| 00:03.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:05.07 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 00:05.44 | Dr-Linux|home | what's special in Gizmo product, which doesn't provide other softphone like eyebeam etc ? :S |
| 00:21.03 | *** join/#asterisk _VoiceMeUp_COM (n=Miranda@145-27.mc.cite.net) |
| 00:22.38 | _VoiceMeUp_COM | hey |
| 00:23.08 | _VoiceMeUp_COM | have a question:. anyway to rewirte exten ? im passing to a macro to try this.. but still shows up with my prfix in cdr logs |
| 00:23.54 | JT | set it to something else? |
| 00:25.11 | blitzrage | Goto(my_rewritten_exten,1) |
| 00:25.23 | blitzrage | my_rewritten_exten,1,Macro(my_macro,arg1) |
| 00:25.29 | blitzrage | ${MACRO_EXTEN} |
| 00:25.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:25.53 | _VoiceMeUp_COM | well hmm |
| 00:26.44 | _VoiceMeUp_COM | i think ill need headers or globals |
| 00:27.42 | _VoiceMeUp_COM | but yeah.. PREFIXEXTEN,set header ( something) then ill push all those to a main XXX with a get header and macro that |
| 00:27.47 | _VoiceMeUp_COM | good idea |
| 00:28.05 | _VoiceMeUp_COM | waste of cycles but ;) |
| 00:29.45 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:30.37 | *** join/#asterisk shodan (n=shodan@ip194.96-113-216.pppoe1.joliette.intermonde.net) |
| 00:31.58 | shodan | if I needed one FXS what should I get these days ? |
| 00:33.09 | JT | what are your other criteria? |
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| 00:34.23 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 00:35.36 | shodan | it has to be reliable and that's about it, I want to know what is a good single or dual port fxs and how much does it cost so I can decide whether voip is still too expensive |
| 00:39.06 | shodan | hmmmm , I mean fxo port ... eheh got that mixed up |
| 00:39.33 | JT | you don't care about if it's a card or an ATA? |
| 00:40.28 | *** join/#asterisk Laureano (i=[U2FsdGV@OL155-33.fibertel.com.ar) |
| 00:40.56 | Laureano | Hello |
| 00:41.12 | shodan | no, although I've had a bad experience with SPA-3102 |
| 00:41.23 | JT | shodan: hrm, really |
| 00:41.56 | shodan | yeah, once in a while it left the line off hook forever |
| 00:42.06 | shodan | happenned twice in 3 weeks |
| 00:42.06 | JT | hrm |
| 00:42.08 | *** join/#asterisk tschafer (n=tschafer@207.241.143.246) |
| 00:42.21 | shodan | also outbound dtmf where bad |
| 00:42.54 | shodan | it had trouble dialing out and with ivr |
| 00:43.44 | shodan | I did a lot of troubleshooting but I stopped short of analysing traffic with ethereal |
| 00:44.54 | shodan | that was a year ago, didn't try it since, didn't want to sink another 400$ on a big card |
| 00:45.30 | *** join/#asterisk neuralwind (n=root@201.222.65.250) |
| 00:45.56 | neuralwind | just in case how do u remove a kernel |
| 00:46.00 | neuralwind | in centos? |
| 00:46.09 | neuralwind | i have this |
| 00:46.24 | neuralwind | kernel-devel-2.6.9-42.0.10.EL |
| 00:46.24 | neuralwind | kernel-2.6.9-42.EL |
| 00:46.24 | neuralwind | kernel-utils-2.4-13.1.83 |
| 00:46.24 | neuralwind | kernel-2.6.9-42.0.10.EL |
| 00:46.25 | neuralwind | kernel-ib-1.0-1 |
| 00:46.37 | neuralwind | i think i have to delete the second one |
| 00:51.50 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 00:51.52 | franck | Hi |
| 00:52.02 | franck | I have set up a queue to pick up call |
| 00:52.39 | franck | I'd like to run statistics on this queue, how many calls are on hold, how many are picked up, how many people give up, what is the average waiting time |
| 00:52.58 | franck | How could I do that? I have freepbx as an interface... |
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| 00:55.24 | neuralwind | franck do u know how to uninstall a kernel in centos? |
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| 01:05.53 | *** join/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 01:06.25 | TUplink | anyone know if tellme 8005551212 has a sip address that i can call |
| 01:07.21 | Strom_M | hahahahahahaha, I doubt it |
| 01:07.45 | TUplink | i really would think they would so they dont kill landlines |
| 01:07.58 | Strom_M | "kill landlines"? |
| 01:08.18 | TUplink | errr ..... SIP is much cheeper |
| 01:08.44 | Strom_M | tellme buys so much traffic from AT&T that they pay ungodly small amounts of money per minute |
| 01:09.18 | TUplink | well if i found a sip addres i wouldnt have to use up my phone line :P |
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| 01:14.42 | *** join/#asterisk ninjagumby (n=gauge@ip70-161-206-163.hr.hr.cox.net) |
| 01:14.44 | ninjagumby | hi |
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| 01:32.47 | demlak | hmm.. voicemail is not sending e-mails... but i donīt see the problem.. |
| 01:33.15 | demlak | is there something special to be activated? |
| 01:34.31 | demlak | i wrote a script to send everything to /dev/console... for getting the exact syntax, which voicemail is using for /usr/sbin/sendmail |
| 01:35.14 | demlak | but voicemail just starts "/usr/sbin/sendmail -t" nothing more... no mail adress.. no more options/parameter.. no attachment, etc.. |
| 01:37.33 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 01:45.29 | *** join/#asterisk volospin (n=volospin@221.124.61.38) |
| 01:45.33 | volospin | Greetings |
| 01:51.02 | BSD_tech | clear |
| 01:54.45 | demlak | does this produce some output on your system? voicemail.conf "mailcmd=echo > /dev/console command\:" |
| 01:58.52 | volospin | Anybody uses Gentoo to install Asterisk? |
| 01:59.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:01.59 | ido | volospin: i do |
| 02:01.59 | ido | is anyone here using openwrt and asterisk 1.2.14? |
| 02:03.02 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 02:06.58 | Qwell | brimstone: what did you do? :( |
| 02:07.02 | brimstone | wasn't me! |
| 02:09.13 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:12.01 | volospin | ido, I just install Gentoo 2007.0 and 1.2.17. Are there any GUI or webmin interface or I need to stick with ssh? |
| 02:13.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
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| 02:14.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:14.48 | volospin | help |
| 02:14.55 | volospin | ehh... sorry... |
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| 02:53.14 | Putzz | zZz? |
| 02:54.08 | wunderkin | ido, i think i am for one client...but i am trying to switch to openser now.. |
| 02:59.54 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 02:59.54 | *** mode/#asterisk [+o angler] by ChanServ |
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| 03:39.25 | BSD_tech | how goes zaptel on 64 bit |
| 03:40.53 | BSD_tech | I have heard it copiles but does not work to it does not compile |
| 03:41.07 | BSD_tech | or it compiles and is very flaky |
| 03:42.47 | sbingner | that's strange, but you could always just run it on a 32-bit kernel if it is |
| 03:44.01 | demlak | any idea on this? http://rafb.net/p/bvUdf426.html |
| 03:44.38 | Juggie | BSD_tech, zaptel runs perfect on 64bit kernel |
| 03:44.50 | BSD_tech | ok |
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| 04:08.06 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 04:08.14 | d4rkst4r75 | hi to all |
| 04:08.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:10.49 | d4rkst4r75 | can someone give me an hint? |
| 04:11.24 | JT | hint: today will be filled with new opportunities |
| 04:11.50 | d4rkst4r75 | thank you JT: now I feel better ;) |
| 04:12.13 | d4rkst4r75 | i want to ask a thing about asterisk-gui |
| 04:12.44 | d4rkst4r75 | i tried to download the cvs version and noticed that it has only basic configuration while the cfgadvanced is missing |
| 04:13.40 | fakhir | cfgbasic.html?advanced=yes |
| 04:14.07 | d4rkst4r75 | i'll give a try fakhir |
| 04:14.45 | d4rkst4r75 | another thing: i noticed that asterisk-gui miss the support for zap devices |
| 04:14.57 | d4rkst4r75 | is this true or i've not seen something? |
| 04:15.23 | fakhir | it does |
| 04:15.38 | fakhir | but dont think it has a tab |
| 04:16.58 | d4rkst4r75 | fakhir, thank you. |
| 04:17.12 | fakhir | :) no problem |
| 04:17.16 | d4rkst4r75 | I think that the best way to configure asterisk is from the command line |
| 04:17.44 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 04:18.09 | fakhir | yeah probably is (at least at this point) |
| 04:18.22 | d4rkst4r75 | but searching a way to let other sys admin to manage * without modify .conf files by hand |
| 04:19.00 | d4rkst4r75 | fakhir, what's the best GUI for * ? |
| 04:19.25 | fakhir | i have only ever used the digium gui |
| 04:19.47 | JT | yeah console is best :) |
| 04:19.55 | d4rkst4r75 | yes JT ;) |
| 04:20.32 | d4rkst4r75 | but i've seen freePBX and I don't like it |
| 04:20.40 | JT | ~freepbx |
| 04:20.54 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:20.54 | JT | :) |
| 04:22.50 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 04:22.53 | d4rkst4r75 | great ;) |
| 04:26.10 | d4rkst4r75 | fakhir, but why in asterisk-gui the "Options" menu remain in "Loading"? |
| 04:27.16 | fakhir | dont know, it works for me |
| 04:27.33 | d4rkst4r75 | uhm.. what revision you have of the code? |
| 04:28.06 | fakhir | 916 |
| 04:28.46 | d4rkst4r75 | i'm at revision 917 |
| 04:29.05 | fakhir | asterisknow? |
| 04:29.08 | d4rkst4r75 | no |
| 04:29.19 | d4rkst4r75 | asterisk installation |
| 04:29.24 | d4rkst4r75 | from source code |
| 04:30.14 | fakhir | hm maybe its broken in 917 then |
| 04:30.30 | d4rkst4r75 | ok, i'll checkout the revision 916 and let you know ,) |
| 04:30.51 | d4rkst4r75 | see you later! and thanks for the support |
| 04:34.36 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:34.37 | *** mode/#asterisk [+o mog] by ChanServ |
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| 06:16.33 | pabs3 | any recommendations for PoE switches for a small office LAN with IP phones? |
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| 06:22.18 | JT | adtran make ethernet gear? |
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| 06:23.48 | mog | yeah |
| 06:23.58 | mog | looks like their telco gear |
| 06:24.02 | JT | hmm |
| 06:24.08 | JT | most be pricy ;) |
| 06:24.23 | mog | probably, but im just geek i dont buy things just get to play with em |
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| 06:33.07 | CpuID2 | telco gear ftw, when its reliable |
| 06:33.13 | CpuID2 | which youd hope it is |
| 06:33.19 | CpuID2 | random things dying ftl |
| 06:33.53 | JT | yeah telco gear is good |
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| 07:09.10 | Bladerunner05 | Hi all using AVM b1 card can't set echo cancellation |
| 07:13.20 | *** join/#asterisk cazze (n=pc@unaffiliated/kammicazze) |
| 07:14.35 | Ifaistos | you are using mIsdn probably right ? |
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| 07:20.10 | Bladerunner05 | <Ifaistos> no I'm using capi_vm |
| 07:21.13 | Bladerunner05 | sorry chan_capi_cm |
| 07:23.12 | Chris-NB | hi |
| 07:23.25 | Chris-NB | anyone used the manager API to redirect calls? |
| 07:24.34 | Chris-NB | I try to redirect an established call into a meetme room |
| 07:25.32 | Ifaistos | Bladerunner05 : Hmmm not sure if it has echo cancellation |
| 07:28.09 | Bladerunner05 | azz |
| 07:28.09 | Ifaistos | Bladerunner05 : if i remember corectly only the Eicon Diva servers have echo cancellation and this is done in hardware |
| 07:31.57 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:32.02 | Bladerunner05 | <Ifaistos> how can I do echo cancellation ? If I set in capi.conf the same paramters of diva's configuration it appers to ignore that |
| 07:32.59 | Bladerunner05 | take a look to http://www.asteriskguru.com/tutorials/avm_b1.html |
| 07:34.45 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 07:34.49 | Ifaistos | Bladerunner05 : i don't think you can do echo cancellation on AVM cards, but i am not 100% sure as i don't use them |
| 07:35.15 | Ifaistos | Bladerunner05 : The Eicon Diva *server* has hardware echo cancellation |
| 07:36.05 | Bladerunner05 | Ok |
| 07:36.20 | Ifaistos | Bladerunner05 : I am using hfc based cards and zaphfc/zaptel that does echo cancellation in software |
| 07:39.27 | Bladerunner05 | ok |
| 07:39.32 | Bladerunner05 | I'll try it |
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| 08:17.45 | puzzled | morning |
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| 08:18.38 | creativx | morning it is indeed |
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| 08:45.11 | ido | is anyone here? |
| 08:45.34 | *** join/#asterisk matsk (n=mk@194.68.102.175) |
| 08:45.35 | ido | if so, what do i need to connect my router with asterisk and usb ports (and ethernet ports) to my analog phone line? |
| 08:45.47 | ido | an fxo? fxs? are there any good USB ones? |
| 08:47.33 | *** join/#asterisk Gringo_ (n=N3TW4LK3@250.102-247-81.adsl-dyn.isp.belgacom.be) |
| 08:47.39 | ido | anyone? |
| 08:47.59 | kiscokid | I think you need a pci card |
| 08:48.07 | Gringo_ | looking for a way to "echo" global variables from the CLI. |
| 08:48.10 | Gringo_ | can this be done? |
| 08:48.13 | ido | i don't have pci slots on this router running asterisk :) |
| 08:48.17 | ido | only USB and rj45 |
| 08:48.23 | ido | usb2.0 |
| 08:48.54 | kiscokid | what kind of router? |
| 08:50.13 | ido | asus wl-500gp |
| 08:50.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:51.58 | kiscokid | I think you can only use VOIP Trunks for this router |
| 08:52.07 | ido | really? |
| 08:52.18 | ido | what about something like the atcom usb gateways? |
| 08:53.38 | kiscokid | does it have any asterisk support like the zap module supports digium cards? |
| 08:54.12 | ido | well, let me ask it another way: are there any USB FXOs supported under asterisk that you know of? |
| 08:54.42 | kiscokid | haven't heard of any but I'm relatively new |
| 08:54.45 | ido | ok |
| 08:59.00 | Chris-NB | oej, think here is a better place : D |
| 08:59.13 | Chris-NB | oej, If u have time? |
| 08:59.15 | ido | so does anyone else know of any USB FXO's for use with asterisk? |
| 08:59.26 | ido | (a PCI card is not an option) |
| 08:59.34 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:59.56 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:02.08 | ido | astribank |
| 09:02.12 | ido | i think i found it |
| 09:02.35 | ido | yikes |
| 09:02.37 | ido | $500 :( |
| 09:02.41 | ido | 8 port fxo/fxs |
| 09:02.49 | ido | but it's usb2 |
| 09:03.08 | kiscokid | looks interesting |
| 09:03.14 | ido | Astribank 8 |
| 09:03.14 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:04.33 | kiscokid | but it only connects phones not phone lines |
| 09:05.29 | ido | it has fxo ports for phone lines |
| 09:08.13 | ido | rather, the one i was looking at does. |
| 09:08.13 | kiscokid | url? |
| 09:08.13 | ido | closed the browser |
| 09:08.13 | ido | it was 6 fxs 2 fxo |
| 09:08.14 | kiscokid | I see one with 8 fxo ports |
| 09:08.14 | ido | another one: Sipura SPA-3000 looks promising |
| 09:08.14 | ido | you mean fxs? |
| 09:08.15 | ido | fxo = to plug into phone line (on/off hook), fxs = to plug telephone into (provides dialtone), right? |
| 09:08.15 | kiscokid | no, it is a choice of 8 fxs or 8 fxo |
| 09:08.22 | ido | yeah, well, 8 fxo would be fine |
| 09:08.22 | ido | i don't need any phones on this thing |
| 09:08.29 | ido | i've got free polycoms :) |
| 09:08.48 | kiscokid | how do you get free polycoms? |
| 09:08.59 | ido | oops, did i say that out loud? ;) |
| 09:10.56 | kiscokid | I won't tell anyone if you send me a few :) |
| 09:11.13 | ido | hehe |
| 09:12.45 | oej | Chris-NB: I'm here |
| 09:14.02 | Chris-NB | oej, here is my redirect. I try to redirect a call between 130 and 132 to 710 (meetme) |
| 09:14.03 | Chris-NB | http://rafb.net/p/LoIQsw70.html |
| 09:14.28 | Chris-NB | at the end of the section there is what is written on the cli |
| 09:16.28 | Chris-NB | oej, the phone 130 (channel in the manager interface) is redirected to 710, 132 (ExtraChannel in manager interface) gets dropped |
| 09:19.20 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-138.dsl.hstntx.swbell.net) |
| 09:19.39 | oej | Chris-NB: Sent into invalid extension 's' in context 'default' on SIP/140-081f4518 |
| 09:20.46 | oej | Seems like the 132 device wasn't in the proper state or something |
| 09:21.06 | Chris-NB | jep. but, why? I wrote Exten 710 and not s? |
| 09:21.06 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 09:21.21 | Chris-NB | what do you mean with proper state |
| 09:21.27 | oej | What was the state of the call SIP/132- etc |
| 09:21.42 | Chris-NB | state? how do I check? |
| 09:22.08 | oej | was it answered or ringing? |
| 09:22.27 | Chris-NB | answered. the call was established |
| 09:23.18 | kippi | hey |
| 09:23.48 | kippi | what is the best way to link in a php site to asterisk, I want to be able to dial numbers from my webpage, whats the best way? |
| 09:24.37 | creativx | kippi: astmanproxy |
| 09:25.04 | oej | Chris-NB: I can't really see what's going on there. For some reason, one of the call legs is sent to /dev/null |
| 09:25.52 | Chris-NB | mhm, but why : / |
| 09:26.12 | Chris-NB | If I do the redircht in two steps, then it works |
| 09:28.33 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:32.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:39.04 | *** join/#asterisk penguinFunk (n=penguin@89-145-196-28.xdsl.murphx.net) |
| 09:39.22 | oej | Chris-NB: that's weird. I can't explain what happens without testing and putting my hands on it, really |
| 09:40.10 | Chris-NB | oej, k. Its really very weird. : / |
| 09:41.03 | oej | Well, we are redesigning the bridge code for 1.6 because of a lot of issues we have with scenarios like this one, and call transfers, and SLA's and and and |
