| 00:11.41 | navaismo | any ideas where to look why the HPEC codec crahs the whole server when i try to call |
| 00:12.08 | WIMPy | backtrace? |
| 00:12.16 | navaismo | s/codec/cancellator/ |
| 00:12.25 | WIMPy | Or do you mean the kernel? |
| 00:12.38 | navaismo | the whole server crash |
| 00:12.47 | navaismo | until i reboot |
| 00:13.13 | WIMPy | Thats ugly. Is it completely dead? Do you have a monitor? |
| 00:13.14 | Greenlight | Faulty card, or motherboard would be my guess |
| 00:13.42 | navaismo | Wildcard AEX410P |
| 00:14.07 | navaismo | dahdi linux complete 2.5 |
| 00:14.29 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681772.dsl.bell.ca) |
| 00:14.35 | navaismo | server HPML110 xeon quad core |
| 00:14.46 | Greenlight | I had a similar issue on a HP recently |
| 00:14.56 | dijib | could someone tell me if this script would do me any good for my SIP filterng needs? |
| 00:14.57 | dijib | http://www.voip-info.org/wiki/view/QoS+Linux |
| 00:15.07 | navaismo | Greenlight: do you sorted it? |
| 00:15.16 | Greenlight | Yea - got new card and new server |
| 00:15.33 | Greenlight | Wasn't sure which one was to blame |
| 00:15.35 | navaismo | i installed the latest version of HPEC-nocona |
| 00:15.37 | WIMPy | dijib: What are your SIP filtering needs? |
| 00:15.49 | dijib | want them priority 0 |
| 00:16.04 | dijib | have other fraffic that needs prioritising |
| 00:16.05 | Greenlight | navaismo: You might try downgrading to DAHDI 2.3/2.4 - This seemed to reduce the frequency of my crashes |
| 00:16.12 | navaismo | Greenlight: no way to do that, another hint? |
| 00:16.13 | dijib | bittorrent =last |
| 00:16.13 | navaismo | ok |
| 00:16.27 | WIMPy | dijib: I assume you want to giv hihest prio to RTP, not to SIP? |
| 00:16.48 | dijib | ok? |
| 00:16.59 | Greenlight | navaismo: It went from crashing withing 5 mins or at boot with HWEC on, to every few days |
| 00:17.57 | navaismo | oops |
| 00:19.10 | WIMPy | dijib: Is Asterisk running on the same machine? |
| 00:19.46 | WIMPy | The easiest way and to be sure to catch all Asterisks traffic is to just take everything from that process. |
| 00:20.00 | dijib | no |
| 00:20.22 | dijib | iptables/tc & asterisk are on two seperate systems |
| 00:21.03 | WIMPy | Then you could filter them by source IP. Unless you have other traffic from the Asterisk box that should not be prioritised. |
| 00:29.19 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 00:31.10 | kaushal | ~book |
| 00:31.10 | infobot | methinks thebook is Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or ~buybook |
| 00:31.59 | *** join/#asterisk radic (~radic@dslb-094-216-235-235.pools.arcor-ip.net) |
| 00:40.14 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 00:43.47 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 01:00.50 | SeRi | dijib, you till in? |
| 01:03.07 | SeRi | dijib, you have support for the script on your openwrt? |
| 01:03.43 | *** join/#asterisk Lifers (3ba787bb@gateway/web/freenode/ip.59.167.135.187) |
| 01:05.08 | Lifers | Hi, I have a client using Asteriks managed by ThirdLane PBX Manager. In many of the options (ie a Hunt List transfering a call to a Queue) it says "Queue command options". |
| 01:05.53 | Lifers | The (bad) ThridLane documentation says to check the Asteriks documentation for more information, but I can't find Queue command options anywhere in the docs |
| 01:06.00 | Lifers | Anyone got some pointers on where to look? |
| 01:08.21 | Greenlight | core show application Queue |
| 01:10.30 | Lifers | Unfortunately I don't have cli access to the asteriks, as a client I only have access to the PBX Manager GUI. |
| 01:10.46 | Greenlight | http://www.lmgtfy.com/?q=asterisk%20queue%20application |
| 01:13.24 | Lifers | finding the exact phrase was (not knowing asteriks). Thanks found exactly what I needed. Now I see what the 'r' was doing. |
| 01:14.10 | Greenlight | No probs, glad you found it ;) |
| 01:15.45 | Greenlight | Also, you'll need to be aware that some of the more obscure options are not in all versions of asterisk |
| 01:18.37 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 01:20.21 | Lifers | thanks. PBX Manager already removes so much functionality so I'm trying to keep it simple (although the customer wants it complicated) |
| 01:20.41 | Greenlight | heh customers always do ;) |
| 01:23.09 | *** join/#asterisk LiuYan1 (~liu.yan@211.154.128.171) |
| 01:27.51 | WIMPy | The fact that it says "command" while we have been calling them "applications" for several years now makes me anxious. |
| 01:36.09 | Lifers | OK, next project: As above, Asterisk PBX managed with ThirdLane PBX Manager with Cisco SPA504G phones. The phones are not managed by PBX Manager. Is there a way of the Phones pulling a Dynamic Directory out of Asterisk/PBX Manager for staff to browse to dial an extension or transfer etc? |
| 01:37.58 | *** join/#asterisk bintut (~bintut@cm61.sigma15.maxonline.com.sg) |
| 01:40.23 | LiuYan | after I logged in AsteriskGUI, the web page keeps reloading, and the dialplan keeps reloading too, why? http://www.fpaste.org/pAW4/ |
| 01:40.36 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
| 01:40.47 | LostyJai | trying to setup conferencing on our asterisk |
| 01:41.03 | LostyJai | http://pastebin.com/UxzEsdk2 |
| 01:41.09 | LostyJai | keeps saying pin is invalid, any ideas? |
| 01:43.53 | WIMPy | Maybe the PIN is invalid? |
| 01:44.21 | *** part/#asterisk shadowapex (~William@adsl-70-232-210-174.dsl.irvnca.sbcglobal.net) |
| 01:44.22 | WIMPy | what does your meetme.conf say about that? |
| 01:44.45 | LostyJai | it's int he pastebin |
| 01:44.47 | LostyJai | at the very to |
| 01:44.49 | LostyJai | top |
| 01:47.30 | *** join/#asterisk bintut_ (~bintut@111.65.28.33) |
| 01:48.39 | WIMPy | Ok, doesn't look too bad, except for the parameters, but I hope that's just the printout hicckup I came across earlier. |
| 01:48.49 | LostyJai | ? |
| 01:51.06 | LiuYan | LostyJai: i only see "conf-invalid" at line #46, not "pin-invalid" |
| 01:51.21 | WIMPy | Hmm it says conf-invalid. Doesn't that mean invalid conference room? At least there's a conf-invalidpin. |
| 01:52.01 | LostyJai | so what does that mean? |
| 01:53.07 | LostyJai | thsi is what my config looks like |
| 01:53.08 | LostyJai | http://pastebin.com/7pkZ6Td4 |
| 01:53.51 | dijib | SeRi, what do you mean? |
| 01:54.00 | dijib | and im allways tillin' |
| 01:56.00 | WIMPy | Ugh. Nasty. 'core restart when convenient' took some time to exit, but didn't restart. That's a new one. |
| 01:57.33 | LiuYan | LostyJai: the conf #900 password is '1111' in config file, but you entered '0000' |
| 01:59.14 | LostyJai | i tried both |
| 01:59.15 | LostyJai | but yeh... |
| 01:59.57 | LostyJai | http://pastebin.com/9z3rJawe |
| 02:06.09 | p3nguin | For my traffic shaper, I've allocated 35% to IAX2, 35% to RTP, and 25% to SIP. Everything else can squeeze into the remaining 5%. |
| 02:06.58 | WIMPy | Why do you reserve anything? |
| 02:07.31 | p3nguin | There's other stuff besides just those three things that need internet access. |
| 02:07.35 | p3nguin | DNS, for one. |
| 02:07.41 | p3nguin | SSH, for two. |
| 02:08.21 | WIMPy | Ok, but why can't they use all remaining bandwidth? |
| 02:09.21 | p3nguin | Like if RTP is idling, let SSH take up part of that 35%? |
| 02:09.54 | WIMPy | yes |
| 02:10.49 | WIMPy | I don;t reserve bandwidth. I just make sure that IAX and RTP get sent first. |
| 02:11.37 | WIMPy | Anything else will only waste bandwidth. |
| 02:13.01 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 02:13.08 | LostyJai | so.. no one knows? |
| 02:13.28 | LostyJai | to include a file in the config, do you need a "#" before include? |
| 02:13.54 | p3nguin | lostyjai: Yes. #Include newfile |
| 02:14.19 | LostyJai | that's what i got... |
| 02:14.20 | p3nguin | wimpy: I have no idea how the Vyatta traffic shaper works. I just know how to configure it. |
| 02:14.24 | LostyJai | yay.... |
| 02:14.54 | LostyJai | got knows why the fk it's still not working |
| 02:14.55 | LostyJai | grrrrr |
| 02:15.25 | p3nguin | The traffic shaper can set bandwidth by percentage or by rate. |
| 02:15.28 | *** join/#asterisk jaminja (~jaminja@unaffiliated/jaminja) |
| 02:16.11 | WIMPy | I don't that thing at all, but reserving bandwidth seems a very bad idea. |
| 02:16.36 | WIMPy | If I don't have any RTP/IAX traffic, I want to be able to use the full bandwidth. |
| 02:16.44 | p3nguin | I understand. |
| 02:17.02 | p3nguin | Traffic shaping isn't exactly the same as QoS, though. |
| 02:18.01 | WIMPy | It can be more, yes. |
| 02:19.24 | WIMPy | And you should make sure your transmit queue is big enough. |
| 02:20.55 | p3nguin | I guess I need to test to see if I am limited to that 5% if there is no other traffic. I think I am. |
| 02:22.42 | LostyJai | so.... no one? |
| 02:25.21 | p3nguin | What's the problem, now? I answered your question. |
| 02:25.32 | LostyJai | keeps saying conference pin is invalid |
| 02:25.36 | LostyJai | even if i entered it correctly |
| 02:25.53 | p3nguin | Show me the dial plan. |
| 02:26.25 | LostyJai | which config file is that? |
| 02:26.39 | p3nguin | extensions.conf |
| 02:27.02 | p3nguin | I'm interested in the extension you have configured for the conference. |
| 02:27.03 | LostyJai | any particular line you're looking for? |
| 02:29.35 | *** join/#asterisk adolfomaltez (~taro@190.62.206.66) |
| 02:38.28 | dijib | the conference. |
| 02:38.53 | LostyJai | nothing about conference in there |
| 02:38.57 | LostyJai | atleast not from what i can see |
| 02:39.15 | dijib | and i need shaping happening, and i would be interested in seeing your Vyolla or whatever its called's conf files |
| 02:40.57 | p3nguin | lostyjai: How do you suspect a conference is being used if there is no conference? If you can't help me help you, we're not going to get anywhere. |
| 02:41.05 | LiuYan | FreePBX could add *_additional.conf files and included in main config files, conference config may exist in _additional file |
| 02:41.21 | p3nguin | We don't support FreePBX here. |
| 02:41.36 | p3nguin | So there's no reason anyone would be asking about any FreePBX crap here. |
| 02:41.37 | LostyJai | "exten => ConferencePhone,1,Goto(from-internal,616,1)" |
| 02:41.39 | LostyJai | taht is all i can see |
| 02:41.43 | WIMPy | dijib: Read the LARTC. It's good. |
| 02:41.55 | SeRi | waz up guys |
| 02:42.04 | p3nguin | About to head home. |
| 02:42.10 | SeRi | you working p3nguin ? |
| 02:42.14 | p3nguin | I feel like crap. |
| 02:42.19 | SeRi | that sucks |
| 02:42.21 | p3nguin | I'm getting nothing accomplished. |
| 02:42.23 | SeRi | sick? |
| 02:42.27 | SeRi | o damn |
| 02:42.28 | LostyJai | dw guys, it's fine |
| 02:42.31 | dijib | sleeep! |
| 02:42.32 | LostyJai | i'll figure something out |
| 02:42.36 | SeRi | one of those days p3nguin :( |
| 02:42.50 | SeRi | dijib, you in the conf? |
| 02:42.55 | dijib | nope.jpg |
| 02:43.23 | p3nguin | I was having really bad abdominal cramping earlier. At least that finally went away, but I'm still having pains around my right side under the rib cage. Maybe my liver is broken. |
| 02:43.29 | dijib | and i changed my ip again |
| 02:43.50 | SeRi | dijib, I am talking about yours :) |
| 02:43.59 | SeRi | p3nguin, d00d that sounds serious |
| 02:44.28 | p3nguin | I had my lab work done yesterday, but the doc's office didn't call today. |
| 02:44.32 | dijib | p3nguin, stop eating denny's |
| 02:44.38 | dijib | make your food from scratch |
| 02:44.52 | LiuYan | LostyJai: the 2 pastes seems different, the 1st one parsed meetme_additional.conf, the 2nd does not |
| 02:44.54 | dijib | itll take a few days im sure |
| 02:44.55 | p3nguin | They either didn't get it processed or they didn't see anything serious enough to call about. |
| 02:44.57 | SeRi | p3nguin, lab usually takes a few days |
| 02:45.22 | dijib | like a week |
