00:01.37 | voxtop | JT: yeah |
00:02.15 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
00:02.28 | voxtop | JT: with "opermode=AUSTRALIA" in /etc/modprobe.d/zaptel and with "wctdm" in /etc/modules , wctdm loads fine. the problem is that asterisk fails to start |
00:03.31 | JT | have you tried reverting to plain wctdm with no opts? |
00:04.16 | voxtop | JT: yeah, and asterisk starts fine then |
00:04.57 | JT | can you pastebin the verbose errors? |
00:07.00 | voxtop | JT: i can't reboot the box at this point to generate the errors, but let me see if i can find my paste from last night. sec.. |
00:07.17 | JT | can't you just start asterisk |
00:07.59 | *** join/#asterisk lucasjb (n=lbarbuto@mail.stabat.com) |
00:09.25 | tzafrir_laptop | voxtop, /etc/zaptel/1 generated? |
00:09.26 | voxtop | JT: found the errors: http://rafb.net/paste/results/e4WnIi76.html |
00:10.00 | voxtop | tzafrir_laptop: hi again :) |
00:10.16 | voxtop | tzafrir_laptop: /dev/zap/{1-4} exist |
00:10.37 | tzafrir_laptop | ztcfg is run successfully? |
00:10.45 | HumpBack | voxtop: try to dmesg|less and at tho botton see if there are any errors |
00:10.48 | lucasjb | Hiyas, I have a user running an IAX soft phone on Windows XP connecting over the Internet (from a different counry) to my Asterisk server. He's complained recently that the audio signal fades in and out constantly with a regular frequency. Can anyone suggest what a likely cause of this might be? |
00:12.30 | voxtop | tzafrir_laptop: yes, ztcfg ran successfully |
00:13.05 | voxtop | HumpBack: unfortunately i can't at this point, as the * box in question is being used and can't be taken down for several hours |
00:13.47 | Druken | does anyone get NAME CID on a pri ?? :) |
00:15.28 | HumpBack | voxtop: because i have a similar problem where ztcfg -vv reports all ok but i cannot use the fxo ports |
00:15.35 | tzafrir_laptop | voxtop, in your zaptel.conf you remmed-out channel 1? |
00:15.54 | tzafrir_laptop | what do you see on /proc/zaptel/1 ? |
00:16.02 | HumpBack | voxtop: s/fxo ports/fxs ports |
00:16.09 | HumpBack | voxtop: and i get errors in dmesg |
00:16.49 | HumpBack | voxtop: http://pastebin.com/824079 |
00:17.46 | voxtop | HumpBack: i only have fxo ports |
00:17.55 | voxtop | HumpBack: never played with fxs ports |
00:18.00 | tzafrir_laptop | HumpBack, did you write to asterisk-users ? |
00:18.27 | HumpBack | tzafrir_laptop: yes I did |
00:18.30 | HumpBack | http://lists.digium.com/pipermail/asterisk-users/2006-November/172459.html |
00:18.33 | HumpBack | That's me |
00:18.39 | tzafrir_laptop | You don't seem to have similar problems. You seem to have physical problems with the card, or similar lower-level problems in the system |
00:18.40 | voxtop | tzafrir_laptop: /proc/zaptel/1 : http://rafb.net/paste/results/KGtkFQ81.html |
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00:19.18 | HumpBack | tzafrir_laptop: i already tried two cards. seems really strange for both to have the same issue |
00:19.21 | tzafrir_laptop | voxtop, in use? Are you trying to run a second instace of asterisk? |
00:19.29 | voxtop | tzafrir_laptop: note that this is with "options wctdm opermode=AUSTRALIA" in /etc/modprobe.d/zaptel and *without* "wctdm" in /etc/modules |
00:19.39 | tzafrir_laptop | HumpBack, time to blame the motherboard, I guess |
00:20.20 | tzafrir_laptop | HumpBack, seriusly, I don't know those cards well enough to give you a good answer |
00:20.49 | tzafrir_laptop | voxtop, the card is listed as 'in use". Someone else, probably a different astersk, uses it |
00:20.58 | tzafrir_laptop | ps aux | grep asterisk |
00:21.21 | tzafrir_laptop | Do you run xen or a similar virtulized setup? |
00:22.01 | voxtop | tzafrir_laptop: i should only have one instance of asterisk running: http://rafb.net/paste/results/NxH5UA51.html |
00:22.20 | voxtop | tzafrir_laptop: no virtualisation; just a standard box |
00:22.28 | HumpBack | tzafrir_laptop: hmmm the only other system i have right now has 4 of the 5 pci slots in use. And i dont know if i can use the last due to irq sharing with the vga card |
00:23.26 | tzafrir_laptop | voxtop, just wondering: you have kernel 2.4, right? |
00:23.33 | voxtop | tzafrir_laptop: 2.6.16 |
00:23.49 | voxtop | tzafrir_laptop: on sarge |
00:23.55 | tzafrir_laptop | so how come we see so many asterisk processes? |
00:24.21 | tzafrir_laptop | That command shouldn't show separate threads differently, AFAIR |
00:25.14 | ucfMethod | night everyone, seeya tomorrow |
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00:27.28 | voxtop | tzafrir_laptop: what does `ps auxf` list on your asterisk boxes? |
00:28.15 | tzafrir_laptop | just a single process |
00:28.36 | voxtop | tzafrir_laptop: even with the 'f' argument? |
00:28.56 | tzafrir_laptop | Could you run it again to see if the processes remain the same? 'f' shows processes as a tree (somewhat like pstree) |
00:31.42 | voxtop | tzafrir_laptop: results from running ps again: http://rafb.net/paste/results/dC4S6967.html |
00:32.53 | tzafrir_laptop | Looks like different threads of the same process (they have exactly the same size) |
00:33.03 | tzafrir_laptop | I still wonder why they are all displayed |
00:34.28 | HumpBack | voxtop: stop the asterisk server and see if you still get processes in ps |
00:35.21 | HumpBack | voxtop: forget what i told makes no sence |
00:36.16 | HumpBack | voxtop: run /lib/libc.so.6 and past the result somewhere |
00:38.10 | JT | mm not bad |
00:38.11 | JT | --- Results after 7252 passes --- |
00:38.11 | JT | Best: 100.000000 -- Worst: 99.816895 -- Average: 99.994708 |
00:52.03 | *** join/#asterisk voxtop (n=voxtop@210.9.120.138) |
00:52.21 | voxtop | hi guys, sorry i disappeared. the keyboard on my laptop died :P |
00:53.58 | voxtop | HumpBack: you asked me to run /lib/libc6.so.6 . however, that file doesn't exist on my * box |
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00:59.10 | HumpBack | voxtop: you should have some /lib/libc*.so or something |
00:59.17 | HumpBack | voxtop: run it |
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01:03.10 | SavageOne | #ext does an unattended transfer, how do i do an attended transfer? |
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01:03.57 | dovid | ~centosbug |
01:03.58 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
01:06.16 | voxtop | HumpBack: what will that do? |
01:06.46 | HumpBack | voxtop: it will "run" the libc that will report some information on the version and extensions |
01:06.53 | SavageOne | hey ll |
01:06.55 | SavageOne | all |
01:07.04 | SavageOne | just trying to find what I need to dial to do an attended transfer |
01:07.09 | HumpBack | voxtop: it is to see if you are using linuxthreads ou the new NPTL |
01:07.14 | SavageOne | or to make it so that # always does attended instead of unattended |
01:07.29 | voxtop | HumpBack: http://rafb.net/paste/results/sF6ml424.html |
01:08.11 | HumpBack | voxtop: linuxthreads-0.10 by Xavier Leroy That is some very VERY old libc |
01:08.43 | HumpBack | and no NPTL. That explains why the threads apper in ps aux |
01:09.14 | HumpBack | voxtop: What distribution are you using? |
01:09.46 | voxtop | HumpBack: Debian 3.1 (Sarge) with 2.6.16-2-686-smp |
01:10.38 | HumpBack | voxtop: strange. I tought sarge already had nptl. Search in the packages to see if there is a libc that uses nptl |
01:10.43 | *** part/#asterisk SavageOne (n=jkhgk@66.212.194.20) |
01:10.59 | voxtop | HumpBack: is it important to use NPTL? |
01:11.24 | HumpBack | voxtop: From a performance perspective the diference in terms of cpu and memory is HUGE |
01:11.26 | voxtop | HumpBack: /lib/libc.so.6 links to libc-2.3.6.so |
01:11.33 | voxtop | ah |
01:11.52 | HumpBack | from the asterisk side I do not know |
01:12.34 | HumpBack | In theory if one uses thread semantics to code the application there should be no diference between linuxthreads and nptl |
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02:03.39 | Katty | allo. |
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02:06.06 | justinu|laptop | mew |
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02:09.12 | Katty | justinu|laptop: (= |
02:13.29 | file | ack it's a mitcheloc |
02:13.51 | file | yay House |
02:17.19 | voxtop | do you set rxgain and txgain for all zap channels, or for each one? |
02:19.45 | aydiosmio | my Zap channels are riding spinnas |
02:19.55 | aydiosmio | twentyfours |
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02:33.59 | voxtop | i changed rxgain from 18 to 68, restarted asterisk, and don't hear any change in the volume. why might that be? |
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02:39.39 | sevard | So |
02:39.51 | sevard | Since when is libIAX a seperate library? |
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02:45.11 | brookshire | i think there is a separate libiax |
02:45.17 | brookshire | i didn't realize it was included |
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02:59.26 | Juggie | libIAX is for writing iax clients. |
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03:09.58 | droops | hey with asterisk 1.2.13, is there a problem with meetme not terminating when no calls are currently going on? |
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03:37.09 | SomethingISODD | hello all question is there a bug in 1.2 that stops people from being about to join a queue? |
03:37.27 | SomethingISODD | i keep getting a message when a person calls saying can not join queue |
03:39.52 | jaike | SomethingISODD: check joinempty and maxlen configuration in queue.conf |
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03:40.50 | SomethingISODD | jaike i have joinempty commented out and maxlen = 0 |
03:41.51 | SomethingISODD | jaike do you see any issue? |
03:41.56 | jaike | nope |
03:42.19 | SomethingISODD | I will paste the queue and extension for that in pastebin, if you could please take a look |
03:42.27 | jaike | SomethingISODD: the member agents cant join the queue? or calls wont go into the queue |
03:42.30 | SomethingISODD | i have been trying to figure this out for two days now with no luck |
03:42.39 | fholmes | Anyone know of any click to call projects working through asterisk? |
03:42.40 | SomethingISODD | calls can`t join the queue |
03:44.39 | jaike | SomethingISODD: try setting joinempty=yes and leavewhenempty=no |
03:45.16 | SomethingISODD | jaike if you have a second i pasted it all |
03:45.17 | SomethingISODD | http://pastebin.com/824714 |
03:45.33 | jaike | fholmes: were doing it, php with manager api |
03:46.09 | fholmes | really? cool. Can you tell me what your using it for? |
03:47.56 | jaike | website, customers enter their tel num and when they click dials their phone and connects to an agent |
03:48.08 | jaike | and vice versa |
03:48.45 | jaike | SomethingISODD: i think you have to explicitly set joinempty and leavewhenempty |
03:48.57 | SomethingISODD | oh ok let me try |
03:49.01 | fholmes | Cool. I was just thinking of maybe making something for affiliates to use to track phone calls made instead of lead forms filled out. |
03:49.20 | SomethingISODD | jaike do i have to set the agent in there as well? |
03:49.29 | SomethingISODD | like i did or can i do the login through the extensions |
03:49.59 | JT | real click to call applications used web soft phones i thought |
03:50.07 | JT | entering a phone number is liable to abuse |
03:51.52 | jaike | JT: its on our website but calls coming in is very rare, in case the customer is to lazy to call us up |
03:52.28 | JT | heh |
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03:52.53 | JT | yeah but it has the potential for misuse, eg. someone putting in the phone number of someone they wish to harras |
03:53.13 | jaike | JT: hmm..good point |
03:53.33 | jaike | SomethingISODD: how do you login your agents? AgentCallbackLogin? AgentLogin |
03:53.35 | fholmes | Yes, it would have a high potential for misuse. |
03:54.21 | SomethingISODD | let me confirm that second |
03:54.24 | JT | you'd at least need to make it have safeguards in place, like a register of numbers to not call, either manually entered, or automatically enterred if calls to one number are being made at a high rate |
03:54.43 | SomethingISODD | AgentCallbackLogin(|${CALLERIDNUM}@tech-queue) |
03:55.10 | fholmes | There would have to be a way to do it somehow with out a softphone on the callers end. |
03:56.38 | fholmes | I want to setup a self hosted application that could be used to track calls to a third party. |
03:56.46 | jaike | SomethingISODD: you have a tech-queue context? |
03:56.51 | SomethingISODD | yes |
03:57.01 | SomethingISODD | that was the queue i pasted i believe in pastebin |
03:57.17 | JT | fholmes: there would be no way that would be foolproof if you let them enter phone numbers to call |
03:57.24 | JT | unless it's a restricted website |
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03:57.27 | JT | that's fairly safe |
03:58.02 | jaike | SomethingISODD: i only see a [tech-queue] queuename, how bout tech-queue context in extensions |
03:58.13 | jaike | anyway try the settings first |
03:58.47 | jaike | has anyone done stress testing on 1.4? |
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04:04.10 | SomethingISODD | jaike ok its not ringing my phone, so when the caller logs in to the queue it has to be under an extension of [tech-queue] ?> |
04:04.21 | SomethingISODD | i mean in the extenstions.conf |
04:07.29 | jaike | the phone extension you are logged in should have an extension in [tech-queue] context, as per your agentcallbacklogin |
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04:12.41 | hd420 | can I use Astrisk in lieu of my vonage router? |
04:12.58 | hd420 | yes, I'm a newbie |
04:20.17 | aydiosmio | hd420: no, not with a standard residential account |
04:20.22 | aydiosmio | you need to use a spftphone account |
04:20.26 | aydiosmio | softphone |
04:20.40 | JT | do they restrict it in some way? |
04:20.41 | hd420 | I have a softphone account |
04:20.44 | hd420 | for travel |
04:20.48 | aydiosmio | there are several asterisk compatible providers though |
04:20.53 | aydiosmio | I'd suggest you switch |
04:21.20 | aydiosmio | hd420: then yes, voip-info.org has tutorials |
04:21.35 | aydiosmio | you can get he sip crednetials form the vonage website under your account |
04:22.03 | hd420 | I'm not sure how everything works |
04:23.13 | aydiosmio | you just set up a trunk with vonage's sip server (sphone.vopr.vonage.net:5061) and your sip credentials (login and password) with a few other specific setigns and you're set |
04:23.29 | aydiosmio | search voip-info.org for vonage |
04:23.39 | hd420 | as indicated above, <hd420> yes, I'm a newbie |
04:24.21 | aydiosmio | okay |
04:24.24 | hd420 | what's a "trunk"? |
04:24.27 | aydiosmio | so you need me to do the goole search for you then? |
04:24.33 | aydiosmio | http://www.voip-info.org/wiki/view/Asterisk+and+Vonage |
04:26.25 | JT | aydiosmio: i'm curious, how do vonage restrict non softphone accounts from connecting to asterisk? |
04:27.10 | aydiosmio | they do not disclose the SIP credentials for their ATA accounts |
04:27.52 | JT | the sip secret? |
04:28.41 | aydiosmio | that's part of it |
04:29.01 | JT | sip username too? |
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04:35.42 | aydiosmio | sip username for softphone accounts is the phone number |
04:35.46 | aydiosmio | not sure about ATA accounts |
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04:36.19 | *** mode/#asterisk [+o Qwell] by ChanServ |
04:37.32 | hd420 | I'm missing something |
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04:38.11 | hd420 | so asterisk is a vonage add-on, in lieu of a vonage replacement? |
04:39.05 | JT | aydiosmio: hrm ok, well otherwise what's the other part? |
04:39.29 | JT | hd420: asterisk is not a vonage add on |
04:39.46 | hd420 | JT: then? |
04:39.59 | hd420 | I have Vonage and a softphone here |
04:39.59 | JT | asterisk is an open source telephony toolkit |
04:40.11 | JT | often used to mak pbxs in software |
04:40.15 | hd420 | JT: I can read as well as the next guy, mate |
04:40.29 | JT | don't get antsy with me |
04:40.30 | JT | you asked |
04:40.32 | JT | i answered |
04:40.37 | JT | if you know, why ask? |
04:40.52 | [TK]D-Fender | JT : I *like* the sound of my voice personally.... |
04:41.09 | JT | heh |
04:41.15 | hd420 | what does "telephony toolkit" mean? |
04:41.19 | [TK]D-Fender | </sarcasm> |
04:41.53 | hd420 | I have Vonage, and a softphone |
04:41.56 | JT | hd420: it allows you to interface with both voip channels and hardware analogue and digital telephony channels |
04:42.08 | JT | so you usually use it in a pbx or ivr capacity |
04:42.43 | [TK]D-Fender | hd420 : Lets sum this up differently : * is a PBX. You can have it process calls coming in from all sorts of interfaces (analog & digital lines, VoIP accounts, etc), and rprocess those calls any which way yo want including sending them out over any fo said technologies and do any kind of prcessing in between that you can come up with as well. |
04:42.56 | JT | hd420: what made you look to using asterisk? is there something in particular you were needing done? |
04:43.14 | [TK]D-Fender | hd420 : Just think of it as a supper-massively customizable PBX limited largely by only your imagination. |
04:44.14 | hd420 | I'd like to be alerted of new voice mails on my mobile phone |
04:44.21 | [TK]D-Fender | hd420 : If you have a soft-phone account with Vonage, it may be possible to use it with * and let * handle voicmail, and offer other routing options to your callers like "conditional fololow-me" and other intelligent stuff |
04:44.39 | JT | hd420: that's actually not very easy to do i think, unfortunately |
04:45.05 | JT | due to voicemail being handled by your mobile |
04:45.08 | [TK]D-Fender | hd420 :That is quite possible with *. You can have * take your VMS for you and upon receipt of a new one send you an SMS message or even e-mail it to you. |
04:45.18 | JT | ahh |
04:45.26 | JT | unless i'm looking at this from the other way around |
04:45.37 | [TK]D-Fender | JT : you are :) |
04:45.47 | hd420 | JT: TKD is explaining how to do what I'd like |
04:46.03 | [TK]D-Fender | JT : But good of you to note that potential issue before it arouse and caught us by surprise ;) |
04:46.19 | JT | [TK]D-Fender: the way he said it could have been taken either way :) |
04:46.51 | [TK]D-Fender | JT : Seemed clear to me, but I'm *special* (thats why they send me the little bus) |
04:47.38 | JT | [TK]D-Fender: i've heard people in here ask for asterisk solutions to check their mobile phone's voice mail |
04:47.41 | JT | not pretty :P |
04:48.32 | hd420 | when a new VM comes in, I want * to send me an SMS saying "new voice mail from +############" |
04:48.36 | hd420 | that's it |
04:48.42 | [TK]D-Fender | JT : Yeah, that would be an extremely daunting task who's requirements would be deemed "not remotely reasonable to validate the wasting of time on" |
04:48.45 | hd420 | is * overkill for this purpose? |
04:48.53 | [TK]D-Fender | hd420 : Pretty easy. |
04:49.26 | [TK]D-Fender | hd420 : not overkill, but try to paint the bigger picture. How is the call going to land on *? |
04:49.39 | hd420 | through the softphone number? |
04:50.01 | [TK]D-Fender | hd420 : (as in : from where? Will * ONLY be taking VM, not performing other duties, etc?) |
04:50.19 | JT | vonage |
04:50.35 | hd420 | TK: it may perform other duties, but at this point, I see no other use for it |
04:51.34 | [TK]D-Fender | hd420 : is your soft-phone account independant of any other accounts you may have? Are you planning on treateing it solely as a VM box in the end? |
04:52.47 | hd420 | I use it when/if I'm on wifi at a cafe or something, where there's no wired connection available or it's inconvenient |
04:53.04 | [TK]D-Fender | hd420 : that will pose a porblem then |
04:53.09 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
04:53.21 | hd420 | Fender: it's not billed independantly |
04:53.41 | [TK]D-Fender | hd420 : If * registers to Vonage as your soft-phone, and then you log in somewhere else remotely, it will "steal" the account away from your * box and it won't get calls anymore. |
04:54.01 | JT | [TK]D-Fender: so he will have to connect to * when he's out and about |
04:54.15 | [TK]D-Fender | JT : That would be the way to do it. |
04:54.31 | hd420 | [TK]D: that sounds simple enough |
04:55.22 | hd420 | I could just get rid of the softphone from Vonage and use some other provider's |
04:55.24 | [TK]D-Fender | hd420 : You could as JT suggested connect * to your Voange soft-phone account, and you would then connect a softphone on a laptop,etc to *. * would then send calls placed through it OUT your Vonage accoun for you. |
04:56.00 | hd420 | that would work fine, Fender |
04:57.10 | hd420 | does t-mobile USA have Internet-email to mobile phone? |
04:57.27 | hd420 | or the other GSM provider? |
04:57.33 | hd420 | (Singular?) |
04:59.14 | [TK]D-Fender | hd420 : Ok, well now that I've confirmed that * may be up to the task you'll need to go about downloading and learning. it. |
04:59.46 | hd420 | Fender: yes, I'm beginning to think it may be overkill for the purpose though |
05:03.06 | Qwell | hd420: cingular |
05:03.23 | Qwell | (that wasn't an answer to your question...just fixing the spelling :) |
05:04.55 | hd420 | Qwell: it was a question, many thanks |
05:05.24 | *** join/#asterisk aadilismail (n=adilisma@41-247-154-202.wol.net.pk) |
05:06.41 | saam | hi |
05:08.53 | riddlebox | can someone help me get the /etc/init.d/asterisk start command to work in ubunt? |
05:11.27 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tj3.cable.mindspring.com) |
05:19.51 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
05:19.51 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
05:19.53 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
05:20.11 | linagee | hahaha. cool. i am nearing completion to a small project to be able to enter in names to asterisk using DTMF! :o) |
05:22.00 | *** join/#asterisk sahafeez (n=sahafeez@m0b0e36d0.tmodns.net) |
05:22.36 | [TK]D-Fender | linagee : "show application directory" |
05:22.49 | linagee | [TK]D-Fender: huh? |
05:23.11 | [TK]D-Fender | linagee : Effectively what you described, including reading the name back |
05:23.14 | linagee | [TK]D-Fender: names. |
05:23.20 | linagee | [TK]D-Fender: people's names. not people in the directory |
05:23.27 | voxtop | when you do 'sip show peers' and it says the status is "Unmonitored", how can you change the status? |
05:23.31 | linagee | [TK]D-Fender: arbitrary names |
05:23.32 | olinux | he probably just rendered all your hard work and planning useless |
05:23.43 | linagee | no |
05:23.49 | [TK]D-Fender | linagee : If you're referring to doing a "pure" entry tool for unknown names,t hen well yeah, thats kinda neat |
05:23.50 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
05:23.59 | linagee | [TK]D-Fender: yes |
05:24.06 | linagee | [TK]D-Fender: i found a big long list of names on the net |
05:24.36 | linagee | [TK]D-Fender: so maybe it doesn't have strange aremeic names, but you can always default and have it say, "we don't know wtf you're trying to say, please speak it instead" and just record. |
05:24.59 | linagee | [TK]D-Fender: it has every name i have thrown at it from friends. not from other countries though, that can get weird. |
05:25.27 | linagee | [TK]D-Fender: similar to how cellphones can guess what words you are trying to say |
05:25.43 | *** part/#asterisk lucasjb (n=lbarbuto@mail.stabat.com) |
05:26.31 | hd420 | hmm... i wonder if I can script the Vonage site using Perl and then just get that to send out the SMS to my mobile |
05:26.57 | hd420 | instead of futzing with asterisk |
05:27.42 | linagee | cool. works with my first name. :) |
05:27.44 | linagee | i get one match |
05:27.51 | linagee | with my last name, it comes up with two matches |
05:28.06 | linagee | anyone want to throw a name in numbers at me and see what it comes up with? :) |
05:28.08 | JT | voice recognition, or dtmf entry? |
05:28.12 | linagee | JT: dtmf |
05:28.26 | linagee | JT: give me some numbers. :) |
05:29.04 | linagee | a male name, i haven't gotten to female yet. (will take a second or two) |
05:30.26 | hd420 | Juliet |
05:30.44 | linagee | hd420: give me the DTMF numbers. that's how it would be when you call in |
05:30.48 | linagee | hd420: and a male name. :p |
05:30.48 | hd420 | linagee: 236 |
05:30.59 | linagee | BEN. :o) |
05:31.01 | hd420 | "Ben", I hope |
05:31.04 | linagee | yup |
05:31.07 | linagee | only one match |
05:31.26 | linagee | if it comes up with more than one, i would have to say, push 1 for Ben, push 2 for ___, etc. |
05:31.37 | linagee | hd420: another! :o) |
05:31.53 | hd420 | 42726 |
05:32.08 | linagee | unknown....? |
05:32.14 | linagee | what was it? |
05:32.20 | hd420 | "Hasan" |
05:32.23 | DrkShdw | linagee: 5333 |
05:32.24 | techie | haha |
