| 00:14.02 | *** join/#asterisk fflush (~lrfurtado@unaffiliated/fflush) |
| 00:19.54 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:31.39 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 00:33.50 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 00:34.48 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:37.02 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:8048:3812:9d88:a161) |
| 00:41.50 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 00:59.24 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 00:59.32 | Micc | does multitenant parking work in 1.8.5? |
| 01:12.37 | *** join/#asterisk code (user@cant.packetflood.me) |
| 01:23.02 | *** join/#asterisk nighty^ (~nighty@74.198.9.163) |
| 01:53.31 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 01:57.04 | capt_cassimir | wtf, asterisk docs need "doxygen"... i look at the dependencies, total disk space required 600MB? |
| 01:57.53 | capt_cassimir | Are man pages not installed without this, or are "progdocs" something else? |
| 02:02.10 | capt_cassimir | sighs... |
| 02:02.17 | capt_cassimir | i really want those man pages. Okay, here we go... |
| 02:05.03 | *** join/#asterisk nightrid3r (~nightrid3@91.176.119.56) |
| 02:14.42 | WIMPy | Yes, it is something else. |
| 02:15.11 | WIMPy | It's the full source code made browsable. |
| 02:16.54 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 03:49.59 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 03:49.59 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.5.0 (2011/07/11), 1.6.2.19 (2011/06/29), 1.4.42 (2011/06/29), *-Addons 1.6.2.4, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 03:50.28 | WIMPy | Does it show somethign unter 'Extension', indicating the channels being in use? |
| 03:51.29 | WIMPy | It could sound like some channel leak, eventually eating up all available channels. |
| 03:52.07 | Linuturk | while this is going on, i don't usually see anything under extension |
| 03:52.15 | Linuturk | after a restart, that information shows up |
| 03:52.57 | WIMPy | So from that perspective it does not looke like no channels available? |
| 03:53.01 | WIMPy | Hmm. |
| 03:53.50 | Linuturk | yes, very weird |
| 03:54.12 | Linuturk | next time (if) it happens, I'll pull a pri debug at 2 |
| 03:55.18 | WIMPy | I wonder if it will start to shuffle channels for inbound calls before it reaches a no go. |
| 03:58.18 | WIMPy | You could try to add some special codes to your dialplan that enable you to dial out specific channels, instead if the group, for testing. |
| 03:58.29 | WIMPy | of |
| 04:01.32 | Linuturk | to see if certain channels are locking? |
| 04:01.38 | WIMPy | jepp |
| 04:02.08 | WIMPy | Or maybe you can still dial out certain chennels. |
| 04:04.33 | Linuturk | well, here's hoping the issue goes away with 1.8.4.4 |
| 04:13.57 | *** join/#asterisk coppice (~chatzilla@116.92.37.56) |
| 04:20.32 | *** join/#asterisk luckman212_ (~irc@pool-74-108-1-53.nycmny.fios.verizon.net) |
| 04:24.01 | ChannelZ | Hmm. Is it only possible to have one account with gtalk in asterisk? |
| 04:25.29 | capt_cassimir | ChannelZ: Someone was saying yesterday they had an asterisk setup with multiple gtalk accounts |
| 04:25.48 | capt_cassimir | one gvoice number routed to one handset, another gvoice to another handset |
| 04:27.57 | ChannelZ | hmm |
| 04:32.30 | capt_cassimir | ahng on, he gave me a link |
| 04:32.32 | capt_cassimir | lemme see if i can find it |
| 04:33.27 | capt_cassimir | ChannelZ: http://pcprob.blogspot.com/2011/03/howto-use-google-and-asterisk-for-free.html |
| 04:33.54 | capt_cassimir | i haven't looked at it yet, i'm still working my way through initial configuration and adding devices |
| 04:44.10 | ChannelZ | thanks, I'll take a look |
| 04:45.22 | ChannelZ | I guess my brain block is the gtalk.conf and how it's handled. From the dist sample it almost looks like sip peers matched with the 'username' but the docs are quite lite, so I'll just have to try it and see |
| 04:49.50 | *** join/#asterisk sourcode (~code@ppp-58-8-89-71.revip2.asianet.co.th) |
| 04:49.59 | ChannelZ | interesting. My newly built * likes to core when I do a 'reload' |
| 04:51.20 | capt_cassimir | revert whatever you changed. :D |
| 04:52.33 | ChannelZ | heh not sure what is going on quite yet |
| 04:53.12 | ChannelZ | I'm getting a missing symbol error when I try to load chan_gtalk so something goofy has happened. |
| 04:53.34 | capt_cassimir | Are you compiling from source or installing? |
| 04:53.39 | capt_cassimir | *installing from package? |
| 04:53.40 | ChannelZ | source |
| 04:53.48 | capt_cassimir | Did you do "make uninstall" |
| 04:53.55 | qop | heya |
| 04:53.56 | ChannelZ | re-configuring and making sure my build wasn't bunk |
| 04:54.16 | capt_cassimir | if you don't make uninstall, it leaves shit |
| 04:54.24 | capt_cassimir | and that old shit might be impacting the new shit |
| 04:54.29 | capt_cassimir | which causes a shit-mess |
| 04:54.29 | qop | hey guys... question here... is it realatively easy to redirect all asterisk trafic into an Avaya pbx? |
| 04:54.48 | ChannelZ | well I have SFA which it whines about but otherwise this is just a point update |
| 04:55.08 | capt_cassimir | ChannelZ: Sounds like you would just take the cable that goes into asterisk and plug it into the pbx instead. :D |
| 04:55.19 | ChannelZ | points at qop |
| 04:55.26 | qop | peep here is having a lot of problems with avaya, but currently have a running Asterisk, would it be so hard to redirect the existing Asterisk traffic into Avaya? o.o |
| 04:55.39 | capt_cassimir | oops, yeah. :D |
| 04:55.48 | qop | pokes ChannelZ |
| 04:55.50 | ChannelZ | what is Avaya? |
| 04:55.54 | qop | haha |
| 04:56.06 | qop | ok |
| 04:56.07 | ChannelZ | I mean is it hardware? |
| 04:56.19 | qop | oh, yes hardware ChannelZ |
| 04:56.31 | capt_cassimir | qop, sounds like a job for the dialplan, although i'm not sure since i'm a newb |
| 04:56.33 | qop | well, both actually |
| 04:56.38 | ChannelZ | so it's being fed T1 or something |
| 04:56.44 | qop | E1 |
| 04:56.47 | ChannelZ | ok |
| 04:57.09 | ChannelZ | So you want it still to be the interface but basically just bridge every DID into Asterisk instead? |
| 04:57.21 | qop | ChannelZ: Two E1's go directly to the Digium card |
| 04:58.05 | qop | ChannelZ: Currently this Asterisk redirects into SIP's, I want it to redirect into an Avaya instead |
| 04:58.14 | ChannelZ | whycome |
| 04:58.18 | ChannelZ | (curious) |
| 04:59.35 | qop | ChannelZ: because the company we server require Avaya atm, we are servicing them using Asterisk currently. The DID that they have provided is traced as 0 in Avaya, and according to the Avaya guy, this trace is conflictting since the "dial" is 0 |
| 04:59.43 | ChannelZ | well I guess it doesn't matter. So are you saying your * is connected to the Avaya system via E1? |
| 04:59.57 | qop | ChannelZ: nops |
| 05:00.31 | qop | ChannelZ: Providers router feeds Asterisk through 2 E1's, then Asterisk redirects to sip extensions |
| 05:00.45 | qop | E1 > Asterisk > 20 sips |
| 05:01.05 | qop | and I want it t be: E1 > Asterisk > Avaya |
| 05:01.17 | ChannelZ | ok - but how do the two connect? |
| 05:01.44 | qop | ChannelZ: I can use Brim, coaxial or EJ45 |
| 05:01.56 | qop | Avaya has brims but I can use abny adappter |
| 05:02.26 | qop | ChannelZ: for example, I could connect Avaya and Asterisk using an eth cable |
| 05:02.30 | qop | no problem |
| 05:03.09 | ChannelZ | I guess what I don't get is via what protocol would they communicate.. or maybe that is your question |
| 05:03.37 | qop | ChannelZ: ah ok... I dunno what Avaya uses :p |
| 05:03.46 | ChannelZ | so that is the question |
| 05:03.52 | ChannelZ | in which case I have no idea |
| 05:03.57 | qop | hehe |
| 05:04.01 | qop | ok :p |
| 05:04.20 | ChannelZ | I mean if the Avaya is capable of connecting to an E1 under normal circumstances, yes? |
| 05:04.41 | qop | ChannelZ: what if I want Asterisk to comunicate to Avaya as if it was a simple E1 router? |
| 05:05.03 | qop | ChannelZ: all I need is Asterisk to be middleware for avaya |
| 05:05.23 | ChannelZ | well you'd need an E1 card for Asterisk to use, and essentially it just "dials out" connecting to the other end connected to your Avaya PBX |
| 05:06.43 | qop | ChannelZ: well, I already have a digium with two empty ports in that card, I could just redirect the traffic from dahdi 1,2 into dahdi3,4 |
| 05:06.47 | qop | right? |
| 05:07.57 | ChannelZ | pretty much, yeah |
| 05:08.25 | ChannelZ | You just make your asterisk dialplan Dial() the appropriate channel(s) going to the Avaya |
| 05:08.32 | qop | ChannelZ: cool! tell me... do you have the link on the how to? |
| 05:08.36 | ChannelZ | I guess you'd just throw them all into a group |
| 05:08.43 | qop | ChannelZ: :) |
| 05:08.56 | ChannelZ | No but it shouldn't be much more complicated than the setup you already have |
| 05:09.05 | qop | ChannelZ: ok, for the details, I'd have to know more, what can I read? |
| 05:09.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 05:09.21 | ChannelZ | You have a bunch of extensions for this customers DIDs that currently dial SIP peers yeah? |
| 05:09.22 | qop | or if you want to guide me through? :p |
| 05:09.40 | qop | ChannelZ: customer has just 1 extension |
| 05:09.51 | ChannelZ | only one number? |
| 05:10.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-plkcqlrqsabswpla) |
| 05:10.45 | ChannelZ | In any case you need to start by configuring another span in your chan_dahdi for the Asterisk <-> Avaya link and get that working (crossover cable between the two?) |
| 05:10.59 | qop | yes, provider has only one number |
| 05:11.41 | ChannelZ | but multiple rollovers I assume..? |
| 05:11.50 | qop | ChannelZ: i dunno :| |
| 05:12.00 | ChannelZ | IE you can call them more than once at a time |
| 05:12.34 | qop | ChannelZ: ok let me explain... we serve the info number, we just take all calls and redirect them to agents who give info |
| 05:12.59 | qop | it is all incoming |
| 05:13.02 | qop | no output |
| 05:13.13 | ChannelZ | so yes then |
| 05:13.18 | qop | ok |
| 05:13.30 | ChannelZ | otherwise it'd be kind of pointless having an E1 with only 1 channel |
| 05:14.18 | qop | ChannelZ: sadly I know a bunch of things but a newb in Asterisk :( |
| 05:14.40 | qop | ChannelZ: what should I look for? |
| 05:14.43 | ChannelZ | anyway like I said you need to config dahdi/system.conf to make a new span for the interconnect |
| 05:14.54 | ChannelZ | and then config chan_dahdi.conf to use those channels |
| 05:15.25 | qop | ah ok! |
| 05:15.45 | ChannelZ | then it's just a matter of programming extensions.conf to dial the new channels instead of your SIP ones |
| 05:15.57 | p3nguin | snickers |
| 05:15.59 | qop | ChannelZ: where do I find propper documentation on how to on those files? |