| 02:45.37 | dijib | maybe its intestinitis |
| 02:45.40 | p3nguin | My old doc always sent me to the hospital's lab and they had the results next day. This doc has his own lab, and I've never used it before. |
| 02:45.56 | LostyJai | it's ok LiuYan, don't worry about it |
| 02:46.56 | p3nguin | The phlebotomist said it might be done by "tomorrow," but she wasn't certain. |
| 02:47.05 | p3nguin | I guess it wasn't done. |
| 02:47.55 | WIMPy | Nothing obvious can be the gallbladder. |
| 02:48.01 | p3nguin | It's gone. |
| 02:48.13 | p3nguin | Haven't had one of those in 17 years. |
| 02:48.17 | SeRi | p3nguin, hang in there you will be fine :) |
| 02:48.18 | WIMPy | That makes it unlikey then. |
| 02:48.26 | dijib | you eat lots of greasy foods/? |
| 02:48.32 | p3nguin | Not really. |
| 02:48.53 | dijib | thats what causes gull bladders to go |
| 02:48.55 | p3nguin | I eat a lot of meat, though. I'm sure to end up with colon cancer. |
| 02:49.12 | dijib | had family with that |
| 02:49.22 | p3nguin | same here. |
| 02:49.46 | WIMPy | gets hungry |
| 02:49.46 | dijib | have to balance the veg/meat/breads |
| 02:49.57 | p3nguin | Heredity + lifestyle = certain death |
| 02:50.04 | dijib | i need a smoke |
| 02:50.07 | *** join/#asterisk bintut_ (~bintut@111.65.28.33) |
| 02:50.21 | p3nguin | My lungs are in tip-top shape! |
| 02:50.29 | dijib | are they for sale? |
| 02:50.39 | p3nguin | Not this week. |
| 02:50.45 | p3nguin | Get back to me after my labs come back. |
| 02:51.51 | p3nguin | Ack, I better hurry or I'm going to miss that horror story show that's coming on. |
| 02:53.13 | dijib | didnt get that |
| 02:53.33 | dijib | oh the lungs |
| 02:53.33 | dijib | should you be getting home? |
| 02:53.33 | SeRi | p3nguin, I hope you feel better |
| 02:53.45 | dijib | you will.. immune system, lots of water |
| 02:56.07 | SeRi | p3nguin, lol |
| 02:56.14 | SeRi | dijib, you still here? |
| 02:56.18 | dijib | yep |
| 02:56.28 | dijib | practising my sleep |
| 02:56.28 | SeRi | did you implemented your script? |
| 02:56.31 | dijib | nope |
| 02:56.35 | dijib | reading.... |
| 02:56.39 | dijib | not on topic |
| 02:57.08 | SeRi | where were you going to use that script? |
| 02:57.21 | dijib | on my router |
| 02:57.24 | dijib | wrt |
| 02:57.37 | dijib | i also need to make backup systems to both router and asterisk |
| 02:57.48 | dijib | but thats another venture |
| 02:58.53 | SeRi | does your router support what script is doing? I didnt look at it but implementing traffic shapping is not simple... |
| 02:59.10 | SeRi | s/what /what the/ |
| 02:59.21 | SeRi | damn it lol |
| 03:00.20 | SeRi | dijib, ^^ |
| 03:00.55 | WIMPy | Ah, it's not that bad. The hard thing is to make up your mind what you want to send in what order. |
| 03:01.03 | dijib | iptables & traffic control (tc) |
| 03:01.15 | dijib | thats what the script was calling for |
| 03:01.19 | WIMPy | And how to identify it. |
| 03:01.55 | WIMPy | iptables has tons of optional stuff. |
| 03:03.36 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 03:04.13 | SeRi | WIMPy, it can be hell if you dont know what you are doing.... thats all I am saying :) |
| 03:04.45 | WIMPy | That's true for round about everything. |
| 03:04.50 | dijib | thats the boat im in mon |
| 03:05.25 | SeRi | WIMPy, agreed. |
| 03:06.22 | WIMPy | I found it easiest to only use iptables marks as queueing input. |
| 03:06.30 | dijib | http://www.youtube.com/watch?v=lvq2Geu7At4 |
| 03:06.52 | dijib | you have a working script? |
| 03:07.00 | WIMPy | Makes the whole thing very flexible and two relatively easy parts. |
| 03:07.29 | WIMPy | Yes, but that's for Asterisk and the router being the same machine. |
| 03:07.51 | dijib | well maybe i just need to do that |
| 03:07.56 | dijib | oh but then wireless |
| 03:07.59 | dijib | nope |
| 03:08.01 | WIMPy | And it's a little more generic. I used it before I started VOIP. |
| 03:08.11 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-chgbferqxavsmfmh) |
| 03:08.32 | p3nguin | I identify IAX2 and SIP traffic by port and RTP by DSCP. |
| 03:09.31 | WIMPy | I don't trust any IP headers. Anyone can set them. |
| 03:11.52 | WIMPy | But TC already did a lot for me before VOIP. Just simpe rules like small packets first, big packets last can make things a lot smoother. |
| 03:14.00 | WIMPy | Sending out ACKs at high prio will boost your downstream. And fast DNS can increase responsiveness. |
| 03:14.08 | p3nguin | So I learned that I can't figure out how to dial SIP URIs in siphon, and that I can't dial SIP URIs with a + in them in iSip. |
| 03:14.43 | WIMPy | Make sure you don't generelly prioritise SSH as SCP uses the same port, so take the packet length in again. |
| 03:14.51 | dijib | WIMPy, i would like to see that script if your ready to share. |
| 03:15.21 | dijib | got installeous? |
| 03:15.23 | dijib | i see |
| 03:15.28 | WIMPy | I think I should tidy that a little. Let me see... |
| 03:15.28 | dijib | hey brb5min smoke break |
| 03:15.36 | SeRi | p3nguin, you like siphon? |
| 03:15.50 | p3nguin | No, but I have installous. |
| 03:16.06 | p3nguin | Other than not being able to dial a SIP URI, it's mostly good. |
| 03:16.18 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 03:16.24 | SeRi | p3nguin, I see. |
| 03:18.51 | SeRi | dijib, Learn first what the script is doing and what are the functions are before you move it in place. |
| 03:19.34 | SeRi | s/what are/what/ |
| 03:19.42 | WIMPy | dijib: http://wimpy.yeti.dk/pastebin |
| 03:20.05 | WIMPy | Start that with device and available BW as parameters |
| 03:26.17 | WIMPy | With ADSL the available BW should about 90% of your nominal BW. But unfortunatly the real BW is not easily predictable. |
| 03:27.03 | WIMPy | florz had somethign that would calculate correct BW, but I never found the time to try and mix that with TC :-( |
| 03:27.28 | florz | you are not supposed to mix it ;-) |
| 03:27.54 | WIMPy | I know, but that would be the ideal solution. |
| 03:28.05 | WIMPy | Good morning BTW :-) |
| 03:28.31 | florz | good morning ;-) |
| 03:28.39 | florz | well, actually I have no clue what kind of mixing you are talking about?! |
| 03:29.43 | WIMPy | Your patch to tun to calculate the correct BW for ADSL. |
| 03:29.48 | *** join/#asterisk yaboo (~jsirucka@b2296.static.pacific.net.au) |
| 03:30.00 | WIMPy | Or which way did that take? |
| 03:30.32 | florz | ahh, you mean the stuff that enables mshaper to shape by netfilter marking? |
| 03:31.04 | WIMPy | Hmm, probably, |
| 03:31.31 | florz | well, yeah, unfortunately works for upstream only anyhow ... |
| 03:31.39 | WIMPy | But you did something to calculate the overhead that would get added at the ATM layer, didn't you? |
| 03:31.46 | WIMPy | Sure. |
| 03:31.55 | florz | that's done by mshaper, yes |
| 03:32.04 | WIMPy | No way to control what you receive. |
| 03:32.10 | florz | well, indirectly |
| 03:32.33 | WIMPy | Only in a very limited way. |
| 03:32.51 | florz | essentially what meanwhile has become somewhat well-known under the buzzword "buffer bloat" ;-) |
| 03:32.56 | WIMPy | Ok, so it's (part of) mshaper that would be ideal for mixing with tc. |
| 03:34.31 | florz | that is: if you reduce bandwidth so as to create an artificial bottleneck on your side, you can influence the typical tcp sender to keep much fewer packets in flight |
| 03:35.06 | WIMPy | Yes, but it only works with tcp and even that only in a limited way. |
| 03:35.39 | florz | well, it should work with any protocol that expects to work reliably over the internet |
| 03:36.28 | florz | if you want to make bulk data transfers at a high data rate you will have to have some probing mechanism that detects congestion in the network |
| 03:36.55 | WIMPy | The main point is that it will not be of any help at all against deliberate congestions. |
| 03:37.09 | florz | no, that of course not |
| 03:37.22 | WIMPy | Yes, but that's usually tcp. |
| 03:38.43 | florz | on the other hand, a shaper at the other end cannot really do that either, as you cannot authenticate the sender there |
| 03:39.12 | WIMPy | Correct. |
| 03:39.27 | SeRi | just finish watching "Limitless" |
| 03:39.27 | WIMPy | But you may be able to predefine certain limits. |
| 03:39.49 | florz | really, the major downside is that you have to limit to a bandwidth well below the line's capacity for this approach to work well |
| 03:40.14 | WIMPy | That's unfortunatly very true. |
| 03:40.37 | florz | other than that it works surprisingly well ;-) |
| 03:40.42 | WIMPy | But it does help. |
| 03:40.56 | SeRi | bah.... I let an appliance do all the math :P Traffic Shapper on pfsense allocates bandwitch base on protocols/packet/service/etc..... it does an awesome job :P |
| 03:41.07 | WIMPy | Unless you don't have the BW you expect to have it's working very well. |
| 03:41.28 | SeRi | s/bandwitch/bandwidth |
| 03:41.44 | WIMPy | SeRi: On ADSL or on something more predictable? |
| 03:42.41 | SeRi | WIMPy, I use cable. BUT it does not matter on pfsense because it does not per say "allocated" fix rates |
| 03:43.07 | SeRi | YOu set you down/up and leaves it open until it gets trigger by a rule |
| 03:43.25 | WIMPy | So how does it find out how much it can send? |
| 03:43.28 | SeRi | Than it starts allocation priority's |
| 03:43.41 | p3nguin | priorities |
| 03:43.53 | florz | and then |
| 03:44.02 | SeRi | thnaks p3nguin |
| 03:44.05 | florz | and even then I fail to parse it |
| 03:44.25 | WIMPy | What use are priorities if you don;t know if your packets get discarded at the next hop? |
| 03:44.44 | florz | or even wors: form a queue :-) |
| 03:44.45 | florz | +e |
| 03:44.54 | dijib | thanks WIMPy |
| 03:45.06 | SeRi | WIMPy, It calculates the need it bandwidth for the protocol. or it can also use fix if need it |
| 03:45.19 | SeRi | WIMPy, it does not matter. the next hope is out side my control |
| 03:45.29 | SeRi | I only control what i can inside my lan |
| 03:45.34 | SeRi | hop* |
| 03:45.51 | WIMPy | No, that's exactely the point. You have to make sure not to exceed the bandwidth to avoid dropped packets. |
| 03:45.56 | florz | SeRi: do I read that correctly that you are reciting some marketing material and don't have the slightest clue how packet scheduling works? |
| 03:45.56 | SeRi | fuck my spelling is worst than ever |
| 03:46.29 | SeRi | florz, marketing no. did I miss understood yes. |
| 03:46.39 | SeRi | WIMPy, I see what you are saying nor |
| 03:46.43 | SeRi | now* |
| 03:46.59 | SeRi | So in ADSL it fluctuates? |
| 03:47.12 | WIMPy | technically no |
| 03:47.41 | *** join/#asterisk master_of_master (~master_of@p57B5556E.dip.t-dialin.net) |
| 03:47.46 | WIMPy | But as te Ethernet frames get tunneled into ATm frames there is some slack. |
| 03:48.20 | florz | well, in principle the problem exists with any transport layer, with the usual ADSLs it's just particularly bad ;-) |
| 03:48.30 | SeRi | ah. ok I see. |
| 03:48.35 | WIMPy | So the available BW depends on the packets sizes and how well they translate. |
| 03:48.42 | SeRi | understood |
| 03:49.00 | SeRi | understands now. |
| 03:49.08 | SeRi | shit no more drinking for me |
| 03:49.22 | florz | The problem ist just that what you buy is kindof the low-level bandwidth of the line. But the packet you are sending is put into a ton of encapsulations stack on one another, which each increase the size of the packet a bit. |
| 03:49.37 | WIMPy | Maybe a PCI modem would be a good idea. But I'm not sure if you get exact results if the ATM layer is done by Linux as well. |
| 03:50.04 | florz | the problem is that it's not a constant factor by which the packets get larger, but rather (more or less) a constant number of bytes |
| 03:50.06 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:50.13 | WIMPy | Yes, that's going to be a long list for VOIP. |
