| 00:00.03 | Tech_Travis | outtolunc Are you referring the authenticate like VoicemailMain uses? |
| 00:00.18 | outtolunc | core show application authenticate |
| 00:00.52 | *** part/#asterisk gavimobile (n=user@bzq-84-108-29-62.cablep.bezeqint.net) |
| 00:01.50 | ManxPower-work | or "show applications like authenticate" |
| 00:02.05 | ManxPower-work | core show applications like authenticate |
| 00:04.39 | *** join/#asterisk mr_ian (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:21.24 | *** part/#asterisk delrocas (n=OWNER32@71-34-217-204.spkn.qwest.net) |
| 00:24.11 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
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| 01:14.35 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 01:22.31 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:29.40 | Katty | hi |
| 01:30.19 | Katty | waits around for face mask to dry |
| 01:32.07 | *** join/#asterisk chendy_ (n=chatzill@219.133.69.182) |
| 01:33.08 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
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| 01:34.48 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 01:45.35 | leifmadsen | Katty: hoi |
| 01:54.42 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
| 01:56.31 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 02:30.10 | jas123 | HI, anyone using fax for asterisk, I have 2 conventional POTS fax machine(panasonic & sharp) faxing to asterisk tdm card, but sharp always show line error message and faxing failure, but panasonic success rate is almost 99%, is anyone face this before implementing fax for asterisk, mind to share |
| 02:35.27 | *** join/#asterisk etnos (n=etnos@adsl-2-204-26.mia.bellsouth.net) |
| 02:40.18 | *** join/#asterisk Katty (n=User@adsl-70-253-169-127.dsl.stlsmo.swbell.net) |
| 02:40.40 | Katty | hello thar! |
| 02:42.26 | n0cturnal | hullo Katty |
| 02:42.33 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 02:43.18 | jas123 | anyone mind to share |
| 02:44.08 | p3nguin | Looks like Fraggle Rock is going to have a new movie. |
| 02:47.21 | JAMMAN2110 | Right |
| 02:47.25 | JAMMAN2110 | hmm |
| 02:48.17 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:50.05 | Katty | what's shakin my bacons |
| 02:50.33 | p3nguin | eh? |
| 02:50.37 | *** part/#asterisk lanning (n=lanning@208.87.235.224) |
| 02:50.47 | p3nguin | shakes katty's bacons |
| 02:51.34 | p3nguin | mmmm... bacon |
| 02:51.59 | Katty | it means what's happenin |
| 02:56.32 | Katty | hugs p3nguin |
| 02:56.36 | b14ck | sup katty |
| 02:56.41 | Katty | anything intereesting going on up north |
| 02:57.15 | Katty | b14ck: hello. |
| 02:57.21 | p3nguin | Here? Just a typical snow-covered night, at 14 degrees. I didn't even leave the house all day. |
| 02:57.31 | Katty | :< |
| 02:57.33 | Katty | i went to work. |
| 02:57.34 | Katty | lucky you! |
| 02:57.49 | b14ck | i finished my jury duty todaY :_ |
| 02:57.51 | b14ck | :) |
| 02:57.59 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 02:58.36 | Katty | ugah, jury duty |
| 02:58.48 | Katty | they tried to get me to do jury duty once |
| 02:59.02 | Katty | but i managed to get out of it cause i'm my company's only server/phone person |
| 02:59.13 | b14ck | heh, i tried that too |
| 02:59.17 | b14ck | but they kept me anyhow |
| 02:59.22 | b14ck | it was only m->th luckily |
| 02:59.22 | b14ck | heh |
| 02:59.42 | *** join/#asterisk simcop2387 (n=simcop23@p3m/member/simcop2387) |
| 03:00.11 | simcop2387 | is this the right channel for asking asterisk related questions or is this one for development of asterisk? |
| 03:00.25 | voipmonk | this is the right place |
| 03:00.28 | voipmonk | ask your question |
| 03:00.33 | simcop2387 | ah cool |
| 03:02.14 | Katty | #asterisk-dev is the other one |
| 03:02.32 | simcop2387 | I'm having issues where all sip calls seem to have their Audio IP pointed at my local ip all of the sudden and i'm not sure why, sip show channel says, Audio IP: 192.168.10.135 (local), and i'm having problems where i'm either getting no audio or they can't hear me and last time this happened it was a problem there, i've got localnet set in the sip.conf and hold on i think i've got an idea.... |
| 03:02.48 | b14ck | nat! |
| 03:02.52 | Katty | sounds like nat. |
| 03:03.01 | simcop2387 | yea i am behind nat, its set to always assume its there |
| 03:03.08 | Katty | infobot: nat |
| 03:03.09 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 03:03.16 | p3nguin | ~sipnat |
| 03:03.17 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:03.23 | Katty | ooh, there it is. thank you |
| 03:03.32 | voipmonk | so now u have a crapload of nat reading to do |
| 03:03.36 | voipmonk | get to it |
| 03:03.38 | voipmonk | :) |
| 03:03.41 | Katty | hello mister monk! |
| 03:03.42 | simcop2387 | yep got it, had an entry in my /etc/hosts that was screwing with it |
| 03:03.44 | Katty | how's the wifie |
| 03:03.50 | voipmonk | brushing her teefers |
| 03:04.01 | Katty | i do likes me a good toofer brushing |
| 03:04.06 | Katty | had one earlier. |
| 03:04.09 | voipmonk | LOL |
| 03:04.12 | Katty | after i did my mint julep mask |
| 03:04.13 | voipmonk | with the right brush |
| 03:04.16 | voipmonk | its very nice |
| 03:04.29 | Katty | you need to get her mint julep mask |
| 03:04.29 | simcop2387 | today's been like this all day long... |
| 03:04.34 | Katty | they carry it at walgreens. |
| 03:04.36 | voipmonk | whispers... I think I need to do that tonight... just cuzzz |
| 03:04.50 | Katty | it's like a minty smelling toothpaste, but it's a clay mask. |
| 03:05.20 | Katty | highly recommend exfoliating first. |
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| 03:46.48 | ChannelZ | I'm afraid to ask what you use for actual toothpaste |
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| 03:51.10 | simcop2387 | forgot to mention i finally found the real problem, stupid router firmware version... |
| 03:51.26 | simcop2387 | now all my stuff is going funny but stablizing and sip works |
| 04:05.17 | *** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-137-89.ks.ks.cox.net) |
| 04:05.56 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 04:08.37 | bcrisp | woot |
| 04:09.12 | ChannelZ | WHARRRRF |
| 04:12.20 | bcrisp | u know anyone that works at google? |
| 04:12.40 | ChannelZ | I used to |
| 04:13.55 | bcrisp | got approached on linked in by a google recruiter.. honestly i dont know anyone that works there |
| 04:14.07 | bcrisp | would be nice to get feedback from them on how it is |
| 04:20.30 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-129-163.ks.ks.cox.net) |
| 04:31.25 | Qwell | bcrisp: all the people I know like working there.. |
| 04:31.36 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:34.00 | bcrisp | ive read mixed things about goog.. long interview periods etc |
| 04:37.39 | bcrisp | mt view CA doesnt sound bad tho heh |
| 04:39.34 | bcrisp | ChannelZ, what did you do there? |
| 04:40.23 | ChannelZ | nothing - I meant I knew someone who used to work there, not me |
| 04:40.53 | ChannelZ | in CO, I don't remember what he actually did there |
| 04:41.07 | ChannelZ | he got a free G1 though |
| 04:41.25 | bcrisp | heh |
| 04:42.05 | ChannelZ | he seemed to like it regardless. I think he came over from doubleclick |
| 04:42.38 | bcrisp | interesting |
| 04:42.54 | bcrisp | its funny because he mentioned interest in MS sql developers... kinda weird |
| 04:43.01 | ChannelZ | but they just downsized a bunch of people out and he was one, but I guess they kept him on for an extra 6 months or so to make the transition |
| 04:43.28 | ChannelZ | brb reboot |
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| 05:00.06 | *** join/#asterisk vk2dgy (n=rossw@ali-syd-3.albury.net.au) |
| 05:00.50 | vk2dgy | can someone offer me a bright idea to help debug an odd problem calling an extension? |
| 05:03.42 | vk2dgy | everyone must be asleep? |
| 05:06.14 | ChannelZ | I suggest you fix it |
| 05:06.55 | vk2dgy | thats a good suggestion... but I'm not sure what the problem is to fix! |
| 05:07.34 | ChannelZ | Me either. |
| 05:07.53 | vk2dgy | basically, I have a number of extensions - all via SIP. |
| 05:08.12 | vk2dgy | all works well - but one extension only - I can call it fine |
| 05:08.39 | vk2dgy | it can call others fine - but if I use a CALL file to call it, it doesn't work |
| 05:08.54 | vk2dgy | any other extension works, but not that one. |
| 05:09.02 | *** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-129-163.ks.ks.cox.net) |
| 05:09.12 | vk2dgy | for some reason, it seems to report as busy (and doesn't have VM) so the call just fails. |
| 05:10.02 | ChannelZ | have to see extensions.conf, sip.conf and your call file |
| 05:11.01 | ChannelZ | Pastebin 'em |
| 05:24.09 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 05:45.55 | voipmonk | back |
| 05:46.55 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 05:47.16 | ChannelZ | front |
| 05:47.38 | vk2dgy | is still here, going greyer :) |
| 05:50.12 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 05:57.27 | voipmonk | ut oh |
| 05:57.30 | voipmonk | why? |
| 05:59.32 | vk2dgy | trying to debug this odd call problem :( |
| 06:00.57 | vk2dgy | we can see the invite packet from * to the ATA, the ATA sends a "SIP/2.0 100 Trying", then "SIP/2.0 603 Decline" |
| 06:01.11 | vk2dgy | I can't (yet) find any hint WHY it's declining though. |
| 06:04.11 | ChannelZ | RE: sip debug a normal call to SIP/231 and see why it's different. There's something different about the requests if it's succeeding there |
| 06:04.43 | voipmonk | vk2dgy: - we need more debug - use pastebin.ca to display your sip debug |
| 06:04.50 | voipmonk | please paste the complete call |
| 06:04.53 | voipmonk | from dialplan execution |
| 06:05.01 | voipmonk | until 3 seconds after you hang up |
| 06:05.16 | vk2dgy | ok, just going to try that now. |
| 06:05.32 | vk2dgy | I have relocated the ATA to a live IP to remove NAT from the whole equation |
| 06:05.42 | vk2dgy | the problem remains, so thats "good" I suppose. |
| 06:06.12 | voipmonk | would you mind explaining the ata's relation to your asterisk system? |
| 06:06.21 | voipmonk | i wasnt present or conscious when you explained that |
| 06:06.40 | vk2dgy | ok, asterisk server is one of my own boxes in a datacentre. |
| 06:06.47 | vk2dgy | it has obviously, a live IP. |
| 06:07.19 | vk2dgy | All extensions are scattered around the place. A mix of live IPs and NAT'd ones (eg, at home behind a adsl modem) |
| 06:07.53 | voipmonk | are you having issues with just the one ata? |
| 06:07.59 | vk2dgy | this particular device is one of two I have - they were odd ones out. |
| 06:08.13 | voipmonk | you've read the nat article, yes? |
| 06:08.27 | vk2dgy | the problem was with only one ATA - but now that I got the other one back, BOTH of them are doing the same thing. |
| 06:08.48 | vk2dgy | yes, I've read the NAT - but this device is now NOT NATed, it's on a live IP. |
| 06:09.05 | vk2dgy | and the problem remains unchanged - so it's not NAT related. |
| 06:09.11 | voipmonk | do you still have the nat settings enabled on the device and in the sip.conf for the device? |
| 06:09.18 | voipmonk | ok - lets backup - what is your problem? |
| 06:09.49 | vk2dgy | ok, the problem is that any extension can call any other extension, and that works. (Including these two) |
| 06:09.52 | Defraz | can seemto find a list of dispositions. |
| 06:10.07 | Defraz | Looking at the cdrs and I can't tell what s or h or any others mean. |
| 06:10.11 | vk2dgy | I have a system that also creates "call files" to connect two extensions |
| 06:10.26 | voipmonk | ok |
| 06:10.53 | vk2dgy | when I create a call file for *ANY* other extensions (including these two problem ones - but only where they are the CALLED devices), it works. |
| 06:11.08 | vk2dgy | so a call file will "patch" arbitary extensions together, if you like. |
| 06:11.39 | vk2dgy | if I change NOTHING ELSE except the "originating" number in the call file, from a (working) extension to one of these two devices, it doesn't work. |
| 06:11.48 | vk2dgy | Asterisk "sees" the extension as busy. |
| 06:12.00 | ChannelZ | Defraz: s or h extensions do you mean? They can be found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf partway down the page |
| 06:12.11 | voipmonk | do u have debug to prove that, vk2dgy ? |
| 06:12.23 | vk2dgy | yes... |
| 06:12.41 | Defraz | that is perfect |
| 06:14.22 | ChannelZ | I'm not sure that's what you were actually talking about |
| 06:33.01 | *** join/#asterisk d00gster (n=doughant@77.31.3.118) |
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| 08:06.21 | shafu | I'm trying to debug a specific peer. when i run sip show peers it shows me a list of peers but when i do on one of them sip set debug peer <name> i get : No such peer |
