| 00:04.54 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
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| 00:44.54 | saftsack | hi, the following case: dial(TELEPHONE1&TELEPHONE2) ... if telephone1 is busy but it is getting available during ringing will it begin ringing after the previous call on this telephone has ended? |
| 00:45.02 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 00:49.59 | *** join/#asterisk ManxPower (n=manxpowe@182.sub-70-221-78.myvzw.com) |
| 00:52.58 | [TK]D-Fender | saftsack, No |
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| 01:01.44 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-30ceb1f9482432a7) |
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| 01:12.45 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:12.56 | ZaVoid | anyone decent at reading backtraces? |
| 01:16.35 | ZaVoid | #4 0x080d7b29 in ast_rtp_bridge (c0=0xb7d33a68, c1=0x92966b0, flags=Variable "flags" is not available.) at rtp.c:3144 <-- is that valid backtrace data at all? |
| 01:17.45 | saftsack | [TK]D-Fender, ok and is there a possibility to fix this? (no pickup ;) ) |
| 01:18.23 | ZaVoid | but bt full gives me this: #1 0x080814e2 in ast_do_masquerade (original=0xb7db2cd0) at /usr/src/asterisk-1.4.15/include/asterisk/lock.h:700 |
| 01:18.23 | ZaVoid | <PROTECTED> |
| 01:20.00 | [TK]D-Fender | saftsack, Sure. Go recode app_dial yourself to constantly retry. Because you see, * can't track what the end-point's response is going to be so it will have to keep trying. |
| 01:20.48 | [TK]D-Fender | saftsack, So what kind of device are you planning on using this with? |
| 01:22.03 | ZaVoid | fender you look at backtraces much? am i looing at the right thing by doing bt full? |
| 01:22.09 | ZaVoid | how you doing too? |
| 01:22.42 | [TK]D-Fender | ZaVoid, if I had something to say I would have.... |
| 01:24.12 | ZaVoid | good point |
| 01:24.18 | ZaVoid | so how you doing? |
| 01:26.16 | saftsack | [TK]D-Fender, sip -> snomphones |
| 01:26.42 | ManxPower | ZaVoid: did you read backtrace.txt included in the Asterisk souce |
| 01:26.43 | [TK]D-Fender | saftsack, and they don't support call-waiting? |
| 01:26.54 | ZaVoid | yes ManxPower |
| 01:27.00 | ManxPower | Just making sure. |
| 01:27.32 | ZaVoid | but i've also been reading a bunch of the bugs posted at bugs.digium.com before i post my crash details and just want to make sure i'm doing it right |
| 01:28.50 | saftsack | what is call waiting? |
| 01:28.57 | saftsack | ok i will look at voip-info :-P |
| 01:29.19 | [TK]D-Fender | saftsack, .... *BEEP* |
| 01:29.31 | [TK]D-Fender | saftsack, You know... the ability to take 1 call while on another.... |
| 01:29.49 | [TK]D-Fender | saftsack, not knowing this stuff is nearly a shoot-on-sight offense in telecom.... |
| 01:30.27 | ZaVoid | since 1.4.15 crashes on me every few days now.. no where near as stable as 1.4.9(never crashed) |
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| 01:34.58 | saftsack | [TK]D-Fender, sorry no one has your knowledge -.- . do you think there are some people from other countries which have another languages and another meaning for some things |
| 01:35.29 | [TK]D-Fender | saftsack, my GRANDPARENTS know what call-waiting is :) |
| 01:36.01 | blitzrage | ZaVoid: have you gotten a backtrace and posted a bug to the bug tracker? |
| 01:36.18 | [TK]D-Fender | saftsack, and having nice SIP phones I'd have thought you'd have gotten calls "beeping through" while on other calls with pretty flashing lights and other painfully obvious signs of "gee I CAN take that call!" |
| 01:36.24 | ZaVoid | i want to make sure i post it right before i post it blitzrage |
| 01:37.39 | saftsack | [TK]D-Fender, this feature is called "klopfen" ;) here in germany. and in the standard setting it is an audio signal and we turned it completely off because it annoyes the persons on the telephones |
| 01:37.48 | *** join/#asterisk BeeBuu (n=beebuu@218.13.89.201) |
| 01:38.43 | [TK]D-Fender | saftsack, yeah, God knows you wouldn't want them interrupting that call to their family for a call from their BOSS. |
| 01:38.56 | ManxPower | Call Waiting is the tool of the devil |
| 01:39.34 | saftsack | here in germany we can trust our people which are on the telephne |
| 01:42.13 | blitzrage | you can't trust anyone |
| 01:42.16 | blitzrage | I barely trust myself |
| 01:42.38 | blitzrage | I have passwords that I don't tell to my other personality all the time just so he doesn't screw up all my hard work |
| 01:43.08 | blitzrage | hey, womens boxing is on.... |
| 01:43.40 | blitzrage | now I'm silky smooth |
| 01:44.59 | coppice | they're putting women in boxes? :-\ |
| 01:45.13 | blitzrage | yep, and shipping them overnight |
| 01:45.23 | ManxPower | coppice: No, I think it's women putting presents in boxes. |
| 01:45.47 | ManxPower | The Elves are unionized, so..... |
| 01:47.01 | coppice | You mean Santa has gone DIY? |
| 01:49.01 | coppice | [TK]D-Fender: you should be careful with those parcels. unless they arrive on time, you might be horrified by how the contents have gone downhill. |
| 01:52.32 | coppice | and if you hear any moaning noises, return it without opening |
| 01:55.09 | [TK]D-Fender | coppice, kill-joy! |
| 01:55.40 | coppice | you let a moany one out of the box, and she wil certainly kill any joy |
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| 02:14.05 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 02:14.11 | jsoftw | Anyone know about ztdummy on freebsd? |
| 02:14.57 | JT | zaptel was written for linux, but there is a bsd zaptel project |
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| 02:23.25 | seanwg123 | anyone know if there are any ways to figure out why a inbound IAX2 call to a ring list fails constantly? |
| 02:23.27 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1088841247.dsl.bell.ca) |
| 02:23.46 | fujin_ | ring list? |
| 02:23.48 | seanwg123 | the ring list contains an extension and an external number |
| 02:24.09 | seanwg123 | yah a ring list in freepbx |
| 02:24.47 | JT | ~freepbx |
| 02:24.47 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:24.47 | fujin_ | #freepbx |
| 02:24.54 | fujin_ | OH SNAP |
| 02:24.57 | fujin_ | hat five JT |
| 02:25.18 | fujin_ | seanwg123: if you were using plain asterisk, I might be more inclined to help - sorry |
| 02:25.31 | JT | vewy niice |
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| 02:58.51 | jsoftw | freebsd is quite... erm.. complex |
| 02:58.56 | jsoftw | erm, freepbx I mean |
| 03:03.58 | fujin_ | s/complex/crap/? |
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| 03:21.03 | Mavvie | Anybody here used a Cisco Call Manager system together with Asterisk doing the voicemail? |
| 03:22.44 | jsoftw | fujin_: might as well be |
| 03:22.58 | fujin_ | Mavvie: soudns overly painful |
| 03:23.03 | jsoftw | Im getting only about 85% on zttest |
| 03:23.07 | jsoftw | using ztdummy :/ |
| 03:23.18 | Mavvie | fujin_: I wouldn't say no to that. |
| 03:23.29 | Mavvie | fujin_: I've most of it working, except for MWI. |
| 03:23.44 | Mavvie | fujin_: but it is euhm... configuration hell to do it right. |
| 03:23.57 | fujin_ | Mavvie: what phones for mwi? |
| 03:23.59 | fujin_ | actually |
| 03:24.03 | fujin_ | I don't think you're going to get mwi tbh |
| 03:24.09 | Mavvie | fujin_: 79xx series. |
| 03:24.22 | fujin_ | unless you can get the 79xx's to subscribe to a mailbox on another Ip address |
| 03:24.25 | Mavvie | fujin_: that's what I was thinking. It's a Proof of Concept anyway. |
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| 03:34.46 | *** join/#asterisk _pepo_ (i=pepo@gateway/tor/x-920b2fc27db9ee25) |
| 03:35.05 | _pepo_ | hi friends |
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| 03:45.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 03:45.30 | Zuchmir | are there any software SIP clients that display the SendText() |
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| 04:02.17 | *** mode/#asterisk [+o mog] by ChanServ |
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| 04:08.14 | craigk | anybody know if i can change the callgroup/pickupgroup of an extension at runtime, or can it only be done statically in the conf files ? |
| 04:08.27 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 04:27.48 | [TK]D-Fender | craigk, only gets set when the configs are read in. |
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| 04:28.13 | craigk | thanks TK ... i thought that would be the case, but just wanted to make sure :) |
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| 04:34.41 | Somax | I have a Asterisk novice question. I'm trying to add simple voice-conferencing features for my web 2.0 service. Is it possible to use open source softphone clients connected to an Asterisk server, for signaling and audio mixing? Which soft-phone is best to start with? Which URLs/tutorials are best, to start with to get this setup? Thanks |
| 04:35.50 | [TK]D-Fender | Somax, I've done 1on1 video, but never 3-way+ |
| 04:35.51 | [TK]D-Fender | ~book |
| 04:35.52 | jbot | [book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 04:35.53 | [TK]D-Fender | ~wikis |
| 04:35.53 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 04:35.57 | [TK]D-Fender | There ---^^ |
| 04:36.43 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 04:42.49 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 04:43.15 | watchy | does * give packets a TOS mark? |
| 04:43.17 | blitzrage | [TK]D-Fender: "Somax, I've done 1on1 video, but never 3-way+" ..... hahaha |
| 04:43.44 | [TK]D-Fender | blitzrage, yup, that come was "fully loaded" wasn't it? ;) |
| 04:43.59 | blitzrage | too funny |
| 04:44.12 | watchy | tk: does * put any TOS stuff in packets? |
| 04:44.49 | WilliamK | watchy, it's in the ip-tos doc |
| 04:44.49 | mosty | watchy, it can |
| 04:44.50 | Somax | Thanks.. quick follow up novice questions. For the 1-1 video, can each user identify who they conference with, based on some other ID (like dynamically generated conference number or URL) than each other's login or some other unique ID? thanks |
| 04:44.51 | [TK]D-Fender | watchy, its rude to target individuals like that |
| 04:44.53 | JT | it does |
| 04:45.16 | watchy | i'm trying to make my mikrotiks see the special packets and give priority |
| 04:45.27 | watchy | i wanna prioritize my VOIP traffic |
| 04:45.36 | JT | watchy: ok |
| 04:45.54 | watchy | you ever done that JT? |
| 04:45.55 | [TK]D-Fender | Somax, Basically you need either registered users (not what you want), or nearly un-authed. You need to go download * and just TRY and see how it works for you. |
| 04:46.09 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 04:46.38 | Somax | ok |
| 04:47.28 | webman | ~pastebin |
| 04:47.28 | 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 |
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| 04:48.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:49.48 | JT | watchy: no |
| 04:51.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 04:53.47 | watchy | hmm |
| 04:53.54 | watchy | man i need to figure this out |
| 04:54.17 | JT | so read the documentation? :) |
| 04:54.52 | watchy | working on it |
| 04:55.02 | watchy | i lack any knowledge of QoS/ToS anything |
| 04:57.30 | webman | can someone look at http://pastebin.com/d110e8cc0 (sip debug) and provide any clues as to what is going wrong? |
| 04:58.32 | webman | It seems to say the codec is not allowed, but they are never asked to send their username/password (their account does allow g729, so then after auth they should be allowed) |
| 05:01.29 | mosty | webman, is the sip client registered? |
| 05:01.57 | webman | mosty: how do I check that ? |
| 05:02.06 | mosty | sip show peers |
| 05:02.58 | webman | it says IP is "unspecified" does that mean it is not registered? |
| 05:03.38 | mosty | yes |
| 05:05.01 | [TK]D-Fender | webman, "From: <sip:220.123.88.111>;tag=1352ADC4-122" <--- no user on incoming |
| 05:05.37 | [TK]D-Fender | webman, Found description format G729 for ID 18 <---- this is a licensed codec, did you pay for and install them? |
| 05:05.58 | [TK]D-Fender | webman, SIP/2.0 488 Not acceptable here <- oh... and * doesn't even AGREE to them OFFERING it. |
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| 05:06.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-107-228.lns10.syd6.internode.on.net) |
| 05:08.44 | *** join/#asterisk gardo (n=gardo@121.97.193.52) |
| 05:10.39 | webman | mosty: even if the user is not registered, normally when I see the sip client send an invite, asterisk sends back a auth required or similar, but it doesn't do that here ? |
| 05:11.36 | webman | also, the specific user section has allow=g729 so I suppose the real problem is they are not authenticating/registering? |
| 05:12.38 | [TK]D-Fender | webman, its not HITTING your user section. The call isn't ID'd as coming from anyone. Check your invite |
| 05:12.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 05:13.43 | webman | ok thanks... |
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| 05:24.30 | gardo | how can you see all the supported audio formats of mixmonitor/monitor? |
| 05:25.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 05:27.38 | [TK]D-Fender | gardo, "show translation" |
| 05:28.18 | *** join/#asterisk mihinomenest (i=v3k4@66.255.220.17) |
| 05:28.22 | gardo | oh that one |
| 05:28.29 | gardo | [TK]D-Fender: thanks! |
| 05:28.36 | gardo | i forgot about that |
| 05:29.45 | gardo | is it possible for mixmonitor to record directly to mp3 format? |
| 05:29.59 | gardo | i usually use the gsm format |
| 05:31.10 | [TK]D-Fender | gardo, Clearly not. |
| 05:31.31 | [TK]D-Fender | gardo, You can record in any format * can transcode, and MP3 isn't it. |
| 05:31.46 | gardo | is there a way to build asterisk w/ mp3 support? |
| 05:31.57 | [TK]D-Fender | gardo, You can always call a post-recording app to convert it for you. |
| 05:32.16 | [TK]D-Fender | gardo, native, sure there's a way, but not legally includable with *. |
| 05:32.32 | gardo | hmm... |
| 05:33.19 | gardo | seems that i have no other choice but to call a post-recording app to do the conversion |
| 05:34.27 | *** join/#asterisk dec (n=tom@unaffiliated/dec) |
| 05:38.23 | dec | is it possible to have Pickup() pick up an extension on a remote asterisk server? |
| 05:40.13 | [TK]D-Fender | dec, no |
| 05:40.51 | *** part/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:41.46 | *** join/#asterisk sg` (n=saurabh@59.160.224.34) |
| 05:42.14 | sg` | hi |
| 05:58.23 | *** join/#asterisk njsf (n=njsf@sxemacs/devel/njsf) |
| 05:58.40 | sg` | how can i interrupt a call and play a custom message to caller and then return him back to call. |
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| 05:59.52 | njsf | quick newbie question. Is asterisk capable of doing a POTS callout on a voice modem? |
| 06:00.26 | njsf | i.e. being a gateway with just a voice modem |
| 06:01.05 | [TK]D-Fender | njsf : no, you need to use a compatible FXO device, and no, some crappy modem will not do |
| 06:01.29 | [TK]D-Fender | sg`, who & how would this be initiated? |
| 06:02.58 | sg` | i would like to do this from an AGI script |
| 06:03.18 | sg` | inform a caller periodically how much balance he has left for a calling card |
| 06:03.30 | [TK]D-Fender | sg`, AGI doesn't just happen in the middle of a call. WHO decides that this recording is to be played? |
| 06:04.29 | njsf | tnx [TK]D-Fender. I suspected as much. Just wanted to confirm |
| 06:04.29 | [TK]D-Fender | sg`, Ah, automated. Use a local channel via 1 ) AMI Originate or 2 ) Call File. This would use Chanspy w/ whisper mode. Go WIKI this all up |
| 06:04.30 | [TK]D-Fender | ~wikis |
| 06:04.31 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 06:04.44 | *** part/#asterisk njsf (n=njsf@sxemacs/devel/njsf) |
| 06:05.55 | sg` | [TK]D-Fender, thanks |
| 06:06.00 | sg` | [TK]D-Fender, checking |
| 06:09.47 | [TK]D-Fender | ~book |
| 06:09.48 | jbot | book is, like, 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 |
| 06:09.49 | [TK]D-Fender | ^^^^^^^ |
| 06:09.54 | [TK]D-Fender | while you're at it |
| 06:14.24 | [TK]D-Fender | ok, checkout time, later all |
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| 07:27.54 | quelo | Hi |
| 07:28.08 | phix | hey |
| 07:28.08 | phix | sup? |
| 07:30.18 | quelo | I've configured a trunk between an Avaya IP406 and an asterisk box and now can I make calls from an asterisk extension to an Avaya extension putting 67 in front of the Avaya extension number |
| 07:30.24 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:30.37 | quelo | Avaya use h323 VoIP protocol |
| 07:31.20 | quelo | but I can't call from an Avaya extension to an asterisk extension |
| 07:32.20 | quelo | http://paste.debian.net/44557 |
| 07:32.37 | quelo | this is the complete CLI output at verbosity 10 |
| 07:32.50 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 07:32.54 | quelo | there is anyone can help me? |
| 07:33.09 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:33.10 | coppice | Why do people keep confusing proprietary with encumbered? I bet people like MS love that. :-\ |
| 07:35.56 | *** part/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 07:38.16 | mosty | quelo, do the calls come in to asterisk? |
| 07:39.52 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 07:40.21 | kaldemar | quelo: complete output? i can't see a Dial line in that output. |
| 07:41.19 | quelo | mosty yes |
| 07:42.50 | quelo | kaldemar I've done from the command prompt asterisk -vvvvvvvvvvr and I've done call from avaya to asterisk (67blabla) |
| 07:42.55 | McDouglas | if i want to connect an E1 line, is this a good card to chose: http://www.digium.com/en/products/digital/te120p.php ? |
| 07:43.08 | mosty | quelo, see what kaldemar said. where is the Dial command in that? can you show us extensions.conf? |
| 07:43.56 | mosty | McDouglas, i would recommend an sangoma a104d (if you need 4 or fewer ports) |
| 07:44.09 | McDouglas | i only need one |
| 07:45.01 | quelo | kaldemar there isnt a DIAL command because the machine that make call is'nt asterisk but avaya, asterisk receive call! |
| 07:45.53 | McDouglas | mosty: how about A101 ? |
| 07:46.03 | mosty | McDouglas, it would work. sangoma cards tend to be better, and have better software, but i don't think they make a single port E1 card |
| 07:46.23 | McDouglas | wellhttp://www.sangoma.com/datasheets/p_a101-specs |
| 07:46.26 | McDouglas | loks like they do |
| 07:46.58 | kaldemar | quelo: yes, but there should be one if asterisk tries to dial your extension. now it just executes a hangup macro. showing your dialplan and h323 config would help. |
| 07:47.02 | mosty | hmm ok, then i'd go for the a101d personally. the d versions have hardware echo cancellation |
| 07:47.02 | quelo | mosty there isnt a DIAL command because the machine that make call is'nt asterisk but avaya, asterisk receive call! |
| 07:47.31 | mosty | quelo, but you need to asterisk to forward the call on to some other device, and to do that you have to use the Dial command |
| 07:47.47 | quelo | one moment |
| 07:47.54 | kaldemar | quelo: are you using trixbox, by the way? |
| 07:51.06 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:51.13 | quelo | kaldemar yes |
| 07:51.43 | quelo | http://paste.debian.net/44559 this is my ooh323.conf |
| 07:52.40 | kaldemar | quelo: trixbox is not supported on this channel. you're more likely to get help in #trixbox or #freepbx. |
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| 08:46.47 | ice_croft | hi |
| 08:47.18 | ice_croft | plz, give me url to html version of o'reilly asterisk? |
| 08:48.32 | tzafrir | ~thebook |
| 08:48.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 08:48.53 | tzafrir | hmm... didn't that have the html version URL? |
| 08:49.17 | ice_croft | yes. i saw it 2 days ago. :) |