| 03:50.32 | florz | which makes small packets easily twice their original size, while large ones get only a few percent larger |
| 03:50.33 | WIMPy | Not even that. |
| 03:50.51 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:51.34 | WIMPy | RTP over UDP over IP over PPP over Ethernet over ATM over DSL. |
| 03:51.45 | SeRi | So how do you guys manage to over come this issues? Is this in general with ADSL? |
| 03:51.51 | florz | which means that your average traffic shaper has no clue how much bandwidth each packet actually uses at the layer at which your line is speced |
| 03:52.04 | WIMPy | Trouble is that you don;t "see" what happens at te ATm layer. |
| 03:52.15 | WIMPy | By wastin BW. |
| 03:52.46 | florz | well, depending on the traffic shaper, it does not even have a clue of the PPPoE |
| 03:52.50 | florz | or PPP |
| 03:53.05 | WIMPy | Yes the limit is done on the ATM layer but you only ever see the Ethernet layer if at all. |
| 03:53.06 | florz | and the "ATM" part actually is multiple layers of encapsulation |
| 03:53.25 | SeRi | dijib, you in? |
| 03:53.32 | dijib | conference or bed... |
| 03:53.37 | dijib | bed yes. conference no |
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| 03:53.56 | SeRi | can you pb your dial plan? |
| 03:54.05 | SeRi | I wanted to see your meetme portion |
| 03:54.19 | florz | to quote from the mshaper web page: "The IP packet (which is what we actually want to transfer) is first encapsulated in PPP (adds 2 bytes), then that's wrapped inside PPPoE (another 6 bytes), which in turn is put into an Ethernet frame (14 bytes, or even 18 bytes if the Ethernet checksum trailer is also transferred over the DSL), after which a layer of RFC 2684 LLC encapsulation is applied (10 bytes), the result of which then is packaged using AAL5 into |
| 03:54.20 | WIMPy | Yes, indeed there is stuff in between the ethernet and the ATM. I completely forgot about that |
| 03:54.38 | dijib | meetme.. k |
| 03:55.37 | WIMPy | pppoe is an extra layer between ppp and Ethernet? |
| 03:55.51 | florz | so that makes at least 45 bytes of overhead beyond the IP packet size |
| 03:55.53 | dijib | http://pastebin.com/eSALue18 |
| 03:55.56 | WIMPy | I thought that was just a ppp packet type on the Ethernet? |
| 03:56.18 | WIMPy | But I have to admit that I never really cared. |
| 03:56.18 | SeRi | dijib, thanks |
| 03:56.26 | florz | WIMPy: nope, that's how you can have multiple PPPoE sessions and all that |
| 03:56.54 | WIMPy | Ok, that makes sense. |
| 03:56.56 | SeRi | dijib, I dont have to play with meetme.conf right? |
| 03:56.58 | florz | WIMPy: it has its own packet type, but it also adds a header to each packet |
| 03:57.00 | SeRi | p3nguin, you in? |
| 03:57.19 | florz | (well, two packet types actually, one for the session management, and one for the payload packet) |
| 03:57.20 | SeRi | or better said are you avail? |
| 03:57.22 | dijib | good question. i dont think so. |
| 03:57.25 | dijib | let me check |
| 03:57.43 | SeRi | dijib, ok. looks like p3nguin is afk. |
| 03:57.49 | SeRi | I want to switch to meetme |
| 03:57.55 | dijib | conf => 1010 |
| 03:58.05 | SeRi | under meetme? |
| 03:58.12 | dijib | ya but im not sure if its needed |
| 03:58.19 | SeRi | s/meetme/meetme.conf |
| 03:58.26 | SeRi | ok |
| 03:58.27 | WIMPy | At least the pppoe part is optional. |
| 03:58.36 | SeRi | any general settings that I need to know off> |
| 03:58.37 | SeRi | ? |
| 03:58.59 | dijib | http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe |
| 03:59.08 | p3nguin | ? |
| 03:59.12 | florz | well, technically, PPP would be optional as well |
| 03:59.22 | dijib | 1acd |
| 03:59.24 | p3nguin | You don't need to touch meetme.conf to use meetme. |
| 03:59.27 | WIMPy | Sure. I meant both. |
| 03:59.37 | dijib | SeRi, the d option is dynamic |
| 03:59.42 | dijib | just starts the conf i guess |
| 03:59.45 | SeRi | I see |
| 03:59.46 | florz | and the ethernet also--there is IP transport defined over AAL5 directly :-) |
| 03:59.58 | SeRi | p3nguin, ok. so leave meetme as is. |
| 04:00.38 | florz | and while you are at it, you could rip out the ATM as well ;-) |
| 04:01.15 | WIMPy | florz: I love how some years back they were all whining about wased BW because of ATM. Now the use the whole lot beginning from RTP with all unneccessary parts included. Absolutely awesome :-( |
| 04:01.30 | dijib | yes SeRi leave it |
| 04:01.39 | WIMPy | Probably but ATM is the least of the problem. |
| 04:01.53 | WIMPy | ATM would be a good solution for NGN. |
| 04:02.08 | WIMPy | Or actually is (elsewhere). |
| 04:02.27 | WIMPy | But the shit they're doing here is just plain braindead. |
| 04:02.53 | dijib | where? |
| 04:02.55 | florz | WIMPy: well, those 5 bytes overhead per 48 bytes payload is some 9 % bandwidth loss ... :-) |
| 04:02.56 | dijib | pppoe? |
| 04:03.03 | WIMPy | de |
| 04:03.22 | SeRi | p3nguin, can I use mixmonitor with meetme? |
| 04:03.26 | dijib | yes |
| 04:03.31 | p3nguin | You could, but you shouldn't. |
| 04:03.35 | dijib | thats the r option |
| 04:03.36 | p3nguin | Use MeetMe's record option. |
| 04:03.41 | WIMPy | Yes, 9% isn't ideal, but with the whole set of tunnels possible easily reach 50% now. |
| 04:03.44 | florz | and with current stuff they at least seem to be moving away from ATM, AFAIK |
| 04:03.53 | SeRi | ok |
| 04:03.53 | WIMPy | +we |
| 04:04.12 | WIMPy | Huh? Where? |
| 04:04.16 | dijib | k guiess im outta here |
| 04:04.18 | dijib | night time |
| 04:04.25 | dijib | im going grey |
| 04:04.29 | WIMPy | They'd be better of skipping everything else. |
| 04:04.36 | WIMPy | +f |
| 04:04.43 | florz | but having ATM underneath and then doing voip through a shared VC with all the other PPPoE is kindof idiotic, obivously ... |
| 04:05.07 | florz | WIMPy: I think the VDSL stuff doesn't do ATM anymore!? |
| 04:05.24 | WIMPy | Oh, you can be even more idiotic. See O2 (ex Hansenet). |
| 04:05.56 | WIMPy | Not sure. I think there's an Ethernet version. But I'm not sure they use that. |
| 04:06.51 | WIMPy | Anyway they do their VOIP with SIP/RTP over pppoe on a dedicated VC. |
| 04:06.54 | florz | IIRC VDSL uses ethernet, with PPPoE on top of that for "normal IP", and then they also do IP multicast for their IPTV offerings |
| 04:07.22 | WIMPy | Ok, didn't think it was that modern :-) |
| 04:07.48 | WIMPy | They use 4 VS in total. All of them with pppoe on top. |
| 04:07.53 | WIMPy | VCs |
| 04:08.13 | florz | that's ... well, just marginally better than mixing it all in one VC ;-) |
| 04:08.41 | WIMPy | That's wasting as much BW as possible. |
| 04:09.05 | WIMPy | And offering the worst possible service. But they all do that here :-( |
| 04:09.07 | florz | no |
| 04:09.29 | florz | for that they would then have to emulate ATM on top of some ethernet transport |
| 04:10.05 | florz | yeah, it's horrible, indeed |
| 04:10.10 | WIMPy | What? ATM over Ethernet over ATM? |
| 04:10.24 | WIMPy | Oh, sorry, you meant for VDSL? |
| 04:11.07 | florz | anything that used ATM emulation on top of ethernet would qualify, I guess ;-) |
| 04:11.49 | WIMPy | I think the ADSL situation is more than qualified. |
| 04:12.07 | florz | *g* |
| 04:12.40 | SeRi | p3nguin, feeling better? up for a beer? :) |
| 04:12.48 | WIMPy | I'd happily give them any prize. |
| 04:13.05 | p3nguin | seri: I still have the pain, but I may have a beer later. |
| 04:13.08 | WIMPy | But for creating the best example of how not to do it. |
| 04:13.32 | SeRi | p3nguin, I am in like on my 6th cup of whiskey. |
| 04:13.47 | SeRi | a beer sounds good about now. |
| 04:14.22 | florz | actually I think all those hosters not grasping ipv6 should get their prizes first =:-) |
| 04:14.59 | WIMPy | Nah, that's definitely 2nd place for me. |
| 04:15.30 | florz | for a simple reason: if they continue like that, they's destroy much of the potential of ipv6 |
| 04:15.34 | WIMPy | But it's also very bad. |
| 04:16.05 | florz | the protocol stacking idiocy just wastes some bandwidth, but doesn't hurt much otherwise (well, except for the brain of anyone thinking about it too much ;-) |
| 04:16.44 | WIMPy | It _does_ hurt. Every time. |
| 04:16.59 | SeRi | p3nguin, man that sounds no good. having that pain must suck bad.... :( |
| 04:17.22 | p3nguin | At least it's not as bad as it has been in the past. |
| 04:17.29 | p3nguin | The last time, I was hospitalized for 5 days. |
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| 04:19.23 | SeRi | damn d00d! Well I sympathize.... I been in the hospital before... last time I was there I spent 2 weeks... Hospitals sucks! |
| 04:19.47 | p3nguin | At least they had wifi for me. |
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| 04:20.16 | *** part/#asterisk hdiogenes (~humberto@201.76.153.10.wimax.digizap.com.br) |
| 04:21.58 | SeRi | same here.... Just couldnt do much of nothing without having a damn breathing tube down my throat |
| 04:22.03 | SeRi | Not fun :/ |
| 04:22.28 | p3nguin | I didn't have that... just IVs hanging out my arm. |
| 04:22.36 | p3nguin | Could still type. |
| 04:22.42 | SeRi | I see. |
| 04:23.46 | SeRi | I was out for most f the times till my lungs recoup... :) I learn my lesson though.... Since I have not missed one day of meds not even one lol |
| 04:25.16 | p3nguin | What kind of med do you take? |
| 04:26.40 | SeRi | Permanent steroids, Xyzal, Prilosec, mor and more steroids :( |
| 04:27.40 | SeRi | it beats having to puff on a damn inhaler every 10min |
| 04:29.19 | vinhdizzo | anyone use nortel 1535 for remote sip connection? |
| 04:31.31 | SeRi | p3nguin, well meetme works :) |
| 04:31.36 | SeRi | nice |
| 04:32.02 | SeRi | there is no meetme help? |
| 04:32.22 | p3nguin | What kind of help are you looking for? |
| 04:32.50 | SeRi | I am trying to udnerstand what is doing just like I can see what each letter means in confbridge |
| 04:33.50 | p3nguin | core show application MeetMe |
| 04:33.55 | SeRi | ah |
| 04:33.56 | SeRi | thnaks |
| 04:34.00 | SeRi | thanks |
| 04:34.32 | SeRi | wow I can see that is better implemented than confbridge |
| 04:34.57 | *** part/#asterisk LostyJai (~blah@202.171.190.130) |
| 04:35.36 | SeRi | p3nguin, is there an admin for meetme like in confbridge? I dont see it listed. |
| 04:35.45 | p3nguin | Yes. |
| 04:35.52 | SeRi | ah MeetMeAdmin |
| 04:35.56 | p3nguin | And meetme also has the CLI commands. |
| 04:36.43 | SeRi | cool I am playing with it right now |
| 04:41.26 | vinhdizzo | Can someone help debug this remote sip connection? http://pastebin.com/uVRuVFru |
| 04:41.59 | vinhdizzo | necessary ports are forwarded. i am able to register from my softphone, but not on my nortel 1535 phon |
| 04:42.12 | vinhdizzo | i would appreciate any insights anyone might have |
| 04:42.18 | *** join/#asterisk radic (~radic@dslb-178-002-234-149.pools.arcor-ip.net) |
| 04:44.16 | SeRi | p3nguin, you used anveo before? |
| 04:44.28 | p3nguin | Never heard of it. |
| 04:44.47 | SeRi | http://blog.anveo.com/ |
| 04:44.50 | SeRi | anveo.com |
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| 04:47.45 | vinhdizzo | p3nguin: can u help me debug a remote sip connection? |
| 05:00.13 | [TK]D-Fender | vinhdizzo: Auth is wrong or your phone isn't responding properly to the 401 challenge |
| 05:00.15 | *** join/#asterisk irroot (~gregory@41.52.78.201) |
| 05:01.48 | vinhdizzo | [TK]D-Fender: i think its the former as i am able to connect when the phone is plugged to the local network |
| 05:02.22 | vinhdizzo | any tips on debugging it? |
| 05:02.35 | [TK]D-Fender | vinhdizzo: compare configs on the PBX vs the phone |
| 05:03.12 | vinhdizzo | what do u mean? on phone i only set username, pw, proxy, domain, stun, etc |
| 05:03.21 | vinhdizzo | what configs do i need to compare? |
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| 05:22.55 | SeRi | cricket cricket |
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| 05:41.57 | SeRi | nom nom nom nachos with cheese home made |
