| 00:00.13 | idespinner | I would guess the suggestion is to specify the IP interface to use in sip.conf |
| 00:00.13 | p3nguin | seri: yes |
| 00:00.28 | SeRi | d00d I am been so bash about this nanp shit |
| 00:00.36 | rocksfrow | idespinner, yeah i was going to say to avoid it is to simply listen on ONE ip |
| 00:00.41 | rocksfrow | not all. |
| 00:00.45 | SeRi | pople say CC is right and I am wrong |
| 00:00.57 | SeRi | I am not sure what to say |
| 00:01.02 | idespinner | SeRi, what people? |
| 00:01.11 | idespinner | prepending a 1 doesnt sound right |
| 00:01.38 | SeRi | http://www.dslreports.com/forum/r26501280-General-CallCentric-Invalid-NANP-CID#26502446 |
| 00:03.08 | p3nguin | I don't know what to say, really. |
| 00:03.10 | SeRi | I have yet to see a call come in my line that would have a 1 infront of it |
| 00:03.16 | idespinner | same here |
| 00:03.23 | idespinner | but they sound legit |
| 00:03.26 | idespinner | I would work around it |
| 00:03.30 | SeRi | people claim its every where and that all POT carriers do it |
| 00:03.40 | p3nguin | +1 maybe, but +1 would make it E.164 caller ID. |
| 00:03.53 | p3nguin | 1 on the front of it does not make it NANP nor E.164. |
| 00:04.28 | p3nguin | +12123234343 = valid (but not NANP) |
| 00:04.49 | p3nguin | 12123234343 = invalid (not NANP and not E.164) |
| 00:05.03 | SeRi | ^^ thats what it displays infornt of my cid |
| 00:05.08 | SeRi | you saw it |
| 00:05.44 | p3nguin | I know what I saw, and it wasn't what I expected to see. |
| 00:06.30 | SeRi | well not sure what to say. I satrted that thread with hopes that a CC would help me... instead I got bashed by a bunch of CC pussys... |
| 00:06.40 | SeRi | CC rep* |
| 00:06.45 | SeRi | lol |
| 00:06.48 | SeRi | o well. |
| 00:08.08 | p3nguin | I've never before seen a caller ID come into my system with just the added 1 on the front of the otherwise valid caller ID number. |
| 00:08.41 | pdtpatrick1 | Question .. here's my music on hold setup |
| 00:08.41 | pdtpatrick1 | http://pastebin.com/xdGFZk8R |
| 00:08.43 | p3nguin | Never on AT&T land line, never on any wireless carriers, never via other ITSPs... |
| 00:08.46 | p3nguin | never. |
| 00:08.49 | pdtpatrick1 | i keep getting error that the file is missing |
| 00:09.04 | pdtpatrick1 | that is because i moved the file. However, i've added new files and i've reloaded moh |
| 00:09.17 | pdtpatrick1 | but it keeps complaining/looking for the old files |
| 00:09.21 | pdtpatrick1 | any suggestions ? |
| 00:13.35 | navaismo | pdtpatrick1: exist an active channel using it? |
| 00:14.01 | navaismo | core show channels verbose or sip show channels show a stuck channel using that file? |
| 00:14.07 | pdtpatrick1 | that's a possibility. Would issuing asterisk -rx "reload" fix that ? |
| 00:14.35 | navaismo | no i think hangup request <channel> |
| 00:15.31 | pdtpatrick1 | i don't see any anything in there with a file name |
| 00:15.50 | navaismo | but using the moh class? |
| 00:16.07 | p3nguin | seri: Make sure in your posts to differentiate between +1NXXNXXXXXX (E.164 format) and the broken 1NXXNXXXXXX format. "E.164 numbers can have a maximum of fifteen digits and are usually written with a + prefix." |
| 00:16.33 | p3nguin | pdtpatrick1: module unload res_musiconhold.so |
| 00:16.36 | pdtpatrick1 | here's my moh class |
| 00:16.38 | pdtpatrick1 | http://pastebin.com/MCrwWBwn |
| 00:17.04 | pdtpatrick1 | p3nguin, [Nov 1 17:16:48] WARNING[6338]: loader.c:505 ast_unload_resource: Soft unload failed, 'res_musiconhold.so' has use count 11 |
| 00:17.13 | p3nguin | There's your issue. |
| 00:17.14 | navaismo | any channel using moh application, follow the p3nguin suggest |
| 00:17.31 | p3nguin | Remove any channels using MoH, then you can reload the config and use new settings. |
| 00:17.43 | pdtpatrick1 | issuing module reload res_musiconhold.so wouldn't fix that ? |
| 00:17.49 | p3nguin | Not right now. |
| 00:18.30 | p3nguin | I have much issue with moh when it comes to this sort of thing. |
| 00:18.47 | p3nguin | If you can get no channels using it, you can unload it fully, then load it again. |
| 00:19.06 | pdtpatrick1 | but if channels are using it .. the best way is to restart the asterisk service itself? |
| 00:19.31 | p3nguin | I'd rather wait for the channels to not be using it and then fix the moh module. |
| 00:20.57 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 00:21.13 | pdtpatrick1 | I see. Well that won't be for another couple of hours. Curious (forgive me, i'm still learning). It keeps looking for these files that were moved. Is it storing the file names in memory and just hanging onto them? I thought it was supposed to read the directory listing whenever someone is put on hold? |
| 00:24.22 | pdtpatrick1 | p3nguin, or anyone - would u guys know the answer to that? |
| 00:28.42 | p3nguin | I'm really not sure. But I do know how I would try to handle fixing it. |
| 00:29.23 | p3nguin | I would do "core restart gracefully" and wait. |
| 00:29.52 | p3nguin | Make sure the conf is set to correct values so it is loaded as desired on the restart. |
| 00:31.00 | p3nguin | Oh, you said read the directory listing each time someone is put on hold.... no, that's not what happens. |
| 00:31.15 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 00:31.27 | p3nguin | The conf is read when moh module is loaded. Files in the provided directory are loaded into memory as being available for use. |
| 00:31.32 | pdtpatrick1 | so it preloads all the files into memory .. and even if u delete or move those files - it thinks they are there and just keeps askign for them ? |
| 00:31.39 | pdtpatrick1 | ohhh |
| 00:31.40 | p3nguin | moh show files |
| 00:32.25 | p3nguin | I would do moh show files and then make sure the files it lists are available (at least temporarily). |
| 00:32.40 | p3nguin | Then I would configure it like it should be, like I want it. |
| 00:32.47 | p3nguin | Then core restart gracefully. And wait. |
| 00:32.49 | pdtpatrick1 | oh nice i didn't know about that . |
| 00:32.58 | pdtpatrick1 | yeah i did core restart gracefull and now waiting |
| 00:33.16 | pdtpatrick1 | basically that would wait for all the channels to clear and then it would restart the service im guessing? |
| 00:33.20 | p3nguin | You can still work with the files and make sure your new settings are like they need to be. |
| 00:34.04 | p3nguin | restart gracefully is going to not permit any new channels to be created; as existing channels go away, eventually no channels will remain; then asterisk restarts. |
| 00:34.14 | pdtpatrick1 | oh nice! |
| 00:34.49 | p3nguin | core restart when convenient - this would allow new channels to still be created, but if at any time there are no existing channels, asterisk restarts. |
| 00:35.18 | p3nguin | gracefully is a little more insistent for an admin to get his system restarted. |
| 00:35.45 | p3nguin | when convenient could be HOURS or even DAYS. |
| 00:36.51 | pdtpatrick1 | yeah |
| 00:37.00 | pdtpatrick1 | i've noticed it not creating new channels |
| 00:37.06 | pdtpatrick1 | now i've only got 5 remaining :) |
| 00:37.57 | p3nguin | During the period that existing calls are dying off, new calls will receive a temp fail (congestion tones). |
| 00:37.59 | Micc | If I need to convert SIP to a plain T1 to work with an existing pbx with a T1, what kind of device would I get? |
| 00:38.12 | Micc | is that called a SIP T1 gateway? |
| 00:38.14 | p3nguin | micc: You could get a gateway. |
| 00:38.54 | Micc | p3nguin, do you have any experience with them? I'd like to get one that I don't have to hope it will work. |
| 00:38.55 | p3nguin | I think you could also get a card to put in an asterisk system and let asterisk be the gateway. |
| 00:39.11 | p3nguin | Adtran is a common provider of such gateway. |
| 00:41.30 | Micc | I have an old PRI card that used to be used in an asterisk server, would that provide plain T1? |
| 00:41.42 | p3nguin | If the T1 is a PRI, probably. |
| 00:41.44 | Micc | It seemed like zaptel could be configured a bunch of different ways. |
| 00:41.58 | Micc | Their pbx won't do PRI, they don't have the license. |
| 00:42.16 | p3nguin | But if you have a PRI and you have a card for it, you can make asterisk into the gateway. |
| 00:42.16 | Micc | so its just plain T1, all 24 channels, no data channel. |
| 00:43.06 | p3nguin | Can you ask the carrier if that is a PRI and if your card is compatible? |
| 00:43.08 | Micc | I want to provide the T1 service, not the sip service. So I wouldn't need a PRI, just a cable. |
| 00:43.15 | p3nguin | oh |
| 00:43.31 | Micc | You see what I"m getting at now? :) |
| 00:43.40 | p3nguin | You're trying to give a PRI to the PBX. |
| 00:43.51 | Micc | yeah, a T1 to the pbx. |
| 00:44.06 | p3nguin | The PBX doesn't talk SIP? |
| 00:44.06 | Micc | we already checked with the carrier/pbx vendor and it doesn't do PRI, just plain T1 |
| 00:44.14 | Micc | it could for more money. |
| 00:44.27 | Micc | probably 2k + some guy to come out and configure it. |
| 00:44.36 | *** join/#asterisk coppice (~chatzilla@m121-203-215-32.smartone-vodafone.com) |
| 00:45.09 | Micc | If I can do it for under 1k, I'd rather do that. |
| 00:45.09 | p3nguin | I'm really the wrong person to discuss analog technology with. I'm a pure VoIP type of person. |
| 00:45.43 | p3nguin | Give me an ITSP, an Asterisk system, and some IP phones, and I'm happy. |
| 00:46.51 | p3nguin | I can't afford to buy that type of equipment to play with to learn it. Cards and gateways are so expensive. |
| 00:46.54 | Micc | Yeah, me too, which is why I've got to ask. |
| 00:47.36 | Micc | p3nguin, who do you work for? Can't they buy you some stuff to play with? |
| 00:47.57 | Micc | You seem like you should be working for some big ITSP or something. |
| 00:48.02 | p3nguin | If it were that simple, we'd have the equipment. |
| 00:49.42 | [TK]D-Fender | Micc: AudioCodes Mediant |
| 00:50.13 | p3nguin | If I were fortunate enough to work for an ITSP, I'd probably have enough spare parts to set up a nice test environment and build out your idea for you. |
| 00:50.32 | [TK]D-Fender | Micc: Mediatrix has stuff but they are kinda bad. Adtran has some rather competitive choices too, might be the best value. |
| 00:50.42 | [TK]D-Fender | Micc: Then there is Cisco, etc.' |
| 00:52.06 | Micc | AudioCodes Mediant 1000 MSBG? |
| 00:52.29 | p3nguin | While I have not had to configure/reconfigure the Adtran devices at the places I have worked, knowing that so many of them have Adtran equipment for their T1s does influence my choice to buy Adtran if I have to deploy my own. |
| 00:52.40 | Micc | looks like about 1200$ from voip supply. prices very a lot. |
| 00:52.46 | Micc | vary |
| 00:57.12 | [TK]D-Fender | Adtran has a very respectable history. |
| 00:57.46 | [TK]D-Fender | Micc: T1 SIP gateways have a hefty premium normally. $1200 is as cheap as I've ever seen one... |
| 00:58.04 | [TK]D-Fender | Used to cost double 5 years ago |
| 00:59.32 | coppice | they are quite complex boxes. they have to EC and transcode and so on |
| 01:00.10 | *** join/#asterisk mil132 (~quassel@ip64-75-183-66.hsia.aloha.net) |
| 01:00.27 | Micc | Would the Adtran Total Access 904 work? |
| 01:00.36 | Micc | seems like thats all I would need for under 1k |
| 01:05.11 | [TK]D-Fender | Micc: Call up their sales to confirm precise functioanlity. I recall some models having certain limitations |
| 01:05.17 | [TK]D-Fender | You'll wan to be precise on it |
| 01:16.37 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 01:19.07 | *** join/#asterisk adolfomaltez (~taro@190.62.226.177) |
| 01:20.47 | SeRi | p3nguin, can I do a test? |
| 01:21.18 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 01:27.18 | *** join/#asterisk neurosys (~neurosys@c-174-48-142-160.hsd1.fl.comcast.net) |
| 01:33.13 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 01:39.44 | *** join/#asterisk coppice (~chatzilla@m121-203-226-241.smartone-vodafone.com) |
| 01:47.10 | p3nguin | seri: I guess. |
| 01:48.33 | SeRi | call your CC? |
| 01:49.04 | p3nguin | seri: Do you have any termination through CallCentric anymore? I'm wondering if they are only sending invalid caller ID within their network, while calls to the PSTN will still terminate with valid caller ID. |
| 01:49.12 | p3nguin | If that's what you're wanting to test, yes. |
| 01:49.25 | SeRi | yes thats fine. |
| 01:51.59 | SeRi | calling |
| 01:52.16 | SeRi | nice IVR! |
| 01:52.18 | SeRi | LMAO |
| 01:52.21 | SeRi | dev null! |
| 01:52.23 | SeRi | rofl! |
| 01:52.29 | SeRi | hahahahaha! |
| 01:52.31 | SeRi | bad ass! |
| 01:53.19 | p3nguin | :P |
| 01:53.56 | SeRi | can you msg me what displayed in the CID? |
| 01:54.39 | SeRi | p3nguin, you mean termination as an did? |
| 01:54.49 | SeRi | I do have a dialing out plan with them |
| 01:54.53 | SeRi | no inbound though |
| 01:54.55 | SeRi | no did |
| 01:54.57 | p3nguin | No, DID is inward, or origination. |
| 01:55.08 | p3nguin | Termination is dialout out to the PSTN. |
| 01:55.12 | SeRi | ok |
| 01:55.21 | SeRi | yes I do have one. |
| 01:55.30 | SeRi | did the cid came out correct? |
| 01:55.32 | p3nguin | Let me give you another number to dial. |
| 01:55.50 | p3nguin | No, that call came from 1832....... |
| 01:56.30 | SeRi | 832 is the correct CID |
| 01:56.39 | SeRi | ok |
| 02:05.02 | mil132 | is there a SIP proverder that anyone can recomend for business use? |
| 02:05.40 | carrar | Verizon/Mci/Level 3 |
| 02:05.59 | mil132 | how are they on rates? |
| 02:06.19 | carrar | high |
| 02:07.19 | carrar | And you need to go through and pass about 80 interopability tests |
| 02:07.26 | mil132 | hmmm |
| 02:08.23 | mil132 | I am looking to put the business that I work for on a SIP provider with a PBX that we have. no more of this analog phoneline crap that we have |
| 02:08.38 | *** part/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 02:09.05 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 02:09.07 | p3nguin | I'm quite satisfied with VoIP.ms. |
| 02:09.16 | p3nguin | There's also Flowroute and Teliax. |
| 02:09.36 | mil132 | think they will port Hawaii numbers? |
| 02:10.12 | *** join/#asterisk jasonbassett (~jasonbass@cpc3-basl7-0-0-cust155.basl.cable.virginmedia.com) |
| 02:10.30 | carrar | How do you want me to answer that? |
| 02:10.31 | p3nguin | Probably. |
| 02:10.48 | mil132 | hmm |
| 02:11.01 | jasonbassett | Good morning folks, I have a dial line with a macro executed on answer with the M(macroname) option |
| 02:11.26 | jasonbassett | The macro is not reading any input dtmf for the Read() application though |
| 02:11.29 | carrar | My Answer: If you offer them FREE 100% KONA Coffee I am sure they will do anythign for you |
| 02:11.33 | jasonbassett | Any ideas? |
| 02:12.24 | carrar | Or if you offer me FREE 100% KONA Coffee, I'll terminate your calls and provide local DID's |
| 02:12.34 | mil132 | hahah |
| 02:12.38 | mil132 | I could arrange that |
| 02:12.56 | carrar | I wan that $45 a pound stuff |
| 02:13.01 | carrar | I'm not cheap |
| 02:13.12 | carrar | Kona Mountain Coffee, organic! |
| 02:13.18 | mil132 | the stuff that is pooped out by some animal then roasted? |
| 02:13.23 | carrar | haha yeah |
| 02:13.30 | coppice | if you want pricy try civet coffee |
| 02:13.30 | carrar | poop makes everything better |
| 02:14.04 | carrar | actually last time I was in Kona, 7 weeks ago, I foudn a kona coffee supplier for cheaper coffee |
| 02:14.36 | carrar | hawaii is too damn hot |
| 02:14.39 | mil132 | I am on Oahu, so I would have to get it from Longs/Times/SafeWay |
| 02:14.57 | mil132 | meh... 72 degrees and resonable humidity right now |
| 02:15.04 | carrar | and it was freezign up at the observatories |
| 02:15.14 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 02:16.17 | carrar | You need someone in Hawaii to terminate your calls |
| 02:16.27 | carrar | otherwise they have to come back to the mainland and then back |
| 02:17.29 | carrar | I bet the telco's rape everyone there |
| 02:17.37 | mil132 | yea... they do |
| 02:17.42 | mil132 | one telco |
| 02:17.46 | mil132 | Hawaii Telecom |
| 02:17.58 | carrar | Start you own via sip over the internet |
| 02:18.03 | carrar | put a box on each island |
| 02:19.37 | carrar | is it a LD call from one island to the next? |
| 02:19.43 | mil132 | yea |
| 02:19.45 | carrar | haha |
| 02:19.55 | carrar | get a pri on each island |
| 02:20.15 | carrar | link em together |
| 02:20.45 | mil132 | honestly that is way above me |
| 02:20.54 | carrar | naw |
| 02:21.01 | carrar | it's pretty easy/simple |
| 02:21.53 | mil132 | well, lets start small |
| 02:22.30 | mil132 | could I have you give me adivce on how to get us on to voip? |
| 02:23.12 | carrar | What do you have today? |
| 02:23.34 | carrar | You can install Asterisk and some SIP phones and there you go |
| 02:23.46 | carrar | phones are all interconnectived via SIP |
| 02:24.52 | mil132 | yea that is what I am thinking |
| 02:25.09 | mil132 | but for the phone service, I want to use a SIP provider, rather than get a T1 |
| 02:25.16 | carrar | why |
| 02:25.29 | mil132 | unlimited domestic calling |
| 02:25.46 | mil132 | but I need one that will port over our hawaii number |
| 02:26.58 | carrar | You want your company to rely on internet connectivity to the US for calls? |
| 02:27.09 | carrar | doesn't seem like the best idea |
| 02:27.21 | mil132 | well, a lot of other compnays here do that because of cost of a t1 |
| 02:27.36 | p3nguin | Tell me the area code and three-digit exchange of a number you wish to port, and I'll check it. |
| 02:27.48 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 02:27.49 | carrar | NPA-NXX |
| 02:27.51 | mil132 | 808-206 |
| 02:27.58 | mil132 | and 808-922 |
| 02:29.10 | carrar | http://www.voip-info.org/wiki/view/Sip+Trunking+Providers |
| 02:29.27 | p3nguin | Not portable to VoIP.ms |
| 02:29.56 | carrar | Hawaii is probably a lot like Alaska |
| 02:30.10 | mil132 | there was one that I was looking at that could do it, but I did not save them it was like grandvox or something |
| 02:31.00 | carrar | mil132, why not ask one of these "a lot of other compnays" |
| 02:31.06 | carrar | what they use |
| 02:31.11 | mil132 | true |
| 02:31.22 | carrar | ask them how they like it |
| 02:34.19 | mil132 | BraodVoice |
| 02:34.27 | mil132 | BroadVoice sry |
| 02:36.18 | mil132 | and Broadvox |
| 02:36.20 | carrar | You could just keep a POTS line for local calls |
| 02:36.27 | carrar | and get a US did for calling the US |
| 02:36.36 | carrar | and don't send hawaii calls to the SIP carrier |
| 02:37.25 | mil132 | well... If we could get interisland calling without the longdistance, then we would probibly not do a hybrid system like that |
| 02:38.55 | mil132 | haha, hawaii is included, but not alaska |
| 02:39.13 | carrar | use google voice |
| 02:39.14 | carrar | heh |
| 02:39.16 | mil132 | I thought we had it bad out here, but alaska sounds worse |
| 02:39.31 | mil132 | google voice does not have 808 numbers |
| 02:39.42 | mil132 | and I need to be able to support ~20 users |
| 02:43.17 | carrar | Should start leaning how to use Asterisk now |
| 02:43.37 | carrar | then when it's time to switch to SIP you will know how |
| 02:43.40 | mil132 | I am going to start playing with a VM of freeSwitch soon |
| 02:43.58 | carrar | Should start leaning how to use Asterisk now |
| 02:44.04 | carrar | not freeswitch |
| 02:44.04 | mil132 | yea |
| 02:44.20 | mil132 | freeSwitch uses astersik right? |
| 02:44.50 | carrar | in their own mangled way |
| 02:44.58 | carrar | unsuported here |
| 02:45.02 | carrar | or by the asterisk people |
| 02:45.13 | mil132 | what about SwtichVox |
| 02:45.19 | carrar | Thats supported by Digium |
| 02:45.22 | mil132 | yea |
| 02:45.42 | mil132 | my manager wants something that comes with support |
| 02:45.44 | carrar | SwitchVox is nice |
| 02:45.48 | carrar | pretty gui |
| 02:45.57 | carrar | but you pay licensing per phone |
| 02:46.04 | carrar | per SIP device |
