses a company like this for bulk term |
15:57.48 | aptura | brad u have customers using those services? |
15:58.14 | brad_mssw | aptura: no 'customers', we just use it for our own business ... |
15:58.19 | aptura | okay |
15:59.17 | [TK]D-Fender | Holos: ${${EXTEN}-10} |
15:59.42 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
15:59.46 | [TK]D-Fender | Holos: $[${EXTEN}-10] rather |
16:03.55 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
16:03.55 | *** mode/#asterisk [+o mog] by ChanServ |
16:04.49 | sudhir492 | Hi aptura |
16:07.12 | TexasJay | is it possible to dial multiple extensions and have them all be able to answer? |
16:07.22 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
16:07.50 | TexasJay | without the first answer trumping the other answers, that is. |
16:10.03 | sudhir492 | msg D-Fender TK, Will you please join #polycom |
16:10.14 | Synoptic | ok |
16:10.36 | Synoptic | i'm in the astrisk CLI, trying to figure out why asterisk thinks my ATA FXS is still in use.. |
16:11.38 | aydiosmio | DEBUG |
16:11.39 | rpm | Synoptic: wrong signalling |
16:13.35 | Synoptic | rpm, when it happens, all calls coming from ISP to all extension have no sound. and I cannot place a cal lanywhere inside the pbx system... |
16:14.23 | *** join/#asterisk ikey (i=ikey@220.226.35.125) |
16:14.31 | Synoptic | do you still think it is wrong signaling? |
16:15.30 | rpm | sounds like rtp is broken then |
16:15.41 | rpm | or your codec is incompatible. |
16:16.47 | fall0ut | first off, is it running SIP or MGCP or what? |
16:17.04 | Synoptic | fall0ut, I'm using an ATA using SIP inside the same network as my * |
16:17.14 | Synoptic | no nat and such |
16:17.19 | fall0ut | what ATA? |
16:18.00 | Synoptic | Generic Cheapo Ebay copy of pap2 style.. |
16:18.19 | fall0ut | so what makes you think it's still in use? |
16:18.29 | *** join/#asterisk adorah (n=admin@87.68.169.166.cable.012.net.il) |
16:18.33 | fall0ut | sip show channels shows active calls? |
16:19.06 | Synoptic | Channel Location State Application(Data) |
16:19.06 | Synoptic | 0 active channels |
16:19.06 | Synoptic | 1 active call |
16:19.17 | Synoptic | andthe phone is hung up |
16:19.45 | Synoptic | and now, I cannot make any call, looks like * is getting confised or something |
16:20.12 | scottmcl | [TK]D-Fender : well you can do blind transfers using the redirect command .....some people seem to suggest using playDTMF then sending *2 ..... or what ever you have set the command too |
16:23.41 | *** join/#asterisk bytefoo (n=bytefoo@207-114-255-134.static.twtelecom.net) |
16:24.21 | bytefoo | in asterisk can i make a voice menu that, when you make a selection will run a script or kick off a cron job? |
16:24.46 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
16:26.54 | aydiosmio | bytefoo: yes, just set the IVR to send to an extension.conf entry that calls an AGI() |
16:27.12 | aydiosmio | then you can make that AGI a perl script that runs a series of commands |
16:27.17 | [TK]D-Fender | scottmcl: Of course you can do blind transfers. You're just "throwing" the channel somewhere else without real intervention. |
16:27.48 | bytefoo | sweet thanks aydiosmio, i don't know what half that meant as i'm still new to all this but I wanted to make sure it was possible ;) |
16:28.01 | [TK]D-Fender | scottmcl: But if you want to do Attended, the AMI has no audio/dtmf interface to allow you to enter where. |
16:37.58 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
16:41.33 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
16:50.54 | _alex_mx_ | hello googling led me to a post of mine of sept 2005 with the same problem that went unresolved. We solved it by getting rid of all our iax devices and going pure sip. Now we have 2 servers under heavy load with iax2 trunk between them and here it is again..."channel.c: Dropping voice to exceptionally long queue on IAX2/" server becomes unresponsive and only solution is to restart. We do voicemail to email but log file doesn't show one being done bef |
16:50.55 | _alex_mx_ | ore the errors start. Running latest 1.2 release on both servers. Any ideas? |
16:54.49 | _alex_mx_ | file, ? posting on the correct channel this time...:P |
16:55.06 | file | ? |
16:55.06 | Bobcat991966 | does naybody have googletalk working with asterisk 1.4 svn? |
16:59.28 | MIXX941 | Hi all. Trying to get outgoing termination going through IAX and am getting the "Call Rejected by x.x.x.x: No Authority Found" error 50. The service doesn't require registration or a username/password, just to send a special pin before the phone number in extensions.conf. Been searching online and haven't found a situation exactly like mine (meaning most of them required usernames/passwords or registering, and that was the problem). Any ideas on wh |
16:59.30 | MIXX941 | at to check? |
17:00.20 | aptura | Bobcat991966 CHECK IN voip-info.org |
17:01.00 | intralanman | MIXX941: what's your dial command look like? |
17:01.29 | intralanman | dial(IAX2/peer/PIN${EXTEN}) or something like that? |
17:01.34 | Bobcat991966 | Ya I hve been there looked at some sample configuration but im not able to get it to work. when I dail a google talk extenshion I get Everyone is busy/congested at this time (1:0/0/1) |
17:02.37 | Bobcat991966 | Im wondering during the inital install ofasterisk if there was a confiuration option that needed to be set prior to doing that make install |
17:04.29 | MIXX941 | intralanman: it was "Dial(IAX2/PIN${EXTEN}@provider)" ...but I tried your way and now get a different error |
17:05.07 | intralanman | well, at least it's a different one |
17:05.12 | intralanman | lol |
17:05.43 | MIXX941 | yeah |
17:05.49 | MIXX941 | "I don't know how to authenticate automation to" |
17:06.17 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
17:07.22 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
17:12.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
17:12.10 | *** mode/#asterisk [+o russellb] by ChanServ |
17:13.58 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
17:18.28 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
17:20.10 | *** join/#asterisk larryrichardson (n=larry@64.56.99.31) |
17:21.21 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
17:21.21 | *** mode/#asterisk [+o denon] by ChanServ |
17:21.42 | MIXX941 | does anyone know what "chan_iax2.c:7205 socket_read: I don't know how to authenticate automation to x.x.x.x" means? online searches turn up nothing for that. |
17:22.11 | *** join/#asterisk alexis101 (n=as@MTRLPQ02-1177996380.sdsl.bell.ca) |
17:23.35 | alexis101 | hi guys i have a little problem with my queue System ... When i called the queue command with the n option asterisk still do a retry on the agent , can someone tell me why? |
17:28.21 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
17:28.29 | *** join/#asterisk mr_canny (n=root@200.138.113.82) |
17:29.54 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
17:30.31 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
17:33.09 | *** join/#asterisk hfb (n=hfb@pool-71-108-114-33.lsanca.dsl-w.verizon.net) |
17:36.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:36.24 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
17:36.39 | diclophis-work | hello all |
17:36.57 | diablopico | anyone know how to give a device ( te212p ) its own IRQ ? |
17:37.07 | diclophis-work | using SIP, is there any way to have more than 2 channels connected together? |
17:38.23 | diclophis-work | diablopico: you can disable all the other devices? |
17:38.50 | larryrichardson | I have a ? on Queue notification. I need to have a queue announce the agent that is accepting the call before it is transferred. It looks like there is a Manager "eventwhencalled", but I don;t know how to tap into this. ANy pointers? |
17:39.11 | diablopico | i am running bare bones as it is |
17:39.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