05:32.25 | linagee | that's a name? |
05:32.34 | hd420 | it's my name, linagee |
05:32.35 | linagee | DrkShdw: jeff. :) |
05:32.57 | hd420 | 46726 |
05:33.02 | linagee | this is cool. it's working. :) |
05:33.07 | DrkShdw | linagee: 627846 |
05:33.14 | linagee | hd420: another empty set |
05:33.20 | hd420 | "Imran" |
05:33.28 | linagee | DrkShdw: martin or marvin |
05:33.34 | hd420 | 6999 |
05:33.35 | linagee | DrkShdw: both are possibilities. :) |
05:33.37 | hd420 | ? |
05:33.46 | linagee | hd420: another empty set. wtf |
05:33.56 | linagee | every name you have given me is weird. hah |
05:34.03 | hd420 | "Ozzy" |
05:34.08 | DrkShdw | not wierd, just not "english" |
05:34.10 | linagee | hd420: it would have to default and say, "wtf? please speak your name" |
05:34.20 | hd420 | Ozzy isn't English? |
05:34.24 | linagee | no |
05:34.36 | linagee | hd420: nobody names their kid ozzy |
05:34.43 | linagee | maybe as a nickname |
05:34.51 | linagee | but not the name on their birth certificate. :p |
05:34.54 | JT | linagee: 22784656639 |
05:35.09 | linagee | JT: nothing...? |
05:35.17 | JT | bartholomew |
05:35.18 | linagee | what was it? |
05:35.36 | linagee | i only have bart and barton. hrmph |
05:35.42 | JT | i purposely used something esoteric :P |
05:35.47 | hd420 | 983883 |
05:35.52 | SomethingISODD | question |
05:35.54 | hd420 | 986663 |
05:35.57 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
05:36.03 | SomethingISODD | is there wanyway to restrict asterisk to only use one ip |
05:36.04 | linagee | my male name database only has 1,500 or so names. like i said, it's suprisingly gotten every friends name i could come up with. |
05:36.07 | SomethingISODD | and not take them all over? |
05:36.20 | DrkShdw | linagee: 25273623 |
05:36.23 | hd420 | 923 |
05:36.40 | linagee | DrkShdw: clarence. :) |
05:36.54 | linagee | DrkShdw: i could not have done that many numbers by hand that fast. :) |
05:37.03 | JT | 628426435 |
05:37.18 | linagee | JT: nathaniel |
05:37.19 | linagee | :) |
05:37.25 | JT | yeah |
05:37.27 | linagee | that's an odd one |
05:37.32 | DrkShdw | linagee: I know, I remember playing with this in the other channel. the female DB is even more impressive |
05:37.43 | linagee | DrkShdw: i have a female DB too. it's a bit bigger. |
05:37.47 | JT | an old long version of nathan |
05:37.51 | linagee | give me a few seconds to do the dtmf cache |
05:37.53 | hd420 | linagee: 983883 (female) |
05:37.57 | linagee | sec |
05:38.04 | hd420 | linagee: sure |
05:38.15 | linagee | creating cache... |
05:38.17 | linagee | done. :) |
05:38.20 | linagee | it's that fast. heh |
05:38.40 | linagee | hd420: wtf? |
05:38.48 | hd420 | wtf wtf? |
05:38.49 | linagee | YUETTE or YVETTE |
05:38.52 | linagee | who has those names. hahaha |
05:38.59 | hd420 | my fiancee is named Yvette |
05:39.03 | linagee | i would guess it's the second one |
05:39.12 | linagee | yuette is just even more weirder. heh |
05:39.33 | linagee | so yep. now i have my female db too. :) |
05:39.36 | hd420 | 98663 (female) |
05:39.49 | linagee | yvone |
05:39.58 | linagee | hd420: the female db is a bit larger |
05:40.08 | SomethingISODD | ?? |
05:40.09 | *** join/#asterisk kph100 (n=kph100@206-248-157-22.dsl.teksavvy.com) |
05:40.36 | linagee | hd420: i plan on running the number through both DBs and saying like, "are you male or female" (for male) and "are you female or male" (for female names) |
05:40.37 | linagee | heh |
05:40.51 | kph100 | anyone can help me setup cdr mysql? |
05:40.55 | linagee | or maybe like, "you're male right? 1)yes, 2)no" |
05:40.59 | linagee | something like that |
05:42.07 | linagee | it would be "fun" to put unknown callers through this. similar to the freepbx privacy manager. |
05:42.20 | linagee | "we don't have a name on file for you..." hah |
05:42.42 | [TK]D-Fender | linagee : Yes, and I've know sever male's witht he name "Lindsay", etc. |
05:42.57 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:43.01 | linagee | [TK]D-Fender: i was thinking transexuals, that's why you just can't assume. hrm |
05:43.02 | JT | linagee: yvette is not an uncommon name |
05:43.13 | linagee | JT: i think i've heard it before. |
05:43.14 | JT | your idea of common names is pretty narrow |
05:43.18 | linagee | my sister had a friend named |
05:43.21 | linagee | JT: hehehe |
05:43.47 | JT | be funnier to replace the names dbs with noun dictionary |
05:43.49 | linagee | JT: my first name is 5 letters and my last name is 5 letters. :-D |
05:43.57 | JT | "did you mean 'toaster"" ARRGH |
05:44.01 | linagee | LOL |
05:44.01 | hd420 | mine is 5 and 5 as well |
05:44.02 | linagee | toaster |
05:44.22 | linagee | JT: hey, that might be a fun game |
05:44.24 | JT | or if it's someone you don't like |
05:44.29 | JT | when it detects their name |
05:44.29 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
05:44.30 | linagee | JT: make an adlib for asterisk. :-D |
05:44.31 | hd420 | 242442 (femalr) |
05:44.33 | JT | "did you mean fuckwit?" |
05:44.37 | hd420 | 242442 (female) |
05:44.45 | linagee | hd420: first unknown |
05:44.58 | linagee | and second is the same |
05:45.14 | [TK]D-Fender | plenty of Yvette's around here :) |
05:45.15 | hd420 | that's because there's no difference in the numbers :) |
05:45.20 | JT | linagee: are you going to be releasing the code? |
05:45.30 | hd420 | but... it's my sister's name "Aicha" |
05:45.41 | linagee | JT: i got the DB from US census data. hehehe. :-D |
05:45.50 | linagee | JT: it took forever to think of who might have that! |
05:45.56 | [TK]D-Fender | hd420 : Only heard that name once elsewhere. |
05:46.11 | linagee | JT: then i put all the names into a DB |
05:46.29 | linagee | JT: and i wrote a few lines to turn the names into DTMF. that way i can do the lookup easier |
05:46.40 | JT | right |
05:46.45 | linagee | pretty simple really. :) |
05:46.46 | JT | but what about the dialplan logic? |
05:46.55 | linagee | JT: haven't gotten there yet. :) |
05:47.04 | linagee | JT: i'm just pushing things at the DB. :) |
05:47.16 | linagee | select * from first_name_male where dtmf="123"; |
05:47.30 | linagee | er, there is no 1 |
05:47.32 | linagee | 234 |
05:47.45 | linagee | and it tells me there is no 234 match. :) |
05:48.01 | JT | does the table have the name in ascii too? |
05:48.12 | linagee | in ascii? |
05:48.18 | JT | text |
05:48.20 | linagee | yep |
05:48.23 | linagee | that's how i do the lookup |
05:48.24 | JT | not dtmf digits |
05:48.27 | hd420 | Fender: where'd you hear the name? |
05:48.30 | linagee | i have four columns |
05:48.35 | JT | i thought you said you converted it to dtmf digits |
05:48.37 | JT | ok |
05:48.39 | linagee | id, name, dtmf, timestamp |
05:48.59 | linagee | JT: i precached the dtmf. it makes the lookup instantaneous. :) |
05:49.16 | linagee | i took the names, turned them into dtmf, stored that in the db. that way, i can query that column. :) |
05:49.24 | JT | i reckon the neatest way to do it would be to have some code in a stored procedure in the db to convert tthe name to dtmf |
05:49.28 | JT | if you often update the db |
05:49.35 | linagee | instead of having to do some weird logic of creating a bunch of possibilieites |
05:49.37 | JT | if the db never changes, it wouldn't help |
05:49.52 | [TK]D-Fender | Stored procedule on DB add <- |
05:49.54 | linagee | JT: it's mysql. no stored procedures. :p |
05:49.58 | [TK]D-Fender | procedure* |
05:50.10 | JT | it's mysql, the db can't do much useful :P |
05:50.10 | linagee | er, it's not mysql 5 i should say |
05:50.19 | [TK]D-Fender | linagee : Time to grow up and use PostgreSQL |
05:50.21 | linagee | JT: it does what i need it to do |
05:50.33 | linagee | [TK]D-Fender: i do use postgresql when i need something slower |
05:50.50 | linagee | like when it's transactional and money is involved or something |
05:50.50 | [TK]D-Fender | linagee : Then you should get a Cisco router for it too! ;) |
05:51.07 | linagee | [TK]D-Fender: same thing. when money is involved. :p |
05:51.20 | JT | if speed is the name of the game you can use sqlite or a flat file :P |
05:51.25 | linagee | [TK]D-Fender: wtf. why postgresql then? |
05:51.34 | linagee | [TK]D-Fender: why not oracle. hahaha. might as well go all the way |
05:51.47 | JT | oracle costs well, heaps |
05:51.48 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
05:52.01 | linagee | JT: mysql is one step above flat files. i don't have to worry about the db logic. :) |
05:52.18 | JT | you don't have to with DBD::CSV |
05:52.32 | linagee | JT: and not have indexes? :p |
05:52.42 | JT | heh |
05:52.58 | linagee | anyhow, it works |
05:53.11 | JT | wtf, 5pm |
05:53.15 | JT | stupid daylights savings time |
05:53.17 | linagee | 9:53pm |
05:53.34 | JT | linagee: living in yesterday :P |
05:53.58 | linagee | can anyone figure that one out? :) |
05:54.36 | JT | nile? |
05:54.37 | bluregard | umm, I'm guessing I can ignore the fact that the Dundi draft says it expires April of 05 right? |
05:54.40 | linagee | JT: mike |
05:55.02 | JT | the asterisk handbook is still a draft, lol |
05:55.34 | bluregard | just making sure. I figured since it was right from dundi.com that it was the most current. |
05:56.26 | *** part/#asterisk hd420 (n=hdiwan@c-71-202-20-219.hsd1.ca.comcast.net) |
05:57.45 | linagee | 1,500 names is not that much. heh |
05:57.55 | linagee | maybe i could make a recording for each. :-D |
05:58.12 | JT | text to speech |
05:58.25 | linagee | JT: doesn't that sound kind of crappy? |
05:58.51 | linagee | maybe i could have a custom pronunciation key for certain names it says weirdly. |
05:59.52 | JT | oh it might sound crappy heh |
06:00.05 | JT | but how does it currently prompt the user if there are multiple options? |
06:00.20 | linagee | JT: like i said, there is no prompting. |
06:00.32 | JT | what if there are multiple answers |
06:00.32 | linagee | JT: right now it's all theoretical beyond having a DB that has the names. :) |
06:00.39 | JT | oh ok |
06:01.00 | JT | recording 1500 name prompts is not an enviable task, anyway |
06:01.06 | linagee | i have a good idea of how it should work, just haven't gotten there yet |
06:01.19 | linagee | would that take forever? heh |
06:01.30 | JT | or lots of $$$ |
06:01.34 | linagee | JT: huh? |
06:01.39 | linagee | JT: why hire someone. :p |
06:01.46 | JT | to get them professionally recorded |
06:01.51 | JT | well it's one of the two |
06:01.54 | linagee | JT: but why professionally recorded |
06:02.08 | JT | and you'd have to fix yours up if you don't want them to sound like crap |
06:02.11 | linagee | nobody likes my voice? :( |
06:02.27 | linagee | my voice is already all over my ivr. hah' |
06:02.29 | JT | samplex beginning and ending at a zero crossing, consistant amounts of silence appended and prepended |
06:02.40 | JT | s/samplex/samples/ |
06:02.53 | linagee | jbot: you are annoying when you do that |
06:03.02 | jbot | linagee: I think you lost me on that one |
06:03.08 | JT | jbot: lart |
06:03.23 | JT | jbot: lart #asterisk jbot |
06:05.50 | linagee | LOL |
06:06.01 | linagee | jbot: lart #asterisk #asterisk |
06:06.28 | linagee | jbot: lart myself |
06:06.58 | JT | oh gawd, what have i done? teaching people infobot tricks |
06:07.02 | linagee | hehehe |
06:07.08 | linagee | jump jbot, jump |
06:07.22 | linagee | jbot: botsnack? |
06:07.22 | jbot | :), linagee |
06:10.25 | linagee | hah. the db has "napolean" |
06:10.29 | linagee | as if you'd be called that |
06:11.22 | linagee | Monty = 66689 |
06:11.28 | linagee | (Monty burns = evil?) |
06:12.01 | linagee | 66689 28767 |
06:12.24 | JT | hey napoleon, what did you say you did last summer? |
06:12.56 | linagee | 2326 7263537 |
06:12.58 | linagee | anyone? :) |
06:13.50 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
06:14.02 | JT | you sure it was napolean and not napoleon? |
06:14.11 | linagee | JT: yes it was napoleon |
06:14.15 | linagee | (sp) :) |
06:14.28 | hoobastooba | has anyone else ever experienced something like a memory leak in asterisk? it seems as if the longer asterisk runs, the less memory i have. |
06:14.50 | linagee | <PROTECTED> |
06:14.56 | Inverted | hoobastooba: the longer you live the less memory YOU have |
06:15.19 | linagee | Inverted: what if you lived forever? |
06:15.42 | Inverted | linagee: then I'd be one dumb & happy fucker :) |
06:15.42 | linagee | would your brain cells keep growing and expanding, or would knowledge just leak out? |
06:17.02 | hoobastooba | helpful... at the end of the day when asterisk is no longer in use, it still has all of my ram consumed and pushed me into swap. |
06:17.22 | Inverted | hoobastooba: that is how things are supposed to work with linux |
06:17.36 | hoobastooba | so after three days of asterisk use, my server should crash? |
06:17.51 | Inverted | you said nothing about crashing |
06:18.04 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
06:18.39 | hoobastooba | usually with linux, it reallocates the swap back to ram, but asterisk is consuming all of my ram. |
06:23.16 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
06:25.45 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
06:26.37 | *** join/#asterisk saftsack (n=saftsack@pD9E07F61.dip.t-dialin.net) |
06:27.28 | *** join/#asterisk mosty (n=mostynm@203.143.64.82) |
06:33.28 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
06:34.58 | mosty | anyone awake? i have a problem with realtime queues, it tries to send calls to lines that are busy, instead of waiting until there is a free agent. is this a bug, or something i have misconfigured? |
06:38.33 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
06:38.43 | gmustafa | hello |
06:39.21 | joelsolanki | Hello |
06:39.56 | joelsolanki | <PROTECTED> |
06:39.59 | joelsolanki | is this possible |
06:40.12 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
06:40.50 | linagee | joelsolanki: yes |
06:41.10 | linagee | joelsolanki: that's child's play with freepbx. with just straight asterisk, you will need voodoo. :) |
06:41.21 | joelsolanki | how ? |
06:41.31 | joelsolanki | i tried but it is not reflecting on my mobile :( |
06:42.01 | linagee | joelsolanki: does your ITSP allow you to set your outgoing callerid? |
06:42.22 | joelsolanki | Yes i can ask them |
06:43.05 | joelsolanki | they will allow callerid to pass |
06:43.15 | joelsolanki | then where to configure in freepbx? |
06:43.23 | linagee | are you using freepbx? |
06:43.26 | joelsolanki | yes |
06:43.36 | linagee | then you should be asking in #freepbx. :) |
06:43.40 | linagee | click the extension |
06:43.42 | linagee | it will be in there |
06:43.46 | joelsolanki | i asked :) |
06:43.49 | linagee | and? |
06:44.08 | joelsolanki | Yes no reply. they might be busy or sleeping |
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07:11.01 | zumbush | anyone got experiance with Cisco 7941G running with asterisk? |
07:12.10 | Qwell | zumbush: should work fine with chan_skinny on 1.4 :D |
07:12.28 | zumbush | ok..thx.. what if i wanna run SIP? |
07:13.10 | hads | Use chan_sip :) |
07:13.21 | zumbush | yea... hehe |
07:13.30 | Qwell | sip sucks on cisco... |
07:13.44 | zumbush | but im having trouble figuring out how to get a hold of the SIP firmware |
07:14.09 | VibroMax | anybody using a cisco 7960 SIP phone know if it is possible to have more than 4 lines configured on the phone. There are six physical line buttons on the phone and I have configured six lines in the phone specific configuration file but only four lines are active. Is the problem my firmware version (P0S302002) or a limitation of hardware ? |
07:15.05 | zumbush | hmm maybee i should try the chan_skinny then |
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07:16.27 | zumbush | thats smartnet shit i hate... why should i have to pay more money just to get firmware for a product i already bought..grr |
07:16.37 | zumbush | :-P |
07:18.56 | zumbush | so if i am to run chan_skinny i need to upgrade to Asterisk 1.4 ? |
07:19.13 | Qwell | "need"...yeah, pretty much |
07:19.25 | Qwell | 1.2 MIGHT work, but I would highly recommend upgrading |
07:19.36 | zumbush | k.. is the 1.4 considered stable? |
07:19.39 | Qwell | no |
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07:19.42 | zumbush | hehe |
07:19.42 | Qwell | it's still beta |
07:19.54 | zumbush | then i need to stick with 1.2 |
07:19.59 | Qwell | but, I don't disagree. Ciscos tactics are...meh |
07:20.07 | Qwell | They don't "get it" :) |
07:20.08 | zumbush | yea |
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07:20.58 | zumbush | it will bite them in the ass |
07:21.19 | Qwell | You'd be amazed if you saw the emails I've gotten from Cisco guys |
07:21.30 | zumbush | yea.. gimme an example |
07:21.44 | zumbush | short verison |
07:21.45 | zumbush | :-P |
07:21.55 | Qwell | "Only 1% of the phones we sell are not used with Call Manager, so we decided that it isn't worth the effort of making them compatible." |
07:21.59 | Qwell | ...uhh...BS |
07:22.05 | zumbush | gah |
07:22.22 | zumbush | stupid |
07:22.33 | zumbush | like saying we dont want to sell more phones |
07:23.25 | zumbush | i just have this one phone from Cisco that i bought and now cant use, rest is Polycom that ive gotten to like |
07:23.38 | Qwell | I asked him bluntly... |
07:23.57 | Qwell | "I own a Cisco 7960. I didn't pay for a license. Am I violating your license?" |
07:24.00 | Qwell | "yes" |
07:24.44 | zumbush | buy this phone for this much money.. but if u dont pay us more u just can use it as a paperweight |
07:24.52 | Qwell | when I asked what I should do; ie, should I sell it on ebay, he gave me no response |
07:25.01 | zumbush | lol |
07:25.10 | Qwell | funny, because he said that if you DID use it as a paperweight, you'd be legal :) |
07:25.18 | zumbush | hahaha |
07:25.29 | Qwell | I can give you a direct quote if you'd like |
07:25.36 | zumbush | do that |
07:25.41 | Qwell | one sec |
07:26.17 | Qwell | Quote 1: "The license is a right to use license, basically the right to operate |
07:26.17 | Qwell | the phone firmware that runs in the phones and register it to a Call |
07:26.17 | Qwell | Agent." |
07:26.44 | Qwell | Quote 2: "If you buy the phone as a spare, and use it to prop open the door, or |
07:26.44 | Qwell | weight down a pile of paper, that's fine. But if you want to operate |
07:26.44 | Qwell | the software in the phone, have it register and process calls, then |
07:26.45 | Qwell | the right to use license should be purchased. " |
07:26.51 | Qwell | Those are DIRECT quotes from a Cisco sales rep |
07:27.08 | zumbush | ROFL |
07:28.05 | Qwell | You'll love this one - re; 1% |
07:28.15 | Qwell | "As to how many people are using Cisco phones on a non Cisco Call Agent, |
07:28.15 | Qwell | it's a much smaller number than many people realize. We have measured this, |
07:28.15 | Qwell | and it comes out to less than 1% of our shipping phones. When we consider |
07:28.15 | Qwell | the resources required to maintain compatibility, and to verify interop |
07:28.15 | Qwell | with the different 3rd party Call Agents, it just doesn't make business |
07:28.18 | Qwell | sense. Yet. " |
07:28.40 | Qwell | The moment I read this, I had completely given up on Cisco |
07:29.06 | zumbush | as complying to SIP would be a waste of resources |
07:29.11 | Qwell | of course |
07:29.31 | zumbush | they are missing an expanding market |
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07:30.03 | Qwell | You're preaching to the choir(sp)... |
07:30.11 | zumbush | hehe |
07:30.20 | zumbush | just need to get it out of my chest |
07:30.34 | Qwell | My response, which was completely ignored: |
07:30.38 | Qwell | "Thanks for the answers - they are quite upsetting (and also quite amusing, to be honest. Cisco *still* just doesn't get it). Since Cisco obviously doesn't care about the "1%", I will no longer be recommending Cisco products for use with Asterisk or any other PBX or provider (I'd reconsider that, if Cisco were willing to stop the silly practices). In addition, should I be selling all of my Cisco hardware on eBay? It seems that I don't have a legal |
07:30.40 | Qwell | <PROTECTED> |
07:31.22 | zumbush | good answer |
07:31.52 | Qwell | and since then, I've acquired 3 more Cisco phones, he |
07:31.53 | Qwell | h |
07:32.09 | zumbush | uh...why? |
07:32.23 | zumbush | works good with chan_Skinny? |
07:32.31 | Qwell | because regardless of the asshattery, I believe skinny is a good protocol |
07:32.36 | zumbush | k |
07:32.37 | Qwell | (Which Cisco DID NOT come up with) |
07:33.16 | Qwell | I actually haven't tested the 2 latest phones.. I need to find a power adapter for them, or get a PoE switch or something |
07:33.30 | zumbush | il try the skinny then on my 1.2 |
07:33.36 | Qwell | don't bother, honestly |
07:33.49 | Qwell | the changes in chan_skinny between 1.2 and 1.4 were immense |
07:34.04 | zumbush | u dont happen to have a cfg example file ? |
07:34.13 | zumbush | k |
07:34.17 | Qwell | I'm biased, because I made about 90% of the changes, but it works a whole hell of a lot better now |
07:34.26 | zumbush | hehe |
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07:34.47 | zumbush | nice |
07:34.58 | Qwell | (which is why I find it extremely funny, but extremely pathetic, of Cisco to pull this BS) |
07:35.13 | zumbush | i can imagine |
07:35.40 | JT | make your own PoE injector :) |
07:36.28 | Qwell | JT: easier said than done... |
07:36.41 | Qwell | google 12SP+ and 30VIP :) |
07:37.01 | JT | it needs some stupid poe signalling does it? :P |
07:37.03 | Qwell | the 12SP I have is actually not even Cisco branded |
07:37.19 | Qwell | the 30VIP has a sticker OVER the Selsius(sp) branding |
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07:37.29 | Qwell | very, very old |
07:41.27 | Qwell | yeah...bed time... |
07:41.55 | zumbush | gn.. nice talking to u |
07:42.04 | jaike | anyone done stress testing on 1.4? am hoping it has better scalability than 1.2 |
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07:42.27 | Qwell | jaike: there is a new "Load Testing Task Force"... |
07:42.45 | Qwell | something like that anyhow. Consists of 3-4 people so far |
07:43.14 | saam | <PROTECTED> |
07:43.19 | saam | ? |
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07:46.22 | quelo | Hi to all |
07:47.26 | quelo | I have a problem is there anyone can help me? |
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07:48.11 | quelo | Hi |
07:48.13 | mazzanet | herro |
07:48.56 | mazzanet | is ${TIMESTAMP} deprecated in 1.4.0-beta3? |
07:49.17 | Qwell | think so, yes |
07:49.53 | mazzanet | that would explain the lack of a timestamp in my monitor files... |
07:49.55 | quelo | I have downloaded asterisk-addons by svn in trunk |
07:50.02 | mazzanet | what's its replacement? |
07:50.11 | quelo | but when I compile it... |
07:50.18 | quelo | it returns this errors... |
07:50.22 | Qwell | mazzanet: let's see |
07:51.06 | quelo | nemo:/usr/src/asterisk-1.2/asterisk-addons# make install |
07:51.15 | Qwell | mazzanet: probably something with ${STRPTIME()} |
07:51.31 | Qwell | erm, STRFTIME? something |
07:51.32 | quelo | http://paste.debian.net/16661 |
07:51.45 | Qwell | mazzanet: core list functions |
07:51.49 | mazzanet | have all been deprecated in favor of their related dialplan functions. |
07:52.10 | mazzanet | what the heck is a dialplan function? |
07:52.21 | Qwell | it's like a variable, but with args |
07:52.27 | quelo | is there anyone can help me? |
07:52.37 | Qwell | ie; ${CALLERIDNAME} is now ${CALLERID(name)} |
07:52.38 | mazzanet | ah i see |
07:53.42 | mazzanet | http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
07:53.44 | mazzanet | ah here we go |
07:54.02 | quelo | I'm using Debian/GNU Linux (etch) |
07:54.21 | quelo | oh no!!! I'm using Debian/GNU Linux (sarge) |
07:54.34 | mazzanet | interestingly, $CALLERIDNUM still works but $TIMESTAMP doesn't |
07:54.52 | Qwell | mazzanet: $CALLERIDNUM was deprecated in 1.2 I believe, but never removed |
07:58.34 | quelo | can I solve installing libtool? |
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08:01.30 | Zeeek | nyuk, nyuk |
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08:04.38 | quelo | I have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge) |
08:04.46 | quelo | http://paste.debian.net/16661 |
08:04.59 | quelo | How can I solve it? |
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08:12.23 | SoftIce | hi, can anyone help me with the snom 190 phone? |
08:12.49 | quelo | I have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge) |
08:12.52 | quelo | http://paste.debian.net/16661 |
08:12.56 | quelo | How can I solve it? |
08:13.49 | Zeeek | do you require h323 ? |
08:16.33 | Zeeek | people of earth attention |
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08:16.44 | Zeeek | look to you skies for a warning |
08:17.24 | quelo | Zeeek how can I solve? |
08:17.46 | Zeeek | no idea, I was asking if you need h323 |
08:17.59 | Zeeek | because if not, it may have something to do with the error |