| 09:43.25 | Chris-NB | mhm. *grml. I should provide a conference feature but ... it's hard to implemente |
| 09:54.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:55.34 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 09:57.28 | *** join/#asterisk emanus (n=emanuele@adsl-241-27.38-151.net24.it) |
| 09:58.07 | emanus | hi |
| 10:00.32 | kippi | is there a howto on setting up astmanproxyt |
| 10:00.32 | kippi | ? |
| 10:04.49 | *** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161) |
| 10:06.26 | creativx | kippi: the docs that follows it |
| 10:06.37 | Bladerunner05 | I use grandstream gxp2000, how can I see when an extension is connected and if a call is in progress ? |
| 10:08.19 | penguinFunk | sip show channels |
| 10:08.25 | penguinFunk | if using sip of course |
| 10:11.31 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 10:12.03 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 10:13.54 | emanus | it is possible write a cdr event for every action of the caller? |
| 10:14.48 | emanus | now, the cdr-csv file change when the call finished |
| 10:15.06 | demlak | any idea on this? http://rafb.net/p/bvUdf426.html |
| 10:19.22 | StyleWarz | Hey guys, i don't get this new Set Language stuff. The asterisk tells me that the old syntax is deprecated and the new is CHANNEL(language) |
| 10:19.31 | StyleWarz | Well is it CHANNEL(de) ? CHANNEL(language=de) ? |
| 10:19.44 | StyleWarz | i don't find any working combination where asterisk doesn't complain |
| 10:21.08 | creativx | set(language()=de) |
| 10:22.10 | emanus | try this Set(CHANNEL(language)=it) |
| 10:23.10 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 10:23.10 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:23.15 | skirmisha | hello |
| 10:23.30 | skirmisha | any idea if there is good support for radius in asterisk |
| 10:25.57 | StyleWarz | thanks |
| 10:26.45 | skirmisha | ? |
| 10:27.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-138.dsl.hstntx.swbell.net) |
| 10:28.22 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:28.49 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 10:28.55 | Uatec | hi there |
| 10:29.30 | Uatec | i was wondering, how much bandwidth (per second) my asterisk box will need for an office of 15 people |
| 10:30.07 | mosty | depends on the codec you use |
| 10:30.21 | mosty | and how many simultaneous calls you want to support |
| 10:30.48 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 10:32.08 | Uatec | 8 external, possibly more internal |
| 10:32.16 | Uatec | using gsm |
| 10:32.29 | mosty | how many simultaneous calls? |
| 10:32.38 | Uatec | 10 |
| 10:33.47 | mosty | plug those values in to this: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 10:34.38 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:37.14 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-185-98.ny325.east.verizon.net) |
| 10:38.57 | Uatec | hmm |
| 10:38.58 | Uatec | excellent |
| 10:39.00 | Uatec | thankyou |
| 10:39.02 | Uatec | :) |
| 10:39.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:40.14 | skirmisha | guys |
| 10:40.24 | skirmisha | is there support of radius at all |
| 10:40.58 | mosty | uatec: beware that external clients will require twice the bandwidth of internal clients, unless there is no nat involved and you can reinvite the audio stream |
| 10:42.43 | defswork | I have a trunk with a dialing pattern of 1411656+. - calls made to 0049etc.. don't seem to match this dial pattern for some reason and therefore don't get allocated to this trunk |
| 10:43.18 | defswork | my outbound route (just 1) has just . for the dial pattern |
| 10:44.09 | Uatec | ok, ty mosty, but our external lines are isdn2 lines |
| 10:44.10 | mosty | paste your dialplan on a paste site |
| 10:45.46 | mosty | uatec: isdn calls don't affect your bandwidth usage, unless the call bridges isdn to a voip provider |
| 10:47.26 | Uatec | mosty, yup, i know, but obviously they affect my internal bandwidth usage |
| 10:48.10 | mosty | well yeah but you could use 10M ethernet and still be swimming in bandwidth |
| 10:52.59 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 10:54.14 | *** join/#asterisk federicoco (n=federico@212.34.251.205) |
| 11:00.47 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:04.46 | Uatec | oh cool :) |
| 11:05.57 | Uatec | :( |
| 11:06.00 | Uatec | that sucks |
| 11:06.08 | Uatec | it means i don't have an excuse to order a gigabit card |
| 11:06.14 | Uatec | although i could still order it |
| 11:06.18 | Uatec | noone would notice it's not there |
| 11:06.19 | Uatec | :) |
| 11:06.24 | Uatec | actually |
| 11:06.28 | Uatec | they're like Ģ16 |
| 11:07.47 | Uatec | what does it mean if my sip phone's dialtone beeps for a couple of seconds then goes continuous |
| 11:11.55 | *** join/#asterisk michael-i (n=michael-@141.41.40.191) |
| 11:13.39 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 11:20.00 | *** join/#asterisk viking78 (i=viking78@24-159-55-68.dhcp.jcsn.tn.charter.com) |
| 11:21.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:30.29 | creativx | bloody hell |
| 11:35.01 | Uatec | ? |
| 11:36.01 | creativx | squid is giving me a hard time :§ |
| 11:39.07 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-227.static.twtelecom.net) |
| 11:40.13 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 11:43.31 | *** join/#asterisk xtr (i=01928375@216.19.191.191.novuscom.net) |
| 11:44.57 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 11:44.57 | *** join/#asterisk friedrich| (n=friedric@e177254003.adsl.alicedsl.de) |
| 11:45.11 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-138.dsl.hstntx.swbell.net) |
| 11:48.58 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 11:53.18 | mosty | Uatec: probably message waiting indicator |
| 11:54.40 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-191-153.buff.east.verizon.net) |
| 11:56.55 | Uatec | oh |
| 12:00.19 | Uatec | hmm |
| 12:00.25 | Uatec | it's not |
| 12:00.31 | Uatec | but it stopped when i restarted the phone |
| 12:00.32 | Uatec | now odd |
| 12:03.15 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:14.30 | *** join/#asterisk Makenshi (n=makenshi@2001:630:1c0:2006:e096:7810:25e:1271) |
| 12:14.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:16.18 | mosty | check your phone's manual |
| 12:19.53 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:19.59 | DrukenLPY | morning everbody |
| 12:20.32 | demlak | morning? what time is it there? =) |
| 12:22.44 | cpm | 0822 |
| 12:22.49 | puzzled | 1422 |
| 12:22.55 | puzzled | still morning for me though |
| 12:23.02 | DrukenLPY | uhmm... 8:22 am |
| 12:23.03 | tzanger | 0822 here too |
| 12:23.14 | demlak | 14:22 here |
| 12:23.47 | cpm | auslander! |
| 12:23.59 | demlak | =) |
| 12:24.02 | vgster | hello, whats the difference between the TE110P and TE120P? |
| 12:24.12 | DrukenLPY | looks like 1 digit :) |
| 12:24.56 | puzzled | lol |
| 12:25.04 | vgster | fishy fish fish fish |
| 12:25.23 | cpm | <PROTECTED> |
| 12:29.55 | DrukenLPY | has anyone used asterisk with a security system? or a home protection system ? |
| 12:32.43 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 12:32.43 | kippi | hey |
| 12:32.56 | tzanger | maybe I'm just extraordinarily lucky... I've never ever had a woman smell like that |
| 12:33.03 | kippi | Has anyone got the firmware for a cisco 7912 handset for sip? |
| 12:33.17 | Qwell | kippi: Cisco does |
| 12:33.27 | kippi | but you can't download it |
| 12:33.31 | demlak | tzanger .avi and .jpg donīt smell =) |
| 12:33.31 | Qwell | yes you can |
| 12:33.40 | kippi | well you have to pay? |
| 12:33.43 | Qwell | yes |
| 12:34.23 | tzanger | demlak: zing! |
| 12:35.04 | demlak | everyday a new shortcut? wth does "zing" mean? |
| 12:37.31 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:40.48 | DrukenLPY | isn't the sip codec from cisco dependant on the mac address of the phone? |
| 12:41.04 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 12:46.18 | kippi | don't know, can't find it anywhere |
| 12:46.46 | DrukenLPY | i'm petty sure it is... |
| 12:46.58 | DrukenLPY | ie, why you have to PAY for it from cisco |
| 12:49.31 | kippi | hmm not sure |
| 12:49.40 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 12:52.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:52.59 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 12:55.20 | dacter | getting an error trying to build zaptel... http://rafb.net/p/HPnCfa69.html ...it's like I'm missing a build tool or something. |
| 12:55.34 | dacter | suggestions? |
| 12:58.05 | tzafrir | dacter, why do you run autoconf? |
| 12:58.27 | dacter | I didn't, that was part of the make script. |
| 12:58.37 | tzafrir | make script from where? |
| 12:59.32 | tzafrir | sorry. Me misreading error message |
| 12:59.34 | dacter | when I say "make" at the command line, it parses the file "Makefile" which has a call to autoconf. |
| 13:00.18 | tzafrir | menuselect/mxml/config.h.in seems to be missing |
| 13:00.32 | dacter | hmmm. |
| 13:00.33 | tzafrir | any chance it was accidentally deleted? |
| 13:01.55 | dacter | it turns out that that file is still there. |
| 13:04.34 | dacter | I wouldn't be a bit surprised if something did get deleted as part of the build process, though. |
| 13:04.35 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:05.07 | dacter | "ahhh, it's not like they're gonna run make more than once, unless they're like, dumb or something..." |
| 13:07.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:18.47 | dacter | compiling zaptel is like... that tree that always eats charlie brown's kite. |
| 13:18.58 | *** join/#asterisk Dantix (n=Andy_CAR@host117248040.arnet.net.ar) |
| 13:19.51 | Dantix | hi all, somone have a bit of experience on doing a doorphone connected to asterisk using an axis camera with audio support? |
| 13:20.21 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 13:21.29 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 13:24.00 | iCEBrkr | Dantix: Sounds like it'd be an interesting project |
| 13:24.55 | *** join/#asterisk qdk (n=qdk@193.226.189.115) |
| 13:30.52 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
| 13:30.52 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:33.05 | tzafrir | dacter, which standard build procedure eat your kite? |
| 13:33.27 | dacter | that would be "make" :) |
| 13:36.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:37.12 | Dantix | iCEBrkr: yes, but I can't achieve to axis tell me what protocol their cameras uses to transport the audio :( |
| 13:39.09 | *** join/#asterisk Peri (n=redanti@ap1.aci.on.ca) |
| 13:43.16 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 13:43.55 | *** join/#asterisk frawd (n=francois@87.220.29.1) |
| 13:45.23 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:45.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:45.33 | *** join/#asterisk dudanogueira (n=duda@unaffiliated/bitdoidao/x-000001) |
| 13:47.15 | frawd | hello, is it possible to force the zaptel timer to ztdummy when we have a zaptel compatible card plugged in? |
| 13:47.25 | SuPrSluG | Axis video products with firmware version 4.xx support G.711 Âĩ-law, G.726 24 |
| 13:47.35 | *** join/#asterisk uphill (n=julian@83-65-191-154.work.xdsl-line.inode.at) |
| 13:47.37 | uphill | hi |
| 13:47.37 | frawd | (I have big IRQ problems impossible to resolve) |
| 13:47.48 | uphill | i have problems using misdn |
| 13:48.09 | cpm | I'm having problems with my colon, it's probably age and stress related |
| 13:48.16 | uphill | sysfs_register |
| 13:50.00 | uphill | register_layer: register_sysfs failed -17 st(00000600) inst(40000600) |
| 13:50.00 | uphill | i always get this error and when i shutdown the system the kernel segfault |
| 13:50.01 | frawd | cpm: i'm sure you're not that old... |
| 13:50.23 | mosty | uphill: what kernel version? |
| 13:50.39 | dacter | frawd, you've probably already done it, but you've tried to change the irq of the card in the bios, right? |
| 13:51.05 | frawd | dacter: yep... seems i have something more like an io problem |
| 13:51.49 | uphill | mosty: 2.6.20-15-generic ubuntu |
| 13:52.00 | uphill | and the current 2.6er from kernel.org |
| 13:52.09 | frawd | dacter: i also tried different io schedulers, preemption options, timers, ... (i have the worst possible setup: xen + smp) |
| 13:52.12 | cpm | frawd, pretty old |
| 13:52.16 | uphill | 2 different machines |
| 13:52.20 | *** join/#asterisk ifnotwhynot (n=davidh@196.209.244.243) |
| 13:52.25 | uphill | everytime the same |
| 13:52.26 | dacter | If you don't mind my asking, what kind of processor? I know that the AMD Geode processors don't implement some APIC stuff which can make irq problems unfixable. |
| 13:52.36 | ifnotwhynot | the channel i dream about at night |
| 13:52.43 | frawd | AMD Athlon 64 X2 |
| 13:52.45 | ifnotwhynot | whats up people? |
| 13:53.06 | SuPrSluG | Dantix: take a gander. I think axis cameras are linux based. http://www.axis.com/files/developer/camera/audio.pdf |
| 13:53.22 | ifnotwhynot | need help again, yes,yes, i know what will i do without #asterisk on mirc |
| 13:53.39 | frawd | whenever i have a bit of load, zttest result can be pretty bad (i had worst values up to -106%), and crackling sound, ... |
| 13:53.59 | frawd | without load, everything perfect |
| 13:54.02 | uphill | misdn 1.1.3 hand-compiled |
| 13:54.17 | uphill | and in the kernel from kernel.org i patched it in |
| 13:54.52 | ifnotwhynot | i have a problem regarding the answer time in my dialplan using 4port digium fxo mod takes to long before it answers: 4 rings before it moves to the next context |
| 13:55.32 | frawd | any possible issue with xen (debian SMP kernel 2.6.18 with xen-dom0, patched with layer7) |
| 13:56.07 | ifnotwhynot | i have a problem regarding the answer time in my dialplan using 4port digium fxo mod takes to long before it answers: 4 rings before it moves to the next context any help will be appreciated thx |
| 13:56.11 | uphill | maybe i should use bristuff |
| 13:56.27 | uphill | bit i don't like bristugg |
| 13:56.49 | ifnotwhynot | i like to refer to it as bristuck |
| 13:56.49 | frawd | anyway, i've been on this for a few weeks now, and not really sure i can do anything on it... how could i force the timer to use ztdummy rtc timer? |
| 13:57.32 | uphill | ifnotwhynot: using misdn? |
| 13:57.39 | ifnotwhynot | no zaptel |
| 13:57.53 | ifnotwhynot | used it a while back on bri ports |
| 13:58.02 | ifnotwhynot | tdm400p to exact |
| 13:58.22 | uphill | i dont need more then one BRI port and 2 B-Channels |
| 13:58.28 | frawd | cpm: no more that 35 i'm sure! |
| 13:58.31 | ifnotwhynot | i have a 4 port fxo mod setup:uphill |
| 13:59.00 | ifnotwhynot | sorry i only got the tail of the problem :uphill whats the problem? |
| 13:59.33 | uphill | with misdn i get kernel errors, it works but quite verry unstable |
| 13:59.54 | uphill | sysfs_register... |
| 14:00.02 | ifnotwhynot | i have it working stable on fedora core 4 and asterisk 1.2 |
| 14:00.02 | uphill | i tried 2 kernels |
| 14:00.06 | uphill | 2 machines |
| 14:00.18 | mosty | uphill: works on debian etch for me |
| 14:00.31 | uphill | i have ubuntu 7.10 |
| 14:00.35 | uphill | server |
| 14:00.48 | uphill | mosty: 2 hfc-cards? |
| 14:01.02 | mosty | no, digium b104p or whatever |
| 14:01.26 | uphill | i hate this ******* |
| 14:01.28 | ifnotwhynot | digium b410p 4 port bri module |
| 14:01.49 | ifnotwhynot | what kerenl are you running? |
| 14:01.54 | tzafrir | uphill, 6.10? 7.04? |
| 14:01.54 | ifnotwhynot | uphill? |
| 14:02.05 | uphill | 7.10 i think |
| 14:02.06 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-b1f12f9bfec53bb4) |
| 14:02.24 | uphill | 2.6.20-15-generic |
| 14:02.30 | tzafrir | uphill, that one will probably be released at around Oct-2007 |
| 14:02.34 | tzafrir | 7.04 |
| 14:02.36 | ifnotwhynot | asterisk version? |
| 14:02.51 | uphill | i also patched a vanilla kernel from kernel.org with misdn |
| 14:03.02 | uphill | Asterisk 1.2.16 |
| 14:03.13 | uphill | asterisk is not the problem |
| 14:03.56 | uphill | if i modprobe the hfcpci module or do misdn-init start, the system has problems with all sysfs operations |
| 14:04.03 | *** join/#asterisk __tobi5_ (n=tobi@telstra.lt-ec.de) |
| 14:04.16 | ifnotwhynot | it took me two weeks to et mine stable, but i didnt have any kernel problems, had probs picking up 2 port on channel |
| 14:04.30 | __tobi5_ | ola |
| 14:04.35 | uphill | for example i get a real big bunch of kernel failures, if make a lspci |
| 14:04.44 | ifnotwhynot | what pc are you running? |
| 14:05.04 | tzafrir | uphill, one guess: find /lib/modules/`uname -r` -name hfcpci.ko |
| 14:05.04 | uphill | Celeron (Coppermine) 557 Mhz |
| 14:05.18 | tzafrir | more than one by any chance? |
| 14:05.21 | ifnotwhynot | i think thats your problem |
| 14:05.27 | uphill | it works, so the module is here |
| 14:05.32 | ifnotwhynot | what bri card are u using? |
| 14:05.38 | uphill | hfcpci |
| 14:05.44 | uphill | 2 pieces |
| 14:06.02 | *** join/#asterisk Wing|wrk (n=wing@newoffice-5.tvcom.ru) |
| 14:06.05 | ifnotwhynot | is it a ducberry? |
| 14:06.12 | uphill | one in te and the other in nt mode |
| 14:06.18 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 14:06.27 | uphill | conrad |
| 14:06.36 | ifnotwhynot | do you have a phone on the te mode? |
| 14:06.49 | uphill | no |
| 14:06.49 | ifnotwhynot | isdn phone that is? |
| 14:07.01 | ifnotwhynot | why do you put it as te mode? |
| 14:07.13 | uphill | 5 phones on nt line with a nt1 and ptp connection to the other card |
| 14:07.31 | ifnotwhynot | is this a multi port card? |
| 14:07.37 | uphill | 2 cards |
| 14:07.46 | uphill | one for internal isdn phones |
| 14:07.46 | ifnotwhynot | how many ports? |
| 14:07.54 | uphill | each 1 port |
| 14:08.20 | ifnotwhynot | are you connecting this to pa pots or a pbx? |