| 08:06.26 | shafu | any one know why? |
| 08:07.57 | *** join/#asterisk chendy (n=chatzill@58.251.100.228) |
| 08:19.17 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 08:26.31 | Tech_Travis | shafu: just a guess, but is the peer you're trying to debug actually registered with your * box? |
| 08:27.36 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:27.54 | shafu | Tech_Travis: yes its registered |
| 08:29.55 | *** part/#asterisk thepopolinux (n=cyril@AToulouse-156-1-163-229.w90-38.abo.wanadoo.fr) |
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| 09:04.04 | ChannelZ | grrph. DSL at work fall down and go boom |
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| 09:28.23 | angryuser | file.c: ast_readaudio_callback: Failed to write frame >> What does it mean ? |
| 09:28.48 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:29.27 | ChannelZ | pretty much what it says, * was unable to write a frame of audio to a channel |
| 09:34.54 | *** join/#asterisk nardul (n=kse@212.37.141.188) |
| 09:35.09 | nardul | Doesn anyone know how to make *45 work to toggle queue joining? |
| 09:35.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 09:36.03 | nardul | Or just any way to make joining and leaving a queue easy. |
| 09:36.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 09:41.45 | *** join/#asterisk jkroon (n=jkroon@dsl-244-7-47.telkomadsl.co.za) |
| 09:44.23 | ChannelZ | Queue members (agents)? |
| 09:45.39 | jkroon | hi guys, asterisk 1.6.1.11, some calls that's going out to SIP channels drops after exactly 30 seconds. |
| 09:46.06 | jkroon | as far as I can tell this only happens with one of my clients. any ideas welcome. |
| 09:48.50 | ChannelZ | well assuming you're not forcing a timeout with a Dial() argument, what does the console say about the call being dropped? |
| 09:49.15 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
| 09:50.16 | jkroon | ChannelZ, no, no timeout on Dial(), and I'm trying to get debug output, this is on a rather busy switch though. |
| 09:54.11 | jkroon | hmm, could be related to MOH. Things are going through app_queue, and it's only calls going through app_queue that gets hung up after 30 seconds. |
| 09:54.23 | jkroon | after being accepted by a SIP/ member. |
| 09:58.55 | jkroon | I don't think this is related, but what's the actual meaning of this: |
| 09:58.57 | jkroon | [Jan 8 11:58:32] NOTICE[20728]: rtp.c:1796 ast_rtp_read: Unknown RTP codec 126 received from '192.9.200.179' |
| 10:00.16 | ChannelZ | hmm. You said this only happens with one of your clients, could it be a specific model of phone? |
| 10:00.17 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 10:01.57 | jkroon | he uses a bunch of softphones, x-lite, eyebeam and bria, all from counterpath. I have another client who also uses x-lite and eyebeam. |
| 10:02.11 | jkroon | other client has made no such reports. |
| 10:02.31 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 10:02.32 | ChannelZ | (and/or is that the point where the call actually terminates? As a NOTICE it should be a bit more of a harmless info) |
| 10:03.07 | jkroon | i see that on other servers too, so not related, I THINK. |
| 10:03.13 | ChannelZ | I see some reference to X-lite liking to set the codec like that periodically |
| 10:03.16 | jkroon | If it is other clients simply haven't reported it. |
| 10:03.20 | ChannelZ | but not that it's fatal |
| 10:03.53 | jkroon | I thought it might be related to https://issues.asterisk.org/view.php?id=15609 but the more I read/see the more I'm very, very confused. |
| 10:04.56 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:05.16 | jkroon | that bug looks like app_queue specifically, and the fact that I only see that at this one client would correlate with that. |
| 10:06.18 | jkroon | ok, installed asterisk 1.6.1.12, will restart with new version as soon as the client hits lunch time... |
| 10:06.31 | jkroon | that's about another 20 minutes. |
| 10:07.00 | ChannelZ | well good luck, I need to be in bed |
| 10:07.29 | jkroon | knock yourself out, and thanks for listening to this madman ranting. |
| 10:23.17 | shafu | is there any known asterisk filter app? asterisk -rcvvv can show a lot of output...how can i filter for specific infos? |
| 10:25.52 | *** join/#asterisk albertoandrade (n=albertoa@189.26.60.233.dynamic.adsl.gvt.net.br) |
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| 10:28.17 | ascenseur | can anyone help me with my AsteriskNOW installation? |
| 10:28.24 | ascenseur | what is the default username and password? |
| 10:28.29 | ascenseur | it's not admin |
| 10:28.45 | ascenseur | woops, dont worry about it - all solved now |
| 10:31.18 | jkroon | ascenseur, see the topic. |
| 10:32.30 | ascenseur | jkroon: woops - solved now anyway |
| 10:32.32 | *** part/#asterisk ascenseur (n=ascenseu@86.24.20.72) |
| 10:33.20 | *** join/#asterisk NET||abuse (n=l@91.123.232.146) |
| 10:44.37 | *** join/#asterisk yoshx (n=yoshx@93.9.150.199) |
| 10:58.43 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:19.40 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 11:34.34 | *** join/#asterisk Uatec (n=uatec@87-194-71-1.bethere.co.uk) |
| 11:34.37 | Uatec | criees |
| 11:34.55 | Uatec | i have a question: why wont my boss let me use asterisk instead of trixbox? |
| 11:36.53 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 11:38.10 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:38.59 | Gido-E | Uatec because he is your bos? |
| 11:39.37 | Gido-E | Start your own company or get used to such stupid dicissions :-) |
| 11:43.20 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 11:43.28 | Uatec | there's got to be a way to convince him |
| 11:43.32 | Uatec | *considers beer* |
| 12:05.21 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 12:14.49 | seanjohn | Uatec: trixbox is asterisk as far as I know. Its just freepbx, asterisk, and trixbox extras bundled |
| 12:15.08 | seanjohn | I would think your boss doesn't know how to program asterisk |
| 12:15.59 | seanjohn | heed this: when the systems start failing because of his lack, it will take down your reputation |
| 12:17.38 | ManxPower-work | ~freepbx |
| 12:17.39 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:28.34 | *** join/#asterisk sazogues (n=sazogues@95.18.51.251) |
| 12:28.39 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 12:33.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 12:35.17 | jbw | anyone know any reason why my fresh install of 1.4.28 is ignoring my extensions.conf ? |
| 12:36.36 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 12:38.08 | ManxPower-work | jbw: That's usually caused by lack of [general] or [globals] or chars before the [general] section. |
| 12:40.30 | *** join/#asterisk Akiraaa (n=Akira@92.81.215.121) |
| 12:40.42 | jbw | [general] is the first line in the file, right after that section there's a [globals] section. Any more ideas ? :( |
| 12:41.01 | *** join/#asterisk Tim_Toady (n=moi@77.49.184.114.dsl.dyn.forthnet.gr) |
| 12:41.54 | ManxPower-work | copy your extensions.conf to pastebin.ca and give is the URL |
| 12:42.08 | Akiraaa | Using telephony PC extension cards remotely requires an asterisk server running on the remote machine? |
| 12:42.33 | ManxPower-work | Akiraaa: Your question makes no sense |
| 12:42.40 | ManxPower-work | ~pb |
| 12:42.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:43.15 | Akiraaa | ManxPower-work: PCI boards with FXO/FXS ports, like the ones sold by Digium |
| 12:43.24 | ManxPower-work | Akiraaa: What about them? |
| 12:44.17 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:44.23 | Akiraaa | They only work with an Asterisk server intalled on the host machine, right? |
| 12:44.32 | ManxPower-work | Akiraaa: correct |
| 12:44.32 | *** join/#asterisk stefanlsd (n=stefanls@ubuntu/member/stefanlsd) |
| 12:44.45 | ManxPower-work | I don't know what you mean by "host machine" |
| 12:45.03 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 12:45.03 | ManxPower-work | You have an Asterisk server. You may or may not have telephony cards in the server. This is not complicated |
| 12:45.04 | Akiraaa | the machine to which the boards are attached |
| 12:45.10 | stefanlsd | hi guys. how do i accept a tone before the whole message is played? im using ivr and only accepts tones at the end of the message |
| 12:45.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:45.35 | Akiraaa | I was wondering about the option of using remote machines with FXO/FXS ports |
| 12:45.39 | ManxPower-work | stefanlsd: Use Background and Waitexten instead of Playback and waitexten |
| 12:45.51 | Akiraaa | other than simply ATA gateways |
| 12:45.54 | ManxPower-work | Akiraaa: if you have Asterisk installed on that remote machine then yes. |
| 12:46.16 | ManxPower-work | stefanlsd: Are you using a GUI for Asterisk? |
| 12:46.44 | stefanlsd | ManxPower-work: im using freepbx |
| 12:46.53 | ManxPower-work | stefanlsd: then ask on the FreePBX channel |
| 12:47.19 | stefanlsd | ManxPower-work: oki. thanks |
| 12:47.57 | ManxPower-work | ~freepbx |
| 12:47.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:48.53 | *** join/#asterisk Akiraaa (n=Akira@92.81.215.121) |
| 12:49.23 | jbw | ManxPower-work, http://pastebin.ca/1742656 |
| 12:53.13 | NET||abuse | i'm seriously lost in trying to get a gui running for my asterisk box. |
| 12:53.49 | NET||abuse | tried checking out the svn of asterisk-gui 2.0,, ./configure'd make install'd and checkconfig'd, it runs the http process, i can hit it, but all i get is a 404??? |
| 12:54.08 | ManxPower-work | NET||abuse: We don't support GUIs here. |
| 12:54.22 | NET||abuse | ubuntu karimc + packaged asterisk + svn checkout of code.. |
| 12:54.41 | ManxPower-work | I don't see how that makes a difference. |
| 12:55.02 | NET||abuse | I shoulda just installed asterisknow. |
| 12:55.04 | jbw | ManxPower-work, it seems like it's something in the file itself.. chopping the file in two shows a semi populated dialplan.. I'll go hunt it down, thanks |
| 12:55.13 | ManxPower-work | jbw: ignorepat => 9 does not belong in [globals] or [general] |
| 12:55.28 | NET||abuse | ok, so if i just want to run the s;ystem with config files.. |
| 12:55.38 | ManxPower-work | remember do NOT use quotes unless you have to. quotes are literals. |
| 12:55.51 | ManxPower-work | NET||abuse: then ask your question |
| 12:56.23 | ManxPower-work | jbw: ignorepat only applies to Zap/DAHDI/Skinny/SCCP/MGCP. It does not apply to SIP |
| 12:56.32 | jbw | ManxPower-work, thanks, that was it :) |
| 12:56.40 | NET||abuse | ah, ok, so to configure the sip service we have for our voip provider, do i just set it up in sip.conf? The book seemed to say that was just for the handsets? |
| 12:56.55 | ManxPower-work | yes, set it up in sip.conf |
| 12:56.56 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:57.29 | NET||abuse | I think at this point i'll give up on gui,,, my setup is simple, little server with one nic. no other ports, using softphones on network, and our voip provider is just a sip service on the net. |
| 12:58.23 | ManxPower-work | jbw: all those extra " will come back and bite you later. |
| 12:58.50 | jbw | i don't need the "'s ? |
| 12:58.59 | NET||abuse | are there any considerations i need to think about to allow the phone system to operate as a group pbx? i want us to be able to do things like transfer calls to eachother's softphone clients |
| 12:59.00 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.202) |
| 12:59.02 | jbw | i see.. i don't, i'll remove them |
| 12:59.30 | Kchehab | suddenly when i reload asterisk dialplan dissapeared |
| 12:59.40 | ManxPower-work | jbw: quotes are literal in Asterisk. So if you set FRED="gone" then $[${FRED} = gone] will NOT match because FRED="gone" not FRED=gone |
| 12:59.56 | ManxPower-work | i.e quotes are literal |
| 12:59.58 | Kchehab | even i cant see that extensions.conf is loaded |
| 13:00.04 | Kchehab | how to fix it ? |
| 13:00.12 | ManxPower-work | Kchehab: fix the error in your extensions.conf |
| 13:00.14 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:00.38 | Kchehab | manxpower is there a way to see my error in extensions.conf since its too large |
| 13:00.45 | ariel_ | Morning folks |
| 13:00.47 | ManxPower-work | Kchehab: try starting Asterisk as "asterisk -cvvv" |
| 13:01.17 | *** join/#asterisk E-bola (n=bola@smtp.techbiz.dk) |
| 13:01.26 | Kchehab | manxpower i will try |
| 13:01.50 | jbw | ManxPower-work, got it - thanks again |
| 13:01.58 | E-bola | Hello all, if i got a queue with announce-frequency = 5 and i dont get any announces at all, what might be wrong? |
| 13:02.11 | E-bola | I have not changed any of the audio files, and tyhey are all chmod'd 777 |
