| 00:02.44 | Ryushin | I'm compiling wanpipe from sangoma. What is: API for Level2 SS7 protocol? |
| 00:06.19 | *** join/#asterisk maxbel (n=maxi@r190-0-143-19.dialup.adsl.anteldata.net.uy) |
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| 00:41.08 | nibbler_de | can i define multiple sip 'server' ips in sip.conf with different ports? |
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| 00:53.44 | bkw_ | I know they have 32k AAL2 and RFC3551 |
| 00:53.47 | bkw_ | in trunk only |
| 00:54.42 | Qwell | bkw_: patches welcome :P |
| 00:59.30 | *** join/#asterisk delmar (n=delmar@203-114-178-98.dsl.sta.inspire.net.nz) |
| 01:02.39 | Ryushin | I'm setting up my PRI, what would be the protocol to use for wanpipe? |
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| 01:20.29 | bkw_ | Qwell, as long as asterisk uses that bitmask for codecs they can't support it |
| 01:20.41 | bkw_ | it takes 8 codec bits to register all the various G726 versions |
| 01:20.59 | Qwell | yeah, there are like 2 free |
| 01:21.07 | Qwell | 3? dunno, something |
| 01:23.01 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:33.29 | bkw_ | FreeSWITCH now supports the G.726 8 Kilohertz Audio codec! (All four bitrates and AAL2 and RFC3551 formats) |
| 01:33.50 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.169) |
| 01:38.46 | Corydon76-home | Qwell: we'll probably switch to a 128 bit codec mask at some point... 64 for audio, 32 for image, 32 for video |
| 01:40.42 | bkw_ | I don't see having 32 image formats |
| 01:40.45 | bkw_ | nor 32 video formats |
| 01:40.58 | bkw_ | using a bitmask will always limit you. |
| 01:41.10 | Corydon76-home | Using a bitmask is quick, though |
| 01:41.42 | Corydon76-home | That's why we use a bitmask to store that information |
| 01:41.56 | bkw_ | well asterisk still does the RTP and SDP wrong |
| 01:42.12 | Corydon76-home | bkw_: you made your bed. Now you have to lie in it |
| 01:42.24 | bkw_ | I didn't make the bed.. :P |
| 01:42.41 | bkw_ | the issue is you have to keep the remote sides RTP map |
| 01:42.59 | bkw_ | Asterisk doesn't really do that correctly. |
| 01:43.20 | bkw_ | i'm sure we'll have to patch it at some point and submit that fix |
| 01:43.41 | Qwell | now girls |
| 01:43.43 | Corydon76-home | And I'm sure you'll bellyache about it the whole time |
| 01:43.55 | bkw_ | Nope we'll fix it so it does it right |
| 01:44.06 | bkw_ | then the patch for it will sit and rot in the tracker as usual. |
| 01:45.20 | Corydon76-home | We all know you're an advocate of openpbx^H^H^H^H^H^H^HFreeswitch, but we'd appreciate you advocating on your own channel |
| 01:45.55 | bkw_ | but we have to interact |
| 01:46.05 | bkw_ | and doing so requires us all to work together |
| 01:46.33 | Corydon76-home | Interaction is fine. But please leave the advocacy of your pet project out |
| 01:48.06 | Qwell | so, completely offtopic |
| 01:48.16 | Qwell | anybody know how close White Castle is to the Boston airport? :P |
| 01:48.29 | bkw_ | and I'll be at Astricon |
| 01:48.40 | Qwell | erm, not Boston.. Cincinnati |
| 01:48.42 | bkw_ | Qwell, i'll have to buy you a drink.. but JUST ONE because I know you can't handle it |
| 01:48.47 | Qwell | bkw_: :P |
| 01:48.52 | bkw_ | Cincinnati airport is ass |
| 01:48.55 | file | I think one is pushing it... |
| 01:48.56 | file | try half |
| 01:49.01 | Qwell | bkw_: which is why I want to find a white castle |
| 01:49.12 | bkw_ | but they atleast have charge stations for laptops |
| 01:51.28 | *** join/#asterisk flujan (n=flujan@201-42-69-138.dsl.telesp.net.br) |
| 01:56.15 | Ryushin | Are the Polycom phones such as the 601 supposed to display the users name? I've configured the phone1.cfg file, entry reg.1.displayName="Name Here" but it doesn't display on the main screen. Just the date and time. |
| 02:06.06 | *** join/#asterisk ShockValue (n=john@71-36-43-11.tukw.qwest.net) |
| 02:06.57 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.05 | ShockValue | hi all.. i was just reading up on asterisk and voip (*newb alert*) and was interested in doing the math to see if i could save some money and/or increase my cool-ness-factor |
| 02:07.30 | ShockValue | what kind of provider can i use as a home-user for voip+asterisk? |
| 02:08.05 | Strom_M | broadvoice is fairly popular |
| 02:08.45 | ShockValue | so its kind of POT -> broadvoice -> internet -> asterisk -> my phone? |
| 02:09.21 | Ryushin | What is SS7 support from Sangoma? Do I need it for voice calls from a PRI? |
| 02:16.57 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 02:19.18 | Corydon76-home | You do not need SS7, no. |
| 02:19.39 | rene- | ShockValue: voice pulse |
| 02:19.44 | rene- | voicepulse connect |
| 02:20.54 | ShockValue | thanks rene |
| 02:22.03 | rene- | np |
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| 02:24.51 | Ryushin | thanks Corydon76-home. I'll remove it then. |
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| 02:48.18 | rene- | shit |
| 02:49.24 | ShockValue | any opinion on telesip/voipxpress ? |
| 03:09.36 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
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| 03:09.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:09.54 | teknoprep | is there a way to dial a number that would change the incomming route for a number from say 1 ivr to another ivr? |
| 03:10.01 | teknoprep | say a feature code? |
| 03:10.28 | teknoprep | so at closing of shop i can dial this number.. and it tells all incomming callers on a new IVR that the store is closed |
| 03:10.34 | teknoprep | then at open i can go back to being open |
| 03:11.11 | shodan-away | anyone has an idea why I'm getting this ? Sep 2 22:54:13 WARNING[5444]: chan_zap.c:1419 zt_train_ec: Unable to request echo training on channel 1 |
| 03:18.51 | shodan-away | I tried mark2/3 steve2 and kb1 so far I always get this :\ |
| 03:20.57 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
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| 03:28.51 | Vco | anyone happen to recall where to find the astricon (2005?) photos are with everyone out on the town at night? |
| 03:29.13 | rene- | shodan: i think the card you are using will not do echo cancellation in software |
| 03:29.22 | Qwell | Vco: who, out where? |
| 03:29.33 | rene- | Qwell: my E1 card sucks |
| 03:29.36 | Vco | trying to remember |
| 03:29.46 | Qwell | Vco: What was the scene? |
| 03:29.46 | rene- | junk-y took those pics |
| 03:30.11 | Vco | ya |
| 03:30.19 | rene- | he had an online gallery |
| 03:30.53 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 03:31.11 | shodan-away | rene-, it's a x100p clone ? well , I managed to make the error go away by setting echotraining=yes instead of echotraining=4000 , but there is still way to much echo to be usable , it's as if the echo canceller is not working at all |
| 03:31.22 | rene- | how many iax2 /g729 calls over a 384 kbps symetrical dsl line? |
| 03:31.26 | Qwell | his gallery site is down |
| 03:31.43 | shodan-away | right now I'm using MG2 and echocancel=128 and I can hear myself clearly , but a good 100ms later |
| 03:31.44 | Vco | k |
| 03:32.24 | Qwell | Vco: http://www.midsouthmarketplace.com/~krice/gallery/ |
| 03:33.45 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 03:34.58 | Qwell | somegeek: your connection sucks..or something |
| 03:35.40 | shodan-away | other than this echo problem my system is ready to go into production , so close , yet so far :( |
| 03:39.29 | shodan-away | is there I way I can find out at least if the echo canceller is ineffective or non-working ? |
| 03:44.54 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 04:02.34 | rene- | shodan-away: how many cards are you using?? |
| 04:03.18 | shodan-away | there are two in the box , but only one is being used |
| 04:03.27 | rene- | try using just one |
| 04:03.33 | shodan-away | ok |
| 04:03.47 | rene- | and see it isnt sharing irqs with anything else |
| 04:03.52 | rene- | lspci -vb |
| 04:06.02 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 04:14.13 | shodan-away | hmm , I remove the 2nd card , but now nothing's working , not good |
| 04:15.28 | rene- | what do u mean? |
| 04:15.39 | rene- | nothing's working? |
| 04:16.54 | shodan-away | I took my sip phone , dialed a line , in the console I have all the normal executing dial ... , called 1/xxxxxx , zap/1-1 answered ... but the number I was calling didn't ring and didn't answer |
| 04:17.22 | shodan-away | then I tried calling my box , it answered but instead of playing the message like the console said it was doing all I had was silence |
| 04:17.42 | shodan-away | just figured out , it's my txgain/rxgain , the old values are not doing the same thing now |
| 04:17.57 | shodan-away | I set them both to 0 now it's working again |
| 04:18.33 | shodan-away | still a lot of echo , also my fxo is sharing irq with the vga and lan |
| 04:18.40 | shodan-away | the echo is still there |
| 04:19.38 | shodan-away | strange ztmonitor is no longer working "unable to open /dev/dsp" |
| 04:22.33 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 04:23.45 | mocker | Anyone have any experience with ViaTalk? |
| 04:23.53 | bkw_ | ViaTalk? |
| 04:23.58 | mocker | Some ITSP.. ;) |
| 04:24.12 | mocker | Just trying to read through reviews of voip providers. |
| 04:38.48 | CunningPike | mocker: They're all crap ;) |
| 04:38.51 | ShockValue | i was just reading up on iax.cc. they seem damn cheap |
| 04:39.01 | ShockValue | makes me scared :) |
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| 05:25.28 | dlynes_laptop | Good evening, cp |
| 05:29.35 | shodan-away | damn this echo :( |
| 05:31.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:35.10 | dlynes_laptop | damn this echo :( |
| 05:35.37 | dlynes_laptop | what echo? |
| 05:35.38 | dlynes_laptop | echo? |
| 05:36.01 | shodan-away | heheh |
| 05:36.40 | *** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.wa.comcast.net) |
| 05:38.25 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 05:38.30 | Rahail | any one here who can make softphone |
| 05:38.46 | dlynes_laptop | why make one? |
| 05:38.53 | dlynes_laptop | there's plenty out there already |
| 05:39.05 | Rahail | I need one with codec g23 |
| 05:39.10 | coppice | if you heat them, they usually soften |
| 05:39.12 | Rahail | g723 and hardcode my serve rip |
| 05:39.23 | Rahail | g723 and hardcode my server ip |
| 05:39.42 | dlynes_laptop | don't most windows softphones already do g723? |
| 05:39.43 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:39.58 | Rahail | i cant find any of them |
| 05:40.07 | dlynes_laptop | ~docs |
| 05:40.09 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:40.10 | coppice | illegally, apparantly |
| 05:40.16 | Rahail | and i dont know how where to get it who can hardcode the server info |
| 05:40.23 | Rahail | and disable register option |
| 05:40.26 | dlynes_laptop | coppice: I thought they were borderline legal, or something? |
| 05:40.47 | Rahail | how |
| 05:41.03 | coppice | its weird. MS only has a licence for that codec to be used with netmeeting, yet its in the general codec pool |
| 05:41.48 | dlynes_laptop | so use netmeeting for your softphone |
| 05:41.51 | dlynes_laptop | and then it's legal |
| 05:42.03 | coppice | yes |
| 05:42.14 | dlynes_laptop | problem solved |
| 05:42.42 | coppice | "use netmeeting" and "problem solved" don't fit well together :-) |
| 05:43.22 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:48.18 | Rahail | what aboug g729 |
| 05:48.21 | Rahail | thats legal right ? |
| 05:49.18 | dlynes_laptop | none of them are, technically |
| 05:49.54 | shodan-away | maybe you live somewhere software patents don't exist ? |
| 05:50.17 | benjk | g729 has been outlawed since it is a threat to homeland security, terrorists use it to blow up telephone exchanges |
| 05:50.39 | Rahail | eh |
| 05:51.00 | Rahail | I cant have my user use ualw alaw |
| 05:51.03 | benjk | shodan, it doesn't matter where you live |
| 05:51.07 | Rahail | they all are on dialup |
| 05:51.16 | *** join/#asterisk marc7 (n=marc@CPE000c4127893d-CM0011ae926e64.cpe.net.cable.rogers.com) |
| 05:51.23 | dlynes_laptop | Rahail: so pay for the codec then |
| 05:51.26 | dlynes_laptop | Rahail: sheesh |
| 05:51.31 | benjk | g729 patents will apply unless you are in a country that has not signed up to any trade agreements whatsoever |
| 05:51.46 | Rahail | sure i will |
| 05:51.55 | benjk | local patent laws do not matter |
| 05:52.07 | bkw_ | hrm wtf is this |
| 05:52.19 | coppice | G.729 is patented pretty much everywhere, anyway |
| 05:53.03 | bkw_ | but is it enforceable |
| 05:53.24 | bkw_ | well hell g723.1 was in the digium CVS for ages |
| 05:53.25 | benjk | maybe some place like pitcairn or tuvalu or antarctica |
| 05:53.30 | bkw_ | infact you can still find it out there in the wild |
| 05:53.58 | bkw_ | anyway |
| 05:54.02 | bkw_ | i'm tired |
| 05:54.03 | coppice | why would it not be enforceable? |
| 05:54.05 | benjk | also, patents don't apply in space |
| 05:54.33 | coppice | dunno. you'll have to check international treaties for that |
| 05:54.34 | benjk | so if you use it between two satellites only or for moon to mars links or some stuff like that, you'll be ok |
| 05:54.46 | bkw_ | coppice, honestly I don't agree with patents in those cases. |
| 05:54.51 | coppice | i would guess areospace companies have ensured that is not true |
| 05:55.01 | bkw_ | but i'm tired |
| 05:55.02 | bkw_ | i'm going to bed |
| 05:55.15 | benjk | codecs should not be patentable anyway |
| 05:55.19 | benjk | not in any form |
| 05:55.33 | benjk | they are nothing other than mathematical formulas |
| 05:55.38 | coppice | bkw_: who asked for your agreement? :-) |
| 05:55.41 | Rahail | any one make softphone here |
| 05:55.53 | benjk | and mathematical formulas are explicitly excluded from patentability |
| 05:56.22 | benjk | its all hypocrisy |
| 05:56.27 | coppice | benjk: only pure maths is excluded. applied has never been |
| 05:56.42 | benjk | still, its hypocrisy |
| 05:57.38 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 05:57.45 | benjk | because all it does is get you patentability of formulas through the back door |
| 05:58.14 | coppice | most patents could be considered like that |
| 05:58.17 | benjk | they should rather allow patenting of everything no matter how stupid |
| 05:58.26 | benjk | but then limit the duration to 1 or 2 years |
| 05:58.34 | benjk | without any possibility to extend |
| 05:58.51 | benjk | petty patents |
| 05:59.07 | coppice | duration is the problem. originally 15 years was a couple of product spins, and that's all they should get now |
| 05:59.18 | benjk | no checking whatsoever, pay a one-time fee get your free patent |
| 06:00.03 | benjk | then they should raise the bar for the conventional patents and be really really really anal on the requirement for non-obviousness |
| 06:00.45 | coppice | I see arguments why algorithms can't be patented, which I agree with. however, if you disagree with patenting a codec, how can you agree with patenting anything? |