| 05:43.15 | p3nguin | ugh |
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| 05:43.44 | p3nguin | I'd beat the neighbor's dog for some nachos with cheese right about now. |
| 05:43.58 | SeRi | lol |
| 05:45.17 | SeRi | I was going to boile budahn or how ever is spelled but nahhhh is late... jalapenos chips with shredded colby jack and sharp chedar will do the job |
| 05:45.58 | SeRi | microwave for like 30sec and you have a heart attack on the spot :) |
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| 05:52.50 | SeRi | p3nguin, you ever used OpenSER? |
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| 06:58.56 | IsUp | morning |
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| 07:33.54 | MariusAgon | Good Morning, Guys. Does penalties works fine with rrmemory strategy? It seems, that anytime call get's into the queue, allways the same agent ring first, if he's busy, calls go further. I'm using Asterisk 1.4.39.1. Thanks. |
| 07:39.22 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:39.25 | schmidts | good morning |
| 07:39.55 | GG111 | Greetings all, I don't seem to be able to bilnd Xfer a call with an analog phone , I'm able to do so only if I put "Tt" in the Dial() of the extension itelft. if I put "DIAL_OPTIONS = Tt " withing the context or within the [GENERAL] in extensions.conf I'm unable to make it work. What am I missing ? |
| 07:42.20 | IsUp | GG111: are you running FreePBX? |
| 07:42.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:44.05 | kaldemar | GG111: your Dial does not have ${DIAL_OPTIONS} in the options part or something writes it over before the Dial. |
| 07:45.10 | IsUp | kaldemar: probably he is using FreePBX, because DIAL_OPTIONS is a global variable in FreePBX as i remember |
| 07:45.28 | kaldemar | GG111: for putting "DIAL_OPTIONS = Tt" in a context, it not valid configuration. variables are not tied to contexts, but channels. only place in extensions.conf for variables is [globals]. |
| 07:46.03 | kaldemar | GG111: which in fact may be one cause for it to not work if you tried putting it in [general], which is the wrong place for global variables. |
| 07:46.48 | GG111 | kaldemar : I see, I'm using Asterisk from source, and yes, I did put the option under "general ", |
| 07:47.22 | kaldemar | move it under [globals] and try again. |
| 07:48.05 | kaldemar | and reload dialplan before dialing also... |
| 07:48.19 | GG111 | kaldemar : my applogies. I did put it under |
| 07:48.47 | GG111 | "general " from the get go, and it did not work, I confused myself and wrote "general " here. |
| 07:49.50 | GG111 | and I know the feature itself works because if I'll put in under the Dial() of the extension itself, it works 100% . |
| 07:50.19 | GG111 | but that only solves the incoming call to that specific channel, and incoming only. |
| 07:55.21 | GG111 | just to clarify, besides setting the right feature key that I want in featurs.conf , all I did is place DIAL_OPTIONS = tT , under "globals" in extensions.conf |
| 07:57.09 | kaldemar | DIAL_OPTIONS is just a variable. you'd need to have the variable in every Dial application in your dial plan. e.g. Dial(Local/foo@bar,30,${DIAL_OPTIONS}). |
| 07:59.03 | GG111 | I see, so there is no " all inclusive " setting for that. unless I use a Macro() to create an extension build. I saw that under the FreePBX extensions.conf , which I used for an example, and I thought that I could do it . |
| 08:00.18 | GG111 | in any case, that will only solve an incoming call, what happens when the phone place an outbound call ? I did try putting "Tt" under the _XNXXXXXXXX for the outgoing channel, but it did not help much.. |
| 08:00.36 | IsUp | GG111: no theres not. you should add that DIAL_OPTIONS variable on every Dial() command. |
| 08:01.40 | kaldemar | GG111: do yourself a favor and don't use freepbx dialplan as an example for anything. |
| 08:02.38 | IsUp | ^^ |
| 08:03.36 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 08:05.27 | GG111 | :( |
| 08:05.33 | GG111 | :) |
| 08:06.50 | GG111 | I see, well, I think I can include this in a Macro that will solve my inbound calls needing to be transferrers, any tips as to how would I apply it on outbound calls ? |
| 08:08.09 | kaldemar | no need for a macro, just put the options where they belong. same goes for outbound calls, use the options in the Dial. if the callee can't transfer, the problem lies elsewhere. for example in DTMF delivery from the callee to asterisk. |
| 08:11.33 | GG111 | OK, so, let me ask further - |
| 08:13.02 | GG111 | incoming call : exten => 1,DIal(SIP/100,25,tT) , all is good, but that will only affect an incoming call. if , exten 100 rings device 100 , that will work great, But, what happens when device 100 place an outbound call ? |
| 08:14.28 | SeRi | p3nguin, you avail? |
| 08:16.04 | kaldemar | GG111: you have another Dial command for outbound calls. put the options in it. |
| 08:16.24 | GG111 | in thory, I could make _XNXNXXXXXX ring an exten, make the exten ring an Outbound trunk by default and that may allow to place a "tT" option over there. |
| 08:16.59 | GG111 | kakdemar : I have a number matching sequne, when I put "tT" over there, it does not work. |
| 08:17.16 | kaldemar | GG111: pastebin a CLI output of a call. |
| 08:18.07 | GG111 | exten = > _XNXNXXXXXX,1,Dial(SIP/${outbount trunk},tT) - does not work |
| 08:18.18 | ChannelZ | you're not dialing anything |
| 08:18.59 | ChannelZ | depending on what ${output trunk} is set to rather. |
| 08:19.04 | GG111 | in that case I need to make it dial in a different way ? |
| 08:19.07 | IsUp | GG111: place an outbound call and pastebin the output |
| 08:19.51 | GG111 | it's set to Dial(${EXTEN}@OUTBOUND-TRUNK) |
| 08:20.02 | ChannelZ | ah |
| 08:20.44 | ChannelZ | that makes only slightly more sense. As is being suggested, pastebin a call |
| 08:21.06 | GG111 | kaldemar : there is little to no info in the CLI of the call that can shed a light on this issue I believe, it only shows the outbound call progress as the most basic setup will. do you still want me to PB it ? |
| 08:21.13 | GG111 | OK' will do |
| 08:21.27 | kaldemar | GG111: your options are in the wrong place. the syntax is Dial(tech/...,timeout,options) |
| 08:21.59 | GG111 | I see, so i need to have two commas ? |
| 08:22.04 | kaldemar | GG111: your tT options are where asterisk expects to find a timeout. two commas, yes. |
| 08:22.20 | GG111 | OK, will try it now, if no go, will pastebin |
| 08:22.43 | IsUp | GG111: exten => 911,1,Dial(SIP/trunkname/911,60,tT) as an example |
| 08:23.30 | ChannelZ | oh.. yeah he forgot the timeout |
| 08:24.16 | kaldemar | GG111: Dial(${EXTEN}@OUTBOUND-TRUNK) also looks invalid, unless your extension really has a channel identifier in it. |
| 08:26.06 | GG111 | kaldemar : mm.. the calls are flowing OK in that way, I have channel defined in sip.conf that has the same name an ${OUTBOUND-TRUNK |
| 08:26.09 | GG111 | } |
| 08:26.23 | GG111 | and it uses that channel and calls are going out OK. |
| 08:26.42 | GG111 | kaldemar :you solved it , the two commas did it. |
| 08:26.56 | GG111 | I'm able to transfer when calling out |
| 08:28.08 | IsUp | GG111: btw, this is not 'channel' if its SIP. probably you mean 'trunk', 'peer', 'provider' at least :p |
| 08:28.09 | kaldemar | GG111: then Dial(${EXTEN}@OUTBOUND-TRUNK) is most likely not what you have in your dialplan. Dial(SIP/${EXTEN}@OUTBOUND-TRUNK) would look more realistic. |
| 08:28.39 | GG111 | did this feature ever work when put in [globals] , in intrigues me because I saw one online discussion of someone that claimd that he got it to work, it was, however , in Asterisk 1.4 and I'm using 1.8.7 |
| 08:28.49 | GG111 | kaldemar : yes, you are right. |
| 08:29.05 | GG111 | in both sentences . |
| 08:30.40 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 08:33.16 | kaldemar | GG111: DIAL_OPTIONS is not a feature of app Dial. it is just another variable. if you have it in globals and app Dial like Dial(tech/..,,${DIAL_OPTIONS}), it will work. |
| 08:34.35 | kaldemar | Dial(tech/..,,${GLOBAL(DIAL_OPTIONS)}) would be even better if you didn't want channel variables to overwrite it. |
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| 08:35.07 | GG111 | oh, I see, so i need to reffer to it from Dial() , and then DIal() will read what ever is noted in [globals] |
| 08:37.14 | IsUp | GG111: exactly |
| 08:38.01 | GG111 | all I was doing is jut placing it under globals, but not referring to it from Dial().. |
| 08:38.18 | GG111 | I guess Asterisk was having a laugh at that time . |
| 08:39.09 | IsUp | app_dial.c:1298 dial_exec_full: Unable to laugh at this time. (cause 0 - Unknown) |
| 08:39.57 | GG111 | :) |
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| 10:42.09 | qakhan | hi all |
| 10:42.29 | qakhan | can anyone tell me how to install free G729 codec |
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| 10:55.38 | kaldemar | qakhan: very unlikely, since G.729 is not free. |
| 10:55.58 | kaldemar | yet. |
| 11:10.20 | qakhan | how much does its license cost? |
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| 11:17.39 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 11:17.42 | _N1x | hi guys |
| 11:18.07 | _N1x | how i can call from my asterisk to siptrunk? |
| 11:19.52 | _N1x | from cli* |
| 11:20.48 | irroot | _N1x originate |
| 11:23.02 | _N1x | irroot: can you show me example? |
| 11:24.59 | kaldemar | core show help channel originate |
| 11:25.53 | kaldemar | qakhan: http://www.digium.com/en/products/g729codec.php |
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| 11:32.24 | _N1x | kaldemar: irroot <tech/data> |
| 11:32.26 | _N1x | what is this field? |
| 11:53.46 | kaldemar | _N1x: where? |
| 11:54.00 | kaldemar | oh, originate.. |
| 11:54.40 | kaldemar | tech as in SIP and data as in the device name from sip.conf. |
| 11:55.04 | kaldemar | if you have [provider] in sip.conf you originate to SIP/provide/number |
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| 12:14.36 | qakhan | hi all |
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| 12:15.57 | qakhan | i want to save custermer info in DB, when he speak his name, number address |
| 12:16.05 | qakhan | is it possible in asterisk? |
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| 13:10.46 | qakhan | if caller speak his name asterisk save his name. is it possible in asterisk???? |
| 13:11.28 | leifmadsen | yes |
| 13:11.46 | leifmadsen | wait, do you mean record the audio, or do speech-to-text? |
| 13:12.06 | leifmadsen | qakhan: because those are two very different things |
| 13:12.34 | *** part/#asterisk hdiogenes (~humberto@201.76.154.190.intranet.digi.com.br) |
| 13:13.10 | qakhan | i want caller callin and speak his name and number and address |
| 13:13.35 | qakhan | his info save in db and my application can fetch his info from db |
| 13:15.17 | irroot | leifmadsen hope i helped bit and not agrivated .... |
| 13:15.27 | leifmadsen | irroot: huh? |
| 13:15.31 | irroot | another day another epic storm .... |
| 13:15.37 | irroot | leifmadsen LDAP |
| 13:17.05 | leifmadsen | qakhan: that requires speech-to-text, and would require you to use a speech-to-text engine (like Sphinx). Yes it is possible. The implementation is probably not trivial though. I've seen some documentation about people using it. Honestly I think you're better off with a code generator, giving people a website, having them enter the code in, then have them enter their data that way. Probably a lot faster and less prone to |
| 13:17.05 | leifmadsen | error. |
| 13:17.11 | leifmadsen | irroot: heh, I haven't tried yet to be honest :) |
| 13:17.14 | leifmadsen | maybe today |
| 13:17.21 | irroot | lol no prob |
| 13:17.59 | irroot | wonder what "HR" will say if i send a techie on the roof to adjust antenna in a lightning storm ..... |
