| 00:00.19 | ZX81 | :-) I'm off to my New Years party in a couple of hours - will be gone until the 3rd |
| 00:00.45 | *** join/#asterisk aday (~aday@belldandy.aday.net.au) |
| 00:00.51 | *** part/#asterisk aday (~aday@belldandy.aday.net.au) |
| 00:00.52 | Strom_TM | thats one hell of a party |
| 00:00.55 | PTG123 | bkw: do you k now enough about unions to know how they occupy memory? |
| 00:01.05 | *** join/#asterisk aday (~aday@belldandy.aday.net.au) |
| 00:01.34 | PatrickDK | unions are exactly what they say |
| 00:01.42 | PatrickDK | variables inside an union overlap |
| 00:01.50 | PatrickDK | so you can convert from one type to another |
| 00:01.59 | PatrickDK | or just use the same memory space for two different things |
| 00:02.02 | ZX81 | Strom_TM: :-) |
| 00:02.03 | ZX81 | yup |
| 00:02.17 | ZX81 | I play a live psytrance set |
| 00:02.19 | ZX81 | :-) |
| 00:02.21 | PTG123 | ok so 3 structs in a union, occupie the amount of memory of the largest struct? |
| 00:02.29 | PatrickDK | yep |
| 00:02.42 | PTG123 | ok thats what i figured, now let me see how i can convert it to pascal :) |
| 00:02.59 | PatrickDK | hmm, I don't think pascal has unions |
| 00:03.06 | Ridgeback | i have just cvs'd asterisk and it's working just fine, iax and sip to 3 other *'s |
| 00:03.06 | PatrickDK | been awhile since I last coded in pascal though |
| 00:03.08 | *** join/#asterisk dano_ (~dano@buggs.crosscountrycourier.com) |
| 00:03.17 | PTG123 | yah they don't :) |
| 00:03.19 | PTG123 | which is the issue |
| 00:03.20 | JerJer | exit stage left |
| 00:03.31 | *** join/#asterisk SexyKen (~sexyken@64-121-7-74.c3-0.snmt-ubr2.sfrn-snmt.ca.cable.rcn.com) |
| 00:03.33 | SexyKen | Dec 28 18:04:23 NOTICE[13379]: Registration from '<sip:921@66.55.69.242>' failed for '24.191.236.37' |
| 00:03.37 | SexyKen | Anyone know what this means? |
| 00:03.38 | PTG123 | which is why i need to know how they occupy memory, trying to use iaxclient in delphi |
| 00:03.55 | dano_ | How can I configure a Dial application to not use a new line in my Zap group, but use the *current* line (for a flash transfer to a legacy PBX)? |
| 00:04.04 | Ridgeback | SexyKen, ummmm failed registration? ;) (probably a bad password or login name) |
| 00:04.09 | triplefff | so theres no way to crank the volume of darn .gsm files ? |
| 00:04.19 | SexyKen | Okie Dokie...anyway to tell what login or password they're trying to use? |
| 00:04.35 | Ridgeback | triplefff, sure there is. any decent audio program should. linux or windows? |
| 00:05.05 | triplefff | well i dont really want to process allmotored call |
| 00:05.09 | Ridgeback | SexyKen, yes you could use ethereal to monitor the SIP packets, not the RTP stream and it will break out the whole packet with login details etc.... |
| 00:05.47 | *** join/#asterisk slak2 (~alex@dsl093-236-218.hfd1.dsl.speakeasy.net) |
| 00:05.48 | slak2 | hi |
| 00:05.49 | SexyKen | Well asterisk is running on the linux server...what would I use then? |
| 00:06.07 | slak2 | ]can someone help me, my ptsn outgoing calls are too quiet |
| 00:06.09 | slak2 | the volume |
| 00:06.18 | slak2 | people keep complaining that they cannot hear |
| 00:06.24 | slak2 | i use a tdm04b digium card |
| 00:06.27 | Ridgeback | you can run ethereal in X windows and trap port 5060 on the *'s ethernet port. or use tethereal if your on a console |
| 00:06.28 | slak2 | tx/rx gains set to 0.0 |
| 00:06.31 | mishehu | slak2: use ztmonitor and adjust your tx/rx gains |
| 00:06.38 | jsharp | set your txgain to something higher. |
| 00:06.48 | slak2 | i thought higher = quiet |
| 00:06.49 | slak2 | er |