| 02:46.18 | mil132 | we have a Aastra system now, but it lacking in a few ways |
| 02:46.40 | carrar | lacking what? |
| 02:46.47 | carrar | (that you ned) |
| 02:46.49 | carrar | need |
| 02:46.54 | mil132 | well, voicemail, for eample, |
| 02:47.07 | carrar | that is lame |
| 02:47.15 | carrar | thats basic stuff |
| 02:47.16 | mil132 | the higher ups want to be able to have all thier voicemail be in thier email |
| 02:47.35 | mil132 | and when they check the voicemail in the email, it removes the notification from thier phone |
| 02:47.57 | mil132 | right now, it goes to email, but some people have ~600 voice messages |
| 02:48.09 | mil132 | and to delete, you need to do it one by one |
| 02:48.24 | dijib | no you dont |
| 02:48.37 | mil132 | pleas for the love of all that is holy how to do it |
| 02:48.39 | dijib | you can do it by deleting them through WinSCP |
| 02:48.45 | mil132 | ??? |
| 02:50.07 | dijib | navigate to /var/spool/asterisk/voicemail/default (or whatever voicemail context you have set) and then the subdirectory of the extension |
| 02:50.13 | carrar | dijib |
| 02:50.17 | carrar | pay attention |
| 02:50.23 | dijib | what? |
| 02:50.46 | carrar | scroll back so you have a clue what you are talking about |
| 02:50.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:51.19 | mil132 | I have ans Aastralink Pro 160, its not a vanilla Astersk install |
| 02:51.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:51.42 | carrar | <mil132> we have a Aastra system now |
| 02:51.46 | SeRi | dijib, is your chan up? |
| 02:51.47 | carrar | I assume a Aastra PBX |
| 02:51.51 | [TK]D-Fender | mil132: Go ask in their support channels then |
| 02:51.52 | dijib | yes |
| 02:51.57 | dijib | but i dont think anyones in it |
| 02:51.59 | SeRi | ill be in soon |
| 02:52.08 | mil132 | hmm |
| 02:52.10 | mil132 | ok |
| 02:52.53 | [TK]D-Fender | mil132: we have no idea what their codebase allows you to get away with |
| 02:52.55 | carrar | mil132, do you want to learn Asterisk or just a phone pbx working and forget it? |
| 02:53.33 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:53.47 | mil132 | basicly, I have 2 weeks to get something together to start implementing |
| 02:54.04 | carrar | Otherwise I would say switchvox will fit you jsut fine, but it's not free of course |
| 02:54.18 | mil132 | spread across 3 locations... 2 in hawaii and one in oregon |
| 02:54.36 | dijib | ok carrar i see he's talking about Aastra now |
| 02:54.38 | carrar | All registering to the same server? |
| 02:54.44 | mil132 | hopefully |
| 02:54.59 | dijib | run asterisk |
| 02:54.59 | carrar | So your pbx should be in a datacenter? |
| 02:55.04 | mil132 | yea |
| 02:55.08 | mil132 | Idealy |
| 02:55.34 | carrar | and if you go with Asterisk you will probably need to outsource it |
| 02:55.41 | carrar | since you haev a lot to learn |
| 02:55.44 | carrar | in 2 weeks :) |
| 02:55.48 | mil132 | although we do have some rackspace and a 100mb fiber link to one of our buildings |
| 02:56.06 | mil132 | I was thinking of putting it in there |
| 02:56.08 | carrar | 100mb to another building doesn't do much good |
| 02:56.22 | carrar | 100mb to the internet will |
| 02:56.23 | mil132 | no it is a 100mb intertube connection |
| 02:56.43 | mil132 | and we might upgade it soon to 300mb |
| 02:57.12 | carrar | Do you need local DID's at all 3 locations |
| 02:57.46 | carrar | definately all very doable |
| 02:57.54 | mil132 | no really, we only have maby 5 differant numbers that would need to be ported, then we would direct the call in the PBX |
| 02:58.31 | mil132 | we also need to have some international calling, but we can take the hit on that |
| 02:59.01 | carrar | Your same SIP provider should be able to provide discounted internationall calling |
| 02:59.15 | mil132 | lets put it this way, we have ~$700 a month right now in long distance |
| 02:59.37 | mil132 | sometimes more if it is our sales season |
| 02:59.48 | mil132 | I have seen it go up to $1200 |
| 03:00.40 | mil132 | so if I can beat that, then we can absorb the cost of equipment |
| 03:01.39 | mil132 | my plan was to setup a centralised PBX, then get a SIP proider for that... then just distribute phones that connect back to the PBX |
| 03:02.20 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 03:02.30 | dijib | you said your in honolulu? |
| 03:02.34 | mil132 | yea |
| 03:02.47 | dijib | do you have toll free? |
| 03:02.59 | mil132 | no, but it would be a plus to have one |
| 03:03.47 | dijib | they get expensive for incomming calls @ $1.95 an hour. but outgoing calls are only $0.75 an hour |
| 03:04.04 | dijib | 1-800 for customers? |
| 03:04.04 | mil132 | it would be more like a sales hotline |
| 03:04.07 | mil132 | yea |
| 03:04.27 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 03:04.32 | carrar | per hour? |
| 03:04.36 | dijib | yes |
| 03:04.37 | carrar | who bills per hour |
| 03:04.53 | [TK]D-Fender | prostitutes |
| 03:04.55 | dijib | its per minute i just do the math to show the cost of hours for easy figuring |
| 03:04.56 | carrar | haha |
| 03:05.03 | carrar | high fives TK |
| 03:05.14 | coppice | [TK]D-Fender: I thought they billed per minute |
| 03:05.16 | dijib | why do you guys keep bringing my mother into this? |
| 03:05.31 | mil132 | We can absorb the cost of a customer callin us |
| 03:05.55 | dijib | whats the avg inbound customer call time? |
| 03:05.57 | [TK]D-Fender | "My mother never saw the irony in calling me a son-of-a-bitch" - Jack Nicholson |
| 03:06.00 | itbroke | Hello, does anyone here use chan_mobile? |
| 03:06.00 | dijib | and how many calls per month |
| 03:06.06 | mil132 | maby 10 min at the most |
| 03:06.16 | mil132 | calls per month is a good question |
| 03:06.23 | mil132 | let me see if I can find out |
| 03:06.29 | dijib | so $0.195 per 10min |
| 03:06.37 | dijib | with the toll free |
| 03:07.05 | dijib | SeRi, let me know when your joinging the conf. |
| 03:07.30 | dijib | anybody else is welcome. Dial(2663@asterisk.serveirc.com); |
| 03:07.37 | SeRi | will do dijib |
| 03:08.14 | p3nguin | mil132: If you don't know how to deploy Asterisk... I do. |
| 03:08.18 | p3nguin | Just sayin' |
| 03:08.22 | carrar | mil132, also check your IRC messages |
| 03:08.26 | mil132 | jeez, I am going to have to guess at ~1200 calls a month, bay |
| 03:09.41 | mil132 | scratch that, more like 6000 |
| 03:09.47 | dijib | pay p3nguin ! |
| 03:09.53 | mil132 | hahah |
| 03:11.07 | dijib | 6000 x 10min x $0.032 = $1920 for the toll free |
| 03:11.12 | dijib | does that make sense? |
| 03:11.13 | mil132 | fuuu |
| 03:11.15 | mil132 | yea |
| 03:11.26 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 03:11.28 | dijib | your using 808's right now? |
| 03:11.32 | mil132 | yea |
| 03:12.02 | dijib | p3nguin, do you know if they have the availability to have more concurrent channels beyond the package with 2? |
| 03:12.18 | p3nguin | dijib: Yes. Don't pay for flat rate. |
| 03:12.19 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 03:12.26 | p3nguin | dijib: With pay-per-minute, you get unlimited channels. |
| 03:12.35 | *** join/#asterisk ks3 (~ks3@cpe-184-57-153-87.cinci.res.rr.com) |
| 03:13.26 | mil132 | so question, with what you just said about pay-per-minuite... if I were to do that, we could theoreticly have 10 people calling that number and it would get routed all to the PBX? |
| 03:13.37 | dijib | yeah but thats expensive... even with the 808# he'ss be paying $894 |
| 03:13.46 | dijib | @ $0.0149 a min |
| 03:14.06 | dijib | mil132, yes |
| 03:14.14 | mil132 | ok |
| 03:14.36 | dijib | what happends now? they get busy signal? |
| 03:14.56 | mil132 | thanks for helping guys, I am so unknowlageable about all this... virtual beer for all of you |
| 03:15.19 | dijib | no thanks.. i had way too much of the real stuff last night |
| 03:15.22 | carrar | I'm virtually drunk |
| 03:15.37 | mil132 | I have some PBR waiting at home for me |
| 03:15.46 | carrar | Professional Bull Riding? |
| 03:15.56 | mil132 | Papst Blue Ribbin |
| 03:15.57 | carrar | They do that on the big island |
| 03:16.07 | dijib | lol |
| 03:16.57 | carrar | Quite the extensive cowboy history on the big island |
| 03:17.03 | carrar | I never knew that before |
| 03:17.19 | carrar | Cowboy's in Hawaii jsut didn't sound right |
| 03:17.20 | mil132 | its where most of the meat here comes from, the big island |
| 03:18.02 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-203.telkomadsl.co.za) |
| 03:18.51 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
| 03:18.54 | mil132 | so I dont think we will do the toll free number, sales would not be able to justify the cost |
| 03:19.12 | carrar | or find a cheaper TF provider :) |
| 03:19.24 | dijib | is their cheaper than voip.ms? |
| 03:23.00 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 03:23.31 | p3nguin | In theory, maybe. |
| 03:24.14 | SeRi | I found my self making to many outbound calls so metered outgoing calls does not work for me |
| 03:24.33 | SeRi | thats why I am at CC... I cant find any thing else cheaper :/ |
| 03:24.40 | SeRi | for out going calls |
| 03:24.57 | SeRi | I use voip.ms for inbound |
| 03:25.01 | SeRi | they rock :) |
| 03:25.14 | p3nguin | What's the rate on unlimited termination via CallCentric? |
| 03:25.30 | p3nguin | $20/mo? |
| 03:25.46 | SeRi | yes |
| 03:25.59 | mil132 | I am out guys... thanks for the info! |
| 03:26.15 | dijib | yeah thats for residential use |
| 03:26.24 | SeRi | yes |
| 03:27.33 | dijib | http://www.callcentric.com/dids/office_unlimited |
| 03:27.47 | p3nguin | We're talking termination, not DIDs. |
| 03:28.02 | SeRi | I spent about 25 dollars last month with voip.ms before I move move brother to voip.ms |
| 03:28.07 | SeRi | ^^ |
| 03:28.20 | SeRi | I also need PR included |
| 03:28.23 | dijib | move move brother? |
| 03:28.31 | SeRi | move my* |
| 03:28.33 | carrar | twice as fast as normal |
| 03:28.33 | p3nguin | I would just call SIP to SIP and forget the ITSP. |
| 03:28.51 | SeRi | p3nguin, that would be hard with the whole fam :( |
| 03:28.58 | carrar | Now offering "MOVE MOVE SIP" |
| 03:29.00 | p3nguin | But that's just me, and we all know no one listens to me. |
| 03:29.05 | SeRi | hells yea! |
| 03:29.24 | p3nguin | If they all have ATAs, it would be easy. |
| 03:29.31 | SeRi | p3nguin, they dont :( |
| 03:29.39 | carrar | I interconnect our whole families via SIP |
| 03:29.42 | p3nguin | Or they could even use soft phones if they have computers. |
| 03:29.47 | carrar | across two continents |
| 03:29.57 | carrar | works well |
| 03:30.53 | p3nguin | That's how I'd do it. |
| 03:31.10 | SeRi | carrar, cool. eventually ill do the same. I had issues creating calls betwenn asterisk to asterisk before due to 5060 having issues with comcast |
| 03:31.31 | carrar | tell comcast to stop blocking |
| 03:31.35 | p3nguin | Every member would have hir own extension. If a DID is needed, that could be added easily. |
| 03:31.47 | carrar | or use a different port |
| 03:32.04 | carrar | or put the traffic over openvpn |
| 03:32.19 | SeRi | I do that with mi otehr brother. |
| 03:32.29 | SeRi | works well over iphone and wifi |
| 03:32.47 | p3nguin | That brings up a question I'd like to understand: How do you get two asterisks interconnected using a non-standard port while keeping asterisk listening on the standard port? |
| 03:33.13 | SeRi | p3nguin, thats the issue we had when we where tryign to make mine work and it did not work. |
| 03:33.23 | p3nguin | Someone recently was asking about something similar, and when he used iptables to forward the non-standard outside port to the normal inside port, things did not work correctly. |
| 03:33.42 | SeRi | me! |
| 03:33.56 | p3nguin | There was someone trying it before I ever knew you. |
| 03:33.56 | SeRi | well not recently it was about a month a go. |
| 03:34.05 | SeRi | I see. |
| 03:34.52 | p3nguin | The problem was that even though iptables was forwarding, asterisk's ports in the packets were not coinciding and so not making the path between the two systems. |
| 03:34.58 | dijib | use the port setting on the register line? |
| 03:35.10 | dijib | nvmd |
| 03:35.29 | p3nguin | That tells your system which port to register to, but if the other system isn't listening on that port, it does no good. |
| 03:35.45 | dijib | ok then |
| 03:37.01 | p3nguin | Take this scenario as an example: I run asterisk on a standard port because the entire world works correctly on the standard port, except for one system I want to interconnect with, which has a port 5060 blockage. |
| 03:37.28 | p3nguin | My asterisk is not capable of listening on two ports. |
| 03:37.52 | p3nguin | So how do I accommodate the other system which needs to use the other port? |
| 03:38.10 | p3nguin | I could use SER, but that seems like a really big hammer for a tiny nail. |
| 03:38.12 | dijib | can you just register to your itsp on 5060 and the other system say 5080 |
| 03:38.13 | dijib | ? |
| 03:38.49 | dijib | redirect 5080 -> 5060 on your system? |
| 03:39.03 | p3nguin | I could register to the abnormal system on 5080 if I wanted to register to it. The problem is that my system is on 5060, and its outbound 5060 is blocked. |
| 03:39.04 | dijib | yes im guessing and no i havent read the book |
| 03:39.34 | p3nguin | s/its out/the abnormal system's out/ |
| 03:40.25 | dijib | cant use port forwarding? |
| 03:40.30 | p3nguin | If I had a SIP proxy, I could do it, but I don't really want to go to such extreme measure. |
| 03:40.49 | SeRi | p3nguin, I use siproxd before |
| 03:40.52 | SeRi | and it works. |
| 03:41.30 | p3nguin | I just got done explaining how forwarding the non-standard port broke communication because it wasn't rewriting the port inside the packet... |
| 03:41.42 | p3nguin | So forwarding does not work the way I had hoped. |
| 03:42.35 | p3nguin | siproxd, huh? I'll take a look. |
| 03:42.54 | [TK]D-Fender | p3nguin: minimalistic proxy. might do you well |
| 03:42.56 | p3nguin | <PROTECTED> |
| 03:43.06 | p3nguin | It sounds like the right tool for the job. |
| 03:43.33 | [TK]D-Fender | SER w/o the psycho mess L( |
| 03:43.49 | p3nguin | I'll have to give it a try next time I encounter that scenario. |
| 03:44.17 | SeRi | dijib, I am calling in now |
| 03:44.49 | SeRi | p3nguin, yeap and this the job well. very simple to setup. |
| 03:44.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:45.03 | SeRi | I had several servers behind it before. |
| 03:45.26 | p3nguin | Does it support a reasonable amount of traffic? |
| 03:45.28 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:45.38 | SeRi | Yes. |
| 03:45.46 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 03:46.09 | SeRi | I tested it at work for students. |
| 03:46.39 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 03:46.51 | SeRi | about 4 different server and 20 ata's behind each server |
| 03:47.21 | SeRi | I cant dial sipuri :( |
| 03:47.22 | dijib | im in conference. |
| 03:47.27 | dijib | and going for a smoke |
| 03:47.46 | SeRi | I am in |
| 03:47.48 | SeRi | :D |
| 03:47.52 | SeRi | it worked |
| 03:51.32 | p3nguin | channel originate SIP/your-phone-id application Dial SIP/2663@asterisk.serveirc.com |
| 03:51.46 | p3nguin | I don't like having to add a line in dial plan for a conference. |
| 03:51.52 | SeRi | yea it worked I for got to make the dial plan for the ata |
| 03:51.56 | SeRi | :) |
| 03:59.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:03.11 | SeRi | p3nguin, I am just testing :) |
| 04:03.45 | SeRi | we need your expertise at the chan p3nguin |
| 04:08.49 | p3nguin | ? |
| 04:09.21 | SeRi | p3nguin, we are talking about loging who comes in to conf calls |
| 04:09.23 | SeRi | jump in |
| 04:09.27 | p3nguin | Busy |
| 04:09.32 | SeRi | ok |
| 04:09.43 | p3nguin | Trying to get some food. |
| 04:09.55 | p3nguin | It's well past my supper time! |
| 04:10.10 | SeRi | ah! cool :) food is good :) |
| 04:10.42 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 04:11.31 | SeRi | ~book |
| 04:11.31 | infobot | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 04:11.43 | SeRi | ~freepbx |
| 04:11.43 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 04:11.47 | SeRi | ~ealstix |
| 04:11.54 | SeRi | ~ealastix |
| 04:12.01 | SeRi | ~elastix |
| 04:12.01 | infobot | it has been said that elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 04:12.03 | SeRi | lol |
| 04:12.05 | SeRi | damn |
| 04:15.30 | tapout | hey p3nguin, i enabled only ulaw ( g711u ) and premium bandwidth, no issues receiving faxes now. Thanks man |
| 04:15.43 | p3nguin | Great! |
| 04:15.54 | tapout | p3nguin, one thing I would like to know tho, is there a way to detect voice... "if no fax, send email saying, someone called on a voice call.. no fax detected, thus no issue with fax receiving?" |
| 04:16.12 | p3nguin | Are you not doing faxdetect on a shared phone number? |
| 04:16.18 | carrar | Y*A*W*N |
| 04:16.43 | p3nguin | Yes, there is a way to do that... it's very easy. |
| 04:16.48 | tapout | no, it's dedicated to only voice |
| 04:16.49 | tapout | err |
| 04:16.51 | tapout | only to fax |
| 04:16.55 | tapout | 800 number for faxes only |
| 04:17.18 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
| 04:17.48 | p3nguin | I'd set up fax detect anyway, and make my dedicated fax number into a shared number -- shared between a Playback saying "this is a fax only number," and the fax extension. |
| 04:19.11 | tapout | how tho? |
| 04:19.12 | p3nguin | You could even go as far as, "This is fax only number. Our voice line is 866-2-call-us. Goodbye." |
| 04:19.33 | tapout | won't that screw up the fax receiving if it detects a voice? |
| 04:19.46 | p3nguin | No, that's what faxdetect is for. |
| 04:20.09 | carrar | Just blast them with FAX TONES |
| 04:20.21 | p3nguin | You'll basically be changing the number over to a voice number which only plays back that message... plus has a fax extension. |
| 04:20.43 | p3nguin | So the primary purpose is still going to be fax only, but it will require the fax tones in order to be a fax number. |
| 04:20.56 | p3nguin | Or like carrar said, just blast them. That's what I do. |
| 04:21.24 | p3nguin | If you can't read "fax" next to my number before you dial, you're an idiot and deserve fax tones in your ear. |
| 04:21.38 | tapout | i blast them, but when my boss gets "failed fax from 1234567890, if you were expecting a fax... |
| 04:22.12 | p3nguin | You could eliminate that part of the hangup extension. |
| 04:22.22 | p3nguin | Then you'll only get successful faxes in email. |
| 04:22.23 | carrar | plug your FAX machine into a POTS line |
| 04:23.36 | SeRi | lol |
| 04:23.37 | SeRi | nice |
| 04:23.42 | SeRi | blast away! |
| 04:23.43 | SeRi | rofl |
| 04:23.46 | carrar | or a t.38 enabled carrier & ata at 9600 |
| 04:24.23 | SeRi | dijib, wtf was that? did sombody blasted your line? |
| 04:24.28 | p3nguin | lol |
| 04:24.29 | SeRi | lol |
| 04:24.52 | SeRi | p3nguin? LOL |
| 04:25.09 | p3nguin | channel originate SIP/2663@asterisk.serveirc.com extension fax@fax-in ? |
| 04:25.32 | SeRi | LOL |
| 04:25.43 | p3nguin | If I were not nice, I might consider that. |
| 04:26.24 | SeRi | he does not have announcement turn on. the line kept clicking as if somebody keep coming in and out |
| 04:27.10 | tapout | thanks so much for helping me p3nguin, I now have to try and figure out why my SSD (ocz failed... failed on my laptop, going to try and see if this bios picks it up) |
| 04:27.17 | SeRi | dijib, you still there or did you go def? |
| 04:27.27 | dijib | i went def |
| 04:27.36 | SeRi | lol |
| 04:27.37 | dijib | my portable analog died |
| 04:27.43 | p3nguin | I think all the clicking is his crappy phone. |
| 04:27.44 | dijib | and made a horrible noise |
| 04:27.52 | SeRi | LOL |
| 04:27.59 | SeRi | p3nguin, probably. |
| 04:28.22 | dijib | some panasonic |
| 04:28.32 | SeRi | :/ |
| 04:29.09 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 04:29.14 | dijib | whats wrong with that and an ATA? |
