| 00:00.28 | jer | No such command 'zap' (type 'help' for help) |
| 00:00.34 | jer | which is extremely odd |
| 00:00.38 | tzafrir_home | X-Filez, what ISDN channel do you use? zap? misdn? anything else? |
| 00:00.41 | jer | it was working before, didn't rebuild * or anything |
| 00:00.54 | tzafrir_home | ok. so chan_zap.so has failed to load |
| 00:01.11 | tzafrir_home | now please pastebin your zapata.conf |
| 00:01.24 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
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| 00:01.36 | jer | ok |
| 00:02.06 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 00:02.11 | jer | http://pastebin.com/m7a3adec5 |
| 00:03.07 | jer | oh sorry, you said zapata.conf |
| 00:03.09 | jer | my mistake |
| 00:04.17 | jer | tzafrir_home, thanks very much; i forgot to change channel => 1 to 4 |
| 00:04.19 | jer | working now |
| 00:08.54 | X-Filez | tzafrir : misdn installed |
| 00:14.17 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:15.46 | X-Filez | tzafrir: misdn.conf --> http://pastebin.com/m62058dea , misdn-init.conf --> http://pastebin.com/m23dffe8f , misdnportinfo --> http://pastebin.com/m49b12f24 and Log by mISDN -->> http://pastebin.com/m7c96b5bf |
| 00:16.35 | X-Filez | ISDN PCI connected to NT device (ISP owner NT) |
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| 00:19.15 | X-Filez | tzafrir_home: you there ? |
| 00:19.42 | tzafrir_home | yes. But I don't really think I can help you much with misdn |
| 00:20.24 | tzafrir_home | sorry |
| 00:20.26 | X-Filez | :( |
| 00:20.52 | X-Filez | zap work in Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] ? |
| 00:20.58 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:22.23 | X-Filez | PPLs, Please help, i cant find problems, where asterisk can't use ISDN channel mISDN, my confuration in : misdn.conf --> http://pastebin.com/m62058dea , misdn-init.conf --> http://pastebin.com/m23dffe8f , misdnportinfo --> http://pastebin.com/m49b12f24 and Log by mISDN -->> http://pastebin.com/m7c96b5bf |
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| 00:28.28 | X-Filez | tzafrir_home you use zap ? |
| 00:28.59 | mvanbaak | lol |
| 00:29.06 | exothermc | How do you set so voicemail does not get emailed out? |
| 00:29.06 | tzafrir_home | sometimes |
| 00:29.25 | exothermc | can you change it via context? |
| 00:29.27 | lotho | X-Filez: show the output of "misdn show port 1" |
| 00:30.05 | *** join/#asterisk mikefoo (n=juso@cpe-24-193-121-198.nyc.res.rr.com) |
| 00:30.06 | X-Filez | lotho: * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:10 |
| 00:30.23 | mikefoo | Anyone know where I could go about purchasing a short code? |
| 00:30.36 | mikefoo | im in the US |
| 00:31.49 | X-Filez | lotho: but : * Port 2 Type TE Prot. PMP L2Link DOWN L1Link:___UP___ Blocked:0 Debug:10 |
| 00:35.03 | lotho | X-Filez: "misdn port up 1" perhaps could this bring the link up |
| 00:36.40 | X-Filez | lotho: this same : * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:10 |
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| 00:37.21 | lotho | X-Filez: you did an "misdn-init config", right? |
| 00:38.03 | X-Filez | lotho: yes, misdn-init config next step misdn-init start |
| 00:38.04 | X-Filez | :) |
| 00:40.30 | X-Filez | lotho: how loaded asterisk and ai tru run misdnportinfo , result -> http://pastebin.com/m447b26a7 |
| 00:41.47 | X-Filez | ai tru = I try :) |
| 00:43.16 | lotho | X-Filez: which asterisk version do you have? |
| 00:43.40 | X-Filez | lotho: 1.4.5 |
| 00:44.41 | lotho | and misdn? |
| 00:44.53 | X-Filez | misdn 1.1.7 |
| 00:45.49 | X-Filez | lotho: load modules : list -> http://pastebin.com/m1a91c2bf |
| 00:50.49 | X-Filez | lotho: have other ideas? |
| 00:51.54 | lotho | X-Filez: i think i had this problem once too |
| 00:52.23 | lotho | but i can not remembder what i did to fix it ;) |
| 00:52.50 | X-Filez | lotho: eh.. |
| 00:53.59 | X-Filez | this proble not in configure misdn-init.conf or misdn.conf? |
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| 01:31.34 | kiscokid | Anyone know how to set a voicemail box back to factory defaults? |
| 01:32.13 | kiscokid | that is, erase all the current messages and reset all greetings? |
| 01:45.09 | tzafrir_home | kiscokid, rm -rf /var/spool/asterisk/voicemail/* ? |
| 01:45.33 | tzafrir_home | or 'mv' if you're not really sure :-) |
| 01:47.13 | kiscokid | trying to do it for one extension so I assume you mean rm -rf /var/spool/asterisk/voicemail/default/8005 |
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| 02:05.42 | jmacz | hi everyone |
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| 02:06.25 | jsmith | Hello |
| 02:06.48 | [gnubie] | what's the benefit of using a template compared of putting the common parameters in the [global] context? |
| 02:06.49 | jmacz | i'm having a problem trying to load zaptel on a hp dl140g3 with a te220b card |
| 02:07.14 | jmacz | i got a "ZT_SPANCONFIG failed on span 1: No such device or address (6)" message |
| 02:08.13 | jmacz | cat /proc/zaptel/* shows nothing and lspci shows: Communication controller: Digium, Inc. Unknown device 0220 (rev 02) |
| 02:08.14 | tzafrir_home | [gnubie], that you can share parameters between only some of your peers (or contexts, or whatever) |
| 02:08.51 | jmacz | i'm running debian stable. Any ideas why ztcfg shows that? |
| 02:09.05 | tzafrir_home | jmacz, what version of zaptel? |
| 02:09.35 | tzafrir_home | also,run: update-pciids |
| 02:09.52 | jmacz | tzafrir_home: zaptel-modules-2.6.18-5-amd64 1.2.15~dfsg-1+2.6.18.dfsg.1-13etch5 |
| 02:09.53 | tzafrir_home | this will give a nicer name to the ID |
| 02:10.26 | jmacz | tzafrir_home: ok, thanks for answering, i'll try that |
| 02:11.43 | jmacz | tzafrir_home: zaptel is zaptel 1.2.11.dfsg-1 |
| 02:12.32 | *** join/#asterisk Rants (n=Rants@ool-182fbb7a.dyn.optonline.net) |
| 02:13.21 | tzafrir_home | jmacz, modinfo wct4xxp | grep 220 |
| 02:13.36 | Rants | How do I block all 800 numbers from incoming lines? I know how to block individual numbers but I want to block 800 numbers and certain area codes from calling in. |