| 05:16.05 | p3nguin | I had to read back to see what happened after I left earlier. |
| 05:16.11 | p3nguin | That's just too funny. |
| 05:16.42 | ChannelZ | well the dist versions of those files show every possible option |
| 05:16.55 | ChannelZ | How did it get configured this far? |
| 05:17.27 | p3nguin | I almost feel sick. I went and had a crapload of shrimp for supper, and I'm stuffed. Still. |
| 05:17.57 | p3nguin | I wanted to drink some beer tonight, but I don't have any room for it. |
| 05:18.08 | ChannelZ | Sounds like someone needs a potty break |
| 05:18.17 | qop | ChannelZ: want the paste? |
| 05:18.27 | ChannelZ | of what? |
| 05:18.34 | qop | hands a glass of water to ChannelZ |
| 05:18.46 | capt_cassimir | p3nguin: The last time i overate was christmas. Prime rib. I went to bed full... and woke up full. |
| 05:18.54 | qop | <ChannelZ> How did it get configured this far? the config? |
| 05:19.08 | qop | dahdi/system.conf ? |
| 05:19.27 | p3nguin | It's not that I need a potty break... I'm literally filled to the brim from supper, so there's no room for beer. I typically don't overeat, but I sure did tonight. |
| 05:19.38 | ChannelZ | it was more of a general question - who did it, if you know nothing of it how did it get running in the first place |
| 05:20.17 | qop | ChannelZ: oh nope, it was setup by someone else |
| 05:20.28 | qop | ChannelZ: and never had to deal with that server |
| 05:20.35 | qop | nor the Avaya stuff |
| 05:20.58 | qop | there was some guy in charge but he was abducted by Aliens or something |
| 05:21.04 | capt_cassimir | Do i smell a recent promotion, qop? |
| 05:21.13 | qop | capt_cassimir: haha |
| 05:21.16 | ChannelZ | I guess what I'm really asking is why does Asterisk need to be involved in this? |
| 05:21.30 | p3nguin | If you guys help him straighten out the crap, he'll be sure to get promoted! |
| 05:21.34 | ChannelZ | You are a service provider using Asterisk but don't know how? (that sounds more rude than I mean it) |
| 05:21.51 | p3nguin | That's typical. |
| 05:21.55 | qop | ChannelZ: exact!! |
| 05:21.57 | qop | :D |
| 05:21.58 | capt_cassimir | yeah no surprises there |
| 05:22.06 | capt_cassimir | i've supported apps i don't have a clue about before |
| 05:22.10 | p3nguin | I've yet to meet an ITSP that knows how to configure Asterisk for the end user. |
| 05:22.14 | qop | I am systems guy yes... but I do not have to do with telephony and stuff |
| 05:22.30 | capt_cassimir | The systems guy wears all hats. :) |
| 05:22.45 | qop | :D |
| 05:22.45 | p3nguin | Even the dirty brown ones? |
| 05:22.55 | qop | when you fall in the fang yes |
| 05:23.11 | capt_cassimir | *especially* the brown and dirty one |
| 05:23.42 | capt_cassimir | grumbles something about solaris |
| 05:23.47 | ChannelZ | well I don't have any T1/E1 knowledge beyond the 10,000ft view so I don't thinK I can help terribly with the details of configuration |
| 05:24.14 | qop | ChannelZ: maybe you dont need to at all |
| 05:24.49 | qop | ChannelZ: all I need to know is how to redirect all traffic from the two first slots in my digium into the 3rd and 4th |
| 05:24.50 | ChannelZ | If it were me personally I'd just copy what I saw and bang on it till it worked :) I guess pastebin your configs and we can look and maybe tell you something that'll work |
| 05:25.11 | ChannelZ | Well it'll be a little higher level than that but yah |
| 05:25.42 | ChannelZ | (but again I ask, why don't you just plug the E1 into the Avaya in the first place, why is Asterisk getting involved?) |
| 05:25.58 | p3nguin | I was about to ask something similar. |
| 05:26.05 | p3nguin | It doesn't sound like it's a job for Asterisk at all. |
| 05:26.32 | capt_cassimir | i'm assuming qop is in some sort of transition phase |
| 05:26.42 | ChannelZ | It could certainly do the job, I just don't know why it needs to if it's not providing any actual services |
| 05:26.45 | capt_cassimir | that's when stupid configurations are necessary, usually |
| 05:26.57 | qop | ChannelZ: all rite :D dahdi/system.conf , chan_dahdi.conf and what else? |
| 05:27.09 | ChannelZ | that'll do to start |
| 05:27.14 | qop | capt_cassimir: you assume right little padawan |
| 05:27.16 | p3nguin | But if it's just redirecting, which was the request, what's the point at all? |
| 05:27.21 | qop | ChannelZ: brb |
| 05:27.22 | p3nguin | in -> out |
| 05:27.25 | ChannelZ | exactly |
| 05:27.31 | p3nguin | No nothing. Just in and out. |
| 05:27.35 | ChannelZ | You're putting a box in the middle that is really doing nothing. |
| 05:28.03 | p3nguin | You could skip that step and do something more productive. |
| 05:29.52 | ChannelZ | damnit what the hell |
| 05:30.10 | ChannelZ | is 1.6.2.19 busted-ass or what |
| 05:30.28 | p3nguin | New release not working? |
| 05:30.49 | ChannelZ | yeah |
| 05:31.01 | ChannelZ | I get a goofy error when I try to load chan_gtalk |
| 05:31.26 | ChannelZ | but more worringly if I just do a 'reload' asterisk crashes |
| 05:41.00 | ChannelZ | well now I did it. |
| 05:41.31 | ChannelZ | I rebooted it from home and now I can't ssh into it. |
| 05:41.58 | capt_cassimir | Woot, my base install and config is tight, i can start adding devices! |
| 05:42.20 | ChannelZ | fark |
| 05:42.44 | p3nguin | Maybe you could telnet it. |
| 05:42.44 | capt_cassimir | ChannelZ: Sounds like you're ready to dispatch a datacenter tech. :D |
| 05:42.59 | ChannelZ | translation: I get in my jeep and drive to work |
| 05:43.16 | capt_cassimir | nah, see if the janitor can power cycle it. :D |
| 05:43.38 | ChannelZ | I am what you call a small business |
| 05:43.51 | capt_cassimir | also, you should invest in a console over IP device like Annex or Cyclade |
| 05:44.06 | capt_cassimir | things are fscking handy - ssh to the cyclade, open a serial connection to the device |
| 05:44.11 | capt_cassimir | we've got em in all our racks |
| 05:44.11 | ChannelZ | as in, I more or less write my own paychecks. There is no janitor |
| 05:44.43 | ChannelZ | Well I don't often bust things from home like this |
| 05:44.50 | p3nguin | The last time I priced an IP KVM, they were outrageous. |
| 05:44.50 | capt_cassimir | you can also do powere management with X11, saw an article on Hack A Day about it |
| 05:45.01 | capt_cassimir | not x11, x10, the home automation system |
| 05:45.06 | ChannelZ | I'm curious what it's doing though, I can ping it but nothing else is running |
| 05:45.18 | capt_cassimir | ChannelZ: Probably hanging out at a root prompt |
| 05:45.41 | p3nguin | ramfs death prompt |
| 05:45.42 | capt_cassimir | KVM is a waste for servers that can be adminstered via serial console |
| 05:45.55 | ChannelZ | Dunno. Either it hung before it ever restarted or something horrible has happened on boot. |
| 05:46.06 | ChannelZ | horrible-ish anyway |
| 05:47.19 | ChannelZ | bollocks. |
| 05:47.19 | capt_cassimir | a dummied down way to do it would be to put a lightweight linux server at the top of the rack with a bunch of serial cards in it |
| 05:47.21 | ChannelZ | will BBL |
| 05:47.32 | capt_cassimir | run serial lines to the various servers and enable getty on them |
| 05:47.46 | capt_cassimir | boom, you can now SSH to the terminal services box and open a console connection to the others |
| 05:48.16 | capt_cassimir | This has limited application with linux servers since the console terminal is a service, not hardware based like Sun / Cisco systems. |
| 05:48.26 | qop | ChannelZ: http://pastebin.com/1mmxj1zb http://pastebin.com/ApHasaAW |
| 05:48.30 | qop | :) |
| 05:48.38 | capt_cassimir | i think he just paged himself to work |
| 05:48.39 | capt_cassimir | :D |
| 05:48.44 | qop | :( |
| 05:49.03 | p3nguin | I wonder if he gets trouble pay for this. |
| 05:49.13 | capt_cassimir | x10 man, i'm telling you. Poor mans powere management. You use those lamp or appliance switches and HEYU in linux |
| 05:49.17 | ChannelZ | well the good news is (maybe) your spans are already configured |
| 05:49.50 | capt_cassimir | i had a script that would monitor the internet connection and power cycle the modem when it detected an outage. |
| 05:49.57 | qop | ChannelZ: who? me? |
| 05:50.00 | ChannelZ | yes |
| 05:50.13 | qop | good :p |
| 05:50.13 | ChannelZ | guess we need to see dahdi-channels.conf and chan_dahdi_additional.conf too though |
| 05:51.05 | ChannelZ | but now I am walking away |
| 05:51.10 | ChannelZ | for real |
| 05:51.26 | ChannelZ | as soon as I find my pants |
| 05:51.29 | qop | ChannelZ: http://pastebin.com/P9e7eCTP |
| 05:51.47 | qop | hides ChannelZ's pants under bed |
| 05:51.49 | ChannelZ | oh so that's all setup too |
| 05:52.09 | qop | ChannelZ: wha? so I just connect 3 and 4 into Avaya then? o.O |
| 05:52.26 | ChannelZ | well, in theory |
| 05:52.34 | qop | gotta test it! |
| 05:52.48 | ChannelZ | then you modify your extensions.conf to take the incoming calls and dial the "outgoing" channel(s) instead |
| 05:53.50 | ChannelZ | You have 2 PRIs coming into your card already for these calls? |
| 05:54.26 | qop | ChannelZ: into the Asterisk? nope, two e1's through an rj45 adapter |
| 05:54.58 | ChannelZ | eh? |
| 05:55.42 | qop | ChannelZ: two E1's are connected into an rj45 adapter that connects into the Asterisk's Digium card port 1 of 4 |
| 05:56.14 | ChannelZ | but you said you're connecting to avaya '3 and 4'...? |
| 05:56.37 | qop | ChannelZ: I was about to plug the digium's card port 4 into the avaya's brim port |
| 05:56.45 | qop | yep |
| 05:56.52 | qop | 1,2>3,4 |
| 05:57.32 | ChannelZ | hmm. blind leading the blind here |
| 05:58.05 | qop | ChannelZ: so that means you've found your pants? |
| 05:58.21 | ChannelZ | yes. back later |
| 05:58.31 | qop | kk |
| 06:00.20 | qop | ok, what port does asterisk use to output dial? |
| 06:00.25 | qop | eth port |
| 06:08.57 | *** part/#asterisk code (user@cant.packetflood.me) |
| 06:10.00 | p3nguin | Whichever one you're using. |
| 06:18.06 | *** join/#asterisk qop (~H@200.94.69.114) |
| 06:31.55 | ChannelZ | Just my luck. The HD in this thing is flaking out. |
| 06:35.33 | capt_cassimir | back up your configs! |
| 06:35.46 | ChannelZ | yeah they do automatically |
| 06:35.53 | ChannelZ | is hunting for his copy of SpinRite |
| 06:41.07 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 06:41.57 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip68-4-0-113.pv.oc.cox.net) |
| 06:49.20 | *** join/#asterisk vikapi (~quassel@124.125.34.134) |
| 07:03.19 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 07:10.24 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 07:56.30 | *** join/#asterisk deltaflyer4747 (5ef24b32@gateway/web/freenode/ip.94.242.75.50) |