| 00:06.53 | mofu | slak2: boost the gains? |
| 00:06.54 | SexyKen | Thanks Ridgeback, I'll check it out real quick. |
| 00:06.58 | Ridgeback | slak2, turn up the txgain in zapata.conf |
| 00:07.03 | Ridgeback | SexyKen, np :) |
| 00:07.03 | slak2 | Ridgeback, to what? |
| 00:07.11 | SexyKen | •Ridgeback• Isn't the password sent in MD5 though? |
| 00:07.26 | Ridgeback | slak2, what ever you want... crank it up, test it and see what happens |
| 00:07.44 | Ridgeback | SexyKen, if you chose md5, then yes. you could switch it to plaintext for testing |
| 00:08.27 | Ridgeback | SexyKen, also you can look at the MD5 hash and run md5sum < password and see if they match |
| 00:09.06 | SexyKen | Ah, gotcha. |
| 00:09.22 | *** join/#asterisk t[a]inted (~point@12.31.187.130) |
| 00:09.39 | Ridgeback | chix dig Asterisk! |
| 00:10.00 | mishehu | astervoipphonesex |
| 00:10.14 | Ridgeback | heck yeah IAXSEX |
| 00:10.20 | ZX81 | ~adn |
| 00:10.22 | jbot | somebody said adn was the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 00:10.26 | hermie | Ridgeback/Sexyken - md5 authentication uses a randomly generated secret that is concatinated to the password and then summed... |
| 00:10.43 | Ridgeback | hermie, oh crap ok then guess hash comparisons wont work |
| 00:11.05 | hermie | Ridgeback: right, otherwise you'd just be sending a plaintext password |
| 00:11.06 | Ridgeback | SexyKen, might have to switch to plain text and check it out |
| 00:11.21 | zigman | g723 is * included right ? |
| 00:11.22 | zigman | <PROTECTED> |
| 00:11.27 | jsharp | exten => xxx,1,Playback(keeptalking). |
| 00:11.34 | jsharp | You can't transcode g723. |
| 00:11.38 | jsharp | Only pass through. |
| 00:11.45 | zigman | you can't |
| 00:11.47 | zigman | ? |
| 00:11.52 | jsharp | Not with *. |
| 00:11.57 | zigman | i thought 729 was the paycodec |
| 00:12.03 | jsharp | It is. |
| 00:12.11 | Nugget | https://slacker.com/~nugget/asterisk9.php <-- was this clever or was this dumb and misguided? |
| 00:12.29 | zigman | trange |
| 00:12.31 | zigman | strange |
| 00:13.04 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 00:14.29 | Chotaire | eyebeam rules. |
| 00:14.47 | Nugget | oh yeah? I've been eager to meet someone who uses it. |
| 00:14.48 | zigman | Dec 29 01:12:39 NOTICE[1805]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
| 00:14.49 | zigman | nice |
| 00:14.56 | Nugget | is the acoustinc echo cancelling in eyebeam effective? |
| 00:15.12 | Ridgeback | whats eyebeam? |
| 00:15.24 | Chotaire | I dunno, I have own echo cancellation on asterisk. |
| 00:15.25 | Nugget | eyebeam is the premium softphone by the people who make x-lite |
| 00:15.40 | Chotaire | a 6 person meetme was fine, also internal conferencing works good. |
| 00:15.43 | Druken | xten |
| 00:15.43 | Chotaire | no echo whatsoever. |
| 00:15.44 | Nugget | Chotaire: well, I mean the mic/speaker echo cancelling, which isn't really addressed by asterisk. |
| 00:16.04 | Chotaire | hm, I think I can only try this when using a desktop speaker. |
| 00:16.06 | Nugget | I've got many cow-orkers who use softphones without headsets and it's annoying. I'll buy them eyebeam if it helps. |
| 00:16.09 | Chotaire | because I never had that problem anyway. |
| 00:16.23 | dano_ | how via the dialplan (or AGI) can I do a flash hook transfer using zap channels? |
| 00:16.38 | Nugget | I have no way to test it myself. |
| 00:16.45 | Strom_TM | dano_, it should just work |
| 00:16.51 | Chotaire | one thing you never do though is put multiple lines in one meetme and then press the conference button. |
| 00:16.52 | Strom_TM | flash, dial other extension, hang up. |