| 02:14.46 | tzafrir_home | jmacz, if it returns nothing, it probably means that your card is not supported by that veersion of Zaptel |
| 02:15.27 | jmacz | tzafrir_home: yep, shows nothing |
| 02:15.36 | Rants | any ideas? |
| 02:16.01 | jmacz | guess i'll have to try using zaptel from sources |
| 02:16.10 | jmacz | tzafrir_home: thanks |
| 02:16.37 | tzafrir_home | If so, I figure you'll need newer zaptel from e.g. pkg-voip.buildserver.net |
| 02:17.56 | jmacz | tzafrir_home: sure, I'll try those first ;-)\ |
| 02:19.36 | Rants | Anyone? |
| 02:21.13 | Rants | is it even possible to block by area code? |
| 02:21.51 | [gnubie] | tzafrir_home: i see.. so with templates, you can define 2 or more templates with almost similar parameters and use them for each sip accounts while declaring it on the global context, it's a flat way to use the defined parameters, am i right? |
| 02:22.58 | tzafrir_home | [gnubie], the point is not to repeat anything twice |
| 02:23.28 | tzafrir_home | If the twwo templates have "almost the same parameters", base them on a common template: |
| 02:23.52 | tzafrir_home | [template-a](!,template-master) |
| 02:24.28 | tzafrir_home | but generally - yes |
| 02:25.55 | jsmith | Rants: You could do something like this: |
| 02:26.08 | *** join/#asterisk Entr4nced (n=root@cpe-75-187-197-217.neo.res.rr.com) |
| 02:26.50 | jsmith | Rants: exten => 123,n,GotoIf($[${CALLERID(num):0:3} = 800]?contextforbadcalls,s,1) |
| 02:27.09 | *** join/#asterisk Entr4nced (n=none@cpe-75-187-197-217.neo.res.rr.com) |
| 02:27.21 | Rants | I see. |
| 02:27.27 | jsmith | Rants: (where 123 is the extension that matches your inbound calls, and where contextforbadcalls,s,1 is where you want to send bad callers) |
| 02:33.12 | [gnubie] | tzafrir_home: thanks.. i should start changing my sip.conf by using templates on my home pbx.. |
| 02:37.21 | Rants | @jsmith: I did this but it doesnt seem to work exten => 600,n,GotoIf($[${CALLERID(num):0:3} = 800]?ext-local,vmu600,1) |
| 02:37.58 | jsmith | Rants: What printed on the CLI when it hit that step in your dialplan? |
| 02:39.35 | Rants | Odd, I dont see it. |
| 02:43.32 | Rants | @jsmith: Got it thanks.. Needed to change some of the context. |
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| 02:44.34 | jsmith | Rants: Glad you got it figured out ;-) |
| 02:46.20 | Rants | What is there is multiple? I just added for 800 and 866 area codes but now it doesnt seem to ring the line from the standard code? |
| 02:46.49 | Rants | Do i have to change n? |
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| 02:47.26 | jsmith | No, you shouldn't have to, unless that's the first priority in your extension |
| 02:47.42 | jsmith | If it's the first priority in that extension, it needs to the "1" instead of "n" |
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| 02:48.13 | Rants | i see.. I'll just move them then. |
| 02:48.53 | jsmith | Rants: Have you read the free Asterisk book? If not, I suggest you download it and check out the dialplan chapters (chapters 5 and 6) |
| 02:48.56 | jsmith | ~thebook |
| 02:48.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:49.05 | Rants | great.. thanks. |
| 02:49.13 | Rants | No i havent.. I will though. |
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| 03:12.51 | [gnubie] | hello all.. |
| 03:13.32 | [gnubie] | my asterisk-1.4.15 crashed again but this time, i got a core dump and the dbg package was installed |
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| 03:48.08 | tessier | Hello all! |
| 03:49.23 | tessier | I have a few Snom 320's (behind NAT in different locations) connected to an asterisk system on a public IP. Most of them work flawlessly all the time. But one of them can make a few calls then can no longer make calls and requires a reboot before it can make calls again. Very annoying. Anyone know what the problem might be? |
| 03:49.27 | Qwell | I'm just gonna go ahead and throw this out there... |
| 03:49.31 | Qwell | I freaking love Sun. |
| 03:50.06 | tessier | Qwell: Why? |
| 03:50.10 | Qwell | Dear Try and Buy Program Participants, Thank you very much for your participation in our survey. We have received overwhelming interest and response. Therefore, we have decided to offer each survey participant a $25 gift certificate - i.e. not limited to the first 300 participants. Thanks again! |
| 03:50.14 | Qwell | that's why |
| 03:50.27 | tessier | Because they bribe you? |
| 03:50.32 | Qwell | post-facto |
| 03:50.39 | Qwell | I did the survey weeks ago |
| 03:50.47 | tessier | Gift certificate for where? |
| 03:50.51 | Qwell | this was entirely uncalled for, but very awesome |
| 03:50.53 | Qwell | Amazon |
| 03:51.05 | tessier | That is cool |
| 03:51.12 | Qwell | tessier: I'm thrilled |
| 03:53.52 | Qwell | Corydon76-dig: they say it might snow here this weekend... |
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| 03:56.21 | Corydon76-dig | Qwell: ugh, I'd have to drive in it on Monday, then |
| 03:56.42 | Qwell | it wouldn't last until then |
| 03:57.26 | Corydon76-dig | I hope not |
| 03:57.42 | Qwell | if last year is anything to go by, anyhow |
| 03:59.28 | Corydon76-dig | If there's one thing I've learned from living in the South, it is not to try to predict the weather |
| 03:59.34 | Qwell | fair enough |
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| 04:04.45 | Corydon76-dig | Welcome to Tennessee. If you don't like the weather, wait 5 minutes; it'll change. |
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| 04:06.53 | tessier | Corydon76-dig: Once you have traveled a bit you will realize they say that everywhere. |
| 04:07.06 | Corydon76-dig | tessier: it's true here |
| 04:07.23 | tessier | That's my point. It's true here too. |
| 04:07.27 | tessier | It's true everywhere. |
| 04:07.40 | Corydon76-dig | tessier: if you're caught in a summer downpour, you can outwait it under a tree. Really. |
| 04:07.40 | tessier | It's a "truism" |
| 04:07.51 | tessier | I've lived in Tennessee. |
| 04:07.58 | tessier | I know. |
| 04:07.58 | Qwell | heh, it'd probably stop by the time you got to the tree |