| 13:18.01 | *** part/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:18.54 | qakhan | @leifmadsen u mean to say. i ask to call type his name, number and address through phone |
| 13:18.57 | qakhan | right? |
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| 13:19.26 | leifmadsen | qakhan: no, that's not what I mean |
| 13:19.43 | leifmadsen | but could also be a solution -- have someone actually answer the call and enter the data |
| 13:19.54 | leifmadsen | alternatively, give the caller/customer a code and have them enter data via a website |
| 13:20.12 | leifmadsen | irroot: depends how valuable that techie is |
| 13:20.17 | _N1x | guys , i making calls from my phone to siptrunk but can't receive calls from siptrunk |
| 13:20.28 | leifmadsen | _N1x: not enough information |
| 13:20.30 | _N1x | and can't see incoming calls in sip set debug |
| 13:20.37 | leifmadsen | then it sounds like a firewall problem |
| 13:20.40 | leifmadsen | or a routing problem |
| 13:20.42 | qakhan | @leifmadsen can u plz give me more detail how? |
| 13:20.47 | leifmadsen | or you haven't registered to the end point |
| 13:21.32 | leifmadsen | qakhan: I'm not sure how much more detail I can give... either have someone answer the caller and enter the data for them, give the caller a code and a website so they can enter their own data, or you'll have to look at something like Sphinx and hack together somethign that will do speech-to-text. |
| 13:22.13 | _N1x | leifmadsen: hm , if my asterisk is registered i think i have not access lists problem |
| 13:22.29 | _N1x | i have not special rules for asterisk |
| 13:22.45 | irroot | leifmadsen he is pretty good ... even read your book ... knows asterisk ... but want to see what happens and if he will smoke |
| 13:22.55 | leifmadsen | run nmap from a remote box and see if you can scan port 5060/udp |
| 13:23.08 | leifmadsen | irroot: sounds like a valiant test |
| 13:23.24 | _N1x | leifmadsen: yep it's open |
| 13:25.20 | leifmadsen | then it sounds like the other side isn't sending traffic at your box if you're not seeing it |
| 13:25.43 | leifmadsen | or your routing isn't setup right -- or if you're behind nat, maybe asterisk isn't setup right (externip and localnet settings in sip.conf) |
| 13:25.49 | leifmadsen | throws out wild guesses |
| 13:25.50 | _N1x | leifmadsen: if im calling from my side to siptrunk? |
| 13:25.55 | leifmadsen | drinks some wild turkey |
| 13:26.08 | _N1x | leifmadsen: hm wanna show my sip.conf? |
| 13:26.11 | leifmadsen | _N1x: I thought you said that part worked |
| 13:26.12 | _N1x | yes im behind nat |
| 13:26.21 | leifmadsen | _N1x: you can show it via a pastebin; maybe someone will look at it |
| 13:26.27 | _N1x | yes |
| 13:27.48 | _N1x | http://pastebin.com/TV3zBh0f |
| 13:27.55 | _N1x | catch the rainbow :) |
| 13:28.20 | tuxx- | rainbow - catch the rainbow? xD |
| 13:28.30 | tuxx- | epic song is epic. |
| 13:28.32 | _N1x | tuxx-: yep : d |
| 13:28.41 | tuxx- | dio & blackmore ftw |
| 13:28.42 | tuxx- | =) |
| 13:29.01 | _N1x | yep ronnie & blackmore :) |
| 13:29.17 | _N1x | so what is my problem guys? |
| 13:33.18 | [TK]D-Fender | _N1x, Just showing a SIP config like that is like showing use a brochure for a 2011 Camry ... and asking us why it crashed. |
| 13:33.21 | [TK]D-Fender | :) |
| 13:33.28 | [TK]D-Fender | _N1x, Where's the body? :) |
| 13:34.27 | [TK]D-Fender | _N1x, Also you aren't registering to anything on the outside. Didi you already hand them your IP directly? Otherwise they should not know how to reach you at all. |
| 13:34.50 | _N1x | [TK]D-Fender: i have registered but not advertised here |
| 13:34.51 | _N1x | :) |
| 13:34.54 | _N1x | sec |
| 13:36.10 | _N1x | http://pastebin.com/tgaK2hSm |
| 13:36.28 | _N1x | this is my full sip.conf , except local registered numbers |
| 13:37.23 | [TK]D-Fender | sip.conf doesn't prove all that much. |
| 13:38.14 | [TK]D-Fender | Where is the debug? |
| 13:38.28 | [TK]D-Fender | tuxx-, RIP Ronnie :( |
| 13:38.35 | _N1x | [TK]D-Fender: i already say , i can't see debug when i calling to my number |
| 13:38.45 | _N1x | jup rip ronnie also :) |
| 13:39.05 | [TK]D-Fender | _N1x, I don't see what you are allowing in there. |
| 13:39.16 | _N1x | sec |
| 13:39.27 | [TK]D-Fender | under general. You are masking all pertinent numbers, you are filtering and we can't see if anything you put is sane |
| 13:39.31 | tuxx- | [TK]D-Fender: ;-( |
| 13:39.39 | [TK]D-Fender | _N1x, You are making this process very painful. |
| 13:39.50 | _N1x | http://pastie.org/2914218 |
| 13:40.13 | [TK]D-Fender | Dialplan is irrelevant if you aren't getting SIP traffic at all |
| 13:40.54 | [TK]D-Fender | Show us that you're registered, and what the attempt looks like |
| 13:41.48 | [TK]D-Fender | You also show as "nat=yes" under general, but have no associated settings to match, and if that peer was for your provider, they should be "nat=no", not "yes" as we see |
| 13:41.55 | _N1x | [TK]D-Fender: i already say. i making calls from asterisk. i have registered asterisk to provider , but can't receive and can't see log |
| 13:42.04 | [TK]D-Fender | Show us |
| 13:42.15 | _N1x | [TK]D-Fender: what? in detail.. |
| 13:42.27 | [TK]D-Fender | Show us that you're registered, and what the attempt looks like <------ |
| 13:42.36 | [TK]D-Fender | and your config appears to missing a lot of settings |
| 13:42.37 | *** join/#asterisk ttyandroid (~tty665@189-015-104-093.xd-dynamic.ctbcnetsuper.com.br) |
| 13:42.46 | ttyandroid | Hi All |
| 13:43.45 | ttyandroid | i tryng using SIP TLS with asterisk 1.8.6 and CsipSample but i'dont know configure certficate |
| 13:44.01 | _N1x | http://pastebin.com/FQbmt5Wg |
| 13:44.08 | _N1x | it's enough? |
| 13:45.02 | [TK]D-Fender | _N1x, No. You are masking all of the IP's, your [general] section is missing a lot of required info, you aren't providing the SIP debuig from the register as requested and I'm having to ask for all of this over and over and over again. |
| 13:45.34 | [TK]D-Fender | _N1x, Please stop wasting our time like this. |
| 13:46.46 | _N1x | [TK]D-Fender: mdam ip is neccessary? |
| 13:47.43 | IsUp | _N1x: paranoid android |
| 13:48.21 | irroot | ttyandroid do not use "verify" bits its is b0rk3d |
| 13:49.06 | IsUp | _N1x: please remove deny=0.0.0.0/0 allow= lines for testing and then do 'sip reload' |
| 13:50.06 | IsUp | _N1x: and please do 'sip debug' (its sip debug peer <peer> in 1.8 i think) and place a call to your number. and see if theres any activity or not. |
| 13:50.13 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:50.45 | [TK]D-Fender | No fintering. Don't dream of trying by "peer". There's far too much tunnel vision going on as it is |
| 13:51.16 | _N1x | IsUp: i making like this but can't see any incomming calls |
| 13:51.24 | _N1x | don't understand what happening |
| 13:51.29 | IsUp | brb |
| 13:52.22 | kaldemar | _N1x: allow is for codecs, not for addresses. "permit" is what you use for ACL. |
| 13:52.39 | [TK]D-Fender | Indeed |
| 13:52.45 | [TK]D-Fender | taht should eb permit/deny |
| 13:53.06 | [TK]D-Fender | gah typing-sync fail... |
| 13:53.20 | [TK]D-Fender | And that IS a "deny all" |
| 13:53.26 | [TK]D-Fender | that should block everything |
| 13:53.35 | [TK]D-Fender | ^^ |
| 13:53.40 | _N1x | i know :) |
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| 14:39.03 | irroot | mandla hi there |
| 14:39.26 | mandla | irroot: im here support. |
| 14:39.44 | irroot | what up |
| 14:40.05 | mandla | irroot: in great my man, and how are you?? |
| 14:40.31 | irroot | no complaints |
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| 15:11.51 | ilj | hey guys, I've a GSM PCI card operated by asterisk using eggsm software module, and here's the thing, When I call the number of a SIM card I can hear ringing sound in my phone, but asterisk never seems to connect two parties. It prints on console that Answer application was run, as well as ... |
| 15:11.56 | ilj | ... subsequent Playback, etc. but it never actually connects the parties. Here's the relevant dialplan snippet and console log: http://pastebin.com/2PNWbRXr |
| 15:13.25 | MrTelephone | Am I missing something about voicemail storage. What about voicemail users? |
| 15:15.28 | bulkorok | MrTelephone: Realtime-Users? |
| 15:18.58 | [TK]D-Fender | ilj, You aren't calling another party. You also call a non-existent dialplan app and the call dies. Do you hear the promprt that it does seem to say should ahve been played back? |
| 15:20.51 | MrTelephone | bilkorok, oh maybe the setting is where you configure sip realtime? hopefully I can just do voicemail first |
| 15:21.17 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:21.58 | elred_ | Hi. I am trying to create some sip.conf template that use predefined variable in an included file. I can include a symbolic link, but how to define variable within it ? Thanks you |
| 15:22.09 | SeRi | Happy Thanksgiving #asterisk! |
| 15:22.48 | [TK]D-Fender | elred_, What kind of variable? defined how? |
| 15:23.04 | MrTelephone | voicemail => odbc,asterisk is for voicemail users and the setting in voicemail.conf is for actual voicemail storage? |
| 15:23.25 | ilj | [TK]D-Fender, you're always around :) No, the Playback app doesn't play back anything. I just hear normal ringing. Do you think the missing Echo app is the cause of this problem? Also, I'm not sure why you say that I'm not calling another party when I can see on console an incoming call from my ... |
| 15:23.30 | ilj | ... cellphone to a sim card in GSM module. |
| 15:23.46 | bulkorok | MrTelephone: Voicemail-Storage must be chosen in make menuconfig! |
| 15:24.03 | [TK]D-Fender | ilj, Do a whole bunch of playbacks in a row. Enough to kill a bunch of time. Make sure that works first |
| 15:24.09 | MrTelephone | yeah I store voicemail in odbc but the user information is in voicemail.conf staticly |
| 15:24.36 | bulkorok | ok... voicemail-users in odbc must be setup in extconfig.conf |
| 15:24.43 | elred_ | [TK]D-Fender, i would like to have for instance $IP = X.X.X.X and then $USER=myaccount etc for later use in sip.conf |
| 15:24.47 | elred_ | is that possible ? |
| 15:24.52 | elred_ | The variable would be in my included file |
| 15:25.11 | ilj | [TK]D-Fender, ok, at this point I'm willing to try anything.. |
| 15:26.13 | bulkorok | MrTelephone: Here are the infos: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail |
| 15:28.21 | [TK]D-Fender | elred_, those don't looks like any kind of variables. |
| 15:28.48 | elred_ | [TK]D-Fender, my question is "where to find documentation about variable definition" ? |
| 15:29.22 | elred_ | I know they don't look like any kind of variables, i am trying to illustrate what i am seeking. Thanks you |
| 15:29.49 | ilj | [TK]D-Fender, nah, it behaves just the same. Answer is executed, then Playback a number of times in a row but all I hear is ringing all along the way. Weird. |
| 15:31.51 | ilj | ha, I ended the call on my cellphone but diaplan still continues to execute Playback application. what the heck? |
| 15:32.45 | [TK]D-Fender | ilj, Then you've got a channel driver issue. I've never heard of EGGSM before.... not sure how many people in here will be able to assist you withit if any |
| 15:32.48 | ilj | is there a way to terminate execution of matching dialplan sequence from console manually? |
| 15:33.08 | ilj | [TK]D-Fender, yeah, I thought so.. |
| 15:34.25 | elred_ | HOST = 85.118.47.87 |
| 15:34.25 | elred_ | host = ${HOST} |
| 15:34.29 | elred_ | is that correct within sip.conf ? |
| 15:34.37 | elred_ | or is that only for extensions.conf ? |
| 15:34.45 | elred_ | (sorry for the IP) |
| 15:35.00 | [TK]D-Fender | elred_, SetVar=HOST=1.2.3.4 |
| 15:35.03 | [TK]D-Fender | like so |
| 15:35.10 | elred_ | in sip.conf ?? |
| 15:35.16 | [TK]D-Fender | and "include" should work fine |
| 15:35.20 | [TK]D-Fender | yes |
| 15:35.27 | elred_ | Ok |