| 06:01.15 | benjk | I think patents are misunderstood by most people |
| 06:01.38 | benjk | you have to look at the reason why patents were introduced in the first place |
| 06:01.55 | benjk | or more precisely why they were limited the way they were originally |
| 06:02.19 | coppice | including you, it seems. the algorithmic patents on G.729 only exist in the US. the patents on the "how to solve problem X in voice compression" patents, which don't mention software at all, have no limit around the world |
| 06:02.20 | benjk | the only motivation for granting patents is to give incentive to inventors to publish |
| 06:02.30 | benjk | no |
| 06:02.41 | benjk | what I am getting at is the motivation to publish |
| 06:02.47 | marc7 | I have a Cisco 7960, I'm just trying to get it setup again with an asterisk server of mine... and for some reason the phone and asterisk can't pair up. I've removed the entry in "sip.conf" for my phone's authentication string, and I've enabled "allowgues, allguest, insecure=very, autocreatepeer=yes, allowexternalinvites=yes" |
| 06:02.57 | marc7 | rebooting the phone gives asterisk the console message: Registered SIP 'marc' at 123.45.67.8 port 5060 expires 3600 // Saved useragent "Cisco-CP7960G/8.0" for peer marc |
| 06:02.59 | benjk | with g729, it was supposed to be a standard used by many different entities |
| 06:03.09 | benjk | so you have to publish the spec anyway |
| 06:03.10 | marc7 | but for all intents and purposes, it doesn't look like my phone's locked uon |
| 06:03.14 | suma | hey any guys around here know how to obtain g729 released by a dutch guy ! |
| 06:03.19 | benjk | consequently there should be no patent granted for it |
| 06:03.19 | marc7 | any ideas as to what I can do to troubleshoot from ehre? |
| 06:03.31 | benjk | because no incentive to publish is needed |
| 06:03.37 | benjk | that should be the only criteria |
| 06:03.43 | suma | i know all the patent stupids, i just want to know whether i can make it to work with asterisk |
| 06:03.49 | benjk | the only question should be this: |
| 06:04.05 | benjk | "Will the public domain suffer if this technology is kept secret?" |
| 06:04.22 | benjk | if the question is yes, then a patent should be granted, otherwise not |
| 06:04.29 | suma | is this asterisk channel or patent channel ! ;) |
| 06:04.45 | marc7 | agreed. benjk is scrolling my perfectly legitimate question off the buffer. ;-) |
| 06:04.47 | dlynes_laptop | #frivolouspatentsrwe |
| 06:04.54 | benjk | because that is the justification for the patent system |
| 06:05.08 | benjk | that inventors have incentive to not keep their inventions secret |
| 06:05.13 | coppice | depends which history book you read. |
| 06:05.22 | benjk | and that should be the only yardstick |
| 06:05.22 | suma | marc7: do you know where to obtain the g729 equivalent codec for asterisk ? |
| 06:05.43 | benjk | no it doesn't |
| 06:06.14 | marc7 | suma: i know that my phone won't even bind to my asterisk server right now, and the g729 codec is irrelevant to my problem |
| 06:06.27 | suma | coppice: do you know where to obtain the g729 equivalent codec for asterisk ? as seen in the mailing lists |
| 06:06.27 | benjk | the reason why patents were limited to novelty and inventive step and by duration was that early patents which were granted as favours got out of hand and led to economic stagnation |
| 06:06.27 | coppice | there are books that say a major driver behind early patent law was the need for things like manuals for big industrial systems to be published. the makers needed protection from everything they published being copied |
| 06:07.14 | *** join/#asterisk marc7 (n=marc@CPE000c4127893d-CM0011ae926e64.cpe.net.cable.rogers.com) |
| 06:07.15 | benjk | that's correct, hence the incentive to publish by granting a time limited monopoly |
| 06:08.01 | benjk | but if the product you are making is by its very nature such that you will have to publish the specs anyway, then there is no need for an incentive |
| 06:08.01 | benjk | consequently there should be no patent granted |
| 06:08.07 | suma | can you please guys rename the asterisk channel to patent channel ;) |
| 06:08.51 | benjk | thus most communication protocols intended for interoperability between different products by different vendors should not be patentable |
| 06:09.09 | benjk | the incentive is already there |
| 06:09.16 | benjk | in form of economies of scale |
| 06:09.28 | marc7 | benjk: </soapbox> already |
| 06:09.55 | coppice | I doubt any of the codec we have today would exist without patents. even things like speex rides on a mass of research which was done under a patent regime. |
| 06:10.10 | benjk | I doubt that |
| 06:10.40 | coppice | anyway, the bottom line is right now codecs are patentable pretty much anywhere, those various speed up algorithms are only patented in the US |
| 06:10.52 | coppice | s/those/though |
| 06:10.54 | suma | coppice: Is the US government having any vacancy for Patent positions ? |
| 06:10.55 | benjk | I think the need for patents as an incentive to publish is nowhere near as imminent as it once was |
| 06:11.21 | marc7 | it's true though... pharmaceutical companies can't continue to dump billions of dollars into R&D without at least having a few years of exclusivity to recoup their costs in an environment that can afford it |
| 06:11.58 | coppice | the economic advantage to the late comer would be huge |
| 06:12.00 | benjk | and if the need isn't as pressing then the whole system should be downgraded to minimise the negative impact it has started to have |
| 06:12.06 | marc7 | that's a touchy issue though, there are a bajillion other issues surrounding drug-related and genomic-related patents... but right now I really need this damn asterisk server up ;P |
| 06:12.31 | shodan-away | it's all my fault sorry "[01:50:29] shodan-away maybe you live somewhere software patents don't exist ?" |
| 06:12.44 | benjk | drugs are one of the very rare examples where the duration might be justified to extend |
| 06:12.55 | coppice | make patents last 5 years and almost all issues go away |
| 06:13.02 | benjk | but for pretty much anything else, the duration should be reduced very significantly |
| 06:13.08 | benjk | indeed |
| 06:14.00 | coppice | which, as i said earler, just gets it back to the original goal - a couple of product spins |
| 06:14.10 | Qwell | alright, I've heard you guys hash this out a hundred times, and the results are the same everytime. :P |
| 06:14.23 | Qwell | "patents suck. codec patents will expire soon. yay us" |
| 06:14.44 | dlynes_laptop | g723 expires soon, right? |
| 06:14.57 | Qwell | yeah, in the next couple years |
| 06:14.57 | coppice | most important codec patents have quite a few years to run |
| 06:15.00 | benjk | if that is what you got out of the discussion, then it would rather seem you didn't get the point, but anyway |
| 06:15.04 | EyeCue | is this under the proviso they arent 'renewed' ? |
| 06:15.07 | dlynes_laptop | oh...still a couple of years |
| 06:15.30 | benjk | EyeCue, you cannot renew patents |
| 06:15.30 | EyeCue | you cant? |
| 06:15.31 | Qwell | only Disney can renew things |
| 06:15.33 | benjk | its 20 years, and 25 years for drugs |
| 06:15.49 | shodan | you can if you "significantly improve it" ? |
| 06:15.53 | coppice | G.723 isn't an interesting codec. G.729 and AMR are the important ones, and they have quite a few years to go |
| 06:15.59 | Qwell | shodan: it'd be a new patent |
| 06:16.06 | benjk | shodan, that is howver, treated as a new application |
| 06:16.51 | benjk | China could upset the codec patents quite a bit if they wanted |
| 06:16.54 | dlynes_laptop | 723 is important though, because most of the carriers are still using it |
| 06:17.15 | benjk | declare those patents illegal unless the patent holders give a blanket license royalty free, worldwide |
| 06:17.34 | EyeCue | and the benefit to patenting then? |
| 06:17.35 | EyeCue | :) |
| 06:17.37 | coppice | China has huge problems with patents |
| 06:18.25 | benjk | that's why I say, the Chinese government should just make them illegal for use and import, only waive if there is a royalty free blanket license |
| 06:18.32 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.233) |
| 06:18.35 | Kernel_Core | hi all |
| 06:19.15 | Kernel_Core | I am useing asterisk 1.2.11 , I can't change default IAX2 port from 4569 to another port ( like 5005) what is wrong ? |
| 06:19.17 | coppice | they can't do that. the whole country is under the thumb of walmart. they sell walmart a DVD player for $19 and pay $2.5 of that in royalties |
| 06:20.08 | benjk | they couldn't do it for all patents, but they could do it selectively for things that are controversial in the Western hemisphere |
| 06:20.19 | Qwell | Kernel_Core: change bindport |
| 06:20.55 | Kernel_Core | Qwell: [general] |
| 06:20.55 | Kernel_Core | bindport=5005 ; bindport and bindaddr may be specified |
| 06:21.20 | Kernel_Core | but it doesn't work , when I issue iax2 show peers ( I get 4569 ) as port |
| 06:21.38 | coppice | their trade arrangements would be completely screwed if they tried to be selective |
| 06:21.43 | dlynes_laptop | Kernel_Core: did you do an unload chan_iax2.so, load chan_iax2.so? |
| 06:21.58 | Qwell | Kernel_Core: That is what port it is connecting to on the other side, I believe |
| 06:22.10 | Kernel_Core | dlynes_laptop: i issued stop now , and restarted manually |
| 06:22.21 | Qwell | that has nothing to do with bindport |
| 06:22.23 | benjk | coppice, I still think they have enough leverage to be smart about it if they wanted to |
| 06:22.24 | dlynes_laptop | Qwell: bind means the port it's listening on, not connecting to |
| 06:22.30 | Qwell | dlynes_laptop: exactly |
| 06:22.35 | Kernel_Core | so... ? |
| 06:22.44 | Qwell | Kernel_Core: so set a port in your peer definition |
| 06:23.22 | Kernel_Core | Qwell: let bindport=4569 unchange in general section ? |
| 06:25.49 | shodan | I installed a new wire direct with no split from the phone company's terminal to my x100p card , still as much echo :( , at least now I've eleminated a wiring fault as the cause |
| 06:25.58 | shodan | .. does polarity matter ? |
| 06:26.51 | benjk | yes, if you swap the wires, you'll hear all audio backwards |
| 06:27.05 | linlin | yup |
| 06:27.37 | shodan | that's inverted not backwards .. |
| 06:27.56 | benjk | no I mean backwards |
| 06:28.36 | benjk | as in "sdrawkcab" |
| 06:29.06 | Qwell | benjk: in the US, it plays upside-down |
| 06:29.12 | shodan | hmm duh ;) |
| 06:29.16 | benjk | heh |
| 06:29.21 | linlin | it plays in spanish for me |
| 06:29.24 | Qwell | as in "umop-apisdn" |
| 06:29.59 | benjk | heh |
| 06:30.19 | benjk | if it plays in Spanish, then there has got to be a fault |
| 06:30.23 | Qwell | heh |
| 06:31.50 | shodan | is there a chance the bell guy screwed up his install ? like improper ground maybe ? |
| 06:32.09 | lowlevel | I've seen bell guys not ground things |
| 06:32.16 | Strom_C | shodan: how long is the loop? |
| 06:32.50 | shodan | less than 1 kilometer for sure |
| 06:33.00 | shodan | probably about 300 meter |
| 06:33.08 | Qwell | it's 800km |
| 06:33.28 | Strom_C | shodan: I'm willing to blame it on the fact that the x100p is...well....junk |
| 06:33.33 | Strom_C | especially the clone x100ps |
| 06:33.45 | benjk | clones? |
| 06:33.49 | benjk | hahaha |
| 06:33.53 | benjk | the naivity |
| 06:34.01 | Strom_C | oh god, don't you fucking start, benjk |
| 06:34.17 | benjk | I didn't know that a heatsink eliminates echo |
| 06:34.26 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 06:34.27 | benjk | funny that |
| 06:34.27 | linlin | $20 of cheap clone goodness |
| 06:34.53 | coppice | not all x100p clones are created equal. some have a terrible analogue front end |
| 06:34.55 | linlin | just in my box for 911 :p |
| 06:35.00 | benjk | I shall buy heatsinks for those T1 framer chips on those Zapata cards |
| 06:35.11 | benjk | maybe that'll do for echo canceling |
| 06:35.27 | Rahail | is there any one here who make oem softphone |
| 06:35.44 | benjk | Rahail, ask zoa |
| 06:35.47 | linlin | idefisk is available in an oem version |
| 06:35.55 | benjk | zoa makes idefisk |
| 06:35.57 | Rahail | costly |
| 06:36.00 | linlin | ah |
| 06:36.24 | benjk | you can always license from Xten |
| 06:36.41 | benjk | I guess that'll get you a new perspective on what's costly |
| 06:36.44 | shodan | it's probably junk , but it's so common place , isn't there someone somewhere that made a tutorial on how to tune the hybrid on these things ? should be easy it's all capacitve/solidstate ? |
| 06:37.02 | Strom_C | shodan: do you know how to use a milliwatt test> |
| 06:37.21 | linlin | Rahail spend a week learning visual basic and make your own |
| 06:37.24 | benjk | shodan, its common place and the general practise is to try it, if it works it works, if it doesn't it doesnt |
| 06:37.32 | Rahail | :( |
| 06:37.42 | linlin | its easy to create a quick one with freely available libraries i think |
| 06:38.03 | shodan | Strom_C, emm no , but I have the equipement to do it (probably) |
| 06:38.20 | Strom_C | shodan: find the number for your local milliwatt test |
| 06:38.30 | Strom_C | it must be located in the same switch that serves your telephone line |
| 06:38.33 | Kernel_Core | Qwell: I defined it in peer section and worked fine :) |
| 06:38.37 | benjk | Rahail, I know a guy who would write you one if you pay his rent for a couple of months |
| 06:38.59 | benjk | he's a student |
| 06:39.07 | Rahail | how much his rent cost |
| 06:39.19 | benjk | not sure but I am happy to put you in touch with him |
| 06:39.25 | Rahail | sure |
| 06:39.28 | Rahail | why not give it try |
| 06:39.34 | shodan | k, is that a number that the 611 guys will give me ? |
| 06:39.57 | Strom_C | shodan: depends; you might get lucky. your telco is Bell Canada? |
| 06:40.01 | shodan | yes |
| 06:40.18 | Strom_C | you can try asking |
| 06:40.26 | Strom_C | i only have test line directory access for AT&T |
| 06:42.48 | *** join/#asterisk rtyler (i=tyler@sand.geekisp.com) |
| 06:43.04 | benjk | hi rtyler |
| 06:46.58 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 06:47.55 | rtyler | Rahail: rumour has it you're looking for a softphone? :P |
| 06:48.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:48.52 | shodan | hehe called the 611 , "no sir that number is for repair man only " |
| 06:48.58 | benjk | did he fall asleep now or what? |
| 06:49.02 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:49.07 | shodan | maybe google will tell |
| 06:49.11 | Rahail | thanx |
| 06:49.47 | Strom_C | shodan: you need to make up a good reason |
| 06:50.03 | shodan | yeah , I didn't have one ;) |
| 06:50.13 | Strom_C | "I'm turning up a PBX at this customer's premises and I need to compensate for loop attenuation" tends to work |
| 06:50.14 | shodan | what would be a good reason ? |