| 07:58.01 | deltaflyer4747 | Hello there, may i ask for assistance / guidance / tips / howto / help? Well... Is there even a chance to create conference call, where one extension is "normal" sip phone and the other is "musiconhold" that way, that only one side of the confference hears the MOH ? Thanks in advance. |
| 07:58.29 | *** join/#asterisk [netman] (~netman@152.252.22.95.dynamic.jazztel.es) |
| 07:58.39 | capt_cassimir | delta, that sounds whack |
| 07:58.52 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 07:59.42 | deltaflyer4747 | yea i know, but i need it for one IPcam that SHOULD use sip for duplex audio, but it uses it only on phone -> camera way, i "hacked" it using MOH < vlc decoding videostream containing audio |
| 08:01.07 | *** join/#asterisk [netman] (~netman@152.252.22.95.dynamic.jazztel.es) |
| 08:01.15 | capt_cassimir | So how many participants are there in the conference call? |
| 08:01.39 | capt_cassimir | cause what you described sounds like someone picks up a hand set and gets music. |
| 08:01.40 | deltaflyer4747 | 2 + one MOH. |
| 08:01.47 | deltaflyer4747 | sortof. |
| 08:02.29 | capt_cassimir | okay lemme see if i understand |
| 08:02.38 | deltaflyer4747 | situation is as follows: camera initiates call (backcall with scriptfile) that dials extension/function that creates a call to sip phone and that sip phone should receive call from that camera (SIP) + MOH together |
| 08:02.50 | ChannelZ | Blech. Blue Pixi Stix are the WORST. |
| 08:06.03 | capt_cassimir | i think i get it now |
| 08:06.11 | deltaflyer4747 | great :) |
| 08:06.53 | deltaflyer4747 | the thing is... if i play that "moh" to the sip phone and to the camera, camera gets its own sound delayed a bit and creating nasty loop |
| 08:07.02 | capt_cassimir | right right |
| 08:07.06 | deltaflyer4747 | so i need to "route" the moh only to the other party |
| 08:07.49 | deltaflyer4747 | btw... even THIS what i done is by camera's manufacturer impossible :] |
| 08:08.03 | capt_cassimir | you fucking hacker |
| 08:08.16 | capt_cassimir | ;-) |
| 08:08.28 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 08:10.22 | deltaflyer4747 | well, i asked alot -censored- support (local, that asked directly manufacturer) and they gave me no-go... few days ago i sat onto that and got it working in ~1h :) The only thing i need is that one side MOH routing - which i hope is possible. |
| 08:10.40 | deltaflyer4747 | whoops :) i won't do adverts here ;) |
| 08:11.01 | deltaflyer4747 | but lets say its perhaps the only dome camera with > 100° wide image |
| 08:13.03 | deltaflyer4747 | capt_cassimir: do you have any idea on how to do that? |
| 08:13.23 | capt_cassimir | I've been pondering it here, and i can't think of anything. But i'm no good with dialplans. |
| 08:14.01 | capt_cassimir | You essentially want to have a conference call where there is an extra channel of input audio to ONE participant, if i understand correctly |
| 08:14.22 | deltaflyer4747 | yes. |
| 08:14.43 | capt_cassimir | and i'm not sure how to do that with asterisk, thinking in terms of "channels" |
| 08:14.50 | deltaflyer4747 | i c |
| 08:14.55 | capt_cassimir | my experience is exceedingly limited though |
| 08:15.01 | capt_cassimir | you should hang out and ask some of the other guys. |
| 08:15.32 | deltaflyer4747 | well, mine is definetly worse. And yes, i will wait, hoping someone will catch on |
| 08:17.31 | capt_cassimir | i've been trying to think of more traditional scenarios that might warrant such a setup, and i can't think of any |
| 08:17.56 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:e414:3bbc:a82:ea51) |
| 08:19.39 | deltaflyer4747 | yep, thats exactly my problem, its such an extraordinary problem i couldn't even google it :) |
| 08:19.47 | deltaflyer4747 | this is my last hope. |
| 08:31.47 | *** join/#asterisk Bipul (75d31983@gateway/web/freenode/ip.117.211.25.131) |
| 08:37.06 | Bipul | capt_cassimir: hellow |
| 08:41.14 | *** join/#asterisk radic (~radic@tmo-096-170.customers.d1-online.com) |
| 08:41.59 | *** join/#asterisk cerberus_za (~coert@dsl-185-78-221.dynamic.wa.co.za) |
| 08:50.19 | deltaflyer4747 | waiting and waiting... :) |
| 08:51.08 | *** join/#asterisk vfabi (~fabi@host-static-109-185-192-57.moldtelecom.md) |
| 08:56.53 | *** join/#asterisk [netman] (~netman@152.252.22.95.dynamic.jazztel.es) |
| 09:18.19 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 09:20.24 | *** join/#asterisk Yabanize (79496fda@gateway/web/freenode/ip.121.73.111.218) |
| 09:20.38 | Yabanize | Hello is anyone willing to help me? |
| 09:24.02 | ChannelZ | maybe |
| 09:27.53 | *** join/#asterisk cerberus_za (~coert@dsl-185-78-221.dynamic.wa.co.za) |
| 09:38.51 | Yabanize | ive got a iax2 trunk set up to a provider and i can call out but not recieve calls |
| 09:39.32 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 09:43.57 | deltaflyer4747 | Yabanize: if is it the same as sip trunk, you need to register yourself at the provider |
| 09:44.26 | Yabanize | i am registered |
| 09:44.48 | Yabanize | sip trunks are a little bit different to iax though |
| 09:46.07 | deltaflyer4747 | so if you open console (asterisk -rvvvvvv) and dial-in, you see proper response? |
| 09:46.16 | kaldemar | Yabanize: how can you not receive calls? do you see something in CLI? |
| 09:46.34 | deltaflyer4747 | kaldemar: ^^ |
| 09:49.51 | ChannelZ | Apparently he has some networking issues. |
| 09:51.44 | deltaflyer4747 | seems alike :] |
| 09:52.21 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 09:52.45 | deltaflyer4747 | and i will wait and wait if someone has any idea :) |
| 09:53.53 | deltaflyer4747 | is looking towards a BIG time spent here... |
| 09:54.53 | *** join/#asterisk madmac2501 (~MAC@247.125.216.87.dynamic.jazztel.es) |
| 09:55.14 | madmac2501 | hi, how can i download documentation from wiki for offline viewing? |
| 09:55.28 | deltaflyer4747 | file -> save page as |
| 09:56.22 | madmac2501 | that will download all? |
| 09:56.33 | deltaflyer4747 | actual page |
| 09:56.59 | madmac2501 | and something more automated to download all? |
| 09:57.58 | deltaflyer4747 | well, you can always make some script and wget <a href>s |
| 09:58.23 | madmac2501 | yes, that will do the trick |
| 09:58.34 | madmac2501 | thanks |
| 10:00.26 | deltaflyer4747 | well, there was some windows program that i cannot recall its name, but it was long time ago |
| 10:01.52 | madmac2501 | don't worry, i think that with wget is enough |
| 10:04.07 | deltaflyer4747 | ok |
| 10:22.44 | *** join/#asterisk vfabi (~fabi@host-static-109-185-192-57.moldtelecom.md) |
| 10:25.32 | *** join/#asterisk jetlag (~jetlag@pool-71-168-195-125.cmdnnj.east.verizon.net) |
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| 10:27.37 | *** join/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
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| 10:35.35 | ahfeel | Hi all |
| 10:35.49 | dym | Im having trouble registering a SIP devive to my asterisk server. UDP 10000:20000 are allowed and so is tcp 5060, but still the sip phone fails to register sometimes. |
| 10:41.43 | ahfeel | I'm doing a streamFile action with escape digits, but I dont get any digit back when trying from a softphone |
| 10:41.52 | ahfeel | does anybody have an hint ? :/ |
| 10:47.13 | *** join/#asterisk jetlag (~jetlag@pool-71-168-195-125.cmdnnj.east.verizon.net) |
| 10:54.38 | deltaflyer4747 | ahfeel: udp 5060 |
| 10:55.01 | deltaflyer4747 | wrong nick |
| 10:55.03 | deltaflyer4747 | dym |
| 10:55.14 | deltaflyer4747 | sorry about that |
| 10:57.11 | dym | oh |
| 10:57.13 | dym | thanks |
| 11:00.41 | deltaflyer4747 | np |
| 11:00.51 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 11:01.00 | deltaflyer4747 | dym: netstat -pant|grep asterisk |
| 11:02.24 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 11:04.50 | *** join/#asterisk jsman (~jsman@unaffiliated/jsman) |
| 11:06.42 | deltaflyer4747 | afk for ~2hrs, shopping |
| 11:12.06 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
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| 11:29.09 | *** part/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 11:33.59 | *** join/#asterisk nightrid3r (~nightrid3@91.176.28.67) |
| 12:01.43 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 12:02.15 | capt_cassimir | hey bipul |
| 12:02.38 | Bipul | capt_cassimir, hy i got my ISP voip service |
| 12:02.50 | Bipul | now how to configure it with my asterick |
| 12:02.55 | capt_cassimir | yeah, is it good for local calls? |
| 12:04.02 | Bipul | nops only ISD calls |
| 12:04.23 | capt_cassimir | ah. So they gave you a username, password and stuff? |
| 12:04.54 | Bipul | yes |
| 12:05.05 | Bipul | but it runs on windows platform |
| 12:05.08 | Bipul | .exe file |
| 12:05.19 | *** join/#asterisk vikapi (~quassel@124.125.34.134) |
| 12:05.55 | capt_cassimir | See, that doesn't help you! You gotta know they SIP level username, password, hostname, etc |
| 12:06.57 | Bipul | i have login alias and user name? |
| 12:07.05 | Bipul | Pin |
| 12:07.11 | Bipul | user pin * |
| 12:09.13 | capt_cassimir | So, even if you want to TRY that in asterisk, you would still need to tell asterisk which internet server to talk to |
| 12:09.21 | capt_cassimir | i believe that's the proxy server |
| 12:09.56 | Bipul | yah |
| 12:10.05 | Bipul | so what shud i do now / |
| 12:12.44 | *** join/#asterisk vikapi (~quassel@124.125.34.134) |
| 12:13.17 | capt_cassimir | like i told you earlier, 1) get docs from your provider, 2) try to figure out the hostname for asterisk and hope that the username / pin you have works, or 3) do an asterisk project WITHOUT interfacing to PSTN |
| 12:14.08 | capt_cassimir | The project i've picked for my first attempt is to create an asterisk server here with two handsets, and then to link it with an asterisk server at my friends house for a private phone network |
| 12:14.14 | Bipul | ok i am ready to do an asterisk project |
| 12:14.30 | capt_cassimir | This doesn't cost us anything other than the handsets and two PCs we had as surplus |
| 12:14.45 | Bipul | but i have a singal computer that having Linux installed |
| 12:14.56 | capt_cassimir | Okay, so just set up two handsets |
| 12:15.03 | deltaflyer4747 | or 4th option |
| 12:15.04 | Bipul | and i do have VOSystem |
| 12:15.23 | deltaflyer4747 | as he has windows app, he may install Wireshark and capture the registrar string ;) |
| 12:15.44 | capt_cassimir | delta, do you see what we're getting stuck on right now? Do you think we need to get into that? :D |
| 12:15.46 | Bipul | deltaflyer4747, ? |
| 12:16.23 | deltaflyer4747 | true :) |
| 12:16.27 | capt_cassimir | Bipul: Will there be internet access for you in the presentation hall? |
| 12:16.35 | capt_cassimir | What about wireless internet access for laptops? |
| 12:16.43 | deltaflyer4747 | Bipul: wireshark is software that captures all data that passes through your network card and allows you to see it as text |