| 15:35.30 | elred_ | Thanks you |
| 15:35.58 | *** join/#asterisk screenn (~screenn@178.251.111.71) |
| 15:37.10 | MrTelephone | will it use the user information from voicemail.conf as well as odbc I wonder? The sippeers and sipusers still allow for local configurations. |
| 15:41.49 | *** join/#asterisk scubes13 (~scubes13@adsl-74-243-172-237.ard.bellsouth.net) |
| 15:44.08 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377161.home.otenet.gr) |
| 15:49.09 | ilj | [TK]D-Fender, oops, turns out the loop was self-induced. it's not channel driver's blame, all mine. but anyway why wouldn't it connect the call? I can' see anything relevant in the logs, which doesn't help things. *sighs* |
| 15:52.21 | ilj | hm, I fixed Goto label, and Hangup is now executed but it then continues to loop anyway, so Hangup doesn't work either. Probably becasue Answer didn't really connect the two parties in the first place. |
| 15:52.24 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:53.10 | irroot | folks have a good turkey day if you are american |
| 15:53.45 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 15:54.25 | ilj | whois irroot |
| 15:54.33 | ilj | oops ^^ |
| 15:54.46 | irroot | is irroot |
| 15:54.49 | ilj | are you american irroot? xD |
| 15:54.55 | ijpalmer | I'm connecting 2 asterisk boxes via sip, I'm using the normal sip.conf on one *1.6 and realtime sip on the other 1.8.5 is it possible to do this, as I'm having trouble getting them to connect |
| 15:55.10 | irroot | is south african |
| 15:55.38 | ilj | cool |
| 15:55.57 | leifmadsen | ijpalmer: sure, doesn't matter that it is using realtime -- the trick is configuring them correctly to work |
| 15:56.17 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:58.52 | ijpalmer | @leifmadsen: thanks I'll continue trying to figure it out, don't suppose you know of a good document outlining this |
| 15:59.10 | leifmadsen | ... |
| 15:59.15 | leifmadsen | ~thebook |
| 15:59.15 | infobot | somebody said thebook was Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or ~buybook |
| 15:59.16 | [TK]D-Fender | ijpalmer, Enable SIP debug. Look at what is happening. Share PASTEBIN's with us... |
| 15:59.36 | leifmadsen | there is a section that shows exactly how to get 2 asterisk boxes working between each other using SIP |
| 15:59.55 | leifmadsen | just do it locally first with the flat files, get it working, then move the configuration to realtime |
| 16:00.38 | ijpalmer | @leifmadsen: thanks, I'll do just that |
| 16:05.26 | MrTelephone | mp3 for voicemail? |
| 16:13.30 | [TK]D-Fender | nope |
| 16:15.07 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:16.52 | MrTelephone | damn |
| 16:17.03 | MrTelephone | what about streaming from wowza for musiconhold |
| 16:22.11 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 16:22.51 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 16:23.59 | p3nguin | If the stream is mp3 or ogg, you can do it. |
| 16:27.55 | MrTelephone | it's from an axis camera product. h264 with aac audio |
| 16:28.10 | MrTelephone | i use it to stream to a web flash player |
| 16:28.18 | [TK]D-Fender | Some people like pain. |
| 16:28.47 | MrTelephone | I monkeyed around with vlc before and it would sample 1 minute and it would use that 1 minute clip from the beginning for every music on hold :( lol |
| 16:29.22 | MrTelephone | I went to one office and the phone technician hacked an alarm clock and hooked it directly to their nortel meridian box :( |
| 16:29.26 | MrTelephone | why can't it be that easy |
| 16:29.34 | [TK]D-Fender | Note to self : Don't let monkeys near the phone system... or any other IT infrastructure. |
| 16:30.02 | MrTelephone | They had a pair of wires right off the speaker inside the alarm clock to their punch down block |
| 16:30.03 | MrTelephone | as if |
| 16:33.04 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:37.43 | *** join/#asterisk hdiogenes (~humberto@201.76.154.190.intranet.digi.com.br) |
| 16:40.38 | MrTelephone | do you have to backquote all enum clumns in mysql while inserting? |
| 16:40.47 | MrTelephone | Armageddon should be coming soon |
| 16:43.40 | p3nguin | backwash |
| 16:43.43 | p3nguin | backfill |
| 16:44.45 | SeRi | p3nguin, waz up d00d |
| 16:44.52 | SeRi | how you feeling today? |
| 16:44.53 | MrTelephone | Mr poppers penguin? |
| 16:45.08 | *** join/#asterisk irroot (~gregory@197.174.109.67) |
| 16:45.11 | p3nguin | I think I'm just getting used to the pain. |
| 16:45.19 | SeRi | p3nguin, :/ |
| 16:45.36 | MrTelephone | you throw your back out like a welfare case? |
| 16:46.26 | SeRi | p3nguin, I found a small bug on voip.ms reseller's interface |
| 16:47.07 | p3nguin | They're everywhere... but what did you find? |
| 16:47.25 | p3nguin | mrtelephone: I think I'm having a liver problem. |
| 16:47.36 | SeRi | I was testing with a su account last night and found that if yuo have the account set to "Hard" not matter what you do the main accounts gets charged the "resellers" rate even if the system marks the number unavailable. |
| 16:48.01 | SeRi | s/su /sub |
| 16:48.06 | SeRi | s/su/sub |
| 16:48.10 | SeRi | lol |
| 16:48.11 | p3nguin | so close! |
| 16:48.15 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 16:48.15 | SeRi | lol |
| 16:48.32 | p3nguin | s/close/far away/ |
| 16:48.41 | SeRi | rofl! hahaha! |
| 16:48.43 | SeRi | nice |
| 16:49.26 | SeRi | any who.... so when you call the number the voip.ms "unavailable" messages comes on and you are getting charged by the min for that call |
| 16:49.40 | MrTelephone | Is exchange 2010 worth buying? |
| 16:50.03 | MrTelephone | seri, because the call status is ANSWERED |
| 16:50.20 | p3nguin | Who is playing the message? |
| 16:50.31 | SeRi | MrTelephone, wrong |
| 16:50.32 | MrTelephone | I have to kill all calls under 5 seconds to prevent accidental billing |
| 16:50.36 | MrTelephone | ? |
| 16:50.37 | SeRi | p3nguin, voip.ms |
| 16:50.56 | p3nguin | Where do you have the number routed? |
| 16:51.13 | MrTelephone | if the remote carrier picks up and plays a message? Do you guys bill your customers for that? |
| 16:51.36 | p3nguin | If there's an answer, there is billing. |
| 16:51.45 | SeRi | to a pap1. But the catch is that the account is set to "Hard" so when the balance is 0 the account is "disable" so calls should not go out or come in. |
| 16:52.14 | p3nguin | What if it is set to soft? |
| 16:52.14 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 16:52.14 | SeRi | voip.ms is answering the call with an unavailable message not me |
| 16:52.25 | SeRi | than the calls come in and or go out |
| 16:52.30 | drudge` | ok.... so i did 'sip show channels' and i get 766 active SIP channels |
| 16:52.33 | SeRi | regardless of the balance |
| 16:52.37 | drudge` | how cna i prevent that from happening? |
| 16:52.41 | lhfnet | Hi, I am having this error trying to do an outside call using MFC/r2 [Nov 24 12:20:55] ERROR[17123]: chan_dahdi.c:4056 dahdi_r2_write_log: Chan 94 - Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 |
| 16:52.41 | lhfnet | DNIS = 04122810524, ANI = 010, MF = 0x20 |
| 16:52.42 | lhfnet | [Nov 24 12:20:55] ERROR[17123]: chan_dahdi.c:3841 dahdi_r2_on_protocol_error: MFC/R2 protocol error on chan 94: Seize Timeout |
| 16:52.49 | SeRi | p3nguin, in the case of soft the reseller gets billed |
| 16:52.58 | SeRi | and that is legit |
| 16:53.00 | p3nguin | ~pb |
| 16:53.00 | 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.us , 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. |
| 16:53.01 | lhfnet | can anyone help me please? |
| 16:53.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:53.22 | p3nguin | In the case of a hard limit, the reseller still gets billed. |
| 16:53.43 | [TK]D-Fender | drudge`, There is nothing inherently wrong with that |
| 16:53.47 | drudge` | they are all coming from 127.0.0.1 as the address |
| 16:53.58 | SeRi | p3nguin, If the calls is not been routed why should the reseller get billed? |
| 16:53.59 | [TK]D-Fender | drudge`, Those do not necessarily repest actual voice channels |
| 16:54.00 | p3nguin | But I agree with you that it should be dead in the water and no billing needs to be done. |
| 16:54.04 | drudge` | is that not bad or what does that indicate? |
| 16:54.07 | [TK]D-Fender | represent* |
| 16:54.25 | drudge` | now there are 922 |
| 16:54.29 | SeRi | p3nguin, exactly. I did open a ticket with voip.ms and they are looking in to it. |
| 16:54.30 | drudge` | whats going on |
| 16:54.34 | p3nguin | core show channels |
| 16:54.43 | [TK]D-Fender | drudge`, go lokat SIP debug. also as to what type those are |
| 16:54.53 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 16:54.57 | lhfnet | here the log file: http://pastebin.com/trSKc1ic |
| 16:55.01 | [TK]D-Fender | drudge`, Perfectly normal to see QUALIFY (options) channels linger a little |
| 16:55.17 | p3nguin | My "sip show channels" does not show "active SIP channels," but says "active SIP dialogs." |
| 16:55.36 | drudge` | that would mean these are attempts? |
| 16:55.42 | p3nguin | core show channels |
| 16:56.16 | drudge` | its not that new, p3nguin |
| 16:56.33 | p3nguin | You're saying an old Asterisk doesn't have core show channels? |
| 16:56.58 | [TK]D-Fender | != core |
| 16:57.00 | MrTelephone | Our carrier doesn't charge if you call a non-existant long distance phone number. |
| 16:57.40 | lhfnet | anyone? |
| 16:57.58 | drudge` | No such command 'core' (type 'help' for help) |
| 16:58.06 | p3nguin | "show channels" then. |
| 16:58.38 | p3nguin | If you're going to run something so outdated, you're going to have to adapt to modern commands without being told. |
| 16:58.40 | drudge` | thats what i have to use on this system |
| 16:59.03 | drudge` | p3nguin, focus dude, iw asnt asking for hte command, i was concerned about the number of channels |
| 16:59.21 | p3nguin | And I'm telling you to look at the channels listed in core show channels. |
| 16:59.28 | drudge` | i started the question of with sip show channels, i know hte goddamn comamnd for this asterisk@home install |
| 16:59.31 | irroot | lhfnet messed with MF/R2 but nothing serious there is some installed in this country but will be replaced with ETSI on request so never needed it |
| 16:59.48 | drudge` | im more concered witht he 700-900 that are listed and if thats cause for concern |
| 16:59.54 | irroot | lhfnet doubt many here will know what MF/R2 is :P |
| 17:00.05 | drudge` | i know they arent active calls cause im looking right at it |
| 17:00.08 | p3nguin | You're apparently not THAT concerned, or you'd look at the channels. |
| 17:00.16 | drudge` | *sigh* |
| 17:00.28 | drudge` | you've got tunnel vision, bruh, nevermind |
| 17:00.37 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 17:00.38 | MrTelephone | no offence but I didn't have time to update to asterisk v12.0.0.1 yet :P |
| 17:00.42 | p3nguin | Good. NEXT! |
| 17:00.55 | MrTelephone | The programmers are racing through versions like I change underwear |
| 17:00.55 | *** part/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 17:01.04 | [TK]D-Fender | <drudge`> i started the question of with sip show channels, i know hte goddamn comamnd for this asterisk@home install <--- A@H?!?! OMMFGWTFBBQ! |
| 17:01.25 | MrTelephone | Fender Bender just happened |
| 17:01.25 | drudge` | yea, a@h.... did everyone start off on trixbox or something?? |
| 17:01.39 | drudge` | no one ever had a@h before or what |
| 17:01.51 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 17:01.54 | enoch | hi all |
| 17:02.02 | p3nguin | I started out on Asterisk. |
| 17:02.09 | [TK]D-Fender | drudge`, No. Most of the sane people here never stuck with any GUI, and those who did... at least upgraded to FreePBX, let alone a version release in the past 4 years |
| 17:02.16 | drudge` | this cheap tight wad company refuses to pay for any engineering time on their current pbx, they just want to pay us monthly to add/remove extensions |