17:40.08 | diablopico | i8 have moved the card to each of the PCI slots on the motherboard , and it always shares an interrupt. |
17:41.44 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
17:46.08 | rob0 | What devices are sharing the IRQ? |
17:46.40 | [TK]D-Fender | diablopico: Tried setting it in your BIOS? What MB are you using? |
17:46.55 | diablopico | the vidio card at the momoent |
17:47.46 | [TK]D-Fender | OUCH.... very bad |
17:47.50 | diablopico | motherboard = SuperMicro ( X7DVA-E |
17:48.09 | [TK]D-Fender | diablopico: Oh well... caught by the Digium IRQ curse..... |
17:48.36 | diablopico | The BIOS wont let me set an irq for the PCI slot |
17:48.56 | diablopico | it all worked until i rebooted |
17:49.09 | japerry | omg |
17:49.11 | tzafrir_home | diablopico, another avenue of magic: kernel parameters |
17:49.14 | japerry | I wanna take a hammer to asterisk |
17:49.27 | tzafrir_home | e.g: acpi=no , or pci=noacpi |
17:49.44 | japerry | seems like the only tool to fix it --- now RANDOMLY incoming calls are getting forwarded to our main extension and not the phone they're supposed to goto |
17:50.09 | [TK]D-Fender | tzafrir_home: I don't think he has the requisite goat to sacrifice for that black an art ;) |
17:50.17 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
17:50.20 | japerry | so IE: I call 555-6001 and it gets routed to 555-6000. but sometimes it will ring 555-6001, but its random |
17:50.23 | diablopico | you mean to recompile the kernel with different parameters ? |
17:50.36 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
17:50.57 | [TK]D-Fender | diablopico: No, he means change your bootloader to call your existing kernel with different options |
17:51.05 | tzafrir_home | japerry, anything interesting in the trace? |
17:51.37 | japerry | thats what I'm attempting to do now |
17:51.41 | tzafrir_home | do you have a full log with decent verbose level ? |
17:52.00 | japerry | tzafrir_home: well I'm using the asterisk CLI with verbosity 10 |
17:52.19 | diablopico | thanks all , it looks like i have some homework to do.... |
17:52.32 | tzafrir_home | japerry, also make sure (in logger.conf) that this is sent to a log file |
17:52.47 | *** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org) |
17:53.57 | japerry | ahh yes yes I've turned on debug into a file |
17:54.11 | japerry | I was only getting notice,warning,error in messages |
17:54.35 | tzafrir_home | japerry, verbose is typically the most useful. Debug is often too verbose |
17:54.42 | japerry | ok |
17:56.29 | jmls | diablopico: did you update the system before the reboot (kernel etc ) |
17:56.59 | japerry | so yah the logs are showing anything different |
17:57.14 | japerry | the odd part is that I think something might be wrong with what number is being sent through DTMF? |
17:57.45 | japerry | because asterisk only gets the last 4 digits, which has been working--however, its now randomly not picking up those numbers |
17:58.06 | japerry | basically it says 'Executing goto --IVRmenu' when it should be saying 'Executing goto ext 601 |
17:58.17 | japerry | but I dial our 602 ext outside, and it works fine |
17:59.22 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
18:00.19 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
18:01.59 | hmmhesays | ok my mediatrix 2102 is freaking out sending out registrations constantly |
18:04.54 | hmmhesays | anyone else run into this? |
18:07.03 | hmmhesays | dead in here today |
18:08.03 | jmls | sorry, not dead. Just the chiili con carne .. |
18:08.12 | jmls | *chilli |
18:08.27 | Bobcat991966 | does anybody know how to install iksemel on asterisk 1.4 svn? |
18:09.08 | jmls | Bobcat991966: do yourself a favour, get the source, and compile it with no optimizations. |
18:09.22 | Bobcat991966 | why so? |
18:09.37 | jmls | Bobcat991966: see bug #7672 |
18:10.27 | jmls | specifically look at the note (0050803) |
18:12.50 | jmls | file: ouch |
18:13.06 | jmls | file: I'm not dead. Honest |
18:13.12 | file | a likely story |
18:13.29 | jmls | I feel happy, perfectly happy. |
18:13.30 | hardwire | freak |
18:13.43 | jmls | frack |
18:15.13 | scottmcl | Does any one know how to do an Attended transfer via the Asterisk Managment API? |
18:15.41 | jmls | scottmcl: I'm sure somebody does. When you find out, let me know ! I want that as well. |
18:17.00 | [TK]D-Fender | scottmcl: Was something in my answer not clear to you? AMI has no audio interface so how are you supposed to talk to the person you want to transfer them to?!?! |
18:17.57 | jmls | [TK]D-Fender: I want to be able to get my application to do the work of the agent (hold->more->transfer->dial-speak->transfer) |
18:18.36 | jmls | [TK]D-Fender: so I want to: Click On Target->Speak-Click and for the call to be transferred to the target |
18:18.37 | [TK]D-Fender | jmls: You can't SIP phones are "smart" and they do their own thing. You can't tell the to do anything. |
18:19.05 | jmls | [TK]D-Fender: I was hoping app_redirect (when it hits the streets) would do this sort of thing |
18:19.06 | [TK]D-Fender | jmls: So forget about forcing them into a transfer they don't initiate. UA = king. |
18:19.55 | Bobcat991966 | jmls: thats an interesting bug, but unless im reading it incorrectly it has nothing to do with adding iksemel.so which is required to make asterisk work with Googletalk. |
18:19.56 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
18:20.14 | scottmcl | D-Fender : i know it is not an audio interface but i want to triger the process from somewhere that is not the phone.... |
18:21.04 | jmls | Bobcat991966: asterisk will make use of the iksemel libraries and calls when using googletalk - so if there is a bug in iksemel, googletalk may find it as well |
18:21.46 | [TK]D-Fender | scottmcl: You can steat the call, but you can't walk them through a transfer. |
18:21.48 | scottmcl | D-Fender : i.e. you are on a call, then you want to click a button on a screen that holds the call then dial a new number.....like *201162227777 |
18:21.50 | mog | hey jmls i should have my rework done to it tonight / tommorrow morning |
18:21.57 | mog | so i will send it to ya |
18:22.05 | jmls | mog: cool! |
18:22.24 | mog | i need to rewrite their makefiles and then it should be good for testing |
18:22.36 | scottmcl | D-Fender : some people say you can do palyDTMF to do it....but not sure if this is the correct way |
18:22.39 | jmls | mog: do you agree with what I'm saying to Bobcat991966, or am I talking male cow dung |
18:22.40 | Bobcat991966 | is bug #7672 what you ment to say because this bugs title is Cannot simultaneously open >1 new browser window |
18:23.22 | jmls | Bobcat991966: 0007672: Asterisk core dump using ast_aji_send |
18:23.22 | mog | that bug is related to ikesemel sucking |
18:23.31 | Bobcat991966 | ahh ok |
18:23.33 | scottmcl | D-Fender: you can start new calls....i have speed dial screen but the transfer process has been pissing me off for weeks |
18:23.36 | mog | and should be fixed with this rework |
18:23.47 | [TK]D-Fender | scottmcl: There is no correct way to do what you want. Is against SIP's concepts |
18:23.53 | mog | Bobcat991966, what happens is their library screws up somewhere inside of gnutls and dies |
18:23.56 | mog | next time we call it |
18:23.57 | aydiosmio | SIP DENIAL |
18:23.58 | mog | we die |
18:24.11 | mog | we could check every time we deref the iksemel parser |
18:24.11 | jmls | Bobcat991966: wtf ? "Cannot simultaneously open >1 new browser window" |
18:24.17 | mog | and maybe should |
18:24.26 | mog | but bug is going to be solved this way |
18:24.36 | mog | as the iksemel parser should never fail |
18:24.41 | jmls | Bobcat991966: are you looking at http://bugs.digium.com/view.php?id=7672 ? |
18:24.47 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
18:24.47 | *** mode/#asterisk [+o denon] by ChanServ |
18:25.07 | Bobcat991966 | jmls,thats beter, makes sence now |
18:25.11 | Bobcat991966 | thanks |
