| 00:03.06 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:03.16 | *** join/#asterisk sjobeck (~sjobeck@2002:ad0c:b8a1:0:226:bbff:fe05:4a1f) |
| 00:04.46 | *** part/#asterisk sjobeck (~sjobeck@2002:ad0c:b8a1:0:226:bbff:fe05:4a1f) |
| 00:07.09 | *** join/#asterisk Whtsup (~sssi@WimaxUser372-58.wateen.net) |
| 00:07.13 | Whtsup | hello |
| 00:07.47 | Whtsup | i have setup 2n voice blue light with asterisk |
| 00:08.07 | Whtsup | but i cannot here voice on outgoing calls |
| 00:08.10 | Whtsup | any idea ? |
| 00:10.16 | hardwire | Amorsen: thanks. This should help me out a bit |
| 00:10.25 | hardwire | At one point I knew that. |
| 00:13.17 | Whtsup | ??? |
| 00:14.07 | hardwire | Whtsup: I wudn't talkin to yeh. |
| 00:17.11 | Whtsup | i m asking something |
| 00:17.15 | Katty | Qwell: ping |
| 00:17.28 | Qwell | pong |
| 00:17.45 | Katty | Qwell: ohai |
| 00:17.47 | Qwell | uno momento |
| 00:17.50 | Katty | Qwell: k |
| 00:18.31 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 00:21.36 | *** part/#asterisk RichardLynch (~RichardLy@c-98-193-36-91.hsd1.il.comcast.net) |
| 00:24.08 | hardwire | Whtsup: wait for an answer. |
| 00:24.19 | hardwire | post to the list |
| 00:24.24 | hardwire | call up voice blue people |
| 00:25.11 | Katty | Qwell: YOUR MINUTE IS UP |
| 00:27.28 | Qwell | I said moment! |
| 00:27.32 | p3nguin_ | moment = minute? |
| 00:28.56 | p3nguin_ | I've often wondered what does it really mean when people say things like "a few moments." If the period is uninterrupted, wouldn't it be a single moment? |
| 00:30.13 | Qwell | Katty: psst! |
| 00:30.20 | Qwell | now who's waiting, eh?! |
| 00:30.42 | Katty | oh |
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| 00:48.17 | afink | could someone please help me with sip registration to my itsp? I just upgraded to 1.6 and I can't figure out why I keep getting timeouts the SIP response 400 "Rejecting Stale Request" |
| 00:52.28 | Katty | bmoraca_work: pingggggggggggggggggggggggggg |
| 00:55.57 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
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| 01:04.49 | ketin | http://www.youtube.com/watch?v=FmiS0RNNp28 |
| 01:09.47 | Octothorpe | Is anyone successfully running Asterisk on Mac OS X? |
| 01:12.11 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
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| 01:41.05 | ManxPower-work | afink, Have you read the UPGRADE*.txt files? |
| 01:41.08 | ManxPower-work | ~answers |
| 01:41.09 | infobot | methinks answers is 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 |
| 01:50.11 | *** join/#asterisk bobJR (~bob@unaffiliated/bobjr) |
| 01:59.51 | ketin | Porn? |
| 01:59.55 | ketin | ~answers |
| 01:59.56 | infobot | from memory, answers is 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 |
| 02:00.03 | ketin | Meh. |
| 02:00.24 | ketin | Bobby, have you been wanking it in the toolshed again? |
| 02:00.29 | ketin | ~answers |
| 02:00.30 | infobot | answers is probably 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 |
| 02:01.32 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
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| 02:18.21 | encinoman | Im having a bit of trouble getting sip trunk to not get routed to sip-from-external. |
| 02:19.13 | encinoman | I had this problem once before, I had to put type=friend. I did that and now its not working again. Wondering if I could have a little help. |
| 02:19.20 | encinoman | I'm posting a pastebin now. |
| 02:20.30 | hipitihop | I have a soptphone peer registering but do get following notice: "Received SIP subscribe for peer without mailbox: username" |
| 02:21.00 | p3nguin_ | hipitihop: You didn't configure the mailbox which you have assigned to the peer. |
| 02:21.47 | ManxPower-work | hipitihop, put a mailbox= line in your sip.conf for that peer or tell the softphone to not subscribe to voicemail notifications |
| 02:22.00 | ManxPower-work | ~freepbx |
| 02:22.01 | 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 |
| 02:22.01 | dzup_ | i need a textention conf example to accept SIP external in diferent LAN segment, please |
| 02:22.04 | ManxPower-work | encinoman, ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 02:22.18 | ManxPower-work | dzup_, your question makes no sense |
| 02:22.47 | hipitihop | p3nguin_, p3nguin_and ManxPower-work thanks, looking up the mailbox= |
| 02:22.55 | encinoman | Thanks Manx, I just realized that. |
| 02:23.21 | p3nguin_ | hipitihop: I think you already have the mailbox= in the peer definition, but you didn't configure the mailbox which you assigned. |
| 02:24.06 | dzup_ | i need to see an example of a SIP extension so i can figure out why am not registering a SIP phone from a diferent ip (out of this IP network), to figure out what am doiung wrong, i connect but got no audio, then i got audio but can register, am lost |
| 02:24.19 | p3nguin_ | dzup_: There is no such thing as a SIP extension. |
| 02:24.27 | dzup_ | SIP phone |
| 02:25.00 | p3nguin_ | dzup_: You can have SIP devices, and you can have extensions, but the two are not connected until you form the relationship in extensions.conf. |
| 02:25.39 | dzup_ | p3nguin_: can i see and example? |
| 02:26.08 | dzup_ | either in sip.conf and extention.conf so i can see a example? |
| 02:26.10 | p3nguin_ | dzup_: http://pastebin.com/m59d17875 |
| 02:26.25 | dzup_ | thanks |
| 02:26.32 | hipitihop | p3nguin_, no mailbox= lines in my sip.conf ... so I gues the zoiper softphones setup is asking for one... just so I understand this should just be some unique id, I don't need to go setup some corresponding dir or storage location ? |
| 02:27.26 | hipitihop | I'm guessing themailbox shold be associated with the user as there may be multiple peers a given user may use but all point to same mailbox |
| 02:27.53 | p3nguin_ | hipitihop: You only need to read and understand the examples in voicemail.conf and create the mailbox for your phone, then add the mailbox= line to the peer. |
| 02:28.52 | hipitihop | p3nguin_, indeed, I need to read more :-) don't mean to waste your time guys, thanks for your help |
| 02:29.14 | p3nguin_ | It's all very simple if you just read the comments in the example configs. |
| 02:29.29 | p3nguin_ | If there is part of it that you don't understand, that's the time to ask. |
| 02:36.14 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 02:37.28 | ManxPower-work | dzup_, there is nothing special you have to do UNLESS you are using NAT or a firewall. |
| 02:38.07 | ManxPower-work | hipitihop, mailbox=vmbox@vmcontext => voicemail.conf [vmcontext]\nvmbox => .... |
| 02:42.10 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 02:47.26 | encinoman | Is there an A2billing irc channel? |
| 02:55.44 | Katty | yawns |
| 02:55.48 | Katty | stretches |
| 03:00.09 | hipitihop | ManxPower-work, p3nguin_ok examples in voicemail.conf ar epretty streight forward but don't yet understand about storage |
| 03:01.25 | ManxPower-work | you should not care about storage at this point. |
| 03:01.25 | ManxPower-work | Asterisk is saying "device asked for their voicemail status and I don't have a voicemailbox configured for that device" |
| 03:02.11 | hipitihop | ok, so I'll setup mailboxes and then point to them in sip.conf |
| 03:02.47 | ManxPower-work | if you want that device to have a mailbox, yes. |
| 03:03.02 | ManxPower-work | if not, just tell your sip client to stop asking about it's mailbox |
| 03:03.45 | hipitihop | sounds like a useful feature so sounds easy to se tup |
| 03:07.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:07.55 | hipitihop | ManxPower-work, p3nguin_ok that seems to have done the trick... reloaded voicmail and sip and no more warnings. thanks |
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| 03:30.40 | drmessano | p3nguin_, |
| 03:32.25 | hipitihop | next issue, technically I need to also support G729a but that has special installation requirements due to license |
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| 03:47.35 | p3nguin_ | drmessano: ? |
| 03:48.59 | drmessano | I believe the cert was the issue |
| 03:49.03 | drmessano | Added the CA |
| 03:49.33 | p3nguin_ | drmessano: Oh, I thought maybe you were referring to something manxpower-work said. |
| 03:49.34 | drmessano | Havent had a chance to test, but I know that was missing |
| 03:51.03 | ketin | :-O |
| 03:51.10 | p3nguin_ | drmessano: I was just going to say that he's a dumb fuck incapable of comprehending that "it's" means "it is" or "it has" rather than "something belonging to it" like he thinks it means. But most people that passed 6th grade English know the difference between "its" and "it's," so I won't even mention it. |
| 03:51.10 | ketin | http://www.firequest.com/G12-013.html |
| 03:51.15 | ketin | Holy crap... |
| 03:51.25 | drmessano | lol |
| 03:53.19 | ketin | They even have bolo rounds! |
| 03:53.44 | TJNII | They're even on sale! |
| 03:55.48 | p3nguin_ | I'm concerned that my 12 ga gun would not withstand the 4000 degrees of fire those produce. |
| 03:55.56 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:56.43 | ketin | They have pepper spray rounds as well. |
| 03:57.07 | ketin | And... confetti rounds? Wtf. |
| 03:57.15 | p3nguin_ | PARTY TIME |
| 03:57.32 | TJNII | HAPPY BIRTHDAY, MOTHER F**KER |
| 03:57.55 | ketin | They sell 37mm grenade launcher attachments and ammo. |
| 03:58.06 | ketin | Might want to consult state laws on that purchase. |
| 03:58.57 | p3nguin_ | drmessano: I didn't have the CA cert that postfix wanted in order to verify gmail, but postfix still delivered the mail to gmail anyway. The mail arrived at the destination successfully after being relayed through gmail. |
| 03:59.55 | p3nguin_ | I would, of course, want a more detailed log to even try to guess where the failure was on your server. |
| 04:05.28 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 04:08.21 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 04:15.31 | *** part/#asterisk encinoman (~matthew@pool-173-63-32-193.nwrknj.fios.verizon.net) |
| 04:35.10 | dddh | ManxPower-work, I do not use gui |
| 04:59.26 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 04:59.40 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:11.01 | ManxPower-work | Ugh. AGI support for PHP sucks compared to Perl. |
| 05:13.46 | ChannelZ | in what way |
| 05:15.56 | ManxPower-work | ChannelZ, Manager Stuff: 1) it does not support all of the 1.4 manager commands. 2) can't seem to handle the multi-line response to the SIPpeers command 3) little documentation or examples out there. |
| 05:18.15 | ChannelZ | It's basically a text interface between the two, how can it "not support" commands? |
| 05:20.12 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 05:20.16 | ManxPower-work | ChannelZ, Exactly! |
| 05:21.14 | ManxPower-work | Specifically it is failing to handle the command RESPONSE for SIPpeers |
| 05:21.24 | ChannelZ | I think you're complaining about AGI wrappers that abstract all the communication for you |
| 05:22.00 | ManxPower-work | ChannelZ, I'd not complain if it was giving me back the data. It's not. |
| 05:27.42 | ChannelZ | So you're using Manager, not AGI |
| 05:35.40 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
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| 05:39.05 | ManxPower-work | the PHP AGI library supports manager connections as well as AGI. |
| 05:46.19 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:49.28 | ChannelZ | sounds like the PHP AGI library you're using is badly written then |
| 05:51.25 | ManxPower-work | *nod* |
| 05:55.03 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 05:56.05 | Katty | hi. |
| 05:57.14 | p3nguin_ | high |
| 05:59.22 | Katty | me and ryan played monopoly on xbox |
