| 00:02.10 | Strom_L | what do you mean "roll over"? |
| 00:04.32 | hsv-al | hello fellow irc addicts |
| 00:04.41 | hsv-al | are we all looking forward to another long & glorious night of irc/internet addiction? :) |
| 00:04.54 | Strom_C | i am not an addict |
| 00:05.04 | Strom_C | NO MY CONNECTION IS DOWN HELP [MADLY PRESSING BUTTONS] |
| 00:05.17 | hsv-al | im about to throw up hahah |
| 00:05.17 | *** join/#asterisk saftsack (n=saftsack@p54A7E3D2.dip.t-dialin.net) |
| 00:05.22 | hsv-al | 7 mile run, then dinner is a five guys burger |
| 00:05.32 | Strom_C | "five guys burger"? |
| 00:05.42 | hsv-al | http://www.fiveguys.com/ |
| 00:06.11 | Strom_C | i'll have to check that out the next time i'm in washington dc |
| 00:06.16 | hsv-al | i give em credit, they were playing this on the speakers in there while eating |
| 00:06.16 | hsv-al | http://youtube.com/watch?v=qSfndZfKFNU |
| 00:06.29 | hsv-al | they play alot of 80's music on their speakers |
| 00:08.26 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 00:08.49 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 00:10.25 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 00:10.26 | hsv-al | strom_c, I got some flack 3 days ago from some cisco management people hahahaha |
| 00:10.26 | Strom_C | ironically, that's one of the singles I /don't/ have in my crates |
| 00:10.30 | Strom_C | why? |
| 00:10.35 | hsv-al | i started this facebook group like 1.8 years ago when i was pursuing ccvp cert |
| 00:10.37 | hsv-al | before I gave up |
| 00:10.43 | hsv-al | and i put a asterisk link up |
| 00:10.44 | hsv-al | http://www.facebook.com/group.php?gid=2348029702 |
| 00:10.47 | hsv-al | 2,400+ members |
| 00:10.59 | hsv-al | im gonna screenshot their message and put it on imageshack.us |
| 00:11.32 | hsv-al | they're essentially offering $5,000 for admin rights to the group |
| 00:11.37 | hsv-al | so i dont spread "asterisk propaganda" |
| 00:11.42 | hsv-al | is what they said essentially HEH |
| 00:12.07 | nick125 | haha |
| 00:12.26 | Strom_C | lool. |
| 00:12.33 | hsv-al | look at the glaring bright orange |
| 00:12.34 | rickross | anyone having any difficulty getting asterisk-addons to build from the latest svn? |
| 00:12.36 | hsv-al | link in the cisco dedicated group |
| 00:12.38 | hsv-al | ahahhaha |
| 00:12.42 | hsv-al | asterisk logo |
| 00:12.44 | hsv-al | THRAWWNG |
| 00:12.49 | rickross | chan_ooh323.c: In function âooh323_newâ: |
| 00:12.49 | rickross | chan_ooh323.c:257: error: invalid operands to binary & |
| 00:12.52 | nick125 | If you don't spread Asterisk, someone else will :) |
| 00:18.00 | hsv-al | heres the moron that sent the harassing letter |
| 00:18.01 | hsv-al | http://www.linkedin.com/in/lasandrabrill |
| 00:18.08 | hsv-al | "Manager, Web & Social Media Marketing at Cisco" |
| 00:18.55 | hsv-al | http://lasandrabrill.blogspot.com/ |
| 00:19.06 | hsv-al | if she sent this type of crap, they must see some value in the group to feel * is a threat heh |
| 00:19.34 | Strom_C | hsv-al: of course cisco sees asterisk as a threat |
| 00:19.44 | Strom_C | it has huge potential to cannibalize their sales of callmanager |
| 00:20.08 | hsv-al | I have two options |
| 00:20.16 | hsv-al | purge teh group, or sell out and let em have reins in it |
| 00:20.35 | nick125 | hsv-al: What if you stand your ground? |
| 00:20.56 | hsv-al | i dont think they can do anything |
| 00:21.01 | hsv-al | i mean its a group I created almost 2 years ago |
| 00:21.05 | hsv-al | when FB was barely starting off |
| 00:21.10 | saftsack | im interested in pri cards. why is there a opinion that digium cards are crap? |
| 00:21.45 | Strom_C | saftsack: zealotry and bad information |
| 00:21.56 | Strom_C | there were some problems years ago, but not anymore |
| 00:23.49 | hsv-al | strom, im waiting for this to come in |
| 00:23.50 | hsv-al | http://pastebin.com/m76d51317 |
| 00:23.56 | saftsack | Strom_C, so it is very cheap in my opinion to build its own PRI <-> SIP gateway |
| 00:24.38 | Strom_C | saftsack: uh, assuming I'm interpreting your sentence correctly...yes |
| 00:25.26 | saftsack | Strom_C, do you think a p3 500mhz is capable in doing this? (no codec translation) just g711a to g711a |
| 00:25.45 | Strom_C | how many T1 spans are you planning on? |
| 00:26.17 | saftsack | 1xE1 with 30 channels. |
| 00:26.19 | hsv-al | hahahahahah |
| 00:26.22 | hsv-al | they must be fuming now |
| 00:26.25 | hsv-al | double whammy |
| 00:26.27 | hsv-al | http://www.facebook.com/group.php?gid=2348029702 |
| 00:26.31 | Strom_C | hsv-al: shut up, please. |
| 00:26.56 | hsv-al | saftsack, hello |
| 00:27.02 | saftsack | hi |
| 00:27.21 | Strom_C | saftsack: you should be OK, I think...I ran a 933mhz P3 with a TDM400 as my pbx for a couple of years, though i never tried it with a T1 |
| 00:27.39 | saftsack | a tdm has 4 channels ... ^^ |
| 00:27.54 | Strom_C | yeah, but i handled plenty of IP calls too |
| 00:27.59 | *** part/#asterisk galeras (n=galeras@190.26.191.166) |
| 00:28.37 | hsv-al | saftsack |
| 00:28.39 | hsv-al | how much ram |
| 00:28.44 | saftsack | 256mb |
| 00:28.48 | hsv-al | sdram? |
| 00:28.53 | saftsack | yes |
| 00:29.06 | hsv-al | seems low, but i guess it'll do |
| 00:29.21 | hsv-al | I have a p4 3.2ghz 2gig pc3200 Ive ran an on for half a year no hitches |
| 00:29.52 | nick125 | It might be worth it to pick up a stick of 512MB or two. |
| 00:31.02 | LiNeTuX | Strom_L: Sorry for the late reply... by 'roll over' I mean if you've got a couple of PRI's (23 channels/ea) and all 23 are in use in the 1st PRI, will the next incoming call go to the 2nd PRI? |
| 00:31.21 | Strom_C | LiNeTuX: you can tell the telco to do that, yes |
| 00:31.31 | saftsack | nick125, not possible. i want to use the card in this environment http://www.pcengines.ch/alix1c.htm |
| 00:31.39 | truent | . |
| 00:31.44 | LiNeTuX | Strom_C: so that's a function of the telco. Got it. |
| 00:32.12 | nick125 | saftsack: Ah, that's a pain. What about using one of those mini ITX VIA boards? |
| 00:32.50 | saftsack | nick125, they are bigger ;) |
| 00:33.13 | hsv-al | saftsack |
| 00:33.23 | hsv-al | these days you can probably get a system 3-4x more optimal then that |
| 00:33.28 | hsv-al | and probably not more then $300-$400 |
| 00:33.49 | nick125 | Even if the VIA boards are marginally larger, they're faster and you can put more RAM into them. |
| 00:33.50 | saftsack | do you have a recommend? |
| 00:34.08 | saftsack | nick125, thats true but it's JUST for this task |
| 00:34.22 | saftsack | just mapping 30s2m ports to a lan cable. not more |
| 00:34.24 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 00:35.09 | hsv-al | sack |
| 00:35.10 | hsv-al | http://www.dell.com/content/products/features.aspx/inspndt_bundles?c=us&cs=19&l=en&s=dhs |
| 00:35.49 | hsv-al | Intelr CoreT2 Q9300 Quad-Core (6MB L2 cache,2.5GHz,1333FSB) |
| 00:35.51 | hsv-al | $719 - base |
| 00:35.53 | hsv-al | thats hot |
| 00:36.26 | saftsack | burns tons of energy every second |
| 00:36.29 | nick125 | Here's a 1.2GHz Jetway board (not including memory, but memory is cheap) for $174 at mini-itx. |
| 00:37.00 | nick125 | 1.5GHz for around the same price |
| 00:37.48 | nick125 | (USD) |
| 00:37.55 | nick125 | http://www.mini-itx.com/store/?c=2#j7f4 |
| 00:38.30 | saftsack | that seems to be nice. but my question was if it would be possible to run these 30 s2m channels on those alix boards |
| 00:38.30 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5e3821430c93e367) |
| 00:38.36 | hsv-al | 1.2GHz C7 Eden CPU |
| 00:38.43 | hsv-al | ? is that intel or amd? |
| 00:39.00 | nick125 | hsv-al: That's VIA. |
| 00:39.19 | hsv-al | never heard of em, personally I wouldnt touch that, but if you have dealt with them, who knows they may be good |
| 00:39.28 | hsv-al | well cpu's they make.. |
| 00:39.38 | saftsack | i have an alix |
| 00:39.48 | saftsack | tomorrow i will generate 30 concurrent calls |
| 00:39.58 | nick125 | saftsack: That's probably the best idea. |
| 00:40.12 | nick125 | Then you know if the board will work or not. |
| 00:40.42 | saftsack | but the pain is that i need a pmx connection but here in germany you pay 280⬠for having it monthly. |
| 00:41.02 | saftsack | there is no test pmx with just 2 enabled b-channels ... you can get just 30 :( |
| 00:41.59 | hsv-al | 30 concurrent calls on a p3 500mhz, 256ram?? |
| 00:42.04 | saftsack | yes |
| 00:42.16 | hsv-al | what codec are you using |
| 00:42.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bcf2683edf8ff011) |
| 00:42.21 | saftsack | should be no problem if you ask me if theres no codec translation |
| 00:42.25 | saftsack | g711a on both sides |
| 00:42.38 | hsv-al | hwec? |
| 00:42.53 | saftsack | what else? ^^ |
| 00:43.34 | saftsack | at this dollar course i would pay about 700⬠for a complete PRI gateway |
| 00:44.01 | saftsack | with hardware ec |
| 00:44.03 | hsv-al | so what is this, two offices connected over frame t1? |
| 00:44.16 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 00:44.18 | saftsack | hsv-al, are you talking to me? |
| 00:44.22 | hsv-al | yes, |
| 00:44.31 | ZX81 | hi all anyone from xorcom here? |
| 00:44.32 | hsv-al | 30call on ancient machine seems like may give headaches. |
| 00:45.03 | saftsack | hsv-al, no connection from an office to the telephone net |
| 00:45.05 | ZX81 | have 3 x 32 Xorcom channel bank thingies and am getting pretty crap sound quality |
| 00:45.14 | ZX81 | just fxs to dialplan |
| 00:45.41 | saftsack | hsv-al, why? the pc has just to unpack the voice from the sip packages and throwing them on the pci bus |
| 00:45.55 | saftsack | nothing more .... should be no problem for those ancient machines |
| 00:47.35 | saftsack | so i go to bed now. thanks for the nice discussion |
| 00:54.16 | *** join/#asterisk ManxPower (n=manxpowe@126.sub-75-202-184.myvzw.com) |
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| 01:18.32 | TJNII | Is there a way to get * to not play hold music when it recieves a "You are on hold" message? |
| 01:20.09 | outtolunc | setmusiconhold(somenonexistantclass) |
| 01:22.33 | TJNII | WARNING[23800]: res_musiconhold.c:646 get_mohbyname: Music on Hold class 'nomohkthxbai' not found |
| 01:22.33 | TJNII | <PROTECTED> |
| 01:22.37 | TJNII | No dice. |
| 01:23.05 | outtolunc | how about disable your default hmmm |
| 01:23.36 | outtolunc | then only 'defined' non-default fall-thru crap only classes would work |
| 01:25.18 | outtolunc | or just create a class that points to a non-existant file to playback |
| 01:25.45 | TJNII | No default worked |
| 01:26.04 | TJNII | Let me try a class with no files |
| 01:30.04 | TJNII | Class with no files doesn't work |
| 01:30.30 | outtolunc | use mode=files, directory=/exists/but/empty/dir |
| 01:30.51 | TJNII | Doesn't work |
| 01:31.00 | TJNII | Just uses a working class |
| 01:31.14 | ManxPower | TJNII: How about not setting an MoH class in the first place |
| 01:31.14 | outtolunc | paste the error, i want to see that |
| 01:31.31 | TJNII | ManxPower: Well, I want MOH |
| 01:31.58 | TJNII | But my ITSP's server sends a "You are on hold message" and no "You're no longer on hold message" when I call them |
| 01:32.11 | TJNII | So whenever I call my ITSP I get put on hold forever. |
| 01:32.26 | TJNII | So in this one special case, I don't want MOH. By default I do. |
| 01:32.28 | ManxPower | TJNII: Then set it in the dialplan for calls where you want MoH |
| 01:33.01 | ManxPower | TJNII: I HIGHLY doubt not setting MoH will do any good. |
| 01:33.04 | TJNII | Yes, that was the solution stated above that worked. |
| 01:33.12 | ManxPower | hold == one-way-audio, IIRC |
| 01:33.17 | TJNII | No default causes MOH to fail and the call works. |
| 01:37.59 | outtolunc | notes placing an empty file such as silence/9.gsm or just pointing your new [silence] class at /usr/lib/asterisk/sounds/silence (if you have one) will work also |
| 01:38.47 | TJNII | No, I don't want a silence, I need * to not try and play MOH |
| 01:39.11 | TJNII | Otherwise it is playing MOH when I should be listening to the call. |
| 01:39.16 | outtolunc | so do a stopmusiconhold prior to that in the dialplan |
| 01:39.28 | Strom_C | TJNII: do they continue to send audio? |
| 01:39.38 | TJNII | Yes |
| 01:39.52 | Strom_C | sounds like they're a whole barrel of stupid |
| 01:39.59 | TJNII | They never send whatever sip message says "You are no longer on hold," AFAIK |
| 01:40.00 | Strom_C | maybe you should switch ITSPs :/ |
| 01:40.05 | TJNII | Strom_C: No argument |
| 01:40.30 | TJNII | Otherwise, though, I really have no complaints. |
| 01:41.12 | TJNII | outtolunc: The dialplan really doesn't help at that point because it is within a dial() command. |
| 01:41.13 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 01:45.49 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 01:47.45 | outtolunc | is beginning to wonder how using 'class' as a var name ever happened <G> |
| 01:50.15 | deeperror | dahdi? |
| 01:52.21 | NovceGuru | yes son |
| 01:53.55 | deeperror | ha exactly |
| 01:57.27 | wonderworld | i have got a digium card, attached to a E1 line. i can start outgoing calls without problems. when trying to make an inbound call i just get "number not available" message from my telco. what might be the cause for that? i created a s extension in the context for the E1 card but calls seem to never reach asterisk. i am not sure if it might be the hardware-setup like wrong cabeling or problems at the telco or problems with my setup. the card |
| 01:57.28 | wonderworld | doesn't get the telco side as timing source, but i set that option in the span in zaptel.conf. any ideas? tnx a lot. |
| 01:59.04 | deeperror | do you see anything in cli when the call comes in? |
| 01:59.17 | wonderworld | no nothing. i set verbosity to 10. |
| 02:01.37 | znoG | does Asterisk 1.4 still use mpg123 for Music On Hold or is it native .wav? |
| 02:05.48 | Strom_C | znoG: you can still use mpg123 although asterisk has support for native moh now |
| 02:06.40 | znoG | Strom_C: yep, cool! native it is .. thanks :) |
| 02:13.07 | TJNII | Aah, the mark of true customer service. Keep you on hold for 10 minutes and then disconnect you. |
| 02:14.28 | deeperror | Its the best, keeps costs lower |
| 02:15.12 | LiNeTuX | fewer calls to answer = less employees. good customers are the ones who buy and never call. |
| 02:15.39 | LiNeTuX | So TJNII, I guess that means you're part of the problem for that company ;) |
| 02:15.57 | TJNII | I suppose...,. |
| 02:16.23 | TJNII | Though I was calling the "new customer" number to ask about transferring a phone number. |
| 02:16.35 | wonderworld | i called today. |
| 02:16.39 | wonderworld | i am a bad guy |
| 02:16.48 | TJNII | Shame on us |
| 02:17.00 | LiNeTuX | Yeah, see, you can't transfer. Transferring means losing business. That's no good. |
| 02:17.01 | TJNII | Harassing the companies whom we pay for services |
| 02:17.36 | deeperror | att is the worst though with that |
| 02:18.25 | deeperror | 1000 departments each knowing less than the previous but more than whoever you are talking to at the moment all of which are experts in closing calls and masters of the transfer |
| 02:20.13 | wonderworld | german telecom used to ignore billing complains completely. the only way for a private person to get money back was to sue them. i heared that they improved but i avoided them whenever i could. |
| 02:20.26 | deeperror | had a tech call them they said he needed to purchase more "trunks"....he was trying to get a number forwarded haha wtf |
| 02:22.07 | deeperror | att hates us and is trying to bill us huge with cancellation fees cause we have basically just up and left them as we are 100% sip now |
| 02:23.21 | ManxPower | did you sign a contract when you got the ATT service? |
| 02:23.26 | TJNII | http://www.qwest.com/smallbusiness/products/voip/index.html |
| 02:23.32 | deeperror | they signed 1 year |
| 02:23.36 | TJNII | VoIP service starting at $625 a month! |
| 02:23.47 | TJNII | s/625/685/ |
| 02:23.50 | ManxPower | and are you breaking your contract? |
| 02:23.57 | *** join/#asterisk Swat2 (n=bler@218-214-169-112.people.net.au) |
| 02:24.14 | ManxPower | Don't worry, you'll go back to some ATT lines after your first major VoIP outage. |
| 02:24.24 | Swat2 | is there any way i can make my phones tune into a online radio station stream when theres no phone call? |
| 02:24.43 | deeperror | that is up next month but we went from 300g to nothing haha that wasn't in the contract but they charged us 20g's for some minimum usage obligation and there was no words in the contract about that so it's hard to fight through the ivr's to get them to remove that and after 15 days it goes right to their collections goons...ha |
| 02:25.04 | ManxPower | deeperror: that's what lawyers are for. |
| 02:25.09 | deeperror | yep they are on it |
| 02:25.10 | ManxPower | and/or your sales rep |
| 02:25.18 | deeperror | haha our sales rep is no longer with them she was fired |
| 02:25.30 | drmessano | majorfail seems like a better nick for you |
| 02:25.38 | drmessano | Let me check it's availability |
| 02:25.48 | deeperror | me? |
| 02:25.54 | ManxPower | we go thru about 5 sales reps per year. At least now we qualify as a large/complex customer, so we get more stable sales reps |
| 02:25.57 | drmessano | Yes, "majorfail" is open |
| 02:26.06 | deeperror | haha |
| 02:26.22 | deeperror | we will fail when the internet fails |
| 02:26.32 | deeperror | so any day now |
| 02:28.07 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 02:28.39 | deeperror | as long as a backhoe doesn't run over our fiber i think were pretty good for now |
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| 02:33.29 | ManxPower | or a dispute between carriers (it's happened before), or a problem at the ITSP, or the ITSP's ISP, or any ISP between you and them. |
| 02:33.38 | ManxPower | or a local loop problem. |
| 02:33.57 | deeperror | yea i'm aware of it all |
| 02:34.12 | deeperror | but we need both internet and phones |
| 02:34.16 | deeperror | so we lose one were down anyways |
| 02:34.24 | drmessano | we're |
| 02:35.05 | deeperror | too late to get bogged down with grammer |
| 02:35.08 | deeperror | haha |
| 02:35.20 | deeperror | and spelln |
| 02:36.19 | drmessano | spelling |
| 02:36.46 | deeperror | thanks |
| 02:37.04 | deeperror | missed grammar |