| 13:02.51 | Kchehab | manxpower the problem didnt appears |
| 13:03.02 | Kchehab | is there another command you know |
| 13:03.09 | Kchehab | -gvvvvvvc ?? |
| 13:04.04 | ManxPower-work | How would that be different from what I gave you? |
| 13:04.23 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 13:05.20 | ManxPower-work | Kchehab: paste the first 100 lines of your dialplan to pastebin.ca |
| 13:05.23 | ManxPower-work | ~pb |
| 13:05.24 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:05.37 | ManxPower-work | Are you using a GUI for Asterisk? |
| 13:06.37 | ManxPower-work | If you are then all this is pointless. |
| 13:07.52 | E-bola | dont understand why his queue works on 1 asterisk server and not on another |
| 13:11.00 | Kchehab | manxpower the problem remains even i reload my old extensions which is running on the other server with same asterisk version |
| 13:11.04 | *** join/#asterisk zorp75ck (n=zorp75ck@146.186.115.44) |
| 13:11.13 | ManxPower-work | Kchehab: Then I cannot help you. |
| 13:11.46 | Kchehab | manxpower what is the name of the module which is responsible to load extension.conf |
| 13:12.31 | phix | E-bola: gremlins |
| 13:13.24 | ManxPower-work | Kchehab: pbx_config.so |
| 13:13.57 | *** join/#asterisk ascenseur (n=ascenseu@86.24.20.72) |
| 13:14.02 | E-bola | is going crazy |
| 13:14.14 | E-bola | the queue just keeps ringing with no interuptions at all |
| 13:14.24 | Kchehab | manxpower problem solved by reloading obx_config |
| 13:14.30 | Kchehab | pbx_config thans |
| 13:14.33 | E-bola | its like its ignorering both the position announcement config and i even tried to config a periodic announcements, neither is ever played |
| 13:14.43 | ascenseur | just a quick question - is it possible to attach a skype name to an extension? |
| 13:15.52 | *** join/#asterisk cxk287 (n=zorp75ck@146.186.115.44) |
| 13:16.40 | leifmadsen | ascenseur: with chan_skype, sure |
| 13:16.50 | ManxPower-work | Kchehab: no, you problem is not solvee. |
| 13:17.35 | ascenseur | leifmadsen: thanks! |
| 13:18.13 | ManxPower-work | Skype is useful -- the poor need phone service too. |
| 13:18.16 | Kchehab | manxpower why ? |
| 13:18.33 | ManxPower-work | Kchehab: because you do not know why that module failed to load. |
| 13:19.28 | Kchehab | manxpower i will clean my script from passwords and post it |
| 13:19.38 | ManxPower-work | Kchehab: Sorry, I have to go to work now. |
| 13:20.01 | ManxPower-work | I did not think it would take 30 mins just to get a copy of that extensions.coinf |
| 13:20.36 | phix | E-bola: yep, it is gremlins then if it is driving you insane, that is what they love to do |
| 13:21.36 | Kchehab | manxpower sorry man i dont mean that |
| 13:21.52 | Kchehab | i didnt* |
| 13:22.30 | *** part/#asterisk ascenseur (n=ascenseu@86.24.20.72) |
| 13:22.49 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:25.54 | Kchehab | manxpower i found my hitch,it was a missing include file in extensions.conf |
| 13:26.44 | ManxPower-work | Kchehab: I'm surprised you didn't notice the #include changes that are listed in UPGRADE*.txt in the Asterisk source code directory. |
| 13:27.12 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
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| 13:38.08 | NET||abuse | wow, just got the asterisk gui to work |
| 13:38.11 | NET||abuse | stupid symlinks. |
| 13:38.19 | NET||abuse | now, on to configuring voip service. |
| 13:38.55 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:44.53 | *** join/#asterisk yoshx (n=yoshx@93.9.150.199) |
| 13:45.16 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
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| 14:03.28 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:03.52 | *** join/#asterisk espent (n=espent@home.kulturit.no) |
| 14:04.19 | *** part/#asterisk espent (n=espent@home.kulturit.no) |
| 14:08.32 | Katty | morning |
| 14:09.44 | *** join/#asterisk _zen_ (n=_zen_@cpe-74-66-140-78.nyc.res.rr.com) |
| 14:11.13 | Katty | MORNING |
| 14:12.14 | ariel_ | ok I am semi awake, Morning Katty oh not so loud please. |
| 14:12.22 | Katty | oh, right. |
| 14:12.26 | Katty | gives ariel_ coffee. |
| 14:13.06 | ariel_ | T/y, but it's cold and I don't think anything will work today, thinking of going back home and going back to bed... |
| 14:13.23 | ManxPower-work | ariel_: That's the best idea I've seen all day. |
| 14:13.48 | ariel_ | It's not suppose to be this cold down here for this long argh, depressing. |
| 14:14.51 | jaytee | morning Katty |
| 14:15.04 | ariel_ | Besides I am working on documentation and trouble shooting white papers, yet another yuk job. |
| 14:15.32 | Katty | hugs jaytee |
| 14:15.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.01 | Katty | holy blackbirds batman |
| 14:16.04 | Katty | infobot: crittercam |
| 14:16.05 | infobot | crittercam is, like, Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 14:16.10 | Katty | they're taking over. |
| 14:16.23 | Katty | anndndd all just flew away. |
| 14:18.00 | jaytee | I'm doing documentation too. I have to write several documents that cover disaster recovery of our Asterisk system, complete from scratch rebuild in case our datacenter is hit by a meteorite, another for quick restore of service using our redundant systems and troubleshooting documentation all at a level of simplicity that even our 2 MCSE's that don't want to really learn linux or asterisk can understand. |
| 14:18.08 | ManxPower-work | .part #squirrel-updates |
| 14:18.11 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 14:24.40 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 14:30.01 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
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| 14:30.48 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 14:32.00 | Katty | brr. |
| 14:32.04 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 14:32.10 | Katty | hi Godfather_ |
| 14:32.27 | Godfather_ | hi Katty :D |
| 14:32.33 | Godfather_ | o/ all |
| 14:32.47 | Katty | what's happenin |
| 14:32.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:33.01 | Katty | hi _ShrikE |
| 14:33.34 | Godfather_ | 3ºC here, and raining a lot |
| 14:34.27 | Katty | 3ºC <- |
| 14:34.30 | Godfather_ | not usually in mallorca :-| |
| 14:34.31 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 14:34.33 | Katty | 3a? |
| 14:34.37 | Katty | what temp is 3a? :P |
| 14:34.39 | Katty | HI MANX |
| 14:34.45 | Katty | THE SQUIRRELS ARE STILL DOING OKAY |
| 14:34.46 | voipmonk | yawns... its... the Godfather ( with an underscore) |
| 14:34.56 | Katty | hi monk. |
| 14:35.11 | Godfather_ | hi voipmonk |
| 14:35.16 | voipmonk | i wanna know what would happen if i let my dog loose near the squirrels |
| 14:35.21 | voipmonk | hi Katty |
| 14:35.23 | Katty | they would go up a tree. |
| 14:35.25 | ManxPower-work | Do it! Do it!~ |
| 14:35.25 | voipmonk | hi Godfather_ |
| 14:35.34 | Katty | riddick occasionally goes nuts :P |
| 14:36.04 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 14:36.17 | Godfather_ | Katty, 38ºF.. |
| 14:36.38 | Katty | brr :< |
| 14:36.41 | Godfather_ | :-D |
| 14:36.49 | _ShrikE | morning Katty |
| 14:38.00 | Katty | checks weather. |
| 14:38.21 | Katty | 9F. |
| 14:38.28 | Katty | 32 is 0C |
| 14:38.32 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 14:38.33 | Godfather_ | lol |
| 14:38.48 | Katty | with windchill is -4F |
| 14:39.04 | Katty | google says -4F = -20C |
| 14:39.54 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 14:40.32 | Katty | p3nguin: what temp is it up north this morning? |
| 14:41.04 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 14:41.39 | Katty | :> |
| 14:41.41 | Katty | hugs coppice |
| 14:42.11 | ariel_ | feels bad since it's was only 44F outside this morning. |
| 14:42.18 | FlaPer87 | hey guys, I have an asterisk installation at work and my own asterisk installation at home... Is it possible to register to my work sip account when I register to my home sip account? |
| 14:42.19 | Katty | ariel_: i will be /right/ over |
| 14:42.24 | Katty | packs her bags. |
| 14:42.42 | Katty | FlaPer87: you can have a phone registered with two servers. |
| 14:42.51 | Katty | FlaPer87: Line 1 and Line 2 |
| 14:43.00 | ManxPower-work | If your phone supports being registered to more than one server. |
| 14:43.12 | ariel_ | or create a link between the two servers. |
| 14:43.14 | voipmonk | FlaPer87: you can have one system register to your itsp and have your phone register to the asterisk registered to your itsp and build a dialplan that rings to wherever you are using time of day, bluetooth proimity detection, or what have you. |
| 14:43.27 | voipmonk | What Katty said, too |
| 14:43.29 | Katty | that would also work. |
| 14:43.40 | voipmonk | :) |
| 14:43.43 | Katty | i have my phone at home simply registered with both servers |
| 14:44.00 | Katty | it was quick. |
| 14:44.02 | voipmonk | probably with two diff rings, too |
| 14:44.08 | Katty | nah |
| 14:44.09 | ariel_ | I keep work at work and my home system seperate... |
| 14:44.15 | ManxPower-work | My home phone is usually registered to 3 servers, 4-lines. |
| 14:44.21 | Katty | ariel_: i work from home occasionally. |
| 14:44.26 | Katty | ariel_: that's the only reason it's setup like that |
| 14:44.33 | leifmadsen | I have the same system in the colo handling my home and business lines |
| 14:44.37 | leifmadsen | why manage separate servers? |
| 14:44.37 | ManxPower-work | 2-lines for work, 1 line for my personal number, one line for testing stuff. |
| 14:44.42 | ariel_ | yes so do I, but it's bad enough, I let the work phone take vm then it emails them |
| 14:44.59 | Katty | oh wow :< |
| 14:45.01 | Katty | that is pretty bad. |
| 14:45.19 | Katty | do you have voicemail.conf to send you txt messages? |
| 14:45.33 | voipmonk | when im away - everything goes to google voice - when im feeling bored, i turn on my iphone and fire up the "Softphone" app from acrobits and log in, immediately my calls are now sent to my iphone. |
| 14:45.40 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:45.52 | Katty | oooh, you have a google voice account? |
| 14:45.54 | voipmonk | when i exit softphone, everything returns to GV |
| 14:45.54 | Katty | i must add you |
| 14:46.07 | ariel_ | Katty: you want an invite? |
| 14:46.23 | Katty | no i already have an account |
| 14:46.29 | Katty | i have 3 invites tho |
| 14:46.32 | Katty | if anyone would like one |
| 14:46.58 | Katty | if you have a google wave invite, i'd take one of those (= |
| 14:47.17 | ManxPower-work | What is google voice good for anyway? |
| 14:47.40 | Katty | ManxPower-work: i gave it to my mom |
| 14:48.05 | Katty | ManxPower-work: so she doesn't have to call 4 different phones to find me. |
| 14:48.06 | ManxPower-work | So it's good for people that are not computer litterate? |
| 14:48.14 | Katty | ManxPower-work: hey now, we're talkin about my mom (= |
| 14:48.20 | Katty | ManxPower-work: you be nice :P |
| 14:48.26 | FlaPer87 | Katty: I work from home occasionally too =P |
| 14:48.36 | ManxPower-work | odd. People only ever have to call one number to get ahold of me? |
| 14:48.52 | ManxPower-work | Maybe it's for people that don't have or don't understand call forwarding? |
| 14:49.04 | voipmonk | http://www.weirdasianews.com/wp-content/uploads/2010/01/Google_Toilet_Paper_wm.jpg |
| 14:49.26 | Katty | voipmonk: that is /awesome/ |
| 14:50.00 | ManxPower-work | Call my work number, if I don't answer it forwards to me cell. Call my personal number, if I don't answer it forwards to my cell |
| 14:50.06 | Naikrovek | i have google wave and voice invites i think |
| 14:50.42 | Naikrovek | yeah |
| 14:50.50 | FlaPer87 | Naikrovek: can you send me a google voice invite? =D |
| 14:51.03 | Naikrovek | sure |
| 14:51.06 | Naikrovek | pm me your email |
| 14:51.08 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:51.10 | *** join/#asterisk jfranco1 (n=ikono@190.146.200.120) |
| 14:51.14 | Katty | hi seanbright |
| 14:51.28 | *** join/#asterisk eppigy (n=Dave@216-139-241-102.aus.us.siteprotect.com) |
| 14:51.33 | seanbright | hi Katty |
| 14:52.07 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:52.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.24 | eppigy | hello |
| 14:52.26 | eppigy | i am dave |
| 14:53.11 | ManxPower-work | Odd that a simple call forwarding service is so in demand. Never underestimate the tech illiteracy of the general public. |
| 14:53.24 | Naikrovek | I'm sorry, Dave. this conversation can no longer serve a useful purpose. good bye. |
| 14:53.57 | Naikrovek | ManxPower-work: well i can have a single number and forward it to wherever I am at the time, it's handy |
| 14:54.03 | Katty | hugs eppigy |