| 00:16.53 | Chotaire | weeeeeeeeee |
| 00:16.58 | Nugget | hah |
| 00:17.07 | Chotaire | nugget: I will try that out. |
| 00:17.08 | Druken | dano_: i don't think you can do it automatically |
| 00:17.10 | Nugget | cool, thanks |
| 00:17.16 | dano_ | strom_tm: i have a group of 2 zap channels. * tries to bridge the next available channel. |
| 00:17.18 | Druken | dano_: why would you want to? |
| 00:17.33 | dano_ | druken: I am integrating a * box as a voicemail box with a legacy PBX. |
| 00:17.43 | Druken | ok, and? |
| 00:17.50 | tzanger | dano_: which PBX |
| 00:17.52 | dano_ | if someone hits "0" right now they tie up both zap channels until done. |
| 00:17.52 | Chotaire | you can also give me a call and see for yourself. |
| 00:17.58 | dano_ | tzanger: avaya magix |
| 00:18.03 | tzanger | dano_: ahh |
| 00:18.11 | *** join/#asterisk C11- (~jonas@p50837419.dip0.t-ipconnect.de) |
| 00:18.14 | mikegrb | dano_: you want to have asterisk send flash to the pbx? |
| 00:18.24 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
| 00:18.24 | mikegrb | there is a code to do that |
| 00:18.29 | mikegrb | don't remember it off hand |
| 00:18.45 | dano_ | mikegrb: yep - if I run a Flash() it works, but then if Dial(...) is the next command, it bridges the lines. |
| 00:18.59 | mikegrb | oh not to sure then |
| 00:19.14 | mikegrb | there is a touch tone command too |
| 00:19.19 | mikegrb | like flash then *0 or something |
| 00:19.28 | dano_ | ohhh |
| 00:19.51 | mikegrb | check the listing of vertical access codes for zap |
| 00:19.54 | mikegrb | got to head out |
| 00:19.55 | mikegrb | good luck |
| 00:19.58 | dano_ | l8r thanks |
| 00:20.23 | dano_ | that won't work. crap. That's only if I'm *calling* the * box. *0 on the line will *0 the pbx. |
| 00:20.34 | macTijn | <- off to bed |
| 00:21.26 | *** join/#asterisk D1ng0 (~dingo@130.205.8.67.cfl.rr.com) |
| 00:23.18 | echinos | well, I can call an extension to hear the date and time, but not much past that. |
| 00:23.48 | dano_ | is it "hook flash" or "flash hook"? |
| 00:24.23 | *** join/#asterisk SexyKen (~sexyken@64-121-7-74.c3-0.snmt-ubr2.sfrn-snmt.ca.cable.rcn.com) |
| 00:25.18 | tzafrir | anybody here uses latest spandsp with stable * ? |
| 00:25.47 | zigman | me |
| 00:25.48 | zigman | pre6 |
| 00:25.49 | Chotaire | re |
| 00:26.09 | Moc | I run spandsp with head |
| 00:26.36 | SexyKen | Hm |
| 00:26.43 | Bile_One | Whew 3 tries to fix this and on the third update it works! |
| 00:27.16 | SexyKen | Okay so most of the users i've setup are working fine with Asterisk...though there is one user who can hear people talk but we can't hear him over the phone |
| 00:27.18 | SexyKen | Any ideas? |
| 00:27.31 | Strom_TM | how is he connecting in? |
| 00:27.40 | tzafrir | OK, I'll give it a shot... |
| 00:27.44 | SexyKen | sip |
| 00:27.47 | ctooley | when Allison records prompts, what format are the audio files provided in? |
| 00:29.05 | Strom_TM | SexyKen, what kind of equipment does he have at his end? |
| 00:29.14 | Strom_TM | is it an actual telephone set or is it a softphone |
| 00:29.44 | SexyKen | It's an actual phone |
| 00:29.46 | SexyKen | A polycom ip 600 |
| 00:30.13 | zigman | i got an optipoint 600 from siemens |
| 00:30.17 | zigman | nice phone |
| 00:30.26 | SexyKen | cool |
| 00:30.26 | zigman | only problem is.. i can't turn off the backlight :) |
| 00:30.46 | ctooley | zigman, at least it has one. |
| 00:30.46 | zigman | which kinda suxx |
| 00:30.55 | SexyKen | My phone doesn't have one |
| 00:31.42 | mofu | SexyKen: could it be a NAT problem? |
| 00:32.01 | Blackvel | gotta run |