| 08:49.35 | ice_croft | the url is on my nb . :( |
| 08:51.21 | mosty | ask mr google |
| 08:51.27 | maagic | http://tfot.leifmadsen.com/ |
| 08:51.35 | ice_croft | o |
| 08:51.36 | ice_croft | thanx |
| 08:51.53 | maagic | np. |
| 08:52.16 | sergee | ~seen wsuff |
| 08:52.18 | jbot | wsuff <n=wsuff@c-76-111-207-155.hsd1.fl.comcast.net> was last seen on IRC in channel #asterisk-doc, 167d 18h 43m 6s ago, saying: 'ya saw it on presale a few places'. |
| 08:54.14 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 08:57.07 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:58.22 | FlatFoot | ~seen blitzrage |
| 08:58.23 | jbot | blitzrage <n=Leif@asterisk/documenteur-extraordinaire/blitzrage> was last seen on IRC in channel #asterisk, 4h 14m 24s ago, saying: 'too funny'. |
| 08:58.57 | *** join/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 08:59.05 | FlatFoot | tzafrir: morning |
| 08:59.18 | FlatFoot | do you know much about cdr_adaptive_odbc ? |
| 08:59.20 | tzafrir | hi |
| 08:59.32 | tzafrir | not much |
| 09:00.02 | FlatFoot | fair enough , just trying to work out how to install it , or find some ref to see if it's in the 1.4 v |
| 09:00.49 | ronr | hi, I'm trying to attach some polycom IP430 phones to my asterisk server, I followed the instructions in the o'reilly book, but it talks about some files like bootROM, application image, sip.cfg etc. and explains what they are, however, where do I get them from?? (and secondly, could the phone specific files be generated automagically somehow) |
| 09:01.25 | _ShrikE | Those files are included in the polycom firmware package |
| 09:02.24 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 09:03.04 | mosty | ronr, you can write a script/program to generate the files |
| 09:03.06 | ronr | _ShrikE: ok, thx |
| 09:03.16 | *** join/#asterisk kn0x (n=pinochle@75.127.83.151) |
| 09:03.34 | mosty | the config files, anyway |
| 09:04.20 | *** join/#asterisk matmoj (n=matmoj@fw.packetfront.com) |
| 09:05.23 | ronr | mosty: and how would I make sure that the script is ran if a phone tries to download non-existing config files? |
| 09:08.33 | matmoj | im having problems detecting my pstn card with my asterisk |
| 09:08.41 | matmoj | i have the wct1xxp module loaded |
| 09:08.58 | matmoj | but running zttool doesent show me my te120p card... |
| 09:09.46 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:15.16 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 09:15.46 | n3glv | matmoj, is that an x100p variant? |
| 09:16.12 | n3glv | there may be a script called 'genzaptelconf' |
| 09:16.28 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 09:16.55 | matmoj | n3glv: i cant say if it's |
| 09:16.59 | matmoj | but yes sounds like it by the name of it |
| 09:17.03 | matmoj | its a digium original card |
| 09:18.01 | mosty | ronr, you can run the script ahead of time, or as a cgi script or similar |
| 09:18.19 | n3glv | digium "original" single port "wildcard" is x100p |
| 09:18.35 | mosty | the te120p is a PRI card |
| 09:18.41 | n3glv | ahh |
| 09:18.54 | n3glv | well, if that script may get you somewhaere |
| 09:18.58 | matmoj | i ran the script |
| 09:19.04 | matmoj | 3 channels configured |
| 09:19.06 | n3glv | what did it do? |
| 09:19.08 | n3glv | ahh |
| 09:19.15 | matmoj | (i also have another card in there) |
| 09:19.20 | matmoj | wich gets detected |
| 09:19.29 | matmoj | but thats not the card im interested right now |
| 09:19.55 | matmoj | since the connection to the pstn i have is a pri one |
| 09:20.15 | n3glv | ahh |
| 09:20.41 | mosty | matmoj, perhaps you want the wcte12xp module for that card? |
| 09:21.00 | matmoj | mosty: ok didnt know there was one |
| 09:21.05 | matmoj | im using debian "stable" as my dist |
| 09:21.29 | mosty | that's from zaptel 1.4.7, i'm not sure it it's in zaptel 1.2 |
| 09:21.51 | matmoj | i should be able to downlod the source for those moduoes from asterisk.org i guess |
| 09:21.56 | matmoj | will the work with 1.2? |
| 09:21.59 | matmoj | modules |
| 09:23.08 | *** join/#asterisk justnulling2 (n=menashe@ool-457bcf75.dyn.optonline.net) |
| 09:24.38 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:24.54 | mosty | i don't think so |
| 09:24.59 | FlatFoot | anyone installed and conversant with frrebsd ? |
| 09:25.10 | FlatFoot | *freebsd |
| 09:25.46 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:25.47 | ice_croft | FlatFoot> err.., what? |
| 09:26.35 | FlatFoot | ice_croft: got * installed on freebsd , i'm not too good with freebsd and i'm trying to change a Makefile to install cdr_adpative_odbc |
| 09:26.45 | FlatFoot | just wondered if anyone had managed this ? |
| 09:28.36 | ice_croft | FlatFoot> well, i didnt use odbc on it, i'm usin mysql |
| 09:28.47 | ice_croft | FlatFoot> work fine |
| 09:37.47 | matmoj | what do you guys think |
| 09:37.51 | matmoj | asterisk 1.4 or 1.2? |
| 09:38.14 | matmoj | also i have been trying to go with the "stable" packages in my distribution but apparently they dont have support for my te120 card ... |
| 09:47.19 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:48.18 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-cd3ddabb862f1b9f) |
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| 09:53.26 | ronr | anyone has an example dhcpd.conf for me that'll pass information like the ftp server to a polycom (IP 430 / IP 601) phone? |
| 09:53.36 | FlatFoot | ice_croft: ta anyway sorry for the slow response got called away |
| 09:54.13 | mosty | matmoj, maybe its supported but with a different module name |
| 09:54.53 | mosty | ronr: use the tftpd-server option, just specify a http://url |
| 09:54.57 | chris_1 | does somebody knows a cheap voip router supporting t38, fallback 2 pstn? |
| 09:55.35 | ronr | mosty: thx |
| 09:55.44 | n3glv | chris_1, look around for Sunrocket ac-211 |
| 09:55.46 | n3glv | used |
| 09:55.52 | n3glv | I got one for $5 |
| 09:56.35 | n3glv | took 2 tries on google to get pw, and 10 min later was on my pbx and safe from upgrades... |
| 10:01.56 | *** join/#asterisk mkl1525 (n=qwertz@p5098c328.dip0.t-ipconnect.de) |
| 10:05.16 | mkl1525 | Hi, (* 1.2) a channel that doesn't get killed by "soft hangup" - is there any other way to kill the channel without restarting the whole *? |
| 10:07.34 | WilliamK | just thinking maybe zap destroy channel ? |
| 10:07.45 | WilliamK | sorry - gotta sleep but that's food for thought |
| 10:08.01 | n3glv | u can't do restart now? |
| 10:08.08 | n3glv | or when convienant? |
| 10:08.27 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 10:08.55 | mkl1525 | restart isn't an option atm - it's a SIP channel btw |
| 10:16.27 | mkl1525 | sip show channels shows "10.0.6.5 254 15e3da8d0bb 00104/00002 alaw Yes (d) Tx: BYE" |
| 10:23.48 | ice_croft | hi ppl |
| 10:23.57 | ice_croft | i have dumb question |
| 10:25.03 | ice_croft | http://pastebin.ca/810845 |
| 10:25.30 | ice_croft | with this dialplan i have handle_request_invite: Call from '1000' to extension '500' rejected because extension not found. |
| 10:25.40 | ice_croft | what i do wrong? |
| 10:34.18 | joelsolanki | Hi Good morning |
| 10:34.41 | joelsolanki | Trying to configure call forwarding. facing some problem. let me pastebin it |
| 10:40.02 | joelsolanki | here it is http://www.pastebin.ca/810853 |
| 10:40.41 | joelsolanki | if i set @huskervoip it works but when i set @digitalphone-unlimited it wont and get some error on cli. plz check pastebhin |
| 10:44.32 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:44.47 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 10:48.54 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
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| 10:51.16 | Dr-Linux | how i can save agents call at my favorite location with selected format? |
| 10:55.38 | mosty | use MixMonitor in your dialplan, or one-touch recording (see features.conf) |
| 10:59.54 | ronr | I got my polycom to download its configuration from the server, is there a howto on how to let the phone know about my asterisk server (the per phone .cfg looks a bit too overwhelming to go read, edit and try) |
| 11:01.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:01.21 | ronr | (I want to get 'working' today, 'working a bit better' tomorrow and 'working good and safe' next week) |
| 11:02.12 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 11:04.03 | Dr-Linux | hi mosty |
| 11:04.54 | Dr-Linux | mosty: agents.conf already has this option, also it provide a choice option if i want my favorite location, but i want my favorite format |
| 11:05.12 | Dr-Linux | here is agent.conf : |
| 11:05.13 | Dr-Linux | ; The optional directory to save the conversations in. The default is |
| 11:05.13 | Dr-Linux | ; /var/spool/asterisk/monitor |
| 11:05.13 | Dr-Linux | ;savecallsin=/var/calls |
| 11:06.33 | mosty | Dr-Linux, the MixMonitor command takes options |
| 11:06.52 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 11:07.30 | Dr-Linux | mosty: i'm not sure how i can put format option in extensions.conf for agents.conf :S |
| 11:08.05 | Dr-Linux | i know commands, but not sure if agents.conf directory communicats |
| 11:08.20 | Dr-Linux | agents.conf communicate with queues.conf agents |
| 11:09.08 | mosty | http://www.voip-info.org/wiki-Asterisk+config+agents.conf see the format options |
| 11:13.24 | Dr-Linux | oh, not recorded call format, i mean i wanna create location by date |
| 11:14.00 | Dr-Linux | like this: |
| 11:14.01 | Dr-Linux | /home/callcenter/MCP/CardFlex/${STRFTIME(${EPOCH},,%Y%m%d)}/${CALLERIDNUM}-${STRFTIME(,,%c)}) |
| 11:15.03 | Dr-Linux | mosty: what it do is, every day create a new folder as date in a specific client's folder |
| 11:15.22 | Dr-Linux | but i wanna record calls from agents.conf |
| 11:15.34 | Dr-Linux | but not sure how can i do that with agents.conf recording |
| 11:15.51 | Dr-Linux | mosty: hopefully my bad english makes you understand. |
| 11:16.46 | mosty | oh you mean the filename format... |
| 11:17.24 | Dr-Linux | mosty: that's correct |
| 11:17.29 | mosty | perhaps you can use the urlprefix setting |
| 11:18.21 | Dr-Linux | mosty: yes, but there i can set only location, but not sure how i can use "${STRFTIME(${EPOCH},,%Y%m%d)}" this :S |
| 11:18.25 | *** part/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:18.56 | Dr-Linux | mosty: i've many clients, so wanna manage this way |
| 11:20.37 | mosty | i doubt you can do it then. but you can use MixMonitor before you send callers to the queue |
| 11:22.46 | Dr-Linux | hhm.. MixMonitor enable recording for me |
| 11:23.04 | Dr-Linux | then again queue queue will do recording for me |
| 11:23.13 | Dr-Linux | then i'm recording from agents.conf |
| 11:23.19 | Dr-Linux | so these will 3 recordings |
| 11:23.36 | Dr-Linux | but i think i can't do that since unless i can use variables in agents.conf |
| 11:25.58 | Dr-Linux | hhm.. |
| 11:26.11 | Dr-Linux | mosty: let's look for other solution then |
| 11:26.20 | McDouglas | I need a card to connect isdn bri into asterisk. Any suggestions? |
| 11:26.31 | Dr-Linux | can we use any variable in dialplan which shows agent's ID? |
| 11:30.18 | mosty | Dr-Linux, so use MixMonitor, and disable it in agents.conf? |
| 11:31.13 | Dr-Linux | mosty: that's what i'm already using, but how can i use agent's ID in call file? |
| 11:31.46 | Dr-Linux | mosty: actually i wanna put agent's name in recorded file name, so i could know who answered the call |
| 11:32.04 | mosty | that won't work with monitor, since you don't know the agent that will answer |
| 11:32.29 | Dr-Linux | yes, but i've configured agents name in agnents.conf |
| 11:32.53 | Dr-Linux | so if i start recording in agents.conf option, than that records the agents name as well, but i don't use variables over there |
| 11:33.09 | Dr-Linux | look here: |
| 11:33.11 | Dr-Linux | agent-wilson-1196951899-2769.wav |
| 11:33.31 | Dr-Linux | now i know wilson answered this call |
| 11:34.01 | R1ck | do you have an agent smith aswell? |
| 11:34.29 | mosty | i have come to the conclusion that asterisk queues are annoying, next time i need to implement a call queue, i will do it manually in dialplan + agi logic |
| 11:35.34 | Dr-Linux | mosty: you are right |
| 11:36.01 | Dr-Linux | mosty: asterisk agents system is deprecated now |
| 11:40.33 | ice_croft | ppl, help me plz!!! |
| 11:40.45 | ice_croft | dialplan - http://pastebin.ca/810845 |
| 11:40.51 | ice_croft | <PROTECTED> |
| 11:40.57 | ice_croft | why's that? |
| 11:41.37 | ice_croft | fresh asterisk installation |
| 11:41.44 | ice_croft | 1.4.15 |
| 11:45.07 | mosty | ice_croft, what context do your sip(?) clients start in? |
| 11:46.13 | ice_croft | no context at all |
| 11:47.08 | mosty | then that's your problem. you haven't set the context in sip.conf |
| 11:47.53 | ice_croft | well, now i set it to default. still have the error |
| 11:48.51 | mosty | of course- there is no extension 500 in the default context |
| 11:48.53 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:49.09 | ice_croft | khm |
| 11:49.25 | ice_croft | then, what should i dial to hear weasels? :) |
| 11:49.41 | coppice | call any lawyer |
| 11:51.18 | ice_croft | mosty> yes, my call's unrouted, but i cant get any errors cought , just "call rejected" |
| 11:52.07 | mosty | chance s to _X. in the default context |
| 11:52.57 | mosty | change, rather |
| 11:53.34 | ice_croft | i did. same picture |
| 11:55.59 | mosty | set verbose 10 and set debug 10 |
| 11:56.16 | mosty | then paste the output when you make the test call |
| 11:57.39 | ice_croft | [Dec 10 15:00:04] NOTICE[35826]: chan_sip.c:13774 handle_request_invite: Call from '1000' to extension '500' rejected because extension not found. |
| 11:57.39 | ice_croft | Really destroying SIP dialog 'MTRlM2RhMjExYmFiOWI5M2IyZjlmMzA5ZTdiODI3ZTc.' Method: ACK |
| 11:57.45 | ice_croft | that's all |
| 11:57.56 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 11:58.11 | mosty | did you do an extensions reload? |
| 11:58.24 | ice_croft | yes |
| 11:59.43 | ice_croft | "dialplan show" shows actual dialplan |
| 12:00.46 | mosty | paste the output from sip show peer 1000 |
| 12:01.54 | ice_croft | http://pastebin.ca/810897 |
| 12:02.50 | *** join/#asterisk SagaZ- (n=danilo@unaffiliated/dbaio) |
| 12:07.30 | ice_croft | mosty> any comments? |
| 12:08.16 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 12:09.18 | mosty | try setting qualify=yes, force the sip client to reregister, then paste the output of sip show peers |
| 12:09.38 | ice_croft | ok |
| 12:13.05 | ice_croft | ast*CLI> sip show peers |
| 12:13.05 | ice_croft | Name/username Host Dyn Nat ACL Port Status |
| 12:13.05 | ice_croft | 1000/1000 10.0.0.168 D 35800 OK (2 ms) |
| 12:13.05 | ice_croft | 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 12:18.19 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 12:19.00 | ice_croft | mosty> maybe my installation' broken? |
| 12:20.06 | mosty | it's a config issue, not an installation issue. you don't appear to have a sip client 500 |
| 12:20.07 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 12:21.00 | ice_croft | mosty> yes, i don't have it, but i expect to hear error message :( |
| 12:21.28 | ice_croft | mosty> tt_weasels |
| 12:22.18 | mosty | paste your dialplan again, and tell me what sip client 1000 is dialing |
| 12:23.09 | ice_croft | http://pastebin.ca/810904 |
| 12:23.20 | ice_croft | sip-client dialing 500 |
| 12:23.29 | ice_croft | or any other number |
| 12:28.31 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 12:30.17 | ronr | is it normal that a polycom IP 430 takes over 30 minutes to boot up (It's at Welcome! Processing configuration now, has been there for about 10 minutes)? |
| 12:33.02 | McDouglas | I need a card to connect isdn bri into asterisk. Any suggestions? |
| 12:36.41 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 12:36.45 | Assid | heya |
| 12:37.37 | Assid | quick question on the polycom phones if i use the sip.ld file instead of the generic application executable. is there any real difference? |
| 12:39.24 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 12:42.00 | *** join/#asterisk Skarmeth (n=Skarmeth@201.9.82.245) |
| 12:42.50 | awk | is there a way I can see why my load average is so high with asterisk? |
| 12:42.59 | awk | its using 99% cpu |
| 12:44.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 12:47.18 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:49.49 | *** join/#asterisk Mw3_ (n=mw3@ip59934bd1.rubicom.hu) |
| 12:53.12 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 12:53.31 | *** join/#asterisk ming_zym (n=ming_zym@124.14.235.143) |
| 12:53.34 | cjk | hi, anyone an idea why ${CDR(accountcode|l)} is no longer working in the latest revision? |
| 12:56.50 | Assid | okay sip 2.2 has 2 ld files for polycom 501's |
| 12:57.19 | Assid | does that mena i use 2345-11500-030.sip.ld in the primary line ?and 2345-11500-040.sip.ld in <APPLICATION_SPIP500 APP_FILE_PATH_SPIP500 ? |
| 13:00.43 | mosty | ronr, no |
| 13:01.31 | mosty | McDouglas, sangoma a500 |
| 13:01.57 | mosty | awk: top / asterisk logs? |
| 13:10.15 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:11.12 | lirakis | hmm.. this is really bizzarre... all of a sudden an AGI executes really really slowly .. but only in some parts of the script. at first i thought maybe a DB connection issue... but its not... it seems to be hanging on set() variables. |
| 13:13.25 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:14.40 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 13:21.52 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 13:25.06 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:25.19 | Dr-Linux | anybody tried cisco 7935 with asterisk? |
| 13:28.24 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 13:31.39 | *** join/#asterisk JoZu (i=asdfasdf@84.120.188.30.dyn.user.ono.com) |
| 13:32.42 | JoZu | someone can giveme the url for the "Asterisk the future..." pdf, please? |
| 13:32.46 | Dr-Linux | Qwell: around? |
| 13:33.11 | Dr-Linux | JoZu: i'd also like to download 2nd edition |
| 13:33.44 | [TK]D-Fender | ~book |
| 13:33.45 | jbot | book is probably 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 |
| 13:33.46 | tzafrir | ~thebook |
| 13:33.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:33.56 | JoZu | thanks, tzafrir |
| 13:36.08 | Assid | [TK]D-Fender sip 2.2 has 2 ld files for polycom 501's |
| 13:36.15 | Assid | does that mean i use 2345-11500-030.sip.ld in the primary line ?and 2345-11500-040.sip.ld in <APPLICATION_SPIP500 APP_FILE_PATH_SPIP500 ? |
| 13:36.25 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 13:36.28 | ronr | how can I free a sip channel in asterisk (one phone is now registered with 2 sip channels, another phone should register with that server, but as it is already occupied, it can't) |
| 13:36.34 | [TK]D-Fender | Assid: Means "read the release notes & admin guide" |
| 13:36.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:38.01 | matmoj | mosty: i just compiled the zaptelsources from asterisk.org and i've gotten further now thnx for the help |
| 13:39.29 | tzafrir | matmoj, what version of Zaptel? |
| 13:39.42 | ice_croft | mosty> so, what should i do? i even can call to other sip client - but cant handle unrouted calls.. |
| 13:40.22 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:40.39 | mosty | ice_croft, what codec are you using? |
| 13:41.07 | Assid | hrmm.. confused |
| 13:42.15 | ice_croft | mosty> g723 |
| 13:42.27 | ice_croft | mosty> standard pack |
| 13:42.44 | ice_croft | oh |
| 13:43.12 | [TK]D-Fender | ice_croft: * doesn't support G.723 in any legal ways except the TC400 transcoder card. |
| 13:43.13 | ice_croft | wait. u sain that i can playback ttweasels message because of codecs? |
| 13:43.25 | [TK]D-Fender | ^^^^ |
| 13:43.49 | mosty | ice_croft, try with g711 |