| 06:50.25 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:50.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 06:50.45 | Strom_C | hi mog |
| 06:50.51 | Strom_C | isn't it balls-late in alabama? |
| 06:50.52 | Qwell | mog: nice interview :p |
| 06:51.01 | Qwell | Strom_C: or balls-early |
| 06:51.23 | shodan | I'll have to call tomorrow, because I don't think they'll buy that at 4h00AM |
| 06:51.31 | Strom_C | no, they wont |
| 06:51.38 | Strom_C | calling during business hours generally works :) |
| 06:51.42 | *** join/#asterisk stopher (n=austchat@cm-24-121-73-66.kingman.az.npgco.com) |
| 06:51.52 | stopher | hellloooo #asterisk :) |
| 06:52.04 | Strom_C | essentially, if you sound like you're some variety of telecommunications professional, they'll usually help you |
| 06:52.17 | Strom_C | if you sound lile some nub subscriber, they won't :) |
| 06:52.30 | mog | thank ya |
| 06:52.35 | mog | some guy read it |
| 06:52.40 | Qwell | "some guy"? |
| 06:52.41 | mog | and is sending me thailand redbull |
| 06:52.44 | Qwell | ha |
| 06:52.51 | Qwell | I was wondering what that was in regards to |
| 06:52.56 | shodan | k, that's pretty much the response I got , like "have you talked to your service provider" |
| 06:53.00 | Qwell | just randomly "hi, here's redbull" |
| 06:53.08 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-74-10.aubnin.fios.verizon.net) |
| 06:53.18 | Qwell | shodan: "You ARE my service provider"... |
| 06:54.38 | Strom_C | bell canada! there is no service at 555-2368. I repeat, there is no service at 555-2368. |
| 06:55.31 | Strom_C | I like her voice |
| 06:55.51 | breakdisk | hi guys,anyone have use Dlink DI-102 voip qos adaptor,because im planning to purchase.Could you recommend this product? |
| 06:57.21 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 06:57.28 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 06:57.44 | shodan | it's the tip that should be at the ground right ? |
| 06:58.03 | Newbie___ | hi all, by playing around with hostname, will it cause asterisk not to pick up the line? |
| 06:58.11 | Strom_C | shodan: neither side should be grounded |
| 06:58.18 | Strom_C | the protector should be grounded |
| 06:58.29 | Strom_C | but shorting tip or ring to ground will just cause a lot of hum on the line |
| 07:00.36 | *** join/#asterisk Dovid (n=dovi5988@pool-70-111-254-94.nwrk.east.verizon.net) |
| 07:00.39 | shodan | ah ok , I just checked neither are grounded |
| 07:00.48 | Dovid | morning all |
| 07:00.57 | Dovid | whats the command to playback digits ? |
| 07:01.05 | Strom_C | SayDigits() |
| 07:01.09 | Dovid | :) |
| 07:02.15 | *** part/#asterisk rtyler (i=tyler@sand.geekisp.com) |
| 07:02.24 | Dovid | any way to check whats in a variable |
| 07:02.43 | Dovid | as far as if there are letters in ir or not ? |
| 07:03.35 | Dovid | nm |
| 07:03.42 | Dovid | asterisk skipps the caharacters |
| 07:03.51 | shodan | is there a way I can test the line impedance of my loop ? |
| 07:04.14 | _Vile | shodan, what kind of loop? |
| 07:04.26 | _Vile | are you facing telco? |
| 07:04.51 | shodan | I have one of these thingy is that usable ? http://www.sencore.com/newsletter/July04/LC103gold_files/image004.jpg |
| 07:05.07 | shodan | it's pstn , yes |
| 07:05.16 | Strom_C | shodan: use the milliwatt test |
| 07:05.31 | _Vile | heh |
| 07:05.38 | _Vile | yes usable |
| 07:05.55 | _Vile | kind of like a t-berd...? |
| 07:06.09 | _Vile | I don't know how to train you to use that device |
| 07:06.16 | _Vile | but yes it is useful |
| 07:06.24 | shodan | can I use another x100p to generate that tone or does it have to come from the office ? |
| 07:06.57 | shodan | (I have 3 lines here , maybe one can call the other and generate the tone ?) |
| 07:07.01 | Strom_C | shodan: it has to come from the office |
| 07:07.04 | shodan | k |
| 07:07.11 | Strom_C | the tone leaves the office at 0dbm |
| 07:07.22 | Strom_C | you measure it on your end and then adjust your receive gain to compensate |
| 07:08.05 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 07:11.14 | shodan | how would that help in reducing echo ? I mean I could set the rxgain blind and reduce the echo until it's acceptable , but I tought I could use that test to make something that would help me match the impedance of the line vs the impedance of the x100p's hybrid ? (or just replace the caps on the x100p to match the line impedance ?) |
| 07:11.48 | Strom_C | shodan: when you properly account fot impedence on the line, your echo reduces |
| 07:12.48 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 07:15.02 | shodan | I'm thinking of getting a linksys spa-3000 and be done with it but... I won't receive it before my system is due to go online :\ |
| 07:15.19 | _Vile | grab the spa-3000 |
| 07:15.24 | _Vile | if you need to |
| 07:15.36 | _Vile | and have on stock regular spa's |
| 07:15.54 | _Vile | then you can grab fxo->fxs converters |
| 07:16.47 | docelmo | spa3000 is a nightmare to setup |
| 07:17.00 | _Vile | doc, shh |
| 07:17.03 | _Vile | :) |
| 07:17.25 | shodan | probably as bad as the spa-2102 I had earlier , but it eventually worked :) |
| 07:17.42 | docelmo | I need a couple hardware FXO's for my asterisk box |
| 07:20.33 | InHisName | Anyone familiar with VMauthenticate() ? Like what can go into the () ? |
| 07:21.38 | docelmo | Are you familiar with "show application" or voip-info |
| 07:22.24 | InHisName | voip-info is too brief in details. |
| 07:22.25 | shodan | InHisName, VMAuthenticate([mailbox][@context]) |
| 07:22.38 | InHisName | trying show application |
| 07:24.40 | InHisName | <PROTECTED> |
| 07:25.16 | _Vile | shodun, go spa-3000s |
| 07:25.29 | _Vile | ignore the x100s |
| 07:25.58 | InHisName | I have tried VMAuthenticate(1234@hostname) and it prompts for account and then pwd. Seems to ignore the account insterted. |
| 07:26.04 | _Vile | if your boss is a cheap ass, let him hear you on both |
| 07:26.24 | shodan | yeah I'll get one , it will also unload my systems a lot too |
| 07:26.52 | _Vile | enjoy |
| 07:26.54 | shodan | in the meantime I guess I'll just attenuate the tx & rx until there is very little echo |
| 07:27.08 | _Vile | yep |
| 07:27.22 | _Vile | i never got it 'good enough' |
| 07:27.24 | _Vile | btw |
| 07:27.31 | coppice | stop the rx, and you'll find echo is very low :-) |
| 07:27.57 | shodan | hehe yeah , also the callers will become a lot less annoying ;) |
| 07:28.12 | coppice | see, wins all round |
| 07:28.17 | _Vile | hha |
| 07:28.34 | *** join/#asterisk ozant (n=ozan@81.215.116.87) |
| 07:31.19 | *** join/#asterisk anjaana_ladka (n=anjaana_@59.176.4.196) |
| 07:32.26 | anjaana_ladka | hello room |
| 07:32.33 | anjaana_ladka | can anybody pls help me out .. |
| 07:32.41 | anjaana_ladka | in working with asterisk.. |
| 07:32.56 | anjaana_ladka | i have defined..... users in sip.conf and extensions.conf |
| 07:33.14 | anjaana_ladka | and at the client level i am using x-lite version 3 |
| 07:33.32 | anjaana_ladka | and all times i get the error request timed out.. |
| 07:34.51 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 07:36.28 | *** join/#asterisk _omer (n=omer@203.128.20.31) |
| 07:36.29 | _omer | hi |
| 07:36.31 | InHisName | VMAuthenticate([mailbox][@context]) why does this ignore the VMaccount and prompt for account number anyway ? |
| 07:36.50 | _omer | how to move to the next EXTEN when call is ANSWERED??? |
| 07:37.06 | anjaana_ladka | hi omer |
| 07:37.08 | Strom_C | the answer is "dead hookers" |
| 07:37.24 | _omer | hi anjaana_ladka |
| 07:37.56 | anjaana_ladka | can u pls help me out in setting up the asterisk pbx??/ |
| 07:38.45 | _omer | sure |
| 07:39.07 | anjaana_ladka | _omer i have tried reading at so many places... but i am not actually able to configure the same.... over here.. |
| 07:39.44 | anjaana_ladka | i have got the basic installation of asterisk done and compiled all the related sources... |
| 07:39.46 | _omer | what exactly you want to configure this asterisk for ? |
| 07:39.51 | anjaana_ladka | including zaptel |
| 07:40.07 | anjaana_ladka | i am purely newbie for the system... |
| 07:40.24 | anjaana_ladka | right now .. i wish to make my machine small pbx for |
| 07:40.36 | anjaana_ladka | inhouse use .... with softphones.. |
| 07:41.19 | _omer | okey... |
| 07:41.45 | anjaana_ladka | i have got configured sip.conf based on my readings from the net and the book.. |
| 07:41.57 | anjaana_ladka | should i put the matter here of the sip.conf? |
| 07:42.15 | _omer | nope.. |
| 07:42.21 | _omer | use www.pastebin.ca |
| 07:42.41 | Sargun | Anyone here have a Laser? |
| 07:43.26 | _Vile | what for? |
| 07:43.26 | *** join/#asterisk j0 (i=dan@CABLE-72-53-45-212.cia.com) |
| 07:43.38 | coppice | yep. one in every CD drive :-) |
| 07:44.18 | Sargun | heh |
| 07:44.22 | coppice | a couple of presentation pointers too. |
| 07:45.07 | anjaana_ladka | http://pastebin.ca/159601 |
| 07:45.08 | _Vile | what kind? |
| 07:45.27 | _Vile | and what for? |
| 07:45.31 | anjaana_ladka | _omer pls check at http://pastebin.ca/159601 |
| 07:46.06 | anjaana_ladka | i have replaced the original sip.conf with the current material.. |
| 07:47.44 | _omer | whats the problem ? |
| 07:47.49 | anjaana_ladka | _omer, are u there????? |
| 07:47.51 | _omer | that you are geting |
| 07:47.54 | _omer | getting* |
| 07:48.16 | anjaana_ladka | well i have started the asterisk after this ... |
| 07:48.32 | _Vile | 'who will save your souls, if you won't save your own' |
| 07:48.42 | anjaana_ladka | and when i try connecting using the .... x-lite soft phone i am getting request timed out error... |
| 07:49.34 | _omer | what about X-lite configuration ? |
| 07:49.43 | _Vile | tried going to the right ip? |
| 07:49.57 | _Vile | and when in sip debug |
| 07:50.12 | _Vile | do you see the attempts? |
| 07:50.21 | anjaana_ladka | in X-lite configuration .... |
| 07:50.27 | _Vile | not on xlite |
| 07:50.28 | anjaana_ladka | i have set |
| 07:50.32 | _Vile | that is the client |
| 07:50.34 | anjaana_ladka | display name abhi |
| 07:50.37 | _Vile | on asterisk |
| 07:50.43 | anjaana_ladka | username abhi |
| 07:50.48 | anjaana_ladka | password welcome |
| 07:50.52 | _Vile | when you do sip debug |
| 07:50.57 | anjaana_ladka | authorisation user name abhi |
| 07:51.08 | _Vile | do you see the client trying to connect? |
| 07:51.12 | _Vile | if not |
| 07:51.19 | anjaana_ladka | domain 192.168.3.11:5060 |
| 07:51.20 | anjaana_ladka | yes |
| 07:51.26 | anjaana_ladka | yes _ville |
| 07:51.33 | anjaana_ladka | i get .... connecting.. |
| 07:51.41 | _Vile | you see the client trying to connect |
| 07:51.45 | _Vile | from within *? |
| 07:51.48 | anjaana_ladka | but in the end i get register failed... error code 408 |
| 07:51.54 | anjaana_ladka | yes |
| 07:52.17 | _omer | whats in SIP PROXY and outbound PROXY ? |
| 07:53.17 | anjaana_ladka | i have selected the option domain... in send outbound via |
| 07:53.28 | _Vile | what's the pastebin for that |
| 07:53.51 | _Vile | it's a client issue |
| 07:54.13 | anjaana_ladka | can i try downloading some other client.. ???? for the same.. |
| 07:54.23 | _Vile | no, the issue is in config |
| 07:54.51 | _Vile | what's the pastebin for that |
| 07:54.57 | anjaana_ladka | then what should i put in..... |
| 07:55.08 | anjaana_ladka | in pastebin i have just placed the sip.conf file.. |
| 07:55.18 | _Vile | go into * |
| 07:55.19 | _Vile | the cli |
| 07:55.24 | anjaana_ladka | to let u know what is the configuration i have set at the asterisk.. |
| 07:55.32 | _Vile | type in sip debug |
| 07:55.41 | _Vile | try to make a call |
| 07:55.54 | _Vile | pastebin that |
| 07:56.41 | anjaana_ladka | i am in cli.. |
| 07:57.06 | anjaana_ladka | i am getting on cli.. |
| 07:57.24 | anjaana_ladka | <PROTECTED> |
| 07:57.24 | anjaana_ladka | <PROTECTED> |
| 07:57.24 | anjaana_ladka | <PROTECTED> |
| 07:57.24 | anjaana_ladka | <PROTECTED> |
| 07:57.46 | _Vile | you're not connecting to * |
| 07:58.04 | anjaana_ladka | what is * |
| 07:58.10 | anjaana_ladka | sorry i am a new bie |
| 07:58.14 | _Vile | ok |
| 07:58.18 | _omer | anjaana_ladka : are you getting this TEXT again and again ??? |
| 07:58.19 | _Vile | in sip.conf |
| 07:58.24 | anjaana_ladka | yes |
| 07:58.26 | j0 | i'm having voice quality issues.. my network isn't the best, ping varies from time to time, but this seems to happen on every isp i've placed my * box... now is iax2 show channels, etc. accurate? it always claims to have no jitter or lag. and this is what i thought was causing the problem.. any ideas? my voice basically has a "digital sound" for half a second from time to time |
| 07:58.27 | anjaana_ladka | yes _omer |
| 07:58.47 | _Vile | anj, it only means you're connecting with asterisk -vvvvvvvvvvvvr etc |
| 07:59.11 | anjaana_ladka | _Vile, now what should i do??? |
| 07:59.16 | _Vile | are you still in the cli? |
| 07:59.22 | anjaana_ladka | yes |
| 07:59.41 | _Vile | type in "sip debug" |
| 07:59.48 | _Vile | try making a call |
| 07:59.51 | anjaana_ladka | i have done that... |
| 07:59.57 | _Vile | good |
| 07:59.59 | anjaana_ladka | how should i make a call ??? |
| 08:00.02 | _Vile | try making a call |
| 08:00.08 | anjaana_ladka | from the client?????? |
| 08:00.08 | _Vile | from your phone |
| 08:00.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:00.27 | anjaana_ladka | but my phone is not registering.. |
| 08:00.54 | _Vile | reboot your phone |
| 08:01.04 | _Vile | see what messages come at you |
| 08:02.01 | anjaana_ladka | my phone first says discovering network.. then registering.. . |
| 08:02.18 | _Vile | do you see anything in the * cli? |
| 08:02.20 | anjaana_ladka | again registration error.. |
| 08:02.53 | anjaana_ladka | <PROTECTED> |
| 08:02.53 | anjaana_ladka | <PROTECTED> |
| 08:02.53 | anjaana_ladka | <PROTECTED> |
| 08:02.53 | anjaana_ladka | <PROTECTED> |
| 08:02.53 | anjaana_ladka | <PROTECTED> |
| 08:02.54 | anjaana_ladka | <PROTECTED> |
| 08:02.56 | anjaana_ladka | <PROTECTED> |
| 08:02.58 | anjaana_ladka | <PROTECTED> |
| 08:03.00 | anjaana_ladka | <PROTECTED> |
| 08:03.00 | _Vile | you're not pointing at * in your phone registration |
| 08:03.02 | anjaana_ladka | <PROTECTED> |
| 08:03.04 | anjaana_ladka | <PROTECTED> |
| 08:03.08 | anjaana_ladka | <PROTECTED> |
| 08:03.10 | anjaana_ladka | <PROTECTED> |
| 08:03.12 | anjaana_ladka | <PROTECTED> |
| 08:03.14 | anjaana_ladka | <PROTECTED> |
| 08:03.16 | anjaana_ladka | <PROTECTED> |
| 08:03.18 | anjaana_ladka | <PROTECTED> |
| 08:03.20 | _Vile | which means go into your registration |
| 08:03.22 | anjaana_ladka | <PROTECTED> |
| 08:03.24 | anjaana_ladka | <PROTECTED> |
| 08:03.26 | anjaana_ladka | <PROTECTED> |
| 08:03.27 | _Vile | point it to your * ip |
| 08:03.28 | anjaana_ladka | <PROTECTED> |
| 08:03.30 | anjaana_ladka | wait _Vile |
| 08:03.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:03.41 | anjaana_ladka | i am in properties window... |
| 08:03.58 | JT | anjaana_ladka: |
| 08:03.59 | anjaana_ladka | there is one place called domain.. |
| 08:04.02 | JT | ~pb |