| 17:02.17 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:02.37 | MrTelephone | be motivated and work for free at home like i do to keep things updated |
| 17:02.40 | drudge` | we have at least two companies like that |
| 17:02.43 | [TK]D-Fender | drudge`, Their day of reckoning will come... |
| 17:03.02 | enoch | i have a little pbx using freepbx and a TDM410P with 1 fxo and to fxs ports, on fxs ports i hear a lot of noises... how can i remove it? |
| 17:03.11 | [TK]D-Fender | drudge`, Hope they aren't at all exposed to the internet at-large. |
| 17:03.18 | drudge` | the other company uses Welcome to the FreePBX Administration 2.1.3 |
| 17:03.30 | [TK]D-Fender | Ancient. |
| 17:03.31 | drudge` | they also refuse to upgrade |
| 17:03.42 | MrTelephone | it's not broke don't fix it attitude? |
| 17:03.45 | [TK]D-Fender | "poor them" when the time comes that something eneds to be done |
| 17:03.50 | [TK]D-Fender | needs* |
| 17:03.58 | drudge` | we set it up for them like 4-5 years ago and its "worked ever since" so they dont want to do anything further |
| 17:04.11 | MrTelephone | I feel that. upgrading can break everything |
| 17:04.26 | irroot | enoch have you got codecs right in europe its ALAW and then make sure impeadance is right |
| 17:04.27 | drudge` | upgrading cna break everythign if you try to upgrade something un-tuched in the last 5 years |
| 17:04.34 | MrTelephone | that's why I install every new version that comes out becuase I don't want to jump from version .9 to 1.8.5 in one shot |
| 17:04.36 | drudge` | regular updates tho, tend to prevent breakages from an upgrade |
| 17:04.54 | MrTelephone | I broke ass just upgrading from 1.2 to 1.8 |
| 17:05.11 | drudge` | that a@h install is using free pbx 2.9 something |
| 17:05.17 | MrTelephone | version 10? do we have to reqrite our 2000 line dialplan? |
| 17:05.36 | MrTelephone | drudge, what was your original problem? |
| 17:05.37 | irroot | enoch have you got a echo cancel module ?? and if so make sure its setup correctly possibly try with vpm turned off on module load |
| 17:06.04 | drudge` | sip show channels currently lists over 1800 sip channels |
| 17:06.09 | drudge` | usually its just 3 or 4 |
| 17:06.45 | enoch | irroot: i haven't an echo cancellation mode... |
| 17:06.47 | *** join/#asterisk ttyandroid (~tty665@189-015-104-093.xd-dynamic.ctbcnetsuper.com.br) |
| 17:06.48 | [TK]D-Fender | Lunch time... |
| 17:07.02 | ttyandroid | Hi ALL someody use SIP TLS |
| 17:07.03 | enoch | irroot: will oslec solve my problems? |
| 17:07.03 | ttyandroid | ? |
| 17:07.20 | drudge` | Transmitting (NAT) to 212.230.165.178:23933: SIP/2.0 401 Unauthorized |
| 17:07.24 | drudge` | thats what sip debug shows |
| 17:07.46 | MrTelephone | wrong username or password? |
| 17:07.51 | p3nguin | mrtelephone: If you're still using numbered priorities, you should change that when you hit 1.4. If you're using any of the apps that have changed, you'll need to update their syntax. All 2000 lines, maybe... but probably not. |
| 17:07.54 | MrTelephone | Is that on a registration reply? |
| 17:08.19 | irroot | enoch probably not where are you located ?? ie check codec/impedance |
| 17:08.34 | MrTelephone | numbering priorities are becoming obsolete? are you joking? |
| 17:08.45 | enoch | irroot: im in italy... what is the configuration file to modify? |
| 17:08.57 | p3nguin | The only numbered priorities I have is number 1 on each extension. |
| 17:09.17 | p3nguin | The rest is n or s, and I use labels like they are supposed to be used. |
| 17:09.24 | MrTelephone | drudge, wrong password/username or multiple authorization headers in sip message? |
| 17:09.43 | MrTelephone | pengiun yeah same here. I started using nameing conventions in brackets too. I hope they keep that |
| 17:09.49 | enoch | irroot: is mg2 an hw echo canceller? |
| 17:09.55 | p3nguin | I'm not sure what that means. |
| 17:10.03 | p3nguin | naming conventions in brackets? |
| 17:10.13 | MrTelephone | exten XXXX,n(jumphere),command |
| 17:10.23 | p3nguin | That's a label, and those are not brackets. |
| 17:10.36 | MrTelephone | what are they, concave mirrors? |
| 17:10.45 | p3nguin | parenthesis |
| 17:10.49 | MrTelephone | lol |
| 17:10.54 | MrTelephone | ok mr popper |
| 17:11.04 | MrTelephone | :P |
| 17:11.04 | p3nguin | [ ] <--- brackets |
| 17:11.15 | p3nguin | { } <--- braces |
| 17:11.25 | MrTelephone | , comma . comma without the tail |
| 17:11.32 | p3nguin | :/ |
| 17:11.49 | MrTelephone | just buggin penguin. I know it's important to be literal |
| 17:12.17 | MrTelephone | {-|-|-|-| |
| 17:12.21 | MrTelephone | braces with teeth |
| 17:12.27 | p3nguin | But anyway, if you are already not numbering your priorities, you're a step ahead. |
| 17:12.47 | SeRi | <3 <---- boobs |
| 17:12.49 | *** join/#asterisk BuenGenio (~Gene@202.180.164.172) |
| 17:12.51 | florz | and the plural is parentheses, btw |
| 17:12.53 | drudge` | those arent boobs |
| 17:12.57 | SeRi | lol |
| 17:12.58 | drudge` | thats an icecream cone |
| 17:13.03 | drudge` | two scoops! |
| 17:13.09 | MrTelephone | My biggest issue is configurations, dynamic, making sure they are the same everywhere. So I'm starting to go mostly database even for the dial plan |
| 17:13.19 | MrTelephone | seri, those look like testicles |
| 17:13.25 | p3nguin | haha |
| 17:13.54 | SeRi | MrTelephone, lol is that whats in your mind? |
| 17:14.07 | MrTelephone | haha yeah I have to go for a piss |
| 17:14.20 | SeRi | lol |
| 17:15.12 | enoch | irroot: are u here? |
| 17:18.38 | irroot | yip |
| 17:19.46 | SeRi | p3nguin, I am switching my main desktop from slack to arch. |
| 17:19.50 | irroot | enoch you need to set the "law" to alaw |
| 17:20.02 | p3nguin | seri: +1 |
| 17:20.04 | SeRi | I will need lots of coffee this weekend :) |
| 17:20.19 | p3nguin | Really? |
| 17:20.36 | p3nguin | Don't delete your home partition! |
| 17:21.30 | irroot | alawoverride=1 enoch |
| 17:21.34 | *** join/#asterisk _N1x (~n1x@95.104.13.145) |
| 17:21.50 | _N1x | well guys. need help |
| 17:22.14 | irroot | opermode=...... not sure what for italy |
| 17:23.24 | ttyandroid | <PROTECTED> |
| 17:24.36 | enoch | irroot: thanks |
| 17:24.39 | SeRi | p3nguin, LOL for sure not. |
| 17:25.09 | irroot | enoch setting ALAW will solve most the problems |
| 17:28.07 | enoch | irroot: nothing changed |
| 17:28.48 | irroot | enoch you "rmmod" |
| 17:29.16 | carrar | gobble gobble gobble |
| 17:29.43 | enoch | irrot wait, dont know exactly what dahdi driver im using :D |
| 17:29.53 | irroot | pulls out grenade tosses it at carrar happy turkey day |
| 17:30.46 | irroot | wctdm or wctdm24xxp |
| 17:30.57 | enoch | irroot: what's better? |
| 17:31.14 | irroot | 4port card is wctdm the 8/24 is wctdm24xxp |
| 17:31.21 | carrar | heh |
| 17:32.28 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:33.00 | enoch | irroot: i have 4 port board and it takes time loading wctdm24xxp |
| 17:33.14 | *** join/#asterisk happylife (~happylife@46.251.83.126) |
| 17:33.30 | irroot | enoch check dmesg output |
| 17:34.27 | enoch | The module parameter alawoverride has been deprecated. Please use the parameter companding=alaw |
| 17:35.46 | enoch | im still having noises btw |
| 17:35.57 | enoch | where can i set impedence |
| 17:36.15 | irroot | opermode |
| 17:36.32 | irroot | the defualt american line interface is 600ohm |
| 17:36.34 | _N1x | guys i making calls from my asterisk but can't receive , here is my sip.conf and extension.conf http://pastebin.com/7CWzURuK , http://pastie.org/2915225 |
| 17:36.36 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 17:37.02 | _N1x | and can't see incomming calls in debug |
| 17:37.11 | _N1x | when i call 904001 |
| 17:37.11 | irroot | here in southafrica we need opermode=SOUTHAFRICA |
| 17:38.06 | enoch | irroot: ok opermode=ITALY... same noises |
| 17:38.30 | irroot | enoch it could be other things important to set up right the other setting we need to change is the rx/tx gain dahdi_monitor will help with this |
| 17:39.03 | SeRi | is helping a customer with Gallery3 |
| 17:39.07 | irroot | also test each port with "echo" or milliwatt test applications |
| 17:39.46 | irroot | enoch is it a DSL line ?? without a pots filter can do this |
| 17:40.26 | irroot | also make sure the modules are seated properly no pins bent and not connected |
| 17:40.42 | [TK]D-Fender | _N1x, nat=no <--- you set every other parm under [general] as if it were behind NAT and do this? makes no sense. Again dialplan is irrelevant if there is no inbound SIP traffic. And you continue to not show any useful SIP debug as repeatedly requested. |
| 17:40.59 | drudge` | after i added that IP address listed in sip debug, all 1800+ promptly went away |
| 17:41.35 | drudge` | (2034 matches) on the access list for that IP so they must be a bored |
| 17:41.38 | enoch | irroot: no pins bent and not connected, what do u mean? |
| 17:41.39 | drudge` | bastard |
| 17:42.23 | _N1x | [TK]D-Fender: my asterisk is behind nat |
| 17:42.32 | _N1x | and should i set nat=no? |
| 17:42.40 | [TK]D-Fender | _N1x, line #9 of your own PB |
| 17:42.47 | irroot | enoch had it once where a techie put the module on card wrong one of the pins was not in the connector bent past it |
| 17:42.53 | [TK]D-Fender | _N1x, You did set it as NAT=no |
| 17:42.54 | irroot | AFK |
| 17:43.02 | _N1x | sec |
| 17:43.03 | *** join/#asterisk FainaUkraina (~Gene@202.180.164.172) |
| 17:43.07 | p3nguin | If you're behind nat, set your general nat to yes. |
| 17:43.26 | _N1x | p3nguin: i have already nat=yes |
| 17:43.38 | [TK]D-Fender | _N1x, line #9 of your own PB <--------- |
| 17:44.05 | p3nguin | It looks like nat=no to us. |
| 17:44.54 | _N1x | sec |
| 17:44.55 | p3nguin | And if the peer which uses the entry named "sip" is not behind a NAT, change its nat to no. |
| 17:45.07 | enoch | irroot: the pins are good |
| 17:45.17 | [TK]D-Fender | p3nguin, I already told him about all of those 3 hours ago |
| 17:45.24 | p3nguin | I suspect it is an ITSP, since you are registering to it. |
| 17:45.35 | enoch | irroot: how to change gains different for every channel? |
| 17:46.32 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.163.50.dsl.dyn.forthnet.gr) |
| 17:51.33 | irroot | enoch in chan_dahdi.conf |
| 17:52.54 | irroot | <punt>Had a user in durban start using T38Gw for there fax line today have been unable to send faxes since asterisk went in they quite happy</punt> |
| 17:57.51 | _N1x | p3nguin: [TK]D-Fender i do , but can't receive calls. and can't see also incomming calls |
| 17:57.56 | MrTelephone | what does that mean? punt? |
| 17:58.16 | [TK]D-Fender | "kick" |
| 17:59.13 | [TK]D-Fender | _N1x, And you're consistently not showing what's been asked fail to follow the advice you'd already been given and provided nothing new to evaluate the sanity of your installation with. |
| 18:00.46 | _N1x | [TK]D-Fender: what you show , i have sip.conf and extensions.conf only and can't see debug when i write sip set debug peer sip |
| 18:00.55 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 18:01.01 | _N1x | whoosh : / |
| 18:02.01 | [TK]D-Fender | _N1x, I've told you repeatedly to enable SIP debug, reload chan_sip and show us the registration attempt, the proof that it's registered and the last configs you've shown were still wrong despite being told to correct several times. |
| 18:03.19 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:05.28 | _N1x | hm sec |
| 18:12.15 | _N1x | [TK]D-Fender: http://pastebin.com/kV3yEnWS see |
| 18:12.42 | _N1x | and i have one debug messages per minute |
| 18:12.45 | [TK]D-Fender | _N1x, You're still masking IP's so we can't prove that everything is right. |
| 18:12.50 | _N1x | i'll show you also sec.. |
| 18:12.59 | _N1x | [TK]D-Fender: every 2.2.2.2 is global ip |