| 13:43.50 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 13:43.51 | Assid | hey.. polycom opened up the site for direct downloads of the bootrom and sip |
| 13:43.58 | Assid | sweet |
| 13:44.01 | ice_croft | ulaw? |
| 13:44.03 | mosty | assid: only for the previous version |
| 13:44.14 | mosty | ice_croft, ulaw if you're in america, alaw most everywhere else |
| 13:44.19 | ice_croft | <PROTECTED> |
| 13:44.20 | Assid | oh yeah |
| 13:44.36 | ice_croft | mosty> so its some of this |
| 13:44.42 | ice_croft | mosty> *these |
| 13:44.42 | Assid | i still dont understand this multiple lines |
| 13:45.52 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 13:46.05 | Zuchmir | is there any SIP software that displays the SendText() |
| 13:46.15 | ice_croft | is there any way to debug dialplan? step by step? |
| 13:46.46 | [TK]D-Fender | ice_croft: watch it in console. "Set verbose 10" |
| 13:47.43 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 13:48.54 | ice_croft | [TK]D-Fender> |
| 13:50.01 | Assid | [TK]D-Fender: okay the admin guide shows the primary line as sip.ld .. and each type below as the ld file for that client |
| 13:50.15 | McDouglas | is it possible to use a fax machine connected to a sip ata? |
| 13:50.23 | ice_croft | damn |
| 13:50.36 | ice_croft | still cant catch unrouted calls |
| 13:50.52 | mosty | McDouglas, fax over voice over ip does not work well, forget that idea |
| 13:51.12 | Dr-Linux | tzafrir: is 2nd edition free to download? |
| 13:51.18 | McDouglas | mosty: oh no, i dont want to fax over ip, i just dont want to purchase analog cards for the fax machine |
| 13:51.58 | McDouglas | mosty: i have a bri card, and tought about using a sip ata to connect fax cals from the pstn to the fax machine |
| 13:52.00 | tzafrir | Dr-Linux, yes |
| 13:52.22 | mosty | McDouglas, even on a lan i wouldn't recommend it. you might be able to use rxfax or something |
| 13:52.26 | ice_croft | mosty> oh, i c now |
| 13:52.50 | Assid | [TK]D-Fender am i right? |
| 13:52.54 | [TK]D-Fender | ice_croft: What the hell is an "unrouted call"? |
| 13:53.02 | [TK]D-Fender | Assid: Keep reading... |
| 13:53.05 | Qwell | [TK]D-Fender: a call that isn't routed |
| 13:53.12 | ice_croft | mosty> is it possible to "Verbose(1|Echo test application)" app cannot be launched? |
| 13:53.22 | [TK]D-Fender | Dr-Linux: The links were sent out TWICE. Read the big print... |
| 13:53.48 | [TK]D-Fender | Qwell: How enlightening. |
| 13:54.03 | Qwell | [TK]D-Fender: any time |
| 13:54.12 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:54.18 | ice_croft | [TK]D-Fender> wrong numbers, etc |
| 13:54.21 | mosty | ice_croft, you should get an error in your logs/console in that case |
| 13:54.25 | [TK]D-Fender | McDouglas: that IS over IP. |
| 13:54.31 | Dr-Linux | [TK]D-Fender: okey thanks! |
| 13:54.49 | Assid | [TK]D-Fender: the link mentioned doesnt work :( |
| 13:54.55 | Dr-Linux | Qwell: any advice cisco 7935? |
| 13:56.43 | ice_croft | mosty> man, what a mess. it's cos of my misunderstandin of dialplan concept |
| 13:57.17 | mosty | ice_croft, you should read the book |
| 13:57.26 | ice_croft | mosty> correct my, after i hang up the phone, dialplan halts? |
| 13:57.56 | ice_croft | mosty> i wrote config exactly by the book |
| 13:57.57 | mosty | ~thebook |
| 13:57.57 | 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 |
| 13:58.22 | ice_croft | anyway, thanx a lot! |
| 13:58.25 | mosty | when you hangup, execution jumps to the h extension |
| 13:58.35 | ice_croft | i c |
| 13:58.37 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 13:58.51 | [TK]D-Fender | ice_croft: Never expect us to trust that you did it right. PASTEBIN <------ |
| 13:58.53 | [TK]D-Fender | ~pb |
| 13:58.54 | 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 |
| 13:59.48 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 13:59.51 | [TK]D-Fender | ice_croft: Taking 2 minutes to let us see it will save us hours of hearing "why doesn't it work?!?!". Like Jerry Mcguire said "SHOW ME THE MONEY!" |
| 13:59.57 | ice_croft | [TK]D-Fender> no need to, i pasted all the configs there already. seems like real misunderstandin of concept for me |
| 14:00.14 | [TK]D-Fender | ice_croft: Where? I don't see you linking it.... |
| 14:00.25 | ice_croft | [TK]D-Fender> u just weren't here, mosty saw my pastes |
| 14:00.37 | [TK]D-Fender | ice_croft: Feel free to share your latest.... |
| 14:01.00 | ice_croft | [TK]D-Fender> http://pastebin.ca/810904 |
| 14:01.20 | ice_croft | [TK]D-Fender> i can't get correct error message from this dialplan |
| 14:02.17 | ice_croft | [TK]D-Fender> "Call from '1000' to extension '400' rejected because extension not found". having this+hangup instead of weasels message |
| 14:02.50 | [TK]D-Fender | ice_croft: '_X' => <-- see this? It means a SINGLE DIGIT. |
| 14:02.57 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:03.04 | [TK]D-Fender | ice_croft: not "any number". it means only *1* digit |
| 14:03.19 | mosty | ice_croft, i said to change s to _X. not _X |
| 14:03.20 | ice_croft | [TK]D-Fender> well, i had "s" there, too |
| 14:03.31 | [TK]D-Fender | ice_croft: s is NOT going to work either |
| 14:04.02 | ice_croft | [TK]D-Fender> so what will? |
| 14:04.03 | ice_croft | mosty> i'm sorry? :)) |
| 14:04.03 | [TK]D-Fender | ice_croft: you are dialing a targeted exten. "s" isn't it. You ocmpletely misunderstand its use. make a dialplan pattern that MATCHES 1000 |
| 14:04.24 | [TK]D-Fender | ice_croft: "_XXXX" will match any 4-digit number. |
| 14:04.37 | ice_croft | [TK]D-Fender> i made it already, and it's work. but i need to handle wrong numbers, etc |
| 14:04.39 | [TK]D-Fender | ice_croft: "_x." will match any number 2 digits or longer |
| 14:06.11 | matmoj | when connecting to a pri, do i need any special wirering or is just a tw good? |
| 14:06.39 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 14:06.42 | ice_croft | [TK]D-Fender> so, what should i do with numbers that r not in any of my contexts>? |
| 14:06.45 | [TK]D-Fender | ice_croft: You clearly fail to understand the basics of getting something to MATCH, forget about dealing with invalid selections for a while. That is far more complicated, and significantly less important. |
| 14:06.56 | [TK]D-Fender | ice_croft: you IGNORE THEM. |
| 14:07.07 | ice_croft | [TK]D-Fender> well, ok. |
| 14:07.53 | [TK]D-Fender | matmoj: Most PRI's are terminated by an RJ48 smart-jack. For this you can use a basic cat5 straight-through cable to connect to most PRI hardware. |
| 14:08.11 | ronr | could anyone tell me what's wrong with this: GotoIf($[${UNIQCHANNEL}=${LOCATION}]?callingself:forward) (it always goes to callingself, even when the variables are not equal) |
| 14:09.06 | ice_croft | [TK]D-Fender> i just tried to make this: http://tfot.leifmadsen.com/ch04s03.html |
| 14:09.11 | [TK]D-Fender | ronr: apstebin a failed attempt including a NoOp of both vars before the call. |
| 14:09.14 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:09.18 | ice_croft | [TK]D-Fender> and totally failed :)) |
| 14:09.41 | [TK]D-Fender | ice_croft: that is ONLY for analog channels. |
| 14:09.50 | [TK]D-Fender | ice_croft: Go read what the "s" exten is for |
| 14:09.56 | [TK]D-Fender | ~stdextens |
| 14:09.57 | jbot | i heard stdextens is "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. The ... |
| 14:09.57 | ice_croft | [TK]D-Fender> ok, will do |
| 14:10.30 | mosty | ronr, i normall put " quotes around variables when comparing them |
| 14:10.33 | mosty | y |
| 14:10.48 | billybongo | ronr: what does it say in the console when it's being compared? |
| 14:11.11 | matmoj | [TK]D-Fender: thnx, i feel like atotal noob (wich i am to asterisk) |
| 14:11.35 | [TK]D-Fender | matmoj: np, easy common question for those just learning about digital PSTN access |
| 14:11.48 | *** join/#asterisk Kobaz (i=kobaz@its.kobaz.net) |
| 14:12.37 | billybongo | ronr: I tend to use the alternative syntax e.g. |
| 14:12.37 | billybongo | GotoIF,$[${UNIQCHANNEL}=${LOCATION}]?X,Y |
| 14:12.48 | billybongo | saves on a set of () |
| 14:13.22 | [TK]D-Fender | billybongo: Don't, and please provide the pastebin I requested of you. |
| 14:13.53 | billybongo | you did? |
| 14:14.18 | billybongo | what's wrong with that syntax? |
| 14:14.39 | ronr | [TK]D-Fender: I can't paste (as the console is on a computer without a mouse or anything), but the NoOp's really show the vars are different, the gotoif shows only one of the 2 vars |
| 14:15.24 | [TK]D-Fender | [09:09]<[TK]D-Fender>ronr: apstebin a failed attempt including a NoOp of both vars before the call. |
| 14:15.27 | mosty | ronr: put spaces around = and quotes around the variables |
| 14:15.37 | ronr | mosty: I'll try |
| 14:15.53 | mosty | ronr, and did you mispell UNIQUECHANNEL? |
| 14:15.55 | billybongo | [TK]D-Fender: hey but I'm not ronr |
| 14:16.17 | [TK]D-Fender | billybongo: Bad aim, sorry |
| 14:16.22 | billybongo | np |
| 14:16.32 | ronr | mosty: nope, I didn't |
| 14:17.18 | billybongo | ronr: can't you ssh into the box and paste from there? |
| 14:17.48 | ronr | billybongo: if I could somehow get the console (F9) output in the ssh shell |
| 14:18.56 | tzafrir | ronr, cat /dev/vcs9 |
| 14:19.00 | [TK]D-Fender | ronr: connect to it from a station via SSH and grab it from there |
| 14:19.33 | ronr | adding " and spaces (like mosty said) did it :) |
| 14:20.11 | [TK]D-Fender | ronr: Good to hear |
| 14:21.28 | *** join/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 14:22.41 | TUplink | hi guys and gals.... i have a prob...... app_voicemail chan_zap arnt loading and neither is some smdi thing i belive the smdi is required by zap and voicemail any ideas? |
| 14:23.09 | mocker | Are you getting any errors? |
| 14:23.12 | TUplink | the error is No SMDI interfaces are available to listen on, not starting SDMI listener. |
| 14:23.34 | TUplink | let me pastebin the whole thing |
| 14:23.51 | mocker | Is this FreeBSD? |
| 14:24.06 | TUplink | yea |
| 14:24.09 | TUplink | http://pastebin.ca/810980 |
| 14:24.18 | mocker | http://www.voip-info.org/wiki/view/Asterisk+FreeBSD |
| 14:24.19 | TUplink | dont tell me its broke in freebsd |
| 14:24.26 | mocker | Someone else had the exact error you did. |
| 14:25.05 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 14:25.52 | TUplink | what is SMDI ? |
| 14:26.19 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:26.25 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 14:26.27 | cjk | hi, is it normal that asterisk-addons does not compile wiht the latest asterisk revision from 1.4/trunk ? |
| 14:26.50 | TUplink | i even got the core dump 11 b4 too :P |
| 14:26.57 | TUplink | wierd thind is that it worked |
| 14:27.00 | TUplink | use to |
| 14:27.07 | [TK]D-Fender | cjk: if the version doesn't match, yes. |
| 14:27.08 | mocker | TUplink: Did you check the comment above? |
| 14:27.22 | mocker | I think he fixed it w/ just a symlink because asterisk wasn't scanning the correct directory. |
| 14:27.35 | cjk | [TK]D-Fender, well how can i check if they match or not i take the latest revision from both |
| 14:28.39 | [TK]D-Fender | cjk: If you took the latest of each they should match. If you're running trunk naturally there is a much higher likelyhood of build errors, etc. thats what you get for trying run bleeding edge releases |
| 14:29.16 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 14:29.32 | mosty | cjk, if asterisk-addons is older than asterisk, it might break |
| 14:29.44 | cjk | ok its older for sure |
| 14:30.17 | TUplink | mocker im reading everything |
| 14:31.17 | [TK]D-Fender | mosty: Um.... its technically always older :) |
| 14:31.47 | [TK]D-Fender | mosty: target would be "the newset version not newer than *" :) |
| 14:32.43 | mosty | [TK]D-Fender, if you're using asterisk-addons 1.4.1 with asterisk 1.4.15 then i would say that the asterisk-addons version is older |
| 14:33.21 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:33.25 | [TK]D-Fender | mosty: I agree. |
| 14:34.02 | mosty | in any case, just read the asterisk-addons changelog, use the most recent version that it says should work with your version of asterisk |
| 14:35.30 | TUplink | ok... i think i got the error to clear..... do you think i need to worie about this one..... it pops up about every 2 min |
| 14:35.31 | TUplink | chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '6c3b7d8443be8bb50dbcf2341aeddacf@75.67.237.149'. Giving up. |
| 14:35.42 | *** join/#asterisk techie (n=techie@adsl-76-214-12-127.dsl.lsan03.sbcglobal.net) |
| 14:35.48 | TUplink | w/ a difrent 1st part |
| 14:36.25 | *** join/#asterisk ming_zym (n=ming_zym@124.14.235.143) |
| 14:37.08 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:38.33 | mocker | TUplink: What was the solution to the error? |
| 14:38.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:41.49 | TUplink | um..... :P |
| 14:42.04 | TUplink | let me get back into the box after it reboots |
| 14:42.10 | TUplink | ill post it on voipinfo |
| 14:42.12 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:50.14 | steliosk | anyone has an idea why when starting asterisk as non root it does not create the asterisk.ctl file unless -c is passed as parameter ? |
| 14:51.33 | TUplink | mocker i posed my solution on there too :P |
| 14:52.14 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 14:53.57 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 14:56.42 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:57.49 | ronr | how can I set the CallerID based on the channel from where the call originates? (eg. I want SIP/1 to get callerID 123456890 and SIP/2 1234543211, etc) |
| 14:58.15 | [TK]D-Fender | ronr: "callerid=" in sip.conf entries. |
| 14:58.16 | tzafrir | Set(), CUT, etc. |
| 14:58.30 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:00.11 | ronr | [TK]D-Fender: I can't use that, as the callerId's will be dynamic, but based on the channel I can find the required calledId in the astdb |
| 15:02.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:02.09 | [TK]D-Fender | ronr: Set(CALLERID(num)=${DB(CIDbyDEVICE/${CALLERID(num)})}) |
| 15:02.18 | [TK]D-Fender | ronr: there's a thought for you |
| 15:02.46 | [TK]D-Fender | ronr: Or use SetVar in your sip peer to set the DB key to lookup by |
| 15:02.51 | [TK]D-Fender | (better idea) |
| 15:03.12 | R1ck | does asterisk support ipv6? |
| 15:03.47 | Qwell | R1ck: no, but there is work being done |
| 15:04.20 | ronr | [TK]D-Fender: thx, I guess SetVar would be better when everything will use SIP (I think at first, it will, but I don't want to restrict myself just yet) |
| 15:04.45 | *** part/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 15:04.54 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:05.19 | R1ck | Qwell: any idea when it will be completed? :) |
| 15:05.24 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 15:06.24 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 15:07.57 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:08.11 | *** join/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 15:08.21 | TUplink | with SetAMAFlags( |
| 15:08.31 | TUplink | with SetAMAFlags() can i set it to anything? |
| 15:08.45 | TUplink | or does it have to be a number |
| 15:12.26 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 15:15.55 | *** join/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
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| 15:17.42 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
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| 15:26.06 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:26.40 | dominic1 | Hi, I am using odb for my asterisk configuration. How is it possible to customize a databaseselect in the dialplan? |
| 15:26.54 | *** part/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 15:29.25 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:29.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:36.02 | *** join/#asterisk nirz (n=nir@bzq-79-177-144-189.red.bezeqint.net) |
| 15:37.30 | Dr-Linux | guys, i wanna use another SIP port as well as 5060. Basically some of my clients provider do not allow sip default port |
| 15:37.34 | Dr-Linux | any suggestion? |
| 15:37.54 | dominic1 | okay func_odbc is my guardian |
| 15:38.33 | file | Dr-Linux: currently no way built in... but some people use iptables |
| 15:39.09 | Dr-Linux | file: file table doesn't work, i already tried |
| 15:39.39 | Dr-Linux | file: iptables job only works for 20 sec call |
| 15:39.54 | Dr-Linux | after that one can't hear |
| 15:40.02 | file | I know people who have it, but I know not how |
| 15:40.06 | file | have it working rather |
| 15:40.48 | [TK]D-Fender | Dr-Linux: Setup another server to register to your provider and pass the calls off internally. |
| 15:41.35 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 15:41.42 | Dr-Linux | hhm... |
| 15:41.49 | Dr-Linux | file: this is the way i tried: |
| 15:41.52 | Dr-Linux | /sbin/iptables -t nat -A PREROUTING -p udp --dport 8989 -j REDIRECT --to-port 5060 |
| 15:41.57 | *** part/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 15:44.04 | Dr-Linux | [TK]D-Fender: your ID make sense but can you make it more clear please |
| 15:44.21 | dominic1 | my isdn provider is signalising a number+DDI, what can I do to use only DDI in my extension |
| 15:44.21 | Dr-Linux | how my both boxes will communicate with each other on different ports :S |
| 15:44.22 | dominic1 | ?? |
| 15:45.14 | [TK]D-Fender | Dr-Linux: Whats to explain? Set up another server binding to the different port, and use that as an intermediary to your existing server |
| 15:45.20 | De_Mon | dominic1 ${EXTEN:-4} would give you the last 4 digits of the extension |
| 15:45.45 | De_Mon | dominic1 you can also use ${CUT()} if there is some sort of separator and the DDI is variable length |
| 15:46.04 | dominic1 | can I set it as new extension? |
| 15:46.04 | FlatFoot | afternoon all |
| 15:46.20 | De_Mon | dominic1 yea s,1,Goto(${EXTEN:-4}) |
| 15:46.27 | Dr-Linux | [TK]D-Fender: hhm.. my user is trying to register from Karachi, i mean from different location, actually his provider blocked SIP port |
| 15:46.50 | dominic1 | will that be used for cdr too? |
| 15:46.53 | De_Mon | dominic1 you need to read The Book if you don't know how to create this basic dialplan |
| 15:46.56 | De_Mon | ~book |
| 15:46.57 | 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 |
| 15:46.57 | [TK]D-Fender | Dr-Linux: Yes, I got that already, I told you what to try. Now get to it.... |
| 15:47.25 | [TK]D-Fender | De_Mon>dominic1 yea s,1,Goto(${EXTEN:-4}) <--- I think YOU need to read the boko a bit yourself :p |
| 15:47.35 | [TK]D-Fender | book* |
| 15:47.47 | Qwell | -4 is valid |
| 15:47.48 | Qwell | :P |
| 15:47.51 | De_Mon | I know I know I left off the ,1 |
| 15:47.55 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:47.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:47.59 | [TK]D-Fender | Qwell: For the VARIABLE yes... look at the GOTO :p |
| 15:48.02 | Qwell | oh, you mean the broken syntax |
| 15:48.04 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:48.14 | De_Mon | I sent him to priority foo instead of extension foo priority 1 |
| 15:48.15 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 15:48.23 | Dr-Linux | [TK]D-Fender: cool, i understand. i've ready other box but it would need a public IP address |
| 15:48.58 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:49.35 | [TK]D-Fender | Dr-Linux: No, all you'd need to do is set a different range for rtp.conf and forward accordingly. |
| 15:50.56 | [TK]D-Fender | Dr-Linux: each * system will run its own port ranges |
| 15:51.44 | Dr-Linux | [TK]D-Fender: currently we established VPN between both location, but problem is that, he can registered but he can't hear me |
| 15:52.32 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:52.49 | [TK]D-Fender | Dr-Linux: Common NAT/localnet issues.... |