| 04:08.04 | Corydon76-dig | tessier: it's not like that in Pennsylvania, where I grew up |
| 04:08.08 | tessier | And the tree might get hit by lightning and fry your ass too. |
| 04:08.26 | Corydon76-dig | I just mean downpour, not a storm |
| 04:08.43 | Corydon76-dig | In a storm, you get your ass inside, because as likely as not, there's a tornado |
| 04:09.18 | tessier | Duck and cover just like a nuclear bomb blast. |
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| 04:09.55 | Corydon76-dig | Except that you actually have a chance at surviving a tornado |
| 04:09.57 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:10.30 | Corydon76-dig | With a nuke, if the blast doesn't kill you, the radiation will |
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| 08:03.55 | NolanG | Hey, quick question. I've got an Office Communications Server 2007, but it won't handle "sip trunking" to Vitelity because it doesn't want to provide a username and password. It can use any sip endpoint as a gateway however. Is it possible to use Asterisk as the sip endpoint, then have it forward the requests through and do the appropriate auth to Vitelity? |
| 08:07.59 | mosty | what do you mean by "sip trunking" ? |
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| 08:11.15 | NolanG | mosty: good question. That's a term I picked up from the msdn boards. I think the only real issue is that OCS doesn't want to provide a username/password to Vitelity, so I need to run it through some kind of translation device that would. |
| 08:12.40 | mosty | so it supports sip, but not authenticated sip? |
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| 08:41.11 | adelas | Hi, i'm kinda new to asterisk, and i want to make myself a small busniess asterisk with pri card |
| 08:41.19 | adelas | would anyone recommend me somewhere? |
| 08:41.33 | adelas | i currently pay and use someone else asterisk version |
| 08:41.38 | adelas | but hes shit and charges too much |
| 08:41.42 | adelas | not good enough support |
| 08:41.46 | adelas | i want to do this myself now |
| 08:41.53 | adelas | any idea where i should start? |
| 08:42.14 | adelas | my thing is based on cisco phones and a pri card and some voip services |
| 08:42.35 | adelas | i'v tried asterisk@home before, but it was difficult and not needing my needs |
| 08:43.13 | mosty | do you want a gui or web frontend for configuring asterisk, or do you want to configure things manually? |
| 08:44.36 | adelas | gui/web front end |
| 08:44.50 | adelas | currently, i have a web gui front end thats really slow like hell |
| 08:44.59 | adelas | and can't customize much except call routings |
| 08:45.37 | mosty | and what was it with asterisk@home that you didn't like? btw i think it's called trixbox now |
| 08:45.56 | adelas | well, i was a total linux noob back then, and it was fruturating |
| 08:46.06 | adelas | it has alot of manually confuration |
| 08:46.16 | adelas | and the syntax of the sip files i couldn't get |
| 08:46.21 | adelas | and the routing |
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| 08:47.16 | adelas | my main goal is to have like a call queue for support, conference rooms, transfer calls, automated answer, logs, monitor, and some custom routing rules |
| 08:47.20 | mosty | you shouldn't need to edit config files if you're using trixbox |
| 08:47.30 | adelas | well, i never tried trixbox yet |
| 08:47.40 | mosty | trixbox is asterisk@home |
| 08:47.54 | adelas | well, the other problem was setting up with the cisco phones |
| 08:48.08 | adelas | the gui wasn't too nice back then |
| 08:48.40 | adelas | well, i'm looking at asteriskNow |
| 08:48.53 | adelas | is that like much different from trixbox? |
| 08:49.04 | mosty | i don't know, i've never used either of them |
| 08:49.41 | mosty | they seem to be the two most commonly used gui's, you should try both. note that this is not the place to ask for help with them though (see the topic) |
| 08:50.02 | adelas | yea |
| 08:50.12 | adelas | well, what about freepbx? |
| 08:50.45 | adelas | i can't really decide now lol |
| 08:51.12 | mosty | i've never looked at freepbx. there's no reason you can't try all three |
| 08:52.23 | adelas | if i recall, did trixbox comprise of freebpx? |
| 08:54.55 | mosty | no idea, sorry |
| 08:55.11 | adelas | i'm soo willing to pay for support |
| 08:55.21 | adelas | but not as much as 1k |
| 08:57.17 | mosty | get a test box together, and try them out, see what you like the look of best |
| 08:58.23 | adelas | i see trixibox i gotta get the cce edition |
| 08:58.27 | adelas | to get features i wanted |
| 08:58.46 | adelas | do you by any chance who developed/backs up AsteriskNow? |
| 08:59.35 | mosty | no. see the topic. we don't deal with the gui's in this particular channel |
| 08:59.57 | kaldemar | trixbox uses freepbx. |
| 09:00.18 | adelas | o, i had another question regaruding faxing |
| 09:00.34 | adelas | has asterisk ever like support faxing via pri line directly? |
| 09:00.50 | adelas | from experience with using the pri to a voip adpator to analog to computer, its sucks |
| 09:01.01 | *** join/#asterisk Hemos\ (n=cyberspa@80.104.209.159) |
| 09:01.21 | adelas | i was hopping theres some way to utlizie the pri card directly? any possibilites out tere? |
| 09:01.50 | mosty | adelas, yes it does, but you should only ever do faxing with asterisk if the fax never goes over a voip connection |
| 09:02.16 | adelas | yes |
| 09:02.22 | mosty | ie do not try to do fax machine - sip ata - asterisk - pstn |
| 09:02.32 | adelas | yea thats the thing righ tnow |
| 09:02.37 | adelas | that i can only do with my current setup |
| 09:02.53 | adelas | i want to go T1Pri - asterisk - machine |
| 09:02.55 | mosty | fax over voice over ip will never work reliably, give up on that now |
| 09:03.09 | adelas | well yea i do that now b/c i have no other choice |
| 09:03.49 | adelas | any chance you have tutorials or such to utlize these asterisk faxing features? |
| 09:04.06 | adelas | or heads up to where to look at? |
| 09:04.17 | mosty | adelas, what do you mean by asterisk - machine ? what kind of interface is in the middle there? |