| 04:29.24 | dijib | you think everyone needs a wifi phone? |
| 04:29.28 | dijib | maybe |
| 04:29.45 | p3nguin | Most VoIP people do not recommend WiFi phones for VoIP. |
| 04:29.45 | SeRi | LOL I dont. I dont use wifi phones. they suck. |
| 04:29.59 | SeRi | ^^ |
| 04:30.09 | dijib | then how to portable? |
| 04:30.25 | carrar | they ok in low wifi usage areas |
| 04:30.35 | p3nguin | ATA and a cordless phone, or a cordless phone with a SIP base |
| 04:30.57 | carrar | use a DECT cordless SIP Phone |
| 04:31.02 | SeRi | I use a simens gigaset IP phone |
| 04:31.04 | dijib | then why chastsize me for my ata and cordless 6ghz? |
| 04:31.29 | SeRi | lol |
| 04:31.32 | dijib | what year is it? |
| 04:31.35 | p3nguin | I don't recall anyone performing that on you for using an ATA and a 6 GHz phone. |
| 04:31.48 | SeRi | ^^ |
| 04:31.59 | p3nguin | What we said was "crappy phone." |
| 04:32.07 | p3nguin | Well, I said it, and seri agreed. |
| 04:32.07 | SeRi | ^^ |
| 04:32.15 | SeRi | LMAO |
| 04:32.17 | SeRi | ^^ |
| 04:32.26 | dijib | you were aware of the phone in question |
| 04:32.43 | p3nguin | It may not have been so crappy before you bounced it off the concrete a couple times yesterday. |
| 04:32.51 | SeRi | rofl! |
| 04:32.55 | SeRi | I heard the storys |
| 04:34.35 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 04:34.36 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 04:35.03 | SeRi | cricket cricket.... |
| 04:36.01 | SeRi | dijib, if it makes you feel better panasonic phones and ata's are awesome. |
| 04:36.24 | dijib | thanks |
| 04:36.26 | carrar | hahah |
| 04:36.27 | SeRi | :) I still use an ata in the house. but no panasonic :) |
| 04:36.34 | dijib | oh yeh did it? |
| 04:36.38 | SeRi | LOL |
| 04:36.41 | dijib | well its a good real world test then |
| 04:36.45 | dijib | like the iphone that hit the rock |
| 04:36.48 | dijib | and everything else |
| 04:36.54 | dijib | it finally died |
| 04:37.23 | dijib | like the laptop that i put a couch on and then sat on it |
| 04:37.31 | dijib | still trucking on |
| 04:38.01 | *** join/#asterisk radic (~radic@dslb-094-216-254-180.pools.arcor-ip.net) |
| 04:38.16 | dijib | im thinking its batteries |
| 04:40.06 | SeRi | do you leave the phone on monitoring the chan? |
| 04:40.21 | dijib | no |
| 04:40.23 | dijib | not at all |
| 04:40.39 | SeRi | just asking :) |
| 04:40.49 | dijib | only my hangup and DID's in and out use monitor. hangup checks callerID and emails the file to that person. |
| 04:41.21 | dijib | so i burn bandwidth right after a call. |
| 04:41.39 | dijib | http://images.4chan.org/adv/src/1320206785943.jpg |
| 04:41.53 | SeRi | I meant like in the conf. are you in the conf monitoring who comes in? |
| 04:42.06 | dijib | no im not, i should |
| 04:42.57 | arnotixe | hi all I have two servers that used to communicate via iax2. One server (remote) is registering to the main. Now, on the remote, "iax2 show peers" shows the ip of the main server. However, on the main, there's (unspecified) as IP. hints on why? |
| 04:43.04 | SeRi | I am out guys. g/n |
| 04:43.16 | SeRi | Thanks for the help today p3nguin |
| 04:43.36 | SeRi | dijib, thanks for the ideas. and yes I know you are white. lol |
| 04:44.08 | dijib | i think every american home needs a sip phone. |
| 04:44.36 | dijib | think of the bandwidth not dedicated to pstn that could be opened up. |
| 04:47.10 | [TK]D-Fender | arnotixe: It hasn't registered |
| 04:48.18 | SeRi | r0m|u, go off line! |
| 04:52.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:00.10 | dijib | i thought you were goig to bed seri |
| 05:01.42 | SeRi | still here |
| 05:01.43 | SeRi | lol |
| 05:01.48 | SeRi | waiting on a call |
| 05:02.09 | dijib | ah |
| 05:02.14 | SeRi | looks like he is not going to call so I am about to really got to bed :) |
| 05:02.25 | SeRi | giving it a few more min |
| 05:02.37 | SeRi | you guys up at the chan? |
| 05:02.41 | dijib | Seri CallerID is the SIP/IP |
| 05:02.58 | dijib | nobody else is |
| 05:02.58 | SeRi | dijib, I figure. cool. |
| 05:03.42 | dijib | why doesnt ip show the configured callerID though? |
| 05:03.59 | SeRi | I am sure because is IP to IP |
| 05:04.12 | dijib | so they dont have those options? |
| 05:04.23 | p3nguin | It might show it if it is sent and received correctly. |
| 05:05.05 | SeRi | p3nguin, what do you mean? |
| 05:05.07 | dijib | it normally does over SIP |
| 05:05.43 | p3nguin | Send the caller ID number to the other side. If the other side parses it and displays it, then there is no problem. |
| 05:06.36 | SeRi | http://pastebin.com/3uD7V8vn |
| 05:06.38 | dijib | parses it out of the sip packet in the header or something? |
| 05:06.40 | SeRi | Thats what I have |
| 05:06.55 | dijib | i dont even know the sip error codes |
| 05:07.19 | p3nguin | I still don't understand why you need to lines to set callerid name and number. |
| 05:07.22 | dijib | i dont see that |
| 05:07.29 | dijib | odly enough my internal is 2666 |
| 05:07.49 | p3nguin | Set(CALLERID(all)=wtf <666>) |
| 05:07.52 | SeRi | for my own records p3nguin I record it on the cdr log |
| 05:07.54 | dijib | we know. |
| 05:07.58 | p3nguin | Do we? |
| 05:08.02 | SeRi | oo yea got cha |
| 05:08.03 | dijib | i did |
| 05:08.43 | p3nguin | But seri clearly did not. |
| 05:09.04 | dijib | i use that way too |
| 05:09.22 | SeRi | I did I just keep forgetting p3nguin I am so use too using two lines |
| 05:09.27 | SeRi | Set(CALLERID(all)=wtf <666>) |
| 05:09.46 | SeRi | I will slap my face to use that |
| 05:10.34 | raden | Katty, ??? you around ??? |
| 05:10.44 | p3nguin | She's in bed. |
| 05:10.54 | raden | well wtf :P |
| 05:11.00 | SeRi | I am out for sure now. |
| 05:11.01 | p3nguin | It's night time. |
| 05:11.02 | SeRi | g/n all |
| 05:11.14 | SeRi | and r0m|u fuck u! |
| 05:11.20 | SeRi | night |
| 05:12.26 | *** join/#asterisk eicto (~morgoth@eicto.broker.freenet6.net) |
| 05:13.04 | dijib | later seri |
| 05:13.29 | eicto | Hello, It is possible to initiate call to queue from command line ? (making alerting robot) |
| 05:13.29 | dijib | i think im going to call it to. meetings tomorrow |
| 05:13.56 | p3nguin | eicto: originate (channel originate) |
| 05:14.25 | p3nguin | What do you want to connect into the queue? |
| 05:14.47 | p3nguin | a SIP phone? an extension? |
| 05:15.02 | eicto | extention with several sip phones |
| 05:15.11 | eicto | may be not use queue, i not sure |
| 05:15.43 | p3nguin | Maybe I don't understand what you are trying to do. |
| 05:15.44 | eicto | Need to initiate call to several persons in round/robin style and play to 1st answered the message |
| 05:16.03 | p3nguin | I thought you wanted to originate a call from the CLI where one phone ends up in your queue. |
| 05:17.05 | eicto | I have several phones and need to be ensure that one of that phones will receive message |
| 05:17.38 | eicto | I also place question to stackoverflow |
| 05:17.51 | eicto | http://stackoverflow.com/questions/6138949/initiate-an-outgoing-call-with-asterisk |
| 05:18.07 | eicto | not that sorry :) |
| 05:18.42 | eicto | http://stackoverflow.com/questions/7975950/initiate-call-from-extention |
| 05:19.03 | p3nguin | Calls do not originate from extensions. |
| 05:19.08 | p3nguin | Calls can originate from phones. |
| 05:19.15 | p3nguin | Calls can be originated from the CLI. |
| 05:19.27 | eicto | but if i do call file, I initiate call |
| 05:19.34 | p3nguin | okay |
| 05:19.46 | eicto | i just need to call some "loop" phone |
| 05:20.16 | eicto | that will always answer, but can't find that it is possible |
| 05:20.33 | carrar | 127.0.0.1 |
| 05:20.51 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:20.51 | carrar | That guy is a jerk though |
| 05:21.08 | p3nguin | Create an extension that only does Answer() and Wait(). |
| 05:21.26 | p3nguin | You could call it extension "answerwait" for example. |
| 05:21.26 | eicto | but how to call it from cmdline ? |
| 05:21.53 | p3nguin | If it is in a context named "crap" ... |
| 05:22.05 | p3nguin | channel originate Local/answerwait@crap ... |
| 05:22.15 | p3nguin | What do you want it to connect to? |
| 05:22.27 | eicto | ah |
| 05:22.47 | eicto | I can call extentions with Local/ |
| 05:22.56 | eicto | thanks, it looks what i need |
| 05:23.12 | p3nguin | Yes, the local channel turns a regular extension in dialplan into a channel which can be called directly. |
| 05:23.12 | eicto | so I do: |
| 05:23.17 | eicto | [Local] |
| 05:23.44 | p3nguin | Your context does not need to be named |
| 05:23.44 | eicto | exten=>answerwait,1,Answer() |
| 05:23.46 | p3nguin | err |
| 05:23.47 | p3nguin | Your context does not need to be named 'Local' |
| 05:23.55 | p3nguin | but it can be if you want. |
| 05:24.02 | eicto | oh, yeah |
| 05:24.08 | eicto | [answerwait] |
| 05:24.13 | eicto | exten=>crap,1 |
| 05:24.15 | eicto | yes? |
| 05:24.18 | p3nguin | no |
| 05:24.23 | p3nguin | extension@context |
| 05:24.37 | p3nguin | That would be crap@answerwait |
| 05:24.41 | eicto | what to place to extention.conf in that case |
| 05:24.53 | p3nguin | The names will be completely arbitrary. |
| 05:24.59 | eicto | It seems i missed something about naming namespaces and extentions |
| 05:25.01 | p3nguin | I just prefer to make things sensible. |
| 05:25.16 | p3nguin | Choose a context where you are going to make your new extension. |
| 05:25.37 | p3nguin | For this, I would probably throw it into my misc context, since it is a miscellaneous item. |
| 05:26.09 | eicto | what is context in two words, i though it like [context] in extentions.conf |
| 05:26.16 | p3nguin | correct |
| 05:26.32 | p3nguin | [this-is-a-context] |
| 05:26.35 | eicto | but what is extention name (crap) in your example |
| 05:26.51 | p3nguin | The example I used was extension named answerwait. |
| 05:26.59 | p3nguin | In a context named crap. |
| 05:27.03 | eicto | ah, ok |
| 05:27.10 | p3nguin | Choose a context. |
| 05:27.17 | p3nguin | It can be any context, new or existing. |
| 05:27.44 | p3nguin | Decide what you will name the new extension. answerwait seemed reasonable to me, since you are going to make it Answer() and then Wait(). |
| 05:28.26 | p3nguin | exten => answerwait,1,Answer() |
| 05:28.32 | p3nguin | exten => answerwait,n,Wait() |
| 05:28.46 | p3nguin | save, exit, "dialplan reload" from the CLI. |
| 05:29.07 | p3nguin | Then call Local/answerwait@whatever-context-you-selected |
| 05:29.17 | eicto | and in call file i using Context: read_text that will do the rest ? |
| 05:29.44 | p3nguin | If the context name is in fact read_text, it's valid. |
| 05:29.52 | eicto | Yes |
| 05:30.21 | eicto | Thatnk you, much more clean asterisk logic now, I deal with it 2nd day only :) |
| 05:30.32 | eicto | *Thank |
| 05:33.47 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:39.21 | eicto | p3nguin, I tried, it not dial, may be i still do something wrong ? http://pastebin.com/12DkX0xQ |
| 05:40.10 | p3nguin | You duplicated priority 1 instead of copying what I typed. |
| 05:40.47 | p3nguin | line 3. Change 1 to n |
| 05:40.48 | eicto | not helps :( |
| 05:41.20 | p3nguin | And line 7, change 2 to 1, then delete line 6. |
| 05:42.00 | p3nguin | In other words, don't Answer() before Dial(). |
| 05:42.37 | arnotixe | hi all I've got the iax between two servers up again now. Don't get it completely, changed the password , iax reload, changed back and iax reload.. Whatever. |
| 05:42.48 | arnotixe | Now, can I make simultaneous IAX calls between the two servers? The first works, but the next simultaneous call fails on the originating server.. ? |
| 05:43.15 | eicto | It dials now, but reset immediatle |
| 05:46.20 | eicto | updated pastebin: http://pastebin.com/JM8eeAhd |
| 05:47.46 | p3nguin | You remembered to dialplan reload? |
| 05:48.33 | p3nguin | Show me a failure. So far all I see if a valid dial plan and a call file. If it fails, show me that it fails so I can see what went wrong. |
| 05:48.44 | p3nguin | s/if/is/ |
| 05:52.27 | eicto | I'll try, the hard thing that it is under freepbx control and I can't find logs, will try to do same on other installation of bare asterisk |
| 05:52.37 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 05:52.45 | carrar | who neds logs! |
| 05:52.48 | carrar | needs |
| 05:52.48 | p3nguin | That's why you should not be using FreePBX. |
| 05:57.50 | eicto | That not me :) |
| 05:58.08 | eicto | In my own asterisk i can't find any failur |
| 05:58.09 | eicto | In my own asterisk i can't find any failure |
| 05:58.25 | eicto | <PROTECTED> |
| 05:59.04 | p3nguin | core set verbose 3 |
| 05:59.07 | p3nguin | Try again. |
| 05:59.13 | p3nguin | Paste the log. |
| 06:01.18 | eicto | btw on bare asterisk it telling me sometime Failed to authenticate on INVITE to '"Alerts" |
| 06:01.26 | eicto | so I can't test often |
| 06:01.30 | eicto | any tricks ? |
| 06:03.18 | eicto | may be i reading incorrect logs ? |
| 06:05.00 | eicto | I trying /var/log/asterisk/messages |
| 06:05.18 | p3nguin | Like I said, core set verbose 3, make the call. |
| 06:05.30 | p3nguin | Doesn't involve /var/log/asterisk/messages |
| 06:07.29 | eicto | I tried |
| 06:07.32 | eicto | same result |
| 06:07.37 | eicto | no verbosity |
| 06:08.27 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:10.17 | eicto | i initiate call by copying call file to outbound |
| 06:14.24 | eicto | wait() need argument :) |
| 06:14.40 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 06:14.53 | p3nguin | No it doesn't. |
| 06:15.34 | p3nguin | It should just wait forever without specifying seconds to wait. |
| 06:17.25 | eicto | but when i add 500 there, it at least waited for my answer |
| 06:17.37 | p3nguin | Okay, that's good. |
| 06:20.13 | eicto | after I answer it not doing next step, but redial me |
| 06:20.16 | eicto | strange |
| 06:28.03 | eicto | looks like answerwait does not make calling state answered |
| 06:28.12 | eicto | it continues redial me |
| 06:28.38 | p3nguin | The Answer() makes it be answered. |
| 06:28.52 | eicto | if I answer it wait several seconds and redial me again while i on line |
| 06:30.25 | carrar | could cause a temporal rip in the time space continuum |
| 06:30.50 | eicto | ok, i fixed |
| 06:31.03 | eicto | I placed Dial to answerwait cotext |
| 06:31.12 | eicto | and my readtext without dial |
| 06:31.15 | eicto | looks working |
| 06:58.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:59.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:18.42 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:24.38 | *** join/#asterisk dom| (~domi@mail.tas.de) |
| 07:24.46 | dom| | hi |
| 07:25.12 | dom| | has anyone here the german voiceprompts from amooma? they are currently not available for download |
| 07:27.45 | irroot | today i think i may be arested |
| 07:27.54 | irroot | morning folks |
| 07:28.16 | WIMPy | I do only have a old collection from different sites. You may try #asterisk-de later. |
| 07:28.35 | WIMPy | Hi irroot. What's going on there? |
| 07:28.46 | dom| | WIMPy, useable for asterisk 1.8 with "eine"? |
| 07:29.07 | irroot | hehe i may murder someone all i can offer in defence is they stupid and annoying |
| 07:29.35 | WIMPy | I'm pretty sure, VM says "eine neue Nachricht". |
| 07:29.54 | WIMPy | Oh, that sort. |
| 07:30.07 | dom| | WIMPy, mhh cool, mine hooks up at "eine" ;) |
| 07:32.48 | dom| | asterisk-prompt-de is installed but digits/1F is missing |
| 07:33.26 | irroot | hehe dont scare me took german for 2 years at school |
| 07:39.55 | dom| | i took english some years at school ;) |
| 07:40.12 | p3nguin | Half the people around here need some more. |
| 07:42.19 | dom| | my english is quite terrible... |
| 07:46.34 | irroot | lol well i failed english in the end :P |
| 07:47.04 | WIMPy | How long ago was that? |
| 07:47.38 | irroot | dom| Namibia is still very german been there couple times finished school 21 years ago |
| 07:48.51 | dom| | ah cool, never been there |
| 08:06.55 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:07.14 | *** join/#asterisk SparFux (~raoul@rl2-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 08:08.33 | *** join/#asterisk Pegasus_RPG (~chatzilla@p5DD42F71.dip.t-dialin.net) |
| 08:24.52 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:25.00 | schmidts | good morning |
| 08:25.41 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:bd2d:c3c:81d:b47e) |
| 08:27.18 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-zapcxrgsgpbyxovn) |
| 08:27.54 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 08:28.07 | SparFux | hi |
| 08:28.36 | WIMPy | Moin SparFux |
| 08:31.42 | SparFux | hi WIMPy |
| 08:32.07 | SparFux | Is it advisable to use procfs in dahdi? AFAIK the /proc is for process info whereas dahdi is a driver. It uses /sys already. |
| 08:33.10 | WIMPy | It uses both so far. |
| 08:33.51 | SparFux | WIMPy: Yes it does. But it seems there is CONFIG_PROC_FS to enable / disable. |
| 08:33.55 | Pegasus_RPG | Hello. I'm having problems with call quality due to crappy internet connections. I've described my scenario fully here: http://pastebin.com/rt4D8vpz Can anyone offer any advice? |
| 08:34.09 | SparFux | I shall use CONFIG_PROC_FS in dahdi_hfcs too. |
| 08:34.14 | WIMPy | Pegasus_RPG: Get more internet? |
| 08:34.29 | Pegasus_RPG | WIMPy: My only other choices are cellular and satellite |
| 08:34.36 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 08:35.02 | Pegasus_RPG | Has anyone tried running * over a HSDPA/UMTS connection? |
| 08:35.24 | WIMPy | Definitely possible. |
| 08:36.21 | SparFux | WIMPy: I understand now. Only if kernel has procfs, dahdi will use it. |
| 08:36.33 | Pegasus_RPG | Possible yes, but what about call quality? I currently have issues with call quality on the edge-of-service DSL connection |
| 08:36.41 | WIMPy | And 60ms packes are pretty bad for call quality as well. |
| 08:37.21 | Pegasus_RPG | OK, I can drop those down. I originally set them high to decrease bandwidth on ulaw |
| 08:37.43 | WIMPy | If tour internet conection is'n good, don't try voip. |
| 08:37.53 | *** join/#asterisk pietro (~pietro@88-149-227-165.dynamic.ngi.it) |
| 08:38.12 | WIMPy | With ulaw you can only get one call through your line. |
| 08:38.22 | WIMPy | I hope that line isn;t used for anything else. |
| 08:38.30 | Pegasus_RPG | That's why I moved * to the data center: so I could use a more efficient codec |
| 08:39.05 | WIMPy | Where is the relation there? |
| 08:41.32 | Pegasus_RPG | The VoIP provider onyl supports ulaw |
| 08:41.56 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-135-115.chyn.qwest.net) |
| 08:42.03 | Pegasus_RPG | So moved my * into the data center (with good internet) so I could use G.723/729 between my office phones and * |
| 08:42.05 | WIMPy | Ah |
| 08:42.28 | Pegasus_RPG | or gsm.. almost anything is better than G.722/ulaw/alaw |
| 08:43.31 | carrar | If you are having issues with 88k for a phone call, you'll have problems with 30k |
| 08:43.42 | carrar | most likely |
| 08:44.00 | Pegasus_RPG | carrar: actually, the 88k is fine as long as the connection isn't used for anything else |
| 08:44.17 | WIMPy | Get a phone line to your Asterisk server and use teh PST to call it. |
| 08:44.22 | WIMPy | PSTN |
| 08:44.46 | WIMPy | If you use the line for other things as well, you need a router with good TC. |
| 08:44.47 | Pegasus_RPG | :P that's expensive...office is in Germany, * is in the US |
| 08:44.55 | carrar | Pegasus_RPG, thats true for any codec |
| 08:45.05 | WIMPy | Oterhwise you don;t have any cahnce of acceptable voice quality. |
| 08:45.13 | Pegasus_RPG | Yeah, so I was thinking of installing Tomato locally |
| 08:45.17 | WIMPy | Put a * in the next village. |
| 08:45.18 | carrar | You need to use QoS on both ends of your wan link |
| 08:45.32 | Pegasus_RPG | WIMPy: haha that's a good idea...they get 7Mbps there |
| 08:45.51 | WIMPy | Good. |
| 08:46.03 | WIMPy | Get a phone flat and use that. |
| 08:46.29 | Pegasus_RPG | phone flat? you man flat rate phone plan? |
| 08:46.42 | WIMPy | yes |