| 15:52.58 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 15:53.21 | Dr-Linux | [TK]D-Fender: both tried, but same issue mmmm |
| 15:54.04 | muiro | hey, any hints on getting the lumenvox speech engine to parse DTMF tones? What seems to be happening is that if DTMF is used, some digits get interpreted multiple times: 15002 becomes 111150000222222 |
| 15:55.54 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 15:56.08 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 15:56.54 | muiro | I know that asterisk doesn't send any dtmf data to the speech engine once it "hears" it, but how can I keep this from happening? Is this the effect of echo? I'm getting this input from a sip trunk, is there any way to do a software echo cancelation on a sip channel? |
| 16:00.04 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-0ea9c317cd831f6e) |
| 16:04.08 | *** join/#asterisk DaveCanoe (n=Dave@66.96.31.47) |
| 16:04.10 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
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| 16:05.34 | *** mode/#asterisk [+o twisted] by ChanServ |
| 16:05.51 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-207-130.dsl.irvnca.pacbell.net) |
| 16:06.01 | UnixDog | hey guys |
| 16:06.28 | *** join/#asterisk destructure (n=kwatz@66.193.229.254) |
| 16:06.30 | UnixDog | has anyone here done dial plan for selective call waiding/forwarding/dnd |
| 16:06.56 | [TK]D-Fender | UnixDog: Yes. |
| 16:07.28 | UnixDog | waiding/waiting |
| 16:09.58 | UnixDog | is there a place to look at one of them say selective call waiitng |
| 16:10.39 | UnixDog | basicly if I disable call waiting but still want a certian client to be able to get threw . |
| 16:11.11 | [TK]D-Fender | UnixDog: This is all dialplan. Think up how big a structure you want fdor this and most of us would use AstDB for this purpose. |
| 16:11.14 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:11.40 | [TK]D-Fender | UnixDog: make a family/key pairing for the various things you want |
| 16:12.44 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 16:12.46 | *** join/#asterisk Corydon76-lap (i=Corydon7@pdpc/supporter/bronze/Corydon76-home) |
| 16:12.46 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
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| 16:12.59 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 16:13.04 | Qwell | UnixDog: you need a fully functional dialplan in order to do that |
| 16:13.37 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 16:14.24 | UnixDog | well thats what I am doing writing dialplan but I am not fully grasping the layout |
| 16:14.31 | UnixDog | hmm |
| 16:14.36 | muiro | anyone have any advice for handling dtmf tones after asterisk receives them during speechbackground()? |
| 16:14.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:17.17 | [TK]D-Fender | UnixDog: Before any extens that dial your devices that you want to apply this logic to, go check for values indicating what you need to consider before following through and ringing the device |
| 16:17.57 | [TK]D-Fender | UnixDog: Like first check if you want to permit the caller through regardless. Then check if DND is on, etc. |
| 16:17.58 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:18.19 | [TK]D-Fender | UnixDog: "show function DB", "show application gotoif" |
| 16:18.38 | [TK]D-Fender | UnixDog: "show function CALLERID" |
| 16:19.07 | nny | anyone have experience with Aastra enough to happily say they can become "remote extensions" over NAT with the saem relative pain in the assness as say Snom or Polycoms? |
| 16:19.11 | *** join/#asterisk ManxPower (n=manxpowe@182.sub-70-221-78.myvzw.com) |
| 16:19.48 | nny | We haver a client that uses a Snom 320 from home with no local gateway and wants to be able to upgrade to something "cordless", hence the Aastra |
| 16:19.58 | Kobaz | how do i set the source address for an iax peer |
| 16:20.08 | UnixDog | ok thats what I missed |
| 16:20.16 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 16:20.18 | UnixDog | the callerid and the db checking |
| 16:20.19 | nny | Just fishing for any horror stories, tales of woe, or general malarky |
| 16:20.23 | ManxPower | nny: Your best bet is an ATA + Cordless Phone |
| 16:20.28 | UnixDog | in the inbund dial pattern |
| 16:21.03 | ManxPower | nny: You should read the mailing list archives then. For the most part all WiFi SIP phones suck badly. |
| 16:21.07 | Seldon75 | hello, our handsets (Polycom301) are configured with DHCP, can someone please tell me how the handset 'knows' what extension it is using? |
| 16:21.08 | nny | ManxPower, hrmm.. ATA eh? That would actually be beneficial in that one ATA could server an entire series of traditional wiring... (In theory?) |
| 16:21.12 | Kobaz | so noone knows? |
| 16:21.15 | nny | ManxPower, nahh aastra is not WiFi |
| 16:21.21 | Seldon75 | is the configuration in the handset or in Asterisk? |
| 16:21.24 | nny | ManxPower, and yeah i hear they all blow various farm animals |
| 16:21.48 | Kobaz | Seldon75: either |
| 16:21.57 | [TK]D-Fender | Seldon75: an extension is a number you can dial in extensions.conf and no phone can know anything about that. |
| 16:21.58 | nny | ManxPower, Aastra 280i CT uses non 802.11 communication from cordless to handset, 900 MHZ mayhaps |
| 16:22.03 | ManxPower | nny: The advantage of an ATA is the fact it works 8-) SIP basestations with DECT wireless is a fairly new thing, but I've not heard many things bad about that setup. |
| 16:22.13 | Kobaz | Seldon75: well asterisk needs to be configd in extensions, but then you need a tftp server for the phone to get it's config |
| 16:22.17 | nny | ManxPower, you mean kirks? |
| 16:22.25 | Kobaz | and, does anyone know how to set the source address of an iax peer? |
| 16:22.28 | ManxPower | nny: Never heard of kirks |
| 16:22.32 | nny | ManxPower, I have two aastra 480i CTs in the wild, and the wireless works great |
| 16:22.33 | UnixDog | ok bbiab |
| 16:22.34 | [TK]D-Fender | Kobaz: "host=" |
| 16:22.39 | UnixDog | off to fix things |
| 16:22.51 | nny | ManxPower, mind you the base station is the isp client, and the handse is just an extension of the base station |
| 16:22.55 | nny | sip* |
| 16:23.11 | ManxPower | *nod* |
| 16:23.12 | Kobaz | [TK]D-Fender: but that's the option for the host to connect to... isnt it? |
| 16:23.24 | nny | ManxPower, polycom kirks.. base is sip client, supports 4 or more wireless phones.. been wanting to get some to test |
| 16:23.25 | ManxPower | Kobaz: permit/deny in iax.conf |
| 16:23.31 | Seldon75 | [TK]D-Fender: ok so where does the Handset find out it should register itself as 'extension 207' |
| 16:23.37 | nny | but i digress |
| 16:23.39 | Kobaz | not permissions, the address it binds the source to |
| 16:23.47 | nny | so yeah..thanks for the advice, will look at an ATA |
| 16:23.49 | Seldon75 | i know Im using the wrong terminology |
| 16:23.51 | ManxPower | Kobaz: that depends on the IAX client. |
| 16:23.56 | [TK]D-Fender | Seldon75: Stop calling DEVICES as EXTENSIONS. |
| 16:24.00 | Kobaz | ManxPower: this is on asterisk itself |
| 16:24.13 | Kobaz | ManxPower: i just want to change the source address |
| 16:24.22 | ManxPower | Kobaz: perhaps you missed the bindaddr= option in iax.conf |
| 16:24.32 | Kobaz | that could be it |
| 16:24.33 | [TK]D-Fender | Seldon75: And how you set up your phone will determine that. Depending on whether you use the phone itself to configure, the web interface, or provisioning. |
| 16:24.55 | ManxPower | Your best bet is to NOT bind to a specific IP address. The source address will then be determined by the OS and routing tables. |
| 16:25.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:25.12 | Kobaz | ManxPower: yeah but for some reason it's not picking the right source address |
| 16:25.22 | Kobaz | ManxPower: every single other piece of software does it right, but asterisk doesn;t |
| 16:25.23 | ManxPower | Kobaz: then you have a network problem |
| 16:25.30 | Kobaz | yeah i know, and i need a quick fix |
| 16:25.43 | ManxPower | Kobaz: there is never a quick fix for your problem. |
| 16:25.48 | Kobaz | yeah there is |
| 16:26.30 | Seldon75 | [TK]D-Fender: whats the correct terminology to use? doesnt a device (handset) register itself to use an extension? |
| 16:26.36 | Kobaz | so it's just bindaddr=ip |
| 16:26.39 | Kobaz | hmm that's not working either |
| 16:26.46 | ManxPower | Seldon75: no, the device does not register itself to use an extension |
| 16:27.07 | [TK]D-Fender | Seldon75: a SIP account is not an EXTENSION. get that through your head. Go download the admin guide for yuor phone to learn how to configure it. |
| 16:27.14 | ManxPower | Seldon75: the device registers itself to a SIP account. extensions.conf is what ties it all togather. |
| 16:28.19 | ManxPower | Seldon75: the correct term is "phone", "device", "peer", "ATA", etc pretty much anything except "extension" |
| 16:28.45 | Seldon75 | ok |
| 16:28.47 | ManxPower | Seldon75: For one thing stop using numbers that look like extensions as your SIP account names. |
| 16:29.22 | [TK]D-Fender | Seldon75: Go download the admin guide & firmware from your reseller, and check the WIKI handbook on provisioning your phone. |
| 16:29.24 | ManxPower | We use the MAC address as the SIP userID |
| 16:29.24 | [TK]D-Fender | ~wikis |
| 16:29.25 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:29.39 | Seldon75 | ok thx |
| 16:30.09 | Seldon75 | just out of interest; is there a concise definition for the term 'extension'? |
| 16:30.28 | Seldon75 | just so I can understand and not get it wrong again |
| 16:30.32 | ManxPower | Seldon75: An extension is a number you dial. |
| 16:30.44 | ManxPower | Really, an extension is JUST a short phone number. |
| 16:31.26 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 16:31.39 | Seldon75 | hmm ok. so I'm not really sure whats wrong with saying a phone registers to use a 'short phone number' |
| 16:31.46 | ManxPower | That phone number is listed in extensions.conf, which specifies what do do when that number is dialed. |
| 16:32.02 | ManxPower | Seldon75: because the phone does NOT register to use a anything. |
| 16:32.29 | ManxPower | The ONLY thing registration does is inform the server what IP address is associated with which SIP user/password. That is why a phone can still make a call even if it is not registered to a server. |
| 16:32.45 | Seldon75 | i see |
| 16:33.09 | ManxPower | It just can't recieve a call from the server, since the device is on a dynamic IP address. If the device is not a dynamic IP address, there is actually no need to register. |
| 16:33.29 | Seldon75 | yeah we're using DHCP |
| 16:33.43 | ManxPower | Seldon75: There are 2 concepts that people have to understand before being able to understand Asterisk. 1) a device is not an extension and 2) contexts. |
| 16:34.24 | [TK]D-Fender | Seldon75: Your DIALPLAN contains EXTENSIONS. An Extens is anumber you can DIAL. This has NOTHING to do with what * will DO when you dial it. |
| 16:35.03 | Seldon75 | right |
| 16:35.15 | Seldon75 | this is an adjustment for ppl coming from old-school PABXs |
| 16:35.22 | dklima | is there a way in PRI to pass courtesy messages from PSTN to SIP instead BUSY tone? |
| 16:39.10 | [TK]D-Fender | dklima: What is generating the busy tone? |
| 16:39.28 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.201) |
| 16:39.30 | bartpbx | hello |
| 16:39.39 | bartpbx | anyone using Sangoma A500 cards here? |
| 16:40.16 | dklima | I'm using DIALSTATUS... I've tried using HANGUCAUSE, but no success |
| 16:40.39 | ice_croft | amm |
| 16:40.43 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-34-163.red.bezeqint.net) |
| 16:40.44 | ice_croft | people |
| 16:40.45 | Dovid | hi |
| 16:40.50 | Dovid | if using a sangoma card |
| 16:40.56 | bartpbx | th a500 |
| 16:41.04 | bartpbx | we have some issues with overlap dialing |
| 16:41.12 | Dovid | the carrier (PRI provider) is asking if I want to use natonial1 or natonial2. for the US which one is it? |
| 16:41.16 | ice_croft | how can i connect gsm phone to *, avoidin cellphone operator? |
| 16:41.24 | dklima | it should be interesting to pass that courtesy message directly.. so if a number do not exist the user will be informed about that |
| 16:42.15 | ice_croft | how can i connect gsm phone to *, avoidin cellphone operator? |
| 16:42.51 | cjk | hi where are fax and phones connected to? to an fxo or fxs port? |
| 16:43.11 | russellb | ~fxofxs |
| 16:43.12 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 16:43.32 | Qwell | ice_croft: get a sip softphone for your phone |
| 16:44.42 | cjk | russellb, thanks |
| 16:44.44 | billybongo | cjk: the way I remember is fxs talks to a *s*tation |
| 16:44.59 | Qwell | the s *does* stand for station |
| 16:45.02 | billybongo | ahh cool |
| 16:45.07 | ice_croft | Qwell> hm. and edge? |
| 16:45.08 | *** join/#asterisk guillote_GNU (n=guillote@host32.200-117-222.telecom.net.ar) |
| 16:45.16 | billybongo | Qwell: does the O stand for office? |
| 16:45.19 | Qwell | it does |
| 16:45.21 | Qwell | ~fxo |
| 16:45.22 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 16:45.23 | Qwell | ~fxs |
| 16:45.24 | jbot | [fxs] foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 16:45.27 | ice_croft | Qwell> interestin.. |
| 16:45.29 | Qwell | system...no |
| 16:45.29 | billybongo | marellous |
| 16:45.31 | Qwell | that's wrong |
| 16:45.42 | billybongo | yeah, jbot is wrong |
| 16:45.44 | dklima | is there a way to do that? to have that courtesy message passing directly instead a busy tone |
| 16:45.51 | ManxPower | ~fxsfxo |
| 16:45.52 | jbot | methinks fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 16:45.59 | Qwell | jbot: no, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 16:46.00 | jbot | Qwell: okay |
| 16:46.01 | Dovid | what goes better with asterisk? |
| 16:46.04 | Qwell | ~fxs |
| 16:46.05 | jbot | rumour has it, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 16:46.05 | Dovid | National 2 or National 1 ? |
| 16:46.07 | UnixDog | ok a few more questions. |
| 16:46.25 | ManxPower | dklima: it depend son what you mean by "courtesy message". a message from Asterisk or a message from the telco? |
| 16:46.37 | ManxPower | Dovid: nobody uses national 1 |
| 16:46.37 | UnixDog | how many numbers should be allowed to be set in the selective callwaiting/dnd |
| 16:46.53 | dklima | ManxPower, from telco |
| 16:46.54 | UnixDog | should it be limited to like say 5 |
| 16:46.56 | Qwell | UnixDog: as many as needed? |
| 16:47.00 | Dovid | Manx: OK. a clients carrier in the US was offering 1 or 2 |
| 16:47.02 | ManxPower | dklima: what version of Asterisk? |
| 16:47.10 | dklima | ManxPower: 1.4.15 |
| 16:47.32 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 16:47.41 | dklima | ManxPower, ie: when a number do not exist |
| 16:48.18 | ManxPower | dklima: "show application hangup" |
| 16:48.31 | ManxPower | the cause code is the standard ISDN (Q.931) cause codes. |
| 16:49.22 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 16:49.27 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:49.52 | *** join/#asterisk pepo-- (n=pepOSX@190.79.246.105) |
| 16:50.33 | dklima | ManxPower, and how I determine that code? |
| 16:50.59 | dklima | ManxPower, using ${HUNGUPCAUSE} for example, always return 0... |
| 16:54.16 | ManxPower | dklima: http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 16:54.25 | ManxPower | that is a list of the standard ISDN cause codes |
| 16:54.32 | ManxPower | you don't screw with hangupcause. |
| 16:54.56 | ManxPower | you do a Hangup(whatever) in your dialplan, where "whatever" is the decimal number of the Q.931 cause codes. |
| 16:55.19 | ManxPower | dklima: in the PAST HANGUPCAUSE was used to send the cause code. That has not been the case since 1.4 was released. |
| 16:55.38 | dklima | ManxPower, hummm good to know |
| 16:55.51 | ManxPower | Now the only thing HANGUPCAUSE is used for is to get the cause code that was sent by the telco |
| 16:56.11 | ManxPower | and it's HANGUPCAUSE, not HUNGUPCAUSE |
| 16:56.55 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 16:57.06 | dklima | ManxPower, hehe sorry my typo |
| 16:57.10 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.201) |
| 16:57.19 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 16:57.36 | ManxPower | dklima: You would use HANGUPCAUSE or DIALSTATUS after a Dial line in extensions.conf. It has nothing to do with calls that come in from the telco that you want to reject. |
| 16:57.39 | magic_hat | anyone have suggestions on a good softphone for use with Linux? Xlite's giving me all kinds of configuration problems on 'nix. |
| 16:57.57 | ManxPower | magic_hat: all softphones suck. |
| 17:00.24 | magic_hat | lol... xlite works okay for us on windows/Os x. |
| 17:00.37 | ronr | does asterisk have some other datastructures for variables (for use in extensions.conf [globals]) as just the MYVAR=myvalue? I'm looking for something like arrays and hashes (an object with attributes and methods would be perfect, but I don't expect that to be possible) |
| 17:01.24 | tzafrir | magic_hat, twinkle? |
| 17:02.18 | tzafrir | amazing how far people go with the star metaphore |
| 17:02.25 | *** join/#asterisk Sentinal1 (n=Sentinal@87-194-204-58.bethere.co.uk) |
| 17:02.26 | FlatFoot | can anyone help with cdr_adaptive_odbc , trying install it on freebsd. Can't seem to find too much in the way of instruction. can anyone help ? |
| 17:02.33 | Sentinal1 | hi! |
| 17:03.36 | Sentinal1 | does anyone know if it would be possible for asterisk to fingerprint an anouncement? |
| 17:03.44 | Qwell | Sentinal1: fingerprint? |
| 17:03.48 | Sentinal1 | if not i'd be very interested in developing such a feature |
| 17:04.09 | Sentinal1 | i mean, sample an anouncement thats played by the telco and be able to identify it |
| 17:04.37 | Sentinal1 | for example 'the number is busy'.. could it sample 'the number is busy' and then say return a cause code 17 |
| 17:05.45 | Sentinal1 | so we would 'fingerprint' each possible anouncement and store that, then fingerprint every anouncement thats heard in normal use |
| 17:05.50 | Sentinal1 | and compare against our list to know what to do |
| 17:07.22 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 17:07.23 | Sentinal1 | i guess it cant do that right now? is there any functionality to sample the media at the moment? i could develop that |
| 17:07.37 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:08.31 | *** join/#asterisk mihinomenest (i=hNUp@66.255.220.17) |
| 17:09.27 | *** join/#asterisk jrobison (n=jrobison@ip67-152-34-15.z34-152-67.customer.algx.net) |
| 17:09.40 | Sentinal1 | hmm what about this voicemail -> email, hows that work? |
| 17:10.03 | Corydon76-lap | Which part? |
| 17:10.04 | jrobison | hello everyone |
| 17:10.17 | Corydon76-lap | Emailing voicemail works fine |
| 17:10.20 | ManxPower | ronr: Asterisk really only has 2 data types in the dialplan. global variables and channel variables. |
| 17:10.26 | Sentinal1 | can at any point asterisk sample the voice? |
| 17:10.35 | ManxPower | You can fate a simple array, but not hashes or anything like that. |
| 17:10.43 | ManxPower | Sentinal1: you mean like "show application monitor" |
| 17:11.02 | Corydon76-lap | ManxPower: I think he means one-touch recording or ChanSpy |
| 17:11.06 | ManxPower | you need to do a "show applications" to get a list of all the secret asterisk applications. BUT DON'T TELL ANYONE |
| 17:11.21 | Sentinal1 | no i mean.. i want it to hear ' the number is busy' on a phone line and understand what that means |
| 17:11.22 | ManxPower | Corydon76-lap: then he was too lazy to be more specific. |
| 17:11.25 | jrobison | I was wondering if anyone has had any experience with Viatalk service in a small office? |
| 17:11.41 | ManxPower | Sentinal1: stop. go back. Ask your question again, this time be more specific. |
| 17:11.48 | Sentinal1 | ok... |