| 09:04.52 | adelas | well i mean asterisk converts the incoming fax into a jpg/gif/w/e and sends it to email for hte machines |
| 09:05.29 | adelas | i have a number for the pri card, that is used for faxing |
| 09:05.59 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 09:06.06 | mosty | adelas, that's the best you can do with asterisk, currently |
| 09:06.27 | adelas | the way i'm doing it? |
| 09:06.32 | mosty | or you could probably hack together a web interface instead of email |
| 09:06.33 | mosty | yes |
| 09:06.49 | adelas | wait i think we have a misunderstanding |
| 09:07.17 | adelas | the way i have it now, its pri -> asterisk -> sip ata -> fax modem -> computer |
| 09:07.39 | adelas | thats the only way? |
| 09:08.25 | mosty | no. that's not the only way. note that it's a particularly unreliable way |
| 09:08.34 | adelas | whats the other way? |
| 09:09.09 | mosty | you can go pri -> asterisk -> email, or pri -> asterisk -> dump the fax to a web site somehow |
| 09:09.17 | adelas | yes thats what i want |
| 09:09.32 | adelas | the current asterisk supports that? |
| 09:09.51 | mosty | yes, but you have to do some of the work yourself i believe. look up faxing with asterisk on voip-info.org |
| 09:09.52 | adelas | like its part of asterisk, not any addon/gui? |
| 09:09.57 | adelas | okay |
| 09:10.46 | mosty | i think you need an addon to do it |
| 09:11.28 | adelas | i fear those addons ;( |
| 09:13.14 | mosty | i don't recommend asterisk for doing faxing to anyone, it's not really good enough yet |
| 09:14.54 | adelas | well, do you know any other way to get fax from a pri line? |
| 09:15.42 | adelas | its usually only for occasional fax incoming |
| 09:15.46 | adelas | not like every day |
| 09:16.32 | mosty | i use hylafax, but i have an entire pri dedicated to faxing |
| 09:16.40 | mosty | no asterisk in the picture there |
| 09:20.48 | tzafrir_home | mosty, and how do you interface hylafax with PRI? |
| 09:21.13 | mosty | tzafrir, eicon diva PRI card |
| 09:22.07 | tzafrir_home | I noticed someone added app_fax (uses spandsp) to asterisk-adons/trunk |
| 09:23.52 | mosty | one benefit of using hylafax is that it has it's own client protocol, so you doint have to mess with voip |
| 09:24.42 | mosty | but at the same time, hylafax has a few downsides |
| 09:31.11 | adelas | mosty, thanks alot i'mma head off and do reading on the new stuff now |
| 09:33.28 | mosty | np |
| 09:45.22 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:50.03 | admgecko | morning |
| 09:50.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 10:05.20 | mosty | tzafrir, do you know if the hrtimer zaptel patches will make ztdummy run with acpi on dells? |
| 10:09.00 | *** join/#asterisk Hemos\ (n=cyberspa@80.104.207.215) |
| 10:18.08 | tzafrir_home | mosty, I suspect it will work |
| 10:20.41 | tzafrir_home | mosty, try it with http://svn.digium.com/svn/zaptel/branches/1.4/live_zap :-) |
| 10:21.44 | mosty | what's live_zap? |
| 10:23.33 | tzafrir_home | This should help you test a newer version of zaptel without fully isntalling it |
| 10:23.53 | tzafrir_home | check the README in the same directory |
| 10:24.16 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 10:31.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:34.49 | X-Filez | hello ppls, mISDN write debug in -> http://pastebin.com/m32e346a9 , why no free channels ? |
| 10:48.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:50.57 | *** join/#asterisk dty (n=dertybiz@195.225.54.221) |
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| 11:04.31 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 11:06.20 | X-Filez | _ys privet :) |
| 11:13.04 | X-Filez | ppl, I installed mISDN and use asterisk 1.4.15, I'm try call from ISDN to asterisk, asterisk don't use exten, debug log in http://pastebin.com/m32e346a9 , Please help... |
| 11:26.35 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 11:26.58 | admgecko | is anyone using an SPA3102 |
| 11:26.59 | admgecko | ? |
| 11:28.23 | mosty | i have in the past, why? |
| 11:28.41 | admgecko | it seems to be randomly disconnecting the call after about 10 minute |
| 11:28.42 | *** join/#asterisk nirz (n=nir@bzq-79-181-155-4.red.bezeqint.net) |
| 11:28.48 | admgecko | inbound and out |
| 11:28.53 | mosty | are you behind nat? |
| 11:29.14 | mosty | between the spa3102 and asterisk? |
| 11:29.30 | admgecko | im behind a firewall, but the asterisk box and spa3102 and phone are on the same lan |
| 11:29.53 | admgecko | no nat inbetween, just stright into the swich |
| 11:30.27 | *** join/#asterisk RoyK (n=roy@ip-101-26-149-91.dialup.ice.no) |
| 11:31.01 | admgecko | if i turn the debug up, i should be able to see it in my syslog server right? (ive got splunk installed on the asterisk box for logging) |
| 11:31.12 | admgecko | should i turn syslog on to see whats going on? |
| 11:32.04 | mosty | sure. set verbose 10 and debug 10, and make sure they're both being sent to syslog (check logger.conf) |
| 11:32.40 | admgecko | i was thinking more on the spa side of things, im monitoring the /var/log/asterisk/full and /var/log/asterisk/messages file already |
| 11:33.25 | mosty | well you wont get anything extra in syslog that isn't in the full log |
| 11:33.46 | admgecko | but if i syslog the spa3102, that might tell me something right? |
| 11:34.03 | mosty | maybe. what does the asterisk full log show when the call drops out? |
| 11:34.07 | mosty | pastebin it |
| 11:36.39 | admgecko | it doesnt have anything in after 13th dec |
| 11:36.51 | admgecko | hmm ill check logger.conf |
| 11:37.48 | admgecko | full was commented out :(, just got messages, ive enabled full doh! |
| 11:39.19 | mosty | disable messages, enable full |
| 11:39.40 | mosty | and when you're done debugging, disable full again, or your box will run out of disk space pretty quick |
| 11:41.41 | admgecko | ok, logging is working again now |
| 11:42.44 | admgecko | mosty: have you use the avaya 4610sw phones? |
| 11:42.58 | X-Filez | ppl, I installed mISDN and use asterisk 1.4.15, I'm try call from ISDN to asterisk, asterisk don't use exten, debug log in http://pastebin.com/m32e346a9 , Please help... |