18:25.17 | jmls | <PROTECTED> |
18:25.36 | jmls | Bobcat991966: what were you looking at ??? |
18:26.14 | Bobcat991966 | https://bugzilla.mozilla.org/show_bug.cgi?id=7672 |
18:26.26 | jmls | ok ..... |
18:26.36 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
18:27.24 | mog | why did you go to mozilla's bug page instead of our projects... |
18:32.27 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:32.33 | Nugget | wow. what an idiot. |
18:32.36 | scottmcl | D-Fender : so injecting DTMF tones will not work? My other thought was to interface with the phone some how but i have elmeg ip290 and very little instruction |
18:35.02 | jmls | no, it's a valid place to go - if you search for iksemel bugs you land up there |
18:35.53 | aptura | Is there any really good free web editors other then nvu? |
18:36.04 | jmls | vi! |
18:36.24 | jmls | like writing the web in assembler :) |
18:36.25 | [TK]D-Fender | scottmcl: Injecting tones? An attended transfer mens you speak to the person you wnt to transfer them to first. You still aren't speaking to anyone. Like I said the most you can expect to be able to do is STEAL the call and throw them somewhere else completely. No DTMF required. |
18:36.55 | [TK]D-Fender | I do all my web programming in Notepad2 |
18:37.06 | mog | vim is nice for any edditting |
18:38.02 | Nugget | I'd expect you to prefer an eddittorr with a sppellcchheckkeerr. ;) |
18:38.07 | file | mog: mog. |
18:38.13 | jmls | mog: really. Hope you didn't use it for irc ;) |
18:38.16 | scottmcl | D-Fender : is there any other way of remotely calling an attended transfer....with out touching the phone....i am sure there must be a way. The old shit phone system could do it..... |
18:38.17 | aptura | ohh common |
18:39.12 | aptura | scott, there is always a way. what old pbx do you use? |
18:39.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
18:39.33 | mog | Nugget, vim7 has spellcheck |
18:39.39 | mog | file: file |
18:39.48 | mog | jmls, i use gajim for all my messaging needs |
18:40.56 | hmmhesays | someone write me a t.38 passthru patch for asterisk 1.2.10 |
18:41.05 | [TK]D-Fender | scottmcl: : Theres something very wrong with your idea. How the hell do you you expect to TALK to the other en first thereby making it an ATTENDED transfer? |
18:42.08 | Holos | Can anyone spot the error in this: Voicemail(${MATH(${EXTEN}-10|i))}@all|u)? It works fine if I take the |i or ,i out, but then I get XX.XXXXXX as a result. |
18:42.11 | jmls | Nugget: I said play nice ! |
18:42.23 | *** join/#asterisk xezz (n=xez@62.103.27.89) |
18:42.25 | xezz | hello |
18:42.37 | xezz | i 've just updated to asterisk 1.2.10 |
18:42.40 | jmls | xezz: welcome to the lions' den |
18:42.40 | xezz | latest i think |
18:42.55 | xezz | and asterisk wont start :/ |
18:42.57 | jmls | isn't 1.2.12 the latest ? |
18:43.03 | xezz | yes |
18:43.04 | hmmhesays | t.38 passthru would be great |
18:43.13 | xezz | i've just updated to that |
18:43.19 | [TK]D-Fender | Holos: Voicemail($[${EXTEN}-10])@all|u) |
18:43.23 | jmls | 12 <> 10 |
18:43.32 | xezz | but asterik isnt'getting up |
18:43.40 | jmls | know how he feels ;) |
18:43.41 | xezz | is there a known bug or something ? |
18:43.56 | jmls | pretty hard to say without you telling us *what* the problem is |
18:43.57 | Holos | [TK]D-Fender: So I don't need the MATH function? |
18:44.13 | [TK]D-Fender | xezz: And you've told us SO much about the error messages its given you that we must clearly have everything we need to help you! |
18:44.19 | [TK]D-Fender | </sarcasm> |
18:44.24 | [TK]D-Fender | Holos: Nope. |
18:44.26 | file | my sarcasm detector exploded |
18:44.47 | jmls | missed the opening <sarcasm>. Must be using vi again .. |
18:45.14 | [TK]D-Fender | jmls: I reopen it regularly and it last for days before I expend enough to reset it ;) |
18:45.25 | jmls | yikes |
18:45.35 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
18:45.36 | xezz | [TK]D-Fender : Cannot connect to Asterisk Manager with admin/amp111 |
18:45.46 | xezz | This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available. |
18:45.48 | jmls | see: we've scared the mog away |
18:45.54 | [TK]D-Fender | xezz: That isn't an * startup problem.... try again... |
18:46.05 | jmls | bzzzzt# |
18:46.06 | [TK]D-Fender | (you KNOW what I smell coming, don't you people?!) |
18:46.10 | [TK]D-Fender | <sarcasm> |
18:46.15 | xezz | Asterisk could not start! |
18:46.15 | xezz | Use 'tail /var/log/asterisk/full' to find out why. |
18:46.16 | jmls | trixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox |
18:46.22 | xezz | haha |
18:47.09 | [TK]D-Fender | xezz : Give us something real to work with...... |
18:47.18 | xezz | how ? |
18:47.23 | xezz | thats what i know... |
18:47.47 | xezz | before updating to to 1.2.12 |
18:47.47 | [TK]D-Fender | xezz: Start * manually and see what it says. |
18:47.50 | jmls | what are you using to run asterisk ? |
18:47.57 | xezz | amportal start |
18:48.09 | xezz | ./etc/init.d/asterisk start |
18:48.13 | jmls | simply try asterisk -vvvvvvvvvvvvc |
18:48.14 | [TK]D-Fender | xezz: Start it normally yourself or ask elsewhere.... |
18:48.17 | jmls | ans see what errors you get |
18:48.26 | jmls | *and |
18:48.29 | jmls | bloody vi |
18:48.43 | jmls | no, not "* and bloody vi" |
18:48.51 | jmls | "*and" (bloody vi) |
18:49.07 | xezz | asterisk -vvvvvvc outputed about 30 pages |
18:49.12 | xezz | :/ |
18:49.13 | jmls | uh huh |
18:49.19 | jmls | what was the last line ? |
18:49.34 | xezz | <PROTECTED> |
18:49.46 | jmls | then back to the # prompt ? |
18:50.04 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
18:50.06 | *** mode/#asterisk [+o mog] by ChanServ |
18:50.22 | jmls | the cat's back! |
18:50.39 | mog | lol |
18:50.40 | [TK]D-Fender | You know you can't jsut upgrade * on a trixbox install don't you? it depends on Zaptel having a matching version, adnthat means recompiling EVERYTHING. And BRI is NOT a normal part of Zaptel. |
18:50.49 | mog | just upgrading my server |
18:50.58 | jmls | (for you non uk people - mog == moggy == cat) |
18:51.17 | mog | http://en.wikipedia.org/wiki/Mog has lots of definitions |
18:51.22 | mog | but none refering to me... |
18:51.28 | mog | yes |
18:51.30 | mog | er yet |
18:51.39 | jmls | I like Myelin oligodendrocyte glycoprotein. |
18:51.43 | larryrichardson | Ok, anybody with an idea on how to announce to a caller (in a queue) that "Agent #xxx is handling your call" when the transfer occurs? |
18:51.44 | jmls | sounds dangerous |
18:52.20 | jmls | larryrichardson: there is an option for this. maybe in a patch. hold. |
18:53.32 | xezz | [TK]D-Fender |
18:53.35 | xezz | thats the error |
18:53.37 | xezz | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
18:53.56 | larryrichardson | thanks jmls |
18:53.56 | [TK]D-Fender | xezz: That error isn't WHY * doesn't start, just REPORTS it. |
18:54.15 | xezz | k |
18:54.27 | jmls | larryrichardson: http://bugs.digium.com/view.php?id=6910 |
18:54.35 | xezz | how can i find out why it isnt starting then ? |
18:55.07 | jmls | xezz: it's a problem with your BRI stuff thingy. Which is not a part of asterisk. |
18:55.07 | [TK]D-Fender | xezz: That error means nothing to us for your problem. I jsut told you that it appears that you are nailed by module dependency and you can't jsut try and upgrade * out from under Trixbox like that without redoing everything. |
18:57.07 | scottmcl | D-Fender : on the remote attended transfer all you are doing is simulating pressing the button on the phone from somewhere else...that is not the phone....so you talk to them the same as you normally do....i do not see why this is so complicated. |
18:57.48 | larryrichardson | jmls: Thank you SOOOO MUCH! 3 days of searching completed! |
18:58.11 | jmls | larryrichardson: chalk up one beer. |
19:01.54 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
19:04.35 | *** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net) |