| 12:16.46 | Bipul | yes |
| 12:17.15 | Bipul | there will i got wirless internet connection |
| 12:17.30 | Bipul | yes i have wireshark install |
| 12:17.36 | Bipul | but i dont know how to use it |
| 12:17.52 | capt_cassimir | don't worry about that |
| 12:18.11 | capt_cassimir | reading packet dumps takes some serious finesse |
| 12:18.15 | deltaflyer4747 | yet its not so hard :) |
| 12:18.36 | capt_cassimir | So do you have a laptop and a server to run asterisk on? |
| 12:18.45 | Bipul | ettercap i think i have |
| 12:18.52 | deltaflyer4747 | capt_cassimir: he just needs to create proper filter... or run wireshark, login to the voip, stop wireshark, save data and send it to you :D |
| 12:19.16 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 12:19.16 | *** join/#asterisk godmachine-x6 (~godmachin@h214.179.90.75.dynamic.ip.windstream.net) |
| 12:19.16 | *** join/#asterisk shine (~stroll@163.5.69.15) |
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| 12:19.16 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
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| 12:19.16 | *** join/#asterisk Dr{Wh0} (~sam@dev.null.nutech.com) |
| 12:19.19 | Bipul | well i can use putty to connect my windows laptop to my peronal computer via SSH |
| 12:19.30 | capt_cassimir | fuck that, i can't even get my handsets configured properly, let alone his janky VOIProvider |
| 12:19.37 | Bipul | and on my personal computer Asterisk is installed |
| 12:19.48 | capt_cassimir | ok ok, that's what i'm getting at |
| 12:19.50 | capt_cassimir | here's what you do |
| 12:20.00 | capt_cassimir | set up asterisk with a profile for two softphones |
| 12:20.15 | capt_cassimir | assuming that the softphones are on two laptops with static IPs |
| 12:20.22 | capt_cassimir | this will be easy to replicate in the lecture hall |
| 12:20.31 | Bipul | Static Ip? Why not dynamic |
| 12:20.43 | capt_cassimir | You can use your laptop with the softphone to test both user accounts that you'll create in asterisk |
| 12:20.57 | capt_cassimir | once you crack open the asterisk conf, you'll see what you want static |
| 12:21.04 | Bipul | ok |
| 12:21.15 | capt_cassimir | Come test day, you fire up your asterisk server on a 4-port ethernet switch / router, whatever |
| 12:21.19 | capt_cassimir | put it on a small lan |
| 12:21.28 | capt_cassimir | with your laptop, logged into the first account you created |
| 12:21.46 | capt_cassimir | Invite one of the other students with a laptop to download and install the software, |
| 12:21.56 | capt_cassimir | then put that laptop on the lan with your asterisk server and assign it an IP |
| 12:22.04 | Bipul | ok |
| 12:22.17 | capt_cassimir | Demonstrate a call between the two laptops using softphones. |
| 12:22.45 | WIMPy | You can do that without Asterisk :-) |
| 12:22.49 | Bipul | THats a Pure VOIP |
| 12:23.20 | capt_cassimir | WIMPy: Class project |
| 12:23.34 | deltaflyer4747 | still noone can help me with my ... whacky problem? :) |
| 12:23.53 | capt_cassimir | right bipul, that's the point: your voip provider didn't give you an asterisk friendly configuration, and may not support asterisk |
| 12:24.28 | capt_cassimir | meanwhile, you can still demonstrate your voip, sip, and asterisk skills without spending money |
| 12:24.28 | WIMPy | deltaflyer4747: The only chance I see for you is to use ChanSpy. |
| 12:24.30 | Bipul | ok |
| 12:24.43 | deltaflyer4747 | WIMPy: will google for that word :) |
| 12:24.57 | Bipul | is there any one who can sponser me for my project ? |
| 12:24.58 | capt_cassimir | Oh, i saw that module! Good call! |
| 12:25.02 | WIMPy | 'core show application chanspy' |
| 12:25.09 | deltaflyer4747 | WIMPy: or google :) |
| 12:25.28 | deltaflyer4747 | WIMPy: might do the trick, ie create call to SIP and then add chanspy for the phone |
| 12:25.31 | deltaflyer4747 | right? |
| 12:26.03 | capt_cassimir | i think that's the gist, you'd chanspy the MOH |
| 12:26.26 | deltaflyer4747 | great :) |
| 12:26.30 | WIMPy | I'm not sure yet, hou you get MOH in there. That probably involves some Bridge()ing. |
| 12:26.40 | WIMPy | And Originate()ing. |
| 12:27.05 | deltaflyer4747 | atleast i have some ground i can stand on |
| 12:27.12 | deltaflyer4747 | thanks alot! |
| 12:28.53 | WIMPy | Ok, read the full story now. Just an Originate of an MOH extension to Chanspy should work, I think. |
| 12:29.15 | deltaflyer4747 | great :) |
| 12:29.18 | deltaflyer4747 | thanks alot :) |
| 12:29.41 | deltaflyer4747 | you know, doing something the author of the product said is impossible is kinda fun :) |
| 12:33.24 | WIMPy | Who said so? |
| 12:39.03 | deltaflyer4747 | manufacturer of that IP camera - they stated, that its impossible to get full duplex audio of that camera |
| 12:39.11 | deltaflyer4747 | yet i did it :) |
| 12:40.51 | deltaflyer4747 | i mean to the sip phone |
| 12:43.45 | *** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net) |
| 12:55.05 | capt_cassimir | registration fron [phone] failed for [ip] - no matching peer found |
| 12:59.52 | deltaflyer4747 | wishes to know asterisk atleast a bit :) |
| 13:01.51 | capt_cassimir | Hey, there we go. |
| 13:02.13 | capt_cassimir | and hello world works |
| 13:02.15 | WIMPy | ~book |
| 13:02.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:02.18 | WIMPy | Try that |
| 13:02.50 | capt_cassimir | LOL wish i had known it was CC before i paid $35 in the nook store... |
| 13:04.38 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 13:04.45 | deltaflyer4747 | is there PDF of that? :D |
| 13:09.13 | WIMPy | No free PDF of the current edition, AKAIK. |
| 13:09.38 | capt_cassimir | WHere in the world does it say it's CC? |
| 13:10.19 | WIMPy | infobot just told you. |
| 13:10.29 | capt_cassimir | infobot lies |
| 13:10.30 | infobot | the cake is a lie! |
| 13:10.38 | capt_cassimir | the links all say copyright leif madson |
| 13:12.04 | deltaflyer4747 | well, i think i need to study it, i don't even know how to create exten that listens to that chanspy and then dials the sip chan :) |
| 13:16.57 | *** join/#asterisk [netman] (netman@152.252.22.95.dynamic.jazztel.es) |
| 13:18.44 | deltaflyer4747 | yea, i know, lame :) |
| 13:45.58 | *** join/#asterisk jakent (~john@c-68-57-33-155.hsd1.va.comcast.net) |
| 13:49.03 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 13:50.50 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:02.28 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:09.12 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 14:13.07 | deltaflyer4747 | WIMPy: any hint on that chanspy/extenspy for the MOH? because i am too lame to find out how to get the moh to the chanspy:( |
| 14:15.17 | WIMPy | You have to create an extension for one of them (I suggest the MOH) and then Originate to that via a local channel and call application chanspy. |
| 14:16.05 | WIMPy | How is the call set up? |
| 14:21.36 | deltaflyer4747 | well. just testing so far as i am really lame at asterisk |
| 14:22.26 | WIMPy | You mentioned a script being involved? |
| 14:23.05 | deltaflyer4747 | thats the plan |
| 14:23.25 | deltaflyer4747 | i know of a "scriptfile" being copied to particular directory initiating a call. |
| 14:23.49 | deltaflyer4747 | i really cant remember details, it was year ago i played with it ... |
| 14:24.40 | WIMPy | You can probably extend that to set up the 2nd call as well. |
| 14:25.18 | deltaflyer4747 | i need it because the camera itself has no button to dial, so i will "hack" the external button via HW interface to the server, launching the script copying the call file ... |
| 14:25.58 | WIMPy | It's your doorphone? |
| 14:26.47 | deltaflyer4747 | security camera for the lift, audio is needed in case of the breakdown of the lift |
| 14:27.12 | deltaflyer4747 | to dial serviceman etc... |
| 14:27.18 | WIMPy | Ok. |
| 14:27.31 | WIMPy | But What is the music good for? |
| 14:27.42 | deltaflyer4747 | okay, from the begining. |
| 14:29.48 | deltaflyer4747 | that IPcam have 2way audio. TO camera you use SIP (just dial the camera extension and speaker connected to the camera will produce sound from SIP chan). FROM camera is more difficult - i had to rig the VLC to convert the streaming video from the camera to the MOH channel. Thats the only way i know how to get audio from the camera to the * |
| 14:30.29 | WIMPy | I see |
| 14:31.06 | deltaflyer4747 | application=/usr/local/bin/vlc rtsp://192.168.2.21/live.sdp -I dummy -q --sout "#transcode{vcodec=none,acodec=s16l,ab=8,channels=1,samplerate=8000}:std{access=file,mux=raw,dst=-}" 2>/dev/null |
| 14:31.17 | deltaflyer4747 | (if anyone needs that ;) ) |
| 14:31.24 | WIMPy | You should only use chanspy, no conference then. |
| 14:32.21 | deltaflyer4747 | the thing is i need to dial(sip/cam1) + chanspy(musiconhold) - dial is outbound only, musiconhold is inbound only. |
| 14:32.43 | Bipul | so no one going to sponser my project :( |
| 14:32.44 | WIMPy | Just place a call from the cam to the phone and let the MOH channel whisper to the phone via ChanSpy. |
| 14:33.17 | WIMPy | It will be two calls. |
| 14:33.32 | deltaflyer4747 | as i said, i don't know how to chanspy musiconhold :) |
| 14:34.12 | WIMPy | Like I already said: Vreate an extension for it and dial that extension via a local channel. |
| 14:34.20 | WIMPy | C |
| 14:35.01 | WIMPy | You should have used a doorphone thing. |
| 14:35.43 | deltaflyer4747 | wimpy: that would be another hardware that needs to be bought with 0 budget :-( |
| 14:36.32 | WIMPy | Because work is worthless. |
| 14:37.10 | deltaflyer4747 | WIMPy: wouldn't dialing that moh extension add its audio to existing call ? (just asking, keep that i-beam down please) |
| 14:37.52 | WIMPy | That's why you should use ChanSpy in whisper mode instead of a conference. |
| 14:38.25 | deltaflyer4747 | now i am lost :-/ |
| 14:38.41 | deltaflyer4747 | nevermind... i'll get it... somehow... |
| 14:38.55 | WIMPy | You set up the call between the cam and the phone. |
| 14:39.31 | deltaflyer4747 | exten => 81,1,Dial(sip/cam1) |
| 14:39.36 | WIMPy | Then you call Chanspy on the channel going to the phone from the moh extension. |
| 14:40.20 | deltaflyer4747 | err... |
| 14:41.36 | deltaflyer4747 | i am not sure how to create that virtual channel |
| 14:42.03 | WIMPy | The same way you create the first call. |
| 14:43.16 | WIMPy | You could do it via the dialplan when the 1st call is answered if you want the advanced version. |
| 14:43.57 | deltaflyer4747 | exten => 81,n,Dial(sip/1002) #1002 = exten for moh |
| 14:44.04 | deltaflyer4747 | thats gonna be the next line? |
| 14:44.47 | *** join/#asterisk irroot (~irroot@41.54.179.201) |
| 14:45.18 | WIMPy | No. That's not a SIP channel. You need a local channel. |