| 14:54.08 | Naikrovek | can do the same with asterisk, yes |
| 14:55.36 | ManxPower-work | Naikrovek: I can do that easily without google voice. |
| 14:55.44 | ManxPower-work | It's called RCF |
| 14:55.46 | Naikrovek | i know |
| 14:56.00 | ManxPower-work | I can do the same without Asterisk. |
| 14:57.20 | Katty | thinks manx is cranky this morning. |
| 14:57.39 | drmessano | I like having one number I never have to give up |
| 14:57.52 | drmessano | Sorry, but porting numbers is expensive and messy |
| 14:57.53 | Chainsaw | Katty: You mean there are days when he's nice? Seriously? |
| 14:58.34 | Naikrovek | Chainsaw: sometimes |
| 14:58.46 | eppigy | warms up next to Katty |
| 14:58.52 | Naikrovek | he has a right to be in whatever mood he's in though, just like any of us |
| 14:59.48 | Katty | Chainsaw: of course. |
| 14:59.56 | Katty | Chainsaw: manx is a very pleasant person, generally. |
| 15:00.29 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 15:00.30 | ManxPower-work | I just think it's a pretty useless service except for the name brand on it |
| 15:00.49 | Naikrovek | it does more than forwarding |
| 15:01.10 | Naikrovek | it records calls, it transcribes voicemails |
| 15:01.11 | ManxPower-work | I'm waiting for the details |
| 15:01.12 | Naikrovek | others |
| 15:01.39 | ManxPower-work | Now THOSE features might be useful to someone without a PBX. |
| 15:01.54 | Naikrovek | you can send/receive sms via the web gui |
| 15:02.07 | Naikrovek | i can send you an invite if you want to fiddle with it |
| 15:02.22 | ManxPower-work | But call forwarding? I was able to remotely change my call forwarding info remotely since like 1985 |
| 15:02.30 | Naikrovek | yeah |
| 15:02.52 | ManxPower-work | Naikrovek: I don't really see the need for wasting my time on yet another "cool must have service" |
| 15:03.01 | Naikrovek | just asking |
| 15:03.07 | ManxPower-work | I like keeping my phone service simple./ |
| 15:03.17 | Naikrovek | understandabel |
| 15:03.42 | Naikrovek | personally, i like knowing what's out there so when friends or family ask about these things i can be informed |
| 15:03.49 | Naikrovek | i don't like being a dick and saying "learn it yourself" |
| 15:03.59 | Naikrovek | not saying that's your position |
| 15:04.05 | Naikrovek | but it is certainly the position of some |
| 15:05.12 | ManxPower-work | It seems to me that the real advantage of Google Voice is that even a moron can use it. |
| 15:05.24 | Naikrovek | that's AN advantage |
| 15:05.53 | Naikrovek | but a moron would never know this service existed |
| 15:06.11 | Naikrovek | he would continually update everyone he knows about his new phone number every time he got a new phone |
| 15:06.27 | ManxPower-work | Not like I do, which is get one number and keep it. |
| 15:06.58 | Naikrovek | because 2 months back he lost his job because he's a moron, and the telco gave his number out to someone else, or because he got a new number when he finally paid his cell phone bill |
| 15:07.06 | ManxPower-work | Too bad 500 number services never took off. |
| 15:07.43 | ManxPower-work | Naikrovek: *nod* As opposed to getting a number for $1.69/month and being able to use it anywhere you have an internet or cellular service. |
| 15:07.52 | Naikrovek | yeqah |
| 15:07.57 | Naikrovek | well google gives you that number for free |
| 15:08.17 | ManxPower-work | Anyone that can't afford $1.69/month should no have a phone. |
| 15:08.20 | Naikrovek | true |
| 15:08.41 | Naikrovek | i don't think we're arguing this time |
| 15:08.45 | ManxPower-work | I see your point. Morons would not use Google Voice. So that leaves...nobody. |
| 15:08.46 | Naikrovek | just going through the motions |
| 15:09.19 | ManxPower-work | hugs his vitelity number |
| 15:09.32 | *** part/#asterisk jfranco1 (n=ikono@190.146.200.120) |
| 15:10.13 | chuckf | am I right that outside of the text to speech and sms all the features of gv can be done in asterisk? |
| 15:10.34 | [TK]D-Fender | chuckf: TTS can be done in *. |
| 15:10.42 | [TK]D-Fender | chuckf: So that leaves SMS |
| 15:10.43 | Naikrovek | not text to speech, speech to text is what gv does |
| 15:10.57 | chuckf | got it backwards |
| 15:11.00 | ManxPower-work | SMS via web works just fine. SMS via FSK analog also works. |
| 15:11.55 | ManxPower-work | Asterisk just can't pretend it's a cell phone on a cellular network and send SMSs |
| 15:12.43 | chuckf | thanks for the info |
| 15:12.50 | voipmonk | GV doesnt do a great job either... its more comedy than function |
| 15:13.05 | voipmonk | the speech to text |
| 15:13.16 | chuckf | but it looks good on paper:) |
| 15:13.47 | Naikrovek | yeah it's not that good |
| 15:13.50 | Naikrovek | but it'll get better |
| 15:13.54 | drmessano | Feature numero uno is getting a number from a company that wont go bankrupt anytime soon |
| 15:14.01 | Naikrovek | yeah |
| 15:14.22 | drmessano | I've lost numbers twice with VoIP services.. |
| 15:14.26 | ManxPower-work | That's the best reason I've seen all day. |
| 15:14.38 | drmessano | Porting sucks |
| 15:14.43 | Naikrovek | i just give out my google voice number and it rings both my cell numbers and my home phone at the same time. |
| 15:15.03 | ManxPower-work | I wish Asterisk could do something like that. |
| 15:15.25 | ManxPower-work | Just admit it. The only reason you have a Google Voice account is because all the cool kids are doing it. |
| 15:15.28 | ManxPower-work | ducks |
| 15:15.39 | Naikrovek | i don't give a damn what a cool kid is or what they're doing. |
| 15:15.41 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:15.43 | drmessano | I use my GV work number to ring my cell and multiple home phones.. nothing asterisk couldnt do, but its a ring group external to my PBX at home and my shitty ATT phone |
| 15:16.18 | Naikrovek | i don't want to set up an asterisk box at home just to do that, when i can do it online for nothing |
| 15:16.23 | Naikrovek | and be done in 5 minutes |
| 15:16.41 | drmessano | I dont just use it for that, duh |
| 15:19.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:19.47 | drmessano | Point being GV is a little more reliable than my connectivity to my ITSP and my cell phone in the case of outage, change in cell provider, or loss of number from ITSP due to bankruptcy or whatever other reason ITSPs are as reliable to be around in 2 years as banks |
| 15:21.31 | drmessano | Of course, when the PSTN dies, having a "phone number" will be useless too.. |
| 15:21.33 | drmessano | Hmmm |
| 15:23.26 | *** join/#asterisk Tim_Toady (n=moi@77.49.184.114.dsl.dyn.forthnet.gr) |
| 15:23.45 | drmessano | Google is also going to be wrapping SIP connectivity into GV at some point thanks to their acquisition of Gizmo5.. So thats another need you can scratch off the list |
| 15:24.37 | chuckf | until the google paranoids out there won't call you... |
| 15:25.01 | *** join/#asterisk tzafrir (n=tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 15:27.37 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 15:27.43 | Chainsaw | leifmadsen: Any hope for #14163 left? |
| 15:28.12 | leifmadsen | Chainsaw: yes, there is hope for all issues |
| 15:28.13 | leifmadsen | :) |
| 15:28.21 | russellb | M14163 |
| 15:28.22 | leifmadsen | Chainsaw: I'm doing my best to get it up the priority stack |
| 15:28.36 | leifmadsen | russellb: bot fail :) |
| 15:28.38 | Chainsaw | leifmadsen: Right, thanks. Some days I wonder if this sruffell still works for Digium. |
| 15:28.41 | russellb | file: perhaps MuffinMan could be here, too? |
| 15:28.59 | file | ummm |
| 15:29.20 | russellb | If it's a problem with the usage of Digium hardware, it's probably best to handle it through Digium technical support |
| 15:29.20 | *** join/#asterisk MuffinMan (n=muffinma@asterisk/issue-tracker-bot/muffinman) |
| 15:29.47 | russellb | that will get it recorded and reported internally, and will likely get it higher on "the list" |
| 15:30.05 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:30.12 | leifmadsen | M14613 |
| 15:30.14 | MuffinMan | [closed] [Asterisk] Applications/app_fax 0014613: Sending and receiving a fax between extensions on the same Asterisk machine fails reported by amessina https://issues.asterisk.org/view.php?id=14613 |
| 15:30.15 | leifmadsen | errr... |
| 15:30.17 | leifmadsen | wrong issue :) |
| 15:30.19 | drmessano | .nick MuffinSpamBot |
| 15:30.23 | leifmadsen | M14613 |
| 15:30.25 | MuffinMan | [closed] [Asterisk] Applications/app_fax 0014613: Sending and receiving a fax between extensions on the same Asterisk machine fails reported by amessina https://issues.asterisk.org/view.php?id=14613 |
| 15:30.29 | russellb | leifmadsen: FAILLLLLLLL |
| 15:30.33 | leifmadsen | holy crap... I'm lexdysic! |
| 15:30.35 | russellb | M14163 |
| 15:30.37 | MuffinMan | [ready for review] [Asterisk] Channels/chan_dahdi 0014163: [patch] UK (BT) lines produce uncleared red alarm on TDM400P during line tests reported by jedi98 https://issues.asterisk.org/view.php?id=14163 |
| 15:30.49 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:30.56 | russellb | there ... so yeah, I would report that via support - http://www.digium.com/en/supportcenter/ |
| 15:31.27 | leifmadsen | which I should have known better as to have recommended a while ago... |
| 15:31.30 | Chainsaw | russellb: They're going to ask for numbers that are on the PCB, aren't they. My production server is >75 miles away from me. |
| 15:31.45 | russellb | ummm, I don't know. maybe there is a way to get it remotely ... |
| 15:32.10 | Chainsaw | russellb: I'll need to know that for sure, or I'll just waste peoples time and annoy them. |
| 15:32.22 | Chainsaw | russellb: That's already happened to me, I don't want to bring that onto others. |
| 15:32.35 | russellb | so, i'm looking at the issue, so you have a patch that fixes it? |
| 15:32.45 | Chainsaw | russellb: Yes. Fixes it up 100% |
| 15:33.00 | drmessano | Write what you need on the back of a $20 and send it to the remote office.. You'll get your numbers |
| 15:33.22 | Chainsaw | russellb: I've forward ported it to 1.6.1 & 1.6.2 |
| 15:33.27 | russellb | I see that, thank you for that |
| 15:33.30 | russellb | keeps looking over it |
| 15:34.20 | russellb | Chainsaw: since you've done all of this work, I"m going to bump it up high on our list to get it looked at as soon as we can |
| 15:34.35 | russellb | almost certainly within the next few weeks, if not sooner |
| 15:34.41 | Chainsaw | russellb: Cheers. |
| 15:36.37 | russellb | leifmadsen had brought this up to me before, but I'm lame and didn't look close enough to fully understand the status of the issue |
| 15:36.41 | russellb | we should knock it out here soon. |
| 15:36.57 | Naikrovek | open source in action |
| 15:38.01 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:38.47 | russellb | blames The_Boy_Wonder |
| 15:39.15 | The_Boy_Wonder | ha, i have no history, all i see is that i'm blamed for something |
| 15:39.45 | russellb | muahaha. |
| 15:44.14 | leifmadsen | The_Boy_Wonder: and you better fix it too! |
| 15:44.20 | leifmadsen | The_Boy_Wonder: it's a DOOZY |
| 15:46.03 | Naikrovek | everyone: http://sleeptalkinman.blogspot.com/ |
| 15:52.36 | *** join/#asterisk clintc (n=clintc@n128-227-2-41.xlate.ufl.edu) |
| 15:53.49 | Katty | i saw that on reddit this morning :> |
| 15:53.57 | Naikrovek | go reddit! |
| 15:54.05 | Katty | :> |
| 15:54.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:55.48 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:03.09 | *** join/#asterisk cool^tom (n=thomas@122.166.133.143) |
| 16:03.20 | cool^tom | Hi |
| 16:05.42 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 16:07.53 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 16:09.52 | Katty | hi. |
| 16:10.09 | Chainsaw | Hello. |
| 16:10.20 | voipmonk | hello |
| 16:15.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:16.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.41 | steve745 | for an * manager account i just want to read the data for that manager, i user api to originate a call and don't want to read any data from other managers or events |
| 16:18.03 | steve745 | what rights do i need to assign |
| 16:19.04 | *** join/#asterisk lowtek (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 16:19.52 | lowtek | Hey guys, is there somewhere to download the current national CNAM database and get updates as necessary? Google isn't much help, plenty of query services, but it would be easier just to roll my own if I could find a good source for the db. |
| 16:21.17 | ManxPower-work | lowtek: No. |
| 16:21.25 | Katty | you could contact your telco about getting a copy of theirs. they will probably charge you if they give it out at all. |
| 16:21.29 | cool^tom | What is the best way to migrate a legacy telephone system to asterisk? |