| 14:08.49 | __tobi5_ | i have some cisco 79xx-phones and call forwarding is not working, i always get this msg in the log: channel.c:2583 ast_request: No channel type registered for 'Local' and app_dial.c:485 wait_for_answer: Unable to create local channel for call forward to 'Local/01703401249@default' (cause = 66) |
| 14:08.59 | uphill | the teptp to a pots |
| 14:09.03 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 14:09.14 | uphill | ntptmp to the phones |
| 14:09.22 | __tobi5_ | call forwarding is the function in the cisco-phones |
| 14:09.44 | uphill | it is working, with clean sound, but it crashes sometimes with this strange sysfs stuff |
| 14:09.57 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:10.14 | ifnotwhynot | looks like irq probs? |
| 14:10.45 | ifnotwhynot | is there any zaptel pro's around? |
| 14:11.13 | uphill | each card have it own irq |
| 14:11.23 | uphill | and i also changed the machine |
| 14:11.45 | ifnotwhynot | if you do a lspci what do you get |
| 14:11.59 | uphill | one moment |
| 14:12.18 | uphill | i have to reboot because if misdn driver are loaded lspci crashes |
| 14:12.38 | ifnotwhynot | something major wrong there |
| 14:12.53 | uphill | i knwo |
| 14:13.18 | uphill | 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 14:13.22 | uphill | 2 times this |
| 14:13.24 | ifnotwhynot | did you reed up if maby someone else have tried setting up two of those cards on one server? |
| 14:13.49 | uphill | i can try only one |
| 14:13.57 | ifnotwhynot | if you remove one card do you still have this error? |
| 14:14.06 | uphill | is it possible to put avm fritz cards in teptp mode? |
| 14:14.30 | uphill | the i use one fritz card and one hfc |
| 14:14.42 | *** join/#asterisk frawd (n=francois@87.220.29.1) |
| 14:15.14 | *** join/#asterisk metabox (n=metabox@modemcable192.65-56-74.mc.videotron.ca) |
| 14:15.20 | ifnotwhynot | i am using those fritz cards for a swbphone on philips is3000 series pbx and i use the capiptp |
| 14:20.26 | __tobi5_ | nobody else with cisco-phones and call forwarding? |
| 14:21.05 | ifnotwhynot | tobi what cisco phone is it? |
| 14:21.13 | __tobi5_ | 7940 and 7960 |
| 14:21.28 | __tobi5_ | but both are working the same |
| 14:21.35 | ifnotwhynot | do you have a html interfase to the phone? |
| 14:21.41 | __tobi5_ | nope |
| 14:21.58 | ifnotwhynot | how do you set it up from the lcd? |
| 14:22.24 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 14:22.30 | neuwald | hi folks |
| 14:22.35 | __tobi5_ | there is a "call forward" button on the screen |
| 14:23.04 | neuwald | I'm having some trouble to put to work one E1 circuit with R2 signalling. anybody can help me ? |
| 14:23.10 | __tobi5_ | which is used to forward everything to a non-local number |
| 14:23.11 | ifnotwhynot | somewhere you need to enable to setup the code for the key? |
| 14:23.34 | ifnotwhynot | where do you edit this number tobi? |
| 14:23.38 | __tobi5_ | hmm, not found sth like that yet, but that would be great |
| 14:23.57 | __tobi5_ | when i press the button, a small line pops up which asks for the number, thats it |
| 14:24.25 | __tobi5_ | after that there is a msg on the screen all the time which says that everything is forwarded |
| 14:24.35 | __tobi5_ | to the number i put in |
| 14:25.13 | __tobi5_ | the problem is more on the asterisk-side, as the phone is forwarding the call to a context and channel which does not exist |
| 14:25.30 | __tobi5_ | Local/01703401249@default |
| 14:25.43 | __tobi5_ | that is an example for a forward that comes from the cisco |
| 14:25.54 | *** part/#asterisk Wing|wrk (n=wing@newoffice-5.tvcom.ru) |
| 14:26.15 | __tobi5_ | the context default is ok, but the Local/ in front of it confuses asterisk |
| 14:26.22 | __tobi5_ | No channel type registered for 'Local' |
| 14:27.28 | mosty | is chan_local.so loaded? |
| 14:27.33 | ifnotwhynot | do you have a context local in your extensions.conf file? |
| 14:27.56 | __tobi5_ | mom |
| 14:28.18 | __tobi5_ | no context local, but the context is "default" if i am not wrong |
| 14:29.02 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:29.35 | neuwald | I got the follow error when I receive a call: |
| 14:29.36 | neuwald | [May 14 11:28:53] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/11 event Detected |
| 14:29.36 | neuwald | [May 14 11:28:55] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/11 event Protocol failure |
| 14:29.36 | neuwald | [May 14 11:28:55] ERROR[2914]: chan_unicall.c:2603 handle_uc_event: Unicall/11 protocol error. Cause 32772 |
| 14:29.44 | __tobi5_ | ok, chan_local was at noload |
| 14:29.47 | ifnotwhynot | create a context local |
| 14:29.55 | neuwald | what does it could be ? |
| 14:29.57 | __tobi5_ | i reload and see what goes |
| 14:31.09 | ifnotwhynot | exten => _X.,1,Dial(Sip/${EXTEN}@default) add this line and do extention reload from your asterisk console |
| 14:32.03 | __tobi5_ | such a default-rule is already existing, but that would call a sip-channel, which would be wrong ;-) |
| 14:32.09 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:32.20 | *** join/#asterisk federicoco (n=federico@212.34.251.205) |
| 14:32.31 | __tobi5_ | my rule is looking for the number and chosses then which channel to dial |
| 14:32.36 | __tobi5_ | but it is like _X |
| 14:32.58 | __tobi5_ | i think it should work if asterisk know a channel-definition "local" |
| 14:33.06 | __tobi5_ | which chan_local should supply |
| 14:33.11 | __tobi5_ | one moment, i restart |
| 14:34.31 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 14:38.22 | __tobi5_ | peerrrrrrrfect, thanks a lot, chan_local.so was the trick |
| 14:38.30 | __tobi5_ | everything working now |
| 14:39.49 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:39.53 | *** part/#asterisk __tobi5_ (n=tobi@telstra.lt-ec.de) |
| 14:39.55 | ifnotwhynot | good one |
| 14:41.34 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 14:47.26 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:47.36 | coolbeans | Anybody seen this in 1.4.4? db.c:66 dbinit: Unable to open Asterisk database It worked fine until a reboot... |
| 14:48.49 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:51.13 | coolbeans | hrm... permissions |
| 14:53.14 | *** join/#asterisk DukeOfURL (n=Michael_@vc7-1-39.dsl.netrack.net) |
| 14:53.45 | DukeOfURL | Hello, where can I find docs for the GUI and users.conf? |
| 14:56.31 | *** join/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 14:56.47 | TUplink | i have no meetme app is there any easy way to fix that? |
| 14:57.08 | TUplink | Malamute*CLI> show application meetme |
| 14:57.08 | TUplink | Your application(s) is (are) not registered |
| 14:57.21 | Qwell[] | install zaptel, then rerun configure and reinstall asterisk |
| 14:57.30 | *** join/#asterisk _VoiceMeUp_COM (n=Miranda@145-27.mc.cite.net) |
| 14:57.34 | TUplink | um.... zaptel is instaled..... |
| 14:57.39 | Qwell[] | then rerun configure and reinstall asterisk |
| 14:57.52 | tzanger | that's it |
| 14:57.56 | tzanger | I'm naming my next box marmite |
| 14:57.59 | *** join/#asterisk locelavi (n=chatzill@221.204-242-81.adsl-dyn.isp.belgacom.be) |
| 14:59.00 | TUplink | ok im running cvsup first so i can get the latest for FreeBSD |
| 14:59.17 | TUplink | should i update zaptel? |
| 14:59.52 | TUplink | and will it keep all of my configs and sound files or should i backit up? |
| 15:02.25 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 15:03.26 | *** join/#asterisk _charly_ (i=nobody@83.246.113.130) |
| 15:03.34 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:03.49 | Dr-Linux | what's so special in Gizmo project, which doesn't have eyebeam etc? :S |
| 15:04.22 | *** join/#asterisk Gregabyte (i=wintermu@nat/digium/x-64f652804afd0a3d) |
| 15:04.34 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:06.02 | _charly_ | hi, may i ask to remove a ban that locks out my irc proxy? i guess the ban was set because my bnc had some connection problems, but those are fixed now |
| 15:06.31 | Qwell[] | _charly_: what hostname? |
| 15:06.53 | _charly_ | the ban is *!*=kroseneg@*.schmidham.net |
| 15:07.25 | *** mode/#asterisk [-b *!*=kroseneg@*.schmidham.net] by Qwell[] |
| 15:07.36 | _charly_ | thank you very much :) |
| 15:08.19 | _charly_ | brb |
| 15:08.37 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 15:09.52 | *** join/#asterisk bbryant (i=brett@nat/digium/x-d1f42944126a5dc8) |
| 15:10.04 | *** part/#asterisk Gringo_ (n=N3TW4LK3@250.102-247-81.adsl-dyn.isp.belgacom.be) |
| 15:10.32 | BrianR___ | Anyone here use vitelity? |
| 15:11.16 | tzanger | BrianR___: no, we typically don't use penis pills here |
| 15:11.36 | BrianR___ | tzanger: Heh. I'm talking about the voip<->pstn service :) |
| 15:11.51 | BrianR___ | The brand name does sound like a penis pill though.. |
| 15:11.54 | tzanger | you gotta admit that that sounds like osme kind of <cough> performance-enhancing drug |
| 15:12.04 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 15:13.13 | BrianR___ | Trying to solve a reinvite problem. I have this one DID which I'm using almost exclusively for forwarding.. It does a little IVR thing and then forwards the call to a PSTN number. Ideally it would do a reinvite so as not to waste bandwidth on my internet connection.. I set canreinvite=yes and so on, but I still see the RTP traffic. |
| 15:13.50 | *** join/#asterisk ifnotwhynot (n=davidh@196.209.244.243) |
| 15:15.43 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 15:19.17 | BSD_tech | ok we have asterisk-1.4.x + libpri-1.4.x + zaptel-1.4.x + asterisk-gui on bsd and its all working |
| 15:19.24 | BSD_tech | and looks nice |
| 15:19.44 | BSD_tech | all ate svn of branches |
| 15:19.51 | BSD_tech | ate/are |
| 15:20.36 | Nugget | Qwell[]: chan_skinny or chan_sccp or chan_sccp2 or what? I'm hopelessly lost. |
| 15:20.40 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:20.47 | Qwell[] | chan_skinny |
| 15:20.51 | Qwell[] | what are we talking about? |
| 15:20.51 | MrTelephone | what does it mean when your phone continually rings? |
| 15:20.58 | *** join/#asterisk littleball (n=littleba@bb220-255-93-109.singnet.com.sg) |
| 15:21.00 | Nugget | trying to get a 7961 talking to asterisk |
| 15:21.01 | MrTelephone | I plugged a phone into the adit 600 and it rang continously |
| 15:21.16 | TUplink | line voltage to high? |
| 15:21.44 | littleball | hello, i am looking for information about how to implement call transfer, waiting etc etc. I am not familiar with such terms. Please recommend some doc for me? |
| 15:21.59 | MrTelephone | what should the line voltage be |
| 15:22.09 | TUplink | http://www.voip-info.org |
| 15:22.30 | TUplink | MrTelephone honestly i dont know but i have mine at like 1 i belive |
| 15:22.34 | TUplink | 10* |
| 15:23.09 | TUplink | but if its high enough it will ring all day long |
| 15:23.42 | Sweeper | anyone using audiocodes mp-114/118 fxo? |
| 15:24.17 | littleball | hello, what is "ACD"? |
| 15:24.35 | [TK]D-Fender | littleball: WIKI time.... |
| 15:24.37 | [TK]D-Fender | ~acd |
| 15:24.40 | jbot | i heard acd is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 15:24.46 | TUplink | you put a cd in the cdrom drive |
| 15:24.50 | Iamnacho | lol |
| 15:24.59 | coppice | I thought it was A Call Drops |
| 15:25.01 | MrTelephone | 75 volts on a ring isn't it? |
| 15:25.23 | littleball | Automatic call distribution system? |
| 15:25.26 | TUplink | its somewhere in there.... i never could find it for sure |
| 15:25.36 | TUplink | call up verizon maybe they can tell you |
| 15:26.04 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 15:26.09 | coppice | 75V is for wimps. real phones use up to 110V AC - 48V DC for an extra kick :-) |
| 15:27.00 | [TK]D-Fender | no acd is <reply> ACD (aka. Call Queueing) Automatic Call Distribution. a call distribution tool that allows callers to be processed sequentially and routed using various strategies to an available agent. *or* All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 15:27.13 | [TK]D-Fender | ~acd |
| 15:27.16 | jbot | ACD (aka. Call Queueing) Automatic Call Distribution. a call distribution tool that allows callers to be processed sequentially and routed using various strategies to an available agent. *or* All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 15:27.19 | [TK]D-Fender | There |
| 15:29.41 | vader-- | is there a page that describes the differences between 1.2 and 1.4 asterisk? |
| 15:29.42 | *** part/#asterisk DukeOfURL (n=Michael_@vc7-1-39.dsl.netrack.net) |
| 15:29.43 | TUplink | Normally there is only the 48V DC voltage from central office battery in the line, and when phone ring there is 70-130V AC voltage in the line. That's what in the line in normal operation. |
| 15:29.43 | TUplink | http://www.epanorama.net/documents/surge/telesurge.html |
| 15:29.43 | [TK]D-Fender | vader--: UPGRADE.TXT |
| 15:29.43 | [TK]D-Fender | vader--: And go check the WIKI |
| 15:29.43 | BrianR___ | Hmm... Aparently vitelity doesn't allow reinvite at all... |
| 15:33.24 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 15:34.07 | MrTelephone | i tested the voltage from the adit and it was 25 volts and the telco is 50 volts |
| 15:34.08 | tuan_modulis | anyone know how to disable CONFIG_RTC using make menuconfig ? just want to know where the option is from that interface |
| 15:34.21 | tuan_modulis | (for kernel) |
| 15:34.45 | MrTelephone | fender that wasn't very nice :( |
| 15:35.38 | TUplink | is asterisk calling the phone. stupid question but maybe.... |
| 15:35.48 | TUplink | its ringing for a reason |
| 15:36.08 | MrTelephone | well i think I have my t1 timings wrong |
| 15:36.19 | MrTelephone | the telco pri is set to normal and i set the adit timing to master with relation to the telco |
| 15:36.23 | TUplink | ok..... you just jumped out of my part |
| 15:36.25 | MrTelephone | i think they should both me normal |
| 15:36.48 | TUplink | know nothing about t1 |
| 15:36.52 | TUplink | :( |
| 15:38.30 | *** join/#asterisk MindTheGap (n=iote@c9502ba2.bhz.virtua.com.br) |
| 15:39.22 | MrTelephone | ringing voltage is 85volts it says in the book |
| 15:39.51 | TUplink | 70-130 |
| 15:39.57 | TUplink | 85 falls inthere |
| 15:40.45 | MrTelephone | the fxs card supports both loopstart and ground start but im using loopstart |
| 15:40.58 | MrTelephone | I think its a t1 problem because the lights arn't green on the card |
| 15:42.19 | MrTelephone | lots of line code violations |
| 15:42.34 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 15:42.45 | rene- | hello, i have a question about the 1.4 jitter buffer |
| 15:43.16 | rene- | if i am taking inbound calls over sip, would it do any good to the call quality to enable the jitter buffer? |
| 15:44.08 | MrTelephone | dtmf connection doesn't work as well I hear |
| 15:44.25 | MrTelephone | when using jitter buffer |
| 15:45.26 | MrTelephone | what happens if you get ring and tip mixed up? |
| 15:46.17 | *** join/#asterisk littleball (n=littleba@bb220-255-93-109.singnet.com.sg) |
| 15:47.09 | littleball | hello, i am looking for some doc for how to implement call queuing with remote agent application, who can recommend? |
| 15:47.16 | rene- | i am not using dtmf in this particular app |
| 15:47.30 | rene- | i guess, that could be avoided using out of band dtmf |
| 15:47.46 | MrTelephone | yeah |
| 15:47.57 | MrTelephone | I would try it because everyones application is a little different |
| 15:48.12 | MrTelephone | that is something that really can't be decided upon |
| 15:48.18 | MrTelephone | is your call quality poor now? |
| 15:48.22 | MrTelephone | do you have a lot of network jitter? |
| 15:48.37 | tuan_modulis | jitter from dtmf? |
| 15:48.44 | MrTelephone | the further away your customers the more likely the packets will come in a rearranged order |
| 15:49.15 | MrTelephone | im waiting for all the calls on the system to stop so I can restart my pri card |
| 15:49.17 | MrTelephone | heh |
| 15:49.53 | *** part/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:50.02 | TUplink | restart gracefully |
| 15:50.59 | MrTelephone | sweet option |
| 15:51.04 | MrTelephone | didn't even know it existed |
| 15:51.13 | MrTelephone | I have to unload and reload the kernel mods though |
| 15:51.49 | TUplink | then stop gracefully |
| 15:52.18 | MrTelephone | I have a dual zeon with 2 pci-64 bit slots.. one with a raid 5 card and one with a sangoma dual pri card.. when I run both spans I get a lot of overruns that show up by using 'ifconfig' |
| 15:52.48 | BSD_tech | Sangoma ruls |
| 15:52.49 | MrTelephone | the 2 pci-64 slots use the same bus and the pci-32 bits are on a different bus but my riser card doesn't support jumping the bus |
| 15:52.51 | BSD_tech | rules |
| 15:53.16 | MrTelephone | but how the hell can I run out of interrups with a dual xeon.. very odd |
| 15:53.27 | MrTelephone | I'm partial to sangoma right now as It has the same irq issues as digium |
| 15:53.48 | *** join/#asterisk wunderkin (n=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 15:54.19 | rene- | MrTelephone i have a lot of audio cuts |
| 15:54.33 | rene- | i used to fixed them easily with the old iax jb |
| 15:54.38 | rene- | but i am using sip now |
| 15:54.44 | rene- | and quality is dismal |
| 15:54.48 | shido6 | hrmm |
| 15:54.59 | MrTelephone | what is your wan quality like? |
| 15:55.01 | shido6 | there are some linux and bios adjustments you'll need to make |
| 15:55.06 | MrTelephone | are you trying to run a server off a cable network? |
| 15:55.25 | MrTelephone | shido6, are you talking about removing the hyperthreading? |
| 15:55.33 | shido6 | no. |
| 15:56.02 | TUplink | my server runs off of cable..... seemes to do fine.... |
| 15:56.07 | MrTelephone | aacraid uses barely any interrupts |
| 15:56.16 | TUplink | only had 6 inbound calls tho.... |