| 14:45.50 | WIMPy | Vou can any any extension in your dialplan like a device via a local channel. |
| 14:47.07 | deltaflyer4747 | ok, s/sip/local |
| 14:48.30 | WIMPy | But you don't Dial, you have to Originate. |
| 14:48.38 | deltaflyer4747 | okay |
| 14:48.45 | WIMPy | It will be two calls. |
| 14:51.46 | deltaflyer4747 | so... exten => 81,n,originate(sip/1002,extension,11@default) |
| 14:51.49 | deltaflyer4747 | right? |
| 14:52.24 | deltaflyer4747 | s/11/81 |
| 14:53.34 | WIMPy | Take the easy one first. Create the 1st call via your .call file. |
| 14:54.19 | deltaflyer4747 | for testing purposses i am dialing directly from phone (exten 11) to the camera (exten 81) |
| 14:55.50 | WIMPy | Then try it manually from *CLI. Originate local/mohext application chanspy sip/1002,Eqw |
| 14:56.30 | deltaflyer4747 | ok |
| 14:57.00 | WIMPy | You need to have that moh exten. |
| 14:57.25 | WIMPy | AFK, BBL |
| 14:57.27 | deltaflyer4747 | 1002 is MOH, 11 is deskphone, 81 is sip to cam |
| 14:58.10 | WIMPy | Originate local/1002 application chanspy sip/11,Eqw |
| 14:58.40 | deltaflyer4747 | ok |
| 15:00.19 | deltaflyer4747 | while dialing the 81 from 11, right ? |
| 15:03.00 | deltaflyer4747 | when i do, call fails. |
| 15:03.40 | deltaflyer4747 | http://pastebin.com/FkhEztgt |
| 15:11.37 | Linuturk | WIMPy: I got lucky this weekend. someone is at the office making calls, so it will be under some sort of load today. I hope I won't see a lockup |
| 15:20.58 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
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| 15:32.01 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 15:41.08 | *** join/#asterisk Elit3 (~Elit3@41.35.199.206) |
| 15:44.49 | *** join/#asterisk cyborg-one (1000@31.31.104.164) |
| 15:58.53 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 16:12.20 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:24.45 | deltaflyer4747 | WIMPy: when you come back, please ping me, i think i got it with those 2 calls, but ... |
| 16:27.27 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 16:46.43 | *** join/#asterisk bavi (~avi@c-67-180-200-95.hsd1.ca.comcast.net) |
| 16:46.58 | deltaflyer4747 | DARN i hate myself :-/ |
| 16:47.42 | p3nguin | bipul: What kind of sponsorship are you looking for? |
| 16:48.02 | bavi | Hey there, an ilbc question; what is the current status of ilbc in asterisk ? not supported ? |
| 16:48.36 | p3nguin | It should be supported just fine. Remember to select the codec in menuconfig when you're building asterisk. |
| 16:49.02 | bavi | supported in what way? currently the codec is half broken |
| 16:49.07 | bavi | not doing 20ms |
| 16:49.17 | bavi | and i am not even 100% sure that it does 30ms correctly |
| 16:49.36 | bavi | maybe there is a 3rd party codec ? |
| 16:49.45 | p3nguin | I wasn't aware it was broken; I just know it is available in the codec list in the menu. |
| 16:50.52 | bavi | I see... What is asterisk's policy with external development ? what If i'd like to fix it? is it possible ? |
| 16:51.39 | bavi | heck i am not even sure whats the project's status, is it developed by the community or just open source ? |
| 16:51.49 | p3nguin | If you want to fix something that can be fully implemented into Asterisk for everyone, you can simply submit your patches and if it's good enough, it'll probably get added. |
| 16:52.10 | bavi | who do i commit my patches to ? |
| 16:52.23 | bavi | w0z 'the dude' ;-) |
| 16:52.36 | p3nguin | This might be a question better asked in #asterisk-dev. |
| 16:52.45 | p3nguin | or here, but on a weekday. |
| 16:52.51 | bavi | outch!! there is asterisk-dev |
| 16:53.04 | bavi | Thank you my lord. |
| 16:53.09 | Bipul | p3nguin, i need voip account for my asterik so that i can make a call |
| 16:53.26 | p3nguin | bipul: Whom do you intend to call? |
| 16:53.41 | Bipul | i can show peoples at presentation hall about this asterisk and in return i will put the name s of that provider |
| 16:53.53 | Bipul | to my freinds at presentation hall |
| 16:54.05 | Bipul | i just make hardly 15 calls |
| 16:54.08 | p3nguin | to mobile phones? |
| 16:54.42 | Bipul | yes but voip provider shud be no-indian |
| 16:55.07 | Bipul | yes mobile phone may be i can call ISD Calls |
| 16:55.12 | p3nguin | Let me look at my rates for calls to India mobile phones. |
| 16:55.24 | Bipul | :) thanks :) |
| 16:55.52 | Bipul | sir p3nguin in india there is restiction to voip provider to hit the local PSTN server |
| 16:56.08 | Bipul | only ISD calls are allowed |
| 16:58.07 | p3nguin | Phone numbers starting with 9192, 9193, 9194, 9197, 9198, and 9199? |
| 16:58.48 | Bipul | 91 is india code |
| 16:59.08 | p3nguin | Are you demonstrating VoIP to PSTN/Mobile... or are you demonstrating Asterisk? |
| 16:59.50 | Bipul | yes sir that's what i want to do :) |
| 17:00.03 | Bipul | Voip TO pstn |
| 17:00.17 | p3nguin | You aren't trying to demonstrate Asterisk? |
| 17:01.41 | p3nguin | If you are not trying to demonstrate Asterisk, I can get you a VoIP account where you will connect your SIP phone and make VoIP calls to the PSTN and mobile phones. |
| 17:01.58 | p3nguin | I can't give you Asterisk access, though. |
| 17:02.26 | Bipul | no i want's to do it through Asterisk |
| 17:02.49 | p3nguin | You'll do it through Asterisk, but you won't have login access to Asterisk. |
| 17:02.50 | Bipul | no sir my aim is dig asterisk |
| 17:03.14 | p3nguin | Are you going to set up Asterisk on your site? |
| 17:03.25 | Bipul | not mine site mine website |
| 17:03.38 | Bipul | sory on my computer |
| 17:04.17 | p3nguin | If you set up Asterisk on your computer, you can configure your asterisk to use me as a provider. That sounds like it should work for you. |
| 17:04.30 | Bipul | alright :) |
| 17:05.08 | p3nguin | What is the date you will do the demonstration? |
| 17:05.33 | Bipul | it is in between 29 to 2 august |
| 17:06.08 | Bipul | in between 29 jul to 2 august |
| 17:07.19 | p3nguin | Okay, I will be your VoIP service provider (ITSP), and you will be able to call India mobile phones. Do you have Asterisk set up already so you can test it today? |
| 17:07.49 | Bipul | yes i do have asterisk installed |
| 17:09.00 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:14.58 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 17:30.30 | p3nguin | bipul: Still here? |
| 17:44.15 | *** join/#asterisk brezular (~brezular@adsl-dyn242.78-98-159.t-com.sk) |
| 17:45.47 | *** join/#asterisk brezular (~brezular@adsl-dyn242.78-98-159.t-com.sk) |
| 17:49.21 | Bipul | yes |
| 17:49.32 | p3nguin | bipul: PM me. |
| 17:49.44 | Bipul | ok |
| 17:50.18 | deltaflyer4747 | is there somebody willing to help me with setting up chanspy via .call file? |
| 18:00.17 | *** join/#asterisk GuySoft (~guysoft@109.226.6.237) |
| 18:00.46 | GuySoft | hi all, - I have a radio plugged in to a sound card - I there a way to make it record only when it passes a certain DB level? |
| 18:01.45 | irroot | GuySoft on asterisk ?? |
| 18:01.59 | irroot | look at the silence/noise functions in main/dsp |
| 18:02.21 | GuySoft | irroot, any software will do, but I thought i might start here . on #hamradio no one knows how to do this.. |
| 18:02.44 | GuySoft | irroot, asterisk is mostly more up to date than most audio software around |
| 18:02.46 | irroot | lol those old farts ... /me ZR6OLG |
| 18:03.18 | GuySoft | irroot, 4Z7GAI :) .. Im new to the hobby and keep bumping in to OMs |
| 18:04.06 | irroot | not sure asterisk is ideal for it but there some decent DSP bits also look at spandsp and the linux ham radio cook book ive been silent for many years though |
| 18:06.27 | GuySoft | irroot, spandsp - is that a program? |
| 18:07.01 | irroot | nah more a lib you can make programs from it |
| 18:07.24 | irroot | http://www.soft-switch.org |
| 18:10.43 | *** join/#asterisk radic (~radic@tmo-096-146.customers.d1-online.com) |
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| 18:45.06 | deltaflyer4747 | damn :( |
| 18:46.35 | deltaflyer4747 | nobody? |
| 18:51.01 | fulcan | what is the easiest method to write to the agi/ami/api? I am getting my butt handed to me with pyst and StartPy because of lack-o-documentation just trying to laund a piece of test code. Haven't tried py-asterisk yet. I guess I could write it in C++, php looks like a pain to setup. Can't someone refer me to a good doc/method of write my first piece of code to the ami/agi/api? |
| 18:52.00 | *** join/#asterisk ttyS1 (~hfly@c-76-26-54-58.hsd1.fl.comcast.net) |
| 18:57.04 | seraphie | fulcan: checkout the asterisk test suite. there are reams of tests written using StarPy. |
| 19:02.03 | *** join/#asterisk radic (~radic@tmo-096-251.customers.d1-online.com) |
| 19:03.11 | deltaflyer4747 | Anyone able and willing to guide me through setup of (exten) + (chanspy on MOH) + exten ? |
| 19:03.22 | deltaflyer4747 | please, i'm going crazy of that :( |
| 19:03.43 | p3nguin | I really have no idea what you're trying to do. Your question makes very little sense to me. |
| 19:03.54 | p3nguin | Can you try to clarify what your intention is? |
| 19:04.07 | deltaflyer4747 | p3nguin: well, i was asking for that since morning... i can repeat... sec, writing, its a bit long-ish |
| 19:04.23 | p3nguin | Just give me the short of it. |
| 19:04.42 | p3nguin | You're trying to connect something to something else. I know that much. |
| 19:04.59 | p3nguin | I briefly read what you were talking about earlier, but I couldn't follow it. |
| 19:08.48 | deltaflyer4747 | http://deltaflyer.cz/ast.txt |
| 19:10.21 | p3nguin | So exten 1001 runs MusicOnHold(camera) ? |
| 19:11.32 | p3nguin | What does exten 81 run? Dial(SIP/camera) ? |
| 19:11.41 | deltaflyer4747 | ano |
| 19:11.42 | deltaflyer4747 | yes |
| 19:11.44 | deltaflyer4747 | sorry... |
| 19:11.47 | p3nguin | And exten 11 runs Dial(SIP/deskphone) ? |
| 19:12.02 | deltaflyer4747 | you can say that, yes |
| 19:12.34 | deltaflyer4747 | + i'd need that through the .call file :] |
| 19:12.54 | p3nguin | I don't know about need. call file could do it, though. |
| 19:14.48 | deltaflyer4747 | i mean - the camera will be the initiator, yet the camera has no buttons to "dial" so i will rig HW button to server calling a script creating an asterisk .call file |
| 19:17.13 | deltaflyer4747 | ofc it can create more than one file :) |
| 19:18.52 | deltaflyer4747 | so... any help will be appreciated |
| 19:22.54 | deltaflyer4747 | p3nguin: still there with me please? |
| 19:25.42 | p3nguin | This process is started by pressing a button near the camera? |
| 19:26.04 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 19:26.14 | gavimobile | what does it mean when I restart amportal and it says Asterisk ended with exit status 0 |
| 19:26.31 | deltaflyer4747 | yes, it does. That calls the script producing the .call file |