19:04.47 | larryrichardson | jlms: Does this mean it is in the current release of * ? Kind of ambiguious... |
19:05.55 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
19:09.07 | sudhir492 | D-Fender: Paging is finally working for me :-) |
19:10.33 | jmls | larryrichardson: No, it's not in yet - the patch has not been accepted into the main source tree just yet. |
19:11.53 | larryrichardson | ahhh... OK, i'll keep an eye on it then... |
19:13.37 | [TK]D-Fender | sudhir492: What did you do? |
19:14.07 | *** join/#asterisk paulhuynh (n=paul@c-68-82-4-138.hsd1.de.comcast.net) |
19:14.17 | paulhuynh | good afternoon |
19:14.24 | paulhuynh | I need help with my asterisk |
19:14.48 | paulhuynh | we use TDM400P on our asterisk and reciever aout of echo |
19:14.49 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
19:14.59 | sudhir492 | There was a typo - at one place it was alertinfo instead of alertInfo . I told you I must be doing something stupid |
19:15.04 | jmls | larryrichardson: although I think it needs a lot of work from reading it. I also think that it should be handled better. For example, the sound file should be passed as part of the addqueuemember or as part of the agent / member definition and added to the queue member structure. this would make coding a lot easier. |
19:15.54 | paulhuynh | any idea on how i can correct my echo issue |
19:16.04 | tmccrary | With presence, Asterisk and Polycom phones, is it possible to have other phones (attendant) see when a phone is off hook? I've gotten it to say when phones are registered or not (online/offline) but I cannot get it to say when the phone is busy |
19:16.09 | paulhuynh | also what si the power connect do for the tdm400p card? |
19:17.17 | Holos | paulhuynh: Powers the card (With FXS) and may be needed for more then 4 FXO's |
19:17.25 | paulhuynh | oh ok |
19:17.45 | paulhuynh | so i have 4 fxo soi do not need power or shoudl i plug it in anyway? |
19:17.52 | Holos | Sangoma cards require external power when there are more then 4 FXO's installed, so Digium may be the same. PCI bus can't supply a lot of power, so I'd plug it in. |
19:18.06 | paulhuynh | OK |
19:18.15 | paulhuynh | how about my echo issue |
19:18.23 | Holos | You're receiving a lot of echo? |
19:18.35 | paulhuynh | very bad echo on our side when call comein via zaptel |
19:19.04 | paulhuynh | yes i can hear myself talk on the phon but my client dont hear any echo at all |
19:19.05 | Holos | Call Digium support first.. they're there for that. Other then that, make sure the card is tuned (zt_tune?) and that the Gain levels are correct. (This is a analog card right?) |
19:19.20 | paulhuynh | correct |
19:19.38 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
19:19.47 | paulhuynh | digium support i though it is just for install and not config? |
19:20.09 | Holos | Echo dectection and cancelation requires the correct gain levels before it really shines. Find a milliwatt test line, and tune the card. |
19:20.11 | paulhuynh | my cureent gain level is set to 0.0 |
19:20.27 | Holos | paulhuynh: Where are you from? |
19:20.30 | paulhuynh | what do you mean? |
19:20.44 | paulhuynh | how do you do a milli watt test on the line? |
19:20.58 | paulhuynh | this line should be very clean |
19:21.14 | paulhuynh | it coming off a T1 my telco provide it to it |
19:21.23 | Holos | http://www.voip-info.org/wiki/view/Asterisk+fxotune |
19:21.39 | paulhuynh | me and plug into a tdm400p 6feet away |
19:22.35 | tmccrary | So, it looks like Asterisk does not support PUBLISH, so you cannot monitor line status from other phones |
19:22.42 | diclophis-work | whats this all about: "Got SIP response 484 "Address Incomplete" " |
19:23.02 | diclophis-work | i am trying to dial from 1 asterisk machine to another using SIP... |
19:23.29 | diclophis-work | the Dial command on the originating machine looks like this Dial(SIP/11231231234@stg-pbx) |
19:23.53 | diclophis-work | and on the recieving machine i have a extension in the dialplan for the context the calls get through .. i think |
19:26.00 | diclophis-work | arg.. my extensions were mismatched |
19:26.05 | diclophis-work | now I am getting a 603 message.. |
19:26.09 | diclophis-work | "declined" |
19:26.51 | diablopico | hello,,,, does anyone know if UNICALL is supposed to work with newer versions of * ??? |
19:29.34 | Holos | diablopico: http://www.sineapps.com/news.php?rssid=1510 says that 1.12.1 supports it. |
19:29.36 | diclophis-work | what is unical |
19:30.17 | diablopico | thanks Holos |
19:31.37 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
19:37.02 | paulhuynh | help i get this |
19:37.14 | paulhuynh | when try to run fxotune |
19:37.15 | paulhuynh | Tuning module 1 |
19:37.15 | paulhuynh | Could not fill input buffer |
19:37.15 | paulhuynh | ..........Failure! |
19:37.15 | paulhuynh | Tuning module 2 |
19:37.34 | syzygyBSD | can asterisk run on a cluster? |
19:37.41 | paulhuynh | some tim eit woudl said module 1 is OK |
19:38.02 | syzygyBSD | say, to offload the compression to other boxes |
19:38.22 | paulhuynh | it randomly through out the different error for the module |
19:40.51 | paulhuynh | any taker? |
19:40.53 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
19:42.03 | aptura | scarry |
19:47.53 | *** join/#asterisk southtel (n=slester@76.17.115.183) |
19:49.16 | inv_arp[work] | in sed how can i match the n'th occurence of a char? ex.. abcd1abcd want to match the 2nd b |
19:50.26 | tzafrir_home | paulhuynh, are you able to use asterisk with the card? |
19:51.23 | tzafrir_home | inv_Arp, you can use a more complicated pattern: |
19:52.04 | tzafrir_home | inv_Arp, s/b[^b]*\(b\)/something/ |
19:53.02 | tzafrir_home | Or do you want the second line that mhas a b? |
19:54.53 | pablus | hmm |
19:57.41 | pablus | a cyber angel |
19:57.46 | pablus | hmmm |
20:02.07 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
20:06.05 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
20:07.31 | afrosheen | anyone familiar with sip-t |
20:08.13 | afrosheen | i.e. a sip t1? |
20:10.21 | afrosheen | arg where are the useful people these days |
20:11.18 | trelane | sip t1? |
20:11.31 | trelane | afrosheen, plz 2 hand over what you're smoking |
20:11.49 | trelane | you hath drunk from the marketing koolaid |
20:12.03 | afrosheen | some itsp's are providing bundles of a t1 plus a sip trunk that goes straight into their pop |
20:12.07 | *** join/#asterisk lirakis (n=tbright@h-68-165-94-219.nycmny83.covad.net) |
20:12.09 | lirakis | hey all |
20:12.14 | afrosheen | commpartners is branding it 'sip direct' |
20:12.24 | trelane | afrosheen, ok then call it a T1 wtih a sip trunk, not a SIP T1 |
20:12.42 | trelane | I have several such packages deployed wtih a local ISP |
20:12.46 | afrosheen | sorry, the admin here has been calling it that all day like it's recognizable |
20:12.47 | trelane | I'm using one in the office |
20:12.52 | lirakis | im trying to figure out how to send the "remote party id" in the packet headers |
20:12.56 | trelane | afrosheen, you know what to do then right? |
20:13.01 | lirakis | has any one done this, or know how to do it? |
20:13.06 | trelane | lirakis, SIPAddHeader() |
20:13.08 | afrosheen | trelane: yeah but my clue bat is too large to swing in here |
20:13.23 | lirakis | i have set "sendrpid=yes" |
20:13.23 | trelane | afrosheen, then go down to maintainance and get a ball-peen hammer |
20:13.27 | afrosheen | haha |
20:13.41 | lirakis | trelane: you will have to forgive my ignorance.. i dont really know where to use that function |
20:13.47 | trelane | afrosheen, after liberal reprogramming with networking tool #4 (the hammer) your admin will be cluefull, or dead |
20:13.51 | lirakis | trelane: in the dial plan some where? |
20:14.03 | lirakis | trelane: extensions.conf? |
20:14.05 | afrosheen | anyway we currently have a normal bonded t1 that everything is sharing including phones and our call quality blows so were going with this bundled deal soon, wondering if there might be any gotchas |