| 15:56.25 | MrTelephone | 6 calls g729? |
| 15:56.33 | TUplink | gsm |
| 15:56.43 | MrTelephone | I get jitter from headoffice to my cable customers sometimes due to plant rf problems |
| 15:57.05 | MrTelephone | someones dog eats the cable wire and they tape the end to their tv |
| 15:57.08 | shido6 | using setpci, cat /proc/interrupts, use hdparm to free up interrupt times, dont share interrupts ( go into the bios and specifically set irq's ) |
| 15:57.11 | TUplink | i just use mine to play with.... |
| 15:57.13 | MrTelephone | then a skidoo drives by and frigs up half the town |
| 15:57.43 | BrokenNoze | Can anyone tell me why when i call the Park() application it announces the parking space number to the calling party? |
| 15:58.02 | rene- | MrTelephone: it is very good, i have a 10mbps connection to the internet |
| 15:58.15 | rene- | i have a sprint virtual t1 and works really well |
| 15:58.26 | MrTelephone | nice |
| 15:58.53 | rene- | they colocated a cisco router that provides me with a real t1 interface and goes out on the internet tru a vpn tru my main connection |
| 15:59.15 | MrTelephone | 209: 32156 0 319551 1 IO-APIC-level aacraid |
| 15:59.15 | MrTelephone | 225: 190917031 37200115 199314379 50221234 IO-APIC-level wanpipe1, wanp |
| 15:59.23 | shido6 | recompile the kernel with APIC so you know everything gets its own IRQ |
| 15:59.31 | MrTelephone | rene how much does that cost? |
| 15:59.33 | shido6 | do it the Manxpower way |
| 15:59.34 | shido6 | :) |
| 15:59.38 | MrTelephone | shido, it does |
| 15:59.52 | MrTelephone | everything has its own irq |
| 15:59.55 | rene- | it is expensive tho, i believe the setup for it was upwards 15,000 usd |
| 15:59.58 | rene- | i am located in Cancun |
| 16:00.04 | shido6 | if everything has its own irq, then go down to the next tweak |
| 16:00.11 | MrTelephone | rene, don't get killed down there |
| 16:00.23 | MrTelephone | whats the next tweak? |
| 16:00.30 | rene- | heh |
| 16:00.43 | rene- | voice quality is flawless |
| 16:00.54 | MrTelephone | the overruns are .008% but i don't think there should be any |
| 16:01.03 | MrTelephone | rene, where do you get jitter then? |
| 16:02.12 | MrTelephone | cancun is a long ways away to have customers say in, canada for instance |
| 16:02.26 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 16:02.39 | MrTelephone | I thought about starting a voip service for other asterisk users but it would be better to rent space for a server in toronto |
| 16:05.45 | tuan_modulis | it helps to have the timer frequency upped too |
| 16:06.31 | tuan_modulis | to 1000hz |
| 16:06.32 | rene- | MrTelephone: i know, our costumers are mostly located in the USA and Canada |
| 16:07.46 | rene- | but the problem is with some calls coming in from the USA tho the actual costumers are located in the central portion of mexico |
| 16:08.38 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:08.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 16:08.55 | tuan_modulis | now as for me, i cant seem to disable CONFIG_RTC... I removed the RTC class from Device Drivers, but it still pops up |
| 16:09.09 | *** part/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 16:09.34 | tuan_modulis | as CONFIG_RTC=y |
| 16:11.40 | Mad|Cow | odes anyone know if meetme has a limiation as to how many people can be part of a conference? |
| 16:17.17 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 16:18.22 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 16:19.32 | *** join/#asterisk PioneerVM3 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 16:19.48 | mosty | how does sip call forwarding work? |
| 16:20.21 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:20.26 | mosty | when i setup my sip phone to forward calls to a mobile phone, asterisk seems to think it's dialed the mobile ok, but the mobile never rings |
| 16:20.32 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 16:21.29 | mosty | i can dial the mobile from my sip phone directly, and it works |
| 16:21.32 | *** join/#asterisk PioneerVM3 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 16:22.09 | mosty | and if i tell my sip phone to forward to a pstn line, it works too |
| 16:24.38 | *** join/#asterisk Fieldy (i=fli9SI81@gentoo/contributor/Fieldy) |
| 16:26.14 | *** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za) |
| 16:27.09 | tuan_modulis | it's supposed to do a dial, and bridge the two channels |
| 16:27.10 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 16:28.22 | mosty | tuan_modulis, it does do that, it even bridges the channels, but the other end never rings even though asterisk says it is |
| 16:29.18 | SoftIce | hi from earlier versions of a2billing I know certain extra tables have been added but with regards to table structure is everything still similar? |
| 16:29.19 | SoftIce | or the same? |
| 16:29.29 | SoftIce | as I need to do an import from an old postgres db to a mysql db |
| 16:29.50 | tuan_modulis | mosty: pretty weird... codec issue maybe? |
| 16:29.53 | SoftIce | and I cant re-type out these thousand entries, so i'm converting the db, just playing it safe by wondering how much with regards to table layout has changed |
| 16:30.37 | mosty | tuan_modulis, but i can dial the mobile number from the sip phone without any problems, so i don't think it's a codec issue |
| 16:35.13 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:40.07 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 16:40.56 | *** join/#asterisk [Mr_X] (n=mrx@88.118.94.171) |
| 16:41.04 | *** join/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 16:41.08 | *** part/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 16:41.32 | *** part/#asterisk [Mr_X] (n=mrx@88.118.94.171) |
| 16:41.38 | *** join/#asterisk [Mr_X] (n=mrx@88.118.94.171) |
| 16:42.18 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 16:42.52 | `Sean | ido |
| 16:42.58 | `Sean | how did you get free polycoms |
| 16:42.58 | `Sean | :P |
| 16:43.49 | LeddyHM | Anyone have any ideas on this http://pastebin.ca/487729 |
| 16:44.06 | LeddyHM | looks like the /etc/init.d/zaptel script is a little goofy |
| 16:44.12 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-0badb18fcd71a71f) |
| 16:44.38 | *** join/#asterisk sharp (n=sharp@pool-72-94-223-131.phlapa.east.verizon.net) |
| 16:51.08 | *** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com) |
| 16:51.49 | LeddyHM | bueller.... bueller |
| 16:52.13 | `Sean | http://www.xorcom.com/products/astribank_8 |
| 16:52.23 | `Sean | not bad for use with a laptop on hte go |
| 16:52.24 | `Sean | :D |
| 16:53.54 | Tond | H I am having a very wierd issue with asterisk 1.4.4 and DTMF. I have tried all different DTMF setting on Asterisk and my Cisco router. but here is the issue it is always skiping the digit dialed so if i dial 1234567 it only receives 123456 and if i dial duplicate numbers it ignores it and also skips the last digit dialed. I.e. 122345678 it gets 1234567 |
| 16:54.02 | Tond | has anyone ever had this issue? |
| 16:55.37 | *** join/#asterisk johann8384 (n=johann83@206.80.75.240) |
| 16:57.16 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:57.30 | hi365 | can you set MOH to play for x amount if time?? |
| 16:59.22 | [TK]D-Fender | hi365: Dial a local channel with an absolute timeout and dump them in MoH. |
| 16:59.44 | hi365 | cool. thanks |
| 17:00.05 | *** join/#asterisk [reed] (n=reed@firefox/gnu.webmaster.reed) |
| 17:00.39 | wunderkin | waitmusiconhold? |
| 17:01.05 | *** join/#asterisk johann8384 (n=johann83@206.80.75.240) |
| 17:01.07 | hi365 | eben better! |
| 17:01.11 | hi365 | even better! |
| 17:01.26 | *** part/#asterisk ifnotwhynot (n=davidh@196.209.244.243) |
| 17:02.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 17:03.58 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 17:04.25 | *** join/#asterisk SnrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 17:04.26 | SnrWup | hey |
| 17:04.32 | [Mr_X] | Hi. |
| 17:06.28 | SnrWup | im using asterisk PBX with digium FXO card to connect to the PSTN with standard analogue telephone lines in the UK. asterisk detects hangups from the zap channels just fine. the digium card is configured with loadzone, progzone, opermode all to UK. However, when I dial out using a zap channel, asterisk dials properly, the call connects, but asterisk does not detect when the person being called by my digium card picks up their handset. The |
| 17:06.41 | SnrWup | The call therefore timesout with status NOANSWER, even though you can hear and talk to someone across the PSTN |
| 17:07.33 | SnrWup | as mentioned, incoming calls are fine, hangups are detected, digium card works properly in every other respect |
| 17:07.38 | SnrWup | any ideas anyone? |
| 17:07.39 | *** join/#asterisk CUF (n=usuario@44.Red-213-98-68.staticIP.rima-tde.net) |
| 17:09.01 | SnrWup | man this channel is *always* dead |
| 17:09.08 | dan__t | 'morning. |
| 17:09.16 | SnrWup | hey |
| 17:09.28 | *** join/#asterisk CUF (n=usuario@44.Red-213-98-68.staticIP.rima-tde.net) |
| 17:09.32 | CUF | Hi all |
| 17:09.54 | CUF | I'm trying to compile Asterisk 1.4.x with chan_misdn channel module |
| 17:10.17 | CUF | all goes ok except for the detail that the module compilation is disabled |
| 17:10.23 | tzafrir | SnrWup, if you're in the UK, why do you use busydetect? |
| 17:10.47 | SnrWup | busydetect? |
| 17:10.55 | CUF | it appears XXX 9. chan_misdn on the 'make menulselect' |
| 17:10.57 | tzafrir | you can detect hangup by polarity reversal (unless you have an x100p) |
| 17:11.13 | SnrWup | tzafrir i have that set to yes |
| 17:11.27 | CUF | it says the module depends on isdnnet(E), misdn(E), suppserv(E) |
| 17:11.44 | CUF | I think misdn is fine because zaptel did compile ok |
| 17:11.49 | SnrWup | answeronpolarityreversal=yes and hanguponpolarityreversal=yes |
| 17:11.58 | SnrWup | or whatever those two options are, theyre both yes |
| 17:12.10 | CUF | however i have no idea what packages do i have to install to satisfy dependencies for suppserv and isdnnet |
| 17:12.17 | CUF | any idea? |
| 17:12.22 | CUF | can anyone help me please? |
| 17:12.31 | CUF | I'm using a Digium B410P |
| 17:12.49 | tzafrir | anyway, I believe that asterisk consideres the call answwered as soon as the FXO port "picks" the phone |
| 17:12.54 | CUF | and debian 4 |
| 17:13.31 | CUF | :'( |
| 17:13.34 | SnrWup | tzafrir you mean as soon as dial is completed? |
| 17:13.42 | tzafrir | CUF, the ones in main are broken, and has thus been removed from Etch |
| 17:13.56 | CUF | I'm not using the Debian packages |
| 17:14.02 | CUF | I'm building from source |
| 17:14.15 | tzafrir | I started building a set of my own, at http://updates.xorcom.com/rapid |
| 17:14.55 | CUF | I have been following Digium's official pdf on b410p installation |
| 17:15.20 | CUF | all works ok except for the fact that chan_misdn doesn't build |
| 17:15.48 | CUF | tzafrir: what do you mean? |
| 17:16.42 | MrTelephone | i can't get dialtone out of the adit 600 :( |
| 17:17.07 | tzafrir | CUF, those are not very successful at this stage, so I'm not really sure you should bother |
| 17:17.46 | tzafrir | but generally instead of saying "A didn't work" it would help stating "I did A expecting B but got C" |
| 17:17.57 | CUF | tzafrir: what do you mean "not very successful at this stage"? the card is bad? |
| 17:18.23 | CUF | tzafrir: well, i compiled the zaptel package first as indicated |
| 17:18.44 | CUF | tzafrir: make clean, make install then make b410p |
| 17:18.51 | tzafrir | I tried building asterisk when they were installed on the system. Got some build errors that I had no time to check. This was a low priority (I needed the asterisk package fast) so I left it |
| 17:19.03 | CUF | after everything ran fine |
| 17:19.13 | CUF | i started building asterisk |
| 17:19.16 | CUF | ./configure |
| 17:19.35 | CUF | /make menuselect |
| 17:19.42 | *** join/#asterisk sharp (n=sharp@pool-72-94-223-131.phlapa.east.verizon.net) |
| 17:20.03 | CUF | and when I go to channel drives the chan_misdn module is disabled |
| 17:20.20 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 17:20.26 | CUF | I can compile asterisk just fine |
| 17:20.37 | CUF | is the chan_misdn channel module what I can not build |
| 17:20.38 | naitram | anyone ever use pjsip with asterisk |
| 17:21.11 | bkruse | CUF: make menuselect and see why its not building |
| 17:22.09 | CUF | bkruse: it says it depends on isdnnet(E), misdn(E), suppserv(E) |
| 17:22.23 | CUF | bkruse: its all it says |
| 17:22.45 | naitram | anyone know of an open source sip client for win ce |
| 17:22.57 | naitram | besides pjsip |
| 17:23.04 | CUF | I have no clue what to install to meet the suppserv dependency |
| 17:23.20 | CUF | nor the isdnnet |
| 17:23.22 | SuPrSluG | CUF:libmisdn-dev |
| 17:23.43 | SuPrSluG | CUF:for debian |
| 17:23.55 | bkruse | CUF: install those |
| 17:24.13 | bkruse | CUF: you on a debian system? |
| 17:24.35 | CUF | bkruse: yes, I'm on debian 4 |
| 17:24.41 | bkruse | yay |
| 17:24.44 | bkruse | apt-cache search misdn |
| 17:24.57 | file | bkruse: let my cake open the door! |
| 17:24.57 | MrTelephone | does anyone here use an adit600 that can help me out with the configuration? |
| 17:25.07 | bkruse | file: it has! |
| 17:25.09 | CUF | SuPrSluG: libmisdn-dev does not exist as a debian package on the debian repositories |
| 17:25.10 | SnrWup | whats the cheapest and most effective way of connecting 8 ISDN PRI lines to asterisk? |
| 17:25.17 | SnrWup | some sort of PCI card? channel bank? what? |
| 17:25.22 | bkruse | CUF: apt-get build-dep asterisk-chan-misdn libmisdn-dev |
| 17:25.26 | SuPrSluG | i show it |
| 17:25.28 | bkruse | that should cover dependencies |
| 17:25.36 | bkruse | ./configure ; and make menuselect again |
| 17:26.52 | SuPrSluG | CUF:it's out there but u may have to add it to your repositories |
| 17:27.15 | bkruse | apt-get update, gah |
| 17:27.50 | CUF | SuPrSluG: any idea of a repository where I can find those libs? |
| 17:28.01 | CUF | SuPrSluG: I'm not a debian expert |
| 17:28.40 | *** join/#asterisk saftsack (n=oliver@pD9E06F9E.dip.t-dialin.net) |
| 17:29.05 | SuPrSluG | CUF:try the testing or experimental ones for your distro. |
| 17:29.39 | CUF | They do not work |
| 17:29.41 | _charly_ | misdn is not included in debian stable/sarge, it's only in unstable/sid |
| 17:29.56 | CUF | problem is that bloody misdn changes every second |
| 17:30.06 | CUF | has to be built from source |
| 17:30.19 | CUF | I had this problem before |
| 17:30.25 | CUF | but managed to build it once |
| 17:30.36 | CUF | :( |
| 17:30.43 | CUF | :'( |
| 17:31.11 | SuPrSluG | All Debian Packages in "oldstable" |
| 17:32.46 | tzafrir | I wouldn't trust the misdn packages from debian/oldstable. They're really bad. Plus, you may actually pull a dependency to "kernel-image" and such |
| 17:33.17 | tzafrir | I also tend to believe that asterisk-chan-misdn requires asterisk-dev to build |
| 17:33.31 | CUF | SuPrSluG: The version on the debian repositories is way too old |
| 17:33.51 | _charly_ | i'm currently trying to build misdn debian etch packages for myself, but this will still need some days, i could give them to you when i'm finished, if you want |
| 17:34.08 | CUF | _charly_: that would be wonderful |
| 17:34.16 | tzafrir | Which I personally consider a good idea (if you install asterisk as well), but some would not like it |
| 17:34.47 | CUF | _charly_: however what I really need is to learn once for all how to compile it myself and make the bloody b410p card work |
| 17:35.13 | CUF | _charly_: Digium's own doc about how to install it is crap |
| 17:35.38 | CUF | _charly_: I never managed to make it work on asterisk 1.2.x, only once on 1.4.x |
| 17:36.01 | tzafrir | _charly_, I'm also interested to hear about such packages |
| 17:36.26 | hi365 | is it posible to exitcute commands after dial?? |
| 17:36.31 | tzafrir | note the different file location between what the original misdn deb package put and what asterisk seems to expect |
| 17:37.35 | tzafrir | I chose to simply put the files where asterisk expects them |
| 17:37.42 | CUF | Hey |
| 17:37.45 | CUF | stop the press |
| 17:37.46 | CUF | <PROTECTED> |
| 17:37.47 | CUF | In short, thinking about ISDN and 1.4? Stick with the 1.2 version for now. As ever ISDN is the poor cousin. |
| 17:37.56 | kippi | hey |
| 17:38.03 | CUF | aaarhgh!!!!! |
| 17:38.06 | CUF | aaargh!!!! |
| 17:38.32 | Nivex | I'll see your "aargh!!!!" and raise you a "gaaah!!!!" |
| 17:38.48 | kippi | how can I program softkeys on cisco handsets? |
| 17:38.58 | CUF | oh my god I want to die |
| 17:39.08 | CUF | how can i tell my customer that!!! |
| 17:39.30 | CUF | I bought the F**** card because is the official digium one!!! |
| 17:40.11 | CUF | I'm dead |
| 17:40.18 | CUF | I'm F***** dead |
| 17:41.02 | danp | you can't just use 1.2? |
| 17:41.21 | danp | oh, i guess you already tried that |
| 17:41.32 | Nivex | I think CUF needs a little bit less stress in h(is|er) life |
| 17:41.59 | CUF | Nivex: I'm my long life I never had to deal with such a pain as asterisk |
| 17:42.14 | wunderkin | CUF, call digium |
| 17:42.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:43.24 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 17:44.21 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:45.38 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:45.57 | DJ-Pyro_ | is there any way to specify multiple ip's or an ip range for a sip peer? |
| 17:47.52 | MrTelephone | if you have a second span with fxols singnalling should you still see the span by typing pri show span 2? |
| 17:47.57 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 17:48.09 | MrTelephone | my first port is span=1,0,0,esf,b8zs |
| 17:48.31 | MrTelephone | and my second port is span=2,0,0,esf,b8zs |