| 02:37.44 | drmessano | Wasn't paying that much attention.. just looking at the failure from afar |
| 02:38.18 | deeperror | majorfail |
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| 02:59.59 | BeeBuu | drop agent when i press * ,how to fix this? |
| 03:01.38 | ManxPower | BeeBuu: why do you need to do this? |
| 03:01.41 | drmessano | Don't press * |
| 03:01.44 | ManxPower | Agents should not be calling IVRs |
| 03:02.19 | BeeBuu | drmessano: i need to transfer~~~ |
| 03:03.23 | BeeBuu | drmessano: wolud you teach me how to fix the problem? |
| 03:03.46 | drmessano | BeeBuu: Good luck with it |
| 03:04.26 | BeeBuu | no way? |
| 03:05.26 | BeeBuu | so a agent can not transsfer? |
| 03:06.16 | deeperror | check features.conf? |
| 03:07.11 | BeeBuu | deeperror: have you try? |
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| 03:32.07 | marc7 | anyone familiar with ATA's that support T.38? |
| 03:32.33 | jaytee | I've used GS HandyTone 286 |
| 03:32.38 | jaytee | worked for me |
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| 04:02.23 | colulu | is there anyway to tell my spa 3102 to detect hangup from the other side? |
| 04:02.38 | colulu | Is this a known bug with Linksys SPA 3102? |
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| 04:05.55 | hsv-al | far-end disconnect issue w/ 3102? |
| 04:07.03 | colulu | hsv-al: yeah |
| 04:07.21 | colulu | hsv-al: is there any known solution for this? |
| 04:07.43 | colulu | hsv-al: it inability to detect farend disconnects screw up my entire system |
| 04:08.54 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:09.23 | hsv-al | http://voipusers.org.nz/pipermail/users/2007-May/003185.html |
| 04:09.38 | hsv-al | http://www.google.com/search?hl=en&q=SPA-3102+far-end |
| 04:10.12 | hsv-al | http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3000-spa3102-disconnect-tone-21529.html |
| 04:11.08 | hsv-al | http://forum.voxilla.com/linksys-sipura-voip-support-forum/translating-your-disconnect-tone-12874.html |
| 04:11.47 | colulu | hsv-al: thanks alot for your help |
| 04:15.43 | hsv-al | get the pdf |
| 04:15.48 | hsv-al | i think its on page 70'ish |
| 04:15.53 | hsv-al | 73,74,75, etc |
| 04:16.02 | hsv-al | some tidbits that will help |
| 04:19.34 | Maliuta | anyone familiar with cisco 79xx firmware upgrades? I want to go from 7.3 to 8.6 and just want to know if there are issues going directly from 1 to t'other or if I need some odd upgrade path |
| 04:23.21 | colulu | hsv-al: which pdf is that? |
| 04:24.24 | Maliuta | ~thebook |
| 04:24.28 | 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 |
| 04:24.37 | Maliuta | that pdf |
| 04:26.22 | colulu | oh. THE BOOK |
| 04:26.43 | nick125 | *The* book. |
| 04:26.48 | Maliuta | no, _The_ Book |
| 04:26.57 | Maliuta | we don't SCREAM in here |
| 04:27.03 | nick125 | haha |
| 04:27.27 | Maliuta | most of us come with a clue too |
| 04:27.47 | colulu | ;) |
| 04:27.49 | Maliuta | and a nice big L.A.R.T. |
| 04:32.43 | *** join/#asterisk HighOctane (n=HighOcta@68-185-143-114.dhcp.jcsn.tn.charter.com) |
| 04:33.18 | HighOctane | Anyone here ever use fonosip |
| 04:35.29 | *** join/#asterisk ftoledo (n=ragnarok@201.250.49.67) |
| 04:36.23 | ftoledo | hi all.. im switching from extension.conf to extension.ael , how to use the hint priority in the ael file? i know hot is it on the extension.conf but i dont found an example |
| 04:36.57 | *** part/#asterisk ftoledo (n=ragnarok@201.250.49.67) |
| 04:37.11 | *** join/#asterisk ftoledo (n=ragnarok@201.250.49.67) |
| 04:38.28 | ftoledo | i found that in extension.conf is 100 => hint,SIP/100 for example. |
| 04:38.29 | codefreeze-lap | ftoledo: you put hint(<spec>) before the extension name... |
| 04:39.27 | codefreeze-lap | so: context c1 { |
| 04:39.27 | codefreeze-lap | <PROTECTED> |
| 04:39.27 | codefreeze-lap | <PROTECTED> |
| 04:40.14 | codefreeze-lap | see http://voip-info.org/wiki/view/Asgerisk+AEL2#Extensions |
| 04:40.30 | codefreeze-lap | s/Asgerisk/Asterisk/ |
| 04:42.20 | ftoledo | codefreeze-lap: good , i now reading |
| 04:42.39 | ftoledo | codefreeze-lap: im usign asterisk 1.2 with AEL not AEL2 , is it aplicable too? |
| 04:48.35 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:49.25 | ftoledo | codefreeze-lap: http://pastebin.proyectoalba.cl/?page=view&id=1212554940 |
| 04:49.38 | ftoledo | codefreeze-lap: in using a macro to dial my extensions |
| 04:50.34 | ftoledo | codefreeze-lap: the hint must be before the "_10XX" extension on my "internos" context? |
| 04:52.31 | codefreeze-lap | No, youre hint extensions can't be patterns. You'll have to define a set of fixed extensions with hints. |
| 04:53.38 | ftoledo | codefreeze-lap: ok, i understand, |
| 04:57.56 | HighOctane | is there any way to see what IP address a particular IP call is coming from in asterisk? |
| 04:59.47 | ftoledo | codefreeze-lap: im put : hint(Sip/1) 1010 => { |
| 04:59.48 | ftoledo | <PROTECTED> |
| 04:59.48 | ftoledo | <PROTECTED> |
| 05:00.20 | ftoledo | but i think that dont work with AEL, i see that is supported in AEL2, true? |
| 05:17.10 | *** join/#asterisk jmazaredo (n=IceChat7@mail.agilys.com.ph) |
| 05:18.13 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 05:21.07 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 05:22.00 | jmazaredo | join #networking |
| 05:22.04 | jmazaredo | oopes |
| 05:31.34 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-239-103.balt.east.verizon.net) |
| 05:58.57 | *** join/#asterisk keulin (n=cray@AMontsouris-158-1-6-236.w90-46.abo.wanadoo.fr) |
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| 06:13.48 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 06:22.38 | drmessano | I wasn't aware the SPA-3102 has issues with far end hangups |
| 06:23.04 | drmessano | If it's configgure correctly, it won't |
| 06:32.24 | unpaidbill | wooooooow pocketsphinx is niiiiiice |
| 06:32.34 | unpaidbill | i had to use a lot of vowels to show you just how nice it is. |
| 06:33.18 | toresbe | That's a lot of vowel! |
| 06:33.24 | unpaidbill | and it's a lot of nice! |
| 06:33.36 | toresbe | It's clear that you do not have a vowel obstruction. |
| 06:33.45 | unpaidbill | nope, i abuse them like redheaded step children |
| 06:33.50 | toresbe | If anything, you have irritating vowel syndrome. |
| 06:33.57 | unpaidbill | haha |
| 06:34.02 | toresbe | :) |
| 06:34.06 | unpaidbill | me and kelsey grammars wife can hang |
| 06:34.13 | toresbe | hahaha |
| 06:34.22 | toresbe | looks "pocketspinhx" up |
| 06:34.33 | *** part/#asterisk jmazaredo (n=IceChat7@mail.agilys.com.ph) |
| 06:34.49 | unpaidbill | voice recognition stuff.. i used to use sphinx2, but pocketsphinx is supposedly faster and has better accuracy |
| 06:34.50 | toresbe | oh, neato. |
| 06:35.21 | unpaidbill | and uses less cpu! |
| 06:35.43 | toresbe | I'll actually try that. |
| 06:36.32 | unpaidbill | if you do.. you'll want to check out pulseaudio, i used that to connect asterisk and pocketsphinx, they dont interface directly |
| 06:36.51 | unpaidbill | i used this howto as a baseline http://www.nabble.com/Misterhouse-Asterisk-%2B-Pocketsphinx-HOWTO-td16800602.html |
| 06:37.00 | unpaidbill | jim duda rocks. |
| 06:38.01 | unpaidbill | this is the easier way in case you get pissed off with pocketsphinx http://www.voip-info.org/wiki-Sphinx |
| 06:38.42 | *** join/#asterisk seldev (i=test@firewall.selmoni.ch) |
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| 07:21.26 | *** join/#asterisk dickyjoe (n=lrguy78@183.69.233.220.exetel.com.au) |
| 07:21.47 | dickyjoe | Hi All |
| 07:22.53 | dickyjoe | I am after some help with zaptel. I run asterisk 1.4.20.1 with zaptel 1.4.10.1 |
| 07:23.26 | dickyjoe | I run a TDM400P card with 1 fxo and 1 fxs. I cannot set opermode=AUSTRALIA |
| 07:23.44 | dickyjoe | has something changed in the setup? I run asterisk over Centos 5 |
| 07:36.58 | Maliuta | loadzone=au defaultzone=au |
| 07:37.07 | Maliuta | what do you need opermode for? |
| 07:38.21 | colulu | does anyone here have experience with vicidial |
| 07:38.47 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-64162596ffab1f84) |
| 07:46.03 | krdian | is back. "Linux: The choice of a GNU generation" |
| 07:46.14 | krdian | hi |
| 07:48.01 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 07:49.51 | krdian | is linksys spa922 good choice for small office ? or you can recommend something better ? |
| 07:50.35 | drmessano | Are you going to use PoE? |
| 07:51.02 | krdian | drmessano: not realy |
| 07:51.29 | drmessano | I would get an SPA 941 |
| 07:51.51 | drmessano | More features, and the xx1 models come with power supplies |
| 07:51.52 | gr0mit | i have not had good experiences with SPA phones |
| 07:52.06 | gr0mit | i do not think the sending audio quality is very good |
| 07:52.07 | drmessano | the xx2 do not have ... |
| 07:52.17 | drmessano | Alrighty then.. |
| 07:52.54 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:53.25 | gr0mit | krdian, I would suggest a Snom or a Thomson 2030 |
| 07:53.45 | drmessano | Snom and Thomson suck |
| 07:53.56 | drmessano | I would use a Linksys before either of those |
| 07:54.08 | krdian | hmmmmm |
| 07:54.10 | drmessano | Snom has crappy tech support |
| 07:54.13 | gr0mit | nah. i have a customer with linksys |
| 07:54.19 | gr0mit | they are problematic |
| 07:54.31 | drmessano | Yah... The Snom and Thomson suck |
| 07:54.42 | drmessano | Linksys seem to work much better |
| 07:54.45 | gr0mit | i have another cusomer with Thomson phones and they are delighted |
| 07:54.50 | drmessano | Well, good for them |
| 07:55.34 | gr0mit | drmessano, can you explain the vacuum caused by Snom and Thomson? |
| 07:55.40 | dickyjoe | Hi Maliuta |
| 07:55.46 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 07:56.29 | Maliuta | that's my nick. don't wear it out |
| 07:56.31 | dickyjoe | I beleive opermode sets the TDM400 card to australia ring cadences, etc |
| 07:56.45 | Maliuta | you don't need it |
| 07:56.50 | drmessano | gr0mit: Not gonna get into a pissing contest over phones with you.. krdian asked about the Linksys, and you decided to jump in with a Linksys rant, for which I really get sick of arguing about |
| 07:56.52 | Maliuta | trust me, I have one |
| 07:57.16 | drmessano | So, krdian, use whatever the fanboys tell you works this week.. if you want a good $90 phone, I love the Linksys SPA941s |
| 07:57.21 | Maliuta | dickyjoe: you can set your own cadences, it's fun |
| 07:58.47 | krdian | drmessano: err, 941 has one rj45 port, i need to connect pc thhrough that |
| 07:59.24 | drmessano | krdian: The 942 and 922 are PoE, you'll need power supplies for them |
| 07:59.43 | gr0mit | drmessano, I am genuinely interested in your views! I am not ranting - I have just had bad experiences with the Linksys sending audio quality. |
| 07:59.59 | gr0mit | they may have solved the issues in later phones? |
| 08:00.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:00.14 | Maliuta | dickyjoe: loadzone should also do this for you |
| 08:00.21 | dickyjoe | iok |
| 08:00.33 | krdian | drmessano: no worries, i can use there switch with poe |
| 08:00.39 | drmessano | gr0mit: In general, no one in is interested in views on phones in here unless it's polycoms... In my personal opinion, the SPA941s sound as good as the Cisco 7960s we use at work |
| 08:01.06 | dickyjoe | in the instructions that came with the tdm, they say opermode=australia is needed to make it work with the australian phone system |
| 08:01.39 | gr0mit | well, I have not yet used polycoms - are they to be recommended or avoided? And as for Ciscos, they sound superb but are a pain to confiure! |
| 08:02.03 | nick125 | is waiting for his 7940G to arrive |
| 08:02.57 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 08:03.31 | drmessano | gr0mit: The Ciscos are good sounding phones.. overpriced and overly complicated.. but a good benchmark of audio quality.. The Polycoms are fine, but in 90% of the cases in here, if you're not using a Polycom phone, you're an idiot.. hence my somewhat blunt and argumentative approach to the convo |
| 08:04.22 | drmessano | Sorry for being so blunt, but it doesn't matter in here what works... only if it rhymes with "wollycom" :) |
| 08:04.45 | gr0mit | drmessano, i take it there are a lot of polly-thusiasts here? |
| 08:05.01 | drmessano | That's putting it lightly |
| 08:05.33 | Maliuta | dickyjoe: out of interest where were you trying to put the opermode? |
| 08:06.37 | drmessano | But in case, before I have to listen for the next three days to how much of an asshole I am in here, I will leave my rant at that. |
| 08:07.11 | dickyjoe | maliuta: when loading wctdm in modprobe.conf you need to pass opermode=australia |
| 08:08.13 | gr0mit | drmessano, ok - thanks for the warning ;-) |
| 08:08.39 | Maliuta | dickyjoe: in /etc/modutils/zaptel -> options wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1 |
| 08:09.13 | dickyjoe | soemthing like that, only problem is it doesnt wotk |
| 08:09.16 | dickyjoe | work |
| 08:09.27 | Maliuta | how can you tell? |
| 08:09.40 | dickyjoe | by running ztcfg -vv |
| 08:10.03 | dickyjoe | i was wondering if there is a log file |
| 08:10.32 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:10.53 | Maliuta | dickyjoe: turn on debug info at the same time |
| 08:11.18 | Maliuta | it should error out if it can't load the tone zone |
| 08:11.29 | dickyjoe | how do i do that |
| 08:12.05 | Maliuta | by setting the flag in the same options line |
| 08:12.14 | dickyjoe | ok |
| 08:12.22 | dickyjoe | debug=on? |
| 08:17.07 | tzafrir_laptop | Maliuta, do you actually use kernel 2.4? |
| 08:17.17 | tzafrir_laptop | if not: /etc/modutils is obsolete |
| 08:17.34 | tzafrir_laptop | (and has no effect) |
| 08:18.29 | tzafrir_laptop | Maliuta, use /etc/modprobe.d/zaptel instead for a 2.6 kernel |
| 08:19.30 | tzafrir_laptop | dickyjoe, if you use a non-ancient (read: 2.6) kernel, you should be able to see the parameters set to the modules under |
| 08:19.39 | tzafrir_laptop | <PROTECTED> |
| 08:19.55 | dickyjoe | ok |
| 08:20.06 | tzafrir_laptop | What kernel version do you use? |
| 08:20.18 | dickyjoe | i use centos 5 which is 2.6 based but built over redhat |
| 08:21.06 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:21.38 | dickyjoe | what do you reckon tzafrir? |
| 08:26.22 | tzafrir_laptop | what's the problem? |
| 08:26.47 | dickyjoe | I said I had 2.6... |
| 08:27.55 | unpaidbill | oh holy fuck, i set up pocketsphinx in super hardmode... |
| 08:28.00 | tzafrir_laptop | So put those parameters in /etc/modprobe.d/zaptel |
| 08:28.09 | dickyjoe | ok then |
| 08:28.15 | tzafrir_laptop | (or actually any other file under /etc/modprobe.d ) |
| 08:28.38 | dickyjoe | will give it a go now... |
| 08:28.48 | dickyjoe | thanks for the help, i am a bit of a linux noob |
| 08:38.11 | tzafrir_laptop | dickyjoe, also use lszaptel or cat /proc/zaptel/* |
| 08:38.48 | dickyjoe | what are those commands used for? |
| 08:40.13 | dickyjoe | ah i see now |
| 08:42.50 | *** join/#asterisk oej (n=olle@213.63.47.140) |
| 08:52.03 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 08:55.07 | Rico29 | in which SIP header will I be sure to find the phone's MAC address ? |
| 08:55.21 | Rico29 | i tried "from", it works with cisco but not with linksys |
| 08:55.35 | florz | Rico29: none? |
| 08:56.01 | florz | Rico29: why should a phone even have a MAC address?! |
| 08:56.26 | Rico29 | an ip phone |
| 08:56.37 | florz | exactly |
| 08:56.47 | Rico29 | an ip phone always have amac addr |
| 08:56.50 | Rico29 | a mac addr |
| 08:56.54 | florz | ip, not ethernet, or firewire, or some such |
| 08:57.01 | florz | nope, it doesn't |
| 08:57.08 | Rico29 | an ethernet phone |
| 08:57.10 | Rico29 | huh ? |
| 08:57.21 | florz | ip works with ip addresses, not with mac addresses |
| 08:57.55 | Rico29 | yes but i need to get the mac addresses of my phones |
| 08:58.00 | Rico29 | and THEY HAVE ONE |
| 08:58.02 | Rico29 | :) |
| 08:58.45 | Rico29 | you can giveme an example of an ethernet (IP...) hardphone with no mac address ? |
| 08:58.47 | florz | well, that may be - but given that an ip phone doesn't need one, I guess you can't expect any mandatory header in an IP telephony protocol that contains a MAC address |
| 08:59.00 | mgdm | an IP phone *definitely does* need a MAC address |
| 08:59.08 | florz | mgdm: no |
| 08:59.09 | mgdm | whether it's advertised in SIP headers is a different matter |
| 08:59.13 | Rico29 | florz> yes |
| 08:59.21 | mgdm | florz: if it didn't have one, it wouldn't be able to talk over Ethernet |
| 08:59.27 | Rico29 | exactly |
| 08:59.45 | Rico29 | florz> you can giveme an example of an ethernet (IP...) hardphone with no mac address ? |
| 08:59.49 | florz | mgdm: Sure, that's the whole point of IP - independence of lower-level protocol |
| 09:00.07 | florz | Rico29: of course not |
| 09:00.10 | mgdm | OK, let me rephrase that. If it has an Ethernet port, it has a MAC address. |
| 09:00.26 | florz | mgdm: well, obviously, yes |
| 09:00.36 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 09:01.16 | Rico29 | and i need tu catch this mac addr |
| 09:01.17 | Rico29 | :) |
| 09:01.25 | Rico29 | in the SIP headers |
| 09:02.46 | florz | Rico29: well, then you'll have to see whether your phone somehow does put that into some header |
| 09:02.58 | florz | Rico29: I guess a packet sniffer will help with that ... |
| 09:03.10 | dickyjoe | bye all and thanks for your help |
| 09:04.09 | Rico29 | florz> I can't sniff, i don't have wireshark or ethereal on my serv |
| 09:04.21 | Rico29 | and I dont have a hub |
| 09:04.59 | mgdm | Rico29: use tcpdump to create a dump file and copy it to somewhere that does have wireshark? |
| 09:06.58 | Rico29 | mgdm> i'll try thanks |
| 09:07.21 | Rico29 | it's a command-line application ? |
| 09:07.23 | mgdm | Looking at my Aastra phones, it appears they don't put the MAC in the header |
| 09:08.06 | mgdm | Rico29: Yep. Do something like "tcpdump -i eth0 -s0 -w output.file port 5060" <-- that'll create a file called output.file which you can load into wireshark |
| 09:08.19 | mgdm | the filter syntax is the same as wireshark's, if you're familiar with that |
| 09:08.28 | Rico29 | ok thanks a lot |
| 09:27.06 | dominic1 | hi does anybody know why I can't use G722 for my internal calls? |
| 09:27.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:27.37 | dominic1 | get this error: |