| 11:43.23 | mosty | adelas, no |
| 11:45.07 | admgecko | mosty: just making a test call now |
| 11:45.21 | admgecko | takes about 9 - 10 minutes to disconnect |
| 11:45.58 | mosty | X-Filez, is there an misdn users mailing list? if so i suggest that you ask there, not many people use misdn |
| 11:49.12 | X-Filez | ISP where my mail, blocked mailing listing random servers :( |
| 11:50.58 | mosty | X-Filez, have you tried the mailing list? you will have trouble finding misdn users here |
| 11:57.28 | *** join/#asterisk RoyK (n=roy@ip-101-26-149-91.dialup.ice.no) |
| 11:59.25 | admgecko | mosty, ive got that file, and stuck it on the webserver, may i pm link? |
| 12:01.27 | mosty | admgecko, paste the link here so i'm not the only person who can help |
| 12:01.50 | admgecko | http://www.boczko.com |
| 12:01.56 | admgecko | whopes |
| 12:02.02 | admgecko | http://www.boczko.com/full |
| 12:02.24 | admgecko | im trying a 10 minute log internal to internal at the minute |
| 12:02.28 | admgecko | to see if that does it |
| 12:03.00 | admgecko | that should say wether its an asterisk problem or an spa problem once an for all |
| 12:03.18 | admgecko | but im only two minutes into the call |
| 12:03.21 | mosty | admgecko, can you paste just the part of the full log from when the call drops out? |
| 12:03.29 | admgecko | yeah holdon |
| 12:05.05 | admgecko | http://www.boczko.com/short |
| 12:05.54 | admgecko | downloaded ok? |
| 12:07.02 | *** join/#asterisk asr33 (n=island@ppp-RAS1-5-233.dialup.eol.ca) |
| 12:07.15 | mosty | admgecko, yeah. so this call goes to asterisk, and then from asterisk out from behind nat to the internet? |
| 12:07.32 | admgecko | no |
| 12:07.44 | admgecko | IP PHONE -> Asterisk _> A |
| 12:08.32 | admgecko | IP PHONE > Asterisk > SPA3102 > FX0 port > PSTN > Exchange > Normal phone |
| 12:08.37 | admgecko | even |
| 12:09.23 | mosty | hmm |
| 12:09.44 | admgecko | thats how it should go at least |
| 12:09.52 | admgecko | and it does, for 9 minutes odd |
| 12:10.05 | admgecko | then disconnects, and throws the phone back to dial tone |
| 12:10.30 | admgecko | the phone is connected to the fxs port on the spa |
| 12:10.41 | admgecko | and one is connected to a pap2t |
| 12:10.59 | admgecko | and on is an Avaya 4610SW Ip phone (with the sip firmware) |
| 12:11.26 | admgecko | the spa and avaya connect to a cataylst switch |
| 12:11.41 | admgecko | also connect to that is a linksys wrt54gl |
| 12:12.13 | admgecko | and the pap2t is connected to another wrt54gl, which is setup as a client bridge to the one connected to the cisco |
| 12:12.16 | *** join/#asterisk [gnubie] (n=chatzill@cm174.gamma179.maxonline.com.sg) |
| 12:12.36 | admgecko | the inernet is connected via a netgear router connect to the cisco |
| 12:12.47 | admgecko | the asterisk is connected to the uplink port of the cisco |
| 12:13.00 | admgecko | there isnt any vlans or any routing defined on the cisco |
| 12:13.08 | [gnubie] | anyone here can point me a good documentation on how to configure an attended transfer feature for asterisk 1.4? |
| 12:13.25 | mosty | [gnubie], lookup features.conf on voip-info.org |
| 12:13.53 | admgecko | 12 minutes on the internel to internal call, so looks like an spa issue |
| 12:14.11 | mosty | admgecko, the spa3102 has an fxs port right? can you test with a analogue phone plugged into the spa? |
| 12:14.37 | [gnubie] | mosty: isn't it the features.conf is only for call parking? |
| 12:15.07 | admgecko | yeah, it does, but the phone on the fxs goes to asterisk, as a sip extention |
| 12:15.36 | mosty | [gnubie], no. or some sip phones can do attended transfer, see the respective manual |
| 12:15.55 | asr33 | IP phones? I understand its very subjective but does anyone have recommendation, my only requirement is low powered because it would be plugged in twenty four hours a day? |
| 12:16.12 | mosty | admgecko: an FXS port is a phone line, not a sip extension |
| 12:16.44 | mosty | asr33, polycom, linksys, snom, aastra seem popular |
| 12:17.19 | admgecko | mosty: yeah, i know, i have an anolog phone plugged into it, and then the spa3102 logs into asterisk, and that phone appears as extenton 103 on the asterisk |
| 12:17.45 | asr33 | snom runs Linux :) |
| 12:18.01 | admgecko | same with the pap2t, plug the phone in, and the pap2t logs in to extention 104 |
| 12:18.02 | mosty | admgecko, make a call using the analogue telephone, with the spa not plugged into asterisk |
| 12:18.15 | admgecko | unplug the network? |
| 12:18.23 | mosty | analogue phone - spa3102 - telephone line |
| 12:18.54 | [gnubie] | mosty: i see. thanks.. |
| 12:19.28 | asr33 | thanks mosty |
| 12:19.52 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 12:20.51 | admgecko | <PROTECTED> |
| 12:21.01 | admgecko | when i pickup the line |
| 12:21.03 | admgecko | it rings, |
| 12:21.33 | admgecko | i get a dial tone, i dial the number, then nothing, then the number you have dialed has not been reconised |
| 12:21.55 | mosty | check the spa3102's dialplan? |
| 12:22.45 | admgecko | im just doing that now |
| 12:23.59 | admgecko | Line 1, fallback dialplan was set to noe |
| 12:24.14 | admgecko | set it to 1, and in dialplan 1, put (xx.) ??? |
| 12:24.46 | mosty | i can't help with the spa3102 dialplan |
| 12:25.01 | admgecko | i think that right, ill give it a go |
| 12:26.27 | admgecko | ok we have the speaking clock :D |
| 12:27.15 | mosty | now see if that call lasts more than 10 minutes |
| 12:30.34 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 12:33.08 | admgecko | ok call started |
| 12:33.41 | admgecko | 2:23 random disconnect |
| 12:35.15 | mosty | ok so it's definitely a problem with the spa |
| 12:35.18 | admgecko | for this call, call before was 4 minutes.... |
| 12:36.56 | admgecko | from the web config Last PSTN Disconnect Reason:PSTN Activity Timeout |
| 12:39.04 | mosty | see if there's a timeout setting you can disable |
| 12:41.15 | admgecko | im just looking for that |
| 12:41.16 | admgecko | :D |
| 12:46.36 | admgecko | im going to try turning off detect pstn long silence |