| 17:48.31 | [TK]D-Fender | DJ-Pyro_: nope. Either add full auth infor to your dialplan and cycle through, or creat multiple peers |
| 17:48.43 | wunderkin | MrTelephone, no because thats not pri |
| 17:48.51 | [TK]D-Fender | MrTelephone: One of those should have "1" as the 2nd parm to indicate a primary timing source |
| 17:48.58 | wunderkin | and that |
| 17:51.24 | [TK]D-Fender | MrTelephone: And we can't tell your signalling from Zaptel, because thats set in ZAPATA. Pastebin BOTH please |
| 17:51.42 | anonymouz666 | dammit my connection is so fucking slow. I am losing dtmf digits. |
| 17:51.43 | anonymouz666 | can't test the IVR |
| 17:51.43 | MrTelephone | primary as in master? |
| 17:51.48 | [TK]D-Fender | anonymouz666: what does connection have to do with DTMF? You using inband on ulaw or something crazy? |
| 17:51.57 | anonymouz666 | something crazy |
| 17:52.19 | MrTelephone | what is the pastebin url |
| 17:52.20 | [TK]D-Fender | MrTelephone: No, "0" says * will act as the timing source and that will cause sync issues. you should have 1,2,3,4 etc across your spans |
| 17:52.57 | MrTelephone | if your using fxo or fxs signalling you can't show pri span on that port? |
| 17:53.33 | anonymouz666 | [TK]D-Fender its rfc2833 with g729. |
| 17:53.55 | anonymouz666 | if I type 123456789. read() will recognize something like 125678 |
| 17:53.57 | anonymouz666 | lots of digits missing |
| 17:53.59 | shido6 | ok |
| 17:55.06 | [TK]D-Fender | anonymouz666: you really shouldn't be losing digits.... thats jsut nuts |
| 17:55.12 | [TK]D-Fender | anonymouz666: Whats on the far side? |
| 17:55.26 | anonymouz666 | I am using relaxdtmf=yes |
| 17:55.36 | anonymouz666 | it's sip to sip |
| 17:55.47 | anonymouz666 | eyeBeam talking to ast |
| 17:56.33 | MrTelephone | eyebeam is good |
| 17:56.40 | *** join/#asterisk Dandan (n=dandan@ip68-9-233-149.ri.ri.cox.net) |
| 17:56.43 | Dandan | hello :) |
| 17:56.50 | MrTelephone | anonymouz, if you hold the button down longer it will detect dtmf |
| 17:57.02 | Dandan | any * developers here? Need to do some small changes |
| 17:57.02 | Dandan | :) |
| 17:57.06 | [TK]D-Fender | anonymouz666: there is no "relaxdtmf" with RFC... its OOB |
| 17:57.30 | Dandan | when logging in to asterisk console, there need to be a small change: show warranty should be changed to: core show warranty :) |
| 17:57.35 | Dandan | for good housekeeping |
| 17:57.36 | anonymouz666 | but there is relaxdtmf in sip.conf |
| 17:57.41 | MrTelephone | fender, asteirsk doesn't recognize a pri unless it has dchan/bchan settings? |
| 17:58.00 | [TK]D-Fender | MrTelephone: You should have that as well |
| 17:58.14 | [TK]D-Fender | MrTelephone: and YES |
| 17:58.26 | MrTelephone | its running a channel bank so I can't show pri span on it |
| 17:58.51 | hi365 | can you control the volume of music on hold? |
| 17:59.04 | kippi | how can I program softkeys on cisco handsets? |
| 17:59.27 | MrTelephone | interesting |
| 18:00.07 | *** part/#asterisk CUF (n=usuario@44.Red-213-98-68.staticIP.rima-tde.net) |
| 18:00.32 | Dandan | high-rez: I do normalize on all my moh .mp3s |
| 18:00.39 | Dandan | hi365: I do normalize on all my moh .mp3s |
| 18:00.41 | Dandan | darn |
| 18:01.03 | hi365 | i guess i could do that, although i meant do it in real time |
| 18:01.23 | [TK]D-Fender | MrTelephone: Most channel banks just talk straight T1 with independent CAS signalling for each channel. |
| 18:01.46 | [TK]D-Fender | MrTelephone: fx[][]s |
| 18:01.48 | Dandan | dunno :) |
| 18:02.27 | [TK]D-Fender | hi365: You can do this if you use an external MoH source, but not Native. |
| 18:02.31 | hi365 | thanks |
| 18:02.35 | MrTelephone | im worried because it was only outputting 25v instead of 48/50v |
| 18:02.38 | [TK]D-Fender | hi365: mpg123's method is well documented already |
| 18:03.55 | MrTelephone | -48 volts must be needed to power a phone |
| 18:04.29 | anonymouz666 | [TK]D-Fender WARNING[31842]: chan_sip.c:1900 retrans_pkt: Maximum retries exceeded on transmission 74434c49f0487639 for seqno 2 (Critical Response) |
| 18:04.39 | *** join/#asterisk ManxPower (n=manxpowe@4.sub-70-221-75.myvzw.com) |
| 18:04.58 | anonymouz666 | this can't be related ? |
| 18:05.10 | anonymouz666 | losing lots of packets |
| 18:05.47 | neuwald | folks |
| 18:05.59 | neuwald | I'm having problem when applying unicall patch on asterisk 1.2.18 |
| 18:06.05 | [TK]D-Fender | anonymouz666: You behind NAT? |
| 18:06.06 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:06.09 | MrTelephone | anonymouz, your probably using inband dtmf which means your losing rtp packets |
| 18:06.20 | [TK]D-Fender | MrTelephone: He's not. Pay attention. |
| 18:06.25 | neuwald | saturn:/usr/src/asterisk1.2/asterisk-1.2.18/channels# patch < channels_Makefile.patch |
| 18:06.25 | MrTelephone | oh |
| 18:06.28 | neuwald | patching file Makefile |
| 18:06.30 | neuwald | Hunk #1 succeeded at 77 with fuzz 1 (offset 14 lines). |
| 18:06.33 | neuwald | Hunk #2 succeeded at 105 (offset 23 lines). |
| 18:06.35 | neuwald | Hunk #3 FAILED at 172. |
| 18:06.37 | neuwald | <PROTECTED> |
| 18:06.40 | neuwald | 2 out of 4 hunks FAILED -- saving rejects to file Makefile.rej |
| 18:06.50 | neuwald | any idea of what does it could be ? |
| 18:07.00 | ManxPower | neuwald: Your patch is for a different version of Asterisk |
| 18:07.08 | ManxPower | You'll need to patch the two failed hunks by hand. |
| 18:07.17 | bkruse | wont be that hard |
| 18:07.51 | ManxPower | You MIGHT be able to use the "fuzz" option to patch. See the man page. |
| 18:08.23 | neuwald | ManxPower ok |
| 18:08.28 | neuwald | now I run asterisk and I got |
| 18:08.32 | neuwald | <PROTECTED> |
| 18:08.32 | neuwald | May 14 15:07:36 WARNING[19662]: loader.c:555 load_modules: Loading module chan_unicall.so failed! |
| 18:08.56 | anonymouz666 | I am not using inband dtmf |
| 18:08.59 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 18:09.00 | anonymouz666 | I am using rfc2833 |
| 18:09.07 | anonymouz666 | and yes I am behind NAt |
| 18:09.25 | *** join/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 18:09.45 | anonymouz666 | but it isn't a problem because my digits (some) gets there and I can hear the prompts |
| 18:09.52 | [TK]D-Fender | anonymouz666: I've double check your settings to be sure |
| 18:10.06 | [TK]D-Fender | anonymouz666: then barring that do some real bandwidth tests |
| 18:10.15 | neuwald | ManxPower any idea of what does it could be ? |
| 18:10.15 | ManxPower | anonymouz666: Using inband with a compressed codec can make SOME of the DTMF work. |
| 18:10.34 | ManxPower | neuwald: A bad patch or a version mismatch between Unicall and Asterisk |
| 18:11.11 | anonymouz666 | there is a checkbox for rfc2833 in eyebeam... I am using with g729 the checkbox of inband dtmf is not active |
| 18:12.04 | ManxPower | anonymouz666: and on the asterisk side? |
| 18:12.44 | anonymouz666 | rfc2833 |
| 18:13.17 | neuwald | ManxPower I'm using libunicall 0.0.3 and asterisk 1.2.18 |
| 18:13.32 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 18:13.38 | ManxPower | neuwald: what verison of does unicall say it requites? |
| 18:13.46 | teknoprep | hi all |
| 18:14.04 | *** join/#asterisk Brandon_W (n=Brandon_@machine76.Level3.com) |
| 18:14.10 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 18:14.12 | anonymouz666 | neuwald: don't use the lastest because won't work |
| 18:14.30 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 18:14.34 | Brandon_W | I have a rather stupid question. What is the best way to see what version of asterisk is currently running? |
| 18:14.42 | anonymouz666 | you can safely download the pre9 packages... and spandsp0.0.3pre22... |
| 18:14.52 | anonymouz666 | that I know that works because I use. |
| 18:15.19 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:15.21 | MrTelephone | show version |
| 18:15.41 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 18:16.32 | ManxPower | show version really only works for released stuff, not svn/cvs |
| 18:17.12 | tmccrary | I have a PRI via digium card going to an Avaya/Lucent Magix PBX and I'd like to be able to dial out. In order to dial out, you have to dial 887, wait for a new dial tone, THEN dial 9 and your number. Is there a way to do this with asterisk? I tried using the w wait character, but it doesn't seem to help. When you dial 887, it doesn't create a new call, but merely starts a new one in band or soemthing |
| 18:17.22 | tmccrary | Any help would be appreciated |
| 18:17.57 | [TK]D-Fender | tmccrary: There is no waiting in you dial with PRI, its DIGITAL |
| 18:18.19 | Brandon_W | trying to translate what SVN-trunk-r7230 is :) |
| 18:18.21 | [TK]D-Fender | tmccrary: Perhaps you need to send the digits as DTMF after sending your prefix. There is a DIAL parameter for this. |
| 18:18.52 | lesouvage | I want to run a mail server and send the subject of the e-mail as sms using Asterisk. e-mail adresses will be like <phonenumber>@meetmecall.nl and the sms will be send to the <phonenumber> part. Does anyone of you know about a allready excisting routine for this. |
| 18:19.13 | tmccrary | ah, duh, this looks perfect. Thanks D-Fender! :) |
| 18:27.01 | BrianR___ | it turns out a ton of voip service providers have reinvite turned off... |
| 18:27.19 | BrianR___ | I guess they're tired of lusers behind nat and so on not remembering to disable it themselves. |
| 18:28.57 | wunderkin | polycom phones don't like it when you have multiple accounts with the same username on different servers.. |
| 18:29.03 | tmccrary | great, the DMTF D() worked, although I had to use the w wait in THERE to make it work (stupid Avaya PBX takes for ever to switch) |
| 18:29.26 | neuwald | lol |
| 18:29.28 | neuwald | google.com is out |
| 18:29.31 | [TK]D-Fender | tmccrary: Avaya Con Dios ;) |
| 18:30.26 | wunderkin | i tried to call the account on the 2nd reg, it placed the call on the 1st reg, it also did not catch the remote hangup.. :P |
| 18:31.34 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 18:34.15 | ThOr101 | Is asterisk a reasonable choice if I want to take an existing POTS line, and an existing Linux FedCore7 box and have it just do voicemail (mailboxes, etc)? |
| 18:36.26 | ThOr101 | To connect the POTS line to the Linux box, will any reasonably current winmodem have the ability to work in this capacity? |
| 18:39.34 | gmfm | Th0r101: you want this modem for data or for Asterisk? |
| 18:39.54 | Hmmhesays | so I have this video ipod, what do I use to convert video for it? |
| 18:41.11 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:46.48 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 18:47.29 | *** join/#asterisk ohadzz (n=ohad@19-231-13-72.static.cosmoweb.net) |
| 18:47.43 | *** join/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 18:47.49 | TUplink | what is tau32pci for.... |
| 18:47.53 | TUplink | its not compiling |
| 18:48.06 | TUplink | tau32pci.c: In function `ec_update': |
| 18:48.27 | TUplink | tau32pci.c:458: error: structure has no member named `tau32_mtx' |
| 18:49.11 | Zipper_32 | Does anybody have experience with Polycom IP Phones? I'm trying to figure out what the powering options for them are. I know that some of the IP501's come with a power adapter, but can they also be used only with a 802.3af PoE Switch connection? |
| 18:49.14 | ThOr101 | gmfm: I was told a winmodem (FXS) would connect my POTS line to the Linux box so I could use * to act as my voicemail system |
| 18:51.34 | gmfm | Th0r101: you'll most likely need a card specifically suited to the purpose... you can use the obsolete but working X100P cards (found on ebay for cheap), or for business purposes I would recommend a TDM400P with as many FXO modules as you have phone lines |
| 18:52.25 | blitzrage | ThOr101: you can't just use any winmodem -- only certain ones -- and they are crap |
| 18:54.05 | TUplink | do i even need tau32pci |
| 18:54.34 | wunderkin | Zipper_32, yes but for the 501 you need the polycom poe cable |
| 18:54.53 | mishehu | the polly com edgar allen poe adapter |
| 18:54.59 | mishehu | and thus spake the raven... nevermore |
| 18:55.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:57.15 | Zipper_32 | So the Polycom 501's don't natively accept the standardized 802.3af? |
| 18:57.44 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:57.45 | Zipper_32 | That's very annoying., |
| 18:58.08 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 18:58.10 | Assid | hey |
| 18:58.12 | [TK]D-Fender | Zipper_32: Well known facts.... |
| 18:58.17 | Assid | hi [TK]D-Fender |
| 18:58.21 | defsdoor | I getting the following - channel.c: Didn't get a frame from channel: SIP/192.168.254.2-0981fb30 on calls coming in via SPA3102 - and the call gets terminated - any ideas ? |
| 18:58.25 | [TK]D-Fender | Zipper_32: 501 is the best choice when yuo DON'T use PoE. |
| 18:58.52 | Zipper_32 | And what kind of suggestions might one make for using PoE? |
| 18:58.56 | Assid | anyone here using voicepulse and having issues with incoming? |
| 18:59.23 | Assid | people are calling in and getting "subscriber you are trying to reach is unavailable" |
| 19:00.26 | ThOr101 | gmfm: Ok, so I should just make my life easier and get one of the X100P cards. That card will accept a POTS line, and send me on my way towards my voicemail goal. True? |
| 19:00.37 | Zipper_32 | I'm looking to purchase ~15 phones. If 501's are not good with PoE, which Polycom's are? |
| 19:00.43 | [TK]D-Fender | Zipper_32: Depends on your needs |
| 19:00.54 | gmfm | Th0r101: correct |
| 19:01.09 | [TK]D-Fender | Zipper_32: For basic users IP 320 is the bang/buck king. @ $95 USD |
| 19:01.14 | Zipper_32 | ThOr101: I picked up mine on ebay for about $18 including shipping to Canada. |
| 19:01.34 | Zipper_32 | [TK]D-Fender: With 802.3af PoE support? |
| 19:01.37 | irule | How can I play a nice message when ZAP is not available? "Executing [931451308@international:1] Dial("SIP/sip501-081e66b8", "Zap/1/ 31451308") in new stack" |
| 19:01.38 | irule | [May 14 13:46:52] WARNING[12150]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/sip501-081e66b8' status is 'CHANUNAVAIL' -- Hungup 'Zap/1-1'" |
| 19:01.52 | [TK]D-Fender | Zipper_32: Clearly. |
| 19:01.53 | ThOr101 | gmfm, thank you very much. The person who told me I needed an FXS. Were they talking about having an analog phone line that worked over IP with my system? |
| 19:02.02 | ThOr101 | Zipper_32 : thanks for the price point |
| 19:02.21 | [TK]D-Fender | irule: put a playback on priority 2 |
| 19:02.39 | gmfm | Th0r101: FXS provides a station port. this is to hook up an analog phone as an extension to your Asterisk box |
| 19:02.45 | [TK]D-Fender | ThOr101: your terminology is all mixed up |
| 19:02.49 | [TK]D-Fender | ~fxofxs |
| 19:03.04 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 19:03.04 | [TK]D-Fender | ~fxsfxo |
| 19:03.08 | jbot | well, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 19:03.08 | *** join/#asterisk Fieldy (i=9Pwd78jG@gentoo/contributor/Fieldy) |
| 19:03.19 | irule | [TK]D-Fender this is within [international] right? |
| 19:03.25 | irule | Executing [931451308@international |
| 19:03.32 | ThOr101 | [TK]D-Fender can you suggest somewhere I can go to get a clue rather than filling this place up with my basic questions? |
| 19:03.53 | [TK]D-Fender | irule: You can clearly see what part of your dialplan is being executed.... you really shouldn't need us telling you that |
| 19:04.06 | ThOr101 | gmfm: Ok, that's what I thought. |
| 19:04.07 | [TK]D-Fender | ThOr101: first place to start : |
| 19:04.09 | [TK]D-Fender | ~book |
| 19:04.12 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:04.45 | Zipper_32 | ThOr101: http://www.voip-info.org/wiki-FXO | http://www.voip-info.org/wiki/view/FXS |
| 19:05.03 | Zipper_32 | And yes, the book is very helpful as well, =) |
| 19:05.38 | ThOr101 | Thank you very much. I'll return when I have better questions. |
| 19:06.32 | Zipper_32 | ThOr101: FXO and FXS information from digium is here as well: http://www.digium.com/en/products/hardware/modulefunc.php |
| 19:11.13 | TUplink | app_meetme.c: In function `conf_run': |
| 19:11.13 | TUplink | app_meetme.c:2241: warning: long int format, time_t arg (arg 10) |
| 19:11.37 | TUplink | but [LD] app_meetme.o -> app_meetme.so looks like it compiled |
| 19:11.44 | TUplink | ok |
| 19:13.31 | Assid | who are some good providers (termination) to europe ? |
| 19:15.53 | Zipper_32 | So the Polycom IP501 is the only Polycom phone which doesn't have Native IEEE 802.3af? It looks like the 320, 330, 430, 550, 601, and 650 all have PoE support integrated. |
| 19:18.40 | [TK]D-Fender | Zipper_32: 300,301,500,501 don't have native PoE and require a special cable. |
| 19:19.54 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
| 19:20.30 | unixlike | anyone have work with blf on polycom 601 ? |
| 19:20.39 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:20.55 | [TK]D-Fender | unixlike: Most of us. Its well documented on the WIKI |
| 19:21.46 | Zipper_32 | [TK]D-Fender: Thanks, I imagine those are the older models. I'm looking to setup an office with 15 new phones and I'm trying to find out what I can support with a POE switch. |
| 19:21.52 | unixlike | just wanted to know if the "docking station" with multiple lines is a sort of blf or only speed dial button |
| 19:22.34 | [TK]D-Fender | Zipper_32: if you don't need the passthrough, the IP 320 is the best bang/buck phone choice out there. |
| 19:22.42 | [TK]D-Fender | unixlike: Both |
| 19:23.09 | unixlike | oh nice, so it worth the cost ? |
| 19:23.24 | [TK]D-Fender | unixlike: you can span line-keys, and speed-dials (with or without Presence) |
| 19:23.31 | [TK]D-Fender | unixlike: What do you NEED? |
| 19:23.39 | Zipper_32 | Thanks [TK]D-Fender. |