20:14.22 | *** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net) |
20:14.22 | trelane | lirakis, before you call Dial() in the dialstring do a SipAddHeader(sip:<ip of sip server>/ :whaver you want in the header> |
20:14.45 | lirakis | trelane: hmmm |
20:14.46 | lirakis | okay |
20:14.48 | trelane | <PROTECTED> |
20:14.49 | afrosheen | we're getting 2 sla's (1 from commpartners and 1 from wiltel) so I think we're safe so far |
20:15.03 | trelane | afrosheen, ooh good so both can blame the other! |
20:15.10 | hmmhesays | can anyone help me out with a crisco as5300? |
20:15.30 | trelane | hmmhesays, yeah, reduce the trans fats in your diet, crisco leads to heart attacks! |
20:15.36 | hmmhesays | LOL |
20:15.51 | hmmhesays | I need to figure out if this thing supports g.729 and if it is possible to enable |
20:15.53 | docelmo | say anyone know what the license is for FCC carriers? |
20:15.58 | lirakis | trelane: hmm.. okay |
20:16.15 | trelane | lirakis, that header allows auto answer |
20:16.20 | lirakis | well.. im just getting ready to head out.. i will have to play with it some more on monday.. or possibly on my own pbx this weekend |
20:16.33 | afrosheen | trelane: so how are the packages working that your local ISP is doing |
20:16.38 | trelane | lirakis, you can do some pretty elite things with phones when you can control what headers you send |
20:16.50 | trelane | afrosheen, on the grounds of self-amusement I plead the fifth |
20:16.53 | trelane | (not well) |
20:17.00 | paulhuynh | i can use the card with asterisk but echo is so bad on it |
20:17.05 | lirakis | trelane: .. so if i wanted to send a Passerted ID i could do that in the header with the same function right? |
20:17.09 | trelane | afrosheen, he's already been a repeat-recipient of the clue-by-four award |
20:17.12 | trelane | lirakis, yep |
20:17.27 | lirakis | trelane: im having a little trouble because we have trixbox here.. and the dialplans are kinda.. huge and split up |
20:17.35 | lirakis | trelane: in know this isnt a trixbox channel.. |
20:18.08 | trelane | lirakis, HEY LETS TAKE A REASONABLE DIALPLAN... AND PUT IN LOTS OF POINTLESS MACROS... AND THEN... AND THEN... WE'LL CALL IT TRIXBOX! |
20:18.08 | lirakis | trelane: .. but im having trouble figuring out where in the darn dial plan to put it... i guess i should watch the console when i make a call to see what it is executing |
20:18.15 | lirakis | trelane: yeah |
20:18.15 | trelane | yep |
20:18.16 | afrosheen | trelane: but a point-to-point T1 into our sip provider's pop should be great right? |
20:18.17 | lirakis | i agree |
20:18.25 | trelane | afrosheen, should be yeah :) |
20:18.26 | lirakis | trelane: .. but its what was here when i got here.. |
20:18.32 | *** join/#asterisk klasstek (n=nunyobiz@c-67-177-199-232.hsd1.co.comcast.net) |
20:18.44 | trelane | lirakis, convince the boss to let you start rewriting the dialplan from scratch |
20:18.49 | lirakis | trelane: okay.. i do need to run.. thanks for your help |
20:18.52 | lirakis | ttyl |
20:18.52 | tmccrary | WHERE CAN I BUY THIS TRICKY BOX OF POINTLESS MACROS, I MUST KNOW |
20:18.59 | lirakis | lol |
20:18.59 | trelane | tmccrary, damn right! |
20:19.03 | afrosheen | THEY HAVE THEM AT THE CAPS LOCK STORE |
20:19.14 | aptura | I have a questions about my second line on my ip500. I want it to be my bussiness line and show my CID. The first line will be my residential zap line. I dont have access to the xml files so if anyone here cares to aid me in this would be appriciated. |
20:19.16 | trelane | afrosheen, I SHOP THERE FREQUENTLY!!!!! |
20:19.48 | trelane | aptura, contact the vendor? |
20:19.50 | aptura | The second business line will be sip to my sip provider |
20:19.53 | [TK]D-Fender | aptura: You can config it in the web gGUI just fine if you're already sticking to that lame-o road. |
20:19.55 | afrosheen | aptura, try hitting the web interface on the phone |
20:20.14 | aptura | afrosheen I rarely use that thing because of its problems. |
20:20.27 | afrosheen | well if you won't use the xml files...or the web interface, what else is left |
20:20.36 | [TK]D-Fender | heck you can do it right on the phone if you really care to. |
20:20.56 | afrosheen | [TK]D-Fender, I know you didn't just advocate the phone's menus for configuration |
20:20.58 | [TK]D-Fender | afrosheen: Direct ont he phone physically |
20:21.22 | [TK]D-Fender | afrosheen: You ASKED. I never said I ADVISED it. Please read what I say carefully :) |
20:21.31 | trelane | heh |
20:21.48 | paulhuynh | i need help with some echo issue |
20:21.52 | trelane | afrosheen, you could throw it out and get a snom |
20:21.53 | aptura | TK this is a area I have not delt with yet. |
20:21.53 | [TK]D-Fender | afrosheen: Anybody configuring Polycom's outside of provisioning should be slapped. |
20:21.54 | paulhuynh | i have tdm400p |
20:21.55 | trelane | paulhuynh, what hardware? |
20:22.03 | trelane | call digium for free support or use fxotune |
20:22.05 | paulhuynh | analog card' |
20:22.11 | paulhuynh | ok |
20:22.18 | paulhuynh | will do that ritgh now |
20:22.40 | trelane | they're really good at getting the hardware working and since they're on the phone with you it's a quicker response (and anyway, you paid for it!) |
20:23.29 | tmccrary | D-fender, have you used Polycom phones before? Have you worked with the 601 expansion module? |
20:24.58 | aptura | I purchaced it from atacom |
20:25.33 | [TK]D-Fender | tmccrary: I own every model short of the new IP650. |
20:25.51 | [TK]D-Fender | tmccrary: I though my rep was universally known around here *sigh* |
20:25.55 | trelane | aptura, atacom or atacomm? |
20:26.19 | trelane | [TK]D-Fender, it's good, between you and me there's a polycom expert and a snom expert |
20:27.02 | tmccrary | Ok, with the 601 expansion module, I have mine setup so it can see when buddies are online or offline (reg'd or unreg'd). However, I cannot seem to get it to display when the line is busy or idle on the expansion module. When I run sip show subscriptions in asterisk, asterisk displays this info. |
20:27.15 | tmccrary | Can the expansion module do this with asterisk? |
20:27.36 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
20:28.29 | [TK]D-Fender | tmccrary: IP601's come stock with SIP 1.6.3. and presence DISABLED entirely. You need to enable it in your provisioning files, and then upgrade that phone to 1.6.6 or higher to be able to monitor more than a handful of people. |
20:28.40 | [TK]D-Fender | tmccrary: It most certainly can. |
20:28.52 | [TK]D-Fender | tmccrary: So upgrade your firmware, and tweak your sip.cfg |
20:28.59 | aptura | atacomm |
20:29.00 | tmccrary | I upgraded to 2.05 or whatever (latest) |
20:29.35 | tmccrary | What function in sip.cfg allows this? I have <feature feature.1.name="presence" feature.1.enabled="1" |
20:30.30 | tmccrary | and the 601 *sees* the other phones as buddies, but does not seem to get notified when they are busy. But asterisk does |
20:31.36 | aptura | Atacomm has changed there page. In bod free shipping anywhere in the country :) |
20:31.42 | aptura | bold |
20:32.07 | [TK]D-Fender | tmccrary: There is a bug between * and SIP 2.0.1 that was only resolved TODAY. YOu'll need the latest SVN for that combo to work. |
20:32.16 | tmccrary | aha, maybe its because reg.1.type is private? |
20:32.28 | [TK]D-Fender | tmccrary: No, your reg is fine |
20:32.34 | *** join/#asterisk bmd (n=bmd@64.50.19.206) |
20:32.45 | tmccrary | damn that sucks |
20:33.10 | C6Vette | Is there a way to have a separate queue_log for each queue? I have 3 queues and would like them separated if possible. |
20:33.45 | [TK]D-Fender | tmccrary: Not SO bad... fix is out... just need to recompile, though I don't like working off anything but FTP official releases myself.... |
20:33.54 | tmccrary | yeah, thats what I meant :) |