| 12:52.00 | mosty | that sounds like a good bet |
| 12:56.21 | *** join/#asterisk mirael_karamir (n=miral_ka@195.189.234.153) |
| 12:57.15 | mirael_karamir | hi all |
| 12:58.30 | mirael_karamir | anybod can tell me, how ENUMLOOKUP works? |
| 13:04.21 | kaldemar | mirael_karamir: read doc/enum.txt in the source package. |
| 13:05.50 | mirael_karamir | i hav problm - incoming e164 calls workd fine, but i cant cinfigure outgoing calls |
| 13:06.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 13:23.39 | mirael_karamir | i read doc/enum.txt and try example 1 |
| 13:24.28 | mirael_karamir | [Dec 15 15:21:45] WARNING[16629]: chan_sip.c:2867 create_addr: No such host: |
| 13:24.28 | mirael_karamir | [Dec 15 15:21:45] WARNING[16629]: app_dial.c:1131 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No |
| 13:24.44 | mirael_karamir | but have this error |
| 13:26.13 | kaldemar | use NoOp to debug what you got with the ENUM lookup. |
| 13:26.22 | mirael_karamir | heres my config |
| 13:26.23 | mirael_karamir | <PROTECTED> |
| 13:26.23 | mirael_karamir | exten => _882999..,1,Set(enumresult=${ENUMLOOKUP(+${EXTEN:3})}) |
| 13:26.23 | mirael_karamir | exten => _882999.,n,Dial(SIP/${enumresult}) |
| 13:26.48 | kaldemar | first get your lookups working, then worry about the dial. |
| 13:27.02 | kaldemar | and don't paste configs here, use pastebin. |
| 13:27.14 | mirael_karamir | ok, what can i do? |
| 13:27.35 | mirael_karamir | i am newbie in IRC, sorry ;) |
| 13:28.03 | kaldemar | ~pastebin |
| 13:28.04 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:29.07 | kaldemar | but, for debugging purpose, insert a priority after 1 that has NoOp(${enumresult}) and check if you're actually getting a valid result with the lookup. |
| 13:31.13 | mirael_karamir | NoOp("SIP/1000-08202370", "") in new stack |
| 13:32.07 | mirael_karamir | no valid result? is it DNS problem? |
| 13:32.31 | kaldemar | looks like you're not getting any response. might be a DNS problem. |
| 13:34.02 | mirael_karamir | how can i check this? |
| 13:39.08 | kaldemar | this is something i've never used myself, but i'd start by checking the domain in enum.conf. |
| 13:39.54 | admgecko | mosty : just done a 30 min call |
| 13:40.02 | admgecko | mosty: so i think we are fixed |
| 13:40.05 | mosty | admgecko, cool |
| 13:40.28 | admgecko | mosty: it wasnt the timeout, it was the disconnection tone |
| 13:40.49 | mosty | wrong setting for your country? |
| 13:41.36 | admgecko | right setting for my country, wrong setting for my screwed up monoplistic telephone provider |
| 13:42.28 | admgecko | http://en.wikipedia.org/wiki/Kingston_Communications#Monopoly_issues |
| 13:42.30 | mirael_karamir | here is my enum.conf http://pastebin.com/d56de8ac2 |
| 13:47.32 | Greek-Boy | any chan_ss7 users here? |
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| 14:41.53 | Greek-Boy | NOTICE[2001]: chan_sip.c:14861 handle_request_register: Registration from '<sip:stelio@172.16.77.69>' failed for '172.16.77.7' - Not a local domain |
| 14:42.02 | Greek-Boy | why wont it register? coz its on same subnet? |
| 14:48.31 | *** join/#asterisk ThoMe (n=tm@keks.be) |
| 14:48.33 | ThoMe | hello |
| 14:48.38 | ThoMe | hola! :) |
| 14:48.46 | ThoMe | have an error if i compile asterisk: |
| 14:48.48 | ThoMe | cc1: error: unrecognized command line option "-march=hppa2.0" |
| 14:48.51 | ThoMe | cc1: error: unrecognized command line option "-march=hppa2.0" |
| 14:48.53 | ThoMe | make[2]: *** [src/add.o] Fehler 1 |
| 14:48.56 | ThoMe | make[1]: *** [gsm/lib/libgsm.a] Fehler 2 |
| 14:48.58 | ThoMe | make: *** [codecs] Fehler 2 |
| 14:49.01 | ThoMe | have a hppa |
| 14:53.08 | ThoMe | hello? |
| 15:05.30 | *** join/#asterisk [gnubie] (n=[gnubie]@cm174.gamma179.maxonline.com.sg) |
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| 15:25.40 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 17:16.24 | shyam_k | hi anyone here from kerala,INDIA? |
| 17:17.32 | d3wayne | Huntsville, AL !! |
| 17:19.07 | jer | Soviet Canuckistan! |
| 17:23.22 | russellb | d3wayne: close enough |
| 17:24.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:25.03 | duki | hello |
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| 17:46.29 | kratzers | is it possible to use channel variables with cdr_custom? |
| 17:46.37 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 18:37.55 | *** join/#asterisk franklucas (n=a@209.8.41.72) |
| 18:38.15 | franklucas | hi there boys and girls |
| 18:38.28 | franklucas | can any of you help a newbie in asterisk? |
| 18:39.30 | franklucas | i only have a couple of questions, so i wont bother you for long |
| 18:39.54 | franklucas | please |
| 18:40.30 | fakhir | you will probably get a better result if you ask specific questions in this channel as well............. |
| 18:40.58 | franklucas | ok, i'll try...but why wont you help me if you know it? |
| 18:41.09 | rob0 | huh? |
| 18:41.42 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 18:42.53 | franklucas | i want to make a difference between SIP and IAX callers (in extensions.conf)...everyone dials the same number |
| 18:43.04 | franklucas | but i want to treat them different |
| 18:43.38 | franklucas | eg: for SIP i do certaing functions (saydigit, saynumber) |
| 18:43.51 | franklucas | and for IAX i do record, dictarte |
| 18:44.47 | rob0 | sounds fairly easy, just have SIP go to one context, IAX to another one. |
| 18:45.15 | franklucas | I tryed that and i didn t succeed |
| 18:46.02 | franklucas | i define the SIP numbers like this 1XXX |
| 18:46.11 | franklucas | and the IAX numbers 2XXX |
| 18:46.19 | franklucas | can I use this differece? |
| 18:46.28 | franklucas | and to send them to the same content? |
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| 19:11.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:13.19 | [TK]D-Fender | franklucas, this is your dialplan it does exactly what you tell it to. |
| 19:14.28 | [TK]D-Fender | franklucas, if you want to treat your calls coming from 1 kind of channel differently than another, then set each device to its appropriate dialplan context calling appropriately coded extens for the effect you want |