| 09:27.39 | dominic1 | <PROTECTED> |
| 09:32.39 | *** join/#asterisk BeeBuu (n=beebuu@218.13.83.123) |
| 09:33.10 | BeeBuu | how can i transfer a call in a queue ? |
| 09:35.11 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 09:38.36 | dominic1 | can I change the codecorder in the dialplan? |
| 09:41.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:42.53 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:43.49 | dominic1 | why do I need codec translation paths? |
| 09:44.04 | dominic1 | is it not possible to change the codecorder dynamically? |
| 09:48.57 | *** join/#asterisk maverick12 (n=root@83.240.229.38) |
| 09:49.08 | maverick12 | hello |
| 09:49.27 | maverick12 | can anybody tell me what are the meaning of pedantic option |
| 09:49.29 | maverick12 | <PROTECTED> |
| 09:55.03 | maverick12 | does anybody knows a way to verify the Remote Crash Vulnerability in SIP channel driver when run in pedantic mode    |
| 10:03.37 | dominic1 | anybody out there |
| 10:03.39 | dominic1 | ??? |
| 10:03.50 | dominic1 | how can I change the codecorder for internal and external calls? |
| 10:04.00 | dominic1 | SIP_CODEC didn't help |
| 10:04.09 | dominic1 | he need to change the allow= |
| 10:07.03 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 10:20.13 | tzafrir_laptop | use different peers? |
| 10:20.38 | dominic1 | peers? |
| 10:20.45 | dominic1 | both peers are snom 370 |
| 10:20.59 | dominic1 | my codecorder ist alaw;g722 |
| 10:21.45 | dominic1 | isn't it so, if I set codec to g722 that he should choose g722 on both ends |
| 10:21.53 | dominic1 | he always wants a translator from me |
| 10:22.01 | dominic1 | is that right? |
| 10:33.26 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:34.35 | maverick12 | does anybody knows a way to verify the Remote Crash Vulnerability in SIP channel driver when run in pedantic mode |
| 10:40.43 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:41.05 | *** join/#asterisk LuisTorres (n=chatzill@bl6-209-225.dsl.telepac.pt) |
| 10:47.50 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 10:49.56 | ltd | Hi all. I've just been playing with a new SIP provider and they are the kind of provider that sends a 302 on initial invite to forward to a media gateway to continue the call. After I receive a 183 Progress from the media gateway it seems asterisk is getting confused and seemingly transmitting an SLIN ring tone down a g.711u RTP stream. Anyone heard of anything like this before? |
| 10:51.16 | ltd | Getting buckets of "Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)" during the ring tone and it's garbled (naturally) |
| 10:52.39 | *** join/#asterisk logi4023 (n=logi4021@76-10-149-62.dsl.teksavvy.com) |
| 10:53.47 | logi4023 | whenever I press a keypad while in an context,extension, the call is dropped. anyone knows why this is happening? |
| 10:58.32 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:12.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:13.00 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-239-103.balt.east.verizon.net) |
| 11:17.19 | dominic1 | can I set different codec orders for internal and external calls? |
| 11:21.28 | krdian | dominic1: SIP ? |
| 11:21.47 | dominic1 | internal sip, external isdn |
| 11:22.37 | krdian | dominic1: try to use variable SIP_CODEC |
| 11:22.49 | dominic1 | did that already |
| 11:22.57 | krdian | dominic1: no success ? |
| 11:23.53 | krdian | oh, its works in ver 1.0 |
| 11:24.55 | *** join/#asterisk Floppe (i=muppet@crew.cluster-lan.org) |
| 11:25.14 | logi4023 | in my dialpan, as soon as a key is pressed on the phone, the call is hung up, even though that exten is defined. |
| 11:25.26 | krdian | is gone. lunch time |
| 11:30.57 | dominic1 | no success |
| 11:31.21 | dominic1 | he always checks the allowed codecs and uses the first one or checks for translation |
| 11:31.34 | dominic1 | but translation is not possible in ast 1.4 :-( |
| 11:31.57 | dominic1 | I think I will start crying.... |
| 11:32.11 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 11:34.44 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 11:39.06 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 11:39.14 | logi4023 | while in a ivr menu, any keypress on phone results in call hang up. anyone knows what could cause this? |
| 11:39.45 | razu | has anyone used cisco 7936 conf phone over SIP ? Is it even possible ? |
| 11:40.50 | *** join/#asterisk stunsch (n=stunsch@138.Red-83-57-10.dynamicIP.rima-tde.net) |
| 11:41.59 | stunsch | I'm having trouble when using the PICKUP function. In some scenarios the call looses one audio channel |
| 11:44.45 | stunsch | I have two types of phones. Thompson 2030 and analog phones connected to Linksys SPA8000 gateways. |
| 11:45.34 | stunsch | The problems arise when the pickup is done from different types of phones. |
| 11:46.44 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 11:47.11 | stunsch | For example, when calling between 2 analog phones, if I pick up the call from a third analog phone, no problem. But if I pick it up from a digital phone, I loose one audio channel. |
| 11:49.06 | *** join/#asterisk LiNeTuX (n=LiNeTuX@proxy.fmpa.com) |
| 11:55.52 | *** join/#asterisk Asterisk-nob (n=Nothing@117.20.21.66) |
| 11:56.18 | Asterisk-nob | what asterisk app/func i can use for parsing? |
| 12:00.48 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 12:01.06 | *** join/#asterisk blinky42 (n=steveb@64.9.15.189) |
| 12:04.17 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:09.37 | ManxPower | Asterisk-nob: parsing what? |
| 12:10.16 | ManxPower | SIP_CODEC only controls the OUTGOING leg of the call. |
| 12:10.32 | ManxPower | Also, it may have been replaced by a function in 1.4/1.6 |
| 12:11.13 | *** join/#asterisk UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
| 12:11.20 | UnixDog | morning guys |
| 12:11.39 | UnixDog | has anyone done a app_callback-request |
| 12:11.43 | ManxPower | UnixDog: Welcome to #ask-a-question-then-go-AFK |
| 12:12.18 | UnixDog | basicly a app to allow a caller to request a callback |
| 12:12.22 | UnixDog | lol |
| 12:12.44 | UnixDog | ManxPower: whats wrong budy |
| 12:12.49 | UnixDog | buddy |
| 12:14.38 | UnixDog | it comes with having to work ManxPower we all cant just stay infront of the kbd all day |
| 12:15.35 | UnixDog | and we did confirm the issue I was discussing the other day. Jerid smith rocks |
| 12:16.45 | dominic1 | hi manxpower thanks for your help |
| 12:16.49 | UnixDog | but back to the dialplan application I was looking for |
| 12:17.07 | UnixDog | app_callback-request |
| 12:17.32 | dominic1 | that's exactly that what my tests told me. I hoped that I can manualle adjust the codecs for specific lines. Internal and external calls |
| 12:18.11 | UnixDog | oyu can set the codecs on each trunk and on each user exten |
| 12:18.19 | UnixDog | to match the device |
| 12:18.30 | UnixDog | and make sure to put them in order of use |
| 12:20.42 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 12:21.13 | UnixDog | and I suggest being gsm is most commonly used for the sound files you include it for soundfile transcoding |
| 12:21.16 | dominic1 | yes, my problem ist the following: If I set the order for my devices to alaw;g722 and set in the dialplan SIP_CODEC=g722 I get the erromessage that the translation ist not possible (on the inernal call). If I set the order to g722;alaw I am able to place a internal call. If I call to external I get the error that my isdn card can not use this codec |
| 12:21.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:22.08 | dominic1 | so the system wants to use the right codec, but checks for translation before. |
| 12:22.13 | UnixDog | asterisk does not yet support g722 |
| 12:22.22 | UnixDog | that I know of |
| 12:22.28 | [hC] | sure it does |
| 12:22.32 | dominic1 | it support g722 |
| 12:22.36 | UnixDog | ok |
| 12:22.37 | dominic1 | but no transcoding |
| 12:23.11 | dominic1 | but I don't want to transcode. I want to use the codec for internal calls. If I call external asterisk should only use alaw |
| 12:23.27 | UnixDog | then its passthru like the default g729 driver |
| 12:23.30 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:23.34 | UnixDog | codec |
| 12:23.38 | UnixDog | hmmmmm |
| 12:23.45 | dominic1 | yes that's right |
| 12:24.43 | dominic1 | if it would be possible to dynamically change allow=g772;alaw or allow=alaw;g722 it should work. |
| 12:25.01 | dominic1 | I did not find out how to do that |
| 12:25.09 | UnixDog | not if it does not transcode |
| 12:25.25 | UnixDog | I recall a issue of this now |
| 12:25.34 | UnixDog | and never found a workaround |
| 12:25.43 | logi4023 | anyone know my call is dropped in the ivr whenever I press any keypad on phone? |
| 12:25.43 | UnixDog | let me look in my logs |
| 12:25.48 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:26.16 | logi4023 | pressing any keypad results in auto fallthrough error |
| 12:26.28 | dominic1 | if I install the transcoding patch it all works. He does not transcode, but just need the tranlastionpath. That's very strange. |
| 12:26.32 | UnixDog | autofallthrew=no |
| 12:26.33 | dominic1 | Thanks... |
| 12:27.01 | UnixDog | paste the patch I would like to test |
| 12:27.17 | UnixDog | I have a polycom 550 that does g722 |
| 12:27.19 | logi4023 | unixdog -- where do i put autofalthroguh=no ? |
| 12:27.43 | *** part/#asterisk AndyGraybeal (n=AndyGray@128-177-27-78.ip.openhosting.com) |
| 12:27.44 | Asterisk-nob | Hi ManxPower |
| 12:27.58 | UnixDog | hold on looking in configs |
| 12:28.01 | Asterisk-nob | nice to see you here, i'd like to ask 2 questions |
| 12:29.32 | UnixDog | extensions.con under [general] |
| 12:29.34 | dominic1 | I am not using the patch anymore, tested that yesterday with my old version of asterisk which was patched |
| 12:29.52 | UnixDog | ok |
| 12:30.05 | dominic1 | the patch was quite unstable, some scratch sounds.... |
| 12:30.06 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:30.10 | UnixDog | well then the patch needs to be updated and put into asterisk |
| 12:30.38 | UnixDog | if you have time post it I would like to look at it |
| 12:30.46 | UnixDog | brb |
| 12:30.58 | Asterisk-nob | ManxPower: actually i will recieve a called number from the web and will dial it which just works fine, but now i want i recieve something like 3-XXXXXXXXX |
| 12:31.03 | *** part/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:31.09 | Asterisk-nob | opss he is gone |
| 12:31.10 | Asterisk-nob | :S |
| 12:31.20 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:31.36 | Asterisk-nob | UnixDog: can i continue my question if you can help |
| 12:32.17 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:32.22 | dominic1 | here is the patch http://users.netplex.net/~andrew/asterisk/ |
| 12:33.22 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:34.02 | Asterisk-nob | [TK]D-Fender: hi |
| 12:34.28 | UnixDog | Asterisk-nob: you can ask your question but multi conversations go on in the channel |
| 12:34.41 | Asterisk-nob | thanks |
| 12:35.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:36.13 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 12:36.18 | hsv-al | heh |
| 12:36.22 | Asterisk-nob | actually i will recieve a called number from the web and will dial it which just works fine, but now i want i recieve something like 10-XXXXXXXXX now i wanna parse it before making the call, 10 is call limit and XXXXXXXXXX is the number |
| 12:37.10 | Asterisk-nob | after parsing i'll put them both in different variables and will make a call according to the mentioned limit |
| 12:37.28 | Asterisk-nob | but i'm not sure what asterisk app/func i should use for parsing? |
| 12:37.31 | Asterisk-nob | CUT? |
| 12:37.45 | Asterisk-nob | or any other suggested |
| 12:38.12 | UnixDog | you can set calllimit=on a user exten and a trunk |
| 12:38.28 | UnixDog | in sip.conf |
| 12:39.05 | Asterisk-nob | UnixDog: yes but there will be different calllimit for different calls |
| 12:39.07 | krdian | Asterisk-nob: yes, look at: show function CUT |
| 12:39.27 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:39.49 | UnixDog | you trying to set a timelimit on the call ? |
| 12:40.25 | Asterisk-nob | UnixDog: yes |
| 12:40.51 | Asterisk-nob | which will be recieved from the web request including number as: |
| 12:41.04 | Asterisk-nob | 10-XXXXXXXXX now i wanna parse it before making the call, 10 is call limit and XXXXXXXXXX is the number |
| 12:41.28 | Asterisk-nob | so for different web users call limit will be different |
| 12:42.14 | UnixDog | then you have to write a timer app |
| 12:42.50 | UnixDog | that monitors the call time and then hangs up the call when it reaches it limit |
| 12:43.11 | Asterisk-nob | UnixDog: what about the way i mentioned above |
| 12:43.32 | UnixDog | or states that the call time limit has expired and to call back again ltr |
| 12:44.19 | Asterisk-nob | like call limit should recived with call, and then i use CUT function and i save time limit and dialed number in different variables? |
| 12:44.41 | krdian | Asterisk-nob: TIMEOUT(absolute) |
| 12:44.53 | drmessano | UnixDog: I asked about the bug we found |
| 12:45.07 | krdian | Asterisk-nob: this variable setting up call duration |
| 12:45.15 | Asterisk-nob | krdian: no that doesn't work if the call is briged |
| 12:45.19 | hsv-al | drmessano, |
| 12:45.25 | hsv-al | looking forward to another long & glorious day on irc? :) |
| 12:45.34 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:45.36 | drmessano | hsv-al: Me, no.. work soon |
| 12:46.05 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:46.31 | Asterisk-nob | UnixDog: in the dial application L option helps for the call limit |
| 12:46.33 | Asterisk-nob | L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are |
| 12:46.33 | Asterisk-nob | <PROTECTED> |
| 12:46.33 | Asterisk-nob | <PROTECTED> |
| 12:46.37 | krdian | Asterisk-nob: r u sure ? what i remember i have used it to change duration durning a call |
| 12:46.51 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:46.51 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:47.52 | Asterisk-nob | krdian: okey but how i'll use multiple calllimits? |
| 12:48.49 | Asterisk-nob | krdian: did you try CUT function? |
| 12:48.55 | krdian | Asterisk-nob: yes |
| 12:49.31 | Asterisk-nob | krdian: can i /msg you? |
| 12:49.32 | krdian | Asterisk-nob: sure |
| 12:49.35 | Asterisk-nob | thanks |
| 12:52.10 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:54.12 | *** join/#asterisk ManxPower (n=manxpowe@52.sub-70-223-243.myvzw.com) |
| 12:56.08 | Mw3 | can somebody give me a sccp firmware for cisco 7912 ip phone? |
| 12:57.07 | *** join/#asterisk vortex` (n=vortex@deathstar.juicedigital.net) |
| 12:57.20 | ManxPower | Mw3: Not legally, no. |
| 12:57.58 | vortex` | assuming my * machine is on a local IP behind a NAT box, would I need to forward port(s) from the outside to the asterisk box to recieve SIP calls from the provider? |
| 12:58.18 | vortex` | have forwarded 5060 external to 5060 on the * box, yet no calls are recieved. |
| 12:58.48 | vortex` | sip show peers says status for the SIP provider is OK (94ms) |
| 12:59.01 | hsv-al | vortex, nice webpage |
| 12:59.02 | hsv-al | :) |
| 12:59.15 | vortex` | :) |
| 12:59.16 | hsv-al | It works! |
| 12:59.30 | vortex` | took a lot of thinking to make that :p |
| 13:00.07 | ManxPower | vortex`: you need localnet/externip set as well as forwarding the RTP AUDIO ports. |
| 13:00.08 | mgdm | vortex`: you'll need the RTP ports as well |
| 13:00.17 | mgdm | and externip, yes |
| 13:00.28 | vortex` | ok thanks. oll check the docs for those. |
| 13:00.34 | ManxPower | ~sipnat |
| 13:00.34 | 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 |
| 13:00.44 | vortex` | too easy, thanks. |
| 13:01.15 | ManxPower | vortex`: only easy if you understand networking. |
| 13:01.31 | ManxPower | Everyone should understand networking before starting Asterisk anyway. |
| 13:01.38 | vortex` | i was referring to getting the answer. making it work is a different story, clearly :) |
| 13:02.01 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 13:02.09 | ManxPower | chortles evily as he goes into full BOFH mode and updates the firewall to the config it's supposed to be. |
| 13:02.34 | ManxPower | Losers...er...Users will be getting into the office any min now to start screaming. |
| 13:02.56 | hsv-al | manxpower. to what degree is the question |
| 13:03.14 | hsv-al | not ccie routing/switching equivalent, but I would say 50% of a ccna :) |
| 13:03.50 | ManxPower | Actually understanding TCP, UDP, and NAT is a start. |
| 13:06.23 | mgdm | I've got a CCNA and to date never used any of the Cisco stuff, but the theory parts are very handy |
| 13:06.30 | hsv-al | there's primarily more phone network experts in here |
| 13:06.40 | hsv-al | then networking experts, ie: ccna,ccnp,ccie r/s |
| 13:07.13 | hsv-al | I'll be honest, alot of the phone terminology, and technology is new to me |
| 13:07.47 | ManxPower | "What codec should I use for a Zap channel" is a classic example of not understanding anything about telecom. 8-) |
| 13:08.00 | hsv-al | heh |
| 13:08.48 | hsv-al | during ccna study, I wouldnt have known there were devices that convert the digital T1 channels |
| 13:08.52 | hsv-al | into possible analog channels |
| 13:11.45 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:13.55 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 13:15.10 | tzafrir_laptop | ManxPower, chan_zap can currently only use ulaw or alaw |
| 13:15.38 | ManxPower | tzafrir_laptop: I was trying to show how uninformed the question is. |
| 13:16.30 | *** join/#asterisk l8router (n=me@122.109.125.252) |
| 13:16.56 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 13:22.00 | UnixDog | ok anyone here have dialplan for checking callerid. |
| 13:22.43 | UnixDog | I have a lay out of what I am working on but been some time since I have writen a app like this |
| 13:23.27 | igascream | Need help I'm trying to receive fax with asterisk but instead I receive an error app_rxfax.c:81 xast_log: WARNING T.30 Non-ECM carrier not found ...and it simply restart the asterisk ??????What can I try ? |
| 13:23.32 | UnixDog | and I need a way to check caller id and if it is set to pass the call and if not set to send the to a request to enter a callerid/callback number |