| 08:47.00 | Pegasus_RPG | hmm, interesting. That precludes doing two calls at once though |
| 08:47.05 | WIMPy | If you haven't got one anyway. |
| 08:47.12 | WIMPy | Why? |
| 08:47.29 | Pegasus_RPG | One analog PSTN channel is one call |
| 08:47.46 | WIMPy | Didn't you say Germany? |
| 08:47.49 | Pegasus_RPG | Yes |
| 08:48.05 | WIMPy | So the standard connection would be BRI. |
| 08:48.23 | Pegasus_RPG | Really? Seems mine is POTS |
| 08:48.38 | carrar | get two internet connections |
| 08:48.40 | carrar | once for voice |
| 08:48.42 | carrar | one for data |
| 08:48.53 | carrar | and never the two shall cross :) |
| 08:49.01 | Pegasus_RPG | that's a good idea too |
| 08:49.07 | WIMPy | Depending on your provider you should be able to upgrade to BRI for 2-4 bucks. |
| 08:49.39 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 08:50.24 | WIMPy | doesn't know who uses POTS except for your granny :-) |
| 08:50.49 | Pegasus_RPG | haha |
| 08:51.05 | Pegasus_RPG | Well, this is DSL which is delivered over POTS |
| 08:51.15 | WIMPy | No. |
| 08:51.18 | Pegasus_RPG | no?? |
| 08:51.32 | carrar | delivered over copper |
| 08:51.34 | WIMPy | In Germany we always use Annex B. |
| 08:51.41 | WIMPy | Annex A does not exist here. |
| 08:51.50 | Pegasus_RPG | Right, my modem is Annex B (and M) |
| 08:52.01 | Pegasus_RPG | I never knew what that meant |
| 08:52.22 | WIMPy | A=DSL/POTS B=DSL/BRI |
| 08:52.55 | Pegasus_RPG | So how could I get a dial tone when I plugged my analog phone set into the same wire pair the DSL connection is coming on? |
| 08:53.05 | WIMPy | So even if you have POTS, it will be Annex B. All the same for everyone. |
| 08:53.08 | Pegasus_RPG | rather how _did_ I? |
| 08:53.50 | WIMPy | How did you order that? :-) |
| 08:54.06 | Pegasus_RPG | Via Telekom's web site, using Google Translate :) |
| 08:54.13 | Pegasus_RPG | is from the US |
| 08:54.26 | carrar | heh |
| 08:54.27 | WIMPy | Obviousely you've got POTS. But you probably didn't want to :-) |
| 08:54.46 | WIMPy | Their website is horrible. |
| 08:54.50 | *** join/#asterisk pietro1 (~pietro@88-149-227-165.dynamic.ngi.it) |
| 08:54.50 | Pegasus_RPG | My original intention was to get bare Internet service and use VoIP |
| 08:54.52 | carrar | googles? |
| 08:54.53 | carrar | yeah |
| 08:54.54 | Pegasus_RPG | yes... yes it is |
| 08:55.01 | Pegasus_RPG | no, T-com |
| 08:55.12 | WIMPy | DTAG |
| 08:55.34 | Pegasus_RPG | I'd done the same in various US locations without a problem, but there I had a minimum of 384K upstream |
| 08:55.46 | WIMPy | That's what you usually get these days anyway. |
| 08:56.03 | Pegasus_RPG | I'm paying for 2M down, 768K up but I'm too far from the CO |
| 08:56.19 | carrar | no cable? |
| 08:56.21 | Pegasus_RPG | nope |
| 08:56.26 | Pegasus_RPG | this is the country :) |
| 08:56.28 | carrar | move out of the BFE |
| 08:56.48 | WIMPy | Cable is usally a good option in the bush. |
| 08:56.49 | Pegasus_RPG | Beautiful scenery, crappy Internet |
| 08:56.58 | WIMPy | But extremely unreliable here :-( |
| 08:57.00 | carrar | You live in 15th century castle on the rhineriver or something? |
| 08:57.08 | Pegasus_RPG | Is there anyone other than Kabel Deutschland? They don't service my address |
| 08:57.36 | WIMPy | No, there is only one cable provider per area. |
| 08:57.45 | Pegasus_RPG | carrar: actually in a small scenic village. Makes the wife happy, but not when we need to make calls :P |
| 08:57.58 | carrar | become a provider |
| 08:58.17 | Pegasus_RPG | carrar: I thought about that too...get a nice OC3 here huh? |
| 08:58.25 | carrar | well you might not get that |
| 08:58.32 | carrar | but might be able to get a E1 |
| 08:58.32 | Pegasus_RPG | All of my neighbors have sat TV, so I don't think cable is an option either |
| 08:58.35 | carrar | and resell it |
| 08:58.36 | WIMPy | Well, go upgrate to BRI ("universalanschluss") and use that. |
| 08:58.38 | carrar | via wifi! |
| 08:59.12 | carrar | string fiber over the clay rooftops |
| 08:59.20 | Pegasus_RPG | carrar: I really thought about that!! |
| 08:59.24 | carrar | once you clean them of goats |
| 08:59.28 | Pegasus_RPG | lol |
| 08:59.29 | WIMPy | Cabel for TV is just too expensive. That's why everyone sets up their own dish. |
| 08:59.54 | carrar | or cardshares |
| 09:00.07 | Pegasus_RPG | WIMPy: so how do I find out who services this area |
| 09:00.23 | carrar | ask the telco erpair people |
| 09:00.40 | Pegasus_RPG | They'll tell me who their competitors are?? |
| 09:00.49 | carrar | probably not |
| 09:00.52 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 09:00.55 | WIMPy | Most probably Kabel Deutschland. |
| 09:00.56 | carrar | but they will give you a start |
| 09:01.04 | carrar | probably only one option anyways |
| 09:01.36 | carrar | setup wifi links on the hilltops to the nearest city |
| 09:01.42 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 09:01.46 | WIMPy | It's pretty easy to persuade KDG to send out diggers. |
| 09:01.49 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:01.54 | carrar | go with 400mbps links! |
| 09:02.03 | WIMPy | But you seriousely don't want to use them for VOIP. |
| 09:02.17 | Pegasus_RPG | carrar: If I had the capital to do that for all the villages around here, that could be very lucrative |
| 09:02.31 | carrar | get some funding |
| 09:02.43 | carrar | make your dreams come true! |
| 09:02.51 | WIMPy | There are many areas where that would make sense. |
| 09:03.14 | Pegasus_RPG | WIMPy: so to be clear, with a BRI, I would set up an * in my office, connect it to the BRI for two voice channels, then have it call the other * in the village with good internet, then call wherever from there? |
| 09:03.37 | WIMPy | The trouble is that if the gig ones find out, someone is planning some alternative supplies thay are fast to connect that area themnselves. :-( |
| 09:03.50 | carrar | win win |
| 09:03.55 | WIMPy | Yes, that'd be my suggestion. |
| 09:04.04 | Pegasus_RPG | oh, so all I have to do is threaten to do it and they'll hook me up :) |
| 09:04.07 | carrar | start the rumors |
| 09:04.36 | WIMPy | But check prices. If you have a DTAG line you can use call-by-call which is often even cheaper than voip. |
| 09:04.37 | Pegasus_RPG | Sounds like lots of dialplan fun |
| 09:04.43 | WIMPy | Might work. |
| 09:04.45 | carrar | get your farmer friends to dig some trenches for fiber |
| 09:04.50 | Pegasus_RPG | DTAG? I'll look that up |
| 09:05.01 | WIMPy | Deutsche Telekom AG |
| 09:05.34 | Pegasus_RPG | Oh. yeah I'm on the per-minute plan now |
| 09:05.59 | carrar | What do you do in a small german town for work? |
| 09:06.03 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:06.10 | Pegasus_RPG | IT support for US clients |
| 09:06.10 | WIMPy | And of you want to send calls via an external Asterisk, you could use an Asterisk at home as well, get the option to use subaddressing and route calls that way. |
| 09:06.13 | carrar | make nut crackers? |
| 09:06.34 | WIMPy | What area are you in? |
| 09:06.43 | WIMPy | Geographical that is. |
| 09:06.49 | carrar | Did you tell your wife you need to live someplace that has high speed internet? |
| 09:07.16 | carrar | clearly in Germany, the women wear the pants! |
| 09:07.28 | schmidts | carrar not only in germany :D |
| 09:07.48 | Pegasus_RPG | I'm in Rhineland-Pfalz, south of Koblenz |
| 09:08.07 | carrar | haha |
| 09:08.32 | carrar | quite different here in Japan |
| 09:08.36 | Pegasus_RPG | carrar: I understand I'm actually lucky to have DSL at all in this area. Frequently there are waiting lists for DSLAM ports so people have only dial-up |
| 09:08.44 | Pegasus_RPG | unless we lived in a city |
| 09:09.07 | carrar | ask them if you can get bonded DSL |
| 09:09.14 | carrar | we offer that in Seattle |
| 09:09.39 | schmidts | carrar for bonded dsl you need even more cooper lines and dslam ports ;) |
| 09:09.42 | carrar | easy way to get 10, 20megs of internet on the cheap |
| 09:09.46 | WIMPy | If the lines are too long, you probaly won't get much more speed from multiple lines. |
| 09:09.50 | carrar | yes |
| 09:09.58 | WIMPy | They might even get slower. |
| 09:10.10 | carrar | two slow DSL's is better hten 1 slow DSL :) |
| 09:10.17 | Pegasus_RPG | I'm actually not so worried about speed. Just quality. If QoS actually worked, I'd be fine |
| 09:10.18 | carrar | two slow bonded that is |
| 09:10.31 | schmidts | Wimpy it depends on how many lines you take :D we are testing a router from patton which can make upt to 45 mbit with 8 copperpairs on around 1,5 km |
| 09:10.46 | WIMPy | Not of two DSL on the same bundle of copper makes it fail all together. |
| 09:11.19 | WIMPy | schmidts: Sure. If the lines are ok. |
| 09:11.31 | WIMPy | But that doesn't seem to be the case for Pegasus_RPG |
| 09:11.34 | Pegasus_RPG | schmidts: so if I picked up a pair of those, got a DSL connection in the next town at 7Mbps, and ran a bunch of copper across the fields, I'd be in good shape? :) |
| 09:11.39 | schmidts | and if you loose all your luck, it could even happen that both copper lines are near together so they will disturb each other, in fact you will have even more problems with bundeling then ;) |
| 09:12.02 | carrar | Pegasus_RPG, run fiber |
| 09:12.03 | WIMPy | That's what I suggested. |
| 09:12.05 | carrar | longer distance |
| 09:12.07 | carrar | less loss |
| 09:12.12 | schmidts | Pegasus_RPG if you have the ability to run something across a field, take a fiber cable ;) |
| 09:12.13 | carrar | use LX Lasers |
| 09:12.18 | Pegasus_RPG | well sure |
| 09:12.19 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 09:12.23 | carrar | do it on the cheap with ebay cisco stuff |
| 09:12.29 | WIMPy | Ask the next farmer to burry some fiber. |
| 09:12.39 | Pegasus_RPG | I could probably do that.. |
| 09:12.42 | carrar | & dinner, bbl |
| 09:12.43 | schmidts | Lasers are not a good idea if there will be snow or fog sometimes |
| 09:12.54 | carrar | under the frost line |
| 09:13.20 | Pegasus_RPG | But at that point, isn't that more expensive than just getting a second DSL connection at €30 per month? |
| 09:13.21 | carrar | need to burryw it obviosuly :) |
| 09:13.39 | WIMPy | A simple wifi link can often be good enough. |
| 09:14.12 | Pegasus_RPG | as in 802.11g? |
| 09:14.18 | Pegasus_RPG | with a big-ass antenna |
| 09:14.27 | WIMPy | I'd go for 11a. |
| 09:14.32 | Pegasus_RPG | n? |
| 09:14.41 | WIMPy | Don't get cought with that :-) |
| 09:14.45 | Pegasus_RPG | :) |
| 09:15.14 | Pegasus_RPG | Yeah for the speeds we're taking, a would be fine |
| 09:17.19 | Pegasus_RPG | But how much does long-distance wifi antenna & amplifier equipment cost? |
| 09:17.52 | schmidts | which long distance we are talking about? |
| 09:18.12 | WIMPy | Try http://www.interprojekt.com/ |
| 09:18.34 | WIMPy | Yes, how far away is the nex acceptable connection? |
| 09:18.52 | Pegasus_RPG | schmidts: less than 1km, over a hill |
| 09:19.04 | Pegasus_RPG | (I'm in a valley) |
| 09:19.22 | WIMPy | Bad situation for wifi. |
| 09:19.23 | Pegasus_RPG | got a clear line of sight to the T-com cell tower though :P |
| 09:19.38 | Pegasus_RPG | I can actually see it clearly from my house |
| 09:20.06 | Pegasus_RPG | I guess cellular is more profitable than DSL |
| 09:20.20 | Pegasus_RPG | Since that tower was added in just the last couple years |
| 09:20.25 | schmidts | maybe you can think about a bundled GSM or EDGE or UMTS connection |
| 09:21.21 | Pegasus_RPG | I did. I haven't tested it though. Would the latency/quality be good for VoIP, running * behind it? |
| 09:22.03 | WIMPy | Probaly. |
| 09:22.15 | WIMPy | But the mobile data plans are unfriendly. |
| 09:22.24 | Pegasus_RPG | They have unlimited ones |
| 09:22.29 | WIMPy | You only get limited volume. |
| 09:22.36 | WIMPy | Not really. |
| 09:23.03 | WIMPy | Yes, it's unlimited, but after 3 or 5 G you get limited to 56kbps. |
| 09:23.06 | schmidts | Wimpy we also have some flat rates for mobil here in austria :P only germany is a little bit behind with these rates |
| 09:23.17 | WIMPy | So you won;t transfer much thereafter. |
| 09:24.17 | Pegasus_RPG | yick |
| 09:24.48 | WIMPy | What about a direct sat uplink? |
| 09:25.12 | Pegasus_RPG | That's possible I think. SkyDSL |
| 09:25.22 | Pegasus_RPG | They say the latency is low |
| 09:25.31 | Pegasus_RPG | but I have a hard time beliving that |
| 09:25.48 | WIMPy | That's obviousely impossible. |
| 09:26.27 | WIMPy | Even if there is ongoing discussion about the speed of light not being the limit. For existing technology it is. |
| 09:27.23 | Pegasus_RPG | haha right |
| 09:28.22 | Pegasus_RPG | https://de.skydsl.eu/index.php?c=order&s=tariff |
| 09:29.19 | WIMPy | Ok, where's the catch? |
| 09:31.13 | Pegasus_RPG | http://de.skydsl.eu/index.php?c=info&s=faq&cs=technic |
| 09:35.13 | Pegasus_RPG | Ah ha... you need a secondary Internet connection |
| 09:35.31 | WIMPy | No |
| 09:35.33 | Pegasus_RPG | So it's mainly for those who need speed. I just need quality |
| 09:35.48 | WIMPy | See th SkyDSL2+ products. |
| 09:37.36 | Pegasus_RPG | oh |
| 09:41.50 | Pegasus_RPG | looks like the 6000 plan is the minimum for VoIP...192kbps upstream cuts it too close |
| 09:42.17 | WIMPy | You always need TC. |
| 09:43.57 | Pegasus_RPG | I guess I better try that before I do anything esle |
| 09:44.09 | Pegasus_RPG | That's different from QoS? |
| 09:44.31 | WIMPy | No, just a more generic term. |
| 09:44.33 | Pegasus_RPG | googles |
| 09:44.48 | WIMPy | there are may approaches. |
| 09:50.10 | carrar | TC is more of a linux term |
| 09:50.17 | carrar | QOS is more of network term |
| 09:52.05 | schmidts | you dont need Qos if you use this for Voip only :D |
| 09:52.25 | Pegasus_RPG | schmidts: well, my first step is to optimize the heck out of what I currently have |
| 09:52.37 | Pegasus_RPG | If it is still not sufficient, then i will look at changing the link |
| 09:52.50 | schmidts | ok ;) |
| 09:53.11 | Pegasus_RPG | (because any optimizations I do for a bad link will work even better on a good one, right?) |
| 09:53.55 | WIMPy | yes |
| 09:54.29 | Pegasus_RPG | Let me just ask this: with a 10Mb connection in the data center (where * is currently) but no QoS on the firewalls there, is that a problem? |
| 09:54.48 | carrar | 10mb isn't much |
| 09:55.16 | Pegasus_RPG | carrar: but for four voice channels? |
| 09:55.19 | WIMPy | Depends on what else uses that link. |
| 09:55.29 | *** join/#asterisk Blackvel (~blackvel@dslb-088-065-101-069.pools.arcor-ip.net) |
| 09:55.35 | carrar | nothing else but voice? |
| 09:55.47 | Pegasus_RPG | lots of other stuff, but the link usually hovers at around 512Kbps-768Kbps |
| 09:56.03 | carrar | try it |
| 09:56.06 | carrar | see how it goes |
| 09:56.25 | Pegasus_RPG | I have tried it. That's what I'm currently using and the called parties hear me break up sometimes, other times its fine |
| 09:56.34 | carrar | like you have any other choices? |
| 09:56.43 | Pegasus_RPG | I did just decrease the packet size on my phones though |
| 09:56.52 | carrar | those breakups is other traffic clobbering your voice traffic |
| 09:56.53 | Pegasus_RPG | I do: I could replace the firewalls |
| 09:57.04 | carrar | welcome to the internet |
| 09:57.10 | Pegasus_RPG | carrar: but how can I tell where that's happening? in the DC or my office internet? |
| 09:57.23 | carrar | graph every port |
| 09:57.40 | carrar | MRTG |
| 09:57.42 | Pegasus_RPG | port like UDP port? |
| 09:57.49 | carrar | physical switch ports |
| 09:58.19 | WIMPy | MRTG is far too slow. |
| 09:58.26 | carrar | MRTG is jsut fine |
| 09:58.37 | carrar | has nothign to do with speed |
| 09:58.42 | carrar | you are polling interface coutners |
| 09:58.50 | Pegasus_RPG | I have been thinking about replacing the switches and firewalls in the data center with all QoS/802.11q-capable equipment |
| 09:58.53 | WIMPy | What use is a 5 minute average is a burst of far less than a second will cause dropouts? |
| 09:59.09 | Pegasus_RPG | WIMPy: the breaking up happens over many seconds |
| 09:59.14 | Pegasus_RPG | it's quite irritating |
| 09:59.16 | carrar | a 5 min burst will show up in counters |
| 09:59.43 | WIMPy | Yes, but if you have 5 Minute bursts you alredy know you're screwed. |
| 09:59.45 | carrar | and for most practical evirments it will give him a clue as to what is using more traffic |
| 10:00.07 | carrar | assuming his switch supports SMTP |
| 10:00.10 | carrar | err |
| 10:00.24 | WIMPy | Maybe, but still no chance to see the evil short peaks. |
| 10:00.26 | carrar | SNTP |
| 10:00.34 | carrar | ok can't type |
| 10:00.41 | WIMPy | SNMP? |
| 10:00.44 | Pegasus_RPG | I know, they do |
| 10:00.52 | Pegasus_RPG | I haven't set it up, but they do |
| 10:00.55 | carrar | snmp |
| 10:01.04 | carrar | set it up |
| 10:01.07 | carrar | graph your ports |
| 10:01.15 | carrar | set it to poll every min if 5 min isn;t enough |
| 10:01.41 | carrar | 1 switch is npt going take much processing power |
| 10:02.49 | Pegasus_RPG | you know, now that you mention it... I bet I know what's causing it. The * server competes with the other servers in the DC LAN that talk to each other at 1Gbps |
| 10:02.57 | Pegasus_RPG | with bursts of traffic |
| 10:03.08 | Pegasus_RPG | I'll set up MRTG to be sure |
| 10:03.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:03.42 | Pegasus_RPG | Does * automatically tag its packets with 802.11q? or is that the switch's job? |
| 10:03.52 | Pegasus_RPG | err QoS flag I means |
| 10:04.03 | WIMPy | See sip.conf |
| 10:04.59 | carrar | switches can look at COS |
| 10:05.46 | carrar | in the 802.1Q ethernet frame |
| 10:14.02 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 10:14.29 | DennisG | hi all |
| 10:14.56 | DennisG | is here someone with a large Asterisk setup? |
| 10:15.19 | carrar | no |
| 10:15.36 | DennisG | large is like 500+ registrations.. |
| 10:16.17 | wdoekes2 | ~ask |
| 10:16.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:16.19 | carrar | Why would someone respond to that |
| 10:16.21 | carrar | heh |
| 10:16.29 | DennisG | hehe carrar |
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| 10:16.55 | wdoekes2 | is there someone drinking coffee at the moment? |
| 10:17.24 | DennisG | well.. i have a question about it. i'm thinking about making a new platform with a minimum of 3 Asterisk boxes. Box A is the registration server and Box B + C is just for the calls |
| 10:17.44 | DennisG | but i don't know if it's a good idea to have 1000+ simultanious |
| 10:17.57 | carrar | Why not use opensips or kamilio |
| 10:17.59 | DennisG | registrations on 1 asterisk box (sorry for the enter) |
| 10:18.00 | kaldemar | sounds like it's not asterisk you want. |
| 10:18.43 | DennisG | yeah i know. opensips looks great but i need the blf functions (presence) and opensips/openser have problems with it |
| 10:19.58 | carrar | How is A not going to be handling all the calls? |
| 10:20.07 | schmidts | does anyone of you use patton isdn voip atas? like the smartnode 4552 |
| 10:20.15 | schmidts | i have a problem with the daylight saving rule |
| 10:20.20 | DennisG | Asterisk have everything what i want. A lot of features and it's very stable. but if i use opensips for the registrations then i believe that i need opensips for the features. |
| 10:20.35 | DennisG | With Dundi carrar |
| 10:20.59 | schmidts | DennisG opensips is a proxy, asterisk is a b2bua which isnt really the same, normally you combine both so you have a proxy in front of an asterisk server |
| 10:23.08 | DennisG | oke. but i need asterisk for the cool features like blf (presence), queues, etc.. |
| 10:23.43 | DennisG | is it possible to use open sips just for registering + load balancing WITH memory and use Asterisk for all other features? |
| 10:27.33 | DennisG | for load balancing i need to put all customers of 1 location on 1 asterisk box (for transferring calls). |
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| 10:33.10 | schmidts | DennisG yes thats exactly whats a proxy is used for |