| 00:32.02 | Blackvel | cu |
| 00:32.29 | SexyKen | •mofu• The user is setup to be dynamic and be used behind nat. |
| 00:32.33 | doolph | what would be the best codec order, I want to do allow=all |
| 00:32.46 | soundguy | im sooooo close |
| 00:32.50 | Strom_TM | doolph, use ulaw |
| 00:32.53 | soundguy | how do I fix a 403 Forbidden message? |
| 00:32.56 | Strom_TM | ulaw ulaw ulaw, and ulaw |
| 00:33.15 | soundguy | When I try to call my grandstream phone from xlite on my desktop I get that message, the phone rings but when i pick it up it is engaged |
| 00:33.27 | doolph | how much kbs does ulaw use |
| 00:33.29 | mofu | SexyKen: port forwarding correctly set on this router? |
| 00:33.29 | soundguy | However, I can call form grandstream to xlite fine |
| 00:33.53 | jsharp | ulaw is 64kbs + overhead. |
| 00:33.59 | doolph | ack |
| 00:34.18 | doolph | that means that dialup users cant use my service |
| 00:34.35 | redder86 | doolph: don't use ULAW |
| 00:34.39 | soundguy | Does anyone have a clue about my problem? |
| 00:35.32 | redder86 | doolph: use something like ilbc and make sure the dialup users don't download anything at the same time as they use VoIP |
| 00:35.36 | jsharp | You'll need ilbc or speex to get it to work worth a crap over dialiup. |
| 00:36.06 | Strom_TM | doolph, let those who can use ulaw use it. they'll be much happier with the quality |
| 00:37.50 | SexyKen | mofu - He doesn't have port forwarding setup...I didn't think he was going to need it....I dont need it on my phone. |
| 00:38.09 | firestrm | i need just 1 fxs circut, what is the best option for a cheap card? anyone? |
| 00:38.29 | Hogie | digium x100p :) |
| 00:38.33 | Hogie | er |
| 00:38.34 | Hogie | fxs |
| 00:38.35 | Hogie | hmm |
| 00:38.37 | blaisen1 | this may be a stupid question but is anyone working on soft-v.90 emulation for Asterisk over PRIs? |
| 00:39.01 | Chotaire | blaisen1: what you need that for? |
| 00:39.01 | blaisen1 | grandstream handytone 286 |
| 00:39.20 | firestrm | Hogie, ya i went with a x100p clone for my fxo $29.00 but i need a fxs |
| 00:39.23 | jsharp | v.90 is still buried in the septic tank of patents. |
| 00:39.25 | Druken | blaisen1: soft v.90 ? |
| 00:39.50 | SexyKen | Actuallyl, mofo I just found out people can't hear me on my phone either. |
| 00:39.51 | blaisen1 | chotaire: well i have 3 pris running on two cisco as5200s... i want to take the last PRI, make it bidirectional (so i can call out) and have an asterisk box hooked to that (since i don't expect large voice volumes) but I still want to use it for my customers to dial in to |
| 00:39.52 | jsharp | quicknet phonejack, maybe? |
| 00:40.02 | dano_ | Druken: I got it. Way easy. |
| 00:40.22 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
| 00:40.26 | mofu | Hogie: tdm400 or external ATA are the only choices I think |
| 00:40.44 | firestrm | phonejack is almost as expensive as a tdm400 |
| 00:40.53 | blaisen1 | .... the other option would be two T1 cards, one that goes to the telco and one that goes to the Cisco AS5200 and then just have asterisk act as an intermediary |
| 00:41.04 | jsharp | That would be your best solution. |
| 00:41.15 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:41.17 | wolfson | blaisen1: you should be able to route calls differently in the as5200 based on the called #, route them to mica modems if its a data did, route via sip if its a voice did |
| 00:41.22 | blaisen1 | jsharp: has anyone does it? does it work? |
| 00:41.32 | jsharp | I've done it. |
| 00:41.34 | blaisen1 | wolfson: its a 5200 not a 5300 |