| 05:59.29 | Katty | it was kinda fun |
| 05:59.51 | ChannelZ | http://ihasahotdog.com/2008/07/27/funny-dog-pictures-on-life-high/ |
| 06:00.42 | Katty | lol |
| 06:16.24 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
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| 06:45.49 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 06:47.03 | hackeron | hey, I haven't used asterisk in a number of years - I remember I had echo problems and used the OSLEC echo canceller which appears to have been added to the kernel at 2.6.28 - I can't find much info about how to get it working with asterisk 1.6 though, is it even required these days? |
| 06:51.33 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 06:57.25 | idespinner | OSLEC? is that similar to HPEC(software high performance echo canceler? |
| 06:57.52 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:59.58 | hackeron | idespinner: think so, but doesnt HPEC require a license? |
| 07:00.07 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 07:01.25 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 07:06.33 | idespinner | hackeron, HPEC does indeed require a license although with any digium TDM card, it is included free |
| 07:06.51 | idespinner | Although why not save yourself some trouble and just buy echo cancelation |
| 07:07.16 | hackeron | idespinner: because it's very expensive :) |
| 07:08.49 | hackeron | idespinner: I have the OpenVOX A400P so I don't have an HPEC license |
| 07:09.11 | hackeron | idespinner: also, from everything I can find, OSLEC appears to be a superior echo canceller? |
| 07:11.48 | idespinner | expensive? we'll compared to free I suppose it is. In light of traditional TDM hardware its pretty cheap... |
| 07:12.08 | idespinner | but, let me see. digium was giving away HPEC licenses free a while ago... |
| 07:12.20 | hackeron | idespinner: the OpenVOX A400P is $100 - a digium card with an echo canceller is $600 or more here |
| 07:13.07 | hackeron | idespinner: and why bother with HPEC is OSLEC is the superior echo canceller? |
| 07:14.57 | idespinner | fair enough... looks like digium only gives em away at astricon |
| 07:15.17 | idespinner | but the build instructions seem fairly straight forward on the OSLEC site |
| 07:16.38 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:16.53 | idespinner | it looks like its not really part of asterisk, but part of zaptel/dahdi, the actual TDM driver |
| 07:17.20 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:18.34 | hackeron | idespinner: well, there's no zaptel anymore it seems, and http://www.asterisk.org/zaptel-to-dahdi is down - also, oslec is now included with the kernel, so I have /lib/modules/2.6.31-19-generic/updates/dkms/dahdi_echocan_oslec.ko but I'm not quite sure what to do with it |
| 07:19.10 | idespinner | so in short, I wouldnt see why OSLEC would be included in a kernel unless DAHDI were included aswell, but if youve got it... you should be good |
| 07:19.44 | idespinner | do you have dahdi installed as part of that kernel? |
| 07:19.45 | hackeron | idespinner: how do I enable it? |
| 07:20.01 | hackeron | idespinner: no, but apt-get install dahdi compiled the module |
| 07:20.42 | idespinner | from the oslec site : cat /proc/oslec/info ? |
| 07:20.46 | idespinner | worth a shot... |
| 07:21.08 | hackeron | hmm, it's not there |
| 07:21.37 | idespinner | no oslec folder? |
| 07:21.50 | hackeron | nope - guess it's not included with the kernel - compiling |
| 07:22.14 | ManxPower-work | See the oslec info in voip-info.org |
| 07:22.16 | idespinner | it may be there but not loaded |
| 07:22.35 | idespinner | try insmod oslec.ko |
| 07:22.43 | ManxPower-work | IIRC if your digium card is modern you might be able to get HPEC for free. |
| 07:22.44 | idespinner | or insmod kernel/oslec.ko |
| 07:23.13 | idespinner | ManxPower-work, : its not a digium.. openvox A400 clone card... :/ |
| 07:23.27 | ManxPower-work | idespinner, then you can purcahase the HPEC if needed. |
| 07:23.43 | ManxPower-work | $10/channel, IIRC. Not good for more than a few channels. |
| 07:24.06 | idespinner | ManxPower-work, not for me, i;m digium all the way, its for hackeron |
| 07:24.09 | hackeron | ManxPower-work: I believe I tried HPEC, but this was as few years back and found OSLEC worked much better - there is a lot of noise on the lines |
| 07:24.41 | ManxPower-work | idespinner, I'm a fan of Sangoma |
| 07:24.56 | hackeron | ManxPower-work: there is no oslec page on voip-info.org? |
| 07:25.07 | idespinner | digium reseller here. part of the agreement is that we cant sell other vendor cards i think... |
| 07:25.20 | hackeron | ManxPower-work: and all the docs talk about zaptel which is not supported in asterisk 1.6 |
| 07:25.24 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 07:25.43 | ManxPower-work | hackeron, http://www.rowetel.com/ucasterisk/oslec.html |
| 07:26.27 | hackeron | ManxPower-work: yeah, I know, but that also talks mostly about zaptel |
| 07:26.42 | ManxPower-work | other than the DAHDI info on the left side menu |
| 07:26.47 | idespinner | hackeron, follow the heading for 'Installing OSLEC with DAHDI" |
| 07:26.54 | idespinner | that should be all you need. |
| 07:27.01 | hackeron | idespinner: I did, now what? |
| 07:27.13 | hackeron | idespinner: I still don't have the oslec module and no /proc/oslec |
| 07:27.40 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 07:27.45 | idespinner | ic, did make end up sucessfull? |
| 07:28.01 | idespinner | and did the insmod commands work at all? |
| 07:28.32 | hackeron | idespinner: what insmod commands? -- ther are no insmod commands there |
| 07:29.47 | idespinner | <PROTECTED> |
| 07:29.51 | idespinner | or insmod kernel/oslec.ko |
| 07:30.16 | idespinner | then look for /proc/oslec |
| 07:31.30 | idespinner | also double check you have dahdi installed first |
| 07:32.32 | idespinner | oh and also i think it mentions you have to stop dahdi before you install oslec |
| 07:32.48 | hackeron | idespinner: there's ./drivers/dahdi/dahdi_echocan_oslec.ko |
| 07:32.55 | hackeron | idespinner: but no oslec.ko and nothing in /proc/oslec |
| 07:33.49 | idespinner | what if you do lsmod | grep dahdi |
| 07:34.08 | hackeron | dahdi 197768 2 dahdi_transcode,dahdi_dummy |
| 07:34.12 | hackeron | and a few other lines, but it's there |
| 07:34.47 | hackeron | there's no oslect though and if I insmod dahdi_echocan_oslec.ko I get unknown symbol and a bunch of errors in dmesg saying things like oslec_create don't exist |
| 07:35.05 | hackeron | so my dahbi is compiled with oslec support, but there's no oslec module |
| 07:35.39 | hackeron | dahdi* |
| 07:35.39 | idespinner | not sure, |
| 07:36.06 | hackeron | well, looking at the guide they have, they don't mention anything about getting the oslec module |
| 07:36.25 | hackeron | only about getting dahdi_echocan_oslec.o compiled |
| 07:37.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:38.08 | idespinner | i hadnt heard of OSLEC before you mentioned it. Its definatley worth a check |
| 07:38.50 | idespinner | you may end up having to work with the OSLEC mailing list for help |
| 07:38.50 | hackeron | yeah, last I installed it in 2007 and it was wonderful, solved all my echo problems (very bad quality lines) |
| 07:39.35 | *** join/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 07:39.39 | mick_laptop | hi everyone |
| 07:39.50 | mick_laptop | anyone use video conferencing w/ asterisk? |
| 07:40.37 | idespinner | I have one customer who uses it with passthrough |
| 07:40.40 | mick_laptop | i have 2 sip peers and I'm using sip communicator -- under the preferences it shows that the video works, but I can't seem to initiate a video conference. anything special that I need to do? |
| 07:40.53 | idespinner | what are the 2 peers? |
| 07:41.13 | ketin | Damn... |
| 07:41.14 | mick_laptop | sip extensions (both using sip communicator) |
| 07:41.19 | ketin | http://files.2theleft.la/Joe_Stack_note.txt |
| 07:41.21 | mick_laptop | one is on OSX the other on Linux |
| 07:42.18 | hackeron | idespinner: ok, it seems there is no /proc/oslec if using oslec with dahdi |
| 07:43.31 | hackeron | idespinner: seems oslec should just work on debian, hmm, haven't tried that :) |
| 07:45.13 | mick_laptop | idespinner: how did you get yours to work? |
| 07:45.23 | mick_laptop | i saw this: http://sourceforge.net/projects/appconference/ |
| 07:45.41 | mick_laptop | seems dated though (and I'm sure it is not relevant anymore) |
| 07:50.12 | idespinner | mick_laptop, wasnt me who got it to work... and to my knowledge asterisk cannot transcode or conference(multiplex) video streams |
| 07:50.16 | idespinner | only passthough... |
| 07:58.16 | mick_laptop | so what is this?: https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/ |
| 08:02.25 | dddh | should I use ssl or not? |
| 08:03.04 | mick_laptop | seems these are my options: http://www.mail-archive.com/asterisk-video@lists.digium.com/msg00302.html |
| 08:03.20 | mick_laptop | i don't suppose anyone has any documentation on these? ;) |
| 08:04.46 | idespinner | so asterisk "can" transcode video via some modified ffdshow sources i believe but i've never tried that personally |
| 08:04.55 | idespinner | there are a ton of patent issues surrounding it |
| 08:06.51 | mick_laptop | i just want to get it running between friends and family, nothing commercial |
| 08:07.00 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 08:07.16 | mick_laptop | so the patent thing isn't an issue for me |
| 08:10.45 | ketin | WTF... http://images.heaven666.org/g/5/extreme-tattoo-and-penis-piercings-3.jpg NSFW. |
| 08:10.50 | idespinner | you just need to ensure your doing passthrough only |
| 08:10.54 | ketin | Permission to puke? |
| 08:12.02 | mick_laptop | idespinner: how do i do that? |
| 08:12.29 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:15.00 | mick_laptop | this patch looks interesting, "This patch adds AMR-NB support to Asterisk 1.4" |
| 08:15.08 | mick_laptop | now wtf is AMR-NB |
| 08:15.11 | mick_laptop | :) |
| 08:15.58 | ManxPower-work | For most people passthru is pretty useless |
| 08:16.52 | mick_laptop | well that one post said that he got two people to do video chat back and forth - so i'm optimistic |
| 08:17.52 | sun28 | moin \o/ |
| 08:18.36 | mick_laptop | this looks really interesting: http://sourceforge.net/projects/vmukti/ |
| 08:18.45 | mick_laptop | sun28: moin, wie gehts? |
| 08:19.44 | mick_laptop | this looks really promising: http://sourceforge.net/dbimage.php?id=224620 |
| 08:20.36 | mick_laptop | oh wtf... "runs on Microsoft Windows Server 2003, WinXP" |
| 08:21.05 | mick_laptop | at least it is written in c#, i might be able to get it to work under mono |
| 08:24.15 | sun28 | MABE U DID NOT COMPEIL IT WIHT TEH RITE OPTIMELIZER PHLAGS!? |
| 08:24.35 | sun28 | so sorry |
| 08:26.23 | mick_laptop | slaps sun28 |
| 08:26.27 | mick_laptop | snap out of it |
| 08:26.42 | ChannelZ | snap into a Slim Jim |
| 08:28.03 | mick_laptop | hopes to get something working cross platform |
| 08:31.14 | mick_laptop | shit this checkout is taking forever |
| 08:31.58 | mick_laptop | though some parts don't look promising. ex: Audio.VistaService |
| 08:32.52 | mick_laptop | and i see references to DirectX |
| 08:32.58 | mick_laptop | yuck |
| 08:34.37 | mick_laptop | Microsoft.Practices.EnterpriseLibrary.Logging.dll ... they must be kidding, good god why not use Log4Net like the rest of the .NET world |
| 08:37.17 | mick_laptop | is digium.com down? |
| 08:39.19 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 08:42.37 | spenguin[work] | http://downforeveryoneorjustme.com/digium.com |