| 10:33.10 | schmidts | blf might be possible only with a proxy but this feature is still very very untested and far away from real production state ;) |
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| 10:51.50 | *** mode/#asterisk [+o ChanServ] by calvino.freenode.net |
| 10:59.09 | TSM | is there any way to notify the user when he enables call recording? |
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| 11:02.42 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 11:03.55 | DennisG | sorry i'm back. problem with freenode |
| 11:06.17 | DennisG | anybody here? :P |
| 11:09.43 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 11:10.12 | DennisG | splitbrained irc? |
| 11:18.11 | schmidts | ;) |
| 11:19.16 | eicto | Debian irc had good intro - don't ask for ask - just ask, so better to ask instead of ask for ask |
| 11:21.07 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 11:21.09 | hetii | Hello |
| 11:21.10 | hetii | :) |
| 11:21.28 | DennisG | did i missed anything about my question btw? ^^ |
| 11:22.32 | hetii | I had a trouble with * and freepbx. I set few trunk(just user type) for incoming trafic. I do it on PEER section cause it is mandatory, i also set the register string on form user:pass@voipProvider/DID. The problem is that when someone call to this trunk * start executing it like did@from-trunk (what is ok) but sometimes like did@from-sip-external. What could be a reason that he put this incoming trafic to from-sip-external instead from-trunk |
| 11:22.50 | arnotixe | DennisG, no, you asked if anyone was around. You're answering your question. You are around. |
| 11:23.34 | kaldemar | hetii: someone at #freepbx probably knows. |
| 11:23.38 | DennisG | arnotixe, i asked a few other things ;) |
| 11:24.07 | hetii | btw. on the register string as we know its possible to provide the DID but this is information for * or for provider? if its for * then is it a real did or some cumtom name of some context ? |
| 11:25.35 | wdoekes2 | hetii: I'm assuming you're talking about the user-part of the Contact: header? a proper registrar shall accept anything you put there |
| 11:26.09 | kaldemar | hetii: it is for the other end to let them know what number they should use when calling asterisk. |
| 11:27.13 | hetii | ok thx, so then it means it could be a name of the trunk and by this the * should know with context should be used |
| 11:27.33 | hetii | but thats the problem somtimes it works sometimes not. |
| 11:28.09 | kaldemar | hetii: no. |
| 11:34.50 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 11:36.35 | nicola_pav | hello. on a busy asterisk server, its it essential to modify the rmem and wmem kernel default values? I am experiencing networks problems. there r disconnections with sip extensions |
| 11:37.07 | nicola_pav | i could find anything in google. is there a manual or guide how to setup those rmem and wmem, max and defaults? please advise |
| 11:37.34 | nicola_pav | i count not* |
| 11:38.06 | DennisG | nicola, what's the load of the asterisk box? |
| 11:40.06 | nicola_pav | DennisG: i have 748 sip ext |
| 11:44.42 | DennisG | nicola_pav, what's the cpu load? (i think that i can't use notice with colloquy haha) |
| 11:47.29 | nicola_pav | DennisG: it has 4 cores |
| 11:47.53 | nicola_pav | DennisG: u mean what i see in htop? |
| 11:48.34 | DennisG | yeah |
| 11:48.52 | schmidts | nicola_pav which asterisk version do you use? |
| 11:49.10 | nicola_pav | cpu seems fine, load average: 2.65 |
| 11:49.22 | nicola_pav | shmidts: asterisk 1.4.36 |
| 11:49.28 | schmidts | with how many concurrent calls? |
| 11:49.53 | nicola_pav | shmidts: more than 50 |
| 11:50.24 | schmidts | imho a little bit high for this amount |
| 11:50.28 | nicola_pav | the server has also 5 pris |
| 11:50.45 | DennisG | are the channels SIP or DAHDI? |
| 11:50.54 | nicola_pav | schmidts: what do u mean by high? which is high? |
| 11:51.01 | DennisG | due network issues with switches or something like that.. |
| 11:51.02 | WIMPy | Any transcoding going on? |
| 11:51.43 | nicola_pav | DennisG: calls are DAHDI and sip |
| 11:51.47 | schmidts | nicola_pav the load is a little bit high for so less calls, i have this amount with around 130 or 150 concurrent calls but these calls are from sip to zap with 8 Pris |
| 11:52.14 | nicola_pav | WIMPy: transcoding would result in high CPU if there is a problem |
| 11:52.15 | schmidts | but this server runs 1.2 so it might be ok for you ;) |
| 11:52.16 | nicola_pav | right? |
| 11:52.21 | nicola_pav | but cpu seems normal |
| 11:52.34 | schmidts | nicola_pav transcoding allways cause high cpu even without a problem ;) |
| 11:53.12 | nicola_pav | schmidts: transcoding involve translating between codecs, right? |
| 11:53.44 | DennisG | you can try to offload some calls to check if it will help :) |
| 11:55.00 | nicola_pav | i ran netstat -s and under Udp i have a lot of RecBufErr |
| 11:55.04 | DennisG | or test with a few calls.. if that solved the problem then check it with more calls |
| 11:55.08 | nicola_pav | is it a problem? |
| 11:55.55 | schmidts | nicola_pav normal calls over a PRI are allways g711u/a if someone use gsm or g729 you have to transcode the audio to make it fit for the pstn world |
| 11:56.49 | DennisG | sorry have to go.. the Internet guy is here for a new connection.. -_- |
| 11:56.50 | WIMPy | They can also use G.722 if any channel supported it. |
| 11:56.54 | nicola_pav | can i run sth in asterisk cli to check for transcoding? |
| 11:56.57 | DennisG | good luck nicola! |
| 11:57.29 | nicola_pav | DennisG: i think i dont have a choice but to unload calls, i will see |
| 11:57.42 | nicola_pav | i just want to makre sure its not the kernel or asterisk open files |
| 11:58.52 | WIMPy | If you ran out of open files you would definitely see that. |
| 11:59.25 | nicola_pav | WIMPy: yeah i remember now, in log files, cannot open socket |
| 11:59.36 | nicola_pav | what about kernel buffer values? receive and send? |
| 12:01.32 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:01.47 | WIMPy | I've never seen RevBufErr, but I'm confident it's bad. |
| 12:07.45 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 12:21.51 | Blackvel | hi all. got varius RTP floods past days (directip). is it enough to limit udp 5060 registers/invites or is there some more firewall protection (limit, recent, string, etc.) for rtp 5004+-x range required? found even a tool for rtp flooding |
| 12:22.09 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:23.02 | Dovid | Blacklevel: You can ajust the ports used in rtp.conf |
| 12:23.06 | Dovid | and then block everything else |
| 12:23.48 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:25.38 | Blackvel | well it happens that there are like 16+ channels open |
| 12:26.06 | Blackvel | as the flood script hammers my asterisk which also seem to answer / pickup the line |
| 12:26.28 | Blackvel | i dont have any open rtp ports for internal phones / isdn gateway connections |
| 12:27.08 | [TK]D-Fender | RTP should be able to get it on it's own and get answered... unless SIP was negotiated to allow it. |
| 12:27.25 | Blackvel | i am thinking about to check if the callerid is unknown and once not to answer the line (do do not to ivr) |
| 12:27.29 | [TK]D-Fender | Now getting flooded with SIP calls.. that is another matter |
| 12:27.30 | *** join/#asterisk irroot (~irroot@197.170.25.120) |
| 12:27.48 | [TK]D-Fender | Because this looks like you're allowing un-authed calls through |
| 12:28.01 | [TK]D-Fender | Which is something you shouldn't be doing anyway. |
| 12:28.18 | Blackvel | my nat router firewall has no limits (for any protocol) and asterisk box fw was not setup for this :) |
| 12:28.22 | Blackvel | ...not yet... |
| 12:28.33 | Blackvel | yes...for directip i seem to accept un-authed calls |
| 12:28.38 | [TK]D-Fender | So far not a FW question though it's a likely solution |
| 12:28.55 | [TK]D-Fender | Have you considered NOT accepting unauthed calls? :) |
| 12:29.10 | McBoingBo | VOIP over VPN connections, anything special to take into consideration? I have users with call quality issues in a remote location, our Asterisk server here seems to be fine, so I have to start thinking that the users connection is at fault |
| 12:29.47 | WIMPy | Don't use the same vpn for anythign else. |
| 12:31.01 | [TK]D-Fender | McBoingBo, VPN just shoves more wreapping around the same packets. Don't expect them to get better for it |
| 12:31.27 | *** part/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:32.41 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:33.02 | McBoingBo | [TK]D-Fender, most definitely dont expect a better result, just making sure that it is not unheard of to use softphone on VPN for a regular home connection |
| 12:33.56 | [TK]D-Fender | McBoingBo, Sure it's been done, but for completely different reasons. Security & to bypass things like ISP filters on SIP at the primary ones |
| 12:34.11 | [TK]D-Fender | are* |
| 12:34.20 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:34.20 | McBoingBo | I used tcpdump + wireshark to take a peak at the conversation we had, and it had loads of jitter/out of sequence but all was from the client to the Asterisk server not a hiccup going from Asterisk to user |
| 12:36.18 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 12:36.19 | *** join/#asterisk mahlon (mahlon@martini.nu) |
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| 12:38.11 | [TK]D-Fender | McBoingBo, Legit client? |
| 12:38.27 | McBoingBo | [TK]D-Fender, Legit in what sense? |
| 12:38.42 | [TK]D-Fender | actual real user or one of those un-authed calls? |
| 12:38.52 | McBoingBo | [TK]D-Fender, Real user |
| 12:39.01 | [TK]D-Fender | McBoingBo, Ok, nothing you can do about that... |
| 12:39.28 | McBoingBo | well there is, like recommending a better upload at the remote office, etc |
| 12:39.58 | [TK]D-Fender | Yeah, but those are things I suppose we shouldn't have to say :) Like you can't "fix" the problem.. only replace the entire scenario |
| 12:41.21 | McBoingBo | yeah but sometimes we "have" to state the obvious |
| 12:41.29 | McBoingBo | well its starting to become obvious to me now |
| 12:42.10 | McBoingBo | do you think I should have a tcpdump at the Firewall machine AND Asterisk server to ensure that the issue is on their end? |
| 12:42.49 | McBoingBo | I doubt thats the issue because its going out through the same path just fine |
| 12:43.52 | [TK]D-Fender | sounds like outbound traffic on their side is just choked up.... |
| 12:44.48 | McBoingBo | [TK]D-Fender, yeah I am getting them to run a simple http://speedtest.net and share that with me, I want to see how pathetic their upload is |
| 12:45.18 | McBoingBo | also, not all remote users are on the G.729 codec, so..yeah |
| 12:45.52 | McBoingBo | so far X-Lite has been hit and miss with quality but overall I like it, would like to try Bria though |
| 12:46.15 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 12:47.30 | Blackvel | is inbound directip supported with a (somewhat guest) username and password? to set allowguest=no for disallowing unauthenticated calls (rtp flooding) |
| 12:47.31 | McBoingBo | any better softphones out there? and is there something I can configure on the softphone side to help the bandwidth choke? |
| 12:49.08 | *** join/#asterisk bakermd (~bakermd@rrcs-69-193-161-242.nyc.biz.rr.com) |
| 12:50.10 | [TK]D-Fender | McBoingBo, It's not the softphone's fault. |
| 12:50.26 | McBoingBo | [TK]D-Fender, yeah I know..../kicks dirt |
| 12:50.27 | [TK]D-Fender | network conditions start from the OS out through the rest of their routing |
| 12:50.47 | [TK]D-Fender | Blackvel, "somewhat guest" user = user |
| 12:52.06 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 12:52.49 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 13:01.34 | Blackvel | no i mean this: |
| 13:01.37 | Blackvel | Dial(IAX2/guestusr:guestpwd@myasteriskserver.com/1234, 30,r) |
| 13:01.43 | Blackvel | that works in the extensions.conf |
| 13:02.29 | Blackvel | does that work for sip too? Dial(SIP/guestusr:guestpwd@myasteriskserver.com/1234, 30,r). so when i giveout my directip number i would just add a special username and password, to be able to add allowguest=no for denying unauthenticated calls |
| 13:04.28 | [TK]D-Fender | guestusr = a user |
| 13:04.48 | [TK]D-Fender | You could call it "Fred" if yuo felt like. |
| 13:04.54 | [TK]D-Fender | It's still an actual account name |
| 13:06.16 | Blackvel | of course |
| 13:06.49 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 13:06.54 | Blackvel | but are even sip direct ip calls possible with a provided username? |
| 13:07.16 | Blackvel | i have no problem to handout the username, if one can enter in his system or voip phone |
| 13:07.34 | [TK]D-Fender | that isn't un-authed. It is simply without creating a peer. Which is technically less secure on their end and means their [general] has to deal with codecs and other stuff better restricted to peers |
| 13:08.24 | [TK]D-Fender | Blackvel, You never needed to be registered to call. |
| 13:09.01 | [TK]D-Fender | Blackvel, So if they choose to make a peer (thy should if they know what's good for them) or just shove it in the Dial (ew), it's up to them |
| 13:09.52 | Blackvel | so either i go improve my FW or improving directip call context if i continue with allowguest=yes |
| 13:10.45 | Blackvel | what way do you guys go with 1-5 company systems? disallowing directip completely? |
| 13:11.03 | Blackvel | i am really not sure right now if voip phone support adding username/password for direct ip calls :) |
| 13:11.43 | Blackvel | most of the times i dont need it...makes only sense to be used for international / EU calls |
| 13:12.17 | [TK]D-Fender | what way do you guys go with 1-5 company systems? disallowing directip completely? <_ ? |
| 13:13.07 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 13:13.34 | Katty | raden: i am now! |
| 13:14.31 | Blackvel | bbl... |
| 13:17.49 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 13:19.41 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 13:22.48 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 13:23.05 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bcomktnczqqxuovd) |
| 13:23.57 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bcomktnczqqxuovd) |
| 13:26.34 | *** join/#asterisk deeperror (~deeperror@adsl-99-102-231-171.dsl.sfldmi.sbcglobal.net) |
| 13:27.21 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 13:27.39 | deeperror | I've blocked an IP from hitting the pbx, but for the past 2 days it's still hammering on the pbx even though all the packets are being dropped. Is there a better way to stop these attacks or just have to wait until they decide it's no use? |
| 13:28.12 | DennisG | deeperror: do you use iptables? if so, do you use deny or drop? |
| 13:28.14 | *** join/#asterisk akiims (akiims@95.130.35.88) |
| 13:28.26 | deeperror | DROP |
| 13:28.32 | DennisG | oke nice :) |
| 13:28.54 | DennisG | but if you have problems with the current situation then the only solution is a dedicated firewall in front of your pbx |
| 13:28.57 | deeperror | iptables -I asterisk -s $IP -j DROP |
| 13:29.02 | TSM | is there any way to notify the user when he enables call recording? |
| 13:29.07 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 13:29.15 | DennisG | because now you still get packets from a blacklisted ip |
| 13:29.29 | deeperror | ok, yea i just see it on my switch charts. I'll add it to the router |
| 13:30.25 | *** join/#asterisk mmoebius (~mmoebius@193.174.22.3) |
| 13:30.29 | DennisG | block it in your gateway :) that's the best solution |
| 13:30.55 | DennisG | now your offloading your pbx (and other servers) |
| 13:31.21 | mmoebius | Hello. Is anybody aware of any simple LDAP server implementation that can serve a phonebook e.g. for SNOM or Ekliga phones e.g. from a file ? |
| 13:32.18 | mmoebius | I don't want to run a full-blown LDAP-Server anymore but I'd rather have a more simple thing, even if it handles only name <--> phone number associations |
| 13:32.42 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:32.55 | WIMPy | uses the browser |
| 13:33.00 | irroot | mmoebius we ship ours with it built in sorry |
| 13:33.02 | hetii | if i will set on some trunk the argument host=example.com will also be valid for subdomain like host=foo.example.com ? |
| 13:33.33 | [TK]D-Fender | deeperror, Well.. that'll stop them 1 layer higher.. but they'll still waste bandwidth. Maybe ask if your ISP can block them |
| 13:33.42 | mmoebius | irroot: asterisk has a "build-in" LDAP ? |
| 13:33.46 | irroot | running ldap with a lighter backend maybe sqllite ?? |
| 13:33.50 | *** join/#asterisk usc911 (~ben@host81-137-209-116.in-addr.btopenworld.com) |
| 13:34.07 | hetii | the problem with my incoming call as i suppose is that i use srv lookup (my voip provider use few server to handle trafic) and as i see they send me some request sometime from some subdomian |
| 13:34.10 | deeperror | [TK]D-Fender: ok that's a good idea and what I was wondering. I'll call them up now. |
| 13:34.11 | irroot | mmoebius its a prebuilt distro |
| 13:34.44 | usc911 | Hey guys, I have been trying to find an IRC chanel for what im doing but was unable. I thoiught this being a phone related chanel that somebody may have experience with panasonic systems and I just need a very quick question? |
| 13:34.45 | hetii | so the question is what i can put on host= value? or should i use for eg. the wildcard like *.example.com ? |
| 13:34.45 | mmoebius | irroot: Currently, i have the berkeley db backent. That is pretty lightweight, from the server perspective. Unfortunately I have no sensible frontend for it to make some .... not too-bright staff people maintain the directory ... |
| 13:35.25 | irroot | mmoebius simple php page goes a long way |
| 13:35.26 | mmoebius | irroot: Which distro is shiping a prebuild asterisk with LDAP ? |
| 13:35.41 | deeperror | DennisG: I'm going to call ATT see what wonderful solutions they have |
| 13:35.43 | irroot | the one we build |
| 13:36.33 | *** join/#asterisk DennisG (~dennisg@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 13:37.14 | mmoebius | irroot: I am sorry for not beeing in the context, but which asterisk-distro do you build ? afaik asterisk itself is only the VoIP/SIP software, no ? |
| 13:38.25 | irroot | mmoebius its a all in one with own gui |
| 13:39.07 | irroot | linux 3.0 with all the bits needed ldap/sql/apache/samba/sendmail/dovecot/.... |
| 13:39.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bcomktnczqqxuovd) |
| 13:39.37 | akiims | hi, when Playback() function plays .alaw audiofile, in the beginning and at the end i can hear some kind of "click" sound. I use asterisk 1.8.4.2 |
| 13:41.22 | mmoebius | irroot: Ist is a buyable product ? Does it have a name and a website ? |
| 13:42.56 | irroot | sure is but more focused on south africa |
| 13:43.53 | [TK]D-Fender | irroot, What's it called? |
| 13:44.29 | *** join/#asterisk lep (~lep@93-50-183-160.ip153.fastwebnet.it) |
| 13:45.01 | irroot | [TK]D-Fender company is distrotech still need a decent name or distro though sell it preloaded on boxes from small atom box upto big servers |
| 13:47.54 | *** part/#asterisk usc911 (~ben@host81-137-209-116.in-addr.btopenworld.com) |
| 13:48.12 | *** join/#asterisk wonderworld (~ww@port-92-201-57-87.dynamic.qsc.de) |
| 13:51.00 | cusco | hi |
| 13:51.16 | WIMPy | lo |
| 13:51.38 | [TK]D-Fender | med |
| 13:51.46 | wonderworld | yo |
| 13:51.54 | cusco | in dialplan how can I hangup a call (2nd call leg) and dial again keeping the channel active? |
| 13:52.17 | [TK]D-Fender | cusco, "core show application dial" <- |
| 13:52.27 | cusco | just a sec |
| 13:53.42 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 13:53.44 | vader-- | hello |
| 13:53.51 | *** join/#asterisk Mackes (~mm@208.69.84.122) |
| 13:54.30 | cusco | ok here is my scenario. call file dials pstn, if answered it goes to a certain exten@context. and I want to hangup that call and dial another number imediatly ? |