| 00:41.37 | echinos | In the * CLI, when I call an extension, I see SIP/xx-e179 and SIP/xx-2e4c and such. ? |
| 00:41.39 | SexyKen | Anyone know why people can't hear us when we talk over our VOIP phones? |
| 00:41.43 | *** join/#asterisk mshades (~mshades@69.146.147.83) |
| 00:41.44 | echinos | Whassat? |
| 00:41.53 | wolfson | i've only dealt with the 5300 |
| 00:41.55 | jsharp | Just have to make sure you set your Dial command to use the low-latency direct bridge. |
| 00:42.01 | echinos | (the extra 4 hex chars)? |
| 00:42.01 | blaisen1 | jsharp: callers still get the same speed v.90 connects and all that? |
| 00:42.02 | jsharp | from PRI port to PRI port. |
| 00:42.03 | *** join/#asterisk JerJer (~jj@65.173.197.109) |
| 00:42.22 | firestrm | SexyKen, usualy its a codec incompatibilty problem |
| 00:42.24 | jsharp | Yep. I never saw a problem. |
| 00:42.25 | pfn | echinos just unique identifiers for the channel name |
| 00:42.39 | blaisen1 | jsharp: cool |
| 00:43.05 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:43.25 | echinos | pfn: They change for each call from that SIP extension, is that normal? |
| 00:43.40 | echinos | (If they are supposed to be registered?) |
| 00:43.48 | Milligan | Is there anyone here with experience in Cisco Universal Gatways... I am getting -- Got SIP response 500 "Internal Server Error" back from 10.1.1.109 on one gateway and the other is working fine.... |
| 00:44.05 | Milligan | The configurations are the same, I just can not figure out what this could be.... |
| 00:44.29 | jsharp | Reload the one that's acting up? |
| 00:44.43 | Milligan | jsharp: Yea I did that too... |
| 00:44.57 | blaisen1 | anybody here using livevoip? |
| 00:45.09 | Milligan | jsharp: even upgraded to the same IOS image as the one that is working correctly |
| 00:45.17 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:45.48 | triplefff | anyone played with dialconfirm ? |
| 00:46.07 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:46.48 | Milligan | jsharp: I know that the request is getting to the access server because the sip-ua statistics are increasing.... |
| 00:47.09 | SexyKen | firestrm what codec should I be using for voip phones? |
| 00:47.19 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:47.53 | dudewhere | Can anyone please tell ne how to fix the following error?: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. |
| 00:47.58 | ctooley | I'm playing around in the sounds directory and I'm finding files like this. I found some files like: vm-tocallnum.gsm. It almost sounds you could call back a number that left a voicemail? |
| 00:48.16 | *** join/#asterisk angler (~angler@24.214.255.242) |
| 00:49.03 | t[a]inted | what does (sic) mean exactly |
| 00:49.11 | Milligan | msg file Do you know anyone that is experienced with Cisco AS5400 Universal Gatways? |
| 00:49.29 | *** join/#asterisk CaMeLL (~dave@alb-66-65-202-87.nycap.rr.com) |
| 00:49.49 | Strom_TM | : intentionally so written -- used after a printed word or passage to indicate that it is intended exactly as printed or to indicate that it exactly reproduces an original <said he seed [sic] it all> |
| 00:50.00 | triplefff | if some one know dialconfirm please pm me |
| 00:50.03 | Bile_One | Thank you channel * for all the help. Happy New Year to you all. |
| 00:50.28 | *** part/#asterisk Bile_One (Bile_One@170.94.90.40) |
| 00:50.40 | cryzeck | Where can i find information and manuals on creating DTMF menus? |
| 00:51.17 | Milligan | I know this is not alot of help, but there is a decent section in the handbook |
| 00:51.18 | t[a]inted | Strom_TM great! and ... |