| 17:11.56 | Sentinal1 | i thought i did.. |
| 17:12.18 | Sentinal1 | imagine we have an asterisk connected to the phone line in my house |
| 17:12.26 | Sentinal1 | so i can make outgoing calls through my landline home service |
| 17:12.50 | Sentinal1 | now imagine i dial my friend.. press 9 to get an outside line, then dial him +1 555 .... |
| 17:12.58 | *** join/#asterisk techie (n=techie@adsl-76-214-12-127.dsl.lsan03.sbcglobal.net) |
| 17:12.59 | Sentinal1 | and i hear an anouncement from the telco.. this number is busy |
| 17:13.01 | ManxPower | Sentinal1: what kind of phone line? standard analog residential/business line |
| 17:13.06 | ManxPower | ? |
| 17:13.10 | Sentinal1 | standard analog residential |
| 17:13.13 | Sentinal1 | that we all have in our homes |
| 17:13.27 | mocker | Sentinal1: Wrong channel to assume that in! :) |
| 17:13.31 | Sentinal1 | lol |
| 17:13.35 | ManxPower | Sentinal1: on analog FXO lines (what you have) the call is considered answered as soon as dialing is finished. |
| 17:13.42 | Sentinal1 | i understand that |
| 17:13.57 | Sentinal1 | what i would like to develop as a feature, if it doesnt exist |
| 17:14.13 | Sentinal1 | is for it to start sampling the voice, and detect ringing, anouncement, progress tones |
| 17:14.27 | kand | Does digium have paid support for the standard version of asterisk? |
| 17:14.30 | Sentinal1 | something like VAD, but more specific.. not just detect any voice, but specific voice |
| 17:15.16 | ManxPower | Sentinal1: PBX makers have tried for 20 years to do what you want to do. They all failed. But I do wish you the best of luck. Take a look at the randomlydisconnectmycalls=yes option....er.....busydetect=, busycount, and callprogress options. |
| 17:15.33 | ManxPower | kand: I think so, but you would have to call them. |
| 17:15.50 | Sentinal1 | i understand that.. the reason it fails is becuase there are many telco's and many noisy lines |
| 17:16.10 | Sentinal1 | i would be working with one line.. each user would fingerprint his own anouncements for his telco |
| 17:16.59 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:17.00 | dklima | ManxPower, HANGUPCAUSE always return 0 for me |
| 17:17.01 | ManxPower | Sentinal1: we don't really deal with coding stuff here, mostly just user stuff. |
| 17:17.07 | Sentinal1 | oh |
| 17:17.20 | Sentinal1 | is there a channel for coding? |
| 17:17.26 | ManxPower | dklima: You are doing something like Noop(HANGUPCAUSE is ${HANGUPCAUSE}) ? |
| 17:17.32 | ManxPower | in your dialplan, after the dial. |
| 17:17.36 | dklima | ManxPower, yes |
| 17:17.42 | ManxPower | Sentinal1: next door. it's labled #asterisk-dev" |
| 17:17.53 | Sentinal1 | lol *hits himself* |
| 17:17.57 | ManxPower | dklima: then you have a signaling issue with your PRI. |
| 17:18.18 | dklima | ManxPower, it what I was afraid to hear |
| 17:18.23 | Sentinal1 | its empty |
| 17:18.34 | ronr | ManxPower: ok, I'll move to AGI then |
| 17:18.41 | ManxPower | Sentinal1: there are 67 people there. |
| 17:18.49 | Sentinal1 | oh lol |
| 17:19.03 | Sentinal1 | it took the " with it |
| 17:20.08 | jrobison | Does anyone know what the simplest solution to getting 3-4 lines into an asterisk box would be? I am trying to switch from Viatalk(which seems to cut us off after an hour) to using analog lines. My manager doesnt trust Viatalk anymore |
| 17:20.23 | Qwell | jrobison: TDM400p |
| 17:20.24 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:20.36 | Qwell | or 800/2400 if you plan on using more than 4 in the future |
| 17:20.45 | ManxPower | jrobison: nobody in their right mind would trust calls sent over the internet |
| 17:21.09 | jrobison | MaxPower, yeah I have realized that. lol |
| 17:21.25 | jrobison | I am asusming I would want FXO ports on the 400P? |
| 17:21.31 | Qwell | for lines? yes |
| 17:21.45 | ManxPower | ~fxofxs |
| 17:21.45 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 17:21.58 | jrobison | ok, sorry. I dont really know very much about this sort of thing. I am a systems engineer, and I dont really know much about Phone systems |
| 17:22.09 | Qwell | jrobison: welcome to #asterisk ;) |
| 17:22.25 | jrobison | I appreciate the help |
| 17:22.26 | Qwell | I'd say well over half of the people here are network/systems guys, rather than phone guys |
| 17:22.53 | Qwell | (or, rather - they were, before they got into asterisk..) |
| 17:23.08 | jrobison | I have to say, now that I have started learning about phone systems, it is intriguing |
| 17:23.52 | jrobison | So, more on the TDM card, how would I configure this? I am using AsteriskNow |
| 17:24.09 | ManxPower | jrobison: you ask on the asterisknow channel |
| 17:24.35 | ManxPower | ~zeeek |
| 17:24.36 | jbot | zeeek is probably someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 17:24.40 | jrobison | We had a Provider upstrairs from us who used to provide an IAX trunk to us, that worked well for the most part |
| 17:25.06 | *** join/#asterisk bluequijote (n=chatzill@user-10cm3kj.cable.mindspring.com) |
| 17:25.07 | jrobison | sorry about that. |
| 17:25.08 | FlatFoot | ManxPower: thats a really good jbot answer |
| 17:25.24 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 17:25.25 | jrobison | funny though |
| 17:25.56 | jrobison | how would I configure it in /etc/asterisk I should have asked. I dont really use the GUI, it was here before my time |
| 17:26.04 | jrobison | I would actually like to move to FreeBSD |
| 17:26.05 | ussrback | Hi all |
| 17:26.20 | ussrback | How can i set participant limit in meetme conference room? |
| 17:26.21 | ManxPower | jrobison: best of luck converting AsteriskNOW to regular Asterisk. |
| 17:26.37 | shido6 | that coming from ManxPower means "work" |
| 17:26.58 | ManxPower | shido6: Hey! How is the training project going? |
| 17:26.59 | FlatFoot | jrobinson: i would build from the start onto FreeBSD. We have and so far it works fine |
| 17:27.27 | jrobison | that would be fine, my manager is just telling me he needs something that works, now |
| 17:27.29 | ManxPower | jrobison: don't make a second newbie mistake and try to learn Asterisk on a production system. |
| 17:27.29 | shido6 | i trashed everything and started over. It all wrong |
| 17:27.30 | FlatFoot | just got stuck on one thing cdr_adaptive_odbc |
| 17:27.37 | jrobison | I am sure you have all heard that speech |
| 17:27.58 | FlatFoot | jrobinson: it took about 3 hours to get a working box |
| 17:28.18 | Qwell | jrobison: If you've never used asterisk, and it needs to be done "now" - don't use freebsd |
| 17:28.25 | jrobison | I have a FreeBSD install working at home, I love it, runs on a sparc |
| 17:28.42 | jrobison | sun blade 100 |
| 17:28.56 | ManxPower | jrobison: if you switch to BSD, almost nobody will be able to help you |
| 17:29.12 | ussrback | @Qwell: do u mean that Freebsd is better to be used then Linux for asterisk? |
| 17:29.18 | jrobison | what do most of you guys/gals use? |
| 17:29.26 | jrobison | I didnt mean to start a flame war |
| 17:29.28 | ManxPower | jrobison: Linux is the supported OS |
| 17:29.32 | Qwell | ussrback: no, I said don't use freebsd |
| 17:29.46 | Qwell | jrobison: "standard" zaptel only works on Linux. |
| 17:29.55 | ussrback | @Qwell: ahh ok. but what about the Solaris OS? |
| 17:30.01 | Qwell | Your hardware won't really be supported if you're running anything besides linux |
| 17:30.09 | Qwell | bbl, lunch |
| 17:30.17 | ussrback | as i know solaris is a best choice for good performance of the sistem |
| 17:30.37 | ManxPower | ussrback: It does not matter how many times you ask, the answer will still be "Asterisk is officially supported on Linux" |
| 17:31.17 | shido6 | unless you are a unix yogi stick with linux |
| 17:31.36 | ussrback | ManxPower: Yes i know, but also i have read the solaris vs linux and facts says that Solaris OS is more stable and load resistant then linux ones |
| 17:31.41 | shido6 | Solaris is a good idea after you've mastered Asterisk on linux |
| 17:32.04 | ManxPower | ussrback: It very well bay me, but that doesn't matter if the software you want to use does not work on Solaris. |
| 17:32.06 | Zuchmir | ussrback: look in the asterisk 1.4 book page #165 |
| 17:32.33 | ussrback | Zuchmir: give me the link |
| 17:32.38 | ManxPower | ~book |
| 17:32.39 | 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 |
| 17:32.47 | shido6 | otherwise you will hit brick walls faster than you can make a left turn at 180mph during a typhoon on a wet road in the philippines |
| 17:33.18 | ManxPower | ussrback: if you want to use Asterisk on Solaris, then you need to start coding to make it work on Solaris. |
| 17:33.40 | De_Mon | i want to use asterisk on mac I hear its more user friendly |
| 17:33.52 | ussrback | yeah thats right cause many features as i know doesnot fit with solaris |
| 17:33.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:34.23 | shido6 | get the picture? |
| 17:35.03 | ussrback | no |
| 17:35.13 | shido6 | then you are on the right track! |
| 17:35.21 | ussrback | :) |
| 17:35.43 | jrobison | well, all of that aside, if I choose to use linux. are there some good resources on setting up a TDM400P on linux? |
| 17:35.45 | ussrback | thats a good |
| 17:35.58 | jrobison | I will use whatever works |
| 17:36.29 | Nugget | I hate Linux just as much as the next guy, but even I suffer through it for my asterisk machines. |
| 17:36.44 | Nugget | Trying to run Asterisk (with Zaptel) on anything other than Linux is the road to misery |
| 17:36.45 | jrobison | yeah, I am a BSD/SOlaris guy |
| 17:36.51 | mort_gib | You use the tools you have at hand |
| 17:37.12 | Nugget | I suggest Slackware. It's the least Linuxy Linux. |
| 17:37.12 | muiro | Ok, let me try to describe the problem I'm having. I'v just recently plugged lumenvox speech recognition into my asterisk system. The grammars I've built are working great for voice. The problem here lies in DTMF. It's my understanding that when SpeechBackground() is being used, if asterisk hears a DTMF tone it stops sending data to the speech engine and simply handles the dtmf tone itself. Here's the issue: Some of these DTMF tones are wr |
| 17:37.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:37.22 | jrobison | I have just never configured a TDM card, or anythiing other than IAX/SIP on an asterisk box |
| 17:38.01 | Nugget | zaptel is a bit nutty, but it's a zillion times simpler than, say, hylafax or something. And Digium offer good support. |
| 17:38.02 | mort_gib | jrobison: NOt difficult, have a peek in the book |
| 17:38.18 | jrobison | the one mentioned above? |
| 17:38.21 | muiro | is it true that asterisk handles those dtmf tones? and if so, is there a way I can clean them up a bit? asterisk dials extensions and everything perfectly fine, it's just the dtmf tones that come during SpeechBackground() that seem to be busted |
| 17:38.34 | mort_gib | Yup, It got me up and working in no time. |
| 17:38.39 | jrobison | thanks man |
| 17:38.52 | mort_gib | Got a bit confused with FXO FXS |
| 17:38.56 | mort_gib | :-/ |
| 17:39.15 | jrobison | oh, now I found that on the asterisk website as well. sorry |
| 17:39.23 | mort_gib | :-) |
| 17:39.33 | shido6 | http://www.voip-info.org/wiki-FXO |
| 17:39.37 | jrobison | my bad, I do at least try to find this stuff first |
| 17:39.46 | shido6 | http://www.voip-info.org/wiki/view/FXS |
| 17:39.46 | mort_gib | Don't worry! |
| 17:39.48 | Nugget | ~fxofxs |
| 17:39.49 | jbot | i heard fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 17:40.02 | shido6 | now...... for the cheap seats |
| 17:40.03 | shido6 | http://www.3cx.com/PBX/FXS-FXO.html |
| 17:40.30 | mort_gib | Yeah, that all good, but it's DARK behind my server and the ports look the*£%$"£$_+ SAME |
| 17:41.02 | jrobison | I will definetly use Linux before I use Windows, I worked for MS for 2 years as a UNIX administrator. |
| 17:41.18 | shido6 | blasphemy |
| 17:42.08 | shido6 | as bkw has been quoted saying: In a world without fences and walls, who needs Gates and Windows. |
| 17:42.09 | mort_gib | jrobison: I LIKE that |
| 17:42.33 | shido6 | wow |
| 17:42.35 | shido6 | i need lunch |
| 17:42.58 | mort_gib | I use OpenBSD for loads of stuff, Linux on my desktop and I support 150+ windows users :-) |
| 17:43.05 | UnixDog | ok has anyone here done a queued pagining setup they would share |
| 17:43.35 | Nugget | I use openbsd for my firewall, linux for my asterisk, freebsd for anything that matters, and os x on the desktop. |
| 17:43.49 | jrobison | mort_gib: yeah, I dont mean to be harsh, I just try to avoide them/thier software whenever I can |
| 17:44.16 | jrobison | Nugget: I am on an MBP right now, lol I love Macs for the Desktop |
| 17:44.27 | tzafrir | Nugget, look for lsh |
| 17:44.37 | mort_gib | I'm opposite, I'm happy that my clients use Windows, I would not be very busy if they used anything else! |
| 17:44.44 | shido6 | leopard here |
| 17:44.45 | tzafrir | beats cmd.exe as a shell :-) |
| 17:44.49 | shido6 | on a Dell :) |
| 17:44.57 | Nugget | heh |
| 17:45.07 | UnixDog | what I am looking for is a paging setup where it answers askes them to record thier page puts it in a queue and then dials the page and plays the page |
| 17:45.17 | shido6 | hrmmmmm |
| 17:45.27 | Nugget | One day as a joke I set a friend's shell to emacs. He won, though, becuse he just left it that way. |
| 17:45.39 | jrobison | lol |
| 17:45.42 | shido6 | i did that sort of as a quick fix for "Blue Light Special on Sugar, aisle 6" announcements using the crisco phones auto answer feature |
| 17:45.46 | shido6 | just made a call file |
| 17:46.01 | *** join/#asterisk uribes (n=Toshiba@189.174.79.124) |
| 17:46.06 | UnixDog | callfile ? |
| 17:46.48 | shido6 | indeed |
| 17:46.50 | shido6 | a call file |
| 17:46.57 | shido6 | but you dont have to do that... |
| 17:47.10 | shido6 | u can do what you need with dialplan logic or a *gulp* agi |
| 17:47.43 | uribes | hi everybody, i'm trying to install asterisk in a slackware distro, but i have a problem when i compiled the zaptel packge, the header "workqueue.h" can not be found.. could you help me? |
| 17:47.48 | shido6 | http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom |
| 17:48.27 | shido6 | um..... |
| 17:48.35 | shido6 | do you have the kernel source installed, uribes? |
| 17:48.56 | shido6 | that should be in *somewhere*//include/linux/* |
| 17:48.57 | uribes | yes.. i think |
| 17:50.06 | uribes | yes.. i'm checking.. i have many headers in /usr/src/linux/include/linux/, but i can see that header |
| 17:50.18 | uribes | do i need to install something else? |
| 17:50.19 | *** join/#asterisk viperdudeuk (n=chatzill@195.74.96.113) |
| 17:50.33 | shido6 | I dont know. |
| 17:51.43 | uribes | so.. what can i do? download the source kernel?! |
| 17:52.53 | tzafrir | uribes, what kernel version? |
| 17:53.13 | uribes | 2.4.33.3 |
| 17:54.08 | tzafrir | could you please pastebin the exact log from the build? |
| 17:54.30 | uribes | ok, hold on |
| 17:54.36 | *** join/#asterisk Zap-W (n=XoX@213.31.43.2) |
| 17:54.38 | Zap-W | hi |
| 17:54.52 | Zap-W | how do i see the list of SIP servers i am connected to and their latency? |
| 17:56.22 | jrobison | sip show peers |
| 17:56.37 | *** join/#asterisk teh_recon (n=Recon@mail.imprinters.com) |
| 17:56.43 | jrobison | but only if you have the flag set to measure thier latency |
| 17:56.56 | jrobison | I could be wrong, but that is my experience. |
| 17:57.24 | jrobison | I would also take a look at sip show peer (name of peer) |
| 17:58.06 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:59.45 | De_Mon | * keeps telling me there's an error in extension logic (missing '}') in this line -- but I don't see anything wrong |
| 17:59.45 | De_Mon | Set(tmp=${DB(Queue/Bridge)}) |
| 18:00.07 | De_Mon | http://pastebin.ca/811172 |
| 18:00.45 | De_Mon | could it be because the db key is empty? |
| 18:01.16 | blitzrage | if the key is empty, so will the value returned |
| 18:01.50 | tzafrir | ~pb |
| 18:01.51 | 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 |
| 18:01.52 | De_Mon | I don't see any missing }'s do you? |
| 18:01.55 | blitzrage | De_Mon: you have an extra semi-colon |
| 18:02.00 | blitzrage | you're escaping the final ) |
| 18:02.06 | De_Mon | oh, I have to escape those... |
| 18:02.09 | blitzrage | why? |
| 18:02.11 | blitzrage | that is wrong |
| 18:02.30 | blitzrage | the format for that line is incorrect |
| 18:02.32 | De_Mon | its wrong to set a db key with a simicolin in it? |
| 18:02.33 | *** join/#asterisk bobkare (i=bob@cakebox.net) |
| 18:02.35 | blitzrage | if you don't have a closing brace |
| 18:02.47 | blitzrage | it is if you don't escape it in asterisk |
| 18:02.49 | De_Mon | We're talking about priority 2 right? |
| 18:02.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:02.52 | blitzrage | ; means something to the dialplan |
| 18:02.57 | blitzrage | priority 3 |
| 18:03.10 | blitzrage | Set(DB(Queue/Bridge)=${DB(Queue/Bridge)}${CHANNEL};) |
| 18:03.21 | blitzrage | you are not closing Set() properly |
| 18:03.29 | blitzrage | because you've commented out the ending ) |
| 18:03.51 | De_Mon | that makes more sense. |
| 18:04.06 | blitzrage | if you need to pass the ;, then you need to escape it |
| 18:04.18 | blitzrage | \; in trunk, \\\; in 1.4 |
| 18:04.22 | De_Mon | would quoting the string work just as well? |
| 18:04.31 | De_Mon | Set(foo=";") |
| 18:04.32 | blitzrage | huh? |
| 18:04.33 | blitzrage | no |
| 18:04.36 | De_Mon | damn |
| 18:04.41 | blitzrage | you'll comment you ") then |
| 18:04.46 | blitzrage | s/you/out |
| 18:04.56 | blitzrage | you escape it as I showed above |
| 18:05.08 | De_Mon | that way looks ugly though |
| 18:05.11 | De_Mon | ;) |
| 18:05.16 | blitzrage | ok |
| 18:05.21 | De_Mon | maybe I should use a different delimiter |
| 18:05.22 | blitzrage | that's how escaping works |
| 18:05.26 | blitzrage | yes -- using something else |
| 18:05.35 | blitzrage | you can't use |, you can't use ; |
| 18:05.39 | blitzrage | those mean something to asterisk |
| 18:05.51 | blitzrage | use osmething like ^ |
| 18:06.03 | blitzrage | or # , or @ |
| 18:06.23 | blitzrage | NEXT! |
| 18:06.25 | De_Mon | waa queue/persistantmenbers/ uses ; to dlimit members |
| 18:07.59 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:08.07 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:09.38 | Katty | herro? |
| 18:10.17 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:10.37 | *** part/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 18:12.28 | Katty | so quiet :< |
| 18:14.04 | *** join/#asterisk aikanaro79 (n={aikanar@89-180-67-54.net.novis.pt) |
| 18:14.43 | aikanaro79 | hi...are SIP event packages supported by asterisk? |
| 18:17.05 | Katty | [TK]D-Fender: MEW?! |
| 18:17.10 | Katty | [TK]D-Fender: you're fallin down on the job. |
| 18:17.58 | [TK]D-Fender | Katty: Mew. |
| 18:18.01 | [TK]D-Fender | Katty: On lunch |
| 18:18.10 | Katty | [TK]D-Fender: oh, i see. k'then |
| 18:18.22 | [TK]D-Fender | Katty: just back now. |
| 18:18.29 | Katty | [TK]D-Fender: cheers. |
| 18:18.41 | Katty | [TK]D-Fender: guess who's looking at engagement rings ^_^ |
| 18:18.49 | De_Mon | blarg no regex replace for asterisk |
| 18:19.04 | bobkare | Is it possible to use chan_mobile with the *1.4 packages in ubuntu gutsy? I really only want chan_mobile and would really like to avoid compiling the entire thing by hand. I've tried, but obviously my attempt must be way off as I got loads of compile errors on standard include files |
| 18:19.24 | Katty | [TK]D-Fender: i found history!!! and bookmarks! one was tungstencarbidedirect.com |
| 18:20.02 | Katty | [TK]D-Fender: and this was bookmarked too: http://www.sapphireweddings.com/sapphire_wedding_ring-small.jpg |
| 18:20.10 | Katty | [TK]D-Fender: i think someone is plotting. |
| 18:20.53 | [TK]D-Fender | Katty: Cubic Zirconium may be right for you! ;) |