08:18.46 | quelo | I need some dev libraries ? |
08:19.43 | quelo | maybe I need h323 |
08:20.06 | Zeeek | are you using h323 ? |
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08:21.33 | quelo | yes |
08:24.46 | linagee | today's random name is randal. :-D |
08:25.46 | Zeeek | quelo I can't help, sorry |
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08:45.14 | mazzanet | hmm |
08:45.16 | mazzanet | is the [globals] context deprecated too? |
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08:56.12 | badcfe | how do i see what context variables are available in some point of the dialplan. actually i need the ${VARIABLES} for AccountCode, |
08:56.29 | badcfe | Destination and StartTime |
08:57.30 | badcfe | ? |
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08:58.28 | Zeeek | there is a file called README.variables os something like that |
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09:06.34 | my007ms | ManxPower, are u there |
09:09.51 | Zeeek | at this hour? :) |
09:11.46 | my007ms | so u know him Zeeek |
09:11.47 | my007ms | :) |
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09:13.05 | Zeeek | ya |
09:13.21 | Zeeek | but he has been seen in these parts at weird hours |
09:13.23 | FTexcom | splitttttttttt |
09:13.44 | tzafrir | join? |
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09:14.12 | Zeeek | From time to time, I get wrong numbers said to come from something@123.456.123.456 |
09:14.34 | Zeeek | I think they come directly to a SIP phone |
09:14.47 | Zeeek | on a line not registered with asterisk |
09:15.19 | FTexcom | Zeeek you can disable anonymous sip calls |
09:15.23 | Zeeek | in fact, not something, but a US phone number like 17076253454@ |
09:15.34 | Zeeek | they come to the phone directly |
09:15.42 | Zeeek | I just wondered how and why? |
09:16.03 | Zeeek | I believe they are really wrong numbers, not hanckers or spammers |
09:16.25 | FTexcom | have you tried calling them back?¿ |
09:16.35 | Zeeek | yeah and the last time I did it was like "no I was calling the bank" |
09:16.53 | Zeeek | so I'm just curious what scenario would make that happen |
09:17.28 | Zeeek | also got one that said "I was calling another room" |
09:17.31 | Zeeek | in fact I think there were two that came from a hotel - it doesn't happen that often |
09:17.44 | FTexcom | you get them everytime? |
09:17.57 | Zeeek | get what everytime? The hotel? |
09:18.18 | Zeeek | it's often late at night (time diff) |
09:18.26 | Zeeek | so I don't usually answer |
09:19.03 | Zeeek | I should check the missed calls list and see if the ip is the same, but like I say it doesn't happen often |
09:19.26 | Zeeek | I was wondering if it had to do with a voip provider I'm registered to on that line |
09:20.05 | Zeeek | This channel is like confession, I think by describing it, I just figured out what it is :) |
09:20.39 | Zeeek | join asterisk-psychiatry-couch |
09:21.45 | FTexcom | it's a pretty funny thing |
09:21.50 | FTexcom | Some years ago |
09:22.06 | FTexcom | I had a debt with a ISP |
09:22.25 | FTexcom | they called me around 2 am with an automatic machine saying "you own us...49 euros" |
09:22.37 | Zeeek | great use for asterisk |
09:23.36 | FTexcom | my response was calling they tech support at 2 am saying "you owe two months of a DSL line!" |
09:23.49 | FTexcom | *owe me |
09:24.28 | Zeeek | because they actually had tech support? |
09:24.56 | FTexcom | 18 year old boys. Working 14 hours for 700 euros...if you want...you can call *THAT* tech support |
09:25.43 | Zeeek | sounds like the US Senate pageboys! Did Senators call them with obscene proposals? |
09:25.49 | FTexcom | lol |
09:26.45 | FTexcom | conservative senators you might add |
09:26.55 | Zeeek | glad that elections happened! |
09:27.06 | FTexcom | you're european like me right? |
09:27.52 | Zeeek | 50-50 USA/French |
09:28.03 | Zeeek | expatriate for 25 years |
09:28.14 | Zeeek | I even started voting |
09:28.14 | FTexcom | expatriate from where? USA or france? |
09:28.22 | Zeeek | US Expat in FR |
09:28.38 | Zeeek | asked for a got fr citizenship a couple years ago |
09:28.50 | FTexcom | you could always join the foreing legion! |
09:28.56 | Zeeek | too old |
09:29.08 | FTexcom | I almost did that... |
09:34.16 | Zeeek | wouldn't be cool for me :) |
09:35.20 | quelo | I have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge) |
09:35.24 | quelo | http://paste.debian.net/16661 |
09:35.33 | quelo | How can I solve it? |
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09:48.34 | shellshark | re |
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09:49.20 | Zeeek | oej |
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09:59.40 | tzafrir | quelo, what version of asterisk do you use? |
10:00.02 | tzafrir | You seem to use add-ons 1.2 and asterisk trunk? |
10:00.59 | tzafrir | that is: the other way around: asterisk 1.2 and addons from trunk |
10:06.39 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
10:09.58 | badcfe | how do i see what context variables are available in some point of the dialplan. actually i need the ${VARIABLES} for AccountCode, Destination and StartTime. |
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10:24.15 | skyhawker | i have a wierd problem .. my asterisk stutters a lot .. when i reboot it seems that it works okay for 15 minutes then back to same problem .. it is not hardware related I think |
10:24.20 | skyhawker | any ideas ? |
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10:29.05 | JT | skyhawker: what interface? |
10:30.13 | skyhawker | it is on a SIP interface |
10:30.16 | skyhawker | to our sip gateway |
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10:35.07 | backblue | any devel here? |
10:35.09 | backblue | russellb: ? |
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10:42.33 | MACscr | if i wanted to temporarily disable a trunk in asterisk, as in i just need to stop it from registering, what file do i need to edit? |
10:42.34 | mosty | how can i get the asterisk console to stop showing sip register commands? |
10:50.15 | MACscr | mosty, have any idea on my question? |
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10:52.48 | MACscr | sweet, i figured it out |
10:55.10 | mosty | depends on the type of trunk |
10:55.45 | MACscr | was just a standard sip trunk, i found out which lines i needed to comment out and it worked perfect |
10:56.05 | MACscr | only one extension used that line and that extension is not in use, so its no biggie |
10:56.37 | bXi | does one of you have experience running hylafax and iaxmodem on an asterisk? |
10:56.37 | quelo | tzafir I use asterisk via svf from trunk |
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10:58.11 | mosty | how can i get asterisk to log everything that appears on the console? |
10:58.32 | quelo | tzafrir I use asterisk via svf from trunk |
10:58.50 | quelo | tzafrir I use asterisk via svn from trunk |
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10:59.07 | tzafrir | take a look at your build logs. It took headers from an 1.2 tree |
11:00.36 | quelo | where is build log ? |
11:02.43 | quelo | tzafrir can you tell me how can I solve? |
11:02.48 | Zeeek | mosty there is a .conf for that in the asterisk directory |
11:03.04 | Zeeek | something hard to guess like logging.conf |
11:03.10 | mosty | zeeek: i think i have it working now, it's logger.conf - thanks |
11:03.19 | Zeeek | np |
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11:16.31 | rnaap | Hi2all. Can I get help? =) |
11:17.38 | tzafrir | quelo, from what you pasted |
11:20.38 | rnaap | ... I have a trouble while I try to compile chan_h323 for Asterisk 1.4 beta3 on FreeBSD 6.1... OpenH323 and PWlib versions are equive README. When I try make Asterisk I recieve next messages: [LD] chan_h323.o h323/libchanh323.a -> chan_h323.so and /usr/bin/ld: cannot find -lexpat |
11:21.19 | rnaap | gmake[1]: *** [chan_h323.so] Error 1 |
11:21.21 | rnaap | etc.... |
11:21.34 | rnaap | Help me, plz... |
11:21.39 | quelo | tzafrir maybe I solve installing libopenh323-dev ? |
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11:21.53 | tzafrir | quelo, no. |
11:22.13 | quelo | sure? |
11:22.31 | tzafrir | quelo, for some reason the asterisk-addons build uses the asters-1.2 version instead of your asterisk trunk version |
11:22.53 | tzafrir | Perhaps you need to explicitly point to the asterisk source tree ? |
11:23.03 | Simplix | hello, i have no hangup detection with a sip client (nor dtmf detection) |
11:23.14 | Tond | Hi.. I have a small issue i have 2 extentions one _9821. and another _98. it seems like when i dial 9821xxxx extention _98. gets matched, although _9821. is configured above it. How can i resolve my issue and get 9821xxxx to match _9821. ? |
11:23.24 | Simplix | dtmf is tranmited with sip info |
11:24.16 | quelo | no I've got asterisk-addons with the command... |
11:25.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
11:25.07 | quelo | svn checkout http://svn.digium.com/svn/asterisk-addons/trunk/ asterisk-addons |
11:26.14 | rnaap | I am put into console pwlibdir=~/pwlib.. export $pwlib... etc.. |
11:26.28 | rnaap | but against.... |
11:27.32 | tzafrir | rnaap, you can pass the path to pwlib to the configure script. If you happen to use Debian, just grab the debs and save yourself the bother |
11:27.41 | quelo | tzafrir how can I point to asterisk-1.2 to build asterisk-addons? |
11:28.01 | tzafrir | quelo, you *should not* point to asterisk 1.2 |
11:28.14 | mosty | Simplix: hangup detection for sip phones should work automatically unless your sip phone is broken |
11:28.15 | tzafrir | You should build addons of trunk with asterisk of trunk |
11:28.41 | tzafrir | (1.4 is probably compatible enough by now) |
11:28.54 | tzafrir | BTW: at this stage consider using branches/1.4 instead of trunk |
11:28.55 | Simplix | mosty: its has worked ..... |
11:29.06 | mosty | tond: you have to include => context1 before context2, the contexts are matched in the order you specify, inside each context there is no guaranteed order of searching |
11:29.11 | quelo | my addons IS from trunk |
11:29.22 | tzafrir | trunk of when? |
11:29.32 | quelo | svn checkout http://svn.digium.com/svn/asterisk-addons/trunk/ asterisk-addons |
11:29.53 | tzafrir | branches/1.4 is now stabalising and mostly stable. trunk is in the process of morphing (and getting broken, actualy) |
11:30.42 | quelo | okok I try |
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11:31.31 | Simplix | mosty: it has worked and i have the problem with 2 differents sip phones |
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11:43.27 | piggie | How does one uninstall Asterisk? make uninstall doesn't work. |
11:44.31 | HarryR | rm -rf $prefix/lib/asterisk and the other stuff :) |
11:44.32 | rnaap | thnx |
11:45.46 | piggie | HarryR: Thanks. |
11:47.59 | quelo | tzafrir I have installed libopenh323-dev and now the error is... |
11:48.02 | quelo | http://paste.debian.net/16665 |
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11:49.39 | rnaap | tzafrir... I am use FreeBSD... |
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11:50.10 | mosty | simplix: what sip phone do you have that cant detect when it's hung up? |
11:51.02 | Simplix | mosty: i have 2 sip phone : a Grandstream GXP2000 and a BT100 |
11:51.47 | mosty | Simplix: what does the asterisk console show when you reproduce the problem? |
11:52.23 | Simplix | <PROTECTED> |
11:52.23 | Simplix | <PROTECTED> |
11:53.52 | Simplix | mosty: the phone ring but when i pick up the phone the caller is still ringing |
11:54.23 | Simplix | mosty: the caller is iax2 |
11:54.51 | Simplix | mosty: all sip phone don't work |
11:55.05 | Aurs | Simplix: are you using queue? |
11:55.17 | Simplix | Aurs: not yet |
11:55.40 | Aurs | Simplix: ok. have had similar problem with sip phones in queue; when 1 picked up, the other phones keeped ringing |
11:56.50 | Simplix | Aurs: the other problem is that if sip phone call someone (voicemail for example) when i hangup .... the voicemail app doesn't stop |
11:59.15 | rnaap | Which type of SIP phone you`re use? |
11:59.33 | Simplix | it's like the phone don't send sip info data or asterisk does't receive them |
11:59.45 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
11:59.46 | Simplix | GXP2000 and BT100 |
11:59.53 | rnaap | oh... sorr |
12:00.29 | rnaap | BindAddr working finelly? |
12:01.04 | Simplix | i can make call with sip phones .... |
12:01.15 | Simplix | and called phone ring |
12:02.05 | rnaap | Can you put here SIP.conf? )))))))) |
12:02.07 | rnaap | sorr )))) |
12:02.14 | mosty | Simplix: run tethereal and see what packets are flying about |
12:02.33 | Simplix | ok ... i send sip.conf after that |
12:02.52 | monsted | mosty: that's tshark if you're using software that isn't outdated :) |
12:04.00 | mosty | i know |
12:04.31 | mosty | simplix: what does the called phone hear when you pickup? |
12:05.25 | Simplix | mosty: SIP=>IAX i can make a conversation |
12:06.29 | Simplix | mosty: but sip hangup is not detected |
12:07.33 | mosty | Simplix: run tethereal/tshark on the asterisk server, showing packets going to the sip phone's ip. then make a call and see what happens when you hang up |
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12:21.24 | Simplix | the sip phone send a BYE command |
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12:21.25 | tparcina | t38 problems, does anybody have them? |
12:21.25 | Simplix | mosty: the sip phone send a BYE command but the server still send data |
12:21.25 | *** join/#asterisk ast_freak (n=jesse@h69-130-171-28.69-130.unk.tds.net) |
12:21.25 | tparcina | I belive that my client has fax that supports t38. and when he tries to send fax to some old fax machine (that doesn't support t38) everything works fine. |
12:21.26 | tparcina | but when he tries to send fax to someone who has fax that also supports t38, then they fax tries to send fax using t38 protocol |
12:21.26 | tparcina | then I have problem... :(( |
12:21.26 | mattfletcher | Hello. http://pastebin.ca/249083 is a copy of the Call Forward code example from http://www.cyber-cottage.co.uk/wiki/index.php/Call_forward (copied to pastebin so I can refer to line numbers) I'm trying to get it to work , and it won't. I spotted that line 23 should refer to ARG2 not ARG1, but it still fails for me. My problem is that different extensions Iwant to forward to are on different technologies (SIP, Zap and UNISTIM). The dial comm |
12:21.26 | mattfletcher | forces me to embed one technology though |
12:21.26 | mattfletcher | is there a way round this? The code refers to the technology "Local" but this gives me errors in the console |
12:21.26 | tparcina | my client has panasonic dx600, and they don't know how (or even can it be done) to tourn off t38 support |
12:22.40 | *** join/#asterisk tsurk0 (n=tsurko@80.72.68.86) |
12:24.02 | mattfletcher | Got it. It was going to context "default" whereas all my extensions are set in "outbound". Changing it to "exten => s-CFIM,n,Dial(Local/${CFIM}@outbound,30,Ttr)" cracked it |
12:28.49 | backblue | anyone use polycoms 601 here? |
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12:34.07 | quelo | tzafrir same error for asterisk-addons from branches/1.4 |
12:35.22 | quelo | nemo:/usr/src/asterisk-1.4/asterisk-addons-1.4# make install |
12:35.22 | quelo | make[1]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4' |
12:35.23 | quelo | make[2]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4/asterisk-ooh323c' |
12:35.26 | quelo | make all-am |
12:35.28 | quelo | make[3]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4/asterisk-ooh323c' |
12:35.31 | quelo | source='src/chan_h323.c' object='chan_h323.lo' libtool=yes \ |
12:35.34 | quelo | depfile='.deps/chan_h323.Plo' tmpdepfile='.deps/chan_h323.TPlo' \ |
12:35.36 | quelo | depmode=gcc3 /bin/sh ./config/depcomp \ |
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12:35.48 | quelo | tzafrir same error for asterisk-addons from branches/1.4 |
12:36.21 | quelo | http://paste.debian.net/16672 |
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12:54.59 | mosty | is there a console command that will show me the license id's i have registered? |
12:55.05 | mosty | (for g729) |
12:55.45 | HarryR | Any help? From a FastAGI (orderlycalls) application, I want to be able to play a sound in the background (IVR menu) then read in DTMF tones... but Background() (and Cepstral()) both block until they've finished playing, or take the first DTMF digit which they shouldn't be |
12:55.53 | rnaap | Anybody have chan_h323.so for asterisk 1.4? |
13:01.01 | *** join/#asterisk SwK (n=Silik0nJ@208.44.30.242) |
13:02.10 | fourcheeze | mosty: show g729 |
13:03.08 | mosty | that just show's the number available and in use, i want to know the keys |
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13:11.39 | fourcheeze | mosty: not sure if that's done from the console or from another app |
13:12.30 | dlynes_laptop | mosty: ls -al /var/lib/asterisk/keys |
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13:13.06 | dlynes_laptop | and on that note...i'm hitting the sack |
13:13.11 | dlynes_laptop | g'night peeps |
13:13.11 | mosty | thanks |
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13:29.13 | Sasch | I have a problem with sip phone ... |
13:29.37 | Sasch | when I recived a call my asterisk direct it into papinicomputer queue |
13:30.10 | Sasch | and after all telephone ring (my 2 wireless telephone and my 2 grandstream telephone) |
13:30.30 | Sasch | when i call with my wireless telephone a grand stream all work |
13:30.45 | Sasch | but when i cal with my grandstream a wireless telephone the Dial() return |
13:31.03 | Sasch | Nov 15 14:25:42 NOTICE[3830]: chan_sip.c:2007 auto_congest: Auto-congesting SIP/10-081a3840 |
13:31.36 | *** join/#asterisk jaike (i=jaike@124.106.190.249) |
13:31.37 | Sasch | can help me |
13:34.55 | mosty | is the phone set to DND? |
13:35.02 | slunk | oej if you are about can we discuss the transdirection patch? |
13:35.33 | Zeeek | transdirectionality is a personal very private issue |
13:37.39 | oej[training] | slunk: I'm here |
13:37.55 | oej[training] | slunk: The transdirection patch was merged to svn trunk |
13:38.08 | oej[training] | slunk: for the various releases |
13:38.14 | slunk | Hi it's stephen dredge. Didn't work for me |
13:38.19 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
13:38.41 | slunk | Tags got mixed up when the incoming channel sent a reinvite |
13:39.09 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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13:39.44 | oej[training] | slunk: With which version? |
13:39.58 | puzzled | morning |
13:40.35 | rnaap | Good evening ) |
13:40.45 | Zeeek | beau soleil |
13:40.48 | slunk | Root cause is the confused use of the SIP_OUTGOING flag. |
13:41.27 | oej[training] | slunk: I know, I know. It's all messed up |
13:41.37 | oej[training] | slunk: But which version are you testing with now? |
13:42.21 | slunk | I had to insert that change into slightly old svn. I have some mgcp channel changes that i need update to latest to test with same |
13:42.55 | *** join/#asterisk willie (n=willie@itscotland.demon.co.uk) |
13:42.56 | oej[training] | slunk: I've changed both 1.2 and 1.4 quite a lot since then. You need to test with new versions, please. Just update chan_sip.c |
13:44.18 | slunk | let me just log into work and check version, I have been though the diffs looking for something which should have made a difference |
13:45.12 | mosty | i can't get realtime queues to work properly with the rrmemory or leastrecent strategies. sometimes the queue sends multiple calls to one phone (i think it might not be updating the stats until an agent finishes a call, when it probably should also update when an agent accepts a call) |
13:45.18 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
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13:47.48 | Sasch | i have two telephone one grand stream and one is samsung wip6000p |
13:47.49 | *** part/#asterisk slvrdrgn (n=Miranda@pd95684d7.dip0.t-ipconnect.de) |
13:47.51 | DrukenHME | 1.2.12 :) |
13:48.13 | Sasch | when i take a samsung and call grand stream all work |
13:48.22 | ManxPower | DrukenHME: too many reported problems with 1.2.12 |
13:48.28 | Sasch | but when i call with grand stream my wip 6000p don't work |
13:48.30 | Sasch | why ?? |
13:48.39 | ManxPower | Sasch: I don't know. |
13:48.42 | jaike | 1.2.12.1 |
13:48.48 | DrukenHME | ManxPower: really? i don't have much problem with it at all..... |
13:48.57 | ManxPower | Sasch: does "sip show peers" show that the phone is lagged. |
13:49.14 | ManxPower | DrukenHME: try using queues, ChanSpy, or Monitor() |
13:49.17 | mosty | sasch: set verbose 10 and set debug 10 in the asterisk console, then watch what it says when you try the call |
13:49.22 | DrukenHME | i do use queues.... |
13:49.36 | Sasch | yes 10/10 192.168.0.6 D N 5060 OK (100 ms) |
13:49.57 | jaike | ManxPower: all those apps work ok on 1.2.12.1 |
13:50.30 | Sasch | http://pastebin.ca/249148 |
13:50.41 | ManxPower | jaike: I'm still too scared to upgrade. |
13:50.47 | Sasch | 10 is my wip 6000p and 13 is my grand stream |
13:51.01 | DrukenHME | 100ms? ouch.... |
13:51.24 | jaike | ManxPower: know how you feel, been there. wondering if the server will explode on the next upgrade |
13:51.37 | DrukenHME | course, my wip300 has a shitty latency as well.... |
13:51.44 | jaike | but 1.2.12.1 proved me wrong, am sticking with it til maybe 2.0 comes out |
13:51.45 | DrukenHME | wireless just can't handle it.. hehehe |
13:51.46 | slunk | version 43649 so very old ( has it been that long ? ) I will test with the latest tommorow |
13:52.04 | Sasch | but if I call my line the call is redirect to papinicomputer's queue ... and wip6000p ring |
13:52.07 | Sasch | why ??? |
13:52.23 | ManxPower | jaike: maybe ?I'll look at 1.2.13 in a few weeks after enough people are running it. |
13:52.27 | Sasch | why when recive a queue ring and when i call with my grand stream don't work |
13:53.11 | Sasch | is a asterisk bug or is my telephone ?? |
13:53.12 | ManxPower | Sasch: What version of asterisk? |
13:53.25 | Sasch | Asterisk SVN-branch-1.2-r46964M built by root @ neo on a i686 running Linux on 2006-11-04 10:38:30 UTC |
13:55.07 | ManxPower | try upgarding to a release |
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13:55.31 | slunk | the base request is updated from a received request in 4 places but the outgoing flag is only cleared at two of them |
13:55.55 | Sasch | ok to upgrade i must make zaptel or i can only make new asterisk ?? |
13:56.21 | slunk | and handle_request i think looks like it expects the flag to be constant for the duration of the call. |
13:56.45 | slunk | has the intention of this flag changed at some point |
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13:57.47 | ManxPower | Sasch: you should always upgrade libpri, zaptel, and asterisk at the same time. |
13:57.58 | Zeeek | or not |
13:58.20 | Sasch | <ManxPower> ok thanks |
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13:59.32 | slunk | ops meant handle_response |
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14:00.59 | ManxPower | slunk: try #asterisk-dev |
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14:04.57 | Chris-NB | hi |
14:05.29 | Chris-NB | anyone experienced asterisk sending rtp to the private IP of the phone (192.168.2.xxx) |
14:06.24 | *** join/#asterisk danielmendez (n=danielme@201.244.247.15) |
14:06.33 | danielmendez | hello there |
14:07.10 | danielmendez | i want to play a DTMF tone when the CALLING party is connected to the CALLED party. |
14:07.15 | danielmendez | anyone has some hints? |
14:08.17 | BrokenNoze | there's an application that does that |
14:08.26 | ManxPower | danielmendez: you mean like the D() option to Dial? |
14:08.32 | BrokenNoze | don't know what its called. |
14:08.49 | tparcina | T.38, anybody? |
14:09.01 | ManxPower | tparcina: no thanks. It doesn't work. |
14:09.09 | BrokenNoze | Anyone know how to set the Callerid in an Originate Manager API function |
14:09.27 | tparcina | yes, I know that, but my fax machine always tries to send fax using T38 |
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14:09.36 | ManxPower | tparcina: tell it not to. |
14:09.44 | BrokenNoze | I'm doing Calledid: <My DDI> but it doesn't set the caller id name??? |
14:09.51 | tparcina | and it can send fax to some numbers (fax machines) but can't to other |
14:09.52 | ManxPower | tparcina: Your fax supports IP? |
14:10.04 | ManxPower | BrokenNoze: the correct format is: Name <number> |
14:10.14 | ManxPower | OF course, the name will not be accepted by carriers |
14:10.26 | tparcina | it seams so - it's panasonic dx600 |
14:10.26 | BrokenNoze | So Callerid: MyDDIName 123456? |
14:10.30 | ManxPower | bro no |