| 16:21.38 | ManxPower-work | cool^tom: Very, very carefully |
| 16:21.39 | lowtek | manxpower: How do these CNAM lookup services get theirs? |
| 16:21.50 | ManxPower-work | lowtek: ask them. |
| 16:21.56 | cool^tom | I was thinking of channel banks. But I have over 120 extensions. |
| 16:21.59 | Katty | cool^tom: personally, i would set up the old system and the new system side by side. |
| 16:22.25 | Katty | cool^tom: and not disconnect the legacy system until the users are happy with it. |
| 16:22.51 | Naikrovek | anyone know what software made this: http://dl.dropbox.com/u/543400/architecture.png |
| 16:22.52 | cool^tom | I did that in an office. Now they want me to replace a system in a resort. |
| 16:22.54 | lowtek | manxpower: lol, I doubt they would want to give up their source ... |
| 16:23.18 | Katty | cool^tom: and how is their happiness any different than the office's? |
| 16:23.27 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 16:23.30 | Katty | cool^tom: if not more so, because it's probably more complicated. |
| 16:24.16 | voipmonk | ? |
| 16:24.34 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:24.34 | cool^tom | Katty: Cabling would be an issue. Would it be better to set up channel banks or have sip phones on WIFi |
| 16:24.47 | Katty | cool^tom: i would not do sip over wifi. |
| 16:24.48 | voipmonk | listen to Katty, cool^tom |
| 16:24.48 | p3nguin | katty: 13 right now; it got down to 7 overnight. |
| 16:24.59 | Katty | cool^tom: that is asking for problems. |
| 16:25.05 | Katty | cool^tom: are the legacy phones analog? |
| 16:25.10 | cool^tom | Yes. |
| 16:25.38 | cool^tom | The cards seem to be going down pretty often. They wanted to change the PBX. |
| 16:26.00 | Katty | cool^tom: well in that case i would configure the phone system side by side in the 'main office' where the majority of the people using the phones are. |
| 16:26.04 | Katty | cool^tom: pick 5 people. |
| 16:26.10 | cool^tom | Siemens tell me that stopped support for the current PBX. |
| 16:26.23 | Katty | cool^tom: leave the legacy system in place until those 5 people are satisified with the new pbx. |
| 16:26.45 | Katty | cool^tom: then swap out the rest of the phones/cabling, and make changes as needed. |
| 16:27.23 | Katty | cool^tom: you should never, ever, take out a working (or mostly working) legacy system until the people are completely happy with the new one... accidents do happen, never get yourself up a tree with a bear at the bottom. |
| 16:27.41 | Katty | cool^tom: cause that will be a whole new world of hurt and hell for you. |
| 16:27.47 | cool^tom | I experience that during my last implementation. |
| 16:28.03 | cool^tom | Fortunately the old system was still in place. |
| 16:29.09 | cool^tom | Guess I will have to get a networking expert. The resort is spread out in around 60 Acres. Will have to lay a fiber backbone for the cabling. |
| 16:29.34 | Katty | i would highly recommend fiber connecting networking closets |
| 16:29.45 | Katty | and networking closets feeding the rest of the buildings |
| 16:29.58 | Katty | you will probably need someone to trench if there's not already fiber ran |
| 16:30.25 | Katty | or vpns. |
| 16:30.53 | Katty | a hardware vpn might be better in this situation, if you're talking about lots of burried fiber and whatnot |
| 16:31.22 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 16:31.30 | Katty | that's under the assumption all the other little locations of their own internet setup |
| 16:31.40 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 16:31.45 | Katty | i'm sure your networking person will give you some options. |
| 16:31.49 | cool^tom | Fiber is the only possibility. The resort has cottages and the only cabling to the cottages are the resorts. |
| 16:32.00 | Katty | k |
| 16:32.17 | Katty | on that thought, some of the polycom phones have two network jacks int he back |
| 16:32.25 | Katty | they function similiar to a fax machine's line out port. |
| 16:33.10 | Katty | we've found them particularly useful in situations where people need an extra network drop, without having to drop it in |
| 16:33.10 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 16:33.31 | Katty | hi ruben |
| 16:33.45 | [TK]D-Fender | Katty: I don't think "network switch" is a complex concept for anyone capable of even logging into freeNode :) |
| 16:33.48 | cool^tom | Katty, Thanks a lot. |
| 16:34.23 | Katty | hi fender. |
| 16:34.34 | [TK]D-Fender | cool^tom: You have cat 3 for your existing PBX to all rooms, right? |
| 16:34.44 | [TK]D-Fender | Katty: Mew. |
| 16:35.07 | ManxPower-work | It sure would be nice if Outlook had a useful search option for searching a message. |
| 16:35.17 | cool^tom | [TK]D-Fender, We have the regular twisted pair. |
| 16:36.01 | ManxPower-work | cool^tom: You can answer "I don't know" rather than saying "regular twisted pair", which clearly indicates you don't know what kind of wiring you have. |
| 16:36.21 | [TK]D-Fender | cool^tom: What kind of functionality do you really neew? |
| 16:36.31 | ManxPower-work | Would that be Cat 2, Cat 3, Cat 4, Cat 5, or Cat 5e "regular twisted pair" |
| 16:37.01 | [TK]D-Fender | ManxPower-work: You can take it as cat3 basic for digital sets... |
| 16:37.23 | cool^tom | Just want voice. |
| 16:37.55 | cool^tom | Maybe voice menus, voice mail. |
| 16:38.01 | *** join/#asterisk wlirc123 (n=wlirc123@unaffiliated/ccbbaa) |
| 16:38.16 | ManxPower-work | I wish VLC's random play was actually random |
| 16:38.26 | ManxPower-work | or even shuffle |
| 16:38.28 | Katty | givges ManxPower-work a block of cheese. |
| 16:38.39 | *** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com) |
| 16:38.46 | [TK]D-Fender | cool^tom: Keep your existing wiring then and run a SIP gatway and use analog sets over it then |
| 16:39.20 | ManxPower-work | Katty: can I use that for searching a message in Outlook? |
| 16:39.29 | [TK]D-Fender | cool^tom: Mediatrix 1124/2124 or AudioCodes MP-124 should do |
| 16:39.43 | Katty | ManxPower-work: it's suppose to enchance your whine experience. |
| 16:39.51 | Katty | s/enchance/enhance/ |
| 16:40.00 | Katty | ManxPower-work: <3 |
| 16:40.14 | Katty | ManxPower-work: and no, i don't know of any useful addon for outlook either. it's just cranky. |
| 16:40.37 | wlirc123 | Hello. How does one work around the limitation of caller id set forcibly in call files? 1.6 causes the Callerid field in the call file to be rewritten so the * server domain is always appended even if the cid is already a valid url in <>'s |
| 16:41.25 | ManxPower-work | You don't put URLs in Callerid ionfo |
| 16:41.48 | ManxPower-work | wlirc123: what is the EXACT line you are using in your .call files to set the CallerID? |
| 16:41.56 | Katty | a call file just dumps you to another area. |
| 16:42.06 | ManxPower-work | Katty: only one leg |
| 16:42.16 | Katty | well yes, but still |
| 16:42.25 | Katty | but you should still be able to set callerid there before doing something |
| 16:42.27 | Katty | like dialing out. |
| 16:42.29 | ManxPower-work | (most callerid problems seem to be people putting in quotes, dashes, etc in their callerid info |
| 16:43.58 | wlirc123 | The url was literally for testing "demo" <demo@192.168.1.10> . The machine was only on a local net. |
| 16:44.22 | ManxPower-work | wlirc123: that is not valid |
| 16:44.24 | bmoraca | ManxPower-work, Google Desktop is great for searching outlook. very fast. |
| 16:44.36 | ManxPower-work | not another reference to google desktop |
| 16:44.53 | bmoraca | would you rather use Windows Search 4.0? |
| 16:45.10 | ManxPower-work | wlirc123: Valid Caller*ID would be something like: demo <12345> |
| 16:45.17 | wlirc123 | This causes * to rewrite it as <demo@192.168.1.10@192.168.1.10> |
| 16:45.19 | ManxPower-work | notice the lack of quotes and only digits in the ,. |
| 16:45.20 | ManxPower-work | <> |
| 16:45.41 | ManxPower-work | bmoraca: It's easier to just copy the message into a text editor for searching. |
| 16:46.09 | bmoraca | oh, you mean searching in a message you've already found...i thought you meant searching for a message based on contents of that message |
| 16:46.10 | Katty | Katty <1234567890> |
| 16:46.12 | [TK]D-Fender | wlirc123: Where is your call file and complete call debug for us to examine? |
| 16:46.14 | [TK]D-Fender | ~pb |
| 16:46.14 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:46.15 | [TK]D-Fender | ^^^^^^^^^ |
| 16:46.16 | ManxPower-work | wlirc123: I'd hope a callerid line like you used would cause Asterisk to just drop the call. |
| 16:46.21 | bmoraca | pardon the misunderstanding |
| 16:46.28 | Katty | bmoraca: NEVAR |
| 16:46.33 | Katty | bmoraca: IT IS TOTALLY UNACCEPTABULHS |
| 16:46.47 | bmoraca | Katty, seems to be the case here :P |
| 16:47.00 | ManxPower-work | bmoraca: outlook isn't very good at searching all messages either, but I just want to search the text of a message I already have. I'm trying to determine easily if a sales rep made a typoe later in his message. |
| 16:47.07 | Katty | bmoraca: just ignore them. i do all th etime. |
| 16:47.17 | ManxPower-work | Like Firefox CTRL-F |
| 16:47.43 | bmoraca | which version of outlook? |
| 16:47.48 | bmoraca | 2007 has a find feature |
| 16:48.03 | *** part/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 16:48.12 | wlirc123 | Actually the manual says o use url notation. And it has to be so. Putting in "demo" <demo> ends up as "demo" <demo@192.168.1.10> but i need it to show another domain. Short of patching * is there a way around it? |
| 16:48.19 | ManxPower-work | bmoraca: Outlook 2003 11.8313.8221 |
| 16:48.30 | ManxPower-work | wlirc123: What specific manual? |
| 16:48.51 | ManxPower-work | wlirc123: I expect you would have to patch Asterisk to accept that totally invalid Caller*ID. |
| 16:48.57 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 16:48.59 | bmoraca | don't have that one to check...though i suspect it does...but in outlook, CTRL+F is the shortcut for "Forward". check the "Edit" menu :) |
| 16:49.30 | ManxPower-work | bmoraca: no find option of any kind on the Edit menu |
| 16:49.56 | ManxPower-work | wlirc123: but I'm still wondering what specific manual told you that you can have URLs in Caller*ID number info. |
| 16:50.44 | ManxPower-work | The only official Asterisk docs are in the doc/ directory of the Asterisk source code. There is no real "manual". |
| 16:52.19 | Poincare | ManxPower-work: an ex-customer doesnt agree to that... he wants me to provide "the full documentation" |
| 16:52.45 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:52.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:52.48 | wlirc123 | Manx examples on the web and the docs. We hardly ever use numbers, almost everything is sip urls with names. Imho * tries to fqdn the url and fails. It looks like a bug. |
| 16:53.11 | wlirc123 | I will hack it later. |
| 16:53.17 | ManxPower-work | wlirc123: I wish you the BEST of luck. If you think it's a bug, report it on issues.digium.com |
| 16:53.53 | [TK]D-Fender | wlirc123: fromdomain <- |
| 16:53.54 | wlirc123 | The fact that it tries to qualify the url suggests a bug |
| 16:54.20 | wlirc123 | D-fender thanks |
| 16:54.23 | ManxPower-work | If you think it's a bug, report it on issues.digium.com |
| 16:55.48 | [TK]D-Fender | If you think its a bug, you should be showing more, and talking less |
| 16:57.32 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 16:57.41 | *** join/#asterisk dptom (n=dptom@c-24-6-168-8.hsd1.ca.comcast.net) |
| 17:00.46 | bmoraca | someone want to try to bounce a call off of a gateway for me? to make sure that it's properly filtering traffic? |
| 17:02.04 | voipmonk | sure |
| 17:02.11 | voipmonk | where? |
| 17:02.14 | voipmonk | what tech & codec? |
| 17:02.46 | bmoraca | going to send to a PM with the details |
| 17:06.26 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 17:06.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:07.27 | wlirc123 | Re: callerid, fromdomain: this patch is not inmainstream, right? https://issues.asterisk.org/view.php?id=1074 |
| 17:08.42 | wlirc123 | I came here to talk about it before doing more. |
| 17:08.54 | leifmadsen | wlirc123: huh? that is a VERY old bug -- it should have been put in Asterisk a few years ago |
| 17:09.13 | leifmadsen | although it is likely it has been refactored many times |
| 17:09.31 | wlirc123 | Well it could be bit rot |
| 17:10.08 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 17:10.15 | wlirc123 | Also the patch does not specifically address callfiles. |
| 17:10.46 | wlirc123 | Maybe this fell between the cracks. |
| 17:11.51 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 17:12.06 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:12.11 | wlirc123 | I mean addressing call files in the patch. |