| 18:21.36 | Katty | [TK]D-Fender: probably. my vision is horrible anyway ;) |
| 18:21.39 | Katty | [TK]D-Fender: but still. |
| 18:21.47 | Katty | [TK]D-Fender: i'd say there's some major plotting going on. |
| 18:22.15 | Katty | [TK]D-Fender: that sapphire site is wedding 'packages' |
| 18:22.15 | [TK]D-Fender | Katty: Thats more like X-mas / Valentines, not engagement. |
| 18:22.35 | Katty | [TK]D-Fender: maybe. |
| 18:22.44 | Katty | [TK]D-Fender: but i've got a hunch |
| 18:23.43 | [TK]D-Fender | Katty: You should work for the DoD/DHS :) |
| 18:24.02 | *** join/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 18:24.16 | TUplink | is there somewhere in a config file that i can set DNS files to use? |
| 18:24.23 | TUplink | DNS servers |
| 18:24.24 | Katty | [TK]D-Fender: Dod? |
| 18:24.26 | Katty | [TK]D-Fender: DHS? |
| 18:24.30 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 18:24.34 | Katty | [TK]D-Fender: english please. |
| 18:24.35 | TUplink | i keep getting DNS lookup errors |
| 18:24.36 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 18:25.04 | TUplink | and im on DHCP so everytime the server reboots it overrited /etc/resolve |
| 18:25.46 | bobkare | you should be able to override whatever's writing to resolv.conf. what distro are you using? |
| 18:25.52 | [TK]D-Fender | Katty: You seem so blissfully unaware of your own government... it's kinda cute ;) |
| 18:26.18 | Katty | [TK]D-Fender: i try. |
| 18:27.35 | uribes | tzafrir: http://pastebin.com/md47d15c.. this is the log, any suggetion it's welcome |
| 18:28.12 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:29.23 | tzafrir | uribes, I think you snipped out the important part |
| 18:31.02 | uribes | so you need to see the warnings? |
| 18:31.53 | tzafrir | At least the first few lines of them |
| 18:32.07 | uribes | ok |
| 18:32.41 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:33.09 | [TK]D-Fender | Katty: Department of Defense / Department of Homeland Security. You know... the one that puts out pretty colour-coded "terror threat-level" warning based on rabidly BS "hunches". |
| 18:33.41 | Katty | no clue. |
| 18:34.39 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 18:34.41 | Katty | or so i claim. |
| 18:34.51 | Katty | can we please not think about politics today. |
| 18:34.53 | Katty | or.. any day. |
| 18:35.58 | jrobison | sweet lets lead the revolution of ignorance and just let the govronment do what they want ;-) |
| 18:37.45 | uribes | tzafrir: http://pastebin.com/m208c8282 here are some warning |
| 18:37.46 | [TK]D-Fender | jrobison: You say that as thought it weren't a day-to-day FACT :p |
| 18:38.01 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:38.30 | jrobison | well, I guess I am the onyl person that is not ok with it and tries to do what I can; |
| 18:38.42 | uribes | and all the warnings are similiar |
| 18:38.45 | tzafrir | Sorry, false warning |
| 18:39.27 | uribes | what i gonna do is to install a new version of the kernel.. |
| 18:39.46 | Assid | err.. anyone know why does this error show up for the polycoms app.log ? 1210132426|cfg |4|03|Edit|Error 0x388002 attempting stat of /ffs0/local/0004f2030f68-phone_cfg.zzz |
| 18:40.04 | TUplink | <PROTECTED> |
| 18:40.19 | uribes | maybe this version has some bug or it's not capable with the zaptel version |
| 18:41.25 | uribes | thanks anyway |
| 18:42.07 | *** join/#asterisk jtexter3 (n=jamest@69-153-182-116.bn02845.tulsok.wayport.net) |
| 18:42.36 | lirakis | ~book |
| 18:42.37 | 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 |
| 18:42.44 | *** join/#asterisk jtexter3 (n=jamest@69-153-182-116.bn02845.tulsok.wayport.net) |
| 18:43.25 | [TK]D-Fender | TUplink: Yup, believeable.... |
| 18:44.34 | bobkare | I have a backported version of chan_mobile for 1.4 release, do I need the entire * src tree to compile it, or can I use just the headers from the asterisk-dev ubuntu package and get the module working? |
| 18:45.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:53.37 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 18:53.57 | *** join/#asterisk cesar_CR (i=cesar@201.198.194.87) |
| 18:53.59 | *** part/#asterisk aikanaro79 (n={aikanar@89-180-67-54.net.novis.pt) |
| 18:54.28 | iratik | How do they (jajah, poivy... etc..) connect one number to another? do they pipe commands directly through to the asterisk CLI? |
| 18:54.42 | tzafrir | uribes, I'm trying to figure out how that code has built on 2.4 before |
| 18:55.42 | *** join/#asterisk techie (n=techie@adsl-76-214-26-129.dsl.lsan03.sbcglobal.net) |
| 18:57.55 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 19:00.08 | mvanbaak | hhmm, 2.4 |
| 19:00.08 | mvanbaak | nice |
| 19:02.56 | Assid | dammit.. the new updates on this sip version kinda messed me up |
| 19:03.01 | Assid | the gains are just messed up |
| 19:03.28 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 19:03.45 | Assid | anyone know what the preamp is for ? |
| 19:05.30 | TUplink | im on freebsd.... when i do kldload /usr/local/lib/zaptel/ztdummy.ko it says that /usr/local/lib/zaptel/ztdummy.ko does not exist..... any ideas |
| 19:05.41 | TUplink | i can ls /usr/local/lib/zaptel/ztdummy.ko and see |
| 19:05.42 | TUplink | it |
| 19:06.54 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:07.16 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:08.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:09.20 | fakhir | haker0 |
| 19:10.23 | tzafrir | uribes, hmm... for some reason on 2.4 it doesn't want to build me wctdm24xxp |
| 19:11.39 | *** join/#asterisk bhima (n=gopi@62.215.80.67) |
| 19:12.30 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 19:13.37 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 19:19.25 | tzafrir | uribes, hmm... it was only successful because I had linux/workqueue.h under /usr/include |
| 19:19.29 | tzafrir | That's bad |
| 19:21.33 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 19:22.01 | *** join/#asterisk naitram (n=naitram@216.77.58.40) |
| 19:22.31 | naitram | can more that 1 client attach to the AMI at the same time? |
| 19:22.52 | [TK]D-Fender | naitram: yes |
| 19:23.28 | uribes | ok, i'm gonna update the kernel.. but can be 2.6 or 2.4 line? |
| 19:24.12 | naitram | [TK]D-Fender: ok, thanks |
| 19:24.57 | [TK]D-Fender | naitram: if you're planning on having many systems connect to it it frequently you might want to run AstManProxy |
| 19:25.57 | tzafrir | uribes, what card do you actually have? |
| 19:26.29 | uribes | actually.. i dont have a card |
| 19:27.28 | naitram | [TK]D-Fender: thnks will look at it |
| 19:28.12 | uribes | that's can be the problem? |
| 19:28.40 | Assid | okay something gone crazily haywire |
| 19:28.45 | Assid | the gains are just messedup |
| 19:31.13 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:31.36 | Assid | err.. the mic is tx right ? and rx = speaker ? |
| 19:37.28 | ManxPower | Assid: no. tx is transmitted audio and rx is received audio. |
| 19:37.41 | ManxPower | WHERE the audio goes and where it comes from does not matter for this. |
| 19:38.29 | Assid | yeah.. so mic = tx |
| 19:38.46 | Assid | i dont get it.. i lowered it to -21 .. and the audio still is low |
| 19:43.39 | *** part/#asterisk naitram (n=naitram@216.77.58.40) |
| 19:48.54 | kaldemar | Assid: tx and rx are from asterisk's point of view. rx from a channel, tx to a channel. |
| 19:53.50 | iratik | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf ---- I have to use auth=rsa with my IAX provider..... i don't know where to put the ".pub" file containing the public key.... where can i put that file so that inkeys= ... will see the file |
| 19:53.52 | iratik | ? |
| 19:54.01 | iratik | nevermind |
| 19:54.46 | uribes | tzafrir: i gotta go thanks |
| 19:54.50 | *** part/#asterisk uribes (n=Toshiba@189.174.79.124) |
| 19:55.28 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:55.31 | tzafrir | duh, missed that. he should have just disabled those specific drivers... |
| 19:58.32 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:01.45 | *** join/#asterisk Giofe (n=Giovanni@mailing.condorviews.com) |
| 20:03.42 | Giofe | hi, help me please,cat /proc/interrupts |
| 20:03.47 | Giofe | <PROTECTED> |
| 20:04.29 | Giofe | how to clear the ERR:1 ? |
| 20:05.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:08.09 | Giofe | ERR:1 is a problem to calls? reboot the synchronization of te22p? |
| 20:08.34 | Giofe | the te220p is on server ML115 HP |
| 20:10.52 | *** join/#asterisk cesar_CR (i=cesar@201.192.86.6) |
| 20:14.19 | *** part/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 20:23.46 | *** join/#asterisk ManxPower (n=manxpowe@81.sub-75-201-135.myvzw.com) |
| 20:25.30 | ManxPower | Assid: -21 would make motorcycle sound like a whisper |
| 20:25.51 | Assid | yeah just learned that |
| 20:26.20 | Assid | so +10 would make it super loud? |
| 20:26.44 | ManxPower | start at 0 for both, then increase or decrease by no more than 2 |
| 20:26.56 | ManxPower | which zap card do you have? |
| 20:27.15 | Assid | err.. everything over ip.. |
| 20:27.33 | Assid | termination and origination over ip |
| 20:27.35 | ManxPower | Assid: Asterisk does not support adjusting audio in VoIP calls. |
| 20:27.56 | Assid | these guys are kinda "deaf" |
| 20:27.57 | ManxPower | Volume adjustment should be done where the call is converted to/from PSTN |
| 20:28.34 | Assid | err.. doing at phone.. polycoms' config |
| 20:28.39 | ManxPower | Assid: I don't care if they are them Pope. You are not going to be able to adjust audio on VoIP calls unless you control the PSTN gateway. |
| 20:28.52 | ManxPower | You should be able to adjust that on the PHONE, of course. |
| 20:29.07 | Assid | yep thats whaty im doing |
| 20:29.34 | ManxPower | which phone do you have? |
| 20:29.48 | *** join/#asterisk swampfox0866 (n=frankb@166.70.132.97) |
| 20:30.10 | *** part/#asterisk Aughey (n=jha@64.219.54.125) |
| 20:30.21 | *** join/#asterisk Aughey (n=jha@64.219.54.125) |
| 20:30.32 | Assid | they have 501's and 601's and im on a 301 remote location |
| 20:30.54 | ManxPower | So you are using sip.cfg or phone1.cfg to adjust the audio |
| 20:31.20 | Assid | yep.. single sip instance |
| 20:31.34 | ManxPower | "single sip instance"? |
| 20:35.45 | Assid | err.. single sip config file |
| 20:36.24 | ManxPower | sip.cfg not sip.conf, right? |
| 20:36.35 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:37.56 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 20:39.45 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 20:39.52 | Sci_05 | afternoon all |
| 20:40.31 | Sci_05 | can anyone point me into the right direction as to what this mean "Failed to write data to channel monitor write stream" I am getting it when I record a call off my ZAP channel |
| 20:41.20 | *** join/#asterisk servergod (n=maverick@70.97.159.120) |
| 20:41.32 | Assid | ManxPower: yessir |
| 20:41.42 | ManxPower | Anyone else getting spammed from "Intelligent Office" advertizing receptionist services? |
| 20:41.45 | mvanbaak_ | gheh, freaking connection |
| 20:42.26 | servergod | has anyone set up a self serv sign-up form for asterisk? |
| 20:43.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:46.28 | ManxPower | servergod: That's trivial. The hard part is the backend that accepts the form data. |
| 20:47.35 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 20:47.44 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 20:48.10 | muiro | if anyone can help me with a lumenvox question: I seem to be failing the environment variables for the lumenvox connector on asterisk startup. My one question is that maybe it's because my licens server is on another machine and one of the environment variables is the license server? Could that be the reason it's failing? I have license_client.conf set to look at the right server |
| 20:48.44 | file | LVBIN |
| 20:48.54 | muiro | LVBIN is right |
| 20:49.40 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 20:50.06 | muiro | oh, you thought I was asking where license_client.conf was. No, I know where it is and I know it's set correctly because that compiled exampled program worked fine |
| 20:51.01 | muiro | file: you don't mean that I should set LV_LICENSE to LVBIN because my license server is elsewhere, do you? |
| 20:51.34 | file | nah, I mean LVBIN is the environment variable used to find the location of license_client.conf |
| 20:51.34 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 20:51.44 | muiro | well, that's set correctly |
| 20:52.16 | file | LVRESPONSE and LVLANG as well. |
| 20:52.41 | muiro | set and set |
| 20:52.47 | muiro | also LVINCLUDE |
| 20:52.56 | muiro | and LD_LIBRARY_PATh and LD_RUN_PATH |
| 20:53.24 | muiro | and just to be on the "this can't be the problems side", since asterisk is running as non-root, I went ahead and made sure that user had perms on the lumenvox stuff |
| 20:54.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:54.22 | muiro | I have all of these env variables set in the safe_asterisk script as well as in both root's and the user's bashrc |
| 20:55.07 | muiro | any further hints? |
| 20:55.07 | Alan_Hicks | I'm having some trouble with getting my Polycom IP 320 phones to auto-answer a call. I've been following the information here: http://tinyurl.com/b9r9f and here: http://tinyurl.com/34m22c but haven't had any success. |
| 20:55.55 | Alan_Hicks | The relevant portion of my extensions.conf file is here: http://pastebin.com/d509fb35. |
| 20:56.15 | Alan_Hicks | When I enter exten 801, the phone rings, but does not auto-answer. |
| 20:56.27 | ManxPower | Alan_Hicks: try th info on voip-info |
| 20:56.45 | Alan_Hicks | ManxPower: Those tinyurl links are to that info. |
| 20:56.48 | [TK]D-Fender | Alan_Hicks: Amazing that you say it even dials... since you have no priority #2 |
| 20:57.01 | [TK]D-Fender | Alan_Hicks: And are using deprecated vars |
| 20:57.08 | Alan_Hicks | [TK]D-Fender: Shoot, I may have butchered that. |
| 20:57.40 | Alan_Hicks | Sorry, that was left over from when I added a SIP header at priority 2, which I later removed. |
| 20:58.01 | ManxPower | Alan_Hicks: and did you read the UPGRADE.txt file in the Asterisk source code so you could figure out how to convert the old outdated stuff that's usually on voip-info into whatever version you are using? |
| 20:58.10 | [TK]D-Fender | Alan_Hicks: Please paste your actual extensions.conf and sip.cfg |
| 20:58.12 | Alan_Hicks | http://pastebin.com/d8010efb |
| 20:58.23 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:58.24 | Alan_Hicks | I'll paste the full extensions.conf if you wish. |
| 20:59.28 | [TK]D-Fender | exten => 801,2,SIPAddHeader(Alert-Info: Auto Answer) |
| 20:59.28 | Alan_Hicks | Full extensions.conf and sip.conf here: http://pastebin.com/d691e5320 |
| 20:59.36 | [TK]D-Fender | Alan_Hicks: I said sip.cfg <----- |
| 20:59.39 | Alan_Hicks | ManxPower: No, I did not know I would need to do such. |
| 20:59.55 | Alan_Hicks | [TK]D-Fender: My mistake. Just a moment. That's a rather large file. |
| 21:00.10 | [TK]D-Fender | Alan_Hicks: look up for the proper way to set the header |
| 21:02.00 | muiro | file: hmm, maybe it's not the env variables. "res_speech_lumenvox.so" was not compiled with the same compile-time options as this version of asterisk. will not be initialized as it may cause instability. |
| 21:03.07 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 21:04.07 | file | if you grab 1.4 from SVN it should be compatible |
| 21:04.53 | muiro | wait, I know what it might be |
| 21:05.02 | muiro | firstly I used 1.4.current from the ftp |
| 21:05.10 | muiro | but I used just a tiny bit older version of the connector |
| 21:05.17 | muiro | like b18 instead of b19 |
| 21:05.31 | file | Not connector |
| 21:05.32 | file | Asterisk |
| 21:05.55 | muiro | no, I know |
| 21:06.09 | muiro | I was telling you exactly what version I had gotten. For larks, I guess. |
| 21:06.17 | muiro | but maybe it's the b18 that's the problem |
| 21:06.24 | file | stuff changed which made binary modules such as that incompatible without recompiling and providing a new one... since Lumenvox hasn't put it up yet you can grab 1.4 from subversion where the logic was changed to allow old binary modules to work |
| 21:06.47 | file | there is nothing you can do short of using 1.4 from subversion, or using 1.4.14 |
| 21:07.02 | muiro | alright, thanks |
| 21:10.36 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 21:10.36 | *** mode/#asterisk [+o twisted] by ChanServ |
| 21:10.40 | twisted | GRRRRR |
| 21:10.45 | twisted | !!!THANKS HSVUTIL |
| 21:10.54 | [hC] | holycrap its twisted |
| 21:11.44 | Corydon76-lap | [hC]: yes, he's so huggable, isn't he? |
| 21:12.04 | *** join/#asterisk jdunck (n=jdunck@adsl-70-247-106-166.dsl.rcsntx.swbell.net) |
| 21:12.05 | Alan_Hicks | sip.cfg: http://pastebin.com/d7b998d73 |
| 21:12.43 | Alan_Hicks | [TK]D-Fender: I'm not sure what you mean by the "proper way to set the header". Isn't SIPAddHeader the "proper way"? |
| 21:12.44 | [TK]D-Fender | aalyup, you feel for the most common error |
| 21:12.48 | [TK]D-Fender | Alan_Hicks: rather |
| 21:13.19 | [TK]D-Fender | Alan_Hicks: <alertInfo voIpProt.SIP.alertInfo.1.value="AA" voIpProt.SIP.alertInfo.1.class="3"/> <alertInfo voIpProt.SIP.alertInfo.2.value="RA" voIpProt.SIP.alertInfo.2.class="4"/> <- you tried doing this in TWO tags. This has to be a SINGLE tag with all the values in it. |
| 21:13.57 | [TK]D-Fender | Alan_Hicks: You did "exten => 801,2,Set(_ALERT_INFO="RA")" , I suggested "exten => 801,2,SIPAddHeader(Alert-Info: Auto Answer)" |
| 21:14.50 | [TK]D-Fender | Alan_Hicks: Now change your dialplan with my new line, consolidate your "<AlertInfo" tags, change the AA & RA or their proper resptive naming, and you should be fine (following reboots |
| 21:14.50 | Alan_Hicks | So I should remove voIpProt.SIP.alertInfo.2.value="RA"? |
| 21:15.01 | [TK]D-Fender | Alan_Hicks: CONSOLIDATE those into 1 tag. |
| 21:15.47 | [TK]D-Fender | Alan_Hicks: Here's a complete replacement : <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class="" voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4" voIpProt.SIP.alertInfo.3.value="Auto Answer" voIpProt.SIP.alertInfo.3.class="3"/> |
| 21:15.49 | Alan_Hicks | I'm not certain what you mean by that, since they have the same info. |
| 21:16.02 | Alan_Hicks | OH! |
| 21:16.04 | [TK]D-Fender | Alan_Hicks: you can't do it in 2, trus me. |
| 21:16.05 | Alan_Hicks | Gotcha. |
| 21:17.05 | [TK]D-Fender | Alan_Hicks: swap out your 2, add mine, change the header call in your dialplan to match, apply, reboot your phone and you should be fine. |
| 21:17.08 | Alan_Hicks | Alright, rbooting. |
| 21:17.29 | Alan_Hicks | Sucks that the info on voip-info is so wrong. |
| 21:17.41 | [TK]D-Fender | Alan_Hicks: I prefer the term "carbon-dated" :) |
| 21:17.50 | Alan_Hicks | Well, that and that I screwed the pooch in sip.cfg. |
| 21:17.54 | mvanbaak | carbon-dated ? |
| 21:17.57 | Alan_Hicks | [TK]D-Fender: haha. I like that one. |
| 21:18.04 | mvanbaak | I think it's from 3 centuries ago ;) |
| 21:18.09 | ManxPower | It's more like "milk dated". |
| 21:18.22 | Alan_Hicks | [TK]D-Fender: You're awesome. |
| 21:18.26 | [TK]D-Fender | mvanbaak: How they age estimate fossils based on carbon decay |
| 21:18.38 | mvanbaak | ;) |
| 21:18.41 | Alan_Hicks | ManxPower: If it was milk dated, I'd hate to see what the milk in your fridge looks like. |
| 21:18.45 | ManxPower | With the 1.2 UPGRADE.txt and the 1.4 UPGRADE.txt, you should be able to translate the info on voip-disinfo.org |
| 21:19.17 | mvanbaak | nice janitor project: fix voip-info |
| 21:19.33 | ManxPower | mvanbaak: it would take a team of janitors years to do that. |
| 21:19.35 | jdunck | ManxPower: all info on voip-info is old, or just some bits? |
| 21:19.39 | [TK]D-Fender | Alan_Hicks: To a familiar Aerosmith tune : "There's something wierd in the fridge today, I don't know what it is. I think that its alliiivvveeee...." |
| 21:19.41 | Alan_Hicks | Now I just need to specify some sort of louder, longer intercom notification for this. |
| 21:19.44 | ManxPower | jdunck: not ALL. |
| 21:20.02 | [TK]D-Fender | Alan_Hicks: I prefer my silent spy option ;) |
| 21:20.07 | mvanbaak | ManxPower: nah, simply put TFOT2 there as replacement |
| 21:20.09 | mvanbaak | :) |
| 21:20.15 | ManxPower | Just most it seems. that's why I usually suggest voip-info as a LAST resort. |
| 21:20.17 | Alan_Hicks | [TK]D-Fender: Not familiar to me. Around these parts, if you asked for aerosmith people would point you to a fletcher. |
| 21:20.32 | Alan_Hicks | [TK]D-Fender: My client won't though. :^) |
| 21:21.21 | mvanbaak | 'as of today voip-info.org is PDF only' |
| 21:21.44 | ManxPower | mvanbaak: perhaps that can be done on April 1 |
| 21:21.50 | mvanbaak | yeah |
| 21:21.52 | mvanbaak | would be fun |
| 21:22.11 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-43-226.lns10.syd7.internode.on.net) |
| 21:22.50 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 21:23.27 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 21:24.45 | [TK]D-Fender | ok, checkout time... BBIAB |
| 21:25.32 | Alan_Hicks | Damn, there goes all my help. :^) |
| 21:25.45 | ManxPower | mvanbaak: BTW, I am currently interviewing potential clients and am accepting new clients,. |
| 21:26.34 | mvanbaak | like being my competitor ? |
| 21:27.59 | ManxPower | mvanbaak: Competition is far overrated. 8- |
| 21:28.05 | mvanbaak | lol |
| 21:28.07 | ManxPower | I was thinking more of cooperation. |
| 21:28.15 | mvanbaak | actually |
| 21:28.22 | mvanbaak | we dont do setups on the clients location |
| 21:28.27 | mvanbaak | we only do hosted voip |
| 21:29.04 | ManxPower | *nod* And I don't do hosted stuff. 8-) I realize you are thousands of miles from me, but I figured it would not hurt to mention it. |
| 21:29.14 | mvanbaak | we have a client in Rotterdam that will need a total redo of their setup |
| 21:29.25 | ManxPower | I've not accepted new clients in at least 5 years, so this is a big change 8-) |
| 21:30.01 | Alan_Hicks | No new clients in 5 years? Ouch! |
| 21:30.03 | ManxPower | mvanbaak: I don't mind traveling if the money is good, but it would be somewhat expensive to import me from the states. |
| 21:30.25 | mvanbaak | ManxPower: how about SSH ? |
| 21:30.27 | ManxPower | Alan_Hicks: You misunderstand. I have had enough clients for the past 5 years, I did not need any more. |
| 21:30.40 | ManxPower | mvanbaak: ssh is how I do most of my work now. |
| 21:30.43 | Alan_Hicks | ManxPower: Yeah, but... you can't grow a business without new clients. |
| 21:30.54 | mvanbaak | ManxPower: and our own setup needs an upgrade |
| 21:30.59 | mvanbaak | it's still 1.0.9 |
| 21:31.15 | ManxPower | Alan_Hicks: I want to grow my business to the point that I have a decent income. I did that a long time ago. |
| 21:31.23 | ManxPower | Alan_Hicks: I've had one client for at least 10 years. |
| 21:31.23 | Alan_Hicks | Why upgrade it then? Everything on voip-info is still correct for that. :-P |
| 21:31.31 | mvanbaak | lol Alan_Hicks |
| 21:31.56 | ManxPower | One of my other long term clients stopped paying their bills, so I had to fire them. |
| 21:32.02 | Alan_Hicks | ManxPower: Myself, I want to grow my business to the point that I'm making enough money to retire in my 40s. |
| 21:32.12 | Alan_Hicks | ManxPower: I hate when that happens. |
| 21:32.14 | ManxPower | Alan_Hicks: that sounds like a lot of work. |
| 21:32.31 | *** join/#asterisk branen (n=branen@dsl-243-61.zhonka.net) |
| 21:32.40 | ManxPower | I work like 10-15 hours per week and get lots of free time. |
| 21:32.41 | Alan_Hicks | ManxPower: You've no idea. :^) It doesn't help that I'm in a bass-ackwards part of the State where technology is concerned. |
| 21:32.58 | Alan_Hicks | ManxPower: I work about.... 40ish, plus research time on my own. |
| 21:33.21 | hmmhesays | LOL: you too Alan_Hicks? |
| 21:33.21 | Alan_Hicks | You remember when I was in here last? |
| 21:33.27 | hmmhesays | You're not in ND are you? |
| 21:33.33 | Alan_Hicks | hmmhesays: Gotta get with it if you're gonna keep up. |
| 21:33.37 | Alan_Hicks | hmmhesays: Dixie. |
| 21:33.51 | ManxPower | Alan_Hicks: I have the short term memory of a butterfly. |
| 21:33.55 | branen | Hi, folks. Might anyone be able to help me troubleshoot IMAP voicemail storage? |
| 21:34.23 | Alan_Hicks | ManxPower: Well it was about a month ago. This Asterisk stuff has the potential to be a big deal for me, but I ain't worked on it in a month 'cause I've been too damn busy. |
| 21:35.13 | Alan_Hicks | branen: How are you storing to IMAP? Are you handing the voicemail off as an e-mail attachment to a mail server? |
| 21:35.32 | mvanbaak | Alan_Hicks: asterisk supports IMAP for voicemail storage |
| 21:35.59 | branen | Alan_Hicks: I'm trying to use the imapserver= directive in voicemail.conf |
| 21:36.11 | Alan_Hicks | mvanbaak: Yeah, but I can't help him if he's doing it that way. :^) |
| 21:36.20 | mvanbaak | lol |
| 21:36.34 | branen | mvanbaak: Have you got it to work? |
| 21:36.34 | Alan_Hicks | Asterisk I'm a noob on. E-mail, I'm reasonably proficient. |
| 21:36.53 | ManxPower | Asterisk 1.4 supports voicemail storage on IMAP, but it is a feature new to 1.4,. not many people use it and it is not well tested. |
| 21:36.56 | mvanbaak | branen: nope, I'm fine with the default vm stuff |
| 21:37.10 | branen | ManxPower: That's what I was afraid of. |
| 21:37.27 | mvanbaak | I write VM to a NFS share to have it available on a cluster of asterisk boxen |
| 21:37.33 | branen | I'm not getting any errors, but neither am I getting any storage to IMAP. |
| 21:37.42 | ManxPower | branen: there are a couple of threads on the mailing list archives talking about VM IMAP. |
| 21:38.33 | mvanbaak | trunk has seen some fixes lately |
| 21:39.03 | mvanbaak | of course not everyone is as brave as me to run -trunk in production |
| 21:39.13 | ManxPower | branen: I'm waiting until 1.6 comes out and is stable, then I'll be looking at VM IMAP. My users are prissy prima donnas that will skin me alive if their voicemail breaks. |
| 21:40.34 | branen | ManxPower: Sounds like a good plan. My users are clamoring for tighter email-voicemail integration, but if it's not working in 1.4, then |
| 21:40.38 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:40.41 | branen | they'll have to wait. |
| 21:41.29 | mvanbaak | hhmm, ODBC storage is something I'm going to investigate real soon |
| 21:41.55 | mvanbaak | I have this 'high priority' feature request to handle voicemail inside our webbased CRM application |
| 21:42.12 | mvanbaak | I think odbc is the way to go here |
| 21:42.51 | blitzrage | that's how I did it |
| 21:42.59 | kand | mvanbaak: I have been running odbc in production and highly recommend it |
| 21:43.13 | mvanbaak | cool |
| 21:43.40 | mvanbaak | I already handle faxes in our CRM with some AGI trickery |
| 21:43.59 | Alan_Hicks | Hmm.... how do I specify a particular ring on these phones when doing auto-answer? |
| 21:44.03 | mvanbaak | grab the TIFF from the line, convert it to PDF and fire it to the database |
| 21:44.05 | kand | mvanbaak: I have been toying with T.38 |
| 21:44.06 | Alan_Hicks | I've got the following in sip.cfg: |
| 21:44.11 | Alan_Hicks | <PROTECTED> |
| 21:44.15 | Alan_Hicks | e.rt.4.ringer="7" se.rt.4.callWait="6" se.rt.4.mod="1"/> |
| 21:44.54 | mvanbaak | kand: I use a CAPI based ISDN BRI card and chan_capi to retreive the faxes |
| 21:44.55 | Alan_Hicks | Ick.... that was supposed to be one line. Anyhow, changing "se.rt.4.ringer" doesn't have an effect. Even if I set it to "1" (no ring) it still gives me a short, quiet, but audible beep. |
| 21:45.54 | mvanbaak | hhmm, for our hosted setup it's even easier to handle |
| 21:46.05 | mvanbaak | because our ITSP offers fax2mail functionality |
| 21:46.21 | mvanbaak | because fax-over-iax2-over-the-internet is still very buggy |
| 21:46.23 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:46.35 | *** join/#asterisk Greek-B0y (n=email@41.221.58.5) |
| 21:46.54 | kand | alan_hicks: <sip><alertinfo voIpProt.SIP.alertInfo.2.value="Internal" voIpProt.SIP.alertInfo.2.class="5"/></sip><sound_effects><ringType><INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="11" se.rt.5.callWait="6" se.rt.5.mod="1"/></ringType></sound_effects> |
| 21:47.49 | mvanbaak | what kind of phone is that ? |
| 21:48.01 | kand | alan_hicks: then exten => s,n,SIPAddHeader(Alert-Info: Internal) |
| 21:48.03 | *** join/#asterisk sheldonh (i=[4Ycn4dP@66.219.59.32) |
| 21:48.06 | kand | mvanbaak: polycom |
| 21:48.06 | jrobison | looks like a polycom |
| 21:48.36 | kand | you can change the names as you see fit (ie "internal" to "my_custom") |
| 21:48.39 | Alan_Hicks | kand: negative on that. I've got "exten => 801,2,SIPAddHeader(Alert-Info: Auto Answer" |
| 21:49.55 | mvanbaak | I never seen a polycom |
| 21:49.59 | kand | alan_hicks: Oh, sorry misunderstood, why do you want to change the ring on a auto answer? |
| 21:50.02 | mvanbaak | only on images on websites |
| 21:50.07 | sheldonh | anyone else having trouble with "asterisk -r" since upgrading to 1.4.15? /var/run/asterisk/asterisk.ctl and its parent dir are readable by group asterisk, and i have group asterisk membership, but connect() gets EACCESS |
| 21:50.21 | kand | mvanbaak: very nice phones, only kind I enjoy working with....lol |
| 21:50.37 | Alan_Hicks | kand: Just want a loud audible ring of some sort that's different from usual so users know that it's a page/intercom call. |
| 21:50.39 | mvanbaak | kand: we use snom and cisco |
| 21:50.48 | JT | mvanbaak: fax over voice codec over any voip protocol over the Internet is "buggy" |
| 21:51.02 | mvanbaak | JT: amen to that |
| 21:51.26 | mvanbaak | that's why we let our ITSP handle faxes on their PRI and mail them to us |
| 21:51.51 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 21:51.58 | JT | T.38 would be the right way to do it in realtime |
| 21:52.37 | kand | I think T.38 is comming of age |
| 21:52.54 | Alan_Hicks | fax over anything digital sucks. |
| 21:53.06 | JT | Alan_Hicks: that statement was full of flaws |
| 21:53.25 | Alan_Hicks | JT: Perhaps, but IME it holds true. |
| 21:53.31 | JT | ... |
| 21:53.32 | kand | I disagree, using QOS on residential connection I have nearly 99% success. |
| 21:53.45 | Alan_Hicks | Granted, my experience is rather limited. |
| 21:53.55 | JT | Alan_Hicks: when you normally send a fax over the PSTN, it almost certainly goes over a digital connection. |
| 21:54.14 | Alan_Hicks | JT: You're correct. |
| 21:54.38 | Alan_Hicks | I'm referring to the hacks I've seen where faxes come into a digital phone system which then attempts to convert it back to analogue. |
| 21:54.39 | mvanbaak | not when you use the pigeon RFC |
| 21:54.45 | Alan_Hicks | And they never come through cleanly. |
| 21:54.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:54.58 | JT | Alan_Hicks: maybe you mean voip |
| 21:55.04 | Alan_Hicks | But disreguard me. I'm talking through my ass. |
| 21:55.16 | Alan_Hicks | JT: No, proprietary digital phone systems. |
| 21:55.34 | mvanbaak | RFC 1149 |
| 21:55.41 | JT | most of those explicitely do not handle faxes |
| 21:56.17 | Alan_Hicks | Go into an Autozone or a Pep Boys and you'll see exactly what I'm talking about. |
| 21:56.51 | *** join/#asterisk Darthclue (n=root@li13-84.members.linode.com) |
| 21:57.11 | JT | don't have any of those here |
| 21:57.23 | Alan_Hicks | But that's got nothing to do with voip so is neither here nor there as far as #asterisk is concerned. |
| 21:57.47 | hmmhesays | this voyager episode with jason alexander is lame |
| 21:57.52 | JT | asterisk isn't #voip |
| 21:58.02 | Alan_Hicks | s/with jason alexander // |
| 21:58.13 | Alan_Hicks | JT: True. |
| 21:58.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581616.dsl.bell.ca) |
| 21:58.59 | Alan_Hicks | I'm gonna shut up now. |
| 21:59.11 | sheldonh | what else can i look at to figure out why asterisk -r isn't working? i've checked perms on the socket file, confirmed that it's attached to the running asterisk process (with lsof) and confirmed i have appropriate group membership (with id, just before running asterisk -r) |
| 21:59.26 | Darthclue | if a call goes into packet2packet mode does that call get tracked anywhere inside of asterisk so that one can determine when it get's terminated? |
| 21:59.32 | mocker | sheldonh: Errors? |
| 21:59.57 | sheldonh | mocker: open on the socket file gets EACCESS |
| 22:00.17 | Alan_Hicks | sheldonh: I assume root has no problems connecting? |
| 22:00.24 | sheldonh | Alan_Hicks: naturally :) |
| 22:00.44 | Alan_Hicks | How did you add the user to the group? In /etc/groups I presume? |
| 22:00.54 | mvanbaak | newgroup |
| 22:00.56 | mvanbaak | ;) |
| 22:01.45 | sheldonh | Alan_Hicks: it was a long time ago (like, months), so i can't say for sure. i probably just edited /etc/groups |
| 22:01.45 | *** join/#asterisk gonza8 (n=gonza@190.2.28.1) |
| 22:01.45 | gonza8 | hello i have a digium card, but i will use SIP Trunks (dont connect my card to any provider). Do i need to configure zaptel.conf zapata.conf? can i use the card as a timing device (for i.e. app_meetme) |
| 22:01.45 | Alan_Hicks | sheldonh: grep that file real quick |
| 22:01.53 | mvanbaak | sheldonh: when logged in as the user that should run asterisk -r run this: id |
| 22:02.12 | mvanbaak | gonza8: the card can be used as timing source |
| 22:02.17 | sheldonh | mvanbaak: scroll up. id confirms i have appropriate group membership :) |
| 22:02.21 | mvanbaak | but you'll need to configure zaptel for that |
| 22:02.28 | sheldonh | i wonder if asterisk 1.4.15 isn't maybe dropping supplemental groups |
| 22:02.47 | mvanbaak | ah, sorry |
| 22:02.58 | sheldonh | no worries. busy channel :) |
| 22:03.04 | mvanbaak | too much things going on here, I missed your line where you mention 'id' |
| 22:03.07 | gonza8 | mvanbaak, i found a lot of tutorials to configure zaptel, but no one to use the card ONLY as a timing device |
| 22:03.31 | mvanbaak | gonza8: just configure it, but dont use it in extensions.conf |
| 22:03.32 | kand | gonza8: you cant yet (in the works) just set it up as if you where going to connect spans |
| 22:03.34 | Alan_Hicks | sheldonh: OS? |
| 22:03.53 | sheldonh | Alan_Hicks: debian linux (4.0, i386) |
| 22:04.32 | sheldonh | Alan_Hicks: although obviously backported, since stable offers asterisk-1.2.x :) |
| 22:05.00 | mvanbaak | that's why I always compile from source |
| 22:05.04 | Alan_Hicks | debian's not my cup of tea, but let's make sure everything is right. pastebin the output of "id" and the perms on the socket please. |
| 22:05.23 | gonza8 | mmm there is a sample config of my card (TE410P) in voip.info i dont know if i can use that zaptel config, becouse it sets signaling, channels... and more |
| 22:05.49 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 22:05.51 | mvanbaak | gonza8: if you are not going to connect it, just use it |
| 22:05.55 | yassine | good evening everyone |
| 22:06.07 | mvanbaak | wont matter because you are not going to use it for accessing lines |
| 22:06.16 | mvanbaak | so the signalling and stuff doesn't matter |
| 22:06.29 | mvanbaak | what matters is a working zaptel config so it will act as a timer |
| 22:06.32 | gonza8 | mvanbaak, so... if the signaling is even "wrong" it doesn't care, as long as i dont... "actually" use it |
| 22:06.36 | yassine | which dtmfmode of (rfc2364 and rfc2543) is better or supported per default by asterisk? |
| 22:06.48 | mvanbaak | gonza8: indeed ! |
| 22:06.56 | kand | rfc2833 |
| 22:07.15 | sheldonh | Alan_Hicks: stand by. i'll include the bottom half of the strace too |
| 22:07.29 | Alan_Hicks | ok |
| 22:07.44 | gonza8 | mvanbaak, thanks, so if i allready have the zaptel module and my card module working i just need to add a "dummy" zaptel.conf so asterisk use it as a timing device |
| 22:07.47 | yassine | kand: my client does onyl support the modes stated above |
| 22:08.12 | mvanbaak | gonza8: yup |
| 22:08.15 | gonza8 | mvanbaak, thanks! |
| 22:08.18 | gonza8 | good bye |
| 22:09.04 | kand | yassine: RFC 2364 - PPP Over AAL5 and RFC 2543 - SIP. Neither are DTMF modes |
| 22:09.35 | sheldonh | Alan_Hicks: proof of weirdness :) http://rafb.net/p/xkMW4x21.html |
| 22:10.02 | kand | yassine: and your options with asterisk are In-band, RFC2833, INFO and auto |
| 22:10.06 | sheldonh | i'm not the source of a private joke because i'm the only guy running 1.4.15, right? :) |
| 22:10.09 | Darthclue | i have a system that shows sip channels going into packet2packet mode. this is fine because it in theory improves system performance but i need to be able to determine when the call gets disconnected. is there any way to do this? |
| 22:10.55 | Alan_Hicks | Shouldn't the socket have group write permission? |
| 22:11.07 | [TK]D-Fender | sheldonh, No, we all see you too.. that makes you a PUBLIC joke ;) |
| 22:11.26 | Alan_Hicks | What's the perms on asterisk.socket? |
| 22:11.36 | sheldonh | [TK]D-Fender: you again :) |
| 22:12.10 | sheldonh | Alan_Hicks: asterisk.socket? you mean asterisk.ctl? i included an ls -ld in the typescript |
| 22:12.43 | Alan_Hicks | Disregaurd that. My eyes are seeing things fuzzy... |
| 22:12.53 | sheldonh | Alan_Hicks: i believe it's setuid asterisk, so if you're a member of group asterisk, you get write access that way :) |
| 22:12.53 | Alan_Hicks | acpid.socket != asterisk.socket |
| 22:13.02 | Alan_Hicks | You're correct. |
| 22:14.54 | sheldonh | and the trace doesn't show it giving up privs |
| 22:15.02 | sheldonh | see why i'm stumped? :) |
| 22:16.32 | Alan_Hicks | Yeah, you need to add group write permissions. |
| 22:17.00 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 22:17.04 | sheldonh | worth a shot, but i'll be amazed |
| 22:17.04 | Alan_Hicks | the "s" designates it as a socket (as opposed to a file, directory, symlink, what have you), NOT its SUIG, SGID properties. |
| 22:17.10 | sheldonh | AHA! |
| 22:17.20 | sheldonh | Alan_Hicks: duh. thanks! |
| 22:17.56 | Alan_Hicks | <snide comment>Debian users....</snipe comment> :^) |
| 22:17.57 | hmmhesays | cash cab suprises me some days |
| 22:18.30 | sheldonh | Alan_Hicks: we're used to it. could be worse. used to be quite into gentoo :) |
| 22:19.15 | mvanbaak | Debian > * |
| 22:19.21 | mvanbaak | on linux that is |
| 22:19.21 | Alan_Hicks | sheldonh: http://noobfarm.org/?467 |
| 22:19.47 | mvanbaak | if you are forced to use linux, might as well pick the least worst |
| 22:19.51 | [TK]D-Fender | Alan_Hicks, Your XML tags STILL never match :) |
| 22:19.58 | sheldonh | :) |
| 22:20.00 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 22:20.03 | Alan_Hicks | mvanbaak: That's Slackware. :^) |
| 22:20.19 | Alan_Hicks | [TK]D-Fender: Shhhh! |
| 22:20.20 | sheldonh | software sucks. choose the software that sucks the way you like |
| 22:20.34 | mvanbaak | sheldonh: BSD |
| 22:21.08 | sheldonh | mvanbaak: BSD sucks too. as sheldonh@freebsd.org, i'll say it sucks the way i like quite a lot :) |
| 22:21.37 | mvanbaak | sheldonh: then run: s/free/open/ and be a happy camper |
| 22:22.02 | sheldonh | mvanbaak: i don't think you're getting it. software sucks :) |
| 22:22.07 | syzygyBSD | what? |
| 22:22.23 | mvanbaak | sheldonh: yup, every OS sux, I agree |
| 22:22.24 | Alan_Hicks | I'll second OpenBSD. |
| 22:22.30 | mvanbaak | but some suck less then others |
| 22:22.51 | Alan_Hicks | Everything since Apple II... Just a buncha crap. |
| 22:23.02 | mvanbaak | and I tried a lot of different ones, and OpenBSD is still the one that is the least worst |
| 22:23.15 | sheldonh | i'm telling you, they need to extend godwin's law to the apple II |
| 22:23.22 | Alan_Hicks | From Microsoft to Macintosh to Lin, Line, Lin, Line-UX. Every computer crashes, 'cause every OS sucks. |
| 22:23.32 | Darthclue | is there any way to track a call once it goes into packet2packet mode? |
| 22:23.52 | sheldonh | Darthclue: are you referring to passthru mode? |
| 22:23.52 | mvanbaak | Darthclue: SIP ? |
| 22:24.06 | Alan_Hicks | mvanbaak: Gotta give props to any group that can produce things like OpenSSH and pf. |
| 22:24.20 | mvanbaak | Alan_Hicks: amen! |
| 22:24.38 | Darthclue | yeah SIP. the call says it is going into packet2packet which means that it is leaving the dial plan (i'm guessing) so the hangup (h) extension doesn't get called |
| 22:25.06 | mvanbaak | I always thought the signalling was still going through your box |
| 22:25.09 | Alan_Hicks | pf makes netfilter cry and suck it's thumb. |
| 22:25.31 | mvanbaak | yeah |
| 22:25.50 | mvanbaak | specially when you look at how easy it is to setup queues to manage your voip bandwidth |
| 22:26.23 | Alan_Hicks | Or dynamically update a table on which lookups are blazingly fast. |
| 22:26.37 | mvanbaak | etc etc etc |
| 22:26.38 | mvanbaak | ipsec |
| 22:26.49 | Alan_Hicks | And table updates don't require the entire netfilter ruleset to be ripped out ad reloaded for any change. |
| 22:26.55 | mvanbaak | hoststated, carp, documentation, trunk |
| 22:27.08 | Alan_Hicks | s/hoststated/relayd/ |
| 22:27.15 | Alan_Hicks | Got a rename. |
| 22:27.29 | mvanbaak | uhhuh |
| 22:27.40 | mvanbaak | but I'm not running -current in production |
| 22:27.40 | Alan_Hicks | http://undeadly.org/cgi?action=article&sid=20071208214322 |
| 22:27.48 | sheldonh | i had astctlpermissions commented out in asterisk.conf. i'll bet the default has changed since 1.4.11 :) |
| 22:28.01 | mvanbaak | so it's hoststated here for another 6 months |
| 22:28.47 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:29.13 | mvanbaak | we use hoststated in production to loadbalance/monitor a cluster of 50 webservers, a cluster of 20 mailservers and a cluster of 10 mysql boxen and it performs great |
| 22:29.14 | sheldonh | now just to get through the next 10 minutes of nagios alerts telling me everything's okay, and today will be over :) |
| 22:29.46 | mvanbaak | lol sheldonh |
| 22:29.48 | Alan_Hicks | It's a shame OpenBSD doesn't scale on SMP systems very well. That's really it's only major flaw. |
| 22:30.08 | mvanbaak | Alan_Hicks: yup. but they are working on it |
| 22:30.12 | sheldonh | Alan_Hicks: and the lack of a decent journaled filesystem :) |
| 22:30.14 | Alan_Hicks | sheldonh: There's a reason why it's called "NAG"ios. |
| 22:30.23 | mvanbaak | sheldonh: FFS2 is great |
| 22:30.36 | sheldonh | Alan_Hicks: and binary updates. but then bsd people consider the latter a feature ;) |
| 22:30.47 | Alan_Hicks | mvanbaak: I should check to see how Dragonfly is coming along. |
| 22:31.33 | mvanbaak | I dont think dragonfly is as quick with fixing bugs in pf and friends as OpenBSD |
| 22:31.36 | Darthclue | ok, here's what shows up in the log...http://pastebin.ca/811444 ... it looks like the call is still in the system during that time, but it leaves the dialplan and doesn't call the h extension...any ideas? |
| 22:31.57 | Alan_Hicks | mvanbaak: They aren't, but they're the ones doing the real work in BSD on SMP performance. |
| 22:32.12 | sheldonh | mvanbaak: ffs2? do you mean ufs2, or have things progressed? :) |
| 22:32.38 | sheldonh | Alan_Hicks: that's not fair :) |
| 22:32.51 | Alan_Hicks | sheldonh: What's not fair? |
| 22:33.01 | mvanbaak | sheldonh: the results of the latest filesystem hacketon are great |
| 22:33.09 | sheldonh | Alan_Hicks: saying dragonfly are the only ones doing real work on bsd smp |
| 22:33.12 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:33.40 | Alan_Hicks | sheldonh: Perhaps, but they're entire reason for being is they believe in a better way to do SMP and actively strive towards that goal. |
| 22:33.49 | hmmhesays | there is a hot british chick on cash cab right now |
| 22:34.18 | Alan_Hicks | hmmhesays: That is perhaps the most random unrelated comment to a discussion that I have ever seen. |
| 22:34.26 | hmmhesays | yeah I want to see her do naughty things |
| 22:34.35 | Alan_Hicks | Now the discussion has to stop while everyone turns the TV to the Discovery Channel. |
| 22:34.44 | sheldonh | Alan_Hicks: yes. at the time, matt was reacting to the fact that freebsd was in bad shape. so he went off and tried something lightweight and has produced great results. but to say he's the only one doing real work in this area slights the progress freebsd is making, i say :) |
| 22:34.50 | hmmhesays | absolutely |
| 22:35.07 | hmmhesays | i wait, she's n ot british |
| 22:35.14 | Alan_Hicks | sheldonh: I can agree to that. |
| 22:35.25 | sheldonh | hmmhesays: we got that when you said she was hot |
| 22:35.34 | Alan_Hicks | hahahahaha |
| 22:35.40 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 22:35.56 | Alan_Hicks | hmmhesays: Sure fire way to tell she's definitely not British. Are her teeth straight? |
| 22:36.08 | mvanbaak | http://undeadly.org/cgi?action=article&sid=20070412145236 |
| 22:36.16 | Alan_Hicks | Apologies to any British people. |
| 22:36.18 | hmmhesays | haha they are |
| 22:36.40 | hmmhesays | oops she just got out of the cab... not so hot |
| 22:36.50 | Darthclue | Alan_Hicks: that isn't absolutely true...there are some non-brits with very unstraight teeth...and a few, very few, with straight teeth |
| 22:37.18 | Alan_Hicks | Darthclue: Oh I know. I'm not British, and my teeth are as crooked as a Kentucky fence. |
| 22:38.02 | bhima | Brits don't do braces as much as americans. |
| 22:38.10 | jsaunders | Is there a way to change how long between keypresses the ivr waits before dialing the extension? |
| 22:38.13 | Alan_Hicks | Can anyone confirm if that's what cabs actually look like on the inside, minus the flashing lights? |
| 22:38.42 | Darthclue | jsaunders ... WaitExten(X) |
| 22:39.26 | [TK]D-Fender | jsaunders, "show function TIMEOUT" |
| 22:39.43 | jsaunders | Darthclue: Not what I'm referring to... Say someone dials 123. If they dial 12 and then pause for too long before pressing 3, the system will dial 12. I'd like to extend the wait between keypresses. |
| 22:39.52 | sheldonh | argh! |
| 22:39.54 | sheldonh | developers |
| 22:39.59 | sheldonh | always fiddling and breaking things |
| 22:40.00 | jsaunders | Fender: I'll give that a try, tnx. |
| 22:40.17 | Alan_Hicks | jsaunders: That's likely a function of your phone. |
| 22:40.33 | jsaunders | Alan_Hicks: ? How so? |
| 22:40.38 | Alan_Hicks | I can't speak to all models, but the Polycoms exhibit this behavior. |
| 22:40.53 | Alan_Hicks | The phones themselves have a dialplan that has its own timeouts. |
| 22:40.57 | Katty | jbot: wocka |
| 22:40.57 | jbot | Fozzie Bear: Wocka Wocka Wocka! (cue: thrown rotten tomatoes from Statler and Waldorf) |
| 22:41.06 | *** join/#asterisk optize (i=tyler@ip70-176-254-41.ph.ph.cox.net) |
| 22:41.12 | jsaunders | Alan_Hicks: I'm talking about POTS |
| 22:41.20 | Alan_Hicks | jsaunders: AH! |
| 22:41.26 | Katty | Darthclue: sorry, i evade bugged. |
| 22:41.31 | [TK]D-Fender | jsaunders, "show function TIMEOUT" <=------- |
| 22:41.42 | optize | For some reason, if a phone comes from a internal IP space, * will try to communicate to it via it's private address space, instead of it's public IP. Does anyone know how to fix that? |
| 22:42.11 | [TK]D-Fender | optize, read this : |
| 22:42.13 | [TK]D-Fender | ~sipnat |
| 22:42.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:42.15 | Alan_Hicks | optize: You mean if the phone is behind a NAT router? |
| 22:43.26 | mvanbaak | meh, res_jabber rox |
| 22:43.44 | optize | Yeah, I tried to do that as well |
| 22:43.45 | optize | hmph |
| 22:44.26 | mvanbaak | I now get incoming call notifications in my bitlbee screen of irssi \o/ |
| 22:44.35 | [TK]D-Fender | optize, please follow that first guide. It explains both when * is behind NAT, and when your remote end is as well |
| 22:44.56 | mvanbaak | brb, time for a whisky |
| 22:46.58 | mvanbaak | back |
| 22:47.49 | hmmhesays | oh facebook api is the most annoying pos |
| 22:48.06 | JT | s/api// |
| 22:48.20 | hmmhesays | yeah thats what they call it anyway |
| 22:48.38 | hmmhesays | they have a list of about 30 php methods you can do to accompish various tasks |
| 22:48.49 | JT | no... facebook is the most annoying pos :) |
| 22:49.00 | hmmhesays | no that wouuld be myspace |
| 22:49.06 | hmmhesays | facebook is completely tolerable in comparision |
| 22:49.08 | JT | they're both annoying |
| 22:49.20 | Darthclue | myspace is worse |
| 22:49.21 | JT | have you read the user agreement/t & cs of facebook? |
| 22:49.23 | Alan_Hicks | I always thought it was Aldelo. |
| 22:50.40 | hmmhesays | I haven't, care to highlight for me? |
| 22:52.04 | JT | you give them an indefinite license to do what they want with all your personal information |
| 22:52.09 | JT | and they may sell it off |
| 22:52.33 | hmmhesays | good thing you don't have to give them much |
| 22:52.44 | JT | but people do |
| 22:52.47 | JT | in their profiles |
| 22:52.54 | hmmhesays | yeah cause they're stupid |
| 22:52.59 | hmmhesays | so let them do as such |
| 22:53.10 | hmmhesays | they have my name and a spam email address |
| 22:54.54 | *** join/#asterisk RoyK (n=roy@ip-10-16-149-91.dialup.ice.no) |
| 22:56.33 | jsaunders | [TK]D-Fender: Tnx fer TIMEOUT() suggestion, perfect. |
| 23:07.30 | ManxPower | This whole social networking thing was after my time. |
| 23:08.00 | [TK]D-Fender | Anti-Social 1.0 baby! |
| 23:08.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:08.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:08.59 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:10.21 | justnulling2 | i am not getting any voicemail emails and not sure what, any ideas how i can debug it? |
| 23:11.11 | blitzrage | tail -f /var/log/maillog |
| 23:11.56 | justnulling2 | blitzrage: nothing is there as if asterisk is just using /dev/null mail agent |
| 23:12.18 | blitzrage | there should be no configuration other than whatever is in the default voicemail.conf file |
| 23:12.23 | blitzrage | unless you don't have sendmail installed |
| 23:12.40 | blitzrage | or you've got some other option preventing asterisk from sending mail |
| 23:13.58 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808846.dsl.bell.ca) |
| 23:15.26 | justnulling2 | is there a debug mode for vm? the asterisk is behind smtp firewall (that to my provider) so it is setup with smarthosts i can email from command line fine but all vm are lost without an error |
| 23:16.06 | bhima | I'm setting up an Asterisk box. I was planning to set up the VoIP phones first, then stick in the T1 PRI interface card and integrate it. Am I on crack? Is there some reason it would be better to set up the T1 card first? |
| 23:16.24 | blitzrage | just enable 'console => notice,warning,error,debug' in the logger.conf, and do, 'logger reload' followed by 'core set debug 4' |
| 23:16.27 | [TK]D-Fender | bhima, no difference |
| 23:16.40 | blitzrage | none |
| 23:16.42 | blitzrage | nada |
| 23:16.43 | blitzrage | zero |
| 23:16.44 | blitzrage | :) |
| 23:16.49 | [TK]D-Fender | Ziltch |
| 23:16.53 | [TK]D-Fender | nien |
| 23:16.57 | [TK]D-Fender | nyat |
| 23:16.58 | [TK]D-Fender | non |
| 23:16.59 | blitzrage | (I could go on for a while, but I've already started the catalyst) |
| 23:17.08 | *** part/#asterisk sheldonh (i=[4Ycn4dP@66.219.59.32) |
| 23:17.43 | Deeewayne | ara (Georgian 'no') |
| 23:18.01 | bhima | critical mass? You mean runaway gravity compression thing, like a black hole? |
| 23:18.03 | Qwell | null, nil |
| 23:19.22 | bhima | nobody said "rien". I guess there aren't any francophones here. :) |
| 23:19.44 | blitzrage | we kicked them all out |
| 23:19.45 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-45-64.red.bezeqint.net) |
| 23:19.46 | bhima | Oh, "non". Sorr. |
| 23:20.03 | bhima | I'm not a francophone. |
| 23:20.06 | Dovid | is it possible to run a TDm400P and a Sangoma A102 in harmony ? |
| 23:20.15 | blitzrage | bhima: :) |
| 23:20.30 | blitzrage | Dovid: I don't know... is it? try it! :) |
| 23:20.36 | [TK]D-Fender | Dovid, Don't see why not. |
| 23:20.57 | [TK]D-Fender | bhima, "non" <---- |
| 23:21.00 | Dovid | Just wondering how to configure zapata.conf and zaptel.conf |
| 23:21.06 | [TK]D-Fender | bhima, I took the otehr form to answer. |
| 23:21.09 | Dovid | never did a PRI b4. time to figure it out ;) |
| 23:21.33 | bhima | [TK]D-Fender: Yeah, sorry, I was a bit slow. It's 2am, I'm allowed to be slow. :) |
| 23:21.40 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:21.51 | [TK]D-Fender | bhima, y-a trop de francophones deja :p |
| 23:22.34 | justnulling2 | so anyway to debug voicemail? |
| 23:23.30 | bhima | [TK]D-Fender: déjà? |
| 23:23.59 | Dovid | TK: Which version of wanpipe is the best ? |
| 23:24.40 | [TK]D-Fender | bhima, Can't be bothered to install the Cdn Multilingual KB on my setup :p |
| 23:24.48 | mvanbaak | latero all |
| 23:24.54 | [TK]D-Fender | bhima, besides I get the accents wrong half the time anyways |
| 23:24.56 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 23:25.03 | [TK]D-Fender | Dovid, all of them :) |
| 23:25.17 | Dovid | lol |
| 23:25.20 | bhima | [TK]D-Fender: I cheated. I pasted that from a translation app cause I wasn't sure about the accents. |
| 23:25.29 | Dovid | just wanted to know if a specific version worked better |
| 23:25.40 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 23:26.30 | justnulling2 | bilzrage: don't be in rage, i missed your comment in that no/zilch/non text |
| 23:27.05 | blitzrage | justnulling2: could have sworn I said it calm... |
| 23:30.37 | fujin_ | I've got a redirection loop going on and can't kill a channel - any reason why? |
| 23:30.38 | fujin_ | 1. Local/601@maxnet-default-2698,2 (wait: 16:23, prio: 0) |
| 23:30.44 | fujin_ | soft hangup Local/601@maxnet-default-2698,2 |
| 23:30.57 | justnulling2 | lucky me, anyways lets try to enable the debug stuff and see if that helps |
| 23:31.06 | bhima | How different are the Digium PCI cards compared to others with the same chipsets? (I intend to buy from Digium because I would like to support them for their work on Asterisk, but somebody was telling me they thought the Digium cards would actually perform better, and I was wondering if this was actually the case) |
| 23:31.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:31.37 | JT | bhima: compared to what? |
| 23:31.41 | fujin_ | I'd just like to kick that call out of the queue |
| 23:31.44 | fujin_ | the channel has already closed |
| 23:32.45 | bhima | JT: T1 cards, I mean. I thought I'd seen some others that claimed full compatibility. |
| 23:34.36 | Dovid | if i put in a card in to a box but do not install the dirvers is there any way to see if the card is in there ? |
| 23:34.42 | Dovid | like cat /pric/? |
| 23:34.46 | Dovid | cat /proc/? |
| 23:34.50 | twisted | haha |
| 23:35.04 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:35.24 | twisted | cat: /pric/: No such file or directory |
| 23:35.29 | twisted | lol |
| 23:35.38 | twisted | i should rename /proc to /pric |
| 23:35.43 | twisted | +k |
| 23:35.46 | bhima | cat: /prick/: No such file or directory |
| 23:35.55 | Dovid | i corrected myself ;) |
| 23:36.11 | *** join/#asterisk saftsack (n=oliver@pD9E057C5.dip.t-dialin.net) |
| 23:36.22 | twisted | but yes, the answer to your question: |
| 23:36.27 | twisted | lspci |
| 23:36.28 | JT | bhima: well it's a poorly kept secret the sangoma cards often have less compatibility/interrupt issues |
| 23:36.56 | Dovid | thanks |
| 23:37.32 | Dovid | twisted: i dont have an lspci |
| 23:37.59 | twisted | Dovid: oh? are you root? |
| 23:38.09 | tzafrir_home | lspci does not require root |
| 23:38.21 | tzafrir_home | install pciutils |
| 23:38.29 | twisted | tzafrir_home i know, but in some distros, like fedora, it's not in the path |
| 23:38.34 | twisted | so the easiest way to direct someone is to be root |
| 23:39.00 | tzafrir_home | lspci in sbin/ ? What's the point? |
| 23:39.19 | Dovid | tzarir: I am usint CentOS: I have pciutils installed on the box |
| 23:39.29 | twisted | why don't you ask the distro maker rather than me :) |
| 23:39.35 | Dovid | but lspci is not in /proc. would it be any where else |
| 23:39.41 | twisted | oh |
| 23:39.42 | twisted | nono |
| 23:39.42 | Dovid | jus twondering if ya knew |
| 23:39.48 | twisted | it's a command |
| 23:40.20 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 23:41.58 | Dovid | TK: when installing the PRI: Is the clock type normal or master ? |
| 23:42.18 | [TK]D-Fender | Dovid, normal |
| 23:42.29 | JT | Dovid: commands are not in /proc |
| 23:42.34 | JT | Dovid: /proc is virtual |
| 23:42.36 | JT | it's not real |
| 23:42.40 | Dovid | i know proc is virtual |
| 23:42.45 | Dovid | was wondering y the other was not there |
| 23:42.51 | JT | it's just an interface to some information brought up from the kernel |
| 23:42.52 | Dovid | TK: thanks |
| 23:42.55 | JT | Dovid: lspci is a command |
| 23:43.01 | JT | Dovid: commands are not in /proc |
| 23:44.16 | Dovid | oh ok |
| 23:44.17 | Dovid | thanks |
| 23:45.04 | JT | Dovid: updatedb; locate lspci |
| 23:45.11 | [hC] | Qwell: hey, is the g729 codec out for the appliance yet? :) |
| 23:45.39 | [TK]D-Fender | [hC], LOL! |
| 23:46.12 | tzafrir_home | well, ls /sbin/lspci /usr/sbin/lspci # if it should be in root's path |
| 23:47.22 | [hC] | [TK]D-Fender: why you laugh, tk?... why... |
| 23:47.53 | [TK]D-Fender | [hC], Gee I don't know... maybe the though of that puny ass CPU trying to survive transcoding perhaps.... |
| 23:47.58 | justnulling2 | ok i see this 'app_voicemail.c:1956 sendmail: Sent mail to X@Y.Z with command '/usr/sbin/sendmail -t'' but nothings is in the mail logs and running this command from the command line (as asterisk ) works fine, any idea? |
| 23:48.08 | [TK]D-Fender | [hC], My watch is more powerful, and I have to wind it up annually! |
| 23:48.19 | twisted | lol |
| 23:48.26 | [hC] | [TK]D-Fender: speaking with qwell last time he gave me the impression they could accomplish it somehow |
| 23:48.27 | twisted | i read that as "i have to wind it up anally" |
| 23:49.04 | [hC] | the pain in the ass being that I cannot use g729 with those things due to the analog ports, they'd need to transcode... my other option to not tax the cpu of course is ulaw which is fantastic for hogging bandwidth |
| 23:49.04 | [TK]D-Fender | [hC], theres a difference between "can be done" and "will emit smoke profusely".... |
| 23:49.22 | [hC] | [TK]D-Fender: by can be done, i mean should work okay. |
| 23:49.52 | [TK]D-Fender | [hC], Yes... faith-based coding.... keep on believing :p |
| 23:50.50 | *** join/#asterisk rcphq (n=rllibre@4.160.229.201.l.sta.codetel.net.do) |
| 23:50.53 | [TK]D-Fender | dammit... whats the centos/RHEL package name for the kernel sources again? |
| 23:50.57 | *** part/#asterisk rcphq (n=rllibre@4.160.229.201.l.sta.codetel.net.do) |
| 23:50.57 | [TK]D-Fender | (el5) |
| 23:53.30 | tzafrir_home | kernel source? or for building something vs. the kernel? |
| 23:53.45 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 23:53.47 | [TK]D-Fender | tzafrir_home, Here's where I'm at : http://pastebin.com/m698c9989 |
| 23:54.12 | [TK]D-Fender | tzafrir_home, I'm jsut feeling silly again... I hate working on incomplete installs... |
| 23:54.26 | tzafrir_home | [TK]D-Fender, ./install_prereq test |
| 23:54.34 | outtolunc | cute |
| 23:55.32 | [TK]D-Fender | tzafrir_home, progress... thanks |
| 23:55.44 | outtolunc | yeah who needs that .1.4 anyways |
| 23:56.17 | justnulling2 | bilzrage: ok from the debug looks like asterisk is not using /dev/null but sendmail -t but mail.log doesn't show anything |
| 23:59.30 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 23:59.56 | Dovid | TK: a few mote |
| 23:59.58 | Dovid | more* |