| 13:54.31 | vader-- | does anyone in heer own a Polycom SoundStation 7000? I have one on an older firmware 4.2 bootrom and 3.2 App firmware and it takes forever to boot. Just trying to see if anyone else experiences this issue? |
| 13:55.26 | WIMPy | cusco: What [TK]D-Fender said. Examine the L region. |
| 13:55.47 | cusco | ow, I was thinking about g |
| 13:55.50 | cusco | kk thanks |
| 13:55.56 | [TK]D-Fender | vader--, how long is "forever"? |
| 13:56.21 | *** join/#asterisk timholum (~tholum@68-117-120-138.static.eucl.wi.charter.com) |
| 13:56.31 | irroot | vader-- is it provisioning ?? it could be trying to dl a boot/sip file ? |
| 13:58.19 | timholum | does anyone know if there is a way to make a catchall voicemail box? so if I dial mailbox 302 ( which does not exsist ) it will go to a mailbox I chouse? |
| 13:58.57 | [TK]D-Fender | timholum, It's your dialplan... do whatever you want |
| 13:59.10 | [TK]D-Fender | timholum, And you don't "dial" a mailbox. |
| 13:59.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:59.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.41 | timholum | fine :) if I Run VoiceMailBox( 302@mycontaxt ); it goes to a default mailbox if it doesnt exsist |
| 14:01.03 | timholum | or will I have to catch that in my dialplan |
| 14:01.23 | [TK]D-Fender | timholum, Do it int he dialplan |
| 14:01.59 | timholum | is there a if mailbox exsists? command |
| 14:02.13 | [TK]D-Fender | timholum, "core show functions" <- |
| 14:05.54 | [TK]D-Fender | \o/ |
| 14:06.44 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:10.00 | *** join/#asterisk master_of_master (~master_of@p57B539A7.dip.t-dialin.net) |
| 14:13.13 | *** join/#asterisk frawd (~francois@189.Red-81-38-88.dynamicIP.rima-tde.net) |
| 14:22.09 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:28.23 | Katty | pokes eppigy |
| 14:30.40 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:31.02 | eppigy | giggles |
| 14:31.11 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 14:31.18 | eppigy | twirls Katty around clumsily |
| 14:33.40 | Blackvel | what is usually the calling id for directip calls? i got "", "asterisk", "sip" and "unknown" |
| 14:35.13 | vader-- | it is taking about 5-10 minutes |
| 14:35.30 | eicto | Blackvel, callerid defined by Callerid: in call file |
| 14:36.30 | Blackvel | on inbound directip calls i mean. just thinking about to check for sip and unknown and let the call not answer |
| 14:36.31 | [TK]D-Fender | Blackvel, Whose? |
| 14:36.40 | Blackvel | mine |
| 14:36.48 | [TK]D-Fender | Blackvel, Stop allowing them in the first palce |
| 14:37.20 | Blackvel | just checked grandstream directip feature: the phone will only allow you to enter the IP address. nothing more :) |
| 14:41.46 | Blackvel | does it make sense that a linux FW box preroutes (nat) 5060 and 5004-x to the asterisk box and then the FW on the asterisk box checks some rules / blocks? dont want to recompile all the newer ip kernel modules on mipsel |
| 14:42.13 | catphish | what proportion of users are likely to run into problems by using consumer firewalls and no stun configuration? we're running into problems with the majority of our users at the moment |
| 14:42.31 | catphish | with nat=yes and no nat on the server side |
| 14:44.02 | *** join/#asterisk Sylnai (~andi@andimiller.net) |
| 14:44.55 | [TK]D-Fender | Blackvel, OS does what you set it up to do. |
| 14:45.15 | [TK]D-Fender | catphish, I have never seen any requirement for STUN before. |
| 14:45.40 | [TK]D-Fender | catphish, And those settings are only a few of those required. |
| 14:45.42 | [TK]D-Fender | ~sipnat |
| 14:45.43 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 14:48.26 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:51.12 | catphish | thanks [TK]D-Fender |
| 14:51.43 | Blackvel | [TK]D-Fender: i mean does it make sense e.g to block traffic on the * box instead of the FW before (as the router still forwards the traffic to * linux server) |
| 14:52.25 | [TK]D-Fender | Blackvel, Makes sense to block them from as far away as you can. |
| 14:52.49 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:53.01 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:53.13 | catphish | all those docs seem to relate to asterisk behind a nat |
| 14:53.18 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
| 14:53.29 | catphish | in my case asterisk is fully open and clients are natted |
| 14:54.10 | TSM | is there any way to notify the user when he enables call recording? |
| 14:54.40 | catphish | i set nat=yes directmedia=no, i don't set externhost or externip, and localnet |
| 14:54.44 | [TK]D-Fender | catphish, And what are your clients? |
| 14:55.12 | [TK]D-Fender | catphish, qualify=yes <- |
| 14:55.24 | [TK]D-Fender | catphish, Something you left off. |
| 14:56.52 | catphish | yeah, i use qualify=yes |
| 14:57.05 | catphish | my clients are hardware sip phones behind consumer nats |
| 15:00.12 | jeffspeff | catphish, for nat to work properly you have to set localnet and your externip |
| 15:02.24 | catphish | can't it determine its own IP? |
| 15:03.57 | Blackvel | [TK]D-Fender thanks |
| 15:04.15 | [TK]D-Fender | catphish, You said your server is public, so it doesn't need those. |
| 15:04.38 | [TK]D-Fender | catphish, Or technically "fully open". Whatever that is supposed to meam |
| 15:04.44 | [TK]D-Fender | mean* |
| 15:05.19 | jeffspeff | catphish, try setting the localnet and externip and see if it works |
| 15:05.42 | catphish | i may add those anyway to be safe |
| 15:05.57 | p3nguin | If it's on a public IP address, it does not need externip/externaddr nor externhost. |
| 15:05.58 | jeffspeff | catphish, and having your server just completely open to all public routes isn't the best security practice. |
| 15:06.04 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:06.16 | p3nguin | Those are for asterisk behind NAT. |
| 15:06.34 | catphish | i thought they were |
| 15:06.44 | p3nguin | Keyword: behind |
| 15:06.44 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:06.54 | p3nguin | If it isn't natted, it already knows its address. |
| 15:07.01 | catphish | thats what i thought |
| 15:07.07 | p3nguin | You were right. |
| 15:07.28 | catphish | i'm having endless trouble with nat'd clients at the moment, not sure if any of the problem is at my side or if i simple need to have everyone use stun |
| 15:07.33 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:07.47 | p3nguin | And if it isn't a gateway system with a LAN attached to the backside, there is no locanet to set either. |
| 15:08.09 | p3nguin | localnet, rather |
| 15:09.22 | jeffspeff | unless you have the modem plugging straight into the server, and not to a router, switch or anything else, then you have to set localnet and externip don't you? |
| 15:10.04 | p3nguin | If there is no RFC1918 addressing on the computer which asterisk is on, there is no localnet and there is no need for externip. |
| 15:10.33 | p3nguin | externip/externaddr is to tell asterisk what external IP address to use when it does not have said external IP address on its interface. |
| 15:10.35 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:10.40 | catphish | jeffspeff: yes it is, routing has no impact on security |
| 15:11.42 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:11.43 | p3nguin | If the asterisk system only has a LAN address, say 192.168.0.200, it has no clue what external address to put in packets when natting out to the world. The externip/externaddr or externhost setting is used to tell it that information. |
| 15:11.49 | jeffspeff | catphish, routing has a huge impact on security. that's why people use routing protocols to restrict/protect servers. |
| 15:12.01 | catphish | jeffspeff: they do? |
| 15:12.05 | catphish | i use a firewall |
| 15:12.14 | p3nguin | If it has a public IP address on the interface, IT ALREADY KNOWS ITS OWN ADDRESS. There is no nat and there is no reason to tell it to use a different address. |
| 15:12.25 | *** join/#asterisk mbrevda_ (~mbrevda@unaffiliated/mbrevda) |
| 15:12.27 | jeffspeff | catphish, a firewall is a common way of saying routing protocols. |
| 15:12.35 | p3nguin | Routing isn't to protect things, firewalls are. |
| 15:12.43 | p3nguin | Routing is to get shit to and from. |
| 15:12.59 | p3nguin | Firewall has nothing to do with routing. |
| 15:13.13 | jeffspeff | p3nguin, then how does a firewall work? |
| 15:13.19 | p3nguin | You can route without a firewall. And you can firewall without routing anything. |
| 15:13.32 | p3nguin | It filters ports. |
| 15:13.40 | catphish | a firewall works by examining the inside of an ip packet |
| 15:13.41 | jeffspeff | based on? |
| 15:13.41 | mbrevda_ | trying to dump calls in to a meetme by doing a dial w/ G, but all except the last phone hangs up with "ANSWERED_ELSEWHERE", and I'm NOT using the c flag. What gives? |
| 15:13.47 | p3nguin | based on ports |
| 15:14.07 | catphish | routing simple redirects packets based on ip headers, it never blocks them as long as they have a valid route |
| 15:14.27 | p3nguin | Routing directs traffic. |
| 15:14.40 | p3nguin | Firewalls block and/or allow traffic. |
| 15:15.01 | jeffspeff | the ports are also related to the ip addresses. if the firewall has no knowledge of routing protocols then it has no ability to block or allow traffic. |
| 15:15.10 | p3nguin | False. |
| 15:15.16 | p3nguin | Firewalls have nothing to do with routing. |
| 15:15.22 | catphish | jeffspeff: wrong |
| 15:15.42 | catphish | firewalls dont need to use the routing table to check the ip or port of a packet |
| 15:15.44 | jeffspeff | unless they lied to me during my CCNA and CCNP, then i think i'm right |
| 15:15.54 | catphish | they likely did |
| 15:15.56 | p3nguin | They either lied to you or you didn't understand it. |
| 15:15.57 | kaldemar | jeffspeff: sounds like they lied. |
| 15:16.06 | catphish | more likely you misunderstood |
| 15:16.24 | jeffspeff | no, they don't use routing tables to check packets, but they use routing tables to block and deny the traffic based on the rules in place for the traffic |
| 15:16.31 | p3nguin | No. |
| 15:16.43 | SwK | jeffspeff: if that statement about routing protocols were true then bridging firewalls would never work |
| 15:16.47 | catphish | no, routing tables are very rarely used to block traffic |
| 15:17.07 | catphish | and firewalls dont care about routing tables |
| 15:17.10 | p3nguin | Routing tables are used to know where to send a packet based on the address. |
| 15:17.21 | jeffspeff | i can use an old cisco 2500 series router and route traffic and do packet inspection on it. |
| 15:17.35 | catphish | sure, packet inspection = firewall |
| 15:17.43 | irroot | catphish unless you set up a blackhole route and mark trafic in iptables to go via this route rule :P |
| 15:17.44 | catphish | ACL = firewall |
| 15:17.54 | p3nguin | Firewalls can allow or deny based on the source address, the destination address, the source port, the destination port, the content of a packet, etc. Nothing to do with routing. |
| 15:17.55 | SwK | ACLs = ghetto firewall |
| 15:17.57 | jeffspeff | what i'm saying is they're one in the same. |
| 15:17.58 | catphish | i had that exception in mind when i said rarely ;) |
| 15:18.20 | catphish | i've seen a cow that could moo and walk |
| 15:18.26 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:18.29 | wcselby | o/ |
| 15:18.29 | catphish | doesn't mean you have to moo to walk |
| 15:18.49 | p3nguin | No matter how many ways you look at it, firewalls have no relation to routing. |
| 15:18.56 | catphish | indeed |
| 15:19.21 | p3nguin | Now firewalls do have the ability to mangle packets and send them where they weren't originally intended to go... |
| 15:19.23 | *** join/#asterisk becca_r (~becca_r@adsl-99-21-18-162.dsl.ksc2mo.sbcglobal.net) |
| 15:19.30 | p3nguin | but the router has to get those packets to the place. |
| 15:19.44 | irroot | catphish just stiring >:-) |
| 15:19.53 | catphish | hehe |
| 15:20.01 | catphish | i'll fetch my troll spray |
| 15:20.07 | SwK | if you look at IPTables, it mangles the packets before it even gets to the routing part |
| 15:20.17 | SwK | then hands it back into the stack as normal |
| 15:20.54 | jeffspeff | so, on our router we have it set to allow traffic over 5060 from our sip provider. if traffic tries to come in through any other source then it's denied. so then doesn't it use routing tables to determine how/what to do with traffic? |
| 15:21.10 | p3nguin | Your "router" is more than a router. |
| 15:21.17 | irroot | iptables has multiple tables and chains in predefined order and once you know where / what traffic does you can do some amazing things but dont attempt it until you understand it |
| 15:21.19 | jeffspeff | catphish, i'm not trolling, we just got off-topic |
| 15:21.41 | catphish | jeffspeff: i walk talking about irroot not you :) |
| 15:21.43 | irroot | jeffspeff think the troll spray was for me |
| 15:21.52 | [TK]D-Fender | All aboard the Crazy Train |
| 15:21.54 | p3nguin | Those store-bought plastic "routers" contain routers, switches, firewalls, often wireless access points. |
| 15:21.55 | [TK]D-Fender | </ozzy> |
| 15:21.55 | catphish | jeffspeff: your router isn't using routing tables to block traffic, its using a firewall |
| 15:22.01 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-224.socal.res.rr.com) |
| 15:22.01 | McBoingBo | TOOT TOOT! |
| 15:22.10 | catphish | jeffspeff: your router is in fact a multipurpose device with both a router and a firewall |
| 15:22.22 | irroot | i think i can i think i can i think i can .... |
| 15:22.44 | jeffspeff | ok, well i'm done... we're not getting anywhere with this, lol |
| 15:22.54 | catphish | i have a feeling my firewall is breaking my natted sip clients, but i really can't work out why |
| 15:23.01 | catphish | it isn't doing any NAT (i hope) |
| 15:23.15 | p3nguin | What kind of firewall? |
| 15:23.15 | [TK]D-Fender | irroot, I made a little statuette of "The Little Engine That Could" for someone ... it had a plaque at the bottom that read "We're not paying you to fucking think" |
| 15:23.20 | schmidts | catphish which kind of firewall? |
| 15:23.22 | catphish | juniper srx240 |
| 15:23.30 | r0m|u | waz up guys |
| 15:23.31 | idespinner | did you disable alg catphish ? |
| 15:23.33 | McBoingBo | [TK]D-Fender, lol nice |
| 15:23.35 | catphish | yes i did |
| 15:23.45 | catphish | i've had a lot more problem since upgrading its firmware |
| 15:24.02 | idespinner | we have sip running through SRX devices without issue |
| 15:24.11 | idespinner | but what firmware are you running? |
| 15:24.22 | [TK]D-Fender | catphish, "Hope"? Telephony & etworking aren't "Faith Based" |
| 15:24.26 | catphish | 10.4R7.5 |
| 15:24.46 | idespinner | a little older but not too old |
| 15:24.50 | irroot | [TK]D-Fender love it like the "intel" award we had to stop giving out it was a 386sx on safety pin the guy who stuffed up the most would have to wear it for a week HR did not like it much .... for the one who needed that extra processing power |
| 15:24.58 | catphish | thats the recommended version |
| 15:25.07 | r0m|u | p3nguin, you in? |
| 15:25.29 | idespinner | catphish, no nat right? pure routing? |
| 15:26.00 | catphish | the router does some nat, but not between my asterisk server and the outside world |
| 15:26.07 | catphish | just dumping some sip conversations now |
| 15:26.11 | wcselby | catphish - after you upgraded your firmware, did you make sure that the SIP alg's didn't get 're-enabled'? |
| 15:26.13 | catphish | think i've caught it failing |
| 15:26.25 | catphish | wcselby: actually sip alg wasnt disabled before |
| 15:26.28 | *** join/#asterisk Vince-0 (~AndChat@41-132-156-89.dsl.mweb.co.za) |
| 15:26.31 | catphish | i only disabled it since having these problems |
| 15:26.36 | wcselby | ahhh |
| 15:27.07 | wcselby | so what is the extent of your problem? i logged in during the catfight over the differences between routers, switches, and firewalls |
| 15:27.17 | catphish | lol |
| 15:27.45 | p3nguin | r0m|u: sort of |
| 15:28.08 | catphish | i have asterisk behind an srx240 (not natted) and a lot of hardware sip phones behind consumer nats |
| 15:28.37 | p3nguin | One way to find out if it's the Juniper would be to take it out of line. |
| 15:28.54 | catphish | p3nguin: that's much easier said that done sadly |
| 15:29.02 | catphish | afaik the juniper isn't mangling sip packets |
| 15:29.15 | r0m|u | p3nguin, you posted yesterday about the differences about a compliant cid and a none compliant. do you have that log? |
| 15:29.22 | wcselby | so you've described the topology to me, what's the issue? |
| 15:29.53 | r0m|u | catphish, can you put a sniffer in front of the juniper? |
| 15:30.02 | p3nguin | r0m|u: I said many things about it. Which thing specifically are you asking about? |
| 15:30.07 | r0m|u | and see if the sip packets are been mangled |
| 15:30.07 | catphish | sadly not, only behind it |
| 15:30.19 | catphish | might give me some clues though |
| 15:31.00 | p3nguin | THe SRX does do NAT, so you may want to examine how things are configured to make sure you really aren't NATing. |
| 15:31.18 | r0m|u | p3nguin, I am looking for what its valid and was its not. you said something like +1NXXNXXXXXXX and none valid 1NXXNXXXXXX or something like that |
| 15:31.29 | p3nguin | r0m|u: Give me a minute. |
| 15:31.34 | r0m|u | Thanks |
| 15:31.49 | idespinner | catphish, my recommendation is to get a pcap before the SRX and after |
| 15:31.56 | idespinner | then you can compare the SIP traffic |
| 15:32.32 | r0m|u | idespinner, not sure about before as it shouldnt be mangled before |
| 15:32.39 | r0m|u | but does not hurt ether :) |
| 15:32.58 | catphish | asterisk is definitely not natted |
| 15:33.08 | r0m|u | catphish, dmz? |
| 15:33.13 | Katty | eppigy: dude. |
| 15:33.16 | Katty | eppigy: i need out of here tday |
| 15:33.21 | catphish | kind of, yes |
| 15:33.30 | r0m|u | yes or no? |
| 15:34.21 | r0m|u | catphish, if you cant put it after the srx than setup an external peer you can call and have that peer setup with wireshark |
| 15:34.53 | catphish | dmz is just a word, it has no technical meaning |
| 15:35.04 | *** join/#asterisk Cubber (~ronny@150.156.193.100) |
| 15:35.44 | r0m|u | It matters though. |
| 15:36.46 | p3nguin | It technically means people don't know what they're doing when they tell me they put their asterisk in DMZ. |
| 15:36.52 | wcselby | catphish - when you say it's behind the srx but not natted, what do you mean? is it assigned a public IP address directly on the asterisk box? |
| 15:37.13 | r0m|u | ^^ |
| 15:37.15 | Cubber | I need to setup a lab solution where one asterisk server is setup to handle sip extensions and act as a regular PBX, the other needs to be able to provide a SIP trunk to the previous server since it has an FXO card installed for outbound connection. The idea is to have a student setup their asterisk server then connect to the other server to obtain an outbound/inbound trunk via SIP. Is this possible? |
| 15:37.24 | wcselby | or is it a private ip on the asterisk box but the srx does 1-to-1 nat (I think juniper calls this a VIP?) |
| 15:37.26 | Cubber | And if so is there any good documentation that I could be pointed to? |
| 15:37.27 | catphish | the asterisk server has an external ip, the SRX routes it |
| 15:37.34 | r0m|u | p3nguin, thats what I am trying to get at. |
| 15:37.45 | [TK]D-Fender | Cubber, .. |
| 15:37.47 | [TK]D-Fender | ~book |
| 15:37.47 | infobot | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 15:38.02 | [TK]D-Fender | Cubber, And yes you can use one * to call another. And handle that call however you want. |
| 15:38.06 | catphish | sadly i'm not even sure what the problem is, or if there is one at all :( |
| 15:38.19 | catphish | so this conversation is mostly pointless at the moment |
| 15:38.20 | Cubber | thanks [TK]D-Fender I will check out the link |