| 08:04.03 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 08:04.07 | anjaana_ladka | should i put * in domain.. |
| 08:04.09 | _Vile | put your * ip there |
| 08:04.20 | anjaana_ladka | sorry jbot |
| 08:04.36 | anjaana_ladka | i will try to keep in mind in future.. |
| 08:04.37 | _Vile | use pastebin from now on. |
| 08:04.45 | _Vile | pastebin.ca |
| 08:04.46 | _Vile | easy |
| 08:05.11 | anjaana_ladka | i know _Vile |
| 08:05.25 | anjaana_ladka | i forgot... sorry .. i will do the same in future.. |
| 08:05.43 | anjaana_ladka | again.. the same error recd.. |
| 08:05.50 | _Vile | *shrug* doesn't bother me, 1am here, noone's around |
| 08:05.50 | anjaana_ladka | i put * on domain.... id.. |
| 08:06.16 | anjaana_ladka | am i disturbing u _Vile |
| 08:06.19 | anjaana_ladka | sorry .. |
| 08:06.22 | _Vile | ok , I have to ask, by putting * on domain, did you put the IP for asterisk or a literal *? |
| 08:06.29 | _Vile | no, I'm here :) |
| 08:06.51 | anjaana_ladka | * |
| 08:06.57 | _Vile | put the ip |
| 08:06.57 | anjaana_ladka | earlier there was ip... |
| 08:07.07 | _omer | * = asterisk |
| 08:07.27 | anjaana_ladka | i have installed asterisk on ... ip 192.168.3.11 |
| 08:07.39 | anjaana_ladka | i placing the same ip on domain... in softphone.. |
| 08:08.13 | anjaana_ladka | i am able to ping the ip from the client.. |
| 08:08.38 | anjaana_ladka | same error recd again.. |
| 08:08.44 | _omer | <_omer> whats in SIP PROXY and outbound PROXY ? |
| 08:08.55 | _omer | of you X-Lite? |
| 08:09.01 | _omer | your* |
| 08:09.15 | anjaana_ladka | 192.168.3.11:5060 |
| 08:09.25 | tzafrir | _omer, those settings are in case you use a separate proxy for signalling and for RTP data |
| 08:09.31 | joat | qls |
| 08:09.35 | joat | arg! |
| 08:09.37 | joat | sorry |
| 08:10.20 | anjaana_ladka | is there any prob... should i remove the ip and port... ??? |
| 08:10.29 | _omer | no |
| 08:10.56 | _omer | thanks tzafrir: any help in call back script ? ;) |
| 08:11.11 | anjaana_ladka | there are three options in send outbound...... 1) domain, 2) proxy 3) target domain |
| 08:11.33 | anjaana_ladka | i have selected proxy and then .. .put the address as 192.168.3.11:5060 |
| 08:11.45 | _Vile | remove :5060 |
| 08:12.44 | anjaana_ladka | same error again.. |
| 08:12.50 | anjaana_ladka | hey _Vile |
| 08:13.01 | anjaana_ladka | can u please help me configure asterisk again.. |
| 08:13.10 | anjaana_ladka | from scratch.. |
| 08:13.30 | _Vile | sure, rm -rf /etc/asterisk, repopulate |
| 08:13.44 | _Vile | what kind of help? |
| 08:13.56 | _Vile | your config is fine on * |
| 08:14.05 | _Vile | your problem is on x10 |
| 08:14.10 | anjaana_ladka | welll ..... i have got asterisk.. installed on ... my machine... 192.168.3.11 |
| 08:14.26 | anjaana_ladka | i wish to start the pbx for the machines connected on lan.. |
| 08:14.29 | tzafrir | _omer, what specifically is your problem? |
| 08:14.45 | anjaana_ladka | and right now there is no fxo card connected.. |
| 08:14.48 | _Vile | tza, he's not pointing his client to * |
| 08:14.53 | _Vile | well that too |
| 08:14.58 | anjaana_ladka | so it will be purely pbx for the connected computers on lan... |
| 08:15.13 | _Vile | he did a sip debug |
| 08:15.24 | *** join/#asterisk Piston (n=what@206M28.oasis.mediatti.net) |
| 08:15.26 | _Vile | restarted his client |
| 08:15.30 | _Vile | no info from * |
| 08:15.36 | _Vile | not talking |
| 08:15.52 | _Vile | pointing possibly to the wrong IP |
| 08:15.59 | _Vile | i would think firewall issue |
| 08:16.06 | _Vile | but he's able to ping the ip |
| 08:16.09 | _Vile | from his box |
| 08:16.19 | anjaana_ladka | hey _Vile there is no firewall at all .. |
| 08:16.20 | Piston | is voipdiscount a good provider if I already have my own SIP device and want a phone number in the USA? |
| 08:16.27 | anjaana_ladka | both machines are connecting on a switch.. |
| 08:16.35 | _Vile | ./|\ |
| 08:17.11 | _omer | tzafrir: I want to create a WEB CALL BACK .....if any sample is available thats gonna be work...Scenerio should be : DIAL 5551212 ...when Answered ....Dial 4441212 |
| 08:17.54 | tzafrir | _omer, what the web has to do with it? Is the trigger a web form post? |
| 08:18.08 | tzafrir | or a dial through a phone line? or a voip call? |
| 08:18.45 | _Vile | if a web form post, you want to use an * api on the back end |
| 08:18.52 | _omer | yes...I want to trigger it from web... |
| 08:18.56 | j0 | i swear asterisk is injecting bad audio into my outgoing streams! heh.. is this possible? |
| 08:19.09 | j0 | my bandwidth is fine.. i have loads of bandwidth |
| 08:19.14 | _Vile | j0 explain |
| 08:19.16 | j0 | incoming call quality is flawless |
| 08:19.31 | anjaana_ladka | _Vile, so how should i start with it.. |
| 08:19.32 | j0 | i've tried every codec, sip and iax calling |
| 08:19.58 | j0 | my outgoing voice has strange 'artifacts' i guess.. the voice distorts every so often for half a second |
| 08:20.24 | _Vile | anja, play with settings on your client |
| 08:20.35 | _Vile | use sip debug on your asterisk box |
| 08:20.45 | anjaana_ladka | okkk |
| 08:20.49 | _Vile | wait until you see something on asterisk box |
| 08:20.54 | _Vile | and then adjust |
| 08:21.12 | anjaana_ladka | can u pls tell me if i can try with .... some other softphone???????/ |
| 08:21.14 | anjaana_ladka | apart from x-lite |
| 08:21.24 | _Vile | use x-lite |
| 08:21.30 | anjaana_ladka | okkk |
| 08:21.30 | _Vile | it is clean. |
| 08:21.42 | j0 | the outgoing audio is bad from my phone, as well as hold music or 'ringing' sounds from asterisk |
| 08:21.54 | _Vile | x100p? |
| 08:22.20 | j0 | _Vile: me? voip only |
| 08:22.28 | benjk | what codec? |
| 08:22.34 | _Vile | interesting |
| 08:22.56 | j0 | tried ulaw and g729, but using gsm mostly |
| 08:23.05 | j0 | all have the exact same problem |
| 08:23.06 | _Vile | what provider? |
| 08:23.10 | benjk | and you get the same with ulaw? |
| 08:23.17 | _Vile | tried diff providers? |
| 08:23.45 | j0 | i've tried peering with other ppls dsl and had what sounded to me like similar problems.. |
| 08:24.02 | j0 | know of a good provider i can test with on the cheap? |
| 08:24.05 | _Vile | have you tried a different provider? |
| 08:24.08 | j0 | good/cheap.. lol |
| 08:24.08 | j0 | no |
| 08:24.09 | benjk | did you have the same with ulaw only? |
| 08:24.15 | j0 | benjk: yes |
| 08:24.28 | benjk | and you are sure the entire route, end to end is all ulaw? |
| 08:24.46 | _Vile | he's probably going back east |
| 08:24.48 | _Vile | from west |
| 08:24.55 | j0 | _Vile: i am.. from bc, canada to toronto |
| 08:25.09 | j0 | benjk: not 100% sure.. i know its ulaw from * to my provider |
| 08:25.19 | j0 | and my phone is set to default to ulaw |
| 08:25.21 | _Vile | traceroute your endpoints? |
| 08:25.42 | _Vile | pastebin |
| 08:25.45 | benjk | it may not be at your end |
| 08:25.56 | _Vile | probably not at your end |
| 08:26.00 | j0 | with a traceroute, any possible problems are on my isps end.. i occassionally get changes of 50ms or so. |
| 08:26.15 | _Vile | bingo |
| 08:26.35 | j0 | i even tried another isp with exactly the same 'glitches' |
| 08:26.35 | _Vile | but a traceroute would be nice anyway |
| 08:26.39 | benjk | for example, your provider may use a codec that compresses audio or they have bandwith issues further down the line |
| 08:27.09 | j0 | benjk: that sounds like a possibility |
| 08:27.21 | j0 | let me give you a traceroute |
| 08:27.38 | _Vile | your provider may use the same outbound provider as your previous one to connect to your endpoint |
| 08:27.47 | benjk | traceroute won't tell you want codecs and what bandwidth there is |
| 08:27.55 | _Vile | not likely, you shifted providers |
| 08:28.20 | _Vile | true |
| 08:28.39 | benjk | you have to test with somebody else's asterisk box or something, peer-to-peer, so you know what's in between |
| 08:29.19 | _Vile | he did shift providers tho |
| 08:29.34 | *** join/#asterisk LoneShadow (n=duh@59.92.141.178) |
| 08:29.40 | benjk | that doesn't mean he knows whats behind those providers |
| 08:29.57 | _Vile | yes |
| 08:30.09 | j0 | http://paste.lisp.org/display/25210 |
| 08:30.09 | benjk | its unscientific to say "its the same with two different experiments, it ought to be the same with n experiments" |
| 08:30.28 | _Vile | scientifically, that will not work |
| 08:30.34 | _Vile | asteriskally, it does |
| 08:30.36 | j0 | i haven't tried different providers yet... but peering with another asterisk box had problems, although he didn't seem to have much bandwidth anyway |
| 08:30.47 | benjk | yeah, asterisk is anti-scientific |
| 08:31.03 | _Vile | you said you tried two different providers |
| 08:31.18 | _Vile | with same result |
| 08:31.19 | _Vile | ahh ok |
| 08:31.22 | j0 | i tried placing my box on 2 diff isps.. sorry |
| 08:31.33 | benjk | i'd try p2p with another * box peered over the internet |
| 08:31.46 | j0 | hmm... in that traceroute it showed big jitter problems on my voip providers isp.. that hasn't happened before |
| 08:32.01 | j0 | darn canada |
| 08:32.37 | j0 | can you reccomend a prepay provider i can easily test with? |
| 08:32.59 | j0 | can u complain about one? |
| 08:32.59 | benjk | you still don't know the infrastructure the provider will use |
| 08:33.07 | _Vile | i can complain about many ;) |
| 08:33.16 | _Vile | benj is right |
| 08:33.19 | benjk | you have to test with two boxes you know exactly how they are set up |
| 08:33.39 | j0 | true, but it's probably easier than getting someone to help me test |
| 08:33.58 | _Vile | not really, choose an ISP |
| 08:34.03 | _Vile | bring * back here |
| 08:34.04 | benjk | its not a conclusive test though |
| 08:34.11 | _Vile | we'll trace you |
| 08:34.20 | _Vile | or vice versa |
| 08:34.38 | j0 | what do u mean? |
| 08:34.42 | j0 | _Vile |
| 08:34.56 | _Vile | get a diff internet provider |
| 08:35.11 | _Vile | most likely they go out the same carrier, right? |
| 08:35.21 | _Vile | keep moving until it works |
| 08:35.41 | anjaana_ladka | hi _Vile |
| 08:35.50 | anjaana_ladka | thanks... |
| 08:35.59 | j0 | we only have 2 isps here.. i've tried both.. they have significantly different networks |
| 08:36.01 | anjaana_ladka | i have come one step forward.. |
| 08:36.02 | _Vile | didja get it? |
| 08:36.10 | anjaana_ladka | yes |
| 08:36.16 | anjaana_ladka | thanks for ur help.. |
| 08:36.17 | _Vile | nice... |
| 08:36.49 | _Vile | j0, that interests me |
| 08:38.06 | _Vile | ok j0, from one * box to another * box, what do you get -- say, next door neighbor? |
| 08:38.23 | j0 | never tried it |
| 08:41.33 | j0 | asked my voip provider a few questions.. lets see what he says |
| 08:47.30 | anjaana_ladka | hey _Vile |
| 08:47.37 | anjaana_ladka | are u there? |
| 08:47.47 | anjaana_ladka | there is one thing i noticed.... |
| 08:48.20 | j0 | http://www.dslreports.com/r3/smokeping.cgi?target=network.19643a04566433b97c4cdd8c4e6577e5&r=849 |
| 08:48.28 | anjaana_ladka | when i tick the open for register with domain and recieve incoming calls.. |
| 08:48.35 | anjaana_ladka | x lite dont register... |
| 08:48.37 | j0 | this shows packet loss for my ip address, but things like ping never shows packet loss |
| 08:48.56 | j0 | even ping -f comes out with 0% |
| 08:49.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:58.52 | *** join/#asterisk stopher (n=austchat@cm-24-121-73-66.kingman.az.npgco.com) |
| 09:00.51 | coppice | i think some people give ping packets preferential treatment so a ping test looks good on their links :-) |
| 09:01.02 | j0 | bastards! |
| 09:03.02 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-74-10.aubnin.fios.verizon.net) |
| 09:06.18 | EyeCue | one can check latecy without using icmp echo'd though right |
| 09:07.09 | j0 | alright.. gn.. try more in the morning <sigh> |
| 09:08.50 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:11.01 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 09:16.31 | coppice | EyeCue: most other ways involve cooperation at the far end, and some applications delays. ICMPs are sent back from deep in the OS and require no cooperation from the far end |
| 09:37.15 | *** join/#asterisk RoyK (n=roy@ti211110a081-1789.bb.online.no) |
| 09:41.40 | *** join/#asterisk RoyK (n=roy@ti211110a081-1789.bb.online.no) |
| 10:22.18 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 10:23.43 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 10:27.08 | DarKnesS_WolF | i want to make a something but can't get it fully orginized on my mind , i want that i can dial an extension and fowd my calls to another extension will have the voice mail for me "u${ARG1}" but it didn't work . |
| 10:27.59 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:29.36 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 10:34.21 | DarKnesS_WolF | can i make an extesion i call like 5001 and i'll be in DND mode so all my calls will go to my voicemail box ? |
| 10:39.10 | Assid | yeah.. see the stdextn macro |
| 10:55.34 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 11:03.26 | *** join/#asterisk stopher (n=austchat@cm-24-121-73-66.kingman.az.npgco.com) |
| 11:21.38 | DarKnesS_WolF | Assid: thx |
| 11:28.37 | DarKnesS_WolF | Assid: i don't think that what i'm looking for :-s |
| 11:29.50 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 11:31.41 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 11:32.08 | Assid | you setting DND on your ip phone right ? |
| 11:33.17 | DarKnesS_WolF | Assid: hm nop :-) |
| 11:33.28 | DarKnesS_WolF | Assid: i want to do it all asterisk thing. |
| 11:34.36 | DarKnesS_WolF | Assid: so the user will dial for ex. **31# and all the calles he will get on his phone will be sent to voicemail box |
| 11:34.49 | DarKnesS_WolF | but i can't think how to do it .. call forwarding or what |
| 11:34.59 | DarKnesS_WolF | i'm sure there is somehting easy and clean |
| 11:36.22 | *** join/#asterisk PoWeRKiLL (n=powerkil@85.64.221.116) |
| 11:36.28 | PoWeRKiLL | Hi |
| 11:37.01 | PoWeRKiLL | I have a problem with forkcdr I got duplicate on my database each time i run forcdr |
| 11:42.49 | Assid | DarKnesS_WolF: okay then.. you could use astdb to send the call to voicemail |
| 11:43.36 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 11:43.37 | DarKnesS_WolF | Assid: how ? |
| 11:46.13 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 11:49.00 | DarKnesS_WolF | Assid: or can u point me to a how to? |
| 11:53.08 | teknoprep | is there a way to dial a number that would change the incomming route for a number from say 1 ivr to another ivr? say a feature code? so at closing of shop i can dial this number.. and it tells all incomming callers on a new IVR that the store is closed... then at open i can go back to being open... |
| 11:56.17 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 11:56.20 | Assid | DarKnesS_WolF: DBGet / DBPut .. |
| 11:56.31 | Assid | DarKnesS_WolF: http://www.voip-info.org/wiki-Asterisk+Database |
| 11:56.38 | DarKnesS_WolF | thx |