| 08:46.07 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 08:46.09 | mick_laptop | svn: URL 'https://vmukti.svn.sourceforge.net/svnroot/vmukti/trunk/ccs0.1' of existing directory 'trunk/CCS0.1' does not match expected URL -- typical windows developers |
| 08:46.21 | mick_laptop | *sigh* |
| 08:55.08 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 08:57.55 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 08:58.25 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 09:04.17 | Dovid | morning ev1 |
| 09:06.13 | phix | Evening Dovid |
| 09:17.29 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 09:18.57 | *** part/#asterisk TecR0c (tecr0c@never.met.a.leet-hacker.net) |
| 09:22.36 | sun28 | mick_laptop: http://downforeveryoneorjustme.com/ ^__^ |
| 09:22.46 | sun28 | oops |
| 09:22.49 | sun28 | spenguin[work]: +1 |
| 09:30.18 | spenguin[work] | :> |
| 09:43.56 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-dgyuhelbeqwxqcsb) |
| 09:43.56 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 09:43.56 | *** mode/#asterisk [+o angler] by ChanServ |
| 09:53.46 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-hzwvfsejvnxqvihb) |
| 10:42.49 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 10:51.53 | jdoe | wtf... |
| 10:52.04 | jdoe | what did I do to this config. |
| 10:52.17 | jdoe | dundi lookups work, but the dialplan persists in trying to dial through the wrong host. |
| 11:02.57 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 11:16.05 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 11:16.20 | jdoe | ah, bad cached results.. |
| 11:19.10 | spenguin[work] | wheres Katty |
| 11:27.04 | *** join/#asterisk birnenkind (~penne@HSI-KBW-109-192-159-250.hsi6.kabel-badenwuerttemberg.de) |
| 11:27.05 | birnenkind | hoi |
| 11:29.31 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 11:32.02 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 11:33.29 | *** join/#asterisk SAngeli (~SAngeli@host242-77-static.82-94-b.business.telecomitalia.it) |
| 11:33.39 | SAngeli | \join |
| 11:47.59 | *** join/#asterisk V4mpire (~gary@82.118.111.252) |
| 12:58.26 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.28.99) |
| 13:01.37 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:35.37 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 13:51.16 | *** join/#asterisk harryv (~harry@67.207.147.205) |
| 13:51.23 | harryv | hrm. skype for asterisk can't be used on mac? |
| 13:53.31 | spenguin[work] | why not |
| 13:53.42 | harryv | register |
| 13:54.06 | harryv | sh: ./register: cannot execute binary file |
| 14:04.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:05.42 | dddh | yay! I finally made my voip work :) |
| 14:06.34 | dddh | I've heard skype for sip is not free? |
| 14:08.18 | harryv | skype for sip is kinda free. but you can't make skype-to-skype calls. |
| 14:08.30 | harryv | or wait, maybe there is some small recurring fee. can't remember. |
| 14:08.40 | harryv | but it's a largely uninteresting product, anyways. |
| 14:14.56 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 14:17.04 | spenguin[work] | you need to pay for the module |
| 14:43.49 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 14:44.57 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 14:49.44 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 14:49.54 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 14:49.55 | Katty | hi |
| 14:50.34 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176462822.dsl.bell.ca) |
| 14:52.34 | *** join/#asterisk ehjay (~abarber@c-66-31-107-214.hsd1.ma.comcast.net) |
| 14:52.52 | spenguin[work] | hey Katty !! |
| 14:53.03 | Katty | howdy |
| 14:53.15 | spenguin[work] | bored on a sunday at work |
| 14:53.45 | Katty | you work on sunday? |
| 14:55.27 | spenguin[work] | once a month |
| 14:55.31 | Katty | ahh |
| 14:56.04 | spenguin[work] | its boring. |
| 14:56.22 | Katty | nods |
| 14:56.38 | spenguin[work] | whats cooking |
| 14:56.53 | Katty | oh nothing. the dog woke me up cause he had to go (= |
| 14:57.25 | spenguin[work] | kk |
| 14:57.27 | Katty | ryan got an xbox, so i've been playin that |
| 14:57.34 | spenguin[work] | has a rooster that crows at 2 am |
| 14:57.36 | Katty | most of the games are boring |
| 14:57.42 | spenguin[work] | hes at US time |
| 14:57.48 | spenguin[work] | heh |
| 14:57.49 | Katty | i think i'd rather play with a rooster |
| 14:57.59 | spenguin[work] | you cant play with a rooster |
| 14:58.07 | Katty | don't be silly |
| 14:58.10 | spenguin[work] | run around and catch it |
| 14:58.14 | spenguin[work] | maybe |
| 14:58.17 | ehjay | I was wondering, can OSLEC be employed with a SPA3XXX, or does it only work with add in cards? |
| 14:58.20 | Katty | a rooster can be tamed :P |
| 14:58.43 | spenguin[work] | to do what |
| 14:58.53 | Katty | to loves of them, of course :P |
| 14:59.02 | spenguin[work] | theyd look stupid if they did roll overs |
| 14:59.06 | Katty | http://www.youtube.com/watch?v=YG3-ZLqtLH4 <- proof! |
| 14:59.06 | spenguin[work] | or play dead |
| 15:01.59 | *** join/#asterisk Heretic (~Fallen@dsl-246-102-216.telkomadsl.co.za) |
| 15:01.59 | Katty | oooh video of a rooster playing soccer |
| 15:02.02 | Heretic | lo all |
| 15:02.31 | Katty | http://www.youtube.com/watch?v=m8Ou7I4PU30 <- rooster playing soccer |
| 15:08.02 | Maliuta | Katty: so what happened to the squirrel fetish? you see one cock with a ball and change sides? ;) |
| 15:23.14 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 15:24.43 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 15:25.34 | *** join/#asterisk CMI-DOS (~ryan_2@5ad9c21d.bb.sky.com) |
| 15:26.56 | CMI-DOS | help i got everything setup as easy as that but how do i add users ive tryed forums and nothing works not even the asterisk docs help |
| 15:27.39 | *** part/#asterisk harryv (~harry@67.207.147.205) |
| 15:33.19 | CMI-DOS | help |
| 15:40.28 | spenguin[work] | CMI-DOS: lol |
| 15:40.52 | spenguin[work] | show us some config |
| 15:46.41 | Maliuta | and define "users" |
| 15:47.44 | spenguin[work] | so were there any tsunamis? |
| 15:47.50 | spenguin[work] | with all that warning |
| 15:47.56 | Maliuta | NFI |
| 15:48.04 | spenguin[work] | NFI? |
| 15:48.07 | Maliuta | spent most of the day in bed |
| 15:48.16 | Maliuta | No Fecking Idea |
| 15:48.34 | Maliuta | I would imagine the coast copped a battering though |
| 15:48.48 | spenguin[work] | was at the beach yesterday |
| 15:48.51 | spenguin[work] | it was normal |
| 15:48.55 | spenguin[work] | low tide. |
| 15:50.02 | Maliuta | http://www.abc.net.au/news/stories/2010/02/28/2832258.htm |
| 15:51.45 | Maliuta | I live in brisbane and far enough inland not to care |
| 15:52.34 | Maliuta | http://www.abc.net.au/news/stories/2010/02/28/2832406.htm says it was a non-starter |
| 15:52.57 | CMI-DOS | pls could someone put there conf file in pastebin ill get it then |
| 15:53.04 | *** join/#asterisk Znuff (Znuff@89.45.83.253) |
| 15:53.07 | Znuff | Hi. |
| 15:53.09 | *** join/#asterisk lucasb (~lucas@S0106001310a1871d.ok.shawcable.net) |
| 15:53.53 | Maliuta | CMI-DOS: there is more than one config file, a whole directory full of them to be specific |
| 15:54.12 | Maliuta | CMI-DOS: the exaple configs and the book are a good place to start |
| 15:54.21 | Maliuta | ~book |
| 15:54.22 | 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 |
| 15:54.49 | Znuff | Does anyone know of a certain utility that allows you to make some "tests", ie: have it call number XXX Y times, and stop after Z minutes then do it all over again? |
| 15:55.55 | Znuff | tried pjsip, but couldn't make it work |
| 15:56.41 | Znuff | Tried sipp, too, but it's way too complicated and I never got it to work |
| 16:02.15 | CMI-DOS | i said i read the docs and they didnt help |
| 16:04.03 | Maliuta | CMI-DOS: well you are not providing us with any insight into the "problem" or what exactly you are trying to do |
| 16:05.17 | CMI-DOS | sorry im repairing a battery i just dont know how to add users and nothing helped me |
| 16:06.33 | Maliuta | CMI-DOS: users to what? an agent? sip? the OS? a dial plan? into the astdb? |
| 16:07.02 | CMI-DOS | sip |
| 16:08.00 | Maliuta | edit sip.conf ... look at the example it's pretty straight forward and is well documents |
| 16:08.08 | Maliuta | s/ts/ted/ |
| 16:08.33 | *** join/#asterisk frek818 (~herman@cpe-98-154-78-151.socal.res.rr.com) |
| 16:08.38 | Maliuta | then do a sip reload at the CLI |
| 16:13.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:14.14 | Znuff | so... anyone? |
| 16:16.45 | *** join/#asterisk encinoman (~matthew@pool-173-63-32-193.nwrknj.fios.verizon.net) |
| 16:17.20 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:30.02 | p3nguin_ | znuff: call file |
| 16:31.26 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 16:31.40 | Znuff | hmmm, doesn't seem to be what I need |
| 16:32.21 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 16:32.25 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:32.42 | p3nguin_ | It should do exactly what you asked for. |
| 16:34.36 | Znuff | Doesn't seem to do multiple lines in the same time |
| 16:35.06 | p3nguin_ | lines? Like multiple phone numbers? |
| 16:35.19 | Znuff | Kind of, yes |
| 16:35.22 | p3nguin_ | That wasn't in your request. |
| 16:35.36 | Znuff | I want to call the same phone number 20-30-40-50-60 times |
| 16:35.40 | Znuff | simultaneously |
| 16:35.57 | p3nguin_ | I guess you need a bunch of call files instead of just one. |
| 16:36.37 | Znuff | Also, can't see how it would hang up after the specified time |
| 16:36.55 | p3nguin_ | That also wasn't in your request. |
| 16:37.06 | p3nguin_ | Next time, be more specific on what you're trying to do. |
| 16:37.29 | Znuff | have it call number XXX Y times, and stop after Z minutes then do it all over again? |
| 16:37.33 | Znuff | that was my initial request =) |
| 16:38.33 | p3nguin_ | And call files will do that. |
| 16:38.35 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 16:38.48 | p3nguin_ | But now you've modified your requirements. |
| 16:39.27 | Znuff | I didn't explain myself properly, then. |
| 16:39.57 | Znuff | So, I want to dial a phone number X times, hold the call for Y minutes then hang-up. |
| 16:40.18 | Znuff | Couldn't find a easy scriptable sip client for that |
| 16:45.22 | CMI-DOS | im in the sip.conf and i want to know if [sip_proxy_out ] is for a external line |
| 16:45.24 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 16:45.55 | p3nguin_ | Only if it contains the info for your ITSP. |
| 16:46.43 | Maliuta | setting up sip is _really_ well documented. |
| 16:47.25 | [TK]D-Fender | CMI-DOS: As opposed to what? |
| 16:47.35 | CMI-DOS | but where about in sip.conf do the users do |
| 16:47.44 | [TK]D-Fender | Znuff: Indeed just queue up a pile of call files... or AMI originate's |
| 16:48.16 | p3nguin_ | cmi-dos: Every device has its own definition in sip.conf. |
| 16:51.26 | CMI-DOS | but what do the users go under |
| 16:51.28 | *** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net) |
| 16:52.07 | p3nguin_ | Your question makes no sense since I already explained that every device has its own definition. |
| 16:52.19 | CMI-DOS | how do i use the digium gui |
| 16:52.35 | [TK]D-Fender | CMI-DOS: SIP is SIP.... What makes one section of sip.conf differnt than another is based on what uses that peer |
| 16:52.38 | p3nguin_ | I guess you have to call Digium and ask them. |
| 16:52.57 | [TK]D-Fender | CMI-DOS: AsteriskGUI is NOT supported here.. and isn't even being maintained. |
| 16:53.05 | [TK]D-Fender | CMI-DOS: and that GUI uses users.conf for EVERYTHING |
| 16:53.13 | [TK]D-Fender | CMI-DOS: it bypasses most of sip.conf, etc. |
| 16:54.16 | Maliuta | as a general rule GUI == blow-goats |
| 16:55.25 | CMI-DOS | is [6000] in users.conf a extension |
| 16:55.33 | p3nguin_ | nope |