| 00:51.33 | *** join/#asterisk iguy-duex (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 00:51.46 | Milligan | cryzeck: It seemed pretty straight forward |
| 00:52.30 | *** part/#asterisk dano_ (~dano@buggs.crosscountrycourier.com) |
| 00:52.52 | cryzeck | Milligan: okay :) |
| 00:52.54 | cryzeck | Thanks |
| 00:53.11 | Milligan | cryzeck: no problem... Anytime... |
| 00:56.48 | blaisen1 | voipjet just making clicking sounds on attempted calls.. thats progress from dead air though |
| 00:57.58 | bkw_ | 64.76.67.64. |
| 00:58.16 | pfn | echinos yes, that's normal |
| 00:58.24 | file | ooooooooh |
| 01:00.40 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:02.58 | *** join/#asterisk Yoda-BZH (~yoda-bzh@pro.game.admin.at.yoda-bzh.net) |
| 01:04.00 | D1ng0 | bkw_, so no luck on this core issue ? |
| 01:05.18 | Druken | shhhh, i'm hunting dwwingo's.... |
| 01:05.22 | D1ng0 | heheh |
| 01:05.24 | tzanger | "Meanwhile, in another blow to the U.S.-led effort in Iraq, Uruguay announces that it intends to pull its troops out of the coalition. Informed that it has no troops in the coalition, Uruguay asks if it can borrow some." |
| 01:05.29 | tzanger | hahahaha |
| 01:05.30 | D1ng0 | im hunter why * is core dumping |
| 01:05.44 | soundguy | how hard is voicemail to setup? |
| 01:05.52 | tzanger | soundguy: very easy actually |
| 01:06.02 | soundguy | just all in voicemail.conf? |
| 01:06.03 | tzanger | check out manxpower's site for some killer macros |
| 01:06.31 | Voipbusiness | whats the link to manxpower's website |
| 01:06.40 | tzanger | ~fnords |
| 01:06.41 | tzanger | ? |
| 01:06.47 | tzanger | ~manxpower |
| 01:06.49 | jbot | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section also see http://www.voip-info.org/wiki-Asterisk also see http://www.fnords.org/~eric/asterisk/ |
| 01:06.57 | tzanger | there you go |
| 01:07.44 | D1ng0 | okay wheres bkw_ this * core dumping issue is pissing me off |
| 01:07.58 | soundguy | I cant find a voceimail docu anywhere |
| 01:08.34 | D1ng0 | why would a xten-lite softphone make asterisk core dump trying to register |
| 01:09.09 | tzanger | D1ng0: because CVS HEAD has had some *major* SIP overhauls? |
| 01:09.16 | tzanger | back up about a week or so and you should be good |
| 01:09.28 | D1ng0 | how can i get a cvs from say 1 week ago |
| 01:09.39 | tzanger | cvs co -D '1 week ago' libpri zaptel asterisk |
| 01:09.43 | tzanger | seriously |
| 01:09.45 | tzanger | cvs rocks for that |
| 01:10.18 | mikegrb | uipgrade |
| 01:10.19 | jsharp | cvs co -D 'when it worked' |
| 01:10.24 | mikegrb | at fuxzzy logiic patch |
| 01:10.37 | mikegrb | anyway, partyi at my house, champagne glass half full |
| 01:10.43 | tzanger | qute time :-) (yes spelled wrong) |
| 01:10.47 | mikegrb | time to top it off agatubn |
| 01:10.53 | tzanger | you'd run qt and it'd say "it's nearly half past four." |
| 01:11.22 | file | when it was really nearly half past two? |
| 01:11.23 | mikegrb | 20:11:17 mikegrb | fuzzyclock |
| 01:11.23 | mikegrb | 20:11:17 sighup | mikegrb: It's about ten past eight in the evening where I am. |
| 01:12.20 | D1ng0 | tzafrir, okay cool, umm you ever seen rxfax/txfax give off a GLIB 2.0 error |
| 01:12.35 | D1ng0 | tzanger, okay cool, umm you ever seen rxfax/txfax give off a GLIB 2.0 error |
| 01:12.38 | D1ng0 | sowwy |
| 01:12.53 | tzanger | D1ng0: no |
| 01:13.55 | *** join/#asterisk Maticks1 (~reece@202.147.104.114) |
| 01:14.01 | D1ng0 | bahhh |
| 01:14.03 | D1ng0 | :) |
| 01:14.08 | Maticks1 | does anyone know what codec SLIN is ? |
| 01:14.27 | Maticks1 | i see IAX2 using it as a format and the payload looks really good for the quality |