14:10.41 | mattfletcher | I want to record EVERY outgoing call made. This involves a lot of different extensions (internal, sip, zap channels etc). Is there a way of calling MixMonitor on every call made out of my "outbound" extension? |
14:10.43 | ManxPower | Callerid: MyDDIName <123456> |
14:10.53 | BrokenNoze | Ah. star cheers guys |
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14:11.05 | ManxPower | mattfletcher: yes, run Monitor before the outgoing Dial |
14:11.09 | tparcina | ManxPower: how can I tell it not to use t38? I didn't find anything usefull in manual |
14:11.44 | ManxPower | tparcina: I don't know. Is the fax machine connected direct to your Ethernet or connected into an ATA? |
14:11.45 | danielmendez | the problem is that the D() opttion from the Dial application send the DTMF to the calling party when the call has not been answered yet |
14:12.21 | ManxPower | danielmendez: it will only do that on analog ports since those are always considered answered as soon as Dialing is finished. And it sends it to the CallED party, not the CallING party |
14:12.21 | tparcina | it's panasonic dx600 - handy tone 386 ata - Asterisk - My SIP provider |
14:12.37 | ManxPower | tparcina: then the Handytone is trying to do the T.38 |
14:12.50 | tparcina | and I can receive fax from everybody (at least it seams like that) |
14:12.54 | danielmendez | i need it to start a process reciveing a # DTMF when the call is answered |
14:13.10 | tparcina | and i can send fax to 70% of people, but those 30% allways fail |
14:13.43 | ManxPower | danielmendez: Which is it? Send DTMF to the called party? Send DTMF to the calling party? Receive DTMF from the called party? or Receive DTMF from the calling party? |
14:14.11 | ManxPower | tparcina: um, FaxOverVoiceOverIP is not reliable. |
14:14.12 | tparcina | ManxPower: you don't think that it's FAX problem, but that it's ATA problem? |
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14:14.24 | ManxPower | tparcina: no, I KNOW it is an ATA problem. |
14:14.32 | ManxPower | your fax machine doesn't know anything about T.38 |
14:15.05 | tparcina | ManxPower: how do you comment that certin numbers allways go thru and some numbers allways fail |
14:15.28 | ManxPower | tparcina: I think it's just the fact that faxovervoiceoverip is not reliable |
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14:15.57 | ManxPower | tparcina: T.38 is an IP protocol. Your fax machine isn't even plugged into the network. |
14:15.58 | tparcina | ManxPower: ok, thank you |
14:16.44 | danielmendez | I need to dial a number, when the call is answered I need to hear a DTMF |
14:16.56 | DrukenHME | i want to see digital voip fax MACHINES!!! |
14:17.04 | piggie | Hi all, I've managed to successfully install zaptel and I went back to asterisk to do make && make install but chan_zap.so is not being compiled and installed, any ideas? I'm using the latest beta for asterisk and zaptel. |
14:17.09 | ManxPower | danielmendez: You need to hear DTMF or the Asterisk Dial Plan needs to hear DTMF. |
14:17.22 | danielmendez | I need |
14:17.40 | ManxPower | danielmendez: that should happen by default and I can't think of anything that would prevent that. |
14:18.09 | ManxPower | danielmendez: So you call me and I answer and then I dial DTMF and you need to hear the DTMF, right? |
14:18.24 | Chris-NB | any ideas why a phone sends his private IP in the sip/sdp paket as it's contact information/owner address? |
14:18.36 | ManxPower | Chris-NB: because you don't have nat=yes |
14:18.44 | Chris-NB | ManxPower, I have nat=yes |
14:18.46 | ManxPower | Chris-NB: oh, and it will always do that. |
14:18.58 | ManxPower | nat=-yes just tells asterik to ignore the ip in the SDP |
14:19.09 | danielmendez | the sYstem must send the DTMF |
14:19.19 | ManxPower | danielmendez: which system? |
14:19.24 | Chris-NB | ManxPower, noop. I've two phones behind a router. one is wired, one is wireless |
14:19.39 | Chris-NB | the wireds sends the public ip, the wireless sends the local IP |
14:19.42 | danielmendez | an LCD dysplay that counts the time of the call |
14:19.54 | Chris-NB | and asterisk sends rtp to the private ip (for the wireless phone) |
14:19.56 | danielmendez | when it is answered |
14:20.05 | Chris-NB | for the wired phone, everything works fine |
14:20.18 | Chris-NB | only in the reg. contact i can see the private IP |
14:20.20 | ManxPower | danielmendez: I can't think of an easy way to do that. |
14:20.27 | Chris-NB | in sip/sdp there is only the public one |
14:20.34 | DrukenHME | Chris-NB: are these both behind the same router? |
14:20.40 | Chris-NB | DrukenHME, jep |
14:20.40 | ManxPower | danielmendez: The dialplan stops until the Dial is finished. |
14:21.30 | Chris-NB | DrukenHME, this router is via dsl connected to the interrnet. my asterisk is behind a firewall connected, connected via another dsl to the internet |
14:21.37 | Chris-NB | everything else works fine |
14:21.39 | ManxPower | danielmendez: Oh! Yes, they updated the D() to support both called and calling |
14:21.51 | Chris-NB | except this wireless sip phone |
14:22.08 | ManxPower | danielmendez: "show application dial" in the Asterisk CLI, pay special attention to the D() option. |
14:22.11 | DrukenHME | Chris-NB: i'd be checking the config of the wireless phone then.... |
14:22.21 | ManxPower | I think the feature you want is in 1.2.x and later. |
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14:22.37 | Chris-NB | DrukenHME, the phone works fine behind a differend router |
14:22.50 | Chris-NB | DrukenHME, so I don't think it's the phone config |
14:22.54 | danielmendez | i use someting like D(:#) but the dial application send the DTMF when the call has not been answered |
14:23.02 | DrukenHME | well, don't think... make sure :) |
14:23.20 | Chris-NB | k, the config of the wired and wireless phone is the same! |
14:23.21 | Chris-NB | : ) |
14:23.29 | ManxPower | danielmendez: did you miss my statement "Asterisk considers analog ports to be answered as soon as the Dialing is finished"? |
14:23.32 | Chris-NB | one works, the other works ... partly : / |
14:23.35 | DrukenHME | diffrent control ports? |
14:23.47 | Chris-NB | DrukenHME, what do you mean? |
14:24.03 | DrukenHME | i mean is the wired and wireless both trying to use 5060 ? |
14:24.12 | Chris-NB | jep |
14:24.21 | DrukenHME | change one of them to 5061 |
14:24.23 | Chris-NB | haven't tried simultanously |
14:24.25 | ManxPower | any decent NAT router will remap the source port of 5060 to a different port. |
14:24.33 | danielmendez | well i actually dont know |
14:25.09 | Chris-NB | the curios thing is, registration and call establishment works fine |
14:25.27 | Chris-NB | except the rtp data is sent to the wrong ip |
14:25.37 | Chris-NB | the private instead of the public one : / |
14:25.50 | ManxPower | If the public IP is set in the SDP then the phone is set for nat in additon ot Asterisk |
14:25.53 | Chris-NB | and i've absolutely no clue why! |
14:26.10 | ManxPower | Chris-NB: the phones SHOULD put the private IP into SDP. |
14:26.22 | ManxPower | nat=yes tells asterisk to work around that. |
14:26.34 | Chris-NB | ManxPower, nat=yes is set |
14:26.54 | Chris-NB | <PROTECTED> |
14:26.56 | ManxPower | Chris-NB: the the incoming call is not matching the sip.conf entry. |
14:26.58 | Chris-NB | from that peer |
14:27.12 | ManxPower | Chris-NB: you don't want both nat=yes AND NAT stuff set on the phone. |
14:27.39 | Chris-NB | ManxPower, what do you mean by that? |
14:27.47 | ManxPower | Chris-NB: put context=INVALID in [general] the put the correct context= line in each of the sip.conf entries. |
14:27.48 | Chris-NB | ManxPower, what nat stuff? |
14:28.12 | ManxPower | Chris-NB: whatever nat stuff the device supports. |
14:28.34 | danielmendez | manxpower: what would happen with a PRI E1 or PRI T1 ? |
14:28.47 | ManxPower | danielmendez: it would not happen with any kind of PRI. |
14:29.00 | Chris-NB | ManxPower, I've nat=yes and context=intern in the [wirelssphone] context |
14:29.22 | ManxPower | Chris-NB: and do you have context=INVALID in [general] |
14:29.24 | ManxPower | have to leave now |
14:29.58 | Chris-NB | ManxPower, context=default in [general] |
14:30.25 | Chris-NB | ManxPower, so the phone has to use the own config?! |
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14:36.44 | piggie | Hi all, I've managed to successfully install zaptel and I went back to asterisk to do make && make install but chan_zap.so is not being compiled and installed, any ideas? I'm using the latest beta for asterisk and zaptel. |
14:37.13 | mattfletcher | ManxPower: I saw you answer my question about MixMonitor'ing outgoing calls, but my PC then crashed so I don't know if there were more replies. You said to add MixMonitor before the Dial command. That misses my point however. I have lots of extensions which all need recording, is there any way to add a dialplan command for every extension in a context is I suppose what I am asking |
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14:37.19 | *** mode/#asterisk [+o anthm] by ChanServ |
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14:43.09 | mattfletcher | Anyone, is there a way to add a dialplan command (MixMonitor) to every extension in a context? |
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14:45.23 | danielmendez | i use someting like D(:#) but the dial application send the DTMF when the call has not been answered |
14:50.50 | *** part/#asterisk mega (i=mega@gateway/tor/x-f9126ab1487a13e4) |
14:51.40 | danielmendez | MAnxpower: checking tje dial application, i ve found the L(x:y:z) parameter, where i can define via a VARIABLE LIMIT_CONNECT_FILE the sound file to be played when the call begins. do you know that option? |
14:52.50 | danielmendez | manxpower: also, do you know what it means "when the call begins" ?? the start of the dial process or the call connected between the parties |
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15:00.15 | b11d | morning all |
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15:08.50 | b11d | :| |
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15:12.23 | dasenjo | Hi! I have a language=es line at [general] in sip/iax/zapata/voicemail.conf and loadzone=es,defaultzone=es in zaptel.conf. ¿Why am i getting Playing 'digits/5' (language 'en') in CLI? |
15:14.44 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
15:14.59 | Dr-Linux|work | hi guys |
15:16.13 | Dr-Linux|work | i need some help regarding variables |
15:16.30 | Dr-Linux|work | my question is also here >> http://networks.pk/forum/viewtopic.php?p=17#17 |
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15:17.19 | Dr-Linux|work | i'm appreciate if someone help |
15:17.43 | Dr-Linux|work | s/i'm/i'll |
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15:22.03 | ManxPower | Dr-Linux|work: SetVar(LANG=${SPANISH}) |
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15:24.21 | dasenjo | I'm using 1.2, ¿should I file a bug? |
15:24.25 | Dr-Linux|work | ManxPower: thanks but in which portion i should use this variable you mentioned? |
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15:25.57 | dasenjo | ¿no help? ¿can you guide debugging the problem? |
15:28.12 | db1310 | installed asterisk on debian sarge via dist package, reading setup from book AsteriskTFOT. Refers to ztdummy, whereis package and do i need it? |
15:28.34 | ManxPower | db1310: ztdummy *should* be part of Zaptel. |
15:28.45 | ManxPower | We really can't help much with distro specific things |
15:29.48 | db1310 | no zaptel card, just trying to setup a simple soft phone for now to experiment. So I do need ztdummy driver |
15:30.16 | dasenjo | db1310, apt-get install zaptel-source; m-a a-i zaptel |
15:30.21 | ManxPower | db1310: I understand. The zaptel source contains ztdummy |
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15:31.23 | db1310 | Ok, so i need to get it build it and load the module. Thanks all |
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15:44.39 | Sasch | to load kernel module is modprobe .. to unload ?? |
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15:46.28 | backblue | rmmod |
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15:53.29 | pifiu | morning everyone |
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15:57.36 | db1310 | reading claims on openpbx.org, how do they compare, seriously. pros / cons Asterisk against OpenPbx |
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16:01.05 | CtRiX | db1310, i cannot speack about openpbx here. I have already been banned once for that reason. |
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16:05.32 | dasenjo | hey, please, I really need help. I think language support in asterisk is almost «poor». Take in account that not all the world speak english |
16:10.08 | CunningPike | dasenjo: There are language bundles available for system prompts, and many people have recorded prompts in their own language. Nothing precludes you from doing this in the language of your choice |
16:10.43 | CunningPike | dasenjo: Language support in asterisk is as good as the prompts that people record for it |
16:12.06 | dasenjo | CunningPike, you did not see my first question |
16:12.15 | dasenjo | Hi! I have a language=es line at [general] in sip/iax/zapata/voicemail.conf and loadzone=es,defaultzone=es in zaptel.conf. ¿Why am i getting Playing 'digits/5' (language 'en') in CLI? |
16:12.56 | dasenjo | CunningPike, I have prompts, I have digits/es es/digits, all the conf ... and asterisk keep playing en messages, ¿why? |
16:13.33 | CunningPike | dasenjo: Hmm- not sure - maybe a buglet - have you tried opening a bug on mantis? |
16:14.32 | dasenjo | uhmmm .. yes .. that was my second ot thrid question .. but karma still make me get confused :P |
16:16.18 | dasenjo | :( |
16:16.23 | *** part/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
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16:28.56 | Katty | hihi. |
16:29.44 | b11d | ihih |
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16:41.45 | L|NUX | mog : hey |
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16:42.43 | file | Katty!!?! |
16:42.54 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
16:43.19 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
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16:48.00 | *** mode/#asterisk [+o mogorman] by ChanServ |
16:48.27 | L|NUX | mogorman : hey |
16:48.29 | L|NUX | mogorman : you are late ;) |
16:48.29 | L|NUX | hehe |
16:48.50 | mogorman | cable guy screwed me |
16:49.11 | L|NUX | oh |
16:49.13 | L|NUX | okay :) |
16:49.39 | L|NUX | btw i just come online |
16:49.42 | L|NUX | almost 10 minutes or so |
16:49.42 | L|NUX | i have been to a doctor |
16:49.45 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
16:49.50 | L|NUX | for my child vaxination |
16:50.20 | L|NUX | mog : its good that you are here now :) |
16:50.37 | mog | ahh okies |
16:50.46 | mog | let me get my box set up shouldnt take but 10 minutes |
16:51.02 | L|NUX | so will you find out what was issue ? |
16:51.04 | L|NUX | ok |
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16:58.40 | DerPraktikan | hi ! i got a problem by installing Fedora Core 6 Zod on a 64Bit system, can somebody please help me? |
16:59.27 | DerPraktikan | i boot from the cd , and startet the grafical installation |
17:01.04 | Katty | file?!?!?!??!!!! |
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17:01.41 | file | Katty: 'chu at work? |
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17:01.53 | Katty | file: yesh. |
17:02.05 | file | thrilling! |
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17:02.32 | Katty | mhmm |
17:04.08 | file | eep |
17:04.38 | zx6r | ...i just want to buy a melon |
17:04.42 | L|NUX | mog : back ? |
17:04.52 | mog | still working yes |
17:05.09 | L|NUX | ok |
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17:11.30 | *** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net) |
17:12.36 | WGFreewill | anyone know how to enable at compile time the new jitterbuffer |
17:12.41 | WGFreewill | in 1.2 |
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17:15.17 | pif | what labels should I use in GotoIf() when using 'n' priorities ? |
17:15.58 | sangee | how do setup g723 passthrough? |
17:17.43 | *** join/#asterisk Pumas (n=KAos@207.248.51.107) |
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17:20.03 | WGFreewill | sangee, for g729 |
17:20.40 | WGFreewill | we just put g729 in the sip.conf or device entry |
17:20.40 | WGFreewill | and set the devices to prefer g729 |
17:20.40 | WGFreewill | and the calls pass right through no license |
17:20.54 | WGFreewill | (I have read that you need to take t and r out of your Dial command) |
17:21.08 | WGFreewill | got g723 |
17:21.19 | WGFreewill | should be just allow=g723 |
17:21.20 | WGFreewill | in the sip.conf |
17:21.22 | WGFreewill | if its supported |
17:21.34 | Pumas | Hi, I have a problem with my asterisk: I have configured asterisk with VOIP service, I must record the incoming calls from VOIP, but when the call has 5 minutes, it stops automatically, |
17:21.49 | Pumas | help me please |
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17:22.41 | mog | ugh L|NUX my jabber server is still down do to cable, let me try connecting elsewhere |
17:22.55 | L|NUX | okies |
17:22.56 | L|NUX | :) |
17:22.57 | L|NUX | brother |
17:23.01 | L|NUX | you will take your time :0 |
17:23.24 | L|NUX | mog : just a favor when you get it fixed email me |
17:23.25 | L|NUX | SecRECV[<iq to="dost4u@gmail.com/talk0F42BBAE" id="307" type="error" from="CreativeLimit@gmail.com/Talk.v100FF1A9D0D"><session type="terminate" id="3885069672" initiator="creativelimit@gmail.com/Talk.v100FF1A9D0D" xmlns="http://www.google.com/session"/><error type="modify"><sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"/><sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">unknown session</sta:text></error></iq |
17:23.25 | L|NUX | shit |
17:23.25 | L|NUX | sorry |
17:24.32 | Pumas | help me please |
17:24.35 | Pumas | Hi, I have a problem with my asterisk: I have configured asterisk with VOIP service, I must record the incoming calls from VOIP, but when the call has 5 minutes, it stops automatically, |
17:24.53 | Pumas | somebody knows why? |
17:25.00 | WGFreewill | how are you setup to record |
17:25.03 | WGFreewill | mixmonitor |
17:25.09 | sangee | thx |
17:26.32 | WGFreewill | Pumas how is your recording setip |
17:26.33 | Pumas | I'm using command record |
17:26.51 | fenlander | L|NUX: that looks like a mismatch on the case of CreativeLimit - Google talk can be picky about the difference betwen creativelimit and CreativeLimit in the initiatior string |
17:27.03 | WGFreewill | are you setting a maxduration |
17:27.08 | WGFreewill | or is the machine out of disk |
17:27.13 | Pumas | but I just have problems when the call comes from VOIP |
17:27.32 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
17:27.38 | Pumas | by the way I'm totally new in asterisk |
17:27.41 | L|NUX | fenlander : okay thanks brother :0 |
17:27.52 | mog | not that again |
17:27.55 | WGFreewill | get some trixbox they use mixmonitor |
17:28.01 | WGFreewill | and it works fine usually |
17:28.03 | mog | i could have swarn i fixed that a week or so ago |
17:28.03 | L|NUX | mog : yeah i am sorry for that brother :( |
17:28.07 | mog | or even farther back |
17:28.33 | Pumas | I have no maxduration in command Record |
17:28.37 | L|NUX | mog : are you talking with me ? |
17:29.01 | WGFreewill | Pumas: you can specify maxduration 0 for no limit |
17:29.19 | *** join/#asterisk TommyTheKid (n=tommy@mpk-edge.cto.sunit.net) |
17:31.23 | TommyTheKid | I have a minor annoyance.. I have a digium te1xxp and te4xxp, both of them continually "reset" (idle?) B channels. I have set the "resetinteval = never" but it just seems to keep on doing it.. is there some other option I am missing? |
17:31.32 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
17:31.33 | Pumas | yes, I did that, but when the call is in 5 minutes exactly it stops |
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17:32.14 | WGFreewill | I see my channels reset |
17:32.21 | WGFreewill | on my 405s |
17:32.40 | WGFreewill | <PROTECTED> |
17:32.45 | TommyTheKid | there are options in zapata.conf that seem to indicate that you can disable it, but as far as I can tell, they are ignored |
17:33.11 | WGFreewill | Pumas: is there silence detection? |
17:33.13 | TommyTheKid | with 96-ish channels it gets a bit annoying when they all hit |
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17:33.26 | WGFreewill | Tommy: I think the network can request the reset as well asn the terminal |
17:33.47 | WGFreewill | Tommy: I've never had it take down calls |
17:33.56 | TommyTheKid | no, I think it only resets idle channels |
17:34.16 | WGFreewill | possibly a q931 debug or something |
17:34.16 | TommyTheKid | hence it being a "minor annoyance" rather than a "major issue" :) |
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17:34.28 | WGFreewill | to see if there is a message from the net |
17:34.29 | Pumas | it is the the command as i use it exten=> *7,n,record(/usr/asteriskneitek/${folder}/${file}:wav) |
17:34.34 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
17:35.16 | *** join/#asterisk yogurt2ungue (n=yogurt2u@host236.200-117-208.telecom.net.ar) |
17:35.18 | WGFreewill | *7,n,record(/usr/asteriskneitek/${folder}/${file}:wav|0|0) |
17:35.35 | WGFreewill | I have learned to ignore it actually, not that its right |
17:36.09 | Pumas | I attached a line to asterisk and I called to it and all works fine, BUT when the call comes from VOIP stops at 5 minutes |
17:36.32 | Pumas | what is the difference between them? |
17:36.39 | WGFreewill | the one I send has |
17:36.48 | WGFreewill | silence 0 and maxduration 0 |
17:36.51 | WGFreewill | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
17:37.21 | Juggie | WGFreewill, when you call over voip, what device are you using? |
17:37.49 | WGFreewill | SJphone, Xten, Polycom Soundpoint, Cisco 79XX, Grandstream |
17:37.56 | Juggie | and, are you actually speaking into it, or laying it on the desk. |
17:38.20 | pif | what code can I use in Hangup() to signal 'normal call clearing' |
17:38.43 | Juggie | just Hangup should do that. |
17:39.04 | WGFreewill | Juggie: dont understand, phones go on desks and tables all around me, or the floor or a bucket |
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17:39.41 | Pumas | in fact I was specified these 2 parameters, but it didnt work with VOIP, |
17:39.42 | Juggie | i was just wondering if it had something to do w/ silence detection. |
17:40.09 | WGFreewill | ahh the silence was for Pumas, record silence detection |
17:40.57 | WGFreewill | are you setting a max threshold on the voip call |
17:41.03 | WGFreewill | to cut it after 5 minutes |
17:41.09 | WGFreewill | can you call and echo test for more than 5 minutes |
17:41.38 | Pumas | where I can configure the threshold? |
17:41.55 | Juggie | Pumas, my initial reponse would be, how are you testing this, |
17:42.04 | Juggie | are you just calling in and putting the phone on the desk. |
17:42.07 | Juggie | and waiting 5 minutes? |
17:42.14 | Pumas | no |
17:42.42 | WGFreewill | Pumas: AbsoluteTimeout(seconds) or setting the timout in the voip Dial command |
17:42.43 | L|NUX | mog : i am going see my email and email me |
17:42.45 | Juggie | are you sending voice down the line for 5 minutes? |
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17:43.56 | Pumas | I'm recording the call, I call to VOIP number and my server takes the call, and starts to record the incoming call but when the call has 5 minutes flat it stops the recording |
17:44.29 | Juggie | right, but what are you recording |
17:44.38 | Juggie | are you talking for 5 minutes, is the phone on the desk, etc. |
17:44.50 | Pumas | in this case my voice |
17:45.17 | WGFreewill | you dont daydream and stop talking? |
17:45.35 | Juggie | Pumas, in your logged.conf enable your full debugging, restart *, make this happen, and then pastebin.ca the 'full' log file. |
17:45.41 | Juggie | er, logger.conf |
17:45.59 | WGFreewill | should tell you why the call went down |
17:47.01 | pif | when getting a BUSY from a zap channel I don't want to relay the busy to the SIP agent, only playtones(busy) and hangup |
17:47.22 | pif | but somehow the SIP client get the dialstatus after my hangup |
17:47.22 | Juggie | pif, why? |
17:47.24 | Pumas | I did another test, I attached a line to Asterisk and call to it, and I'm able to record with no time limit, |
17:48.00 | Pumas | why just the voip has this problem? |
17:48.15 | Juggie | pumas, follow my instructions and we'll find out. |
17:48.23 | Pumas | ok, thanks |
17:48.25 | Juggie | enable full debug, make it happen, then www.pastebin.ca the output. |
17:48.25 | WGFreewill | yes pumas the logs |
17:48.27 | *** join/#asterisk Pug|Work (n=jamesj@128.227.123.36) |