| 19:24.19 | [TK]D-Fender | Zipper_32: http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm |
| 19:24.37 | [TK]D-Fender | Zipper_32: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-57838514688.htm |
| 19:24.44 | ThOr101 | are there X100P clones that I need to steer clear of on ebay? |
| 19:24.57 | [TK]D-Fender | ThOr101: Yes... ALL OF THEm |
| 19:25.07 | [TK]D-Fender | ThOr101: ... |
| 19:25.07 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:25.09 | [TK]D-Fender | ~ygwypf |
| 19:25.23 | jbot | extra, extra, read all about it, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 19:25.24 | ThOr101 | so they should be digium wildcard x100p |
| 19:25.26 | [TK]D-Fender | ThOr101: No, all X100p type cards should be avoided |
| 19:26.30 | ThOr101 | <gmfm> Th0r101: you'll most likely need a card specifically suited to the purpose... you can use the obsolete but working |
| 19:26.31 | ThOr101 | <PROTECTED> |
| 19:26.53 | unixlike | [TK]D-Fender: well the front office girl needs to know other phone's activity. this is why i'm asking if the polycom 601 extension module is good for that |
| 19:27.02 | ThOr101 | And Zipper_32 said he/she picked one up on ebay. So now I shouldn't purchase this to connect 1 POTS line to my * box? |
| 19:27.22 | [TK]D-Fender | ThOr101: Correct. |
| 19:27.26 | Zipper_32 | ThOr10: Is the purpose only for voicemail (which I believe you said it was)? |
| 19:27.39 | ThOr101 | Yeah, I want to make my on IVRS |
| 19:27.45 | [TK]D-Fender | unixlike: How many phones? |
| 19:27.47 | Zipper_32 | I've used them for voicemail, which is fine since there is no problem with echo on a one-sided call. |
| 19:28.09 | unixlike | [TK]D-Fender: about 10 |
| 19:28.28 | Zipper_32 | at least, there have been no problems with mine, but when you get oth parties talking, it gets ugly. It really is a bad piece of equipment, but as TK said, you get what you pay for. |
| 19:28.32 | [TK]D-Fender | ThOr101: Digium TDM400P, Sangoma A200(d), or Linksys SPA-3102 are all better choices |
| 19:28.39 | ThOr101 | If I ever decide to go full tilt VOIP, I'll get a better suited box, and more appropriate hardware. This is just for voicemail. |
| 19:28.43 | ThOr101 | [TK]D-Fender I |
| 19:28.50 | ThOr101 | [TK]D-Fender I'll go research those as well. Thanks. |
| 19:30.54 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 19:30.54 | Zipper_32 | TDM01B is what you're looking for if you want a single FXO. It'll be around $170CAN. |
| 19:31.21 | *** join/#asterisk bts3685|work (n=brent_sa@pool-71-244-104-187.phlapa.fios.verizon.net) |
| 19:31.22 | TUplink | May 14 15:29:40 Malamute kernel: pid 77115 (asterisk), uid 0: exited on signal 11 (core dumped) |
| 19:31.40 | bts3685|work | hey, is there a way to connect a sip point to an asterisk server at a different location? |
| 19:31.53 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:32.40 | Zipper_32 | bts3685|work: Can you clarify that question? |
| 19:32.47 | Zipper_32 | "Sip Point"? |
| 19:33.07 | Zipper_32 | Sip Device? |
| 19:33.12 | TUplink | how do i find out why asterisk is dumping the core? |
| 19:33.19 | ThOr101 | Zipper_32 : I'll go research the TDM01B, but it looks like it is a one line version of the TDM400P which looks like it is the same price, with 4 lines. No? |
| 19:33.33 | bts3685|work | Zipper_32: yeah, sorry. :) okay. so if i have a phone connected at point A, can it go over the WAN and connect to the asterisk server at point B at a different WAN location? |
| 19:33.38 | Zipper_32 | ThOr101: The TDM01B is a TDM400P with 1 FXO. |
| 19:34.13 | Zipper_32 | bts3685|work: Yes, you can have the SIP device at any location in the world, and for its configuration settings, specify the IP address of the Asterisk server, which again, can be anywhere in the world. |
| 19:34.26 | ThOr101 | Zipper_32 Ahh. The difference between the red and the green modules. |
| 19:34.42 | Zipper_32 | If your SIP device doesn't have to go through NAT, or if your Asterisk server isn't behind NAT, you're good to go. |
| 19:35.16 | Zipper_32 | ThOr101: That's right. You can have 4 modules in a TDM400P. For what you're looking for, you only need 1 FXO. |
| 19:35.27 | bts3685|work | Zipper_32: if it's behind a nat at each point, i just need to forward the ports, right? |
| 19:36.08 | Zipper_32 | If it's behind a NAT at each point, there are known problems. They're all outlined in the wiki here: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 19:36.25 | SoftIce | TUplink: gdb it and see what is causes it to crash |
| 19:36.49 | TUplink | gdb? |
| 19:36.59 | bts3685|work | okie. thank you so much Zipper_32, you've saved me hours of confusion. |
| 19:37.06 | lesouvage | In Holland push to talk is introduced (late, I know) and it seems nothing else then a sip connection over gprs with a send/receive switch. I tried sip over gprs and is was working fine but with a delay of a couple of seconds. With a switch (send/receive) asterisk seems just the perfect software for a private push to talk server. Does this make sence? |
| 19:37.13 | ThOr101 | TUplink : sometimes you can even run strings against it and see if anything interesting pops up. Another option is to run it under something like strace. |
| 19:37.24 | TUplink | cdr_pgsql.so => (PostgreSQL CDR Backend) |
| 19:37.24 | TUplink | <PROTECTED> |
| 19:37.24 | TUplink | Segmentation fault (core dumped) every time |
| 19:37.31 | TUplink | how do i turn that off |
| 19:37.51 | SoftIce | well you need to find out why its dumpimng |
| 19:37.58 | SoftIce | so listen to what i said, use gdb and trace the dump |
| 19:38.21 | Zipper_32 | bts3685|work: No worries. Just remember, that if you want to have a device at a 'different location' as you mentioned, you only need your IP address configuration to establish connectivity. That's the beauty of IP telephony. |
| 19:38.41 | TUplink | i added it to modules.conf |
| 19:38.47 | TUplink | as a no load |
| 19:39.59 | TUplink | not is dumps at cdr_odbc.so |
| 19:40.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:40.13 | TUplink | but what is gdb? |
| 19:41.27 | ThOr101 | TUplink it is a debugger. It analyzes core files. |
| 19:41.41 | TUplink | ok...... so where do i start |
| 19:42.38 | Zipper_32 | [TK]D-Fender: A question of personal opinion; I've deployed the IP501's before in a small office, and I believe that the 320's would feel too much like a budget-phone. Have you any opinion on the IP430? |
| 19:42.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:42.43 | ThOr101 | 1. install gdb 2. scroogle.org gdb |
| 19:42.59 | TUplink | <PROTECTED> |
| 19:43.05 | TUplink | but i have never used it |
| 19:43.21 | TUplink | do i do somthing like asterisk | gdb? |
| 19:43.35 | SoftIce | dont be Ģ%*"(^%"Ģ lazy |
| 19:43.36 | SoftIce | read |
| 19:43.46 | SoftIce | i've allready pointed you in the right direction, if you get stuck then ask |
| 19:43.51 | SoftIce | but its all over the web how to use it |
| 19:43.54 | TUplink | ok |
| 19:44.07 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:48.10 | *** join/#asterisk Viscid2 (i=limpvisc@go.dreamhost.com) |
| 19:48.16 | Viscid2 | HI GUYS |
| 19:48.40 | Viscid2 | How are you! Are you an International Community? |
| 19:49.29 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 19:51.23 | unixlike | anyone knows if polycom ip 601's extension module is good for blf or not ? |
| 19:53.40 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-433606c128efe302) |
| 19:53.40 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:54.03 | ThOr101 | so looking at at TDM400P, probably a good idea to get 2 FXO and 2 FXS and then I can play around with it. That will allow for 2 POTS lines, 2 analog phones, and then whatever else I want to do with VOIP phones on my internal network. |
| 19:54.30 | Sweeper | oh FUCKING HELL |
| 19:54.31 | Sweeper | get thi |
| 19:54.47 | Sweeper | I've got an audiocodes 8-fxo |
| 19:55.03 | Sweeper | it _reboots_ when I make an _analog_ _call_ to it |
| 19:55.52 | TUplink | ok im going back to where i riged this....... what is tau32pci.c: for..... |
| 19:55.56 | TUplink | it wont compile |
| 19:57.51 | Zipper_32 | ThOr101: What i prefer to do is simply use the 2x FXO and 1xFXS and wire ALL of my phones to the FXS. That way, I have a system like in a regular home where when I pickup one phone, it is the same extension as all of the other phones. If you have two FXS's, you can have two analog extensions, with as many phones on each extension as you want. |
| 19:57.56 | ThOr101 | Just a suggestion... Most C Files have at least a header that gives a small hint to what they do. My guess is that it is a 32 bit PCI C file that does something with TAU, and probably not the greek symbol. |
| 19:58.08 | Zipper_32 | ThOr101: Just remember, that the FXS can be tied to as many analog phones as you want with a simple splitter. =) |
| 19:58.23 | Viscid2 | HI I KEEP PRESSING ENETER BUT IT DOES NOT RING |
| 19:58.25 | TUplink | i think its somthing with zap timing |
| 19:58.39 | ThOr101 | Zipper_32 and then you have one free space with no module then right? |
| 19:58.47 | Viscid2 | OH WAIT SOMEONE IS CALLING NOW ITS OK |
| 19:58.48 | Zipper_32 | That's correct. |
| 19:58.54 | *** part/#asterisk Viscid2 (i=limpvisc@go.dreamhost.com) |
| 19:59.24 | Zipper_32 | ThOr101: Most often, you won't need 2x FXS. |
| 19:59.44 | Zipper_32 | Unless you want two distinct ring areas of your home. |
| 19:59.47 | Zipper_32 | or office. |
| 19:59.54 | ThOr101 | Zipper_32 : yeah, now that is clear to me. I might do it though so I can have a business "circuit" and a home "circuit" |
| 20:00.01 | ThOr101 | or, exactly what you just typed :-) |
| 20:00.28 | ThOr101 | Wow, I didn't know there were places still left on IRC that were this friendly and useful. |
| 20:01.05 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:01.09 | Zipper_32 | They're around, we just sleep a lot. |
| 20:01.10 | [TK]D-Fender | Zipper_32: IP 430 is nearly pointless for purchase |
| 20:01.22 | Zipper_32 | Why is that TK? |
| 20:01.26 | [TK]D-Fender | Zipper_32: Polycom = good regardless of the model. |
| 20:01.43 | [TK]D-Fender | Zipper_32: ip320/330 cover virtaully everything that the 430 does. |
| 20:05.08 | TUplink | [May 14 16:04:33] NOTICE[78534]: codec_zap.c:365 find_transcoders: No Zaptel transcoder support! |
| 20:05.16 | ThOr101 | So the people on e-bay selling TDM01B cards with 1FXO, and the picture has 2 greem nodules and 2 red modules are displaying the wrong product (or at least the wrong configuration) |
| 20:05.26 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 20:05.41 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 20:06.05 | Zipper_32 | ThOr101: Correct |
| 20:06.13 | irule | what should I investigate to understand more clearly the fact that my #$^%$&%$&^ phone will not dial 234 inmediatelly unless I use 234#? Id like to clearly answer the question to my boss |
| 20:06.37 | BSD_tech | has anyone here writen a queue exten that ask the user to input thier account number then when a agent answers it plays the account number back |
| 20:06.39 | TUplink | dialplan on the phone? |
| 20:07.07 | Zipper_32 | irule: What type of phone? |
| 20:07.10 | slmnhq | quit |
| 20:07.33 | Zipper_32 | ThOr101: The naming convention for the TDM bundles is as follows: TDM X Y B. Where "TDM" denotes that the card is TDM, "X" denotes the number of FXS modules, "Y" denotes the number of FXO modules, and "B" indicates that that this product is a bundle. |
| 20:08.31 | ThOr101 | Ahhh. Thanks again. |
| 20:08.46 | Zipper_32 | np, =) |
| 20:09.06 | *** join/#asterisk fall0ut (i=tim@realfuckingnews.com) |
| 20:09.26 | Zipper_32 | irule: What kind of phone? |
| 20:09.36 | irule | Zipper_32 it is a plain old telephone connected to a sipura ata 2000 |
| 20:10.34 | Zipper_32 | irule: Check your ATA's dialplan. http://www.netphonedirectory.com/pap2_dialplan2.htm |
| 20:10.59 | Zipper_32 | Erm, go back to page1 of that link I sent you: http://www.netphonedirectory.com/pap2_dialplan.htm |
| 20:11.00 | Mercestes | My controller card failed and now my phones don't work. Should I file a bug? |
| 20:11.13 | Zipper_32 | It explains what the dialplan is, and how to change it to suit your needs. |
| 20:11.33 | Sweeper | Mercestes: file 10. assign them all to rick |
| 20:12.15 | Mercestes | Cool |
| 20:12.24 | TUplink | ThOr101 im not getting it |
| 20:13.33 | anonymouz666 | anyone in here already convert the date in format to MM-DD-YYYY to Unixtime using strftime? |
| 20:13.52 | ThOr101 | TUplink that's a bummer. You might want to take your questions to #UnixHelp or a linux channel, as that is the topic of your question. Debugging core files is not easy. You might want to try turning on more log files or increasing verbosity. |
| 20:14.17 | ThOr101 | Walking someone through GDB for the first time is not an exercise best done over IRC... or for free for that matter. |
| 20:14.39 | TUplink | yea i did -vvvvvvvvvvvvvvvvvvvvvvvvvv -ddddddddddddddddddddddddddddd |
| 20:15.01 | ThOr101 | You need moe more v |
| 20:15.20 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:15.36 | *** part/#asterisk bts3685|work (n=brent_sa@pool-71-244-104-187.phlapa.fios.verizon.net) |
| 20:15.43 | ThOr101 | linux works best when you intermingle the v's and the d's as well, as in -vdvdvdvdvd |
| 20:16.17 | ThOr101 | Seriously though, you need to troubleshoot this problem. If you don't have the skills to do that, then you need to find someone who can help you, most likely locally. |
| 20:16.55 | ThOr101 | Once I get to the "I'll add 50 v's to the verbosity level" I know it is time to call in a favor, or leave it for the night and mull it over a steak and a beer. |
| 20:17.04 | ThOr101 | then a scotch, some wine, another beer, etc. |
| 20:17.40 | TUplink | hehe.... |
| 20:18.04 | TUplink | what a fin bummer i removed my good version ..... to get metme to work..... and now this bull sh** |
| 20:19.00 | TUplink | zaptel wont even build all the way...... ill bet thats the prob |
| 20:19.17 | TUplink | tau32pci messes up every time |
| 20:19.24 | TUplink | and there is nothing online about it |
| 20:20.51 | *** join/#asterisk elpropagandista (n=elpropag@208-106-81-64.dsl.static.sonic.net) |
| 20:21.24 | anonymouz666 | anthm: the %s in ${STRFTIME} should return the unixtime seconds? |
| 20:22.49 | anthm | ? |
| 20:23.01 | ThOr101 | TDM22P for $110 shipped sound reasonable? |
| 20:23.07 | anthm | it works like the time command on unix |
| 20:23.22 | ThOr101 | TDM22B I guess actually. |
| 20:23.23 | Zipper_32 | ThOr101: Sounds very good. |
| 20:23.25 | anthm | format is applied to epoch and returned as a string |
| 20:23.36 | Zipper_32 | USD? |
| 20:24.06 | ThOr101 | Yep |
| 20:24.18 | Zipper_32 | Sounds good. |
| 20:24.22 | Zipper_32 | Very cheap |
| 20:25.22 | anonymouz666 | anthm: I understand I need something to return the epoch as unixtime seconds |
| 20:25.22 | lesouvage | anthm: see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 20:25.56 | lesouvage | 666: I have that somewhere in my asterisk box. wait a moment. |
| 20:26.03 | anonymouz666 | lesouvage: ok, thanks. |
| 20:26.28 | anthm | so like you said %s |
| 20:27.51 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 20:27.51 | anthm | ,,%s |
| 20:28.13 | DeeJayTwo | in [general] of sip.conf, which context is used between context=defaultcontext and domain=domain.com,defaultcontext2 ? |
| 20:28.14 | DeeJayTwo | I mean.. when is context used...and when is domain's context used? |
| 20:28.40 | DeeJayTwo | which has precedence over the other... |
| 20:28.50 | lesouvage | 666: exten => s,n,Set(TIME_START_MAIN_MENU=${STRFTIME(${EPOCH},GMT+1,%s)}) |
| 20:29.33 | anonymouz666 | ${STRFTIME(${expiration_date},GMT-5,%s)}) |
| 20:29.44 | anonymouz666 | expiration_date has 01-01-2010 |
| 20:29.53 | anonymouz666 | this strftime returns 1! |
| 20:29.57 | JunK-Y | You do not appear to have the sources for the 6.1-RELEASE kernel installed. how do ya fix that on FreeBSD? |
| 20:30.13 | lesouvage | 666: this line let you count the seconds h,n,Set(SECONDEN_MAIN_MENU=$[ ${TIME_STOP_MAIN_MENU} - ${TIME_START_MAIN_MENU}]) |
| 20:30.18 | irule | I just noticed that my sipura actually has its own dial plan? WOW! I also noticed that I can do this <1:2>, so I press on 1 and 2 will be dialed? WOW! so, does this apply equally into the *'s dialplan in extensions.conf? thanks |
| 20:31.42 | anonymouz666 | lesouvage: do I need that? |
| 20:31.51 | *** join/#asterisk mick_home (n=chatzill@clamwin/admin/mickhome) |
| 20:31.59 | anonymouz666 | %s does not convert the 01-01-2010 to unixtime ? |
| 20:33.01 | lesouvage | 666: I don't know, most of the times one needs the time in seconds since a certain moment (what moment?) to calculate the difference between two timestamps. It's just an example. |
| 20:33.36 | anonymouz666 | I need less than that |
| 20:33.42 | anonymouz666 | Just to convert some date to unixformat |
| 20:33.47 | MrTelephone | who here runs an adit 600? |
| 20:34.30 | lesouvage | 666: just check the webpage on the wiki. I think the answer is there. |
| 20:36.53 | _VoiceMeUp_COM | irule ? lol |
| 20:37.08 | _VoiceMeUp_COM | exten => 1,dial(sip/2@peer) |
| 20:37.11 | anthm | oh you want to turn an arbitrary date string into epoch sec ? |
| 20:37.28 | anthm | like 2004-01-01 12:00 |
| 20:38.23 | anonymouz666 | Set("SIP/6001-0a1d7478", "getserviceexp=01-01-2015") Set("SIP/6001-0a1d7478", "calltime=1179175019") |
| 20:38.25 | anonymouz666 | working |
| 20:38.33 | anonymouz666 | anthm: little typo here... |
| 20:38.41 | anonymouz666 | I used ${} inside strftime |
| 20:39.33 | anonymouz666 | oops I think the result is wrong |
| 20:39.34 | anonymouz666 | lol |
| 20:39.35 | anonymouz666 | haha |
| 20:39.55 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:42.39 | irule | _VoiceMeUp_COM? |
| 20:43.57 | anonymouz666 | its saying the current time |