| 13:24.10 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
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| 13:26.00 | [TK]D-Fender | UnixDog: "core show application gotoif" , "core show function CALLERID" |
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| 13:34.38 | hi365 | where are the options that you pick in make menuselect saved? |
| 13:35.31 | errr | menuselect.makeopts iirc |
| 13:37.35 | l8router | how do i view jitter/round trip delay in CLI |
| 13:38.32 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
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| 13:41.20 | hi365 | thanks errr |
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| 13:51.02 | stunsch | I am loosing audio when I pick up phones in certain scenarios. (When mixing pick ups between Thompson 2030 phones and analog phones behind a SPA8000 Linksys gateway. |
| 13:51.13 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:51.29 | glaz | ~nat |
| 13:51.30 | jbot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 13:51.49 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:51.56 | ManxPower | stunsch: I've using Asterisk since 2001 and have never ever seen that problem SPECIFIC to parallel dialing |
| 13:52.18 | ManxPower | one way audio problems are common but they happen regardless of how you Dial |
| 13:52.41 | stunsch | jbot_ |
| 13:52.46 | maverick12 | does anybody knows a way to verify the Remote Crash Vulnerability in SIP channel driver when run in pedantic mode |
| 13:52.59 | stunsch | jbot_: Is your answer for me? |
| 13:53.18 | ManxPower | maverick12: no. try #asterisk-dev |
| 13:53.41 | stunsch | ManxPower: I suppose this is not normal, but I don't know what else to try |
| 13:54.23 | ManxPower | stunsch: Neither do I. If it happened all the time I'd suggest you look into NAT problems, but since it only happens with paralell dial, that would not be the problem |
| 13:55.24 | ManxPower | Since your problems seems to be a one-of-a-kind, I recommend you write up a problem description including what you tried and didn't work and post to the asterisk-users mailinglist. |
| 13:55.49 | stunsch | ok |
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| 14:00.51 | *** part/#asterisk maverick12 (n=root@83.240.229.38) |
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| 14:13.16 | LuisTorres | Hi |
| 14:14.58 | LuisTorres | Im setting up a iax2 trunk between to asterisk boxs.., should I use trunk=yes? or not? |
| 14:17.18 | [TK]D-Fender | LuisTorres: are you tight on bandwidth for the # of calls you are planning? |
| 14:17.32 | LuisTorres | nop |
| 14:17.40 | LuisTorres | have enought |
| 14:17.46 | [TK]D-Fender | LuisTorres: then you shouldn't use IAX period |
| 14:17.55 | LuisTorres | rgr |
| 14:18.00 | LuisTorres | thanks :) |
| 14:18.14 | LuisTorres | just SIP? |
| 14:18.18 | [TK]D-Fender | yup |
| 14:18.37 | LuisTorres | ok cool |
| 14:20.32 | jackson__ | [TK]D-Fender, I'm interested to hear why SIP is preferable to trunk with if bandwidth isn't a factor. Care to expand on it? |
| 14:20.49 | igascream | Anybody knows what cause this problem with rxfax: WARNING T.30 Non-ECM carrier not found ????? |
| 14:21.36 | *** join/#asterisk ajohnstone (n=ajohnsto@host81-137-217-96.in-addr.btopenworld.com) |
| 14:22.09 | [TK]D-Fender | jackson__: IAX has been known to encounter more audio & jitter issues, is harder to debug, and in the case of trunking 1 issue will kill ALL calls being trunked. |
| 14:22.41 | anonymouz666 | [TK]D-Fender: so you don't use/recommend IAX2? |
| 14:22.56 | ManxPower | igascream: looks like you have a bad version of spandsp or rxfax or you are trying to do FaxOverVoiceOverIP |
| 14:23.14 | [TK]D-Fender | anonymouz666: only time I recommend it is if yuo NEED it. |
| 14:23.19 | ManxPower | LuisTorres: it's not a trunk if you don't have trunk=yes, otherwise it's a peer |
| 14:23.32 | gr0mit | also, iax2 is very good at getting through firewalls |
| 14:23.43 | anonymouz666 | [TK]D-Fender: need what? audio & jitter issues? |
| 14:23.55 | ManxPower | iax2 is no different than DNS, SNMP, or Bittorrent when it comes to firewalls |
| 14:23.55 | gr0mit | this is probably its strongest selling point |
| 14:23.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:24.13 | ManxPower | gr0mit: I can't imagine why. SIP is trivial to work thru NAT and firewalls. |
| 14:24.35 | gr0mit | well it is a lot better than it used to be, ManxPower |
| 14:24.54 | gr0mit | but with IAX you only need to open the one port |
| 14:24.55 | jackson__ | ManxPower, but SIP seems more of a handfull to qos with... |
| 14:24.58 | anonymouz666 | ManxPower: I'd recommend this to read: draft-ietf-sipping-nat-scenarios-08 - Best Current Practices for NAT Traversal for SIP |
| 14:24.59 | LuisTorres | so Im on a dilame now :) |
| 14:25.24 | LuisTorres | should I Iax or should I SIp :) ? |
| 14:25.27 | gr0mit | iax was much easier to use when trying to get round voip-avoidance |
| 14:25.37 | gr0mit | LuisTorres, try both! |
| 14:25.41 | gr0mit | Learn both |
| 14:25.56 | gr0mit | then use whicever you feel happier with |
| 14:26.01 | [TK]D-Fender | anonymouz666: the only point for IAX2 is in trunk more or if there is a shit firewall in your path |
| 14:26.13 | LuisTorres | maybe its better :) |
| 14:26.35 | *** join/#asterisk ronr (n=ron@82-204-104-172.fttx.bbeyond.nl) |
| 14:26.55 | LuisTorres | which one in you oprinios are less choppy? |
| 14:27.11 | [TK]D-Fender | LuisTorres: I've already answered that. |
| 14:27.12 | ManxPower | LuisTorres: neither |
| 14:27.30 | [TK]D-Fender | LuisTorres: Feel free to try IAX2, but it'll only be a learning experience... |
| 14:27.38 | LuisTorres | :) you did |
| 14:27.44 | ManxPower | jackson__: QoS is not much different either |
| 14:28.35 | ManxPower | anonymouz666: I recommend trying it in the real world. No matter how complicated people want to make SIP+NAT, it is not complicated for the vast majority of people -- as evidenced by practically every single VoIP provider out there supporting SIP with clients behind NAT. |
| 14:30.51 | gr0mit | in most cases it just works |
| 14:31.04 | codefreeze-lap | ~AEL2 |
| 14:31.16 | codefreeze-lap | ~ael2 |
| 14:31.19 | codefreeze-lap | ~ael |
| 14:31.20 | jbot | from memory, ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 14:31.34 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 14:32.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:33.41 | mmlj4 | SIP+NAT ne trivial |
| 14:33.54 | mmlj4 | doable, sure |
| 14:34.54 | [TK]D-Fender | mmlj4: yeah, because ITSP's have such a hard time rolling out customers because of it... |
| 14:35.01 | *** join/#asterisk mintee (n=mintone@72-165-177-94.dia.static.qwest.net) |
| 14:35.27 | mintee | is there a way to encrypt the sip authenication? |
| 14:35.36 | UnixDog | md5 |
| 14:35.47 | UnixDog | authmode=md5 |
| 14:36.11 | mintee | that's as good as it gets? no des3 or anything? |
| 14:36.27 | mmlj4 | don't most of them use STUN or similar? |
| 14:36.58 | ManxPower | mmlj4: no need to |
| 14:37.00 | UnixDog | stun is a server not a uth mode |
| 14:37.23 | mmlj4 | UnixDog: 2 different threads here |
| 14:37.25 | ManxPower | mintee: you can use RSA Auth |
| 14:37.29 | ManxPower | at least for IAX2 |
| 14:37.49 | mintee | yeah, i'm looking specifically for sip |
| 14:38.00 | mmlj4 | well, right, IAX2 is just one port for both signalling and media |
| 14:38.03 | mintee | just the authenication... not the actual data itself |
| 14:38.24 | mmlj4 | SIP is one port, then some random port that the firewall's not expected to know about |
| 14:38.27 | ManxPower | mintee: Well what auth methods do the SIP RFCs support? |
| 14:38.45 | ManxPower | mmlj4: no, SIP UDP/5060 + ports as configured in rtp.conf for audio |
| 14:38.57 | *** join/#asterisk Corazu (n=Justin@bas3-toronto12-1177559202.dsl.bell.ca) |
| 14:38.59 | mmlj4 | hrm |
| 14:39.26 | Corazu | Hi all |
| 14:39.35 | anonymouz666 | anybody here has a Linksys SPA 941 around? |
| 14:39.49 | ManxPower | two ports are used per call, set the port range in rtp.conf and set up your port forwarding and externip and localnet, but that's only for Asterisk being behind NAT, for clients all you need to do is add nat=yes to the SIP peer config. |
| 14:40.00 | *** join/#asterisk gaetronik (n=gaetan@200.111.138.170) |
| 14:41.03 | gaetronik | hi all |
| 14:41.21 | UnixDog | ;Request CallBAck |
| 14:41.22 | UnixDog | ;Stage 1 Check called ID |
| 14:41.23 | UnixDog | ;Stage 2 IF called ID = Annon or not Set request caller enter a callerid/callback number |
| 14:41.23 | UnixDog | ;Stage 3 request Time for callback |
| 14:41.23 | UnixDog | ;Stage 4 Create callback file |
| 14:41.24 | UnixDog | ;Stage 5 Hangup |
| 14:41.27 | UnixDog | sorry |
| 14:41.31 | Corazu | I've got a question if anyone has experience with it. I'm looking at setting up a system using 2 phone lines and a VOIP line. And I want to use an IP-phone. If the IPPhone only has 2 lines on it, can I still connect to the other lines from the phone by dialing some sort of number (can that be configured?) |
| 14:41.32 | UnixDog | hit my wheel |
| 14:41.42 | UnixDog | shoots his mouse |
| 14:42.05 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 14:42.58 | gaetronik | i'm looking to a method to verify the order in which asterisk does the call distribution |
| 14:43.17 | [TK]D-Fender | Corazu: You do not seem to understand the concept of "lines" with regards to things like SIP phones. |
| 14:43.31 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 14:43.35 | [TK]D-Fender | Corazu: typically this implies distict "registrations" or "identities" |
| 14:43.51 | gaetronik | is anyone developed something like this in a script language |
| 14:44.03 | [TK]D-Fender | Corazu: this may have some relation to the number of simultaneous calls your phon is capable of, but that isn't implicit |
| 14:44.25 | [TK]D-Fender | gaetronik: ... huh? |
| 14:44.37 | gaetronik | [TK]D-Fender, in the case of queues and agents |
| 14:44.46 | gaetronik | i use the leastrecent distribution |
| 14:45.24 | gaetronik | and according to some persons the distribution is not really like that |
| 14:45.27 | [TK]D-Fender | gaetronik: it ranks by the agents weight and then goes in order of ... least recent |
| 14:45.41 | [TK]D-Fender | gaetronik: Why don't you go TEST and tell us? |
| 14:45.43 | gaetronik | yes but i want to be sure there is not a bug |
| 14:46.00 | gaetronik | [TK]D-Fender, in a production env with 80 agents |
| 14:46.08 | gaetronik | it's quite difficult to see |
| 14:46.24 | [TK]D-Fender | gaetronik: how does it feel? |
| 14:46.41 | gaetronik | it sounds like some agents are over used and other no |
| 14:47.14 | [TK]D-Fender | Corazu: You can process calls from your phone any which way you want. Thats what the dialplan is for. and you can send whatever calls you want to it as well. |
| 14:47.17 | ix33 | is IAXTel considered 'dead'? |
| 14:48.13 | gaetronik | leastrecent means the one which received least recent not which hangup leastrecent? |
| 14:49.28 | Corazu | [TK]D-Fender: Can other phones then connect to the same call? For example, with the phones we have now, we can pick up on "line" 1 if there is a call active, if for example I answer a support call on line 1 and need to transfer the call to another person in the office I merely hold and they pick up Line 1 on the phone. I assumed that this would work in Asterisk this way, but how would I accomplish it. Is it available as default be |
| 14:49.28 | Corazu | haviour or is that something you do in a dialplan? |
| 14:49.32 | [TK]D-Fender | gaetronik: in order of who TOOK a call last, not ended last. |
| 14:49.53 | gaetronik | ok |
| 14:49.53 | gaetronik | for me it's clear |
| 14:49.54 | gaetronik | but not for the agent's supervisor |
| 14:50.13 | [TK]D-Fender | Corazu: No, * is a not a "key system" and does not support SIP-B. There is a hack-ish attempt at SLA, good for little, but available in 1.4 |
| 14:50.36 | [TK]D-Fender | Corazu: Normally you "park" a call using a parking extension, and then pick it up from the other station. |
| 14:51.00 | [TK]D-Fender | gaetronik: well if its clear for you, then you need some communication skills to convey this to them. |
| 14:51.13 | gaetronik | they want a formal proof |
| 14:51.22 | gaetronik | they can't understand with words |
| 14:52.30 | Corazu | [TK]D-Fender: By normally do you mean in regards to a regular SIP system or * itself? If the functionality exists in * then that is what I'm looking for, if it doesn't then I'll have to find a way to get similar functionality. |
| 14:53.51 | Corazu | [TK]D-Fender: For example: call forwarding, if that's possible - or conference and hangup - which probably isn't an ideal solution. I might be missing the point here completely though. |
| 14:55.03 | jackson__ | Corazu, if you simply want to put someone on hold, then have someone else pick up that call, then you want to research call parking as [TK]D-Fender stated. |
| 14:55.23 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:55.42 | [TK]D-Fender | Corazu: Yup... mostly missing the point. I'm not sure how many SIP solutions fully implement SIP-B (SLA-like functionality). Pretty much all of us do jsut fine with parking, transfers,, etc |
| 14:55.59 | Corazu | jackson_: That's what I figured; I'm looking at that and Call Transferring, which is what I think I meant by forwarding. |
| 14:56.12 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 14:56.24 | Corazu | So basically Parking covers the functionality that I want. |
| 14:56.25 | grandpapadot | Hi all. Does anyone know how to convert a ulaw recording to wav with sox? |
| 14:56.44 | [TK]D-Fender | Corazu: you park the call, and they pick it up by dialing the lot # |
| 14:57.20 | Corazu | Alright, I'll look that up. |
| 14:57.31 | *** join/#asterisk anthm (n=anthm@adsl-69-210-52-254.dsl.milwwi.ameritech.net) |
| 14:58.07 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:58.26 | dominic1 | how can I check if command authenticate was succesful or not |
| 14:58.48 | [TK]D-Fender | dominic1: "core show application authenticate" |
| 14:59.42 | dominic1 | hi fender, okay the option j says jump to proirity 101, but in ael I have no priorities or? |
| 14:59.56 | [TK]D-Fender | dominic1: Says a lot more than that. Read it again. |
| 15:00.04 | Corazu | In regards to the actual stated lines on the phone (For example the IP Phone says 2 lines) that refers to how many calls I can have at once on that phone? So in my example I could have 2 - I'm not sure I follow on that precisely. I'm going to attempt to look up more on it, if you know any resources on it that I could start with though, that would be appreciated. |
| 15:00.24 | [TK]D-Fender | Corazu: depends on the phone. |
| 15:00.25 | dominic1 | ok, sorry I try again :-D |
| 15:01.14 | ManxPower | Corazu: IP phones DO NOT HAVE LINES |
| 15:01.16 | [TK]D-Fender | Corazu: The Polycom IP 6XX series is a 6-line phone. Taht means 6 distinct registrations, using 1 "line key" each. However each line-key can support up to 8 simultaneous calls meaning you can juggle up to 24 calls at a time off it. |
| 15:01.18 | ManxPower | They have call appearances |
| 15:01.47 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:01.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:02.06 | dominic1 | okay found out that it disconnects me if I set the wrong number for three times |
| 15:02.09 | dominic1 | thanks fender |
| 15:03.12 | [TK]D-Fender | Corazu: How many calls are you really looking to be able to handle at a time on one phone? |
| 15:05.48 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 15:06.43 | *** join/#asterisk vooduhal (n=vooduhal@static-71-127-151-243.bltmmd.fios.verizon.net) |
| 15:07.15 | vooduhal | Is there a variable or command I can use to determine if a call originated from an extension that has successfully authenticated? |
| 15:07.33 | Corazu | [TK]D-Fender: I don't think I will need more than one or 2. Is there inherent hold on the phone/* or is it handled by parking the call? |
| 15:08.03 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:08.07 | [TK]D-Fender | Corazu: For SIP phones, "hold" is a phone-based feature. being able to grab it from ANOTHER phone would mean "parking". |
| 15:08.34 | [TK]D-Fender | vooduhal: You allow un-authed calls on your system? And don't set the context to somethign different? |
| 15:09.07 | vooduhal | [TK]D-Fender: I want to force all calls through an unauthenticated context and route calls accordingly. |
| 15:09.24 | [TK]D-Fender | vooduhal: And you can use "SetVar" in your sip.conf entry for your phones to set a variable for all calls that your peers originate. |
| 15:09.37 | [TK]D-Fender | vooduhal: That'd work. |
| 15:09.38 | Corazu | [TK]D-Fender: Right, I understand that. However, if I had a support call come in from Caller A and then another one came in from Caller B - that would be the 2nd call. Would I be able to place Caller A on hold (on my phone) and pick up Caller B? Or is it handled internally by * - or depends on phone? |
| 15:09.44 | vooduhal | Sweet, thank you. |
| 15:10.11 | [TK]D-Fender | Corazu: handled on the phone. typically when you press whatever you ahve to press to take the call, it auto-holds the other one. |
| 15:11.26 | UnixDog | asterisk needs a dial plan wiki where people can post and share dial plan |
| 15:11.45 | [TK]D-Fender | UnixDog: We have a WIKI. Use it. |
| 15:11.56 | UnixDog | wich one |
| 15:12.01 | [TK]D-Fender | ~wikis |
| 15:12.02 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:12.18 | UnixDog | its loaded is to much crap |
| 15:12.27 | UnixDog | and some of the info is way outdated |
| 15:12.36 | UnixDog | it needs to be restarted fresh |
| 15:13.57 | *** join/#asterisk trythis (n=name@p54A14131.dip.t-dialin.net) |
| 15:14.20 | Corazu | [TK]D-Fender: So in regards to the number of calls on one phone it's only for the active/held calls. I don't think there's much need for more than 2, maybe 3 on phone. I suppose that if all of the 'lines' on that phone are taken up I won't be able to pick it up, and someome else can? |
| 15:14.57 | [TK]D-Fender | UnixDog: Make 1 new page on it. link it in the topic. Make sure to inspec all changes to to and pages linked to it. Then you have a whole sanitized section. |
| 15:15.15 | [TK]D-Fender | UnixDog: Just because your kitchen is messy doesn't mean your bedroom has to be as well. |