17:48.31 | Pumas | I''ll do it |
17:48.32 | Pug|Work | So, about trixbox... |
17:48.43 | WGFreewill | #amportal |
17:48.43 | *** part/#asterisk Pug|Work (n=jamesj@128.227.123.36) |
17:48.48 | hmmhesays | #freepbx |
17:48.49 | Juggie | for trixbox join #freepbx/#amportal |
17:48.53 | WGFreewill | ahhh yes |
17:48.56 | WGFreewill | new irc and everything |
17:49.13 | hmmhesays | it must be cold where this chick is performing her stand up |
17:49.14 | WGFreewill | how about that digium GUI |
17:49.26 | WGFreewill | slick ajax |
17:49.30 | Juggie | WGFreewill, as for the b channels resetting thats nothing to be conerned with. |
17:49.39 | WGFreewill | yes I never care |
17:49.48 | WGFreewill | but I see it happen on all ten servers I have |
17:49.53 | Juggie | its a tie over from days when asterisk and switches were less friendly to each other, it could probally be removed, but its not a problem. |
17:50.07 | Juggie | yeah, asterisk resets all free b channels on the top of every hour. |
17:50.27 | WGFreewill | what I really need is that new jitterbuffer |
17:50.36 | Juggie | it wont affect calls in progress or anything like that. |
17:50.40 | WGFreewill | I implemented a voice over wireless |
17:50.47 | WGFreewill | and we are getting killed |
17:50.59 | TommyTheKid | ok, another minor annoyance... when I dial out to a SIP gateway (cisco) everything is "normal".. however when I dial out using ZAP (pri) it sounds like I am getting a double ring sound... one will start maybe 1/4th of a second before the other then both of them at once then the second one finishes. I have seen many posts about this in mailing lists, but never really seen a resolution. It almost seems like (even without the "r") the Dial comman |
17:51.37 | Juggie | TommyTheKid, thats because analog sucks, |
17:51.44 | WGFreewill | your signalling |
17:51.48 | TommyTheKid | analog? who is using analog? |
17:52.04 | WGFreewill | try switching the switch and the ZAP |
17:52.16 | Juggie | oh, i didnt read closely hah, that sounded like an analog problem. |
17:52.16 | WGFreewill | isdn is supposed to tell you inband or out of band ring |
17:52.32 | Juggie | yeah, make sure everything is set to out of band. |
17:52.32 | WGFreewill | you seem to get both |
17:52.33 | TommyTheKid | Digium 412p to an Avaya PBX. |
17:52.49 | Juggie | TommyTheKid, what switch type? |
17:52.53 | TommyTheKid | .. and I am not saying for sure its not all the Avaya |
17:53.11 | WGFreewill | I line national, dms-100, 5ess |
17:53.17 | TommyTheKid | 1,1,0,esf,b8zs,yellow |
17:53.21 | WGFreewill | but I have had to switch both sides |
17:53.30 | TommyTheKid | oh, national |
17:53.31 | WGFreewill | in some cases to make it work right |
17:53.43 | Juggie | TommyTheKid, make sure you are using out of band signalling. |
17:53.51 | WGFreewill | DMS-100 doesnt like national |
17:53.55 | WGFreewill | even though both ends support it |
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17:54.01 | WGFreewill | yes |
17:54.05 | Juggie | i use all dms100 |
17:54.15 | WGFreewill | I use mostly dms-100 |
17:54.21 | WGFreewill | but you know |
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17:54.25 | Juggie | canada is still mostly dms100 |
17:54.25 | WGFreewill | I have an EICON card |
17:54.34 | WGFreewill | wont link to the freaking thing |
17:54.37 | Juggie | though i think we are moving to 5ess |
17:54.41 | Juggie | EICOM? |
17:54.42 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
17:54.58 | WGFreewill | yeah think real expensive PCI T1 card |
17:55.07 | Juggie | yah, they bought dialogic |
17:55.12 | WGFreewill | (has neato t.30 fax DSPs) |
17:55.45 | ucfMethod | is anyone here in the DC area? |
17:55.58 | WGFreewill | how many miles? |
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17:56.38 | *** join/#asterisk Strom_C (i=strom@nat/digium/x-ba966754fc753e43) |
17:56.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
17:56.48 | WGFreewill | juggie: i have the eicon connected to a sangoma, then sangoma to the nortel works |
17:57.54 | Juggie | WGFreewill, you should report that to digium then. |
17:58.00 | Juggie | if you cant make them link. |
17:58.14 | TommyTheKid | so, I inherited the "zap" portion http://pastebin.ca/249333 of asterisk... I was told this is what works.. back when we had a singe PRI card.. now we have the 412p.. I don't remember the double ring before, but I may not have been paying as much attention before I was the one managing this... does that look setup correctly? (I added the extra "channel" secons and duplicated the "span/bchan/dchan" sections |
17:59.48 | Strom_C | TommyTheKid, are you setting up four completely separate PRI spans? |
17:59.59 | TommyTheKid | I think so |
18:00.22 | Strom_C | well, better to find out definitively yes or no before you go any further :) |
18:00.23 | TommyTheKid | they are in one trunk group on the PBX, but as far as I know, they are not sharing D-channels or anything |
18:00.49 | TommyTheKid | "the PBX" in this case should be considered as my "telco" :) |
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18:01.11 | Strom_C | TommyTheKid, verify how you have things set up on that pbx |
18:01.50 | TommyTheKid | what do I ask them.. I need specific questions, cause I have to break thru like 2 or 3 tiers of "support" before I get to the people who have a clue :) |
18:01.58 | quidpro | Hmm... anybody using a SPA-3000/3102 here? |
18:02.24 | Strom_C | TommyTheKid, "Are these individual PRIs, or do these multiple T1 spans comprise an NFAS group?" |
18:02.45 | Strom_C | "NFAS" is pronounced "En-Fass" |
18:02.53 | Strom_C | er, EN-fass |
18:03.07 | arcanine | can i use a generic fax/modem card in replace for the rhino fxo crd |
18:03.39 | Juggie | TommyTheKid, take a look at the sample zapata.conf file, and look at some of the options you are missing, like inband/outband settings, make sure you setup out of band etc. |
18:07.14 | TommyTheKid | OK, I am sure its not NFAS, the people who set it up are paranoid :) also the PRIs were originally NOT all plugged into the same place. We did change them to all be part of the same "trunk group".. but decided not to setup shared D- channels. |
18:07.29 | Strom_C | well....define "trunk group" |
18:07.32 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
18:07.38 | TommyTheKid | I will take a look at the zapata.conf.sample... it seemed to have a lot of "nonsense" it in for analog type stuff |
18:07.47 | Strom_C | because "trunk group" to me would seem to suggest NFAS |
18:08.32 | TommyTheKid | thats what I thought too, essentially if we have 400 DID's and 4 PRI's one of those DIDs can use all 92-ish lines |
18:08.57 | Strom_C | TommyTheKid, find out for certain |
18:09.05 | TommyTheKid | we have only ever had about 60 concurrant calls tho ;) |
18:09.11 | TommyTheKid | concurrent ;) |
18:09.18 | arcanine | what most suited digium card for inbound toll-free |
18:09.24 | arcanine | i hav 4 analog 1-800 lines |
18:09.49 | Strom_C | arcanine, for four analog lines, either the tdm400p or the tdm2400p |
18:10.21 | file | Strom_C: where... are you! |
18:10.46 | Strom_C | file, with Mancini and Jan |
18:11.01 | file | yay Jan |
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18:13.47 | Pumas | in logger.conf I have this line: full => notice,warning,error,debug,verbose, but how can I tell asterisk where to write the log output? |
18:15.02 | Pumas | are you there WGFreewill? |
18:16.04 | *** join/#asterisk mhz121 (n=mhz121@128.238.244.190) |
18:16.25 | TommyTheKid | i have a ticket in.. meantime I will poke around in zapata.conf.sample looking for inband/outofband stuff .. unfortunately, it seems you can't just "zap reload" :) .. so I can't really test anything till I can do a full restart |
18:16.33 | WGFreewill | yeah |
18:16.39 | WGFreewill | sorry |
18:16.57 | Katty | umm, umm, file. |
18:16.58 | brif8 | Hi all, are 1.4 questions asked here or on asterisk-gui I can't assign Service Providers ? |
18:16.59 | Pumas | in logger.conf I have this line: full => notice,warning,error,debug,verbose, but how can I tell asterisk where to write the log output? |
18:17.23 | Katty | file: is there a way to make the cli dump out active extensions? |
18:17.36 | Pumas | or in what file is written the output? |
18:17.45 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
18:18.35 | WGFreewill | Pumas: /var/log/asterisk/full |
18:18.49 | TommyTheKid | [asterisk@iml-v20z-11 asterisk]$ grep log asterisk.conf |
18:18.50 | TommyTheKid | astlogdir => /var/log/asterisk |
18:19.06 | Pumas | ah ok, thanks |
18:19.09 | *** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com) |
18:19.36 | Strom_C | I have twelve T1 red alarms all blinking in unison on the back of this box |
18:19.39 | Strom_C | it's like christmas |
18:20.01 | mhz121 | Anybody know if there will be a performance hit by putting a TDM400 and a TE110P in the same server? I have a client that uses modems to dial into Air conditioning units and wants to use his new T1/PRI line to dial out with the modems (did any of that make sense?) |
18:20.38 | WGFreewill | pumas: you'll probably spot the problem |
18:20.50 | TommyTheKid | modems dial into air conditioning units.. |
18:20.54 | Katty | Strom_C: yay christmas! |
18:21.01 | Strom_C | mhz121, explain that one more time? |
18:21.03 | mhz121 | Yep, I was surprised also. |
18:21.17 | Strom_C | what do they want to do - connect the modems to the tdm400? |
18:22.08 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
18:22.24 | TommyTheKid | guessing.. connect the tdm400 to PC's (modems) with some software .. and use the PRI lines to dial out? |
18:29.41 | *** join/#asterisk SwK (n=Silik0nJ@208.44.30.242) |
18:29.41 | TommyTheKid | we had some sort of "virtual modem" software that linked into some expensive fax thing on a NT PC at one company I worked at.. that would be kinda cool |
18:29.41 | mhz121 | Client wants to purchase a T1 line and send it directly into the * server, I was thinking of using a TE110P to handle that. However he has analog POTS lines that he would love to ditch and instead use channels on the T1 when needed. so I thought of also putting in a TDM400 (with FXS modules) so that when the modem dials it creates a channel on the T1. |
18:29.41 | luke-jr_ | Any way to test if the local telephone company is connected to my apt's wall wiring? |
18:29.42 | luke-jr_ | short of dialing 911 |
18:29.42 | Strom_C | mhz121, ah ok |
18:29.42 | Strom_C | mhz121, if you're doing any kind of data calls at all, you dont want to bridge across the pci bus |
18:29.42 | aydiosmio | luke-jr_: listen for a dialtone. |
18:29.42 | Strom_C | mhz121, so i'd suggest a TE2xxP and a channel bank |
18:29.42 | luke-jr_ | mhz121: I've heard of dynamic phone/data balancing that uses PPP to establish bridged data paths over voice lines |
18:29.42 | luke-jr_ | aydiosmio: lack of dialtone only means no service, not disconnected wiring |
18:29.42 | Strom_C | luke-jr_, check for talk battery |
18:29.42 | luke-jr_ | aydiosmio: usually, the wiring would still be connected and dialing 911 would still work |
18:29.42 | luke-jr_ | talk battery? |
18:29.43 | Strom_C | luke-jr_, yes |
18:29.43 | Strom_C | plug an analog phone in and blow into the transmitter |
18:29.43 | Strom_C | if you get sidetone, you have talk battery |
18:29.43 | luke-jr_ | hm... an analog phone :) |
18:29.43 | luke-jr_ | what's sidetone? |
18:29.44 | luke-jr_ | eg, if I can hear myself? |
18:29.44 | Strom_C | yes |
18:29.52 | file | Strom knows too much... kill him! |
18:29.53 | aydiosmio | luke-jr_: usually. but one would assume no dial tone, no dialing |
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18:29.56 | L|NUX | bye all |
18:30.20 | luke-jr_ | Strom_C: if I can't, does that more-or-less guarantee they're unwired? |
18:30.21 | mhz121 | So, if I understand, better bet is to get a two channel T1 card (TE2xxP) and use one channel for the inbound feed (from service provider) and another channel as a feed to a channel bank? |
18:30.21 | luke-jr_ | aydiosmio: usually you can dial 911 despite lack of dialtone |
18:30.21 | mhz121 | Any recommendations on a channel bank? |
18:30.21 | aydiosmio | luke-jr_: this telus guide says that if you can hear yourself blowing into the reciever, there is service |
18:30.21 | Strom_C | mhz121, adtran |
18:30.21 | Strom_C | luke-jr_, well, make sure its not a phone that relies on an external power source |
18:30.21 | aydiosmio | assumign you're using a line powered phone |
18:30.22 | aydiosmio | heh |
18:30.22 | luke-jr_ | Strom_C: the one I have does, but works w/o external power minimally |
18:30.22 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
18:30.22 | TommyTheKid | it would be cool if there was some sort of "fake modem driver" that could make a "sip/iax2" connection into asterisk and dial out :) |
18:30.22 | luke-jr_ | TommyTheKid: there is in development |
18:30.22 | luke-jr_ | but it doesn't speak SIP/IAX2 natively |
18:30.22 | aydiosmio | To check whether your telephone circuit is busy or has been destroyed, blow into the receiver mouthpiece. If you cannot hear yourself in the earpiece and your telephone is connected to the wall jack, the circuit may be out of service and you will have to try another phone. |
18:30.22 | vader-- | anyone ever link asterisk with nagios in here? |
18:30.22 | vader-- | im looking for a way my nagios server can send something to asterisk so asterisk can call me if there is a problem |
18:30.22 | *** join/#asterisk p3n (n=p3n@dxb-as56546.alshamil.net.ae) |
18:30.23 | TommyTheKid | sorry.. back to my config (shuts up) |
18:32.59 | luke-jr_ | yay no AT&T junk |
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18:42.33 | DrAk0SX | anyone here using Sugar CRM with Asterisk integration? |
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18:49.25 | ucfMethod | anyone know why cell phones wont display CALLERID(name)? |
18:49.37 | TommyTheKid | they don |
18:49.54 | TommyTheKid | oh yea, mine does cause it looks them up in the phone book :) |
18:49.58 | Strom_C | ucfMethod, because mobile phones have never had caller ID name delivery |
18:50.09 | ucfMethod | Strom_C: thanks... |
18:50.39 | zx6r | ucfMethod, depends on your provider. |
18:51.10 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
18:52.12 | hoobastooba | on the polycom phones it only shows 9 of the 10 digits of a telephone number, because (I am assuming) the call timer takes up the rest of the char spaces... is there a way to change this? or am i the only person with this issue? |
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18:54.56 | diclophis-work | hello all |
18:55.06 | diclophis-work | i have a quuestion about timezones |
18:55.24 | aydiosmio | um |
18:55.38 | diclophis-work | wrong window |
18:56.11 | TommyTheKid | hoobastooba: I have a 500sp that displays it fine |
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18:58.28 | mhz121 | Strom: Looks like channel banks start at $1000, we're trying to keep the costs down (hence the $350 TDM400 idea). Do you not think that's reasonable? |
18:59.30 | mhz121 | With only four modems that are used occasionaly it's really not a justifiable expense |
18:59.38 | *** join/#asterisk Pumas (n=KAos@207.248.51.107) |
19:03.04 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
19:03.09 | drcode | hi all |
19:03.21 | drcode | I want to call to zap card |
19:03.36 | drcode | I need to load somthing in module.conf? |
19:04.29 | mhz121 | Thanks all!!! |
19:04.33 | mhz121 | Have agood day |
19:04.37 | *** part/#asterisk mhz121 (n=mhz121@128.238.244.190) |
19:05.06 | CunningPike | Is there a way to expand a regexp within Asterisk to dial all extensions that match a pattern? |
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19:14.13 | hmmhesays | do you capitalize a month in a sentence? |
19:16.31 | CunningPike | hmmhesays: ?? |
19:17.14 | aydiosmio | hmmhesays: yes |
19:17.43 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
19:18.02 | CunningPike | hmmhesays: Misread your question - yes, you do, as aydiosmio says |
19:18.19 | hmmhesays | it is considered a pronoun right? |
19:18.44 | CunningPike | hmmhesays: I think it's a regular noun |
19:19.00 | hmmhesays | hmm |
19:19.04 | hmmhesays | been too long sing english class |
19:19.09 | CunningPike | hmmhesays: iirc, a pronoun is a word like 'it', that refers to something that is itself a noun |
19:19.16 | *** join/#asterisk sjobeck (n=sjobeck@66-182-49-26.atgi.net) |
19:20.22 | hmmhesays | the wikipedia confirms that |
19:21.30 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
19:21.58 | QbY | i have mailboxes in a context, is it possible to send a message to every mailbox in that context? there are about a hundred or so |
19:22.00 | Katty | hmmhesays: mew. |
19:22.29 | hmmhesays | Katty: how are you? |
19:22.49 | Katty | hmmhesays: i'm meh. |
19:22.56 | hmmhesays | yeah about the same here |
19:23.14 | hmmhesays | got a couple projects that aren't going as well as planned |
19:23.44 | Katty | :< |
19:24.05 | hmmhesays | msn is acting up today too |
19:25.19 | *** join/#asterisk gmfm (n=aaron_pi@rtr.enterprisemtg.net) |
19:26.11 | gmfm | Nov 15 10:33:48 WARNING[26275]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/23 already in use on span 3. Hanging up owner. |
19:26.12 | gmfm | Nov 15 10:33:48 WARNING[26274]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. |
19:26.29 | gmfm | i'm getting wierd PRI errors and can't make outbound calls reliably today |
19:26.41 | gmfm | just started today on a system that's been working for months |
19:26.45 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
19:30.58 | hmmhesays | i'm not gonna lie, i'll not be a gentlemen |
19:31.03 | hmmhesays | behind the boathouse |
19:31.07 | hmmhesays | I'll show you my dark secret |
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19:33.25 | hmmhesays | this ds9 episode where they are flying the defiant manually is funny |
19:33.40 | hmmhesays | they can warp but they can't get the com system up |
19:36.29 | hmmhesays | hey file |
19:36.31 | zx6r | woah.. random possum kingdom :| |
19:36.43 | hmmhesays | zx6r: yeah I had to learn it for practice tonight |
19:36.43 | hmmhesays | woo |
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19:40.54 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
19:40.59 | drcode | hi all |
19:41.08 | drcode | I got unable to connect ZAP device |
19:41.13 | drcode | any idea? |
19:41.19 | drcode | I compile ZAP module |
19:41.22 | drcode | 1.2 |
19:41.41 | drcode | and I am using Astrisks from debian on ubuntu |
19:44.03 | *** join/#asterisk etaoins (n=ryan@209.139.250.204) |
19:44.34 | etaoins | Anyone have problems with one way audio on the Polycom SoundStation 4000? |
19:45.11 | etaoins | If I call from a Grandstream to the 4000, audio works both ways |
19:45.33 | etaoins | If I call the the 4000 to the Grandstream, audio only goes from the 4000 to the Grandstream |
19:45.59 | etaoins | If I dial out from the 4000 through zap, audio only goes from Zap to the 4000 |
19:46.08 | _rnz- | etaoins |
19:46.14 | _rnz- | why use asterisk, when you can use Call Manager? |
19:46.29 | _rnz- | ._. |
19:46.44 | _rnz- | is the budget a little low? :) |
19:46.52 | etaoins | Are you trolling? |
19:47.15 | _rnz- | No Ryan, but you may start cumming |
19:48.58 | etaoins | Way to use NickServ |
19:49.07 | etaoins | Anyone else have any ideas? |
19:49.46 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
19:50.20 | CunningPike | etaoins: Could it be a codec issue? |
19:51.06 | ucfMethod | anyone know if there is a website that has all the standard asterisk sounds so I dont have to constantly create a test extension and use Playback to hear the sounds? |
19:51.25 | etaoins | CunningPike: Both ends are set up for ulaw |
19:51.37 | etaoins | CunningPike: I can send you sip debug if you'd like |
19:51.49 | CunningPike | etaoins: Pastebin it so others can see also |
19:52.39 | _rnz- | pastebin.com is mad slow |
19:52.41 | _rnz- | the past few months |
19:52.46 | _rnz- | i use pastebin.co.uk now |
19:52.54 | drcode | why I can't run zap show channel? |
19:52.56 | drcode | any idea |
19:53.01 | drcode | "zap show channels" |
19:53.07 | drcode | no such command |
19:53.24 | etaoins | <PROTECTED> |
19:53.42 | etaoins | That's a call from the 4000 to a Grandstream |
19:54.00 | etaoins | Where audio only passes from the 4000 to the Grandstrean |
19:54.06 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tj3.cable.mindspring.com) |
19:55.54 | etaoins | 192.168.97.231 is the Polycom |
19:56.00 | etaoins | 192.168.98.8 is the Asterisk server |
19:57.49 | drcode | any idea why "zap show" dosnt work?? |
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19:58.11 | dlynes_laptop | drcode: load chan_zap.so |
19:59.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
20:00.10 | drcode | no such file |
20:00.12 | drcode | dlynes_laptop: |
20:00.38 | dlynes_laptop | drcode: that's why you don't have zap show channels then |
20:00.47 | dlynes_laptop | drcode: you didn't build the module |
20:00.50 | drcode | I use astrisks from debian |
20:00.57 | FastFeet | I need a recommendation for some hardware. I need a device which will allow me to use my existing office telephones (3 users), and allow output to a PSTN line. |
20:00.57 | drcode | I need to recompile it or somthing ? |
20:01.01 | dlynes_laptop | drcode: Did you install zaptel and libpri? |
20:01.22 | drcode | zttools give me ok |
20:01.26 | drcode | libpri? |
20:01.29 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-49cc8f76fec0dd96) |
20:01.29 | drcode | its module? |
20:01.34 | dlynes_laptop | drcode: I didn't say zttools |
20:01.38 | dlynes_laptop | drcode: i said zaptel |
20:01.48 | dlynes_laptop | drcode: i have no idea what zttools is |
20:01.50 | drcode | yes |
20:01.56 | drcode | what is libpri? |
20:02.10 | dlynes_laptop | drcode: it's a library for T1 lines |
20:02.24 | drcode | I use x100p |
20:02.27 | dlynes_laptop | drcode: but if I remember correctly, chan_zap.so doesn't get compiled if it's not there |
20:02.47 | drcode | its in zaptel module? |
20:02.53 | dlynes_laptop | drcode: no...it's separate |
20:03.09 | drcode | how I can download it? |
20:03.19 | dlynes_laptop | drcode: install zaptel, then libpri, then asterisk, then asterisk-addons |
20:03.32 | dlynes_laptop | drcode: you can install asterisk-sounds at any part of that cycle |
20:03.49 | dlynes_laptop | drcode: www.asterisk.org has all the code available for download |
20:04.08 | dlynes_laptop | drcode: you can also try asking on #debian to find out what the package name is you have to download for it, if you want to use apt-get |
20:04.14 | *** join/#asterisk martin5519 (n=marty@host-209-50-87-3.dyn.295.ca) |
20:04.43 | FastFeet | I need a device recommendation that will allow me to use my internal phones (3 users), PSTN connect, and Ethernet. I was looking at the SIPura SPA3000, but I am unable to locate them, as they have been replaced by Linksys models. Will they work? |
20:04.54 | martin5519 | Hello... can anyone tell me if they have ever used Asterisk to deliver VOIP to an appartment building? |
20:05.26 | dlynes_laptop | martin5519: what's the difference between an apartment building and any other building? |
20:05.31 | monsted | martin5519: someone has - anything special you want to do? |
20:05.32 | FastFeet | lol |
20:06.02 | dlynes_laptop | FastFeet: you're wanting to use how many analog phone lines and how many analog phones? |
20:06.14 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
20:06.24 | FastFeet | I I have 3 analog phones. |
20:06.31 | FastFeet | I telco line. |
20:06.34 | FastFeet | 1 |
20:06.45 | martin5519 | monsted... nothing fancy.. |
20:06.53 | dlynes_laptop | FastFeet: I would get a tdm400p with 3 fxs ports and 1 fxo port, myself |
20:07.00 | monsted | will the phones be sharing a single FXS port or three seperate? |
20:07.05 | dlynes_laptop | FastFeet: throw that into a pc |
20:07.24 | dlynes_laptop | FastFeet: or if what monsted says, then one fxs port and one fxo port |
20:07.33 | FastFeet | Support PSTN? |
20:07.34 | martin5519 | monsted.. essecntially, just deliver on a Amphonol interface and make sure the Asterisk box supports all the features and that users can active those features from their phone and that Asterisk can pass all calls to a VOIP gateway at anothr elocation |
20:08.40 | martin5519 | We had problems with Quintum... Mitel and Cisco are outrageously expensive |
20:08.43 | dlynes_laptop | martin5519: so you're wanting to use the existing cabling infrastructure, and hook up to the 24 pair demarc via amphenol tail? |
20:08.48 | FastFeet | BTW, I have a low budget... |
20:08.50 | martin5519 | exactly! |
20:09.03 | dlynes_laptop | martin5519: or 50 pair demarc, depending on the size of the building |
20:09.16 | FastFeet | My intenstions for this Asterisk box, is for a College Project. |
20:09.18 | dlynes_laptop | martin5519: yeah...no difference between an apartment building and any other building for that |