| 19:26.49 | deltaflyer4747 | or files |
| 19:27.43 | *** join/#asterisk TimeRider (steve@5ac31889.bb.sky.com) |
| 19:27.50 | p3nguin | When the button is pressed, what happens? |
| 19:28.02 | p3nguin | Yes, I know the call file is produced. |
| 19:28.09 | p3nguin | But what happens? |
| 19:29.18 | gavimobile | what does it mean when I restart amportal and it says Asterisk ended with exit status 0 |
| 19:29.24 | p3nguin | You already said that. |
| 19:29.36 | p3nguin | No one new has joined. |
| 19:29.46 | gavimobile | my bad |
| 19:29.56 | p3nguin | And there hasn't been any significant scroll. |
| 19:31.04 | deltaflyer4747 | p3nguin: well, thats what i need to figure. |
| 19:31.27 | deltaflyer4747 | i need to call 11 -> 81 and enabling chanspy on 1001 for 11 |
| 19:32.31 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 19:32.44 | seraphie | gavimobile: exit status 0 means Asterisk shut down normally |
| 19:33.04 | deltaflyer4747 | btw sorry for my bad english, but i am quite tired after whole day trying to get it done :( |
| 19:33.42 | gavimobile | seraphie: so its normal? |
| 19:33.45 | *** part/#asterisk jsman (~jsman@unaffiliated/jsman) |
| 19:33.51 | p3nguin | deltaflyer4747: When someone presses the button at the camera, you want your phone to call the camera? |
| 19:33.54 | gavimobile | I don't remember seeing it previously |
| 19:33.54 | seraphie | Yep |
| 19:34.03 | gavimobile | seraphie: ok, great thanks |
| 19:34.54 | deltaflyer4747 | p3nguin: or the other way... doesn't matter, problem is that routing. But basically yes. |
| 19:35.25 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 19:36.12 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 19:36.23 | p3nguin | deltaflyer4747: Are you wanting help building the call file to establish the call between the camera and the phone? |
| 19:37.58 | deltaflyer4747 | thats easy. |
| 19:38.10 | p3nguin | Then what part are you having trouble with? |
| 19:38.12 | deltaflyer4747 | i need a help on routing extensions (dialplan) |
| 19:38.24 | p3nguin | Well, extensions aren't something that gets routed. |
| 19:39.11 | p3nguin | So let's start with the button being pressed and work inward. Tell me where you're stuck. |
| 19:39.26 | p3nguin | Someone presses the button. It activates a script, which generates a call file. |
| 19:39.30 | deltaflyer4747 | you see, i can easily create a call between 11 and 81 thus person at the camera will be able to hear me at deskphone. But the problem is that i need to route chanspy on the same channel as this call (11-81) listening to 1001 |
| 19:40.00 | p3nguin | The call file causes two devices to ultimately be connected. |
| 19:40.20 | deltaflyer4747 | yes, i could create conf. call calling 81 and 1001 together, but then the 81 will get audio from 1001 as well, which is something i must avoid. |
| 19:40.30 | deltaflyer4747 | p3nguin: true |
| 19:40.55 | p3nguin | Now that SIP/camera has been connected to SIP/deskphone, you have one-way audio? |
| 19:41.05 | deltaflyer4747 | yes. |
| 19:41.20 | deltaflyer4747 | only from deskphone to camera. |
| 19:41.23 | p3nguin | SIP/camera can hear SIP/deskphone, but SIP/deskphone hears nothing? |
| 19:41.28 | deltaflyer4747 | yes. |
| 19:41.40 | p3nguin | Where is the audio from SIP/camera? |
| 19:41.42 | deltaflyer4747 | to be able to hear camera, you have to launch music on hold |
| 19:41.52 | deltaflyer4747 | moh(camera) - exten 1001 |
| 19:42.07 | p3nguin | Okay, so SIP/camera's audio has already been inserted into the MoH at this point? |
| 19:42.16 | deltaflyer4747 | yes |
| 19:42.25 | deltaflyer4747 | took me awhile rigging that VLC for that |
| 19:42.29 | p3nguin | And now you just need to get SIP/deskphone to listen to the stream. |
| 19:43.23 | p3nguin | If you try to make a second call from SIP/deskphone, SIP/camera isn't going to be able to hear the deskphone anymore. |
| 19:43.44 | deltaflyer4747 | unless i join them on the deskphone |
| 19:43.49 | p3nguin | not without some type of conference, since the deskphone can only have one active call at a time. |
| 19:43.55 | deltaflyer4747 | but then the 81 will "hear" the MOH as well. |
| 19:44.00 | deltaflyer4747 | which i have to avoid. |
| 19:44.03 | p3nguin | and you don't want that. |
| 19:44.07 | deltaflyer4747 | well... |
| 19:44.42 | deltaflyer4747 | imagine placing a mic connected to amplifier near the speaker of the same amplifier. |
| 19:44.46 | deltaflyer4747 | that would be the result. |
| 19:44.55 | deltaflyer4747 | sound loop. |
| 19:45.44 | p3nguin | Where does the feedback come from? |
| 19:45.53 | p3nguin | camera speaker and mic together? |
| 19:45.55 | deltaflyer4747 | thats why i need to route the audio from MOH(camera) only to the deskphone (11). WIMPy suggested Chanspy / Extenspy, but that made me totally silly. |
| 19:46.00 | deltaflyer4747 | yes |
| 19:46.17 | deltaflyer4747 | well, i need to use it as another "phone" |
| 19:46.24 | p3nguin | You could use ChanSpy() from SIP/camera to listen to a channel without SIP/camera being heard. I don't know if that would work for you or not. |
| 19:46.41 | deltaflyer4747 | (security cam inside elevator cabin used in case of emergency to call for service) |
| 19:46.58 | p3nguin | ChanSpy() would allow a caller to spy on another channel, and have one-way audio. |
| 19:47.06 | deltaflyer4747 | p3nguin: yes. But thats what i don't know why. |
| 19:47.13 | deltaflyer4747 | how to put it all together. |
| 19:47.56 | p3nguin | Now that I think about it, that won't work either. |
| 19:48.09 | deltaflyer4747 | why not? |
| 19:48.09 | p3nguin | Because the cam will still hear audio and the mic would still pick it up. |
| 19:48.16 | deltaflyer4747 | nope |
| 19:48.20 | p3nguin | no? |
| 19:48.30 | deltaflyer4747 | chanspy gives the audio only to one party, right? |
| 19:48.38 | p3nguin | yes |
| 19:48.41 | deltaflyer4747 | so |
| 19:48.55 | deltaflyer4747 | chanspy output is "routed" to 11 |
| 19:48.58 | deltaflyer4747 | only |
| 19:49.05 | deltaflyer4747 | right? |
| 19:49.20 | p3nguin | There's no routing going on. This isn't networking. |
| 19:49.23 | deltaflyer4747 | let me check that, fortunatelly i can do that checking |
| 19:49.37 | p3nguin | ChanSpy() connects to a channel that is active. |
| 19:50.15 | p3nguin | If I pick up my phone and call an extension that runs ChanSpy() on a channel, I hear that channel on my phone. |
| 19:51.03 | deltaflyer4747 | i see the problem now |
| 19:51.41 | deltaflyer4747 | darn |
| 19:52.11 | p3nguin | I still can't quite figure out how your deskphone is going to send its audio to SIP/camera AND also listen to MoH at the same time unless they are conferenced toghether. |
| 19:52.25 | deltaflyer4747 | they can be |
| 19:52.33 | deltaflyer4747 | but the audio from MOH cannot go to camera. |
| 19:52.43 | deltaflyer4747 | thats the only problem i'm facing. |
| 19:53.47 | p3nguin | MeetMe() can do that, I guess. |
| 19:53.57 | deltaflyer4747 | really? |
| 19:54.03 | p3nguin | You can make SIP/camera join a MeetMe conf using the t option. |
| 19:54.09 | p3nguin | <PROTECTED> |
| 19:54.22 | p3nguin | But I'm not sure if that helps. |
| 19:54.52 | deltaflyer4747 | it won't |
| 19:55.06 | p3nguin | That might be the wrong thing to use... since you actually want SIP/camera to hear the deskphone. |
| 19:55.12 | deltaflyer4747 | yes. |
| 19:57.04 | deltaflyer4747 | but what about option m |
| 19:57.14 | deltaflyer4747 | 'm' - The conference is in so called monitor mode ( Only listen, no talking) |
| 19:57.28 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 19:57.36 | deltaflyer4747 | is there an option for manager of the conference to listen to only one person? |
| 19:57.38 | p3nguin | That will allow everyone to listen to ... what? |
| 19:57.48 | deltaflyer4747 | to manager |
| 19:57.49 | deltaflyer4747 | ? |
| 19:57.56 | p3nguin | You could probably allow one device to talk. |
| 19:58.02 | deltaflyer4747 | to everybody |
| 19:58.07 | p3nguin | yes |
| 19:58.15 | p3nguin | But I'm not sure if that will help either. |
| 19:58.29 | deltaflyer4747 | manager = deskphone |
| 19:58.46 | deltaflyer4747 | other two parties = 1001 (moh) and 81 (sip to camera) |
| 19:58.56 | deltaflyer4747 | everybody will listen to the manager |
| 19:59.04 | deltaflyer4747 | = audio from deskphone to camera |
| 19:59.09 | p3nguin | If you allow the deskphone to talk in the conf, the camera will hear it and the deskphone will hear nothing. |
| 19:59.16 | deltaflyer4747 | yes. |
| 19:59.31 | deltaflyer4747 | is there an option that the manager can hear to ONE participant ? |
| 19:59.32 | p3nguin | And you don't have the MoH stream available to anyone |
| 19:59.41 | p3nguin | I don't know. |
| 20:00.07 | deltaflyer4747 | ic |
| 20:00.17 | p3nguin | I have another idea. |
| 20:00.27 | deltaflyer4747 | hit me |
| 20:00.39 | p3nguin | It requires creating a channel where MoH is playing. |
| 20:00.50 | deltaflyer4747 | can do |
| 20:00.54 | deltaflyer4747 | (tested) |
| 20:01.07 | p3nguin | Then you have to use ChanSpy() on that MoH channel to connect to the channel where the deskphone is... |
| 20:01.12 | p3nguin | using the w option. |
| 20:01.13 | deltaflyer4747 | ie calling from 1001 to another MOH(silent) on 1101 |
| 20:01.39 | p3nguin | Pretend that MoH is a phone for a minute. |
| 20:01.45 | deltaflyer4747 | thats the original idea |
| 20:01.46 | p3nguin | SIP/deskphone is on a call to SIP/camera. |
| 20:01.50 | deltaflyer4747 | yes |
| 20:02.02 | deltaflyer4747 | wait |
| 20:02.04 | deltaflyer4747 | not w |
| 20:02.05 | deltaflyer4747 | but W |
| 20:02.16 | deltaflyer4747 | pushing the audio TO the extension 11 |
| 20:02.29 | deltaflyer4747 | that MIGHT work ! |
| 20:02.32 | *** join/#asterisk okei (bc81c390@gateway/web/freenode/ip.188.129.195.144) |
| 20:02.42 | okei | hello guyys, i have some question |
| 20:02.53 | deltaflyer4747 | will try it in a few |
| 20:03.02 | okei | in my extension when i write special extension < i > this is match any invalid extension yes? |
| 20:03.05 | p3nguin | You want SIP/deskphone to listen to moh, so option w. |
| 20:03.19 | deltaflyer4747 | what i meant is opposite direction |
| 20:03.19 | p3nguin | If SIP/moh uses ChanSpy, it can listen to the call, and using w talk to SIP/deskphone only. |
| 20:03.25 | deltaflyer4747 | 11 calls 81 |
| 20:03.38 | deltaflyer4747 | 1001 connects to 1101 (MOH(silent) ) |
| 20:04.22 | deltaflyer4747 | scratch that, i don't know how to connect two channels |
| 20:04.28 | p3nguin | When the button is pressed, SIP/camera is joined with SIP/deskphone in a call. |
| 20:04.34 | deltaflyer4747 | yes |
| 20:04.40 | p3nguin | Camera's audio is on MoH. |
| 20:04.56 | deltaflyer4747 | Channel:Local/11 context:default extension:81 |
| 20:05.02 | deltaflyer4747 | (test.call) |