| 20:44.01 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 20:44.04 | anonymouz666 | not the 01-01-215 |
| 20:44.18 | anthm | yah that is all it can do |
| 20:44.35 | anthm | you would need a new function to do what you want if you want to turn any string into epoch |
| 20:45.11 | anonymouz666 | what new function? |
| 20:47.02 | *** join/#asterisk bkunyiha (n=bgitonga@66-113-78-53.rev.ibsinc.com) |
| 20:49.24 | anthm | one you must code |
| 20:50.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net) |
| 20:50.23 | anonymouz666 | I can do that easily through AGI |
| 20:50.29 | anonymouz666 | using PHP functions |
| 20:54.18 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 20:54.44 | mocker | So, I can make calls w/ a DUNDi lookup, but I can't get the 'dundi lookup' command to work. ;) |
| 20:54.56 | anthm | yes that is your best bet then |
| 20:58.19 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 20:59.28 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 20:59.47 | irule | how can I require password before dialing a number?? |
| 21:00.04 | nowork | hi, where can I get cheap&good DID number, any suggestions?thanks? |
| 21:01.07 | karlhaines | nowork: less.net |
| 21:01.08 | karlhaines | any gentoo users in here? |
| 21:01.08 | mocker | nowork: I've had good luck w/ vitelity, although their customer support is kinda lacking. |
| 21:01.19 | karlhaines | same with less.net |
| 21:01.34 | nowork | Karlhaines, Mocker: thank you.. |
| 21:01.57 | *** part/#asterisk TUplink (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 21:02.12 | *** join/#asterisk MoFeedle (n=feedle@host-69-145-69-143.bln-mt.client.bresnan.net) |
| 21:02.34 | mocker | karlhaines: Same with every ITSP it seems. :( |
| 21:02.41 | *** part/#asterisk elpropagandista (n=elpropag@208-106-81-64.dsl.static.sonic.net) |
| 21:02.46 | karlhaines | lol |
| 21:03.24 | nowork | how about we apply as ITSP? |
| 21:03.35 | karlhaines | i'm starting a website, asterisk-gentoo.org, devoted to, obviously, using asterisk on gentoo, if anyone is interested in helping, please let me know! |
| 21:03.41 | johann8384 | I personally think NetLogic is a good place to get a quality/cheap DID (I work there) |
| 21:03.55 | karlhaines | johann8384: you're biased then :) |
| 21:03.56 | karlhaines | lol |
| 21:04.01 | nowork | :) |
| 21:04.16 | MoFeedle | Anybody ever had any success with 2BCT on PRIs on Asterisk? I can't get it to work for nothin'. |
| 21:04.27 | johann8384 | yep, but when I worked for BluegrassNet in Louisville, Ky, I thought NetLogic was a good place to get a cheap DID as a customer of NetLogic's before I worked here |
| 21:04.52 | karlhaines | johann8384: where do you live? |
| 21:05.30 | johann8384 | St. Louis, MO |
| 21:05.40 | johann8384 | moved here 3 weeks ago |
| 21:05.55 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 21:07.19 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 21:08.09 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 21:08.27 | nowork | Karlhaines: should be les.net not less.net:) |
| 21:08.35 | mocker | So can someone help me digest DUNDi? If I set it up using the http://www.voip-info.org/wiki-DUNDi+Enterprise+Configuration+IAX guide does that mean it's using IAX trunking to complete the calls? |
| 21:08.50 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 21:09.26 | johann8384 | mocker: yeah it does |
| 21:09.49 | mocker | johann8384: Ok. |
| 21:10.03 | mocker | It's just weird because my normal IAX trunks don't look like that at all in iax.conf. |
| 21:10.09 | mocker | I guess that's the point though? |
| 21:11.19 | johann8384 | mocker: basically. It allows you accept calls via your/the DUNDi network |
| 21:12.39 | mocker | So in theory if you were setting up say 10 remote offices, you'd use this instead of just plain trunks? |
| 21:12.45 | karlhaines | nowork: oh yeah, lol, sorry |
| 21:12.48 | mocker | Or am I off base? |
| 21:13.01 | johann8384 | mocker: yeah, that's the theory |
| 21:13.19 | mocker | Ok, making sure I'm barking up the right tree. :) |
| 21:13.58 | johann8384 | mocker: I personally have been using ENUM and my own DNS server to route calls instead of a private DUNDi network |
| 21:14.38 | mocker | johann8384: Any reason? |
| 21:14.39 | johann8384 | there are benefits/problems etc. with each method. DUNDi is distributed which is nice |
| 21:16.15 | mocker | johann8384: Either way, it means I have to rewrite my dialplan! |
| 21:16.18 | mocker | *sigh* |
| 21:16.20 | mocker | :) |
| 21:17.02 | *** join/#asterisk Hymie (n=Hymie@l8r.net) |
| 21:18.28 | Hymie | any polycom users have hint/presence working so the registered sip connection can be used for such? |
| 21:19.48 | mocker | Hymie: Yup. |
| 21:20.57 | karlhaines | johann8384: i'm in nashville, tn |
| 21:21.05 | karlhaines | johann8384: not too far from you! |
| 21:21.19 | Hymie | mocker: what did you do? I easily have my 501 working so that I have hints showing various extensions, but what did you do to make a hint show at the primary sip line? |
| 21:22.06 | mocker | Hymie: Nothing really special. :( |
| 21:22.11 | johann8384 | karlhaines: cool. I went down to a 2600 meeting there in January |
| 21:22.13 | mocker | Never used a 501, just a 601. |
| 21:22.36 | irule | can I just compile 1.4 over an installed 1.0 *? or should I uninstall first? |
| 21:22.42 | karlhaines | johann8384: cool.. i dont get along with the other nashville geeks, so i dont ususally go to things like that |
| 21:22.52 | karlhaines | johann8384: lol, the LUG hates me |
| 21:23.10 | Hymie | mocker: could you give me an idea? how did you get a buddy to show as the first sip regisitration? whenever I add buddies, they all are appended after... |
| 21:23.47 | mocker | Hymie: Are you talking about in the directory on your Polycom? |
| 21:24.06 | Hymie | mocker: no, I'm talking about buddies that show on the line buttons... |
| 21:24.13 | mocker | Right.. |
| 21:24.17 | Hymie | mocker: my first line button is my sip registration |
| 21:24.26 | Hymie | mocker: I'd prefer if that first button showed a hint |
| 21:24.30 | mocker | Ahhhh. |
| 21:24.38 | Hymie | mocker: that way, I could tie that to my outside line |
| 21:24.50 | Hymie | mocker: and show a hint as to when the outside line group is busy.... |
| 21:24.53 | mocker | Sorry, that's the way I have it. |
| 21:25.01 | Hymie | doh |
| 21:25.07 | mocker | I thought you didn't have presense working at all. ;) |
| 21:25.25 | mocker | I have my line appearance and then 4 buddies beneat that. |
| 21:25.28 | Hymie | ah |
| 21:26.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:26.14 | Hymie | I guess I could label the first "Intarnal" and then add two lines after that.. hmm |
| 21:26.25 | Hymie | it's too bad it isn't a bit more flexible |
| 21:26.34 | coolbeans | Does line appearence work in 1.2 or just 1.4? |
| 21:26.44 | mocker | coolbeans: I'm on 1.2 |
| 21:26.46 | Hymie | 1.2 is fine |
| 21:26.56 | mocker | Not using 1.4 for awhile because I'm paranoid. ;) |
| 21:27.21 | Hymie | mocker: don't blame you.. I upgraded, fonud and reported 2 bugs, then had to downgrade because all of the others |
| 21:27.24 | Hymie | but, 1.4 is young |
| 21:27.56 | mocker | Hymie: That's why I'm avoiding it.. |
| 21:28.04 | mocker | I'll let others gnash their teeth. :) |
| 21:28.23 | Hymie | it's hard not to, with telephony |
| 21:28.51 | mocker | People seem to get angry when their phones don't work. |
| 21:28.58 | mocker | It's not like when a website goes down.. |
| 21:29.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:29.28 | coolbeans | Thanks. |
| 21:30.52 | *** join/#asterisk zm23 (n=chatzill@zombie.cc.columbia.edu) |
| 21:31.56 | zm23 | hello all. need help with transfering calls to asterisk. Asterisk is starting a new call instead of completing transfer. |
| 21:33.16 | ManxPower | zm23: press the Trnsfr button a 2nd time or press the Blnd button after you press the Trnsfr button. This is documented in the manual for your Polycom phone. |
| 21:33.53 | wunderkin | ManxPower are you psycho? |
| 21:34.08 | ManxPower | wunderkin: it would depend on who you ask. |
| 21:34.21 | mcab | wunderkin: psychic? |
| 21:34.30 | wunderkin | :) he never mentioned polycom, maybe he was here earlier |
| 21:34.41 | ManxPower | wunderkin: since he is an idiot and didn't bother to say what phone he was using or what method of transfer he is using, I had to assume. |
| 21:34.52 | ManxPower | or what version of Asterisk for that matter. |
| 21:34.55 | zm23 | ManxPower my polycom phone is registered with another sip proxy and call is directed to asterisk. When i press transfer again, the call is not disconected, and the transferred party gets a new session with asterisk. |
| 21:35.19 | ManxPower | zm23: are you trying to do supervised or blind transfers |
| 21:35.44 | mcab | ManxPower: not one to mince words, are you? :-) |
| 21:35.45 | *** join/#asterisk srd2 (i=srd@wombat.frictious.net) |
| 21:35.49 | zm23 | manxpower, this idiot is using polycom, |
| 21:36.12 | wunderkin | i think i had that problem before too, i have not completed my new openser config yet |
| 21:36.20 | zm23 | manxpower using asterisk 1.2 but have tried with 1.4 also.. this happens with both type of call transfer |
| 21:37.37 | zm23 | wunderkin so in your case, was teh problem on openser side or asterisk? |
| 21:38.11 | wunderkin | i assume because of the proxy |
| 21:38.35 | ManxPower | I agree |
| 21:39.36 | zm23 | wunderkin i see that call comes to asterisk with REFER header... on polycom to polycom transfer, 3rd phone will accept invite from phone 1 replace/end session with phone 2 (the phone that transfered the call) |
| 21:40.24 | zm23 | wunderkin however asterisk does not do a replace.. instead starts a new session on that invite with REFER .. and never disconnects teh phone that started teh transfer |
| 21:42.03 | wunderkin | hopefully we can figure that out |
| 21:43.21 | zm23 | wunderkin you'll be my hero for doing that :) |
| 21:44.48 | zm23 | have i overlooked any asterisk config setting to enable this? |
| 21:49.13 | zm23 | wunderkin, manxpower? helppppppp pppleasseeeee |
| 21:49.13 | srd2 | how would I set a global variable in a context without using exten => ? |
| 21:49.13 | ManxPower | zm23: I cannot help you with proxies |
| 21:49.14 | ManxPower | I have never in my life ever need a SIP/RTP proxy. |
| 21:49.14 | ManxPower | srd2: you can 't. They are GLOBAL |
| 21:49.40 | srd2 | what is global? |
| 21:50.16 | zm23 | manxpower ok thanks |
| 21:51.54 | ManxPower | global is anywhere |
| 21:52.10 | srd2 | what I'm wanting to do is, each sip user has it's own context, within that context it declares what the users std/area code is, voicemail userid, outgoing callerid etc and then go onto other contexts using include => |
| 21:52.53 | srd2 | so if I type 1571, asterisk will look in the context, say this users mailbox is 123 and then pass it on to the voicemail context |
| 21:53.59 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:54.16 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:55.07 | *** join/#asterisk dunder (n=dolphin@24-119-81-201.cpe.cableone.net) |
| 21:57.22 | irule | is it possible to Authenticate() a [context] and include the trunks in there? thanks |
| 21:58.36 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:58.41 | *** join/#asterisk corramor (i=corramor@nat/digium/x-c742400a586f2cbe) |
| 21:58.42 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 22:00.19 | ManxPower | irule: no. |
| 22:00.31 | ManxPower | Authenticate auths calls, not contexts or extensions |
| 22:00.34 | *** join/#asterisk bbryant (i=brett@nat/digium/x-9220429943bc7ae3) |
| 22:03.19 | irule | so it must be exten = s,n,Authenticate(1) and then exten = s,1,Dial...? If I use this to authenticate 2 users, it will be a huge dialplan, are there aoptions? thanks |
| 22:03.53 | irule | <PROTECTED> |
| 22:03.53 | *** join/#asterisk CVirus (n=GoD@196.205.192.96) |
| 22:04.06 | CVirus | How can I know if asterisk is compiled with pri support or not ? |
| 22:04.33 | *** part/#asterisk corramor (i=corramor@nat/digium/x-c742400a586f2cbe) |
| 22:05.03 | *** join/#asterisk ZedStarr (n=chatzill@port-83-236-134-210.static.qsc.de) |
| 22:05.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:07.53 | wunderkin | CVirus, see if you have the pri command on the asterisk cli |
| 22:08.07 | wunderkin | irule, you need a 1 priority first, then you can use n on each additional one after |
| 22:09.25 | CVirus | wunderkin: I type p then tab and it completes the command "pri" but when I press enter, it says No such command 'pri' (type 'help' for help) |
| 22:09.37 | CVirus | it autocompletes it |
| 22:09.58 | irule | wunderkin my concern is, how may I password-protect making calls for many users? I can easilly add Authenticate to each trunk but, multiplied by n users it will become a nightmare in the future |
| 22:09.59 | wunderkin | press space then tab, so yes |
| 22:10.24 | CVirus | excuse me ? |
| 22:10.48 | wunderkin | pri space tab that will give you the rest of the commands |
| 22:11.07 | wunderkin | irule, where are you planning on storing this info |
| 22:11.08 | CVirus | wunderkin: got that .. thanks |
| 22:11.20 | *** join/#asterisk limbeaux (n=IceChat7@69.17.126.244) |
| 22:11.22 | wunderkin | is each user going to have a pass or a single pass for each provider, what are you doing |
| 22:11.49 | irule | wunderkin in the dialplan? :s |
| 22:14.19 | limbeaux | newbie here, can someone point me to an easy to follow install and configure guide. I would prefer to use ubuntu or debian as i am most familiar with the way they work. |
| 22:14.57 | limbeaux | if it is easier to install and configure on another flavor, i can take the time to learn another linux flavor |
| 22:15.45 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:17.43 | DJ-Pyro_ | I have a calling app that works great 98% of the time, but sometimes it fails multiple times with OutgoingSpoolFailed, if I sip debug I see asterisk sending a CANCEL invite to my gateway and then 'Call failed to go through, reason 0' |
| 22:17.49 | DJ-Pyro_ | anyone see that type of behavior before? |
| 22:18.16 | jm|laptop | limbeaux: use newly released Debian etch (stable) then install asterisk (1.4) from apt |
| 22:18.31 | jm|laptop | then check voip-info.org for config help |
| 22:19.21 | wunderkin | irule, what are you trying to accomplish?? |
| 22:20.10 | irule | <PROTECTED> |
| 22:20.36 | *** part/#asterisk DJ-Pyro_ (n=abrezins@gateway.brevient.net) |
| 22:20.42 | *** join/#asterisk DJ-Pyro_ (n=abrezins@gateway.brevient.net) |
| 22:21.04 | limbeaux | thanks jm|laptop !! |
| 22:24.25 | *** join/#asterisk Stridernzl (n=neville@125-239-161-233.jetstream.xtra.co.nz) |
| 22:27.29 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 22:29.57 | bkunyiha | How can i prevent asterisk from SIP Seeding peer from astdb in sip.conf |
| 22:30.29 | bkunyiha | I want to use real time instead |
| 22:35.13 | mutilator | anyone looking to buy a te110p? i have one for $400 |
| 22:35.45 | Cresl1n | I'll give you $20 |
| 22:35.49 | Cresl1n | and that's my final offer |
| 22:35.55 | Qwell[] | I'll give you $18 |
| 22:36.17 | Cresl1n | crap |
| 22:36.25 | mutilator | winnar! |
| 22:36.31 | Qwell[] | w00t |
| 22:36.31 | Cresl1n | I'll give you $17.50 |
| 22:36.34 | Qwell[] | too late! |
| 22:36.36 | file | $16.99! |
| 22:36.40 | mutilator | heh |
| 22:37.17 | Cresl1n | I'll give you 16.98 plus free email support for 3.14159 days |
| 22:37.25 | Qwell[] | mmm....pie |
| 22:37.33 | file | Cresl1n: I... will.... soon see you! |
| 22:37.37 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 22:37.42 | file | be afraid. |
| 22:37.47 | Cresl1n | file: was that a warning? |
| 22:37.49 | Cresl1n | :-P |
| 22:37.51 | file | yes |
| 22:37.51 | jm|laptop | I have a TMD100P going |
| 22:37.59 | Qwell[] | jm|laptop: a what? |
| 22:38.01 | JT | what's that? |
| 22:38.10 | Cresl1n | what's a TDM100P? |
| 22:38.15 | JT | who knows |
| 22:38.21 | jm|laptop | I'm much happier with my 3102 |
| 22:38.33 | Cresl1n | I like my 2783 |
| 22:38.42 | JT | what did you mean to say instead of TDM100P? |
| 22:39.20 | jm|laptop | X100P ? |
| 22:39.36 | jm|laptop | X101P? I don't now anymore. |
| 22:39.40 | jm|laptop | know* |
| 22:39.54 | sandorp | I just installed a zaptel card and I see a ZAP/g0 trunk has been defined; I have 2 outbound routes defined, each with one pattern -- 9|. and . ; however, I can't seem to be able to dial out ... internal extensions that don't use the card work fine; what am I missing? |
| 22:40.00 | JT | right, one of those |
| 22:40.22 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 22:41.03 | jm|laptop | the only 'problem' with my 3102 is that it does Answer() before it tries its trunk call (well dur) but I means I lose my Dial(,,,m) |
| 22:41.39 | jm|laptop | and free calls (x)800 or enum numbers play the soupdragons |
| 22:42.19 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 22:43.22 | luckyone | when I start asterisk with the script in /etc/init.d/asterisk, how do I attach to it and access the command line? |
| 22:43.22 | sandorp | I get an "all circuits are busy" recording which sounds like the voice for the other built in recordings |
| 22:43.25 | Qwell[] | luckyone: asterisk -r |
| 22:43.25 | *** join/#asterisk Strom_M (n=strom@12.175.45.205) |
| 22:43.49 | luckyone | Qwell[]: thanks for answering my noob ?! |
| 22:44.32 | Strom_M | catsex |
| 22:45.07 | luckyone | Qwell[]: can you specify username and password? |
| 22:45.11 | jm|laptop | luckyone: asterisk -vr ? |
| 22:45.23 | jm|laptop | what Qwell[] said |
| 22:45.57 | luckyone | jm|laptop: I was not being smart when saying thanks, I am so scared to ask such dumb questions in this channel |
| 22:46.17 | jm|laptop | bless. |
| 22:46.39 | luckyone | Qwell[]: I just discovered asterisk -help, I am golden now! Thanks |