20:09.25 | dlynes_laptop | martin5519: how old is the building? |
20:09.59 | monsted | martin5519: cisco may be more expensive, but you usually get what you pay for |
20:10.00 | martin5519 | Well, the boards I saw on Digium that support the Amphenol are 24 ports - that's cool anyways... we can add later.. oh, these buildings have CAT3 and are probably about oh.. I do't know 30 to 50 years old maybe |
20:10.16 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
20:10.29 | dlynes_laptop | martin5519: Yeah...you'll want to get the tdm2400p's with hardware echo cancellers then |
20:10.41 | martin5519 | monsted: Big difference though between $30,000 solution and $3,000 box that I can even customize in Perl |
20:10.52 | dlynes_laptop | martin5519: You'll probably also want to get the single port or dual port pri cards with hardware echo canceller, too |
20:10.54 | martin5519 | dlynes: And echo cancellers? |
20:11.20 | martin5519 | Well, we don't need PRI cards though as we're passing off the calls via Ethernet to a VOIP Peer |
20:11.23 | dlynes_laptop | martin5519: then you can have a pri/t1 coming in, instead of a 25 pair amphenol |
20:11.40 | dlynes_laptop | martin5519: ah...ok...nvm then |
20:11.43 | monsted | martin5519: oh, i didn't mean call manager, but their gear in general |
20:12.12 | martin5519 | monsted... show be a router that can do all that for less than 5 gran.. and I'll be interested - my expertise is actually Cisco |
20:12.30 | dlynes_laptop | martin5519: there are other solutions from sangoma that use regular rj9 jacks instead of amphenol tails, as well |
20:12.55 | SheriF_SpacE | dose asterisk 1.4 supporsts video confrancing ? |
20:13.06 | martin5519 | dlynes.. yes but the problem is the lines come down from teh appartments onto BIX blocks so then we just punch down an amphonel cable on the bix |
20:13.23 | monsted | martin5519: 24 port FXO thingies? no idea, i only work on the server end of things :) |
20:13.28 | monsted | err, FXP |
20:13.37 | dlynes_laptop | martin5519: yeah..I would just use cross connect wire to hook up the rj9's |
20:13.47 | dlynes_laptop | martin5519: you must be in Canada :) |
20:13.54 | martin5519 | monsted - not sure what you call that interface - all I know it's a 25-pair Amphenol... |
20:14.07 | dlynes_laptop | martin5519: And it must be all new wiring |
20:14.08 | martin5519 | LOL! I am actually! how did you know? |
20:14.17 | dlynes_laptop | martin5519: because americans don't use bix blocks :) |
20:14.28 | martin5519 | oh.. how does it get wired? |
20:14.46 | martin5519 | Come to think of it, I've never seen a Telecom Closet down south |
20:14.46 | dlynes_laptop | martin5519: all the old crap in Canada is all 110 block or 66 block |
20:14.57 | monsted | martin5519: i'm pretty sure they have a media gateway that does SIP/MGCP/Skinny/H323 with just about any kind of interface you need |
20:15.00 | martin5519 | We;re behind.. |
20:15.04 | dlynes_laptop | martin5519: all the newer stuff is bix, and all the really new stuff is a newer standard |
20:15.27 | dlynes_laptop | martin5519: Telus just started experimenting with a new standard that has gel in the slots |
20:15.32 | dlynes_laptop | martin5519: and it's toolless |
20:15.44 | dlynes_laptop | martin5519: so even alarm guys can figure out how to use it |
20:15.49 | martin5519 | ok.... well, what do you suggest? |
20:16.10 | martin5519 | I'm no cabling expert by any means... if there's a better way, i'm all ears |
20:16.22 | dlynes_laptop | martin5519: well, it's a new enough job...check to see if the cabling's new enough to be cat5 |
20:16.25 | file | only ears? no body? |
20:16.42 | martin5519 | Right now, in the buildings, we've been redoing the cross-connects so that it connect to our amphenol - that's all I know. |
20:16.51 | dlynes_laptop | martin5519: most bix strips are cat5e compliant |
20:17.13 | martin5519 | We can't start running cat 5 to every appartment - too expensive |
20:17.34 | dlynes_laptop | martin5519: ok, so it's all cat3 for sure? |
20:17.51 | martin5519 | I'd have to double check but probably yes |
20:18.10 | dlynes_laptop | martin5519: if so, you're going to need echo can for sure...if it's cat5e you might be able to scrape buy with just a software echo can |
20:18.12 | martin5519 | It just comes down and currently punched down on the Bell BIX... I have pics too... on my laptop though |
20:19.17 | dlynes_laptop | martin5519: the cables are generally labelled |
20:19.28 | martin5519 | ok... well not CAT5E or CAT5 - so I'll get echo cancellation ... but do you thin kI can do all this with an Asterisk box? I mean... I need the low tech guy in teh office to go to a web browsser or some type of managmenet interface and be able to enable/disable ports and manage features for users. |
20:19.40 | dlynes_laptop | martin5519: whether the cables are blue or grey or white doesn't mean anything either...you can get cat3/cat5/cat5e in all colors |
20:19.52 | martin5519 | I'm in Ottawa - buildings are in Toronto though.. I'll check it out next tiem I'm there |
20:20.00 | monsted | martin5519: can't be worse than most PBX'es :) |
20:20.12 | dlynes_laptop | martin5519: easily |
20:20.24 | dlynes_laptop | martin5519: the only question is how many problems you'll run into with call quality |
20:20.28 | martin5519 | Really eh! that's awesome - are you an expert on ASTERISK? |
20:20.48 | dlynes_laptop | martin5519: I wouldn't say I'm an expert, but I have been using it for a couple of years now |
20:21.03 | martin5519 | Ah... call quality - that's were I come in - I am a networking expert ;) -... serously though... so no prob there... I've done VOIP before withouth issue |
20:21.27 | dlynes_laptop | martin5519: I'm talking about between the analog phones and the asterisk box |
20:21.33 | dlynes_laptop | martin5519: not on the voip end |
20:21.36 | martin5519 | oh.... |
20:21.58 | martin5519 | Well... it's just analog though at that point... |
20:22.02 | dlynes_laptop | martin5519: 99% of your call quality issues when dealing with analog devices is usually on the analog end |
20:22.12 | martin5519 | yeesh... |
20:22.17 | martin5519 | Good to know actually |
20:22.36 | dlynes_laptop | martin5519: that's why hardware echo can is a must if you want to be guaranteed of less problems right from the get go |
20:22.42 | martin5519 | Well, we have on ebuilding right now operating off a Quintum with 12 users .... had only 1 issue but I see what you're saying |
20:22.56 | martin5519 | So, what are these hardware devices called? |
20:23.03 | martin5519 | how do they fit on the line? |
20:23.07 | dlynes_laptop | tdm2400p/tdm400p/a200d |
20:23.10 | martin5519 | Is it a board I put inside the server? |
20:23.18 | dlynes_laptop | erm tdm400p doesn't have hardware echo can |
20:23.21 | martin5519 | .... Oh well yea... the tdm2400p is actually what I'm looking to get |
20:23.33 | dlynes_laptop | The 2400p has a hardware echo can option |
20:23.41 | martin5519 | Because that tdm2400p is the proper physical interface too... we have a winner! |
20:23.45 | dlynes_laptop | You can buy it with or without hardware echo can |
20:23.50 | martin5519 | I' |
20:23.54 | martin5519 | d buy with obvously |
20:23.58 | martin5519 | We want a rock solid service |
20:24.25 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
20:24.49 | martin5519 | See th eproblem we ran into with Quintum is it deosn't have a good interface to enable/disable ports by non-tech admin. and it deosn't do call forward and something else... and the users are not able to program features via their phones |
20:24.54 | Corydon-w | If you wanted rock solid, you wouldn't be using analog |
20:25.01 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
20:25.01 | Corydon-w | You'd be using PRI |
20:25.26 | martin5519 | Corydon - Yeah I know... but that' $500/month |
20:25.41 | dlynes_laptop | Corydon-w: if he wanted rock solid, he wouldn't be using a voip line, either |
20:25.57 | martin5519 | From an engineering standpoint, you can pull all the best stuff out to be stable... but there's a cost issue there.. |
20:26.00 | Corydon-w | martin5519: so is 12 lines of analog |
20:26.06 | martin5519 | yes |
20:26.41 | Corydon-w | Yet a PRI gives you nearly double the channels |
20:26.45 | martin5519 | yep - our current building is 12 analog lines being delivered on an amphonol interface into the Quintum which passes the calls over a DSL line (no PPP header and no data) to a VOIP Peer |
20:27.35 | Corydon-w | Would you like it good or cheap? ;-) |
20:27.39 | martin5519 | Corydon - w: Sorry, the lines come in as wire pairs form the appartments.... are you suggesting we put those wire pairs on a digital interface? |
20:28.00 | martin5519 | Then, we'd have to get Digital phones for all the units??? |
20:28.07 | FastFeet | Any alternatives to the TDM400P? Honestly, it is a little over my budget. looking to spend no more than $200 if I can help it. |
20:28.08 | aydiosmio | good, cheap, fast / good, cheap, legal |
20:28.34 | dlynes_laptop | FastFeet: are all analog phones going to be wired in parallel? Or are they going to be three separate stations? |
20:28.34 | Corydon-w | FastFeet: nope, not really |
20:28.47 | FastFeet | parallel |
20:29.00 | dlynes_laptop | FastFeet: go with the sipura 3000 or the grandstream ata-186 then |
20:29.20 | FastFeet | I was looking at the sipura 3000, but I see Linksys bought them |
20:29.21 | dlynes_laptop | FastFeet: erm...forget the grandsteam ata-186...it's not a gateway |
20:29.26 | FastFeet | are their devices the same? |
20:29.28 | dlynes_laptop | FastFeet: Sipura still exists |
20:29.35 | dlynes_laptop | FastFeet: it's just a division of Linksys now |
20:30.10 | dlynes_laptop | FastFeet: They're gradually rebranding all sipura devices as Linksys, but not all of them have been rebranded yet |
20:30.13 | martin5519 | Corydon... if we delivered the appartements on a PRI interface, wouldn't each unit need a digital phone? |
20:30.21 | dlynes_laptop | FastFeet: I think sipura 3000 is one of the ones that hasn't been rebranded yet |
20:30.26 | dlynes_laptop | martin5519: no |
20:30.34 | dlynes_laptop | martin5519: you can give them all analog phones |
20:30.43 | FastFeet | OK, any recommendations one where to purchase one? |
20:30.54 | dlynes_laptop | FastFeet: What country are you in? |
20:31.15 | FastFeet | Canada, but would be willing to look within the US as well. |
20:31.23 | dlynes_laptop | FastFeet: Are you in the east, or the west? |
20:31.37 | FastFeet | East |
20:31.39 | Corydon-w | martin5519: generally for that kind of setup, I'd do a PRI to the apartments and a channel bank out to each individual |
20:31.45 | dlynes_laptop | FastFeet: Try voxilla.ca |
20:31.53 | dlynes_laptop | FastFeet: or voipdepot.ca |
20:32.17 | martin5519 | Corydon-w: A channel bank - what's that? |
20:32.31 | FastFeet | Thanks, a big help! |
20:32.34 | Corydon-w | martin5519: splits T1 CAS into individual analog lines |
20:32.35 | dlynes_laptop | FastFeet: Or, we might even have a used one kicking around...I'd have to see how much the boss would let it go for, though |
20:32.40 | dlynes_laptop | FastFeet: We're in Vancouver |
20:33.08 | FastFeet | cool.... |
20:33.18 | FastFeet | Just looking at the linksys once again.... |
20:33.18 | martin5519 | Corydon-w: You got my curiosity... but if you break that into anaolog - aren't you back to square 1? |
20:33.22 | FastFeet | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1149639756784&pagename=Linksys%2FCommon%2FVisitorWrapper |
20:33.32 | FastFeet | This is the same thing as the Sipura 3000? |
20:33.41 | martin5519 | By the way, what deos CAS stand for? |
20:33.42 | Corydon-w | martin5519: nope, you have a superior interface to the phone company |
20:33.52 | Corydon-w | Channel Associated Signalling |
20:34.13 | FastFeet | just renamed?? |
20:34.13 | dlynes_laptop | FastFeet: nope...that one's a sipura 3000 built into a router |
20:34.22 | dlynes_laptop | FastFeet: the sipura 3000 doesn't have a router functionality |
20:34.23 | FastFeet | Confussed... |
20:34.28 | FastFeet | OK |
20:34.32 | Corydon-w | martin5519: analog signalling to the phone company is bad when you figure that they charge if you don't hangup properly |
20:34.34 | dlynes_laptop | FastFeet: that one's probably more expensive |
20:34.45 | martin5519 | Corydon-w: Ok... I think you may have missed something... we are not sending the calls to a phone company, we are sending the calls via IP to a VOIP Peers over Ethernet. We don't deliver the calls to any LE or CO |
20:34.50 | FastFeet | I C.. Your probley right.... |
20:35.03 | martin5519 | Here's a sorta map |
20:35.08 | FastFeet | I will check those links out..... |
20:35.37 | _rnz- | heh |
20:35.40 | _rnz- | i just socially engineered |
20:35.41 | martin5519 | appartements -------- cat3 ------- amphenol (VOIP BOX )Ethernet ------------- ATM CLOUD ------------ VOIP Peer |
20:35.42 | _rnz- | #freebsd |
20:35.47 | _rnz- | into a religous argument |
20:36.22 | dlynes_laptop | FastFeet: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1125638798035&pagename=Linksys%2FCommon%2FVisitorWrapper |
20:36.50 | FastFeet | That would be really sweet |
20:37.10 | FastFeet | if you had one kicking around. |
20:37.11 | dlynes_laptop | FastFeet: that's the rebranded sipura 3000; the model number is still spa 3000 |
20:37.20 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
20:37.22 | FastFeet | ok.... |
20:37.30 | dlynes_laptop | FastFeet: we've got the old sipura branded device |
20:37.59 | dlynes_laptop | FastFeet: basically the only difference is the linksys looks nicer, and it's in a cheaper feeling enclosure |
20:38.13 | FastFeet | ok, I wasn't sure... |
20:41.01 | FastFeet | <---| Newbie... |
20:41.01 | _rnz- | dlynes |
20:41.01 | _rnz- | that reminds me |
20:41.01 | FastFeet | I am looking to set up and show off Asterisk for a college senior project. |
20:41.01 | _rnz- | i have a wrt54g at home |
20:41.02 | _rnz- | i saw linksys |
20:41.02 | FastFeet | thanks for your help. |
20:41.02 | _rnz- | arent wrt54g's intensely configurable |
20:41.02 | _rnz- | , ie: to hack around |
20:41.02 | martin5519 | Corydon? |
20:41.02 | FastFeet | yup, I have linux running on mine. |
20:41.02 | Corydon-w | martin5519: what? |
20:41.03 | martin5519 | Did you see what I meant? |
20:41.03 | Corydon-w | Yes |
20:41.03 | martin5519 | so - what do you think? |
20:41.03 | martin5519 | We have this working fine right now - jsut poor features on VOIP box we're using |
20:41.03 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
20:41.03 | Corydon-w | I'd probably still go with a channel bank |
20:41.03 | martin5519 | which side the the VOIP box in the diagram - you mean facing the appartments? |
20:41.03 | Corydon-w | Yep |
20:41.14 | Corydon-w | channel bank + t1 interface |
20:41.32 | martin5519 | But does that mean each appartment gets digital and that we need a digital phone in each unit? |
20:41.36 | martin5519 | or do you mean something like: |
20:41.39 | Corydon-w | what features are you looking for? |
20:41.47 | Corydon-w | No, each apt is still analog |
20:41.52 | dlynes_laptop | _rnz-: only certain hardware versions |
20:42.02 | martin5519 | appartments----------- cat3-------- channelbank-----T1(VOIPBOX).......... |
20:42.12 | Corydon-w | Yep |
20:42.13 | dlynes_laptop | _rnz-: certain hardware versions allow you to replace the OS on them |
20:42.35 | martin5519 | All features typical to voice ... call forward... call waiting... users who can activate/deactivate features from their phone etc... |
20:42.41 | martin5519 | give me 2 seconds... client in here |
20:42.43 | FastFeet | That Grandstream ATA-186 device.... have they been bought out by Cisco as well? |
20:43.02 | Corydon-w | Generally T1 interface + channel bank is cheaper than a TDM2430 |
20:43.08 | dlynes_laptop | _rnz-: there's a mailing list dedicated to throwing asterisk onto those wrt54g's and the Linksys network storage devices |
20:43.43 | dlynes_laptop | FastFeet: If you'll scroll up, you'll see where I said forget about the ata-186, because it's just an ata, not a gateway...and afaik, it hasn't been bought by cisco...but I could be wrong |
20:44.12 | FastFeet | Your right, I missed that comment. |
20:44.15 | FastFeet | Sorry. |
20:44.28 | FastFeet | Sipura it is... |
20:45.02 | *** join/#asterisk bungalow (n=tasat-@c-24-7-62-81.hsd1.ca.comcast.net) |
20:45.11 | martin5519 | back... |
20:45.48 | martin5519 | So Corydon... oh right.. I see... makes sense.. so what type of channel bank could I buy? |
20:46.08 | bungalow | hi -- I'm getting chan_sip warning: Maximum retries exceeded on transmision ... for seqno ### (Critical Response) -- can someone explain what's happening and suggest a way to fix the problem? Is this me, my sip provider, or both? |
20:46.28 | martin5519 | And the Channel bank in that case would have the AMphenol interface... |
20:46.35 | Corydon-w | martin5519: correct |
20:46.52 | *** join/#asterisk Skarmeth (n=Skarmeth@201009095121.user.veloxzone.com.br) |
20:47.14 | martin5519 | Would it then just be easier for me to buy a Cisco Router with a PRI interface that can be configured for Voice? |
20:47.26 | martin5519 | ... considering I have a lot of experience with Cisco products |
20:48.39 | Corydon-w | I have no idea what Cisco would do this |
20:48.56 | martin5519 | Right... ok.. |
20:49.17 | martin5519 | So Corydon-w: have you done this with a Channelbank and PRI on a box? |
20:49.20 | Corydon-w | Cisco is generally far more expensive than anything else |
20:49.27 | Corydon-w | Forget PRI |
20:49.28 | martin5519 | Agreed... |
20:49.34 | Corydon-w | PRI is for the connection to the telco |
20:49.50 | martin5519 | Well, I could have done it facing the appartments as well |
20:50.29 | *** join/#asterisk letoto (n=paul@tla.xelerance.com) |
20:50.35 | martin5519 | how much deos a channel bank cost and how much does a PRI card from Digium cost? |
20:50.47 | Corydon-w | You can usually get the CAC1 channel bank for just under $400 |
20:50.57 | letoto | easy question. Does Asterisk work inside a Xen virtual machines, including timings for conference calls? |
20:51.19 | Corydon-w | letoto: probably not |
20:51.42 | letoto | so i need to give the xenU the USB bus. That's possible |
20:51.51 | Corydon-w | letoto: I wouldn't even trust it within VMware |
20:52.01 | letoto | I trust xen more then vmware |
20:52.19 | *** join/#asterisk dahunter3 (n=dahunter@168.sub-75-214-69.myvzw.com) |
20:52.19 | martin5519 | AND PRI for about the same price I imagine yet the card I need is $900 - kind of works out to the same |
20:52.19 | letoto | but what are timers used for conference? Is it still usb timings? or did it use other kernel internals? |
20:52.54 | Corydon-w | letoto: no, it uses RTC |
20:53.14 | Corydon-w | It used USB only under 2.4 |
20:53.19 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
20:53.32 | letoto | ah. so there is no point gating the usb bus to the xenu |
20:53.44 | Corydon-w | letoto: not unless you're running 2.4 |
20:53.56 | CunningPike_ | bungalow: Some critical call signaling packets are going missing - likely due to latency |
20:54.12 | *** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
20:54.20 | martin5519 | Anyone on the board know who's the biggest ASTERISK expert? |
20:54.36 | Corydon-w | This isn't a board; it's a channel |
20:54.42 | martin5519 | channel.. |
20:54.54 | monsted | hmm, these channel banks would take a PRI "FXS" port and turn it into 24/30 POTS ports? |
20:55.03 | monsted | not an item i've come in contact with before :) |
20:55.29 | Corydon-w | monsted: You're combining different technologies |
20:55.45 | martin5519 | I need an Asterisk expert |
20:55.53 | Corydon-w | martin5519: see voip-info.org and find a local consultant |
20:56.54 | bungalow | hi -- I'm getting chan_sip warning: Maximum retries exceeded on transmision ... for seqno ### (Critical Response) -- can anyone suggest an approach to debug this? |
20:56.57 | monsted | Corydon-w: ok, i must admit that ISDN in general confuses me :) |
20:57.14 | martin5519 | oh no... I don't want to go through a company... I'd rather find someone local I can pay via PAYPAL once I know this is all doable with Asterisk |
20:57.39 | Corydon-w | martin5519: it's all doable with Asterisk |
20:57.39 | monsted | i'd rather pay a company that could actually service the crap when it breaks ;) |
20:57.42 | letoto | martin5519: companies use paypal too :P We accept paypal for our Openswan contracts |
20:58.16 | martin5519 | letoto: Right, but I just want an individual Asterisk expert I can work with |
20:58.43 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
20:58.44 | monsted | martin5519: why not someone who can cover you even if one person is sick or on vacation? |
20:59.03 | martin5519 | ok |
20:59.15 | martin5519 | maybe a company would be a good thing |
21:00.34 | *** join/#asterisk jjasper (n=jjasper@h-66-112-162-129.connactivity.com) |
21:01.17 | CunningPike_ | martin5519: Do you have a specific problem, or are you just looking for support in general? |
21:01.40 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
21:02.13 | brif8 | Hi All, Do we ask here about 1.4 GUI or asterisk-gui I have installed 1.4 to test and can't configure Service Providers or Calling Rules. I have got Users configured but that's about it ?? |
21:02.15 | martin5519 | CunningPike: I have a requirement that I know I'm going to need help with getting rolling. I |
21:02.26 | aydiosmio | craigslist! |
21:03.11 | CunningPike_ | brif8: Try #asterisk-gui |
21:03.14 | martin5519 | CunningPike: Essntially, I need someone who knows Asterisk so well that they can assurem it can deliver what we need. Then, we order th part, bring up the server, give the consultant a an account via SSH and log in and help us out. I'm very techincal but it will take me 10 times the amoutn of time figuring things out |
21:03.49 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
21:04.04 | Strom_C | martin5519, I do consulting, and let me tell you - having someone on-site is EXTREMELY useful |
21:05.05 | martin5519 | Strom_C: For us, we have our setup as if we were on site: Remote Reboot box (APS Masterswitch) and Console Access via backup dial-up |
21:05.19 | Strom_C | and what about the phones? |
21:05.20 | martin5519 | We can't be everywhere... :)... our buildings are in Toronto - w'ere in Ottawa |
21:05.28 | martin5519 | The phones? |
21:06.14 | CunningPike_ | martin5519: I'm in BC - drop me an email and I'll see if I can help you |
21:06.17 | hoobastooba | i see that in "/var/spool/asterisk/voicemail/context/extension" there are four versions of the recording... gsm,wav,WAV and text. I have set in the [general] context format=wav and in the context i am using for those extensions format=wav. is that all i have to change? or is there something else... I only want the .wav files in voicemail |
21:08.30 | CunningPike_ | hoobastooba: That should be it - if there are multiple versions of a file present, I think asterisk will pick the 'least cost' one - the .conf settings only affect the recording of new files. You will need to delete the others if you want the wav ones to play |
21:09.21 | hoobastooba | and once the recordings are listened to are they supposed to be moved from the INBOX to a new location? |
21:10.59 | CunningPike_ | hoobastooba: Yes - they should end up in the 'Old' folder |
21:11.24 | hoobastooba | ok, thanks |
21:11.36 | *** join/#asterisk adorah (n=admin@87.68.144.118.cable.012.net.il) |
21:11.37 | *** join/#asterisk dahunter3 (n=dahunter@168.sub-75-214-69.myvzw.com) |
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21:15.45 | letoto | martin5519: Xelerance has a presence in Ottawa and Toronto. www.xelerance.com |
21:20.44 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
21:26.53 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
21:27.32 | martin5519 | CunningPike - did you get my e-mail? |
21:27.47 | CunningPike | Yes - I did, thanks |
21:29.19 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com) |
21:33.35 | CunningPike | martin5519: Check your pm |
21:37.46 | arcanine | what is digium's equivalent for rhino r4fxo card |
21:38.04 | *** join/#asterisk icel (n=dan@63.78.162.77) |
21:38.29 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
21:38.35 | Strom_C | arcanine, four FXO ports? |
21:38.56 | arcanine | tdm04b |
21:38.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net) |
21:39.06 | Strom_C | yes |
21:39.12 | arcanine | k thanks |
21:39.48 | *** join/#asterisk icel (n=icel@63.78.162.99) |
21:41.13 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
21:41.36 | hoobastooba | is it possible to make it so that when voicemail creates a directory for an extension, it creates it with different read write permissions that what is currently done by default? I need each directory to be 744 and i dont want to have to go back and modify it each time one is created. |
21:41.39 | hoobastooba | possible? |