| 15:38.24 | wcselby | catphish - what is happening? |
| 15:38.27 | wcselby | are calls not completing? |
| 15:38.32 | [TK]D-Fender | Cubber, All users register and call through box#1. Box#1 sends all outbound calls to box#2 to actually get out. |
| 15:38.32 | wcselby | are calls one-way audio? |
| 15:38.37 | wcselby | are calls disconnecting? |
| 15:38.42 | [TK]D-Fender | Cubber, All very simple. |
| 15:38.54 | *** part/#asterisk deeperror (~deeperror@adsl-99-102-231-171.dsl.sfldmi.sbcglobal.net) |
| 15:39.00 | Cubber | [TK]D-Fender I figured it had to be doable since that is what the internet SIP providers are probably doing |
| 15:39.29 | catphish | wcselby: one sided calls are happening, not i've had very mixed information about whether stun is required or not for 2-way audio to work with a sip phone behind a nat |
| 15:39.44 | catphish | s/not// |
| 15:39.57 | p3nguin | r0m|u: http://pastebin.com/Ka55VnxK |
| 15:40.32 | r0m|u | Thanks! :) |
| 15:40.32 | catphish | what's annoying me is that our customers seem to have had no problems until i upgraded our firewall |
| 15:40.35 | wonderworld | Cubber: peer -> SIP -> Asterisk 1 -> IAX -> Asterisk 2 -> PSTN |
| 15:40.39 | r0m|u | exactly what I need it. |
| 15:40.42 | wcselby | catphish - stun is an option for nat on the phones. how do you have them setup in your sip.conf peer, if you've got them setup as "nat=yes", it should take care of it |
| 15:40.43 | *** join/#asterisk frem (~chatzilla@65.183.105.202) |
| 15:40.49 | catphish | since then they've all needed stun and to disable their own ALG |
| 15:41.00 | catphish | i do have nat=yes |
| 15:41.01 | p3nguin | STUN is not *required* for for 2-way audio to work with a SIP phone behind NAT. |
| 15:41.19 | wcselby | STUN is just an option for nat issues, but I've personally never needed it |
| 15:41.31 | p3nguin | I have asterisk behind a NAT. I have phones behind other NATs. There is no STUN in my life. |
| 15:41.44 | catphish | yeah, things worked well here until this router upgrade |
| 15:41.47 | r0m|u | me nether. and I am behind a pfsense. the worst monster you could ever have behind SIP! |
| 15:41.48 | catphish | so i'm rather puzzled |
| 15:41.53 | wcselby | setup SIP debug on one of the peers on the asterisk box, make some test calls, and pastebin the results |
| 15:42.11 | p3nguin | ALG should *always* under all circumstances be disabled. Do not argue it, just disable it. |
| 15:42.20 | [TK]D-Fender | catphish, If they have their own ALG then they are screwing themselves and you should not be using nat=no. If their gatway is rewriting the rule, then you have to follow them |
| 15:42.40 | [TK]D-Fender | "should be nat=no" |
| 15:43.03 | p3nguin | ALG breaks asterisk's ability to nat things by itself. |
| 15:43.13 | wcselby | nat=yes just means ignore the IP address in the SIP header and respond to the actual IP the packet came in from |
| 15:43.39 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-181.mobile.uci.edu) |
| 15:44.04 | *** join/#asterisk moy_ (~moy@64.231.55.213) |
| 15:44.06 | wcselby | nat=no means respond to the IP in the SIP header |
| 15:44.30 | p3nguin | Under most circumstances, when using asterisk, all ALG should be disables, all phones need to not be configured for NAT traversal, and asterisk should deal with the NAT stuff internally. |
| 15:45.31 | p3nguin | Some plastic routers do not work with SIP/RTP, though, and there is no way to make them work. |
| 15:45.33 | frem | question: if there's an SPA device acting as a POTS trunk at a remote location, will people making calls from that location just have to go through asterisk for the number routing? or will the entire call need to pass through the PBX? |
| 15:45.54 | catphish | p3nguin: thats annoying :) |
| 15:45.58 | p3nguin | Yes it is. |
| 15:46.05 | p3nguin | I sold my Cisco SOHO router because of it. |
| 15:46.25 | p3nguin | the SIP part worked fine, but RTP would never use the public IP address. |
| 15:46.58 | catphish | puzzling, i thought that was the purpose of nat=yes |
| 15:47.26 | p3nguin | Me too. |
| 15:47.58 | p3nguin | I tried and tried. No one else had any ideas either, so I got rid of it and went back to a Linux router. |
| 15:48.28 | [TK]D-Fender | frem, In all likelyhood you won't be able to re-invite through their router so yes it wil have to pass throughthe PBX |
| 15:49.25 | catphish | would nat=yes cause a problem if it wasn't required? |
| 15:49.31 | catphish | ie when using stun |
| 15:51.02 | frem | [TK]D-Fender: So the sip audio stream would come from the trunk, travel 100 miles, go through asterisk, go 100 miles back, then down to the IP phone? Eww. |
| 15:51.44 | vader-- | hmmm i upgraded the firmware on the polycom soundstation IP 7000 and now boot time is around 3 minutes... It stays on Processing Configuration... Is 3 minutes about normal for these phones to boot? |
| 15:51.56 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:52.04 | [TK]D-Fender | frem, Might be workable if you have a SIP proxy on the public interface on the end that has the phones & SPA |
| 15:52.18 | [TK]D-Fender | frem, But that also opens them up to attacks more. |
| 15:53.11 | [TK]D-Fender | vader--, Sure |
| 15:53.15 | p3nguin | catphish: Usually it does not break things to have it set to yes when it doesn't need to be, but do not rely on this. |
| 15:53.15 | vader-- | ok |
| 15:53.22 | vader-- | it was taking more like 10 minutes before |
| 15:53.28 | [TK]D-Fender | vader--, They also boot faster if you're not use the composite sip.ld |
| 15:53.34 | [TK]D-Fender | using* |
| 15:53.37 | frem | [TK]D-Fender: ok, thanks. |
| 15:54.00 | vader-- | TK, i downloaded the split firmware? did i download the wrong one? |
| 15:54.43 | wcselby | frem - you could try setting up a local asterisk box at the customer site that all the local phones register to, as well as the spa, and have that be the link back to the original asterisk, and then set directmedia=yes for the trunk between the asterisk boxes |
| 15:54.56 | vader-- | on my server i have 3111-40000-001.bootrom.ld, 3111-40000-001.sip.ld, bootrom.ld |
| 15:55.30 | frem | wcselby: that's looking like what has to happen. thanks! |
| 15:56.22 | *** join/#asterisk Greenlight (~Wullie@cpc2-dund11-2-0-cust994.sgyl.cable.virginmedia.com) |
| 15:56.33 | *** join/#asterisk jasonbassett (~jasonbass@cpc3-basl7-0-0-cust155.basl.cable.virginmedia.com) |
| 15:56.38 | jasonbassett | Hello folks |
| 15:57.17 | jasonbassett | I asked this question last night but when I got up any pointers had scrolled away into the ether, so I will ask again,,, |
| 15:57.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:57.24 | Greenlight | Hiya all. Just got a new TE410P Digium card though, and in the box is a small PCB with what looks like something between an RJ45 and an RJ11 connection on it - I've not seen one before and just wondered what this thing is? |
| 15:57.58 | jasonbassett | I have a Dial() line which executes a macro upon answer using the M(macroname) option |
| 15:58.11 | [TK]D-Fender | vader--, And what did you specify for use in the phone's provisioning? |
| 15:58.47 | irroot | Greenlight loopback ?? |
| 15:58.49 | jasonbassett | Any DTMF digits pressed when in the macro are not being read, I am trying to use the Read() application. |
| 15:58.51 | jasonbassett | Any ideas? |
| 15:58.57 | catphish | who invented sip and why do they hate their fellow man? |
| 15:59.44 | wcselby | Greenlight- pics? |
| 15:59.51 | wcselby | catphish- IETF |
| 16:00.06 | wcselby | jasonbassett- does DTMF work on just a regular call? |
| 16:00.10 | [TK]D-Fender | catphish, Don't go feelin' all special or nothin' ... but it's just YOU 8| |
| 16:00.14 | wcselby | outside of the macro? |
| 16:00.33 | jasonbassett | Yeh, any call without a macro running on answer is fine. I have never had to make use if macro on answer before. |
| 16:00.34 | [TK]D-Fender | jasonbassett, show us the call & dialplan |
| 16:00.36 | [TK]D-Fender | ~pb |
| 16:00.36 | 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. |
| 16:00.37 | [TK]D-Fender | ^^^ |
| 16:00.40 | catphish | :) |
| 16:01.06 | vader-- | tkd, not sure |
| 16:01.07 | Greenlight | WOuld a loopback not be RJ45 though? |
| 16:01.18 | vader-- | it's been like 2-3 years since i setup the cfg files |
| 16:02.03 | [TK]D-Fender | vader--, Go be sure. |
| 16:02.15 | Greenlight | Can I paste images to pastebin? |
| 16:02.34 | Qwell | Greenlight: pasteimage.com, but it is indeed a loopback plug |
| 16:02.43 | [TK]D-Fender | Greenlight, tinypic.com |
| 16:03.06 | willzzz | [Nov 2 12:02:40] NOTICE[27404]: pbx_spool.c:360 attempt_thread: Call failed to go through, reason (1) Hangup |
| 16:03.09 | Greenlight | Ahh okay cool - so what sort of connection would I need or was it my eyes deceiving me that it wasn't RJ45? |
| 16:03.13 | willzzz | is there a way for asterisk to ignore the hang-up |
| 16:03.39 | Qwell | Greenlight: It would be RJ45 (it's not *actually* called RJ45, but it's the same connector) |
| 16:03.46 | jasonbassett | http://pastebin.com/gkQf4QVg |
| 16:04.10 | jasonbassett | Something like that, changed slightly to hide my modesty :-) |
| 16:04.23 | Greenlight | Ahh cool - thanks very much |
| 16:04.23 | willzzz | basically i have a callback script |
| 16:04.33 | willzzz | its suppose to hang-up the user |
| 16:04.38 | jasonbassett | It just site waiting at the Read line |
| 16:04.40 | willzzz | and then callback the user using a different extension |
| 16:04.47 | jasonbassett | and then timesout |
| 16:07.32 | [TK]D-Fender | willzzz, Your callout failed. Why should it ignore it? |
| 16:07.35 | Greenlight | Hmm - how can i check what timing source dahdi is using? It's not picking up my new card apparently, but its still giving timing under dahdi_test. Does dahdi_dummy now show under dahdi_scan? |
| 16:07.52 | Qwell | Greenlight: If you have hardware, it's using that. |
| 16:08.16 | Greenlight | Even it it's not listing any under dahdi_tool or dahdi_scan? |
| 16:08.34 | Qwell | maybe not |
| 16:08.46 | Qwell | sounds like the driver isn't loaded |
| 16:08.50 | Greenlight | Indeed |
| 16:09.09 | Greenlight | That's what I thought, but couldn't understand why its still giving timing |
| 16:09.20 | Qwell | because dahdi provides timing on its own |
| 16:09.38 | Greenlight | Wouldn't that need dahdi_dummy? |
| 16:09.48 | Qwell | not anymore |
| 16:09.51 | Greenlight | Ahhhhh |
| 16:09.54 | Greenlight | That'll explain it! |
| 16:09.55 | Greenlight | :) |
| 16:09.55 | p3nguin | There's no dahdi_dummy now. |
| 16:10.00 | Greenlight | gotcha |
| 16:10.01 | p3nguin | There's only dahdi. |
| 16:10.20 | tzanger | there is no dahdi. there is only zuul. |
| 16:10.28 | Greenlight | ^^ |
| 16:11.34 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 16:16.06 | [TK]D-Fender | mmmm... 80's Annie Potts.... |
| 16:20.34 | WIMPy | Why is it calles sip.conf and not chan_sip.conf? |
| 16:20.38 | willzzz | <PROTECTED> |
| 16:21.15 | [TK]D-Fender | WIMPy, backwards compatability |
| 16:21.35 | p3nguin | You wouldn't want everything to have the same naming scheme, now would you? |
| 16:21.37 | [TK]D-Fender | WIMPy, DAHDI was new so it got a new looking name. |
| 16:21.41 | WIMPy | I think the official term is "historical reasons". |
| 16:22.06 | p3nguin | The reasons aren't really historical, though. |
| 16:22.29 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:23.17 | willzzz | is there a way i can tell HangUp to NOT hang up until my .sh script has finished executing or exited on its own? |
| 16:23.50 | p3nguin | Don't run Hangup() until you are ready. |
| 16:24.29 | p3nguin | I'm pretty sure there is no Hangup(just-playing). |
| 16:24.43 | [TK]D-Fender | willzzz, And what is this script doing? |
| 16:25.10 | WIMPy | A hangupandplayback() would be a very good thing, however. |
| 16:25.36 | p3nguin | Use Playback() followed by Hangup() for that. |
| 16:25.42 | willzzz | d-fender, callback |
| 16:25.45 | willzzz | http://voipspectator.com/wordpress/2010/03/how-to-implement-an-automatic-callback-with-asterisk/ |
| 16:26.03 | WIMPy | No, hangupandplayback() not playbackandhangup(). |
| 16:26.09 | vader-- | TKD, i have this line APP_FILE_PATH="sip.ld" |
| 16:26.12 | WIMPy | order does matter. |
| 16:26.36 | p3nguin | You can't playback something after the channel is gone. |
| 16:27.01 | WIMPy | Yes, that's exactely the issue. |
| 16:27.07 | p3nguin | Once I'm off the line, I won't hear anything you tried to play even if it would work. |
| 16:27.20 | WIMPy | Usually you play announcements after clearing the call. |
| 16:27.36 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 16:27.43 | WIMPy | Only one side can be first. |
| 16:29.52 | wonderworld | you can put both callers transparently in a conference, kick the one who didn't hangup from the conf and let him go on in the dialplan |
| 16:30.38 | wonderworld | ok, not really a "clean" way to do it :) |
| 16:30.39 | p3nguin | Or just use the correct option in Dial(). |
| 16:30.58 | WIMPy | That's not the point. The point is that I cannot end the connection (i.e. billsecs) befor playing an announcement. |
| 16:31.44 | WIMPy | Or if a connection cannot be established, I have to use early media instead which is not a clean solution. |
| 16:33.17 | willzzz | http://pastebin.com/Vr7JPLET |
| 16:35.39 | WIMPy | And the last discussion on outgoing overlap on SIP seems to be 3.5 years ago. |
| 16:36.10 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-135-115.chyn.qwest.net) |
| 16:36.37 | [TK]D-Fender | <willzzz> [Nov 2 12:02:40] NOTICE[27404]: pbx_spool.c:360 attempt_thread: Call failed to go through, reason (1) Hangup |
| 16:36.55 | willzzz | i want the first call to hang up |
| 16:36.58 | willzzz | i want to make a new call |
| 16:37.01 | willzzz | the calls are seperate |
| 16:37.03 | [TK]D-Fender | willzzz, the outbound attempt fired off.. aand it failed. This has nothing to do with your script not having done it's job from what I can tell. |
| 16:37.46 | willzzz | well let me change trunks on the outbound |
| 16:37.54 | [TK]D-Fender | willzzz, we also don't see what it's calling, and I'm suspecting it shouldn't be dialing right away |
| 16:38.07 | willzzz | its waiting 30 seconds in the script |
| 16:38.24 | willzzz | that script should be completely seperate from the 1st call |
| 16:38.27 | [TK]D-Fender | willzzz, You aer also forcing your caller to wait through that audio prompt in full for no good reason. All sorts of unncessary delays before starting the outbound process |
| 16:38.34 | willzzz | whose purpose is to get the CLI only |
| 16:38.47 | [TK]D-Fender | willzzz, taht script is not separate, its executed from the first call. |
| 16:39.00 | [TK]D-Fender | willzzz, the CALL FILE however is separate once it's moved over |
| 16:39.23 | [TK]D-Fender | willzzz, And we don't see the whole picture |
| 16:41.09 | Blackvel | how many invites on a new call per second / minute would be just normal (same ip)? 1-2? 3+ would be more than one call, right? |
| 16:42.42 | p3nguin | For a new call, one invite per call leg seems normal to me. |
| 16:43.15 | p3nguin | If you're just calling to asterisk, one invite. If you're calling through asterisk, one invite to asterisk and one invite to the other peer. |
| 16:43.46 | WIMPy | Some ITSPs send multipel INVITEs. |
| 16:43.49 | willzzz | <PROTECTED> |
| 16:44.08 | p3nguin | What would be the purpose of multiple invites? |
| 16:44.32 | WIMPy | Fail safe. They come from different networks. |
| 16:48.35 | Blackvel | but for sip providers the ip is all the time the same? so if i get 5 different calls to * (over the same provider), i definitely got 5 invites per hour |
| 16:48.47 | Blackvel | from the same ip |
| 16:49.25 | p3nguin | Invites aren't typically measured over time. |
| 16:49.32 | WIMPy | Some use only one IP, others use many. |
| 16:49.38 | [TK]D-Fender | Blackvel, What is the goal of this count? |
| 16:50.32 | p3nguin | If you get five calls from any peer, expect five (or more) invites. |
| 16:54.40 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:55.07 | vader-- | TKD, i have this line APP_FILE_PATH="sip.ld" |
| 16:55.13 | vader-- | is that the line you were referencing? |
| 16:55.21 | navaismo | Hi, I'm unable to use the originate application on the cli on asterisk 1.8.7.1 http://pastebin.com/e4DmHDxz |
| 16:55.36 | [TK]D-Fender | vader--, Yes |
| 16:55.56 | [TK]D-Fender | vader--, Go specify the model-specific version of your firmware |
| 16:56.03 | p3nguin | navaismo: channel originate SIP/000011112222 extension 3145551212@phones |
| 16:56.08 | p3nguin | Is that how you are using it? |
| 16:56.25 | [TK]D-Fender | navaismo, You don't use DIALPLAN APPS on the CLI |
| 16:56.42 | navaismo | nope originate dahdi/1/ZXXXXX application musiconhold default |
| 16:57.06 | [TK]D-Fender | navaismo, "Originate' is the old CLI command name. It was renamed. |
| 16:57.24 | vader-- | so it should read? APP_FILE_PATH="3111-40000-001.sip.ld" |
| 16:57.27 | p3nguin | navaismo: Then you didn't enable cli aliases. Use channel originate or enable aliases so you don't have to specify the channel prefix. |
| 16:57.37 | [TK]D-Fender | vader--, if that's the one, yes, that should do it. |
| 16:58.16 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:58.24 | p3nguin | navaismo: cli_aliases.conf and res_clialiases.so |
| 16:58.34 | navaismo | ok let me try enable aliases, thx [TK]D-Fender and p3nguin |
| 16:59.06 | p3nguin | It would be a good idea to get in the habit of "channel originate ..." anyway. |
| 16:59.26 | navaismo | yes i'll keep in mind that sintaxis |
| 17:00.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:00.58 | mort_gib | Hi, has anyone had stability problems with 1.6.2.20 |
| 17:01.25 | navaismo | ja my mistake i dont execute the make samples grrrr |
| 17:01.33 | mort_gib | Like the cli works, and you can see "sip show peers" but no calls are passed onto the phones... |
| 17:01.40 | p3nguin | Don't do it now! |
| 17:01.55 | Qwell | mort_gib: 1.6.2 is no longer supported. You should be upgrading to 1.8. |
| 17:02.08 | mort_gib | Uhm |
| 17:02.10 | p3nguin | mort_gib: So are you really asking about stability? |
| 17:02.17 | [TK]D-Fender | mort_gib, that doesn't offer anything to go on |
| 17:02.17 | mort_gib | Should still work |
| 17:02.17 | navaismo | nope |
| 17:02.19 | p3nguin | It sounds like you are asking, "Does it work at all?" |
| 17:02.43 | mort_gib | No, that's not what I'm asking |
| 17:02.54 | mort_gib | Problem is I get nothing in the logs |
| 17:03.02 | mort_gib | Just stops |
| 17:03.03 | [TK]D-Fender | mort_gib, "not being passed on" <- show us |
| 17:03.15 | mort_gib | It's very intermitted |
| 17:03.17 | [TK]D-Fender | mort_gib, What stops? Details.... |
| 17:03.37 | mort_gib | But I suppose the answer to my question is -no we have never seen or heard anything like that |
| 17:04.05 | p3nguin | If you're asking if 1.6.2.20 works and allows calling, the answer is yes. |
| 17:04.07 | mort_gib | I use a sangoma card (A500) with Dahdi support |
| 17:04.21 | mort_gib | p3nguin, no |
| 17:04.26 | mort_gib | Not what I'm asking |
| 17:04.27 | Qwell | mort_gib: The real answer is "we no longer care if there is such an issue with 1.6.2" |
| 17:04.39 | mort_gib | Fair enough |
| 17:04.51 | mort_gib | I still have users on 1.4 |
| 17:05.09 | mort_gib | Not happy about that but they don't want to update |
| 17:05.39 | p3nguin | 1.4 didn't magically get less stable or reliable just because new branches were developed. |
| 17:05.52 | navaismo | thx p3nguin [TK]D-Fender now its working |
| 17:06.43 | mort_gib | Tk, I can ssee the call coming in on Dahdi, but it then drops, without being passed onto the phones |
| 17:06.57 | [TK]D-Fender | mort_gib, Know what we see? |
| 17:07.15 | mort_gib | But That's fine if nobody has seen that before, that answereed my question |
| 17:07.40 | mort_gib | No, TK what do you see?? |
| 17:08.12 | WIMPy | Hi mort_gib. Long time no see. |
| 17:08.46 | mort_gib | Hi WIMPy -Yes have had a rough time |