| 11:56.50 | Assid | and then make your dialplan such that it reads the astdb key/value .. and acts accordingly |
| 11:57.00 | DarKnesS_WolF | Assid: but i was giving it a deep think |
| 11:57.01 | DarKnesS_WolF | it's wrong |
| 11:57.12 | *** join/#asterisk RoyK (n=roy@ti211110a081-1789.bb.online.no) |
| 11:57.14 | Assid | ? |
| 11:57.18 | DarKnesS_WolF | i already have a CIFM |
| 11:57.21 | DarKnesS_WolF | and using astdb |
| 11:57.39 | Assid | you make another tree |
| 11:57.43 | Assid | or key |
| 11:57.58 | DarKnesS_WolF | what i'm doing is .. **21*5001 this will fwd from the exte. i'm dialing from to 5001 |
| 11:58.23 | DarKnesS_WolF | and at 5001 i done this exten 5001,1,voicemail($ARG1}@context) |
| 11:58.29 | DarKnesS_WolF | but it's not working it cant' get ARG1 |
| 11:58.35 | Assid | so you use ${EXTEN:5} |
| 11:58.41 | DarKnesS_WolF | hm |
| 11:58.43 | DarKnesS_WolF | let me try |
| 11:59.02 | Assid | ($ARG1}@context) is wrong |
| 11:59.06 | Assid | your missing { |
| 11:59.37 | DarKnesS_WolF | no tis is a typo :-D |
| 11:59.37 | Assid | and you need to check the database before you send it to voicemail |
| 11:59.47 | DarKnesS_WolF | this didn't work ur method it need mailbox number |
| 11:59.56 | DarKnesS_WolF | yes it's already checking the database |
| 12:00.37 | DarKnesS_WolF | i'm soo lost -:-s |
| 12:00.46 | Assid | your def. doing something wrong |
| 12:01.00 | DarKnesS_WolF | yex |
| 12:01.01 | DarKnesS_WolF | yes |
| 12:01.47 | *** join/#asterisk markQing (n=mark@82-197-202-242.dsl.cambrium.nl) |
| 12:02.30 | Assid | **21*X,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:5}) |
| 12:02.32 | Assid | there |
| 12:02.38 | Assid | that will put it to the astdb for you |
| 12:02.57 | Assid | **21*X,1,DBput(CFIM/${CALLERIDNUM}=1) |
| 12:02.59 | Assid | or use that |
| 12:03.05 | Assid | a simple true/false |
| 12:03.27 | Assid | then when your call comes in.. dbget.. and check if its 1 .. |
| 12:03.34 | RoyK | is it possible to use the db interface with an rdms like mysql? |
| 12:03.52 | Assid | **21,1,DBput(CFIM/${CALLERIDNUM}=1) |
| 12:04.00 | Assid | you dont need the wildcard at the back |
| 12:04.12 | Assid | since the callerid is set anwyasy |
| 12:04.30 | Assid | dont think so royk |
| 12:05.34 | DarKnesS_WolF | Assid: seems really i coudn't explan my problem :-s |
| 12:05.42 | DarKnesS_WolF | i'm already haveing CFIM |
| 12:06.38 | DarKnesS_WolF | and i have DBget / DBput |
| 12:07.03 | DarKnesS_WolF | but i want to make a special extension so it can be fwd to the vm box directly |
| 12:08.10 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:08.32 | Assid | did you read what i said |
| 12:11.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:13.59 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:15.06 | DarKnesS_WolF | yep i did |
| 12:27.27 | DarKnesS_WolF | Assid: thx alot for ur help but really i can't do it i'm soo confuised but thx alot |
| 12:28.48 | Piston | holy shit |
| 12:28.53 | Piston | new house season starts on tuesday |
| 12:28.58 | Piston | :o |
| 12:29.08 | Piston | and with that, I will sleep with a smile on my face |
| 12:29.10 | Piston | goodnight rh |
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| 12:37.11 | Assid | dr. house? |
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| 13:15.21 | chkbsd | hi |
| 13:17.19 | chkbsd | i use asterisk 1.2.10 and chan_capi 0.7 and it works finde, but ive no "ring ring" when i call some via my isdn card. Its only quiet and when some is take off then i can speak with him |
| 13:17.28 | chkbsd | the problem is i want to hear the ringing |
| 13:17.40 | chkbsd | before some takes off |
| 13:21.02 | chkbsd | got it |
| 13:21.03 | chkbsd | works |
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| 13:45.11 | *** join/#asterisk sonium (n=Sonium@p5492C72D.dip0.t-ipconnect.de) |
| 13:45.46 | sonium | hi, to install zaphfc on ubuntu is it nescessary to have a self compiled kernel? |
| 13:48.47 | *** part/#asterisk chkbsd (n=chkbsd@p54AA3E8C.dip0.t-ipconnect.de) |
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| 13:55.34 | benjk | sonium, works fine here with Ubuntu Server 6.06 and stock kernel |
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| 13:56.05 | sonium | benjk: how did you do it? |
| 13:56.14 | benjk | what? |
| 13:56.29 | sonium | I'm trying now for three day to get the fucking shit compiled |
| 13:56.52 | benjk | I used the script that comes with it and then just make ; make install |
| 13:57.34 | sonium | did you use the asterisk version from britstuff? |
| 13:57.45 | benjk | BRIstuff stable yes |
| 13:58.06 | sonium | hrm.. |
| 13:58.29 | sonium | and you just runned the installer and after that from the zaphfc dir make install? |
| 13:58.57 | benjk | I did everything exactly as the readme file from BRIstuff says |
| 13:59.13 | sonium | and do you have linux-source installed? |
| 13:59.30 | benjk | kernel sources yes |
| 13:59.44 | sonium | but not compiled? |
| 13:59.54 | benjk | no I didn't compile the kernel |
| 14:00.07 | sonium | did you copy of ther .config file? |
| 14:00.09 | benjk | may have done make dep or something, don't remember |
| 14:00.23 | benjk | but I just followed the instructions that came with BRIstuff |
| 14:00.41 | sonium | kk. maybe I should no use the unstable bristuff |
| 14:00.56 | benjk | you use the experimental tree? |
| 14:01.00 | sonium | yes |
| 14:01.01 | benjk | I didn't try with that |
| 14:01.08 | benjk | this is a production system |
| 14:01.15 | sonium | kk. then I will try the production tree |
| 14:01.16 | sonium | thx |
| 14:02.12 | benjk | I did upgrade the kernel but only using apt-get |
| 14:02.28 | benjk | so no recompile |
| 14:07.28 | *** join/#asterisk katriel (n=penguin_@bzq-88-155-228-254.red.bezeqint.net) |
| 14:08.36 | katriel | Hello |
| 14:09.18 | katriel | Can anyone help me with a hardware related issue? |
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| 14:27.21 | puowvip | okey dokey. |
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| 14:40.17 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:49.32 | stephane_ | Rez, |
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| 14:58.59 | Feral_Kid | Anyone using the Grandstream GXP-2000? |
| 14:59.23 | Merlin2006 | anyone that can help me with chan_misdn install? |
| 15:02.09 | Ciber311 | yes Feral_Kid, they suck :P |
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| 15:44.15 | cuco | anyone using a siphone.com account? it seems that i cannot dial my FWD account from siphone.com |
| 15:44.24 | cuco | does it work for anyone? |
| 15:48.13 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:49.51 | Lorbas | cuco: (how) is that related to asterisk? |
| 15:49.58 | Ciber311 | lol |
| 15:51.23 | cuco | Lorbas: both accounts are on an asterisk box...? (ya i konw lame excuse) |
| 15:54.44 | Lorbas | cuco: you mean: fwd and siphone both use an asterisk as regsitration server? |
| 15:56.26 | cuco | nope. the opposite. |
| 15:57.24 | rbordeaux | hi, I install a second (generi) x100p but I cannot configure it correctly. I checked zaptel.conf and zapata.conf. On the cli a zap show status shows both boards as OK but a zap show channel 2 gives me a "Unable to find given channel 2". Any idea to see what I am missing? |
| 15:57.34 | Lorbas | cucp: you use an asterisk to register yourself with fwd and siphone? |
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| 15:58.46 | markQing | hello |
| 16:00.30 | cuco | Lorbas, yes. i can make calls from my asterisk (acutally 2 different servers) into both services echo tests. it seems that the problem is on the FWD setup. |
| 16:01.29 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 16:01.31 | Lorbas | cuco: any nat constellation? |
| 16:03.12 | *** part/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 16:04.12 | cuco | yes, exactly my toughts. even tough that the FWD is using iax2, so this should not be a problem right? |
| 16:06.58 | Lorbas | cuco: so can you call from anywhere else into your fwd account? |
| 16:08.24 | cuco | from my FWD yes, it seems that dialing INTO is the problem |
| 16:09.46 | Lorbas | cuco: so my question was |
| 16:10.19 | Lorbas | cuco: were you able to call into fwd from anywhere else but siphone? |
| 16:10.33 | cuco | nope. |
| 16:11.26 | *** join/#asterisk Lorbas (n=b@pd956943c.dip0.t-ipconnect.de) |
| 16:13.26 | Lorbas | cuco: sorry, my irc client crashed (should you have answered) |
| 16:13.46 | cuco | the answer was no |
| 16:14.24 | Lorbas | hm |
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| 16:14.49 | Lorbas | but you can verify that its registered? |
| 16:19.28 | blitzrage | yah, if not registered, FWD won't know how to route the call to you |
| 16:20.43 | Lorbas | but it seems like it doesn't, you know. |
| 16:21.13 | Lorbas | maybe i misses something from you error description |
| 16:21.36 | Lorbas | but to me it sounds like it doesn't route the call to your * box |
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| 16:41.52 | rbordeaux | hi, I install a second (generic) x100p but I cannot configure it correctly. I checked zaptel.conf and zapata.conf. On the cli a zap show status shows both boards as OK but a zap show channel 2 gives me a "Unable to find given channel 2". Any idea of what I am missing? |
| 16:41.56 | DeaLER- | Can someone help me find a good working MGCP/NCS channel source? |
| 16:42.20 | DeaLER- | isn't x100p a single channel card? |
| 16:42.33 | rbordeaux | yes |
| 16:42.40 | rbordeaux | I install a second card |
| 16:42.51 | rbordeaux | s/install/installed |
| 16:43.18 | DeaLER- | and channel 1 works? |
| 16:43.21 | rbordeaux | yes |
| 16:43.29 | DeaLER- | are u sure it sees 2 cards |
| 16:43.41 | DeaLER- | what does dmesg say |
| 16:44.01 | rbordeaux | rbast*CLI> zap show status |
| 16:44.01 | rbordeaux | Description Alarms IRQ bpviol CRC4 |
| 16:44.01 | rbordeaux | Generic Clone Board 1 OK 0 0 0 |
| 16:44.01 | rbordeaux | Generic Clone Board 2 OK 0 0 0 |
| 16:44.01 | rbordeaux | rbast*CLI> |
| 16:44.23 | dlynes_laptop | rbordeaux: zap show channels shows what? |
| 16:44.31 | DeaLER- | you probably don't have ur zapata.conf properly set up |
| 16:45.00 | DeaLER- | instead of going channel => 1-2 do them seperate |
| 16:45.10 | rbordeaux | rbast*CLI> zap show channel 2 |
| 16:45.10 | rbordeaux | Unable to find given channel 2 |
| 16:45.10 | rbordeaux | rbast*CLI> |
| 16:45.16 | rbordeaux | I tried them separte |
| 16:45.18 | DeaLER- | zap show channels |
| 16:45.22 | dlynes_laptop | zap show channels, not zap show channel 2 |
| 16:45.24 | Strom_C | rbordeaux: please stop flooding the irc channel |
| 16:45.29 | Strom_C | use pastebin |
| 16:45.31 | Strom_C | ~bp |
| 16:45.36 | jbot | hmm... bp is pb backwards |
| 16:45.37 | Strom_C | er |
| 16:45.37 | Strom_C | ~pb |
| 16:45.38 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 16:45.43 | DeaLER- | does asterisk business edition support NCS 1.0? |
| 16:46.00 | rbordeaux | rbast*CLI> zap show channels |
| 16:46.00 | rbordeaux | <PROTECTED> |
| 16:46.00 | rbordeaux | <PROTECTED> |
| 16:46.00 | rbordeaux | <PROTECTED> |
| 16:46.00 | rbordeaux | rbast*CLI> |
| 16:46.18 | DeaLER- | pseudo? |
| 16:46.22 | dlynes_laptop | You only have one channel configured |
| 16:46.24 | rbordeaux | ok, sorrry |
| 16:46.36 | dlynes_laptop | DeaLER-: pseudo is always configured when you're doing zap |
| 16:47.04 | dlynes_laptop | rbordeaux: pastebin your zapata.conf file |
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| 16:48.13 | DeaLER- | what is the difference between asterisk business and asterisk non-business |
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| 16:52.37 | rbordeaux | dlynes_laptop: http://pastebin.com/782809 (sorry for the delay, browser issues) |
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| 16:55.07 | dasenjo | Hi, I used to load the wcfxs module in the kernel to handle the FXO in the TDM400 ... as I can see, now it is called wxtdm, but when I try to modprobe it, I get an error saying me that wcfxs can't be found. ¿What can I do? |
| 16:55.24 | SwK | dealer: you dont get the source with ABE and you pay a per channel fee for it and it comes with paid support... the free version is GPL'd but you have to pay for all professional support |
| 16:55.55 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:56.59 | dasenjo | sorry .. s/wxtdm/wctdm/ |
| 16:57.32 | dasenjo | I'm using zaptel-1.2.8 |
| 17:01.43 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 17:05.17 | teknoprep | is there a way to dial a number that would change the incomming route for a number from say 1 ivr to another ivr? say a feature code? so at closing of shop i can dial this number.. and it tells all incomming callers on a new IVR that the store is closed... then at open i can go back to being open... |
| 17:13.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.111) |
| 17:13.31 | benjk | exten => *888,1,DBput(ivr/mode=bizhours) and exten => *999,1,DBput(ivr/mode=afterhours) |
| 17:14.38 | benjk | the in your incoming context you get the value with DBget and route to one or the other IVR context |
| 17:14.43 | teknoprep | ahh nice |
| 17:15.01 | *** join/#asterisk liran_ (i=liran@212.199.177.203.static.012.net.il) |
| 17:15.03 | teknoprep | now i must understand what you just told me lol |
| 17:15.16 | teknoprep | since i am a freepbx user i will have to probably read the asterisk manual to get this working |
| 17:15.21 | benjk | read the documentation for DBget and DBput |
| 17:15.23 | teknoprep | but thanx for at least telling me i could doit |
| 17:15.29 | teknoprep | will do |
| 17:15.43 | benjk | on the console: show application DBget |
| 17:15.51 | benjk | likewise for DBput |
| 17:16.17 | benjk | for branching you can use GotoIf() |
| 17:17.09 | teknoprep | so ivr/mode=foo is pointing to an IVR config that is named foo? |
| 17:17.43 | teknoprep | and the other part is the *foo ... is configuring that route to be setup for either *foo or *foo2 depending on what is dialed |
| 17:17.48 | benjk | GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 17:17.49 | teknoprep | is this a correct assumption |
| 17:18.01 | teknoprep | yeh i can't do this with a time/day scenario |
| 17:18.14 | teknoprep | as the times changes as its based on what time ppl schedule appointments |
| 17:18.19 | teknoprep | so the closing times change too much |
| 17:18.28 | benjk | its not based on time |
| 17:18.36 | teknoprep | GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) ? |
| 17:18.38 | benjk | its based on the setting of ivrmode |
| 17:18.49 | benjk | and that setting you set with DBget |
| 17:18.58 | dasenjo | tzafrir, can you help me? why loading wctdm depends on loading wcfxs if it does not exist in 1.2.8? |
| 17:18.58 | teknoprep | ahh |
| 17:19.05 | benjk | and that fetches the setting from the database, which you control by dialing *888 or *999 |
| 17:19.28 | teknoprep | wonder if freepbx has a config for this type of setup? |
| 17:19.30 | teknoprep | would be nice |
| 17:19.43 | benjk | I have no clue |
| 17:19.50 | teknoprep | yeah i know... just wondering |
| 17:19.58 | benjk | and I doubt anybody here has any clues about freepbx |
| 17:19.59 | teknoprep | i still need to read the rest of the asterisk manual... |
| 17:20.13 | teknoprep | yes i know.. i only typed it out as a general out-loud though |
| 17:20.14 | teknoprep | t |
| 17:20.20 | benjk | you will have more luck asking in #freepbx |
| 17:20.27 | teknoprep | yes i know.. i am in there also lol |
| 17:20.59 | benjk | in any event this is very simple to do |
| 17:21.08 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 17:21.16 | benjk | step 1, get the setting from the database (DBget) into a dialplan variable |
| 17:21.23 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 17:21.54 | benjk | step 2, check the dialplan variable and branch to the respective ivr context (using GotoIf) |
| 17:22.23 | RoyK | ~dtmf |
| 17:22.25 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 17:22.25 | benjk | assuming you have one context for you biz hours ivr and another for your after hours ivr |
| 17:22.39 | teknoprep | benjk of course.. |
| 17:23.07 | benjk | step 3, there is no step 3 |
| 17:23.10 | teknoprep | lol |
| 17:24.06 | teknoprep | one thing i don't understand |
| 17:24.16 | teknoprep | how does the system know of different inbound routes |
| 17:24.25 | teknoprep | i add say exten => *888,1,DBput(ivr/mode=bizhours) and exten => *999,1,DBput(ivr/mode=afterhours) |
| 17:24.32 | teknoprep | but i have 3 inbound routes for differnet numbers |
| 17:24.40 | teknoprep | i would only want to do this for 1 number say... |
| 17:25.17 | *** join/#asterisk marta (n=a@host184-5.pool8711.interbusiness.it) |
| 17:25.19 | *** join/#asterisk JohnRobert (n=john@i-195-137-116-41.freedom2surf.net) |
| 17:25.20 | benjk | the *888 and *999 are only there for you to change the ivrmode |
| 17:25.21 | marta | hello |
| 17:25.29 | JohnRobert | is IAX2 backwards compatiable with IAX1? |
| 17:25.41 | benjk | no, its not |
| 17:25.52 | Lorbas | teknoprep: you can chek the callerid in a gotoif statement |
| 17:25.54 | benjk | IAX1 is dead |
| 17:26.18 | JohnRobert | hmm |
| 17:26.25 | benjk | has been for years |
| 17:26.26 | teknoprep | ohhh |
| 17:26.29 | teknoprep | i understand |
| 17:26.37 | teknoprep | now i got it |
| 17:26.45 | benjk | teknoprep, there are two independent parts |
| 17:26.50 | teknoprep | yeah i understand now |
| 17:26.56 | benjk | one is the setting of the ivrmode in the database |
| 17:27.03 | Lorbas | teknoprep: or if the phone you want to dial from is voip phone you could just choose a drifferent incoming context |
| 17:27.11 | benjk | the other is the checking of that value and branching when a call comes in |
| 17:27.16 | teknoprep | you are just setting a variable to x with the first statement... then x is used to declare which is true in the second statement |
| 17:27.37 | JohnRobert | doesn't matter, the server is IAX2 anyway |
| 17:27.43 | marta | does anyone know if the 'callid' field (in chan_sip.c) is uniq in the iflist? |
| 17:29.02 | DeaLER- | anyone where a mgcp or ncs expert? |
| 17:29.51 | teknoprep | Lorbas they are all voip phones yes |
| 17:31.25 | JohnRobert | can anyone recommend a good softphone that supports IAX2? |
| 17:32.01 | JohnRobert | (nice gnome one where possible) |
| 17:32.54 | Lorbas | teknoprep: so you could declare context=service in protocol.conf and include=>incoming in "service" |
| 17:34.22 | Lorbas | then you can define some other service tasks in "service" for that specific phone, too |
| 17:34.54 | marta | I found the answer, tnx. |
| 17:39.30 | hmmhesays | bah |
| 17:42.27 | *** join/#asterisk XGizzmo (n=gizzmo@unaffiliated/xgizzmo) |
| 17:42.31 | *** join/#asterisk dacleric (n=dacleric@p54820827.dip0.t-ipconnect.de) |
| 17:52.43 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 18:05.30 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 18:05.42 | bkw_ | RoyK, you share the love? |
| 18:06.44 | RoyK | the love for parasites? |
| 18:06.54 | bkw_ | that site I showed you :P |
| 18:11.15 | dasenjo | help me please .. why zaptel-1.2.8 is trying to load wcfxs when modprobing wctdm? |
| 18:11.49 | dasenjo | I really need to set up my lines after tomorrow .. :p |
| 18:19.07 | dlynes_laptop | dasenjo: because you have it set that way in your startup to automatically load all zaptel devices |
| 18:19.21 | dlynes_laptop | dasenjo: fix your /etc/modprobe.conf file |
| 18:21.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:21.34 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 18:21.36 | dasenjo | dlynes_laptop, I have thie line "alias wctdm wcfxs" in /etc/modprobe.d/zaptel |
| 18:21.43 | *** part/#asterisk DeaLER- (n=Tester@Toronto-HSE-ppp3697785.sympatico.ca) |
| 18:21.45 | dasenjo | should I remove it? |
| 18:22.37 | dasenjo | I have no /etc/modprobe.conf |
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| 18:24.27 | *** part/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 18:26.48 | dlynes_laptop | dasenjo: why would you have that line? what kind of card are you using? |
| 18:27.16 | dasenjo | dlynes_laptop, I'm using a TDM400 |
| 18:27.34 | dlynes_laptop | dasenjo: yeah...lose the alias line, then |
| 18:27.37 | dasenjo | I commented it,run depmod -a, but the error is still there |
| 18:28.17 | dlynes_laptop | dasenjo: pb the error |
| 18:28.28 | dlynes_laptop | dasenjo: i.e. the whole dump of depmod -a |
| 18:29.20 | dasenjo | depmod -a does not output anythind |
| 18:29.36 | dlynes_laptop | You just finished saying it outputted an error |
| 18:29.43 | dlynes_laptop | So it's not outputting an error then? |
| 18:29.47 | *** join/#asterisk remiss (i=bofh@46.80-203-38.nextgentel.com) |
| 18:29.57 | dasenjo | icom-ppop:~# modprobe wctdm |
| 18:29.58 | dasenjo | FATAL: Module wcfxs not found. |
| 18:29.58 | dasenjo | FATAL: Error running install command for wcfxs |
| 18:30.19 | dasenjo | the error appears when I try to load wxtdm |
| 18:30.26 | dasenjo | !wctdm |
| 18:30.59 | dlynes_laptop | because you have an alias of wctdm -> wcfxs |
| 18:31.05 | dlynes_laptop | so it's trying to load wcfxs, not wctdm |
| 18:32.23 | *** join/#asterisk saftsack (n=saftsack@p54A7EC7B.dip.t-dialin.net) |
| 18:32.27 | Juggie | he did say he commented it out |
| 18:33.38 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:34.14 | saftsack | whats about wifi ip telephones? are they working properly? are they less reliable than dect telephones ore more? |
| 18:34.29 | benjk | they work but only just about |
| 18:34.31 | dlynes_laptop | saftsack: how reliable is wifi, period? |
| 18:34.45 | benjk | and the phones themselves suck too |
| 18:34.47 | dlynes_laptop | saftsack: i don't even find it all that reliable for computers, never mind telephones |
| 18:35.27 | benjk | its ok if you have it as a secondary extension that rings together with your deskphone |
| 18:35.34 | Ryushin | Okay, I got my first polycom phone to register with asterisk. Now on to the next step. I'm going through the demo stuff that asterisk installed. When I try and make a call out the PRI, I get "pbx_extension_helper: Cannot find extension context 'from-sip'". I assume the problem is that I haven't defined the from-sip context. Where would I define this? |
| 18:35.57 | dlynes_laptop | dasenjo: pastebin modprobe -c |
| 18:35.59 | wunderkin | Ryushin: extensions.conf |
| 18:36.09 | saftsack | dlynes_laptop, how do you mean? |
| 18:36.15 | Ryushin | vunderkin: Thanks. |
| 18:36.43 | dlynes_laptop | saftsack: I've had computers right beside the wap, and they can't get a signal, but yet another computer 15 feet away from it can get full signal strength |
| 18:37.34 | saftsack | so it would be the best way to further use dect telephones? |
| 18:37.49 | dasenjo | dlynes_laptop, I hava the mentioned line in /etc/modprobe.d/zaptel .. it is being generated automatically .. :s |
| 18:38.18 | dasenjo | "alias wctdm wcfxs" |
| 18:38.22 | benjk | the best would be a DECT base station that can speak SIP |
| 18:38.24 | dlynes_laptop | dasenjo: what's generating it automatically? |
| 18:40.02 | dasenjo | I don´t know, but comenting it does not work even |
| 18:41.05 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
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| 18:41.56 | l-fy | hey lilo |
| 18:41.59 | l-fy | long time |
| 18:42.37 | RoyK | benjk: indeed, preferably with one able to do g.726 passthrough to the sip gate |
| 18:43.18 | benjk | even if its only g711 that'll be good enough as you could always transcode on your server |
| 18:43.49 | RoyK | benjk: DECT uses g.726 |
| 18:44.17 | RoyK | so doing passthrough would help to avoid transcoding in the first place |
| 18:44.48 | benjk | how many DECT phones could you possible have on a single base |
| 18:44.57 | RoyK | dunno |
| 18:44.58 | RoyK | a few |
| 18:45.03 | RoyK | ten-ish i thinki |
| 18:45.07 | RoyK | s/i$// |
| 18:45.51 | benjk | well, I generally use G711 for just about everything unless I call somebody in the Middle East or India or Africa |
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| 18:46.00 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:46.25 | benjk | and there its mostly GSM 06.10 |
| 18:46.28 | RoyK | benjk: still, DECT uses g.726 between handsets and base |
| 18:46.44 | RoyK | gsm isn't really that good, though |
| 18:47.04 | l-fy | RoyK > than is a stupid dect |
| 18:47.05 | RoyK | also, with the new packetization code, you can tune down overhead a lot :) |
| 18:47.05 | benjk | it has to transcode anyway if it uses a POTS interface, no big deal |
| 18:47.13 | RoyK | l-fy: by standard |
| 18:47.25 | benjk | who cares |
| 18:47.46 | RoyK | pots is evil :) |
| 18:47.48 | benjk | if you call Africa or India you don't make any comment like "isn't really good" |
| 18:47.58 | l-fy | no dear |
| 18:48.00 | benjk | you can be happy you can call at all |
| 18:48.04 | l-fy | stupid people are evil |
| 18:48.09 | RoyK | l-fy: Like G.711, G.726 has its roots in the PSTN network. It is primarily used for international trunks to save bandwidth. Where G.711 uses 64 Kbps, G.726 uses 32 Kbps, providing nearly the same quality. It is also the standard codec used in DECT wireless phones. |
| 18:48.14 | benjk | instead of smoke signals or sending a courier on a camel |
| 18:48.49 | benjk | RoyK, G726 isn't really just 32K |
| 18:48.59 | benjk | it has a whole bunch of modes |
| 18:49.00 | RoyK | i know.... |
| 18:49.12 | benjk | what you are probably talking about is ADPCM |
| 18:49.15 | RoyK | "The bitrate can be 16, 24, 32 or 40 Kbps, but 32 Kbps is the defacto standard." |
| 18:49.18 | dlynes_laptop | I thought g726 was wideband, too? |
| 18:49.20 | benjk | which is like G711 half rate |
| 18:49.24 | RoyK | benjk: just quoted from http://www.voip-info.org/wiki/view/ITU+G.726 |
| 18:49.26 | RoyK | dlynes_laptop: no. |
| 18:49.32 | RoyK | dlynes_laptop: g.722 is, though |
| 18:49.33 | benjk | PHS uses that too |
| 18:49.41 | dlynes_laptop | Ah...getting it mixed up with g.722 then |
| 18:49.43 | dasenjo | dlynes_laptop, http://pastebin.ca/159959 is my modprobe -c |
| 18:49.57 | benjk | it is very cheap to convert from G711 to ADPCM |
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| 18:54.36 | dlynes_laptop | dasenjo: can you turn udev and hotplug off? |
| 18:55.08 | dasenjo | dlynes_laptop, I thnk so, I'm gonna try |
| 18:56.03 | dasenjo | # apt-get remove hotplug .. :p |
| 18:56.46 | dlynes_laptop | I said disable it, not remove it :) |
| 18:57.48 | dasenjo | it is a telephony server .. I don't need it |
| 19:01.00 | dasenjo | dlynes_laptop, I have never disabled udev .. is it going to be removed with hotplug ? |
| 19:02.06 | dlynes_laptop | I believe hotplug loads it |
| 19:02.16 | Strom_M | do any of you have experience with the mysql cdr addon? |
| 19:02.18 | dlynes_laptop | the kernel already has udev loaded though, too |
| 19:02.27 | dlynes_laptop | but the userspace stuff is loaded by hotplug |
| 19:02.48 | dlynes_laptop | at least on my distribution...i'm guessing it's the same on all |
| 19:02.55 | saftsack | whats the reason why voip telephones are such expensive in comparison to BRI telephones? |
| 19:03.11 | Strom_M | saftsack, less demand? |
| 19:03.21 | Strom_M | more horsepower required? |
| 19:06.02 | dasenjo | dlynes_laptop, i removed hotpug but modprobe -c did not change, in http://pastebin.ca/159969 is the /etc/modprobe.d/zaptel file |
| 19:06.09 | saftsack | hmm for a normal BRI telephone i pay about 60eu. for a normal ip telephone 140eu |
| 19:06.17 | dlynes_laptop | dasenjo: did you reboot? |
| 19:06.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:06.29 | dasenjo | yes, I rebooted |
| 19:07.11 | dasenjo | actually, the machine rebooted, I just typed the command :p :) |
| 19:15.57 | dasenjo | dlynes_laptop, did you the zaptel file? |
| 19:17.51 | benjk | saftsack, economies of scale still in favour of BRI phones but about to change |
| 19:19.51 | saftsack | ok so sip telephones will be cheaper in the future? |
| 19:23.19 | benjk | most likely so |
| 19:26.33 | rene- | bri boards are expensive |
| 19:26.56 | rene- | i think i have a bad digium board |
| 19:27.15 | rene- | i have created a loop between the two E1 interfaces of it |
| 19:27.19 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 19:27.46 | rene- | i saw a couple of HDLC errors but i left running the box all night but do data was logged for most of the period |
| 19:27.54 | rene- | s/do/no/ |
| 19:28.18 | rene- | like 10 hours or so are missing from both debug and messages log |
| 19:28.51 | rene- | anyway how can i stress test the card to see if i can get more of this HDLC errors and make a claim with digium that my board has gone bad?? |
| 19:29.16 | rene- | in production with a live E1 line i am experiencing half hour blockings of the card each 5 1/2 hours or s |
| 19:29.20 | rene- | o |
| 19:30.20 | rene- | the port goes online by itself but it leaves me without connectivity to PSTN for a total of over 2 hours each day |
| 19:30.33 | rene- | which sucks |
| 19:31.10 | rene- | and well digium has been somewhat responsive but well i havent been able to reach them since friday evening and they wont be available tomorrow either |
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| 19:35.56 | dasenjo | dlynes_laptop, no clues? |
| 19:37.46 | rene- | it is so lonely in here |
| 19:40.30 | l-fy | hey rene- |
| 19:41.11 | rene- | hello I-fy |
| 19:41.14 | rene- | how is it going |
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| 19:58.23 | Ryushin | Are there any specific howtos for setting up a pri, polycom phones, etc. for a office? It would be nice to have a step by step document for a base. |
| 19:58.57 | rene- | setting up a pri is well documented |
| 19:59.02 | Strom_M | so you want a consultant in a book? |
| 19:59.16 | rene- | the polycoms are abit harder but there are administrative guides to download from polycom web site |
| 19:59.31 | rene- | Strom_M =~ Strom_C? |
| 19:59.38 | Strom_M | yes |
| 19:59.38 | Strom_M | :) |
| 20:00.22 | rene- | do you have any advice for my E1 woes? |
| 20:00.47 | rene- | how can i stress my box to see if the card is missing frames? |
| 20:01.05 | l-fy | rene- > loop |
| 20:01.22 | Ryushin | More like the howtos for setting up Cyrus, Postfix, Amavis, MythTV, etc. They're are plenty of howtos for stuff like that. It's just nice to see the steps that laid out and I can customize from that point. |
| 20:03.06 | rene- | Ryushin: the voip-info wiki should help you with the E1 part |
| 20:03.23 | rene- | and anyway Email is a very old techonology and it is better understood than voip |
| 20:03.54 | rene- | voip is a lot more complex than any of the above you mentioned |
| 20:04.14 | Strom_M | there's a shitload of theory you need to understand before you should set up a PBX for an office |
| 20:04.41 | rene- | as they said it is kind of a black art |
| 20:05.28 | rene- | and following a recipe will get you only so far |
| 20:06.08 | rene- | you need to invest some time in learning how to do it properly, for the PRI part you have it easy just look into the wiki |
| 20:06.19 | Strom_M | telephony != baking a cake |
| 20:13.30 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 20:14.31 | Ryushin | rene-: I disagree that you consider voip harder to understand than enterprise grade mail. Mail can be VERY complex. These are just two technologies that have a steep learning curve. |
| 20:14.56 | Ryushin | I am learning these procedures. It's just nice to have a document that someone has gone through. |
| 20:15.07 | Ryushin | I'll go through the wiki some more. |
| 20:16.23 | *** join/#asterisk niZon (n=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 20:16.49 | Ryushin | rene-: And the recipe is only supposed to take you so far. It's a foundation. Not a complete solution. |
| 20:17.10 | *** join/#asterisk micky (n=micky@80.96.92.40) |
| 20:18.55 | benjk | setting up asterisk is very much like baking a cake |
| 20:19.05 | _Vile | more like baking cookies |
| 20:19.23 | benjk | you can do the simple stuff by recipe, but for sophisticated stuff you need lots of experience |
| 20:19.34 | Ryushin | Just like I can install postfix and be able to send mail out of a system. Configure the system to deliver mail to a local account. It's a start. |
| 20:20.14 | *** join/#asterisk j23a (n=nikhil@ip68-229-76-11.ri.ri.cox.net) |
| 20:20.17 | _Vile | but filtering and plaing with i.e. maia, need to know a little more |
| 20:20.45 | j23a | I'm having trouble with asterisk-perl and having 'callback' work so that when i'm doing a record_file and the user hangs up stuff is excuted after the hangup. It doesn't seem to call the subroutine i defined. does anyone have any ideas? |
| 20:25.59 | *** join/#asterisk Shoragan (n=shoragan@datenfreihafen.org) |
| 20:26.59 | micky | hello, does asterix need to run a new daemon or child for each incomming call or icecast stream ? i'm asking this because i'm trying to figure out why my server starts 100 processes .. have a look at http://pastebin.com/782950 (almost the whole ps-aux .. pastebin did not accept the whole thing) btw the server is a 3GHZ Xeon with 256 RAM. |
| 20:28.06 | *** join/#asterisk Libila (n=vye@ip68-6-130-152.sd.sd.cox.net) |
| 20:28.50 | Libila | I've got an SIP conference going here at the house. Some of the phones are echoing, others arent. Do I look in the asterisk config files for fixing this problem or is it the phone settings? |
| 20:29.10 | hmmhesays | what kind of phones? |
| 20:29.11 | _Vile | what kind of phones |
| 20:29.18 | hmmhesays | haha |
| 20:29.19 | Libila | polycom / grandstream |
| 20:29.23 | _Vile | u beat me :) |
| 20:29.30 | _Vile | grandstream will have issues |
| 20:29.35 | Libila | grandstream seems to be fine, the polycom are echoing |
| 20:29.35 | _Vile | poly should be golden |
| 20:29.39 | Libila | haha |
| 20:29.42 | _Vile | whoah |
| 20:30.22 | _Vile | that should not be |
| 20:31.10 | _Vile | polycom sip phones? |
| 20:32.44 | Libila | _Vile: yeah it's the SoundPoint IP 501 |
| 20:32.53 | Libila | both are |
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| 20:33.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:33.19 | _Vile | that should be golden, where's the echo? outbound or phone to phone? |
| 20:33.45 | Libila | it's speaker phone, maybe that has something to do with it |
| 20:33.55 | Libila | all the phones are in seperate rooms but they're all on speaker phone. |
| 20:33.56 | _Vile | built in echo can, where's the echo? |
| 20:34.04 | _Vile | every room? |
| 20:35.32 | Libila | in and out of the polycom is echoish |
| 20:35.47 | Libila | the grandstream can even hear the echos from the polycom phones when it gets bad |
| 20:36.13 | Libila | and I've just added a x-lite soft phone in another room and I can hear the echo bad from that phone on my polycom |
| 20:36.30 | _Vile | http://threebit.net/mail-archive/asterisk-users/msg00358.html |
| 20:36.35 | Libila | although, the person talking on the grandstream sounds crystal clear |
| 20:37.38 | _Vile | http://threebit.net/mail-archive/asterisk-users/msg00354.html |
| 20:37.40 | gambolputty | Hi. I am testing call parking and I don't get a voice back indicating the extension the call got parked on. Could someone help? |
| 20:37.45 | rene- | Ryushin: i agree than mail is a very complex technology, i myself being mostly ignorant about it but i think you will agree with us as voip being somewhat more involved |
| 20:39.03 | j0 | all i need is some magical software program to troubleshoot quality issues.. then the rest will be like baking a cake to me |
| 20:39.13 | rene- | exactly |
| 20:39.18 | kronos003 | i just realized that freeswitch has an interesting voice conference - how cool! |
| 20:39.33 | rene- | if it wasnt for quaility issues voip wouldnt be half as hard |
| 20:40.45 | micky | so .. any idea in why / if asterisk needs +1 instance for each inbound call or icecast stream ? ps -aux > http://pastebin.com/782950 |
| 20:40.46 | rene- | anyway how can i stress test between two ports of a digium E1 card? |
| 20:42.07 | Libila | _Vile: man I'm still going through this looong thread |
| 20:44.04 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
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| 20:51.31 | Strom_M | gambolputty, you need to do an attended transfer for call parking to function |
| 20:52.22 | Libila | _Vile: guess what, I hung up the grandstream and ALL the echo went away. |
| 20:52.30 | Libila | _Vile: turns out you were right |
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| 20:52.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:53.30 | gambolputty | the person at the other end wont have a phone just voicemail |
| 20:53.35 | gambolputty | I did a blind transfer |
| 20:53.42 | rene- | grandstream = crap |
| 20:53.51 | gambolputty | snom = not crap |
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| 20:55.07 | gambolputty | is there a way to have the park extension announced with a blind transfer? |
| 20:56.58 | Strom_M | gambolputty, no |
| 20:57.21 | Strom_M | gambolputty, you need to do an attended transfer so that the party doing the transfer knows what extension the call is parked on |
| 20:57.46 | Strom_M | with a blind transfer, you cant keep the audio path up after initiating the transfer |
| 20:58.10 | gambolputty | I blind transfer the call to 700 the first default park extension |
| 20:58.19 | Strom_M | you're not listening to me |
| 20:58.40 | gambolputty | how can I do an attended transfer to 700 when 700 is not a phone? |
| 20:58.54 | Strom_M | it doesnt matter whether or not it's a phone |
| 20:59.05 | Strom_M | you can do attended transfers to voicemail, and voicemail is not a phone |
| 21:00.32 | gambolputty | just tried doing an attended transfer to 700 and it didn't work |
| 21:01.27 | Strom_M | what happens when you just dial 700? |
| 21:01.49 | gambolputty | 700 get declined |
| 21:01.51 | gambolputty | for trying to dial it |
| 21:04.37 | gambolputty | http://pastebin.ca/160090 please see that |
| 21:04.39 | Strom_M | did you include the callparking context in your dialplan? |
| 21:04.45 | gambolputty | yes |
| 21:04.48 | gambolputty | see the pastebin |
| 21:05.02 | Strom_M | no no no |
| 21:05.04 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 21:05.08 | gambolputty | ? |
| 21:05.09 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:05.10 | Strom_M | take the parkedcalls context out of your dialplan |
| 21:05.18 | *** join/#asterisk Shoragan (n=shoragan@datenfreihafen.org) |
| 21:05.37 | Strom_M | just leave the include |
| 21:05.51 | Strom_M | the [parkedcalls] in extensions.conf is overriding the definition in features.conf |
| 21:08.28 | gambolputty | looks like it works now |
| 21:08.52 | gambolputty | just tested it |
| 21:08.54 | gambolputty | thanks |
| 21:08.59 | Strom_M | woot |
| 21:09.02 | Strom_M | very welcome |
| 21:10.24 | *** join/#asterisk cian (n=cian@cian.ws) |
| 21:16.25 | shodan | is there a reason why I have to use an extra number in front of all outgoing calls ? (like 9) can't I let the use just dial the number and send it out and instead all the extentions could begin with * or 0 ? |
| 21:16.41 | j0 | heres a voice recording of my garbled audio.. it happens only on outgoing audio, with any codec, i'm just wondering if its 100% caused by a bad network somewhere, and if its packet loss or jitter that would cause it. http://upload.ohshare.com/v/976446/voip_test_2.wav.html |
| 21:17.15 | Strom_M | shodan, * conflicts with vertical service codes; 0 conflicts with operator and TOPS/OSPS dialing |
| 21:17.36 | Strom_M | shodan, make all calls 1+ and have extensions beginning with 2-9 |
| 21:17.51 | gambolputty | I don't use 9, but mine is programmed to take it if it is dialed. I go off the length of the digit sequence, and make decisions. |
| 21:17.53 | Strom_M | or have all calls prefixed with 9 and have extensions beginning with 1-8 |
| 21:18.26 | Strom_M | gambolputty, that's a fairly bad way of doing it; you must not have any zap phones on your system |
| 21:18.53 | gambolputty | I have IP phones, and these statements would work with any phone |
| 21:18.58 | gambolputty | no matter what the type |
| 21:19.06 | Strom_M | ok.... |
| 21:19.37 | Strom_M | zap phones match dialplan as you dial digit-by-digit |
| 21:19.54 | gambolputty | didn't know tht |
| 21:19.55 | gambolputty | that |
| 21:20.04 | Strom_M | so if you're trying to dial 555-2368 but you have an extension called 5552, you'll never be able to reach 555-2368 |
| 21:20.09 | gambolputty | I am used to pressing an OK button |
| 21:20.14 | shodan | yeah but 1 is the prefix for long distance calls , the users might be worried they are making long distance call accidently ? |
| 21:20.28 | gambolputty | ok |
| 21:20.33 | Strom_M | shodan, so prefix calls outside the PBX with 9 |
| 21:20.35 | gambolputty | use 9 for analog phones then |
| 21:20.52 | Strom_M | gambolputty, you've still got a bad dialplan design |
| 21:21.12 | shodan | hmm |
| 21:21.18 | Strom_M | no dialable number should ever be a subset of any other dialable number |
| 21:21.27 | gambolputty | if need be mine can change |
| 21:22.02 | gambolputty | by a "zaptel" phone are you referring to an analog phone hooked up to a fxs card? |
| 21:22.09 | Strom_M | yes |
| 21:22.20 | gambolputty | I won't encounter that at work |
| 21:22.25 | gambolputty | but I will keep it in mind |
| 21:22.44 | Strom_M | lord, i hope you're not setting up production systems :) |
| 21:23.03 | gambolputty | this is just my personal system |
| 21:23.06 | gambolputty | works quite fine |
| 21:23.13 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 21:23.28 | Strom_M | yeah, I hate the "works quite fine" mentality |
| 21:23.50 | Strom_M | you should have seen the mess I was presented with yesterday morning because the consultant thought it worked just fine |
| 21:23.57 | gambolputty | if I have errors I fix them |
| 21:24.02 | gambolputty | what happened then? |
| 21:24.25 | Strom_M | the network was such a mess that I had to literally tear it apart and rebuild it from scratch |
| 21:24.47 | gambolputty | ok |
| 21:24.52 | gambolputty | thats apart from a pbx |
| 21:24.55 | gambolputty | but I can relate |
| 21:24.59 | Strom_M | this /was/ the PBX |
| 21:25.16 | kusznir | Hi all: I would like to set up asterisk to offer traditional-style phone services to other parties where they post-pay for usage. More importantly, the costs of the calls differs by the number dialed. I know cdr's are used for this, but so far, asterisk's cdr's only give what actually happened. What would I use to then process this and turn it into a bill? |
| 21:25.47 | Strom_M | kusznir, you write custom software to parse the CDRs and determine call costs by terminating rate center |
| 21:26.22 | kusznir | Nobody's written something open source that will do the processing given a csv or other dB of the per-min rate? |
| 21:26.37 | Strom_M | possibly |
| 21:26.40 | kusznir | It seems like a standard-enough use for the cdr.... |
| 21:27.13 | kusznir | I'm not even shure what to google for....asterisk cdr processor only returns unrevevant links.... |
| 21:27.43 | Strom_M | beats me; ive never looked into it |
| 21:28.54 | mitcheloc | Strom_M ~ Strom_C? |
| 21:28.59 | Strom_M | yes |
| 21:29.05 | Strom_M | didn't you hear? I come in six packs! |
| 21:29.24 | mitcheloc | whoo! what's the other 4 called? |
| 21:29.25 | Strom_M | :) |
| 21:29.51 | Strom_M | Fruit Punch, Cola, Vodka, and Butterscotch |
| 21:30.15 | mitcheloc | no thanks =/ |
| 21:32.28 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 21:32.49 | IronMan2000 | hello |
| 21:33.03 | Strom_M | good afternoon |
| 21:33.34 | IronMan2000 | can anyone here help me with my asterisk server? |
| 21:33.40 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 21:33.40 | *** join/#asterisk hotroot (n=michael@pD9E95CCB.dip.t-dialin.net) |
| 21:33.43 | Strom_M | IronMan2000, just ask a question already |
| 21:34.14 | IronMan2000 | I am trying to configure my extension.conf to be able to make 800 and other toll free calls. |
| 21:34.26 | IronMan2000 | not working |
| 21:34.33 | IronMan2000 | need a sample config |
| 21:34.55 | Strom_M | exten => _1800NXXXXXX,1,Dial(whatever/${EXTEN}) |
| 21:34.57 | Strom_M | and so on |
| 21:36.30 | hotroot | hi @ all. Can someone tell me how I cam make asterisk send a stuttered dialtone to my isdn-phone when there are messages waiting for the specified mailbox (zapata.conf mailbox=XXXXX)? |
| 21:36.31 | IronMan2000 | this is what I have |
| 21:36.34 | IronMan2000 | exten => _1800NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:38.26 | IronMan2000 | do I just replace the "Macro" with "Dial" ?? |
| 21:38.57 | Un1x | yes 45% os slackware done :D |
| 21:44.24 | IronMan2000 | I am trying to set my asterisk up where I can make 800 and other toll free calls, but I cannot get it to work. Here is what I have in my extension.conf. Can someone tell me if this is set correctly? |