20:33.55 | [TK]D-Fender | tmccrary: Or downgrade to 1.6.7 |
20:34.05 | [TK]D-Fender | BBIAB, heading home |
20:34.58 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
20:35.32 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
20:36.45 | *** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
20:37.29 | brif8 | Hi All, On a linksys PAP2,, anyone know the admin username and password by default ? I just bought one and it won't let me in ?? |
20:38.08 | trelane | brif8, RTFM? |
20:38.22 | aptura | Atacomm does not provide any configuration support |
20:38.40 | aptura | Guess its up to me anyway :) |
20:38.47 | trelane | aptura, none of this stuff is THAT hard to configure! |
20:38.57 | brif8 | trelane: I have repeatedly it only talks about my routers access which isnt a linksys |
20:39.23 | trelane | so you bought a PAP2... and the PAP2 instructions don't contain the login and password infor for the PAP2? |
20:39.56 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
20:41.07 | *** join/#asterisk SexyKen (n=Ken@c-71-202-149-39.hsd1.ca.comcast.net) |
20:41.13 | SexyKen | Hey guys -- have you heard of Trixbox? |
20:41.25 | TrixVox | nope |
20:41.30 | TrixVox | :) |
20:41.46 | trelane | SexyKen, read the topic |
20:42.03 | trelane | <PROTECTED> |
20:42.22 | SexyKen | I see. |
20:42.23 | SexyKen | :-) |
20:42.53 | trelane | brif8, where did you get the PAP2? |
20:42.55 | SexyKen | Sorry about that :-) |
20:42.58 | trelane | no worries |
20:43.21 | brif8 | trelane: radioshack I think it wants me to get a vonage account and I don't want one |
20:43.31 | trelane | yeah it appears locked to vonage |
20:43.53 | trelane | which will be going out of business any day now |
20:44.05 | trelane | worst ipo ever |
20:44.20 | aptura | vonage is giving voip a bad name |
20:45.19 | tmccrary | I looked into become a vonage reseller a few years ago |
20:45.27 | tmccrary | you had to have 300k customers+ |
20:45.34 | *** part/#asterisk bmd (n=bmd@64.50.19.206) |
20:45.56 | pifiu-laptop | <[TK]D-Fender> pifiu-laptop: EW..... hope you didn't spend personal money on it.... |
20:45.58 | pifiu-laptop | no i didnt |
20:46.04 | pifiu-laptop | but why are they that bad? lol |
20:46.56 | tmccrary | What? |
20:48.22 | brif8 | trelane: yeah so any ideas ? |
20:49.13 | trelane | brif8, good luck, you'll need it |
20:49.50 | *** part/#asterisk acrg (n=aragon@decoder.geek.sh) |
20:52.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
20:53.07 | *** join/#asterisk Norskman (n=jack@jackbuechler.plus.com) |
20:53.38 | Norskman | anyone know anything about SIP versions of the Cisco 7970 V8.04SR1 - I need to know where ot place my files |
20:54.17 | anthonyl | are you asking about the firmware |
20:54.20 | *** join/#asterisk ikey (i=ikey@220.226.35.125) |
20:54.31 | *** join/#asterisk AvoidingDeadlock (n=ASSERT_K@adsl-69-219-50-18.dsl.chcgil.ameritech.net) |
20:55.11 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
20:55.20 | Norskman | THe SEP000F90CEF9BF.cnf.xml file - my TFTP server says it can;t find it even thouh its in the same directory as the f/w and it loaded that into the phone just fine |
20:55.38 | *** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
20:56.13 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
20:57.04 | anthonyl | is the file name correct and matching the one the phone is sending via the tftp request? |
20:57.16 | Norskman | yes 100%. |
20:58.07 | Norskman | e.g. it can send and the phone receives the XmlDefault.cnf.xml file which is in the directory |
20:58.08 | anthonyl | the error is on the server side? |
20:58.19 | anthonyl | it could be a permissions thing |
20:58.31 | funkmaster | is any1 here making use of skypho? |
20:58.47 | Norskman | Not 100% - the file is in the right place. No SEPfile from the TFTP server, no file in the phone recieved =no login to server |
20:59.08 | Norskman | where on the permissions, I am using a windows Pc to send the fiules. |
21:00.24 | Norskman | a read request is being made of that file to the TFTP server. then the TFTP server says File <Sep....> error 2 in system call Create file cannot find the filespecified |
21:00.36 | Norskman | and now its looping looking for the config file. |
21:00.41 | anthonyl | well if the request is not specificying any directory it is just trying to grab it from your tftp-root |
21:00.47 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
21:01.07 | Norskman | TFTProot being the directory pointed at by the TFTP server? |
21:01.59 | anthonyl | ya |
21:02.19 | Norskman | I thought so.. |
21:02.53 | Norskman | SO I am doing it right and its absolutely driving me mad. I have spent weeks trying to get this work and still no luck. Can;t work out where the problem is. |
21:03.03 | Norskman | Loads of weird error messages. |
21:03.17 | *** join/#asterisk hardwire (n=hardwire@89-208-58-66.gci.net) |
21:03.19 | hardwire | BOO |
21:03.21 | anthonyl | have you used wireshark to double check to make sure things are 100%? |
21:03.36 | Norskman | whats wireshark please? |
21:03.55 | hmmhesays | so who wants to help me with this as5300 |
21:04.04 | anthonyl | Norskman, it's a packetsniffer |
21:04.09 | anthonyl | wireshark.org |
21:04.11 | hmmhesays | i'm having a helluva time with it |
21:04.34 | Norskman | ok will look. I have ethereal loaded - presumably thats ok as well |
21:04.39 | Norskman | Same sort of ting |
21:06.48 | *** part/#asterisk Norskman (n=jack@jackbuechler.plus.com) |
21:09.12 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
21:09.20 | Optic | i'm having weird segfaults :( |
21:09.44 | Optic | Connected to Asterisk SVN-branch-1.2-r44580 currently running on operator (pid = 18701) |
21:09.51 | Optic | backtrace here: http://pastebin.ca/193400 |
21:09.56 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
21:10.11 | anthonyl | wireshark is the new-er etherreal |
21:10.13 | Optic | any ideas? |
21:10.49 | anthonyl | Optic, humm |
21:11.03 | anthonyl | what version of asterisk? |
21:11.08 | Optic | see above :) |
21:11.13 | Optic | SVN-branch-1.2-r44580 |
21:11.42 | Optic | i just svn update'd and rebuilt today because of the crashing, but i've had a crash on a rebuilt verson too |
21:11.48 | Optic | basically it's today's svn stable |
21:12.03 | Optic | it seems to be crashing every couple of hours |
21:12.39 | C6Vette | no bt full |
21:12.51 | Optic | i can do a bt full for you :) |
21:12.51 | anthonyl | Optic, has this machine been stable for a while before? |
21:12.58 | Optic | yes |
21:13.32 | anthonyl | i would submi a report to bugs.digium.com with the bt full attached |
21:14.36 | anthonyl | submit* |
21:14.37 | tzafrir_home | Optic, do you get any core file? |
21:14.39 | Optic | yes |
21:14.55 | Optic | is there a way to turn off gdb's pager or send the bt full to stdout non-interactively? |
21:14.58 | Optic | it's long :) |
21:17.05 | C6Vette | I just turn on the logger in the ssh client. If your not local that is. |
21:17.43 | Optic | ah |
21:17.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
21:17.55 | Optic | cat >gdb.cmds |
21:17.59 | Optic | set pagination off |
21:18.01 | Optic | bt full |
21:18.06 | Optic | ^D |
21:18.13 | Optic | gdb -x gdb.cmds asterisk corefile |
21:18.14 | Optic | :) |
21:20.06 | *** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
21:21.21 | Optic | http://pastebin.ca/193413 |
21:21.26 | Optic | now w/bt full |
21:22.24 | Optic | yes |
21:22.42 | Optic | yes, if i'm getting a backtrace :) |
21:22.51 | Optic | oops :) |
21:25.03 | C6Vette | yea,.. I would post a bug with the bt full |
21:25.25 | Optic | ok |
21:25.48 | Optic | i mean, it's possible that it's a hardware problem with the box or something |
21:25.51 | Optic | but I feel like it isn't :) |
21:26.31 | C6Vette | If it was working finr before the update i would ASSume it wasnt the hardware. |
21:27.06 | C6Vette | See some memory errors |
21:27.13 | C6Vette | looks like to me. |