21:42.28 | CunningPike | hoobastooba: umask? |
21:42.36 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
21:42.58 | *** join/#asterisk db1310 (n=danny@216.77.58.40) |
21:43.09 | hoobastooba | CunningPike: what is umask? |
21:43.19 | CunningPike | hoobastooba: Scratch that - won't work anyway :) |
21:43.30 | CunningPike | hoobastooba: What are you trying to accomplish? |
21:44.03 | CunningPike | martin5519: Did you get my pm? |
21:44.05 | db1310 | can someone please tell me where the xilink files for the zapata open Tormenta 2 card is? |
21:44.17 | hoobastooba | i need to be able to send files from another server/user to the "/var/spool/asterisk/<context>/<extension>" directory for an application I am writing. |
21:44.41 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
21:50.05 | *** join/#asterisk bblack (n=bblack@129.174.111.120) |
21:50.36 | db1310 | anyone know where i might get support for the zapata hardware project? |
21:51.11 | CunningPike | hoobastooba: Hmm - well, we actually store our voicemail on an NFS mount, so we use the anonuid argument in /etc/exports to make sure that the remote server uses a particular uid to create the files |
21:51.34 | justinu|laptop | db1310: good luck |
21:51.45 | Waverly360 | I need NTP help. My NTP..it no worky..and I'd like to know why. |
21:52.02 | CunningPike | hoobastooba: So, your other server could NFS to your asterisk server using the same uid that owns the existing files |
21:52.04 | Waverly360 | What's wrong you ask? |
21:52.07 | *** join/#asterisk emp (n=emp@host-69-144-157-39.bln-mt.client.bresnan.net) |
21:52.18 | CunningPike | Waverly360: What's wrong? |
21:52.20 | CunningPike | :) |
21:52.22 | Waverly360 | :) |
21:52.25 | aydiosmio | No, I ask "Why are you askign here?" |
21:52.26 | pigpen | Can anyone tell me if Asterisk can kill itself, then relaunch on it's own? |
21:52.40 | db1310 | justinu|laptop: so, former developers to busy making money on the project? |
21:52.53 | Waverly360 | Phones aren't getting time...they're just flashing. They're polycoms...I've had port 123 opened from the asterisk box to the outside world for udp |
21:52.56 | ucfMethod | Anyone know what format wav file, if any the blackberry's support? The emails w/ wav attachments that are being sent from Asterisk are not playable by the phones. |
21:52.57 | CunningPike | pigpen: No - it can relaunch itself if something else kills it, but it doesn't kill itself ;) |
21:53.16 | aydiosmio | you can send signal HUP |
21:53.19 | emp | what's a good distro to run asterisk? for the lazy :) trixbox? something else? |
21:53.21 | justinu|laptop | db1310: that's a good question... i haven't heard about tormenta cards in a while |
21:53.34 | CunningPike | emp: The best distro is the one you know the best |
21:53.35 | Waverly360 | Unfortunately for me..I'm not really sure how the ntp stuff works. Does asterisk talk to the ntp server, and then serve time to the phones? or do the phones talk to the ntp server themselves? |
21:53.36 | aydiosmio | that usually will do a clean reload on a process |
21:53.51 | CunningPike | Waverly360: The phones speak directly to an NTP server |
21:53.55 | aydiosmio | NTP is sepearate |
21:54.01 | pigpen | hmm...I had a box kill itself off, then bring it back up. Originally, it was running as a process "/usr/sbin/asterisk -U asterisk" now it is "asterisk" and my new MOH music is playing.... |
21:54.10 | Waverly360 | Ok..how do I tell the phones to talk to the ntp server I have running on my pbx? |
21:54.27 | pigpen | ie: I added several new music files, but I had not reloaded yet....old music was left intact. |
21:54.46 | CunningPike | Waverly360: In your sip.cfg, you need to set the IP address of the NTP server - hang on a sec..... |
21:54.52 | hoobastooba | <PROTECTED> |
21:54.55 | Waverly360 | CunningPike: kk |
21:55.21 | pigpen | Sorry to sound like a dumbass...but .... "safe_asterisk" ? |
21:55.40 | Waverly360 | pigpen: it's a process that watches asterisk. If asterisk dies..it restarts it. |
21:55.46 | pigpen | hmm.... |
21:55.53 | CunningPike | Waverly360: <SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="132.246.168.164" tcpIpApp.sntp.gmtOffset="-28800" etc etc etc |
21:55.57 | pigpen | works for me...that would at least explain it. |
21:56.09 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
21:56.27 | pigpen | It was down for about 5 min...but of course 250 phones will take a few min to re-register... |
21:56.27 | Waverly360 | CunningPike: Ok...so even better question...if tcpIpApp.sntp.address="" then where would my phones get their time from? |
21:56.41 | CunningPike | Waverly360: Nowhere :D |
21:56.49 | Waverly360 | then how...do my phones have time? |
21:57.10 | CunningPike | Waverly360: Does your DHCP server serve an NTP option? |
21:57.25 | Waverly360 | CunningPike: *gasp* I think so....forgot about that. |
21:57.37 | Waverly360 | CunningPike: hold please :) |
21:57.40 | CunningPike | Waverly360: ;) |
21:57.41 | hoobastooba | pigpen: check the wiki on safe_asterisk |
21:57.50 | pigpen | k |
21:57.50 | hoobastooba | easier than explaining. |
21:58.04 | Waverly360 | CunningPike: It does! oooh...ok..it's all coming together..either that or I'm blacking out again..we'll see |
21:58.11 | CunningPike | Waverly360: heh heh |
21:59.17 | *** join/#asterisk CharlesR (n=charlesr@cpe-76-188-71-88.neo.res.rr.com) |
21:59.59 | db1310 | anyone know who to ask to get the full set of design files for the pci zapata card, its open license but all of the files are not available. Doesn't that violate the license? |
22:00.52 | _rnz- | (BREAKING NEWS)(AP/REUTERS) - Iran has launched 4 nuclear missiles into Israel causing unprecedented, and widescale devastation in Tel Aviv. Details Soon. |
22:01.52 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
22:02.46 | erubright | exit |
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22:11.22 | Dovid | _rnz-: where did u see that report ? |
22:11.36 | pigpen | yeah...I can't find anything. |
22:11.59 | Dovid | some one must be bored |
22:12.20 | *** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com) |
22:12.28 | Dovid | i am in israel now - if it did happen i would know about it |
22:13.11 | pigpen | lets hope it didn't happen... |
22:13.16 | justinu|laptop | that's a joke |
22:13.20 | justinu|laptop | a poor idea of a joke, at least |
22:13.46 | Tond | hi in my extentions i have a _98. extention as well as _9821. extentions pattern, but when i dial 982144 it matches _98. pattern, whyc does it do that? Besides _9821. pattern is created before _98. ! |
22:14.16 | Dovid | where is a mod when u need some one to bounce an ass ? i freaked out for a sec. but on the other hand i am 30 min from TLV. had it happend i would of heard it |
22:14.41 | bblack | Tond: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
22:14.46 | puzzled | Tond: if you do "show dialplan" from the cli then you can see how asterisk interprets the order |
22:15.06 | Tond | Thanks guys... |
22:15.07 | Tond | :) |
22:16.09 | Dovid | test |
22:16.19 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
22:17.33 | _rnz- | Dovid |
22:17.36 | _rnz- | it was a fake headline |
22:17.40 | _rnz- | I take it by your name: Dovid |
22:17.42 | _rnz- | you are israeli :) |
22:17.45 | Dovid | NOPE |
22:17.47 | Dovid | american |
22:17.58 | _rnz- | why o, and not a? |
22:18.00 | Dovid | but that was real sweet for scrain the shit outa sme one |
22:18.02 | _rnz- | Dovid sounds cripled, heh |
22:18.04 | _rnz- | lol |
22:18.08 | _rnz- | you get scared? : |
22:18.08 | Dovid | its my legal name |
22:18.19 | _rnz- | Dahvid = Dovid |
22:18.23 | _rnz- | David = Day-vid |
22:18.28 | Dovid | for a moment then realized had there been one i woulda hear it from here |
22:18.34 | Dovid | Dovid = Duh-Vid |
22:19.40 | icel | what is the best way to hook up multiple (>100) digital phones? Surely you don't have to have a fxs module for each one? |
22:20.39 | monsted | icel: IP phones? a switch port :) |
22:20.42 | CunningPike_ | icel: What do you mean by 'digital' phones? |
22:20.49 | icel | not a soft phone |
22:21.04 | monsted | there are physical IP phones too |
22:21.24 | icel | that might be a good option then |
22:21.42 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
22:21.49 | monsted | looks like a regular phone but has a standard ethernet port (or two) in the back |
22:21.56 | Dovid | then u use a switch |
22:22.11 | Dovid | thats when asterisk gets cheap |
22:22.16 | monsted | with some of them, you can even use the switch port peoples PC is in and plug the PC into the phone |
22:22.17 | Dovid | u dont have to pay for ports |
22:22.18 | icel | CunningPike: digital as in receives power from phone switch |
22:22.35 | JT | huh |
22:22.42 | monsted | icel: PoE switch ports and a compatible phone |
22:22.44 | JT | that's got nothing to do with the definition of digital |
22:22.57 | CunningPike_ | icel: So, IP phones? |
22:23.02 | icel | So if I have a voice T1, a digium card to use it, then my cheapest option is to use IP phones and throw them onto a switch |
22:23.11 | JT | both analogue phones, digital key system phones, and ip phones receive power from the switch |
22:23.12 | monsted | icel: correct |
22:23.24 | JT | a PoE capable switch |
22:23.26 | icel | this is like school, it is good for me |
22:23.26 | Dovid | icel: be carefull with what phones u get. in general cheap phones suck (IMHO) |
22:23.42 | _rnz- | NEW YORK - In an account his publisher considers a confession and some media executives call revolting, O.J. Simpson plans a book and TV interview to discuss how, hypothetically, he could have killed his ex-wife and her friend. |
22:23.44 | JT | it's probably not the *cheapest* solution, but one of the best solutions |
22:23.46 | _rnz- | http://www.msnbc.msn.com/id/15723351/ |
22:24.05 | icel | Even with good phones it should be cheaper than having to buy billions of modules and hardware though I would think |
22:24.12 | Dovid | yup |
22:24.17 | Dovid | thats y we luv asteris |
22:24.19 | Dovid | asterisk* |
22:24.19 | _rnz- | One expert noted that the justice system’s protection against double jeopardy means Simpson’s book, explosive as it may be, should not expose him to any new legal danger. |
22:24.40 | monsted | icel: it's certainly a lot easier |
22:24.41 | JT | icel: it wouldn't be cheaper than getting cheap analogue phones, T1/E1 card and channel bank |
22:24.58 | icel | JT: ok |
22:25.04 | JT | _rnz-: do you really need to paste all that? |
22:25.17 | icel | is he still talking? I did an ignore a while ago |
22:25.22 | JT | _rnz-: it's offtopic, and that last paragraph is obvious to most people who know anything about the law |
22:25.49 | icel | Is there any good documentation about ways to design the whole bloody phone system? |
22:26.01 | JT | ~thebook |
22:26.02 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
22:26.07 | icel | I find lots of good documentation, but it seems to lack in architecture |
22:26.08 | icel | k |
22:26.14 | JT | it won't tell you how to design it really |
22:26.19 | JT | that's still up to you |
22:26.24 | JT | what sort of architecture? |
22:26.45 | icel | I just want to know good and feasible ways that I can connect hardware to software in a 100+ phone environment |
22:27.02 | justinu|laptop | icel that comes from experience, and i have yet to see a single good telephony architecture handbook |
22:27.05 | Dovid | icel: read the book |
22:27.08 | *** part/#asterisk letoto (n=paul@tla.xelerance.com) |
22:27.14 | justinu|laptop | try newtons telecom dictionary for a bit of help maybe |
22:27.15 | JT | you'd really want to go VoIP phones unless there was an overriding reason not to |
22:27.39 | bblack | icel: a good read: http://www.amazon.com/Carrier-Grade-Voice-Over-IP/dp/0071406344/sr=8-10/qid=1163629613/ref=pd_bbs_sr_10/104-6750231-8639109?ie=UTF8&s=books |
22:27.43 | _rnz- | icel, a good solution to that is |
22:27.47 | _rnz- | Call Manager Express |
22:27.52 | icel | I figured that would be good so I messed around with SIP a lot and got softphones working how I want, but now its time to try hardware |
22:28.11 | monsted | with 100 phones, i'd go for a full Call Manager (or asterisk) |
22:28.18 | icel | thanks so far on the uRLS and ideas, keep em comin! |
22:28.34 | _rnz- | monsted, im using the ccvp call manager express simulator |
22:28.39 | _rnz- | and it has the scenarios which he mentions |
22:28.44 | Waverly360 | CunningPike: Oh..by the way..I believe you solved my problems. Mucho appreciation. |
22:29.11 | icel | whaddya mean a full call manager or asterisk |
22:29.13 | CunningPike_ | Waverly360: Great! |
22:29.20 | Waverly360 | CunningPike: Oh..and btw...remember WAAAAY back when I was asking all kinds of questions about presence with the Polycoms in asterisk? |
22:29.21 | JT | call manager would take the problems out of your hands |
22:29.23 | JT | but cost++ |
22:29.33 | JT | icel: they're spruiking cisco call manager |
22:29.34 | icel | do you mean something different from the asterisk software pbx i am using? |
22:29.53 | justinu|laptop | yeah, they're talking about cisco's integreated voip pbx |
22:29.55 | Waverly360 | CunningPike: Well..turns out the Polycoms will send out the information properly...but only after some weird freaking subscribe is sent to it |
22:30.08 | monsted | icel: Cisco Call Manager is a similar bit of software, but a bit more "slick", CCM Express is a smaller version on Cisco routers |
22:30.13 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
22:30.24 | Waverly360 | CunningPike: Live Communication Server is currently the only thing that supports it, and the way polycom is doing presence is different than any of the other phones out there. |
22:30.24 | _rnz- | monsted, express is for a small business primarily |
22:30.27 | _rnz- | that has a lower budget |
22:30.33 | monsted | _rnz-: yep |
22:30.52 | AsteriskMonkey | has any one experinece a problem with voicemail where is continually fills up with 5 second blank voicemails from the same caller id? |
22:30.54 | CunningPike_ | Waverly360: Are we still talking about NTP? |
22:31.02 | _rnz- | what is digiums relevance to asterisk again? |
22:31.05 | _rnz- | their HQ is in town here |
22:31.06 | Waverly360 | CunningPike: no...presence from a long time ago |
22:31.08 | _rnz- | in huntsville, AL |
22:31.08 | monsted | _rnz-: they made it :) |
22:31.18 | Waverly360 | CunningPike: in reference to the MyStat softkey on the phones |
22:31.19 | CunningPike_ | Waverly360: Oh - sorry :) |
22:31.29 | _rnz- | when you say they? the one guy, or a team of engineers? |
22:31.34 | _rnz- | if its open source now, what does the company do? |
22:31.40 | JT | makes hardware |
22:31.41 | JT | seriously |
22:31.43 | _rnz- | fabricate hardware asterisk integration solutions? |
22:31.44 | JT | www.digium.com |
22:31.45 | justinu|laptop | they made parts of it, other parts were contributed by ppl not connected with digium at all |
22:31.45 | CunningPike_ | Waverly360: Did you find out what the 'weird freaking subscribe' was? |
22:32.27 | _rnz- | where can I download it? id be curious to lok at it at home |
22:32.34 | _rnz- | and compare it to ccm |
22:32.36 | Waverly360 | CunningPike: My friend has a copy of it. He's currently working to implement it in Freeswitch..but he's having a helluva time. |
22:32.37 | JT | www.asterisk.org |
22:32.48 | JT | next url you need is www.google.com |
22:32.51 | justinu|laptop | Waverly360: who's your friend? |
22:32.52 | justinu|laptop | pdt? |
22:32.59 | _rnz- | in the 3 of 5 tests ive passed in the CCVP test(s) so far |
22:33.03 | _rnz- | i havent heard asterisk mentioned once |
22:33.09 | Waverly360 | justinu|laptop: yep |
22:33.11 | justinu|laptop | Waverly360: great guy |
22:33.29 | Waverly360 | justinu|laptop: Yep...killer smart too |
22:33.29 | monsted | _rnz-: well, are you really surprised that Cisco doesn't talk about Asterisk in their tests? |
22:33.31 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) [NETSPLIT VICTIM] |
22:33.50 | JT | cisco only supports cisco for their cisco phones |
22:34.05 | bblack | Waverly360: did you also play with agent login on Polycoms? |
22:34.05 | monsted | Cisco only supports Cisco for anything :) |
22:34.06 | JT | if you only listen to stuff cisco says, you'll get a very narrow world view :] |
22:34.14 | justinu|laptop | heh |
22:34.16 | Katty | which conf file is responsible for the master.csv? |
22:34.27 | JT | monsted: oh, they're at least forced to make their gear talk tcp/ip properly for other stuff |
22:34.29 | justinu|laptop | cdr-csv.conf? |
22:34.31 | justinu|laptop | something like that |
22:34.34 | _rnz- | cisco supports only cisco for the cisco file, because cisco ciscos the cisco chip on the cisco phone, that only works on the cisco software? |
22:34.36 | Waverly360 | bblack: I used to, until we realized it was locking up our PRI |
22:35.08 | JT | their sip phones don't talk the sip standard very well though |
22:35.09 | monsted | there are three realities in the cisco tests - there's the way it's supposed to work, there's the way cisco works and there's the way the real world works ;) |
22:35.21 | _rnz- | cisco = $$$ |
22:35.24 | _rnz- | open source = potential $$$ |
22:35.31 | _rnz- | salary talk |
22:35.38 | Katty | justinu|laptop: do you think that the postgres conf file will play nicely with mysql? |
22:35.43 | _rnz- | havent heard one major defense contractor in town |
22:35.46 | _rnz- | that uses asterisk here |
22:35.46 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
22:35.46 | bblack | Waverly360: I saw the work done by bweschke in 6119 but it's out of date for 1.4 and didn't really have time to see what's going on and update it |
22:36.06 | monsted | Nobody ever got fired for buy IBM^H^H^HCisco :) |
22:36.08 | justinu|laptop | Katty: play nice how? you want to log CDR to a mysql database, and a postgres database? |
22:36.09 | _rnz- | Im thinking asterisk is used by individuals or companies |
22:36.14 | _rnz- | who want to be "bleeding edge" |
22:36.14 | monsted | buying, even |
22:36.15 | _rnz- | or "alternative" |
22:36.22 | CunningPike | Waverly360: Does your friend happen to have the signaling that Polycoms need to display the ID of the called party? |
22:36.24 | Katty | justinu|laptop: to a mysql database....we don't have postgres installed on any machines. |
22:36.35 | JT | _rnz-: i'm thinking you like to talk a lot |
22:36.36 | monsted | _rnz-: it's used by thousands of companies who want a good, cheap soft-pbx |
22:36.37 | Katty | justinu|laptop: and another guy's going to write a lil website app to query it, etc |
22:36.41 | JT | and have little knowledge |
22:36.47 | justinu|laptop | Katty: for mysql, i think you need a cdr logging module out of asterisk-addons?? |
22:37.04 | Waverly360 | CunningPike: It's possible. |
22:37.05 | _rnz- | monsted, the 8 companies ive setup the ccm here, none of em even think twice |
22:37.09 | _rnz- | about having a free solution |
22:37.15 | _rnz- | they want a product that has a good foundation/history :) |
22:37.29 | Waverly360 | CunningPike: Sadly I don't know near as much about it as he does. |
22:37.35 | monsted | _rnz-: i know - i make CCM solutions for a living too |
22:37.37 | JT | _rnz-: get the pole out of your arse |
22:37.41 | CunningPike | Waverly360: Does he some here? |
22:37.42 | _rnz- | It's like telling a windows user: Use linux, because its free |
22:37.43 | JT | _rnz-: seriously |
22:37.49 | JT | _rnz-: did you come here just to troll? |
22:37.51 | monsted | (well, CCM and Nortel CS1000 stuff) |
22:38.04 | Waverly360 | He used to, but he sticks to the freeswitch channels mostly these days. |
22:38.05 | JT | err there are other reasons to use asterisk apart from price |
22:38.10 | JT | and you can pay for it if you want to |
22:38.24 | CunningPike | It's funny when you have /ignore set -you can see all the replies but none of the trolling :) |
22:38.27 | Waverly360 | CunningPike: I can maybe convince him to login if you want. |
22:38.28 | Katty | justinu|laptop: umm umm, i found it. |
22:38.36 | Katty | justinu|laptop: any documentation on how to actually use it? (= |
22:38.37 | CunningPike | Waverly360: That would be great........ |
22:39.03 | justinu|laptop | Katty: i can only think to check the README files, and the sample config files |
22:39.09 | CunningPike | Waverly360: What we are interested in is sending whatever message to the Polycom that it needs to display the CID of the called party |
22:39.43 | justinu|laptop | Katty: they might also tell you to set it up using ODBC, but that's more complicated, and if you ask me ODBC is evil |
22:41.03 | Waverly360 | CunningPike: Hmm..y'know I think he wrestled with something similar awhile back...not sure he found an answer though :( |
22:41.21 | Katty | justinu|laptop: i found a lil doc on it, dankou! |
22:41.23 | Waverly360 | CunningPike: Lemme forward that stuff onto him, and see if he won't log in and chat with you. |
22:41.26 | CunningPike | Waverly360: OK - we're bugging our tech rep about it |
22:41.30 | justinu|laptop | no prob ;) |
22:41.30 | CunningPike | Waverly360: Cool |
22:41.57 | justinu|laptop | CunningPike: i know how to do that. |
22:42.08 | CunningPike | justinu|laptop: You do? Do tell........ |
22:42.14 | JT | _rnz is afk - tossing |
22:42.23 | justinu|laptop | CunningPike: who's "we"? ;) |
22:42.32 | JT | _rnz-: it's "asterisk" btw, no s on the end |
22:42.39 | monsted | I would like to point out that not all cisco geeks are such insufferable twats ;) |
22:42.49 | JT | yeah |
22:42.50 | CunningPike | justinu|laptop: We are the District of North Vancouver in BC, Canada? Why |
22:43.01 | justinu|laptop | just curious... |
22:43.07 | justinu|laptop | i'll dig up the specs |
22:43.08 | JT | i've done the ccna |
22:43.17 | JT | not an excuse not to use my own brain though :) |
22:43.18 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
22:43.21 | CunningPike | justinu|laptop: Cool |
22:43.32 | CunningPike | justinu|laptop: That would be awesome |
22:43.37 | justinu|laptop | basically, it's done by sending the phone a particular Remote-Party-ID tag in the 180/183/200 OK message |
22:43.44 | justinu|laptop | of the party you connected to |
22:43.47 | justinu|laptop | or are ringing to |
22:45.12 | bblack | CunningPike: maybe you want to check this: http://bugs.digium.com/bug_view_page.php?bug_id=6643 |
22:45.50 | *** join/#asterisk jtf0518 (n=jaytee@c-69-137-243-25.hsd1.in.comcast.net) |
22:45.51 | justinu|laptop | that sounds like the right patch, yeah |
22:46.07 | _rnz- | if everyone in here actually has any legitimate voip desires/interests |
22:46.12 | _rnz- | ccvp would be beneficial to anyone |
22:46.19 | justinu|laptop | i even got credit for help in that patch ;) |
22:46.25 | _rnz- | just because theres an anti cisco sentiment in here |
22:46.27 | _rnz- | is no reason to take it |
22:46.30 | puzzled | justinu|laptop: does #6643 work on 1.2 also? |
22:46.32 | _rnz- | to not take it rather |
22:46.40 | CunningPike | bblack, justinu|laptop: Great - thanks! |
22:46.53 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
22:46.53 | justinu|laptop | puzzled: yeah, it was made for 1.2 |
22:47.15 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
22:47.18 | bblack | puzzled: patch made for 1.2.4 |
22:47.20 | jtf0518 | I have a couple questions about Asterisk. First one is how well does it scale? |
22:47.31 | justinu|laptop | lol, talk about a loaded question! |
22:47.36 | _rnz- | jtf, when compared to call manager, |
22:47.36 | shido6 | to the sky |
22:47.38 | JT | _rnz-: why would you convince anyone to take the ccvp after your prior behaviour in here? |
22:47.40 | _rnz- | its not looking to good :) |
22:47.53 | CunningPike | jtf0518: Search the wiki for Asterisk-at-large |
22:47.58 | bblack | jtf0518: no offence, but you can write a book about it ;) |
22:48.00 | JT | jtf0518: ignore _rnz-, he's just a cisco troll |
22:48.02 | jtf0518 | Thanks CunningPike |
22:48.05 | puzzled | justinu|laptop: cool. which one is for 1.2? gork-calledparty.diff or gork-calledrpid-trunk.diff? |
22:48.17 | jtf0518 | JT, I kinda figured that already |
22:48.19 | _rnz- | jtf, ignore JT, hes just bitter |
22:48.23 | _rnz- | :) |
22:48.31 | JT | who won't even enter into any discussion with people |
22:48.32 | _rnz- | hes to lazy to go for a ccvp, or ccie in voice |
22:48.32 | bblack | puzzled: first |
22:48.35 | JT | he just keeps trolling |
22:48.41 | JT | "too" |
22:48.41 | puzzled | bblack: thanks |
22:48.42 | _rnz- | he sticks to generic networking knowledge |
22:48.49 | justinu|laptop | man, gork didn't even get any karma points for that patch |
22:49.00 | _rnz- | if JT asked someone how RIP or ospf worked, he would say: IP routing works by routing a packet of kluged packets through a maze of mess |
22:49.03 | _rnz- | without even knowing theory |
22:49.04 | _rnz- | :) |
22:49.04 | AsteriskMonkey | no zconfig.h anymore? |
22:49.06 | justinu|laptop | and he's still active on the issue, it seems |