| 15:15.32 | trythis | How can I make a menu in the dialplan for a zapata device? For example a zap device calls asterisk takes the call and after that the user has some options like press 0 to make a call above VOIP or 1 to make a call to a SIP client |
| 15:15.57 | [TK]D-Fender | Corazu: Another misconception. You haven't layed out any sense of flow. And now you make a sparse comment about other phones getting "your" calls. |
| 15:16.35 | [TK]D-Fender | Corazu: If you don't answer a call (because the phone rejects for being to busy to even beep, or you jsut not answering), * carries on and does whatever you told it to do next. |
| 15:17.16 | [TK]D-Fender | trythis: Go read up on "ivr" on the WIKI and in the BOOK. |
| 15:17.17 | [TK]D-Fender | ~wikis |
| 15:17.18 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:17.19 | [TK]D-Fender | ~book |
| 15:17.20 | 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, or see ~buybook |
| 15:18.05 | trythis | ([TK]D-Fender): Thanks |
| 15:18.51 | Corazu | [Tk]D-Fender: Right, sorry, what I meant was on incoming calls - which would ring on all of the phones in the office excluding the sales' phones. Those other phones would be able to pick it up instead of me, and I wouldn't be able to pick up another call until I dropped some of my callers or parked/transferred them. |
| 15:19.19 | ManxPower | Corazu: you need to read The Good Book, you don't even enough enough to use the right terms at this point. |
| 15:19.46 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:19.46 | ix33 | Corazu: i successfully replaced a key system using call queues with station logon/off and call parking. |
| 15:20.05 | ix33 | Corazu: i got pretty much everything i needed from voip-info.org and the * readmes |
| 15:20.32 | Corazu | ManxPower: Are you referring to the O'Reilly book? |
| 15:20.36 | [TK]D-Fender | Corazu: You can pick up any calls your phone gets while still being able to accept another. And if you ring a bunch of phones at the same time, then yes any of them could answer as well. |
| 15:20.43 | ManxPower | Corazu: The Wiki is full of outdated, inaccurate, and just plain wrong information. Use it as a last resort |
| 15:21.03 | ManxPower | Corazu: yes, the O'Reily book |
| 15:21.27 | Corazu | ManxPower: How about the forums as a resource - I'd assume it has some decent information as well - in addition to the readmes. |
| 15:21.40 | ManxPower | Corazu: only newbies use the forums. |
| 15:21.44 | ManxPower | Try the Mailing List instead |
| 15:22.16 | Corazu | ManxPower: Well, for all intensive purposes - on this issue, I am a newbiew for now ;) |
| 15:22.21 | [TK]D-Fender | Corazu: For you, go get the BOOK, install * and get your hands dirty. You are looking at this from too far away |
| 15:22.40 | ManxPower | Corazu: If you want help from other newbies that known next to nothing, then feel free to use the Forums |
| 15:22.52 | Corazu | ManxPower: Good point. |
| 15:22.54 | [TK]D-Fender | Corazu: Sure I'd be cautious about spending money on stuff, but you don't need to buy anything to learn *. |
| 15:24.12 | Corazu | [TK]D-Fender: I've got the PDF downloaded, will look at it and install it on a spare box. |
| 15:24.31 | Corazu | Thanks for all the help guys, I think I've got enough to work with. |
| 15:24.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:26.22 | vooduhal | [TK]D-Fender: Is there any way for the client to set that variable in there SIP request? |
| 15:29.26 | errr | is there a way to see how long a call has been connected from cli? |
| 15:30.42 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 15:31.01 | kannan | hello all |
| 15:32.34 | tuxx- | hiya |
| 15:32.34 | tuxx- | X |
| 15:32.35 | tuxx- | :X |
| 15:32.41 | [TK]D-Fender | vooduhal: If the client can add a header, you can retreive it int he dialplan. |
| 15:32.58 | *** join/#asterisk cjk (n=cjk@vodsl-11071.vo.lu) |
| 15:33.00 | [TK]D-Fender | errr: "show channel [channel]" |
| 15:33.55 | kannan | :)\ |
| 15:34.55 | *** join/#asterisk l8router (n=me@122.109.125.252) |
| 15:36.57 | errr | [TK]D-Fender: ah thanks |
| 15:37.14 | *** join/#asterisk mike-ekim (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 15:37.33 | mike-ekim | can I generate a text file with sip.conf information as far as providers, and have the main sip.conf file include that text file? |
| 15:39.17 | [TK]D-Fender | mike-ekim: "#include myotherfile" |
| 15:39.46 | dandre | hello, |
| 15:39.51 | vooduhal | [TK]D-Fender: No, I meant, if in my sip.conf file sets a variable called AUTHED, and my dial plan logic allows outbound calls based on the value of AUTHED, can the SIP client set AUTHED in their SIP request? |
| 15:40.05 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 15:40.07 | ManxPower | vooduhal: THAT DEPENDS ON THE SIP CLIENT!!!!! |
| 15:40.15 | timeshell | Greetings. |
| 15:40.32 | ManxPower | vooduhal: SIP clients can only set headers, not variables. |
| 15:40.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:40.49 | vooduhal | Sweet. That's what I needed to know. Thank you. |
| 15:40.50 | timeshell | I recently installed oslec on my zap. It works great!! I was wondering if/when it will be included with the zap? |
| 15:41.04 | ManxPower | Also remember that each time you change sip.conf or any included files, you have to reload chan_sip.so |
| 15:41.14 | hsv-al | iax soft phones > idefisk = ? |
| 15:41.15 | timeshell | Before oslec, I had a heck of a time with lag issues. |
| 15:41.28 | mike-ekim | oh |
| 15:41.35 | mike-ekim | [TK]D-Fender: it wouldnt be like including a context? |
| 15:41.36 | ManxPower | timeshell: I imagine as soon as the OSLEC copyright holders license it to Digium |
| 15:41.45 | mike-ekim | include => file |
| 15:41.47 | dandre | How can I follow a call in cdr records? |
| 15:41.47 | dandre | For instance, I have an inbound call that is redirected to several extensions and then only one picksup the call. I have lots of cdr data for that call. How could I limit them or filter? |
| 15:41.55 | ManxPower | mike-ekim: you can't include => file |
| 15:42.02 | ManxPower | that would include the context called [file] |
| 15:42.05 | trythis | Ok now I have tested do use the demo menu in the dial plan. But nothing happens if I press 2 on my zap phone so I cant navigate in the menu. What can be the problem? |
| 15:42.23 | ManxPower | trythis: we don't know |
| 15:42.24 | mike-ekim | oh ok |
| 15:42.31 | mike-ekim | gotcha |
| 15:42.31 | trythis | hehe thanks |
| 15:42.38 | mike-ekim | so #include for files |
| 15:42.43 | mike-ekim | and include => context for context inclusion |
| 15:43.00 | ManxPower | trythis: now if you put the IVR part of your dialplan and the CLI output of a failed call onto pastebin.ca then we might be able to help you. |
| 15:43.16 | ManxPower | But *I* am not going to help you since I'm pretty sure you did not read The Good Book |
| 15:43.17 | [TK]D-Fender | mike-ekim: No. you don't include "contexts" in anything other than extensions.conf |
| 15:43.35 | [TK]D-Fender | mike-ekim: "#include myotherconfigfile" |
| 15:44.03 | hsv-al | == Spawn extension (phones, 500, 2) exited non-zero on 'SIP/1000-081f3248' |
| 15:44.11 | [TK]D-Fender | ManxPower: And here I thought not following spoon-fed instructions wouldd have been reason enough. |
| 15:44.13 | hsv-al | *'s way of stating sip connection/client = hung up? |
| 15:44.43 | RoyK | hsv-al: ${DIALSTATUS} |
| 15:45.11 | trythis | (ManxPower): The problem is that I can hear the demo but then when it says to press 2 to get more information and I do this nothing happens...in the CLI it goes to Timeout on Zap..., going to t |
| 15:45.15 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 15:48.16 | rch1 | morning folks, I've got a voicemail problem - namely it isn't loading "[Jun 4 05:38:46] ERROR[9112]: app_voicemail.c:7935 load_module: app_voicemail.so depends upon res_adsi.so |
| 15:48.28 | rch1 | which is a bit odd, considering res_adsi is loaded |
| 15:49.28 | rch1 | I'm racking my brain and can't come up with anything, can anyone point me in the right direction? |
| 15:49.33 | [TK]D-Fender | rch1: "module load res_adsi.so" |
| 15:49.43 | rch1 | yeah |
| 15:49.45 | rch1 | asterisk*CLI> module reload res_adsi |
| 15:49.45 | rch1 | <PROTECTED> |
| 15:49.45 | rch1 | <PROTECTED> |
| 15:49.59 | [TK]D-Fender | rch1: and then "module reload app_voicemail.so" |
| 15:50.16 | rch1 | asterisk*CLI> module load app_voicemail |
| 15:50.16 | rch1 | [Jun 4 05:45:47] ERROR[9161]: app_voicemail.c:7935 load_module: app_voicemail.so depends upon res_adsi.so |
| 15:50.28 | [TK]D-Fender | rch1: Something is really fubar'd. What ver of *? |
| 15:50.29 | spokra | anyone running 1.6beta9 and agi? |
| 15:50.38 | rch1 | 1.4.19 |
| 15:51.04 | rch1 | 1.4.19.1 to be exact |
| 15:51.58 | [TK]D-Fender | rch1: I might try upgrading.... |
| 15:52.21 | rch1 | yeah, that was next on my things to try, was just wondering if there was something else I may have overlooked |
| 15:54.05 | rch1 | I'll give it a shot - thanks! |
| 15:56.05 | codefreeze-lap | dandre: in 1.4 up, you can set the "unanswered=no" in cdr.conf to avoid getting CDR's for unanswered channels. |
| 15:56.46 | dandre | ok I'll try |
| 15:56.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:00.13 | dandre | I had alraedy it: |
| 16:00.13 | dandre | CDR output unanswered calls: no |
| 16:00.14 | dandre | on cdr status command |
| 16:02.12 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 16:02.39 | *** join/#asterisk javar (n=javar@200.118.170.56) |
| 16:09.41 | ix33 | anyone know the status of iaxtel? |
| 16:09.52 | russellb | it doesn't exist. |
| 16:09.58 | russellb | it has been dead for a long time |
| 16:10.25 | ix33 | why does digium still show iaxtel numbers for support, et al |
| 16:11.40 | dominic1 | hi, I know I posted it alreasy but I hope now somebody is able to help me. I want to use G722 for internal calls and G711 for external calls. Can I set the codecorder dynamically? If I set SIP_CODEC=G722. He needs a translationpath if allow=alaw,allow=g722. If allow=g722,alaw I am not able to place calls via isdn. |
| 16:14.35 | russellb | ix33: i don't know, i asked for that stuff to get removed a long time ago, heh |
| 16:17.03 | *** join/#asterisk ManxPower (n=manxpowe@253.sub-70-222-48.myvzw.com) |
| 16:18.29 | x86 | 10:56 < codefreeze-lap> dandre: in 1.4 up, you can set the "unanswered=no" in cdr.conf to avoid getting CDR's for unanswered channels. |
| 16:18.34 | x86 | codefreeze-lap: you serious?! |
| 16:18.37 | x86 | that's awesome! |
| 16:19.02 | ix33 | where sarge is the name of the guy i have in iax.conf |
| 16:19.09 | codefreeze-lap | dandre: hmmm |
| 16:19.26 | ix33 | oops wrong window |
| 16:20.08 | ManxPower | I guess you never read upgrade.txt, x86 |
| 16:20.26 | ManxPower | Might want to think about doing that to see what else you didn't know is new. |
| 16:22.48 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 16:25.32 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:26.03 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:26.28 | [TK]D-Fender | ManxPower: s/new/2.5 years ago |
| 16:27.26 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:37.23 | ThoMe | have a isdncard in my computer and the card is connected with a phonebox |
| 16:37.33 | ThoMe | is te_ptmp ok? |
| 16:37.47 | *** join/#asterisk LuisTorres (n=chatzill@bl6-208-64.dsl.telepac.pt) |
| 16:38.39 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 16:39.13 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:39.22 | *** join/#asterisk hardwire (n=hardwire@rdbk-9581.mtaonline.net) |
| 16:41.30 | gaetronik | [TK]D-Fender, there was right |
| 16:41.40 | gaetronik | there is a bug in the least recent |
| 16:42.07 | hardwire | blah |
| 16:42.20 | gaetronik | s/there/they |
| 16:43.06 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 16:43.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:46.50 | *** part/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 16:48.10 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:52.51 | *** join/#asterisk dijungal (n=kdaniel@216.110.126.201) |
| 16:52.57 | dijungal | how do i change the default location of recorded calls? |
| 16:53.48 | hardwire | dijungal: changing default locations sometimes makes administration hairy |
| 16:54.02 | hardwire | I wonder if you can simply specify an absolute path for recorded files per-app. |
| 16:56.10 | *** join/#asterisk zippytech (n=ron@244.zippytech.com) |
| 16:56.20 | dijungal | i've created a ramdrive to store recordings, i would like to tell asterisk/trixbox to store the recordings to this ramdrive |
| 16:56.32 | zippytech | any one here ever used a polycom ip 300 |
| 16:57.12 | dijungal | hmmm |
| 16:57.20 | dijungal | sad phone :P polycom ip 300 |
| 16:57.25 | [TK]D-Fender | zippytech: I have an IP 301 |
| 16:57.34 | dijungal | no speaker phone.. :S |
| 16:57.51 | zippytech | how's it |
| 16:58.13 | [TK]D-Fender | zippytech: Does everything its supposed to. Why are you looking at this model? |
| 16:58.26 | zippytech | i found them for 90 each |
| 16:58.27 | zippytech | new |
| 16:58.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:58.47 | *** join/#asterisk westis (n=Daniel@vxj-161-196-233-83.3.cust.bredband2.com) |
| 16:59.14 | [TK]D-Fender | zippytech: IP 320 Goes for $80 new and has a full-duplex speakerphone, native PoE, pixel display, Microbrowser support, and inexpensive 2.55mm headset jack. |
| 16:59.18 | dijungal | so back to my question... how do i change the default call recording location in asterisk??? |
| 16:59.27 | [TK]D-Fender | dijungal: asterisk.conf |
| 17:00.26 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:00.42 | westis | hi. I just heard about Asterisk and have a question. We currently have three phone lines via IP. Using Asterisk, does that mean we can cut that down to one, so that we only have one number and use Asterisk to configure different phones internally? |
| 17:01.04 | dijungal | asterisk.conf changes the location of the spool directory |
| 17:01.20 | [TK]D-Fender | westis: You can use any number of resources with * in any combination you want. |
| 17:01.34 | [TK]D-Fender | dijungal: and thats as specific as "default" gets. |
| 17:01.59 | dijungal | hmmm |
| 17:02.17 | [TK]D-Fender | westis: You can use * with 100 phones and NO ITSP if you want, ot 1 phones with 100 different ITSP providers. |
| 17:05.41 | westis | [TK]D-Fender: ok, thanks. I don't fully grasp Asterisk yet, but will continue to read about it. What we are missing with our current setup is a way to forward calls from one phone to another (and forward to mobile phones when out-of-office) and to have a good answering machine. |
| 17:06.36 | [TK]D-Fender | westis: Basic PBX use. Sure, * is more than capable. |
| 17:06.45 | [TK]D-Fender | westis: Here: |
| 17:06.47 | [TK]D-Fender | ~book |
| 17:06.52 | 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, or see ~buybook |
| 17:07.41 | westis | thanks |
| 17:09.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:09.48 | kannan | ok, now i hit the bud id bug_id=11141 , but i cant know what the workaround is , tho its listed right there |
| 17:10.07 | kannan | http://bugs.digium.com/print_bug_page.php?bug_id=11141 |
| 17:12.32 | kannan | i had put latest stable kernel ,because the ethernet card that was rtl igabit on a dell machine was otherwise not detected (default kernel 2.6.18) |
| 17:13.09 | russellb | run "make menuselect" and disable modules you don't need |
| 17:13.10 | [TK]D-Fender | kannan: Status: closed Last Update:01-25-08 16:33 |
| 17:13.12 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 17:13.14 | kannan | of course, i have now put a 10/100 realtek card , and boot into the default kernel and zaptel compiles OK then |
| 17:13.16 | russellb | the problem may only be in modules you don't need |
| 17:13.17 | [TK]D-Fender | kannan: Old news. UPGRADE |
| 17:13.41 | kannan | upgrade Asterisk? |
| 17:13.51 | [TK]D-Fender | kannan: Read the big print. |
| 17:14.41 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:14.43 | [TK]D-Fender | kannan: Would help if you actually read the bug tracker info. |
| 17:14.47 | *** join/#asterisk luisjose (n=luisjose@nelug/coreteam/luisjose) |
| 17:15.07 | kannan | [TK}D-Fender, i am going thru more now, |
| 17:17.57 | *** part/#asterisk westis (n=Daniel@vxj-161-196-233-83.3.cust.bredband2.com) |
| 17:18.30 | *** join/#asterisk Super_Cat_Frog (n=bob@87-194-183-38.bethere.co.uk) |
| 17:18.57 | Super_Cat_Frog | hi - anybody know where i can buy a PCI-X ISDN BRI card from? Its not for use with asterisk, but i figure its the best place to ask |
| 17:19.13 | Super_Cat_Frog | ( for UK ISDN ) |
| 17:19.57 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 17:20.12 | [TK]D-Fender | Super_Cat_Frog: http://www.google.ca/search?hl=en&q=PCI-X+ISDN+BRI+buy&btnG=Google+Search&meta= |
| 17:20.32 | Super_Cat_Frog | hmm, i tried googling, but apparently i'm rubish at it |
| 17:20.35 | Super_Cat_Frog | thanks |
| 17:21.49 | Qwell | ThoMe: Don't do that |
| 17:21.57 | ThoMe | Qwell: ups |
| 17:22.00 | ThoMe | idiot |
| 17:22.01 | ThoMe | sorry! |
| 17:22.27 | Strom_C | wonders what united parcel service has to do with it |
| 17:22.38 | Qwell | Strom_C: he clarified in his next line |
| 17:22.47 | ThoMe | Qwell: which codec is use for the demo on asterisk? |
| 17:22.52 | ThoMe | ([demo]) ? |
| 17:22.59 | ThoMe | alaw? gsm? |
| 17:23.07 | [TK]D-Fender | ThoMe: Where is that? |
| 17:23.10 | kannan | [TK}D-Fender, does the fixed means that i have to get SVN for the issue not to happen? |
| 17:23.12 | Qwell | the best prompts available for the channel it's in |
| 17:23.31 | [TK]D-Fender | kannan: it means anything relased AFTER the fix should be FIXED. |
| 17:24.05 | kannan | [TK]D-Fender, oh ok , thanks |
| 17:24.19 | *** join/#asterisk jgoddess (n=womkim@leon.g-cipher.net) |
| 17:24.30 | kannan | so i got it as i installed the older versions from my pen drie |
| 17:24.41 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-167-96.red.bezeqint.net) |
| 17:24.50 | Dovid | i have something wierd |
| 17:25.00 | Strom_C | does it involve misspelling "weird"? |
| 17:25.07 | Dovid | my server refuses to send out any calls over ulaw (when set to) it will only send out over g729 |
| 17:25.23 | Dovid | Sttrom_C: been a long day. didnt have time to heat yet |