| 01:14.40 | jsharp | slin = signed linear |
| 01:15.08 | *** join/#asterisk jcims (~chatzilla@cpe-69-135-121-57.columbus.rr.com) |
| 01:15.14 | jsharp | raw, uncompressed audio samples. |
| 01:15.22 | Maticks1 | hrmm |
| 01:15.43 | Maticks1 | that might explain why its got virtually no latency |
| 01:16.16 | Maticks1 | also why i couldnt find any information in regards to slin as a codec :) |
| 01:16.17 | Maticks1 | thanks |
| 01:16.18 | Maticks1 | :) |
| 01:16.49 | Chotaire | anyone know how many concurrent incoming calls I can have on iaxtel? |
| 01:17.12 | jcims | is it a complicated setup to have * acting as SIP proxy client to two different systems (* and stanaphone) and route inbound calls from both to a single extension? kind of like having two phone lines to different telcos? |
| 01:17.37 | jsharp | Not at all. |
| 01:17.44 | jcims | (actually, add a third, local telco) |
| 01:18.06 | jsharp | I've got 3 providers and a local telco on my home * system. |
| 01:18.07 | jcims | are they all extensions in this case? or do i 'register' to the two sip proxies |
| 01:18.28 | Maticks1 | has anyone heard anymore about ss7 and asterisk intergation |
| 01:19.00 | jcims | i'm just kind of swimming in the soup right now and cant get my head around how to tie them together... |
| 01:19.06 | jsharp | that depends on the proxies. You'll probably register to them. |
| 01:19.31 | *** join/#asterisk khb (~phoenix@ool-43516e13.dyn.optonline.net) |
| 01:19.44 | jcims | can i do inbound _and_ outbound that way? there was some note to use two extensions if you're going to do inbound and outbound (peer and user) vs. a single 'friend' type |
| 01:20.06 | jcims | are these even extensions? |
| 01:20.19 | jsharp | No, those aren't extensions. Those are just user profiles. |
| 01:20.23 | jcims | my local sjphone sip connection to my local * box is running like a champ |
| 01:22.45 | jcims | so do i configure the * client connections to the remote sip proxies as 'register' statements in the [general] section of sip.conf? |
| 01:24.50 | *** join/#asterisk QRPartner (~tenikiwon@ns1.accu-com.com) |
| 01:24.55 | QRPartner | Hello Hello |
| 01:25.31 | QRPartner | Anyone know of known issues with calling a pager? It seems to dial but never connect. Yet it doesn't Fail. This only happens with certain pagers |
| 01:28.08 | PatrickDK | hmm, pages are 1200baud modems |
| 01:30.15 | *** join/#asterisk crimd_ (~crimd@66-65-89-219.nyc.rr.com) |
| 01:30.23 | soundguy | what port(s) do u need to open on the firwall to let other people connect to the asterisk sip server? |
| 01:30.42 | PatrickDK | soundguy, ALOT of them |
| 01:30.47 | PatrickDK | 5060 for sip |
| 01:30.58 | soundguy | is there a list somewhere? tcp or udp or both? |
| 01:31.05 | PatrickDK | then if you actually want audio to work, the rtp ports you have defined |
| 01:31.08 | PatrickDK | udp only |
| 01:31.09 | mikegrb | and other ports for rtp |
| 01:31.19 | soundguy | how do i found out the rtp ports? |
| 01:31.19 | PatrickDK | default rtp in asterisk is 10000-20000 |
| 01:31.25 | PatrickDK | rtp.conf :) |
| 01:31.30 | D1ng0 | soundguy, check the wiki for firewalls it give rules for most firewall types right there |
| 01:31.31 | soundguy | ok, coz i havnt changed anything with that ;) |
| 01:31.58 | PatrickDK | you need two rtp ports per phone call |
| 01:32.12 | PatrickDK | at a min |
| 01:32.37 | soundguy | i'll just open 10000-20000 ;) |
| 01:33.34 | progcaribou | does anybody know about this switch: Alcatel OmniAccess 512 VoIP ? |
| 01:33.43 | progcaribou | is it any good? |
| 01:36.20 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |