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01:40.58 | Beirdo | !wx ksea |
01:41.09 | Beirdo | argh, wrong channel :) |
01:44.05 | Kobaz | do de do |
01:44.16 | Kobaz | what's the deadline to register for astricon as a speaker? |
01:44.49 | Maliuta | I would hope their PA has enough speakers ;P |
01:44.58 | Kobaz | heh |
01:45.05 | Kobaz | i see there's still open slots in the speaker schedule |
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01:51.58 | Kobaz | awwww |
01:52.02 | Kobaz | The cutoff date for entering data here is June 30, 2010. |
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02:54.12 | nightwalk | [208084.003233] wctdm24xxp 0000:01:07.0: Missed interrupt. Increasing latency to 22 ms in order to compensate |
02:55.04 | nightwalk | Can anyone tell me if that's a symptom of a symptom of a bad motherboard or a bad card? Or maybe the a card/motherboard incompatibility of some sort? |
02:56.44 | nightwalk | Total IRQ misses as reported by /proc/dahdi/1 is 202, so this seems to be happening fairly frequently. I'm guessing it's the cause of the random disconnects since supposedly 30ms is the max before it cuts off. |
02:58.07 | nightwalk | ...and as the line above demonstrates, it's already over 20. |
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03:02.17 | nightwalk | My money is on it being a motherboard problem, since this motherboard doesn't play all that nicely with linux, either. Just wondered if it was possible to establish that as fact given the above error. |
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03:06.21 | ChannelZ | hard to say |
03:07.09 | ChannelZ | Tried moving the card into a different slot? It could be sharing resources with something else |
03:08.07 | p3nguin | How can a main board care what operating system is employed? |
03:08.39 | nightwalk | Oops, guess they changed the max. Just got this: |
03:08.42 | nightwalk | wctdm24xxp 0000:01:07.0: ERROR: Unable to service card within 25 ms and unable to further increase latency |
03:08.49 | ChannelZ | it might have a wonky chipset that doesn't play well |
03:09.13 | nightwalk | ChannelZ: The board only has one usable PCI slot, unfortunately |
03:09.26 | ChannelZ | hmph. |
03:10.04 | nightwalk | p3nguin: It's not about the board caring what the OS is, it's about the mainboard not being standards-compliant |
03:10.20 | drmessano | LOL |
03:10.37 | nightwalk | hates non-standards-compliant vendors...grrrr |
03:11.15 | drmessano | There's no such thing as "standards" when it comes to hardware |
03:11.52 | nightwalk | Oh, really? I must've dreamed acpi and even x86 then :P |
03:12.26 | drmessano | Really, do you know how many vendor specific implementations of ACPI are in the kernel? |
03:12.36 | drmessano | ACPI != ACPI |
03:13.03 | nightwalk | Yes, but that just means there are lots of vendors not following the standard |
03:13.11 | drmessano | REALLY NOW?!?! |
03:13.37 | p3nguin | like SIP, perhaps? |
03:13.42 | nightwalk | Extensions are one thing, but core functionality should NOT deviate from the standards unless there's an extremely good reason. |
03:14.20 | drmessano | "should NOT" is wonderful and almost never applies in the real world |
03:15.26 | drmessano | Which is why every "standard" is succeeded by yet another "standard" that's determined to not go wrong where the previous went wrong |
03:15.40 | drmessano | Which is, most of the time, implementation |
03:17.11 | drmessano | So your motherboard not working because it's not "standards compliant" is highly unlikely.. Chances are if it DID stick to some standard, it wouldn't work, because a non-patched or non-amended implementation of the "standard" probably hasn't been tested due to lack of real world examples |
03:18.12 | drmessano | p3nguin, Could you imagine if someone implemented a truly standards compliant SIP stack? NOTHING would work with it.. |
03:20.36 | nightwalk | And yet, there are tons of boards made by manufacturers that carefully draw inside the lines whose boards I have no problem with. Just boards from companies that think they know best and do it their own way (Apple), or companies that decide being able to list more "features" is better than bothering to conform properly to the standards |
03:21.51 | nightwalk | Actually, it's not even the fact that they don't conform to standards so much as it's that they don't conform to the standards *and* they fail to make any sort of meaningful contribution to the oss driver base for it. |
03:22.10 | p3nguin | If everyone made their hardware exactly the same, where would the differentiation of the brands be? |
03:22.17 | nightwalk | price |
03:22.24 | *** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com) |
03:23.22 | p3nguin | If everyone made their hardware exactly the same, the cost would be the same and the profit margin between cost and retail price _should_ be the same. In the case where one brand has a higher retail price, you could safely by the other brand (since the hardware is exactly the same). |
03:23.28 | DogBoy | well it would be like with slogans printed on them |
03:23.33 | DogBoy | like with t-shirts |
03:23.44 | p3nguin | buy, that is. |
03:25.03 | nightwalk | Ah, but some would be more efficient, and thus be able to make the boards more cheaply. |
03:29.07 | nightwalk | Anyway, none of you have any idea whether or not the latency thing can be traced back to a faulty/non-standards-compliant motherboard, huh? |
03:30.57 | nightwalk | This problem seems to come up a lot, but so far I haven't found any pages that explain what the root problem might be. |
03:59.14 | ChannelZ | put it in another computer and see what happens |
03:59.25 | ChannelZ | then you'll have your answer |
04:12.05 | drmessano | Which "Standard" would this motherboard be in violation of to be causing these errors? |
04:13.05 | drmessano | Lets get down to some science here, rather than a Google search for the glossary to the A+ Study Guide |
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04:14.06 | drmessano | http://zhulizhong.blogspot.com/2010/04/troubleshooting-of-bri-cards.html |
04:14.15 | drmessano | Check out the "IRQ miss" question |
04:14.29 | coppice | A common problem is BIOSes chasing benchmark results |
04:15.28 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
04:19.50 | coppice | This page says far more about PCI issues http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
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04:53.02 | gamedna | evening all. |
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05:12.04 | Alton35 | Ok ChannelZ or others, still trying to use the local channel, |
05:12.15 | Alton35 | my problem is that it places the call twice, and neither works, not helpful, |
05:12.26 | Alton35 | however, I have learned how to use pastebin :-) Here: http://pastebin.com/ZEMvtjM1 |
05:48.04 | gamedna | anyone know if its possible to run multiple instances of asterisk on the same machine? |
05:50.06 | joobie | gamedna, why do u want tod o that? |
05:50.17 | gamedna | development, testing, etc... |
05:50.29 | gamedna | experimentation |
05:50.42 | gamedna | dont want the overhead of a VM |
05:51.20 | joobie | i duno man |
05:51.25 | joobie | it's just weird what you're suggesting tho |
05:51.41 | joobie | running a dev and production build onthe same machine |
05:51.45 | gamedna | no no |
05:51.46 | joobie | usually u'd seperate these out |
05:52.08 | gamedna | this is not production |
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05:52.21 | JerJer | xen / openvz |
05:52.33 | gamedna | JerJer: trying to stay away from VM |
05:52.39 | JerJer | good luck then |
05:52.53 | joobie | what's wrong wiht 1 dev instance then? |
05:53.11 | JerJer | you might be able to tell asterisk to use all different config files / sockets / etc |
05:53.13 | p3nguin | What good would it do to have a second asterisk running? |
05:53.40 | Alton35 | virtualbox is good |
05:53.45 | gamedna | well, i dont have to run VMs for testing interaction between two asterisk boxes |
05:54.15 | p3nguin | alton35: The problem with your call file is that you are calling the phone number twice. |
05:54.17 | gamedna | packet sniffing on local interface, instead of going on the ethernet |
05:54.32 | gamedna | um... many other advantages |
05:54.37 | p3nguin | alton35: What are you trying to accomplish? |
05:54.54 | Alton35 | using the local channel, to be able to make the call from within my agi |
05:55.05 | Alton35 | I'm still stuck. |
05:55.25 | p3nguin | alton35: Your call file says to call 9565815577, and after it answers, call 9565815577, then bridge them. |
05:56.09 | joobie | gamedna> packet sniffing on local interface, instead of going on the ethernet |
05:56.15 | Alton35 | let me see |
05:56.17 | joobie | wtf.. |
05:56.17 | p3nguin | alton35: Using a local channel is for when you need to call another location in the dialplan. |
05:56.43 | Alton35 | well, it was suggested last night and I kinda though it was a good idea, since I thought you had to have a channel up to invoke an AGI. |
05:56.46 | gamedna | joobie: yea why not? say im on my laptop not connected to a net |
05:57.09 | p3nguin | alton35: Do you simply want to call a phone number and after it answers play the sound file? |
05:57.22 | Alton35 | no, there are database things to do in the agi |
05:57.54 | p3nguin | I can't see any reason to call the same number twice. Especially if you intend to bridge the two calls together. |
05:58.13 | joobie | gamedna, dood you are a tripper |
05:58.15 | Alton35 | yeah, I just don't see how I'm doing it, been trying to get rid of that. |
05:58.16 | joobie | .. use VM's |
05:58.42 | Alton35 | gamedna, yeah, I believe asterisk was not written to be run twice on the same machine, might not be a good idea. |
05:58.57 | p3nguin | alton35: You're calling the number via local channel, then calling it via context/extension. |
05:59.05 | p3nguin | Pick one. |
06:00.00 | Alton35 | I didn't think the local channel called it. I would prefer that it didn't, but if I don't call from there, then how to get my AGI invoked? |
06:00.02 | gamedna | joobie: i do use VM's... but what about running 4+ instances on the same laptop... not really "efficient" |
06:00.28 | joobie | gamedna, why the heck do you want to run 4 instances of asterisk on the same box? |
06:00.52 | Alton35 | maybe a philosophical aversion to VMs :-) |
06:00.56 | gamedna | joobie: application development |
06:01.10 | joobie | gamedna, that's great - but why do you need 4 instances of asterisk on the same box? |
06:01.11 | gamedna | Alton35: not at all |
06:01.15 | gamedna | Alton35: love VM's |
06:01.26 | gamedna | just finding them rather inconvenient for what i want to do |
06:01.48 | JerJer | I run 3 guests on my macbook - usually 2 linux and one winblows |
06:02.28 | Alton35 | p3nguin: I have never found out how to dial out from within an AGI under asterisk, which doesn't seem to be such an unreasonable thing to do. |
06:02.30 | p3nguin | alton35: If you have to call the number via local channel to execute that part of the dialplan, don't include Context: and Extension: as well as Channel: in the call file. |
06:02.38 | gamedna | joobie: working on something that deals w/ asterisk boxes talking to each other |
06:02.39 | gamedna | cant say more than that ATM |
06:02.42 | Alton35 | ok, let me see here |
06:03.15 | gamedna | JerJer: i usually have 2 running... Windows and Linux. |
06:03.30 | joobie | gamedna, then go somewhere else for help if you dont want to explain why |
06:03.38 | JerJer | gamedna: if you look into the so called HA solutions they all use some form of virtualization technology |
06:03.54 | joobie | gamedna, what you're suggesting to do sounds stupid.. and you cant explain why you need the 4 instances, so bugger off |
06:04.25 | Alton35 | I have seen too much server sprawl with virtualization, but if it's not too much trouble to get going, it can be useful here and there. |
06:04.48 | gamedna | joobie: why such hostility? just wanted to know if anyone knew how to do it? |
06:04.58 | p3nguin | alton35: Or if you find that the call file doesn't work without those other fields being present, at least use a context/extension that isn't the same as the number you've already called by a local channel. It's nonsense to call the same number twice at the same time. |
06:04.58 | gamedna | joobie: so bugger off yourself. |
06:04.59 | xheliox | "cant say more ATM" -- translation "I couldn't find a clue with two hands and a flash light" |
06:05.12 | gamedna | sheesh |
06:05.15 | joobie | gamedna, sorry we don't support retarded configurations |
06:05.56 | gamedna | joobie: too bad you are so shortsighted. |
06:06.05 | Alton35 | Hah, gamedna, like I say, asterisk not particularly written to be run more than once on the same machine, so it's not good practice to try doing it. |
06:06.24 | xheliox | distracts gamedna with a shiney metal object |
06:06.32 | gamedna | JerJer: like i said before, this is just for convenience in development. Not for production... |
06:06.55 | p3nguin | I'm still waiting to hear what good it would do to have a second asterisk running. |
06:07.05 | gamedna | Alton35: have you tried it? |
06:07.07 | p3nguin | There's nothing it can do. |
06:07.11 | joobie | he cant tell you.. it's a secret |
06:07.15 | Alton35 | No, because I know it's not written for that. |
06:07.17 | p3nguin | The ports will already be in use by the first one. |
06:07.27 | ChannelZ | bind it to another IP |
06:07.29 | Alton35 | Maybe try one asterisk with 4 clearly-separated contexts or something. |
06:07.39 | gamedna | looking at the source and the conf files, looks like it can be done. |
06:07.40 | *** join/#asterisk thansen (~thansen@S0106001c1092cd20.cg.shawcable.net) |
06:07.47 | JerJer | duno then |
06:07.53 | gamedna | just needs ssome major configuration |
06:07.56 | xheliox | You could probably setup 3 chroot environments and bind it bind to a different IP for each instance, I'm clueless as to why you'd do that. |
06:08.01 | Alton35 | still not a good idea, you have a lot of people here telling the same thing |
06:08.22 | gamedna | Alton35: yea, but nobody saying, i tried it at it does not work. |
06:08.36 | joobie | because no one is retarded enough to go down that path |
06:08.36 | Alton35 | that's because we know better than to even try it :-) |
06:08.38 | gamedna | xheliox: good idea. |
06:08.45 | xheliox | gamedna: Because shockingly, no one is dumb enough to try. |
06:08.46 | gamedna | hehe |
06:09.03 | joobie | it's like loading multiple instances of apache to run 2 sites.. when you can just use virtual hosts |
06:09.06 | gamedna | gamedna: hahhaha... only dumb until it works. |
06:09.06 | xheliox | Which is saying a lot if you spend a couple of minutes watching this channel. |
06:09.22 | Alton35 | part of learning about computers in general is knowing which path to take. don't take the cursed and dark path. |
06:09.22 | xheliox | Um, nope.. even if it works, it's still dumb. |
06:09.47 | joobie | .. or the path that is prone to failure and carries a large management overhead |
06:09.49 | gamedna | joobie: people have done that before. |
06:10.01 | joobie | gamedna, like i said, we dont support retarded configurations in here |
06:10.11 | gamedna | xheliox: yea but if it sames me time, then its not |
06:10.19 | ChannelZ | that's what #freepbx is for |
06:10.22 | b14ck | Hey guys. I'm currently using dahdi ONLY for dahdi_dummy support. Is there a way I can completely remove the chan_dahdi.conf file without having Asterisk complain? Right now it's just an empty file, seems kinda pointless to have it. |
06:10.26 | joobie | heh |
06:10.41 | Alton35 | they're saying that if you need to do it, then you're probably something something else wrong in the first place. and they're offering to help you with the whole thing, but you won't say anything more. |
06:10.57 | gamedna | Alton35: i understand taht... |
06:11.12 | gamedna | Alton35: im not trying to be a prick, i just cant say anymore.... |
06:11.15 | xheliox | gamedna: In the time you've had this discussion, you could have had 3 VMs running. And even then I'm not sure what the point is. |
06:11.21 | Alton35 | haha |
06:11.22 | Alton35 | true |
06:11.35 | gamedna | xheliox: i cant run 3 vms.. im already resource bound on this machine |
06:11.43 | Alton35 | well, what they're also saying is, if there's so much money at stake, then go somewhere else and pay someone. |
06:12.02 | Alton35 | oh come on, VMs will run with 64mb of ram each. |
06:12.12 | Alton35 | install slackware or debian 5 or something. |
06:12.17 | gamedna | trying to work around a CPU / memory resource limitation on my computer, so that i can test mutlpe asterisk system interactions. |
06:12.30 | p3nguin | b14ck: Is that file causing some problems to exist? Why remove it if there is no problem? |
06:12.49 | b14ck | p3nguin, the only thing that bothers me about removing it is: asterisk complains in the full logfile when restarting. |
06:12.58 | b14ck | It has no other effects. |
06:13.01 | p3nguin | leave it alone, then. |
06:13.10 | b14ck | Am I missing something? |
06:13.19 | b14ck | Seems like it wouldn't be complaining unless I did something bad =p |
06:13.20 | p3nguin | An empty file surely doesn't take up much disk space. |
06:13.28 | b14ck | Yah, it's the thought though. |
06:13.34 | b14ck | It's a matter of pride, or something. |
06:14.04 | xheliox | Ahh.. I love the smell of stupid in the morning. |
06:14.14 | b14ck | ... |
06:14.24 | b14ck | I just want to remove the file and not have asterisk complain. |
06:14.43 | b14ck | Incase I'm missing something, and it is actually causing a problem that I haven't yet caught. |
06:14.45 | b14ck | How is that stupid. |
06:15.07 | p3nguin | More nonsense. |
06:15.08 | xheliox | Hey guys.. I have this file on my Linux box /boot/vmlinuz-2.6.23 and when it boots I get an error, can I remove it???? |
06:15.49 | p3nguin | It's not taking up space to have it, asterisk complains if it's gone... the answer should be clear: LEAVE IT. |
06:18.23 | b14ck | Sigh. |
06:18.30 | b14ck | The file has been empty up until this point. |
06:18.51 | b14ck | So, I'm assuming that the error I'm getting is pointing to some functionality I've broken. |
06:18.58 | b14ck | It doesn't make me happy to just leave an empty file there. |
06:19.05 | b14ck | I want to know what I'm breaking. |
06:19.21 | ChannelZ | Just get on with your life |
06:19.30 | b14ck | Might as well just read the source. |
06:19.32 | b14ck | brb |
06:19.37 | xheliox | lol |
06:20.00 | b14ck | Don't know why everyone has a problem answering a question. Might as well just say "I don't know". |
06:20.36 | ChannelZ | you're loading a module which has a config file. You want to delete the config file. It complains. You are freaked out that it complains. What answer is it you want? |
06:21.09 | b14ck | I'd like to know if by providing either an empty config file, or no config file at all, if it impacts the ability of dahdi_dummy to provide timing to my environment. |
06:21.24 | ChannelZ | Here's a thought, try it. |
06:21.38 | b14ck | I have (as I mentioned). |
06:21.46 | b14ck | But I'm not sure whether there is some other breakage happening behind the scenes. |
06:21.48 | ChannelZ | It takes 2 seconds to make a dialplan with MeetMe() or something. Or presumably you already have something setup using it or else you wouldn't be needing dahdi_dummy |
06:21.55 | b14ck | Already done that. |
06:22.09 | xheliox | "I don't want to live on this planet anymore.." -Professor Farnsworth |
06:22.14 | ChannelZ | lol |
06:22.17 | ChannelZ | blasts off |
06:22.26 | xheliox | Take me with you! |
06:32.50 | *** join/#asterisk boobsbr (~george@201008039169.user.veloxzone.com.br) |
06:32.54 | boobsbr | howdy |
06:33.27 | boobsbr | Is there a way to test if the softphone I'm using is sending the DTMF to Asterisk? |
06:34.37 | xheliox | What DTMF mode are you using? |
06:35.45 | *** join/#asterisk JerJer_ (~PhatJ@asterisk/original-h323-guy/JerJer) |
06:36.05 | xheliox | If you're using rfc2833, you should see the DTMF packets in rtp debug. If you're using inband, I'm not entirely sure. |
06:38.09 | boobsbr | xheliox, inband with ulaw |
06:38.42 | boobsbr | xheliox, I'll give rfc2833 a try |
06:39.17 | xheliox | afaik, that's the more reliable way to go. |
06:44.28 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
06:45.03 | xheliox | ChannelZ: wb - just had to reboot into Windows 98 and steal my idea, eh? :) |
06:45.11 | ChannelZ | yes |
06:45.16 | xheliox | Bastid. |
06:45.22 | xheliox | I want royalities. |
06:48.21 | ChannelZ | I can offer you a cookie. |
06:48.36 | xheliox | What kind of cookie? |
06:48.51 | boobsbr | xheliox, I managed a little test with ReadDigits() and SayDigits() and the DTMF is working. Thanks. |
06:49.02 | ChannelZ | Chocolate chip. |
06:49.06 | xheliox | Ok - that seems fair. |
06:49.13 | xheliox | boobsbr: Excellent. |
06:50.40 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
06:57.58 | *** join/#asterisk af_ (~getsmart@78.134.22.178) |
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07:03.21 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
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07:29.22 | boobsbr | xheliox: I can't get Asterisk to say the POUND digit with my little test routine, STAR and the numbers plays OK though. |
07:30.02 | xheliox | Ah.. the classic # bug.. |
07:30.12 | xheliox | lol - that's quite strange. |
07:30.26 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
07:31.26 | xheliox | boobsbr: I haven't a clue why that might be, but if you're using rfc2833 now.. you should see the dtmf packets in rtp debug, verify it's actually sending it. |
07:31.42 | xheliox | Other than that, I've had way too many drinks to offer much more assistance for this evening. :) |
07:31.59 | boobsbr | yes, the packets are going through with rfc2833, a bunch of them |
07:32.21 | xheliox | Which soft phone are you using? |
07:32.41 | boobsbr | Twinkle, the only one I could get working on Ubuntu |
07:34.30 | boobsbr | and X-Lite on a Windows VM, but the sound is horrible |
07:34.46 | ChannelZ | I think it defaults to gsm being first on the list |
07:34.47 | xheliox | you're experiencing the same thing with both clients? |
07:35.24 | boobsbr | no, X-Lite doesn't seem to send the DTMF packets |
07:35.47 | boobsbr | and I changed the codecs on Twinkle to alaw and ulaw only |
07:36.09 | xheliox | if you're using rfc2833, the codec matters none. |
07:37.07 | Kyosh | boobsbr: are you running asterisk on a vm or x-lite or both? |
07:37.48 | boobsbr | asterisk on a debian vm and xlite on a windows vm, both on virtualbox |
07:38.37 | Kyosh | hmm |
07:38.46 | Kyosh | im curious why x-lite on a vm? |
07:38.52 | Kyosh | or "in" a vm |
07:39.43 | Kyosh | rfc2833 works fine for x-lite, thats for sure. |
07:39.53 | boobsbr | because it failed to work in ubuntu, even though I installed the damned 32 bit libstdc++.so.5 library it required. i think it's an alsa/oss/pusleaudio problem |
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07:40.40 | Kyosh | well you have to narrow down if the problem is on the host or the client (for ast of course, not virtualbox) |
07:40.53 | Kyosh | then determine if virtualbox is playing a role |
07:41.14 | Kyosh | anyone will telly ou that you need to narrow down the problem and once you do, then you can begin troubleshooting it |
07:41.29 | gamedna | i second that.. |
07:41.36 | gamedna | start in one direction and work your way to the other side |
07:41.37 | Kyosh | currently your problem has too broad implicating factors for problems |
07:41.43 | Kyosh | bingo |
07:41.50 | gamedna | and dont make any assumptions |
07:42.02 | Kyosh | gotta rule out, not assume |
07:42.04 | Kyosh | sup game |
07:42.30 | gamedna | hey kyosh.. |
07:42.54 | boobsbr | well, I'll try removing virtualbox outta the equation once I get a new machine next week. |
07:42.55 | Kyosh | nada, working on some iax2 integration |
07:43.19 | Kyosh | so far i got some nice fax over ip working with ulaw |
07:43.49 | boobsbr | so, can I use the * key to make blindxfers? |
07:43.51 | Kyosh | my partner is workign on the hylafax tests |
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07:44.11 | Kyosh | boobs: if thats how your features.conf is configured |
07:45.06 | Kyosh | msn mobile on my bberry bold killed my battery from 30% to 0 in 30 mins |
07:45.13 | Kyosh | so stupid |
07:45.16 | boobsbr | so, I only need to change the blindxfer option, to, let's say, *0 ? |
07:46.03 | gamedna | in band fax or using T.38? |
07:46.25 | Kyosh | inband |
07:46.34 | Kyosh | sip+ulaw |
07:46.44 | Kyosh | speed sucks, between 4800 and 9600 |
07:47.00 | Kyosh | however we;ve tested hylafax and it uses iax at 14400 |
07:47.11 | Kyosh | so im gonna make an integration |
07:47.41 | Kyosh | sadly hylaxfax has no database support except for log files |
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07:47.47 | Kyosh | gonna have to change that |
07:47.57 | shido6 | where do you get your linksys srtp private key from? |
07:48.09 | shido6 | voxilla's generator is broken |
07:48.32 | Kyosh | my who? |
07:50.31 | gamedna | Kyosh: nice work. |
07:52.27 | Kyosh | what work/? |
07:52.33 | Kyosh | what did i break now? |
07:52.47 | gamedna | no, the fax stuff |
07:53.12 | gamedna | the hylafax iax @ 14,400 |
07:54.20 | Kyosh | oh i dunno. it just happened :-p |
07:54.23 | WIMPy | Fax at 64000 might be more interesting nowadays. |
07:54.43 | Kyosh | oh man id be happy with that |
07:55.16 | WIMPy | Thing is that it should be a lot easier than the old stuff. |
07:55.29 | Kyosh | should be |
07:58.19 | WIMPy | Trouble is that you coldn't even have it on the same number with Asterisk. |
07:58.55 | WIMPy | Damn. The more I get into it again the more I think that there's still a very long way to go :-( |
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08:10.06 | boobsbr | alright, I got Asterisk to understand that the pound sign is used for transfering |
08:10.49 | boobsbr | but now it says the number i'm dialing is not a valid extension |
08:10.52 | boobsbr | =( |
08:12.11 | boobsbr | is there a way to find out what number Asterisk thought I was dialing? |
08:16.56 | boobsbr | wow, it worked |
08:17.08 | boobsbr | i don't know what i did, but it worked |
08:18.11 | boobsbr | damn, this stuff is pretty cool |
08:18.24 | Kyosh | i gotta make a new ivr for a customer |
08:18.28 | Kyosh | i hate doing that |
08:20.11 | WIMPy | Does anyone have a good idea how to patch BC based routing into Asterisk? Maybe by defining different contexts per BC in the channel config? |
08:20.20 | WIMPy | Any comments? |
08:20.39 | WIMPy | (probably the wrong day to ask) |
08:20.44 | boobsbr | sorry, what is a BC? |
08:21.06 | WIMPy | Bearer Capability |
08:21.18 | WIMPy | i.e. the type of service. |
08:21.37 | boobsbr | oh. |
08:21.55 | boobsbr | I'm pretty new to this, so, sorry again. |
08:22.12 | boobsbr | Kyosh: do you have to record the messages too? |
08:24.28 | Kyosh | sadly yea. my voice is too raspy and nasaly |
08:24.47 | Kyosh | maybe i can get a voice emulator |
08:31.39 | boobsbr | Is there a way to test call quality? |
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08:32.24 | boobsbr | Like a standard sound file to be played at one end and recorded at the other, then compared and analyzed. |
08:33.06 | Kyosh | umm |
08:33.11 | Kyosh | standard? |
08:33.41 | Kyosh | hmm |
08:33.48 | Kyosh | for a single call? |
08:33.52 | boobsbr | yeah |
08:33.56 | Kyosh | you can perform an echo test |
08:34.34 | boobsbr | but how can the input and output be analyzed? |
08:35.06 | boobsbr | i tried the echo test, and it works perfectly, the calls also work |
08:35.41 | boobsbr | I need to write on a paper that the call quality is analog or superior to a POTS call |
08:36.02 | boobsbr | and my professor said I need a way to quantify this quality |
08:36.19 | boobsbr | just saying it's good is not proof enough |
08:36.39 | boobsbr | got the idea? |
08:36.54 | Kyosh | yea dude |
08:36.56 | Kyosh | umm |
08:37.04 | Kyosh | hmm |
08:37.18 | Kyosh | the inbound call quality or the outbound call quality? |
08:37.47 | boobsbr | now you got me |
08:37.50 | Kyosh | ok |
08:37.55 | Kyosh | have asterisk record the test call |
08:38.06 | boobsbr | that's an interesting idea |
08:38.10 | Kyosh | then grab the file from the /sounds dir and run it thru a spectrum analyzer |
08:38.31 | Kyosh | also record the same phrase into a wav file and run it thru the analyzer |
08:38.58 | Kyosh | make sure the wav file is 8bit, same bitrate |
08:39.01 | boobsbr | I thought about that |
08:39.13 | Kyosh | prolly the only way i can think of |
08:39.18 | boobsbr | but I didn't know about the record function |
08:39.18 | Kyosh | i aint that smart |
08:39.24 | Kyosh | ya |
08:40.35 | Kyosh | in sip.conf, under the extension you are calling, set record_out=Always and record_in=Always |
08:40.54 | Kyosh | or fromt he ext you are using |
08:40.55 | Kyosh | either way |
08:41.04 | boobsbr | got it |
08:41.09 | boobsbr | thanks for the help |
08:41.14 | Kyosh | hope it does help |
08:41.23 | Kyosh | using asterisk for class project? |
08:41.29 | Kyosh | what kinda class is that? |
08:41.46 | boobsbr | it's a graduation project |
08:41.54 | Kyosh | what kinda class is that? |
08:42.05 | boobsbr | electrical engineering , telecom |
08:42.19 | Kyosh | no dude |
08:42.21 | Kyosh | no way |
08:42.23 | Kyosh | no fukin way |
08:42.28 | boobsbr | what? |
08:42.35 | Kyosh | graduation projects for my bsee was way different |
08:42.57 | Kyosh | waaaay diff |
08:42.57 | boobsbr | harder? |
08:43.06 | Kyosh | we didnt get to play with shit like that |
08:43.17 | Kyosh | then again, this is going back a bit |
08:43.22 | boobsbr | when? |
08:43.40 | Kyosh | oh umm |
08:43.48 | Kyosh | you may have been out of diapers :-p |
08:44.26 | boobsbr | 1983? |
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08:47.26 | boobsbr | here goes another question, has anyone used pjsua? http://www.pjsip.org/pjsua.htm |
08:47.59 | Kyosh | nah dude i was a freshman in hs then |
08:49.32 | Kyosh | what is that link? |
08:49.48 | boobsbr | a command line implementation of a sip client |
08:50.46 | Kyosh | oh |
08:50.52 | Kyosh | for the purpose of? |
08:51.05 | Kyosh | its command line |
08:51.36 | Kyosh | last smoke before bed |
08:52.16 | boobsbr | placing and receiving calls from a remote location, using a command line shell. saves bandwidth |
08:52.46 | boobsbr | for testing purposes, it's not practical for regular use. |
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09:01.55 | boobsbr | well, gotta go to bed too, 6am already. thanks everyone for the help! |
09:02.06 | boobsbr | \q |
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09:30.23 | gamedna | have multiple asterisk instances running on the same machine w/o vm... no problems |
09:30.31 | gamedna | dont need SUID |
09:30.36 | gamedna | dont need chroot |
09:31.06 | gamedna | and no virtual macines.. |
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12:10.58 | rethus | i have solvved all problems to compile app_cconference for asterisk 1.62. But i have still little problem to install it |
12:11.42 | tzafrir_laptop | rethus, just drop the module into the modules directory, right? |
12:11.48 | rethus | if i tyüe make install, i got "install -m 755 app_conference.so /usr/lib/asterisk/modules", but ffor make i have added the Parm: "ASTERISK_INCLUDE_DIR=/usr/lib64/asterisk/" |
12:12.18 | tzafrir_laptop | rethus, do you have a .so file? |
12:12.32 | tzafrir_laptop | If so: just copy it to /usr/lib/asterisk/modules/ |
12:12.43 | tzafrir_laptop | A bit simpler than fighting makefiles |
12:12.43 | rethus | tzafrir_laptop: jep |
12:13.00 | rethus | on my system its lib64/asterisk... |
12:13.05 | tzafrir_laptop | Did it need patching? (if so, please submit fixes) |
12:13.05 | rethus | thats right. |
12:13.26 | rethus | yes, the patch is still createt on sorurceforge. |
12:13.31 | rethus | i get it from there |
12:13.45 | rethus | now the make-precess runs like a charm |
12:14.06 | rethus | isn't there a parm for the makefile, where i can change the install-dir? |
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12:57.14 | rethus | k fund it. and works well |
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13:56.53 | yidiyuehan | anybody knows how to kick out some people in a conference from auto attendant? |
13:57.41 | EmleyMoor | The SIP VoIP mode on my N97 has stopped working :( |
13:58.07 | yidiyuehan | I know there are commands available in CLI for meetme kick, but how to control if I remotely dial into my * box? |
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14:07.14 | rethus | i have installed Conference as additional applikation. in cli i have now the command conference. |
14:07.27 | rethus | did anyone know, whit which command i could start a conference? |
14:07.39 | rethus | core show help conference gives no output for this module |
14:09.03 | Gugge | where did you get the application? |
14:09.55 | *** join/#asterisk guilhermebr (~Guilherme@189.63.75.25) |
14:11.29 | [TK]D-Fender | rethus: app_conference.so gives you "Conference" |
14:11.58 | [TK]D-Fender | rethus: "core show applications" <- go check your list |
14:12.12 | rethus | Gugge: i got it on sourceforge. |
14:12.12 | rethus | [TK]D-Fender: it is into this list |
14:12.28 | [TK]D-Fender | rethus: Then that's what you call. |
14:12.36 | Gugge | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference <- so maybe this gives you an idea rethus |
14:13.02 | rethus | if i say, help output nothing, means nothing usable... on section syntax is only "Not available" |
14:13.34 | Gugge | "core show application Conference", like all other apps |
14:13.54 | rethus | i have even try this: http://help.cloudvox.com/faqs/reference/reference-for-php#manageconf |
14:14.09 | rethus | didn't work. jump directly from the agi-script back to the dial-plan |
14:14.34 | rethus | Gugge: like i sayed above... there is no usable output |
14:15.16 | Gugge | paste the output |
14:15.25 | Gugge | on pastebin |
14:15.27 | [TK]D-Fender | [10:13]<rethus>i have even try this: http://help.cloudvox.com/faqs/reference/reference-for-php#manageconf <- this has nothing to do with app_conference |
14:15.47 | rethus | http://pastebin.com/4fRSbPpw |
14:16.21 | Gugge | if that is the output you get from "core show application Conference" you have a seriously fucked up install :P |
14:16.21 | rethus | [TK]D-Fender: this is the same command... paste over phpagi |
14:16.41 | [TK]D-Fender | rethus: So you have a broken AGI. What does this have to dowith app_conference? |
14:16.41 | rethus | no, thats the output if i try to dial in |
14:16.44 | rethus | ;) |
14:17.10 | Gugge | why dont you just make a dialplan with Conference(123) |
14:17.11 | [TK]D-Fender | rethus: You have not called it in there |
14:17.23 | [TK]D-Fender | rethus: FIX YOUR AGI |
14:17.35 | rethus | here the output of help for the module: http://pastebin.com/wx83MdCd |
14:17.35 | [TK]D-Fender | ]reththis is not a problem with app_ference |
14:18.07 | [TK]D-Fender | rethus: [10:12]<Gugge>http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference <- so maybe this gives you an idea rethus <--- he GAVE you a link for this |
14:18.38 | Gugge | actually, that link is the first google hit on "asterisk conference" :P |
14:18.48 | Gugge | no sorry, the second hit :) |
14:18.51 | Gugge | the first hit is meetme |
14:18.59 | rethus | thanxs, i have seen this now. Only want to show, that there is no syntax-explenation on the module. |
14:19.15 | Gugge | im gonna bet there is a readme |
14:19.17 | [TK]D-Fender | rethus: WE KNOW. Big deal. They didn't put the instructions in there. |
14:19.22 | [TK]D-Fender | rethus: DEAL WITH IT |
14:19.29 | rethus | so i now the parms now, but not the exact syntax. |
14:19.40 | rethus | thats what i'm searching for. |
14:19.48 | [TK]D-Fender | rethus: Your problem (well, just one of them anyway) is that you aren't even CALLING the application. |
14:19.51 | Gugge | what dont you get from the wiki link? |
14:19.53 | rethus | but now i first look for my agi |
14:20.21 | rethus | how could u see, that i not called in into my agi ? |
14:20.29 | rethus | [TK]D-Fender |
14:20.51 | [TK]D-Fender | rethus: You see when dialplan apps get called in CLI. it ISN'T |
14:21.05 | [TK]D-Fender | rethus: I can see it isn't called... because it ISN'T |
14:21.08 | rethus | ah, ok |
14:21.29 | [TK]D-Fender | rethus: We see the RX & TX and there ISN'T anything. |
14:22.11 | Gugge | rethus: try opening the README file from the source, and search for "Using app_conference" |
14:22.21 | Gugge | it actually is documented in the file you downloaded ..... |
14:23.24 | rethus | Gugge: ahh, k. Thanks. I've found it |
14:23.36 | [TK]D-Fender | Gugge: For reference he was working on just installing it for WEEKS |
14:24.09 | Gugge | impressive ... |
14:29.17 | rethus | so, if i logged in on the agi-interface, i should see this on CLI with agi-debug ?! |
14:29.56 | rethus | isn't that "<SIP/dev-00000006>AGI Tx >> agi_request: auth_congregation.php" the indication, that this script send a request to agi ? |
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14:30.37 | [TK]D-Fender | no, that is a refernce to some PHP file |
14:30.42 | [TK]D-Fender | rethus: ^ |
14:31.53 | rethus | this means, my connection data (username and password) for connecting to agi... may be wrong? |
14:35.29 | [TK]D-Fender | rethus: AGI doesn't HAVE passswords. What the fuck are you talking about? |
14:36.13 | [TK]D-Fender | rethus: rethus You dont' seem to have the slightest clue about ANYTHING you are working on. I suggest you go hire a consultant. |
14:37.02 | rethus | if u want to use agi, u have to set webenabled true in manager.conf and there u can give a "secret" means password?! |
14:37.39 | [TK]D-Fender | rethus: that is ***AMI**, not AGI |
14:37.52 | [TK]D-Fender | rethus: And what does AMI have to do with your use of app_conference? |
14:37.54 | Gugge | AMI/AGI ... who can tell the difference :P |
14:38.10 | [TK]D-Fender | Gugge: Everything on the outside looks the same! |
14:38.16 | Gugge | :) |
14:40.40 | [TK]D-Fender | rethus: And if you are using AMI via PHP_AGI as called by the AGI dialplan app... you aren't even ACCESSING it via the "web". |
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14:44.40 | [TK]D-Fender | rethus: And you also don't call dialplan apps via AMI <--- |
14:44.54 | [TK]D-Fender | rethus: Stop pouring windshield washer fluid in your GAS TANK |
14:48.55 | rethus | <PROTECTED> |
14:49.32 | [TK]D-Fender | rethus: You are talking about things in the same sentence that have nothing to do with each other. |
14:49.40 | rethus | "In everything, do to others as you would have them do to you" |
14:49.42 | [TK]D-Fender | rethus: You can't call dialplan apps via **AMI** |
14:49.56 | [TK]D-Fender | rethus: AGI doesn't have passwords. |
14:50.02 | [TK]D-Fender | rethus: Nothing you say makes any sense |
14:50.17 | [TK]D-Fender | [10:36]<[TK]D-Fender>rethus: rethus You dont' seem to have the slightest clue about ANYTHING you are working on. I suggest you go hire a consultant. |
14:51.06 | rethus | is that reason enough ffor you to disrespect me... cause i asking for help |
14:51.33 | [TK]D-Fender | rethus: We can't even tell what you are actually needing help with. The pieces don't even add up |
14:51.47 | [TK]D-Fender | rethus: And you never stated your goals. |
14:52.07 | [TK]D-Fender | rethus: No clue. No goals. No CODE. No debug. NOTHING |
14:52.18 | [TK]D-Fender | rethus: You have done nothing to help yourself in this process |
14:52.38 | [TK]D-Fender | rethus: What little you have every shown or asked has made no sense |
14:52.50 | [TK]D-Fender | rethus: How can anyone help you? |
14:53.19 | [TK]D-Fender | Well I've wasted enough time on this... I'm off for a while |
14:53.25 | rethus | thats not a reason do disrepect or kidding me. if you don't wan't to help me... only ignore me, but stop kidding me. |
14:53.43 | jamko | what is the reason for using openser if you can use dundi? |
14:54.10 | [TK]D-Fender | rethus: None of the pieces you have mentioned belong in the same SENTENCE. How can we "help" that? You are talking about things that don't belong together |
14:54.23 | Gugge | jamko: what does openser and dundi have in common? |
14:54.29 | [TK]D-Fender | jamko: OpenSER has nothing to do with DUNDi |
14:54.38 | jamko | I understand that |
14:54.49 | [TK]D-Fender | HOLY FUCKING SHIT THE CRAZIES ARE OUT IN FORCE |
14:55.05 | jamko | My question is, if you are not having issues with load balancing, or nat, why would one use openser? |
14:55.08 | Gugge | maybe we are just having a nightmare [TK]D-Fender :) |
14:55.23 | jamko | Hey fender go fuck yourself. Anyone have an intelligent answer? |
14:55.30 | [TK]D-Fender | jamko: LOTS of reasons. |
14:55.40 | jamko | lets hear one to start you asshole |
14:56.20 | Gugge | actually, i dont see any reason to run openser |
14:56.29 | Gugge | as it no longer exists :P |
14:56.36 | jamko | opensips then |
14:56.56 | [TK]D-Fender | jamko: To proxy *'s connection to multiple providers, to route out multiple public interfaces since * + multi-homed = PITA, to support LARGE installs where * can't handle setup of so many calls or clients at all. So you can use * as a BACKEND since SER does RADIUS billing so much better. |
14:57.05 | [TK]D-Fender | jamko: HUNDREDS of reasons. |
14:57.17 | [TK]D-Fender | jamko: TOTALLY different scale. |
14:57.39 | [TK]D-Fender | jamko: And SER is a proxy, * is NOT. Proxies can do things * cannot. |
14:57.42 | Gugge | But of cause, if asterisk does what you need, there is never a reason to mix in another app |
14:57.59 | [TK]D-Fender | jamko: Yes they can relate to each other and be used together benificially. |
14:58.09 | [TK]D-Fender | jmakbut they are VERY different |
14:59.17 | [TK]D-Fender | jamko: And starting out of the gate with relating SER to DUNDi is a bad start. DUNDi isn't even really a load balancing tool. |
14:59.56 | [TK]D-Fender | jamko: It is a "misc peering". There is no weighting, prioritizing,, logging, etc. |
15:00.47 | [TK]D-Fender | DUNDi = nearly worthless. Also a poor hack at trying to be a "general E.164" |
15:01.34 | [TK]D-Fender | is out for a while. |
15:01.35 | WIMPy | woot? |
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15:01.54 | WIMPy | dundi is great for a set of decentral servers. |
15:05.44 | jamko | ok.. So in a ser + * scneario, * would be used for pbx function, and SER for signalling.. but does ser stay in the middle after the call is setup, or does it reinvite the endpoints together? |
15:09.38 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
15:09.56 | bougyman | jamko: that's up to you. |
15:09.56 | p3nguin | <jamko> Hey fender go fuck yourself. <--- I wonder if this guy thinks he's the first one to think of that remark. |
15:10.31 | bougyman | in that scenario, * kind of is the endpoint. |
15:10.46 | jamko | p3nguin: I wonder if you haver ever seen a vagina. Go fuck yourself. |
15:11.06 | bougyman | jamko: this is completely uncalled for |
15:11.17 | jamko | Is it necessary to be condescending assholes in a chat room where people come for help? NO, it's not.. so again go fuck yourself. |
15:11.33 | bougyman | take your own advice and ignore the people who you aren't comfortable with. |
15:11.38 | p3nguin | Oh, a real wise guy, trying to complicate human anatomy with telephony. |
15:12.01 | jamko | oh a real wise guy... Think of that yourself? |
15:12.17 | p3nguin | No, I read it on a blog. |
15:12.35 | bougyman | jamko: what led you to this question, may I ask? you are talking a scenario that would usually only come up in a high-traffic or complex routing needs situation. |
15:12.40 | jamko | thanks for the tip bougyman, but I se here and watch these guys act like pricks over and over to other people. It is completely uncalled for. |
15:12.53 | bougyman | jamko: agreed, but so is what you are doing. |
15:13.00 | jamko | true. |
15:13.07 | bougyman | especially when someone is spending their time trying to understand your problem so I can help. |
15:13.17 | jamko | lol .. thanks.. |
15:14.54 | jamko | I'm just trying to visualize the big picture. Right now I have maybe 25 phones registered to a single asterisk box, but starting thinking bigger, and thought of bandwidth limitations, failover, load limitiations on a single asterisk server, etc. |
15:15.06 | bougyman | ah. |
15:15.21 | bougyman | yes, to scale asterisk, you have to go horizontal. |
15:15.27 | jamko | and I like to setup labs and see how stuff works. |
15:15.40 | bougyman | the asterisk side is the easy part. |
15:16.06 | bougyman | using kamalio or ser (both the same sip-router code) takes a pretty solid knowledge of the stack. |
15:16.29 | bougyman | i'd be hanging out in #kamalio to glean that special knowledge. |
15:16.40 | jamko | Well when dealing with NAT, you can't reinvite a peer and have good results. But does keeping ser in the middle defeat the purpose? |
15:17.07 | bougyman | jamko: that's where a proxy comes in. |
15:17.14 | jamko | and can it even stay in the middle? I was under the assumption it could not pass rtp traffic. |
15:17.29 | bougyman | it does not deal with rtp at all |
15:17.39 | bougyman | rtpproxy or a similar tool hanles the rtp |
15:18.02 | jamko | wow.. the rabbit hole... |
15:18.13 | bougyman | i'm _not_ trying to be an ass, but once again this is a kamalio/ser/sip-router problem, not an asterisk one. |
15:18.20 | bougyman | maybe that's why the * guys in here went off? |
15:19.12 | jamko | no, they went off because I asked the question openser v. dundi, and because they are elitist know it alls, decided to be pricks. |
15:19.28 | jamko | dundi is part of asterisk, no? |
15:19.54 | bougyman | ok maybe some of both, then.. dundi is part of * but your problem isn't an asterisk one. |
15:20.31 | jamko | maybe.. but my question boils down to asterisk clustering with failover. |
15:20.42 | jamko | and I believe asterisk has something to do with that, no? |
15:20.47 | bougyman | there are as many ways to do that as there are people who have done it. |
15:20.54 | jamko | ok and.. |
15:21.10 | jamko | my theory is (as naive as it might be)... is to |
15:21.29 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
15:21.32 | cusco | hi |
15:21.41 | bougyman | openvz live migration? check. sip-router handling external load balancing? check. SBC facing endpoints? check. |
15:21.46 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
15:21.46 | cusco | can we use case in AEL, or only if else ?? |
15:21.47 | bougyman | there's just a ton of choices, jamko |
15:21.49 | [sr] | howdy people |
15:22.21 | WIMPy | [sr]! |
15:22.23 | jamko | have a bunch of asterisk boxes, and when the amount of phones surpasses the supported load in asterisk, you register the next group of phones to another asterisk box, and use dundi to "find out" about the others. |
15:22.25 | Gugge | cusco: http://www.voip-info.org/wiki/view/Asterisk+AEL2 case is described there |
15:22.31 | [sr] | WIMPy: how r u? |
15:23.09 | WIMPy | Debugging... |
15:23.17 | cusco | thanks Gugge |
15:23.55 | WIMPy | But I had to debug the debugger first. |
15:24.03 | [sr] | WIMPy: thats the best time for it, on a sunday afternoon (here) |
15:24.34 | _Guhit | I'm running version 1.6.2 and have the problem, when a second call connects to the asterisk box things get laggy, key presses are missed, and the sound in confbridge is broken and choppy. The asterisk box is only using about 1% CPU. I'm using iax or sip channels, there is no dahdi hardware enabled on the box. The asterisk box is a co-lo, so there is plenty of bandwidth. |
15:24.40 | WIMPy | It's a test setup, so doesn't matter. |
15:25.11 | Gugge | _Guhit: what kind of hardware do you run it on? |
15:25.47 | _Guhit | Gugge: It's an old 1.33ghz athlon, 2GB RAM |
15:26.12 | *** join/#asterisk hopper75 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
15:26.26 | Gugge | _Guhit: and no virtualization on it? |
15:27.05 | Alton35 | Maybe "core show translation" and see if any of the numbers are too large. |
15:27.11 | [sr] | WIMPy: i see |
15:27.39 | *** join/#asterisk fr00d (~andi@unaffiliated/fr00d) |
15:27.39 | _Guhit | Gugge: I'm running in a FreeBSD jail, but there is nothing else running on the box. |
15:27.44 | fr00d | Hello! |
15:28.19 | Gugge | _Guhit: strange: i have no idea then |
15:29.35 | fr00d | Is that possible to use asterisk with KabelDeutschland? I have an account where the two wires at the cable modem are diabled and phone normally just works with the shipped Fritz!Box 7270 via TAE, S0 or DECT. What I want is to install asterisk on my OpenWRT and connect a USB ISDN interface to use the s0 bus. Is that possible? |
15:29.59 | fr00d | Oh just strip one "is that possible" ;) |
15:30.23 | WIMPy | fr00d: Just get the config out of the FritzBox and put them into sip.conf. |
15:30.43 | WIMPy | Google will give you howtos for that. |
15:31.31 | _Guhit | Alton35: what should I be looking for in that output? |
15:32.01 | _Guhit | How large? |
15:32.44 | fr00d | Hmm, damn. My Fritz!Box which I just got yesterday broke this morning and I thought I can get faster a USB ISDN interface to get telephony work again. |
15:32.51 | Alton35 | _Guhit: let me see |
15:33.40 | WIMPy | fr00d: If you can't get at it's configuration, you're screwed. |
15:34.07 | *** part/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
15:34.09 | WIMPy | Otherwise the USB thing is possible. |
15:35.19 | Alton35 | _Guhit: Do you know which codecs you are using? If you are using different ones, the computer has to translate between them. |
15:35.28 | fr00d | Ok, then I'll search for a howto on that and try to setup the configuration so I have a basic undertanding and hofully I just need to insert my credentials to get this work when I got a working Fritz!Box from KD. |
15:35.40 | _Guhit | Alton35: gsm and ulaw |
15:36.05 | Alton35 | And the number(s) in your table where those intersect? |
15:36.08 | _Guhit | Alton35: My numbers are in the 5000 range for that |
15:36.24 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
15:36.28 | fr00d | WIMPy: Could you advice me a usb isdn interface which would work? |
15:36.33 | WIMPy | fr00d: Mind you, for me KDGs telephony has been pretty horrible. |
15:37.05 | Alton35 | _Guhit: Golly, that shouldn't kill you. Trying to think of what else it could be. |
15:37.24 | fr00d | We will see. :D |
15:37.24 | WIMPy | fr00d: It needs to be based on a CCD HFC chip. |
15:37.55 | fr00d | I saw a howto with a conceptronics c128u device. |
15:38.16 | WIMPy | fr00d: Haven't heard of that one. |
15:38.27 | fr00d | Don't know if there's a really great variety of devices. |
15:39.18 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
15:39.26 | WIMPy | fr00d: I've got Trust and a X-Tensions mouse-shaped ones that work. |
15:40.12 | _Guhit | Alton35: How much latency on the SIP trunks would be too much, I'm getting between 85-180ms. |
15:40.51 | fr00d | WIMPy: Where do you come from and which provider do you use for that? |
15:41.35 | WIMPy | fr00d: de and I tried KDG, but had to cancel that early as it was not usable. |
15:41.45 | Alton35 | _Guhit: That's high. I can't think of how it would slow down the computer or asterisk though. |
15:42.43 | WIMPy | fr00d: Actually using sipgate worked much better than KDGs own telephony service. |
15:42.48 | WIMPy | (at least for me) |
15:43.25 | fr00d | Yes I know, I've got a sipgate account, too. |
15:43.37 | fr00d | But never tried that with asterisk. |
15:43.50 | WIMPy | fr00d: Ok, then you do have a working phone connection at least. |
15:44.10 | fr00d | Hehe |
15:44.25 | fr00d | But I don't have any credits on my sipgate account, just the free one. |
15:44.46 | fr00d | I'd really like to use my flat at KDG. |
15:44.48 | WIMPy | So do I. |
15:46.09 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
15:46.42 | fr00d | Hmm, this would add 8,90 € per month. |
15:46.50 | fr00d | But it would work... |
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15:52.34 | *** part/#asterisk rethus (~suther@p5087A9D5.dip.t-dialin.net) |
15:55.35 | p3nguin | _guhit: 5000? Holy cow! Between 1 and 5 might be a normal number for the translation between gsm and ulaw. |
15:57.01 | _Guhit | p3nguin: I just switched everything to gsm and it seems to be working better, any ideas those high number, or anything I can do to diagnose? |
15:57.28 | Alton35 | p3nguin: just verifying that you've seen my message. |
15:57.54 | p3nguin | huh? |
15:58.11 | Alton35 | I sent you 2 messages last night and one a while ago. |
15:58.22 | Alton35 | like so |
15:58.49 | p3nguin | Yeah, I don't receive unsolicited private messages. |
15:58.55 | Alton35 | aha |
15:59.26 | Alton35 | basically offering to stuff your paypal full of money for some help. ok, not quite, but something. |
15:59.34 | Alton35 | can't afford leif's $200/hour |
16:00.07 | _Guhit | On my desktop I'm seeing about 1000/400 for that translation...the numbers are in microseconds. |
16:00.09 | p3nguin | You want to discuss it privately? I'll open it up for you. |
16:00.13 | Alton35 | ok |
16:00.34 | p3nguin | Try now. |
16:26.15 | *** join/#asterisk steve (steve@bouncer.stephen.marsh.name) |
16:26.17 | steve | hi all |
16:27.15 | steve | I'm trying to figure out which IP a specific provider sends inbound SIP connections from, so I can add them to my firewall... I know where they go to, but can't seem to crack where they're coming from - can anyone point me to a specific portion of the log that would reveal this please? |
16:30.10 | Alton35 | should be the same IP. |
16:30.19 | Alton35 | it would be unusual for it to be different |
16:30.28 | steve | it's definitely not the same IP |
16:30.46 | Alton35 | or turn off your firewall and do: sip set debug peer xxxxx |
16:30.54 | steve | ah, good idea |
16:31.11 | steve | will that put info into asterisk/full? |
16:31.23 | Alton35 | I don't know, but it sure will put it on the console. |
16:31.46 | Alton35 | You'll have to scroll back to read it all. |
16:34.05 | *** part/#asterisk _Guhit (~amistry@adsl-99-174-178-97.dsl.wotnoh.sbcglobal.net) |
16:35.36 | Alton35 | steve: I'm not sure how you'd see which IP to use, maybe look in your firewall log? or ask the guys at the provider. seems like the right way to go about things. |
16:38.39 | jamko | steve: you should be able to see this from the cli when a call comes in. The only way you would not is if you don't have 5060 open in your firewall. Even if the call is rejected, you will see the ip address in teh console. You should not need to turn on debugging to see this. |
16:39.28 | jamko | just make sure your core verbosity is at least 3 |
16:40.21 | steve | yeah I actually found the problem not where I expected it to be |
16:40.30 | steve | firewall wasn't permitting loopback it would seem |
16:40.38 | steve | asterisk wasn't pleased :P |
16:48.33 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
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17:00.44 | cusco | hi.. |
17:01.47 | cusco | when I am performing an outbound call... |
17:02.31 | cusco | after the Answer, it is still ringing on the other end... is there any flag on Dial() that allows me to identify for how long it actually rang, |
17:02.46 | cusco | or how long I actually connected? |
17:03.09 | jamko | yes. |
17:04.01 | jamko | but is the other end a sip device? |
17:04.10 | jamko | sounds like you have a nat issue. |
17:04.47 | cusco | no |
17:04.59 | cusco | the other end is PSTN trough dahdi |
17:05.02 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
17:05.24 | cusco | what is the flag i am looking for? or the cdr var? |
17:06.45 | jamko | Dial(SIP/4025,15) |
17:06.53 | jamko | the 15 would be your timeout |
17:07.07 | cusco | oh no, Im not looking for a timeout |
17:07.19 | jamko | oh i see. |
17:07.23 | jamko | i misread what you said. |
17:07.24 | cusco | Im trying to count the time on-line |
17:08.09 | cusco | we insert info on sql database, when dialing out and when hang up |
17:08.54 | cusco | and the time between dial and hangup would be ok, unles the called party does not answer or answers after 5 secs ringing |
17:09.06 | cusco | so.. how can I count only the connected time? |
17:10.19 | jamko | sorry no clue man. pass |
17:11.10 | *** join/#asterisk madpr (~madpr@81.200.7.6) |
17:11.16 | *** join/#asterisk SWFu (~SWFu@unaffiliated/swfu) |
17:11.16 | Gugge | cusco: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
17:11.33 | Gugge | "CDR Variables" on that page |
17:12.17 | SWFu | Could any give me some pointers about UK providers please? |
17:12.54 | cusco | thanks |
17:12.54 | jamko | swfu, what kind of pointers? |
17:13.11 | SWFu | Like who to use |
17:13.23 | madpr | how can i escape a variable in dialplan application arguments? (coz it may contain a comma, which makes asterisk crazy) |
17:13.30 | SWFu | Of what I understand a SIP provider? |
17:13.54 | SWFu | Only started looking into the idea this morning |
17:19.11 | *** join/#asterisk shapr (~shapr@c-76-29-246-212.hsd1.al.comcast.net) |
17:19.19 | jamko | is this for calls in the UK, or you just want the provider to be located in the UK? |
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17:38.43 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
17:45.17 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
17:50.40 | [TK]D-Fender | jamko: Somewhat belated but watch the anatomical vulgarity. |
17:52.26 | [TK]D-Fender | [11:19]<jamko>no, they went off because I asked the question openser v. dundi, and because they are elitist know it alls, decided to be pricks. [11:19]<jamko>dundi is part of asterisk, no? <- and no, we weren't fractionally as harsh as your retorts. |
17:53.24 | [TK]D-Fender | jamko: And DUNCi has no relationship with SER, isn't a "load balancer", etc. Some larger points of what SER can add to a hybrid install were already provided |
17:53.30 | [TK]D-Fender | DUNDi* |
17:57.59 | jamko | right I understand it has no relation to SER. I am just want a bunch of asterisk servers to be able to talk to each other, without overcomplicating the issue. By "talk to each other" |
17:58.27 | jamko | I mean, find a sip registration, or extension that is not on the local box. |
17:58.44 | [TK]D-Fender | jamko: So far that doesn't say "load balance" either... |
17:59.00 | [TK]D-Fender | jamko: Do you need to "search" to find where it is appropriate? |
17:59.35 | [TK]D-Fender | jamko: Or is this a scenario where you know 2XX = PBX1, 3XXX =PBX2, etc? |
17:59.45 | jamko | right but do I need load balancing, if I only put say 100 phones per box? |
18:00.23 | jamko | right. |
18:00.37 | [TK]D-Fender | jamko: A single * can handle hundreds of calls depending how you do it |
18:00.55 | [TK]D-Fender | jamThousands even. I think the record is a bit over 10000 on a single box |
18:01.22 | jamko | wow. |
18:02.02 | [TK]D-Fender | jamko: Sangoma sellsan 8 Port PRi card... that's 240 calls just to support that 1 card. Of course * can handle more. |
18:02.41 | jamko | well, then my problem gets to be bandwidth restraints on a single internet connection in front of asterisk, which is why I was thinking multiple boxes, behind multiple internet connections. |
18:03.03 | jamko | then using dundi to link them. |
18:03.42 | jamko | so say with g729 I max out at 25 simultaneous calls, but want 500 calls to act as if on 1 pbx. |
18:04.44 | [TK]D-Fender | jamko: That gets a LOT messier... |
18:04.45 | jamko | and I am pure sip, no need for cards. |
18:04.52 | [TK]D-Fender | jamBecause of *'s transcoding,e tc |
18:06.18 | jamko | Fender : would opensips help at all in sorting out that mess? |
18:06.34 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
18:06.45 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
18:07.29 | [TK]D-Fender | jamko: it might help.... it is a complicated beast though... not sure I'd advise it yet. I'd have to see a clearer map of your layout. |
18:11.35 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
18:13.12 | *** join/#asterisk rethus (~suther@p5087A9D5.dip.t-dialin.net) |
18:16.04 | Kobaz | anyone know if you can still squeek into astricon as a speaker? |
18:16.16 | Kobaz | on the site it says the deadline is june 30... but there's still empty slots in the schedule |
18:19.01 | rethus | someone here who is using phpagi? |
18:19.17 | rethus | try to find a way to create a new conference via agi. |
18:20.37 | ChannelZ | Well you can use 'exec' to call a dialplan app like ConfBridge or MeetMe |
18:20.54 | [TK]D-Fender | rethus: You don't create conferences with AGI |
18:21.23 | [TK]D-Fender | rethus: You call dialplan appas with AGI. |
18:24.26 | rethus | so this should work for creating a new conference on channel 1 as admin-user? |
18:24.26 | rethus | $agi->exec('meetme', '1', 'a'); |
18:25.22 | [TK]D-Fender | rethus: that is JOINING a conference. This is not "creating" anything |
18:25.31 | rethus | k |
18:25.34 | Alton35 | I use phpagi. Not that I can help you much. |
18:25.53 | Alton35 | Stick with these guys for now. |
18:26.19 | bougyman | only one way to find out, Kobaz |
18:26.23 | [TK]D-Fender | Alton35: It isn't even an AGI program. |
18:26.27 | bougyman | there are usually cancellations at these things, too. |
18:29.02 | rethus | my problem is, that asterisk jumps into the phpagi-script... but doesn't matter what i do, it come back (while don't entering this conference) - say "AGI Script auth_congregation.php completed, returning 0" and switch back to dialplan. |
18:29.22 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
18:29.38 | rethus | so i checked the asterisk-logs and start agi debug, but get no error which show me, why the conference is not enterd |
18:29.54 | [TK]D-Fender | rethus: Your AGI is BROKEN. We've rtond you thins many times. |
18:30.00 | [TK]D-Fender | rethus: You're doing it wrong. |
18:31.22 | rethus | so how can i check it step by step to get it to work again? |
18:31.53 | rethus | and what do you mean with agi... my php-script? |
18:32.04 | [TK]D-Fender | YES |
18:32.28 | rethus | k, but it works on another machine (ubunbtu) like a charm (not here on SUSE) |
18:32.28 | [TK]D-Fender | rethus: Your AGI script is BROKEN. Please get this in your head. BROKEN. Bad code. Errors. FAIL. |
18:33.02 | [TK]D-Fender | rethus: rethus We can't see anything so how are we supposed to know WHY it is failing? |
18:39.31 | Kobaz | brokes |
18:39.44 | Kobaz | teh borken |
18:41.26 | jamko | rethus: maybe you want to stick with ubuntu? |
18:41.50 | [TK]D-Fender | Because clearly his DISTRO is at fault.. yeah.. uh huh |
18:43.02 | jamko | either way, it is just logical to stick with what is working. |
18:43.18 | jamko | unless you can fix what is not, which apparently he can't/ |
18:44.04 | [TK]D-Fender | Nope. |
18:44.05 | Kobaz | or he can learn about the system he's developing on |
18:44.31 | [TK]D-Fender | Kobaz: this has been going on for weeks. |
18:45.46 | Kobaz | hehe |
18:47.39 | b14ck | Hey, is anyone using the new dialplan pattern matcher in production? extenpatternmatchnew=yes ? |
18:48.44 | Kobaz | b14ck: what's it look like |
18:49.13 | b14ck | Kobaz, what? |
18:49.41 | Kobaz | the patterns |
18:49.44 | Kobaz | or is it just a new engine |
18:49.57 | rethus | jamko: i would, but is not my server |
18:50.00 | b14ck | It's a new algorithm that you can (optionally) use. |
18:50.06 | Kobaz | how's it work |
18:50.10 | b14ck | I wonder if it's been well-tested or not. |
18:50.14 | b14ck | And if it is production safe. |
18:50.18 | b14ck | Might as well give it a try. |
18:52.30 | b14ck | Nice, seems to work! |
18:52.42 | b14ck | We've got like ~60,000 extensions =p |
18:52.46 | b14ck | So that's a big speedup. |
18:52.47 | b14ck | heh |
18:53.53 | Kobaz | oh... efficiency improvements? |
18:54.14 | b14ck | Yah, apparently, with 10,000 extensions, the speedup is 374x |
18:54.24 | b14ck | With 1000, it is 25x |
18:54.35 | Kobaz | heh |
18:57.51 | rethus | so, i have found the error now. it was a problem with the used PATH in Asterisk. in php-script i had an include()... i didn't know why, but asterisk always seems to start from /tmp as basedir. |
18:58.49 | [TK]D-Fender | rethus: * does what you tell it to. |
18:59.59 | rethus | cause no errors @ all was thrown, it was an error which is hard to find. And there ist one thing, that makes this problem more difficult. If u start asterisk with the -cv command out of another directory... this direcory is the base-dir.... so if u start asterisk as deamon via init.d, it has another basedir than starting it with -cv |
19:01.22 | rethus | if some phpagi-users here... how do you debug your scripts. is there a way to make asterisk more verbose on showing this kind of php-script-errors? |
19:01.24 | [TK]D-Fender | rethus: Maybe you should look at what the STARTUP script is doing. That is your job |
19:01.40 | rethus | your right, good idea |
19:03.19 | rethus | mhh, no path-parms in the init-script. |
19:06.46 | [TK]D-Fender | rethus: asterisk.conf <- |
19:07.14 | *** join/#asterisk shapr (~shapr@c-76-29-246-212.hsd1.al.comcast.net) |
19:07.37 | rethus | so no parms in the config-files of asterisk, (on asterisk.conf its only set as record_cache_dir). so i realy wonder, why the handling on ubuntu is deifferent from suse |
19:08.06 | [TK]D-Fender | rethus: Maybe you should look at BOTH system similarly. |
19:08.20 | rethus | is there an parm for asterisk, which set the basedir manualy from cli? |
19:08.55 | [TK]D-Fender | rethus: Go look for yourself |
19:09.12 | [TK]D-Fender | rethus: Takes less to find out than it did for you to ask. |
19:11.34 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
19:14.06 | jamko | maybe you should keep your systems uniform and stay with ubunutu. I think you would have saved a lot time, unless you simply are just doing this for the sake of doing it. |
19:17.53 | shapr | Does dahdi-linux-complete put its bunch of kernel modules into the modules.conf? |
19:21.23 | jamko | 1 |
19:23.13 | *** join/#asterisk Z_God (~julius@stud169209.mobiel.utwente.nl) |
19:23.54 | *** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com) |
19:24.36 | rethus | jamko: i would stay on ubuntu, but its not my server where my asterisk-webapp is installed... and so i must get this running on SUSE |
19:25.53 | Mark22 | Hello, I am trying to get LCDial (http://www.voip-info.org/wiki/view/Application+LCDial#ApplicationLCDialLeastCostRoutingFailOve) to work (mainly because it gives some failover options). log information: http://yourpaste.net/5932/ dialplan (part related to this call): http://yourpaste.net/5933/ I did check if I could connect with the login details in lcdial.conf to mysql from the commandline and it did work without any problem |
19:26.12 | Mark22 | what could be the solution? |