| 19:15.36 | DarKnesS_WolF | voip-info down !? |
| 19:15.45 | DarKnesS_WolF | [TK]D-Fender: hello dude ;-) |
| 19:21.07 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:27.07 | X-Filez | DarKnesS_WolF: yep, down |
| 19:28.04 | [TK]D-Fender | Damn.... now where are we supposed to get our randomly outdated information?!?! |
| 19:28.37 | *** join/#asterisk CBR1000RR6 (n=Anthony@217.128.127.150) |
| 19:28.52 | CBR1000RR6 | Hello, I use Asterisk 1.2 And I found in my log: too many open files. I saw on Internet to do this: ulimit -n 65536 for example.. But How Asterisk can close these sockets without to change ulimit value? |
| 19:28.54 | DarKnesS_WolF | [TK]D-Fender: i don't know :( |
| 19:30.44 | *** join/#asterisk sacitec (n=tobi@189.149.59.150) |
| 19:34.23 | [TK]D-Fender | ~pb |
| 19:34.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:34.26 | [TK]D-Fender | ^^^^^^^ |
| 19:37.39 | *** join/#asterisk vrtk (n=vyrotiko@201.50.245.75) |
| 19:37.52 | sacitec | hello |
| 19:38.03 | sacitec | anyone has this problem when compiling asterisk 1.2.x ? |
| 19:38.04 | sacitec | http://pastebin.com/m283ef72a |
| 19:38.49 | vrtk | well.. permission denied puts it very clear.. |
| 19:39.12 | [TK]D-Fender | sacitec, I think you're missing automake... |
| 19:39.21 | vrtk | it's very weird because you're already root though |
| 19:39.39 | sacitec | automake u mean.. |
| 19:39.47 | russellb | automake wasn't used at all in asterisk 1.2 |
| 19:39.54 | russellb | furthermore, why would you be compiling asterisk 1.2? |
| 19:40.21 | sacitec | testing |
| 19:41.30 | *** join/#asterisk ManxPower (n=manxpowe@248.sub-75-203-24.myvzw.com) |
| 19:41.38 | sacitec | i'm doing a downgrade from asterisk 1.5 |
| 19:41.40 | sacitec | mean 1.4 |
| 19:41.45 | sacitec | finger error |
| 19:41.47 | vrtk | did you change permission ? |
| 19:42.48 | vrtk | this error is usually reported when you are trying to run something that isn't executable |
| 19:44.11 | sacitec | you're right, problem solved, there was the executable file missing |
| 19:45.11 | vrtk | awesome.. i had an similar issue yesterday |
| 19:45.21 | *** join/#asterisk dennisonicc (n=dennis@cpc1-seve11-0-0-cust650.popl.cable.ntl.com) |
| 19:45.26 | dennisonicc | Hi |
| 19:45.27 | sacitec | now i have tons of warnings |
| 19:45.48 | dennisonicc | can anyone recommend me a VOIP provider for asterisk in UK |
| 19:45.50 | dennisonicc | ? |
| 19:45.56 | vrtk | because, god knows how, i had a file that had it attributes corrupted (see chattr man page) |
| 19:46.21 | vrtk | sacitec: something seems to be missing then.. |
| 19:46.45 | vrtk | it seems we're attacking the sympthons, not the cause.. |
| 19:48.11 | sacitec | triying to locate what's the issue |
| 19:48.30 | sacitec | any value that will change when downgrading from 1.4 to 1.2 ? |
| 19:48.35 | vrtk | warnings aren't likely to point out what the issue is about... |
| 19:49.36 | vrtk | what you mean with value ? |
| 19:51.05 | sacitec | when compiling |
| 19:51.55 | vrtk | you mean ./configure parameters ? |
| 19:52.15 | sacitec | exactly |
| 19:54.23 | vrtk | don't know, i think for a default installation the answer is no.. |
| 19:54.50 | vrtk | you don't need to bother about those values.. |
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| 20:16.47 | CBR1000RR6 | Hello, I use Asterisk 1.2 And I found in my log: too many open files. I saw on Internet to do this: ulimit -n 65536 for example.. But How Asterisk can close these sockets or files who are not used by asterisk without to change ulimit value? |
| 20:18.31 | vrtk | lsof can help you detect what are those files, to know exactly where you need to change something.. |
| 20:20.08 | mvanbaak | CBR1000RR6: do you use AGI scripts ? |
| 20:20.21 | CBR1000RR6 | with lsof command and wc -l it retrun the number of open file. But how can close automatically theses files? |
| 20:20.51 | CBR1000RR6 | mvanbaak which AGI scripts? |
| 20:22.01 | mvanbaak | CBR1000RR6: are you using AGI ? |
| 20:22.28 | mvanbaak | if so, check if those use files and wether they close the files and exit nicely with an exit code |
| 20:22.50 | CBR1000RR6 | ah yes I use AGI script |
| 20:23.05 | mvanbaak | ok |
| 20:23.08 | mvanbaak | so check them |
| 20:23.14 | CBR1000RR6 | ok thanks |
| 20:23.33 | mvanbaak | most common cause of running out of filedescriptors is agi scripts that dont close them |
| 20:23.57 | CBR1000RR6 | ok |
| 20:25.09 | CBR1000RR6 | I will check my AGI scripts |
| 20:25.29 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:31.09 | X-Filez | ppls, voip-info.org not work :( where i can found manual configure SNOM 320 and Asterisk - Conference Button ? Please help |
| 20:33.13 | mvanbaak | voip-info.org works fine here |
| 20:34.43 | mvanbaak | X-Filez: http://lunteren.vanbaak.info/tmp/snom320.htm |
| 20:34.54 | mvanbaak | taken from voip-info.org |
| 20:35.58 | mvanbaak | if that's not the page, you can always use google's cache |
| 20:37.13 | X-Filez | *** Address: www.voip-info.org |
| 20:37.13 | X-Filez | *** Failed: Address Can't Be Resolved ! |
| 20:37.15 | X-Filez | :( |
| 20:38.41 | CBR1000RR6 | www.voip-info.org works fine for me |
| 20:39.24 | CBR1000RR6 | X-Filez: may be a DNS problem for you |
| 20:40.39 | X-Filez | wait, flushdns |
| 20:41.03 | mvanbaak | X-Filez: can I remove that file ? |
| 20:41.31 | X-Filez | wow, worked |
| 20:41.35 | X-Filez | mvanbaak: yep.. |
| 20:41.37 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:41.40 | mvanbaak | cool |
| 20:45.09 | X-Filez | hm, for use button conference need meetme ? |
| 20:45.21 | [TK]D-Fender | X-Filez, No. |
| 20:45.38 | [TK]D-Fender | X-Filez, that is for 3-way calling directly done by your phone. |
| 20:45.55 | [TK]D-Fender | X-Filez, if you want MORE than 2 other people with you, THEN you will need meetme |
| 20:46.40 | X-Filez | [TK]D-Fender: what me need for 3-way ? |
| 20:46.56 | [TK]D-Fender | X-Filez, ? |
| 20:47.19 | [TK]D-Fender | X-Filez, for 3-way calling, your PHONE does all the work, and you don't need anything configured in * |
| 20:47.34 | X-Filez | [TK]D-Fender: name module or small help where configure in asterisk |