| 21:44.26 | IronMan2000 | exten => _1800NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:44.26 | IronMan2000 | exten => _1866NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:44.27 | IronMan2000 | exten => _1877NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:44.27 | IronMan2000 | exten => _1888NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:47.44 | Ryushin | In the extensions.conf file, how do I change it so when a user calls their own extension, that it goes right to voicemail? |
| 21:48.29 | IronMan2000 | goodluck gettng someone to answer you Ryushin. |
| 21:48.43 | hotroot | I'd try something linke GotoIf($["${EXTEN}" = "${CALLERID(number)}"]) |
| 21:49.48 | Ryushin | IronMan2000, well, it is a long weekend in the US, so I can understand it being quiet. |
| 21:50.09 | Ryushin | hotroot: Can I put the GotoIF anywhere in the file? |
| 21:50.45 | Ryushin | I have voice mail set for x011 right now. |
| 21:51.09 | Ryushin | So do I make it something like GotoIf($["${EXTEN}" = "${CALLERID(011)}"]) ? |
| 21:53.08 | hotroot | In my case I'd write somthing like this: |
| 21:53.08 | hotroot | exten => _1X,1,GotoIf($["${EXTEN}" = "${CALLERID(number)}"]?3) |
| 21:53.08 | hotroot | exten => _1X,2,Dial(SIP/${EXTEN}) |
| 21:53.09 | hotroot | exten >= _1X,3,Voicemail(${EXTEN}) |
| 21:53.42 | hotroot | if eg User 10 calls himself no Dial would be skipped |
| 21:54.18 | shodan | anyone knows how to tell a polycom 301 to use a distinctive ring ? probably something with sipaddheader(alert-info: ???????) |
| 21:54.19 | Ryushin | Cool. Thanks. |
| 21:56.02 | IronMan2000 | can anyone help me setup my extensions.conf to be able to do Toll Free calling? I need to dial 800 numbers, but it won't work with what I currently have in my extensions.conf. |
| 21:56.25 | IronMan2000 | exten => _1800NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 21:56.41 | IronMan2000 | what does this need to look like |
| 21:56.44 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.119) |
| 22:01.08 | *** part/#asterisk XGizzmo (n=gizzmo@unaffiliated/xgizzmo) |
| 22:01.10 | IronMan2000 | hotroot can you please help with this problem? |
| 22:02.04 | hotroot | What does the macro outgoingdial25 do? |
| 22:02.55 | IronMan2000 | I don't know. It's what that is in my config. how do I find out and I will look. |
| 22:03.38 | hotroot | is there a context [macro-outgoingdial25] in your extensions.conf? |
| 22:05.20 | IronMan2000 | yes |
| 22:05.27 | hotroot | can you post it? |
| 22:05.29 | IronMan2000 | [macro-outgoing] |
| 22:05.29 | IronMan2000 | ; |
| 22:05.29 | IronMan2000 | ; ${ARG1} - Outgoing device |
| 22:05.29 | IronMan2000 | ; ${ARG2} - Outgoing number |
| 22:05.29 | IronMan2000 | ; |
| 22:05.30 | IronMan2000 | exten => s,1,Ringing |
| 22:05.32 | IronMan2000 | ; DO NOT CHANGE THE NEXT 3 LINES MANUALLY !!!!!!! |
| 22:05.34 | IronMan2000 | exten => s,2,NoOp ;# CALLERIDNAMEANDNUMBER |
| 22:05.36 | IronMan2000 | exten => s,3,NoOp ;# CALLERIDNAMEONLY |
| 22:05.38 | IronMan2000 | exten => s,4,NoOp ;# CALLERIDNUMBERONLY |
| 22:05.40 | IronMan2000 | exten => s,5,Dial(${ARG1}/${ARG2},,${OUTDIALOPT}) |
| 22:05.42 | IronMan2000 | exten => s,6,Hangup |
| 22:05.44 | IronMan2000 | exten => s,106,Congestion |
| 22:05.51 | hotroot | ok |
| 22:06.18 | hotroot | your problem is, that ARG1 is in your call the Number and not the device |
| 22:06.20 | file | that's the 'outgoing' macro not outgoingdial25 |
| 22:06.44 | hotroot | ARG2 should be the number ans ARG1 should be something like ZAP/g1 |
| 22:07.04 | *** join/#asterisk T42X (n=T42X@88.118.148.158) |
| 22:07.23 | IronMan2000 | what number in ARG2? |
| 22:07.37 | hotroot | Phonenumber |
| 22:07.50 | hotroot | but as file mentioned this is not the macro you call... |
| 22:08.41 | IronMan2000 | I understand about putting the ZAP device on ARG1, but what number goes on ARG2? |
| 22:08.48 | IronMan2000 | sorry |
| 22:08.51 | hotroot | the Phone-number |
| 22:08.53 | micky | hello, does asterix need to run a new daemon or child for each incomming call or icecast stream ? i'm asking this because i'm trying to figure out why my server starts 100 processes .. have a look at http://pastebin.com/782950 (almost the whole ps-aux .. pastebin did not accept the whole thing) btw the server is a 3GHZ Xeon with 256 RAM. |
| 22:09.00 | IronMan2000 | the number im rtring to dial |
| 22:09.19 | file | micky: Asterisk is multithreaded, depending on your system threads may show up as processes |
| 22:09.41 | hotroot | IronMan2000: yes |
| 22:11.23 | micky | file thanks i'm using Debian linux. |
| 22:11.41 | file | which version? |
| 22:12.19 | micky | 3.1 |
| 22:13.01 | file | interesting |
| 22:13.02 | dflow | micky: you have 2.6 or 2.4 kernel? |
| 22:14.24 | micky | kernel 2.4.27-2-386 (default install) |
| 22:16.27 | dflow | micky: then everything is ok, on 2.4 kernel threads are show as processes , 2.6 kernel have NPTL thread and you will have one asterisk process |
| 22:16.47 | dflow | s/thread/threads/ |
| 22:17.31 | dflow | hehe |
| 22:21.48 | *** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130) |
| 22:22.17 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 22:25.55 | *** join/#asterisk RoyK (n=roy@ti211210a080-1432.bb.online.no) |
| 22:27.42 | *** join/#asterisk IronMan2001 (n=kent@65.124.236.24) |
| 22:27.56 | *** join/#asterisk Infamy (n=infamy@CPE005004804d2e-CM014090207037.cpe.net.cable.rogers.com) |
| 22:29.05 | Infamy | any cisco sip gurus present? |
| 22:31.44 | Un1x | heh never :P |
| 22:32.32 | rene- | ~NTPL |
| 22:32.33 | jbot | it has been said that ntpl is the Native Posix Thread Library |
| 22:32.43 | rene- | thanks jbot |
| 22:34.12 | rene- | has anyone used ztclock? |
| 22:46.05 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.119) |
| 22:51.40 | IronMan2001 | s |
| 23:00.37 | IronMan2001 | We are a very small VoIP service, and we offer VoIP service using asterisk to our high speed customers. We sell two seperate packages, (1) consist of free Long Distance, and the other we offer just local and statewide. My question is, How can I stop my standard package customers from being able to make LD calls. I need to be able to setup certain extensions from being able to make LD. Anyone know how this could be done? |
| 23:04.29 | *** join/#asterisk Rahail (i=rahail1@209.19.88.232) |
| 23:05.35 | snappy | We currently have an adtran 612 CSU/DSU, panasonic tda100&tva50. The panasonic gear appears to never been configured and the company I work for always appear to be too busy to get someone certified to configure it. I've decided to take matter into my own hands and configure an Asterisk PBX. We have one t1 trunk, what I'm wondering is what kind of hardware would be sufficient. The adtran connects to seems what is a small little box, that then connects |
| 23:08.19 | Strom_M | IronMan2001, you use contexts |
| 23:09.04 | Strom_M | snappy, you get a TE110P and connect it to the telco smartjack |
| 23:09.57 | shodan | anyone knows the alert-info string for setting distinctive rings ? |
| 23:11.16 | snappy | Strom_M: Does the x100p compare to the te110p at all? Sorry I'm unfamiliar with the hardware aspects when interfacing with the telephones and the T1 |
| 23:11.33 | Strom_M | the x100p is for a single analog line and costs $10-$20 |
| 23:11.51 | snappy | ah, it's like a modem card then? |
| 23:11.51 | Strom_M | the TE110P is for a single T1 span and costs $500-$600 |
| 23:12.09 | snappy | I see |
| 23:13.09 | snappy | So regardless I am looking at spending ~500 to get started. But I'm thinking this is negligible, as we don't even have manuals, software, etc. for our Panasonic equipment. Can't seem to get it off Panasonic's site either. Looks like they want certified professionals to come by and configure it. |
| 23:13.12 | IronMan2001 | Storm_M, can you please give me an example of contexts? Maybe some example confs |
| 23:14.12 | Strom_M | IronMan2001, and you're reselling service? |
| 23:14.24 | IronMan2001 | yes |
| 23:14.27 | Strom_M | i would seriously recommend that you read the book before you do anything else |
| 23:14.31 | Strom_M | ~book |
| 23:14.32 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:14.49 | Strom_M | either that or hire a consultant or something |
| 23:14.57 | *** join/#asterisk rickb|server (i=_11W_Voi@cpe-71-66-110-248.neo.res.rr.com) |
| 23:16.16 | rickb|server | Hello, I am having a problem with Asterisk. It wont authenticate SIP clients. I have everything setup within asterisk, i go to connect. in the Asterisk cli i get a Unauthorized attempt thing.. The password is correct, I am positive. Any ideas please!! :) |
| 23:16.18 | rene- | snappy: i agree: basically there is some background info that you must have before either attempting to do it by yourself or hire a consultant, |
| 23:17.07 | IronMan2001 | have you checked your asterisk logs to see if it's even attmepting to hit your asterisk server? |
| 23:17.26 | snappy | Right, yeah considering the consultant part. First though, unsure if our existing phones would be any useful. |
| 23:17.48 | rickb|server | IronMan2001, was that to me/ |
| 23:17.53 | IronMan2001 | yes |
| 23:17.54 | rickb|server | k |
| 23:17.55 | *** join/#asterisk shodan (n=shodan@ip186.96-113-216.pppoe1.joliette.intermonde.net) |
| 23:17.59 | rickb|server | It is attempting. |
| 23:18.18 | shodan | hmm power outtage :\ at least the phone system still works |
| 23:18.38 | Strom_M | snappy, if you cant administer the system that runs your panasonic phones, then there's no point in trying to keep them around |
| 23:18.39 | IronMan2001 | do you have your client set to hit port 5060 |
| 23:18.49 | shodan | are sip headers declared with "variable; value" or "variable: value" ? |
| 23:19.08 | Strom_M | snappy, you will wind up spending more in labor trying to get the panasonic system to work than you would if you just bought ip phones for everyone |
| 23:19.39 | rickb|server | I believe so. |
| 23:19.58 | IronMan2001 | what are you trying to connect to your SIP with? |
| 23:20.01 | snappy | Strom_M: Right thats what I'm thinking. It's not that I'm looking for a free or cheap system, it's just that Asterisk actually has documentation, and is quite interoperable, or at least thats what it seems. |
| 23:20.07 | rickb|server | EyeBeam |
| 23:20.09 | rickb|server | Softphone |
| 23:21.17 | IronMan2001 | check the configuration to see if you have it set to hit port 5060 for your SIP server, and outbound Proxy |
| 23:21.38 | IronMan2001 | it may not know which port to hit when it tries to negociate a connection. |
| 23:21.42 | rickb|server | What should outbnound proxy be set as? |
| 23:22.10 | IronMan2001 | the IP number of your SIP and the port. Example. 192.168.???.???:5060 |
| 23:22.18 | hotroot | is someone here who knows how to make asterisk send a stuttered dialtone to ISDN-Phones if there are new voicemails? a mailbox=XXX-Line in zapata.conf does exists. |
| 23:23.04 | rickb|server | ok.. |
| 23:23.20 | rickb|server | I did that. |
| 23:23.27 | rickb|server | the CLI comes back as this |
| 23:23.28 | rickb|server | Transmitting (no NAT) to 192.168.1.25:6251: |
| 23:23.28 | rickb|server | SIP/2.0 401 Unauthorized |
| 23:24.05 | rickb|server | But the problem with that is... The server this is on a dedicated server. |
| 23:24.13 | rickb|server | so its not on the same ip network. |
| 23:24.54 | snappy | Alright, well thanks for the info Strom_M, rene. |
| 23:26.01 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:26.37 | snappy | Oh lastly, are there any voip boxes that can be configured to convert analogue line phones to do voip? |
| 23:26.44 | Strom_M | yes |
| 23:27.03 | Strom_M | you can get either analog terminal adapters or use something like a digium tdm400p / tdm2400p |
| 23:27.19 | file | mmm tdm2400p |
| 23:27.20 | TripleFFFF | questin in realtime.. the permit=1.2.3.4 how can we add 2 ? i mean we sepearte with ; ? or & ? |
| 23:27.22 | *** join/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 23:27.54 | Strom_M | do any of you know if sixtel has a limit on concurrent outbound calls? |
| 23:28.34 | snappy | Hmm, that t2400p |
| 23:28.48 | snappy | It appears to use the same connector that our Panasonic tda100 has |
| 23:28.53 | rickb|server | IronMan2001, any ideas |
| 23:29.13 | Strom_M | snappy, it's called an amphenol connector and has been industry standard since at least the 1960s |
| 23:29.26 | snappy | ah, thanks |
| 23:29.54 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 23:32.17 | snappy | So to connect existing phones, I would want something like a tda2400p. Then to connect to our T1, use a te110p. Assuming I can connect directly to our telco from the t1 trun line they provide, can we avoid using the Adtran CSU/DSU? If thats the case though, would I be able to configure the te110p as a network interface? |
| 23:32.27 | Strom_M | yes |
| 23:32.41 | Strom_M | asterisk interfaces directly with the t1 via the te110p |
| 23:33.38 | snappy | Excellent, because we have an ethernet port on our adtran which gives us our data. So I assume I can use bridging in linux, and have a 10/100 NIC go out to the switch that the current network cable is connected to. |
| 23:33.44 | snappy | Sweet, again, thanks. |
| 23:34.14 | shodan | wow , polycom phones are harder to setup than asterisk itself ! ;( |
| 23:34.14 | Strom_M | yeah, you can do HDLC with zaptel |
| 23:34.19 | Strom_M | shodan, duh |
| 23:35.15 | Infamy | first time asterisk user i got a phone connect to asterisk |
| 23:35.20 | Infamy | now how can i do a test.. |
| 23:35.30 | Infamy | is there a dial pack number or something by default? |
| 23:39.12 | *** part/#asterisk hotroot (n=michael@pD9E95CCB.dip.t-dialin.net) |
| 23:41.59 | shodan | this is probably the most perverse use of xml I've ever seen ! ;) |
| 23:43.46 | AndyCap | shodan: what is? |
| 23:44.21 | shodan | polycom config files |
| 23:44.31 | Strom_M | shodan, yeah, tell me about it |
| 23:44.39 | Strom_M | it made me want to kill someone the first time i saw it |
| 23:44.45 | Rahail | hey do you guiess know how to disbale registration option on iax2 |
| 23:45.57 | shodan | I mean I have a dual head monitor , both 1600x1200 , I opened the xml file in notepad and stretched the notepad window accross both screen and the xml "values" are so long that they still wordwrap !!! |
| 23:46.23 | shodan | I'm thinking , "do I care this much about distinctive rings ?" |
| 23:53.39 | *** join/#asterisk aRJAy (n=aRJAy@218-214-130-112.people.net.au) |
| 23:53.51 | aRJAy | ehlo |
| 23:53.52 | aRJAy | :) |
| 23:54.14 | TripleFFFF | shodan lol |
| 23:54.18 | TripleFFFF | get a 3rd screen ? |
| 23:55.54 | aRJAy | I have a general VOIP question I'm hoping someone can help with... As with AIM, ICQ, and other instant messangers.... Is there a way to provide a link on a site that initiates a VOIP session? The web user needs to have no particular software on his/her puter.. |
| 23:56.18 | Strom_M | aRJAy, look into something called "click to call" |
| 23:57.40 | aRJAy | oh? |
| 23:58.18 | aRJAy | hmm |