| 17:25.24 | Strom_C | Dovid: IAX2 or SIP? |
| 17:25.27 | Dovid | SIP |
| 17:25.38 | Dovid | 1.4.18 |
| 17:25.42 | Strom_C | does the other end also support ulaw? |
| 17:26.23 | Super_Cat_Frog | sniff the packets, look for the media type negotiation (there's a sip filter in wireshark that excludes the RTP) |
| 17:26.43 | Dovid | Strom_C: yes it does |
| 17:26.51 | Dovid | i did a trace on the other end |
| 17:26.54 | Dovid | nothing even comes in |
| 17:26.58 | Strom_C | Dovid: pastebin the relevant bits of sip.conf |
| 17:27.05 | Dovid | seems that when set to ulaw the server does not even send out anything |
| 17:27.18 | kannan | [TK]D-Fender, thanks , its fine, the source i had was older one to the Fixed date |
| 17:27.20 | Strom_C | do you have access to the other end as well? |
| 17:28.13 | kannan | it is the first time i read the bug tracker, and then i got in from google only |
| 17:28.44 | Dovid | http://pastebin.com/m3575d38f |
| 17:28.56 | Dovid | Strom_C: Yes I do. running Ngrep there |
| 17:29.30 | Strom_C | pastebin that one too |
| 17:31.26 | Strom_C | also, not using the sample configs as a base for your production configs will go an extremely long way towards making debugging easier |
| 17:32.14 | Dovid | yea. its a test box |
| 17:32.15 | Dovid | brb |
| 17:33.51 | Dovid | Strom_C: Nothing to show from ngrep |
| 17:34.28 | Strom_C | could you...you know....pastebin the config, please? |
| 17:34.43 | zippytech | does asterisk work with video calls |
| 17:34.45 | Dovid | basicly if the call is sent from server A with G729 Server B gets something on port 5060 (but will not work cause that server does not have g729 licencing) if i send with ulaw only the reciving server rejects it (I tried sending to multiple servers) |
| 17:34.49 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 17:34.53 | Dovid | extensions.conf ? |
| 17:35.00 | Strom_C | Dovid: sip.conf |
| 17:35.07 | Strom_C | zippytech: for passthrough, yes |
| 17:35.10 | Dovid | recieving end ? |
| 17:35.28 | Strom_C | Dovid: obviously -- you already pastebinned the transmitting end, right? |
| 17:35.35 | Dovid | just wanted to make sur e;) |
| 17:36.30 | *** part/#asterisk UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
| 17:37.12 | Dovid | http://pastebin.com/m2543ef37 |
| 17:37.38 | Dovid | Strom_C: I tried sending it from this sever to multiple servers and it refuses to go out on anything but G729. real interesting |
| 17:38.50 | Strom_C | what kind of station equipment? |
| 17:39.07 | Dovid | all asterisk |
| 17:39.14 | Strom_C | ... |
| 17:39.16 | Strom_C | station equipment |
| 17:39.20 | Strom_C | not switching equipment |
| 17:39.51 | Dovid | station equipment = ? |
| 17:39.55 | Dovid | a bit tired. |
| 17:40.02 | Strom_C | telephone sets and so on |
| 17:40.07 | Dovid | i think i need to re-compile asterisk |
| 17:40.20 | hsv-al | ST302 IP phones decent? |
| 17:40.20 | Dovid | ITSP -> Server A -. Server B - > ITSP |
| 17:40.56 | Strom_C | what happens if you hook up a phone to server A and server B and try to make a call from one to the other? |
| 17:41.28 | Strom_C | because your diagram makes no sense if you only have one SIP entry on both server A and server B |
| 17:41.48 | Strom_C | surely you'd have at least two -- one for the ITSP and one for the other server |
| 17:41.59 | Dovid | Strom_C: y does it not make sense ? I amjust routing the call |
| 17:42.09 | Dovid | oh. i just didnt post that ;) |
| 17:42.22 | Strom_C | slams head really hard into the desk |
| 17:42.23 | Dovid | my issue is that it does not even leave Server A if set to ulaw |
| 17:42.26 | Dovid | haha |
| 17:42.31 | Dovid | i have too much in there |
| 17:42.32 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 17:42.33 | Strom_C | please, don't be a tard. |
| 17:42.34 | [TK]D-Fender | Dovid: pastebin the failed call withs ip debug |
| 17:42.35 | Dovid | gona see how i built asterisk last |
| 17:44.08 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:44.15 | Assid | hi |
| 17:44.44 | Assid | how do i know the codec capability if its 262 |
| 17:44.49 | Assid | or 1294 |
| 17:46.49 | *** join/#asterisk sniper_sniper (i=michofr@62.84.81.85) |
| 17:47.19 | Dovid | TK: http://pastebin.com/m189a723a |
| 17:47.57 | luisjose | is there a way to send a "flash" event via dtfm? |
| 17:48.06 | luisjose | dmtf |
| 17:48.36 | Strom_C | luisjose: why? |
| 17:48.41 | luisjose | DTMF |
| 17:48.43 | [TK]D-Fender | luisjose: Read up on "features.conf" on the WIKI, and do this with a "dynamic feature" |
| 17:48.51 | [TK]D-Fender | Dovid: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0xdee (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|ilbc)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) |
| 17:49.20 | [TK]D-Fender | Dovid: and you didn't show a peer for "rich" anywhere in your PB, OR this other peer originating the call. |
| 17:49.31 | luisjose | Strom_C, i have problem with transfer type on softphones that dont have "flash" button, so i need it to send with * or something |
| 17:49.47 | ThoMe | what i should use? |
| 17:49.49 | Strom_C | luisjose: softphones don't transfer with "flash" |
| 17:49.51 | ThoMe | nt_ptmp or te_ptmp ? |
| 17:49.55 | [TK]D-Fender | Dovid: -- Executing [0788187064@rm1:1] Dial("SIP/66.34.32.212-090c0028", "SIP/0788187064@rich") in new stack [Jun 4 14:11:07] WARNING[7845]: chan_sip.c:3008 sip_call: No audio format found to offer. Cancelling call to 0788187064 |
| 17:49.55 | Strom_C | there's a transfer button. |
| 17:50.04 | Dovid | TK: I have rich in there. |
| 17:50.08 | Dovid | let me post |
| 17:50.20 | [TK]D-Fender | Dovid: and seeing this without any attempt at negotiation tells me taht the peer itself doesn't have a compatible codec. |
| 17:50.42 | Dovid | http://pastebin.com/m41d1e40b |
| 17:50.59 | Dovid | the recieving peer ? |
| 17:51.00 | ThoMe | [TK]D-Fender: nt_ptmp or te_ptmp ? |
| 17:51.24 | luisjose | Strom_C, yes i know, but flash transfer have a diferent beheavor than the other transfer |
| 17:51.28 | [TK]D-Fender | ThoMe: If I had something to say, I would have. Don't direct questions at bystanders like that, its very rude. |
| 17:51.42 | luisjose | you can jump multiple times between parties and even stay as a conference |
| 17:51.47 | Dovid | TK: It wont work to any other server including if i send it to the server that it came from |
| 17:51.55 | ThoMe | [TK]D-Fender: ok :-) |
| 17:52.00 | Strom_C | luisjose: that's not transfer -- that's threeway calling |
| 17:52.25 | ThoMe | [TK]D-Fender: sory, sir. |
| 17:52.46 | Dovid | TK: I am a retard. NM |
| 17:53.09 | [TK]D-Fender | Dovid: Make a single comprehensive pastebin of EVERYTHING, and FFS remove all that commented out crap. |
| 17:53.21 | [TK]D-Fender | Dovid: Good... 11 steps to go! |
| 17:53.28 | Dovid | hahahah |
| 17:53.39 | Dovid | i missed the obvious. |
| 17:53.47 | Dovid | no more working 15 hours on 3 hours of lseep |
| 17:54.06 | [TK]D-Fender | Dovid: Oh, don't let that stop you... just don't come HERE :p |
| 17:55.13 | anonymouz666 | Dovid: 3 hours of sleep? [TK]D-Fender sleeps, join IRC and plays guitar at the same time. |
| 17:55.17 | anonymouz666 | lol |
| 17:56.30 | Assid | err.. anyone by chance know the latest polycom sip and bootrom out? |
| 17:56.43 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:57.42 | _ShrikE | I think the latest bootrom is 4.1.0 |
| 17:58.03 | _ShrikE | And the latest sip is 3.0.1 |
| 17:58.10 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-76.unitz.ca) |
| 17:59.08 | Assid | their site isnt updated |
| 17:59.21 | Assid | or the 301's dont support the latest |
| 17:59.54 | _ShrikE | That is just the most recent I see availalbe for download from my vendor |
| 18:00.41 | Super_Cat_Frog | hi - i'm getting this error when converting a table to ndbcluster: "The total number of locks exceeds the lock table size" - does anybody know what i can change to increase the size of the lock table ? |
| 18:01.08 | Assid | _ShrikE: any idea if the updates affect 301's ? |
| 18:01.40 | [TK]D-Fender | _ShrikE: 3.0.2C |
| 18:02.19 | _ShrikE | Im still not sure about upgrading to the 3.x version |
| 18:02.51 | Super_Cat_Frog | argh crap, wrong channel, sorry |
| 18:03.07 | _ShrikE | Assid: Im not sure |
| 18:05.20 | Assid | okay this is weird.. i called from 1 extension of my house.. to anotehr extension |
| 18:05.21 | Assid | http://assid.pastebin.com/de8409d3 |
| 18:05.59 | Assid | it shows 729 for the pap2 .. but ulaw for the polucom |
| 18:06.02 | Assid | polycom even |
| 18:06.14 | Assid | does this make sense? |
| 18:06.23 | *** join/#asterisk jcaceres (n=jcaceres@200.6.115.10) |
| 18:07.29 | Assid | but when i sip show peer on that polycom user: i get : Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729:20,ulaw:20,alaw:20) |
| 18:07.50 | Assid | so the question remains.. why does it show ulaw there |
| 18:08.10 | *** part/#asterisk dijungal (n=kdaniel@216.110.126.201) |
| 18:12.06 | Assid | anyone? |
| 18:13.41 | *** join/#asterisk tinne (n=tinne@p5B325DAE.dip.t-dialin.net) |
| 18:18.39 | tinne | Hi there. One of our customers want's to connect his SPA9000 (Linksys) to our Asterisk PSTN Gateway. He uses a single SIP Account on our Server. How is it possible to use the caller-id of each of its phones connected to its spa9000? Linksys says "The INVITE sent to the ITSP carries a <ref> URI parameter in the FROM header that specifies which is the extension". How can we take this out of the URI? |
| 18:19.16 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:20.16 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 18:20.41 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 18:21.34 | Tourinho | hello all.. I need to study VXML. Any indications ? I saw a vxml from i6net.. do you know an open project that interacts with asterisk? |
| 18:23.12 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-74-56.vif.net) |
| 18:23.22 | *** part/#asterisk Super_Cat_Frog (n=bob@87-194-183-38.bethere.co.uk) |
| 18:23.23 | [TK]D-Fender | Tourinho: there's at least 1 solution out for *. Go check the WIKI |
| 18:24.20 | [TK]D-Fender | tinne: "core show function SIP_HEADER" |
| 18:24.51 | tinne | [TK]D-Fender: thank you! i'll try. |
| 18:26.02 | *** join/#asterisk Samurai_dio (n=diovani@201.41.41.235) |
| 18:26.08 | SamuraiDio | hi |
| 18:26.19 | kannan | Assid , whats the codec set on the polycom? |
| 18:27.47 | Assid | kannan: i set it to 729 ulaw alaw order |
| 18:28.21 | kannan | in asterisk ? and in the polycom phone? |
| 18:28.39 | kannan | Assid :) |
| 18:28.47 | Assid | same |
| 18:30.19 | Tourinho | [TK]D-Fender: ok, Ill look now |
| 18:30.35 | Assid | and how do i query the codecs the polycom is capable off? from within asterisk |
| 18:30.57 | tinne | [TK]D-Fender: thanks so far. IIRC i can use "SIP_HEADER(From,1)" to get the first occuring From-Header. Can i get only the "ref" parameter out of it? Header-Example: "From: PBX Line 1 <sip:37683101@matrix.sipura.com>;tag=4625e6e97e37072ao2;ref=5031" |
| 18:31.18 | [TK]D-Fender | tinne: "cor show function CUT" |
| 18:31.22 | [TK]D-Fender | tinne: "core show function CUT" |
| 18:32.01 | Tourinho | [TK]D-Fender: there is only vxml from i6net |
| 18:32.09 | tinne | [TK]D-Fender: Thanks! |
| 18:33.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:35.48 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 18:37.37 | kannan | does any one have the full list of country codes for prefix in csv or xml format? |
| 18:37.49 | Strom_C | kannan: one moment |
| 18:38.59 | Strom_C | you want the country codes themselves, right? (i.e. 1 for north america, 44 for UK, 27 for south africa, etc) |
| 18:39.37 | kannan | Srom_C , yes, + if you have the mobile and landline prefix for them also, like 447 fr UK |
| 18:39.42 | kannan | Strom_C |
| 18:40.09 | Strom_C | kanaan: you'll have to actually pay money for that much level of detail |
| 18:40.19 | Strom_C | www.trainfo.com |
| 18:40.58 | kannan | Strom_C, ok , thanks |
| 18:41.21 | Assid | kannan: ? |
| 18:41.42 | kannan | Assid , no idea on your prob, sorry |
| 18:41.48 | Assid | L| |
| 18:42.32 | *** join/#asterisk miduyak (n=josh@bzq-219-135-48.static.bezeqint.net) |
| 18:43.08 | [TK]D-Fender | Assid: you'll see it in an invite, and in sip.cfg |
| 18:43.28 | kannan | Assid, I cannot understand how Asterisk will not use g729 if thats the first pref in the asterisk config and the polycom, and you have the license in the * also |
| 18:43.54 | miduyak | I have a wildcard x101p that is dropping called midway but when I run asterisk with verbosity and debug I dont see it handup just the sound goes mute. Any suggestions on how to further debug this? |
| 18:43.59 | Assid | well its supposed to use it cause of the passthrough anyways |
| 18:44.48 | Strom_C | miduyak: dump the piece of junk card and get something better |
| 18:45.06 | miduyak | ok but the card has been working fine up until a week ago |
| 18:45.07 | gaetronik | miduyak, openvox? |
| 18:45.08 | kannan | oh, will Asterisk aloow g729 even without any licensed channels? |
| 18:45.17 | logi4023 | while in an ivr, the call is terminated as soon as a key is pressed on the phone keypad even though that exten is defined. |
| 18:45.22 | Strom_C | kannan: it will allow passthrough, but it wont transcode |
| 18:45.25 | miduyak | how could I verify if its the card or a problem with the machine |
| 18:45.30 | miduyak | openvox? |
| 18:45.37 | kannan | Strom_C, i see , ok |
| 18:45.50 | miduyak | no asterisk SVN-branch-1.4-r116466 |
| 18:45.52 | Strom_C | miduyak: four years of experience tell me that it's almost certainly your card |
| 18:45.56 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 18:45.56 | gaetronik | logi4023, look at the console to see what happen |
| 18:46.24 | gaetronik | miduyak, the vendor of the card |
| 18:46.27 | miduyak | really, I've had the card for about 3 years now and never had a problem |
| 18:46.28 | logi4023 | gae -- I only get a 'auto fallthrouhh, channel 'sip/fgsfgff' status is unknown' |
| 18:47.04 | gaetronik | if you put more debug logi4023 |
| 18:47.26 | miduyak | what about the vendor of the card? |
| 18:47.43 | [TK]D-Fender | logi4023: thats a sign taht you hit the end of priorities on your exten without an explicit hangup |
| 18:47.55 | miduyak | which card is an appropriate replacement card for that? |
| 18:48.09 | Strom_C | miduyak: TDM410P |
| 18:48.27 | kannan | Strom_C, that telecordi has a full list of 750000 or so dial codes, thanks, its what i am looking for actually |
| 18:49.10 | logi4023 | pressing '1' on the keypad while in 's' should cause a jump to exten '1'. instead it does a hangup. with 'auto fallthrough' message |
| 18:49.19 | miduyak | Strom_C, before I buy a new card, do you have any suggestions on how to verify that its indeed the card and not a problem with the computer? |
| 18:49.35 | Strom_C | miduyak: roll back to an older version of asterisk and zaptel |
| 18:49.45 | Strom_C | logi4023: pastebin the relevant section of extensions.conf |
| 18:50.01 | [TK]D-Fender | logi4023: pastebin your ivr context and [general] and [globals] |
| 18:50.03 | [TK]D-Fender | ~pb |
| 18:50.04 | 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 |
| 18:50.06 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:50.14 | miduyak | Stron_C, cpu is fine, theres only ~4000k of memory |
| 18:50.19 | Strom_C | miduyak: roll back to an older version of asterisk and zaptel |
| 18:50.24 | logi4023 | i use asterisk extensions.conf, cant pastebin |
| 18:50.39 | Strom_C | logi4023: if you can't pastebin, we probably can't help you |
| 18:51.10 | miduyak | Strom_c any particular version that you'd suggest? |
| 18:51.10 | logi4023 | i use realtime extensions.conf. |
| 18:51.10 | Assid | [TK]D-Fender: http://assid.pastebin.com/d4cf7dc6e -- the 2 places i see any codecs |
| 18:51.10 | Strom_C | miduyak: the last version which you know worked |
| 18:51.59 | miduyak | Strom_C I have a TDP400P lying around, are there significant differences between that and the TDM410P? |
| 18:52.24 | Strom_C | miduyak: it's an entirely different architecture...but i'd try swapping in the TDM400 and see if that works |
| 18:52.50 | [TK]D-Fender | Assid: and not the other place I told you to look for. |
| 18:53.03 | [TK]D-Fender | Assid: look at the codec capabilities match line. |
| 18:53.22 | miduyak | Strom_C except that I need to buy an FXS module if I'm going to do that, so the question is if its better to go with the TDM410 or the TDM400 will work fine |
| 18:53.26 | Strom_C | logi4023: can you pastebin the output of "dialplan show"? |
| 18:53.48 | Strom_C | miduyak: i thought you said this was an FXO card |
| 18:54.23 | Assid | [TK]D-Fender: a=rtpmap:0 PCMU/8000 ? |
| 18:54.36 | [TK]D-Fender | Assid: look at the "codec capabilities" match line. <---- |
| 18:55.03 | miduyak | Strom_C no its a X101P |
| 18:55.58 | logi4023 | strom -- extensions.conf has only 'switch => realtime/@extensions' |
| 18:56.07 | Assid | [TK]D-Fender: which file? grep shows me nothing |
| 18:56.16 | Strom_C | logi4023: you're not listening to me |
| 18:56.23 | Strom_C | "dialplan show" at the asterisk CLI |
| 18:56.38 | Strom_C | miduyak: yeah, X101P is an FXO as far as I'm aware |
| 18:56.55 | [TK]D-Fender | Strom_C: He said he's using realtime, that woulodn't show |
| 18:57.06 | Strom_C | [TK]D-Fender: ah, alright...for some reason I assumed it would |
| 18:57.15 | Strom_C | deducts five points from score |
| 18:57.26 | ManxPower | Using Realtime pretty much eliminates any possible help from this channel |
| 18:57.31 | [TK]D-Fender | Strom_C: Nope, that only shows parsed PBX_ module stuff |
| 18:57.44 | [TK]D-Fender | ManxPower: not that bad. |
| 18:57.50 | kannan | ManxPower, why does realtime not get support? |
| 18:57.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.08 | ManxPower | I hear it also causes sterility in males. |
| 18:58.15 | ManxPower | kannan: you are seeing why right now. |
| 18:58.21 | [TK]D-Fender | logi4023: And if you aren't competant enough to do a database dump and pastebin it you don't deserve to be running your dialplan in a database in the first place. |
| 18:58.21 | kannan | heheh |
| 18:58.27 | [TK]D-Fender | logi4023: No excuse for that. |
| 18:58.39 | logi4023 | the only message i get is: 'Auto fallthrough, channel 'SIP/xxxxx-0950f3c8' status is unknown |
| 18:58.45 | ManxPower | kannan: incredibly difficuly to support. |
| 18:59.03 | Assid | [TK]D-Fender: i cant find any "codec capabilities" mentioned anywhere |
| 18:59.11 | kannan | ManxPower, sorry to hear, as i was planning to try this next :) |
| 18:59.22 | [TK]D-Fender | ManxPower: Well he doesn't have a realtime issue. I'm sure THAT works. The dialplan he PUT it in it is another matter entirely. |
| 18:59.31 | ManxPower | kannan: wait until you NEED it -- at 10,000 users or so. |
| 18:59.33 | [TK]D-Fender | Assid: pastebin the entire friggen call |
| 18:59.36 | Corazu | ~book |
| 18:59.37 | 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, or see ~buybook |
| 19:00.26 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 19:00.31 | ManxPower | kannan: VoIP and Asterisk is complicated enough without adding a database to the mix. |
| 19:00.40 | Assid | [TK]D-Fender: http://assid.pastebin.com/d4ef09476 - ran sip debug peer on me |
| 19:01.12 | kannan | ManxPower, yes , i know that , coz its been hard so far for many things |
| 19:02.16 | kannan | i even wanted to go off to trixbox , but now i got most things i need working fine |
| 19:02.20 | [TK]D-Fender | Assid: Where is the invite from your polycom? |
| 19:02.51 | x86 | I've got a Toshiba PBX with some analog station ports going over to a Rhino FXO channel bank which goes into Asterisk... it seems like if a call is bridged between the two systems, it goes through ok and both parties can hear each other, but I randomly get a tone that almost sounds like a call waiting tone |
| 19:02.53 | kannan | but have not touched things like AEL or realtime |
| 19:02.57 | x86 | any ideas what this could be? |
| 19:03.00 | kannan | so far |
| 19:03.02 | Assid | thats all i got.. i didnt answer the call.. both phones are in different rooms.. |
| 19:03.22 | [TK]D-Fender | x86: intermittant blip? |
| 19:03.38 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 19:03.44 | Assid | [TK]D-Fender: do you want me to try it while answering the call ? |
| 19:03.50 | Assid | hold on |
| 19:03.56 | pjz | does the AA50 use AsteriskNow? |
| 19:03.56 | x86 | [TK]D-Fender: i'm not sure if it's intermittant or not... I think it could be on a regular recurring basis... |
| 19:03.58 | [TK]D-Fender | Assid: Place a call from your polycom. |
| 19:04.14 | ManxPower | x86: call waiting? |
| 19:04.16 | x86 | [TK]D-Fender: it's not at all like the sound you hear when there is packet loss |
| 19:04.17 | kannan | Assid, can you not auto anser it from extensions itself |
| 19:04.26 | [TK]D-Fender | x86: I mean its occurance during the call. is it a CONTINUOUS tone, or an infrequent blip? |
| 19:04.34 | x86 | ManxPower: I don't have callwaiting=yes for that span... should I explicitly say callwaiting=no? |
| 19:04.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:04.50 | ManxPower | x86: I meant on the TOSHIBA |
| 19:05.01 | ManxPower | pjz: perhaps you could ask on the correct channel? |
| 19:05.36 | pjz | ManxPower: I was trying to determine whether that was the corect channel before I went there, actually. |
| 19:05.38 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 19:06.00 | ManxPower | (2:05:42 PM) current topic is: Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) .... asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) |
| 19:06.05 | Strom_C | pjz: no, the AA50 runs a custom-built linux OS designed specifically for the AA50 |
| 19:06.09 | ManxPower | For AAxx questions you should talk to Digium sales |
| 19:06.18 | logi4023 | here's pastebin: http://pastebin.com/d27122086 |
| 19:06.53 | pjz | Strom_C: ah, okay, thanks! |
| 19:07.06 | x86 | [TK]D-Fender: it's intermittent |
| 19:07.09 | Strom_C | pjz: but I echo ManxPower -- you should contact digium sales |
| 19:07.19 | Assid | okay found : Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) |
| 19:07.25 | x86 | [TK]D-Fender: say every 30 or so seconds... I'd almost say it's on a schedule |
| 19:07.29 | ManxPower | The AAxx is a closed system anyway -- we could not support it even if we wanted to. |
| 19:07.33 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583373.dsl.bell.ca) |
| 19:07.37 | [TK]D-Fender | x86: I've heard tones like that put on by PBX's to indicate that they are (or might be) recording the call. |
| 19:07.38 | x86 | ManxPower: hmm, I don't know for sure |
| 19:07.50 | logi4023 | the problem is occuring with asterisk 1.6beta9 ... |
| 19:07.51 | [TK]D-Fender | x86: This would be quite plausibly it. |
| 19:08.04 | x86 | [TK]D-Fender: we don't do any recording on the toshiba though |
| 19:08.16 | ManxPower | logi4023: How long are you planning on having this text/development system? |
| 19:08.16 | logi4023 | anyone can go over the pastebin? |
| 19:08.21 | ManxPower | test, not text |
| 19:08.37 | Assid | [TK]D-Fender: http://assid.pastebin.com/d6bd5c89b |
| 19:08.44 | ManxPower | I assume it's not a production system since you are running BETA software. |
| 19:08.46 | Assid | a call from my polycom to the other extension |
| 19:08.53 | logi4023 | manx - what do you mean test/development ? |
| 19:09.03 | ManxPower | x86: double check the sync source for your spans. |
| 19:09.13 | ManxPower | logi4023: you are running the development/beta/test version of Asterisk. |
| 19:09.38 | x86 | [TK]D-Fender: I also am having some issues where voice will have little blips of garbling... lasting only 1-2 seconds, and seemingly random... going over the same trunk lines to the Toshiba (no IP involved at all) |
| 19:09.50 | x86 | ManxPower: hmm ok |
| 19:09.53 | ManxPower | x86: sounds like a sync problem to me. |
| 19:09.55 | [TK]D-Fender | Assid: Line 14. Next time, wake up. No really... Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) |
| 19:10.15 | deeperror | x86, are you using any rhino cards? |
| 19:10.30 | ManxPower | I find it funny that Assid is using both ulaw AND alaw AND G729 |
| 19:10.39 | ManxPower | you never want to enable both alaw and ulaw |
| 19:11.05 | x86 | deeperror: nope, just the rhino channel bank |
| 19:11.26 | x86 | ManxPower: Asterisk is set to master on that span... you think the rhino would try to be master as well? |
| 19:12.06 | ManxPower | x86: I doubt it, but you'd be a fool to not verify the Channel Bank is getting it' |
| 19:12.11 | ManxPower | s sync from Asterisk |
| 19:12.35 | x86 | I didn't even know a Rhino had selectable clock source... just always thought it defaulted to slave/line timing |
| 19:13.05 | ManxPower | x86: All should have a selectable sync source, all should default to getting their timing from the T-1 |
| 19:13.12 | x86 | ahh... that span has 69981 overruns, and 69981 framing errors |
| 19:13.27 | Assid | [TK]D-Fenderright.. but its sitting on 729 as first option |
| 19:13.38 | Assid | dont see why it isnt doing it as first option |
| 19:13.42 | x86 | but that could be from auto-configuration last time I power-cycled the Rhino, I guess |
| 19:13.54 | deeperror | x86, i had the blips when using rhino r4t1 cards so not sure on that they always corresponded with a kernel message in messages |
| 19:14.02 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 19:14.07 | x86 | Assid: take everything else but g729 out and get that working first |
| 19:14.48 | x86 | deeperror: are you talking about the "call waiting tone" blip or the "garbled voice for 1-2 seconds while in a call" blip? :) |
| 19:15.03 | ManxPower | Assid: G729 will NEVER be selected if any other codecs are allowed |
| 19:15.06 | deeperror | x86, and they occurred when a configured channel bank was not plugged in like every 10 seconds |
| 19:15.31 | Assid | ManxPower: shouldnt having the order set in the mac-sip.cfg work ? |
| 19:15.33 | deeperror | x86, no this was a blip or chirp is what our agents said..."sounds like birds in the line" |
| 19:15.44 | x86 | yeah that's not the same deal |
| 19:15.55 | ManxPower | Assid: ASTERISK will NEVER select G729 is there are any other codecs allowed |
| 19:16.07 | Assid | oh |
| 19:16.13 | x86 | ManxPower: mine always did |
| 19:16.29 | x86 | when I was running g729, it would try that first |
| 19:16.30 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 19:16.39 | x86 | of course my other choice was g723.1 ;) |
| 19:16.53 | ManxPower | jbot: slap x86 |
| 19:16.53 | jbot | ACTION slaps x86, keep your grubby fingers to yourself! |
| 19:17.16 | ManxPower | As soon as I typed that I knew some asshole would mention G723.1 |
| 19:17.20 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 19:18.18 | *** join/#asterisk doolph (i=c9daf159@gateway/web/ajax/mibbit.com/x-62ecda8cf93ab685) |
| 19:18.23 | doolph | hello |
| 19:19.20 | doolph | anyone know what's the best way to do this, only this extension 100 can make calls starting by 8 |
| 19:19.33 | ManxPower | I *never* have codec problems, but I never allow more than 1 codec for each peer/user |
| 19:19.50 | doolph | if not extension 100 then unvailable |
| 19:19.52 | ManxPower | doolph: contexts are your friend. |
| 19:20.09 | doolph | yeah |
| 19:20.24 | doolph | i know but right now i forgot all the commands |
| 19:21.04 | [TK]D-Fender | doolph: "commands"? huh?! |
| 19:21.08 | Assid | ManxPower: okay .. when i call from my polycom.. i get 729-729.. but if another extension calls me.. i get ulaw-729 (ulaw on polycom) |
| 19:21.23 | [TK]D-Fender | doolph: Go read the book... chapter 5 is your friend... |
| 19:21.25 | [TK]D-Fender | ~book |
| 19:21.26 | 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, or see ~buybook |
| 19:21.35 | doolph | ummm context |
| 19:21.38 | doolph | oh i remember |
| 19:21.41 | ManxPower | Assid: I can't imagine why you think this is off. |
| 19:22.07 | Assid | ManxPower: why cant it use 729 ? for incoming as well? |
| 19:22.19 | ManxPower | Assid: one leg is not using g729 |
| 19:22.30 | ManxPower | are you dialing by IP? |
| 19:22.39 | Assid | nah.. extension based |
| 19:22.49 | ManxPower | you can't dial by extension, you can only dial by peer |
| 19:22.58 | Assid | err. yeah i meant peer |
| 19:23.02 | ManxPower | Then say peer. |
| 19:23.03 | Assid | both peers on the same server |
| 19:23.13 | ManxPower | and both peers have only G729 allowed? |
| 19:23.15 | Assid | and oh yeah.. both peers behind the same nat |
| 19:24.09 | Assid | no.. the linksys pap2 - has ulaw available. BUT when i run sip show channel ... it shows 729 for that leg |
| 19:24.29 | ManxPower | turn off ulaw in the peer for the pap2 |
| 19:24.32 | Assid | http://assid.pastebin.com/d6453b11b -- one of the calls |
| 19:24.59 | Assid | but that shows - its 729 not ulaw |
| 19:25.06 | ManxPower | But you just said it was ulaw |
| 19:25.47 | ManxPower | Also you CANNOT have ulaw on the polycom unless you have it allowed in the peer definition or are dialing by IP. |
| 19:26.20 | Assid | no no.. linksyspap2 shows 729 - polycom is ulaw.. when i call from pap2 to polycom |
| 19:26.24 | Assid | and peer based calling |
| 19:26.30 | ManxPower | Assid: show me the Dial line for that call. |
| 19:26.35 | ManxPower | or just paste the CLI output |
| 19:26.40 | ManxPower | WITHOUT sip debug |
| 19:27.08 | Assid | http://assid.pastebin.com/d607ad88a |
| 19:27.31 | Assid | -satish2 is the pap2 , calling -satish which is the polycom |
| 19:27.53 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 19:28.10 | ManxPower | nowpastebin the sip.conf entries for [spherelinx-satish2] abd [spherelinx-satish] |
| 19:28.33 | Assid | i am now trying 729 only on both ends |
| 19:28.41 | Assid | hold |
| 19:28.46 | Assid | err. hold on please |
| 19:28.48 | sniper_sniper | Hi all...I need to configure my asterisk server to playback number 2 when the extension 12345 is dialed...I added the following to extensions.conf but stil the extension 12345 ringing when trying to call it |
| 19:29.12 | sniper_sniper | [incoming] |
| 19:29.16 | Strom_C | oh christ |
| 19:29.17 | ManxPower | exten => 12345,1,SayDigits(2) |
| 19:29.18 | Strom_C | PASTEBIN |
| 19:29.24 | sniper_sniper | exten => 12345,1,Playback(digits/2 |
| 19:29.40 | Assid | okay when i make a 729 ONLY on both ends its sticking to 729 only |
| 19:29.53 | ManxPower | Assid: That is what I have been telling you for the past 10 mins |
| 19:29.57 | sniper_sniper | ManxPower, it still answering |
| 19:30.13 | ManxPower | sniper_sniper: remove all other reverences to extension 12345 |
| 19:30.16 | Assid | ManxPower: sorry.. i apologize |
| 19:30.28 | Strom_C | sniper_sniper: also, add the "noanswer" flag to playback*( |
| 19:30.34 | Strom_C | s/*(/()/ |
| 19:30.39 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 19:30.53 | Assid | err.. thanks ManxPower and [TK]D-Fender |
| 19:31.07 | ManxPower | sniper_sniper: first you said it's ringing, now you are saying it's answering. They are two different things. Which is it? |
| 19:31.35 | Assid | i wonder if i should update my bootrom and sip |
| 19:31.37 | BBHoss | hi all, how can i manually log a message as an error? I've tried using exten => *87,n,Log(ERROR|Error Logged) and it doesn't work |
| 19:31.45 | sniper_sniper | ManxPower, sorry...when tryong to dial the extension it's ringing |
| 19:32.16 | ManxPower | sniper_sniper: pastebin your extensions.conf and the CLI output of a failed call |
| 19:32.42 | sniper_sniper | ManxPower, OK i'll |
| 19:32.46 | ManxPower | BBHoss: What does "core show application log" tell you? |
| 19:33.33 | BBHoss | Log(<level>|<message>) level must be one of ERROR, WARNING, NOTICE, DEBUG, VERBOSE, DTMF |
| 19:33.41 | BBHoss | thats what i have |
| 19:33.51 | ManxPower | try using a , instead of a | |
| 19:33.51 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 19:33.53 | Assid | ManxPower: is there a way i can know if the call is routed internally over the lan ? or over the net? |
| 19:34.02 | ManxPower | Assid: I cannot help you further |
| 19:34.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:34.42 | Assid | was it something i said? |
| 19:35.07 | Assid | sniffs.. ok i didnt fart |
| 19:35.36 | sniper_sniper | ManxPower, http://pastebin.com/m15fcc24c |
| 19:35.43 | sniper_sniper | ManxPower, This is CLI outut |
| 19:35.51 | x86 | hmm |
| 19:36.00 | BBHoss | nm i got it, i think i forgot to reload dialplan :) :) |
| 19:36.09 | x86 | almost all of my spans are taking errors... although they are taking them very slowly |
| 19:36.15 | ManxPower | sniper_sniper: we cannot help you with Trixbox/Amp/etc and we resent you trying to con us into helping you with it when it CLEARLY says in the topic the correct place to go |
| 19:36.33 | x86 | and they're all internal T1's except one going out to the PSTN (which, surprisingly, is not taking errors at all) |
| 19:37.38 | sniper_sniper | ManxPower, I know that it should be straight forward and it should work directly |
| 19:37.48 | x86 | all the errors are framing and overruns... what's that mean? |
| 19:38.53 | x86 | I'm using either fxo_ks or fxs_ks on all of the spans, and the channel banks support gs and ls... they're set on ls right now, which should be compatible with ks right? |
| 19:38.56 | *** join/#asterisk GueSss`Who (n=m@41.234.221.100) |
| 19:39.20 | x86 | should I be using loopstart instead of kewlstart? |
| 19:39.47 | GueSss`Who | Good evening aLL |
| 19:40.22 | *** join/#asterisk truent (i=0c224403@gateway/web/ajax/mibbit.com/x-1d010918c0dd4f8a) |
| 19:40.36 | truent | is there no more setup wizard in asterisknow? |
| 19:41.01 | Strom_C | truent: #asterisknow |
| 19:41.33 | ManxPower | sniper_sniper: Not one single thing is simple in Trixbox |
| 19:41.48 | sniper_sniper | ManxPower, it'snot trixbox |
| 19:41.48 | ManxPower | sniper_sniper: That pastebin should have been like 5 lines long in a normal Asterisk system |
| 19:41.58 | sniper_sniper | ManxPower, it's an ASTERISK Server |
| 19:44.16 | *** part/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 19:44.35 | *** join/#asterisk ManxPower (n=manxpowe@253.sub-70-222-48.myvzw.com) |
| 19:45.39 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:46.54 | *** join/#asterisk glaz (i=strke@mofu.ca) |
| 19:51.02 | *** join/#asterisk ix33 (n=ix@69.73.18.50) |
| 19:51.21 | [TK]D-Fender | ManxPower: Careful... you should assassinate him on using FreePBX, not trixbox. |
| 19:51.54 | ManxPower | [TK]D-Fender: They all look alike to me -- much like spammers and politicians |
| 19:51.56 | [TK]D-Fender | ManxPower: its not the distro we have a problem with, its the demon-spawn output of the GUI it comes with (can can be yanked out) |
| 19:52.08 | ManxPower | [TK]D-Fender: my issue is the config files. |
| 19:52.22 | [TK]D-Fender | ManxPower: then make sure to accuse him of the proper thing :) |
| 19:52.39 | ManxPower | I shall try |
| 19:52.55 | ManxPower | Did you send him packing? |
| 19:54.07 | [TK]D-Fender | ManxPower: no, he left all by himself just before I caught up to the conversation. |
| 19:54.56 | ManxPower | He's on ##Linux and #openser now. I bet he's asking for FREEPBX help there. |
| 19:55.05 | [TK]D-Fender | ManxPower: Not our problem |
| 19:55.20 | ManxPower | Nope. |
| 19:55.52 | ManxPower | Why can't we just change the channel to #Asterisk_Yes_We_Have_No_GUIs |
| 19:56.10 | Corazu | #asterisk-gui :P |
| 19:56.37 | ManxPower | No GUIs on #Asterisk-gui? |
| 19:56.44 | Corazu | :P |
| 19:56.58 | Corazu | I find non-gui programs to be more powerful, from a developer's perspective |
| 19:57.21 | Corazu | you can create your own GUI for your own use..which allows for more flexibility for what you need |
| 19:57.26 | ManxPower | Corazu: I don't care if someone uses a GUI or not. Just don't come to this channel expecting support for a GUI |
| 19:57.54 | ManxPower | If your GUI is not giving you the support you need then perhaps it's time to switch to something else. |
| 19:58.02 | Corazu | ManxPower: if ciyrse |
| 19:58.05 | Corazu | course* |
| 19:58.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.37 | ManxPower | The most common excuse I hear here is "But nobody helped me on #trixbox|AsteriskGUI|FreePBX!" |
| 19:58.37 | Corazu | Anyways, I'm out for the day. Thanks again for the help earlier. |
| 19:59.39 | *** join/#asterisk javar (n=javar@200.118.170.56) |
| 19:59.59 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 20:00.44 | kannan | and no one there to help on #a2billing |
| 20:01.56 | Qwell | kannan: see above. |
| 20:03.27 | kannan | Qwell, yes , just joking |
| 20:06.37 | *** join/#asterisk Bananaskin (n=mike@78-105-246-198.zone3.bethere.co.uk) |
| 20:08.28 | LiNeTuX | How do you hang up a Zap channel from the CLI? |
| 20:09.56 | Strom_C | "zap destroy channel" IIRC |
| 20:10.04 | LiNeTuX | thz |
| 20:10.29 | LiNeTuX | y, you're right. "zap destory channel 26" for me |
| 20:10.57 | kannan | is it possible to have both MYSQL_FRIENDS in sip.conf, as well as other sip peers defined in clear text? |
| 20:11.02 | *** join/#asterisk Torikun (n=386426@75-171-135-108.hlrn.qwest.net) |
| 20:11.21 | Torikun | Hello |
| 20:12.20 | ManxPower | I thought zapdestroy killed the channel until the next load/unload of chan_zap. |
| 20:12.57 | Torikun | Can I use asterisk with Sip phone? |
| 20:13.05 | Strom_C | Torikun: yes |
| 20:13.22 | *** part/#asterisk Geert (i=geert@irssi/staff/geert) |
| 20:13.46 | Torikun | I want to make a PBX but I do not have the hardware |
| 20:13.49 | [TK]D-Fender | ManxPower: it does |
| 20:14.06 | [TK]D-Fender | Torikun: Go read... THE BOOK. |
| 20:14.08 | [TK]D-Fender | ~book |
| 20:14.12 | 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, or see ~buybook |
| 20:14.15 | Torikun | ty |
| 20:14.25 | [TK]D-Fender | Torikun: And you'll have a better idea of what * is capable of |
| 20:14.28 | Torikun | ty |
| 20:15.04 | ManxPower | prefers to call it The Good Book |
| 20:15.13 | Strom_C | likes to hit people with it |
| 20:15.42 | ThoMe | is it posible with asterisk+iax+hylafax multiplle fax recieved? |
| 20:15.56 | hsv-al | why did i need to add a /s string |
| 20:16.12 | hsv-al | after an IAX2 string to make a call using zoiper -> asterisk -> outside world(iax2) |
| 20:16.56 | *** join/#asterisk BlackBishop (i=adi@unaffiliated/blackbishop) |
| 20:17.03 | hsv-al | exten => 501,1,Dial(IAX2/guest@bleh.foo.bar.com/s) |
| 20:17.10 | hsv-al | wasn't working until I added /s at the end |
| 20:17.12 | BlackBishop | any simple how to on how to get 2 users to register and then be able to talk to eachother ? ( using sip ) |
| 20:17.32 | ManxPower | hsv-al: that sends the call to extension "s" on bleh.foo.bar.com |
| 20:18.11 | hsv-al | manxpower |
| 20:18.17 | hsv-al | its not required for same style, but using SIP |
| 20:18.37 | ManxPower | it COULD be required -- it all depends on the destination sustem |
| 20:18.38 | [TK]D-Fender | BlackBishop: Easy enough. Get started by reading the book, and here's another simple level guide for "inspiration" value : |
| 20:18.40 | [TK]D-Fender | ~book |
| 20:18.40 | 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, or see ~buybook |
| 20:18.40 | ManxPower | and system too |
| 20:18.43 | [TK]D-Fender | ~jerjerguide |
| 20:18.43 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 20:21.24 | Torikun | IS A LINKSYS pap2 a PSTN ? |
| 20:21.37 | Strom_C | no |
| 20:21.49 | ManxPower | ~fxofxs |
| 20:21.49 | jbot | i guess fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 20:21.58 | Torikun | ah |
| 20:21.59 | ManxPower | PSTN is a generic term for the global telephone network |
| 20:22.04 | Strom_C | the PSTN is the entire worldwide telephone network |
| 20:22.43 | ManxPower | hits Strom_C with "it" |
| 20:22.52 | hsv-al | manxpower |
| 20:23.06 | Strom_C | yum |
| 20:23.14 | hsv-al | what's the reason in extensions.conf, a sip extension doesnt need / at the end, but the iax2 target did, purely a remote server configuration? |
| 20:23.25 | hsv-al | : /s rather |
| 20:23.44 | ManxPower | hsv-al: show me a sample SIP Dial line you have. |
| 20:23.58 | ManxPower | without variables please |
| 20:24.24 | hsv-al | exten => 502,1,Dial(SIP/5602@x.x.x.x) |
| 20:24.25 | ManxPower | Dial(SIP/extension@peer) Dial(IAX2/username@peer/extension) |
| 20:24.27 | hsv-al | worked without /s |
| 20:24.46 | ManxPower | you are giving it 5602 instead of s |
| 20:24.49 | hsv-al | but Dial(IAX2/guest@blah.foo.bar) |
| 20:24.51 | hsv-al | need bar/s) |
| 20:25.13 | ManxPower | You just don't get it. They are differently formatted lines, the info is in different places. |
| 20:25.47 | hsv-al | typos |
| 20:25.56 | hsv-al | exten => 502,1,Dial(IAX2/guest@x.x.x.x/s) - worked |
| 20:25.58 | hsv-al | exten => 502,1,Dial(IAX2/guest@x.x.x.x) - did not work |
| 20:26.00 | ManxPower | The /extension at the end of the IAX2 Dial line is the destination extension to request. The stuff before the @ in the SIP Dial is the destination extension to request |
| 20:26.11 | ManxPower | correct, you are not requesting a destination extension in your 2nd one |
| 20:26.41 | ThoMe | have "exten => _X.,1,Dial(IAX2/iaxmodem)" |
| 20:26.45 | ManxPower | x.x.x.x could easily require /12345 or /fred at the end of it, it is all how that side is set up |
| 20:27.03 | ThoMe | i would like, if "iaxmodem" is busy then iaxmodem1, if iaxmodem1 busy then iaxmodem2 |
| 20:27.06 | hsv-al | I guess I havent came to the dial language section yet |
| 20:27.07 | ThoMe | iaxmodem3. .iaxmodem3 |
| 20:27.09 | ThoMe | is it posible? |
| 20:27.09 | hsv-al | to pick up on this |
| 20:27.30 | [TK]D-Fender | ThoMe: Yes, you can have 20 if you want to. |
| 20:27.31 | ThoMe | any ideas for me? |
| 20:27.42 | ThoMe | [TK]D-Fender: ah fien. have 4 add in iax.conf |
| 20:27.50 | ManxPower | hsv-al: (3:24:25 PM) ManxPower: Dial(SIP/extension@peer) Dial(IAX2/username@peer/extension) |
| 20:28.10 | ThoMe | [TK]D-Fender: and hylafax has 4. now is for me the question: how i can if if if if on the dialplan? |
| 20:28.20 | [TK]D-Fender | ThoMe: The doncs on IAXModems site and in the tarball, etc show you how to add 2. Use some brains and realize that may as well have been 20. |
| 20:28.42 | [TK]D-Fender | ThoMe: if its in use, its no different than any other device you'd call. |
| 20:28.58 | [TK]D-Fender | ok, heading home. Later all |
| 20:45.50 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:46.52 | hsv-al | yep |
| 20:50.54 | *** part/#asterisk vortex` (n=vortex@deathstar.juicedigital.net) |
| 20:53.42 | *** join/#asterisk Amun (n=Jonsi@c-71-205-221-174.hsd1.mi.comcast.net) |
| 20:59.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:59.37 | *** part/#asterisk Torikun (n=386426@75-171-135-108.hlrn.qwest.net) |
| 21:03.03 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:03.17 | anonymouz666 | [TK]D-Fender at home! |
| 21:03.43 | anonymouz666 | how fast |
| 21:08.43 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:08.57 | BlackBishop | ... ok .. I got them to call one another .. only problem .. there's no sound ... and ekiga keeps flashing "PCMU" on input/output |
| 21:11.11 | BlackBishop | <PROTECTED> |
| 21:11.11 | BlackBishop | <PROTECTED> |
| 21:11.18 | BlackBishop | <PROTECTED> |
| 21:11.18 | BlackBishop | <PROTECTED> |
| 21:11.25 | BlackBishop | <PROTECTED> |
| 21:11.30 | BlackBishop | ... not working .. no sound .. |
| 21:11.41 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 21:11.57 | [TK]D-Fender | BlackBishop, where is the call coming from? |
| 21:13.12 | BlackBishop | localnetwork .. |
| 21:13.50 | BlackBishop | dexter ( me ) is 10.12.0.241 and lovebeer ( the computer next to me ) is 10.12.0.243 |
| 21:15.49 | [TK]D-Fender | BlackBishop, well we can see the call is coming in unauthenticated. Why isn't it being matched against a peer in sip.conf? |
| 21:16.10 | [TK]D-Fender | BlackBishop, Dial("SIP/10.3.0.123-08c52288", "SIP/lovebeer") <- when you see it coming from an IP like that you know its un-authed. |
| 21:16.23 | BlackBishop | hmmm |
| 21:16.26 | BlackBishop | but I am authed ! |
| 21:16.36 | [TK]D-Fender | BlackBishop, No, the first leg clearly isn't |
| 21:16.53 | BlackBishop | <PROTECTED> |
| 21:16.53 | BlackBishop | <PROTECTED> |
| 21:17.24 | [TK]D-Fender | BlackBishop, SIP/10.3.0.123-08c52288 <-- this is NOT 10.12.0.241 |
| 21:17.28 | truent | . |
| 21:17.36 | [TK]D-Fender | BlackBishop, Time to get a new pair of eyes |
| 21:17.50 | BlackBishop | 10.3.0.123 is the asterisk server both me and lovebeer are connected |
| 21:18.10 | BlackBishop | same thing if lovebeer tries to call me .. |
| 21:18.25 | [TK]D-Fender | BlackBishop, wait, you're running a softphone on your * server? |
| 21:18.39 | [TK]D-Fender | BlackBishop, but regardless if you see an IP, its NOT coming in authed. |
| 21:18.39 | BlackBishop | no |
| 21:18.51 | BlackBishop | so .. how do I make it go "authed" ? |
| 21:19.03 | [TK]D-Fender | BlackBishop, pastebin your sip.conf masking only passwords. |
| 21:19.04 | [TK]D-Fender | ~pb |
| 21:19.04 | 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 |
| 21:19.05 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 21:20.00 | *** join/#asterisk jicksta_ (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 21:20.48 | [TK]D-Fender | BlackBishop, Next, at * CLI enter "sip debug" and try the call again, and pastebin that. |
| 21:21.33 | BlackBishop | oki |
| 21:22.12 | BlackBishop | think I fixed it |
| 21:22.12 | BlackBishop | <PROTECTED> |
| 21:22.12 | BlackBishop | <PROTECTED> |
| 21:23.10 | BlackBishop | http://rafb.net/p/w4rw9L54.html ( sip.conf ) |
| 21:23.41 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:24.05 | [TK]D-Fender | BlackBishop, never do "allow=all". Always do "disallow=all", and then only allow the specific codec(s) you need. Generally that should only be 1 codec. |
| 21:24.26 | [TK]D-Fender | BlackBishop, and where are each of your peers located relative to your server? |
| 21:25.58 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:26.59 | BlackBishop | we're all on the same network ofcourse .. |
| 21:27.09 | BlackBishop | the server is another room .. |
| 21:27.26 | pjz | anyone know what codec polycom 330s will work with? |
| 21:27.28 | [TK]D-Fender | BlackBishop, ok, so 3 different systems? |
| 21:27.34 | BlackBishop | yup |
| 21:27.41 | [TK]D-Fender | pjz, ulaw,alaw,g.729,g.723 |
| 21:27.50 | BlackBishop | asterisk is running on a linux machine .. me and lovebeer are running windows ( both using ekiga ) |
| 21:28.21 | [TK]D-Fender | BlackBishop, ok, add the fix I told you to do. Then for every system thats a PC running a softphone VERY your sound card. make sure the speakers & mic are 100% functional on each. |
| 21:28.38 | [TK]D-Fender | BlackBishop, the stupidist things happen when people don't check their PC's sound settings |
| 21:29.07 | BlackBishop | http://mail.d3xt3r01.tk/~dexter/sip_debug ( this is the sip_debug ) |
| 21:29.16 | *** join/#asterisk jamuse (n=josh@bzq-219-135-48.static.bezeqint.net) |
| 21:29.25 | BlackBishop | the sound works .. I did the sound tests on both ! |
| 21:29.55 | jamuse | any ideas what could be causing the following error message: pbx.c:2979 ast_register_application: Already have an application 'Directory' |
| 21:30.01 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 21:30.03 | BlackBishop | ok, what's a good codec for local lan ? |
| 21:30.12 | [TK]D-Fender | BlackBishop, "ulaw" |
| 21:30.22 | [TK]D-Fender | BlackBishop, which is indeed what it seems to have negotiated. |
| 21:31.43 | [TK]D-Fender | BlackBishop, Verify your mic's, and recording sources |
| 21:32.02 | BlackBishop | I did, 3 times ! |
| 21:32.11 | BlackBishop | Is it ok that PCMU to flicker ? |
| 21:33.11 | BlackBishop | that sip debug log all happened in a couple of secconds. and it kept getting bigger as the flicker continued. |
| 21:34.47 | [TK]D-Fender | BlackBishop, set for GSM on all sides & sip.conf. test again |
| 21:35.03 | [TK]D-Fender | BlackBishop, and set "canreinvite=no" for your peers |
| 21:36.33 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 21:36.51 | Titanous | does anyone know whether RAGI works with 1.4? |
| 21:36.51 | BlackBishop | seems not to flicker anymore |
| 21:37.21 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:41.38 | BlackBishop | OK .. now I can hear stuff .. but nothing that I can understand .. |
| 21:44.21 | BlackBishop | any other codec that should work in a lan and on a net ( dsl for example ) |
| 21:44.22 | BlackBishop | ? |
| 21:45.50 | [TK]D-Fender | BlackBishop, Ok, I've had some real quality issues with ekiga before. Go install Zoiper and try again. |
| 21:45.52 | [TK]D-Fender | ~zoiper |
| 21:45.52 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 21:47.48 | BlackBishop | Ok, lets see. |
| 21:51.03 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 21:57.00 | BlackBishop | while lovebeer is installing zoiper .. another q: is asterisk secure ? I mean, if someone can sniff the packets, is it using ssl and stuff ? |
| 21:57.41 | [TK]D-Fender | BlackBishop, Negotiation is MD5 (not the most secure), but general voice isn't |
| 21:57.53 | [TK]D-Fender | BlackBishop, * 1.6 should offer TLS, etc. |
| 21:58.15 | [TK]D-Fender | BlackBishop, This is for SIP. IAX2 offers encryption IIRC. |
| 22:01.41 | BlackBishop | ... ooookie .. |
| 22:02.22 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 22:03.04 | pjz | anyone have a recommended SIP Application version number for polycom 330s ? They came with 2.1.2 |
| 22:03.24 | pjz | recommended to work well with Asterisk, or versino to avoid or anyhing? |
| 22:03.57 | russellb | i always run the latest on my phone :) |
| 22:04.28 | pjz | fair enough; I know in the past on older phones there were newer versions that didn't work but older versions that did |
| 22:04.34 | pjz | so I figured I'd ask to be safe |
| 22:07.47 | [TK]D-Fender | pjz, if it works and you don't a specific reason to change, then don't |
| 22:11.24 | pjz | [TK]D-Fender: ah, but it's not working, and I'm trying to make it work, and wondering if upgrading would help |
| 22:13.26 | [TK]D-Fender | pjz, Wow.... could you give a little LESS detail? |
| 22:14.07 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 22:14.37 | pjz | [TK]D-Fender: It's not registering with the call director, it *is* managing to get message-waiting status though |
| 22:15.15 | pjz | [TK]D-Fender: and feel free not to answer, I've already been told to buzz off here since it's an AA50 |
| 22:15.16 | *** join/#asterisk s0lid (n=s0lid@124.106.208.232) |
| 22:15.28 | pjz | er, since the server is an AA50 |
| 22:15.41 | pjz | nevermind that I can ssh into it and read the config files |
| 22:16.01 | [TK]D-Fender | pjz, the firmware works. Your configs don't. |
| 22:16.27 | pjz | [TK]D-Fender: fair enough. But I can't seem to even manually config it to work via the web interface on the phone |
| 22:16.33 | pjz | I've got softphones working just fine |
| 22:16.54 | pjz | so it's got to be some kind of config issue with the phone itself, no? |
| 22:17.11 | [TK]D-Fender | pjz, general consensuc is anyone attempting to configure a polycom phone via its web interface should be dragged out and shot. |
| 22:17.23 | [TK]D-Fender | pjz, And the "error" could be on either side |
| 22:17.36 | [TK]D-Fender | pjz, the fact they don't AGREE is the problem |
| 22:20.42 | pjz | [TK]D-Fender: http://pastebin.ca/1039049 |
| 22:21.19 | pjz | ack, gotta run, leaving this so I can read your answer later |
| 22:21.49 | [TK]D-Fender | pjz, registersip=yes - wrong |
| 22:22.11 | [TK]D-Fender | pjz, but the whole pile doesn't tell me anything useful. |
| 22:31.22 | *** join/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 22:32.18 | Bananaskin | hey [TK]D-Fender |
| 22:32.54 | Bananaskin | [TK]D-Fender, you still looking for the ability to bulk fax documents ?? |
| 22:33.18 | [TK]D-Fender | Bananaskin : yup |
| 22:33.49 | Bananaskin | check out - astassistant.com - bottom of page |
| 22:34.30 | Bananaskin | the Hylaassistant |
| 22:35.32 | [TK]D-Fender | thanks. will look at shortly. |
| 22:35.38 | Bananaskin | nps |
| 22:38.09 | *** join/#asterisk xzcvczx (n=simon@gentoo/user/xzcvczx) |
| 22:38.44 | *** join/#asterisk juanjoc (n=juanjoc@host65.190-225-194.telecom.net.ar) |
| 22:38.57 | xzcvczx | is there any reason why i would have a ~45ms time to sip peers on the internet, but a 101ms time to a softphone on a dynamic host on a 100mbit network? |
| 22:39.23 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 22:39.31 | Qwell | how are you measuring it? |
| 22:40.02 | xzcvczx | sip show peers |
| 22:41.03 | file | a SIP stack may give an OPTIONS packet a lower priority in handling it |
| 22:41.04 | *** join/#asterisk Zipper_32 (n=None@d75-155-200-227.bchsia.telus.net) |
| 22:42.42 | xzcvczx | file: wouldnt that affect all the peers not just the softphone? |
| 22:43.16 | file | I'm not talking about the SIP stack in Asterisk, I'm talking about the remote one |
| 22:43.22 | file | disappears |
| 22:43.35 | xzcvczx | ah ok cheers |
| 22:44.36 | *** join/#asterisk anthm (n=anthm@mb50736d0.tmodns.net) |
| 22:50.25 | deeperror | xzcvczx, did you try mtr to each ip maybe something slowing down on the network? |
| 22:55.50 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 22:56.36 | andrewy | how can I do something like Dial(SIP/foo,A(bar)), but keep the call bridged while bar is playing? |
| 23:00.24 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:01.51 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:02.42 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 23:02.59 | Titanous | Is it possible to use AMI to create a 3-way call? |
| 23:03.08 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 23:03.23 | CrashSys | Anyone familiar with what kind of T1/E1/J1 they would have in Hong Kong? |
| 23:03.41 | russellb | Titanous: originate a few calls into a MeetMe |
| 23:03.52 | russellb | Titanous: or redirect 2 channels in an existing call into a MeetMe |
| 23:03.57 | russellb | and orignate the 3rd into the MeetMe |
| 23:04.09 | Titanous | ok, gotcha |
| 23:06.56 | ManxPower | You CANNOT infer network latency by using qualify=n |
| 23:13.35 | *** join/#asterisk RoyK (n=roy@ip-82-48-149-91.dialup.ice.no) |
| 23:14.11 | *** join/#asterisk RoyK (n=roy@ip-82-48-149-91.dialup.ice.no) |
| 23:20.31 | xzcvczx | deeperror: mtr and ping both give me 0.* ms for pings to my local softphone computer |
| 23:26.28 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:26.53 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 23:30.11 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 23:32.24 | esk | does anyone know why SMDI isn't used for e&m signaled lines? |
| 23:39.43 | codefreeze-lap | Interesting but random thought: There are 25 people that can call me "Uncle". And if you count their spouses, 26-- a number that could grow to 50 with time. |
| 23:42.35 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 23:49.33 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 23:58.29 | *** part/#asterisk Zipper_32 (n=None@d75-155-200-227.bchsia.telus.net) |
| 23:59.56 | Qwell | uncle codefreeze? |