| 16:55.48 | p3nguin_ | extensions are found in extensions.conf only. |
| 16:56.14 | p3nguin_ | [6000] in users.conf is a user by the name of 6000. |
| 16:56.20 | CMI-DOS | whats vmsecret |
| 16:56.31 | p3nguin_ | voicemail pass code |
| 16:56.53 | CMI-DOS | so wheres the extension or user name |
| 16:57.00 | p3nguin_ | (1055.45) <p3nguin_> extensions are found in extensions.conf only. |
| 16:57.04 | p3nguin_ | Pay attention better. |
| 16:57.19 | [TK]D-Fender | CMI-DOS: [thisisthename] |
| 16:57.52 | [TK]D-Fender | [11:55]<p3nguin_>extensions are found in extensions.conf only. <- not entirely true.... |
| 16:57.56 | encinoman | Ya know, thats because if you use the command-line you are much more 1337 too. |
| 16:58.40 | p3nguin_ | [tk]d-fender: Where else can you find extensions? |
| 16:59.51 | [TK]D-Fender | p3nguin_: users.conf |
| 17:00.08 | [TK]D-Fender | p3nguin_: and extensions.ael clearly... |
| 17:01.30 | p3nguin_ | [tk]d-fender: By creating a user in users.conf, an extension magically is built? |
| 17:01.39 | [TK]D-Fender | p3nguin_: Yes |
| 17:02.45 | *** join/#asterisk mace (~mace@debian/developer/mace) |
| 17:03.45 | CMI-DOS | do you know of another pbx im going to leave asterisk for a bit its getting on my nerves |
| 17:04.00 | encinoman | freeswitch |
| 17:04.08 | encinoman | analog pots |
| 17:04.14 | CMI-DOS | thanks |
| 17:04.43 | encinoman | You can get a nice desk with a bunch of cables to manually connect the trunks and extensions. jk |
| 17:05.13 | [TK]D-Fender | CMI-DOS: Picking AsteriskGUI is probably the worst choice you could start with. |
| 17:05.21 | [TK]D-Fender | CMI-DOS: Nothing wrong with * itself. |
| 17:05.54 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 17:05.57 | [TK]D-Fender | CMI-DOS: And asking for GUI support in here is also not a great idea as they are not supported here. They each have their own support channels |
| 17:06.27 | *** join/#asterisk jmcdowell (~nooe@173-112-151-245.pools.spcsdns.net) |
| 17:08.27 | CMI-DOS | the gui for remote uses isnt it |
| 17:09.34 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:09.46 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 17:09.50 | [TK]D-Fender | CMI-DOS: Sory, that doesn't make any sense |
| 17:10.14 | CMI-DOS | the gui for asterisk is web based |
| 17:10.25 | [TK]D-Fender | CMI-DOS: Pretty much all of them, yes |
| 17:10.50 | [TK]D-Fender | CMI-DOS: Where by "web" we really mean "via HTTP" |
| 17:10.50 | CMI-DOS | well i wont be needing it then |
| 17:11.12 | [TK]D-Fender | CMI-DOS: Then feel free to not use one, and just learn * itself |
| 17:12.19 | encinoman | Wow, you're still ripping on CMI-DOS? |
| 17:13.02 | [TK]D-Fender | encinoman: Since when has what I've provided qualified as "ripping". |
| 17:13.05 | [TK]D-Fender | ? |
| 17:13.39 | encinoman | I think he gets it, don't use the GUI blah blah blah. |
| 17:13.44 | [TK]D-Fender | encinoman: I must have missed the part where you have contributed any useful information to this conversation somewhere along the way. How far back would I have to scroll for that? |
| 17:13.59 | [TK]D-Fender | encinoman: I certainly didn't say that |
| 17:14.03 | encinoman | Don't bother, I have none. |
| 17:21.24 | CMI-DOS | im sticking to asterisk |
| 17:22.42 | CMI-DOS | why can i dial from the console but i need to register a user to dial from a softphone |
| 17:23.03 | [TK]D-Fender | CMI-DOS: Who said you need to register? |
| 17:29.39 | *** join/#asterisk fofware (~chatzilla@190.224.76.151) |
| 17:32.57 | [TK]D-Fender | CMI-DOS: You could allow completely un-authed calls to your system... but what kind of scenario are you looking at where that would not be a "bad" thing? |
| 17:34.13 | jmcdowell | He has not been war dialed by some Chinese POS yet.. |
| 17:34.20 | jmcdowell | Once that happens, he will change his mind forever. |
| 17:37.15 | CMI-DOS | whats the password for ext 1234 |
| 17:37.29 | jmcdowell | What ever you made it |
| 17:37.44 | jmcdowell | Anyone here have experience with Mitel |
| 17:37.44 | jmcdowell | ? |
| 17:38.39 | p3nguin_ | extensions don't have passwords. |
| 17:39.18 | jmcdowell | They have secrects |
| 17:39.24 | p3nguin_ | negative |
| 17:39.25 | jmcdowell | So they can authenticate |
| 17:39.34 | p3nguin_ | extensions don't authenticate, either. |
| 17:39.43 | jmcdowell | Mine do, so either we aren't on the same page.. Orr.... |
| 17:40.06 | p3nguin_ | You're likely thinking of DEVICES having secrets and authenticating. |
| 17:40.24 | p3nguin_ | Extensions are not devices. |
| 17:40.38 | jmcdowell | So, extension 1301 has a secret of 0000000 |
| 17:40.48 | p3nguin_ | negative |
| 17:40.52 | jmcdowell | I can't authenticate or use my phone without it. |
| 17:41.14 | p3nguin_ | SIP device 1301 might have a secret of 0000000, but that has nothing to do with an extension. |
| 17:41.22 | *** part/#asterisk bobJR (~bob@unaffiliated/bobjr) |
| 17:42.14 | p3nguin_ | For all I know, extension 1301 could be Playback(tt-weasels). Again, that's not related to the device. |
| 17:42.37 | jmcdowell | I don't understand what you mean. |
| 17:42.43 | p3nguin_ | apparently |
| 17:42.57 | [TK]D-Fender | [12:37]<CMI-DOS>whats the password for ext 1234 <- pardon? |
| 17:43.09 | p3nguin_ | exten => 1301,1,Playback(tt-weasels) |
| 17:43.12 | [TK]D-Fender | CMI-DOS: This is your system. Why should we know your passwords? |
| 17:43.38 | p3nguin_ | exten => 1301,1,Playback(tt-weasels) <-- THIS is extension 1301. It has nothing to do with your SIP device. |
| 17:44.05 | jmcdowell | [1401] deny=0.0.0.0/0.0.0.0 secret=123456 |
| 17:44.22 | p3nguin_ | That's a peer definition. |
| 17:44.25 | jmcdowell | Hmm.. |
| 17:44.31 | p3nguin_ | 1401 is the device's name. |
| 17:44.36 | jmcdowell | Clearly, there is more about asterisk that I need to develop an understanding of. |
| 17:44.53 | ManxPower-work | jmcdowell, you are denying everything. |
| 17:45.07 | jmcdowell | It's like 2 virgins, it just doesn't work. |
| 17:45.08 | jmcdowell | :D |
| 17:45.16 | jmcdowell | That's strange, because it works.. |
| 17:45.20 | jmcdowell | But perhaps that is the paging issue. |
| 17:45.27 | jmcdowell | Where it beeps and just hangs up. |
| 17:45.30 | ManxPower-work | jmcdowell, I hope that line is not your entire sip.conf entry for 1401 |
| 17:45.32 | Kobaz | there's no way to have sip peers by name right.. everything is by ip address? |
| 17:45.45 | jmcdowell | It's not, it's just the first couple of lines. |
| 17:45.52 | ManxPower-work | Kobaz, Put down the bong and step away from the keyboard. |
| 17:46.06 | jmcdowell | lol |
| 17:46.09 | Kobaz | ManxPower-work: i remember reading that a while ago |
| 17:46.20 | CMI-DOS | whats context |
| 17:46.38 | [TK]D-Fender | CMI-DOS: A section of a dialplan |
| 17:46.47 | [TK]D-Fender | CMI-DOS: You need to sit down with teh Book and read the basics.. |
| 17:46.49 | [TK]D-Fender | ~book |
| 17:46.50 | 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 |
| 17:46.52 | [TK]D-Fender | ^^^^^^^ |
| 17:47.04 | jmcdowell | Read till yours eyes bleed, get treatment and do it again.. |
| 17:47.07 | Kobaz | i mean, everything i've done, confirms that |
| 17:47.11 | jmcdowell | Asterisk is nice, but very complex. |
| 17:47.15 | [TK]D-Fender | \cmiContexts are used to separate what extens a call has access to. |
| 17:47.30 | [TK]D-Fender | CMI-DOS: Contexts are used to separate what extens a call has access to. |
| 17:47.33 | Kobaz | if you have two peers with the same ip, it'll just use the first one |
| 17:48.40 | *** join/#asterisk LemensTS (~LemensTS@71.86.32.146) |
| 17:48.49 | ManxPower-work | Kobaz, millions of NAT users would disagree with you |
| 17:49.13 | Kobaz | oh right |
| 17:49.15 | Kobaz | hmm |
| 17:49.19 | Kobaz | something else is wiggity then |
| 17:49.48 | p3nguin_ | Where's the paste? |
| 17:50.03 | LemensTS | can i see if a variable is all alpha characters in the dialplan? |
| 17:50.10 | LemensTS | ...no numerical |
| 17:50.16 | ManxPower-work | In fact all people that register more than one line to Asterisk from the same phones would also disagree with you. |
| 17:50.20 | p3nguin_ | lemensts: Verbose() or NoOp() |
| 17:50.26 | jmcdowell | wow |
| 17:50.29 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:50.30 | jmcdowell | I crashed the mitel phone.. |
| 17:50.33 | ManxPower-work | LemensTS, not that I'm aware of. |
| 17:50.42 | jmcdowell | it had all kinds of funky stuff on the screen. |
| 17:51.27 | LemensTS | p3nguin: i mean to have the dialplan see if the variable is all alpha characters, if so GOTO etc... |
| 17:51.30 | jmcdowell | Just sayin, that was a first. |
| 17:51.48 | p3nguin_ | lemensts: You can probably use some type of regex test. |
| 17:52.42 | LemensTS | p3nguin: yea ill do that, didnt know if there was another way to check or not. thansk! |
| 17:53.22 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 17:53.24 | LemensTS | p3nguin: you get pm's? |
| 17:53.28 | p3nguin_ | nope |
| 17:53.50 | LemensTS | are you in illinois |
| 17:53.53 | p3nguin_ | Yes |
| 17:54.04 | LemensTS | anyway to contact u outside here? |
| 17:54.31 | p3nguin_ | I guess that depends on your reason for wanting to contact me. :) |
| 17:54.53 | LemensTS | Im in Missouri |
| 17:55.11 | p3nguin_ | Hmm, that's a popular state for #asterisk. |
| 17:55.21 | LemensTS | really? |
| 17:55.22 | p3nguin_ | Several people in here live there. |
| 17:55.48 | LemensTS | Who like? |
| 17:56.01 | ManxPower-work | LemensTS, What about FILTER FILTER(<allowed-chars>|<string>) Filter the string to include only the allowed characters |
| 17:56.30 | LemensTS | p3nguin: im just looking at business oppurtunies with people relatively close |
| 17:56.42 | [TK]D-Fender | LemensTS: Compare the string to the filtered version of the string. Single GotoIF |
| 17:56.48 | p3nguin_ | lemensts: Looking for work, or looking for someone to do work for you? |
| 17:57.42 | LemensTS | ManxPower: cool that may be a little easier thanks |
| 17:58.01 | LemensTS | p3nguin: looking for someone to do work |
| 17:58.34 | LemensTS | email me at customgt @ hotmail.com and we can talk further |
| 17:58.41 | ManxPower-work | LemensTS, "core show functions" is your friend. |
| 18:20.49 | *** join/#asterisk LemensTS (~LemensTS@71.86.32.146) |
| 18:25.21 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 18:28.37 | *** join/#asterisk LemensTS (~LemensTS@71.86.32.146) |
| 18:28.58 | jblack | lol. Who gets into "business opportunities" with someone with a hotmail address? |
| 18:29.21 | p3nguin_ | That's just a "public" address, obviously. |
| 18:29.30 | jblack | That's like getting into "importing" with a middle school dropout that lives in the slums |
| 18:29.46 | LemensTS | jblack: rather not post my business address to people i dont know :) |
| 18:29.51 | p3nguin_ | I wouldn't post my business email address here, either. |
| 18:33.11 | jblack | You're doing business with a hotmail address. |
| 18:33.27 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 18:33.49 | jblack | If you think that's appropriate, then that's fine by me. |
| 18:34.30 | jblack | Don't expect me to stop pointing and laughing at you, though. |
| 18:34.46 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:34.49 | LemensTS | jblack: :) i would be broke if i did not do business with random people around the world who i only talk thru im and email |
| 18:35.16 | p3nguin_ | How is using a hotmail address for an initial contact any different than giving someone your office phone number before giving them your cell or home number? |
| 18:36.08 | jblack | p3nguin_: Surely you can answer that question yourself? |
| 18:36.19 | p3nguin_ | I can... it is no different. |
| 18:36.22 | jblack | No. |
| 18:36.55 | jblack | Giving a business email address is equivilant to giving a business phone number. Giving a hotmail email address is like giving the number to the payphone down the block from your house. |
| 18:37.10 | jblack | I can't believe you don't realize that. |
| 18:37.34 | LemensTS | as long as you get paid i dont see a difference? |
| 18:37.41 | LemensTS | its all about $$$ |
| 18:38.04 | jblack | Like I already implied. More power to ya. |
| 18:41.18 | jblack | I'm sorry. I'm acting silly. If it works for you, that's great! |
| 18:52.08 | *** join/#asterisk LemensTS (~LemensTS@71.86.32.146) |
| 18:53.04 | *** join/#asterisk lbt (~david@Maemo/community/contributor/lbt) |
| 18:54.43 | *** join/#asterisk Zettatronic (~nick.croc@mail.maverickforms.biz) |
| 18:55.29 | LemensTS | . |
| 18:55.43 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 19:08.53 | CMI-DOS | hi i have a wind98 pc and which has a pci commecations adapter inside i wanted to know if i plug that into the asterisk server could i use my win98 box has a phone |
| 19:12.53 | *** join/#asterisk vkdelta (~vkdelta@rtr-trident.wireless.indra.com) |
| 19:13.25 | vkdelta | hello |
| 19:13.31 | vkdelta | anyone here ? |
| 19:13.42 | *** part/#asterisk vkdelta (~vkdelta@rtr-trident.wireless.indra.com) |
| 19:13.48 | *** join/#asterisk vkdelta (~vkdelta@rtr-trident.wireless.indra.com) |
| 19:14.31 | Qwell | vkdelta: Do not message people without permission. |
| 19:14.37 | vkdelta | Sorry |
| 19:14.45 | Qwell | ~ask |
| 19:14.46 | 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. |
| 19:15.09 | p3nguin_ | #asterisk: Total of 238 nicks [7 ops, 0 halfops, 0 voices, 231 normal] |
| 19:15.14 | p3nguin_ | Nope, no one here. |
| 19:15.47 | vkdelta | Ok.. I want to know about asterisk roadmap for SIPconnect1.1 .. If anyone can explain it to me.. |
| 19:16.36 | vkdelta | I am loggin to IRC after 10 years .. sorry if I am come across as wierd / rude / in-appropriate |
| 19:16.49 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.30 | CMI-DOS | hi i have a wind98 pc and which has a pci commecations adapter inside i wanted to know if i plug that into the asterisk server could i use my win98 box has a phone |
| 19:21.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:22.25 | vkdelta | I want to know about asterisk support for SIPconnect1.1 .. If anyone can explain it to me.. |
| 19:23.11 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 19:23.48 | ManxPower-work | vkdelta, we don't usually deal with specific non-asterisk products here. If it runs SIP it should run with Asterisk unless the vendor has taken steps to not work with Asterisk. |
| 19:24.49 | vkdelta | It is not a product. It is an update to SIPconnect specifications for supporting enterprise users |
| 19:25.14 | ManxPower-work | What's the RFC? |
| 19:25.47 | vkdelta | It is a specification from SIPforum |
| 19:25.49 | vkdelta | http://www.sipforum.org/sipconnect |
| 19:26.09 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 19:26.26 | vkdelta | I can give you direct link to the spec if needed |
| 19:27.15 | p3nguin_ | Sounds like regular SIP to me. |
| 19:27.19 | vkdelta | Also there is lot of work going on IETF around this area.. MARTINI and BLISS is working on it.. |
| 19:27.34 | vkdelta | it is regular SIP .. but with few differences.. |
| 19:27.52 | vkdelta | or rather some strict requirements.. |
| 19:28.22 | ManxPower-work | You might try asking on the mailing lists |
| 19:28.24 | ManxPower-work | ~mailinglist |
| 19:28.25 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 19:28.49 | vkdelta | Sure. Thanks. |
| 19:29.19 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:32.31 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:35.34 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 19:36.45 | Micc | I keep seeing rtcp halted operation not permitted in my logs, and it happens on random calls, Is there a setting in asterisk for RTCP? I checked the phones and they are all setup the same way for regular udp rtp. |
| 19:38.37 | jmcdowell | anyone have any good reads on mitel phones and their provisioning habbits ? |
| 19:38.49 | CMI-DOS | hi i have a wind98 pc and which has a pci commecations adapter inside i wanted to know if i plug that into the asterisk server could i use my win98 box has a phone |
| 19:39.13 | jmcdowell | CMI-DOS : No, based on what I understand. |
| 19:39.27 | CMI-DOS | k |
| 19:39.48 | jmcdowell | You need an FXO card to take analog to asterisk |
| 19:41.03 | CMI-DOS | no not from computer to landline from computer to computer via a 56k pci modem |
| 19:41.15 | Kobaz | ~modem |
| 19:41.15 | infobot | modem is probably (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc. Random disconnects? S10=255 sure to do the trick! |
| 19:41.28 | Kobaz | er |
| 19:41.30 | Kobaz | hmm |
| 19:41.42 | Kobaz | which one is the other one... saying modems are bad to use as fxo |
| 19:41.52 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:41.56 | Chainsaw | Kobaz: You wanted the "I just wanted to save some money using a soldering iron and a few old modems" one, right? |
| 19:42.06 | Kobaz | yeah |
| 19:42.08 | jmcdowell | Umm... |
| 19:42.11 | jmcdowell | Hrrrmm |
| 19:42.12 | Kobaz | something like that |
| 19:42.26 | CMI-DOS | i modem is a fax modem not a dialup modem |
| 19:42.31 | jmcdowell | But NICS come standard on every board these days. |
| 19:42.35 | Chainsaw | CMI-DOS: A fax modem is a special kind of dial-up modem. |
| 19:42.44 | Chainsaw | CMI-DOS: So no, you can not use that for telephony. |
| 19:42.45 | jmcdowell | is lost.. |
| 19:42.56 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:43.00 | CMI-DOS | k |
| 19:43.00 | Kobaz | CMI-DOS: don't use modems with asterisk... they generally don't have the proper hardware and interfaces |
| 19:43.02 | Chainsaw | jmcdowell: Your answer is correct. |
| 19:43.10 | Chainsaw | jmcdowell: We're just clarifying it further. |
| 19:43.22 | Kobaz | CMI-DOS: and if you do get it to work, the audio quality is going to suck |
| 19:43.42 | *** join/#asterisk Nephfl (~none@rrcs-67-78-149-118.se.biz.rr.com) |
| 19:43.46 | jmcdowell | Now that you are all breathing again.. |
| 19:43.53 | jmcdowell | Anyone with Mitel provisioning experience ? |
| 19:43.53 | Kobaz | goes back to sleep |
| 19:44.16 | Chainsaw | jmcdowell: Not me, no. Just Cisco handsets (avoid, *AVOID*) |
| 19:44.37 | jmcdowell | My phone keeps putting random header on the front of the file it's looking for from the webserver.. |
| 19:44.49 | jmcdowell | which makes it unable to locate the <mac>.cfg file. |
| 19:44.59 | jmcdowell | r u saying avoid Cisco ? |
| 19:45.10 | *** join/#asterisk jpvoip (~bd55b6c4@gateway/web/freenode/x-hsclgrybhvpfsrkd) |
| 19:45.31 | Chainsaw | jmcdowell: Yes. |
| 19:45.40 | Nephfl | im needing to set up provisioning for some cisco phones, but dont have a support contract, where can I go for firmware? |
| 19:45.45 | Chainsaw | jmcdowell: Their switches aren't great either, especially if you take value for money into account. |
| 19:46.05 | Chainsaw | Nephfl: Normally a google search will turn something up. |
| 19:46.14 | jmcdowell | Chainsaw : I have no experience with Cuxsco, but can say that Polycom has proven the same thing to me. They sound GREAT but their standards are flakey at best. |
| 19:46.16 | Chainsaw | Anyhow, my train is here. Back online later. |
| 19:46.18 | jpvoip | Hello guys, im starting my final project that is about XMPP vs SIP/SIMPLE protocols in unified communitcations solutions. Someone have sugestions? |
| 19:46.19 | p3nguin_ | ~savemoney |
| 19:46.19 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 19:46.23 | Nephfl | google didnt seem to know |
| 19:46.27 | Nephfl | but ill check aain |
| 19:46.29 | Nephfl | again |
| 19:46.30 | Chainsaw | Nephfl: Then you did not ask the right question. |
| 19:46.37 | Chainsaw | Nephfl: Something along the lines of Cisco 7960 SIP |
| 19:46.42 | Chainsaw | Nephfl: Correct as appropriate. |
| 19:47.21 | Nephfl | did that 7940 actually, but havent checked today, will look again |
| 19:47.52 | ManxPower-work | Nephfl, Get it from the same place you got your unlicensed copy of MS Office. |
| 19:48.19 | Nephfl | unlicensed copy of office? |
| 19:48.32 | jmcdowell | LOL |
| 19:48.33 | ManxPower-work | Nephfl, Cisco sells their firmware, they do not give it away for free. |
| 19:48.46 | *** part/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 19:49.30 | jmcdowell | what is this garbage? |
| 19:49.34 | jmcdowell | \ /var/www/html/\x04@+=MN_08000F194B14.cfg |
| 19:49.36 | Nephfl | says they dont sell the firmware, they sell their support contracts |
| 19:49.56 | ManxPower-work | Nephfl, I wish you the BEST of luck. |
| 19:50.15 | CMI-DOS | im going to try and hook up the fax modem |
| 19:50.19 | jmcdowell | lol |
| 19:50.24 | jmcdowell | CMI-DOS : For what? |
| 19:50.38 | jmcdowell | CMI-DOS : What exactly are you trying to accomplish? |
| 19:51.01 | CMI-DOS | because i can then plug a phone into the pc |
| 19:51.19 | jmcdowell | CMI-DOS : And talk through it? |
| 19:53.14 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 19:54.47 | jmcdowell | CMI-DOS : As stated before, unless you have a USR voice modem or something like it. There is no chance of you getting voice through that thing. Even if it did work, it would sound HORRIBLE. |
| 19:55.49 | CMI-DOS | so how can i use it to send incoming fax to my printer |
| 19:56.10 | *** join/#asterisk jmacz (~jmacz@190.25.7.33) |
| 19:56.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:56.50 | jmcdowell | For faxing, that would be something else all together. |
| 19:56.57 | jmcdowell | But are you using this over VOIP? |
| 19:57.05 | CMI-DOS | ya |
| 19:57.09 | CMI-DOS | no |
| 19:57.17 | jmcdowell | Over a regular land line? |
| 19:57.23 | CMI-DOS | no |
| 19:57.32 | jmcdowell | Jesus dood.. |
| 19:57.45 | jmcdowell | Over IP or Over a regular land line.. Which one is it ? |
| 19:57.50 | *** join/#asterisk bminish (~bminish@pdpc/supporter/professional/bminish) |
| 19:57.52 | CMI-DOS | im going to use it with a fax server |
| 19:58.01 | CMI-DOS | over ip |
| 19:58.11 | jmcdowell | I have never gotten t.38 to work |
| 19:58.18 | jmcdowell | but that's where you need to at least start looking. |
| 19:58.32 | CMI-DOS | k |
| 19:58.36 | jmcdowell | Without t.38 100% working, you don't have a prayer |
| 19:59.04 | jmcdowell | even then, you have to get an ATA device that will take your fax modem into asterisk, and that has to support t.38 as well. |
| 19:59.26 | bminish | having a bit of bother with hints in @1.6.1.17: two things, first on startup hints for parked calls always startup as State:inuse |
| 19:59.41 | bminish | second SIP devices never show inuse |
| 19:59.58 | CMI-DOS | im going to use hylafax-server |
| 20:00.47 | drmessano | What you use to process the fax isnt nearly as important as how well the TRANSPORT is going to work |
| 20:01.04 | jmcdowell | CMI-DOS : even still, your provider has to also support t.38 |
| 20:01.07 | Kobaz | heh. i just crashed a polycom phone |
| 20:01.07 | CMI-DOS | drm dont worry |
| 20:01.26 | jmcdowell | CMI-DOS : My sip provider provides a virtual fax service to me for free. |
| 20:01.35 | CMI-DOS | get the fax sent over tcp |
| 20:01.50 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 20:01.50 | drmessano | tcp using what? |
| 20:01.51 | jmcdowell | CMI-DOS : They get the fax, and e-mail it to me. |
| 20:02.06 | CMI-DOS | a fax server provider |
| 20:02.30 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 20:03.22 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 20:03.30 | *** join/#asterisk [netman] (~netman@193.153.154.9) |
| 20:03.59 | drmessano | So not voip? |
| 20:04.06 | doolittlework | hi there need some help please I have setup a sip trunk to a cisco router that in turns has a pri link to a celleler mux for outbound calls, when i send the call to the the cisco it connects to the mux and i can make a call, my problem is i dont get any ringing when i dial ou |
| 20:04.34 | doolittlework | t |
| 20:04.48 | jmcdowell | Hiy TESTICAL TUESDAY doolittlework! |
| 20:05.02 | jmcdowell | Sounds complicated |
| 20:05.51 | drmessano | doolittlework, so the call works, there's just silence until it connects? |
| 20:06.00 | *** join/#asterisk smooth_penguin (~smoove@59.95.3.100) |
| 20:06.20 | Nephfl | so, not many people get fax working these days? |
| 20:06.52 | doolittlework | drmessano: spot on no ringing |
| 20:06.57 | jmcdowell | Nephfl : So far, I have been unablo to do so but haven't tried very hard yet. |
| 20:07.17 | drmessano | You can use the "r" option in the dial command for that, I believe |
| 20:07.40 | drmessano | doolittlework: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial <--- Scroll to lower-case r |
| 20:07.43 | Nephfl | i tried for a few minutes but havent focused on fax yet myself, but it is on the list |
| 20:07.47 | doolittlework | drmessano: tried that i don't get ringing |
| 20:07.52 | drmessano | Hmm |
| 20:08.35 | bminish | anyone any 'hints' on hints ? |
| 20:08.59 | drmessano | Sounds like the call is answered by the Cisco, then passed to the mux.. so you're basically on a connected call waiting for the bridging in the Cisco |
| 20:09.53 | jmcdowell | Nephfl |
| 20:09.59 | doolittlework | drmessano: do you think the problem is on the cisco |
| 20:10.03 | CMI-DOS | i needed to run the server on the pc with the fax modem |
| 20:10.07 | jmcdowell | Nephfl: Doesn't look very good at the moment. |
| 20:11.59 | Nephfl | wonder why that is, seems like an important part of a phone system |
| 20:11.59 | drmessano | doolittlework, yes.. If the cisco is answering the call, then as far as Asterisk goes, call is completed.. the Cisco would need to generate ringing on the line from that point |
| 20:12.41 | jmcdowell | Nephfl : It's mostly commercial |
| 20:12.58 | jmcdowell | Nephfl : Hence why I won't likely go to far down that path. |
| 20:12.59 | drmessano | Nephfl, getting fax working in Asterisk isnt hard at all.. Getting FaxOverVoiceOverIP working is yet another matter |
| 20:13.08 | jmcdowell | Agreed |
| 20:13.51 | jmcdowell | The word is that with asterfax, it just works, and pretty well.. But read their crap documentation, and then try and impliment something based on the said documentation. |
| 20:14.07 | CMI-DOS | im not using asterisk im using nch axon pbx |
| 20:14.14 | jmcdowell | Correction, asterfax and asterisk 1.6 |
| 20:14.34 | jmcdowell | CMI-DOS : Man are you about to get it.. |
| 20:14.35 | jmcdowell | .lol |
| 20:14.53 | Nephfl | is it relatively easy to get fax reception working? |
| 20:15.12 | drmessano | Nephfl, yes.. and the Digium apps are well documented |
| 20:15.13 | Nephfl | i couldnt get it to detect incoming fax |
| 20:15.32 | drmessano | But again.. That doesnt address the FOVOIP issue |
| 20:15.45 | jmcdowell | If astefax is digum, then someone dropped the ball |
| 20:15.49 | Nephfl | does asterisk for faxing out? |
| 20:16.10 | p3nguin_ | http://www.english-for-students.com/Hence-Why.html |
| 20:16.26 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:16.33 | drmessano | http://www.digium.com/en/products/software/faxforasterisk.php |
| 20:16.34 | Nephfl | something is missing there...does asterisk have the ability to send faxes via voip through a print driver or email/web |
| 20:16.44 | jmcdowell | Asterfax doesn't appear to digium |
| 20:17.20 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.34 | drmessano | Asterisk, with the correct app, can send faxes.. But we're back to the "over voip" thing |
| 20:17.42 | drmessano | Fax does not work well over VOIP |
| 20:18.23 | doolittlework | any cisco experts here that know how to set the callprogress tones to asterisk answered channel? |
| 20:20.38 | ManxPower-work | doolittlework, your question makes no sense. |
| 20:20.45 | CMI-DOS | im going to carry on tomorrow |
| 20:21.32 | *** join/#asterisk ddickenson (~Drew@24-155-15-78.dyn.grandenetworks.net) |
| 20:22.28 | ddickenson | How do you set different moh classes for different contexts or users? |
| 20:22.35 | ManxPower-work | When calling from a SIP phone, before the line is answered, the phone provides the counds to the caller. |
| 20:23.00 | ManxPower-work | When calling from a SIP phone, after the line is answered, the far end provides the sounds to the caller. |
| 20:23.36 | ManxPower-work | ddickenson, You can't. But you set it on a per sip.conf entry basis. |
| 20:23.37 | ddickenson | I have two sites running off one server and one of them has some announcements that don't apply to the other site |
| 20:23.44 | ddickenson | bummer... |
| 20:23.51 | ddickenson | although that helps and I can do that |
| 20:23.54 | ddickenson | thanks |
| 20:24.00 | ManxPower-work | Obviously you can set it in the dialplan. |
| 20:24.29 | drmessano | ManxPower-work: Sounds like the Cisco router is answering the call, passing to the connected cellular mux, and bridging when the call completes.. he's getting silence in the meantime and wants ringing |
| 20:24.57 | ManxPower-work | drmessano, Sorry, I forgot to load chan_psychic.so |
| 20:25.03 | drmessano | lol |
| 20:25.30 | drmessano | He did go into some detail if you scroll a bit |
| 20:25.44 | drmessano | I had to work out the rest between the lines |
| 20:25.48 | p3nguin_ | ddickenson: You can set it per extension with SetMusicOnHold(). |
| 20:26.10 | ddickenson | ManxPower-work: I basically have different contexts in the dialplan and have everything split up and have two different macros, one for each site. Do you think it would be easier to set it up in the dialplan somehow or the sip.conf? I also have call queues for both sites but I think I can set the call queue part in queues.conf. |
| 20:26.33 | p3nguin_ | ddickenson: It is set per channel with SetMusicOnHold(). |
| 20:26.40 | ddickenson | p3nguin_: I have tried SetMusicOnHold() but it first of all says that it's a depreciated command and doesn't seem to change anything... |
| 20:27.29 | ManxPower-work | ddickenson, perhaps you should read the UPGRADE*.txt files to know what the new commands are for the old out dated commands you are trying to use. |
| 20:27.30 | ddickenson | p3nguin_: I put it in the macro just prior to the dial command thinking that would do it but no dice for some reason. Still goes with default moh |
| 20:28.16 | ddickenson | ManxPower-work: Set(CHANNEL(musicclass)=...) but not getting any luck... I tried it yesterday |
| 20:28.48 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 20:29.48 | ManxPower-work | ddickenson, Does "moh show classes" in the CLI show you what you expect to see? |
| 20:30.07 | ddickenson | ManxPower-work: hmm lemme check |
| 20:30.45 | doolittlework | drmessano: if the router guy says he can not provide ring, i can't disagree with him, for i dont know cisco, does this mean I have to get use to the silence? |
| 20:31.36 | ddickenson | ManxPower-work: no, just shows default even though I've defined it in musiconhold.conf and reloaded moh |
| 20:31.40 | doolittlework | core set verbose 8 |
| 20:31.45 | ddickenson | and res_musiconhold |
| 20:31.47 | doolittlework | lol wrong terminal |
| 20:31.59 | p3nguin_ | ddickenson: Did you reload after making changes to the moh conf? |
| 20:32.02 | drmessano | Unless I am missing some way of telling Asterisk "Answered? Hell no.. Lies.. Keep ringing.." |
| 20:32.10 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 20:32.32 | drmessano | DialAndRingForever() isn't in 1.6.x.x |
| 20:32.44 | ddickenson | p3nguin_: yeah, I've tried reloading the actual resource and moh reload as well as just reloading asterisk all together |
| 20:32.54 | drmessano | Plus, i would imagine the constant ringing on the channel would get old |
| 20:33.30 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 20:33.32 | p3nguin_ | ddickenson: Maybe there is something wrong in the conf. Paste your entire musiconhold.conf into pastebin.com. |
| 20:34.00 | doolittlework | lol thx will leave it up to them to provide ring, strange if i dial from a softphone i get ring(does the softphone provide the ring) or am i missing the plot. |
| 20:35.14 | ddickenson | p3nguin_: http://pastebin.com/N1YiwEhN |
| 20:36.26 | ddickenson | p3nguin_: sorry it's kinda messy because this is the only file I didn't start from scratch on... |
| 20:36.36 | ddickenson | which could be the problem... |
| 20:39.04 | p3nguin_ | Here's the cleaned up version: http://pastebin.com/RRSD4pX9 |
| 20:39.57 | *** join/#asterisk smooth_penguin (~smoove@59.95.8.102) |
| 20:40.11 | p3nguin_ | I don't even have a [general] in my sample conf. Is that new? |
| 20:41.41 | smooth_penguin | <CMI-DOS> im not using asterisk im using nch axon pbx |
| 20:41.49 | smooth_penguin | lol how many of these do you get per day |
| 20:42.08 | ddickenson | p3nguin_: dunno... lemme look at a clean install |
| 20:43.49 | jdoe | did something change in followme for 1.6.2? |
| 20:44.23 | ddickenson | yeah it seems to be even in my oldest install which is 1.6.0.9, nothing defined under it but it's there |
| 20:44.43 | jdoe | because with the same configs it's not working for me now. |
| 20:44.59 | jdoe | it dials out to an external number, but I never get a prompt, just dead air. |
| 20:45.41 | ddickenson | p3nguin_: sweet, that did it... Dunno what the deal was but maybe I had a bad comma or something in there! thanks |
| 20:46.11 | p3nguin_ | ddickenson: Just cleaning up fixed it, or removing the [general] did it? |
| 20:46.17 | ddickenson | cleaning it up |
| 20:46.22 | p3nguin_ | interesting |
| 20:46.38 | jpvoip | Hello guys, im starting my final project that is about XMPP vs SIP/SIMPLE protocols in unified communitcations solutions. Someone have sugestions? |
| 20:47.05 | ddickenson | well i cleaned it up then actually did a total "service asterisk restart" because no one is in the building today using the phones |
| 20:48.58 | ddickenson | p3nguin_: now I have to add some more lines to make it work on outbound calls too... |
| 20:49.48 | p3nguin_ | ddickenson: I don't see why Asterisk would care which direction the call is going, it will play MoH on the channel. |
| 20:52.21 | ddickenson | p3nguin_: I guess I don't know how to set it up properly then. I have two sites running off one t1 and same DID range. I have everything split in the dialplan so they can not talk to eachother without going out to the pstn, like they were seperate systems. (may not be the most efficient way to do it but it works) |
| 20:55.21 | ddickenson | p3nguin_: basically one of the systems has to play the "default" moh and the other has to play the new one we just added. maybe it'd be easier to explain if I just showed you...http://pastebin.com/5GN4p0Z0 |
| 20:58.17 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:00.01 | jdoe | let's try that a different way. |
| 21:00.08 | jdoe | does anyone here have followme working in 1.6.2? ;) |
| 21:00.50 | p3nguin_ | ddickenson: For starters, I do not like that you have contexts called "parkedcalls" and "features" in extensions.conf. |
| 21:01.31 | p3nguin_ | ddickenson: Next, where is the peer definition you're using to get the inbound calls? |
| 21:03.33 | ddickenson | don't you have to have the parkedcalls context to make call park work?? I agree with the features part, especially on this one but it was sorta a habit I picked up from * advanced class before I knew how anything worked |
| 21:03.49 | p3nguin_ | include => parkedcalls |
| 21:04.13 | p3nguin_ | As long as you have it configured in features.conf, that is how you enable it. |
| 21:04.20 | ddickenson | why peer definition? it's a t1 coming into the box, no sip trunks, isn't that why you would need peer definition? or maybe I misunderstood |
| 21:04.40 | ddickenson | oh, so you don't actually have to make a context just tell it to include... good to know |
| 21:04.43 | p3nguin_ | Well, what context do the incoming calls go into? |
| 21:05.55 | ddickenson | p3nguin_: the "incoming" context at the top that has about 4 include statements in it. As I said this is probably not best practice it was just how I could wrap my head around it as I'm still really learning how to put one of these together properly. Also this is the first time I've done one that services two different businesses |
| 21:06.32 | p3nguin_ | So all inbound calls from the T1 go into [incoming]? |
| 21:07.10 | ddickenson | p3nguin_: yeah |
| 21:08.04 | p3nguin_ | Give me an example of the phone number that comes in when someone calls. |
| 21:08.39 | *** join/#asterisk brezular (~brezular@adsl-dyn163.78-99-73.t-com.sk) |
| 21:09.01 | p3nguin_ | In other words, what extension does your provider send a call to? |
| 21:09.47 | ddickenson | they send 4 digits for the DID's |
| 21:10.17 | p3nguin_ | Fine. Give me one of those 4-digit extensions as an example of the number you get when a call comes in. |
| 21:10.19 | ddickenson | and they are all DID's, there is also a "main" number that I have pointed to a voice menu on one and a queue on the other |
| 21:10.27 | ddickenson | http://pastebin.com/D1JFZF87 |
| 21:10.47 | ddickenson | 1537 |
| 21:11.26 | ddickenson | 7300 is the "main number" for the [burbidge] contexts and 1688 for the [wacolung] contexts |
| 21:12.15 | p3nguin_ | I asked for that because I needed to visually follow your dialplan logic. |
| 21:12.26 | ddickenson | p3nguin_: yeah I figured |
| 21:14.08 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 21:14.15 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 21:14.49 | p3nguin_ | line 395... did you try exten => s,n,SetMusicOnHold(burbidge) instead of what is there now? |
| 21:16.52 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:17.03 | ddickenson | p3nguin_: yeah, that's when the output to the CLI told me it was being depreciated and gave me that command to put there instead. And that one works but it's only for calls that hit the macro which I don't think the outbound ones do |
| 21:17.31 | ddickenson | p3nguin_: again, it's probably not real efficient |
| 21:17.42 | p3nguin_ | Macro is also being deprecated, but you're still using it. |
| 21:17.45 | ddickenson | p3nguin_: looking for guidance! haha |
| 21:17.54 | ddickenson | really... well crap |
| 21:18.20 | p3nguin_ | Gosub() and Return() are now favored over Macro(). |
| 21:18.36 | ddickenson | sounds like I have some reading to do |
| 21:19.48 | p3nguin_ | So if SetMusicOnHold(burbidge) is working on incoming calls that reach the macro, you'll want to put it on calls that aren't subjected to the macro. |
| 21:20.32 | p3nguin_ | Err, put it on extens that don't end up going to the macro where it would otherwise be set. |
| 21:20.35 | ddickenson | p3nguin_: right, so I was thinking that if I put it in the pattern matching for things that get sent out to the dahdi interface that should work.. |
| 21:21.49 | p3nguin_ | Are you really able to set your own CallerID name on outbound calls? |
| 21:21.49 | ddickenson | p3nguin_: which brings me to another question... all that mess I had to put in to get CID and stuff set on each of the patterns I am matching etc is there a better way to do that? or will I need to also put the moh line there on each different pattern like I have been for the Set(Callerid... |
| 21:22.13 | ddickenson | p3nguin_: yeah, talked to the translation people for our service provider and they let me |
| 21:23.08 | ddickenson | well I say that. I haven't been able to test the name part... number yes for sure but I can't seem to find anyone to call about the name part, may be just wasting time with that one |
| 21:23.11 | p3nguin_ | When a SIP phone is placing a call outbound, it needs to obey the settings in sip.conf... |
| 21:23.37 | ManxPower-work | ddickenson, you can set moh class in sip.conf for each device |
| 21:23.49 | p3nguin_ | so mohinterpret=burbidge in every phone's burbidge sip definition should take care of it. |
| 21:24.09 | p3nguin_ | in every burbidge phone's sip definition, rather. |
| 21:24.26 | ddickenson | that may be the best way for a rather small install like this one |
| 21:24.34 | p3nguin_ | mohinterpret=otherclass for the other phones. |
| 21:24.54 | ManxPower-work | then you can override it in the dialplan if you need to on a case-by-case basis |
| 21:25.33 | p3nguin_ | mohinterpret= and mohsuggest= are probably useful for other conditions, as well. |
| 21:25.35 | ddickenson | so the dialplan takes priority over sip.conf always or just in this case |
| 21:25.52 | ddickenson | what does mohsuggest do? |
| 21:26.11 | p3nguin_ | The explanation is in the sample sip.conf. |
| 21:26.17 | ddickenson | ok |
| 21:29.20 | p3nguin_ | I would probably use both of those settings on every peer and not force it in the dialplan... then test. Wouldn't it be nice if that took care of it completely? |
| 21:30.08 | ddickenson | yeah that certainly would be. Any other thoughts about my methods in the dialplan? Things I could have done better? |
| 21:33.08 | ddickenson | sweet! |
| 21:33.18 | p3nguin_ | Well, I don't like that you allow the inbound caller to do dtmf transfers. |
| 21:33.39 | hackeron | hey, I've set up dahdi and it's showing up in asterisk - but what exactly do I put in my extensions.conf? -- currently I have exten => 8,8,Dial(ZAP/1/07595897203,120) - but naturally that doesn't work :) |
| 21:34.29 | p3nguin_ | SIP phones typically have Transfer keys, so no dtmf transfer needs to be enabled unless there is a special case. |
| 21:34.49 | p3nguin_ | hackeron: Where are your previous 7 priorities for extension 8? |
| 21:35.08 | hackeron | p3nguin_: various logging and monitoring |
| 21:35.23 | p3nguin_ | We generally don't use numbered priorities anymore. |
| 21:35.30 | ddickenson | p3nguin_: which one is that the big T or the little t |
| 21:35.48 | p3nguin_ | Explicitly set 1, then n for each progressing priority. |
| 21:36.02 | p3nguin_ | ddickenson: Both, since SIP phones have transfer keys. |
| 21:36.16 | p3nguin_ | ddickenson: No dtmf transfers should be done. |
| 21:36.17 | ddickenson | we have channel banks with analog lines for wall phones in lobby etc |
| 21:36.30 | ddickenson | although it probably wouldn't be missed |
| 21:36.34 | p3nguin_ | ddickenson: Are you using your macro for analogs, too? |
| 21:36.46 | ddickenson | good point... duh |
| 21:36.53 | hackeron | p3nguin_: ok, I will change that - but I tried to change ZAP/1 to DAHDI/1 and it still says app_dial.c:1721 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 21:37.09 | p3nguin_ | hackeron: Do you run Zaptel or Dahdi? |
| 21:37.26 | hackeron | p3nguin_: was using Zaptel, but upgraded to 1.6, so guess now I don't have a choice but to use Dahdi :) |
| 21:37.43 | hackeron | p3nguin_: followed this to configure Dahdi" http://www.experts-exchange.com/articles/Networking/Telecommunications/IP_Telephony/Asterisk_/Who%27s-your-DAHDI-How-to-go-from-Zaptel-to-DAHDI-in-5-Easy-Steps.html |
| 21:37.50 | hackeron | p3nguin_: but it doesn't mention what to put in extensions.conf |
| 21:38.35 | hackeron | p3nguin_: dahdi show status shows: Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 21:39.32 | ManxPower-work | You're not going to be able to dial when you have a YELlow alarm |
| 21:39.53 | ManxPower-work | *blink* How can you have a yellow alarm on an analog board? |
| 21:40.20 | hackeron | ManxPower-work: hmmm? - I donno? |
| 21:40.31 | hackeron | ManxPower-work: worked with zaptel |
| 21:41.48 | ManxPower-work | pastebin the output of dahdi_cfg -vvv |
| 21:42.07 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 21:42.35 | hackeron | ManxPower-work: http://pastie.org/847232 |
| 21:43.56 | hackeron | ManxPower-work: any ideas? |
| 21:45.09 | ManxPower-work | also make sure you don't have any zaptel config files laying around. |
| 21:46.06 | hackeron | ManxPower-work: why would that make a difference? -- I'm using 1.6, doesn't it ignore all zaptel configs? |
| 21:46.14 | ManxPower-work | put that output on something other than pastie.org |
| 21:46.15 | hackeron | ManxPower-work: it doesn't even have the zap command |
| 21:46.24 | hackeron | ManxPower-work: where? |
| 21:46.31 | ManxPower-work | nevermind, I managed to get thru to the site. |
| 21:46.50 | ManxPower-work | hackeron, zap command or no zap command, dahdi will read your old zaptel config files if they are there. |
| 21:46.52 | hackeron | ManxPower-work: better? < http://dpaste.com/166108/ |
| 21:47.29 | ManxPower-work | now pastebin "dahdi show channels" |
| 21:47.29 | p3nguin_ | pastebin.com is nicer because it allows editing of the info and then reposting it. |
| 21:47.54 | p3nguin_ | With these two, you have to copy the paste, fiddle with it locally, then paste it back. |
| 21:48.37 | hackeron | ManxPower-work: p3nguin_: http://pastebin.com/qRXSxWgq |
| 21:49.05 | ddickenson | p3nguin_: thanks again |
| 21:49.11 | p3nguin_ | Doesn't appear that you have your channel driver loaded. |
| 21:49.23 | hackeron | p3nguin_: how do I load it? |
| 21:49.37 | p3nguin_ | ddickenson: Things are going better, I take it. |
| 21:50.13 | ManxPower-work | yu have no channels loaded. |
| 21:50.23 | ManxPower-work | since the dahdi command worked, chan_dahdi.so is loaded. |
| 21:50.32 | hackeron | it's an OpenVox A400P04 and I added wctdm to /etc/dahdi/modules and restarted -- do I need to do anything else? |
| 21:50.33 | p3nguin_ | Well, chan_dahdi is probably present since you have the psuedo channel. |
| 21:50.33 | ManxPower-work | hackeron, look at /etc/asterisk/chan_dahdi.conf |
| 21:50.41 | p3nguin_ | But you didn't configure the channels. |
| 21:50.56 | ManxPower-work | hackeron, dahdi_cfg showed no errors. Therefore your kernel modules and dahdi are setup correctly. |
| 21:51.10 | ddickenson | p3nguin_: yep, sip.conf lines helped most, then I put a Set(CHANNEL(musicclass) line before my meetme and found parkedmusicclass= in features.conf... Think I have all my bases covered |
| 21:51.33 | ddickenson | p3nguin_: possibly have to figure out multiple parking lots at some point though... |
| 21:51.49 | p3nguin_ | ddickenson: I was getting ready to ask about that. |
| 21:51.54 | hackeron | p3nguin_: ManxPower-work: ah, ok, I see - so what do I need to put in chan_dahdi.conf? - I haven't touched that file |
| 21:52.18 | ddickenson | p3nguin_: which? the multiple lots or the meetme and features.conf stuff |
| 21:52.52 | p3nguin_ | ddickenson: You only mentioned parking lots. |
| 21:53.05 | ddickenson | p3nguin_: I don't want that to be the only way calls could get accidentally from one system to the other |
| 21:53.37 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 21:54.29 | ddickenson | p3nguin_: someone decides to park call and it parks on 701 and someone from the other site parks a call but instinctively picks up 701 instead of 702 and the call is on the other sys... |
| 21:55.02 | ManxPower-work | hackeron, pretty much what you had in /etc/asterisk/zaptel.conf. See the documentation. |
| 21:55.06 | p3nguin_ | ddickenson: Yep, that's going to be an issue. Plus the parked calls will be sharing the same music class, won't they? |
| 21:55.38 | hackeron | ManxPower-work: ah, ok, copied zaptel.conf to chan-dahdi.conf - seems to work |
| 21:56.03 | ddickenson | yep, that's another issue, but I could just send that to the music only class so they aren't getting site specific messages, but still doesn't help the problem of picking up a call from the other site accidentally |
| 21:56.22 | ddickenson | p3nguin_: any easy fixes for that one?? |
| 21:57.20 | p3nguin_ | I think russellb had something on his site about that, actually. |
| 21:57.44 | Micc_ | something is really getting me stumped here. I'm trying to do a dial with a macro that says wether to bridge the call or not. In the example here http://www.voip-info.org/wiki/view/Asterisk+tips+campon it shows a call to dial then right after that the voicemail command. |
| 21:58.00 | p3nguin_ | ddickenson: See if there is anything here: http://www.russellbryant.net/blog/2008/04/22/asterisk-16-now-with-multiple-parking-lots/ |
| 21:58.10 | Micc_ | The problem is, when I hangup on the called side, it goes to voicemail command anyways. even though I've answered and had my conversation. |
| 21:58.46 | Micc_ | I don't think the writers of the wiki article would have wrote that article if it was doing this to them. |
| 21:59.08 | ddickenson | p3nguin_: yeah, I've read that but didn't see anything about how it works or how to implement |
| 21:59.21 | Micc_ | And the dialstatus doesn't help me, its ANSWERED, but that doesn't tell me if they pressed 1 or not. |
| 21:59.41 | p3nguin_ | ddickenson: Set the PARKINGLOT variable on the channels much like you did with the MoH class. |
| 21:59.50 | Micc_ | I'm guessing i've got some kind of asterisk dialplan setting that is different than the guy that wrote the wiki article. Like auto fallthrough. |
| 21:59.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:00.01 | ddickenson | ahhhh |
| 22:00.03 | ManxPower-work | You cannot use WaitExten or background to get input when in a macro that is run as part of Dial. Use Read. |
| 22:00.27 | ddickenson | ok, is there a place you define the parkinglots? I would guess features? |
| 22:01.33 | p3nguin_ | parkinglots are done in features.conf, yes. |
| 22:02.56 | ddickenson | set up as seperate contexts like the [general] that is there already? like [burbidge] and then below that set all the features for that specific lot? |
| 22:04.08 | p3nguin_ | ddickenson: Maybe this will be of some use: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226114.html |
| 22:04.42 | Micc_ | When I set a variable in a macro that gets called from Dial with M option, how can I see that variable from the rest of the dialplan after the dial command is executed? |
| 22:05.29 | Micc_ | It seems to me in my testing that even if I define __FOO=1 before the dial command and set FOO=0 inside the macro, its still 1 outside the macro. |
| 22:05.41 | ddickenson | p3nguin_: ahhh yes, thanks |
| 22:06.15 | Micc_ | I'm finding it impossible to communicate anything to the rest of the dialplan from a macro called by the dial command. |
| 22:07.36 | hackeron | I remember with zaptel, I had a command to see the input and output sound levels during a call - is there an equivalent for dahdi? |
| 22:08.27 | Micc_ | This is realy starting to piss me off. |
| 22:09.54 | ManxPower-work | hackeron, have you read ANY of the docs included with DAHDI? |
| 22:10.16 | hackeron | ManxPower-work: nope, just the zaptel ones |
| 22:16.18 | ManxPower-work | hackeron, well go read them then |
| 22:16.40 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:17.41 | ManxPower-work | BTW, there should not be any zaptel docs in the DAHDI tarball |
| 22:23.18 | Micc_ | the screening samples in the dial command wiki are broken I think. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 22:23.33 | ManxPower-work | Micc_, um, you should EXPECT that. |
| 22:24.52 | Micc_ | ManxPower-work, I suppose, it is an example from 2006. |
| 22:25.19 | Micc_ | I was getting so excited about the M option in dial, until I learned it doesn't work right. |
| 22:25.23 | Kobaz | yeah, don't trust the wiki for much of anything |
| 22:25.30 | Kobaz | Micc_: M works perfectly fine |
| 22:26.05 | Micc_ | Kobaz, yeah, but how do I communicate back to the channel that called the Dial command to tell it what happened? |
| 22:26.05 | Kobaz | Micc_: i use it all the time |
| 22:26.22 | Micc_ | Kobaz, I can't find any way to set a variable that I can read outside the macro. |
| 22:26.32 | Gugge | have you tried setting _FOO in the macro, and not just FOO ? |
| 22:26.45 | Kobaz | Micc_: Set(SHARED(result,otherchannel)=myresult) |
| 22:26.46 | Micc_ | Gugge, yes and __FOO too. |
| 22:27.48 | Kobaz | Set(__ParentChannel=${CHANNEL}); Dial(foo,,M(macro)); NoOp(${SHARED(result)) macro: Set(SHARED(result,${ParentChannel})=myresult); |
| 22:28.27 | Micc_ | Kobaz, thats exactly what I needed. thanks |
| 22:28.38 | Kobaz | see, Macro works fine :P |
| 22:29.06 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 22:29.23 | Micc_ | Kobaz, your right. now that I can actually communicate outside of the macro. |
| 22:29.50 | Kobaz | there's always a way |
| 22:30.04 | Kobaz | and if there isn't a way, then you can always write another dialplan app/function to do what you want |
| 22:30.17 | Kobaz | which then makes it so that there is always a way |
| 22:33.17 | orangepower | LOLOLOL @ canada gonna lose |
| 22:34.40 | Micc_ | Kobaz, true, I try to avoid making custom changes to the asterisk source though. I don't want to have to patch every time I upgrade to a new version. |
| 22:34.49 | Micc_ | Kobaz, I've done that before and its a pain. |
| 22:35.12 | Kobaz | Micc_: submit your new features to the tracker then... that's what i do |
| 22:35.35 | Kobaz | and if you set it up right, merging your own changes from asterisk to asterisk is very very easy |
| 22:35.44 | Micc_ | Kobaz, they were custom things that no one else would need though. |
| 22:36.14 | Kobaz | i have almost 1000 lines of my own code in asterisk, that i move from version to version |
| 22:36.15 | Gugge | you never know what others might need :) |
| 22:36.21 | Kobaz | yeah |
| 22:36.23 | Micc_ | Kobaz, I haven't figured out how to patch easily. I just haven't used it much. |
| 22:36.36 | Kobaz | Micc_: what you need to do, is make your code generic enough that others can make use of it |
| 22:36.54 | Micc_ | Kobaz, yeah, I could have done that if I spent more time on it. |
| 22:36.58 | Kobaz | Micc_: the easiest thing is work with the svn checkouts of tags... run svn diff, and poof, you have a patch |
| 22:38.50 | lesouvage | I'm searching for a way to use a certain line added to the log file as a trigger to generate a .conf file that can be included in sip.conf Any suggestion on how to do this? |
| 22:39.42 | Kobaz | lesouvage: the suggestion would be to rethink your approach... that sounds like a horrible idea |
| 22:40.56 | Micc_ | Kobaz, thanks for your help, my screening macro is working perfectly now. |
| 22:41.09 | Kobaz | nice |
| 22:41.24 | *** part/#asterisk lbt (~david@Maemo/community/contributor/lbt) |
| 22:42.32 | lesouvage | Kobaz: I will explain. If Openbts (combined with a usrp it makes a gsm access point) discovers a new SIM card (actually an IMSI number) it will try to register at Asterisk but there isn't a SIP etry in sip.conf yet. So if this happens the sip entry has to be generated, asterisk needs a reload and after that the discoverd sim card is ready to go. |
| 22:43.09 | p3nguin_ | Where's the authentication mechanism in that? |
| 22:44.09 | lesouvage | p3nguin_: There is no authentication (I know, that is not good) |
| 22:45.31 | ManxPower-work | "authenticate" via IP address. |
| 22:48.38 | lesouvage | ManxPower-work: the IP adress is, with OpenBTS and Asterisk on the same machine, 127.0.0.1 And the the name of the sip entry is [IMSI<imsi number> . |
| 22:48.48 | lesouvage | ] |
| 22:50.09 | lesouvage | So I'm trying to find a way to add a sip entry after a failed registration attempt with IMSI<imsi number> |
| 22:53.41 | Kobaz | lesouvage: well, if you really want to do that... and i siggest you find a better way... write a shell script that looks at your log files and then adds the apropriate entries |
| 22:54.02 | Kobaz | suggest |
| 22:56.07 | lesouvage | Kobaz: what other option do you suggest? |
| 22:56.13 | Kobaz | if you want to do your log file approach... or you can hack up chan_sip.c, and add an event for failed registrations, since I don't think there is one |
| 22:56.30 | Kobaz | lesouvage: have a seperate system for registering your devices, which adds the apropriate items to your sip.conf |
| 22:59.03 | *** join/#asterisk andres833 (~andres833@190.159.5.103) |
| 23:00.12 | lesouvage | Kobaz: thanks, I like the chan_sip.c approach but I don't have the expertise and skills to do it :-( |
| 23:00.49 | jdoe | I don't suppose there are any developers here, or anyone with some familiarity with the source? A patch back in 1.6.1.13 has broken something for me and I'm trying to figure out if it's an issue with the patch or with something I'm doing. |
| 23:01.07 | Kobaz | jdoe: #asterisk-dev |
| 23:01.18 | jdoe | thanks./ |
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| 23:36.27 | ManxPower-work | lesouvage, since you know the IP address of the device, why use registration at all? |
| 23:36.49 | ManxPower-work | registration is to notify asterisk of the ip address of the remote device associated with that userid/password |
| 23:38.49 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 23:43.27 | lesouvage | ManxPower-work: OpenBTS is between the actual accespoint and Asterisk and all the mobile phones (actually all SIM cards) have IP number 127.0.0.1 so I guess you really need a registration. But the interfacing between OpenBTS and Asterisk is a kind of black box (at least for me) |
| 23:45.39 | florz | actually, registration is not about "ip addresses", but about SIP URIs |
| 23:46.06 | p3nguin_ | And what do those URIs consist of? |
| 23:47.12 | florz | mostly an optional userpart, a hostpart, an optional portnumber, and optional options |
| 23:47.31 | florz | have I forgotten anything? |
| 23:48.18 | p3nguin_ | Since everything seems optional except for one, what can be found in the URI is only the mandatory part is present? |
| 23:48.28 | p3nguin_ | if only, rather |
| 23:48.55 | florz | a host part (and the procotol specifier is mandatory, too, actually) |
| 23:49.07 | *** join/#asterisk andres833 (~andres833@190.159.5.103) |
| 23:49.08 | p3nguin_ | So the entire thing is optional? |
| 23:49.20 | florz | the entire what? |
| 23:49.42 | p3nguin_ | Where have you been during this conversation? We were clearly talking about the SIP URI. |
| 23:50.09 | florz | Where have you been during this conversation? I have clearly stated that the host part is not optional. |
| 23:51.17 | p3nguin_ | Back to my previous question: Since everything seems optional except for one part, what will be found in the URI if only the mandatory part is present? |
| 23:51.36 | florz | Back to my previous answer: a host part (and the procotol specifier is mandatory, too, actually) |
| 23:51.52 | lesouvage | good night, see you! |
| 23:52.02 | p3nguin_ | So then IP addressing is a rather important piece after all. |
| 23:52.22 | florz | sure, so what? |
| 23:53.59 | p3nguin_ | You tried to say that registration wasn't about the IP address, but was about the SIP URI... but if the only mandatory part of the URI is the host identifier (likely an IP address), you've clearly not made a very good argument. |
| 23:54.16 | florz | your logic is faulty |
| 23:54.49 | p3nguin_ | Sounds more like yours was from the time you tried to make the point about URIs. |
| 23:55.45 | florz | you did read what that was in reply to, didn't you? |
| 23:58.29 | florz | you see, the only mandatory part in a postal address is a zip code |
| 23:58.58 | p3nguin_ | That might get the mail to the post office, but that's pretty useless. |
| 23:59.33 | florz | so, if I state that postal addresses are not just about zip codes, that's clearly not a very good argument, as the only mandatory part of the postal address is the zip code? |