| 22:47.01 | sandorp | this noob could use some hand holding right now too; I'm not too scared to look dumb because I usually get good help in return :) |
| 22:47.24 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-433606c128efe302) |
| 22:50.11 | *** join/#asterisk Curus (n=Curus@10.8.185.213.dk-amb.res.sta.perspektivbredband.net) |
| 22:53.35 | *** join/#asterisk XVampireX (n=serge@unaffiliated/xvampirex) |
| 22:53.41 | XVampireX | How's everyone doing? |
| 22:53.57 | MrTelephone | shitty |
| 22:54.03 | MrTelephone | I can't get my adit 600 to work |
| 22:54.10 | XVampireX | Oh |
| 22:54.23 | XVampireX | Well I actually joined here if anyone knows of any good SIP services like Tellme |
| 22:54.31 | XVampireX | maybe even based on Asterisk |
| 22:55.45 | JT | what sort of service do you actually need? |
| 22:56.36 | XVampireX | Interactive one |
| 22:56.40 | XVampireX | that's fun to dial to :) |
| 22:56.48 | XVampireX | I keep checking my horoscope with Tellme |
| 22:56.53 | JT | what |
| 22:56.57 | JT | err okay |
| 22:57.04 | JT | so time wasters, right |
| 22:57.09 | XVampireX | I dial, I say "horoscopes" and then say my sign "leo" |
| 22:57.13 | XVampireX | yeah |
| 22:57.20 | JT | horoscopes are stupid |
| 22:57.24 | JT | i hate them so much |
| 22:57.26 | XVampireX | Heh |
| 22:57.32 | XVampireX | Well they got ringtones there |
| 22:57.35 | XVampireX | and a blackjack game |
| 22:57.37 | XVampireX | and so on |
| 22:57.48 | [hC] | voice recog blackjack? how annoying |
| 22:58.00 | XVampireX | heh |
| 22:58.01 | JT | what use is ringtones on sip? it's not like it can easily send them to your sip device |
| 22:58.11 | XVampireX | no idea |
| 22:58.20 | XVampireX | Actually, what's a great hardware sip phone? |
| 22:58.24 | Strom_M | polycom |
| 22:58.25 | XVampireX | that doesn't cost too much |
| 22:58.28 | Strom_M | ip430 |
| 22:58.39 | XVampireX | I kinda started liking SIP now that I fixed my net |
| 22:58.44 | XVampireX | works wonders |
| 22:59.09 | XVampireX | mobile too? |
| 22:59.43 | JT | mobile...? |
| 23:00.04 | XVampireX | Yeah mobile |
| 23:00.14 | XVampireX | I heard there's mobile sip phones using wireless |
| 23:00.19 | JT | what has mobile phones got to do with sip? |
| 23:00.22 | JT | oh, cordless |
| 23:00.34 | JT | they're rubbish, the ones that use wifi |
| 23:01.20 | XVampireX | Why? |
| 23:04.43 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:04.52 | JT | because the audio sucks, and they have no battery life |
| 23:05.14 | JT | moving devices using wifi is very bad for voip |
| 23:11.43 | XVampireX | Is there ever gonna be an easy way to call for free other than WIRED or Softphone ways? |
| 23:12.07 | JT | you can still use a conventional cordless phone and an ATA |
| 23:12.19 | JT | or even get a bluetooth headset for your SIP phone |
| 23:13.40 | zm23 | has anyone used asterisk with imap ? just want to know how well is it working out |
| 23:14.28 | XVampireX | ATA? |
| 23:14.44 | Zipper_32 | XVampireX: Call for free? To the PSTN? You'll always need somebody to connect to the PSTN if that's your desired destination. |
| 23:15.01 | JT | ~ata |
| 23:15.23 | jbot | ata is, like, Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 23:15.23 | Zipper_32 | XVampireX: Analog Cordless Phone, and an Analog Telephone Adapter, |
| 23:15.23 | Zipper_32 | ATA = Analog Telephone Adapter |
| 23:15.23 | JT | not quite |
| 23:15.39 | XVampireX | Ah, okey |
| 23:15.44 | Zipper_32 | Not quite? |
| 23:15.48 | XVampireX | Zipper_32: No, just SIP or some new protocol.... |
| 23:16.12 | XVampireX | jbot: You're slow. |
| 23:16.16 | jbot | i'll take that as a complement |
| 23:16.16 | JT | Analogue Terminal Adapter |
| 23:16.41 | XVampireX | ;) |
| 23:17.00 | XVampireX | voip-info seems a rather large wiki on voip... |
| 23:17.18 | JT | my goodness, who'd have thought ;) |
| 23:17.52 | XVampireX | heh |
| 23:18.33 | *** join/#asterisk vonkleist (n=gera@189.155.128.168) |
| 23:18.45 | vonkleist | Hi |
| 23:19.16 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 23:19.23 | XVampireX | anyway what protocol do you people use most with asterisk? |
| 23:19.36 | luckyone | XVampireX: my provider recommends sip |
| 23:20.03 | XVampireX | Yeah seems like that's the best one and most standard open protocol |
| 23:20.18 | XVampireX | But even then, can't quite connect to every single other SIP user. |
| 23:20.45 | luckyone | I am trying to get my new phone number to connect to my asterisk box |
| 23:20.53 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:20.59 | luckyone | I am not sure which side is not configured correctly |
| 23:21.05 | XVampireX | I.E: from gizmo account dialing to sip:333@voip.wengo.fr doesn't work, and from wengo dialing sip:411@gizmoproject.com doesn't work (Static) |
| 23:21.17 | JT | jbot: no, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 23:21.25 | jbot | JT: okay |
| 23:21.45 | XVampireX | jbot is learning :D |
| 23:23.11 | luckyone | ok, so in my sip.conf, I have defined an [provider-inbound] context |
| 23:23.35 | luckyone | it is set to point to the [inbound] context in my extensions.conf file |
| 23:25.17 | luckyone | [inbound] has an extension that looks like exten => <my number>,1,Answer |
| 23:27.35 | *** part/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 23:27.35 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:27.35 | Mercestes | XVampireX, the voice from tellme is hot |
| 23:28.02 | rene- | heh |
| 23:28.08 | XVampireX | Mercestes: Maybe :P |
| 23:28.17 | XVampireX | well for you it is, heh |
| 23:28.20 | blitzrage | the voices in my head are hawt |
| 23:28.39 | rene- | if i had a sip provider that says his calls can come from anywhere on a /24 network should i require him to register? |
| 23:28.55 | blitzrage | sure |
| 23:29.08 | blitzrage | since you won't know where to send the call back to |
| 23:29.12 | XVampireX | Oh, is there anyway to change the sound people dial to me? |
| 23:29.16 | XVampireX | like have a ringtone thing? |
| 23:29.20 | blitzrage | if you're not sending calls to them, then registration is unnecessary |
| 23:29.25 | XVampireX | with SIP |
| 23:29.31 | JT | thankfully not, usually |
| 23:29.32 | rene- | i am just terminating their calls |
| 23:29.42 | rene- | thankx blitzrage |
| 23:29.53 | JT | music instead of ringing indication is super annoying |
| 23:29.55 | Mercestes | XVampireX, change the sound people dial to you? Umm....I guess you could change the locale on the phone and hope it reprograms DTMF for you |
| 23:29.58 | blitzrage | rene-: then just use deny=0.0.0.0/0.0.0.0 ; permit=192.168.1.0/255.255.255.0 |
| 23:30.07 | XVampireX | ah |
| 23:30.08 | blitzrage | (obviously replace 192.168.1.0 with their network) |
| 23:30.34 | rene- | blitzrage: but still host=dynamic right? to force registration ? |
| 23:30.35 | blitzrage | XVampireX: do you mean the ringing you hear on incoming calls, or the ringing sounds the caller hears when waiting for you to answer? |
| 23:30.49 | blitzrage | rene-: yes, but why force registration if you're not sending them calls? |
| 23:30.55 | XVampireX | blitzrage: Both, actually :D |
| 23:31.00 | blitzrage | if they are just placing calls, registration is unnecessary |
| 23:31.23 | rene- | blitzrage so i can get away without putting a host= option in the sip stanza? |
| 23:31.57 | blitzrage | XVampireX: see SIP_HEADER(ALERT_INFO,...) function |
| 23:32.17 | blitzrage | rene-: don't even include the host=, you're not sending them calls |
| 23:32.23 | XVampireX | blitzrage: Thank you :) |
| 23:32.37 | JT | blitzrage: i think XVampireX wants to do silly things like "caller tones" too |
| 23:32.41 | blitzrage | XVampireX: for changing ringing... see the Dial(....,,M()) option |
| 23:33.25 | rene- | cool blitzrage |
| 23:33.27 | *** part/#asterisk rene- (n=rene@200.34.66.137) |
| 23:33.29 | XVampireX | wow, now if I setup asterisk I can have a pretty damn nice system for VoIP with SIP :D |
| 23:33.36 | blitzrage | yes, yes you call |
| 23:33.44 | blitzrage | s/call/can/ |
| 23:34.01 | blitzrage | man I'm bored.... I hate that |
| 23:34.07 | XVampireX | jbot is a cute bot :D |
| 23:34.10 | blitzrage | I should try and get my DSL connection up and running on this Soekris box |
| 23:34.11 | Mercestes | you can fix mysystem |
| 23:34.21 | blitzrage | Mercestes: what's wrong with it? |
| 23:34.29 | blitzrage | I should really be reviewing this email from a client... |
| 23:34.31 | Mercestes | blitzrage, Controller card failure |
| 23:34.38 | Mercestes | that's replaced bu tnow I have to rebuild my PBX |
| 23:34.48 | blitzrage | hardware sucks |
| 23:34.50 | Mercestes | some dumbass (me) built it on a raid 0 instead of a raid 5. |
| 23:34.55 | blitzrage | heh |
| 23:35.56 | Mercestes | so now I get to backup all my stuff and wipe my disks and start al over and rebuild my system and load up my backups and hope like hell I didn't forget anything. |
| 23:36.33 | blitzrage | you will... |
| 23:36.34 | Mercestes | I think before I start I'll do a nice tar -zcvf everything.tgz / |
| 23:36.34 | XVampireX | One final question for now, relating to one of the first questions: What's a good hardware sip phone? the ones you told me about are for enterprise, I need something decent for just calling... something just basic mostly... |
| 23:36.35 | blitzrage | I always do |
| 23:36.39 | blitzrage | ya, good idea |
| 23:36.52 | luckyone | I am heading home, thanks everyone in this channel - talk to you in a few minutes! |
| 23:36.55 | blitzrage | Linksys SPA-942 |
| 23:37.00 | Mercestes | maybe a ...tar -tcvf |scp bob@server:/home/bob/everything.tgz / |
| 23:37.02 | JT | XVampireX: polycom is a good recommendation |
| 23:37.05 | Mercestes | how would I do that..? |
| 23:37.11 | JT | it doesn't matter if enterprise or not |
| 23:37.15 | JT | i have 3 polycoms at home |
| 23:37.28 | XVampireX | JT: Is that the brand? or is that the name of the device? |
| 23:37.37 | JT | it's the brand |
| 23:37.42 | JT | there's a number of models |
| 23:37.48 | blitzrage | I have a 7960, 7912, 7970, IP501, Mitel 5220, Linksys SPA-942... :) |
| 23:37.51 | XVampireX | What's a decent one? |
| 23:38.03 | blitzrage | Polycom IP430 |
| 23:38.09 | blitzrage | that's a good 2 line phone |
| 23:38.17 | JT | depends how much you want to spend, but you probably don't need anything greater than an IP430 |
| 23:38.31 | JT | and IP301 or IP320 will also do fine |
| 23:38.31 | blitzrage | useful speakerphone compared to the useless speakerphone on the Linksys SPA-942 (but Linksys has a backlight) |
| 23:38.35 | XVampireX | Can I set them up with whatever codec? |
| 23:38.38 | XVampireX | or does it use GSM? |
| 23:38.41 | JT | unless you need speakerphone, 301 doesn't have that |
| 23:38.45 | blitzrage | G.711 and G.729 |
| 23:38.49 | JT | they can do g.711 or g.729 |
| 23:38.50 | XVampireX | oh |
| 23:38.55 | XVampireX | G711U is grand |
| 23:39.03 | blitzrage | not sure how many phones actually do GSM |
| 23:39.04 | XVampireX | I haven't tried g729 series yet |
| 23:39.15 | JT | not many phones do gsm |
| 23:39.19 | XVampireX | Yeah |
| 23:39.30 | JT | gsm sounds awful anyway |
| 23:39.46 | blitzrage | its ok for what it is |
| 23:39.53 | XVampireX | Ah, I found the twinkle ringtone option ! |
| 23:39.55 | blitzrage | sounds like my cell phone :P |
| 23:39.55 | XVampireX | Cool |
| 23:40.14 | JT | blitzrage: g.729 sounds better though |
| 23:40.20 | blitzrage | well ya |
| 23:40.22 | XVampireX | Wav format :D |
| 23:40.26 | blitzrage | wav isn't a codec |
| 23:40.35 | XVampireX | blitzrage: how is g729 compared to g711? |
| 23:40.47 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 23:40.47 | blitzrage | XVampireX: very close -- but much less data bit rate |
| 23:40.48 | JT | XVampireX: g.729 is a highly compressed patent encumbered codec |
| 23:40.50 | XVampireX | blitzrage: I know it's not a codec, that's what I need the ringtones to be |
| 23:40.53 | JT | a far cry from wav |
| 23:40.56 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au) |
| 23:40.58 | XVampireX | Oh |
| 23:41.10 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 23:41.14 | XVampireX | Now lets get us a Castlevania one ;) |
| 23:41.19 | blitzrage | 64kbs vs. 8kbps |
| 23:41.27 | JT | g.711 is uncompressed, but companded |
| 23:41.34 | XVampireX | The best client for SIP seems to be Twinkle |
| 23:41.39 | XVampireX | It is SO customizable |
| 23:41.55 | XVampireX | Has anyone heard of iSAC? |
| 23:41.58 | JT | that would be why i've seen zero recommendations for it here :P |
| 23:42.03 | JT | twinkle |
| 23:42.04 | Daejeo1 | loading ztdummy: running ztcfg: notice: configuration file is /etc/zaptel.conf line:21 no such tine zone known: kr 1 error detected.by default it was US, but i changed it to kr |
| 23:42.22 | blitzrage | looks like 'kr' isn't a valid zone |
| 23:42.35 | Daejeo1 | what should I do? |
| 23:42.40 | blitzrage | change it something valid |
| 23:42.42 | Mercestes | Daejeo1, move to the US |
| 23:42.45 | Daejeo1 | I am in south korea |
| 23:43.11 | Daejeo1 | physical move? |
| 23:43.19 | Mercestes | of course! |
| 23:43.21 | Daejeo1 | physically move |
| 23:43.29 | Daejeo1 | oh |
| 23:43.47 | Daejeo1 | it is not possible at this moment |
| 23:44.25 | Daejeo1 | can I use any country code? |
| 23:44.37 | Daejeo1 | does it matter? |
| 23:45.11 | Mercestes | it's not a country code. |
| 23:46.02 | Mercestes | trying to find a time zone list for you |
| 23:46.12 | JT | time zone? |
| 23:46.17 | JT | it's a tone zone |
| 23:47.31 | Daejeo1 | then what is the valid tone zone for korea? |
| 23:47.31 | Daejeo1 | try all? |
| 23:47.33 | Zipper_32 | JT: You mentioned that you have 3 polycoms at home; which models? |
| 23:47.35 | Mercestes | http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 23:47.41 | JT | try whatever you think has the closest tones |
| 23:47.52 | JT | i just checked the zaptel source code, kr is not in there |
| 23:48.08 | JT | zonedata.c in the zap sources |
| 23:48.29 | JT | you can always add your own if you have the information on what the frequencies of your tones are |
| 23:48.31 | Mercestes | oh |
| 23:48.45 | Mercestes | that would explain why google zaptel timezone was largely unfruitful |
| 23:49.01 | JT | quite possibly |
| 23:49.12 | JT | Zipper_32: IP301, IP430, IP501 |
| 23:49.57 | Daejeo1 | JT: how can i find the frequency of my phone tone |
| 23:50.21 | JT | Daejeo1: perhaps it's published by a government regulatory agency or telco, i have no idea |
| 23:50.51 | JT | otherwise you could record all the tones and analyse them with spectral analysis software |
| 23:50.58 | JT | ie. most good audio software |
| 23:52.27 | Zipper_32 | How do you like the 430's? I'm looking to purchase some for an office which has used 501's in the past, but will likely move to PoE in the future. |
| 23:53.03 | JT | they're quite nice for rollouts |
| 23:53.11 | JT | no stupid PoE cable |
| 23:53.18 | JT | has a dc jack built in |
| 23:53.30 | Zipper_32 | And a DC adapter with the phones? |
| 23:53.31 | Hymie | JT: you're a POE cable.. YOU"RE A POE CABLE!@ |
| 23:53.32 | JT | and comes with very compact inline switchmode power supplies |
| 23:53.39 | JT | Hymie: ? |
| 23:53.52 | [TK]D-Fender | Zipper_32, If you're planning no PoE now, and YES to later, then the IP 430 is a good choice |
| 23:54.10 | Zipper_32 | Hi again [TK], |
| 23:54.36 | Zipper_32 | And the main difference between the 430's and the 501's is... exactly what Hymie is asking. |
| 23:54.39 | [TK]D-Fender | Hymie, if you're NOT planning PoE, the 501 is a better choice |
| 23:55.18 | Zipper_32 | Other than that, it's a 3 line vs. 2 line deal. As far as I see it. |
| 23:55.20 | Hymie | only two lines though |
| 23:55.35 | [TK]D-Fender | Hymie, most people don't even need 2. |
| 23:55.37 | [TK]D-Fender | let alone 3 |
| 23:55.54 | [TK]D-Fender | do NOT confuse having 2 "lines" and meaning supporting 2 CALLS. |
| 23:56.02 | Hymie | I AM NOT MOST PEOPLE, I AM A LOUD BASTARD, I NEED 1000 |
| 23:56.03 | [TK]D-Fender | I will explain this upon my return in 2.5h |
| 23:56.12 | JT | Zipper_32: 501 has a bigger screen, greyscale shading |
| 23:56.18 | Zipper_32 | ha, bye TK, |
| 23:56.18 | JT | 430 has no greyscale |
| 23:56.19 | Hymie | [TK]D-Fender: yes, I know each "line" can support 6 or 8 calls .. or whateer depending upon the model |
| 23:56.23 | JT | 430 is a much newer design |
| 23:56.29 | JT | the 501 is older vintage |
| 23:56.51 | Zipper_32 | 430's it is then, these are for retail. |
| 23:56.58 | Hymie | smaller LCD though |
| 23:56.59 | Hymie | hmm |
| 23:57.26 | JT | also, the 501s are pretty old scool with PHYSICAL LABELS for line keys ;) |
| 23:57.29 | JT | school |
| 23:57.43 | wunderkin | Zipper_32, check the 320 and 330 then |
| 23:58.02 | Zipper_32 | wunderkin: Any particular reason? |
| 23:58.11 | JT | cheaper |
| 23:58.15 | wunderkin | other than fender told you earlier? |
| 23:58.20 | Zipper_32 | That's about it? |
| 23:58.28 | JT | newest design series |
| 23:58.28 | Hymie | there's only $5 difference for me between the 430 and the 501 |
| 23:58.41 | wunderkin | plus PoE cable |
| 23:58.49 | Zipper_32 | Well Fender mentioned just that they're cheaper. That was it. |
| 23:58.59 | JT | wunderkin: that's not a difference to the 430 |
| 23:59.04 | JT | 430 doesn't need special cables |
| 23:59.19 | JT | Zipper_32: the 430 is the lowest model with speakerphone afaik |
| 23:59.25 | wunderkin | yes but you have to buy the cable for 501... he is doing a comparison |
| 23:59.30 | Hymie | er |
| 23:59.35 | Hymie | you do? |
| 23:59.37 | Hymie | I mean |
| 23:59.46 | Zipper_32 | Ahh, the 330 doesn't have speakerphone. That's my main difference then, |
| 23:59.50 | Hymie | mine came with an A/C adapter and an ethernet cable to inject the power into |