| 17:12.47 | *** join/#asterisk Tim_Toady (n=moi@77.49.184.114.dsl.dyn.forthnet.gr) |
| 17:13.14 | leifmadsen | wlirc123: call files may not have existed... hard to say |
| 17:13.25 | leifmadsen | the link you're referencing is from 2004 |
| 17:13.28 | leifmadsen | it's not 2010 |
| 17:13.32 | leifmadsen | s/not/now/ |
| 17:13.33 | [TK]D-Fender | wlirc123: I'm seeing a lot more talk, and no show. |
| 17:13.42 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:13.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:14.22 | wlirc123 | I cant paste with my blackberry. Maybe monday. |
| 17:14.43 | [TK]D-Fender | wlirc123: I suggest asking for help when you're in a position to act on it... |
| 17:15.08 | [TK]D-Fender | wlirc123: Because right now this is an ice-fishing expedition.... in a skating rink |
| 17:15.10 | ariel_ | #vuc |
| 17:15.11 | leifmadsen | I'm still not sure what the question is |
| 17:15.23 | wlirc123 | I hacked the problem for now using rewriting... |
| 17:16.07 | ManxPower-work | leifmadsen: as far as I can tell he's annoyed that asterisk modifies the callerid number when it's set to a url. i.e. CallerID: timmy <robert_dobbs@192.168.0.1> in his .call file. |
| 17:16.12 | *** part/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 17:16.40 | [TK]D-Fender | ManxPower-work: which was never legal to begin with. |
| 17:16.47 | wlirc123 | D-fender, canadian anecdote ? ^^ |
| 17:16.53 | ManxPower-work | [TK]D-Fender: Yeah, I said that in the first place, but he didn't like the answer. |
| 17:17.11 | [TK]D-Fender | * is NOT a SIP PROXY. It is a B2BUA that interacts with a LOT more than jsut SIP. *'s handling of CID, etc is AGNOSTIC |
| 17:17.26 | ManxPower-work | [TK]D-Fender: I figured that getting his bug report closed by the developers might be a smack on the head to listen |
| 17:17.55 | [TK]D-Fender | ManxPower-work: I figure that it was closed OVER 5 YEARS AGO should have been a bigger hint |
| 17:18.00 | ManxPower-work | And on the off chance it really is a bug, at least it would be reported. |
| 17:18.23 | [TK]D-Fender | ManxPower-work: Keep on floggin' than equine! |
| 17:19.29 | wlirc123 | Ok, i'll patch * and keep it private. I don't want to offend the inquisition ^^. Thanks for the fromdomain hint. |
| 17:19.45 | p3nguin | This is interesting. Buchheit sells Cat 5E by the 1000' box. |
| 17:20.16 | *** part/#asterisk wlirc123 (n=wlirc123@unaffiliated/ccbbaa) |
| 17:20.22 | ManxPower-work | [TK]D-Fender: There isn't even a callerid field in SIP is there? |
| 17:20.33 | ManxPower-work | It's just faked based on the Contact: header? |
| 17:20.49 | [TK]D-Fender | ManxPower-work: No in those terms directly. Mixed between Asserted, From, etc |
| 17:21.23 | [TK]D-Fender | manxpowerrpid... |
| 17:21.32 | [TK]D-Fender | ManxPower-work: rpid... |
| 17:23.40 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:24.37 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:25.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:25.38 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 17:27.17 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:28.14 | *** join/#asterisk moy (n=moy@74.12.129.52) |
| 17:29.30 | *** join/#asterisk michael-i (n=michael-@208.53.198.95) |
| 17:35.16 | ManxPower-work | [TK]D-Fender: I never heard of a use for rpid, other than the use that does not actually work. |
| 17:35.45 | [TK]D-Fender | ManxPower-work: Inter-server CID that doesn't interfere with *'s peer matching. |
| 17:38.14 | ManxPower-work | [TK]D-Fender: Ah. The only use I had heard of was to have the phone display the Called Party callerid info, which does not actually work. |
| 17:39.08 | [TK]D-Fender | ManxPower-work: RPID != CPID |
| 17:39.10 | [TK]D-Fender | ~cpid |
| 17:39.11 | infobot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 17:39.13 | [TK]D-Fender | ^^^^6 |
| 17:39.33 | [TK]D-Fender | ManxPower-work: Still annoyingly a separate patch.... maybe by 2.4 we'll get it merged ;) |
| 17:39.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:40.08 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 17:41.20 | *** join/#asterisk Tim_Toady (n=moi@77.49.184.114.dsl.dyn.forthnet.gr) |
| 17:52.28 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 17:54.06 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 18:02.10 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:02.31 | *** join/#asterisk kihote (n=aefefuul@118.69.139.88) |
| 18:02.44 | kihote | pls help me ? Can i install opensips and mediaproxy on difference Server.I means, Server 1 will install opensips , server 2 will install mediaproxy s |
| 18:04.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:04.42 | ManxPower-work | ~ask |
| 18:04.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:06.46 | *** join/#asterisk puzzled (n=patrick@535335AA.cable.casema.nl) |
| 18:15.48 | [TK]D-Fender | kihote: And what does ANY of that have to do with *? I think you're in the wrong channel |
| 18:16.30 | *** join/#asterisk Get_The_Fish (n=IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 18:34.20 | Pan3D | I have a question on potatoes au gratin |
| 18:35.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:35.53 | jaytee | I have a question about over the counter medication. I bought a bottle of Bayer 325mg 100 ct at CVS today and after opening the box discovered the foil seal was broken already on the bottle and the box lid had been sealed with double sided tape. Should I return it or risk poisoning? |
| 18:36.28 | p3nguin | How far is it to CVS? |
| 18:36.41 | jaytee | it's on the way home from work |
| 18:37.08 | p3nguin | Do you need the aspirin right now, or can you wait until you pass by CVS again? |
| 18:37.13 | beek | Was it on sale and would it cost you additional $$$ to replace it? |
| 18:37.15 | jaytee | I can wait :-) |
| 18:38.01 | jaytee | While I did get a bottle that looks like it had been tampered with I was just asking in here for shits and giggles :-) |
| 18:38.13 | jaytee | I'm planning on bringing it back anyways |
| 18:38.26 | jaytee | but let's get back to the potatoes au gratin |
| 18:43.55 | beek | Is there a nice, juicy steak to go with it? |
| 18:43.56 | bmoraca | did Polycom discontinue the 650 phone? |
| 18:43.56 | ariel_ | ahh steak |
| 18:43.56 | Qwell | jaytee: take it back. you don't want to screw around with pills |
| 18:43.56 | Qwell | if taking it back isn't an option - toss it. |
| 18:43.56 | jaytee | Qwell, yeah I know. I'm old enough to remember the Tylenol scare back in the 80's |
| 18:43.56 | p3nguin | Ah, the '80s. |
| 18:43.57 | p3nguin | Good times. |
| 18:43.57 | p3nguin | Not. |
| 18:43.57 | jaytee | yeah, I burned all my polyester disco shirts and never looked back |
| 18:43.57 | Nugget | heh |
| 18:43.58 | jaytee | androgynous rock bands |
| 18:43.58 | ariel_ | bmoraca, not as far as I can see, http://www.polycom.com/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip650.html |
| 18:43.58 | bmoraca | ariel_, yeah, i see that...but my distributor doesn't list them anymore |
| 18:43.58 | jaytee | "Wow! That singer is sooo hot!!" "Dude, the singer is a guy!" "no way!" "way!" |
| 18:43.58 | [TK]D-Fender | jaytee: I miss Poison :p |
| 18:43.59 | p3nguin | and Ratt, and Great White, and ... |
| 18:43.59 | jaytee | [TK]D-Fender, yea, well, in retrospect there were some great bands back then |
| 18:43.59 | ariel_ | bmoraca, just check with mine and they have them still in inventory and in stock. |
| 18:43.59 | jaytee | Ah, Great White! Pyrotechnics anyone? |
| 18:43.59 | [TK]D-Fender | jayIf I played any more hair-metal I should buy stock in Revlon |
| 18:44.00 | bmoraca | ariel_, which distributor do you use? |
| 18:44.00 | ariel_ | Ingram |
| 18:44.00 | carrar | Save all your Love!! |
| 18:44.00 | bmoraca | really? |
| 18:44.00 | bmoraca | weird |
| 18:44.00 | jaytee | one of my friends back home lost his sister in the nightclub fire |
| 18:44.00 | bmoraca | Techdata doesn't list them anymore |
| 18:44.00 | [TK]D-Fender | jaytee: Metallica's pyrotechnics were far better ;) |
| 18:44.00 | bmoraca | i'll check Ingram |
| 18:44.00 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:44.00 | ariel_ | just checked insight they also have them |
| 18:44.01 | bmoraca | bad day for buying... "The U.S. Ingram Micro web site is not available" |
| 18:44.07 | [TK]D-Fender | bmoraca: SELL HIGH! |
| 18:44.29 | ariel_ | humm, my GP inventory is active to them and I see them. |
| 18:44.51 | Katty | :> |
| 18:44.53 | Katty | HERROES. |
| 18:44.59 | bmoraca | great plains? |
| 18:45.00 | jaytee | hiya! |
| 18:45.18 | ariel_ | bmoraca, yes that is what my corp uses. |
| 18:45.27 | bmoraca | i hate great plains |
| 18:45.29 | bmoraca | uhg |
| 18:45.38 | bmoraca | have you ever looked at the database? there has to be 5000 tables |
| 18:45.39 | ariel_ | so do I, but I am just a little fish here |
| 18:45.48 | bmoraca | and none of them are intitively named |
| 18:46.07 | ariel_ | bmoraca, no I have not and don't care too. |
| 18:46.38 | bmoraca | i've had to write a few apps that pull data from it...not fun |
| 18:47.25 | ariel_ | well, I see 10 on hand in Miami w/h, and 2 in there NY, don't see much more inventory other then those. |
| 18:48.07 | bmoraca | techdata has 7 550s in CA...but no more on order |
| 18:48.15 | ariel_ | you said 650 |
| 18:48.51 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:49.08 | ariel_ | my GP does not show future items.... argh I hate gp every time I need to look I have to enter the cap code... |
| 19:04.02 | *** join/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 19:05.06 | aces1up | i just had an issue with one our asterisk servers where the cpu was throttled at 100% according to freepbx status, now i had to reboot to repair just to get it back up cause i was getting hammered with calls, so does that freepbx show asterisk using 100% cpu or is that just the system? |
| 19:05.34 | ManxPower-work | ~freepbx |
| 19:05.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:10.19 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:11.30 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) |
| 19:12.29 | ph8 | hi everyone, i'm trying to configure my voicemail - i actually thought i'd done it properly the first time round when I configured my asterisk server a few months ago - it appears not! in /etc/asterisk/voicemail.conf I have: [myvoicemail]\n101 => 1234,My name,my@email.com,my@otheremail.com |
| 19:13.00 | voipmonk | ph8, how about you you use pastebin.ca |
| 19:13.06 | ph8 | i was just thinking that :) |
| 19:13.10 | ph8 | thx |
| 19:13.50 | ManxPower-work | ph8: so you have something like Voicemail(1234@myvoicemail) or VoicemailMain(@myvoicemail)? |
| 19:14.19 | ManxPower-work | But you might actually want to tell us the actual problem first. |
| 19:14.30 | ph8 | yes |
| 19:14.32 | ph8 | like this: http://pastebin.com/m502a9f32 |
| 19:14.39 | ph8 | sorry, i dial 666 and don't get voicemail as i expect |
| 19:14.52 | ph8 | I get a 404 not found / wrong number / beeping tone thing |
| 19:15.02 | ph8 | my phone says 404 but i'm not sure if that's an asterisk thing |
| 19:15.08 | ph8 | it's a grandstream so it could be anything :p |
| 19:15.13 | ManxPower-work | ph8: the phone you are calling from does not have access to the my-voicemail extensions.conf context |
| 19:15.26 | ph8 | oh i see |
| 19:15.30 | ph8 | how can i grant that access? |
| 19:15.44 | ManxPower-work | ph8: include => my-voicemail |
| 19:15.52 | ManxPower-work | you're not using a GUI, are you? |
| 19:16.03 | ph8 | no |
| 19:16.13 | voipmonk | what context are you using that is missing my-voicemail ? |
| 19:16.18 | ManxPower-work | then you should read up on include => |
| 19:16.25 | ph8 | well my phone is on as SIP/Ph0 |
| 19:16.33 | ph8 | just got to figure out where to put that include |
| 19:16.36 | voipmonk | thats great, what context does it use? |
| 19:16.37 | ph8 | will do, i've got a book here |
| 19:17.09 | ph8 | ah i see |
| 19:17.11 | p3nguin | Each SIP device has a definition in sip.conf, where you define a context. |
| 19:17.13 | ph8 | i have a context as my-outbound |
| 19:17.14 | ManxPower-work | ph8: Great! So you can see the sip.conf entry for [Ph0] has a context= line, that is where the call will land in the dislplan |
| 19:17.15 | ph8 | which is my IAX stuff |
| 19:17.26 | ph8 | maybe i've misunderstood the context there |
| 19:17.35 | ph8 | so the my-outbound could have the voicemail included AND an outbound trunk |
| 19:17.39 | ph8 | and that would be what a context is |
| 19:17.44 | ph8 | right? |
| 19:17.49 | ManxPower-work | ph8: contexts are one of the top most important things to understand. |
| 19:17.56 | ph8 | ok i'll look it up, thanks for the help |