| 20:05.12 | p3nguin | MoH needs to ChanSpy(SIP/deskphone,w) |
| 20:05.17 | deltaflyer4747 | p3nguin: yes |
| 20:05.22 | deltaflyer4747 | yes |
| 20:05.28 | deltaflyer4747 | exactly |
| 20:05.39 | p3nguin | If MoH will ChanSpy(SIP/deskphone,w), then deskphone will hear moh, but camera will not. |
| 20:05.52 | deltaflyer4747 | hopefully. |
| 20:05.55 | *** join/#asterisk okei (~n1x@188.129.195.144) |
| 20:06.02 | deltaflyer4747 | btw W is send audio only |
| 20:06.05 | okei | p3nguin: can u help me? :) |
| 20:06.11 | deltaflyer4747 | ie no listening, but that won't bother me |
| 20:06.19 | p3nguin | Okay, W could be better. |
| 20:06.21 | deltaflyer4747 | okei: yes, i is standard extension name for invalid |
| 20:06.39 | p3nguin | Since moh doesn't need to listen to SIP/deskphone, W is probably the right option. |
| 20:06.39 | deltaflyer4747 | so |
| 20:06.40 | okei | yes but dont working and need help :P |
| 20:06.53 | deltaflyer4747 | okei: what are you trying? |
| 20:07.09 | okei | i'll show u ext.conf |
| 20:07.17 | deltaflyer4747 | pastebin.com |
| 20:07.20 | okei | y |
| 20:07.28 | p3nguin | So ChanSpy(SIP/deskphone,W) |
| 20:07.38 | deltaflyer4747 | p3nguin: just thinking how to chanspy that moh |
| 20:07.43 | deltaflyer4747 | that call |
| 20:07.47 | p3nguin | Yeah, it seems like a problem. |
| 20:07.57 | deltaflyer4747 | originate |
| 20:07.59 | p3nguin | If moh is a real device, it's no problem. |
| 20:08.25 | deltaflyer4747 | really? how can i listen to inactive device? |
| 20:08.29 | deltaflyer4747 | only by calling it. |
| 20:08.31 | p3nguin | originate it |
| 20:08.42 | deltaflyer4747 | you can originate MOH too |
| 20:08.46 | deltaflyer4747 | (tested) |
| 20:09.34 | p3nguin | If it were a real device, you could originate SIP/moh application ChanSpy SIP/deskphone,W |
| 20:09.52 | deltaflyer4747 | let me test it ... will got result in a while |
| 20:09.55 | okei | deltaflyer4747: http://pastebin.com/2yE7XVkh |
| 20:10.12 | okei | when i trying to call number 123 |
| 20:10.35 | okei | output is rejected because extension is not found in context test |
| 20:10.35 | okei | :/ |
| 20:11.37 | deltaflyer4747 | pls include output from CLI |
| 20:11.51 | okei | kk |
| 20:12.34 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:12.40 | okei | deltaflyer4747: http://pastebin.com/09GrTVE2 |
| 20:13.08 | p3nguin | Oh! We can easily make MoH into a device. I just realized. |
| 20:13.12 | deltaflyer4747 | i meant a bit longer part ;) |
| 20:13.17 | deltaflyer4747 | p3nguin: hit me |
| 20:13.23 | p3nguin | extension 1001 is the moh? |
| 20:13.26 | deltaflyer4747 | yes |
| 20:13.51 | p3nguin | What context is 1001 in? |
| 20:14.09 | okei | p3nguin: test |
| 20:14.45 | *** join/#asterisk jenna (~jjones@unaffiliated/jenna) |
| 20:14.49 | p3nguin | deltaflyer4747: ^^ |
| 20:14.57 | deltaflyer4747 | p3nguin: okei same as all of them |
| 20:15.05 | deltaflyer4747 | ie default |
| 20:15.05 | jenna | Hi, anyone with pbxinaflash experience while trying out asterisk ? |
| 20:15.16 | p3nguin | deltaflyer4747: originate Local/1001@default application ChanSpy SIP/deskphone,W |
| 20:15.21 | deltaflyer4747 | okei: send me bit longer part of CLI output |
| 20:15.33 | deltaflyer4747 | p3nguin: let me test the first way pls :) |
| 20:15.44 | p3nguin | What first way? |
| 20:15.55 | okei | deltaflyer4747: what output? |
| 20:16.01 | p3nguin | This is the first idea where I reached the end. |
| 20:16.04 | okei | see http://pastebin.com/09GrTVE2 |
| 20:16.10 | deltaflyer4747 | okei: LONGER ;) |
| 20:16.36 | okei | deltaflyer4747: exactly? |
| 20:16.42 | p3nguin | Get SIP/deskphone and SIP/camera on a call toghether. |
| 20:16.58 | okei | sry i'm newbie in * |
| 20:17.02 | p3nguin | Then, on the CLI: originate Local/1001@default application ChanSpy SIP/deskphone,W |
| 20:17.15 | p3nguin | That will test if it works. |
| 20:17.21 | deltaflyer4747 | okei: just login to CLI, clear it, dial 123 and copypaste everything there is |
| 20:17.23 | p3nguin | To automate it, I'd probably use a macro. |
| 20:17.29 | deltaflyer4747 | p3nguin: w8 pls :) |
| 20:17.34 | p3nguin | Dial(M()) |
| 20:17.40 | p3nguin | weight? |
| 20:17.47 | okei | wait : ) |
| 20:17.52 | okei | : d |
| 20:17.55 | p3nguin | 8 = eight |
| 20:17.57 | okei | okay wait both |
| 20:18.03 | p3nguin | w8 = weight |
| 20:18.40 | p3nguin | What the hell... my mouse isn't working anymore. |
| 20:18.43 | p3nguin | |
| 20:18.43 | p3nguin | |
| 20:19.05 | okei | p3nguin: dont use mouse |
| 20:19.06 | okei | well wait |
| 20:19.20 | p3nguin | It's kind of important at some times. |
| 20:19.30 | deltaflyer4747 | pls a moment :) |
| 20:20.17 | okei | i have same message http://pastebin.com/09GrTVE2 |
| 20:20.25 | okei | deltaflyer4747: |
| 20:20.53 | p3nguin | When the environment is designed around the fact that a mouse is available, it's a key piece of hardware. |
| 20:21.08 | p3nguin | I've never had this happen before. |
| 20:21.45 | okei | anyone help me? |
| 20:22.39 | deltaflyer4747 | okei: for the 3rd time. WAIT A SEC ;) |
| 20:22.45 | okei | kk |
| 20:22.54 | p3nguin | deltaflyer4747: Did you want okei to wait a second? |
| 20:23.00 | deltaflyer4747 | you too :)) |
| 20:23.02 | deltaflyer4747 | both of you :D |
| 20:23.30 | okei | p3nguin: if u have answer no |
| 20:23.39 | okei | if u have't meh |
| 20:23.42 | p3nguin | I haven't been paying attention to you. |
| 20:24.05 | p3nguin | That's the nice thing about IRC... I can choose what to read and not to read. |
| 20:24.16 | deltaflyer4747 | yep |
| 20:24.33 | okei | p3nguin: okay and wait :) deltaflyer4747 is working :) |
| 20:24.34 | p3nguin | And since I was trying to help deltaflyer4747 solve a problem, I wasn't paying attention to other stuff. |
| 20:24.52 | deltaflyer4747 | right |
| 20:24.56 | deltaflyer4747 | i am testing it... |
| 20:25.02 | deltaflyer4747 | so far no luck, but ... |
| 20:25.07 | deltaflyer4747 | this is the right course i hope |
| 20:25.13 | okei | : d |
| 20:25.15 | p3nguin | My idea will work. |
| 20:25.18 | deltaflyer4747 | not chanspy but extenspy |
| 20:25.19 | p3nguin | I'm sure. |
| 20:25.23 | deltaflyer4747 | yes |
| 20:25.26 | p3nguin | ChanSpy() is just fine. |
| 20:25.37 | okei | deltaflyer4747: if you testing my conf please test |
| 20:25.38 | okei | :D |
| 20:25.41 | okei | <PROTECTED> |
| 20:25.41 | deltaflyer4747 | what i did is |
| 20:25.47 | p3nguin | ExtenSpy() might also work, but I'd rather use ChanSpy() I think. I'll reconsider. |
| 20:25.48 | deltaflyer4747 | call from 11 to 81 |
| 20:26.03 | deltaflyer4747 | then use test.call containing this |
| 20:26.08 | deltaflyer4747 | Channel:Local/1001 Application: Extenspy Data: 11,qW |
| 20:26.21 | p3nguin | That's not valid. |
| 20:26.38 | deltaflyer4747 | how come |
| 20:26.50 | okei | valid ext to invalid ext |
| 20:27.03 | p3nguin | ExtenSpy() requires that you spy an extension. |
| 20:27.09 | deltaflyer4747 | yes |
| 20:27.13 | deltaflyer4747 | 11 is and extension |
| 20:27.32 | p3nguin | I guess your lack of context threw me off. |
| 20:27.34 | p3nguin | Carry on. |
| 20:27.42 | deltaflyer4747 | 11 is extension |
| 20:27.47 | deltaflyer4747 | i spy on extension 11 |
| 20:28.04 | p3nguin | If you wrote ExtenSpy 11@default, I would have considered it valid. |
| 20:28.09 | deltaflyer4747 | but on reverse - ie pushing audio from caller through spy to spied extension |
| 20:28.19 | p3nguin | I guess leaving off @default means @default. |
| 20:28.26 | deltaflyer4747 | yes |
| 20:28.30 | p3nguin | Carry on. |
| 20:28.51 | deltaflyer4747 | but i cannot hear a thing. |
| 20:29.05 | p3nguin | I would have used ChanSpy(SIP/deskphone,W) |
| 20:29.12 | okei | p3nguin: do u know any forum to write my problem? |
| 20:29.34 | deltaflyer4747 | okay, i can try. |
| 20:29.43 | p3nguin | okei: No. Like I already told you, I haven't paid any attention to what you've been saying. |
| 20:29.44 | *** part/#asterisk GuySoft (~guysoft@109.226.6.237) |
| 20:29.52 | deltaflyer4747 | okei: please wait, i will try to help you atm i get my problem done |
| 20:30.03 | deltaflyer4747 | i feel i am close now |
| 20:30.18 | okei | deltaflyer4747: no i need only p3nguin's help |
| 20:30.20 | okei | : d |
| 20:30.48 | deltaflyer4747 | p3nguin: that dropped the 11-81 call |
| 20:30.50 | deltaflyer4747 | :D |
| 20:31.06 | p3nguin | deltaflyer4747: This is a two-step process. It requires only one call file to test. Create your call file for the camera to call the deskphone. |
| 20:31.22 | deltaflyer4747 | okay |
| 20:31.51 | p3nguin | When the camera can hear the deskphone, then use your CLI and run: originate Local/1001@default application ChanSpy SIP/deskphone,W |
| 20:31.56 | p3nguin | And tell me what happens. |
| 20:32.11 | deltaflyer4747 | Channel:SIP/11 Context:default Extension:81 |
| 20:32.47 | deltaflyer4747 | and... |
| 20:32.58 | deltaflyer4747 | nothing. |
| 20:33.19 | deltaflyer4747 | don't hear a thing. |
| 20:33.53 | deltaflyer4747 | http://pastebin.com/GcRArw5c |
| 20:37.28 | p3nguin | Did you make sure your moh class was working? Try calling 1001 from another phone. |
| 20:37.39 | deltaflyer4747 | works of course |
| 20:38.03 | p3nguin | So you did check it after the chanspy failed? |
| 20:38.08 | deltaflyer4747 | yes |
| 20:38.10 | deltaflyer4747 | just now |
| 20:38.27 | p3nguin | I'm going to have to replicate your scenario. |
| 20:38.36 | deltaflyer4747 | poor guy :) |
| 20:38.37 | p3nguin | just to make sure. |
| 20:39.08 | deltaflyer4747 | [ahfm] mode=custom application=/usr/bin/mpg123 -q -s -r 8000 -f 8192 -mono --ignore-mime http://nl3.ah.fm:9000 |
| 20:39.16 | deltaflyer4747 | use this as moh :)) |
| 20:39.22 | deltaflyer4747 | or any other MP3 stream ;) |
| 20:39.24 | deltaflyer4747 | :D |
| 20:39.45 | deltaflyer4747 | btw - way how to listen to live stream as MOH ;) |
| 20:40.05 | deltaflyer4747 | okei: now i have time. Please... paste the long CLI output and your config again. |
| 20:41.24 | okei | http://pastebin.com/2yE7XVkh |
| 20:41.34 | okei | http://pastebin.com/09GrTVE2 |
| 20:41.41 | okei | blah |
| 20:41.46 | deltaflyer4747 | long one ;) |
| 20:41.58 | okei | deltaflyer4747: wtF? what is long output |
| 20:42.02 | okei | her eis one error |
| 20:42.19 | deltaflyer4747 | okei: asterisk -rvvvvvvvvvvvvv |
| 20:42.21 | deltaflyer4747 | okei: ctrl+l |
| 20:42.27 | deltaflyer4747 | okei: dial 123 |
| 20:42.32 | okei | mda |
| 20:42.33 | deltaflyer4747 | wait second or two |
| 20:42.43 | okei | deltaflyer4747: and you have output http://pastebin.com/09GrTVE2 |
| 20:42.45 | okei | here!!!!! |
| 20:43.29 | deltaflyer4747 | okay okay |
| 20:43.50 | deltaflyer4747 | keep the I-beam down |
| 20:44.15 | okei | so tired |
| 20:44.19 | okei | p3nguin: can u help me right now? |
| 20:44.51 | deltaflyer4747 | okei: testing your problem now |