21:30.07 | Optic | yeah |
21:30.15 | C6Vette | Im having memory leak problems on one of mine. at the end of the day I have only a few k available |
21:30.17 | Optic | there, posted :) |
21:30.22 | Optic | http://bugs.digium.com/view.php?id=8109 |
21:30.30 | Optic | this is a pretty busy site too, 30 or so users |
21:30.34 | Optic | and it's normally pretty stable :( |
21:30.51 | C6Vette | Im running aroung 70 users |
21:30.55 | Optic | nice :) |
21:31.04 | Optic | what's your PSTN connection? |
21:31.11 | Optic | we're 10 channels on a PRI here |
21:31.13 | C6Vette | VOIP sip |
21:31.21 | C6Vette | running about 12000 calls an hour |
21:31.26 | Optic | wow |
21:31.33 | Optic | that's awesome :) |
21:32.10 | anthonyl | Optic, cool |
21:32.12 | C6Vette | its a handfull sometimes |
21:32.58 | aptura | man |
21:33.25 | aptura | and asterisk is the box handeling these calls? |
21:34.03 | C6Vette | yea. Im wanting to split it into 2 servers one for handing all the sip clients and one to handle the outbound/inbound calls |
21:34.23 | aptura | now that i have my ftpserver shutdown the ip500 is not locking up. Time to make the changes |
21:34.23 | Optic | ok, i'm out |
21:34.26 | Optic | thanks for your help ;)_ |
21:35.25 | C6Vette | np L8r |
21:36.30 | *** join/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
21:37.46 | slobberknocker | has anyone experienced poor quality music on hold with the files included with asterisk sounds? I have played them on the pc and the sound fine, but when listening to them on hold they have intermittent periods that sound like static. |
21:38.28 | *** join/#asterisk xnon (n=xnon@200.8.86.187) |
21:40.59 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
21:41.00 | TripleFFFF | <PROTECTED> |
21:41.05 | TripleFFFF | egetting lots of these |
21:41.51 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
21:42.24 | xnon | anybody know any easy to install billing system for asterisk? |
21:42.27 | *** join/#asterisk skopii (n=skopii@fuckcingularwireless.com) |
21:42.40 | TripleFFFF | From: "Unknown" <sip:Unknown@74.13.210.125>;tag=as723c47f2 |
21:42.41 | TripleFFFF | lol |
21:42.44 | TripleFFFF | ok !!! |
21:42.54 | TripleFFFF | someone trying to connect as a guest lol darn sniffers |
21:43.24 | aydiosmio | it begins |
21:43.39 | aydiosmio | add tor and you've got a giant newwork of fraud-phones |
21:43.44 | aydiosmio | network |
21:45.01 | TripleFFFF | tor ? |
21:45.20 | TripleFFFF | ill push all the non auth invites to FBI Lol |
21:45.28 | skopii | Hello, I am looking to setup a pbx with sales,and support queues. I have got asterisk installed and am playing around with freepbx. What I dont get is whether or not I need to buy a PRI+Digium Wildcard or if I can use a SIP trunk for both inbound and outbound calls. |
21:45.45 | TripleFFFF | text to speeach ("Im a hacker my ip is X.X.X.X please arrest my dumb ass for being so lame ") |
21:46.17 | slobberknocker | anyone on the moh question? |
21:46.48 | C6Vette | slobberknocker, using mpg123? |
21:46.54 | slobberknocker | native |
21:46.55 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
21:46.59 | *** join/#asterisk VOIPoTD (n=td@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
21:47.06 | C6Vette | good |
21:47.09 | [TK]D-Fender | skopii : You can do it any way you want. VoIP only / PRI / Analog / Whatever. |
21:47.16 | fiber0pti | Just developed a new operator panel for asterisk.. looking for feedback. Anyone want to install it? |
21:47.28 | slobberknocker | fiber0pti: |
21:47.31 | slobberknocker | i am interested |
21:47.41 | C6Vette | fiber0pti: how does it connect ? |
21:47.56 | fiber0pti | It connects via the Asterisk manager |
21:48.01 | skopii | can you tell me why in gods name anyone would want to use a PRI if they could use VOIP then? |
21:48.21 | fiber0pti | slobberknocker: if you would like to try it out please go to i9technologies.com/isymphony |
21:48.25 | fiber0pti | download the server and the client |
21:48.32 | fiber0pti | They both require java. |
21:48.32 | *** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net) |
21:48.38 | slobberknocker | ok, i will play with it tonight. |
21:49.23 | awannabe | has anyone used a adtran atlas 550 for testing FXO and PRI cards in *? |
21:49.34 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
21:50.08 | fiber0pti | slobberknocker: I should be on all weekend.. Please feel free to send feedback |
21:50.24 | fiber0pti | We're going to be changing the UI within a week.. should be much better than it is now |
21:50.40 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
21:50.43 | slobberknocker | ok |
21:51.43 | slobberknocker | fiber0pti: is this first revision? |
21:52.21 | fiber0pti | slobberknocker: it is. |
21:53.33 | slobberknocker | no ideas on my moh sound quality then eh? |
21:54.30 | [TK]D-Fender | skopii VoIP is considerably less reliable, you have to pay for both bandwidth and the service you run on it. |
21:55.01 | [TK]D-Fender | skopii :Higher sound quality (max), and many other reasons |
21:59.55 | *** part/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
22:01.46 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
22:02.36 | skopii | [TK]D-Fender: is it possible to setup rendunant calls on the inbound sort of active/active load-balanced config? |
22:02.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
22:03.02 | skopii | then use the same setup for outbound with a PRI as a failover? |
22:04.02 | wmandra | is anyone here using asterisk with LCS?? |
22:04.43 | xnon | anybody know php-pcntl? |
22:05.00 | xnon | i cant find these pack! |
22:05.47 | wmandra | i'm working on a CSTA Gateway for * that allows pc to pstn and remote call control with LCS and communicator, and just wanted to see if there was any interest ....... |
22:05.48 | xnon | anybody know any biling system for asterisk easy install! |
22:05.56 | *** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net) |
22:06.09 | *** join/#asterisk schirpich (n=kvirc@66.238.64.20) |
22:06.32 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
22:08.35 | qdk | wmandra: LCS? |
22:09.37 | wmandra | live communications server |
22:10.54 | qdk | wmandra: and what's that? |
22:11.27 | wmandra | qdk: http://www.microsoft.com/office/livecomm/prodinfo/default.mspx |
22:11.54 | wmandra | it's basically an internal IM server, but alos has phone control capabilities |
22:12.28 | wmandra | it's actually really nice.... i've been using it here and i love it |
22:13.57 | *** join/#asterisk necudeco (n=chatzill@190.40.219.33) |
22:13.58 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
22:14.02 | necudeco | hi |
22:14.19 | necudeco | I need help to install asterisk in debian |
22:14.19 | qdk | wmandra: sounds expensive... and i still have no idea of what its good for. |
22:15.38 | wmandra | the communicator client is basically an IM client / softphone rolled into one. the nice thing is how it integrates into outlook |
22:15.44 | qdk | necudeco: so right about now you expect people to spoonfeed you? |
22:16.07 | qdk | wmandra: so you can place a call from outlook? |
22:16.41 | wmandra | other users that have me added to their contact list can see my presence status in real time (ie. in a meeting, on the phone) |
22:17.28 | wmandra | if a call comes into my * deskphone a popup appears with the callerid and i have the option to answer the call or forward it to another number like my cell |
22:18.04 | qdk | wmandra: ok, sounds a bit like a feature i have planed for my solution using *, web and other quite free tools. |
22:22.01 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
22:22.13 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
22:24.06 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
22:24.44 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