19:26.34 | p3nguin | shapr: modules.conf is deprecated and not present on current Linuxen. |
19:27.08 | p3nguin | shapr: Actually, disregard what I just said. |
19:27.55 | p3nguin | shapr: I was thinking of something else rather than /etc/asterisk/modules.conf. |
19:28.11 | [TK]D-Fender | "Couldn't connect to database server '127.0.0.1'." <- looks blatant enough to me |
19:28.19 | [TK]D-Fender | Mark22: ^^ |
19:28.42 | Mark22 | [TK]D-Fender: it looks good, however why couldn't it connect is my problem (so I could find a solution) |
19:29.04 | [TK]D-Fender | Mark22: It's wrong. It says its wrong. It isn't lying. |
19:32.19 | Mark22 | if I do "mysql -h 127.0.0.1 -u asterisk -p asterisk" and after that enter the password (same user/database/password as listed in lcdial.conf) it works, why lcdial can't use it is my problem :S (passwords contains A-Za-z0-9) |
19:32.33 | Mark22 | I don't say the log is lying, I only can't find the problem |
19:34.59 | [TK]D-Fender | Mark22: We sill can't see enough to point it out to you either. |
19:35.10 | [TK]D-Fender | Rock, hard place. Hard place, rock. |
19:35.16 | [TK]D-Fender | has settled the introductions. |
19:35.53 | Mark22 | do you know how I can get more information about it? |
19:36.39 | [TK]D-Fender | Mark22: We don't see all configs, database dumps, etc |
19:36.57 | [TK]D-Fender | comparative local logins, etc |
19:38.29 | drmessano | Sounds like either a RAM failure or maybe a bad PCI slot |
19:38.51 | drmessano | That's as good of a guess as any, with no data |
19:39.15 | drmessano | Maybe check the LED's and the color of the power cord |
19:41.06 | shapr | p3nguin: So, dahdi modules to modprobe do show up in /etc/asterisk/modules.conf? |
19:41.39 | rethus | have anyone an idea, how i can make a benchmark while "converence_app" vs. "meetme" ? |
19:41.42 | p3nguin | shapr: I wouldn't expect them to, since /etc/asterisk/modules.conf is for asterisk modules and you're asking about kernel modules. |
19:49.31 | *** join/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
19:50.24 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
19:51.33 | *** join/#asterisk Godfather_ (~godfather@233.Red-88-8-14.dynamicIP.rima-tde.net) |
19:51.34 | Godfather_ | jhi |
19:54.08 | shapr | hhi |
19:54.44 | xheliox | jello |
19:56.21 | [TK]D-Fender | beats xheliox with a Pudding Pop |
19:56.51 | xheliox | Seems like a perfectly good waste of a Pudding Pop. |
19:59.19 | shapr | One or more of the dahdi kernel modules has make my system extremely flaky. I want to only load the wctdm module. Does anyone know how to keep the other modules from loading? |
20:01.57 | [TK]D-Fender | shapr: remove them |
20:03.28 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
20:06.40 | *** join/#asterisk fr00d (fr00d@unaffiliated/fr00d) |
20:10.02 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
20:22.57 | cusco | what would be the most common cause for a call to disconnect while on hold for 30 secs |
20:23.00 | cusco | ? |
20:26.29 | jamko | rtpholdtimeout= can cause issues, not sure about the most common though. |
20:30.59 | *** join/#asterisk shapr (~shapr@c-76-29-246-212.hsd1.al.comcast.net) |
20:31.59 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
20:32.47 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
20:32.53 | tompaw | Good evening, everyone. |
20:35.58 | ChannelZ | maybe your MOH music is really terrible and people just hang up |
20:37.09 | xheliox | ChannelZ: Disco lives forever! |
20:38.37 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-64-197-9.mia.bellsouth.net) |
20:40.10 | Kobaz | hmmm |
20:40.16 | shapr | I'm going through the asterisk book (but using asterisk 1.6-current). I've reached chapter 4, where it talks about setting up a dialplan for test calls... |
20:40.31 | shapr | How do I know that my TDM400 is setup correctly? |
20:40.39 | Kobaz | in 1.6.2... it seems if you don't answer the call... cell phones calling into a t1 won't hear ringing |
20:41.11 | shapr | Also, I know the red card is an FXO, and the green card is an FXS, but I don't remember which slot on the card maps to which RJ11, does anyone know? |
20:41.13 | tompaw | Guys, I am trying to test my a2b installation under full load. But before the cpu and ram are consumed, I'm getting a bunch of errors like these: http://pastebin.com/Hfq7Sewn |
20:41.38 | GameGamer43|Mac | tompaw: a2billing is not supported here per the channel topic |
20:41.55 | Kobaz | even if i do Answer()... Ringing()... cell phones still wont hear ringing |
20:41.58 | tompaw | GameGamer43|Mac: these are not a2b errors, it's related to * only. |
20:42.07 | Kobaz | i have to play music on hold with a recorded wav of ring |
20:42.28 | tompaw | my ulimit is set to unlimited btw... |
20:43.41 | tompaw | It looks like there are some sorts of limits hidden somewhere in */agi configuration that I cannot find. Syslog is clean, there is free cpu and ram... |
20:45.44 | tompaw | Googling for "rtp.c: Unable to allocate RTP socket: Too many open files" takes me way back to 2006 and is mostly related to ulimit, which is not the case here... |
20:46.14 | [TK]D-Fender | [16:40]<shapr>How do I know that my TDM400 is setup correctly? <- USE IT |
20:46.44 | shapr | [TK]D-Fender: I would love to, but I don't yet know how to do that. |
20:47.11 | [TK]D-Fender | shapr: You don't know how to DIAL? Very sad |
20:47.26 | [TK]D-Fender | tompaw: [Aug 15 22:15:33] ERROR[26266] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files <- FS lockout. Somrhitng is opening too many files. |
20:48.37 | *** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com) |
20:48.44 | tompaw | [TK]D-Fender: checked the ulimit, /proc/sys/fs/file-max, running * as root, no OS errors in syslog... maybe I should try disabling the cdrs? |
20:49.16 | [TK]D-Fender | tompaw: No, the lack of ability to open files killed your ability to open SOCKETS as well because they are effectively the same |
20:49.42 | shapr | [TK]D-Fender: I still have lots of basic questions like, should I hear a dialtone when I have a phone plugged into the FXS port of my TDM400? If not, should I get some reaction when I dial? |
20:49.43 | [TK]D-Fender | tompaw: [Aug 12 20:16:48] ERROR[5549] utils.c: write() returned error: Broken pipe <- also an error common to AGI/AMI screwups |
20:49.55 | tompaw | correct. I only wonder on which "layer" is this limit enforced, sinced it's not the filesystem. |
20:50.02 | [TK]D-Fender | shapr: You should. Did you conenct the molex to the card? |
20:50.11 | shapr | I did, yes. |
20:51.37 | shapr | [TK]D-Fender: I don't get a dialtone on the FXO port, but I do get power to the phone and am able to dial. |
20:54.15 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com) |
20:54.38 | [TK]D-Fender | shapr: What do you mean no tone on the FX port? |
20:54.42 | [TK]D-Fender | FXO* |
20:54.50 | tompaw | lol, someone is having a similar problem to mine and asked at a2b forum. the response was: http://forum.asterisk2billing.org/viewtopic.php?f=2&t=2661&p=9530&hilit=too+many+open+files#p9530 |
20:55.02 | tompaw | "THIS IS NOT ASTERISK SUPPORT FORUM!!!" |
20:55.03 | tompaw | =) |
20:55.19 | shapr | [TK]D-Fender: I mean that, picking up the handset does not give dialtone, but the phone is powered. |
20:58.40 | [TK]D-Fender | shapr: Phones should plugged into FXS ports, not FXO |
20:58.55 | shapr | [TK]D-Fender: Ah right, sorry... I meant the FXO port. |
20:59.23 | shapr | I don't have access to the PSTN at the moment, so the FXO doesn't do anything for me. |
20:59.36 | p3nguin | heh |
20:59.50 | shapr | Er, I meant the FX*S* port... |
20:59.56 | shapr | grumbles |
21:00.54 | *** join/#asterisk Alagar (~Administr@122.164.34.24) |
21:03.12 | shapr | I'm just trying to get asterisk talking to my analog handset. Once I have a vaguely functional config, I can poke around in the system more productively. |
21:05.09 | tompaw | whoa |
21:05.31 | tompaw | [TK]D-Fender: you're actually right, looks like centos ignores my limits, just checked /proc/<*_uid>/limits... 1024 ! |
21:08.11 | [TK]D-Fender | imagine that.... |
21:10.10 | tompaw | fixed. |
21:10.50 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
21:11.09 | tompaw | it went up to 190 calls :-) now I'm stuck with this broken pipe. If only asterisk reported which part causes the error... |
21:25.09 | *** join/#asterisk Diffen2 (~diffen2@c-2875e555.042-17-73746f11.cust.bredbandsbolaget.se) |
21:30.16 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
21:33.40 | *** part/#asterisk rethus (~suther@p5087A9D5.dip.t-dialin.net) |
21:42.30 | *** join/#asterisk ruied (~ruied@95.69.57.109) |
21:53.12 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
21:53.30 | Diffen2 | hello. i have two asterisk and i have created a user on the first one and then im trying to use that user as a siptrunk on the second asterisk. is that possible? |
21:54.22 | *** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com) |
21:57.50 | [TK]D-Fender | Diffen2: Yes. |
22:04.21 | Diffen2 | ok nice i think i have managed it to register as a peer but when i call in i get username mismatch from the second asterisk |
22:04.41 | Diffen2 | guess i have missed something out in sip.conf? |
22:05.21 | b14ck | Can someone explain to me what allowoverlap does (in sip.conf)? What is 'overlap dialing support'? I googled it, but couldn't find any straightforward answers. |
22:07.46 | *** join/#asterisk Pegasus_RPG (~chatzilla@p4FF90751.dip.t-dialin.net) |
22:08.34 | Pegasus_RPG | hello there. I'm having a problem with * using Broadvoice VoIP service. Inbound calls are going straight to the BV voice mail system and Asterisk's console doesn't ever see the calls |
22:08.49 | Pegasus_RPG | It used to work fine a year ago |
22:09.00 | Pegasus_RPG | I've since dist-upgraded to the latest Debian Testing |
22:09.21 | Pegasus_RPG | Any ideas on what to try? |
22:13.10 | shapr | Is there a guide that translates the Asterisk book's 1.4 configuration directives into 1.6 directives? |
22:13.14 | *** join/#asterisk fofware (fabian@190.225.15.129) |
22:13.28 | jamko | PEGASUS: Sounds like the provider is sending calls to the wrong ip, or not at all. |
22:13.29 | shapr | For example, there's no zapata.conf file in 1.6 |
22:15.47 | jamko | PEGASUS: Or you don't have the sip port specified in your sip.conf, open in your firewall. |
22:16.01 | Pegasus_RPG | checks his firewall |
22:16.35 | *** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com) |
22:18.15 | carrar | shapr, that would be the all the text in ChangeLog since that release :) |
22:21.09 | Pegasus_RPG | jamko: My firewall looks good. I have 69, 5060-5063 and 10K-20K all UDP forwarded to my * box. Also if I have my SIP hard phone connect directly to Broadvoice, it works fine |
22:22.43 | Pegasus_RPG | without forwarding anything special in the firewall |
22:23.26 | jamko | what about your linux firewall? is it off? |
22:23.55 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
22:27.03 | Pegasus_RPG | jamko: I never installed one |
22:27.41 | Pegasus_RPG | jamko: though iptables is installed...but there's no "iptables" process running |
22:29.58 | jamko | are you setup for ip auth through your provider? |