22:49.12 | _rnz- | or if someone asked JT that rather |
22:49.12 | JT | i have more networking and telecommunications knowledge than your small little head could fathom, _rnz- |
22:49.33 | _rnz- | if someone asked JT how rip or ospf worked, he would say: IP routing works by routing a packet of kluged packets through a maze of mess |
22:49.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
22:49.39 | _rnz- | he wouldnt know BGP if it smacked him in the face |
22:49.44 | JT | that makes no sense |
22:49.49 | jtf0518 | I have a Nortel Meridian Option 11C and I'd like to connect an Asterisk server to it so I could route internal calls in our organization both ways with 4 digit dialing. I don't want to go with the Nortel solution and I've been looking at alternatives. |
22:49.51 | AsteriskMonkey | nor an AS number :P |
22:49.52 | aydiosmio | so |
22:49.53 | JT | worst pay out i've ever seen |
22:49.57 | aydiosmio | are you two gonna shut up? |
22:50.00 | _rnz- | aydiosmio |
22:50.02 | _rnz- | ask him |
22:50.03 | _rnz- | dont know :) |
22:50.12 | monsted | _rnz-: just because someone doesn't use Cisco and has no Cisco certs doesn't mean they know nothing about networking |
22:50.17 | JT | AsteriskMonkey: who was that in reference to? |
22:50.17 | shido6 | thats on the wiki jtf0518 |
22:50.37 | jtf0518 | thanks shido, that would be www.asterisk.com? |
22:50.42 | shido6 | no |
22:50.46 | shido6 | www.voip-info.org |
22:50.50 | Pumas | I have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation |
22:50.55 | jtf0518 | thank you, shido6 |
22:51.00 | Pumas | help me please |
22:51.00 | CunningPike | jtf0518: We did exactly that :) |
22:51.15 | monsted | i find the Nortel (CS1000-friendly) phones quite horrible |
22:51.17 | jtf0518 | CunningPike, can I ask you a specific question? |
22:51.24 | CunningPike | jtf0518: Of course |
22:51.44 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
22:51.58 | hoobastooba | what is the dial string for dialing more than one extension at the same time? |
22:52.02 | CunningPike | justinu|laptop: Is SIPCalledRPID (http://bugs.digium.com/bug_view_page.php?bug_id=6643) in 1.4? |
22:52.03 | shido6 | whoops, http://www.pham.org/asterisk/asterisk-meridian-a1.pdf is missing. |
22:52.04 | jtf0518 | I've looked at the Digium products for Asterisk. How did you interface your Option 11C with Asterisk? |
22:52.18 | CunningPike | hoobastooba: Dial(SIP/ff&SIP/bar) |
22:52.49 | shido6 | what kind of interfaces do u have jtf0518 ? |
22:52.53 | justinu|laptop | CunningPike: apparently not |
22:52.59 | CunningPike | jtf0518: We have a TE411P and run 2 PRIs to the Nortel - we needed 2 to get the CID to pass in both directions |
22:53.03 | justinu|laptop | one of those things that wasn't very important, I guess |
22:53.18 | justinu|laptop | it required a revised ast_indicate API to work "right" |
22:53.23 | justinu|laptop | but that is in 1.4, I believe |
22:53.44 | jtf0518 | which Nortel PRI cards are you using? |
22:54.00 | CunningPike | jtf0518: Now you're asking tough questions lol |
22:54.01 | hoobastooba | did not understand that... if I want to dial sip/7000 and sip/7001 what would that look like? I thought it was Dial(SIP/7000),(SIP/7001) but its not working... looks like by your example i am way wrong |
22:54.13 | CunningPike | jtf0518: Hang on while I ask |
22:54.21 | Pumas | I have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation |
22:54.23 | CunningPike | hoobastooba: Dial(SIP/7000&SIP/7001) |
22:54.30 | hoobastooba | ah, ok,thanks |
22:55.26 | CunningPike | justinu|laptop: I wonder is there a SIP-Header that would do the same thing..... |
22:56.32 | justinu|laptop | there's another way to do it, with P-Asserted-Identity |
22:56.36 | justinu|laptop | i think that's the RFC compliant way |
22:56.45 | justinu|laptop | i know for a fact, polycom supports the RPID method. |
22:57.00 | justinu|laptop | but it might also support the P-Asserted-identity method |
22:57.16 | JT | AsteriskMonkey: haven't had that problem myself, but who was that comment regarding ASNs aimed at, before? |
22:58.04 | AsteriskMonkey | dont know just saw someone making fun of someone who knew nothing about bgp so i threw in the asn comment :) |
22:58.05 | AsteriskMonkey | lol |
22:58.35 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
22:58.48 | JT | AsteriskMonkey: you aided a troll who knew nothing about the knowledge of the person they were bagging out, actually :P |
22:59.36 | JT | no matter :) |
23:00.07 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
23:00.26 | monsted | he was just pissed when he found out that having cisco certs didn't mean you could actually do anything useful |
23:00.34 | justinu|laptop | CunningPike: i consider it a pleasure to help out the District of North Vancouver, BC, Canada ;) |
23:00.44 | JT | oh i think he was here for no good since the beginning |
23:01.06 | aydiosmio | amg r0d3nt|m |
23:01.09 | monsted | JT: possibly just attempting a dicksize competition |
23:01.15 | JT | possibly |
23:01.22 | monsted | JT: "Yeah, that's fine, but i'm better than you because i know Cisco!" |
23:01.31 | CunningPike | justinu|laptop: ;) - bblack just sent me this: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header |
23:02.07 | JT | monsted: grasping at straws really "* _rnz- chuckles at asterisks fetid vermin ROI" |
23:02.13 | r0d3nt|m | amg aydiosmio |
23:02.23 | AsteriskMonkey | lol last time i used a cisco cert was when i ran out of rolling paper |
23:02.24 | justinu|laptop | CunningPike: ok, but my feeling is that will make aserisk send RPID/Asserted Identity on INVITEs when making the outbound call |
23:02.24 | AsteriskMonkey | :P |
23:02.33 | monsted | too bad he didn't try for real... i've got the login for enough stuff to make most trolls fold ;) |
23:02.40 | justinu|laptop | CunningPike: you need asterisk to send your phones RPID on 183/180/200 OK responses |
23:02.44 | *** join/#asterisk yassine (n=yassine@xdsl-87-78-21-126.netcologne.de) |
23:02.48 | yassine | hi everyone |
23:02.48 | CunningPike | justinu|laptop: OK |
23:02.58 | yassine | anyone here running asterisk on debian ? |
23:03.05 | monsted | yassine: many |
23:03.16 | CunningPike | justinu|laptop: I'll try it and see what happens |
23:03.49 | yassine | monsted, good news are there any prepackages packages or debian is only being used as a platform and everything is done manually ? |
23:04.08 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
23:04.23 | jtf0518 | If we try out this Asterisk solution we're thinking of either Suse Linux or Ubuntu server. I'd like to hear anyone's opinion on those platforms as a choice. |
23:04.32 | justinu|laptop | gentoo |
23:04.38 | jtf0518 | now way! |
23:04.42 | justinu|laptop | ubuntu then |
23:04.50 | JT | i always say go for the platform you are most comfortable with administering, jtf0518 |
23:04.52 | mosty | i have just registered some more g729 licenses, can i get asterisk to use them without restarting? |
23:04.56 | AsteriskMonkey | lol apparently someone is trying to war dial my asterisk box lol... 4 channels vs my 8 pris... |
23:05.00 | jtf0518 | I have very little hair on top of my head justinu|laptop :-) |
23:05.12 | justinu|laptop | gentoo is really slick one you get the hang of it |
23:05.17 | Nugget | Linux is poo. |
23:05.42 | justinu|laptop | jtf0518: if you're in a hurry and don't want to compile, go with ubuntu |
23:05.42 | jtf0518 | I've heard but it's a nightmare to setup for most unless you're 133t |
23:05.50 | Katty | Nugget: you! |
23:06.00 | aydiosmio | I've used CentOS and Debian for Asterisk |
23:06.04 | justinu|laptop | jtf0518: you're end up learning a lot of stuff you really should have known in the first place! |
23:06.09 | aydiosmio | I prefer Debian, as it's the platform Digium develops on |
23:06.28 | Katty | all my stuff runs on debian. |
23:06.33 | Katty | i'm an apt-get addict. |
23:06.34 | justinu|laptop | s/you're/you'll/ |
23:06.35 | dlynes_laptop | Katty!!! |
23:06.45 | Katty | dlynes_laptop: allo (= |
23:06.51 | puzzled | aydiosmio: iirc mark developed on fedora core for at least a while |
23:06.57 | jtf0518 | I believe you justinu and I've heard good things about Gentoo (in fact I work at a zoo and we have real Gentoo penguins here) but time is of the essence. |
23:07.08 | JT | my choice is debian |
23:07.09 | Katty | jtf0518: penguins :> |
23:07.10 | JT | but meh |
23:07.21 | justinu|laptop | jtf0518: understandable... got any pics of your Gentoo's btw?? |
23:07.23 | jtf0518 | We also have Rockhoppers too but no one's made a distro by that name yet. |
23:07.44 | jtf0518 | justinu, not handy with me but I could get some and post them some other time. |
23:07.50 | justinu|laptop | that would be cool |
23:07.58 | jtf0518 | I'll make a point to do that. |
23:08.11 | Katty | Nugget: i've got new projects going :> |
23:08.11 | Waverly360 | Hmm..for you linux gurus out there..I know this really isn't the forum for it...but anyone have a clue what would keep services from starting up in the order they are specified in /etc/rc3.d on a fedora core box? |
23:08.16 | Katty | Nugget: including bananapos :> |
23:08.24 | jtf0518 | I can also get stuffed penguins at our gift shop at a discount too if someone wants a linux mascot toy. |
23:08.25 | Nugget | yay |
23:08.33 | *** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) |
23:08.34 | Katty | Nugget: and xen :> |
23:08.43 | Katty | Nugget: xen is hotnessss |
23:08.53 | suma | hi how can i make asterisk to access extensions directly from mysql database ? |
23:09.03 | dlynes_laptop | suma: real time extensions |
23:09.12 | Nugget | suma: you want the (catastrophically mis-named) asterisk "realtime" stuff. |
23:09.13 | Corydon-w | jtf0518: any 36 inch plush penguins? |
23:09.26 | puzzled | Waverly360: maybe you need to do a reset or resetpriorities on the funky service(s). check man chkconfig |
23:09.26 | dlynes_laptop | Nugget: no doubt |
23:09.41 | jtf0518 | Corydon, the ones I've seen are about 12 - 18 " but I can check for you. |
23:09.46 | suma | dlynes_laptop: all the configurations taken from mysql ? |
23:09.56 | Nugget | all the ones that matter. |
23:09.57 | dlynes_laptop | suma: if that's what you want to use, yes |
23:10.19 | suma | dlynes_laptop: any doc how to configure the same? please |
23:10.19 | dlynes_laptop | suma: basically any database supported by unixODBC |
23:10.25 | dlynes_laptop | ~doc |
23:10.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
23:10.29 | dlynes_laptop | ~docs |
23:10.30 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
23:10.31 | dlynes_laptop | ~wiki |
23:10.32 | jtf0518 | Today we played around and got Asterisk working in a VM running Suse 10.1 with softphones. |
23:10.36 | Corydon-w | jtf0518: I haven't been able to find 36" penguins since linuxmall.com went out of business |
23:10.36 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
23:10.53 | Corydon-w | jtf0518: and mine is beginning to show age |
23:10.55 | justinu|laptop | Katty: all well in mysql land? |
23:10.55 | dlynes_laptop | there ya go, suma |
23:11.06 | Katty | justinu|laptop: i'm aptgetting teh package of sqlly goodness. |
23:11.07 | aydiosmio | I had Debian running in VMWare with * for a long time |
23:11.10 | aydiosmio | worked really well |
23:11.50 | Katty | justinu|laptop: 2 minutes to go (= |
23:11.51 | Waverly360 | puzzled: doesn't the linkname of the startup script in that directory specify which order it's started though? Or am I mistaken in believing that it's based on directory listing order? |
23:12.03 | justinu|laptop | Katty: what are you on, dialup?? |
23:12.10 | Katty | t1 |
23:13.08 | puzzled | Waverly360: I think it's the link name but the priority is also noted in the service file name in /etc/rc.d/init.d |
23:14.25 | jtf0518 | darn, that pdf file link in the wiki for Legacy Integration for an Option 11C is broken! |
23:14.28 | monsted | Waverly360: it's in numerical order (that is S01blah starts before S02blah) |
23:15.26 | jtf0518 | Is that O'Reilly book on Asterisk any good? |
23:15.51 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
23:16.04 | Nugget | imho asterisk development is too chaotic for dead tree books to be useful. |
23:16.09 | Nugget | maybe in another year |
23:16.23 | Waverly360 | monsted, puzzled: Well here's the problem. I have 'S19sangoma_config -> /etc/init.d/sangoma_config', then 'S20wanrouter -> /usr/sbin/wanrouter', then 'S40asterisk -> ../init.d/asterisk' |
23:16.29 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-114-42-210.satx.res.rr.com) |
23:16.40 | Waverly360 | That should run the sangoma_config script, then wanrouter, then asterisk |
23:16.48 | monsted | Waverly360: correct |
23:16.58 | Waverly360 | but in the var/log/messages file, it shows asterisk trying to startup before wanrouter |
23:17.16 | Waverly360 | which would cause asterisk to bomb out, because there are no channels for it to find |
23:17.21 | Waverly360 | er..zaptel devices |
23:17.25 | Waverly360 | whatever you wanna call it |
23:17.42 | *** part/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
23:17.53 | Waverly360 | what's killing me |
23:17.57 | Waverly360 | is that I have two identical servers |
23:18.03 | Waverly360 | both running the same stuff..setup the same way |
23:18.07 | Waverly360 | one works right..the other doesn't |
23:18.21 | Waverly360 | my problem is..asterisk isn't starting when the box has a power outage, or when it's rebooted |
23:18.36 | Waverly360 | I can't recreate the problem on my test box...ONLY on the live one..of course |
23:19.10 | monsted | are the links in the right rcX.d directory? |
23:19.11 | justinu|laptop | same filesystem? |
23:19.22 | Waverly360 | yep |
23:19.25 | Waverly360 | identical |
23:19.27 | *** join/#asterisk bblack (n=bblack@129.174.112.186) |
23:19.33 | justinu|laptop | odd |
23:19.39 | Waverly360 | tell me about it *headdesk* |
23:20.35 | Waverly360 | I'm no linux noob..but this is killin me |
23:20.36 | monsted | note that they probably all start just about simultaneously - you could try putting in a "sleep 10" in the wanrouter script to give it a chance to find all the hardware before the script runs off to start asterisk :) |
23:20.36 | Nivex | wacky idea: Can I use an USB ACM capable cellphone as a channel for Asterisk? |
23:20.39 | Waverly360 | well..and I can do that..but that still doesn't explain why it works on one box..and not the other...unless.. |
23:20.54 | Waverly360 | unless maybe the lab test box doesn't have the same amount of pri/analog cards in it. |
23:21.07 | CunningPike | justinu|laptop: Ya - I tried the RPID and P-Asserted-ID and it sends them to the dialed phone |
23:21.09 | Waverly360 | I guess wanrouter would take longer to startup if there are more devices for it to configure? |
23:21.22 | monsted | Waverly360: i'd think so |
23:21.55 | justinu|laptop | CunningPike: yeah, so you need gork's patch |
23:22.39 | Waverly360 | and here's another little bit of weirdness that I don't understand. In my var/log/messages file on the broken box..I have dates and times of Nov 15 13:46...but at one point..the time changes to 07:45...then back to 13:45 |
23:23.07 | Waverly360 | is /var/log/messages not guaranteed to be in order? |
23:23.32 | justinu|laptop | only reason I can thing for that is that syslog might have restarted, and dumped the klog messages twice?? |
23:23.44 | justinu|laptop | but that doesn't sound right |
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23:24.30 | Waverly360 | the time changes on this command rc.sysinit: -e |
23:26.05 | CunningPike | Waverly360: I'm going to bet you're in CST |
23:26.08 | *** join/#asterisk sjobeck (n=sjobeck@66-182-49-26.atgi.net) |
23:26.16 | Waverly360 | Yes I am |
23:26.25 | *** join/#asterisk Choubaka (n=odiq@c66.110.141-241.clta.globetrotter.net) |
23:26.27 | CunningPike | Waverly360: Your timezone is changing |
23:26.43 | Waverly360 | .... |
23:26.44 | CunningPike | Waverly360: From local to GMT and back |
23:26.45 | Waverly360 | hah |
23:26.48 | Choubaka | anyone can help me with zaptel module? |
23:26.49 | justinu|laptop | how does that happen? |
23:27.21 | CunningPike | justinu|laptop: Possibly something borked in the setup |
23:27.25 | Waverly360 | hmm...I suppose that would explain it. |
23:27.45 | CunningPike | Waverly360: When you installed linux, did you opt to use UTC for the system clock? |
23:28.10 | Waverly360 | CunningPike: to be honest, I don't know. It was an image that's auto installed on the box. |
23:28.29 | CunningPike | Waverly360: That would be my suspicion anyway....... |
23:28.31 | Waverly360 | hmm |
23:28.47 | Waverly360 | How would I figure that out now? |
23:28.55 | Waverly360 | Any thoughts? |
23:29.35 | Waverly360 | here's the thing..this has been happening on several boxes |
23:29.41 | Waverly360 | but not my test box.. |
23:29.47 | Waverly360 | I need to figure out what's different |
23:30.05 | Waverly360 | Do you think my problems with services starting at the wrong time could be time related? |
23:30.17 | Nugget | I doubt it. |
23:30.58 | Nugget | if any of the running processes has a TZ environment variable set to a different timezone than the system time (as defined by /etc/localtime) this will result in conflicting times reported by syslog logs. |
23:31.09 | Nugget | but there's no actual problem, just awkward logs. |
23:32.49 | Waverly360 | well, then there's not much point in chasing that problem down |
23:34.17 | Nivex | hmm, looks like I'd have better luck with chan_bluetooth |
23:36.19 | Choubaka | I can't place outgoing call, I've got this: fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf |
23:36.22 | Choubaka | anyone? |
23:36.38 | Nugget | "can't" is a little vague. |
23:36.58 | Nugget | does that file exist? |
23:37.05 | Choubaka | nop |
23:37.18 | Nugget | what's supposed to be in it? |
23:37.23 | Choubaka | dunno :P |
23:37.40 | yassine | im trying to set up asterisk on debian etch and while trying to run : m-a a-i zaptel i get this error : http://rafb.net/paste/results/WnPWpw63.html any help is apreciated |
23:38.32 | Nugget | google indicates that localprefixes.conf is an AMP thing. |
23:38.57 | Nugget | or perhaps asterisk@home? |
23:39.02 | Choubaka | Nugget, yep I wee it.. but I think is not very important |
23:39.09 | JT | yassine: that's a really easy error |
23:39.17 | Choubaka | I'm using trixbox v2 beta |
23:39.18 | JT | yassine: exactly like it says, you need to install gcc |
23:39.29 | Nugget | you won't get much help for that in here. |
23:39.39 | yassine | JT i also think so but i have gcc installed a newer version :s |
23:39.55 | JT | hmm |
23:40.02 | JT | it's looking for that specific one |
23:40.31 | yassine | JT gcc version 4.1.2 20061028 (prerelease) (Debian 4.1.1-19) |
23:41.58 | JT | there might be a package that allows you to install gcc-3.3 concurrently |
23:42.43 | Choubaka | I got this message now.. localprefixes is now ok.. All circuit is busy now, please try again later.. |
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23:43.20 | yassine | JT im wondering how i can set the RELAX_CC_CHECK VAR |
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23:44.01 | JT | debian often has packages for different versions of gcc |
23:44.05 | JT | i suggest you check into it |
23:44.19 | Nugget | Choubaka: I suggest you ask on the trixbox forums. |
23:44.23 | JT | much more likely to work, in case it doesn't like gcc4 for whatever reason |
23:44.27 | Nugget | I don't think anyone here will be able to help |
23:44.29 | Choubaka | k, cool |
23:44.38 | JT | or #freepbx |
23:45.28 | e-horn | I'm having a problem where I'm trying to use * (a) to drop people into their voicemail menu when the greeting starts playing (so when traveling people can check their voicemail)... it's working fine when you call from a voip phone, but on a outside line (landline/cell/etc), the DTMF tones are ignored... however I know it's not a problem with dtmfmode or anything like that because calling into my IVR and choosing options works, any ideas? |
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23:46.01 | *** join/#asterisk bulatitoy (n=rmn@adsl-70-231-130-250.dsl.snfc21.sbcglobal.net) |
23:46.34 | bulatitoy | i need help again |
23:46.45 | shido6 | how are the calls coming in, e-horn? |
23:47.24 | bulatitoy | can i use a single phone line, X100P card then have 2-3 extensions? |
23:48.07 | JT | yes |
23:48.55 | bulatitoy | what kind of card do i use to connect the phones? |
23:49.05 | bulatitoy | sorry im really new to this |
23:49.16 | JT | if you need physical phones |
23:49.32 | justinu|laptop | ethernet card |
23:49.49 | JT | the main options are analogue phones + TDM400P with FXS modules, T1 + channel bank, or SIP VoIP phones via ethernet |
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23:50.20 | bulatitoy | so i will install the X100P on the linux machine running asterisk |
23:50.29 | rollergrrl | Has anyone heard of a new federal DID tax? |
23:50.32 | JT | SIP VoIP phones usually offer the best flexibility |
23:50.49 | JT | bulatitoy: do the extensions need to be different, or is shared ok? |
23:51.05 | bulatitoy | JT: shared is fine |
23:51.28 | bulatitoy | i just want to know how everything works and I will do more experiments when i get this working |
23:51.46 | JT | right so there bare minimum for that if you went with an analogue setup is a TDM400P with 1 FXS module |
23:51.53 | JT | and if you did that |
23:51.59 | JT | you may as well get an FXO module too |
23:52.03 | JT | because X100Ps are crappy |
23:52.07 | bulatitoy | i see |
23:52.09 | JT | and no longer produced by digium |
23:52.22 | JT | you might get one that works, you might not |
23:52.26 | bulatitoy | so the setup would be like this |
23:53.00 | bulatitoy | PSTN---X100P or TDM400P on Linux/Asteris machine --- phones |
23:53.05 | bulatitoy | is that correct? |
23:53.09 | JT | of course you can always us PC based softphones for testing |
23:53.14 | JT | yeah pretty much |
23:53.36 | JT | phones would connect via the TDM400P too if they were analogue |
23:53.41 | JT | or you could go voip |
23:53.43 | bulatitoy | if i use softphones, is it safe to say, i was like testing phone hardware? |
23:53.57 | e-horn | shido6: from a SIP provider |
23:53.58 | JT | bulatitoy: what do you mean? |
23:54.15 | shido6 | do u see "*" come in when u debug? |
23:54.34 | e-horn | debug will show that? |
23:54.34 | bulatitoy | are they the same? softphone and the actual phone device/unit? |
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23:54.59 | JT | bulatitoy: they aren't exactly the same, but they can still access the phone lines and what not |
23:55.09 | JT | the differences will be in the features, and how you use them |
23:55.32 | e-horn | like I said it works fine when other voip phones call the number and hit *... so you would think if it does nothing when pressing * on outside line that it's just not coming in, but if maybe it's ignoring it for some reason.... I don't know |
23:55.35 | JT | analogue phones don't have that many features, but are easy to use to an extent |
23:55.53 | e-horn | and whether it's a voip phone or outside call it all drops into the same context/extension/etc |
23:55.59 | bulatitoy | i would like to try first analog, just to see how they are interconnected |
23:56.04 | JT | you'd need to use feature codes to do anything much pabx like on analogue phones |
23:56.04 | e-horn | so there's nothing different |
23:56.20 | JT | ok, that's fine as long as you're prepared to pay for the TDM400P :) |
23:56.27 | JT | softphones are cheap (free mostly) |
23:56.42 | bulatitoy | i am having a hard time looking for docs that shows how to connect the necessary hardware |
23:56.58 | JT | have you found voip-info.org? |
23:57.17 | bulatitoy | ill check on that |
23:57.19 | JT | and the book |
23:57.21 | JT | ~thebook |
23:57.30 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
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23:57.46 | bulatitoy | thanks for the link! |
23:57.52 | JT | np |
23:57.59 | JT | digium also put out a handbook too |
23:58.04 | JT | and there's other random stuff online |
23:58.18 | bulatitoy | so i can go TDM400P and a single phone line (my home phone) + softphones? |
23:58.23 | JT | voip-info is a real resource provided you can find what you're looking for |
23:58.29 | JT | bulatitoy: yes |
23:58.39 | JT | the TDM400P is a board with a space for 4 modules |
23:59.01 | JT | there are 2 types of modules, FXO and FXS, for exchange connection, and handsets, respectively |
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23:59.20 | e-horn | shido6: I just tried turning on sip debug for a peer, and it didn't print anything when I pressed * calling in from outside |
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23:59.54 | e-horn | shido6: but it doesn't show anything even if I do the debug on a working phone, so... |
23:59.57 | bulatitoy | i can see that some TDM400P has 4 FXO |