| 20:47.54 | [TK]D-Fender | X-Filez, this ISN'T anything to configure for your phone to do 3-way calling. |
| 20:51.16 | X-Filez | brrr |
| 20:51.22 | X-Filez | don't understand.. |
| 20:51.50 | [TK]D-Fender | X-Filez, The is nothing to configure in Asterisk to use 3-way calling on your phone! |
| 20:51.57 | [TK]D-Fender | X-Filez, Your PHONE does all of the work. |
| 20:52.31 | mvanbaak | at least it's supposed to do it ;) |
| 20:53.03 | [TK]D-Fender | mvanbaak, Don't start, this guy is difficult enough as it is. |
| 20:53.21 | X-Filez | aa |
| 20:53.27 | lesouvage | X-file: SIP is kind of confusing. The endipoints like phones are smart by themselve and it is a choice to make use of that. When you set up a trhee way call using the conference button on the snom320 Asterisk ins't involved at all. This something between the phones. |
| 20:53.36 | X-Filez | [TK]D-Fender: thanks :) i how understand :) |
| 20:56.24 | *** join/#asterisk Telamon (n=telamon@blk-7-252-110.eastlink.ca) |
| 20:57.12 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:00.50 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 21:01.52 | *** join/#asterisk LeddyHM (n=none@polar.artica.net) |
| 21:07.17 | X-Filez | [TK]D-Fender: have small question, A_Snom is talking (busy) B_Snom is free, one time call B_Snom but can't answer, I can answer from A_Snom click in pad where LED On is B_Snom ? |
| 21:07.57 | *** join/#asterisk chat_jokey (n=acehunky@123.201.46.202) |
| 21:08.12 | X-Filez | or need send *8 to asterisk ? |
| 21:08.31 | otaku42 | hi all. anyone know a pstn-to-voip service with did numbers in greece? |
| 21:09.49 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 21:14.05 | deever | how can i use the gotoif application in the extensions.ael? |
| 21:14.45 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:14.49 | Qwell | deever: You shouldn't need to use it with ael |
| 21:15.01 | Qwell | If you do, then your dialplan is flawed.. |
| 21:17.18 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:17.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:25.06 | deever | Qwell: i want to HangUp() if Dial() sets ${DIALSTATUS} to "CANCEL"... |
| 21:25.19 | deever | ...how to do that with ael? |
| 21:25.20 | Qwell | so then just use an If |
| 21:27.46 | *** join/#asterisk TuY (n=murta@189.12.47.198) |
| 21:28.04 | TuY | alguem Brasil??? |
| 21:29.03 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:32.08 | mvanbaak | or a switch |
| 21:32.22 | mvanbaak | switch (${DIALSTATUS}) |
| 21:32.28 | mvanbaak | <PROTECTED> |
| 21:32.34 | mvanbaak | <PROTECTED> |
| 21:32.40 | mvanbaak | <PROTECTED> |
| 21:41.18 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:43.06 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 21:43.06 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 21:46.07 | deever | Qwell, mvanbaak: thank you! ;) |
| 22:07.29 | X-Filez | Ppls, in snom have xfer , i press xfer and enter number 206, snom say "redirected to 206" , but asterisk this don't understand, what problem ? please help |
| 22:07.29 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:12.23 | X-Filez | hm, why use LOCAL ? -- Now forwarding SIP/205-b7448510 to 'Local/206@sip_default' (thanks to SIP/105-081b3218) |
| 22:13.21 | [TK]D-Fender | X-Filez, thats just what a transfer looks like |
| 22:13.47 | X-Filez | i want transfer from A SNOM to B SNOM |
| 22:14.11 | [TK]D-Fender | X-Filez, what you are not following is that you don't transfer a call to a PHONE. You are transferring a call to an EXTENSION. |
| 22:14.36 | [TK]D-Fender | X-Filez, An extension is something you can DIAL in your dialplan. If this leads to * dialing another phone, they fine |
| 22:15.00 | [TK]D-Fender | X-Filez, but you do not transfer phones to other phones. |
| 22:15.15 | [TK]D-Fender | X-Filez, You transfer them to EXTENSIONS that would DIAL a phone. |
| 22:15.32 | X-Filez | in dialplan have 206 Dial |
| 22:15.50 | [TK]D-Fender | X-Filez, thats fine. what happens AFTER those lines you pasted? |
| 22:16.42 | X-Filez | hangup |
| 22:16.56 | [TK]D-Fender | X-Filez, pastebin your dialplan. |
| 22:16.57 | [TK]D-Fender | ~pb |
| 22:16.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:16.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 22:17.01 | X-Filez | wait |
| 22:18.03 | X-Filez | [TK]D-Fender: http://pastebin.com/m5a17a400 |
| 22:18.41 | [TK]D-Fender | X-Filez, now please pastebin that CLI output of a complete call attempt including a transfer |
| 22:19.06 | X-Filez | wait sec |
| 22:19.37 | X-Filez | [TK]D-Fender: http://pastebin.com/m4966d14b |
| 22:21.24 | [TK]D-Fender | X-Filez, I want the COMPLETE call. |
| 22:21.33 | X-Filez | ok wait |
| 22:26.39 | mvanbaak | gheh |
| 22:26.45 | X-Filez | sorry for long time, mob phone call |
| 22:26.50 | X-Filez | http://pastebin.com/m7a4d2968 |
| 22:27.06 | mvanbaak | define(COMPLETE, !INCOMPLETE); |
| 22:28.08 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 22:29.59 | X-Filez | [TK]D-Fender: |
| 22:30.07 | [TK]D-Fender | X-Filez, that doesn't even have a DIAL in it. Complete waste |
| 22:30.19 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 22:30.33 | [TK]D-Fender | X-Filez, Sorry, time has run out for me to assist you. Good luck. |
| 22:31.04 | X-Filez | [TK]D-Fender: thanks, for help |
| 22:31.42 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584024.dsl.bell.ca) |
| 22:35.19 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 22:36.51 | deever | which modules need to be loaded for voicemail/voicemailmain? |
| 22:37.20 | deever | when i enter the VoiceMailMain app, nothing happens... |
| 22:37.27 | deever | ...neither an error message |
| 22:38.32 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
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| 22:54.20 | Odie_flocon | hello all. |
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| 22:57.24 | X-Filez | found , need chan_local.so include to modules.conf |
| 22:57.25 | X-Filez | :) |
| 23:01.39 | mikefoo | Anyone know where I could go about purchasing a short code? |
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