| 20:45.25 | okei | i'm here since 12:00 am nad haven't answer one little problem |
| 20:45.27 | okei | wtf : / |
| 20:48.43 | deltaflyer4747 | okei: okay, got it . |
| 20:49.05 | deltaflyer4747 | the problem is well known, just tested the working scenario, pasting right now |
| 20:50.36 | deltaflyer4747 | http://pastebin.com/aVpfcfKM |
| 20:50.39 | deltaflyer4747 | there you go |
| 20:50.45 | deltaflyer4747 | tested & working |
| 20:52.09 | okei | deltaflyer4747: and it's working? |
| 20:52.12 | deltaflyer4747 | (btw thanks, i just put it in my config) |
| 20:52.12 | okei | hm |
| 20:52.16 | deltaflyer4747 | okei: yes, it is :) |
| 20:52.29 | okei | well thanks but why my conf is not true |
| 20:52.50 | deltaflyer4747 | because you need to call that i by something ... |
| 20:53.04 | deltaflyer4747 | like "menu" |
| 20:53.24 | okei | hm |
| 20:53.24 | deltaflyer4747 | http://www.voip-info.org/wiki/view/Asterisk+i+extension |
| 20:53.28 | okei | thanks |
| 20:53.30 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-185-miami.txt.hfc.comcastbusiness.net) |
| 20:53.46 | deltaflyer4747 | np :) thanks to you, got nice message of nonexistant number |
| 20:53.57 | okei | :) |
| 20:54.30 | deltaflyer4747 | extension "123" - i am sorry, but thats not valid extension. Please try again. |
| 20:54.52 | *** join/#asterisk wonderworld (~ww@port-92-201-47-241.dynamic.qsc.de) |
| 20:55.14 | deltaflyer4747 | taken from http://www.planetwayne.com/forums/viewtopic.php?t=218 |
| 20:55.20 | p3nguin | When I use originate Local/moh@misc extension 762@phones, my phone rings and I hear the moh when I pick up. If I am on the phone already and I use originate Local/moh@misc application ChanSpy SCCP/myphone, I hear nothing. |
| 20:55.36 | deltaflyer4747 | p3nguin: yep, thats it |
| 20:55.43 | deltaflyer4747 | same as here |
| 20:57.00 | p3nguin | I guess I will see if ExtenSpy does anything. |
| 20:57.42 | deltaflyer4747 | okay |
| 20:58.02 | okei | deltaflyer4747: but i think _X. this is true string :P |
| 20:58.52 | deltaflyer4747 | okei: isnt _. listing all other numbers not mentioned exclusively? |
| 20:58.58 | deltaflyer4747 | (real question, i don't know) |
| 21:00.48 | p3nguin | no, it isn't. |
| 21:00.57 | p3nguin | _. is a terrible pattern to use. |
| 21:01.21 | p3nguin | It matches EVERYTHING, including extensions that you shouldn't be matching. |
| 21:01.45 | deltaflyer4747 | thats the reason it should be in separate context :) |
| 21:02.08 | deltaflyer4747 | it searches local context firts then goes to included :) |
| 21:02.15 | p3nguin | _X. is better if you want to match most numbers. One digit followed by at least one more character. |
| 21:02.36 | deltaflyer4747 | will not work if you dial "1" |
| 21:02.38 | deltaflyer4747 | :) |
| 21:02.49 | deltaflyer4747 | this does ;) |
| 21:02.51 | p3nguin | You're not telling me anything I don't already know. |
| 21:03.04 | deltaflyer4747 | i know :) just stating the obvious for okei :) |
| 21:03.13 | deltaflyer4747 | but... do as you wish :) |
| 21:03.27 | deltaflyer4747 | i tested it and works for local & outbound calls |
| 21:03.28 | p3nguin | _. shouldn't be used. It matches extensions s, h, i, and t, as well as everything else. |
| 21:03.47 | deltaflyer4747 | but only in that context, right? |
| 21:04.08 | *** join/#asterisk nightrid3r (~nightrid3@91.176.68.134) |
| 21:06.22 | deltaflyer4747 | nevermind, important thing is that it works :) |
| 21:06.29 | deltaflyer4747 | hopefully i will get the same result |
| 21:10.11 | deltaflyer4747 | p3nguin: any news for me? |
| 21:10.33 | p3nguin | ExtenSpy also didn't inject any sound. I'm trying to figure out why. |
| 21:11.34 | okei | how i can play music instead tone when calling one phone to another |
| 21:11.40 | okei | ivr? |
| 21:11.58 | p3nguin | Dial's m option. |
| 21:12.18 | p3nguin | m = musiconhold instead of ringing |
| 21:12.46 | okei | p3nguin: example please |
| 21:13.06 | p3nguin | exten => 54321,1,Dial(SIP/jack,30,m) |
| 21:13.22 | deltaflyer4747 | Dial(type/identifier, timeout, options, URL) |
| 21:13.54 | deltaflyer4747 | so that 30 is timeout on how long will the phone be jumping on the desk |
| 21:14.13 | p3nguin | What version introduced URL as a dial option? |
| 21:14.23 | p3nguin | dial parameter, rather |
| 21:16.25 | deltaflyer4747 | idk, copypaste from voip-info.org |
| 21:17.28 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:17.41 | p3nguin | Nevermind, it doesn't matter. It's available in my version, but I have no reason to use it. |
| 21:18.20 | deltaflyer4747 | true |
| 21:18.35 | deltaflyer4747 | yet my deskphone would probably understand it |
| 21:29.10 | deltaflyer4747 | p3nguin: still no luck, right |
| 21:30.11 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
| 21:35.33 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 21:38.37 | *** part/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 21:39.46 | okei | p3nguin: can u play mp3 files? |
| 21:39.53 | okei | can I* |
| 21:40.06 | *** join/#asterisk dfamorato (~dfamorato@c-75-74-235-36.hsd1.fl.comcast.net) |
| 21:40.26 | deltaflyer4747 | you can in MOH |
| 21:40.39 | okei | deltaflyer4747: only in moh? |
| 21:40.41 | deltaflyer4747 | you can actually play anything you want in MOH :) |
| 21:40.43 | okei | playback? |
| 21:41.04 | deltaflyer4747 | i am affraid that playback is limited to PCM WAV |
| 21:41.11 | okei | gsm |
| 21:41.12 | okei | ;p |
| 21:41.18 | deltaflyer4747 | okay... |
| 21:41.26 | deltaflyer4747 | but you know what i mean |
| 21:41.31 | okei | yep :) |
| 21:44.26 | okei | but i did't understand why i extension is not working |
| 21:44.27 | okei | : / |
| 21:45.17 | deltaflyer4747 | http://www.voip-info.org/wiki/view/Asterisk+i+extension |
| 21:45.30 | deltaflyer4747 | read that |
| 21:45.46 | p3nguin | Extension i will be used when you've used BackGround() or WaitExten() and inputted an extension that isn't valid. |
| 21:47.02 | deltaflyer4747 | or that ^^ |
| 21:55.07 | *** join/#asterisk Micc_ (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 21:55.30 | Micc_ | I'm getting a not a local domain when registering to 1.8.5.0 when the domain is in the domain list. |
| 21:55.36 | deltaflyer4747 | p3nguin: and any good news for me? |
| 21:56.19 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 22:03.30 | *** join/#asterisk esperanto (~rusty@46.115.20.72) |
| 22:04.36 | esperanto | hey fellas, I am trying to generate a sip packet that would take 1.6.2.16-1 down |
| 22:04.42 | esperanto | vulnerability http://web.nvd.nist.gov/view/vuln/detail?vulnId=CVE-2011-2529 |
| 22:05.01 | esperanto | does anybody know how exactly sip packet should look like? |
| 22:06.09 | deltaflyer4747 | for i in range (10,0): print i |
| 22:10.10 | esperanto | I don't think I get it, I am using sipp utility to generate sip packet |
| 22:10.47 | esperanto | could u please elaborate a bit? |
| 22:13.11 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 22:16.44 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 22:17.25 | esperanto | deltaflyer? |
| 22:17.44 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 22:20.16 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:26.48 | Micc_ | I've got an aastra 57iCT and a mitel both registering to the same server. When the aastra registers I get an error that its not a local domain, but the mitel registers fine. Asterisk version is 1.8.5.0, and it registers fine to 1.6.2.19 |
| 22:28.00 | Micc_ | anyone want me to paste the sip debug to take a look at what the difference could be? |
| 22:33.40 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 22:40.14 | p3nguin | deltaflyer4747: I cannot get the local moh channel to spy correctly, playing the music on the spy channel. It just will not play, and I can't see any reason. |
| 22:40.52 | p3nguin | I can make it spy. The spy channel goes up. moh starts. But there is no sound from moh to the spied channel or exten. |
| 22:50.03 | Micc_ | that seems really odd that I would have to add domain=host.domain.com:5060 to get aastra phones to register now. That should be in the upgrade docs at least if its supposed to be that way, or its a major bug in 1.8.5.0 |
| 22:50.29 | Micc_ | I never had to put the port on the end before. |
| 22:51.49 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:17.53 | dym | Why cant I grab the Phone Number of an external caller with ${CALLERID(num)} but only internal SIP Lines, etc.? |
| 23:25.03 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:30.51 | ChannelZ | maybe the channel has no callerID info associated with it |
| 23:32.50 | dym | ChannelZ: well, the call is from outside the PBX, and it has a CallerID |
| 23:33.00 | dym | cause when i call my regular phone with my mobile, i do see the number. |
| 23:33.19 | ChannelZ | So perhaps your DAHDI is misconfigured and is not getting the caller ID |
| 23:33.25 | ChannelZ | What kind of interface are you using |
| 23:33.34 | dym | SIP only |
| 23:33.38 | dym | SIP Trunk |
| 23:33.55 | ChannelZ | check the SIP debug and see if your provider is even sending the right thing |
| 23:34.33 | dym | odd |
| 23:34.39 | dym | it arrives as anonymous via the Trunk |
| 23:34.46 | ChannelZ | and/or make sure you aren't hot-wiring caller ID in your sip.conf for the peer your calls come in on |
| 23:34.52 | dym | From: "anonymous" <sip:anonymous@213.239.205.120>;tag=as14c104cc |
| 23:35.26 | p3nguin | Did you trustrpid? |
| 23:36.36 | dym | no, but will try now. |
| 23:37.23 | dym | still anonymous |
| 23:37.24 | dym | mhh |
| 23:37.27 | ChannelZ | maybe they charge extra |
| 23:37.43 | dym | well, if i use my landline to call, the ID is transferred |
| 23:37.50 | ChannelZ | or it's an option to turn on and off on their control panel for some unknown reason |
| 23:37.51 | dym | just my mobile thats causing problems |
| 23:37.57 | dym | doubt it |
| 23:38.15 | ChannelZ | oh.. then that's something with your mobile |
| 23:38.28 | ChannelZ | blocking there, or for some reason they have difficulty with your itsp |
| 23:39.14 | dym | but then again - if i call my landline with my mobile - i can see the number. |
| 23:40.04 | ChannelZ | shrugs - ask call your ITSP |
| 23:40.10 | ChannelZ | ask/call |
| 23:40.35 | ChannelZ | Argh. Finally got my own * back up. PITA |
| 23:41.47 | dym | mhh |
| 23:41.58 | dym | even if i pass the call straight on with Dial() its "unknown". |
| 23:42.00 | dym | oddmuch |
| 23:44.14 | ChannelZ | if you're not getting it from them, it's not going to magically appear |
| 23:52.29 | ChannelZ | damnit why aren't my cli aliases working |
| 23:53.11 | Micc_ | Is there any way to tell asterisk 1.8 to handle quotes in Set the same way as before? It seems like now when I do Set(var="something") it actually keeps the quotes as part of the assignment. |
| 23:53.30 | p3nguin | Don't use the quotes. |