22:30.23 | Pegasus_RPG | jamko: Don't know what that is |
22:30.45 | carrar | iptables -n -L |
22:31.34 | jamko | how do you authenticate your asterisk box to your origination provider? |
22:34.54 | Pegasus_RPG | carrar: no rules are listed |
22:35.10 | Pegasus_RPG | jamko: using the steps given here: http://broadvoice.com/support_install_asterisk.html |
22:36.06 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
22:36.30 | [TK]D-Fender | Pegasus_RPG: enable SIP DEBUG, and show us your REGISTRATION, and your CALL ATTEMPT. |
22:36.31 | [TK]D-Fender | ~pb |
22:36.32 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
22:36.34 | [TK]D-Fender | ^^^ |
22:40.17 | Pegasus_RPG | http://pastebin.ca/1917743 |
22:41.22 | [TK]D-Fender | Pegasus_RPG: What router are you using? |
22:41.40 | Pegasus_RPG | Linksys WRT310N v1 |
22:41.50 | Pegasus_RPG | Firmware Version: v1.0.09 |
22:42.23 | [TK]D-Fender | Pegasus_RPG: Ok, not seeing the problem, just yet. Confirm that your WIN IP on your router is indeed : 79.249.7.81 |
22:42.26 | Pegasus_RPG | I have it set to try to do QoS on the * IP |
22:43.01 | [TK]D-Fender | Don't screw with the traffic in any way. Just ensure 5060, 10000-20000 all UDP is forwarded and that's all on the router side. No SIP ALG transform, etc |
22:43.01 | Pegasus_RPG | yup 79.249.7.81 |
22:43.25 | Pegasus_RPG | and the firewall doesn't have anything SIP-specific. Just QoS |
22:43.34 | Pegasus_RPG | (which doesn't work all that well, I might add) |
22:43.48 | [TK]D-Fender | Pegasus_RPG: Ok, BTW, I just got to the bottom... ensure that your broadvoice peer has "insecure=port,invite" |
22:43.59 | Pegasus_RPG | ahh, it's using insecure=very |
22:45.14 | Pegasus_RPG | ok that helped...now it's just not matching the extension |
22:45.32 | Pegasus_RPG | where's a list of wildcard characters? |
22:45.42 | [TK]D-Fender | ~book |
22:45.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
22:45.46 | [TK]D-Fender | ^^^ |
22:45.46 | carrar | heh |
22:46.09 | Pegasus_RPG | thanks |
22:46.46 | Pegasus_RPG | http://www.asteriskdocs.org/ doesn't seem to work, btw |
22:46.53 | Pegasus_RPG | digs up his old PDF copy |
22:47.14 | carrar | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
22:47.26 | golikwid|mac | Pegasus_RPG: works fine here |
22:48.01 | Pegasus_RPG | gaah, I hate it when ISP DNS servers try to be "helpful" with their host-not-found catch-alls |
22:48.32 | golikwid|mac | yea i dont use my isp's dns server for that reason |
22:48.35 | Pegasus_RPG | ah, so ! means what _ used to in v1.0.x? |
22:49.33 | [TK]D-Fender | Nope |
22:49.51 | [TK]D-Fender | You also shouldn't need a wildcard match |
22:52.06 | Pegasus_RPG | wheee it works now! |
22:52.36 | Pegasus_RPG | (Somehow my first exten =>_<number> had n instead of 1) |
22:52.46 | Pegasus_RPG | thank you so much everyone |
22:53.26 | [TK]D-Fender | Pegasus_RPG: you're welcome |
22:53.36 | Pegasus_RPG | I do also have a number reformatting question. |
22:53.43 | [TK]D-Fender | Shoot |
22:53.58 | Pegasus_RPG | I live in Germany, but use BroadVoice for US calls |
22:54.37 | [TK]D-Fender | Pegasus_RPG: Don't worry, we've already alerted the authorities.... |
22:54.41 | Pegasus_RPG | I want * to take any dialed number starting with 0, strip off the 0, add 01149 and the rest of the remaining digits, and tell BV to call that |
22:54.45 | Pegasus_RPG | lol |
22:55.43 | golikwid|mac | (01149)+0|NXXXXXX |
22:55.47 | golikwid|mac | something like that i think |
22:59.05 | Pegasus_RPG | also, sicne I call a number of european countries with variable-length numbers, how can I have it match any length for int'l calls? _011X. ? |
22:59.32 | golikwid|mac | yea the . |
23:00.25 | golikwid|mac | whats the number pattern used un europe |
23:00.28 | golikwid|mac | ? |
23:01.20 | florz | there is non, essentially |
23:01.22 | Pegasus_RPG | there is no defined rule |
23:01.22 | florz | +e |
23:01.29 | [TK]D-Fender | [18:54]<Pegasus_RPG>I want * to take any dialed number starting with 0, strip off the 0, add 01149 and the rest of the remaining digits, and tell BV to call that <--- exten => _0.,1,Dial(SIP/broadvoice/01149${EXTEN:1}) |
23:01.36 | Pegasus_RPG | even within Germany, the number lengths can vary |
23:01.48 | golikwid|mac | weird |
23:02.00 | golikwid|mac | sounds like more complicated dialplans than around here |
23:02.31 | florz | only because asterisk is kindof not adapted to it |
23:02.46 | golikwid|mac | im sure there is a way |
23:02.55 | golikwid|mac | just more lines of patterns to match |
23:03.13 | florz | that doesn't really help you |
23:03.30 | florz | variable length means you can't know when a number is complete |
23:04.33 | Pegasus_RPG | [TK]D-Fender: thanks, that works perfectly |
23:04.45 | [TK]D-Fender | Pegasus_RPG: Or at least precisely as you asked |
23:04.49 | Pegasus_RPG | yes... |
23:04.57 | [TK]D-Fender | Pegasus_RPG: Now go read up on viables & patterns |
23:05.00 | Pegasus_RPG | it mangles if I manually dial 011xx... of course |
23:05.02 | Pegasus_RPG | I will |
23:06.47 | Pegasus_RPG | thanks alot for your help and patience with me |
23:08.30 | Pegasus_RPG | uh, I can't find an explicit negative match notator |
23:08.44 | Pegasus_RPG | that is a "don't match" operator |
23:08.57 | Pegasus_RPG | or "match everything but 123" |
23:09.29 | [TK]D-Fender | Pegasus_RPG: No such thing |
23:09.57 | Pegasus_RPG | that would explain why I can't find it. :) |
23:10.03 | [TK]D-Fender | Pegasus_RPG: You have to have a "match ALL that takes precedence LAST |
23:10.14 | Pegasus_RPG | all |
23:10.17 | Pegasus_RPG | er ahh |
23:10.39 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
23:12.38 | xSmurf | hey guys, I'm working with the manager api and using the Originate action to connect a sip extension and an outbound call. There are two things I cannot figure out. The first one being how to send a caller id to the sip extension (ex: Outbound <5551234567>) and the second is it possible to play back "please wait while I try to connect you" when the sip extension has answered and the outbound call is being dialed? |
23:14.05 | xSmurf | for a ref, this is my php script connecting to the manager api using sockets http://pastie.textmate.org/private/5p96sjkihuop6m8yyn91cq |
23:14.55 | xSmurf | nothing fancy |
23:15.35 | [TK]D-Fender | xSmurf: does it Originate? |
23:15.46 | xSmurf | yeah it works great :) |
23:16.10 | [TK]D-Fender | xSmurf: then that part is almost irrelevant |
23:16.44 | xSmurf | I know I know, I just posted it for reference, some people are more picky and like seeing the big picture ;) |
23:17.25 | xSmurf | and others might just enjoy the script, I'm just exploring after I wrote an Address Book.app plugin for Click to dial |
23:17.30 | [TK]D-Fender | xSmurf: I suppose... |
23:19.00 | xSmurf | back the the relevant part, is there anyway for to display CID information to the sip extension? |
23:19.37 | *** join/#asterisk root52 (~root52@ip68-228-177-7.cl.ri.cox.net) |
23:23.27 | root52 | hey all... http://pastebin.ca/1917791 So This problem just started yesterday. Seems that my dahdi channel goes into red alarm and then comes out a min or two later. Then when it is not in alarm and I make a test call I get a message I have not seen before and it never picks up the call like it says it does. Any Thoughts. The odd message is line 11 and 12 in the pastebin. |
23:23.54 | [TK]D-Fender | xSmurf: Which? the Channel: or the EXTEn? |
23:24.33 | xSmurf | the exten, basically I wanna display the number from the channel to the exten when originate does the local dial |
23:24.52 | xSmurf | CallerID doesn't seem to be display, and anyhow, I don't want that passed to the channel! |
23:27.29 | [TK]D-Fender | xSmurf: then set it in your dialplan |
23:28.48 | xSmurf | yeah mean create a custom extension that the call is directed to and that then connects the exten and the chan? yeah I was just thinking of that, though I'd like to avoid having to play with the dialplan (don't always have access to it) |
23:30.14 | [TK]D-Fender | xSmurf: And why not? |
23:30.42 | [TK]D-Fender | xSmurf: Next set a SIMPLE variable in your Originate and set the CID to that. |
23:30.56 | xSmurf | yeah I figured |
23:30.57 | [TK]D-Fender | xSmurf: Do not attempt to set it directly |
23:31.04 | xSmurf | as I said don't always access to the dialplan |
23:31.56 | xSmurf | what about the playback of some file to the exten while the call is connecting? same thing?? |
23:32.05 | [TK]D-Fender | xSmurf: and I asked you why not... and why would you need continuos access to it? |
23:32.30 | xSmurf | not continuous... access period |
23:32.39 | [TK]D-Fender | [19:31]<xSmurf>what about the playback of some file to the exten while the call is connecting? same thing?? <- this you clearly need control over the dialplan for. |
23:33.00 | [TK]D-Fender | [19:32]<xSmurf>not continuous... access period <- please answer this simple question in some meaningful way. |
23:33.00 | xSmurf | ok |
23:33.05 | xSmurf | say I can ask an admin to add a manager for an API, but not to add custom stuff in the dialplan |
23:33.32 | xSmurf | doesn't matter anywayI pretty much have my answer, it that case that would be a no go |
23:35.16 | [TK]D-Fender | xSmurf: If you have access to the API you could add your own support dialplan regardless of them |
23:35.33 | xSmurf | true |
23:35.46 | xSmurf | well |
23:35.58 | xSmurf | depending on the permissions of the user |
23:37.12 | [TK]D-Fender | xSmurf: If your environment is hostile, don't bother asking us "how" when you are being faought against. These restraints are ARTIFICIAL. |
23:37.32 | [TK]D-Fender | xSmurf: ":p |
23:37.57 | xSmurf | so if someone is in a restricted environment they shouldn't attempt to do anything with asterisk? makes sense |
23:38.04 | xSmurf | ;) |
23:40.11 | root52 | hahah |
23:40.35 | root52 | sorry wrong window ;-) |
23:40.50 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
23:41.29 | [TK]D-Fender | xSmurf: No, asking how and then saying. "yeah that'll work, but I'm not allowed" makes it sould like telling you how is a waste of time... |
23:41.36 | [TK]D-Fender | xSmurf: But you knew that already... |
23:41.58 | xSmurf | no it's not a waist of time, at least I learnt how it's done |
23:42.30 | xSmurf | mentioning just allows me to express that although this is a solution, if there are others I'd be interested in knowing them |
23:42.34 | [TK]D-Fender | oh well |
23:43.31 | xSmurf | indeed |
23:50.01 | *** join/#asterisk cnu (~cnu@the.ultimate.lamer.la) |
23:52.11 | root52 | FWIW the problem I mentioned above has cleared itself up. all is well after i left the line from the telco unplugged for about two min. Odd but i'll take it. Perhaps the telco's switch got "confused" about what state the line was in and it just needed a min or two to "reset". |
23:53.04 | *** part/#asterisk Pegasus_RPG (~chatzilla@p4FF90751.dip.t-dialin.net) |
23:55.42 | fenrus | it's not unheard of that sdh over different kinds of sdsl etc takes a minute to synchronize |
23:56.18 | fenrus | and just pure sdh/pdh aswell, have been waiting for minutes for loops in the sdh to appear in my routers when the transmission people have set them. |