| 19:18.03 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 19:18.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:18.06 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 19:18.53 | p3nguin | You should be creating an hierarchical structure with your contexts. |
| 19:19.35 | ph8 | a sort of component based context assembly? so all phones include a general context which includes outbounds, voicemails etc? |
| 19:20.11 | p3nguin | For example, my phones have contexts of "phones" and the [phones] context in extensions.conf includes outbound and internal. That allows the phones to have access to the trunk as well as other internal phones. |
| 19:20.33 | ph8 | ah i see |
| 19:20.38 | ph8 | cool |
| 19:21.23 | p3nguin | Then in inbound context can include internal, too, but never outbound... so that keeps people from being able to call in and get a line back out. |
| 19:22.31 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 19:26.33 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 19:26.54 | Katty | http://www.uniquedaily.com/wp-content/uploads/2009/10/fendsoff-bullying-birds.jpg |
| 19:28.23 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 19:29.27 | mesfet | Hi all. Just a question: does Asterisk 1.6.1.12 support SIP messaging (used by some phones to exchange instant messaging)? |
| 19:33.26 | [TK]D-Fender | mesfet: No. |
| 19:36.23 | mesfet | [TK]D-Fender: Ok, many thanks! |
| 19:37.05 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 19:45.40 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:48.33 | bmoraca | woo...AS5400 live and taking calls now |
| 19:49.19 | Katty | calls it |
| 19:53.33 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 19:53.43 | ariel_ | AS5400 are ok, and can do allot, but over price system. |
| 19:53.54 | bmoraca | that's why i bought it used |
| 19:54.17 | ariel_ | we have many of them here |
| 19:54.50 | bmoraca | it's a bear to set up, but i like it. |
| 19:55.15 | voipmonk | heh |
| 19:56.07 | ariel_ | yuk |
| 19:56.20 | ariel_ | if you don't get the dial pairs in order they really mess up |
| 19:57.14 | ariel_ | And they are not so great at transcoding |
| 19:59.08 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:03.17 | bmoraca | what problem do they have transcoding? |
| 20:03.43 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:03.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:03.46 | bmoraca | i'm using mine as a media gateway between asterisk and the PSTN...so it shouldn't actually be doing any transcoding |
| 20:06.03 | *** join/#asterisk moy (n=moy@74.12.129.52) |
| 20:07.25 | ariel_ | bmoraca, that is what we use them mostly for, How are you connecting them to asterisk? |
| 20:07.35 | bmoraca | SIP |
| 20:07.56 | bmoraca | most frustrating thing evar, lol |
| 20:08.42 | ariel_ | We connect our as H323 Gateways to our provider then via E1 boards to our asterisk systems |
| 20:09.01 | bmoraca | backwards! |
| 20:09.13 | ariel_ | our providers are still on h323 |
| 20:11.03 | *** join/#asterisk ascenseur (n=ascenseu@86.24.20.72) |
| 20:11.27 | ascenseur | does anyone know whether this would be a good phone to use with asterisk 1.5? http://cgi.ebay.co.uk/3Com-NBX-100-1102-Grey-Display-Speaker-Phone-3C10121_W0QQitemZ260522990868QQcmdZViewItemQQptZUK_Computing_Networking_SM?hash=item3ca8615d14#ht_556wt_1167 |
| 20:12.00 | ascenseur | *sorry - 1.6 |
| 20:12.37 | carrar | ascenseur, use a Polycom |
| 20:12.53 | ascenseur | carrar: thanks - any reason why? |
| 20:13.11 | bmoraca | ascenseur, 3com SIP phones aren't standard. |
| 20:13.14 | ariel_ | Polycom's are the best and work everytime |
| 20:13.27 | carrar | Polycom 430, 601, 650 |
| 20:13.43 | carrar | 3Com NBX 1102 is not even a SIP phone from what I can tell |
| 20:13.50 | ariel_ | 330/331 are nice low cost ones |
| 20:13.55 | ascenseur | yes - but the deciding factor is price im afraid! |
| 20:14.11 | carrar | the deciding factor should be what works |
| 20:14.40 | ariel_ | cost is related to what works and how long does it take to get it to work |
| 20:14.52 | carrar | ascenseur SPA-942 is ok also |
| 20:14.57 | carrar | LinkSys SPA-942 |
| 20:15.16 | carrar | You can find those phones cheaper refurb |
| 20:15.19 | bmoraca | i want to try out the new SPA500 phones from Cisco |
| 20:15.21 | carrar | same with polycom |
| 20:15.25 | ariel_ | looks at one on his desk and the IP331 and says IP331 far better |
| 20:15.36 | ascenseur | £105 in the UKâ¦. hmmmm |
| 20:15.47 | carrar | I like Cisco phones too but they lack in features that some poeple want |
| 20:16.05 | carrar | I use 7941 on my desk |
| 20:16.21 | ascenseur | so, you dont think that the one on ebay was SIP compatible? |
| 20:16.31 | carrar | 7970 work great too |
| 20:16.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:16.44 | carrar | ascenseur, doesn't look like it |
| 20:16.52 | ascenseur | aah⦠thanks anyway |
| 20:17.33 | carrar | You can find old 7940 & 7960's cheap |
| 20:17.45 | ascenseur | 7910? |
| 20:17.54 | ascenseur | i really only need a basic phone |
| 20:17.59 | carrar | I would not use that |
| 20:18.16 | ascenseur | any specific reason? |
| 20:18.22 | carrar | PITFA |
| 20:18.33 | carrar | and it's SCCP only |
| 20:18.34 | carrar | not SIP |
| 20:18.53 | ascenseur | aah |
| 20:19.03 | carrar | need to do your homework |
| 20:19.16 | ascenseur | i am! |
| 20:19.19 | carrar | this for play or work pbx? |
| 20:19.50 | ascenseur | play mainly, maybe work when its reliable enough |
| 20:19.57 | carrar | You can just use xlite free software phones if you want cheap (FREE) |
| 20:20.51 | carrar | 7940's are $27 on ebay |
| 20:20.58 | carrar | get 1 or two of those |
| 20:21.10 | p3nguin | carrar: I use 7900 series phones with SIP images... what "features" are missing? What would I get if I converted them to SCCP? |
| 20:21.19 | ascenseur | although Xlite is good, its not mac compat. (using Telephone, so thats all good) |
| 20:21.34 | carrar | p3nguin, you can't have different incoming calls have different ring types |
| 20:21.36 | p3nguin | Use zoiper for Mac -- X-Lite sucks. |
| 20:21.39 | ascenseur | Unfortunatley, the UK ebay isnt quite so brimming with deals.. |
| 20:21.42 | carrar | auto answer on some calls |
| 20:21.44 | carrar | etc... |
| 20:21.59 | ascenseur | p3nguin: using Telephone 0.1.4 all good |
| 20:22.03 | carrar | intercom feature |
| 20:22.16 | carrar | Cisco lacks A LOT of programming features |
| 20:22.23 | carrar | but Cisco is a great working phone |
| 20:22.27 | carrar | looks nice too |
| 20:22.33 | p3nguin | A lot of that is done in the Call Manager, though. |
| 20:22.51 | bmoraca | p3nguin, you lose BLF, too, at least on the 79x0 phones |
| 20:22.52 | carrar | Which we are using Asterisk |
| 20:22.53 | p3nguin | If I went to an SCCP image and used chan_skinny, will I regain those features? |
| 20:22.55 | ascenseur | another unit: Cisco 7906G - any ideas on that |
| 20:22.57 | carrar | not CCM |
| 20:23.25 | carrar | ascenseur, 7940 or HIGHER |
| 20:23.39 | carrar | 7940/7960 are the easyest |
| 20:23.45 | ascenseur | carrar: ah, ok |
| 20:23.50 | bmoraca | the 7912G's work well, but are not simple to configure |
| 20:24.01 | p3nguin | The 7912 doesn't have a speakerphone with the SIP image (it is only a monitor speaker)... does the SCCP image make it into a speakerphone again? |
| 20:24.03 | carrar | 7941 or 7970 higher are XML based, more of pain but work |
| 20:24.10 | bmoraca | and the SIP image has a stupid, stupid bug which makes you unable to ring the phone for more than 19 seconds, lol |
| 20:24.26 | p3nguin | bmoraca: SIP image on what? |
| 20:24.28 | bmoraca | p3nguin, no, it doesn't have a mic |
| 20:24.34 | ascenseur | thanks |
| 20:24.35 | bmoraca | p3nguin, the 7912G |
| 20:24.36 | *** part/#asterisk ascenseur (n=ascenseu@86.24.20.72) |
| 20:24.58 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 20:25.03 | carrar | http://cgi.ebay.com/Two-Polycom-SoundPoint-IP-430-SIP-phones-2201-11402-001_W0QQitemZ290388502209QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item439c80e6c1 |
| 20:25.07 | carrar | get that and be done with it |
| 20:25.31 | bmoraca | if only that were buy-it-now, lol |
| 20:25.33 | p3nguin | Hmm. I have a 7912G right here. I should test your theory. |
| 20:25.56 | carrar | There is a lot of 25 buy it now :) |
| 20:26.23 | p3nguin | 36 seconds and counting |
| 20:26.35 | bmoraca | weird |
| 20:26.51 | bmoraca | do you have a voicemail box configured with that phone? |
| 20:26.52 | p3nguin | It rang for 54 seconds before hitting followme. |
| 20:27.01 | p3nguin | yes |
| 20:27.11 | bmoraca | with the messages uri in the config? |
| 20:27.12 | p3nguin | Let me check the SIP version. |
| 20:27.36 | p3nguin | It sounds like you have configured a timeout in the config. |
| 20:27.39 | bmoraca | because any time that I have that, it gives me a weird-ass beep and shoots the call off to the voicemail application. that said, i didn't really try to solve it...but whatever |
| 20:27.49 | p3nguin | It's a config option. |
| 20:27.49 | bmoraca | p3nguin, none that i could see |
| 20:27.50 | ariel_ | I have no issues with BLF on any of the Cisco with sip images and they can ring for every. we have a few 7912G, 7940G 7960G and 7970's |
| 20:28.31 | ariel_ | there a pain to configure but work fine, speaker phone part of the polycom's are far better |
| 20:28.51 | bmoraca | i'm trying to get away from the 7940s...the screens make them look like they're from 1995 |
| 20:29.03 | bmoraca | i'd like to get a couple of the SPA504Gs |
| 20:29.09 | carrar | go 7941 7970 |
| 20:29.12 | bmoraca | and see about moving to them |
| 20:29.17 | bmoraca | i can't stand the 7941s |
| 20:29.21 | bmoraca | too many weird SIP problems |
| 20:29.29 | bmoraca | like not being able to connect through a NAT |
| 20:29.33 | ariel_ | but for the price the polycom's are a far better phone |
| 20:29.35 | bmoraca | deal breaker for my application |
| 20:30.01 | bmoraca | ariel_, yeah, but my boss prefers the Cisco name...thinks people respond better to it, even if it is a worse product |
| 20:30.01 | ariel_ | besides you need to get the license for the Cisco's to use them as sip. |
| 20:30.27 | carrar | All part of doing biz |
| 20:30.40 | bmoraca | yep, which is why i just got 50 more 7940s in |
| 20:30.43 | p3nguin | # Parameter: ForwardToVMDelay |
| 20:30.47 | ariel_ | bmoraca, even if you get a Cisco Sip phone on ebay your suppose to buy the sip lic to use them |
| 20:31.01 | p3nguin | # Description: Number of seconds before forwarding a call to the |
| 20:31.03 | p3nguin | # VoiceMailNumber, if configured. |
| 20:31.06 | p3nguin | ForwardToVMDelay:120 |
| 20:31.19 | p3nguin | There's why mine does not send to VM at 19 seconds. |
| 20:31.24 | bmoraca | i'll give that a try |
| 20:31.39 | bmoraca | i didn't see that in any of the docs or samples i've found for this phone |
| 20:32.00 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 20:32.01 | p3nguin | It's in the gkMAC config file. |
| 20:32.05 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 20:32.33 | ManxPower-work | "I'm sorry your phone sucks. We were going to use Polycom, some of the best phones on the market, but the boss liked the Cisco name so we used them." |
| 20:32.33 | bmoraca | i never got around to actually downloading the firmware package from Cisco |
| 20:33.07 | bmoraca | ManxPower-work, it's not that I CAN'T sell the Polycoms...it's that he refuses to push them and will only demo the Ciscos for people |
| 20:33.08 | p3nguin | It could also be the SigTimer parameter, bits 14-19. |
| 20:33.27 | ariel_ | get a new boss |
| 20:33.32 | p3nguin | Parameter: SigTimer 14-19 RING TIMEOUT Timeout in ringing the phone after which the incoming call is rejected |
| 20:33.35 | bmoraca | ariel_, if only, lol |
| 20:33.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:33.40 | ManxPower-work | bmoraca: I had forgotten you sell hardware. Sucks. |
| 20:34.15 | ariel_ | secret demo with the polycom's speaker on and the Cisco on the other you will see the diff |
| 20:34.38 | *** join/#asterisk prometheanfire (n=mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net) |
| 20:34.55 | bmoraca | this is the same boss that made me set up a hosted PBX for a company with 29 phones and wouldn't let me buy the g729 licenses for them |
| 20:35.00 | prometheanfire | can I use uuids to save files with the recievefax aplication? |
| 20:35.22 | bmoraca | they had to threaten to stop paying their $900/mo bill in order for him to authorize the $290 purchase of the licenses |
| 20:38.47 | ManxPower-work | prometheanfire: your question makes no sense |
| 20:39.10 | ManxPower-work | You can do anything you want with the fax file name. Just code your dialplan correctly |