| 18:13.00 | [TK]D-Fender | Retransmitting #1 (NAT) to 2.2.2.2:5060: <-- and we can see that you STILL haven't fixed your peer |
| 18:13.04 | _N1x | it isn't necessary... |
| 18:13.31 | _N1x | [TK]D-Fender: what you mean exactly? |
| 18:13.35 | p3nguin | REGISTER sip:2132.2.2.2/2.0 ... no... THAT's not edited... |
| 18:13.43 | [TK]D-Fender | NAT <------------ |
| 18:13.53 | [TK]D-Fender | You've been told that for providers this is supopsed to be "NO" |
| 18:13.55 | [TK]D-Fender | Many times |
| 18:14.07 | [TK]D-Fender | And you have not shown updated configs. |
| 18:14.46 | _N1x | p3nguin: :) |
| 18:14.55 | _N1x | i have another debug sec.. |
| 18:18.16 | _N1x | http://pastebin.com/cazUs0Hk |
| 18:18.19 | _N1x | this |
| 18:18.28 | _N1x | i see in every one minute |
| 18:19.15 | [TK]D-Fender | _N1x, That isn't your registration attemp, and you are still masking IP's. Stop wasting our time. |
| 18:19.50 | _N1x | [TK]D-Fender: guy! what is difference between global ips? |
| 18:19.58 | _N1x | 2.2.2.2 is 55.22.11.22 |
| 18:20.15 | _N1x | 2.2.2.2 is my public ip |
| 18:20.27 | p3nguin | So these packets are from the same address as they are to. |
| 18:20.37 | _N1x | sorry providers ip... |
| 18:20.39 | _N1x | not mine... |
| 18:21.53 | p3nguin | Myth #1: Your provider's *PUBLIC* IP address is a secret. |
| 18:22.03 | [TK]D-Fender | _N1x, pastebin the complete debug. Do not mask anything at all. |
| 18:22.47 | _N1x | [TK]D-Fender: ok :) |
| 18:22.57 | _N1x | [TK]D-Fender: this is complete debug |
| 18:23.01 | _N1x | but change default ips |
| 18:23.05 | _N1x | if it is problem |
| 18:23.07 | _N1x | mda |
| 18:23.08 | _N1x | :) |
| 18:25.38 | [TK]D-Fender | NEW DEBUG |
| 18:25.44 | p3nguin | Does anyone know what "mda" means? |
| 18:25.52 | _N1x | http://pastebin.com/s4C4QirM |
| 18:27.05 | [TK]D-Fender | _N1x, Where's the registration? |
| 18:27.17 | [TK]D-Fender | _N1x, I have never asked for qualify packets... |
| 18:27.38 | _N1x | what? |
| 18:28.14 | p3nguin | (1227.05) <[TK]D-Fender> _N1x, Where's the registration? |
| 18:28.14 | p3nguin | (1227.17) <[TK]D-Fender> _N1x, I have never asked for qualify packets... |
| 18:28.18 | p3nguin | Seems clear enough to me. |
| 18:30.04 | [TK]D-Fender | 13:19 <[TK]D-Fender> _N1x, That isn't your registration attemp, and you are still masking IP's. Stop wasting our time. |
| 18:30.22 | [TK]D-Fender | 13:02 <[TK]D-Fender> _N1x, I've told you repeatedly to enable SIP debug, reload chan_sip and show us the registration attempt, the proof that it's registered and the last configs you've shown were still wrong despite being told to correct several times. |
| 18:30.34 | _N1x | mda |
| 18:30.39 | p3nguin | Also very clear. |
| 18:30.50 | p3nguin | And I still don't know what mda means. |
| 18:30.56 | _N1x | damn fcking pbx , CME rulez :0 |
| 18:31.20 | _N1x | p3nguin: take easy ;) it's geeky |
| 18:31.27 | _N1x | many peoples don't know what is mda |
| 18:31.33 | [TK]D-Fender | 08:45 <[TK]D-Fender> _N1x, No. You are masking all of the IP's, your [general] section is missing a lot of required info, you aren't providing the SIP debuig from the register as requested and I'm having to ask for all of this over and over and over again. |
| 18:31.43 | [TK]D-Fender | 08:42 <[TK]D-Fender> Show us that you're registered, and what the attempt looks like <------ |
| 18:31.56 | _N1x | i already showed |
| 18:32.05 | _N1x | if you can , help me |
| 18:32.06 | p3nguin | You're still interested in this after this many hours? |
| 18:32.19 | [TK]D-Fender | _N1x, last thin g you sowed was QUALIFY, not REGISTER |
| 18:32.34 | p3nguin | five hours |
| 18:33.09 | [TK]D-Fender | 13:25 <_N1x> http://pastebin.com/s4C4QirM <-- not a registration attemp. Not what I have reuested time and again over the past 5 hours |
| 18:35.13 | _N1x | (01:12:15 PM) _N1x: [TK]D-Fender: http://pastebin.com/kV3yEnWS see |
| 18:35.57 | [TK]D-Fender | And you're masking again. |
| 18:36.19 | [TK]D-Fender | Retransmitting #1 (NAT) to 2.2.2.2:5060: |
| 18:36.25 | [TK]D-Fender | ^^^^^^ na dits NAT AGAIN |
| 18:37.07 | _N1x | [TK]D-Fender: Retransmitting #1 (NAT) to 213.131.55.42:5060 |
| 18:37.08 | _N1x | :) |
| 18:37.15 | _N1x | resolve? |
| 18:37.16 | _N1x | whoala |
| 18:37.16 | _N1x | mnda |
| 18:37.17 | _N1x | : \ |
| 18:37.23 | [TK]D-Fender | 13:12 <_N1x> [TK]D-Fender: http://pastebin.com/kV3yEnWS see |
| 18:37.28 | _N1x | i think i spend time without reason. |
| 18:37.39 | [TK]D-Fender | _N1x, And now you've tried passing me an OLD pastebin. |
| 18:37.49 | [TK]D-Fender | you are wasting MY time |
| 18:37.53 | [TK]D-Fender | Showing me old pastebin's |
| 18:38.19 | [TK]D-Fender | Constantly tampering with the evidence and asking "why it doesn't work" |
| 18:38.34 | [TK]D-Fender | Do you lie to your doctor when you co in when you're sick? |
| 18:38.38 | [TK]D-Fender | g0* |
| 18:39.00 | ChannelZ | g0g0 |
| 18:39.10 | [TK]D-Fender | Do you refuse to open the hood of your car when you ask your mechanic to find out what's wrong with the starter motor? |
| 18:39.38 | _N1x | eh |
| 18:39.49 | _N1x | im going can't help |
| 18:39.51 | _N1x | :) |
| 18:39.55 | p3nguin | Finally! |
| 18:39.57 | _N1x | thanks for catch my time |
| 18:40.50 | _N1x | never ask quesiton on cisco channel you [TK]D-Fender and p3nguin you also ! :) |
| 18:40.55 | *** join/#asterisk linuxplatform (~centoslin@88.87.48.115) |
| 18:41.15 | p3nguin | If I have a Cisco question that I can't answer, I'll be sure to ask there. Thanks for the invitation. |
| 18:42.15 | _N1x | p3nguin: and answer is verry helpfull like me |
| 18:42.25 | _N1x | bb |
| 18:42.26 | *** part/#asterisk _N1x (~n1x@95.104.13.145) |
| 18:42.31 | [TK]D-Fender | I hope not.... |
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| 19:43.54 | krotos | hi all :) |
| 19:44.37 | krotos | Today my asterisk box crashes after a udptl debug |
| 19:44.46 | krotos | in the log file i can see only this |
| 19:44.54 | krotos | (about 2K rows): |
| 19:45.09 | krotos | [Nov 24 10:03:32] WARNING[29603] udptl.c: (no tag): UDPTL read error: Bad file descriptor |
| 19:45.19 | krotos | after that, asterisk crash |
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| 19:47.05 | pabelanger | ~backtrace |
| 19:47.05 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 19:47.08 | pabelanger | krotos: ^ |
| 19:48.46 | krotos | thankyou now i read the wiki |
| 19:51.47 | krotos | ok, now i start asterisk with -g option..and if it crash i can post the core dump |
| 19:52.05 | krotos | thankyou again |
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| 21:25.33 | RobertLaptop | Is there a version of AsteriskNow that uses Asterisk 1.8? |
| 21:27.40 | [TK]D-Fender | RobertLaptop, Not included, but the upgrade is easy |
| 21:27.51 | [TK]D-Fender | #asterisknow <- |
| 21:28.19 | RobertLaptop | [TK]D-Fender, TY |
| 21:30.54 | [TK]D-Fender | checkout time, BBIAB |
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| 23:28.25 | Greenlight | How would I go about setting up voicemail that's not linked to an extension - eg I just want it to be a destination that I can use that'll send the voicemail as an email |
| 23:29.36 | Greenlight | Can I just configure an arbitary extension which doesn't actually exist in voicemai.conf? |
| 23:30.17 | dwmw2 | it needs to exist in voicemail.conf but it can be an arbitrary number |
| 23:30.24 | dwmw2 | I just use '1' for ours |
| 23:30.35 | dwmw2 | and a few different extensions will route to it if unanswered |
| 23:30.55 | Greenlight | Ahh perfect, that's nice and simple then. Just something I've never done before :) |
| 23:30.59 | Greenlight | Many thanks |
| 23:32.44 | dwmw2 | one problem I had was that the standard stdexten macro doesn't cope well. It assumes that the mailbox# will equal the extension# |
| 23:32.47 | dwmw2 | I fixed that thus: |
| 23:32.53 | dwmw2 | -exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})}) |
| 23:32.59 | dwmw2 | +exten => _X.,n,Set(LOCAL(mbx)=${IF($[${REGEX("@" ${cntx})}]?${cntx}:${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})}) |
| 23:33.44 | dwmw2 | which means I can later invoke the stdexten macro thus: |
| 23:33.53 | Greenlight | It's way to late in the day for me to try and work out what that's doing, but copy and paste for later I will |
| 23:33.55 | dwmw2 | +exten => 586671,1,Gosub(586671,stdexten(SIP/c475,1@other)) |
| 23:34.04 | Greenlight | Ah |
| 23:34.11 | dwmw2 | the stdexten macro takes a voicemail *context* optionally |
| 23:34.45 | dwmw2 | my change makes it take either a context or a full mailbox ('1@other' instead of just 'other') |
| 23:35.05 | Greenlight | Gotcha, nice one |
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| 23:45.14 | p3nguin | greenlight: Extensions are not related to voice mail boxes until you make an association in extensions.conf. |
| 23:45.31 | p3nguin | You pick anything you want for a mailbox number and put it in voicemail.conf. |
| 23:46.04 | Greenlight | Ahh makse sense I guess - I'd just always had that association with the actual device in extensions.conf |
| 23:46.06 | p3nguin | And remember extensions are not phones. |
| 23:46.21 | p3nguin | And phones are not extensions. |
| 23:46.43 | p3nguin | Device configuration isn't done in the extensions file, either. |
| 23:48.40 | p3nguin | So if you want a mail box number, create it in voicemail.conf. If you want to be able to reach it, you'll create an extension that runs VoiceMail() with that new mail box number as the app data. |
| 23:49.10 | Greenlight | Think it's more just getting my head into that way of thinking, after years of being used to other PBX's |
| 23:49.20 | Greenlight | It does really make it so flexible though |
| 23:49.48 | p3nguin | Every mail box is capable of emailing new messages, too. |
| 23:50.35 | Greenlight | Mostly all the work I've been doing with asterisk has been resitricted to relativey small sections of it (been using AMI to originate calls for a dialler), just now starting to explore more of it |
| 23:51.22 | Greenlight | Yea - I've had the emailing working for a bit, but that was through FreePBX (gasp!), I'm trying to pick up how to do all these things directly now |
| 23:51.57 | p3nguin | I can't tell you how to undo what FreePBX has done, but I can tell you how to do it all in Asterisk. |
| 23:52.48 | Greenlight | I've got combinations of 'clean' installs and FreePBX based installs with my custom extensions.conf files to provide functionality for my dialler |
| 23:53.55 | Greenlight | One thing I was wondering, when I press the "transfer" button on the physical handset does that get thrown into the same context that the hadnset would be in if it's dialling out normally, and would I still require the "t" in Dial ? |
| 23:54.40 | p3nguin | The t dial option is for DTMF transfers via features.conf -- not related to a SIP transfer. |
| 23:54.52 | p3nguin | But yes, the transfer will be subjected to the context of the phone. |
| 23:55.23 | Greenlight | Cool, I knew I needed it for DTMF transfers but wasn't sure if it was require to allow SIP REFER to work too |
| 23:55.44 | p3nguin | I only use the t option for phones that do not have a real transfer button. |
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| 23:57.26 | Greenlight | Yea I'd been using xlite quite a lot and so was using t option to enable transferring |
| 23:58.29 | p3nguin | I'd be surprised if it doesn't have a transfer button. |
| 23:58.46 | Greenlight | I beleive you need the 'paid' version |
| 23:58.52 | p3nguin | Oh. |
| 23:58.53 | Greenlight | Not the free one |
| 23:58.54 | p3nguin | That could be. |
| 23:59.04 | Greenlight | eyeBeam which is the paid Xlite has it |
| 23:59.07 | p3nguin | I'm not a Windows user, so I don't get to mess with all the Windows soft phones. |
| 23:59.52 | Greenlight | It's free, and looks very much like a 'real' phone so generally users don't get too confused |