22:25.31 | BlepsoaF | Hello all, I've had experience with Cisco 79XX phones before, but soon I will be moving to a new office and have the chance to purchase new equip. Can anyone make any recommendations about good SIP phones for the price to use with asterisk, or has anyone had experience with Polycom phones? |
22:26.53 | aptura | polycom are good |
22:27.07 | aptura | aastra are also good |
22:28.13 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
22:28.26 | afrosheen | yeah the polycom ip601's and 501's are nice |
22:28.44 | afrosheen | they're a pain in the ass to learn at the beginning but once you get provisioning down they're easy enough to work with |
22:28.51 | aptura | just beware that if you dont do line two right it may lockup. im trying to undoo mine right now. |
22:29.12 | aptura | mine is locking up now afrosheen |
22:29.29 | afrosheen | your polycom? |
22:30.17 | ShadowHntr | i'm considering grandstream phones here |
22:30.21 | BlepsoaF | hmm |
22:30.40 | ShadowHntr | the polycoms look nice |
22:30.43 | aptura | yea |
22:30.44 | ShadowHntr | but they're out of my price range |
22:30.46 | aptura | my ip500 |
22:31.05 | afrosheen | ShadowHntr, are you doing this for your home? |
22:31.15 | aptura | i added a second line..going out to my provider. I just deleted it and rebooting the phone. Hopfully this fixed it. |
22:31.16 | ShadowHntr | afrosheen: yeah - home and/or home office. |
22:31.27 | ShadowHntr | i've been researching |
22:31.32 | afrosheen | the ip300 or the newer 430 are probably what you want then |
22:31.41 | ShadowHntr | and it seems like the deal is VoicePulse |
22:31.51 | ShadowHntr | and maybe move to Voicepulse Connect for Asterisk later on |
22:32.17 | ShadowHntr | my ultimate goal is to put an IP phone in each room |
22:32.25 | ShadowHntr | total of 3 or 4 phones |
22:32.29 | ShadowHntr | using Asterisk |
22:33.20 | afrosheen | well if you're going to be cheap just buy some iaxy's from digium, plug a standard phone in |
22:33.30 | afrosheen | that way you get cordless phones also |
22:33.31 | ShadowHntr | nah i'd like them all to be SIP phones |
22:33.44 | afrosheen | well they'll be iax phones which is even better :) |
22:34.25 | ShadowHntr | well |
22:34.33 | ShadowHntr | i'm not gonna be ready for this for probably a few months |
22:34.37 | afrosheen | or if you want purely sip phones, the polycom 430 is around $149 each, 2 lines and a speakerphone |
22:34.41 | justinu|laptop | iaxy's aren't exactly cheap |
22:34.43 | *** join/#asterisk robin_z (n=you@adsl.redpoint.org.uk) |
22:35.10 | afrosheen | iaxy's are about $85 each |
22:35.23 | afrosheen | I think you should get at least one just for the portable phone aspect |
22:35.29 | justinu|laptop | i'd get a real SIP ATA for less |
22:35.56 | justinu|laptop | just for the fact that real ATAs can resolve a DNS name, and the iaxy can't |
22:35.59 | afrosheen | we have an iaxy we give to our road warriors, you never know what kind of firewall they'll end up with at whatever hotel or coffee shop |
22:36.39 | afrosheen | and you can trust that an iaxy will gaijin smash anything on it's way to the server |
22:38.50 | aptura | have your road warriers come up against any walls and could not get though? |
22:39.57 | aptura | This is really sucking. Man If I had the forsight of this phone locking up a head of time I would not have had to make these changes. |
22:40.39 | afrosheen | aptura, not yet, double nat is a joke for iaxy also |
22:41.28 | robin_z | just avoid Sipura SPA2102s when buying ATAs |
22:42.04 | robin_z | about as much fun as slamming your fingers in the desk drawer |
22:42.08 | aptura | okay i had to race though these menues on the phone before it locked up..a 10th time. |
22:42.24 | aptura | Okay looks like it may have worked. |
22:42.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
22:43.25 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
22:43.30 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
22:43.33 | iCEBrkr | <PROTECTED> |
22:43.36 | iCEBrkr | huh? |
22:43.54 | *** join/#asterisk alkiser (n=support@bdsl.66.14.163.224.gte.net) |
22:44.55 | kink0 | hello |
22:45.57 | iCEBrkr | Anyone else having problems with VoicePulse? |
22:47.59 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
22:48.24 | iCEBrkr | hrrm, apparently, it's fixed now |
22:48.32 | iCEBrkr | It just said it's reachable |
22:48.42 | iCEBrkr | but I'm still getting Got SIP response 481 "Call/Transaction Does Not Exist" back from 2 |
22:49.54 | aptura | new issues? |
22:50.34 | *** join/#asterisk chief3rd (i=ExUser@cuscon30833.tstt.net.tt) |
22:50.46 | TrixVox | iCEBrkr: Working okay here... looks like they just lowered their rates or something though. |
22:51.01 | iCEBrkr | It seems as if they just fixed it |
22:51.21 | iCEBrkr | But whatever that Call/Transaction does not exist message is new to me. |
22:51.37 | iCEBrkr | I can make calls |
22:51.44 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
22:51.45 | iCEBrkr | oh well |
22:52.21 | TrixVox | nice, most of europe was lowered... canada too |
22:56.01 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
22:56.14 | TripleFFFF | got booted off lol |
22:58.12 | *** join/#asterisk vilito (n=lsackett@68.80.4.115) |
23:01.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
23:01.04 | *** mode/#asterisk [+o anthm] by ChanServ |
23:02.56 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
23:03.24 | *** join/#asterisk spr1te (n=spr1te@quarta.synapse.net.ua) |
23:07.07 | fiber0pti | Anyone want to give me feedback on a java operator panel? |
23:07.22 | rpm | fiber0pti: java operator panel? |
23:07.25 | TripleFFFF | hate hjava |
23:07.49 | fiber0pti | rpm: Nod. It runs on java.. both server and client |
23:07.58 | fiber0pti | TripleFFFF: What operator panel do you use? |
23:11.03 | TripleFFFF | ? vi |
23:11.13 | TripleFFFF | vi is the best panel there is.. |
23:11.18 | TripleFFFF | its like notepad but better |
23:11.32 | TripleFFFF | and get the job done with less cpu then any other means |
23:13.22 | *** join/#asterisk AdmoIRC (n=Miranda@user-0c93s2v.cable.mindspring.com) |
23:15.01 | vilito | hi all, i have an interesting problem. i built and install asterisk 1.2.12.1 and everything works fine except that everytime i reboot, the /var/run/asterisk dir is removed |
23:15.34 | vilito | i just did a df and i see that varrun is not on the regular fs, but seems to be like a tmp fs. |
23:16.07 | vilito | my question is, how do i prevent or ensure that /var/run/asterisk is always present before asterisk runs? |
23:16.49 | *** join/#asterisk linuxmigration_ (n=jeremy@dsl254-075-124.nyc1.dsl.speakeasy.net) |
23:19.02 | *** part/#asterisk robin_z (n=you@adsl.redpoint.org.uk) |
23:20.27 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
23:27.12 | iCEBrkr | isn't /var/run/asterisk the lock file so it knows that it IS running |
23:31.35 | aptura | man people can be like sheep not so observant that the tenant was dead for three years in the apt and no one knew. The persons pension continiosly paid the rent. |
23:34.14 | *** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za) |
23:37.16 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
23:38.37 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
23:40.04 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
23:42.51 | BlepsoaF | vilito: I'm not sure why you would wnat to do that, but write a wrapper script in bash |
23:50.53 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
23:50.55 | EyeCue | asterisk-1.2.9.1_1 < needs updating (port has 1.2.12.1) |
23:50.56 | EyeCue | ooo :D |
23:56.51 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
23:57.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
23:58.38 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
23:58.40 | BlepsoaF | how is 1.4 anyhow |
23:58.54 | BlepsoaF | hopefully it goes production ready by the time I need to move into my new office |
23:59.34 | VOIPoTD | Any consultants here or someone with a few hours know AEL2 very well? |
23:59.43 | VOIPoTD | If so, pm me |