| 20:39.18 | p3nguin | bmoraca: For what it's worth, I found those parameters in the config file for SIP Version 8.0.0(060111A). |
| 20:39.49 | ManxPower-work | bmoraca: I wonder how many thousands of dollars a year your boss gets in "incentives" from Cisco |
| 20:39.53 | p3nguin | which is the one I use. |
| 20:40.43 | leifmadsen | bmoraca: you need a new boss :) |
| 20:41.24 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 20:41.49 | prometheanfire | basically I need a way of generating uuids or at least uids in asterisk so I don't overwrite files |
| 20:42.31 | ManxPower-work | prometheanfire: ${UNIQUEID} |
| 20:42.38 | prometheanfire | that builtin? |
| 20:43.13 | ManxPower-work | prometheanfire: I think you need to learn Asterisk. Yes, it's one of the standard channel variables as documented in /path/to/src/asterisk/doc/tex/channelvariables.tex |
| 20:43.18 | prometheanfire | and does that stay the same throughout the call |
| 20:43.28 | ManxPower-work | prometheanfire: Go read. |
| 20:44.35 | ManxPower-work | ~book |
| 20:44.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:44.42 | ManxPower-work | better yet, go read the book |
| 20:44.43 | prometheanfire | I have that book |
| 20:44.49 | voipmonk | good start |
| 20:44.51 | prometheanfire | I've read it |
| 20:44.53 | voipmonk | read it a few times :) |
| 20:45.12 | ManxPower-work | prometheanfire: good, now look at the OFFICIAL documentation that is in the Asterisk source doc/ directory |
| 20:46.47 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:46.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:49.25 | ariel_ | people have it so good now, we have lots of books and wiki's, with info about asterisk, I remember starting and the general reply was go read the code to see what it does. |
| 20:49.56 | prometheanfire | why back in my day... |
| 20:50.08 | voipmonk | 2001 or 2002? |
| 20:50.16 | prometheanfire | heh |
| 20:50.31 | prometheanfire | I was still in middle school back then |
| 20:50.33 | ariel_ | my first setup was with asterisk .5 |
| 20:51.10 | prometheanfire | I would have been 14 in 2001 |
| 20:51.42 | ariel_ | I have children older then you |
| 20:52.16 | prometheanfire | lol |
| 20:54.02 | ManxPower-work | ~answers |
| 20:54.02 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 20:54.45 | prometheanfire | I even have that one agi book |
| 20:55.26 | prometheanfire | I just try not to touch asterisk or any software, just manage it, not configure it |
| 20:56.06 | [TK]D-Fender | prometheanfire: What does "managing without configuring" imply? |
| 20:56.08 | *** join/#asterisk andres833 (n=andres83@200.119.44.131) |
| 20:58.09 | prometheanfire | I try to manage the servers, not configure them. I feel that I would have to specialize in to many areas to do that effectivly |
| 20:58.42 | ariel_ | you have to know how to configure them first before you are able to just manage |
| 20:59.48 | prometheanfire | I can do basic stuff for it all, I need time :D |
| 21:00.56 | ariel_ | ever heard of a lab? test system? a tinker room? |
| 21:01.24 | prometheanfire | I have that and am learning |
| 21:01.30 | ariel_ | good |
| 21:01.41 | prometheanfire | I have a full VM setup at home with a few things |
| 21:02.24 | prometheanfire | I really need to get a voip service provider so I can start working on that more. |
| 21:02.36 | prometheanfire | I could probably just get a free one from work |
| 21:03.44 | ariel_ | or you can get ones like ipkall for testing inbound |
| 21:04.07 | ariel_ | it's free |
| 21:04.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:05.23 | prometheanfire | true |
| 21:05.44 | prometheanfire | I kinda wanna do a dialajoke thing |
| 21:08.16 | voipmonk | well you kinda wont get it done with a kinda sorta attitude, just do it |
| 21:08.45 | voipmonk | step by step woulda coulda is a waste of time, dive in head first and get it done ;) |
| 21:08.49 | Katty | kinda sorta maybe possibly thinking about the idea of perhaps doing a dialajoke thing |
| 21:08.59 | voipmonk | hah |
| 21:09.00 | voipmonk | ! |
| 21:09.37 | Katty | i wanna go home. |
| 21:09.57 | Katty | no kinda sorta maybe possiblies. |
| 21:10.50 | prometheanfire | heh |
| 21:12.14 | ariel_ | I also want to go home |
| 21:16.05 | mocker | signs up for a trail at bandwidth.com |
| 21:16.15 | mocker | trial even. |
| 21:16.19 | prometheanfire | they do trials now? |
| 21:17.20 | mocker | eh, sign up and then have an out on the contract. |
| 21:18.13 | ariel_ | get a 5 dollar voipjet account for outbound testing |
| 21:26.02 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:28.19 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 21:31.20 | prometheanfire | the fax docs say to check out the KB for fax stuff, the link they put in the readme links to a kb category with not kb articles in it :D |
| 21:32.29 | ManxPower-work | "fax docs" |
| 21:32.35 | ManxPower-work | What specific file says that? |
| 21:34.11 | [TK]D-Fender | BBIAB |
| 21:34.20 | prometheanfire | http://downloads.digium.com/pub/telephony/fax/README |
| 21:34.32 | prometheanfire | and the digium-fax module |
| 21:34.48 | ariel_ | I have not played with that yet |
| 21:34.57 | ariel_ | have had no need for it |
| 21:35.43 | prometheanfire | big boss man wants a fax to email server, so I made one with freeswitch. It works. Now he wants it with asterisk. |
| 21:36.54 | ariel_ | why? |
| 21:37.03 | ariel_ | if it works |
| 21:37.05 | ariel_ | don't change |
| 21:37.20 | prometheanfire | support, he wants someone to yell at if it goes bad |
| 21:38.08 | ManxPower-work | prometheanfire: Who, exactly, will he yell at if you switch to Asterisk? |
| 21:38.21 | ManxPower-work | I doubt Digium support would put up with it. |
| 21:38.46 | ariel_ | fax and faxing fall in the 80/20 rule |
| 21:38.52 | ariel_ | no matter what you do |
| 21:39.07 | prometheanfire | faxing fail you mean? |
| 21:39.32 | ariel_ | 80% works fine, but when you really need it, it falls in the 20% that does not work |
| 21:39.33 | prometheanfire | ManxPower-work: we have a support contract, it is more a piece of mind type of thing |
| 21:39.55 | ariel_ | support contact from digium? |
| 21:40.00 | prometheanfire | I have had about 95% good at least with the freeswitch solution |
| 21:40.10 | prometheanfire | yep, plus we are a reseller and that bit |
| 21:40.16 | ariel_ | good reason to stay |
| 21:41.10 | prometheanfire | I know but he wants proof so I'll give him it |
| 21:41.24 | ariel_ | lab |
| 21:42.41 | *** join/#asterisk mspiceland (n=mike@nat/digium/x-sirnfcmvefnxtnti) |
| 21:42.46 | prometheanfire | that is what I am doing now |
| 21:42.53 | prometheanfire | gonna send a test fax now |
| 21:43.00 | ariel_ | your keeping the one that works right? |
| 21:43.09 | prometheanfire | yep |
| 21:43.13 | prometheanfire | diferent VMs |
| 21:43.14 | ariel_ | good man |
| 21:43.25 | ariel_ | oh on vm's |
| 21:43.51 | prometheanfire | we have high call volumes on VMs and it is good :D |
| 21:43.55 | ariel_ | hides as he knows vm and asterisk and faxing are far less reliable |
| 21:44.00 | prometheanfire | I did that side and am proud of it |
| 21:44.46 | ariel_ | actually I don't use any VM's, but I do use XEN |
| 21:45.18 | prometheanfire | we are using vmware right now but I think I am gonna push for kvm/redhat |
| 21:45.55 | prometheanfire | cheaper and we get official redhat licence/support for unlimited VMs instead of centos |
| 21:46.07 | ariel_ | why |
| 21:46.19 | ariel_ | Citrix Xen in my view is far better |
| 21:46.20 | *** join/#asterisk DMeloUK (n=Administ@64.129.95.226) |
| 21:47.00 | prometheanfire | kvm is being pushed by redhat and the kernel dev people |
| 21:47.14 | ariel_ | yes and every year you pay |
| 21:47.35 | prometheanfire | true |
| 21:48.58 | ariel_ | We use Xen 5.5 here and it's been working fine, we are moving all our VM's to the Xen cluster |
| 21:49.03 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:49.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:49.28 | prometheanfire | live migration and fault tolerance? |
| 21:49.35 | ariel_ | we can even move a xen from one server to another while it's running |
| 21:49.46 | prometheanfire | we can in kvm too :D |
| 21:49.47 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 21:49.57 | ariel_ | did I say it's free |
| 21:50.06 | prometheanfire | all that kvm is missing is fault tolerance |
| 21:50.12 | ariel_ | we have that |
| 21:50.15 | prometheanfire | ariel_: linky link is linked? |
| 21:50.31 | ariel_ | google Citrix xen 5.5 |
| 21:50.38 | prometheanfire | ariel_: running the same VM on 2 machines so if it goes down you don't loose calls? |
| 21:50.51 | ariel_ | you loose calls yes |
| 21:51.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:52.06 | prometheanfire | I didn't loose calls when I tested fault tolerance with vmware |
| 21:52.09 | prometheanfire | or kvm |
| 21:52.18 | prometheanfire | oh wait, not kvm |
| 21:55.15 | ariel_ | any calls that are on a box that goes down will go down, but there are ways you can build minimal down time |
| 21:57.14 | ariel_ | Enswitch is a combo of OpenSer/asterisk and mysql that has a fairly nice fault recover setup, but any calls you might have on a system's pri lines or analog pots that goes down you will loose |
| 21:57.30 | prometheanfire | ariel_: when I tested it on vmware there was 2-3 sec of silence then the calls picked back up agian |
| 21:58.13 | ariel_ | some times you can do that via sip, and canreinvite |
| 21:58.40 | prometheanfire | it was a basic asterisk instalation with a call going into a moh |
| 21:58.59 | *** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) |
| 21:59.24 | prometheanfire | the VM runs on both servers at once and if the primary fails the seccondary picks it up at the same point |
| 21:59.35 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 22:02.58 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:03.03 | ariel_ | I am really happy you were able to get that working. But your going to hit the 80/20 rule with it. |
| 22:03.26 | ariel_ | It's that time of the day that I am out of here folks, hope everyone has a great weekend. |
| 22:03.42 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 22:08.03 | prometheanfire | alright, thanks for your help |
| 22:08.09 | prometheanfire | cya, I'm out too |
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| 22:31.21 | nvicf | hello guys, I have a question, I have two numbers (incoming) and I want when those numbers are calling to redirect the call to a given extension, is that possible? i've read it was but no idea how |
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| 22:34.41 | voipmonk | [ yes |
| 22:34.48 | voipmonk | its very possible |
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| 22:39.37 | DarkFibre_Drittb | voipmonk: For a relative newbie would you recommend apstel visual designer to test dialplan generation? |
| 22:41.45 | voipmonk | no |
| 22:41.53 | voipmonk | i would recommend reading the book and work from there |
| 22:41.57 | voipmonk | the future of telephony |
| 22:42.03 | voipmonk | no additional tools required |
| 22:42.15 | voipmonk | and there are docs that come with asterisk , too |
| 22:46.18 | DarkFibre_Drittb | I have that book |
| 22:46.42 | DarkFibre_Drittb | but sometimes its nice to have a visual representation for planning with clients |
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| 22:49.44 | voipmonk | I see |
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| 23:16.27 | ManxPower-work | DarkFibre_Drittb: If it will help you visualize dialplan code that you write, then that's great. |
| 23:17.24 | ManxPower-work | If you are trying to use it to visually design a dialplan, then all you are learning is how to make machine generated code, which bears little resemblance to any humans dialplan. |
| 23:17.28 | carrar | haha |
| 23:17.29 | carrar | http://9gag.com/photo/16103_full.jpg |
| 23:17.32 | carrar | WS |
| 23:24.22 | DarkFibre_Drittb | manxpower-work - would it read it a human written dialplan and diagram it? |
| 23:24.38 | DarkFibre_Drittb | I am just not very good with diagrams |
| 23:26.19 | DarkFibre_Drittb | and for documentation the operations people want a pretty picture |
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| 23:26.44 | DarkFibre_Drittb | spending hours drawing boxes seems to be a waste of time :) |
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| 23:37.48 | VxJasonxV | Anyone have any documentation on res_phoneprov? Google is turning up mostly checkin pages, mailing lists, ml resyndicates... etc. etc. etc. |
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| 23:44.29 | carrar | DarkFibre_Drittb, just draw clouds and arrows |
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