| 17:08.59 | [TK]D-Fender | mort_gib, Nothing :) |
| 17:09.03 | WIMPy | What happened? |
| 17:09.17 | mort_gib | Just a lot of work and a lot of sad meetings |
| 17:09.29 | mort_gib | Though going solo would get me out of that, but no sir |
| 17:10.09 | mort_gib | <[TK]D-Fender> :-) THere is not much to show you |
| 17:10.18 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:10.23 | [TK]D-Fender | mort_gib, i hear it.. and never believe it |
| 17:10.36 | WIMPy | The main use for meetings is not to work, isn't it? |
| 17:10.38 | mort_gib | That's fine |
| 17:10.43 | pdtpatrick1 | Question ..is this usually caused by agi script not exiting properly? I've searched google for a while and everyone seems to have different opinions. Maybe it is one of those that can be anything. So i thought i'd ask in here |
| 17:10.44 | pdtpatrick1 | [Nov 2 10:09:34] ERROR[28884]: utils.c:1175 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 17:11.13 | mort_gib | WIMPy Yes, or so sad people that are NOT working can feel important and hear their own voice |
| 17:11.26 | Faustov | leifmadsen: ping |
| 17:11.44 | WIMPy | And keep others from working as well. |
| 17:11.53 | mort_gib | Correct |
| 17:12.04 | [TK]D-Fender | pdtpatrick1, Oftn by outputting to interface when you shouldn't excess script noise, etc |
| 17:13.18 | mort_gib | <[TK]D-Fender> I haven't tried to up debugging yet, but there is nothing that even looks like a problem, but I'm happy with nobody else having seen the issue |
| 17:14.29 | pdtpatrick1 | [TK]D-Fender, thanks |
| 17:23.14 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:26.42 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:31.21 | vader-- | TKD that line didn't help the boot speed of the phone |
| 17:31.24 | vader-- | still kinda long |
| 17:31.46 | catphish | is 10000-20000 the default rtp port range? |
| 17:32.33 | WIMPy | rtp.conf |
| 17:32.40 | catphish | i don't have one |
| 17:33.18 | WIMPy | Do you have peers with nat=yes? |
| 17:33.39 | wcselby | pdtpatrick1 - check if your script is properly following all the AGI rules. if it's not correctly parsing the AGI environment and sending back the expected responses, or if it's not sending back the expected responses after requesting data, it's very likely to get the error message you got. |
| 17:33.39 | catphish | i do |
| 17:33.57 | WIMPy | The default configuration is 10000-20000, yes, NFI what you get if not configured. |
| 17:34.06 | WIMPy | Then you have a security issue. |
| 17:34.10 | [TK]D-Fender | vader--, How long exactly? 2 Minutes is perfectly normal |
| 17:34.29 | catphish | sorry what do you mean? |
| 17:35.24 | wcselby | catphish i think he means that if you have 10000 UDP ports open in your firewall to allow UDP traffic, you've very likely created a security issue. |
| 17:35.33 | WIMPy | You should set strictrpt=yes, unless you know exactely what you're doin. |
| 17:35.33 | wcselby | i mean to allow rtp traffic |
| 17:35.38 | WIMPy | So get out an rtp.conf. |
| 17:35.57 | catphish | what's the problem with opening 10,000 ports? |
| 17:36.05 | catphish | the only thing that would listen on those is asterisk |
| 17:36.09 | WIMPy | No. I don't see a problem there. |
| 17:36.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:36.24 | wcselby | heh |
| 17:36.36 | catphish | strictrtp is wise, though we're not using it for now until we're happy with everything else |
| 17:36.47 | WIMPy | Ok, there's one more condition. Do you have feature transfers enabled? |
| 17:41.15 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 17:41.22 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 17:42.32 | vader-- | tkd, ya it's like 3 minutes exactly... it was around 10 minutes before |
| 17:42.45 | vader-- | the firmware upgrade helped |
| 17:46.13 | [TK]D-Fender | vader--, then you're done. |
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| 18:04.52 | *** join/#asterisk freezer (~freezer@91-64-201-149-dynip.superkabel.de) |
| 18:04.56 | freezer | hi |
| 18:05.17 | freezer | Anyone can recommend a good conference voip phone? |
| 18:05.44 | freezer | for up to 10people to use in a room about 20sqm in size |
| 18:06.21 | TangoElectro | HI i have installed asterisk on a 8gb box, but it only seems to be using 32mb of RAM, any idea what could be restricting it |
| 18:06.40 | willzzz | is there a asterisk command to prevent hangup |
| 18:06.50 | willzzz | until the caller intiates it |
| 18:07.14 | WIMPy | 'core show application Dial' see operator mode. |
| 18:07.39 | willzzz | no i'm talking extensions programming |
| 18:07.57 | WIMPy | Then please explain. |
| 18:10.18 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
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| 18:17.41 | willzzz | http://pastebin.com/vXRtVQdu |
| 18:18.59 | F2Knight | Q: Stopping sip scans.... Is it even possible? I have a few clients with SIP phone out in the wild... (remote extensions). Lately they have been getting calls in the middle of the night by wild array of numbers... (calls meaning just ringing) I noticed I can run svwar on a local phone and make it ring just by sending it an -m INVITE packet... so the question now is this... How do I prevent the device from ringing from any old person th |
| 18:18.59 | F2Knight | at happens to be running a sip attack on a public node? |
| 18:20.34 | eppigy | Katty: lets leave then |
| 18:22.13 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 18:23.00 | *** join/#asterisk tc (~travis@rrcs-67-78-243-170.se.biz.rr.com) |
| 18:23.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:23.23 | WIMPy | Use phones that require a correct (possibly random) URI. |
| 18:23.38 | *** part/#asterisk tc (~travis@rrcs-67-78-243-170.se.biz.rr.com) |
| 18:27.32 | [TK]D-Fender | willzzz, -- Attempting call on SIP/000@zayo-trunk-1 for s@callback:1 (Retry 1) <--- 000 = no good |
| 18:27.43 | [TK]D-Fender | willzzz, -- Got SIP response 604 "Does not exist anywhere" back from 000:5060 |
| 18:28.35 | [TK]D-Fender | willzzz, -- Executing [5069@incomingall:4] System("SIP/zayo-trunk-1-00001045", "echo 'Channel: SIP/000@zayo-trunk-1' >> /var/spool/asterisk/callback.tmp.call") in new stack <- you're also ttrying to creaet the spool file directly in SPOOL *live*. This is a forbidden |
| 18:29.17 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:29.19 | *** join/#asterisk plundra (1000@v0.article.se) |
| 18:29.26 | [TK]D-Fender | willzzz, And as I told you yesterday you have no need for call files for this at all. |
| 18:29.35 | [TK]D-Fender | willzzz, Originate() the new channel |
| 18:31.47 | willzzz | ok i will replace with originate() |
| 18:31.49 | willzzz | interesting |
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| 18:41.13 | r0m|u | guys is it safe to disable dahdi if I dont have an interface card? |
| 18:41.22 | r0m|u | FXO/FXS |
| 18:45.43 | [TK]D-Fender | r0m|u, Do you need it for anything? |
| 18:46.14 | p3nguin | If you want to use MeetMe, you'll consider keeping it. |
| 18:46.57 | r0m|u | [TK]D-Fender, not that I know off. MeetMe? |
| 18:47.31 | [TK]D-Fender | that's one |
| 18:47.40 | *** join/#asterisk bipul (~h4x0r@unaffiliated/bipul/x-4918593) |
| 18:47.59 | WIMPy | And Page uses MeetMe. |
| 18:48.25 | WIMPy | Or has that been changed, yet? |
| 18:48.27 | r0m|u | ok so dahdi is not just for hardware? |
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| 18:50.00 | p3nguin | "Dahdi: It's Not Just for Hardware" |
| 18:51.22 | r0m|u | Thanks p3nguin |
| 18:51.48 | Qwell | nobody uses meetme anymore |
| 18:52.12 | r0m|u | I see |
| 18:52.38 | p3nguin | Unless ConfBridge from 10 is fabulous and can be backported into 1.8, I'd imagine a bunch of people use MeetMe still. |
| 18:52.51 | mjordan | half of your statement is true |
| 18:52.56 | r0m|u | I am just trying to unload what I dont need to gain a bit more mem. |
| 18:53.10 | p3nguin | s/and/AND/ |
| 18:53.31 | WIMPy | Qwell: including Page? |
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| 18:54.25 | Qwell | pfft, who pages phones? |
| 18:54.50 | WIMPy | It can be handy. |
| 18:54.52 | Qwell | but really, multicast RTP |
| 18:55.31 | WIMPy | That makes most sense. |
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| 19:00.36 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 19:00.43 | wcselby | o/ |
| 19:01.08 | r0m|u | guys I am trying to disable things that have error in my system IE: ERROR[2708]: chan_misdn.c:11176 in load_module: Unable to initiali |
| 19:01.09 | r0m|u | ze mISDN |
| 19:01.42 | r0m|u | ERROR[2708]: codec_dahdi.c:578 in find_transcoders: Failed to open |
| 19:01.42 | r0m|u | <PROTECTED> |
| 19:01.49 | wcselby | so do you use misdn? |
| 19:01.54 | r0m|u | no sr. |
| 19:02.00 | r0m|u | Just want to make sure is safe |
| 19:02.07 | r0m|u | to disable |
| 19:02.22 | wcselby | then in modules.conf do a noload chan_misdn.so, or whatever is the appropriate module name |
| 19:02.33 | wcselby | do you get the dahdi transcode error more than once/ |
| 19:02.35 | wcselby | ? |
| 19:02.35 | WIMPy | You can disable everything you don't need. That's the idea. |
| 19:02.51 | r0m|u | wcselby, no. just at boot up |
| 19:03.00 | wcselby | i've always gotten that, but let it ride. i think it just means there's no hardware installed, but I've honestly never investigated it |
| 19:03.10 | wcselby | dahdi is important for other functions too, so I've always left it |
| 19:03.19 | r0m|u | WIMPy, I am trying to get there :) |
| 19:03.29 | wcselby | but yeah, I noload all the unneeded modules that got compiled. |
| 19:03.43 | r0m|u | cool |
| 19:03.45 | wcselby | you can opt to not compile those modules in the future by unchecking them in the make menuselect menu |
| 19:04.23 | r0m|u | yea I knew that one. but it was my first time at asterisk :) |
| 19:04.39 | r0m|u | so wanted to see what works and what does not. |
| 19:05.20 | wcselby | so yeah, i usually noload chan_mgcp, chan_iax2, config_ael, config_lua, etc, and then a few other items that I may have compiled but ended up not using. i intentionally leave out compiling things I know I won't use, like misdn, etc, so I don't have to even noload those. |
| 19:05.36 | wcselby | i don't remember if those are the proper names or not |
| 19:06.52 | p3nguin | If you are using autoload=yes and you don't have configs for the modules, they shouldn't be loaded anyway. |
| 19:07.28 | r0m|u | I see |
| 19:08.31 | WIMPy | At least most of them. |
| 19:09.02 | WIMPy | So removing vonf files can be a good idea as well. |
| 19:09.15 | r0m|u | I see |
| 19:09.30 | r0m|u | thanks for the info guys |
| 19:10.38 | *** join/#asterisk moy_ (~moy@209.226.201.250) |
| 19:12.20 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 19:13.16 | raden | hugs Katty |
| 19:13.27 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 19:15.54 | wcselby | what is this "new look" tomfoolery that gmail is throwing at me |
| 19:21.03 | r0m|u | I am using it |
| 19:21.07 | r0m|u | dont like it much |
| 19:21.19 | r0m|u | its a bit block likr |
| 19:21.25 | r0m|u | like* |
| 19:29.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:33.32 | wcselby | yeah I'm not liking it much either |
| 19:33.34 | wcselby | but meh |
| 19:33.48 | r0m|u | lol |
| 19:36.28 | [TK]D-Fender | Yuo can change the view size back towrds the traditional styling however the text-labels dfor buttons is gone. |
| 19:36.37 | [TK]D-Fender | like[-1] |
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| 19:37.09 | r0m|u | I guess we just have to get use to it |
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| 20:21.45 | jaytee | anyone here use Bandwidth.com? |
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| 20:31.31 | r0m|u | jaytee, not me but I have heard good things. |
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| 21:07.08 | willzzz | who here has done automated callback /w asterisk |
| 21:07.20 | hardwire | me! |
| 21:07.30 | hardwire | any other questions? |
| 21:09.00 | r0m|u | lol |
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| 21:34.01 | dschuett | does anyone know how to get a DND soft button on the cisco 7940 using SIP? |
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| 21:46.51 | paulc | dschuett: I haven't played with one of those for AGES but I'm sure I had one.. (it just makes the phone return "Busy here" when a call arrives) |
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| 22:02.15 | beccara | well this is a first, an error google can't find ANYTHING on |
| 22:02.17 | beccara | [Nov 3 11:00:55] ERROR[16553]: pbx.c:3385 ast_func_write: Function FILE cannot be written to |
| 22:02.32 | beccara | The file is -rwxrwxrwx 1 root root 0 2011-11-03 10:52 /tmp/foo.txt |
| 22:02.51 | beccara | <PROTECTED> |
| 22:03.48 | beccara | any ideas? |
| 22:05.43 | *** part/#asterisk dschuett (~dschuett@mail2.hoovestol.com) |
| 22:06.53 | [TK]D-Fender | beccara: it is only for reading, not writing. |
| 22:07.25 | beccara | ah so the manual is wrong |
| 22:07.27 | wdoekes2 | indeed.. asterisk 1.6.x and lower |
| 22:07.45 | wdoekes2 | which manual? .. the 1.8 manual? |
| 22:08.38 | beccara | http://www.voip-info.org/wiki/view/Asterisk+func+FILE |
| 22:08.51 | wdoekes2 | ~voip-info |
| 22:08.52 | infobot | somebody said voip-info was the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 22:09.14 | wdoekes2 | hm.. that was not the message I was expecting ;) |
| 22:09.20 | beccara | lol |
| 22:09.30 | beccara | I know it's not the beall and endall of info but it' |
| 22:09.33 | beccara | s a good place to start |
| 22:09.50 | beccara | so you can only write files in * 1.8+? |
| 22:09.52 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 22:10.00 | wdoekes2 | using the FILE function, yes |
| 22:10.35 | beccara | okie dokie have to goto sql then :) |
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| 22:27.05 | jeffspeff | how do i set the option to go forward or backward through the moh? i've set it before, i just can't find it or remember. like if you're listening to moh and you press 9 it goes to the next song. |
| 22:32.42 | wdoekes2 | beccara: you can use System (or Exec, or whatever it's called) |
| 22:47.41 | beccara | cheers wdoekes2 |
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| 23:10.56 | shadowapex | Hello everyone. I am currently configuring "media bypass" in Microsoft Lync with our current Asterisk server. In Asterisk terminology "media bypass" appears to be the same thing as "canreinvite" (or "directmedia"), but there is currently an option in the Lync configuration asking how many early dialogs the PSTN gateway (our Asterisk server) can support. |
| 23:12.35 | shadowapex | I've scoured the internet and haven't been able to find how many early dialogs Asterisk can support, and if it is even an adjustable option. From what I've read, early dialogs are the number of forked responses to an INVITE message. Anyone know how many forked responses Asterisk supports? |
| 23:14.10 | *** part/#asterisk pietro (~pietro@88-149-227-41.dynamic.ngi.it) |
| 23:18.11 | *** join/#asterisk garryfre (~gfreemyer@mxgw.videomaker.com) |
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| 23:20.34 | garryfre | I am tasked with making a voice mail system. My boss is demanding that it does flash hook transfers and that my code checks the result of the transfer - was it answered, busy on other end or no answer. I've tried all kinds of tricks to test. Is it even possible to find out what happened from code at all after a flash hook transfer? |
| 23:22.10 | garryfre | I NEED to know if this is even possible,b/c if not, I'm going to have to look for another job or suffer from more months of this torture of being forced to create something I know too little about. |
| 23:22.20 | garryfre | anyone? anyone at all please?!!? |
| 23:25.08 | garryfre | asterisk 1.6 code is flash(), Background(silence/1) SendDTMF(${EXTEN},background(silence/1) |
| 23:25.19 | SeRi | p3nguin, Thanks for the info! |
| 23:25.21 | garryfre | beauler .... beautler? |
| 23:26.22 | garryfre | I repeat, is it possible to get the success or failure of a flash hook attempt done with code in a custom dialplan? asterisk 1.6 code is flash(), Background(silence/1) SendDTMF(${EXTEN},background(silence/1) |
| 23:26.58 | garryfre | I'm not asking how I'm asking if it's possible. Anyone, Charles Manson? Marylyn Monroe? Obama? Beauler? Anyone? |
| 23:27.26 | navaismo | ~book @ garryfre |
| 23:27.39 | navaismo | ~ book @garryfre |
| 23:27.46 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 23:28.01 | navaismo | http://ofps.oreilly.com/titles/9780596517342/ |
| 23:28.49 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 23:28.53 | paulc | garryfre: I don't understand the question.. why are you hook flashing? |
| 23:28.58 | garryfre | Oh yes, the lost book I once knew of. Online and for free yet. I've been in such agony I forgot about this book. |
| 23:29.39 | garryfre | because the boss demands it. we got four lines into the asterisk server and I wanted to use dial, he demands I use flash hook to save line usage and congestion. |
| 23:30.17 | navaismo | i dont understan the " he demands I use flash hook to save line usage and congestion." |
| 23:31.25 | garryfre | he wants to get it transfered via our telco instead of connecting through asterisk - I didn't say I fully understand it. He - boss, me peon, do or be fired |
| 23:31.41 | navaismo | actually i dont understand anything |
| 23:32.16 | garryfre | when we do a transfer with hook flash the two can talk after asterisk hangs up and not be going through the asterisk server after that. |
| 23:33.20 | navaismo | ok, i can't help you, sorry. |
| 23:33.53 | garryfre | I'm sorry if I don't know enough to ask a question that makes sense. I'm forced to do this. yes I'm sorry too plus every day I'm reminded how I'm not equal to this task and I don't know squat |
| 23:34.07 | shadowapex | @garryfre http://www.voip-info.org/wiki/view/Asterisk+cmd+Flash |
| 23:34.32 | shadowapex | Looks like Flash() returns 0 or -1 whether or not it was successful. |
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| 23:36.04 | shadowapex | For my question, does anyone happen to know how many forked outbound INVITE requests Asterisk can handle? |
| 23:37.44 | garryfre | thing is if you look at the code it does a flash ... ok, that sets the caller to wating, then the senddtmf(ext) sends it to the other extension. The flash has already returned before the destination ext is even processed. My problem is I need to know if the destination ext answered or not. |
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| 23:39.41 | navaismo | (I think) you cant if asterisk is not handle the call |
| 23:40.06 | garryfre | Yep, that's what I suspect and keep trying to tell him. |
| 23:40.16 | shadowapex | Yeah, you'll only be able to check if flash() and senddtmf() executed correctly. |
| 23:40.28 | navaismo | you only know if succes like shadowapex say: Looks like Flash() returns 0 or -1 whether or not it was successful. |
| 23:47.02 | garryfre | looks like time to try to find another job. ... I hate to do it. I tend to be loyal, but I like my head in a round shape instead of pulverizing it against an impassible brick wall. Been at this since july and every time I think I've got something that will work its rejected |
| 23:47.52 | navaismo | some people cant understand that some cars cant fly |
| 23:47.53 | garryfre | He want it to act exactly like the old system. Sort of like expecting to switch from mac to linux and expecting everything to be exactly the same. |
| 23:48.05 | garryfre | I think I did see a pig fly once. |
| 23:48.33 | garryfre | I like that bout car's can't fly |
| 23:51.33 | garryfre | I think his congestion issue could be solved by using sip phones. It was sinfully easy for me to set a few up. I can call a sip from my digital office phone and talk, but if my sip tries to call it, the phone rings, but asterisk can't see that I picked the receiver up. |
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